[Asterisk-Users] Fax and codecs preferences to PSTN

[René Mayorga]

Hi,

I already have a G729 license and the voice is OK
also I tried to use G729 codecs on SIP and AS5400 side, without any g729
license and works fine with a pass-trought configuration, so you only
need to check out voip-info.org to solve that issue.
on the stun and NAT part, I have not reached to that part yet 


On Wed, 2005-06-01 at 21:06 -0500, Obaid Siddiqui wrote:
> Hi,
> I am using the exact scenario you are using (As5400-asterisk-ata's). But I
> have not reached to fax
> issue yet.
> I tried codec g729 for voice call, it is not working. I think you have to
> buy g729 from digium.
> 
> Since both my Asterisk and SIP clients are in different NATS , I have to do
> the painful part of port forwarding for every client.  Specially if I have
> more then one client on same NAT, what needs to be done.
> 
> Voip-wiki suggest using SER or STUN. Can SER and  MyStun can be installed on
> same Server where Asterisk is residing.
> 
> Will somebody suggest anything.
> 
> Regards,
> OMS
> Prizm Communications
> 
> 
> 
> ----- Original Message ----- 
> From: "René Mayorga" <rmayorga at saltel.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Sent: Wednesday, June 01, 2005 12:05 PM
> Subject: [Asterisk-Users] Fax and codecs preferences to PSTN
> 
> 
> Hi,
> I have an asterisk running with a passtrought conf with G729,
> when I try to send a fax from SIP to SIP  the ATAs make a good codec
> negociation and the fax transmicion is OK,
> 
> But when I try to send the fax to PSTN fax machine
> (SIP --> AS5400 --> PSTN)
> The ATA Device try to send the RTP with G711ulaw and the Cisco keep
> answereing with G729
> a snip some part of my confs.
> 
> < sip.conf >
> [general]
> port=5060
> bindaddr=0.0.0.0
> context=default
> allow=g729
> .
> .
> .
> [22194007]
> type=friend
> host=dynamic
> secret=22194007
> canreinvite=yes
> callerid=ATA Sipura FAX <22194007>
> .
> .
> .
> 
> [as5400]
> type=friend
> host=XXX.XXX.XXX.XXX
> canreinvite=yes
> insecure=yes
> insecure=very
> qualify=yes
> 
> </sip.conf>
> 
> 
> <as5400>
> 
> dial-peer voice 999001 pots
>  description PRUEBAS SIP
>  max-conn 3
>  destination-pattern 65732.%
>  progress_ind alert enable 8
>  port 7/5:D
> !
> dial-peer voice 999000 voip
>  description PRUEBAS SIP
>  destination-pattern 2219400.
>  session protocol sipv2
>  session target sip-server
>  dtmf-relay h245-alphanumeric
>  fax-relay ecm disable
>  fax rate 9600
>  fax nsf 000000
>  fax protocol pass-through g711alaw
>  no vad
> 
> </as5400>
> 
> Thanks in advance
-- 
René Mayorga
Internet & Data 
El Salvador Telecom S.A. de S.V.
Tel:(503) 2247-7246
    (503) 2247-7156
Cel:(503) 7962-8205