Unable to do conference in Cisco IP Phone

Hello;

Actually I faced same problem in letting Cisco IP Phones doing conference (3 way conference), I found a post from Mr. Citex but I am not able to place the URL because I am new user.

But did not mention what was done to be resolved. Appreciate for the help please.

From the other side, I believe that Cisco IP Phone is sending custom messages that is sent for Cisco Call Manager which are not standard so Asterisk can not understand it. I am going to place the sip trace from Asterisk for the conference if someone can help me in resolving this problem:

[Nov 7 21:40:16] Reliably Transmitting (NAT) to 192.168.60.3:5060:
[Nov 7 21:40:16] OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
[Nov 7 21:40:16] Via: SIP/2.0/UDP 192.168.50.12:5060;branch=z9hG4bK48ed9af0;rport
[Nov 7 21:40:16] Max-Forwards: 70
[Nov 7 21:40:16] From: “asterisk” <sip:asterisk @192.168.50.12>;tag=as1695d502
[Nov 7 21:40:16] To: <sip:1001 @192.168.60.3:5060;transport=udp>
[Nov 7 21:40:16] Contact: <sip:asterisk @192.168.50.12:5060>
[Nov 7 21:40:16] Call-ID: [email protected]:5060
[Nov 7 21:40:16] CSeq: 102 OPTIONS
[Nov 7 21:40:16] User-Agent: Asterisk PBX 13.38.2-vici
[Nov 7 21:40:16] Date: Mon, 07 Nov 2022 19:40:16 GMT
[Nov 7 21:40:16] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 7 21:40:16] Supported: replaces, timer
[Nov 7 21:40:16] Content-Length: 0
[Nov 7 21:40:16]
[Nov 7 21:40:16]
[Nov 7 21:40:16] —
[Nov 7 21:40:16]
[Nov 7 21:40:16] <— SIP read from UDP:192.168.60.3:5060 —>
[Nov 7 21:40:16] SIP/2.0 200 OK
[Nov 7 21:40:16] Via: SIP/2.0/UDP 192.168.50.12:5060;branch=z9hG4bK48ed9af0;rport
[Nov 7 21:40:16] From: “asterisk” <sip:asterisk @192.168.50.12>;tag=as1695d502
[Nov 7 21:40:16] To: <sip:1001 @192.168.60.3:5060;transport=udp>;tag=00aa6e0ef0f90007092fc16e-14dca2f1
[Nov 7 21:40:16] Call-ID: [email protected]:5060
[Nov 7 21:40:16] Session-ID: 1a9cc6a600105000a00000aa6e0ef0f9;remote=00000000000000000000000000000000
[Nov 7 21:40:16] Date: Mon, 07 Nov 2022 19:40:16 GMT
[Nov 7 21:40:16] CSeq: 102 OPTIONS
[Nov 7 21:40:16] Server: Cisco-CP7821/14.1.1
[Nov 7 21:40:16] Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
[Nov 7 21:40:16] Allow-Events: kpml,dialog,refer
[Nov 7 21:40:16] Accept: application/sdp,multipart/mixed,multipart/alternative
[Nov 7 21:40:16] Accept-Encoding: identity
[Nov 7 21:40:16] Accept-Language: en
[Nov 7 21:40:16] Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0
[Nov 7 21:40:16] Content-Length: 299
[Nov 7 21:40:16] Content-Type: application/sdp
[Nov 7 21:40:16] Content-Disposition: session;handling=optional
[Nov 7 21:40:16]
[Nov 7 21:40:16] v=0
[Nov 7 21:40:16] o=Cisco-SIPUA 2023 0 IN IP4 192.168.60.3
[Nov 7 21:40:16] s=SIP Call
[Nov 7 21:40:16] t=0 0
[Nov 7 21:40:16] m=audio 0 RTP/AVP 0 8 116 18 101
[Nov 7 21:40:16] b=TIAS:64000
[Nov 7 21:40:16] a=rtpmap:0 PCMU/8000
[Nov 7 21:40:16] a=rtpmap:8 PCMA/8000
[Nov 7 21:40:16] a=rtpmap:116 iLBC/8000
[Nov 7 21:40:16] a=fmtp:116 mode=20
[Nov 7 21:40:16] a=rtpmap:18 G729/8000
[Nov 7 21:40:16] a=fmtp:18 annexb=yes
[Nov 7 21:40:16] a=rtpmap:101 telephone-event/8000
[Nov 7 21:40:16] a=fmtp:101 0-15
[Nov 7 21:40:16] <------------->
[Nov 7 21:40:16] — (18 headers 14 lines) —
[Nov 7 21:40:16] Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

Regards
Bilal

Truth of the matter, Cisco Phones work fine within the Cisco ecosystem they just aren’t powerful enough to work without that so just ‘not a very good SIP phone’ better to spend your time and resources finding an alternative.