I recently switched from BroadVoice to VoicePulse Connect on my Asterisk box. Too many Meetme quality complaints (whether real or perceived).
I had to make a choice to use IAX2 or SIP with VoicePulse. I first tried to go with SIP because I already had it working and all of our devices are SIP. Problem is that every time I turn my back, the Asterisk registration with the VoicePulse SIP server times out. So, I switched to IAX2. Now, everything works fine 95% of the time . . . but every once in a while, perhaps 5 seconds into a call or 20 minutes into a call, the call will simply drop. This occurs several times per week with no observable pattern. I have attached an excerpt from the log file at the end of this message. Has anyone else experienced this? Know what is causing it? Has anyone gotten VoicePulse Connect to work with SIP? Thanks, Adam Mar 17 11:50:23 VERBOSE[9987]: -- Executing Dial("SIP/2034-771f", "IAX2/[EMAIL PROTECTED]/19043317785") in new stack Mar 17 11:50:23 VERBOSE[9987]: -- Called [EMAIL PROTECTED]/19043317785 Mar 17 11:50:23 VERBOSE[24993]: -- Call accepted by 66.234.228.160 (format ulaw) Mar 17 11:50:23 VERBOSE[24993]: -- Format for call is ulaw Mar 17 11:50:23 VERBOSE[9987]: -- IAX2/voicepulse-out-01/6 stopped sounds Mar 17 11:50:23 VERBOSE[9987]: -- IAX2/voicepulse-out-01/6 is making progress passing it to SIP/2034-771f Mar 17 11:50:23 DEBUG[24993]: Ooh, voice format changed to 4 Mar 17 11:50:33 VERBOSE[9987]: -- IAX2/voicepulse-out-01/6 answered SIP/2034-771f Mar 17 11:50:33 DEBUG[24992]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 2: Found Mar 17 11:51:03 DEBUG[24993]: Immediately destroying 7, having received INVAL Mar 17 11:51:43 DEBUG[24993]: Immediately destroying 4, having received INVAL Mar 17 11:51:43 DEBUG[24993]: Raw Hangup 69.73.19.178:4569, src=4, dst=285 Mar 17 11:52:32 DEBUG[24992]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 156: Found Mar 17 11:52:43 DEBUG[24993]: Sending VNAK Mar 17 11:52:48 DEBUG[24992]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 156: Found Mar 17 11:53:04 DEBUG[24993]: Immediately destroying 6, having received INVAL Mar 17 11:53:04 DEBUG[9987]: Didn't get a frame from channel: IAX2/voicepulse-out-01/6 Mar 17 11:53:04 DEBUG[9987]: Bridge stops bridging channels SIP/2034-771f and IAX2/voicepulse-out-01/6 Mar 17 11:53:04 DEBUG[9987]: We're hanging up IAX2/voicepulse-out-01/6 now... Mar 17 11:53:04 DEBUG[9987]: Really destroying IAX2/voicepulse-out-01/6 now... Mar 17 11:53:04 VERBOSE[9987]: -- Hungup 'IAX2/voicepulse-out-01/6' Mar 17 11:53:04 DEBUG[9987]: Exiting with DIALSTATUS=ANSWER. Mar 17 11:53:04 VERBOSE[9987]: == Spawn extension (intl-access, 919043317785, 2) exited non-zero on 'SIP/2034-771f' Mar 17 11:53:04 DEBUG[9987]: update_user_counter(2034) - decrement inUse counter Mar 17 11:53:04 DEBUG[24992]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users