asterisk users - Feb 2010 - Losing local SIP phones when internet goes down?

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.

Setup: PC with two ethernet cards: eth0 goes to local network, including 
two SIP phones (Aastra 9112i, wired, and Nokia E75, over WIFI); eth1 goes 
to router and thence to the internet over ADSL.  PC also has one Zap 
channel.

the SIP phones use DHCP but have defined IP addresses (DHCP server running 
on the PC).  The PC is also running a firewall (FIAIF), but not a DNS 
server.

Version of Debian Asterisk package: 1:1.4.21.2~dfsg-3+lenny1

Problem: When the internet connection goes down (which has been happening 
sporadically of late), connections to the two SIP phones on the local 
network get lost; ongoing calls from one of these phones over the Zap 
channel may get terminated, despite not using the internet.

I can reproduce this by switching off my ADSL router; however, if I simply 
take down the eth1 interface completely (by using "ifdown eth1", which
executes "route del default gw ... eth1" and "ifconfig eth1
down"), the
connections to the two SIP phones continue with no problems at all.

I enclose an extract from my sip.conf below.  Also, the logs indicate that 
Asterisk thinks the SIP phones are no longer reachable (ping timing out), 
while a manual ping from the same machine shows no trouble at all: the 
wired phone is responding in less than 2 ms each time, while the wireless 
one was a max of about 120 ms.

Any thoughts much appreciated!  Hopefully it's something obvious that
I've
overlooked...

Oh, BTW, the local phones are on a private net (10.9.8.xxx), but as it's 
the Asterisk box that's doing the NAT'ing, I used nat=no; I presume
that's
correct.  eth0 has address 10.9.8.1, while eth1 has a global internet IP 
address.

Cheers,

Nikhil.

-----

Extract from sip.conf:

[general]
context=incoming
srvlookup=yes
realm=nikhil-nair.net
; Various "register=>" statements, not relevant to the local phones

[101] ; Aastra 9112i at 10.9.8.101
type=friend 
secret=...
qualify=yes ; Qualify peer is no more than 2000 ms away 
nat=no ; This phone is not natted 
host=dynamic ; This device registers with us 
canreinvite=no ; Asterisk by default tries to redirect 
context=local
disallow=all
allow=alaw
allow=ulaw

[111] ; Nokia E75 via WIFI access point, at 10.9.8.111
type=friend 
secret=...
qualify=yes ; Qualify peer is no more than 2000 ms away 
nat=no ; This phone is not natted 
host=dynamic ; This device registers with us 
canreinvite=no ; Asterisk by default tries to redirect 
context=local
disallow=all
allow=alaw
allow=ulaw
allow=gsm

On 02/05/10 02:05, 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.
>
>Setup: PC with two ethernet cards: eth0 goes to local network, including
>two SIP phones (Aastra 9112i, wired, and Nokia E75, over WIFI); eth1 goes
>to router and thence to the internet over ADSL.  PC also has one Zap
>channel.
>
>the SIP phones use DHCP but have defined IP addresses (DHCP server running
>on the PC).  The PC is also running a firewall (FIAIF), but not a DNS
>server.
>
>Version of Debian Asterisk package: 1:1.4.21.2~dfsg-3+lenny1
>
>Problem: When the internet connection goes down (which has been happening
>sporadically of late), connections to the two SIP phones on the local
>network get lost; ongoing calls from one of these phones over the Zap
>channel may get terminated, despite not using the internet.
>
>I can reproduce this by switching off my ADSL router; however, if I simply
>take down the eth1 interface completely (by using "ifdown eth1",
which
>executes "route del default gw ... eth1" and "ifconfig eth1
down"), the
>connections to the two SIP phones continue with no problems at all.
Does your router runs DHCPD, assigning network addresses on on your LAN?
If yes, and you switch the ADSL route OFF then you kill your DHCPD so no
connection to asterisk server.

-- 
Joseph

On 02/05/10 02:05, Nikhil Nair wrote:
>
>Extract from sip.conf:
>
>[general]
>context=incoming
>srvlookup=yes
>realm=nikhil-nair.net
Your resolve authentication to an outside server, isn't it?
So here might be your problem; if there is no connection to the Internet no
authentication.

-- 
Joseph

I solved similar issues by setting srvlookup=no, having bind running locally and
just the line "nameserver 127.0.0.1" on /etc/resolv.conf.


Atenciosamente,

Vin?cius Fontes
Gerente de Seguran?a da Informa??o
Canall Tecnologia em Comunica??es
Passo Fundo - RS - Brasil
+55 54 2104-7000

Information Security Manager
Canall Tecnologia em Comunica??es
Passo Fundo - RS - Brazil
+55 54 2104-7000

----- "Randy R" <randulo2008 at gmail.com> escreveu:
> 2010/2/5 Vin?cius Fontes <vinicius at canall.com.br>:
> > Have you tried to set srvlookup=no on your sip.conf?
> 
> I think that just stops SRV lookups, not regular DNS.
> 
> /r
> 
> -- 
> _____________________________________________________________________
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> 
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Could be. Important thing is the problem was solved :)



Atenciosamente,

Vin?cius Fontes
Gerente de Seguran?a da Informa??o
Canall Tecnologia em Comunica??es
Passo Fundo - RS - Brasil
+55 54 2104-7000

Information Security Manager
Canall Tecnologia em Comunica??es
Passo Fundo - RS - Brazil
+55 54 2104-7000

----- "Jeff LaCoursiere" <jeff at jeff.net> escreveu:
> On Fri, 5 Feb 2010, Vin?cius Fontes wrote:
> 
> > I solved similar issues by setting srvlookup=no, having bind running
> 
> > locally and just the line "nameserver 127.0.0.1" on
> /etc/resolv.conf.
> >
> 
> Your local bind is what solved the problem.  The srvlookup=no didn't 
> actually help IMO.
> 
> j
> 
> >
> > Atenciosamente,
> >
> > Vin?cius Fontes
> > Gerente de Seguran?a da Informa??o
> > Canall Tecnologia em Comunica??es
> > Passo Fundo - RS - Brasil
> > +55 54 2104-7000
> >
> > Information Security Manager
> > Canall Tecnologia em Comunica??es
> > Passo Fundo - RS - Brazil
> > +55 54 2104-7000
> >
> > ----- "Randy R" <randulo2008 at gmail.com> escreveu:
> >
> >> 2010/2/5 Vin?cius Fontes <vinicius at canall.com.br>:
> >>> Have you tried to set srvlookup=no on your sip.conf?
> >>
> >> I think that just stops SRV lookups, not regular DNS.
> >>
> >> /r
> >>
> >> --
> >>
> _____________________________________________________________________
> >> -- 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
> >
> > -- 
> >
> _____________________________________________________________________
> > -- 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
> >
> -- 
> _____________________________________________________________________
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> 
> asterisk-users mailing list
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>    http://lists.digium.com/mailman/listinfo/asterisk-users

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

I have the same problem.  I have asterisk on the public internet and 
other ips on the private lan.  When the internet goes down my private 
asterisk network is compromised.  My thought is that it has something to 
do with the ports/ips on which asterisk is trying to communicate.  It 
may be a configuration issue but as of yet I haven't figured it out.

On 2/4/2010 9:05 PM, 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.
>
> Setup: PC with two ethernet cards: eth0 goes to local network, including
> two SIP phones (Aastra 9112i, wired, and Nokia E75, over WIFI); eth1 goes
> to router and thence to the internet over ADSL.  PC also has one Zap
> channel.
>
> the SIP phones use DHCP but have defined IP addresses (DHCP server running
> on the PC).  The PC is also running a firewall (FIAIF), but not a DNS
> server.
>
> Version of Debian Asterisk package: 1:1.4.21.2~dfsg-3+lenny1
>
> Problem: When the internet connection goes down (which has been happening
> sporadically of late), connections to the two SIP phones on the local
> network get lost; ongoing calls from one of these phones over the Zap
> channel may get terminated, despite not using the internet.
>
> I can reproduce this by switching off my ADSL router; however, if I simply
> take down the eth1 interface completely (by using "ifdown eth1",
which
> executes "route del default gw ... eth1" and "ifconfig eth1
down"), the
> connections to the two SIP phones continue with no problems at all.
>
> I enclose an extract from my sip.conf below.  Also, the logs indicate that
> Asterisk thinks the SIP phones are no longer reachable (ping timing out),
> while a manual ping from the same machine shows no trouble at all: the
> wired phone is responding in less than 2 ms each time, while the wireless
> one was a max of about 120 ms.
>
> Any thoughts much appreciated!  Hopefully it's something obvious that
I've
> overlooked...
>
> Oh, BTW, the local phones are on a private net (10.9.8.xxx), but as
it's
> the Asterisk box that's doing the NAT'ing, I used nat=no; I presume
that's
> correct.  eth0 has address 10.9.8.1, while eth1 has a global internet IP
> address.
>
> Cheers,
>
> Nikhil.
>
> -----
>
> Extract from sip.conf:
>
> [general]
> context=incoming
> srvlookup=yes
> realm=nikhil-nair.net
> ; Various "register=>" statements, not relevant to the local
phones
>
> [101] ; Aastra 9112i at 10.9.8.101
> type=friend
> secret=...
> qualify=yes ; Qualify peer is no more than 2000 ms away
> nat=no ; This phone is not natted
> host=dynamic ; This device registers with us
> canreinvite=no ; Asterisk by default tries to redirect
> context=local
> disallow=all
> allow=alaw
> allow=ulaw
>
> [111] ; Nokia E75 via WIFI access point, at 10.9.8.111
> type=friend
> secret=...
> qualify=yes ; Qualify peer is no more than 2000 ms away
> nat=no ; This phone is not natted
> host=dynamic ; This device registers with us
> canreinvite=no ; Asterisk by default tries to redirect
> context=local
> disallow=all
> allow=alaw
> allow=ulaw
> allow=gsm
>
>

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
> >
>
>
>
>
>
>
>
> ------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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>
> End of asterisk-users Digest, Vol 67, Issue 19
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Its a long and old thread, haven't read it all, but just to let you know
this happens when there is no reply from the DNS. So change DNS or install
it locally on your asterisk server. At least caching name server should be
installed.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-09-24 1:51 PM, "Gopalakrishnan A.N" <saigop at
gmail.com> wrote:

Still I have the connection loss when internet goes down, I have to restart
the Asterisk machine or need to remove the VoIP trunk accessing internet...

DNSmasq is the only option by losing the connection when internet goes
down...is there any other way...

Thanks



On Fri, Feb 12, 2010 at 4:20 AM, Matt Riddell <lists at venturevoip.com>
wrote:
>
> On 9/02/10 12:59 ...
-- 
Thank you  with regards,
Gopalakrishnan A.N,



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