Asterisk PBX with Cisco SIP Phone conference not working

Asterisk Asterisk Support
1

Hi All,

We are using asterisk pbx for a long time . Recently we converted some cisco phone to SIP firmware and all models are working fine except Model 6921 .
From this phone inbound and outbound call are working fine but while creating conference call we are getting “Unable to complete conference” message on the screen . Meantime we checked asterisk log and found below error .
Please help us to resolve this issue…

9295bc997000c00005387-000039ec –To-tag
[2017-04-26 15:40:09] DEBUG[1466]: acl.c:946 ast_ouraddrfor: For destination ‘10.254.254.41’, our source address is ‘10.254.254.140’.
[2017-04-26 15:40:09] DEBUG[1466]: chan_sip.c:3903 ast_sip_ouraddrfor: Setting AST_TRANSPORT_UDP with address 10.254.254.140:5060
[2017-04-26 15:40:09] DEBUG[1466]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting ‘10.254.254.41:5060’ into…
[2017-04-26 15:40:09] DEBUG[1466]: netsock2.c:226 ast_sockaddr_split_hostport: …host ‘10.254.254.41’ and port ‘5060’.
[2017-04-26 15:40:09] DEBUG[1466]: chan_sip.c:8981 __sip_alloc: Allocating new SIP dialog for f029295b-c9970004-00007f8d-000045ec@10.254.254.41 – REFER (No RTP)
[2017-04-26 15:40:09] DEBUG[1466]: chan_sip.c:28628 handle_incoming: **** Received REFER (9) – Command in SIP REFER
[2017-04-26 15:40:09] DEBUG[1466]: chan_sip.c:26793 handle_request_refer: Call f029295b-c9970004-00007f8d-000045ec@10.254.254.41: Declined REFER, outside of dialog…
[2017-04-26 15:40:09] DEBUG[1466]: chan_sip.c:3746 __sip_xmit: Trying to put ‘SIP/2.0 603’ onto UDP socket destined for 10.254.254.41:5060
[2017-04-26 15:40:09] DEBUG[1466]: chan_sip.c:3402 sip_alreadygone: Setting SIP_ALREADYGONE on dialog f029295b-c9970004-00007f8d-000045ec@10.254.254.41
[2017-04-26 15:40:09] DEBUG[1466]: chan_sip.c:6579 sip_pvt_dtor: Destroying SIP dialog f029295b-c9970004-00007f8d-000045ec@10.254.254.41
[2017-04-26 15:40:09] DEBUG[10600][C-00000021]: res_rtp_asterisk.c:3519 ast_rtp_write: No remote address on RTP instance ‘0x7fcf24008df0’ so dropping frame
[2017-04-26 15:40:09] DEBUG[1466]: chan_sip.c:9394 __find_call: = Looking for Call ID: OutOfDialog–000b-000076ad-00003db3@10.254.254.41 (Checking From) –From tag f029295bc9970026000027b4-00001f78 –To-tag
[2017-04-26 15:40:09] DEBUG[1466]: acl.c:946 ast_ouraddrfor: For destination ‘10.254.254.41’, our source address is ‘10.254.254.140’.
[2017-04-26 15:40:09] DEBUG[1466]: chan_sip.c:3903 ast_sip_ouraddrfor: Setting AST_TRANSPORT_UDP with address 10.254.254.140:5060
[2017-04-26 15:40:09] DEBUG[1466]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting ‘10.254.254.41:5060’ into…
[2017-04-26 15:40:09] DEBUG[1466]: netsock2.c:226 ast_sockaddr_split_hostport: …host ‘10.254.254.41’ and port ‘5060’.
[2017-04-26 15:40:09] DEBUG[1466]: chan_sip.c:8981 __sip_alloc: Allocating new SIP dialog for OutOfDialog–000b-000076ad-00003db3@10.254.254.41 – REFER (No RTP)
[2017-04-26 15:40:09] DEBUG[1466]: chan_sip.c:28628 handle_incoming: **** Received REFER (9) – Command in SIP REFER
[2017-04-26 15:40:09] DEBUG[1466]: chan_sip.c:26793 handle_request_refer: Call OutOfDialog–000b-000076ad-00003db3@10.254.254.41: Declined REFER, outside of dialog…
[2017-04-26 15:40:09] DEBUG[1466]: chan_sip.c:3746 __sip_xmit: Trying to put ‘SIP/2.0 603’ onto UDP socket destined for 10.254.254.41:5060
[2017-04-26 15:40:09] DEBUG[1466]: chan_sip.c:3402 sip_alreadygone: Setting SIP_ALREADYGONE on dialog OutOfDialog–000b-000076ad-00003db3@10.254.254.41
[2017-04-26 15:40:09] DEBUG[1466]: chan_sip.c:6579 sip_pvt_dtor: Destroying SIP dialog OutOfDialog–000b-000076ad-00003db3@10.254.254.41
voip*CLI>
[2017-04-26 15:40:09] DEBUG[10600][C-00000021]: res_rtp_asterisk.c:3519 ast_rtp_write: No remote address on RTP instance ‘0x7fcf24008df0’ so dropping frame
[2017-04-26 15:40:09] DEBUG[10600][C-00000021]: res_rtp_asterisk.c:3519 ast_rtp_write: No remote address on RTP instance ‘0x7fcf24008df0’ so dropping frame

SIP Debug logs are,

<— SIP read from UDP:10.254.254.41:5060 —>
REFER sip:10.254.254.140 SIP/2.0
Via: SIP/2.0/UDP 10.254.254.41:5060;branch=z9hG4bK00006b2d
From: “3005” <sip:3005@10.254.254.140>;tag=f029295bc9970024000035ba-00006805
To: <sip:3005@10.254.254.140>
Call-ID: OutOfDialog–0009-000006d3-00002326@10.254.254.41
Max-Forwards: 70
Date: THU, 27 APR 2017 06:49:42 GMT
CSeq: 101 REFER
User-Agent: Cisco-CP6921/9.4.1
Contact: <sip:3005@10.254.254.41:5060;transport=udp>
Referred-By: “3005” <sip:3005@10.254.254.140>
Refer-To: cid:000059d5@10.254.254.41
Require: norefersub
Authorization: Digest username=”3005″,realm=”asterisk”,uri=”sip:10.254.254.140″,response=”cb6d379c811542e1302188af107bf534″,nonce=”51dcc2f2″,algorithm=MD5
Content-Length: 862
Content-Type: application/x-cisco-remotecc-request+xml
Content-Disposition: session;handling=required
Content-Id: <000059d5@10.254.254.41>
<?xml version=”1.0″ encoding=”UTF-8″?>
<x-cisco-remotecc-request>
<softkeyeventmsg>
<softkeyevent>Cancel</softkeyevent>
<dialogid>
<callid>f029295b-c9970007-0000741b-000072dc@10.254.254.41</callid>
<localtag>f029295bc997002200001788-000051f9</localtag>
<remotetag>as404a8f7f</remotetag>
</dialogid>
<consultdialogid>
<callid>f029295b-c9970008-000016e4-000018e3@10.254.254.41</callid>
<localtag>f029295bc997002300003895-0000576c</localtag>
<remotetag>as11d997b9</remotetag>
</consultdialogid>
<joindialogid>
<callid></callid>
<localtag></localtag>
<remotetag></remotetag>
</joindialogid>
<linenumber>0</linenumber>
<participantnum>0</participantnum>
<userdata></userdata>
<softkeyid>0</softkeyid>
<applicationid>0</applicationid>
<eventdata><invocationtype>implicit</invocationtype></eventdata>
<state>false</state>
</softkeyeventmsg>
</x-cisco-remotecc-request>
<————->
— (18 headers 28 lines) —
Sending to 10.254.254.41:5060 (NAT)
Call OutOfDialog–0009-000006d3-00002326@10.254.254.41 got a SIP call transfer from caller: (REFER)!
<— Transmitting (NAT) to 10.254.254.41:5060 —>
SIP/2.0 603 Declined (No dialog)
Via: SIP/2.0/UDP 10.254.254.41:5060;branch=z9hG4bK00006b2d;received=10.254.254.41;rport=5060
From: “3005” <sip:3005@10.254.254.140>;tag=f029295bc9970024000035ba-00006805
To: <sip:3005@10.254.254.140>;tag=as4edc8a57
Call-ID: OutOfDialog–0009-000006d3-00002326@10.254.254.41
CSeq: 101 REFER
Server: OMBUTEL PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
2

There are no ERRORs in the log.

You have received a Cisco proprietary request, which will have to be eliminated on the Cisco side.

3

Hi ,

Thanks for your reply,

But why its giving below error . Other 79xx series phones are working fine with conferences. This phone also working fine with single channels calls . I checked audio codecs , that part also working fine.
We changed almost all available firmware for this phone . No hope.
Is it related to any bridging issue ?

4

Because it is using a Cisco proprietary format which Asterisk is unable to understand and process.

Specifically it is using a REFER request with a Refer-To which is not a SIP URI, but CID URI, i.e. a reference to a part in the body of the request. Asterisk understands neither CID: URIs, nor the proprietary XML in the body, so has no option but to refuse the REFER.

5

It is also entirely possible that it is trying to establish a conference server side using this proprietary method, which Asterisk would not support.

6

Hello Jamesarems,

Good day!..

7962, 7931
I would be really appreciate you as you already configured Cisco 79xx series phones with CONFERENCES, recently we have implemented asterisk and we have configured 7962,7931 SIP firmware via TFTP everything are working fine except conferences.

My humble request you to that could you please help me out on this we can i make the conference calls as i already make test conference calls through Xlite 4.8 new version.

Please waiting for your kind response.

Thank you in Advance.