Digital Receptionist Transfer from external call failure.

[jlag]

Tested using Linksys SPA1001, SoundWin 4000 series as hard ATA's, 3CX Soft Phone as PC client and CSIPSimple, SIPDroid as Android Clients, problem is consistent.

Using only SPA1001 as test ATA now for variable elimination.

Problem is: DR answers call, caller enters extension for transfer and transfer fails 95% of the time, no audio.

Have played with "Supports Re-Invite" and "Support 'Replaces' Header" to no avail on the SPA1001 extension.

If the external caller receives "music on hold" after entering the extension for transfer all seems to work.

If the external caller is receives silence after entering the extension, the extension rings, answers, and there is no audio.

Music on hold appears to happen randomly after caller enters extension in the DR.

Sometimes the external caller enters the extension, is presented with silence, the extension rings and answers, with no audio to the extension or external caller. If the extension then places the call on hold, the external caller is presented with music on hold, if the extension then rejoins the call, all works fine.

I have directed this ATA to my trixbox and DR transfers work fine to the ATA extension.

Perhaps port issue, but cant resolve as of yet.

Log from succesful External->DR->Extension:

12:28:08.383  [CM503008]: Call(3): Call is terminated
12:28:07.502  Currently active calls - 1: [3]
12:28:00.081  [MS210001] C:3.1:Answer received. RTP connection[unsecure]: 50.16.57.187:6086(6087)
12:28:00.081  Remote SDP is set for legC:3.1
12:27:59.701  [CM503007]: Call(3): Device joined: sip:[email protected]:5060
12:27:59.701  [MS210002] C:3.1:Offer provided. Connection(transcoding mode): 173.79.212.122:9000(9001)
12:27:59.701  [MS210001] C:3.3:Answer received. RTP connection[unsecure]: 172.16.0.95:12340(12341)
12:27:59.691  Remote SDP is set for legC:3.3
12:27:59.691  [CM505001]: Ext.2243: Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA1001-3.1.19(SE)] PBX contact: [sip:[email protected]:5060]
12:27:59.691  [CM503002]: Call(3): Alerting sip:[email protected]:5060
12:27:55.014  [MS210003] C:3.1:Answer provided. Connection(transcoding mode[unsecure]):173.79.212.122:9000(9001)
12:27:55.004  [MS210000] C:3.1:Offer received. RTP connection: 50.16.57.187:6086(6087)
12:27:55.004  Remote SDP is set for legC:3.1
12:27:54.844  [CM503025]: Call(3): Calling Ext:Ext.2243@[Dev:sip:[email protected]:5060]
12:27:54.844  [MS210002] C:3.3:Offer provided. Connection(transcoding mode): 172.16.0.120:7010(7011)
12:27:54.794  [CM503004]: Call(3): Route 1: Ext:Ext.2243@[Dev:sip:[email protected]:5060]
12:27:54.774  [CM503010]: Making route(s) to <sip:[email protected]:5060>
12:27:54.774  Refer: from=<sip:[email protected]:5060>;tag=06682f0e; to="+15986552467:4334220401"<sip:[email protected]:5060>;tag=845f1827; RefTo=<sip:[email protected]:5060>
12:27:54.584  [MS210003] C:3.2:Answer provided. Connection(transcoding mode[unsecure]):127.0.0.1:7008(7009)
12:27:54.584  [MS210001] C:3.1:Answer received. RTP connection[unsecure]: 50.16.57.187:6086(6087)
12:27:54.574  Remote SDP is set for legC:3.1
12:27:54.033  [MS210002] C:3.1:Offer provided. Connection(transcoding mode): 173.79.212.122:9000(9001)
12:27:54.023  [MS210000] C:3.2:Offer received. RTP connection: 127.0.0.1:40610(40611)
12:27:54.023  Remote SDP is set for legC:3.2
12:27:43.127  Session 73 of leg C:3.1 is confirmed
12:27:43.017  [CM503007]: Call(3): Device joined: sip:[email protected]:40600;rinstance=cc996086cfc88b0b
12:27:43.007  [CM503007]: Call(3): Device joined: sip:[email protected]:5060
12:27:43.007  [MS210005] C:3.1:Answer provided. Connection(proxy mode):173.79.212.122:9000(9001)
12:27:43.007  [MS210001] C:3.2:Answer received. RTP connection[unsecure]: 127.0.0.1:40610(40611)
12:27:42.997  Remote SDP is set for legC:3.2
12:27:42.987  [CM505001]: Ext.8000: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX IVR;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] PBX contact: [sip:[email protected]:5060]
12:27:42.987  [CM503002]: Call(3): Alerting sip:[email protected]:40600;rinstance=cc996086cfc88b0b
12:27:42.847  [CM503025]: Call(3): Calling Ext:Ext.8000@[Dev:sip:[email protected]:40600;rinstance=cc996086cfc88b0b]
12:27:42.847  [MS210004] C:3.2:Offer provided. Connection(proxy mode): 127.0.0.1:7008(7009)
12:27:42.817  [CM503004]: Call(3): Route 1: Ext:Ext.8000@[Dev:sip:[email protected]:40600;rinstance=cc996086cfc88b0b]
12:27:42.817  [CM503010]: Making route(s) to <sip:[email protected]:5060>
12:27:42.817  [MS210000] C:3.1:Offer received. RTP connection: 67.231.8.106:10950(10951)
12:27:42.817  Remote SDP is set for legC:3.1
12:27:42.817  [CM505003]: Provider:[FreePBX] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [FreeSWITCH-mod_sofia/1.0.4-exported] PBX contact: [sip:[email protected]:5060]
12:27:42.797  [CM503001]: Call(3): Incoming call from +15986552467@(Ln.10002@FreePBX) to <sip:[email protected]:5060>
12:27:42.496  [CM503012]: Inbound any hours rule (4334220401) for 10002 forwards to DN:8000
12:27:42.476  Looking for inbound target: called=4334220401; caller=+15986552467
12:27:42.466  [CM500002]: Info on incoming INVITE:
  INVITE sip:[email protected]:5060;rinstance=3c643cbf0185a4a8;received=10.112.189.140:5060 SIP/2.0
  Via: SIP/2.0/UDP pb2proxy-pro-aws03.phoneboothdev.info;branch=z9hG4bK0706.5e20a9d7.0;received=184.72.227.214
  Via: SIP/2.0/UDP 10.110.26.133;received=10.110.26.133;rport=5060;branch=z9hG4bKHDUec03K0UQDD
  Max-Forwards: 53
  Record-Route: <sip:pb2proxy-pro-aws03.phoneboothdev.info;lr=on;did=3c5.7dec4337>
  Contact: <sip:[email protected]:5060>
  To: <sip:[email protected]:5060;rinstance=3c643cbf0185a4a8;received=10.112.189.140:5060>
  From: "+15986552467"<sip:[email protected]>;tag=6Dc0j3r23tDFD
  Call-ID: 80b0ff1c-d403-122e-9d90-1231391c1c77
  CSeq: 10330561 INVITE
  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
  Supported: precondition, path, replaces
  User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported
  Allow-Events: talk, refer
  Privacy: none
  P-Asserted-Identity: "+15986552467" <sip:[email protected]>
  Content-Length: 0

Log from unsuccesful External->DR->Extension:

12:31:37.795  Currently active calls [none]
12:31:35.712  [CM503008]: Call(5): Call is terminated
12:31:22.122  [CM503007]: Call(5): Device joined: sip:[email protected]:5060
12:31:22.112  [MS210003] C:5.1:Answer provided. Connection(transcoding mode[unsecure]):173.79.212.122:9004(9005)
12:31:22.112  [MS210001] C:5.3:Answer received. RTP connection[unsecure]: 172.16.0.95:12344(12345)
12:31:22.112  Remote SDP is set for legC:5.3
12:31:22.112  [CM505001]: Ext.2243: Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA1001-3.1.19(SE)] PBX contact: [sip:[email protected]:5060]
12:31:22.112  [CM503002]: Call(5): Alerting sip:[email protected]:5060
12:31:14.922  [MS210001] C:5.2:Answer received. RTP connection[unsecure]: 127.0.0.1:40614(40615)
12:31:14.912  Remote SDP is set for legC:5.2
12:31:14.872  [CM503025]: Call(5): Calling Ext:Ext.2243@[Dev:sip:[email protected]:5060]
12:31:14.872  [MS210002] C:5.3:Offer provided. Connection(transcoding mode): 172.16.0.120:7018(7019)
12:31:14.821  [CM503004]: Call(5): Route 1: Ext:Ext.2243@[Dev:sip:[email protected]:5060]
12:31:14.821  [CM503010]: Making route(s) to <sip:[email protected]:5060>
12:31:14.811  Refer: from=<sip:[email protected]:5060>;tag=5c01fa0e; to="+15986552467:4334220401"<sip:[email protected]:5060>;tag=373dbb00; RefTo=<sip:[email protected]:5060>
12:31:14.711  [MS210004] C:5.2:Offer provided. Connection(proxy mode): 127.0.0.1:7016(7017)
12:31:14.711  [MS210000] C:5.1:Offer received. RTP connection: 50.16.57.187:15836(15837)
12:31:14.711  Remote SDP is set for legC:5.1
12:31:14.661  [MS210003] C:5.2:Answer provided. Connection(transcoding mode[unsecure]):127.0.0.1:7016(7017)
12:31:14.661  [MS210001] C:5.1:Answer received. RTP connection[unsecure]: 50.16.57.187:15836(15837)
12:31:14.651  Remote SDP is set for legC:5.1
12:31:14.471  [MS210002] C:5.1:Offer provided. Connection(transcoding mode): 173.79.212.122:9004(9005)
12:31:14.471  [MS210000] C:5.2:Offer received. RTP connection: 127.0.0.1:40614(40615)
12:31:14.461  Remote SDP is set for legC:5.2
12:31:07.751  Currently active calls - 1: [5]
12:31:04.677  Session 137 of leg C:5.1 is confirmed
12:31:04.607  [CM503007]: Call(5): Device joined: sip:[email protected]:40600;rinstance=cc996086cfc88b0b
12:31:04.577  [CM503007]: Call(5): Device joined: sip:[email protected]:5060
12:31:04.577  [MS210005] C:5.1:Answer provided. Connection(proxy mode):173.79.212.122:9004(9005)
12:31:04.577  [MS210001] C:5.2:Answer received. RTP connection[unsecure]: 127.0.0.1:40614(40615)
12:31:04.567  Remote SDP is set for legC:5.2
12:31:04.567  [CM505001]: Ext.8000: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX IVR;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] PBX contact: [sip:[email protected]:5060]
12:31:04.567  [CM503002]: Call(5): Alerting sip:[email protected]:40600;rinstance=cc996086cfc88b0b
12:31:04.386  [CM503025]: Call(5): Calling Ext:Ext.8000@[Dev:sip:[email protected]:40600;rinstance=cc996086cfc88b0b]
12:31:04.376  [MS210004] C:5.2:Offer provided. Connection(proxy mode): 127.0.0.1:7016(7017)
12:31:04.336  [CM503004]: Call(5): Route 1: Ext:Ext.8000@[Dev:sip:[email protected]:40600;rinstance=cc996086cfc88b0b]
12:31:04.326  [CM503010]: Making route(s) to <sip:[email protected]:5060>
12:31:04.326  [MS210000] C:5.1:Offer received. RTP connection: 67.231.8.106:24868(24869)
12:31:04.326  Remote SDP is set for legC:5.1
12:31:04.326  [CM505003]: Provider:[FreePBX] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [FreeSWITCH-mod_sofia/1.0.4-exported] PBX contact: [sip:[email protected]:5060]
12:31:04.296  [CM503001]: Call(5): Incoming call from +15986552467@(Ln.10002@FreePBX) to <sip:[email protected]:5060>
12:31:04.246  [CM503012]: Inbound any hours rule (4334220401) for 10002 forwards to DN:8000
12:31:04.246  Looking for inbound target: called=4334220401; caller=+15986552467
12:31:04.236  [CM500002]: Info on incoming INVITE:
  INVITE sip:[email protected]:5060;rinstance=3c643cbf0185a4a8;received=10.112.189.140:5060 SIP/2.0
  Via: SIP/2.0/UDP pb2proxy-pro-aws03.phoneboothdev.info;branch=z9hG4bK5d3c.2bdf9115.0;received=184.72.227.214
  Via: SIP/2.0/UDP 10.110.26.133;received=10.110.26.133;rport=5060;branch=z9hG4bKU7Q6XecQ3mp8g
  Max-Forwards: 53
  Record-Route: <sip:pb2proxy-pro-aws03.phoneboothdev.info;lr=on;did=399.c4bb5972>
  Contact: <sip:[email protected]:5060>
  To: <sip:[email protected]:5060;rinstance=3c643cbf0185a4a8;received=10.112.189.140:5060>
  From: "+15986552467"<sip:[email protected]>;tag=cccZ3KpyeStHr
  Call-ID: f8f5c8fd-d403-122e-9d90-1231391c1c77
  CSeq: 10330662 INVITE
  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
  Supported: precondition, path, replaces
  User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported
  Allow-Events: talk, refer
  Privacy: none
  P-Asserted-Identity: "+15986552467" <sip:[email protected]>
  Content-Length: 0

 

[jlag]

Re: Digital Receptionist Transfer from external call failure

Here is an easier situation to see perhaps what is the issue.

Here the caller dials in to PBX, DR answers, caller enters extension, extension rings, picks up, no audio but at the same time the caller is presented with music on hold. The extension then places the call on hold, and then re-joins the call and all is working after this.

Maybe somebody can see the difference in ports used after the hold and re-join?

08:10:08.652  [MS210003] C:29.3:Answer provided. Connection(transcoding mode[unsecure]):172.16.0.120:7112(7113)
08:10:08.652  [MS210001] C:29.1:Answer received. RTP connection[unsecure]: 50.16.57.187:11084(11085)
08:10:08.652  Remote SDP is set for legC:29.1
08:10:08.251  [MS210002] C:29.1:Offer provided. Connection(transcoding mode): 142.72.29.17:9028(9029)
08:10:08.241  [MS210000] C:29.3:Offer received. RTP connection: 172.16.0.95:50570(50571)
08:10:08.241  Remote SDP is set for legC:29.3
08:10:04.676  [MS210003] C:29.3:Answer provided. Connection(transcoding mode[unsecure]):172.16.0.120:7112(7113)
08:10:04.676  [MS210001] C:29.1:Answer received. RTP connection[unsecure]: 50.16.57.187:11084(11085)
08:10:04.676  Remote SDP is set for legC:29.1
08:10:04.546  [MS210002] C:29.1:Offer provided. Connection(transcoding mode): 142.72.29.17:9028(9029)
08:10:04.546  [MS210000] C:29.3:Offer received. RTP connection: 0.0.0.0:50570(50571)
08:10:04.546  Remote SDP is set for legC:29.3
08:09:57.265  [MS105000] C:29.2: No RTP packets were received:remoteAddr=127.0.0.1:40650,extAddr=0.0.0.0:0,localAddr=127.0.0.1:7110
08:09:55.863  [CM503007]: Call(29): Device joined: sip:[email protected]:5060
08:09:55.863  [MS210003] C:29.1:Answer provided. Connection(transcoding mode[unsecure]):142.72.29.17:9028(9029)
08:09:55.853  [MS210001] C:29.3:Answer received. RTP connection[unsecure]: 172.16.0.95:50570(50571)
08:09:55.843  Remote SDP is set for legC:29.3
08:09:55.843  [CM505001]: Ext.2243: Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA1001-3.1.19(SE)] PBX contact: [sip:[email protected]:5060]
08:09:55.843  [CM503002]: Call(29): Alerting sip:[email protected]:5060
08:09:51.697  [MS210001] C:29.2:Answer received. RTP connection[unsecure]: 127.0.0.1:40650(40651)
08:09:51.687  Remote SDP is set for legC:29.2
08:09:51.617  [CM503025]: Call(29): Calling Ext:Ext.2243@[Dev:sip:[email protected]:5060]
08:09:51.617  [MS210002] C:29.3:Offer provided. Connection(transcoding mode): 172.16.0.120:7112(7113)
08:09:51.597  [CM503004]: Call(29): Route 1: Ext:Ext.2243@[Dev:sip:[email protected]:5060]
08:09:51.597  [CM503010]: Making route(s) to <sip:[email protected]:5060>
08:09:51.597  Refer: from=<sip:[email protected]:5060>;tag=6049d855; to="+15039984434:4885540400"<sip:[email protected]:5060>;tag=1e27e15d; RefTo=<sip:[email protected]:5060>
08:09:51.547  Currently active calls - 1: [29]
08:09:51.497  [MS210004] C:29.2:Offer provided. Connection(proxy mode): 127.0.0.1:7110(7111)
08:09:51.497  [MS210000] C:29.1:Offer received. RTP connection: 50.16.57.187:11084(11085)
08:09:51.497  Remote SDP is set for legC:29.1
08:09:51.457  [MS210003] C:29.2:Answer provided. Connection(transcoding mode[unsecure]):127.0.0.1:7110(7111)
08:09:51.457  [MS210001] C:29.1:Answer received. RTP connection[unsecure]: 50.16.57.187:11084(11085)
08:09:51.457  Remote SDP is set for legC:29.1
08:09:51.187  [MS210002] C:29.1:Offer provided. Connection(transcoding mode): 142.72.29.17:9028(9029)
08:09:51.177  [MS210000] C:29.2:Offer received. RTP connection: 127.0.0.1:40650(40651)
08:09:51.177  Remote SDP is set for legC:29.2
08:09:41.903  Session 8519 of leg C:29.1 is confirmed
08:09:41.823  [CM503007]: Call(29): Device joined: sip:[email protected]:40600;rinstance=cc996086cfc88b0b
08:09:41.793  [CM503007]: Call(29): Device joined: sip:[email protected]:5060
08:09:41.793  [MS210005] C:29.1:Answer provided. Connection(proxy mode):142.72.29.17:9028(9029)
08:09:41.793  [MS210001] C:29.2:Answer received. RTP connection[unsecure]: 127.0.0.1:40650(40651)
08:09:41.783  Remote SDP is set for legC:29.2
08:09:41.783  [CM505001]: Ext.8000: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX IVR;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] PBX contact: [sip:[email protected]:5060]
08:09:41.783  [CM503002]: Call(29): Alerting sip:[email protected]:40600;rinstance=cc996086cfc88b0b
08:09:41.603  [CM503025]: Call(29): Calling Ext:Ext.8000@[Dev:sip:[email protected]:40600;rinstance=cc996086cfc88b0b]
08:09:41.593  [MS210004] C:29.2:Offer provided. Connection(proxy mode): 127.0.0.1:7110(7111)
08:09:41.553  [CM503004]: Call(29): Route 1: Ext:Ext.8000@[Dev:sip:[email protected]:40600;rinstance=cc996086cfc88b0b]
08:09:41.543  [CM503010]: Making route(s) to <sip:[email protected]:5060>
08:09:41.543  [MS210000] C:29.1:Offer received. RTP connection: 67.231.8.106:17402(17403)
08:09:41.543  Remote SDP is set for legC:29.1
08:09:41.543  [CM505003]: Provider:[FreePBX] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [FreeSWITCH-mod_sofia/1.0.4-exported] PBX contact: [sip:[email protected]:5060]
08:09:41.513  [CM503001]: Call(29): Incoming call from +15039984434@(Ln.10002@FreePBX) to <sip:[email protected]:5060>
08:09:41.463  [CM503012]: Inbound any hours rule (4885540400) for 10002 forwards to DN:8000
08:09:41.463  Looking for inbound target: called=4885540400; caller=+15039984434
08:09:41.463  [CM500002]: Info on incoming INVITE:
  INVITE sip:[email protected]:5060;rinstance=b7b0265111244517;received=10.112.189.140:5060 SIP/2.0
  Via: SIP/2.0/UDP pb2proxy-pro-aws03.phoneboothdev.info;branch=z9hG4bKfe39.c4465b97.0;received=184.72.227.214
  Via: SIP/2.0/UDP 10.110.26.133;received=10.110.26.133;rport=5060;branch=z9hG4bKttUHj5gB9tS5g
  Max-Forwards: 53
  Record-Route: <sip:pb2proxy-pro-aws03.phoneboothdev.info;lr=on;did=c67.d0ea63c5>
  Contact: <sip:[email protected]:5060>
  To: <sip:[email protected]:5060;rinstance=b7b0265111244517;received=10.112.189.140:5060>
  From: "+15039984434"<sip:[email protected]>;tag=NB2pt2XaD9cyN
  Call-ID: a324d3f5-d4a8-122e-9d90-1231391c1c77
  CSeq: 10366024 INVITE
  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
  Supported: precondition, path, replaces
  User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported
  Allow-Events: talk, refer
  Privacy: none
  P-Asserted-Identity: "+15039984434" <sip:[email protected]>
  Content-Length: 0
  
08:09:21.514  Currently active calls [none]

 

[jlag]

Re: Digital Receptionist Transfer from external call failure

Thanks jlag for the answer.

Actually, that was not the answer, it was a different explantion of my problem. I thought it was port mapping problems due to NAT, but I put the PBX on a outside IP directly and the problem remains, can't transfer from the DR to any extensions from an external call unless the caller is put on hold after the called extension picks up the call.

 

[SY]

Re: Digital Receptionist Transfer from external call failure

You need to turn off "Supports re-INVITEs" option in the VoIP provider configuration.
It should resolve the problem.

 

[jlag]

Re: Digital Receptionist Transfer from external call failure

You need to turn off "Supports re-INVITEs" option in the VoIP provider configuration.
It should resolve the problem.

Yes, my original post states that I have tried those settings to no avail. Strange because I have checked numerous times that REINVITE is off on that extension, "Supports Reinvite" above.

 

[SY]

Re: Digital Receptionist Transfer from external call failure

You need to turn off "Supports re-INVITEs" option in the VoIP provider configuration.
It should resolve the problem.

Yes, my original post states that I have tried those settings to no avail. Strange because I have checked numerous times that REINVITE is off on that extension, "Supports Reinvite" above.

All of the logs, you provided above, say that the configuration of VoIP provider(line 10002. "User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported") is:
Supports re-invites=on
PBX delivers audio=off

Do you think that the comment was made using a "blind manner"?

P.S. "received" parameter still appears in unexpected places...

 

[jlag]

Re: Digital Receptionist Transfer from external call failure

You need to turn off "Supports re-INVITEs" option in the VoIP provider configuration.
It should resolve the problem.

Yes, my original post states that I have tried those settings to no avail. Strange because I have checked numerous times that REINVITE is off on that extension, "Supports Reinvite" above.

All of the logs, you provided above, say that the configuration of VoIP provider(line 10002. "User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported") is:
Supports re-invites=on
PBX delivers audio=off

Do you think that the comment was made using a "blind manner"?

P.S. "received" parameter still appears in unexpected places...

The comment was most certainly not made blind but I do owe you an apology. I was thinking this was strictly an extension side issue, not a provider side issue. I have switched REINVITE to off in the Provider settings and, yes, the problem is solved.

However, when I use this trunk on a different PBX, I have the CANREINVITE for the provider = 'true', so I assumed that this would be the same with the 3CX PBX.

I have also verified that the trunk does support and use a direct RTP media path on the other PBX. So I am thinking that this is perhaps related to the default REINVITE structure that is going out to the provider from 3CX. Perhaps there is a way I could structure this reinvite so that we can establish a direct media path, but I don't see the reinvite packet structure in the verbose logging in 3CX.

Please let me know where this does and/or does not make sense if you have the time.

PS my comment above:

tried those settings to no avail. Strange because I have checked numerous times that REINVITE is off on that extension, "Supports Reinvite" above.

Was supposed to read:

tried those settings to no avail. Strange because I have checked numerous times that REINVITE is off on that extension, but you can still see that "Supports Reinvite" is enabled above for that extension.
08:09:55.843 [CM505001]: Ext.2243: Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA1001-3.1.19(SE)] PBX contact: [sip:[email protected]:5060]

Are these capabilities reported from 3CX config for that extension or is this somehow read from the device? I have verified that Reinvite for this extension is OFF.

 

[SY]

Re: Digital Receptionist Transfer from external call failure

The comment was most certainly not made blind but I do owe you an apology. I was thinking this was strictly an extension side issue, not a provider side issue. I have switched REINVITE to off in the Provider settings and, yes, the problem is solved.

Thanks for the update and explanation. I'm sorry for my previous question. It was too "straight".

I have also verified that the trunk does support and use a direct RTP media path on the other PBX. So I am thinking that this is perhaps related to the default REINVITE structure that is going out to the provider from 3CX. Perhaps there is a way I could structure this reinvite so that we can establish a direct media path, but I don't see the reinvite packet structure in the verbose logging in 3CX.

"Server Activity Log" page of the management console doesn't show all requests. They can be found in 3CXPhonesystem.trace.log, but wireshark can provide much better information. Install it on PBX host and make a capture of the call. You will see all the requests sent and received by PBX. It will be interesting to look on it.

Thanks