asterisk users - May 2007 - Problem on incoming call from Zap channel to SIP phones...

I have an Asterisk 1.2.16 server running CentOS 4.4 with a TE110P card
and an OpenVox A1200P card.  Up to today everything was working
perfectly.  The OpenVox card has 8 FXS and 2 FXO ports.  The two faxo
ports are used for a GSM adapter and for an ATA connected to Vonage.

	The problem we started noticing today was that the Vonage line will
receive a call and then cannot connect to any of the SIP phones.  The
port is configured to answer, dial SIP/603 and if that line is busy it
will dial to SIP/604.  Here is the output from the CLI:

    -- Starting simple switch on 'Zap/40-1'
    -- Executing Answer("Zap/40-1", "") in new stack
    -- Executing Set("Zap/40-1", "TIMEOUT(response)=5") in
new stack
    -- Response timeout set to 5
    -- Executing Dial("Zap/40-1", "SIP/603|20") in new stack
    -- Called 603
May 29 17:15:35 NOTICE[28703]: chan_sip.c:2012 auto_congest:
Auto-congesting SIP/603-0080d790
    -- SIP/603-0080d790 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)

	As you can see I get the message that says that the phone is busy even
when it is not.  This happens no matter which SIP phone I try to dial on
the server.  I can dial from any other phone to the same extension
without any problems.  Only calls coming from the Vonage line have this
problem.  I can make outgoing calls using that same line.  This was
working until mid day today and there were no changes made to the
configuration until after the problem was detected.

	The only thing I notice is that if I do a "zap show channel 40" it
tells me "Hookstate (FXS only): Offhook" even when the line is not in
use.  The other port connected to the GSM adapter says Onhook.  Also,
when Asterisk answers the call from Vonage I hear a loud click on the
phone and after it has tried both extensions I can hear the Voicemail
message play.

-- 
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez Prats
Director de Tecnolog?a
+52-55-91169161 ext 2001
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On Tue, May 29, 2007 at 05:22:45PM -0500, Carlos Chavez
wrote:
> 	I have an Asterisk 1.2.16 server running CentOS 4.4 with a TE110P card
> and an OpenVox A1200P card.  Up to today everything was working
> perfectly.  The OpenVox card has 8 FXS and 2 FXO ports.  The two faxo
> ports are used for a GSM adapter and for an ATA connected to Vonage.
> 
> 	The problem we started noticing today was that the Vonage line will
> receive a call and then cannot connect to any of the SIP phones.  The
> port is configured to answer, dial SIP/603 and if that line is busy it
> will dial to SIP/604.  Here is the output from the CLI:
> 
>     -- Starting simple switch on 'Zap/40-1'
>     -- Executing Answer("Zap/40-1", "") in new stack
>     -- Executing Set("Zap/40-1", "TIMEOUT(response)=5")
in new stack
>     -- Response timeout set to 5
>     -- Executing Dial("Zap/40-1", "SIP/603|20") in new
stack
>     -- Called 603
> May 29 17:15:35 NOTICE[28703]: chan_sip.c:2012 auto_congest:
> Auto-congesting SIP/603-0080d790
>     -- SIP/603-0080d790 is circuit-busy
>   == Everyone is busy/congested at this time (1:0/1/0)
> 
> 	As you can see I get the message that says that the phone is busy even
> when it is not.  This happens no matter which SIP phone I try to dial on
> the server.  I can dial from any other phone to the same extension
> without any problems.  Only calls coming from the Vonage line have this
> problem.  I can make outgoing calls using that same line.  This was
> working until mid day today and there were no changes made to the
> configuration until after the problem was detected.
> 
> 	The only thing I notice is that if I do a "zap show channel 40"
it
> tells me "Hookstate (FXS only): Offhook" even when the line is
not in
> use.  The other port connected to the GSM adapter says Onhook.  Also,
> when Asterisk answers the call from Vonage I hear a loud click on the
> phone and after it has tried both extensions I can hear the Voicemail
> message play.
What is the output of:

  show channels

Any chance that there is actually another call that keeps that channel
busy?

-- 
               Tzafrir Cohen       
icq#16849755                    jabber:tzafrir@jabber.org
+972-50-7952406           mailto:tzafrir.cohen@xorcom.com       
http://www.xorcom.com  iax:guest@local.xorcom.com/tzafrir

On Wed, 2007-05-30 at 02:49 +0300, Tzafrir Cohen wrote:
> 
> What is the output of:
> 
>   show channels
> 
> Any chance that there is actually another call that keeps that channel
> busy?
> 
	No, the line is not busy with another call.

-- 
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez Prats
Director de Tecnolog?a
+52-55-91169161 ext 2001
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On Tue, 2007-05-29 at 18:54 -0500, Carlos Chavez wrote:
> On Wed, 2007-05-30 at 02:49 +0300, Tzafrir Cohen wrote:
> 
> > 
> > What is the output of:
> > 
> >   show channels
> > 
> > Any chance that there is actually another call that keeps that channel
> > busy?
> > 
> 	No, the line is not busy with another call.
> 
	In fact at this moment there are 0 active calls in the system.  The
phones are all Aastra 9133i which can support up to 9 calls.  I have
restarted the service, rebooted the server and even upgraded to the
latest zaptel drivers and made sure CentOS is patched and up to date.

-- 
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez Prats
Director de Tecnolog?a
+52-55-91169161 ext 2001
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On Tue, 2007-05-29 at 19:14 -0500, Carlos Chavez wrote:
> On Tue, 2007-05-29 at 18:54 -0500, Carlos Chavez wrote:
> > On Wed, 2007-05-30 at 02:49 +0300, Tzafrir Cohen wrote:
> > 
> > > 
> > > What is the output of:
> > > 
> > >   show channels
> > > 
> > > Any chance that there is actually another call that keeps that
channel
> > > busy?
> > > 
> > 	No, the line is not busy with another call.
> > 
> 	In fact at this moment there are 0 active calls in the system.  The
> phones are all Aastra 9133i which can support up to 9 calls.  I have
> restarted the service, rebooted the server and even upgraded to the
> latest zaptel drivers and made sure CentOS is patched and up to date.
> 
	I just made another test by dialing to a Zap channel instead of a SIP
phone and the call goes through without any problem.  It is just when
you try to dial to a SIP phone that you get the auto-congestion message.

-- 
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez Prats
Director de Tecnolog?a
+52-55-91169161 ext 2001
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Carlos Chavez wrote:
> On Tue, 2007-05-29 at 19:14 -0500, Carlos Chavez wrote:
>> On Tue, 2007-05-29 at 18:54 -0500, Carlos Chavez wrote:
>>> On Wed, 2007-05-30 at 02:49 +0300, Tzafrir Cohen wrote:
>>>
>>>> What is the output of:
>>>>
>>>>   show channels
>>>>
>>>> Any chance that there is actually another call that keeps that
channel
>>>> busy?
>>>>
>>> 	No, the line is not busy with another call.
>>>
>> 	In fact at this moment there are 0 active calls in the system.  The
>> phones are all Aastra 9133i which can support up to 9 calls.  I have
>> restarted the service, rebooted the server and even upgraded to the
>> latest zaptel drivers and made sure CentOS is patched and up to date.
>>
> 	I just made another test by dialing to a Zap channel instead of a SIP
> phone and the call goes through without any problem.  It is just when
> you try to dial to a SIP phone that you get the auto-congestion message.
> 
"sip show peers" will list the IP address of the phones.  If it is not
listed then you have a problem with your phones not registering to 
Asterisk.

On Tue, 29 May 2007 20:20:13 -0500, Eric \"ManxPower\" Wieling wrote
> >>
> > 	I just made another test by dialing to a Zap channel instead of a SIP
> > phone and the call goes through without any problem.  It is just when
> > you try to dial to a SIP phone that you get the auto-congestion
message.
> >
> 
     All other phones in the system are working properly, they are all
registered and you can send and receive calls from anywhere except that zap
channel.

--
Carlos Chavez
Director de Tecnolog?a
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Tel: +52-55-91169161 Ext 2001

Carlos Chavez wrote:
> On Wed, 2007-05-30 at 14:49 -0500, Eric "ManxPower" Wieling
wrote:
> 
>> Can two SIP phones on that system call each other?
> 
> 	Everything else in the system works, all sip phones can call each other
> and the PSTN.  They have a GSM adapter on the same card and they can
> place and receive calls.  Only calls coming from the Vonage ATA have
> this problem.
I missed the first part of the thread.  Can you paste the CLI output of 
a successful call (SIP phone to SIP phone) and an unsuccessful call (Zap 
to SIP)?