Here's the old conversation........
That's for digging for that one Robert! Sounds like passing back through to Cisco engineering would be worth it if anybody here has a Cisco support contract on their phones. Mike -----Original Message----- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Sven Evensen Sent: Wednesday, May 12, 2010 6:39 PM To: JOLY, ROBERT (ROBERT) Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Media drops out after 5-6 minutes Bob, Thanks for the excellent explanation. Do you think/know if the newer Cisco phones 7941/7961 etc are better or are they same cr**? Regards, Sven -----Original Message----- From: JOLY, ROBERT (ROBERT) [mailto:rj...@avaya.com] Sent: 12 May 2010 18:09 To: Sven Evensen Subject: RE: [sipx-users] Media drops out after 5-6 minutes > > Bob, > > Have you had a chance to look at the logs I sent. Ok, I took a look at the logs and I finally found what was wrong. Nowadays, when I look at traces I tend to assume that the fundamentals of SIP are sound and focus my troubleshooting efforts on the trickier parts of the protocol but every once in a while there comes a sad phone implementation that reminds me that I shouldn't make such foolish assumptions... The Route: header that Cisco puts in its ACK is truncated. It must contain the entire Route as set by the Record-Route: header it gets in the 200 OK but Cisco chooses the silently chop it which causes vital information to be missing from the message. The call is then treated as a stale call and the NAT traversal cleanup routine sweeps the media part of the call away after 5 minutes or so. I suggest you use a real phone if you have one. Cheers, bob > -----Original Message----- > From: Sven Evensen > Sent: 10 May 2010 17:19 > To: 'JOLY, ROBERT (ROBERT)' > Subject: RE: [sipx-users] Media drops out after 5-6 minutes > > Hi Bob, > > Here is a (hopefully) full snapshot. > The call is from 8051 to 907795951717. > The media dropped out after approx 5:05, I hung up 10 seconds later. > > Regards, > Sven > > > > -----Original Message----- > From: JOLY, ROBERT (ROBERT) [mailto:rj...@avaya.com] > Sent: 05 May 2010 18:16 > To: Sven Evensen > Subject: RE: [sipx-users] Media drops out after 5-6 minutes > > Sven, > I looked at the traces and from a media perspective, the ITSP and > Cisco are well-behaved which leaves sipXecs as the prime suspect. I > need to assign you more homework. Please bump the logging level of > the Proxy, Media Relay and SIP Trunking to DEBUG and restart those > services. > Reproduce the problem yet again and take a snapshot that will cover > the entire call time period. Send the resulting snapshot directly to > me. > > Thanks, > bob > > > -----Original Message----- > > From: Sven Evensen [mailto:sven.even...@onrelay.com] > > Sent: Wednesday, May 05, 2010 11:12 AM > > To: JOLY, ROBERT (ROBERT) > > Subject: RE: [sipx-users] Media drops out after 5-6 minutes > > > > Hi Robert, here is the attachment. > > > > Sven > > > > -----Original Message----- > > From: sipx-users-boun...@list.sipfoundry.org > > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of JOLY, > > ROBERT (ROBERT) > > Sent: 04 May 2010 20:35 > > To: Matt White; sipx-users@list.sipfoundry.org > > Subject: Re: [sipx-users] Media drops out after 5-6 minutes > > > > > > > > I believe there is a cisco issue. We had another customer > > report this > > > issue with a set of cisco handsets yesterday. I have not > > heard back > > > from our techs what they found yet, but the issue reported > > to us was > > > not via sipxbridge but for a remote worker calling to an internal > > > extension...so sipxrelay. > > > > > > Calls drop at exactly 5:33 seconds. > > > > > > >>> "Sven Evensen" 05/04/10 11:29 AM >>> > > > > > > > > > Hi, > > > > > > > > > > > > We have a scenario where a Cisco 7640 dials out through > our UK SIP > > > trunk to a mobile. After 5 -6 minutes the media > > > > > > suddenly ends. The calls stays up until the parties hang up. > > > This does not happen on the Polycom phone > > > > > > > > > > > > The only clue I see is that at the exact time the media > > goes away, I > > > can see in sipxrelay.log an entry "pauseBridge". > > > > The pauseBridge entry would be there as a consequence of the call > > being terminated by one end. So, it is the effect, not the cause. > > Your best bet is to get a network trace of the problem and > send it in. > > If you can run 'tcpdump -n -nn -s 0 -i any - w cisco_drop.cap' from > > the sipXecs command line, make a call, wait for it to fail and then > > send it to me, I'll check that the media and signaling are > what they > > are supposed to be. > > > > > > > > > > > > > > I am really not sure which logging to turn on to dig deeper > > into this, > > > can some one advise. > > > > > > > > > > > > This is sipX 4.0.4 > > > > > > > > > > > > Regards, > > > > > > Sven > > > > > > > > > > > > > > > > > > Sven Evensen, Technical Support Consultant > > > > > > OnRelay > > > Elizabeth House | 39 York Road, London SE1 7NQ, UK | +44 (0) > > > 207 902 8123 | sven.even...@onrelay.com | > <http://www.onrelay.com/> > > > www.onrelay.com <http://www.onrelay.com/> > > > > > > OnRelay in the News: > > > > > > OnRelay Named in Tech Media Invest Top 100 > > > > > > http://www.guardian.co.uk/tech-media-invest-100/top-100 > > > > > > This electronic message transmission contains information from > > > OnRelay, Ltd., that may be confidential or privileged. > > > The information is intended solely for the recipient and > use by any > > > other party is not authorised. If you are not the intended > > recipient, > > > be aware that any disclosure, copying, distribution or use of the > > > contents of this information or any attachment, is > > prohibited. If you > > > have received this electronic transmission in error, please > > notify us > > > immediately by electronic mail (i...@onrelay.com) and delete this > > > message, along with any attachments, from your computer. > > > Registered in England No 04006093 | Registered Office 1st > > Floor, 236 > > > Gray's Inn Road, London WC1X 8HL > > > > > > > > > > > > > > _______________________________________________ > > sipx-users mailing list sipx-users@list.sipfoundry.org List > > Archive: http://list.sipfoundry.org/archive/sipx-users > > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > > sipXecs IP PBX -- http://www.sipfoundry.org/ > > > _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/ _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/ From: Michael Scheidell [mailto:scheid...@secnap.net] Sent: Wednesday, June 23, 2010 1:55 PM To: Picher, Michael Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Voice Transmission stops after 29 Minutes I have been on a cisco phone, fw 8-12, sipxecs 4.2.0 for over an hour, with no problems. and I don't remember any recent discussions on this. are you thinking polycom phones with firmware 3.2.3, the known problem with 'spotty' networks? I upgraded the firmware while doing regression testing. I found 8-12 fixed a bunch of things that cisco didn't document fixing. fix your codex to G.711 on phone and itsp, just in case. On 6/23/10 1:47 PM, Picher, Michael wrote: Cisco phones going out through sipXbridge will drop off the call in roughly 10 minutes. On sipXecs 4.2.0. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Scheidell Sent: Wednesday, June 23, 2010 1:38 PM To: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Voice Transmission stops after 29 Minutes On 6/23/10 11:57 AM, Picher, Michael wrote: You left out some critical information... Type of phones and how is user A reaching user B (ie., sipXbridge, gateway)? Are you running Cisco phones? If you are, there's a known bug... what known bug? what firmware has the bug? -- Michael Scheidell, CTO Phone: 561-999-5000, x 1259 > | SECNAP Network Security Corporation Certified SNORT Integrator 2008-9 Hot Company Award Winner, World Executive Alliance Five-Star Partner Program 2009, VARBusiness Best Anti-Spam Product 2008, Network Products Guide King of Spam Filters, SC Magazine 2008 ________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ________________________________ -- Michael Scheidell, CTO Phone: 561-999-5000, x 1259 > | SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best Anti-Spam Product 2008, Network Products Guide * King of Spam Filters, SC Magazine 2008 ________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ________________________________
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