Could not write to sip:XXXX@sip.operator.com

[1am]

Hi

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:XXXX@sip.operator.com - 
  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?

[jwhardcastle]

You may need to add some time to wait in there. I found that my "enterprise" VOIP servers require at least 8 seconds of wait time to be reliable, and I'm using more where I can.

sipcmd -x "w13000;c1863;w250;d#;w250;h;" -o sip.log

[mikhailnov]

probably port 5060 is already in use

[PenzinAlexander]

You may need to add some time to wait in there. I found that my "enterprise" VOIP servers require at least 8 seconds of wait time to be reliable, and I'm using more where I can.

sipcmd -x "w13000;c1863;w250;d#;w250;h;" -o sip.log

God bless you, man! I was looking for solution like 3 days already. And wait before call did the trick! Thank you @jwhardcastle !