Hello my friends, I'm having a problem like this post...the difference is that my asterisk goes down and i have to reboot my server in order to make it up again...
following you will see some errors that i can see in the Asterisk /var/log/messages qhen asterisk goes down: [Feb 5 10:32:45] WARNING[6519] chan_sip.c: Maximum retries exceeded on transmission 1850202354 at 10.4.1.152 <http://lists.digium.com/mailman/listinfo/asterisk-users> for seqno 21 (Critical Response) [Feb 5 10:32:45] WARNING[6519] chan_sip.c: Hanging up call 1850202354 at 10.4.1.152 <http://lists.digium.com/mailman/listinfo/asterisk-users> - no reply to our critical packet. [Feb 5 10:33:04] NOTICE[6519] chan_sip.c: Call from '346' to extension '3415554' rejected because extension not found. [Feb 5 10:35:31] NOTICE[6519] chan_sip.c: Disconnecting call 'SIP/301-09ad3be8' for lack of RTP activity in 301 seconds [Feb 5 10:36:17] NOTICE[6519] chan_sip.c: Disconnecting call 'SIP/317-b7735220' for lack of RTP activity in 301 seconds [Feb 5 10:38:19] NOTICE[6519] chan_sip.c: Peer '353' is now Reachable. (1ms / 2000ms) [Feb 5 10:42:59] NOTICE[6519] chan_sip.c: Peer '353' is now Reachable. (7ms / 2000ms) [Feb 5 10:51:09] NOTICE[6519] chan_sip.c: Peer '358' is now Reachable. (1ms / 2000ms) [Feb 5 10:53:08] NOTICE[6519] chan_sip.c: Peer '366' is now UNREACHABLE! Last qualify: 108 But later, at 2 pm, Asterisk went down again but with no weird message in /var/log/asterisk/message (just some unreachable messages of some extensions that has always been in the console since i installed Asterisk, but it never crash Asterisk untill last weeks ago): [Feb 5 13:54:11] NOTICE[6536] chan_sip.c: Peer '328' is now Reachable. (1ms / 2000ms) [Feb 5 13:55:18] NOTICE[6536] chan_sip.c: Registration from '< sip:300 at 10.4.1.6 <http://lists.digium.com/mailman/listinfo/asterisk-users>:5060>' failed for '10.4.2.3' - No matching peer found [Feb 5 13:57:40] NOTICE[6536] chan_sip.c: Call from '346' to extension '04265417457' rejected because extension not found. [Feb 5 13:59:15] NOTICE[6536] chan_sip.c: Peer '341' is now Reachable. (2ms / 2000ms) [Feb 5 13:59:25] NOTICE[6536] chan_sip.c: Peer '328' is now Reachable. (1ms / 2000ms) [Feb 5 14:01:43] NOTICE[6536] chan_sip.c: Peer '339' is now UNREACHABLE! Last qualify: 101 [Feb 5 14:04:22] NOTICE[6536] chan_sip.c: Peer '339' is now Reachable. (44ms / 2000ms) [Feb 5 14:04:39] NOTICE[6536] chan_sip.c: Peer '328' is now Reachable. (1ms / 2000ms) [Feb 5 14:09:53] NOTICE[6536] chan_sip.c: Peer '328' is now Reachable. (1ms / 2000ms) What could be the problem my friends? Thanks in advance 2010/2/7 <[email protected]> > Send asterisk-users mailing list submissions to > [email protected] > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > [email protected] > > You can reach the person managing the list at > [email protected] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > > 1. Re: Losing local SIP phones when internet goes down? (sean darcy) > 2. Re: A2Billing and other prepaid Billing like ASTCC, who is > better? (bilal ghayyad) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sun, 07 Feb 2010 11:19:39 -0500 > From: sean darcy <[email protected]> > Subject: Re: [asterisk-users] Losing local SIP phones when internet > goes down? > To: [email protected] > Message-ID: <[email protected]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Nikhil Nair wrote: > > Hi, > > > > I'm getting some strange behaviour on Asterisk 1.4 running on Debian > > Stable (Lenny). I suspect it's something to do with my setup, rather > than > > a bug, but I'm struggling to see it, and would appreciate any input. > > > > Thanks for posting this. And for persistently following up. I've had > this problem before, but never posted the problem - once the internet > came back up! > > I've now configured dnsmasq on my * box. This week I'll test it. > > sean > > > > > ------------------------------ > > Message: 2 > Date: Sun, 7 Feb 2010 09:41:00 -0800 (PST) > From: bilal ghayyad <[email protected]> > Subject: Re: [asterisk-users] A2Billing and other prepaid Billing like > ASTCC, who is better? > To: [email protected] > Message-ID: <[email protected]> > Content-Type: text/plain; charset=us-ascii > > Does your billing work with gnugk? Do u have a documentation on how it can > be used with gnugk? > > Does the free version work with the gnugk? > > Regards > Bilal > > ------------------------ > > > Please try our billing which has easier managing interface > > and works ok with > > H323: http://www.voip-info.org/wiki/view/MOR > > > > FREE version is available over this link: > > http://www.kolmisoft.com/billing-and-routing/mor-voip-billing-demo/ > > > > Regards, > > Mindaugas Kezys > > http://www.kolmisoft.com > > VoIP Billing and Routing Solutions > > > > -----Original Message----- > > From: [email protected] > > [mailto:[email protected]] > > On Behalf Of bilal ghayyad > > Sent: 2010 m. vasario 7 d. 01:20 > > To: [email protected] > > Subject: [asterisk-users] A2Billing and other prepaid > > Billing like ASTCC, > > who is better? > > > > Hi All; > > > > I used A2Billing, basically it is nice and fine, but > > management > > possibilities is not that rich, so a lot of staff are need > > to be repeated > > that let the admin facing a problem of the needed time to > > do the task. > > > > Anyone advise for another open source prepaid billing that > > is rich by the > > management features? > > > > Also, I hope to find an open source Billing (prepaid and > > postpaid) that can > > work with Asterisk and Gnugk at the same time (instead of > > using one billing > > for asterisk and one billing for gnugk, specially that > > gnugk is good for > > h323 functionalities that are missing in asterisk). > > > > Appreciate any help and advise in that direction. > > > > Regards > > Bilal > > > > > > > > > > ------------------------------ > > _______________________________________________ > --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 > > End of asterisk-users Digest, Vol 67, Issue 19 > ********************************************** >
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