About every 10th call coming into my x1000p is not getting the audio it should. You can see the messages scrolling on the console as they usually would, playing the thankyou, then and menu messages. internal phones ring, but when answered there is no audio. The caller gets a full volume echo with about 1/2 second latency. At first I thought it might be related to using the aggressive suppressor echo canceller...I recompiled it out, and the problem is still there...doesn't seem to matter if the caller is generating noise when connecting or not. didn't get it to happen when calling from an analog line...seems to happen when calling from a cell phone...I don't see how that would look any different to the x1000p though. perhaps there is a latency difference. I am urgently trying to solve this problem. If I can't solve this problem, it will certainly be the death of my * installation. Has anyone seen this problem before?
I'm running a large number (125) remote sip phones for FEMA in the Gulf area over satellite. I've run into a major problem and need some assistance. When dialing the FEMA voice response system, it appears that it never actually answers the phone. I never get audio when dialing via SIP through a provider and when dialing over my PRI it actually times out with a phone not ansered message, though the audio is passed. Apparently the FEMA system does not issue an 'ANSWERED' or 'CONNECTED' code back to the PSTN as it should. The link stays in an in-progress state until timeout occurs or the user hangs up. Is there any way to get SIP to pass audio prior to getting a call complete message? This is Asterisk CVS-HEAD 08-01-2004. Please respond to tmckee@sdnglobal.com as I don't have good access to the email account serving this list during the day. Many thanks, Tim McKee -------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef Size: 1770 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050930/3dd47654/winmail.bin
Tim McKee wrote:> Is there any way to get SIP to pass audio prior to getting a call complete > message? This is Asterisk CVS-HEAD 08-01-2004.There is another possibility... in your dialplan context that is handling the call from the SIP phone out to the PRI, issue an Answer() before placing the outbound call. This will force the SIP session into full-duplex audio and not cause the SIP phone to drop the call.