simple commandline example to start app properly about sipcmd HOT 12 CLOSED

Same probem. How do I get this to do anything at all?

$ ./sipcmd -u "[email protected]" -R "sip.voipbuster.com;pepijndevos;MyPassw0rd;*" -w "sip.voipbuster.com" -P sip -x "[email protected]"
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
Using SIP registrar sip.voipbuster.com for sip:[email protected]
Created LocalEndPoint
Main
sleep 2000 ms to allow time for registration ... 
Done!
## Call ##
TestPhone::Main: calling "[email protected]" at Thu Nov 15 09:56:03 2012

Setting up a call to: [email protected]
LocalEndpoint::MakeConnection
LocalEndpointCreateConnection
LocalConnection
OnIncomingConnection: token=L8d75ccc22
connection set up to [email protected]
OnReleased: reason: EndedByHostOffline
TestPhone::Main: shutting down
OnReleased: reason: EndedByHostOffline
OnClearedCall
~LocalConnection
TestPhone::Main: exiting...
Exiting...
~Manager

from sipcmd.

There was a bug in register implementation, try again using the description in README.

from sipcmd.

Hi, I search for some code like spicmd, but I miss a really basic example ...
I tried the previous post' command with my values -
$ ./sipcmd -u "[email protected]" -R "sip.odorik.cz;411171;password;*" -P sip -x "+420608544321"

and I got

Starting sipcmd
in debug mode
Manager
Init
initialising SIP endpoint...
TestChanAudio
TestChanAudio
Created LocalEndPoint
Main
TestPhone::Main: shutting down
TestPhone::Main: exiting...
Exiting...
~Manager

...after some time it exists. No idea what is up, how to use.

Thanks for a comment

from sipcmd.

Example of making a call

./sipcmd -P sip -u <username> -c <password> -w <server> -x "c<number>;w200;d12345"

from sipcmd.

In your specific case try
./sipcmd -P sip -u 411171 -c -w sip.odorik.cz -x "c420608544321;w200;d12345"

from sipcmd.

./sipcmd -P sip -l 135.60.10.21 -u "1777" -c milga -w "cc.com" -x "c197253;w5000;h" -o bluga.txt

Starting sipcmd
in debug mode
Manager
Init
0:00.027 Opal Garbage:0x36443700 PTLib Started thread 0x7dfcf0 (31086) Opal Garbage:0x36443700
TestChanAudio
TestChanAudio
initialising SIP endpoint...
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 "197253" using gateway "cc.com" at Thu Apr 18 10:42:29 2013
Setting up a call to: sip:[email protected]
LocalEndpoint::MakeConnection
LocalEndpointCreateConnection
LocalConnection
OnIncomingConnection: token=Lb4478fb52
connection set up to sip:[email protected]
OnReleased: reason: EndedByHostOffline
Wait: waiting for 5000ms
OnReleased: reason: EndedByHostOffline
OnClearedCall
~LocalConnection
Wait: wait done
Hangup
Hangup: at Thu Apr 18 10:43:07 2013
TestPhone::Main: shutting down
TestPhone::Main: exiting...
Exiting...
~Manager

The opal log shows the SIP register and invite messages being sent out at the same time. It does not wait for the response from the registration. I tried this successfully with PJSUA and got back a "407 Proxy Authentication Required".

0:02.107 sipcmd SIP Sending PDU (640 bytes) to: >rem=udp$204.11.192.160:5080,local=udp$135.60.10.21:5060,if=135.60$
REGISTER sip:cc.com SIP/2.0
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 135.60.10.21:5060;branch=z9hG4bK5864313f-aca6-e211-9bc7-001e0b2c62d8;rport
User-Agent: sipcmd/1.0.1

0:02.211 sipcmd SIP Sending PDU (1979 bytes) to: rem=udp$204.11.192.161:5080,local=udp$135.60.10.21:5060,if=135.6$
INVITE sip:[email protected] SIP/2.0
CSeq: 1 INVITE
v: SIP/2.0/UDP 135.60.10.21:5060;branch=z9hG4bK1250403f-aca6-e211-9bc7-001e0b2c62d8;rport
User-Agent: sipcmd/1.0.1

Can sipcmd do this Proxy Authentication stuff? Please help.

Thx!

from sipcmd.

Could you attach the whole opal log please. From what i remember off the
top of my head there should be a flag set to force a wait after
registrayion (or maybe before invites). I will try and look into this soon
:)

On 18 April 2013 16:55, dark4archon [email protected] wrote:

./sipcmd -P sip -l 135.60.10.21 -u "1777" -c milga -w "cc.com" -x
"c197253;w5000;h" -o bluga.txt

Starting sipcmd
in debug mode
Manager
Init
0:00.027 Opal Garbage:0x36443700 PTLib Started thread 0x7dfcf0 (31086)
Opal Garbage:0x36443700
TestChanAudio
TestChanAudio
initialising SIP endpoint...
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 "197253" using gateway "cc.com" at Thu Apr 18
10:42:29 2013
Setting up a call to: sip:[email protected]
LocalEndpoint::MakeConnection
LocalEndpointCreateConnection
LocalConnection
OnIncomingConnection: token=Lb4478fb52
connection set up to sip:[email protected]
OnReleased: reason: EndedByHostOffline
Wait: waiting for 5000ms
OnReleased: reason: EndedByHostOffline
OnClearedCall
~LocalConnection
Wait: wait done
Hangup
Hangup: at Thu Apr 18 10:43:07 2013

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

The opal log shows the SIP register and invite messages being sent out
at the same time. It does not wait for the response from the registration.
I tried this successfully with PJSUA and got back a "407 Proxy
Authentication Required".

0:02.107 sipcmd SIP Sending PDU (640 bytes) to: >rem=udp$
204.11.192.160:5080,local=udp$135.60.10.21:5060,if=135.60$
REGISTER sip:cc.com SIP/2.0
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 135.60.10.21:5060
;branch=z9hG4bK5864313f-aca6-e211-9bc7-001e0b2c62d8;rport
User-Agent: sipcmd/1.0.1

0:02.211 sipcmd SIP Sending PDU (1979 bytes) to: rem=udp$
204.11.192.161:5080,local=udp$135.60.10.21:5060,if=135.6$
INVITE sip:[email protected] SIP/2.0
CSeq: 1 INVITE
v: SIP/2.0/UDP 135.60.10.21:5060
;branch=z9hG4bK1250403f-aca6-e211-9bc7-001e0b2c62d8;rport
User-Agent: sipcmd/1.0.1

Can sipcmd do this Proxy Authentication stuff? Please help.

Thx!

—
Reply to this email directly or view it on GitHubhttps://github.com//issues/3#issuecomment-16585316
.

from sipcmd.

Opal log below

from sipcmd.

Opal Log:

0:00.027 sipcmd Version 1.0.1 by Command line VoIP testphone on Unix Linux (3.2.0-34-generic-x86_64) with PTLib (v2.10.2 (svn:
25898)) at 2013/4/18 10:42:27.853
0:00.027 sipcmd OpalMan Attached endpoint with prefix sip
0:00.027 sipcmd OpalEP Created endpoint: sip
0:00.027 sipcmd PWLib File handle high water mark set: 8 PUDPSocket
0:00.027 sipcmd IfaceMon Initial interface list:
127.0.0.1 <00-00-00-00-00-00> (lo)
135.60.10.21 <00-1E-0B-2C-62-D8> (eth0)
fe80::21e:bff:fe2c:62d8 <00-1E-0B-2C-62-D8> (eth0)
::1 <00-00-00-00-00-00> (lo)

0:00.027 sipcmd PTLIB Opened NetLink socket
0:00.027 sipcmd PWLib File handle high water mark set: 13 Thread unblock pipe
0:00.027 sipcmd PTLib Created thread 0x7e2530
0:00.027 sipcmd PTLib Thread high water mark set: 3
0:00.027 sipcmd PWLib File handle high water mark set: 15 Thread unblock pipe
0:00.027 sipcmd PTLib Created thread 0x7e2650 Housekeeper
0:00.027 Network In...0x36402700 PTLib Started thread 0x7e2530 (31087) Network Interface Monitor:0x36402700
0:00.027 sipcmd PTLib No permission to set priority level 4
0:00.027 Network In...0x36402700 IfaceMon Started interface monitor thread.
0:00.028 sipcmd PTLib Thread high water mark set: 4
0:00.028 Housekeeper:0x363c1700 PTLib Started thread 0x7e2650 (31088) Housekeeper:0x363c1700
0:00.028 sipcmd OpalMan Attached endpoint with prefix sips
0:00.028 sipcmd SIP Created endpoint.
0:00.028 sipcmd OpalMan Added route "local:.=sip:"
0:00.028 sipcmd OpalMan Added route "sip:.=local:"
0:00.028 sipcmd PWLib File handle high water mark set: 16 PUDPSocket
0:00.028 sipcmd MonSock Created socket bundle for all interfaces.
0:00.028 sipcmd PWLib File handle high water mark set: 17 PUDPSocket
0:00.028 sipcmd MonSock Created bundled UDP socket 135.60.10.21:5060
0:00.028 sipcmd PWLib File handle high water mark set: 18 PUDPSocket
0:00.028 sipcmd MonSock Could not listen on fe80::21e:bff:fe2c:62d8:5060 - Invalid argument
0:00.028 sipcmd PWLib File handle high water mark set: 19 Thread unblock pipe
0:00.028 sipcmd PTLib Created thread 0x7dc660 Opal Listener
0:00.028 sipcmd PTLib Thread high water mark set: 5
0:00.028 sipcmd PTLib No permission to set priority level 4
0:00.028 Opal Liste...0x36380700 PTLib Started thread 0x7dc660 (31089) Opal Listener:0x36380700
0:00.028 sipcmd SIP Start REGISTER
aor=1777
remote=cc.com
local=
contact=
authID=
realm=
expire=0
restore=30
minRetry=default
maxRetry=default
compatibility=FullyCompliant
0:00.028 Opal Liste...0x36380700 Listen Started listening thread on udp$_:5060
0:00.029 sipcmd PWLib File handle high water mark set: 20 PUDPSocket
0:00.029 sipcmd SIP Constructed REGISTER handler for sip:[email protected]
0:00.029 sipcmd SIP Executing state change to Subscribing for REGISTER handler, target=sip:[email protected], id=f276f43
d-aca6-e211-9bc7-001e0b2c62d8@actnew
0:00.029 sipcmd SIP Changing REGISTER handler from Unavailable to Subscribing, target=sip:[email protected], id=f276f43d
-aca6-e211-9bc7-001e0b2c62d8@actnew
0:00.029 sipcmd DNS SRV Lookup cc.com service _sip._udp
0:02.102 sipcmd DNS SRV Record found _sip.udp.cc.com
0:02.103 sipcmd SIP Attempting SRV record entry 0: 204.11.192.161:5080
0:02.103 sipcmd OpalUDP Binding to interface: 0.0.0.0:5060
0:02.103 sipcmd SIP Created transport udp$204.11.192.161:5080<if=udp$:5060>
0:02.103 sipcmd OpalUDP Started connect to 204.11.192.161:5080
0:02.103 sipcmd OpalUDP Writing to interface 0 - "135.60.10.21%eth0"
0:02.104 sipcmd OpalMan Listener interfaces: associated transport=udp$135.60.10.21:5060
udp$135.60.10.21:5060,udp$[fe80::21e:bff:fe2c:62d8]:5060
0:02.105 sipcmd SIP Transaction created.
0:02.106 sipcmd DNS SRV Lookup cc.com service _sip._udp
0:02.106 sipcmd DNS SRV Record found _sip._udp.cc.com
0:02.106 sipcmd SIP Attempting SRV record entry 0: 204.11.192.160:5080
0:02.106 sipcmd SIP Transaction remote address is udp$204.11.192.160:5080
0:02.106 sipcmd SIP Set new remote address udp$204.11.192.160:5080 for transport udp$204.11.192.160:5080<if=udp$135.60.10.21:5060>
0:02.107 sipcmd SIP Sending PDU (640 bytes) to: rem=udp$204.11.192.160:5080,local=udp$135.60.10.21:5060,if=135.60.10.21%eth0
REGISTER sip:cc.com SIP/2.0
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 135.60.10.21:5060;branch=z9hG4bK5864313f-aca6-e211-9bc7-001e0b2c62d8;rport
User-Agent: sipcmd/1.0.1
From: sip:[email protected];tag=767ff43d-aca6-e211-9bc7-001e0b2c62d8
Call-ID: f276f43d-aca6-e211-9bc7-001e0b2c62d8@actnew
Organization: Command line VoIP testphone
To: sip:[email protected]
Contact: sip:[email protected]:5060;q=1, sip:1777@[fe80::21e:bff:fe2c:62d8]:5060;q=0.500
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
Expires: 3600
Content-Length: 0
Max-Forwards: 70

0:02.107 sipcmd OpalUDP Setting interface to 135.60.10.21%eth0
0:02.107 sipcmd SIP Transaction timers set: retry=10.000, completion=16.000
0:02.107 sipcmd OpalUDP Skipping incompatible interface 1 - "[fe80::21e:bff:fe2c:62d8]%eth0"
0:02.107 sipcmd OpalMan Attached endpoint with prefix local
0:02.107 sipcmd OpalEP Created endpoint: local
0:02.107 sipcmd LocalEP Created endpoint.
0:02.108 sipcmd OpalMan Set up call from local:* to sip:[email protected]
0:02.108 sipcmd Call Created Call[Ca3d6169a1]
0:02.108 sipcmd OpalMan Set up connection to "local:"
0:02.108 sipcmd OpalCon Created connection Call[Ca3d6169a1]-EP[Lb4478fb52]
0:02.108 sipcmd LocalCon Created connection with token "Lb4478fb52"
0:02.108 sipcmd OpalCon SetPhase from UninitialisedPhase to SetUpPhase for Call[Ca3d6169a1]-EP[Lb4478fb52]
0:02.108 sipcmd OpalMan OnIncoming connection Call[Ca3d6169a1]-EP[Lb4478fb52]
0:02.108 sipcmd Call GetOtherPartyConnection Call[Ca3d6169a1]-EP[Lb4478fb52]
0:02.108 sipcmd OpalMan Searching for route "local:bigboss sip:[email protected]"
0:02.108 sipcmd OpalMan Matched regex "^local:. .$" ("local:.")
0:02.108 sipcmd OpalMan Set up connection to "sip:[email protected]"
0:02.109 sipcmd OpalCon Created connection Call[Ca3d6169a1]-EP[f4d6313f-aca6-e211-9bc7-001e0b2c62d8]
0:02.109 sipcmd RFC2833 Handler created
0:02.109 sipcmd RFC2833 Handler created
0:02.109 sipcmd SIP Created connection.
0:02.109 sipcmd LocalCon Outgoing call routed to sip:[email protected] for Call[Ca3d6169a1]-EP[Lb4478fb52]
0:02.109 sipcmd Call OnSetUp Call[Ca3d6169a1]-EP[Lb4478fb52]
0:02.109 sipcmd SIP SetUpConnection: sip:[email protected]
0:02.109 sipcmd DNS SRV Lookup cc.com service _sip._udp
0:02.109 sipcmd DNS SRV Record found _sip._udp.cc.com
0:02.109 sipcmd SIP Attempting SRV record entry 0: 204.11.192.170:5080
0:02.109 sipcmd SIP Connecting to sip:[email protected] via sip:[email protected]:5080
0:02.109 sipcmd SIP Setting new transport for destination "sip:[email protected]:5080"
0:02.196 sipcmd OpalUDP Binding to interface: 0.0.0.0:5060
0:02.196 sipcmd SIP Created transport udp$204.11.192.170:5080<if=udp$*:5060>
0:02.196 sipcmd OpalUDP Started connect to 204.11.192.170:5080
0:02.202 sipcmd Call GetMediaFormats for Call[Ca3d6169a1]-EP[f4d6313f-aca6-e211-9bc7-001e0b2c62d8]
GSM-AMR
GSM-06.10
G.711-uLaw-64k
G.711-ALaw-64k
H.263
H.261
CELT-48K
CELT-32K
G.722-64k
G.722.1-24k
G.722.1-32k
G.722.2
SILK-16
SpeexIETFWide-20.6k
SpeexWB
SpeexWide-20.6k
G.726-16k
G.726-24k
G.726-32k
G.726-40k
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
H.263P
RFC4175_YCbCr-4:2:0
theora
MSRP
SIP-IM
T.140
H.224/H323AnnexQ

0:02.202 sipcmd SIP Remote media formats set to GSM-AMR,GSM-06.10,G.711-uLaw-64k,G.711-ALaw-64k,H.263,H.261,CELT-48K,CELT-32K,G.72
2-64k,G.722.1-24k,G.722.1-32k,G.722.2,SILK-16,SpeexIETFWide-20.6k,SpeexWB,SpeexWide-20.6k,G.726-16k,G.726-24k,G.726-32k,G.726-40k,LPC-10,MS-GSM,MS-IMA-ADPCM,S
ILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIETFNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.3
8,UserInput/RFC2833,NamedSignalEvent,H.261-CIF,H.261-QCIF,H.263P,RFC4175_YCbCr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ
0:02.202 sipcmd OpalUDP Writing to interface 0 - "135.60.10.21%eth0"
0:02.202 sipcmd SIP Getting local URI from registeration: sip:[email protected]
0:02.202 sipcmd SIP Updating dialog tag from "" to "f4d6313f-aca6-e211-9bc7-001e0b2c62d8"
0:02.203 sipcmd SIP Remote dialog address from target: sip:[email protected]
0:02.203 sipcmd DNS SRV Lookup cc.com service _sip._udp
0:02.203 sipcmd DNS SRV Record found _sip._udp.cc.com
0:02.203 sipcmd SIP Attempting SRV record entry 0: 204.11.192.161:5080
0:02.204 sipcmd SIP INVITE transaction id=z9hG4bK1250403f-aca6-e211-9bc7-001e0b2c62d8 created.
0:02.204 sipcmd SIP Creating INVITE request
0:02.204 sipcmd SIP Offering all configured media:
GSM-AMR,GSM-06.10,G.711-uLaw-64k,G.711-ALaw-64k,H.263,H.261,CELT-48K,CELT-32K,G.722-64k,G.722.1-24k,G.722.1-32k,G.722.2,SILK-16,SpeexIETFWide-20.6k,SpeexWB,Sp
eexWide-20.6k,G.726-16k,G.726-24k,G.726-32k,G.726-40k,LPC-10,MS-GSM,MS-IMA-ADPCM,SILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIET
FNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.38,UserInput/RFC2833,NamedSignalEvent,H.261-CIF,H.261-QCIF,H.263P,RFC4175_YCb
Cr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ
0:02.204 sipcmd SIP Offering media type audio in SDP
0:02.204 sipcmd Call IsMediaBypassPossible Call[Ca3d6169a1]-EP[f4d6313f-aca6-e211-9bc7-001e0b2c62d8] session 1
0:02.204 sipcmd OpalMan IsMediaBypassPossible: session 1
0:02.204 sipcmd OpalCon IsMediaBypassPossible: default returns false
0:02.204 sipcmd RTP Cannot find media session 1
0:02.204 sipcmd RTP Cannot find RTP session 1
0:02.205 sipcmd VoIP Metrics RTCP_XR_Metrics created.
0:02.205 sipcmd RTP_UDP Session 1, created with NAT flag set to 0
0:02.205 sipcmd PWLib File handle high water mark set: 21 PUDPSocket
0:02.205 sipcmd PWLib File handle low water mark set: 20 PUDPSocket
0:02.205 sipcmd RTP_UDP Session 1 created: 135.60.10.21:5000-5001 ssrc=3349705597
0:02.205 sipcmd PWLib File handle high water mark set: 22 PUDPSocket
0:02.205 sipcmd RTP Creating new session RTP_UDP
0:02.205 sipcmd RTPEp Session 1, remembering local RTP port 5000 on connection Call[Ca3d6169a1]-EP[f4d6313f-aca6-e211-9bc7-001e
0b2c62d8]
0:02.205 sipcmd RTP Found existing media session 1
0:02.205 sipcmd MediaFormat Validation of merge for media option "BitRate" failed.
0:02.205 sipcmd SDP SDP not including SpeexIETFWide-20.6k as it is not a SIP transportable format
0:02.206 sipcmd SDP SDP not including SpeexWide-20.6k as it is not a SIP transportable format
0:02.206 sipcmd SDP SDP not including MS-GSM as it is not a SIP transportable format
0:02.206 sipcmd SDP SDP not including MS-IMA-ADPCM as it is not a SIP transportable format
0:02.206 sipcmd SDP SDP not including SpeexIETFNarrow-11k as it is not a SIP transportable format
0:02.206 sipcmd SDP SDP not including SpeexIETFNarrow-15k as it is not a SIP transportable format
0:02.206 sipcmd SDP SDP not including SpeexIETFNarrow-18.2k as it is not a SIP transportable format
0:02.206 sipcmd SDP SDP not including SpeexIETFNarrow-24.6k as it is not a SIP transportable format
0:02.206 sipcmd SDP SDP not including SpeexIETFNarrow-5.95k as it is not a SIP transportable format
0:02.206 sipcmd SDP SDP not including SpeexIETFNarrow-8k as it is not a SIP transportable format
0:02.206 sipcmd SDP SDP not including SpeexWNarrow-8k as it is not a SIP transportable format
0:02.206 sipcmd MediaFormat Merging UserInput/RFC2833 into UserInput/RFC2833
0:02.206 sipcmd RFC2833 Set tx pt=[pt=101], events="0-16,32,36" for UserInput/RFC2833
0:02.206 sipcmd RFC2833 Set rx pt=[pt=101], events="0-16,32,36" for UserInput/RFC2833
0:02.206 sipcmd MediaFormat Merging NamedSignalEvent into NamedSignalEvent
0:02.206 sipcmd RFC2833 Set tx pt=[pt=100], events="192-193" for NamedSignalEvent
0:02.206 sipcmd RFC2833 Set rx pt=[pt=100], events="192-193" for NamedSignalEvent
0:02.206 sipcmd SIP Offering media type video in SDP
0:02.206 sipcmd Call IsMediaBypassPossible Call[Ca3d6169a1]-EP[f4d6313f-aca6-e211-9bc7-001e0b2c62d8] session 2
0:02.206 sipcmd OpalMan IsMediaBypassPossible: session 2
0:02.206 sipcmd OpalCon IsMediaBypassPossible: default returns false
0:02.206 sipcmd RTP Cannot find media session 2
0:02.206 sipcmd RTP Cannot find RTP session 2
0:02.206 sipcmd VoIP Metrics RTCP_XR_Metrics created.
0:02.206 sipcmd RTP_UDP Session 2, created with NAT flag set to 0
0:02.206 sipcmd PWLib File handle high water mark set: 23 PUDPSocket
0:02.206 sipcmd PWLib File handle low water mark set: 22 PUDPSocket
0:02.206 sipcmd RTP_UDP SetOption(8,1048576) failed, even though it said it succeeded!
0:02.206 sipcmd RTP_UDP SetOption(8,524288) failed, even though it said it succeeded!
0:02.206 sipcmd RTP_UDP SetOption(8,262144) failed, even though it said it succeeded!
0:02.206 sipcmd RTP_UDP Session 2 created: 135.60.10.21:5002-5003 ssrc=711983677
0:02.206 sipcmd PWLib File handle high water mark set: 24 PUDPSocket
0:02.207 sipcmd RTP Creating new session RTP_UDP
0:02.207 sipcmd RTPEp Session 2, remembering local RTP port 5002 on connection Call[Ca3d6169a1]-EP[f4d6313f-aca6-e211-9bc7-001e
0b2c62d8]
0:02.207 sipcmd RTP Found existing media session 2
0:02.207 sipcmd SDP SDP not including H.261-CIF as it is not a SIP transportable format
0:02.207 sipcmd SDP SDP not including H.261-QCIF as it is not a SIP transportable format
0:02.207 sipcmd SIP Transaction remote address is udp$204.11.192.161:5080
0:02.207 sipcmd SIP Set new remote address udp$204.11.192.161:5080 for transport udp$204.11.192.161:5080<if=udp$135.60.10.21:5060>
0:02.207 sipcmd OpalPlugin to_customised_options:
Format Name = SILK-16
Media Type = audio
Payload Type = [pt=120]
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

0:02.208 sipcmd OpalPlugin to_customised_options:
Format Name = SILK-8
Media Type = audio
Payload Type = [pt=119]
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

0:02.209 sipcmd OpalPlugin to_customised_options:
Format Name = H.263
Media Type = video
Payload Type = H263
Encoding Name = h263
Annex F - Advanced Prediction (R/W) = 1 FMTP name: F (0) Boolean
CIF MPI (R/W) = 1 FMTP name: CIF (33) UnsignedInt
CIF16 MPI (R/W) = 1 FMTP name: CIF16 (33) UnsignedInt
CIF4 MPI (R/W) = 1 FMTP name: CIF4 (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) = 327600 UnsignedInt
Max Rx Frame Height (R/W) = 1152 UnsignedInt
Max Rx Frame Width (R/W) = 1408 UnsignedInt
Max Tx Packet Size (R/O) = 1444 UnsignedInt
MaxBR (R/W) = 0 FMTP name: maxbr (0) UnsignedInt
Media Packetization (R/O) = RFC2190 String
Min Rx Frame Height (R/W) = 96 UnsignedInt
Min Rx Frame Width (R/W) = 128 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
SQCIF MPI (R/W) = 1 FMTP name: SQCIF (33) UnsignedInt
Target Bit Rate (R/W) = 327600 UnsignedInt
Tx Key Frame Period (R/W) = 125 UnsignedInt

0:02.209 sipcmd OpalPlugin to_customised_options changed option "MaxBR" from "0" to "3276"
0:02.209 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

0:02.210 sipcmd OpalPlugin to_customised_options:
Format Name = H.263P
Media Type = video
Payload Type = [pt=94]
Encoding Name = h263-1998
Annex D - Unrestricted Motion Vector (R/O) = 1 FMTP name: D (0) Boolean
Annex F - Advanced Prediction (R/W) = 1 FMTP name: F (0) Boolean
Annex I - Advanced INTRA Coding (R/W) = 1 FMTP name: I (0) Boolean
Annex J - Deblocking Filter (R/O) = 1 FMTP name: J (0) Boolean
Annex K - Slice Structure (R/O) = 0 FMTP name: K (0) UnsignedInt
Annex N - Reference Picture Selection (R/O) = 0 FMTP name: N (0) Boolean
Annex T - Modified Quantization (R/O) = 0 FMTP name: T (0) Boolean
CIF MPI (R/W) = 1 FMTP name: CIF (33) UnsignedInt
CIF16 MPI (R/W) = 1 FMTP name: CIF16 (33) UnsignedInt
CIF4 MPI (R/W) = 1 FMTP name: CIF4 (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) = 327600 UnsignedInt
Max Rx Frame Height (R/W) = 576 UnsignedInt
Max Rx Frame Width (R/W) = 704 UnsignedInt
Max Tx Packet Size (R/O) = 1444 UnsignedInt
Min Rx Frame Height (R/W) = 96 UnsignedInt
Min Rx Frame Width (R/W) = 128 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
SIF MPI (R/W) = 320,240,1 FMTP name: CUSTOM () String
SIF4 MPI (R/W) = 640,480,1 FMTP name: CUSTOM () String
SQCIF MPI (R/W) = 1 FMTP name: SQCIF (33) UnsignedInt
Target Bit Rate (R/W) = 327600 UnsignedInt
Tx Key Frame Period (R/W) = 125 UnsignedInt

0:02.210 sipcmd OpalPlugin to_customised_options changed option "CIF16 MPI" from "1" to "33"
0:02.210 sipcmd OpalPlugin to_customised_options:
Format Name = theora
Media Type = video
Payload Type = [pt=125]
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

0:02.211 sipcmd SIP PDU is too large (2029 bytes) trying compact form.
0:02.211 sipcmd SIP PDU is likely too large (1979 bytes) for UDP datagram.
0:02.211 sipcmd SIP Sending PDU (1979 bytes) to: rem=udp$204.11.192.161:5080,local=udp$135.60.10.21:5060,if=135.60.10.21%eth0
INVITE sip:[email protected] SIP/2.0
CSeq: 1 INVITE
v: SIP/2.0/UDP 135.60.10.21:5060;branch=z9hG4bK1250403f-aca6-e211-9bc7-001e0b2c62d8;rport
User-Agent: sipcmd/1.0.1
f: "bigboss" sip:[email protected];tag=f4d6313f-aca6-e211-9bc7-001e0b2c62d8
i: d6db313f-aca6-e211-9bc7-001e0b2c62d8@actnew
k: 100rel,replaces
Organization: Command line VoIP testphone
t: sip:[email protected]
m: sip:[email protected]
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
l: 1394
c: application/sdp
Max-Forwards: 70

v=0
o=- 1366299749 1 IN IP4 135.60.10.21
s=sipcmd/1.0.1
c=IN IP4 135.60.10.21
t=0 0
m=audio 5000 RTP/AVP 112 3 0 8 114 113 9 116 117 118 120 110 124 123 122 121 111 119 109 101 100
a=sendrecv
a=rtpmap:112 AMR/8000/1
a=rtpmap:3 gsm/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:114 CELT/48000/1
a=rtpmap:113 CELT/32000/1
a=rtpmap:9 G722/8000/1
a=rtpmap:116 G7221/16000/1
a=fmtp:116 bitrate=24000
a=rtpmap:117 G7221/16000/1
a=fmtp:117 bitrate=32000
a=rtpmap:118 AMR-WB/16000/1
a=fmtp:118 octet-align=1
a=rtpmap:120 SILK/16000/1
a=rtpmap:110 Speex/16000/1
a=rtpmap:124 G726-16/8000/1
a=rtpmap:123 G726-24/8000/1
a=rtpmap:122 G726-32/8000/1
a=rtpmap:121 G726-40/8000/1
a=rtpmap:111 lpc10/8000/1
a=rtpmap:119 SILK/8000/1
a=rtpmap:109 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 34 31 94 97 125
b=AS:186624
b=TIAS:186624000
a=sendrecv
a=rtpmap:34 h263/90000
a=fmtp:34 F=1;CIF=1;CIF16=1;CIF4=1;maxbr=3276;QCIF=1;SQCIF=1
a=rtpmap:31 h261/90000
a=fmtp:31 CIF=1;QCIF=1
a=rtpmap:94 h263-1998/90000
a=fmtp:94 D=1;F=1;I=1;J=1;CIF=1;CIF4=1;QCIF=1;CUSTOM=320,240,1;SQCIF=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:125 theora/90000
a=fmtp:125 height=576;width=704

0:02.211 sipcmd OpalUDP Setting interface to 135.60.10.21%eth0
0:02.211 sipcmd SIP Transaction timers set: retry=10.000, completion=32.000
0:02.211 sipcmd OpalUDP Skipping incompatible interface 1 - "[fe80::21e:bff:fe2c:62d8]%eth0"
0:02.212 sipcmd OpalCon SetPhase from UninitialisedPhase to SetUpPhase for Call[Ca3d6169a1]-EP[f4d6313f-aca6-e211-9bc7-001e0b2c62
d8]
0:02.212 sipcmd OpalCon OnSetUpConnectionCall[Ca3d6169a1]-EP[f4d6313f-aca6-e211-9bc7-001e0b2c62d8]
0:02.212 sipcmd OpalEP OnSetUpConnection Call[Ca3d6169a1]-EP[f4d6313f-aca6-e211-9bc7-001e0b2c62d8]
0:02.212 sipcmd OpalMan SetUpCall succeeded, call=Call[Ca3d6169a1]
0:12.108 Housekeeper:0x363c1700 SIP REGISTER transaction id=z9hG4bK5864313f-aca6-e211-9bc7-001e0b2c62d8 timeout, making retry 1, timeout 20.000, s
tate 1
0:12.108 Housekeeper:0x363c1700 SIP Set new remote address udp$204.11.192.160:5080 for transport udp$204.11.192.160:5080<if=udp$135.60.10.21:5060>
0:12.108 Housekeeper:0x363c1700 SIP Sending PDU (640 bytes) to: rem=udp$204.11.192.160:5080,local=udp$135.60.10.21:5060,if=135.60.10.21%eth0
REGISTER sip:cc.com SIP/2.0
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 135.60.10.21:5060;branch=z9hG4bK5864313f-aca6-e211-9bc7-001e0b2c62d8;rport
User-Agent: sipcmd/1.0.1
From: sip:[email protected];tag=767ff43d-aca6-e211-9bc7-001e0b2c62d8
Call-ID: f276f43d-aca6-e211-9bc7-001e0b2c62d8@actnew
Organization: Command line VoIP testphone
To: sip:[email protected]
Contact: sip:[email protected]:5060;q=1, sip:1777@[fe80::21e:bff:fe2c:62d8]:5060;q=0.500
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
Expires: 3600
Content-Length: 0
Max-Forwards: 70

from sipcmd.

Odorik:
Thanks for the clear advice on comandline. Still I cannot call, but now it is more clear what happens. I get an error 401 ... SIP/2.0 401 Unauthorized.
BTW, I see - it probes all interfaces, but there is only 192.168.0.136%wlan0" that makes sense.

0:00.145 Opal Liste...0xb4e62b40 SIP Queueing PDU "1 REGISTER <401>", transaction=z9.......

well, now I dont know.......

from sipcmd.

Hi,
Every time I pull up the phone the connection is broken with:
...
...
TestChannel::Write
TestChannel::Write
TestChannel::Write
TestChannel::Write
OnEstablished
TestPhone::Main: shutting down
OnEstablished
OnEstablishedCall
In call with sip:[email protected]:5060 using local:hm token=[C042e8de51]
Close [ Call[C042e8de51]-EP[L6c9bcfb82] - 0xb7321870 ]
TestChanAudio::CloseChannel
StopAudioPlayback
StopAudioRecording
OnClosedMediaStream
OnClosedMediaStream
Close [ Call[C042e8de51]-EP[L6c9bcfb82] - 0xb7332e90 ]
TestChanAudio::CloseChannel
StopAudioPlayback
StopAudioRecording
OnClosedMediaStream
OnClosedMediaStream
OnReleased: reason: EndedByLocalUser
~LocalConnection
OnReleased: reason: EndedByLocalUser
OnClearedCall
TestPhone::Main: exiting...
Exiting...

What the heck?... Any suggestion? Just a little?

from sipcmd.

Readme updated

from sipcmd.