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 -- -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131120/74d1dc38/attachment.html>
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 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 > > > -- > > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131120/f6f4dfde/attachment.html>
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 at denwaip.com>
Sent: Thursday, November 21, 2013 8:25 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at 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 at
kriskinc.com>
wrote:
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 at
denwaip.com>
wrote:
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 at vivoxie.com> wrote:
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 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
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Why? On Wednesday, 20 November 2013, Damian Gonzalez wrote:> 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 > > > -- > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131121/29910077/attachment.html>
On 21/11/2013 3:32 AM, Damian Gonzalez wrote:> 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 > >Perhaps you could add the following to the peer configuration faxdetect=no Larry.
On 20/11/13 20:32 , Damian Gonzalez wrote:> 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.I have had the same problem with a carrier, where some calls we receive from them have an image and an audio stream in the initial INVITE, even though the call is intended to use the audio stream. Responding back accepting T.38 will fail and *all* other options trying to reject the T.38 using known SIP supported methods will also fail. The *only* option is to just ignore the image stream, which is not allowed by the current set of SIP RFCs... Asterisk used to ignore the image stream, but since the 1.8(?) timeframe its behaviour has changed more towards standards compliance in this area. And now we're between a rock and a hard place. The only way out that I could find is to put something in front of Asterisk that just removes the image stream from initial INVITEs when received from the carrier. (OpenSIPS has this nice method called remove_stream() since a couple of versions) Complaining about this didn't help, "Asterisk is not certified because Open Source", was basically their answer. -- Andreas Sikkema