Mayur
2008-Jan-12 19:09 UTC
[asterisk-users] Asterisk RFC2833 to SIP INFO DTMF conversion erros.
Hi, I am using asterisk 1.4.17 which is connected to a SIP trunk supporting rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and for SIP clients I have set dtmfmode=info. So when I make a call to a cell number using the sip trunk and then press digits I can see the 2833 dtmf events coming to asterisk in the rtp captures. Asterisk seems to detect those and give SIP INFO to the SIP client. However it fails to detect some of the digits (which is random) hence the correct sequence of digits is not received at the SIP client. I have tried setting relaxdtmf=yes in sip.conf but that does not seem to help. Can anyone help me out here? Regards, Mayur -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080112/346eb71c/attachment.htm
If I enable the firewall on my Server, which ports should I open for Asterisk to work properly. Is it enough to just open the SIP ports? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080112/2f06a90a/attachment.htm
dave cantera
2008-Jan-13 05:45 UTC
[asterisk-users] Asterisk RFC2833 to SIP INFO DTMF conversion erros.
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mayur,<br>
did you try inband? with sip?<br>
daveC<br>
<tt>;dtmfmode=inband
; Choices are inband, rfc2833, or
info<br>
;allow=ulaw
; dtmfmode=inband only works with ulaw
or alaw!<br>
;dtmfmode=rfc2833
; Choices are inband, rfc2833, or info<br>
;allow=ulaw
; dtmfmode=inband only works with ulaw
or alaw!</tt><br>
<br>
Mayur wrote:
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<p class="MsoNormal"><font face="Arial"
size="2"><span
style="font-size: 10pt; font-family:
Arial;">Hi,<o:p></o:p></span></font></p>
<p class="MsoNormal"><font face="Arial"
size="2"><span
style="font-size: 10pt; font-family: Arial;">
I am using asterisk
1.4.17 which is connected to a SIP trunk
supporting rfc2833 dtmf events. Asterisk stays in the media path. In
sip.conf I
have set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk
account) and
for SIP clients I have set dtmfmode=info. So when I make a call to a
cell
number using the sip trunk and then press digits I can see the 2833
dtmf events
coming to asterisk in the rtp captures. Asterisk seems to detect those
and give
SIP INFO to the SIP client. However it fails to detect some of the
digits (which
is random) hence the correct sequence of digits is not received at the
SIP
client.<o:p></o:p></span></font></p>
<p class="MsoNormal"><font face="Arial"
size="2"><span
style="font-size: 10pt; font-family: Arial;">I have tried setting
relaxdtmf=yes in sip.conf but that does
not seem to help. Can anyone help me out
here?<o:p></o:p></span></font></p>
<p class="MsoNormal"><font face="Arial"
size="2"><span
style="font-size: 10pt; font-family:
Arial;"><o:p> </o:p></span></font></p>
<p class="MsoNormal"><font face="Arial"
size="2"><span
style="font-size: 10pt; font-family:
Arial;">Regards,<o:p></o:p></span></font></p>
<p class="MsoNormal"><font face="Arial"
size="2"><span
style="font-size: 10pt; font-family:
Arial;">Mayur<o:p></o:p></span></font></p>
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