I have a client who is on google cloud hosted 3CX. Verson 16.0.493
Everything is working good and they have asked me to install a paging system for their yard.
I have the algo successfully registering with the PBX with a manual setup. not through the SBC.
But when i try to intercom from a phone i get the following in the logs of the 8180. And a busy signal after 2 rings from any phone.
It keeps trying to sip send with the Local PBX IP
I have tried modifying the Put Public IP in SIP VIA Header and Select which IP to use in 'Contact' (SIP) and 'Connection'(SDP) fields
and it still does not intercom.
What setting may i be missing?
Everything is working good and they have asked me to install a paging system for their yard.
I have the algo successfully registering with the PBX with a manual setup. not through the SBC.
But when i try to intercom from a phone i get the following in the logs of the 8180. And a busy signal after 2 rings from any phone.
Code:
May 11 14:25:37 phone: [pudp] sip recv External PBX IP:5060, size = 1188
May 11 14:25:37 phone: [pudp] - INVITE sip:[email protected]:5060;intercom=true SIP/2.0[]Via: SIP/2.0/UDP Local PBX IP:5060;branch=z9hG4bK-524287-1---fa88d75a5d8c5d24;rport[]Max-Forwards: 70[]Contact: <sip:250@External PBX IP:5060>[]To
May 11 14:25:37 phone: [pudp] - : <sip:250@External FQDN>;intercom=true[]From: "PG803::Pager"<sip:803@External PBX IP:5060>;tag=99934908[]Call-ID: zS6DRluXaRfmkO_vhelKXw..[]CSeq: 1 INVITE[]Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
May 11 14:25:37 phone: [pudp] - REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE[]Call-Info: <sip:3cx.pbx>;answer-after=0[]Content-Type: application/sdp[]Supported: replaces, timer[]Content-Length: 605[][]v=0[]o=3cxPS 7164
May 11 14:25:37 phone: [pudp] - 651020222464 18767249082941441 IN IP4 External PBX IP[]s=3cxPS Audio call[]c=IN IP4 External PBX IP[]t=0 0[]m=audio 10488 RTP/AVP 0 8 3 112 9 13 18 97 111 110 101[]a=rtpmap:0 PCMU/8000[]a=rtpmap:8 PCMA/80
May 11 14:25:37 phone: [pudp] - 00[]a=rtpmap:3 GSM/8000[]a=rtpmap:112 opus/48000/2[]a=fmtp:112 maxplaybackrate=16000;sprop-maxcapturerate=16000[]a=rtpmap:9 G722/8000[]a=rtpmap:13 CN/8000[]a=rtpmap:18 G729/8000[]a=fmtp:18 annexb=no[]
May 11 14:25:37 phone: [pudp] - a=rtpmap:97 SPEEX/8000[]a=rtpmap:111 iLBC/8000[]a=fmtp:111 mode=30[]a=rtpmap:110 iLBC/8000[]a=fmtp:110 mode=20[]a=rtpmap:101 telephone-event/8000[]a=fmtp:101 0-15[]a=ptime:20[]a=sendrecv[]
May 11 14:25:37 phone: [psdp] sdp audio addr = External PBX IP, rtp = 10488, rtcp = 10489
May 11 14:25:37 phone: [sipbg] cmd: <execute>
May 11 14:25:37 phone: [bridge] mq cmd: <page media remote>
May 11 14:25:37 phone: [audbg] cmd: <call open remote External PBX IP 10488 10489 0 20 101>
May 11 14:25:37 phone: [pudp] sip send Local PBX IP:5060, size = 639
May 11 14:25:37 phone: [pudp] - SIP/2.0 200 OK[]Via: SIP/2.0/UDP Local PBX IP:5060;branch=z9hG4bK-524287-1---fa88d75a5d8c5d24;rport[]From: "PG803::Pager" <sip:803@External PBX IP:5060>;tag=99934908[]To: <sip:250@External FQDN>;inter
May 11 14:25:37 phone: [pudp] - com=true;tag=1083630554[]Call-ID: zS6DRluXaRfmkO_vhelKXw..[]CSeq: 1 INVITE[]Contact: <sip:[email protected]:5060>[]Content-Type: application/sdp[]Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY,
May 11 14:25:37 phone: [pudp] - OPTIONS[]Content-Length: 204[][]v=0[]o=- 1426686933 1426686933 IN IP4 192.168.40.156[]s=apsip[]c=IN IP4 192.168.40.156[]t=0 0[]m=audio 3000 RTP/AVP 0 8 101[]a=rtpmap:0 PCMU/8000[]a=rtpmap:8 PCMA/80
May 11 14:25:37 phone: [pudp] - 00[]a=rtpmap:101 telephone-event/8000[]
May 11 14:25:37 phone: [sipbg] cmd: <execute>
May 11 14:25:37 phone: [sipbg] cmd: <tr delete 0xc94c008>
May 11 14:25:37 phone: [pudp] sip send Local PBX IP:5060, size = 639
May 11 14:25:37 phone: [pudp] - SIP/2.0 200 OK[]Via: SIP/2.0/UDP Local PBX IP:5060;branch=z9hG4bK-524287-1---fa88d75a5d8c5d24;rport[]From: "PG803::Pager" <sip:803@External PBX IP:5060>;tag=99934908[]To: <sip:250@External FQDN>;inter
May 11 14:25:37 phone: [pudp] - com=true;tag=1083630554[]Call-ID: zS6DRluXaRfmkO_vhelKXw..[]CSeq: 1 INVITE[]Contact: <sip:[email protected]:5060>[]Content-Type: application/sdp[]Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY,
May 11 14:25:37 phone: [pudp] - OPTIONS[]Content-Length: 204[][]v=0[]o=- 1426686933 1426686933 IN IP4 192.168.40.156[]s=apsip[]c=IN IP4 192.168.40.156[]t=0 0[]m=audio 3000 RTP/AVP 0 8 101[]a=rtpmap:0 PCMU/8000[]a=rtpmap:8 PCMA/80
May 11 14:25:37 phone: [pudp] - 00[]a=rtpmap:101 telephone-event/8000[]It keeps trying to sip send with the Local PBX IP
I have tried modifying the Put Public IP in SIP VIA Header and Select which IP to use in 'Contact' (SIP) and 'Connection'(SDP) fields
and it still does not intercom.
What setting may i be missing?
