Hi Steve, Thanks again for the response-- the answer you gave was more or less the answer that I was expecting.
I was logging all packets to and from the phone, and I never saw an ACK from the phone for the OK to Asterisk on the VM calls -- not an ACK directed to a different location, just no ACK period. I noted in my other reply that as a test I had added a call to Ringing() followed by Wait(1) before dropping into Voicemail for the voicemail extension in the dialplan, since I noticed that the only difference that appeared to exist between a SIP-POTS or SIP-SIP call and a SIP-Voicemail call, aside from the missing ACK from the phone is that Asterisk reported session progress of "100 Trying" and "180 Ringing" to the phone, where it didn't report either of these when calling Voicemail, instead jumping straight to "200 OK with session description". In the 24 hours since I did that we haven't been able to get any of dozens of calls to Voicemail to fail, when normally it would borderline on greater than one in every two call. I'm still not convinced it's fixed, but I'm feeling fairly good about the solution, so it seems to my untrained eye like there may be an issue in the Cisco 79x1 firmware if the PBX "accepts" a call without providing any intermediate status? That seems like it would manifest itself in other places, and I'm kind of grasping at straws but... Thanks again to everyone who took the time to read and or respond to this issue -- I'll post again if it turns out that that wasn't actually the fix, but for now management is happy that they can actually listen to their entire voicemail messages. Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440.729.4640 x1107 F: 440.729.0884 I:http://www.controlworks.com Crestron Authorized Independent Programmer -----Original Message----- From: Steven J. Douglas [mailto:stev...@moij.biz] Sent: Wednesday, February 04, 2009 12:43 AM To: Lincoln King-Cliby Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail Hi Lincoln, The fact that you can hear and respond to the voice mail (even if its for the first 20 seconds), means that your phone has received the OK message properly. The problem is the missing ACK after receiving OK. When asterisk did not receive the ACK after a few retries of the OK, it terminated the call. This resulted in your RTP streams getting the icmp error messages. Assuming that you are capturing every packet that goes on between Asterisk and the phone, there are two possibilities. 1. The phone has a bug. 2. The ACK was sent somewhere else. Normally the ACK message destination is constructed from the response to the INVITE. In this case, it will be the OK message. If you suspect its the second case, you can capture the traffic for both a good voicemail call and a failed voicemail call. Then by comparing the messages, you might get a hint. If you need help, you can send the packet capture to me privately (not through the list as it might be a large file) and I can help vet it for you. Unfortunately there is no flag that you can set to confirm a session based on OK being transmitted and not wait for ACK. Regards, Steve <snipped my original reply> _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
