My configs that I've reworked in the process of trying to fix this SIP problem actually started from Freepbx. I removed and reinstalled Asterisk last night, things seem to be working smoother, I'll no by noon if the problem is fixed or not. Thanks for the help from everyone, Ken
________________________________ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Deepak Naidu Sent: Wednesday, May 09, 2007 11:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] SIP Problems continue... A small way to make little easy, I dont know it people are ok to that, try integrating freepbx & asterisk so you know what the sip configs should look like when things are all well. Things might stop working if there is a bug or change in configs. -- Deepak Ken Williams <[EMAIL PROTECTED]> wrote: I mean that SIP phones cannot answer incoming calls or make outgoing calls. When a call comes in on ZAP, it actually rings all the phones like normal, but when you try to answer no one is there. In addition, when you try to dial out you eventually get a message on the phones saying unable to communicate with the server. So there is some traffic still traveling on the SIP channel (the server's dialing extensions from an incoming ZAP call) but no further communication...almost as if it's a one way street of communication. The server can send data out on SIP but isn't receiving any. As for your issue, we haven't really had that (thankfully), so I don't think you're heading down the horrible spot we're in right now. Tonight I'm going to remove all aspects of Asterisk and reinstall fresh, if that fails I'll format & reinstall the entire box. -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Moffett Sent: Wednesday, May 09, 2007 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Problems continue... I also get the mysterious SIP INVITE channels. 10.101.2.204 xxx 748e8b0a625 00102/00000 unkn No Init: INVITE And I also am running 1.4.4 on CentOS4. Is that a pattern or just coincidence? The other symptom you mention is this "...the SIP phones couldn't communicate with the server, though there was no error message on the server and everything appeared fine on the server." Do you mean no calls in or out until you reboot? I don't have that thankfully, but I do have a guy telling me that incoming audio just goes away for a few seconds at a time. He says also that it sometimes goes away for long enough time that he was mistaking it for a dropped call. But if he waits long enough it pretty generally always comes back. I have consistent solid network performance from the asterisk server to the ATA (and believe me, I've looked very hard for a network problem), and I don't know what to look at next. Incidentally, the guy hasn't called me since I rebooted last week. Is this similar to how your situation started? ********************************* Adam Moffett Plexicomm, LLC [EMAIL PROTECTED] ph: 866-759-4678x104 ********************************* _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ________________________________ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now <http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc 2VjA21haWwEc2xrA3RhZ2xpbmU> .
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