[asterisk-users] Losing local SIP phones when internet goes down?

Danny Dias ing.diasdanny at gmail.com
Sun Feb 7 14:03:26 CST 2010 

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 <asterisk-users-request at lists.digium.com>

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> 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 <seandarcy2 at gmail.com>
> Subject: Re: [asterisk-users] Losing local SIP phones when internet
>        goes    down?
> To: asterisk-users at lists.digium.com
> Message-ID: <hkmp6q$e4n$1 at ger.gmane.org>
> 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 <bilmar_gh at yahoo.com>
> Subject: Re: [asterisk-users] A2Billing and other prepaid Billing like
>        ASTCC,  who is better?
> To: asterisk-users at lists.digium.com
> Message-ID: <73114.13099.qm at web53906.mail.re2.yahoo.com>
> 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: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com]
> > On Behalf Of bilal ghayyad
> > Sent: 2010 m. vasario 7 d. 01:20
> > To: asterisk-users at lists.digium.com
> > 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
> >
>
>
>
>
>
>
>
> ------------------------------
>
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