[Asterisk-Users] Cisco GW and DTMF problems

[Billy Huddleston]

Problem was with asterisk.. Mark had made a change in chan_sip.c that 
affected noncodec capabilities, it's been fixed.

Thanks,
Billy

----- Original Message ----- 
From: "Tenorio, Leandro" <LTenorio at intelaction.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Thursday, September 09, 2004 6:42 PM
Subject: RE: [Asterisk-Users] Cisco GW and DTMF problems


I got a AS5350, and change the config to reflect your config as follows

AS5350
dial-peer voice 50101 voip
 preference 1
 destination-pattern xxxxxxxxxxx
 voice-class codec 1
 session protocol sipv2
 session target ipv4:xxx.xxx.xxx.xxx
 dtmf-relay rtp-nte

Etc....

And in *

sip.conf
[context]
type=friend
host=ip of AS5350
nat=yes
canreinvite=no
insecure=very
dtmfmode=rfc2833
disallow=all
allow=g729
;allow=ulaw
;allow=alaw

Etc..

Works just fine,

Please attach you config info if you want further help



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Billy
Huddleston
Sent: Wednesday, September 08, 2004 11:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco GW and DTMF problems

c2600-is5-mz.123-9

rfc2833

----- Original Message -----
From: "Tenorio, Leandro" <LTenorio at intelaction.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Wednesday, September 08, 2004 10:37 PM
Subject: RE: [Asterisk-Users] Cisco GW and DTMF problems


What version of IOS 're u using, and what's your dtmfmode in *?

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Billy
Huddleston
Sent: Wednesday, September 08, 2004 6:29 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Cisco GW and DTMF problems

I'm stuck running a old copy of asterisk because something strange is
going on in later versions of the CVS..

When I call in from a PSTN into my cisco 2610XM gateway which then
routes the call to my asterisk box via sip..  Asterisk can no longer
process DTMF tones generated by the calling party.  This affects DISA,
prompts and menus.. Has anyone else had this problem?? and use.. I DO
have dtmf-relay rtp-nte toggled in my dial peer..

Thanks, Billy


             +--------------------------------------------------+
             | Billy Huddleston   Senior Systems Administrator  |
             | Net-Express                  http://www.nxs.net  |
             | 114 Sherway Rd.             Voice: 865-691-2011  |
             | Knoxville, TN  37922          Fax: 865-691-9894  |
             | billy at nxs.net                                    |
             +--------------------------------------------------+

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