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sipcmd

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sipcmd's Introduction

sipcmd — the command line SIP/H.323/RTP softphone

Introduction

Command line soft phone that makes phone calls, accepts calls, enters DTMF digits, plays back WAV files and records them. A useful testing tool for VoIP systems. Runs on Linux.

News

Upgraded to latest versions of ptlib and opal avaliable on apt repos on Ubuntu 12.04. (3.10.2 and 2.10.2, respectively).

Install

Dependencies

Ubuntu

apt-get install opal-dev ptlib-dev

Ubuntu 18.04 Bionic

apt-get install libopal-dev sip-dev libpt-dev

Debian

apt-get install libopal-dev libpt-dev

Download

Get source tarball from GitHub.

Compile

make

To disable debug messages, comment out DEBUG flag from Makefile

Environment

If you compile the dependencies from source, make sure that libpt and libopal are in your LD_LIBRARY_PATH. The default installation location is /usr/local/lib.

Usage

testphone options:

-u  --user          username (required)
-c  --password      password for registration
-a  --alias         username alias
-l  --localaddress  local address to listen on
-o  --opallog       enable extra opal library logging to file
-p  --listenport    the port to listen on
-P  --protocol      sip/h323/rtp (required)
-r  --remoteparty   the party to call to
-x  --execute       program to follow
-d  --audio-prefix  recorded audio filename prefix
-f  --file          the name of played sound file
-g  --gatekeeper    gatekeeper to use
-w  --gateway       gateway to use
-m  -mediaformat    one or more codecs to use, separated by semicolon; wildcards are supported (e.g. -m "G.711*;G.722*")

-l or -p without -x assumes answer mode. Additional -r forces caller id checking. -r without -l, -p or -xassumes call mode. To register to a gateaway, specify -c, -g and -w Example:

./sipcmd -P sip -u [username] -c [password] -w [server] -x "c;w200;d12345"

WAV file requirements:

mono
8 kHz sampling rate
16 bits sample size

The EBNF definition of the program syntax:

prog	:=  cmd ';'  |
cmd	:=  call | answer | hangup
	  | dtmf | voice | record | wait
	  | setlabel | loop
call	:=  'c' remoteparty [ timeout ]
answer	:=  'a' [ expectedremoteparty ]
hangup	:=  'h'
dtmf	:=  'd' digits
voice	:=  'v' audiofile
record	:=  'r' [ append ] [ silence ] [ iter ] millis audiofile
append	:=  'a'
silence	:=  's'
closed	:=  'c'
iter	:=  'i'
activity:=  'a'
wait	:=  'w' [ activity | silence ] [ closed ] millis
timeout	:=  'w' millis
setlabel:=  'l' label
loop	:=  'j' [ how-many-times ] [ 'l' label ]

Example:

"l4;c333;ws3000;d123;w200;lthrice;ws1000;vaudio;rsi4000f.out;j3lthrice;h;j4"

Parses to the following:

do this four times:
1. call to 333
2. wait until silent (max 3000 ms)
3. send DTMF digits 123
4. wait 200 ms
5. do this three times:
  1. wait until silent (max 1000 ms)
  2. send sound file 'audio'
  3. record until silent (max 4000 ms) to files 'f-[0-3]-[0-2].out'
6. hangup
7. wait 2000 ms

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sipcmd's Issues

how to disable debug mode

Hi,
How can I disable sipcmd debug mode ?

error on call

Get 2 errors on trying to call with sip cmd:

root@ubuntu:/usr/src/sipcmd-master# ./sipcmd -P sip -u 599 -c 1234 -o opal.log -x "c503;ws3000;d123;w200;h" -w 192.168.4.203
Starting sipcmd
in debug mode
Manager
Init
initialising SIP endpoint...
TestChanAudio
TestChanAudio
Listening for SIP signalling on 0.0.0.0:5060
Assertion fail: Function pthread_setschedparam failed, file ptlib/unix/tlibthrd.cxx, line 757, Error=1

bort, ore dump, hrow exception, gnore? i

Ignoring.
SIP listener up
registered as sip:[email protected]
Created LocalEndPoint
Main

Call

TestPhone::Main: calling "503" using gateway "192.168.4.203" at Fri Oct 4 11:19:05 2013

Setting up a call to: sip:[email protected]
LocalEndpoint::MakeConnection
LocalEndpointCreateConnection
LocalConnection
OnIncomingConnection: token=L4410ea3e2
connection set up to sip:[email protected]
OnReleased: reason: EndedBySecurityDenial

Wait: waiting for 3000ms

OnReleased: reason: EndedBySecurityDenial
OnClearedCall
~LocalConnection
Wait: wait done

DTMF "123"

no call found with token=C49c691cd1
Problem running command sequence ("c503;ws3000;d123;w200;h"):

TestPhone::Main: shutting down
TestPhone::Main: exiting...
Exiting...

~Manager

Follow the opal log.

0:00.043 sipcmd Version 1.0.1 by Command line VoIP testphone on Unix Linux (2.6.32-22-pve-x86_64) with PTLib (v2.10.4 (svn:26606)) at 2013/10/4 11:19:02.312
0:00.043 sipcmd OpalMan Attached endpoint with prefix sip
0:00.043 sipcmd OpalEP Created endpoint: sip
0:00.044 sipcmd PWLib File handle high water mark set: 8 PUDPSocket
0:00.044 Opal Garbage:0x8b36700 PTLib Started thread 0x88a340 (20072) Opal Garbage:0x8b36700
0:00.044 sipcmd IfaceMon Initial interface list:
127.0.0.1 <00-00-00-00-00-00> (lo)
127.0.0.2 <00-00-00-00-00-00> (venet0)
192.168.4.208 <00-00-00-00-00-00> (venet0:0)
::1 <00-00-00-00-00-00> (lo)

0:00.044 sipcmd PTLIB Opened NetLink socket
0:00.044 sipcmd PWLib File handle high water mark set: 14 Thread unblock pipe
0:00.044 sipcmd PTLib Created thread 0x88eae0
0:00.044 sipcmd PTLib Thread high water mark set: 3
0:00.044 sipcmd PWLib File handle high water mark set: 16 Thread unblock pipe
0:00.044 sipcmd PTLib Created thread 0x88dae0 Housekeeper
0:00.044 sipcmd PTLib Thread high water mark set: 4
0:00.045 Housekeeper:0x8ab4700 PTLib Started thread 0x88dae0 (20074) Housekeeper:0x8ab4700
0:00.045 sipcmd OpalMan Attached endpoint with prefix sips
0:00.045 Network In...:0x8af5700 PTLib Started thread 0x88eae0 (20073) Network Interface Monitor:0x8af5700
0:00.045 sipcmd SIP Created endpoint.
0:00.045 sipcmd OpalMan Added route "local:.=sip:"
0:00.045 sipcmd OpalMan Added route "sip:.=local:"
0:00.045 sipcmd PWLib File handle high water mark set: 17 PUDPSocket
0:00.045 sipcmd MonSock Created socket bundle for all interfaces.
0:00.045 sipcmd PWLib File handle high water mark set: 18 PUDPSocket
0:00.045 sipcmd MonSock Created bundled UDP socket 192.168.4.208:5060
0:00.045 sipcmd PWLib File handle high water mark set: 20 Thread unblock pipe
0:00.045 sipcmd PTLib Created thread 0x894740 Opal Listener
0:00.046 sipcmd PTLib Thread high water mark set: 5
0:00.045 Network In...:0x8af5700 IfaceMon Started interface monitor thread.
0:00.046 sipcmd PWLib Assertion fail: Function pthread_setschedparam failed, file ptlib/unix/tlibthrd.cxx, line 757, Error=1
0:00.046 Opal Listener:0x8a73700 PTLib Started thread 0x894740 (20075) Opal Listener:0x8a73700
0:00.046 Opal Listener:0x8a73700 Listen Started listening thread on udp$:5060
0:03.672 sipcmd SIP Start REGISTER
aor=599
remote=192.168.4.203
local=
contact=
authID=
realm=
expire=0
restore=30
minRetry=default
maxRetry=default
compatibility=FullyCompliant
0:03.673 sipcmd PWLib File handle high water mark set: 21 PUDPSocket
0:03.673 sipcmd SIP Constructed REGISTER handler for sip:[email protected]
0:03.673 sipcmd SIP Executing state change to Subscribing for REGISTER handler, target=sip:[email protected], id=a22c29fa-752b-e311-8d69-f9c4e96053ec@ubuntu
0:03.673 sipcmd SIP Changing REGISTER handler from Unavailable to Subscribing, target=sip:[email protected], id=a22c29fa-752b-e311-8d69-f9c4e96053ec@ubuntu
0:03.674 sipcmd OpalUDP Binding to interface: 0.0.0.0:5060
0:03.674 sipcmd SIP Created transport udp$192.168.4.203:5060<if=udp$:5060>
0:03.674 sipcmd OpalUDP Started connect to 192.168.4.203:5060
0:03.674 sipcmd OpalUDP Writing to interface 0 - "192.168.4.208%venet0:0"
0:03.675 sipcmd OpalMan Listener interfaces: associated transport=udp$192.168.4.208:5060
udp$192.168.4.208:5060
0:03.675 sipcmd SIP Transaction created.
0:03.676 sipcmd SIP Transaction remote address is udp$192.168.4.203:5060
0:03.676 sipcmd SIP Sending PDU (554 bytes) to: rem=udp$192.168.4.203:5060,local=udp$192.168.4.208:5060,if=192.168.4.208%venet0:0
REGISTER sip:192.168.4.203 SIP/2.0
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 192.168.4.208:5060;branch=z9hG4bK8a9429fa-752b-e311-8d69-f9c4e96053ec;rport
User-Agent: sipcmd/1.0.1
From: sip:[email protected];tag=524029fa-752b-e311-8d69-f9c4e96053ec
Call-ID: a22c29fa-752b-e311-8d69-f9c4e96053ec@ubuntu
Organization: Command line VoIP testphone
To: sip:[email protected]
Contact: sip:[email protected]:5060;q=1
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
Expires: 3600
Content-Length: 0
Max-Forwards: 70

0:03.676 sipcmd OpalUDP Setting interface to 192.168.4.208%venet0:0
0:03.676 sipcmd SIP Transaction timers set: retry=10.000, completion=16.000
0:03.676 sipcmd OpalMan Attached endpoint with prefix local
0:03.676 sipcmd OpalEP Created endpoint: local
0:03.676 sipcmd LocalEP Created endpoint.
0:03.677 sipcmd OpalMan Set up call from local:* to sip:[email protected]
0:03.677 sipcmd Call Created Call[C49c691cd1]
0:03.677 sipcmd OpalMan Set up connection to "local:"
0:03.677 sipcmd OpalCon Created connection Call[C49c691cd1]-EP[L4410ea3e2]
0:03.677 sipcmd LocalCon Created connection with token "L4410ea3e2"
0:03.677 sipcmd Call GetOtherPartyConnection Call[C49c691cd1]-EP[L4410ea3e2]
0:03.677 sipcmd OpalCon SetPhase from UninitialisedPhase to SetUpPhase for Call[C49c691cd1]-EP[L4410ea3e2]
0:03.677 sipcmd OpalMan OnIncoming connection Call[C49c691cd1]-EP[L4410ea3e2]
0:03.677 sipcmd Call GetOtherPartyConnection Call[C49c691cd1]-EP[L4410ea3e2]
0:03.677 sipcmd Call GetOtherPartyConnection Call[C49c691cd1]-EP[L4410ea3e2]
0:03.677 sipcmd OpalMan Searching for route "local:root sip:[email protected]"
0:03.677 sipcmd OpalMan Matched regex "^local:. .$" ("local:.")
0:03.677 sipcmd OpalMan Set up connection to "sip:[email protected]"
0:03.677 sipcmd OpalCon Created connection Call[C49c691cd1]-EP[cee129fa-752b-e311-8d69-f9c4e96053ec]
0:03.677 sipcmd RFC2833 Handler created
0:03.678 sipcmd RFC2833 Handler created
0:03.678 sipcmd SIP Created connection.
0:03.678 sipcmd LocalCon Outgoing call routed to sip:[email protected] for Call[C49c691cd1]-EP[L4410ea3e2]
0:03.678 sipcmd Call OnSetUp Call[C49c691cd1]-EP[L4410ea3e2]
0:03.678 sipcmd SIP SetUpConnection: sip:[email protected]
0:03.678 sipcmd SIP Connecting to sip:[email protected] via sip:[email protected]
0:03.678 sipcmd SIP Setting new transport for destination "sip:[email protected]"
0:03.678 sipcmd SIP Found registrar on domain 192.168.4.203, using interface 192.168.4.208%venet0:0
0:03.678 sipcmd Opal Illegal IP transport port/service: "tcp$192.168.4.208%venet0:0"
0:03.678 sipcmd OpalUDP Binding to interface: 0.0.0.0:5060
0:03.678 sipcmd SIP Created transport udp$192.168.4.203:5060<if=udp$*:5060>
0:03.678 sipcmd OpalUDP Started connect to 192.168.4.203:5060
0:03.688 sipcmd Call GetMediaFormats for Call[C49c691cd1]-EP[cee129fa-752b-e311-8d69-f9c4e96053ec]
G.722.2,GSM-AMR,GSM-06.10,G.726-40k,G.726-32k,G.726-24k,G.726-16k,G.711-uLaw-64k,G.711-ALaw-64k,H.261,G.722-64k,G.722.1-24k,G.722.1-32k,SILK-16,SpeexIETFWide-20.6k,SpeexWB,SpeexWide-20.6k,LPC-10,MS-GSM,MS-IMA-ADPCM,SILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIETFNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.38,UserInput/RFC2833,NamedSignalEvent,H.261-CIF,H.261-QCIF,RFC4175_YCbCr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ
0:03.688 sipcmd SIP Local media formats set:
G.722.2,GSM-AMR,GSM-06.10,G.726-40k,G.726-32k,G.726-24k,G.726-16k,G.711-uLaw-64k,G.711-ALaw-64k,H.261,G.722-64k,G.722.1-24k,G.722.1-32k,SILK-16,SpeexIETFWide-20.6k,SpeexWB,SpeexWide-20.6k,LPC-10,MS-GSM,MS-IMA-ADPCM,SILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIETFNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.38,UserInput/RFC2833,NamedSignalEvent,H.261-CIF,H.261-QCIF,RFC4175_YCbCr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ
0:03.688 sipcmd SIP Remote media formats set:
G.722.2,GSM-AMR,GSM-06.10,G.726-40k,G.726-32k,G.726-24k,G.726-16k,G.711-uLaw-64k,G.711-ALaw-64k,H.261,G.722-64k,G.722.1-24k,G.722.1-32k,SILK-16,SpeexIETFWide-20.6k,SpeexWB,SpeexWide-20.6k,LPC-10,MS-GSM,MS-IMA-ADPCM,SILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIETFNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.38,UserInput/RFC2833,NamedSignalEvent,H.261-CIF,H.261-QCIF,RFC4175_YCbCr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ
0:03.688 sipcmd OpalUDP Writing to interface 0 - "192.168.4.208%venet0:0"
0:03.688 sipcmd SIP Getting local URI from registeration: sip:[email protected]
0:03.688 sipcmd SIP Updating dialog tag from "" to "cee129fa-752b-e311-8d69-f9c4e96053ec"
0:03.688 sipcmd SIP Remote dialog address from target: sip:[email protected]
0:03.689 sipcmd SIP INVITE transaction id=z9hG4bKa2932bfa-752b-e311-8d69-f9c4e96053ec created.
0:03.689 sipcmd SIP Creating INVITE request
0:03.689 sipcmd SIP Offering all configured media:
G.722.2,GSM-AMR,GSM-06.10,G.726-40k,G.726-32k,G.726-24k,G.726-16k,G.711-uLaw-64k,G.711-ALaw-64k,H.261,G.722-64k,G.722.1-24k,G.722.1-32k,SILK-16,SpeexIETFWide-20.6k,SpeexWB,SpeexWide-20.6k,LPC-10,MS-GSM,MS-IMA-ADPCM,SILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIETFNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.38,UserInput/RFC2833,NamedSignalEvent,H.261-CIF,H.261-QCIF,RFC4175_YCbCr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ
0:03.689 sipcmd SIP Offering media type audio in SDP
0:03.689 sipcmd Call IsMediaBypassPossible Call[C49c691cd1]-EP[cee129fa-752b-e311-8d69-f9c4e96053ec] session 1
0:03.689 sipcmd OpalMan IsMediaBypassPossible: session 1
0:03.689 sipcmd OpalCon IsMediaBypassPossible: default returns false
0:03.689 sipcmd RTP Cannot find media session 1
0:03.689 sipcmd RTP Cannot find RTP session 1
0:03.690 sipcmd PTLib Created PXConfig 0x8b2120
0:03.690 sipcmd VoIP Metrics RTCP_XR_Metrics created.
0:03.690 sipcmd RTP_UDP Session 1, created with NAT flag set to 0
0:03.690 sipcmd PWLib File handle high water mark set: 22 PUDPSocket
0:03.690 sipcmd PWLib File handle low water mark set: 21 PUDPSocket
0:03.690 sipcmd RTP_UDP Session 1 created: 192.168.4.208:5000-5001 ssrc=916538279
0:03.690 sipcmd RTP Creating new session RTP_UDP
0:03.690 sipcmd RTPEp Session 1, remembering local RTP port 5000 on connection Call[C49c691cd1]-EP[cee129fa-752b-e311-8d69-f9c4e96053ec]
0:03.690 sipcmd RTP Found existing media session 1
0:03.691 sipcmd MediaFormat Validation of merge for media option "BitRate" failed.
0:03.691 sipcmd SDP SDP not including SpeexIETFWide-20.6k as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including SpeexWide-20.6k as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including MS-GSM as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including MS-IMA-ADPCM as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including SpeexIETFNarrow-11k as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including SpeexIETFNarrow-15k as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including SpeexIETFNarrow-18.2k as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including SpeexIETFNarrow-24.6k as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including SpeexIETFNarrow-5.95k as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including SpeexIETFNarrow-8k as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including SpeexWNarrow-8k as it is not a SIP transportable format
0:03.691 sipcmd MediaFormat Merging UserInput/RFC2833 into UserInput/RFC2833
0:03.691 sipcmd RFC2833 Set tx pt=[pt=101], events="0-16,32,36" for UserInput/RFC2833
0:03.691 sipcmd RFC2833 Set rx pt=[pt=101], events="0-16,32,36" for UserInput/RFC2833
0:03.691 sipcmd MediaFormat Merging NamedSignalEvent into NamedSignalEvent
0:03.691 sipcmd RFC2833 Set tx pt=[pt=100], events="192-193" for NamedSignalEvent
0:03.691 sipcmd RFC2833 Set rx pt=[pt=100], events="192-193" for NamedSignalEvent
0:03.692 sipcmd SIP Offering media type video in SDP
0:03.692 sipcmd Call IsMediaBypassPossible Call[C49c691cd1]-EP[cee129fa-752b-e311-8d69-f9c4e96053ec] session 2
0:03.692 sipcmd OpalMan IsMediaBypassPossible: session 2
0:03.692 sipcmd OpalCon IsMediaBypassPossible: default returns false
0:03.692 sipcmd RTP Cannot find media session 2
0:03.692 sipcmd RTP Cannot find RTP session 2
0:03.692 sipcmd VoIP Metrics RTCP_XR_Metrics created.
0:03.692 sipcmd RTP_UDP Session 2, created with NAT flag set to 0
0:03.692 sipcmd PWLib File handle high water mark set: 23 PUDPSocket
0:03.692 sipcmd PWLib File handle high water mark set: 24 PUDPSocket
0:03.692 sipcmd PWLib File handle low water mark set: 23 PUDPSocket
0:03.692 sipcmd RTP_UDP SetOption(23,8,1048576) failed, even though it said it succeeded!
0:03.692 sipcmd RTP_UDP SetOption(23,8,524288) failed, even though it said it succeeded!
0:03.692 sipcmd RTP_UDP SetOption(23,8,262144) succeeded.
0:03.692 sipcmd RTP_UDP Session 2 created: 192.168.4.208:5002-5003 ssrc=1440891105
0:03.692 sipcmd RTP Creating new session RTP_UDP
0:03.692 sipcmd RTPEp Session 2, remembering local RTP port 5002 on connection Call[C49c691cd1]-EP[cee129fa-752b-e311-8d69-f9c4e96053ec]
0:03.692 sipcmd RTP Found existing media session 2
0:03.692 sipcmd SDP SDP not including H.261-CIF as it is not a SIP transportable format
0:03.692 sipcmd SDP SDP not including H.261-QCIF as it is not a SIP transportable format
0:03.693 sipcmd SIP Transaction remote address is udp$192.168.4.203:5060
0:03.693 sipcmd OpalPlugin to_customised_options:
Format Name = SILK-16
Media Type = audio
Payload Type = [pt=103]
Encoding Name = SILK
Channels (R/W) = 1 UnsignedInt
Clock Rate (R/O) = 16000 UnsignedInt
Complexity (R/O) = 1 UnsignedInt
Frame Time (R/O) = 320 UnsignedInt
Max Bit Rate (R/O) = 30000 UnsignedInt
Max Frame Size (R/O) = 75 UnsignedInt
Max Frames Per Packet (R/O) = 5 UnsignedInt
Needs Jitter (R/O) = 1 Boolean
Protocol (R/O) = SIP String
Rx Frames Per Packet (R/W) = 5 UnsignedInt
Tx Frames Per Packet (R/W) = 2 UnsignedInt
Use DTX (R/O) = 0 FMTP name: usedtx (0) Boolean
Use In-Band FEC (R/O) = 1 FMTP name: useinbandfec (1) Boolean

0:03.694 sipcmd OpalPlugin to_customised_options:
Format Name = SILK-8
Media Type = audio
Payload Type = [pt=102]
Encoding Name = SILK
Channels (R/W) = 1 UnsignedInt
Clock Rate (R/O) = 8000 UnsignedInt
Complexity (R/O) = 1 UnsignedInt
Frame Time (R/O) = 160 UnsignedInt
Max Bit Rate (R/O) = 20000 UnsignedInt
Max Frame Size (R/O) = 50 UnsignedInt
Max Frames Per Packet (R/O) = 5 UnsignedInt
Needs Jitter (R/O) = 1 Boolean
Protocol (R/O) = SIP String
Rx Frames Per Packet (R/W) = 5 UnsignedInt
Tx Frames Per Packet (R/W) = 2 UnsignedInt
Use DTX (R/O) = 0 FMTP name: usedtx (0) Boolean
Use In-Band FEC (R/O) = 1 FMTP name: useinbandfec (1) Boolean

0:03.695 sipcmd OpalPlugin to_customised_options:
Format Name = H.261
Media Type = video
Payload Type = H261
Encoding Name = h261
Annex D (R/O) = 0 FMTP name: D (0) Boolean
CIF MPI (R/W) = 1 FMTP name: CIF (33) UnsignedInt
Clock Rate (R/O) = 90000 UnsignedInt
Content Role (R/W) = No Role Enum
Content Role Mask (R/W) = 0 UnsignedInt
Frame Height (R/W) = 288 UnsignedInt
Frame Time (R/W) = 1500 UnsignedInt
Frame Width (R/W) = 352 UnsignedInt
Max Bit Rate (R/W) = 621700 UnsignedInt
Max Rx Frame Height (R/O) = 288 UnsignedInt
Max Rx Frame Width (R/O) = 352 UnsignedInt
Max Tx Packet Size (R/O) = 1444 UnsignedInt
Min Rx Frame Height (R/O) = 144 UnsignedInt
Min Rx Frame Width (R/O) = 176 UnsignedInt
Protocol (R/O) = SIP String
QCIF MPI (R/W) = 1 FMTP name: QCIF (33) UnsignedInt
Rate Control Enable (R/W) = 0 Boolean
Rate Controller (R/W) = String
Target Bit Rate (R/W) = 621700 UnsignedInt
Tx Key Frame Period (R/W) = 125 UnsignedInt

0:03.695 sipcmd OpalPlugin to_customised_options:
Format Name = theora
Media Type = video
Payload Type = [pt=124]
Encoding Name = theora
CAP Delivery (R/W) = in_band FMTP name: delivery-method (in_band) String
CAP Height (R/W) = 576 FMTP name: height (15) UnsignedInt
CAP Sampling (R/W) = YCbCr-4:2:0 FMTP name: sampling (YCbCr-4:2:0) String
CAP Width (R/W) = 704 FMTP name: width (15) UnsignedInt
Clock Rate (R/O) = 90000 UnsignedInt
Content Role (R/W) = No Role Enum
Content Role Mask (R/W) = 0 UnsignedInt
Frame Height (R/W) = 288 UnsignedInt
Frame Time (R/W) = 1500 UnsignedInt
Frame Width (R/W) = 352 UnsignedInt
Max Bit Rate (R/W) = 768000 UnsignedInt
Max Rx Frame Height (R/O) = 720 UnsignedInt
Max Rx Frame Width (R/O) = 1280 UnsignedInt
Max Tx Packet Size (R/O) = 1444 UnsignedInt
Min Rx Frame Height (R/O) = 144 UnsignedInt
Min Rx Frame Width (R/O) = 176 UnsignedInt
Protocol (R/O) = SIP String
Rate Control Enable (R/W) = 0 Boolean
Rate Controller (R/W) = String
Target Bit Rate (R/W) = 768000 UnsignedInt
Tx Key Frame Period (R/W) = 125 UnsignedInt

0:03.696 sipcmd SIP PDU is too large (1747 bytes) trying compact form.
0:03.697 sipcmd SIP PDU is likely too large (1697 bytes) for UDP datagram.
0:03.697 sipcmd SIP Sending PDU (1697 bytes) to: rem=udp$192.168.4.203:5060,local=udp$192.168.4.208:5060,if=192.168.4.208%venet0:0
INVITE sip:[email protected] SIP/2.0
CSeq: 1 INVITE
v: SIP/2.0/UDP 192.168.4.208:5060;branch=z9hG4bKa2932bfa-752b-e311-8d69-f9c4e96053ec;rport
User-Agent: sipcmd/1.0.1
f: "root" sip:[email protected];tag=cee129fa-752b-e311-8d69-f9c4e96053ec
i: 20ea29fa-752b-e311-8d69-f9c4e96053ec@ubuntu
k: 100rel,replaces
Organization: Command line VoIP testphone
t: sip:[email protected]
m: "root" sip:[email protected]
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
l: 1141
c: application/sdp
Max-Forwards: 70

v=0
o=- 1380899945 1 IN IP4 192.168.4.208
s=sipcmd/1.0.1
c=IN IP4 192.168.4.208
t=0 0
m=audio 5000 RTP/AVP 123 125 3 116 117 118 119 0 8 9 121 122 103 115 104 102 114 101 100
a=sendrecv
a=rtpmap:123 AMR-WB/16000/1
a=fmtp:123 octet-align=1
a=rtpmap:125 AMR/8000/1
a=rtpmap:3 gsm/8000/1
a=rtpmap:116 G726-40/8000/1
a=rtpmap:117 G726-32/8000/1
a=rtpmap:118 G726-24/8000/1
a=rtpmap:119 G726-16/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:9 G722/8000/1
a=rtpmap:121 G7221/16000/1
a=fmtp:121 bitrate=24000
a=rtpmap:122 G7221/16000/1
a=fmtp:122 bitrate=32000
a=rtpmap:103 SILK/16000/1
a=rtpmap:115 Speex/16000/1
a=rtpmap:104 lpc10/8000/1
a=rtpmap:102 SILK/8000/1
a=rtpmap:114 Speex/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=maxptime:22
m=video 5002 RTP/AVP 31 97 124
b=AS:186624
b=TIAS:186624000
a=sendrecv
a=rtpmap:31 h261/90000
a=fmtp:31 CIF=1;QCIF=1
a=rtpmap:97 raw/90000
a=fmtp:97 rate=90000;height=288;width=352;colorimetry=BT601-5;depth=8;sampling=YCbCr-4:2:0
a=rtpmap:124 theora/90000
a=fmtp:124 height=576;width=704

0:03.697 sipcmd OpalUDP Setting interface to 192.168.4.208%venet0:0
0:03.697 sipcmd SIP Transaction timers set: retry=10.000, completion=32.000
0:03.697 sipcmd OpalCon SetPhase from UninitialisedPhase to SetUpPhase for Call[C49c691cd1]-EP[cee129fa-752b-e311-8d69-f9c4e96053ec]
0:03.697 sipcmd OpalCon OnSetUpConnectionCall[C49c691cd1]-EP[cee129fa-752b-e311-8d69-f9c4e96053ec]
0:03.697 sipcmd OpalEP OnSetUpConnection Call[C49c691cd1]-EP[cee129fa-752b-e311-8d69-f9c4e96053ec]
0:03.697 sipcmd OpalMan SetUpCall succeeded, call=Call[C49c691cd1]
0:03.924 Opal Listener:0x8a73700 OpalUDP Binding to interface: 192.168.4.208:5060
0:03.924 Opal Listener:0x8a73700 SIP Waiting for PDU on udp$192.168.4.203:5060<if=udp$192.168.4.208:5060>
0:03.925 Opal Listener:0x8a73700 SIP PDU received: rem=udp$192.168.4.203:5060,local=udp$192.168.4.208:5060,if=192.168.4.208%venet0:0
SIP/2.0 404 Not found
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 192.168.4.208:5060;branch=z9hG4bK8a9429fa-752b-e311-8d69-f9c4e96053ec;received=192.168.4.208;rport=5060
User-Agent: Asterisk PBX
From: sip:[email protected];tag=524029fa-752b-e311-8d69-f9c4e96053ec
Call-ID: a22c29fa-752b-e311-8d69-f9c4e96053ec@ubuntu
Supported: replaces
To: sip:[email protected];tag=as432f22a7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Content-Length: 0

Localaddress option is broken

Hi! I've tried to run sipcmd on a machine with several Network Cards, all in different subnets. I expected -localaddress to bind the application to one of them. Instead, it seems to pick up the NC randomly.

sipcmd terminates when call is picked up

Hi,
when executing sipcmd to call someone. sipcmd terminates when the call is picked up.
System: Raspbian GNU/Linux 8
This is my command:
sudo /opt/sipcmd-master/sipcmd -P sip -u 70 -c <password> -w 192.168.1.10 -x 'c20;w10000;h'

Could you please help.
Thanks

This is the output:

Starting sipcmd
in debug mode
Manager
Init
initialising SIP endpoint...
Listening for SIP signalling on 0.0.0.0:TestChanAudio
TestChanAudio
5060
SIP listener up
registered as sip:[email protected]
Created LocalEndPoint
Main
## Call ##
TestPhone::Main: calling "20" using gateway "192.168.1.10" at Fri Nov 18 14:46:20 2016

Setting up a call to: sip:[email protected]
LocalEndpoint::MakeConnection
LocalEndpointCreateConnection
LocalConnection
OnIncomingConnection: token=Lf47b23482
connection set up to sip:[email protected]
TestPhone::Main: calling "sip:[email protected]" for 0
TestPhone::Main: calling "sip:[email protected]" for 0
TestPhone::Main: calling "sip:[email protected]" for 0
TestPhone::Main: calling "sip:[email protected]" for 1
TestPhone::Main: calling "sip:[email protected]" for 1
TestPhone::Main: calling "sip:[email protected]" for 1
TestPhone::Main: calling "sip:[email protected]" for 1
TestPhone::Main: calling "sip:[email protected]" for 1
TestPhone::Main: calling "sip:[email protected]" for 1
TestPhone::Main: calling "sip:[email protected]" for 1
TestPhone::Main: calling "sip:[email protected]" for 1
TestPhone::Main: calling "sip:[email protected]" for 2
TestPhone::Main: calling "sip:[email protected]" for 2
TestPhone::Main: calling "sip:[email protected]" for 2
TestPhone::Main: calling "sip:[email protected]" for 2
OnOpenMediaStream
recording media from sip
CreateMediaStream
TestChannel[ Call[Cba18a7b31]-EP<local>[Lf47b23482] - 0x6e129a88 ]
OnOpenMediaStream
streaming media to local
TestPhone::Main: calling "sip:[email protected]" for 4
TestChannel::Write
pi@raspberrypi:~ $

Problem Running Command Sequence

I have a local PBX on my Debian linux machine. I am attempting to test calls between the extensions. The call isn't going through, and it isn't obvious to me through the console, nor the logs, as to what is causing the call to fail. Below is the command I've used:

admin@ip-172-26-2-113:~/.local/lib/sipcmd-master$ ./sipcmd -P sip -u UID -c myPassword -w 127.0.0.1 -x "c0004;ws30;vclip"

So, I am attempting to use the loopback address to place a call. Both extensions are showing as registering, and when I dial 0004 on the phone client, it successfully makes the call, and everything seems to be working fine. However, when I try to do it through sipcmd, it fails, and I'm not certain why.

Here is the console's output (without the logs):

    Starting sipcmd
    in debug mode
    Manager
    Init
    initialising SIP endpoint...
    TestChanAudio
    TestChanAudio
    Listening for SIP signalling on 0.0.0.0:5060
    SIP listener up
    registered as sip:[email protected]
    Created LocalEndPoint
    Main
    ## Call ##
    TestPhone::Main: calling "0004" using gateway "127.0.0.1" at Wed Mar 13 14:36:35 2019
    
    Setting up a call to: sip:[email protected]
    LocalEndpoint::MakeConnection
    LocalEndpointCreateConnection
    LocalConnection
    OnIncomingConnection: token=L4e2c6dc42
    Call setup to sip:[email protected] failed
    Problem running command sequence ("c0004;ws30;vclip"):
    
    TestPhone::Main: shutting down
    OnReleased: reason: EndedByNoAccept
    OnReleased: reason: EndedByTransportFail
    OnClearedCall
    ~LocalConnection
    TestPhone::Main: exiting...
    Exiting...
    ~Manager

And the following are the logs when I used the -o flag:

	Starting sipcmd
	in debug mode
	Manager
	Init
	sipcmd: Could not open trace output file "/home/admin/.local/lib/sipcmd-master/~./call.log"  0:00.038                        sipcmd         Version 1.0.1 by Command line VoIP testphone on Unix Linux (4.9.0-8-amd64-x86_64) with PTLib (v2.10.11 (svn:30295)) at 2019/3/13 11:12:36.666
	initialising SIP endpoint...
	  0:00.038                       sipcmd OpalMan Attached endpoint with prefix sip
	  0:00.038                       sipcmd OpalEP  Created endpoint: sip
	  0:00.039                       sipcmd PTLib   Created read/write mutex 0x55d893db4438
	  0:00.039                       sipcmd PWLib   File handle high water mark set: 8 PUDPSocket
	  0:00.039                       sipcmd IfaceMon        Initial interface list:
	127.0.0.1 <00-00-00-00-00-00> (lo)
	172.26.2.113 <02-D4-F5-4A-EF-8C> (eth0)

	  0:00.040                       sipcmd PTLIB   Opened NetLink socket
	  0:00.040                       sipcmd PWLib   File handle high water mark set: 15 Thread unblock pipe
	  0:00.040                       sipcmd PTLib   Created thread 0x55d893db4910
	  0:00.040                       sipcmd PTLib   Thread high water mark set: 3
	  0:00.040                       sipcmd PTLib   Created read/write mutex 0x55d893db46b8
	  0:00.040                       sipcmd PWLib   File handle high water mark set: 17 Thread unblock pipe
	  0:00.041                       sipcmd PTLib   Created thread 0x55d893db4bc0 Housekeeper
	  0:00.041                       sipcmd PTLib   No permission to set priority level 4
	  0:00.041                       sipcmd PTLib   Thread high water mark set: 4
	  0:00.041                       sipcmd OpalMan Attached endpoint with prefix sips
	  0:00.042                       sipcmd SIP     Created endpoint.
	  0:00.042                       sipcmd OpalMan Added route "local:.*=sip:<da>"
	  0:00.042                       sipcmd OpalMan Added route "sip:.*=local:<db>"
	TestChanAudio
	TestChanAudio
	Listening for SIP signalling on 0.0.0.0:5060
	  0:00.042                       sipcmd PTLib   Created read/write mutex 0x55d893dbada0
	  0:00.043                       sipcmd PWLib   File handle high water mark set: 18 PUDPSocket
	  0:00.043                       sipcmd MonSock Created socket bundle for all interfaces.
	  0:00.043                       sipcmd PWLib   File handle high water mark set: 19 PUDPSocket
	  0:00.043                       sipcmd MonSock Could not listen on 172.26.2.113:5060 - Address already in use
	  0:00.043                       sipcmd PWLib   File handle high water mark set: 20 Thread unblock pipe
	  0:00.044                       sipcmd PTLib   Created thread 0x55d893db9c80 Opal Listener
	  0:00.044                       sipcmd PTLib   Thread high water mark set: 5
	  0:00.044                       sipcmd PTLib   No permission to set priority level 4
	SIP listener up
	  0:00.044                       sipcmd SIP     Start REGISTER
			  aor=UID
		   remote=127.0.0.1
			local=
		  contact=
			proxy=
		   authID=
			realm=
		   expire=0
		  restore=30
		 minRetry=default
		 maxRetry=default
	compatibility=FullyCompliant
	  0:00.045                       sipcmd SIP     Normalised REGISTER
			  aor=sip:[email protected]
		   remote=sip:[email protected]
			local=
		  contact=
			proxy=
		   authID=UID
			realm=
		   expire=3600
		  restore=30
		 minRetry=default
		 maxRetry=default
	compatibility=FullyCompliant
	  0:00.046                       sipcmd PTLib   Created read/write mutex 0x55d893dbb580
	  0:00.046      Housekeepe...3168ef4700 PTLib   Started thread 0x55d893db4bc0 (24057) Housekeeper:0x7f3168ef4700
	  0:00.046                       sipcmd PWLib   File handle high water mark set: 21 PUDPSocket
	  0:00.047                       sipcmd SIP     Constructed REGISTER handler for sip:[email protected]
	  0:00.047                       sipcmd SIP     Executing state change to Subscribing for REGISTER handler, target=sip:[email protected], id=6636871a-1044-e911-9313-02d4f54aef8c@ip-172-26-2-113
	  0:00.047                       sipcmd SIP     Changing REGISTER handler from Unavailable to Subscribing, target=sip:[email protected], id=6636871a-1044-e911-9313-02d4f54aef8c@ip-172-26-2-113
	  0:00.048                       sipcmd PTLib   Created read/write mutex 0x55d893dbd2f0
	  0:00.048                       sipcmd OpalUDP Binding to interface: 0.0.0.0:5060
	  0:00.048                       sipcmd SIP     Created transport udp$127.0.0.1:5060<if=udp$*:5060>
	  0:00.048                       sipcmd OpalUDP Started connect to 127.0.0.1:5060
	  0:00.049                       sipcmd SIP     Changing REGISTER handler from Subscribing to Unavailable, target=sip:[email protected], id=6636871a-1044-e911-9313-02d4f54aef8c@ip-172-26-2-113
	  0:00.049                       sipcmd SIP     Retrying REGISTER after 30 seconds.
	registered as sip:[email protected]
	  0:00.049      Opal Liste...3168eb3700 PTLib   Started thread 0x55d893db9c80 (24058) Opal Listener:0x7f3168eb3700
	  0:00.049      Opal Liste...3168eb3700 Listen  Started listening thread on udp$*:5060
	  0:00.049      Opal Garba...3168f76700 PTLib   Started thread 0x55d893db36e0 (24055) Opal Garbage:0x7f3168f76700
	  0:00.049                       sipcmd OpalMan Attached endpoint with prefix local
	  0:00.049                       sipcmd OpalEP  Created endpoint: local
	  0:00.050                       sipcmd LocalEP Created endpoint.
	Created LocalEndPoint
	Main
	## Call ##
	TestPhone::Main: calling "0004" using gateway "127.0.0.1" at Wed Mar 13 11:12:36 2019
	  0:00.050      Network In...3168f35700 PTLib   Started thread 0x55d893db4910 (24056) Network Interface Monitor:0x7f3168f35700
	  0:00.050      Network In...3168f35700 IfaceMon        Started interface monitor thread.

	Setting up a call to: sip:[email protected]
	  0:00.050                       sipcmd OpalMan Set up call from local:* to sip:[email protected]
	  0:00.050                       sipcmd PTLib   Created read/write mutex 0x55d893dbfdc0
	  0:00.050                       sipcmd Call    Created Call[C5416e2a61]
	  0:00.050                       sipcmd OpalMan Set up connection to "local:*"
	LocalEndpoint::MakeConnection
	LocalEndpointCreateConnection
	  0:00.050                       sipcmd PTLib   Created read/write mutex 0x55d893dc03a0
	  0:00.050                       sipcmd OpalCon Created connection Call[C5416e2a61]-EP<local>[Lfe18107c2]
	  0:00.051                       sipcmd LocalCon        Created connection with token "Lfe18107c2"
	LocalConnection
	  0:00.051                       sipcmd Call    GetOtherPartyConnection Call[C5416e2a61]-EP<local>[Lfe18107c2]
	  0:00.051                       sipcmd OpalCon SetPhase from UninitialisedPhase to SetUpPhase for Call[C5416e2a61]-EP<local>[Lfe18107c2]
	OnIncomingConnection: token=Lfe18107c2
	  0:00.051                       sipcmd OpalMan OnIncoming connection Call[C5416e2a61]-EP<local>[Lfe18107c2]
	  0:00.051                       sipcmd Call    GetOtherPartyConnection Call[C5416e2a61]-EP<local>[Lfe18107c2]
	  0:00.051                       sipcmd Call    GetOtherPartyConnection Call[C5416e2a61]-EP<local>[Lfe18107c2]
	  0:00.051                       sipcmd OpalMan Searching for route "local:admin        sip:[email protected]"
	  0:00.051                       sipcmd OpalMan Matched regex "^local:.*        .*$" ("local:.*")
	  0:00.051                       sipcmd OpalMan Set up connection to "sip:[email protected]"
	  0:00.052                       sipcmd PTLib   Created read/write mutex 0x55d893dc6f30
	  0:00.052                       sipcmd OpalCon Created connection Call[C5416e2a61]-EP<sip>[480d881a-1044-e911-9313-02d4f54aef8c]
	  0:00.052                       sipcmd RFC2833 Handler created
	  0:00.052                       sipcmd RFC2833 Handler created
	  0:00.052                       sipcmd SIP     Created connection.
	  0:00.053                       sipcmd LocalCon        Outgoing call routed to sip:[email protected] for Call[C5416e2a61]-EP<local>[Lfe18107c2]
	  0:00.053                       sipcmd Call    OnSetUp Call[C5416e2a61]-EP<local>[Lfe18107c2]
	  0:00.053                       sipcmd SIP     SetUpConnection: sip:[email protected]
	  0:00.053                       sipcmd SIP     Connecting to sip:[email protected] via sip:[email protected]
	  0:00.053                       sipcmd SIP     Setting new transport for destination "sip:[email protected]"
	  0:00.053                       sipcmd SIP     Found registrar on domain 127.0.0.1, using interface
	  0:00.053                       sipcmd PTLib   Created read/write mutex 0x55d893dcb340
	  0:00.054                       sipcmd OpalUDP Binding to interface: 0.0.0.0:5060
	  0:00.054                       sipcmd SIP     Created transport udp$127.0.0.1:5060<if=udp$*:5060>
	  0:00.054                       sipcmd OpalUDP Started connect to 127.0.0.1:5060
	  0:00.060                       sipcmd Call    GetMediaFormats for Call[C5416e2a61]-EP<sip>[480d881a-1044-e911-9313-02d4f54aef8c]
		G.722.2,GSM-AMR,iLBC,GSM-06.10,G.726-40k,G.726-32k,G.726-24k,G.726-16k,G.711-uLaw-64k,G.711-ALaw-64k,H.264,H.264-1,H.264-0,MPEG4,H.263,H.263plus,H.261,G.722-64k,G.722.1-24k,G.722.1-32k,SILK-16,SpeexIETFWide-20.6k,SpeexWB,SpeexWide-20.6k,LPC-10,MS-GSM,MS-IMA-ADPCM,SILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIETFNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.38,iLBC-13k3,iLBC-15k2,UserInput/RFC2833,NamedSignalEvent,RFC4175_YCbCr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ
	  0:00.060                       sipcmd SIP     Local media formats set:
		G.722.2,GSM-AMR,iLBC,GSM-06.10,G.726-40k,G.726-32k,G.726-24k,G.726-16k,G.711-uLaw-64k,G.711-ALaw-64k,H.264,H.264-1,H.264-0,MPEG4,H.263,H.263plus,H.261,G.722-64k,G.722.1-24k,G.722.1-32k,SILK-16,SpeexIETFWide-20.6k,SpeexWB,SpeexWide-20.6k,LPC-10,MS-GSM,MS-IMA-ADPCM,SILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIETFNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.38,iLBC-13k3,iLBC-15k2,UserInput/RFC2833,NamedSignalEvent,RFC4175_YCbCr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ
	  0:00.060                       sipcmd SIP     Remote media formats set:
		G.722.2,GSM-AMR,iLBC,GSM-06.10,G.726-40k,G.726-32k,G.726-24k,G.726-16k,G.711-uLaw-64k,G.711-ALaw-64k,H.264,H.264-1,H.264-0,MPEG4,H.263,H.263plus,H.261,G.722-64k,G.722.1-24k,G.722.1-32k,SILK-16,SpeexIETFWide-20.6k,SpeexWB,SpeexWide-20.6k,LPC-10,MS-GSM,MS-IMA-ADPCM,SILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIETFNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.38,iLBC-13k3,iLBC-15k2,UserInput/RFC2833,NamedSignalEvent,RFC4175_YCbCr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ
	  0:00.061                       sipcmd OpalCon SetPhase from UninitialisedPhase to SetUpPhase for Call[C5416e2a61]-EP<sip>[480d881a-1044-e911-9313-02d4f54aef8c]
	  0:00.061                       sipcmd SIP     Could not write to sip:[email protected] -
	  0:00.061                       sipcmd OpalCon SetPhase from SetUpPhase to ReleasingPhase for Call[C5416e2a61]-EP<sip>[480d881a-1044-e911-9313-02d4f54aef8c]
	  0:00.061                       sipcmd OpalCon Call end reason for Call[C5416e2a61]-EP<sip>[480d881a-1044-e911-9313-02d4f54aef8c] set to EndedByTransportFail
	  0:00.061                       sipcmd OpalCon Releasing asynchronously Call[C5416e2a61]-EP<sip>[480d881a-1044-e911-9313-02d4f54aef8c]
	  0:00.062                       sipcmd PWLib   File handle high water mark set: 22 Thread unblock pipe
	  0:00.062                       sipcmd PTLib   Created thread 0x55d893dd4570 OnRelease
	  0:00.062                       sipcmd PTLib   Thread high water mark set: 6
	  0:00.062                       sipcmd OpalCon SetPhase from SetUpPhase to ReleasingPhase for Call[C5416e2a61]-EP<local>[Lfe18107c2]
	  0:00.062                       sipcmd OpalCon Call end reason for Call[C5416e2a61]-EP<local>[Lfe18107c2] set to EndedByNoAccept
	  0:00.063                       sipcmd OpalCon Releasing asynchronously Call[C5416e2a61]-EP<local>[Lfe18107c2]
	  0:00.063                       sipcmd PWLib   File handle high water mark set: 24 Thread unblock pipe
	  0:00.063                       sipcmd PTLib   Created thread 0x55d893dd4820 OnRelease
	  0:00.063                       sipcmd PTLib   Thread high water mark set: 7
	  0:00.064                       sipcmd Call    Clearing Call[C5416e2a61] reason=EndedByTransportFail
	  0:00.064                       sipcmd OpalCon Already released Call[C5416e2a61]-EP<sip>[480d881a-1044-e911-9313-02d4f54aef8c]
	Call setup to sip:[email protected] failed
	Problem running command sequence ("c0004;ws30;vclip"):

	TestPhone::Main: shutting down
	  0:00.064                       sipcmd OpalMan Clearing all calls and waiting, primary thread.
	  0:00.064                       sipcmd Call    Clearing Call[C5416e2a61] reason=EndedByLocalUser
	  0:00.065      OnRelease:...31497ae700 PTLib   Started thread 0x55d893dd4820 (24060) OnRelease:0x7f31497ae700
	  0:00.065      OnRelease:...31497ae700 OpalCon OnReleased Call[C5416e2a61]-EP<local>[Lfe18107c2]
	  0:00.065      OnRelease:...31497ae700 OpalCon Media streams closed.
	  0:00.065      OnRelease:...31497ae700 OpalEP  OnReleased Call[C5416e2a61]-EP<local>[Lfe18107c2]
	OnReleased: reason: EndedByNoAccept
	  0:00.065      OnRelease:...31497ae700 OpalMan OnReleased Call[C5416e2a61]-EP<local>[Lfe18107c2]
	  0:00.066      OnRelease:...31497ae700 Call    OnReleased Call[C5416e2a61]-EP<local>[Lfe18107c2]
	  0:00.066      OnRelease:...31497ae700 OpalCon Already released Call[C5416e2a61]-EP<sip>[480d881a-1044-e911-9313-02d4f54aef8c]
	  0:00.066      OnRelease:...31497ae700 OpalCon SetPhase from ReleasingPhase to ReleasedPhase for Call[C5416e2a61]-EP<local>[Lfe18107c2]
	  0:00.066      OnRelease:...31497ae700 OpalCon Connection Call[C5416e2a61]-EP<local>[Lfe18107c2] released
			Initial Time: Wed, 13 Mar 2019 11:12:36 -04:00
			  SetUpPhase: 0.000
		 ProceedingPhase: N/A
		   AlertingPhase: N/A
		  ConnectedPhase: N/A
		EstablishedPhase: N/A
		 ForwardingPhase: N/A
		  ReleasingPhase: 0.011
		   ReleasedPhase: 0.015
		 Call end reason: EndedByNoAccept

	  0:00.067      OnRelease:...31497ae700 OpalCon OnRelease thread completed for Call[C5416e2a61]-EP<local>[Lfe18107c2]
	  0:00.067      OnRelease:...31497ae700 PWLib   File handle high water mark set: 25 PTextFile
	  0:00.067      OnRelease:...31497ae700 PTLib   Thread ended: name="OnRelease:0x7f31497ae700", real=0.002, kernel=0.000 (0%), user=0.000 (0%), both=0.000 (0%)
	  0:00.067      OnRelease:...3154078700 PTLib   Started thread 0x55d893dd4570 (24059) OnRelease:0x7f3154078700
	  0:00.068      OnRelease:...3154078700 SIP     OnReleased: Call[C5416e2a61]-EP<sip>[480d881a-1044-e911-9313-02d4f54aef8c]
	  0:00.068      OnRelease:...3154078700 OpalCon OnReleased Call[C5416e2a61]-EP<sip>[480d881a-1044-e911-9313-02d4f54aef8c]
	  0:00.068      OnRelease:...3154078700 OpalCon Media streams closed.
	  0:00.068      OnRelease:...3154078700 OpalEP  OnReleased Call[C5416e2a61]-EP<sip>[480d881a-1044-e911-9313-02d4f54aef8c]
	OnReleased: reason: EndedByTransportFail
	  0:00.068      OnRelease:...3154078700 OpalMan OnReleased Call[C5416e2a61]-EP<sip>[480d881a-1044-e911-9313-02d4f54aef8c]
	  0:00.069      OnRelease:...3154078700 Call    OnReleased Call[C5416e2a61]-EP<sip>[480d881a-1044-e911-9313-02d4f54aef8c]
	OnClearedCall
	  0:00.069      OnRelease:...3154078700 OpalCon SetPhase from ReleasingPhase to ReleasedPhase for Call[C5416e2a61]-EP<sip>[480d881a-1044-e911-9313-02d4f54aef8c]
	  0:00.069      OnRelease:...3154078700 OpalCon Connection Call[C5416e2a61]-EP<sip>[480d881a-1044-e911-9313-02d4f54aef8c] released
			Initial Time: Wed, 13 Mar 2019 11:12:36 -04:00
			  SetUpPhase: 0.009
		 ProceedingPhase: N/A
		   AlertingPhase: N/A
		  ConnectedPhase: N/A
		EstablishedPhase: N/A
		 ForwardingPhase: N/A
		  ReleasingPhase: 0.009
		   ReleasedPhase: 0.017
		 Call end reason: EndedByTransportFail

	  0:00.070      OnRelease:...3154078700 OpalCon OnRelease thread completed for Call[C5416e2a61]-EP<sip>[480d881a-1044-e911-9313-02d4f54aef8c]
	  0:00.070      OnRelease:...3154078700 PTLib   Thread ended: name="OnRelease:0x7f3154078700", real=0.003, kernel=0.000 (0%), user=0.000 (0%), both=0.000 (0%)
	  0:02.050      Opal Garba...3168f76700 SIP     Deleting connection.
	  0:02.050      Opal Garba...3168f76700 SIP     Setting new transport for destination ""
	  0:02.050      Opal Garba...3168f76700 Opal    Transport clean up on termination
	  0:02.050      Opal Garba...3168f76700 Opal    Transport Close
	  0:02.050      Opal Garba...3168f76700 Opal    Transport clean up on termination
	  0:02.050      Opal Garba...3168f76700 Opal    Transport Close
	  0:02.050      Opal Garba...3168f76700 Opal    Deleted transport udp$127.0.0.1:5060<if=udp$*:5060>
	  0:02.050      Opal Garba...3168f76700 PTLib   Destroying read/write mutex 0x55d893dcb340
	  0:02.050      Housekeepe...3168ef4700 PTLib   Destroyed thread 0x55d893dd4820 OnRelease:0x7f31497ae700(id = 0)
	  0:02.050      Housekeepe...3168ef4700 PTLib   Destroyed thread 0x55d893dd4570 OnRelease:0x7f3154078700(id = 0)
	  0:02.050      Opal Garba...3168f76700 OpalCon Connection Call[C5416e2a61]-EP<sip>[480d881a-1044-e911-9313-02d4f54aef8c] destroyed.
	  0:02.050      Opal Garba...3168f76700 PTLib   Destroying read/write mutex 0x55d893dc6f30
	~LocalConnection
	  0:02.050      Opal Garba...3168f76700 LocalCon        Deleted connection.
	  0:02.050      Opal Garba...3168f76700 OpalCon Connection Call[C5416e2a61]-EP<local>[Lfe18107c2] destroyed.
	  0:02.051      Opal Garba...3168f76700 PTLib   Destroying read/write mutex 0x55d893dc03a0
	  0:03.051      Opal Garba...3168f76700 Call    Destroyed Call[C5416e2a61]
	  0:03.051      Opal Garba...3168f76700 PTLib   Destroying read/write mutex 0x55d893dbfdc0
	  0:03.051                       sipcmd OpalMan All calls cleared.
	TestPhone::Main: exiting...
	Exiting...
	~Manager
	  0:03.051                       sipcmd OpalMan Shutting down endpoints.
	  0:03.051                       sipcmd OpalMan Clearing all calls and waiting, primary thread.
	  0:04.051                       sipcmd OpalMan All calls cleared.
	  0:04.051                       sipcmd SIP     Shutting down.
	  0:04.052                       sipcmd SIP     Changing REGISTER handler from Unavailable to Unsubscribing, target=sip:[email protected], id=6636871a-1044-e911-9313-02d4f54aef8c@ip-172-26-2-113
	  0:04.052                       sipcmd SIP     Not retrying REGISTER due to error response 1 Transport Error
	  0:04.052                       sipcmd SIP     Changing REGISTER handler from Unsubscribing to Unsubscribed, target=sip:[email protected], id=6636871a-1044-e911-9313-02d4f54aef8c@ip-172-26-2-113
	  0:04.152                       sipcmd OpalEP  sip endpoint shutting down.
	  0:04.152                       sipcmd Listen  Stopping listening thread on udp$*:5060
	  0:04.153      Opal Liste...3168eb3700 Listen  UDP read error.
	  0:04.153      Opal Liste...3168eb3700 PWLib   File handle low water mark set: 18 PTextFile
	  0:04.153      Opal Liste...3168eb3700 PTLib   Thread ended: name="Opal Listener:0x7f3168eb3700", real=4.105, kernel=0.000 (0%), user=0.000 (0%), both=0.000 (0%)
	  0:04.163                       sipcmd PTLib   Destroyed thread 0x55d893db9c80 Opal Listener:0x7f3168eb3700(id = 7f3168eb3700)
	  0:04.163                       sipcmd OpalEP  local endpoint shutting down.
	  0:04.163                       sipcmd PTLib   Destroying read/write mutex 0x55d893db46b8
	  0:04.163                       sipcmd PTLib   Destroying read/write mutex 0x55d893db4438
	  0:04.163                       sipcmd Opal    Transport clean up on termination
	  0:04.163                       sipcmd Opal    Transport Close
	  0:04.163                       sipcmd Opal    Transport clean up on termination
	  0:04.163                       sipcmd Opal    Transport Close
	  0:04.163                       sipcmd Opal    Deleted transport udp$127.0.0.1:5060<if=udp$*:5060>
	  0:04.163                       sipcmd IfaceMon        Awaiting thread termination
	  0:04.163      Network In...3168f35700 IfaceMon        Finished interface monitor thread.
	  0:04.163      Network In...3168f35700 PTLib   Thread ended: name="Network Interface Monitor:0x7f3168f35700", real=4.116, kernel=0.000 (0%), user=0.000 (0%), both=0.000 (0%)
	  0:04.163                       sipcmd PTLib   Destroyed thread 0x55d893db4910 Network Interface Monitor:0x7f3168f35700(id = 7f3168f35700)
	  0:04.164                       sipcmd PTLib   Destroying read/write mutex 0x55d893dbada0
	  0:04.164                       sipcmd PTLib   Destroying read/write mutex 0x55d893dbd2f0
	  0:04.164                       sipcmd SIP     Destroyed REGISTER handler for sip:[email protected]
	  0:04.164                       sipcmd PTLib   Destroying read/write mutex 0x55d893dbb580
	  0:04.164                       sipcmd OpalEP  sip endpoint destroyed.
	  0:04.165                       sipcmd LocalEP Deleted endpoint.
	  0:04.165                       sipcmd OpalEP  local endpoint destroyed.
	  0:04.165      Opal Garba...3168f76700 PWLib   File handle low water mark set: 8 PTextFile
	  0:04.165      Opal Garba...3168f76700 PTLib   Thread ended: name="Opal Garbage:0x7f3168f76700", real=4.118, kernel=0.000 (0%), user=0.000 (0%), both=0.000 (0%)
	  0:04.165                       sipcmd PTLib   Destroyed thread 0x55d893db36e0 Opal Garbage:0x7f3168f76700(id = 7f3168f76700)
	  0:04.165                       sipcmd OpalMan Deleted manager.
	  0:04.166                       sipcmd PTLib   Destroying read/write mutex 0x55d893dae3e8
	  0:04.166                       sipcmd OpalPlugin      Using default handler for plugin codec spandsp_ptplugin
	  0:04.166                       sipcmd OpalPlugin      Using default handler for plugin codec mpeg4_ffmpeg_ptplugin
	  0:04.166                       sipcmd OpalPlugin      Using default handler for plugin codec h263_ffmpeg_ptplugin
	  0:04.167                       sipcmd OpalPlugin      Using default handler for plugin codec theora_ptplugin
	  0:04.167                       sipcmd OpalPlugin      Using default handler for plugin codec h264_x264_ptplugin
	  0:04.167                       sipcmd OpalPlugin      Using default handler for plugin codec h261_vic_ptplugin
	  0:04.167                       sipcmd OpalPlugin      Using default handler for plugin codec ima_adpcm_ptplugin
	  0:04.167                       sipcmd OpalPlugin      Using default handler for plugin codec g7221_ptplugin
	  0:04.167                       sipcmd OpalPlugin      Using default handler for plugin codec gsm0610_ptplugin
	  0:04.167                       sipcmd OpalPlugin      Using default handler for plugin codec speex_ptplugin
	  0:04.168                       sipcmd OpalPlugin      Using default handler for plugin codec lpc10_ptplugin
	  0:04.168                       sipcmd OpalPlugin      Using default handler for plugin codec g722_ptplugin
	  0:04.168                       sipcmd OpalPlugin      Using default handler for plugin codec gsmamrcodec_ptplugin
	  0:04.168                       sipcmd OpalPlugin      Using default handler for plugin codec silk_ptplugin
	  0:04.168                       sipcmd OpalPlugin      Using default handler for plugin codec g726_ptplugin
	  0:04.168                       sipcmd OpalPlugin      Using default handler for plugin codec iLBC_ptplugin
	  0:04.168                       sipcmd OpalPlugin      Using default handler for plugin codec g7222_ptplugin
	  0:04.168                       sipcmd UDLL    Closing /usr/lib/opal-3.10.10/codecs/audio/g7222_ptplugin.so
	  0:04.168                       sipcmd UDLL    Closing /usr/lib/opal-3.10.10/codecs/audio/iLBC_ptplugin.so
	  0:04.169                       sipcmd UDLL    Closing /usr/lib/opal-3.10.10/codecs/audio/g726_ptplugin.so
	  0:04.169                       sipcmd UDLL    Closing /usr/lib/opal-3.10.10/codecs/audio/silk_ptplugin.so
	  0:04.169                       sipcmd UDLL    Closing /usr/lib/opal-3.10.10/codecs/audio/gsmamrcodec_ptplugin.so
	  0:04.169                       sipcmd UDLL    Closing /usr/lib/opal-3.10.10/codecs/audio/g722_ptplugin.so
	  0:04.169                       sipcmd UDLL    Closing /usr/lib/opal-3.10.10/codecs/audio/lpc10_ptplugin.so
	  0:04.169                       sipcmd UDLL    Closing /usr/lib/opal-3.10.10/codecs/audio/speex_ptplugin.so
	  0:04.169                       sipcmd UDLL    Closing /usr/lib/opal-3.10.10/codecs/audio/gsm0610_ptplugin.so
	  0:04.169                       sipcmd UDLL    Closing /usr/lib/opal-3.10.10/codecs/audio/g7221_ptplugin.so
	  0:04.169                       sipcmd UDLL    Closing /usr/lib/opal-3.10.10/codecs/audio/ima_adpcm_ptplugin.so
	  0:04.170                       sipcmd UDLL    Closing /usr/lib/opal-3.10.10/codecs/video/h261_vic_ptplugin.so
	  0:04.170                       sipcmd UDLL    Closing /usr/lib/opal-3.10.10/codecs/video/h264_x264_ptplugin.so
	  0:04.170                       sipcmd UDLL    Closing /usr/lib/opal-3.10.10/codecs/video/theora_ptplugin.so
	  0:04.170                       sipcmd UDLL    Closing /usr/lib/opal-3.10.10/codecs/video/h263_ffmpeg_ptplugin.so
	  0:04.170                       sipcmd UDLL    Closing /usr/lib/opal-3.10.10/codecs/video/mpeg4_ffmpeg_ptplugin.so
	  0:04.170                       sipcmd UDLL    Closing /usr/lib/opal-3.10.10/fax/spandsp_ptplugin.so
	  0:04.170                       sipcmd UDLL    Closing /usr/lib/ptlib-2.10.11/devices/sound/oss_pwplugin.so
	  0:04.170                       sipcmd UDLL    Closing /usr/lib/ptlib-2.10.11/devices/sound/alsa_pwplugin.so
	  0:04.170                       sipcmd UDLL    Closing /usr/lib/ptlib-2.10.11/devices/sound/pulse_pwplugin.so
	  0:04.171                       sipcmd UDLL    Closing /usr/lib/ptlib-2.10.11/devices/videoinput/v4l2_pwplugin.so
	  0:14.166      Housekeepe...3168ef4700 Housekeeping thread ended
	  0:14.166      Housekeepe...3168ef4700 PWLib   File handle low water mark set: 6 PTextFile
	  0:14.166      Housekeepe...3168ef4700 PTLib   Thread ended: name="Housekeeper:0x7f3168ef4700", real=14.120, kernel=0.000 (0%), user=0.000 (0%), both=0.000 (0%)
	  0:14.173                       sipcmd PTLib   Destroyed thread 0x55d893db4bc0 Housekeeper:0x7f3168ef4700(id = 7f3168ef4700)

Any idea what it is the issue is here?

Any VoIP provider it works with?

Hello,
this would help a lot - is there ANY public VoIP service provider that sipcmd is known to work with? If yes - would it be possible to give an explicit example ? Thnks

permissive build on Ubuntu 12.10 server

Hey, Tuomo !

Ubuntu 12.10 x86_64
gcc 4.7.2
libopal  3.10.4
libpt 2.10.4

sipcmd will compile only with -fpermissive which, i suppose, not valid behavior

Unknown option -a

Hi,
it seems that option -a is out of function, isn't it?
Can you debug this pls.

Thanks in advance.

Update documentation with simple exmaples

Update docs to ensure all options are correct and include some simple examples

[Feature Request] Call next number if remote party not answering

Or, if this is not possible, use diffrent exit codes that can be used by linux shell scripts...
this would be very nice.

Cheers!

License question

@tmakkonen hi! Could you please choose license for project and i will add respective PR?

EndedByQ931Cause

hi
Where is my fault?
I am listening udpand asterisk by tshark. No data from 127.0.0.1

root@baturorkun:/usr/local/src/sipcmd# ./sipcmd -P sip -u 661 -c abc123 -w 127.0.0.1 -x "c333"
Starting sipcmd
in debug mode
Manager
Init
initialising SIP endpoint...
TestChanAudio
TestChanAudio
Listening for SIP signalling on 0.0.0.0:5060
SIP listener up
registered as sip:[email protected]
Created LocalEndPoint
Main

Call

TestPhone::Main: calling "333" using gateway "127.0.0.1" at Thu Nov 10 09:52:20 2016

Setting up a call to: sip:[email protected]
LocalEndpoint::MakeConnection
LocalEndpointCreateConnection
LocalConnection
OnIncomingConnection: token=L769f2f3e2
OnReleased: reason: EndedByNoAccept
Call setup to sip:[email protected] failed
Problem running command sequence ("c333"):

TestPhone::Main: shutting down
OnReleased: reason: EndedByQ931Cause
OnClearedCall
~LocalConnection
TestPhone::Main: exiting...
Exiting...
~Manager

Some dependencies are no longer available in Ubuntu 14.10.

// ,

➜ sipcmd git:(master) sudo apt-get install opal-dev ptlib-dev
[sudo] password for v6:
Reading package lists... Done
Building dependency tree
Reading state information... Done
E: Unable to locate package opal-dev
E: Unable to locate package ptlib-dev

Point-to-point configuration?

Is it possible to configure sipcmd to connect against another sipcmd instance running on the local host, without connections to server/proxy? What should I specify as 'c' - command argument?

playback of .wav file sounds distorted over phone

i've hooked up sipcmd with text2wave from festival, but unfortunately the sound quality is miserable / distorted.

text2wave generates a wav-file that sounds fine, when played on the computer and it fits your specs: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz

Could you please specify opal/ptlib versions which are known to work?

I have troubles compiling sipcmd on NixOS. We are using opal-3.6.8 and ptlib-2.6.7 at the moment

FAIL - Ubuntu 16.04.3

Installation fail on Ubuntu 16.04.3. How to fix?

root@ubuntu-s-1vcpu-1gb-ams2-01:~# apt-get install libopal-dev
root@ubuntu-s-1vcpu-1gb-ams2-01:~/sipcmd# make
g++ -c -Wall  src/main.cpp -o src/main.o -I/usr/include/opal -I/usr/include/ptlib -Isrc/ -g -DDEBUG
In file included from /usr/include/opal/h323/h323.h:39:0,
                 from src/includes.h:25,
                 from src/main.h:24,
                 from src/main.cpp:24:
/usr/include/opal/h323/h323ep.h:148:30: warning: converting to non-pointer type 'unsigned int' from NULL [-Wconversion-null]
       unsigned int options = NULL,      ///<  options to pass to conneciton
                              ^
src/main.cpp: In member function 'bool Manager::SendDTMF(const PString&)':
src/main.cpp:446:18: warning: comparison between signed and unsigned integer expressions [-Wsign-compare]
         for (; i < dtmf.GetSize() - 1; i++) {
                  ^
src/main.cpp:459:17: warning: comparison between signed and unsigned integer expressions [-Wsign-compare]
         ok = (i == dtmf.GetSize() - 1 ? true : false);
                 ^
g++ -c -Wall  src/commands.cpp -o src/commands.o -I/usr/include/opal -I/usr/include/ptlib -Isrc/ -g -DDEBUG
In file included from /usr/include/opal/h323/h323.h:39:0,
                 from src/includes.h:25,
                 from src/main.h:24,
                 from src/state.h:25,
                 from src/commands.cpp:29:
/usr/include/opal/h323/h323ep.h:148:30: warning: converting to non-pointer type 'unsigned int' from NULL [-Wconversion-null]
       unsigned int options = NULL,      ///<  options to pass to conneciton
                              ^
src/commands.cpp: In member function 'virtual bool Wait::ParseCommand(const char**, std::vector<Command*>&)':
src/commands.cpp:439:30: warning: format '%u' expects argument of type 'unsigned int*', but argument 3 has type 'size_t* {aka long unsigned int*}' [-Wformat=]
   sscanf(*cmds, "%u", &millis);
                              ^
g++ -c -Wall  src/channels.cpp -o src/channels.o -I/usr/include/opal -I/usr/include/ptlib -Isrc/ -g -DDEBUG
In file included from /usr/include/opal/h323/h323.h:39:0,
                 from src/includes.h:25,
                 from src/channels.h:25,
                 from src/channels.cpp:27:
/usr/include/opal/h323/h323ep.h:148:30: warning: converting to non-pointer type 'unsigned int' from NULL [-Wconversion-null]
       unsigned int options = NULL,      ///<  options to pass to conneciton
                              ^
g++ src/main.o src/commands.o src/channels.o -o sipcmd -lopal -lpt

Could not write to sip:[email protected]

I've been trying to follow the examples from readme but unfortunately trying to run something as simple as:

./sipcmd -P sip -u $login -c $pwd -w @gate -x "cXXXXXX;ws2000;h" -o /tmp/sipcmd.log

fails. I'm not sure what's the reason since why sipcmd can't write to sip... the relevant part from sipcmd.log looks like this:

  0:00.187                   sipcmd OpalUDP Started connect to 213.218.117.66:5060
  0:00.220                   sipcmd Call    GetMediaFormats for Call[C53886b6b1]-EP<sip>[c0513abe-b5cc-e411-8904-0800273c7ef6]
    G.722.2,GSM-AMR,iLBC,GSM-06.10,G.726-40k,G.726-32k,G.726-24k,G.726-16k,G.711-uLaw-64k,G.711-ALaw-64k,H.264,H.264-1,H.264-0,MPEG4,H.263,H.263plus,H.261,G.722-64k,G.722.1-24k,G.722.1-32k,SILK-16,SpeexIETFWide-20.6k,SpeexWB,SpeexWide-20.6k,LPC-10,MS-GSM,MS-IMA-ADPCM,SILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIETFNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.38,iLBC-13k3,iLBC-15k2,UserInput/RFC2833,NamedSignalEvent,RFC4175_YCbCr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ
  0:00.222                   sipcmd SIP Local media formats set:
    G.722.2,GSM-AMR,iLBC,GSM-06.10,G.726-40k,G.726-32k,G.726-24k,G.726-16k,G.711-uLaw-64k,G.711-ALaw-64k,H.264,H.264-1,H.264-0,MPEG4,H.263,H.263plus,H.261,G.722-64k,G.722.1-24k,G.722.1-32k,SILK-16,SpeexIETFWide-20.6k,SpeexWB,SpeexWide-20.6k,LPC-10,MS-GSM,MS-IMA-ADPCM,SILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIETFNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.38,iLBC-13k3,iLBC-15k2,UserInput/RFC2833,NamedSignalEvent,RFC4175_YCbCr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ
  0:00.223                   sipcmd SIP Remote media formats set:
    G.722.2,GSM-AMR,iLBC,GSM-06.10,G.726-40k,G.726-32k,G.726-24k,G.726-16k,G.711-uLaw-64k,G.711-ALaw-64k,H.264,H.264-1,H.264-0,MPEG4,H.263,H.263plus,H.261,G.722-64k,G.722.1-24k,G.722.1-32k,SILK-16,SpeexIETFWide-20.6k,SpeexWB,SpeexWide-20.6k,LPC-10,MS-GSM,MS-IMA-ADPCM,SILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIETFNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.38,iLBC-13k3,iLBC-15k2,UserInput/RFC2833,NamedSignalEvent,RFC4175_YCbCr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ
  0:00.226                   sipcmd OpalCon SetPhase from UninitialisedPhase to SetUpPhase for Call[C53886b6b1]-EP<sip>[c0513abe-b5cc-e411-8904-0800273c7ef6]
  0:00.230                   sipcmd SIP Could not write to sip:[email protected] - 
  0:00.230                   sipcmd OpalCon SetPhase from SetUpPhase to ReleasingPhase for Call[C53886b6b1]-EP<sip>[c0513abe-b5cc-e411-8904-0800273c7ef6]
  0:00.231                   sipcmd OpalCon Call end reason for Call[C53886b6b1]-EP<sip>[c0513abe-b5cc-e411-8904-0800273c7ef6] set to EndedByTransportFail
  0:00.231                   sipcmd OpalCon Releasing asynchronously Call[C53886b6b1]-EP<sip>[c0513abe-b5cc-e411-8904-0800273c7ef6]
  0:00.232                   sipcmd PWLib   File handle high water mark set: 23 Thread unblock pipe
  0:00.232                   sipcmd PTLib   Created thread 0x25996e0 OnRelease
  0:00.232                   sipcmd PTLib   Thread high water mark set: 6
  0:00.232                   sipcmd OpalCon SetPhase from SetUpPhase to ReleasingPhase for Call[C53886b6b1]-EP<local>[L7fa9f6312]
  0:00.232                   sipcmd OpalCon Call end reason for Call[C53886b6b1]-EP<local>[L7fa9f6312] set to EndedByNoAccept
  0:00.232                   sipcmd OpalCon Releasing asynchronously Call[C53886b6b1]-EP<local>[L7fa9f6312]
  0:00.232                   sipcmd PWLib   File handle high water mark set: 25 Thread unblock pipe
  0:00.233                   sipcmd PTLib   Created thread 0x259bc10 OnRelease
  0:00.233                   sipcmd PTLib   Thread high water mark set: 7
  0:00.233                   sipcmd Call    Clearing Call[C53886b6b1] reason=EndedByTransportFail

The accounts are working since I've made calls using them with other apps. Could someone share their wisdom with me on the topic?

sipcmd can't make call

Hello

I'm trying to make test call but no success
sipcmd from github
opal-3.10.2
ptlib-2.10.2
centos 6, ubuntu 12.04, the same story

sipcmd even not trying to connect
please help

./sipcmd -o /tmp/opal.log -p 12345 -P sip -x "c123;w1000;h" -w 192.168.101.10

Starting sipcmd
in debug mode
Manager
Init
0:00.024 Opal Garbage:0xb778fb70 PTLib Started thread 0x928d4b8 (22881) Opal Garbage:0xb778fb70
TestChanAudio
TestChanAudio
initialising SIP endpoint...
Created LocalEndPoint
Main

Call

TestPhone::Main: calling "123" using gateway "192.168.101.10" at Thu Nov 21 16:06:43 2013

Setting up a call to: sip:[email protected]
LocalEndpoint::MakeConnection
LocalEndpointCreateConnection
LocalConnection
OnIncomingConnection: token=Lf307adea2
OnReleased: reason: EndedByCallerAbort
OnClearedCall
Call setup to sip:[email protected] failed
Problem running command sequence ("c123;w1000;h"):

TestPhone::Main: shutting down
~LocalConnection
TestPhone::Main: exiting...
Exiting...
~Manager

/tmp/opal.log:

0:00.024 sipcmd Version 1.0.1 by Command line VoIP testphone on Unix Linux (2.6.32-358.14.1.el6.i686-i686) with PTLib (v2.10.2 (svn:25898)) at 2013/11/21 16:06:43.351
0:00.025 sipcmd OpalMan Attached endpoint with prefix sip
0:00.025 sipcmd OpalEP Created endpoint: sip
0:00.025 sipcmd PWLib File handle high water mark set: 8 PUDPSocket
0:00.025 sipcmd IfaceMon Initial interface list:
127.0.0.1 <00-00-00-00-00-00> (lo)
176.X.X.X.X <6C-62-6D-46-AB-D8> (eth0)
10.X.X.X <00-00-00-00-00-00> (tun0)
fe80::6e62:X <6C-62-6D-46-AB-D8> (eth0)
::1 <00-00-00-00-00-00> (lo)
2a01:X <6C-62-6D-46-AB-D8> (eth0)

0:00.025 sipcmd PTLIB Opened NetLink socket
0:00.025 sipcmd PWLib File handle high water mark set: 12 Thread unblock pipe
0:00.025 sipcmd PTLib Created thread 0x928bf08
0:00.025 sipcmd PTLib Thread high water mark set: 3
0:00.025 sipcmd PWLib File handle high water mark set: 14 Thread unblock pipe
0:00.025 sipcmd PTLib Created thread 0x9290880 Housekeeper
0:00.025 sipcmd PTLib Thread high water mark set: 4
0:00.025 Network In...0xb774eb70 PTLib Started thread 0x928bf08 (22882) Network Interface Monitor:0xb774eb70
0:00.025 Network In...0xb774eb70 IfaceMon Started interface monitor thread.
0:00.025 Housekeeper:0xb77deb70 PTLib Started thread 0x9290880 (22883) Housekeeper:0xb77deb70
0:00.025 sipcmd OpalMan Attached endpoint with prefix sips
0:00.025 sipcmd SIP Created endpoint.
0:00.025 sipcmd OpalMan Added route "local:.=sip:"
0:00.025 sipcmd OpalMan Added route "sip:.=local:"
0:00.026 sipcmd OpalMan Attached endpoint with prefix local
0:00.026 sipcmd OpalEP Created endpoint: local
0:00.026 sipcmd LocalEP Created endpoint.
0:00.026 sipcmd OpalMan Set up call from local:* to sip:[email protected]
0:00.026 sipcmd Call Created Call[Cd1c1dbc71]
0:00.026 sipcmd OpalMan Set up connection to "local:"
0:00.026 sipcmd OpalCon Created connection Call[Cd1c1dbc71]-EP[Lf307adea2]
0:00.026 sipcmd LocalCon Created connection with token "Lf307adea2"
0:00.026 sipcmd OpalCon SetPhase from UninitialisedPhase to SetUpPhase for Call[Cd1c1dbc71]-EP[Lf307adea2]
0:00.026 sipcmd OpalMan OnIncoming connection Call[Cd1c1dbc71]-EP[Lf307adea2]
0:00.026 sipcmd Call GetOtherPartyConnection Call[Cd1c1dbc71]-EP[Lf307adea2]
0:00.026 sipcmd OpalMan Searching for route "local:root sip:[email protected]"
0:00.026 sipcmd OpalMan Matched regex "^local:. .$" ("local:.")
0:00.026 sipcmd OpalMan Set up connection to "sip:[email protected]"
0:00.026 sipcmd OpalMan Searching for route "local:root sip:[email protected]"
0:00.026 sipcmd OpalMan Matched regex "^local:.* .$" ("local:.")
0:00.026 sipcmd OpalMan Searching for route "local:root sip:[email protected]"
0:00.026 sipcmd OpalMan Did not match regex "^sip:.* .$" ("sip:.")
0:00.026 sipcmd OpalMan Set up connection to "sip:[email protected]"
0:00.026 sipcmd OpalMan Searching for route "local:root sip:[email protected]"
0:00.026 sipcmd OpalMan Did not match regex "^sip:.* .$" ("sip:.")
0:00.026 sipcmd OpalMan Set up connection to "sip:[email protected]"
0:00.026 sipcmd OpalCon SetPhase from SetUpPhase to ReleasingPhase for Call[Cd1c1dbc71]-EP[Lf307adea2]
0:00.026 sipcmd OpalCon Releasing Call[Cd1c1dbc71]-EP[Lf307adea2]
0:00.026 sipcmd OpalCon Call end reason for Call[Cd1c1dbc71]-EP[Lf307adea2] set to EndedByCallerAbort
0:00.026 sipcmd OpalCon OnReleased Call[Cd1c1dbc71]-EP[Lf307adea2]
0:00.026 sipcmd OpalCon Media streams closed.
0:00.026 sipcmd OpalEP OnReleased Call[Cd1c1dbc71]-EP[Lf307adea2]
0:00.026 sipcmd OpalMan OnReleased Call[Cd1c1dbc71]-EP[Lf307adea2]
0:00.026 sipcmd Call OnReleased Call[Cd1c1dbc71]-EP[Lf307adea2]
0:00.026 sipcmd OpalCon SetPhase from ReleasingPhase to ReleasedPhase for Call[Cd1c1dbc71]-EP[Lf307adea2]
0:00.026 sipcmd OpalCon Connection Call[Cd1c1dbc71]-EP[Lf307adea2] released

BUG: does not build on ubuntu 12.04 LTS server with opal-dev and ptlib-dev installed.

Platform:

arif@khost:~/sipcmd$ uname -ar
Linux khost 3.2.0-32-generic #51 SMP Mon Oct 8 18:50:35 BDT 2012 x86_64 x86_64 x86_64    
GNU/Linux

Libopal version:

 libopal-dev    3.10.2~dfsg-0u OPAL library header files

LIbpt version

 libpt-dev      2.10.2~dfsg-0u Portable Tools Library development files

g++ version:

g++ (Ubuntu/Linaro 4.6.3-1ubuntu5) 4.6.3

clang++ version:

clang version 3.0-6ubuntu3 (tags/RELEASE_30/final) (based on LLVM 3.0)

g++ "make" output:

g++ -c -Wall src/main.cpp -o src/main.o -I/usr/include/opal -I/usr/include/ptlib -Isrc/ -g -DDEBUG
In file included from src/state.h:26:0,
                 from src/main.cpp:26:
src/channels.h: In constructor âTestChannel::TestChannel(OpalConnection&, TestChanAudio&)â:
src/channels.h:112:35: error: cast from âTestChannel*â to âunsigned intâ loses precision [-fpermissive]
src/channels.h: In destructor âvirtual TestChannel::~TestChannel()â:
src/channels.h:116:59: error: cast from âTestChannel*â to âunsigned intâ loses precision [-fpermissive]
src/main.cpp: In member function âbool Manager::SendDTMF(const PString&)â:
src/main.cpp:454:37: warning: comparison between signed and unsigned integer expressions [-Wsign-compare]
src/main.cpp:467:37: warning: comparison between signed and unsigned integer expressions [-Wsign-compare]
src/main.cpp: In member function âvoid RTPSession::SelectAudioFormat(RTPSession::Payload)â:
src/main.cpp:486:28: warning: unused variable âfmtâ [-Wunused-variable]
/usr/include/opal/opal/mediatype.h: At global scope:
/usr/include/opal/opal/mediatype.h:328:1: warning: âPFactoryLoader::T38PseudoRTP_Handler_loaderâ defined but not used [-Wunused-variable]
/usr/include/opal/rtp/rtp.h:1328:1: warning: âPFactoryLoader::RTP_Encoding_loaderâ defined but not used [-Wunused-variable]
/usr/include/opal/opal/pres_ent.h:717:1: warning: âPFactoryLoader::SIP_Presentity_loaderâ defined but not used [-Wunused-variable]
/usr/include/opal/opal/recording.h:194:1: warning: âPFactoryLoader::OpalWAVRecordManager_loaderâ defined but not used [-Wunused-variable]
/usr/include/opal/h323/h235auth.h:228:1: warning: âPFactoryLoader::H235AuthSimpleMD5_loaderâ defined but not used [-Wunused-variable]
/usr/include/opal/h323/h235auth.h:269:1: warning: âPFactoryLoader::H235AuthCAT_loaderâ defined but not used [-Wunused-variable]
/usr/include/opal/h323/h235auth.h:310:1: warning: âPFactoryLoader::H235AuthProcedure1_loaderâ defined but not used [-Wunused-variable]
make: *** [src/main.o] Error 1

clang++ make output: (Too long to post the whole thing:))

994 warnings and 2 errors generated.
make: *** [src/main.o] Error 1

segmentation fault on receive/send audio

On receive, for example:
Main

Answer

Answer: starting at Tue Feb 14 16:29:03 2017

OnIncomingConnection: token=d0eb5b9d-2ff1-e611-9bc7-005056a2792a
LocalEndpoint::MakeConnection
LocalEndpointCreateConnection
LocalConnection
Answer: connection established

Wait: waiting for 900ms

CreateMediaStream
TestChannel[ Call[Cf307adea1]-EP[L8b564a892] - 0x150b9a40 ]
OnOpenMediaStream
recording media from local
OnOpenMediaStream
streaming media to sip
OnOpenMediaStream
recording media from sip
CreateMediaStream
TestChannel[ Call[Cf307adea1]-EP[L8b564a892] - 0x150c7dc0 ]
OnOpenMediaStream
streaming media to local
OnEstablished
TestChannel::Write
Segmentation fault

Segmentation fault

Running this command will more likely crash with a segfault.
$ ./sipcmd -P sip -u 1183 -c ***** -w 192.168.20.240 -x "c1234;ws2000;d9876;w1000;h" -o ./opal.log

The opal.log has no hints for me, and the segfault will happen about 80% of the time.
Using Ubuntu 14.04 x86_64

Feature request : Change build to "scons".

It seems the author shares the same hatred towards autotools as me. I'm willing to change the build to scons.

http://www.scons.org/

Which is more powerful than autotools but very easy to comprehend. The best part is, the build scripts are actually python scripts. So no more weird syntax.

FreeVoipDeal can't call Chinese no. ?

I have FreeVoipDeal account, I can log on and call other counties such German, Romania, but why can't call Chinese? It is said: your call is restricted. Btw, I'm in China, before I can call my Chinese friendly without any problem.
Than you for your help.

where is RTP_UDP?

In file included from src/main.cpp:24:0:
src/main.h:76:35: error: expected class-name before ‘{’ token
class RTPSession : public RTP_UDP {

I have been looking for it and I don't find it. What library should provide it?

Can't get it compiled under CentOS

Please provide assistance to make it work under CentOS.
Thanks you!

sipcmd IVR invoke a cmd or script with DTMF received from caller?

For a home automation project I'd like sipcmd:

wait for a incoming call
play greet+prompt
receive DTMF digits + # or *, e.g; "12#"
invoke script, e.g. activate.pl "12#"
(re)open and play the audio file named or rewritten by the script
hangup

Where can I find an IVR example and in particular, a description of the x-command?

Help with configurations

I was looking for a simple cli sip client and foiund yours which seems to have all the functionalities that a i need.

I've download it and compiled with success but so far i had no success making calls to a landline and i have some doubts with the conifugrations.

In all other clients i've used i had configured two parameters:
server: ims.vodafone.pt
proxy: proxythomson.ims.vodafone.pt

can you help me map this parameters to the -w and -g options?

Regarding the user name should i put [email protected] or +3512340XXXXX only?

Thanks

Can't call using freevoipdeal.com

Hi.

I just create a new account with credit on freevoipdeal.com. I test on to my android phone using the native sip configuration, and work ok. But i can't make calls using sipcmd.
./sipcmd -P sip -u "myuser" -c "mypass" -w "sip.freevoipdeal.com:5060" -x "c0034999999999" -o log.log

I also make some test, putting and remove the port. And the execute options. Also, i change the destination phone including 00349999999 +349999999 a phone never ring.

No protocol specified
xcb_connection_has_error() returned true
Starting sipcmd
in debug mode
Manager
Init
initialising SIP endpoint...
TestChanAudio
TestChanAudio
Listening for SIP signalling on 0.0.0.0:5060
SIP listener up
registered as sip:[email protected]:5060
Created LocalEndPoint
Main

Call

TestPhone::Main: calling "00349999999" using gateway "sip.freevoipdeal.com:5060" at Sat Mar 15 13:33:03 2014

Setting up a call to: sip:[email protected]:5060
LocalEndpoint::MakeConnection
LocalEndpointCreateConnection
LocalConnection
OnIncomingConnection: token=Lcdb4886a2
connection set up to sip:[email protected]:5060
TestPhone::Main: calling "sip:[email protected]:5060" for 0
...
TestPhone::Main: calling "sip:[email protected]:5060" for 10
Problem running command sequence ("c00349999999"):
Call: Dial timed out
TestPhone::Main: shutting down
OnReleased: reason: EndedByLocalUser
OnReleased: reason: EndedByLocalUser
OnClearedCall
~LocalConnection
TestPhone::Main: exiting...
Exiting...
~Manager

Because log file is a little big, i upload into pastebin.
http://pastebin.com/Z2wuhtdE

Any help?

How to set ringing timeout?

Question: Is there anyway we can change sipcmd to ring a number for X seconds before giving up with a graceful message like "sipcmd gave up because remote party did not answer for X seconds"?

Commentary: I use sipcmd to activate a MWI (message waiting indicator) on IP phones. To activate the light, I use:

sipcmd -u xxxx -c xxxx -P sip -w 192.168.1.1 -x "c98;h"

Where is the IP phone that I want to turn on the MWI light.

So sipcmd makes a SIP call to the Cisco router (192.168.1.1), the router turns on the MWI light on the specific IP phone when the line 98 is rung by sipcmd. But, the router does not actually answer the call. Simply calling the number 98 is all that is required to turn the MWI light on. So the problem is that sipcmd just keeps ringing until the router eventually gives a fast busy tone (two minutes later). For the entire time that sipcmd is ringing, the MWI feature is locked on the router (you can't turn on or off any other extension MWI while sipcmd is ringing because sipcmd is keeping the router MWI line 98.. busy). And, the script that I use to turn MWI on is also held up for a full two minutes until sipcmd gives up after the router finally gives up (2 minutes later) terminates the ringing and unanswered call.

Can't disable debug output

Hi,

i tried to disable the debug output by commenting out the #DEBUG=-g -DDEBUG Option in the makefile but this didnt help.

i did run a make clean and then make again. the new bin file was a lot smaller but i getting debug messages all over again:

TestChannel[ Call[Cdea7631a1]-EP[Lec6d32502] - 0x7fec2c039530 ]
OnOpenMediaStream
streaming media to local
TestChannel::Write
TestChannel::Write
CreateMediaStream
TestChannel[ Call[Cdea7631a1]-EP[Lec6d32502] - 0x7fec2c052ba0 ]
OnOpenMediaStream
recording media from local
OnOpenMediaStream
streaming media to sip
TestChannel::Write
TestPhone::Main: calling "sip:[email protected]" for 0
TestChannel::Write
TestChannel::Write
TestChannel::Write
TestPhone::Main: calling "sip:[email protected]" for 1
TestChannel::Write
TestChannel::Write
TestPhone::Main: calling "sip:[email protected]" for 1
TestChannel::Write
TestChannel::Write
TestPhone::Main: calling "sip:[email protected]" for 1
TestChannel::Write
TestChannel::Write
TestChannel::Write
TestPhone::Main: calling "sip:[email protected]" for 1
TestChannel::Write
TestChannel::Write
TestChannel::Write
TestPhone::Main: calling "sip:[email protected]" for 1
TestChannel::Write
TestChannel::Write
TestPhone::Main: calling "sip:[email protected]" for 1
TestChannel::Write
TestChannel::Write
TestChannel::Write
TestPhone::Main: calling "sip:[email protected]" for 1
TestChannel::Write
TestChannel::Write
TestPhone::Main: calling "sip:[email protected]" for 1

Did i something wrong?

Not usable with Fritz!OS 6.80 anymore - possible to reduce codec list in SDP?

Hi,
do you see a way to disable all codecs except the codecs I want to use?

For me it is okay to use PCM16 and G.711a-law.
I would prefer an option to set sipcmd to use just these but I'm lost in C++

The Invite is quite huge due to the very long list:

INVITE sip:**[email protected] SIP/2.0
CSeq: 1 INVITE
v: SIP/2.0/UDP 169.254.80.70:5060;branch=z9hG4bK580685e7-f7f4-e611-8ef2-b827ebc64bf5;rport
User-Agent: text2sip/1.0.1
f: "Haustür" <sip:[email protected]>;tag=9a517ae7-f7f4-e611-8ef2-b827ebc64bf5
i: a87f7ae7-f7f4-e611-8ef2-b827ebc64bf5@loxberry
k: 100rel,replaces
Organization: LoxBerry Text2SIP
t: <sip:**[email protected]>
m: "Haustür" <sip:[email protected]:5060>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
l: 1669
c: application/sdp
Max-Forwards: 70

v=0
o=- 1487498425 1 IN IP4 169.254.80.70
s=text2sip/1.0.1
c=IN IP4 169.254.80.70
t=0 0
m=audio 5000 RTP/AVP 120 93 126 3 121 122 123 124 0 8 9 117 118 89 116 125 92 115 101 100
a=sendrecv
a=rtpmap:120 AMR-WB/16000/1
a=fmtp:120 octet-align=1
a=rtpmap:93 AMR/8000/1
a=rtpmap:126 iLBC/8000/1
a=fmtp:126 mode=20
a=rtpmap:3 gsm/8000/1
a=rtpmap:121 G726-40/8000/1
a=rtpmap:122 G726-32/8000/1
a=rtpmap:123 G726-24/8000/1
a=rtpmap:124 G726-16/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:9 G722/8000/1
a=rtpmap:117 G7221/16000/1
a=fmtp:117 bitrate=24000
a=rtpmap:118 G7221/16000/1
a=fmtp:118 bitrate=32000
a=rtpmap:89 SILK/16000/1
a=rtpmap:116 Speex/16000/1
a=rtpmap:125 lpc10/8000/1
a=rtpmap:92 SILK/8000/1
a=rtpmap:115 Speex/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=maxptime:30
m=video 5002 RTP/AVP 103 104 91 34 105 31 97 90
b=AS:240000
b=TIAS:240000000
a=sendrecv
a=rtpmap:103 H264/90000
a=fmtp:103 packetization-mode=1;max-br=240000;max-fs=6336;max-mbps=380160;profile-level-id=42801e
a=rtpmap:104 H264/90000
a=fmtp:104 max-br=240000;max-fs=6336;max-mbps=380160;profile-level-id=42801e
a=rtpmap:91 MP4V-ES/90000
a=fmtp:91 profile-level-id=5
a=rtpmap:34 H263/90000
a=fmtp:34 F=1;CIF=1;CIF16=1;CIF4=1;maxbr=3276;QCIF=1;SQCIF=1
a=rtpmap:105 H263-1998/90000
a=fmtp:105 D=1;F=1;I=1;J=1;CIF=1;CIF4=1;maxbr=3276;QCIF=1;SQCIF=1
a=rtpmap:31 h261/90000
a=fmtp:31 CIF=1;QCIF=1
a=rtpmap:97 raw/90000
a=fmtp:97 rate=90000;height=288;width=352;colorimetry=BT601-5;depth=8;sampling=YCbCr-4:2:0
a=rtpmap:90 theora/90000
a=fmtp:90 height=576;width=704

BR
Christian

URGENT - Segmentation fault

Following, dials on my phone, send/receives no DTMF on the receiver phone, call remain connected forever (does not disconnect after certain duration).


root@ubuntu-s-1vcpu-1gb-ams2-01:~/sipcmd# ./sipcmd -P sip -u xxx -c xxx -w sip1.nomado.eu -x "c003247171xxxx;ws3000;d**;w200;h"
Starting sipcmd
in debug mode
Manager
Init
initialising SIP endpoint...
TestChanAudio
TestChanAudio
Listening for SIP signalling on 0.0.0.0:5060
SIP listener up
registered as sip:[email protected]
Created LocalEndPoint
Main
## Call ##
TestPhone::Main: calling "003247171xxxx" using gateway "sip1.nomado.eu" at Thu Feb 22 08:17:58 2018

Setting up a call to: sip:[email protected]
LocalEndpoint::MakeConnection
LocalEndpointCreateConnection
LocalConnection
OnIncomingConnection: token=L513285842
connection set up to sip:[email protected]
TestPhone::Main: calling "sip:[email protected]" for 0
TestPhone::Main: calling "sip:[email protected]" for 0
TestPhone::Main: calling "sip:[email protected]" for 0
TestPhone::Main: calling "sip:[email protected]" for 0
TestPhone::Main: calling "sip:[email protected]" for 1
TestPhone::Main: calling "sip:[email protected]" for 1
TestPhone::Main: calling "sip:[email protected]" for 1
TestPhone::Main: calling "sip:[email protected]" for 1
TestPhone::Main: calling "sip:[email protected]" for 1
TestPhone::Main: calling "sip:[email protected]" for 1
TestPhone::Main: calling "sip:[email protected]" for 1
TestPhone::Main: calling "sip:[email protected]" for 1
TestPhone::Main: calling "sip:[email protected]" for 1
TestPhone::Main: calling "sip:[email protected]" for 1
TestPhone::Main: calling "sip:[email protected]" for 2
TestPhone::Main: calling "sip:[email protected]" for 2
TestPhone::Main: calling "sip:[email protected]" for 2
TestPhone::Main: calling "sip:[email protected]" for 2
TestPhone::Main: calling "sip:[email protected]" for 2
TestPhone::Main: calling "sip:[email protected]" for 2
TestPhone::Main: calling "sip:[email protected]" for 2
TestPhone::Main: calling "sip:[email protected]" for 2
TestPhone::Main: calling "sip:[email protected]" for 2
TestPhone::Main: calling "sip:[email protected]" for 2
TestPhone::Main: calling "sip:[email protected]" for 3
TestPhone::Main: calling "sip:[email protected]" for 3
TestPhone::Main: calling "sip:[email protected]" for 3
TestPhone::Main: calling "sip:[email protected]" for 3
TestPhone::Main: calling "sip:[email protected]" for 3
TestPhone::Main: calling "sip:[email protected]" for 3
TestPhone::Main: calling "sip:[email protected]" for 3
TestPhone::Main: calling "sip:[email protected]" for 3
TestPhone::Main: calling "sip:[email protected]" for 3
TestPhone::Main: calling "sip:[email protected]" for 3
TestPhone::Main: calling "sip:[email protected]" for 4
TestPhone::Main: calling "sip:[email protected]" for 4
TestPhone::Main: calling "sip:[email protected]" for 4
TestPhone::Main: calling "sip:[email protected]" for 4
TestPhone::Main: calling "sip:[email protected]" for 4
TestPhone::Main: calling "sip:[email protected]" for 4
OnOpenMediaStream
recording media from sip
CreateMediaStream
TestChannel[ Call[Cbb3c0ddc1]-EP<local>[L513285842] - 0x7f96bc03dd30 ]
OnOpenMediaStream
streaming media to local
TestChannel::Write
TestChannel::Write
CreateMediaStream
TestChannel[ Call[Cbb3c0ddc1]-EP<local>[L513285842] - 0x7f96bc0572d0 ]
OnOpenMediaStream
recording media from local
OnOpenMediaStream
streaming media to sip
Segmentation fault (core dumped)

hang up on aswer

Hello, is it possible to make a command sequence to make sipcmd hangup when the call is answered?

Mac build instructions

Please include build instructions for Mac

simple commandline example to start app properly

Hello, tmakkonen!

I try to use your app (sipcmd) and cant understand how to start. Is it possibly to show how to connect:

Direct call to 300 (192.168.1.1:5060), my num is set to 200. After 300 hook off send dtmf 1 2 3 ?

2.The same, but using EBNF ?

I try use registrar (kamailio-192.168.1.1) + 2 sip client with num 100,300. Between 100 <-> 300 conn OK.

./sipcmd -P sip -R "192.168.1.1;200;;" -x "c300"
According to wireshark no one packet has been sent ((

Thanks.

Can not record while playing audio

Using the record mode of the execute plan it is not possible to record an entire call sequence. This limits the functionality for "userland" test cases. It would be useful to have a mode that would record an entire call.

Need mac support

can you give the detail for installation and some working examples on mac system ( mac sierra )

H.323 calls fail in all possible configurations

First, great piece of software!

After hours of work, I have tried every possibile H.323 combination, but the error message "EndedByCallerAbort" keeps popping up

Sip (both with and without authentication (if you provider supports it)) works truly great.

The following is (almost standard) output from the H.323 call setup:

Setting up a call to: h323:00316xxxxxxxx@server
LocalEndpoint::MakeConnection
LocalEndpointCreateConnection
LocalConnection
OnIncomingConnection: token=L02cfcc502
OnReleased: reason: EndedByCallerAbort
OnClearedCall
Call setup to h323:00316xxxxxxxx@server failed
Problem running command sequence ("c00316xxxxxxxx;w200;d12345"):

Opal.log:

0:00.039 sipcmd OpalMan Attached endpoint with prefix h323
0:00.039 sipcmd OpalEP Created endpoint: h323
0:00.039 sipcmd OpalMan Attached endpoint with prefix h323s
0:00.039 sipcmd H323 Created endpoint.
0:00.039 sipcmd OpalMan Added route "pc:.=h323:"
0:00.039 sipcmd OpalMan Added route "h323:.=pc:"
0:00.040 sipcmd OpalMan Attached endpoint with prefix local
0:00.040 sipcmd OpalEP Created endpoint: local
0:00.040 sipcmd LocalEP Created endpoint.
0:00.040 sipcmd OpalMan Set up call from local:* to h323:00316xxxxxxxx@server
0:00.040 sipcmd Call Created Call[C7cb09e8b1]
0:00.040 sipcmd OpalMan Set up connection to "local:"
0:00.040 sipcmd OpalCon Created connection Call[C7cb09e8b1]-EP[L02cfcc502]
0:00.040 sipcmd LocalCon Created connection with token "L02cfcc502"
0:00.040 sipcmd Call GetOtherPartyConnection Call[C7cb09e8b1]-EP[L02cfcc502]
0:00.041 sipcmd OpalCon SetPhase from UninitialisedPhase to SetUpPhase for Call[C7cb09e8b1]-EP[L02cfcc502]
0:00.041 sipcmd OpalMan OnIncoming connection Call[C7cb09e8b1]-EP[L02cfcc502]
0:00.041 sipcmd Call GetOtherPartyConnection Call[C7cb09e8b1]-EP[L02cfcc502]
0:00.041 sipcmd Call GetOtherPartyConnection Call[C7cb09e8b1]-EP[L02cfcc502]
0:00.041 sipcmd OpalMan Searching for route "local:username h323:00316xxxxxxxx@server"
0:00.041 sipcmd OpalMan Did not match regex "^pc:. .$" ("pc:.")
0:00.041 sipcmd OpalMan Did not match regex "^h323:.* .$" ("h323:.")
0:00.041 sipcmd OpalMan Set up connection to "h323:00316xxxxxxxx@server"
0:00.041 sipcmd OpalCon SetPhase from SetUpPhase to ReleasingPhase for Call[C7cb09e8b1]-EP[L02cfcc502]
0:00.041 sipcmd OpalCon Releasing Call[C7cb09e8b1]-EP[L02cfcc502]
0:00.041 sipcmd OpalCon Call end reason for Call[C7cb09e8b1]-EP[L02cfcc502] set to EndedByCallerAbort
0:00.041 sipcmd OpalCon OnReleased Call[C7cb09e8b1]-EP[L02cfcc502]
0:00.041 sipcmd OpalCon Media streams closed.
0:00.041 sipcmd OpalEP OnReleased Call[C7cb09e8b1]-EP[L02cfcc502]
0:00.041 sipcmd OpalMan OnReleased Call[C7cb09e8b1]-EP[L02cfcc502]
0:00.041 sipcmd Call OnReleased Call[C7cb09e8b1]-EP[L02cfcc502]
0:00.041 sipcmd OpalCon SetPhase from ReleasingPhase to ReleasedPhase for Call[C7cb09e8b1]-EP[L02cfcc502]
0:00.041 sipcmd OpalCon Connection Call[C7cb09e8b1]-EP[L02cfcc502] released
Initial Time: Tue, 18 Mar 2014 23:48:29 +01:00
SetUpPhase: 0.000
ProceedingPhase: N/A
AlertingPhase: N/A
ConnectedPhase: N/A
EstablishedPhase: N/A
ForwardingPhase: N/A
ReleasingPhase: 0.000
ReleasedPhase: 0.000
Call end reason: EndedByCallerAbort

0:00.041 sipcmd Call Clearing Call[C7cb09e8b1] reason=EndedByCallerAbort
0:00.042 sipcmd OpalMan Clearing all calls and waiting, primary thread.
0:00.042 Opal Garbage:0x2a741700 PTLib Started thread 0x258e520 (6057) Opal Garbage:0x2a741700
0:02.042 Opal Garbage:0x2a741700 LocalCon Deleted connection.
0:02.042 Opal Garbage:0x2a741700 OpalCon Connection Call[C7cb09e8b1]-EP[L02cfcc502] destroyed.
0:02.043 Opal Garbage:0x2a741700 PWLib File handle high water mark set: 10 Thread unblock pipe
0:02.043 Opal Garbage:0x2a741700 PTLib Created thread 0x258bed0 Housekeeper
0:02.043 Opal Garbage:0x2a741700 PTLib No permission to set priority level 4
0:02.043 Opal Garbage:0x2a741700 PTLib Thread high water mark set: 3
0:02.043 Housekeeper:0x2a700700 PTLib Started thread 0x258bed0 (6058) Housekeeper:0x2a700700
0:03.043 Opal Garbage:0x2a741700 Call Destroyed Call[C7cb09e8b1]
0:03.043 sipcmd OpalMan All calls cleared.
0:03.043 sipcmd OpalMan Shutting down endpoints.
0:03.043 sipcmd OpalMan Clearing all calls and waiting, primary thread.
0:04.044 sipcmd OpalMan All calls cleared.
0:04.044 sipcmd OpalEP h323 endpoint shutting down.
0:04.044 sipcmd OpalEP local endpoint shutting down.
0:04.044 sipcmd OpalEP h323 endpoint destroyed.
0:04.044 sipcmd LocalEP Deleted endpoint.
0:04.044 sipcmd OpalEP local endpoint destroyed.
0:04.044 Opal Garbage:0x2a741700 PWLib File handle high water mark set: 13 PTextFile
0:04.044 Opal Garbage:0x2a741700 PTLib Thread ended: name="Opal Garbage:0x2a741700", real=4.002, kernel=0.000 (0%), user=0.000 (0%), both=0.000 (0%)
0:04.054 sipcmd PTLib Destroyed thread 0x258e520 Opal Garbage:0x2a741700(id = 7f932a741700)
0:04.054 sipcmd OpalMan Deleted manager.
0:04.055 sipcmd OpalPlugin Using default handler for plugin codec spandsp_ptplugin
0:04.055 sipcmd OpalPlugin Plugin Codec DLL mpeg4_ffmpeg_ptplugin contains no codec definitions
0:04.055 sipcmd OpalPlugin Using default handler for plugin codec theora_ptplugin
0:04.055 sipcmd OpalPlugin Using default handler for plugin codec h261_vic_ptplugin
0:04.055 sipcmd OpalPlugin Using default handler for plugin codec h263_ffmpeg_ptplugin
0:04.056 sipcmd OpalPlugin Using default handler for plugin codec gsmamrcodec_ptplugin
0:04.056 sipcmd OpalPlugin Using default handler for plugin codec lpc10_ptplugin
0:04.056 sipcmd OpalPlugin Using default handler for plugin codec g7221_ptplugin
0:04.056 sipcmd OpalPlugin Using default handler for plugin codec silk_ptplugin
0:04.056 sipcmd OpalPlugin Using default handler for plugin codec ima_adpcm_ptplugin
0:04.056 sipcmd OpalPlugin Using default handler for plugin codec speex_ptplugin
0:04.056 sipcmd OpalPlugin Using default handler for plugin codec g722_ptplugin
0:04.056 sipcmd OpalPlugin Using default handler for plugin codec g7222_ptplugin
0:04.056 sipcmd OpalPlugin Using default handler for plugin codec g726_ptplugin
0:04.056 sipcmd OpalPlugin Using default handler for plugin codec gsm0610_ptplugin
0:04.056 sipcmd UDLL Closing /usr/lib/opal-3.10.4/codecs/audio/gsm0610_ptplugin.so
0:04.056 sipcmd UDLL Closing /usr/lib/opal-3.10.4/codecs/audio/g726_ptplugin.so
0:04.056 sipcmd UDLL Closing /usr/lib/opal-3.10.4/codecs/audio/g7222_ptplugin.so
0:04.056 sipcmd UDLL Closing /usr/lib/opal-3.10.4/codecs/audio/g722_ptplugin.so
0:04.056 sipcmd UDLL Closing /usr/lib/opal-3.10.4/codecs/audio/speex_ptplugin.so
0:04.056 sipcmd UDLL Closing /usr/lib/opal-3.10.4/codecs/audio/ima_adpcm_ptplugin.so
0:04.057 sipcmd UDLL Closing /usr/lib/opal-3.10.4/codecs/audio/silk_ptplugin.so
0:04.057 sipcmd UDLL Closing /usr/lib/opal-3.10.4/codecs/audio/g7221_ptplugin.so
0:04.057 sipcmd UDLL Closing /usr/lib/opal-3.10.4/codecs/audio/lpc10_ptplugin.so
0:04.057 sipcmd UDLL Closing /usr/lib/opal-3.10.4/codecs/audio/gsmamrcodec_ptplugin.so
0:04.057 sipcmd UDLL Closing /usr/lib/opal-3.10.4/codecs/video/h263_ffmpeg_ptplugin.so
0:04.057 sipcmd UDLL Closing /usr/lib/opal-3.10.4/codecs/video/h261_vic_ptplugin.so
0:04.057 sipcmd UDLL Closing /usr/lib/opal-3.10.4/codecs/video/theora_ptplugin.so
0:04.057 sipcmd UDLL Closing /usr/lib/opal-3.10.4/codecs/video/mpeg4_ffmpeg_ptplugin.so
0:04.057 sipcmd UDLL Closing /usr/lib/opal-3.10.4/fax/spandsp_ptplugin.so
0:04.057 sipcmd UDLL Closing /usr/lib/ptlib-2.10.4/devices/sound/alsa_pwplugin.so
0:04.057 sipcmd UDLL Closing /usr/lib/ptlib-2.10.4/devices/sound/pulse_pwplugin.so
0:04.057 sipcmd UDLL Closing /usr/lib/ptlib-2.10.4/devices/sound/oss_pwplugin.so
0:04.057 sipcmd UDLL Closing /usr/lib/ptlib-2.10.4/devices/videoinput/v4l2_pwplugin.so
0:04.058 Housekeeper:0x2a700700 Housekeeping thread ended
0:04.058 Housekeeper:0x2a700700 PWLib File handle low water mark set: 6 PTextFile
0:04.058 Housekeeper:0x2a700700 PTLib Thread ended: name="Housekeeper:0x2a700700", real=2.015, kernel=0.000 (0%), user=0.000 (0%), both=0.000 (0%)
0:04.068 sipcmd PTLib Destroyed thread 0x258bed0 Housekeeper:0x2a700700(id = 7f932a700700)

Any clues?

Problem with connection state in sip protocol

Hello,
The state of the connection doesn't change when the remote answer or hangup.
State stays in CONNECTING.

Here is the parameters I used:

./sipcmd -P sip -u 0033972nnnn -c mypasswd -w sip3.ovh.fr -x "w250;c00336nnnnnn;vtest8.wav;ws500;h"

sipcmd is counting up to 10, called phone is ringing at about 5.
Whatever I answer or hang up on remote side, sipcmd complains with a Dial timeout error.

What's wrong please?

Question: Answering calls

Hi,

I'm trying to design an automated end to end call testing suite. I have managed to work out the calling part, but i'm having difficulty getting calls answered. I wondered if you could either provide a simple example of call answering or provide some more detail in the README.

Thanks in advance

Ed Holland

Audio Stream

Is it possible to use audio from alsa in real time?
Thanks

Error when using free2.voipgateway.org

I'am trying to use sipcmd with the free2.voipgateway.org server using the following commandline:

sipcmd -u 123546789 -c mypw -P sip -w free2.voipgateway.org -x "c987654321;ws3000;vtest.wav;h"

I always get the error "OnReleased: reason: EndedBySecurityDenial". What could be the problem? Here is the whole output:

Starting sipcmd
in debug mode
Manager
Init
initialising SIP endpoint...
Listening for SIP signalling on 0.0.0.0:TestChanAudio
TestChanAudio
5060
SIP listener up
registered as sip:[email protected]
Created LocalEndPoint
Main
## Call ##
TestPhone::Main: calling "987654321" using gateway "free2.voipgateway.org" at Wed Jan 13 16:51:04 2016

Setting up a call to: sip:[email protected]
LocalEndpoint::MakeConnection
LocalEndpointCreateConnection
LocalConnection
OnIncomingConnection: token=L5d00ab8b2
connection set up to sip:[email protected]
TestPhone::Main: calling "sip:[email protected]" for 0
OnReleased: reason: EndedBySecurityDenial
OnReleased: reason: EndedBySecurityDenial
OnClearedCall
## Wait: waiting for 500ms ##
Wait: wait done
## Voice audiofile=smokealert.wav ##
PlaybackAudioFile
PlaybackAudioFile: state 3
## Hangup ##
Hangup: at Wed Jan 13 16:51:05 2016

Call keeps going if application terminated

Due to segment fault, the application terminates, but the called number still has the call in progress! This happens usually.

Invalid datagram from udp > no answer ?

Hello,
First thanks for this nice piece of soft, that's just what I was looking for an art project
but till now, I did'nt manage to make it work
I'd like to make a call threw CLI with sipcmd
My SIP gateway works (I can give a call with linphone)
with sipcmd the call stop in an EndedByQ931Cause
I use my ISP SIP service (freephonie.net), the Public IP of my box is 82...**, I tried many parameter combinations as "To register to a gateaway, specify -c, -g and -w ", with no success ...
Any input is welcome
small log below, full log here :http://hastebin.com/mohexoqatu.rb
thks
b

$ ./sipcmd -P sip -u 09********@freephonie.net -c passwd -w freephonie.net -g 82.**.**.** -x "c02********;ws3000;d123;h" -o log.txt
Starting sipcmd
in debug mode
Manager
Init
initialising SIP endpoint...
TestChanAudio
TestChanAudio
Listening for SIP signalling on 0.0.0.0:5060
SIP listener up
registered as sip:09********@freephonie.net
Created LocalEndPoint
Main
## Call ##
TestPhone::Main: calling "02*********" using gateway "freephonie.net" at Wed Mar 26 21:59:59 2014

Setting up a call to: sip:02*********@freephonie.net
LocalEndpoint::MakeConnection
LocalEndpointCreateConnection
LocalConnection
OnIncomingConnection: token=Lee8e79702
connection set up to sip:02*********@freephonie.net
TestPhone::Main: calling "sip:02*********@freephonie.net" for 0
OnReleased: reason: EndedByQ931Cause
## Wait: waiting for 3000ms ##
OnReleased: reason: EndedByQ931Cause
OnClearedCall
~LocalConnection
Wait: wait done
## DTMF "123" ##
no call found with token=C77fd21301
Problem running command sequence ("c02*********;ws3000;d123;h"):

TestPhone::Main: shutting down
TestPhone::Main: exiting...
Exiting...
~Manager

none

segmentaion fault

valgrind says:

CreateMediaStream
TestChannel[ Call[C02e1f8fb1]-EP[L22cf95bf2] - 0x3c0550f0 ]
assert.cxx(112) PWLib Assertion fail: Attempt to do simultaneous reads from multiple threads: os_handle=24, thread ID=0x1f385700, file ptlib/unix/channel.cxx, line 110OnOpenMediaStream
recording media from local

TestPhone::Main: calling "sip:403#######@192.168.0.12" for 10
Assertion fail: Attempt to do simultaneous reads from multiple threads: os_handle=24, thread ID=0x1f385700, file ptlib/unix/channel.cxx, line 110

I have the latest version of ptlib_dev, any ideas ?

Call recording

I am having trouble converting the recorded PCM file to a decent sounding MP3. I have played around with various settings using lame but have yet to reproduce the quality of the call.

I have found this to be usable but slow...
lame -r -s 8 -m m --bitwidth 16 call.out call.mp3

What settings should I be using to convert the file?

Thanks

UDP packet too small for all the configuration data

Good day,
I'm running a Kamailio SIP server (UDP transport) with rtpproxy on the same machine to assist in NAT traversing. I can successfully set up calls from mobile phones that run Zioper SIP client over the internet using the Kamailio/rtpproxy. Instant messaging also works.

I'm running sipcmd on Ubuntu 14.04LTS with libopal 3.10.10

I've tried to use sipcmd to initiate a call to a SIP client and play a wave file. The examples are clear enough, but there seems to be a problem when sipcmd (via Opal) tries to customise the parameters of the available codecs during the INVITE phase. I get this error message in the log:
0:01.535 sipcmd SIP PDU is too large (2295 bytes) trying compact form.
0:01.535 sipcmd SIP PDU is likely too large (2245 bytes) for UDP datagram.

My assumtion is that a partial INVITE message arrives at the server, which then rejects setting up the call.

Is there a way to limit the number of codecs that Opal:GetMediaFormats() returns?

Regards,
Frix

Paralles calls

HOWTO make more then one call in the same time using the same dialing number and executable rules ?

tmakkonen / sipcmd Goto Github PK

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sipcmd — the command line SIP/H.323/RTP softphone

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Call

Wait: waiting for 3000ms

DTMF "123"

~Manager

Content-Length: 0

Call

./sipcmd -o /tmp/opal.log -p 12345 -P sip -x "c123;w1000;h" -w 192.168.101.10

Call

Answer

Wait: waiting for 900ms

Call

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