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/xxxxxxxxxx@voicepulse-out-01/19043317785") in new stack Mar 17 11:50:23 VERBOSE[9987]: -- Called xxxxxxxxxxx@voicepulse-out-01/19043317785 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 'aba2d05a-50073364-7bdeb47d@192.168.2.52' 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 'a449c34f-33721eb9-bb93e01a@192.168.2.57' of Request 156: Found Mar 17 11:52:43 DEBUG[24993]: Sending VNAK Mar 17 11:52:48 DEBUG[24992]: Stopping retransmission on '90a2f084-85313396-7ec1c37b@192.168.2.57' 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 'aba2d05a-50073364-7bdeb47d@192.168.2.52' 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.
Adam Robins wrote:>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? > >I have also seen this in the last couple of weeks... I've had some long (90 min) calls.. and had it happen a few times during the same call... though I've not taken the time to pull any logs. My net service at home can be spotty sometimes... so I thought it might have been caused by that. It's not been bad enough to terminate a call... just 10-15 seconds of silence. Jared
Adam Robins wrote:>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? > > >Hi Admin, I use the connect service from voicepulse ( as I am sure you do, just specifying for future searches ), and I haven't had any of these problems you have mentioned. I do have a problem when the call is connected, there's about half a second of silence about half a second into the call, on every call. I mention it here in case it's related. Honestly, my first instict says this is a firewall problem. Is that at all possible with your setup? Sean
Jared Watkins wrote:>Adam Robins wrote: > >>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 20minutes>>into a call, the call will simply drop. This occurs several times per >>week with no observable pattern > >I have also seen this in the last couple of weeks... I've had some >long (90 min) calls.. and had it happen a few times during the same >call... though I've not taken the time to pull any logs. My netservice>at home can be spotty sometimes... so I thought it might have been >caused by that. It's not been bad enough to terminate a call... just >10-15 seconds of silence.I have had similar problems with VoicePulse for 2 - 3 months now. Along with random dropped calls, we have also experienced dial-in problems. VoicePulse tech support has been utterly silent despite several emails and phone calls. We're now trying out SixTel and Live VoIP. Live VoIP has recently had a problem with our (and others) DID numbers, but they have been very responsive and it looks like the problem will be solved shortly. Joe
I don't see how it could be firewall issues. I have firewall ports 4569 and 5036 open for UDP traffic to and from the Asterisk server. Yesterday we conducted a conference call that lasted several hours without a drop, just periodic "dead spots" for a few seconds. Other calls disconnect entirely. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Sean Kennedy Sent: Wednesday, March 23, 2005 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VoicePulse Issues Adam Robins wrote:>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? > > >Hi Admin, I use the connect service from voicepulse ( as I am sure you do, just specifying for future searches ), and I haven't had any of these problems you have mentioned. I do have a problem when the call is connected, there's about half a second of silence about half a second into the call, on every call. I mention it here in case it's related. Honestly, my first instict says this is a firewall problem. Is that at all possible with your setup? Sean _______________________________________________ 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 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.
Adam Robins wrote:>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 > >In cases where a provider offers a choice of SIP or IAX2 I have been wondering if the provider would be using a * server in either case. I prefer IAX2 as long as I know the provider has adequate server capacity for the load. Otherwise I would think that getting my handoff directly from SIP gateways would be more reliable.