Matthew,
Anything new! I am getting the external call drop at 15 seconds too, multiple scenarios. For us it is after dialing inbound. This is on a test environment. We tried a desktop Bria3 (just updated) calling extension to extension with a remotely registered 3G iPhone using the Bria iPhone client. Call dropped at 15 seconds. The goal of the test was to take Polycom and our carrier out of the equation. So we still get the exactly 15 sec drop with no Polycom phone or ITSP carrier in the mix. In contrast the same two end points, with the iPhone (using a Bria SIP client) registered over the local WiFi in our building. Worked like a charm, calls connect, two way audio, did not drop on call exceeding 2 minutes. We have a new SBC being tested (Audiocodes MSBG 6.2), but your configuration seems to be using just SipX bridge. We also have a one way and sometimes zero way audio problem we are chasing as well. On this, when we have one way audio it is the UDP RTP stream coming in from the external device that does not get through. We have identified that the BGP routers are not getting the udp/rtp stream sent to them, and the FW sees no packets coming in (nor dropped). Don From: [email protected] [mailto:[email protected]] On Behalf Of Matthew Kitchin (public/usenet) Sent: Monday, August 01, 2011 11:37 AM To: sipx-users Subject: [sipx-users] Fwd: Re: Polycom Spectralink 8020 - drops at 15 seconds Polite bump request. Does anyone see anything obvious in the sip trace? I'm 99.9% positive this has not been occurring all the time. If it has, my testing is absolutely awful. Outbound calls drop at 15 seconds. This is the latest version of Sipx 4.4 with all updates. The server in this example is virtual, but it is happening on all my physical boxes as well. 8020 = 10.81.3.253 Sipx = 10.81.3.5 Verizon VoIP = 172.30.216.x There is no IP address NAT, but there is an inbound 5060=>5080 NAT. -------- Original Message -------- Subject: Re: [sipx-users] Polycom Spectralink 8020 - drops at 15 seconds Date: Thu, 28 Jul 2011 14:46:17 -0500 From: Matthew Kitchin (public/usenet) <[email protected]> <mailto:[email protected]> To: [email protected] 1 way audio was a false alarm. The phone was fine after a rebbot. The dropped call at 15 second issue still exists on external calls. I am using sipxbride. The sip trace is attached. The test call is the only call in the logfiles. Any ideas? On 7/28/2011 2:41 PM, Tony Graziano wrote: "verizon is not seeing an ack on the call and thinks rtp is not actually established." is now "sipxecs" is not verizon is not seeing an ack on the call and thinks rtp is not actually established. On Thu, Jul 28, 2011 at 3:33 PM, Matthew Kitchin (public/usenet) <[email protected]> wrote: I just realized they are getting 1 way audio even on internal calls too. Something has to be wrong on our end. I will keep investigating. On 7/28/2011 2:21 PM, Tony Graziano wrote: verizon is not seeing an ack on the call and thinks rtp is not actually established. so then either the proxy is not sending the ack or the phone is not. got a siptrace? On Thu, Jul 28, 2011 at 3:15 PM, Matthew Kitchin (public/usenet) <[email protected]> wrote: Can anyone help me interpret the attached wireshark? I'm 99.9% positive this has not been occurring all the time. If it has, my testing is absolutely awful. Outbound calls drop at 15 seconds. This is the latest version of Sipx 4.4 with all updates. The server in this example is virtual, but it is happening on all my physical boxes as well. 8020 = 10.81.3.253 Sipx = 10.81.3.5 Verizon VoIP = 172.30.216.x There is no IP address NAT, but there is an inbound 5060=>5080 NAT. I'm not an expert at reading these, but I believe the BYE is coming from Verizon and I do not know why. I think I am going to have to open a ticket with them, but I wanted to check here first and see if I am missing something. Thanks, Matthew _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.326.5325 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our voip fax services! _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.326.5325 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our voip fax services! _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ "The information in this electronic mail message is the sender's confidential business and may be legally privileged. 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