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@nxs.net |
+--------------------------------------------------+
What version of IOS 're u using, and what's your dtmfmode in *?
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Billy
Huddleston
Sent: Wednesday, September 08, 2004 6:29 PM
To: asterisk-users@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@nxs.net |
+--------------------------------------------------+
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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@lists.digium.com
[mailto:asterisk-users-bounces@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@intelaction.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Billy
Huddleston
Sent: Wednesday, September 08, 2004 6:29 PM
To: asterisk-users@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@nxs.net |
+--------------------------------------------------+
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
asterisk-users-bounces@lists.digium.com wrote:> Problem was with asterisk.. Mark had made a change in chan_sip.c > that affected noncodec capabilities, it's been fixed.Do you have a bug number? Or something else to find it in the bug database? -- Andreas Sikkema Rits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540
>Do you have a bug number? Or something else to findit in the bug database? bug #2394 Seems, the minor issue with "Non-codec capabilities" in sip debug still exists. Arsen. __________________________________ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail