SIP Trunk to Cisco, names are not displayed

asuhov

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Hello everyone,

We have configured a SIP trunk between 3CX and Cisco Call Manager. All works fine except that the names are not displayed on Cisco phones when call is initiated from 3CX

Is there any way to make it send the names
 
Hello asuhov,

can you make a 'Capture' SIP trace, when you make a call out on the trunk to Cisco.
See if the name is passed, and in what SIP field?

Paulo
 
Hello,
Hello asuhov,

can you make a 'Capture' SIP trace, when you make a call out on the trunk to Cisco.
See if the name is passed, and in what SIP field?

Paulo
Hello Paulo,

I was able to fix it. Now the other problem is when call is going from Cisco to 3CX, 3CX shows only number. Below is the trace captured on 3CX - seems like Cisco is not sending Name information even though all required settings are configured on Cisco side

4719 is Cisco, 8300 is 3CX

P.S. Just noticed, that 127.0.0.1 is shown for 4719, I believe it should be our Cisco's server IP

L:269.5[Extn:8300{7b198376cf42}] initiating call with InvOUT Send Req INVITE from 0.0.0.0:0 tid=fd1e22082c37a601 Call-ID=qe6giYLGSsFsEkq-chSESQ..:INVITE sip:[email protected]:5060;rinstance=0-8913f699-4b62-40f0-93c4-849393f4c00d SIP/2.0Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---fd1e22082c37a601;rportMax-Forwards: 70Route: <sip:[email protected]:5080;user=proxy;tunnelip=192.168.206.147;uri=clnt.0-8913f699-4b62-40f0-93c4-849393f4c00d_21269380718>Contact: <sip:[email protected]:5060>To: <sip:[email protected]>From: "4719"<sip:[email protected]:5060;nf=e>;tag=5b0d1270Call-ID: qe6giYLGSsFsEkq-chSESQ..CSeq: 1 INVITEAlert-Info: <http://www.notused.invalidtld>;info=externalAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATEContent-Type: application/sdpSupported: replaces, timerContent-Length: 427v=0o=3cxPS 275465109504 88701140993 IN IP4 127.1.2.3s=3cxPS Audio callc=IN IP4 127.1.2.3t=0 0m=audio 2876 RTP/AVP 0 8 9 18 112 102 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:9 G722/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:112 opus/48000/2a=fmtp:112 maxplaybackrate=16000;sprop-maxcapturerate=16000a=rtpmap:102 telephone-event/48000a=rtpmap:101 telephone-event/8000a=sendrecv
 
Hello asuhov,

I think this is NOT a trace from the SIP TRUNK, this is from internal call handeling.

Paulo
 

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