VoIP Mailing List Archives

[dgonzalez at denwaip.com]

Hello,


I have a problem with movistar in Mexico with a sip calls. Movistar send to me T38 and G729 in the INVITE and they say that I have to ignore T38 and use G729 in the voice call.


When a fax call is made Movistar send only T38 in the INVITE. 


Invite example:



v=0
o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2
s=sip call
c=IN IP4 192.168.1.2
t=0 0
m=audio 6370 RTP/AVP 18 101
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=image 6372 udptl t38
a=T38FaxVersion:0
a=T38FaxMaxBuffer:1100
a=T38FaxMaxDatagram:612
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy



How can I  ignore T38 and use only G729 for this call?.



Thanks for your help.


Damian




--

[alyed at vivoxie.com]

Think you only need to make sure you have in your sip.conf file these configs:


[your-device-name]

.....
.....
disallow=all

allow=g729
.....
.....


Alyed

2013/11/20 Damian Gonzalez <dgonzalez@denwaip.com (dgonzalez@denwaip.com)>

[dgonzalez at denwaip.com]

Hello,


Thanks for the quickly response. I have only G729 in the peer but I have t38pt_udptl= yes


If I put t38pt_udptl=no , asterisk reject the call with 488 code.


The problem is that Movistar send T38 codec in all calls and I need ignore only if in the SDP I have G729 and T38 (18 and 101), but if I have only T38 I have to negociate a fax call.



Thanks.



On Wed, Nov 20, 2013 at 4:46 PM, Alyed <alyed@vivoxie.com (alyed@vivoxie.com)> wrote:

[kris at kriskinc.com]

It is possible that Asterisk requires an rtpmap even for static payload types (I'm not sure about this).  The INVITE from your provider omits rtpmap for payload type 18 (G729) which is perfectly valid.



On Wed, Nov 20, 2013 at 2:56 PM, Damian Gonzalez <dgonzalez@denwaip.com (dgonzalez@denwaip.com)> wrote:

[dgonzalez at denwaip.com]

Any posible solution?


On Wed, Nov 20, 2013 at 6:03 PM, Kristian Kielhofner <kris@kriskinc.com (kris@kriskinc.com)> wrote:

[BryantZ at zktech.com]

Can you funnel them through a specific inbound dial context. Then force a re-invite to g729?

Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003


From: "Damian Gonzalez" <dgonzalez@denwaip.com>
Sent: Thursday, November 21, 2013 8:25 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Movistar sip Mexico

Any posible solution?


On Wed, Nov 20, 2013 at 6:03 PM, Kristian Kielhofner <kris@kriskinc.com (kris@kriskinc.com)> wrote:

[alyed at vivoxie.com]

Which version of Asterisk are you using?


According to http://www.voip-info.org/wiki/view/Asterisk%20T.38 unless you are using Asterisk 10, there's quite some patching (or buying) you'll need to be doing.


Alyed


2013/11/21 Bryant Zimmerman <BryantZ@zktech.com (BryantZ@zktech.com)>

[dgonzalez at denwaip.com]

Hi,


I have Asterisk 10.12.1. I can not figure out the solution. 


Thank you for your help.


Best Regards



On Thu, Nov 21, 2013 at 7:07 PM, Alyed <alyed@vivoxie.com (alyed@vivoxie.com)> wrote:

[alyed at vivoxie.com]

Have you followed the instructions in: https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway

and: http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway
??


If possible try with a different ATA since it seems not all of them work fine with fax pass trough.

Alyed


2013/11/21 Damian Gonzalez <dgonzalez@denwaip.com (dgonzalez@denwaip.com)>

[dotnetdub at gmail.com]

Why? 

On Wednesday, 20 November 2013, Damian Gonzalez wrote:

[lmoore at omninet.net.au]

On 22/11/2013 6:49 AM, Alyed wrote:

[lmoore at omninet.net.au]

On 21/11/2013 3:32 AM, Damian Gonzalez wrote:

[h323 at ramdyne.nl]

On 20/11/13 20:32 , Damian Gonzalez wrote: