[asterisk-users] Asterisk Avaya SIP Trunking One Way Audio

Skyler skchopperguy at gmail.com
Thu Apr 7 22:38:13 CDT 2011


First, I'm pretty sure avaya peer needs to friend. Try adding the below to
sip.conf and do a reload.

 

[general]

externip = the.wan.ext.ip

localnet = 192.168.1.0/255.255.255.0

 

 If that doesn't work, add nat=yes to avaya peer=friend

 

Skyler

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Lyle Giese
Sent: Thursday, April 07, 2011 6:39 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Asterisk Avaya SIP Trunking One Way Audio

 

On 04/07/11 03:00, Shariq Khan wrote:
> I am facing one way audio problem in sip trunking between asterisk and
> avaya.
>
>                +-------------+       +----+
>                | avaya sip   |-------| P1 |
>                +-------------+       +----+
>                       |
>                       |
>                       |
>                +-------------+
>                |  Asterisk   |               WAN
> -------------------------------------------------
>                |             |               LAN
>                +-------------+
>                   |
>                   /
>         +----+   /
>         | P2 |--+
>         +----+
>
> When P1 dial P2, P2 hears voice clear but P1 could not hear any voice.
>
> My sip.conf is
>
> [avaya]
> type=peer
> fromdomain=xx.xx.xx.xx
> host=xx.xx.xx.xx
> disallow=all
> allow=ulaw
> dtmfmode=rfc2833
> canreinvite=yes
>
>
> --
> Regards,
> Shariq Khan
> 0333-3501125
>
>
>
> --
> _____________________________________________________________________
> -- 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

Turn off reinvite on all extensions and SIP trunks involved and try again.

Lyle Giese
LCR Computer Services, Inc.

--
_____________________________________________________________________
-- 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 

  _____  

No virus found in this message.
Checked by AVG - www.avg.com
Version: 10.0.1204 / Virus Database: 1498/3523 - Release Date: 03/22/11
Internal Virus Database is out of date.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110407/d5644dc2/attachment.htm>


More information about the asterisk-users mailing list