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IP Phone CP-7821 & Conference problem with Asterisk IP Telephony

bghayyadh
Level 1
Level 1

‎11-08-2022 02:17 PM

Hello;

I succeeded to let CP-7821 to register with Asterisk IP Telephony and doing calls and doing transfer using the soft button. But I tried to do conference (3 parties), actually it did not succeed to complete it. Actually when doing conference, it should keep the parties connected (not placing them on hold as long I did not press hold button), even, the phone behavior was not like this, but rather it was place the parties on hold. Below is the Asterisk SIP trace for the conference:

 

Spoiler

[Nov  7 21:39:16] <--- SIP read from UDP:192.168.60.3:5060 --->

[Nov  7 21:39:16] SIP/2.0 200 OK

[Nov  7 21:39:16] Via: SIP/2.0/UDP 192.168.50.12:5060;branch=z9hG4bK19e020b0;rport

[Nov  7 21:39:16] From: "asterisk" <sip:asterisk@192.168.50.12>;tag=as0630b625

[Nov  7 21:39:16] To: <sip:1001@192.168.60.3:5060;transport=udp>;tag=00aa6e0ef0f900063218fbc8-0ebee137

[Nov  7 21:39:16] Call-ID: 1a2f48e00d0f536132cb6d8721542227@192.168.50.12:5060

[Nov  7 21:39:16] Session-ID: 1a9cc6a600105000a00000aa6e0ef0f9;remote=00000000000000000000000000000000

[Nov  7 21:39:16] Date: Mon, 07 Nov 2022 19:39:16 GMT

[Nov  7 21:39:16] CSeq: 102 OPTIONS

[Nov  7 21:39:16] Server: Cisco-CP7821/14.1.1

[Nov  7 21:39:16] Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE

[Nov  7 21:39:16] Allow-Events: kpml,dialog,refer

[Nov  7 21:39:16] Accept: application/sdp,multipart/mixed,multipart/alternative

[Nov  7 21:39:16] Accept-Encoding: identity

[Nov  7 21:39:16] Accept-Language: en

[Nov  7 21:39:16] 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 21:39:16] Content-Length: 299

[Nov  7 21:39:16] Content-Type: application/sdp

[Nov  7 21:39:16] Content-Disposition: session;handling=optional

[Nov  7 21:39:16]

[Nov  7 21:39:16] v=0

[Nov  7 21:39:16] o=Cisco-SIPUA 6337 0 IN IP4 192.168.60.3

[Nov  7 21:39:16] s=SIP Call

[Nov  7 21:39:16] t=0 0

[Nov  7 21:39:16] m=audio 0 RTP/AVP 0 8 116 18 101

[Nov  7 21:39:16] b=TIAS:64000

[Nov  7 21:39:16] a=rtpmap:0 PCMU/8000

[Nov  7 21:39:16] a=rtpmap:8 PCMA/8000

[Nov  7 21:39:16] a=rtpmap:116 iLBC/8000

[Nov  7 21:39:16] a=fmtp:116 mode=20

[Nov  7 21:39:16] a=rtpmap:18 G729/8000

[Nov  7 21:39:16] a=fmtp:18 annexb=yes

[Nov  7 21:39:16] a=rtpmap:101 telephone-event/8000

[Nov  7 21:39:16] a=fmtp:101 0-15

[Nov  7 21:39:16] <------------->

[Nov  7 21:39:16] --- (18 headers 14 lines) ---

[Nov  7 21:39:16] Really destroying SIP dialog '1a2f48e00d0f536132cb6d8721542227@192.168.50.12:5060' Method: OPTIONS

[Nov  7 21:39:16]

[Nov  7 21:39:16] <--- SIP read from UDP:192.168.60.3:5060 --->

[Nov  7 21:39:16] SIP/2.0 200 OK

[Nov  7 21:39:16] Via: SIP/2.0/UDP 192.168.50.12:5060;branch=z9hG4bK5fac5e6f;rport

[Nov  7 21:39:16] From: "asterisk" <sip:asterisk@192.168.50.12>;tag=as601b0692

[Nov  7 21:39:16] To: <sip:1001@192.168.60.3:5060;transport=udp>

[Nov  7 21:39:16] Call-ID: 14de64323a028d1641860fd778ba24f9@192.168.50.12:5060

[Nov  7 21:39:16] Session-ID: 1a9cc6a600105000a00000aa6e0ef0f9;remote=00000000000000000000000000000000

[Nov  7 21:39:16] Date: Mon, 07 Nov 2022 19:39:16 GMT

[Nov  7 21:39:16] CSeq: 102 NOTIFY

[Nov  7 21:39:16] Content-Length: 0

[Nov  7 21:39:16]

[Nov  7 21:39:16] <------------->

[Nov  7 21:39:16] --- (9 headers 0 lines) ---

[Nov  7 21:39:16] Really destroying SIP dialog '14de64323a028d1641860fd778ba24f9@192.168.50.12:5060' Method: NOTIFY

vicibox10*CLI>

vicibox10*CLI>

vicibox10*CLI>

vicibox10*CLI>

vicibox10*CLI>

vicibox10*CLI>

[Nov  7 21:39:48] Really destroying SIP dialog '00aa6e0e-f0f90002-0492104b-758d7877@192.168.60.3' Method: REGISTER

[Nov  7 21:40:01]   == Manager 'sendcron' logged on from 127.0.0.1

[Nov  7 21:40:01]   == Manager 'sendcron' logged on from 127.0.0.1

[Nov  7 21:40:01]   == Manager 'sendcron' logged off from 127.0.0.1

[Nov  7 21:40:02]   == Manager 'sendcron' logged off from 127.0.0.1

[Nov  7 21:40:06]   == Manager 'sendcron' logged on from 127.0.0.1

[Nov  7 21:40:06]   == Manager 'sendcron' logged off from 127.0.0.1

[Nov  7 21:40:16] Reliably Transmitting (NAT) to 192.168.60.3:5060:

[Nov  7 21:40:16] OPTIONS sip:1001@192.168.60.3:5060;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: 2437ecd14190056e01e8c27238134e42@192.168.50.12:5060

[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: 2437ecd14190056e01e8c27238134e42@192.168.50.12:5060

[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 '2437ecd14190056e01e8c27238134e42@192.168.50.12:5060' Method: OPTIONS

vicibox10*CLI>

 

It seems there is something custom for Cisco, how I can overcome this problem?
What is need to be added or removed in asterisk configuration or the IP Phone XML configuration.

Looking to hear from you.

Regards

Bilal

 

 

 

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