I am trying to connect to a vocal server from an asterisk server. A call is received via iax2 to my asterisk server. I then initiate a SIP connection to the vocal server. everything works great except dtmf doesnt work. A cisco 5300 can connect to this vocal server and do dtmf without a problem. I have my dtmf set to rfc2833 in the general section of the sip.conf . I can confirm that the channel is in rfc2833 during the call via show channel. With SIP debug though I dont see any event for dtmf. I do see dtmf in IAX though.
I think that you are thinking of SIP INFO messages if you are expecting to see something in the SIP messaging. RFC2833 is sent as part of the RTP packets so you are not going to see a plain text 1,2,3,4,etc in a trace when using it. http://www.faqs.org/rfcs/rfc2833.html Sean ----- Original Message ----- From: "Scott England" <scott@homelan.com> To: <asterisk-users@lists.digium.com> Sent: Monday, November 17, 2003 5:58 AM Subject: [Asterisk-Users] DTMF> I am trying to connect to a vocal server from an asterisk server. A call > is received via iax2 to my asterisk server. I then initiate a SIP > connection to the vocal server. everything works great except dtmf > doesnt work. A cisco 5300 can connect to this vocal server and do dtmf > without a problem. I have my dtmf set to rfc2833 in the general section > of the sip.conf . I can confirm that the channel is in rfc2833 during > the call via show channel. With SIP debug though I dont see any event > for dtmf. I do see dtmf in IAX though. > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
I dont expect to see an ascii code or such since the tones are in a rtp stream. But when I place the dtmf type to "info" in the sip.conf and make a call I see this under asterisk with sip debug on. DEBUG[122896]: File rtp.c, Line 942 (ast_rtp_raw_write): Difference is 960, ms i s 140 I assume this is asterisk sending the dtmf tone, but if I switch to rfc2833 I dont see anything. What I am looking for is a way to verify * is sending the dtmf to the vocal server, since it does not see the dtmf even though the audio portion is in operation and I know dtmf works between the vocal server and a cisco AS5300. Scott England Sean P. Robertson wrote:>I think that you are thinking of SIP INFO messages if you are expecting to >see something in the SIP messaging. RFC2833 is sent as part of the RTP >packets so you are not going to see a plain text 1,2,3,4,etc in a trace when >using it. > >http://www.faqs.org/rfcs/rfc2833.html > > >Sean >----- Original Message ----- >From: "Scott England" <scott@homelan.com> >To: <asterisk-users@lists.digium.com> >Sent: Monday, November 17, 2003 5:58 AM >Subject: [Asterisk-Users] DTMF > > > > >>I am trying to connect to a vocal server from an asterisk server. A call >>is received via iax2 to my asterisk server. I then initiate a SIP >>connection to the vocal server. everything works great except dtmf >>doesnt work. A cisco 5300 can connect to this vocal server and do dtmf >>without a problem. I have my dtmf set to rfc2833 in the general section >>of the sip.conf . I can confirm that the channel is in rfc2833 during >>the call via show channel. With SIP debug though I dont see any event >>for dtmf. I do see dtmf in IAX though. >> >>_______________________________________________ >>Asterisk-Users mailing list >>Asterisk-Users@lists.digium.com >>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 > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031117/ef6ed5ee/attachment.htm
I apologize. When you said that you were looking in the SIP debug, I thought
that you were expecting the rfc2833 to be a SIP message.
You might could do something with Ethereal that would show you what is going on.
Sean
----- Original Message -----
From: Scott England
To: asterisk-users@lists.digium.com
Sent: Monday, November 17, 2003 1:10 PM
Subject: Re: [Asterisk-Users] DTMF
I dont expect to see an ascii code or such since the tones are in a rtp
stream. But when I place the dtmf type to "info" in the sip.conf and
make a call I see this under asterisk with sip debug on.
DEBUG[122896]: File rtp.c, Line 942 (ast_rtp_raw_write): Difference is 960, ms
i
s 140
I assume this is asterisk sending the dtmf tone, but if I switch to rfc2833 I
dont see anything.
What I am looking for is a way to verify * is sending the dtmf to the vocal
server, since it does not see the dtmf even though the audio portion is in
operation and I know dtmf works between the vocal server and a cisco AS5300.
Scott England
Sean P. Robertson wrote:
I think that you are thinking of SIP INFO messages if you are expecting to
see something in the SIP messaging. RFC2833 is sent as part of the RTP
packets so you are not going to see a plain text 1,2,3,4,etc in a trace when
using it.
http://www.faqs.org/rfcs/rfc2833.html
Sean
----- Original Message -----
From: "Scott England" <scott@homelan.com>
To: <asterisk-users@lists.digium.com>
Sent: Monday, November 17, 2003 5:58 AM
Subject: [Asterisk-Users] DTMF
I am trying to connect to a vocal server from an asterisk server. A call
is received via iax2 to my asterisk server. I then initiate a SIP
connection to the vocal server. everything works great except dtmf
doesnt work. A cisco 5300 can connect to this vocal server and do dtmf
without a problem. I have my dtmf set to rfc2833 in the general section
of the sip.conf . I can confirm that the channel is in rfc2833 during
the call via show channel. With SIP debug though I dont see any event
for dtmf. I do see dtmf in IAX though.
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
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
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I can't resist asking. What do you think of Vocal as compared to *? Anything Vocal has but missing in *? ---------- Original Message ---------------------------------- From: Scott England <scott@homelan.com> Reply-To: asterisk-users@lists.digium.com Date: Mon, 17 Nov 2003 02:58:55 -0800>I am trying to connect to a vocal server from an asterisk server. A call >is received via iax2 to my asterisk server. I then initiate a SIP >connection to the vocal server. everything works great except dtmf >doesnt work. A cisco 5300 can connect to this vocal server and do dtmf >without a problem. I have my dtmf set to rfc2833 in the general section >of the sip.conf . I can confirm that the channel is in rfc2833 during >the call via show channel. With SIP debug though I dont see any event >for dtmf. I do see dtmf in IAX though. > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >-- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --
Hello all, Is it possible to send dtmf tones to an answering terminal (after answering the call)? I have for example a external voicemail system that I want to connect to *. Now for the right integration I need to send dtmf tones to the analog ports that answered the call. Cheers. Robin -------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef Size: 1306 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041213/19e0d692/winmail.bin
Hi, i need to set the dtmf mode on my quintum tenor a400 gateway. i cant dial any extension thru my normal digital phone which is connected to asterisk thru the quintum gateway. it always falls to 's' extension. So plz help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060706/2934b12b/attachment.htm
try setting your dial plan in sip.conf using dtmf = rfc2833 -------------- Original message -------------- From: El Flynn <el_flynn@lanvik-icu.com>> Rizwan Hisham wrote: > > Hi, > > i need to set the dtmf mode on my quintum tenor a400 gateway. > > You might want to check the a400 manual on how to do that. > > > i cant dial > > any extension thru my normal digital phone which is connected to asterisk > > thru the quintum gateway. it always falls to 's' extension. So plz help > > > > This is most likely a misconfiguration of your dialplan and/or sip.conf files. > it would help if you post it here? > > Flynn > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > 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/20060706/70692b94/attachment.htm
Hi Guys, I need a little help in using DTMF settings. Im using SIP and H323 channels, both are set to use dtmf=rfc2833. 2 days ago it was working fine, it still works fine when im in conference, for example when i use the following extension: exten=>1234,1,MeetMe(1234|X|) by using this extension im able to jump to any extension i want by dialing that extension. The problem occurs when i use the Dial() application: exten=>1,1,Dial(SIP/200,,tT) when i press # nothing happens i have no idea how to solve this problem -- Regards Rizwan Hisham Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060717/cc058924/attachment.htm