[Asterisk-Users] SIP and DTMF

Scott England scott at homelan.com
Fri Nov 14 18:06:45 MST 2003


Being relatively new to * I has what may be a simple question, I haven't
been able to find it in the archives though, or at least been able to
recognize it.

I have a 400P the is acting as a pstn gateway. It then forwards via IAX2
to another * server at another site. The calls then get routed via
callerid to a sip client with an exten  statement in extensions.conf.
However I cant seem to get DTMF to forward to the sip extension. With
IAX debugging I can see the dtmf call at both iax points but I dont see
it happen at the sip point. Do I have to do something different then a
simple exten => 6000,1,SIP/6000 at 192.168.0.108   ?

-- 
Scott England
General Manager, ControlNet Inc.
voice 408-850-4904
fax   408-866-4211

------------------------------------------------------------------------------
The information contained in this message may be privileged and 
confidential and protected from disclosure. If the reader of this 
message is not the intended recipient, or an employee or agent 
responsible for delivering this message to the intended recipient, you 
are hereby notified that any dissemination, distribution or copying of 
this communication is strictly prohibited. If you have received this 
communication in error, please notify us immediately by replying to the 
message and deleting it from your computer. Thank you.
------------------------------------------------------------------------------






More information about the asterisk-users mailing list