Re: [asterisk-users] no sound between extensions

2010-06-02 Thread Gary Baribault 
I don't know if this makes any difference, I created a lot of this
configuration with the Asterisk-GUI (SVN-branch-2.0-r4980) and when I
edit the users.conf file, there are two entries 'type = peer' for each
extension and they are highlighted in red!

Gary Baribault

On 06/02/2010 08:32 AM, taimur hasan wrote:
>
> Also check the codecs as if you are using g729 or g723, there is a
> chance that they are not available in codecs directory (
> /usr/lib/asterisk/modules).
>
> *-THQ-  !!!ONE*
>
>
>
>
>
> 
> Date: Tue, 1 Jun 2010 19:24:41 -0400
> From: zisha...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] no sound between extensions
>
> Do you agree something is blocking the audio in one direction? Can you
> do a 'rtp debug' and then initiate a SIP call and see if there is two
> way audio traffic. Also make sure these extensions have 'canreinvite=no'.
>
> Zeeshan A Zakaria
> --
> Sent from my Android phone with K-9 Mail.
>
> On 2010-06-01 7:02 PM, "Gary Baribault"  <mailto:g...@baribault.net>> wrote:
>
> As I stated, the incoming calls are on TDM DS0s connected to the
> Digium card, and the extensions are on the same local network as
> the Asterisk server. There is currently no NAT anywhere.
>
> Gary Baribault
>
> On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote:
> >
> > Output of 'iptables -L -n' would also be helpfu...
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> 
> Hotmail: Trusted email with powerful SPAM protection. Sign up now.
> <https://signup.live.com/signup.aspx?id=60969>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] no sound between extensions

2010-06-02 Thread Gary Baribault 
I have checked, the users have ulaw, then alaw, the phones are set to
711u then 711a which is the same thing (I think).

Gary Baribault

On 06/02/2010 08:32 AM, taimur hasan wrote:
>
> Also check the codecs as if you are using g729 or g723, there is a
> chance that they are not available in codecs directory (
> /usr/lib/asterisk/modules).
>
> *-THQ-  !!!ONE*
>
>
>
>
>
> 
> Date: Tue, 1 Jun 2010 19:24:41 -0400
> From: zisha...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] no sound between extensions
>
> Do you agree something is blocking the audio in one direction? Can you
> do a 'rtp debug' and then initiate a SIP call and see if there is two
> way audio traffic. Also make sure these extensions have 'canreinvite=no'.
>
> Zeeshan A Zakaria
> --
> Sent from my Android phone with K-9 Mail.
>
> On 2010-06-01 7:02 PM, "Gary Baribault"  <mailto:g...@baribault.net>> wrote:
>
> As I stated, the incoming calls are on TDM DS0s connected to the
> Digium card, and the extensions are on the same local network as
> the Asterisk server. There is currently no NAT anywhere.
>
> Gary Baribault
>
> On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote:
> >
> > Output of 'iptables -L -n' would also be helpfu...
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> 
> Hotmail: Trusted email with powerful SPAM protection. Sign up now.
> <https://signup.live.com/signup.aspx?id=60969>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] no sound between extensions

2010-06-02 Thread Gary Baribault 
I have remote access to the server so I checked the canreinvite .. they
are all set to no. I can't try the call from here, I will get back to you.

Gary Baribault


On 06/01/2010 07:24 PM, Zeeshan Zakaria wrote:
>
> Do you agree something is blocking the audio in one direction? Can you
> do a 'rtp debug' and then initiate a SIP call and see if there is two
> way audio traffic. Also make sure these extensions have 'canreinvite=no'.
>
> Zeeshan A Zakaria
>
> --
> Sent from my Android phone with K-9 Mail.
>
>> On 2010-06-01 7:02 PM, "Gary Baribault" > > wrote:
>>
>> As I stated, the incoming calls are on TDM DS0s connected to the
>> Digium card, and the extensions are on the same local network as the
>> Asterisk server. There is currently no NAT anywhere.
>>
>> Gary Baribault
>>
>>
>>
>> On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote:
>> >
>> > Output of 'iptables -L -n' would also be helpfu...
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> 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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] no sound between extensions

2010-06-02 Thread taimur hasan 


Also check the codecs as if you are using g729 or g723, there is a chance that 
they are not available in codecs directory ( /usr/lib/asterisk/modules).
-THQ-  !!!ONE



Date: Tue, 1 Jun 2010 19:24:41 -0400
From: zisha...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] no sound between extensions

Do you agree something is blocking the audio in one direction? Can you do a 
'rtp debug' and then initiate a SIP call and see if there is two way audio 
traffic. Also make sure these extensions have 'canreinvite=no'.



Zeeshan A Zakaria

--

Sent from my Android phone with K-9 Mail.


On 2010-06-01 7:02 PM, "Gary Baribault"  wrote:




  


As I stated, the incoming calls are on TDM DS0s connected to the Digium
card, and the extensions are on the same local network as the Asterisk
server. There is currently no NAT anywhere.



Gary Baribault

On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote:
>
> Output of 'iptables -L -n' would also be helpfu...



--

_

-- Bandwidth and Colocation Provided by http://www.api-digital.com --

New to Asterisk? Join us for a live introductory webinar every Thurs:

   http://www.asterisk.org/hello



asterisk-users mailing list

To UNSUBSCRIBE or update options visit:

   http://lists.digium.com/mailman/listinfo/asterisk-users

  
_
Hotmail: Trusted email with powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] no sound between extensions

2010-06-01 Thread Zeeshan Zakaria 
Do you agree something is blocking the audio in one direction? Can you do a
'rtp debug' and then initiate a SIP call and see if there is two way audio
traffic. Also make sure these extensions have 'canreinvite=no'.

Zeeshan A Zakaria

--
Sent from my Android phone with K-9 Mail.

On 2010-06-01 7:02 PM, "Gary Baribault"  wrote:

 As I stated, the incoming calls are on TDM DS0s connected to the Digium
card, and the extensions are on the same local network as the Asterisk
server. There is currently no NAT anywhere.

Gary Baribault



On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote:
>
> Output of 'iptables -L -n' would also be helpfu...

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] no sound between extensions

2010-06-01 Thread Gary Baribault 
As I stated, the incoming calls are on TDM DS0s connected to the Digium
card, and the extensions are on the same local network as the Asterisk
server. There is currently no NAT anywhere.

Gary Baribault

On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote:
>
> Output of 'iptables -L -n' would also be helpful. I am sure its a NAT
> issue if incoming and ougoing calls are on ZAP channels.
>
> Zeeshan A Zakaria
>
> --
> Sent from my Android phone with K-9 Mail.
>
>> On 2010-06-01 3:53 PM, "Danny Nicholas" > > wrote:
>>
>> My assumption is that inbound/outbound calls are DAHDI and that internal
>> calls are SIP.  Can OP post "core show channels" from working and
>> non-working calls?
>>
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> 
>> [mailto:asterisk-users-bou. ..
>>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] no sound between extensions

2010-06-01 Thread Gary Baribault 
This is done while the calls are active? I just issued the command and
got nothing, but there where no active calls.

Gary Baribault

On 06/01/2010 03:45 PM, Danny Nicholas wrote:
> My assumption is that inbound/outbound calls are DAHDI and that internal
> calls are SIP.  Can OP post "core show channels" from working and
> non-working calls?
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Baribault
> Sent: Tuesday, June 01, 2010 2:20 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] no sound between extensions
>
> Hello all,
>
>I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a
> Digium 8 port FXO card. The local network is 100Mbps Ethernet and my
> phones are Linksys SPA-921 or Linksys Analog adaptors.
>
>The phones are setup with DHCP, and are on the same flat non-routed
> network. There is no NAT involved.
>
>If I call from extension 6000 to extension 6001, or vice-versa both
> are SPA-921s. The 6001 rings, but when the phone is picked up, I have
> no sound. I have the same problem between any phones in the system,
> but this is the simplest example.
>
>Incoming calls and outgoing calls work fine, sound is correct.
> Voice mail works fine as well, the IVR works great.
>
>Any ideas?
>
> Gary Baribault
>
>
>
>   

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] no sound between extensions

2010-06-01 Thread Gary Baribault 
Incomming calls are on TDM lines connected to the Digium card. Calls
between extentions are on the LAN for SIP registered users/ip phones.

Gary Baribault



On 06/01/2010 03:32 PM, Zeeshan Zakaria wrote:
>
> Incoming and outgoing calls are on SIP or on ZAP?
>
> Zeeshan A Zakaria
>
> --
> Sent from my Android phone with K-9 Mail.
>
>> On 2010-06-01 3:28 PM, "Gary Baribault" > > wrote:
>>
>> Hello all,
>>
>>   I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a
>> Digium 8 port FXO card. The local network is 100Mbps Ethernet and my
>> phones are Linksys SPA-921 or Linksys Analog adaptors.
>>
>>   The phones are setup with DHCP, and are on the same flat non-routed
>> network. There is no NAT involved.
>>
>>   If I call from extension 6000 to extension 6001, or vice-versa both
>> are SPA-921s. The 6001 rings, but when the phone is picked up, I have
>> no sound. I have the same problem between any phones in the system,
>> but this is the simplest example.
>>
>>   Incoming calls and outgoing calls work fine, sound is correct.
>> Voice mail works fine as well, the IVR works great.
>>
>>   Any ideas?
>>
>> Gary Baribault
>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> 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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] no sound between extensions

2010-06-01 Thread Zeeshan Zakaria 
Output of 'iptables -L -n' would also be helpful. I am sure its a NAT issue
if incoming and ougoing calls are on ZAP channels.

Zeeshan A Zakaria

--
Sent from my Android phone with K-9 Mail.

On 2010-06-01 3:53 PM, "Danny Nicholas"  wrote:

My assumption is that inbound/outbound calls are DAHDI and that internal
calls are SIP.  Can OP post "core show channels" from working and
non-working calls?


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-bou...
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] no sound between extensions

2010-06-01 Thread Danny Nicholas 
My assumption is that inbound/outbound calls are DAHDI and that internal
calls are SIP.  Can OP post "core show channels" from working and
non-working calls?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Baribault
Sent: Tuesday, June 01, 2010 2:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] no sound between extensions

Hello all,

   I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a
Digium 8 port FXO card. The local network is 100Mbps Ethernet and my
phones are Linksys SPA-921 or Linksys Analog adaptors.

   The phones are setup with DHCP, and are on the same flat non-routed
network. There is no NAT involved.

   If I call from extension 6000 to extension 6001, or vice-versa both
are SPA-921s. The 6001 rings, but when the phone is picked up, I have
no sound. I have the same problem between any phones in the system,
but this is the simplest example.

   Incoming calls and outgoing calls work fine, sound is correct.
Voice mail works fine as well, the IVR works great.

   Any ideas?

Gary Baribault



-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] no sound between extensions

2010-06-01 Thread Zeeshan Zakaria 
Incoming and outgoing calls are on SIP or on ZAP?

Zeeshan A Zakaria

--
Sent from my Android phone with K-9 Mail.

On 2010-06-01 3:28 PM, "Gary Baribault"  wrote:

Hello all,

  I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a
Digium 8 port FXO card. The local network is 100Mbps Ethernet and my
phones are Linksys SPA-921 or Linksys Analog adaptors.

  The phones are setup with DHCP, and are on the same flat non-routed
network. There is no NAT involved.

  If I call from extension 6000 to extension 6001, or vice-versa both
are SPA-921s. The 6001 rings, but when the phone is picked up, I have
no sound. I have the same problem between any phones in the system,
but this is the simplest example.

  Incoming calls and outgoing calls work fine, sound is correct.
Voice mail works fine as well, the IVR works great.

  Any ideas?

Gary Baribault



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users