> Did this fix make it into 1.8.4? Getting registration errors on Cisco 79XX in > 1.8.4, going back to 1.8.3.3 everything works. I did open > https://issues.asterisk.org/view.php?id=19264 and included a SIP trace.
Sorry all, I did not follow up adequately. Definitely a problem with 1.6.2.18 and the issue # is 18951. Fixed in 1.8.3.3; Cisco 79xx registered fine. I don't know about 1.8.4 yet; haven't installed it for testing yet. Cassius > > > > On Fri, May 6, 2011 at 12:24 PM, Julian Lyndon-Smith <[email protected]> > wrote: >> It was my problem ;) >> >> https://issues.asterisk.org/view.php?id=18951 >> >> fixed in svn >> >> On 6 May 2011 16:45, Steve Davies <[email protected]> wrote: >>> > On 6 May 2011 16:30, Eric Wieling <[email protected]> wrote: >>>>> >>> -----Original Message----- >>>>> >>> From: [email protected] >>>>> >>> [mailto:[email protected]] On Behalf Of >>>>> >>> Cassius Smith >>>>> >>> Sent: Friday, May 06, 2011 11:23 AM >>>>> >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>>>> >>> Subject: [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not >>>>> >>> registering >>>>> >>> >>>>> >>> Hi all, >>>>> >>> I have a production server running with about 90 Cisco >>>>> >>> 79[46]1's and SIP release 8.5(2)SR1 from last year. I was >>>>> >>> running Asterisk 1.6.2.9 and upgraded last night after hours. >>>>> >>> (Seemed low risk to me!) >>>>> >>> >>>>> >>> Much to my surprise, not a single one of the Cisco 79XX >>>>> >>> phones would register. Since it's a production server, I >>>>> >>> rolled back to 1.6.2.9 and everything was fine. All my >>>>> >>> Linksys SPA phones and Polycom speaker phones registered just fine. >>>>> >>> >>>>> >>> I am now setting up test servers with both 1.6.2.18 and >>>>> >>> 1.8.3.3 to collect some debug. >>>>> >>> >>>>> >>> I am just curious - has anyone else had SIP issues with these >>>>> >>> phones and updating Asterisk broke them? >>>>> >>> >>>>> >>> I will post results of my findings after I have time to collect them. >>>>> >>> >>>>> >>> Cassius Smitha >>>>> >>> >>>> >> >>>> >> I seem to recall this issue mentioned on asterisk-dev. Check >>>> issues.digium.com <http://issues.digium.com> and see if there is anything >>>> similar to your issue. >>>> >> >>> > >>> > I also remember this being mentioned - I believe it was fixed in the >>> > chan_sip Via: header handling code. The fix is in branches/1.6.2 >>> > already, so you should be able to grab the patch without too much >>> > trouble. >>> > >>> > Regards, >>> > Steve >
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