I did a bit of searching around and found this class in chan_sip.c:
I am going to test the Duration at 500, and see how this effect
things. If anyone has already played with these values, and had any
bad gotchas please let me know.

==================
static int add_digit(struct sip_request *req, char digit)
{
       char tmp[256];
       int len;
       char clen[256];
       snprintf(tmp, sizeof(tmp), "Signal=%c\r\nDuration=250\r\n", digit);
       len = strlen(tmp);
       snprintf(clen, sizeof(clen), "%d", len);
       add_header(req, "Content-Type", "application/dtmf-relay");
       add_header(req, "Content-Length", clen);
       add_line(req, tmp);
       return 0;
}
==================


James Sizemore wrote:

I have a gateway using a Digium card to convert a PRI
call to a sip call then I transport the sip call to a Cisco
IAD where it is converted back to a PRI. This all works
well except DTMF is sent with a duration of .25sec.
PRI specs says this should be .25sec to .5sec so this
is with in spec, however the PBX on the other side of
the IAD does not reliable work with the DTMF tones
the minimum allowable length. I found in the INFO packets
where the DTMF is set to a duration or 250, I would like
to change this to 500.

Which file and class would be the correct place to change
this value at?

==============
INFO packet options I would like to change
==============
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=5
Duration=250
==============
_______________________________________________
Asterisk-Dev mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-dev
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-dev


_______________________________________________
Asterisk-Dev mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-dev
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-dev

Reply via email to