Cisco 79xx/89xx specific local conference bridge issue

Hoping someone might be able to shed some light on this (otherwise it will at least make the issue known because I didn’t get any similar sightings googling).

For some reason I’m unable to set up a local conference call on a Cisco 7945 and 8961. (using the phones internal bridge that is). The phones will nicely announce on screen they couldn’t set up the conference when trying to do so.

Versions used;
Asterisk + Cisco 7945 fw 9.0.3
Asterisk + Cisco 8961 fw

I’m using TCP transport in both setups. I’ve tried insecure port and invite and allow reinvites, but to no avail.

I’m having no issues in another environment running Asterisk 1.4.22 + Cisco 7945 fw 8.5.2 in udp.

I know I should be able to change the transport to udp in the config files of the phones. I’ll try that as soon as I have some time.

Maybe Cisco altered the way they do this from fw v9.x on.

It sure would be nice to retain this feature for users with the current releases.

Below the SIP REFER request from the 8961 that I believe should set up the conference and is declined by lack of a dialog by Asterisk. I have no reference yet to the SIP messaging in a working old setup, but I’ll try to obtain that information when I can.

To: <sip:>^M
Call-ID: [email protected]^M
Date: Thu, 30 Dec 2010 07:00:35 GMT^M
CSeq: 1903 REFER^M
User-Agent: Cisco-CP8961/9.0.1^M
Accept: application/x-cisco-remotecc-response+xml^M
Expires: 60^M
Max-Forwards: 70^M
Contact: <sip:[email protected]:51904;transport=TCP>^M
Referred-By: "displayname" <sip:[email protected]>^M
Refer-To: cid:[email protected]^M
Content-Id: <[email protected]>^M
Content-Length: 961^M
Content-Type: application/x-cisco-remotecc-request+xml^M
Content-Disposition: session;handling=required^M
<?xml version="1.0" encoding="UTF-8"?> ^M
   <callid>[email protected]</callid>^M
   <callid>[email protected]</callid>^M

[Dec 30 08:00:37] VERBOSE[8103] chan_sip.c: --- (19 headers 30 lines) ---
[Dec 30 08:00:37] VERBOSE[8103] chan_sip.c:
<--- Transmitting (NAT) to --->
SIP/2.0 603 Declined (no dialog)^M
Via: SIP/2.0/TCP;branch=z9hG4bK53432641;received=^M
From: "displayname" <sip:[email protected]>;t



Thank you SkykingOH,

I’m sorry, I wasn’t clear; the calls actually get set up without issues, it’s just when doing the local bridge that fails. Actions;

  1. Call party 1
  2. Hit the confrn button on the phone
  3. Call party 2
  4. Hit the confrn button again to complete
  5. All parties should be joined, but aren’t, from fw v9 on a message is displayed that the conference couldn’t be set up. The ongoing 2 calls remain as they were.

I tested with Asterisk 1.6.15 + a Cisco 7941 v8.5.2, which also failed to do the conference. This also wrote of TCP as relevant because it used UDP.

I checked chan_sip.c of 1.6 and noticed that Asterisk doesn’t support the REFER method out of dialog. Looking at the SIP trace again the REFER request Call-ID doesn’t equal one of the two calls previously set up. So the out of dialog handling makes perfect sense.

I checked chan_sip.c of 1.4 and 1.8 to see if it was any different, but doesn’t seem to be.

I’ll check a 1.4 install + an even older Cisco firmware just to verify my own sanity… :wink: (plus it will hopefully provide me a clue on what a working dialog looks like/requires)

With best regards,


I see you got the ‘603 declined’, really need to have a look at the SDP message in the SIP invite to see why Asterisk declined to setup the call. Usually this is caused by a CODEC mismatch.

Did you ever get this to work? I am having a similiar issue with my cisco 8961 phones. When I try to do a conference I get an error on the phone that says “Unable to complete conference”. Any help would be greatly appreciated.