Massimiliano Stucchi
2009-Apr-16 14:08 UTC
[asterisk-users] Problem transferring calls between Cisco 7940 with SIP firmware
Hi all, I'm having a strange problem with a bunch of cisco 7940G with SIP firmware. The problem arises when transferring a call coming in from a SIP account to another phone. The call connects, but for the first 10 seconds there is a situation with one-way audio, then it turns into a fully working call. I've googled extensively, but couldn't find much about this situation. The problem is also present using the SCCP firmware, though. All the phones are using a separate vlan with the server as well, so there's no nat that has to be implied in here, and I can't understand what's causing the problem. What's more interesting is that if you send any dtmf over from the phone which received the transferred call, the one-way audio goes away and the call is fine immediately. Here are some config files: sip.conf ---- [902] username=902 secret=**** context=prova host=dynamic type=friend canreinvite=no qualify=yes nat=never dtmfmode=rfc2833 ---- I've already checked codec config and any other thing that could come to my mind, but I'm getting out of ideas, so if anybody has any hint on how to fix the issue, I'd be glad. The problem is present both with asterisk 1.4 and 1.6, both latest versions, and even with both sccp and sip on both versions. If you need any debugging session, please ask and I'll provide. Ciao -- Max
David Backeberg
2009-Apr-16 16:33 UTC
[asterisk-users] Problem transferring calls between Cisco 7940 with SIP firmware
On Thu, Apr 16, 2009 at 10:08 AM, Massimiliano Stucchi <stucchi at willystudios.com> wrote:> firmware. ?The problem arises when transferring a call coming in from a > SIP account to another phone. ?The call connects, but for the first 10 > seconds there is a situation with one-way audio, then it turns into a > fully working call. > Here are some config files: > sip.conf > > ---- > > [902] > canreinvite=noTry enabling reinvite