Re: [Asterisk-Dev] INFO and Duration=250
Doubling the value to 500 did not seem to effect the length of the
tone played at allhm. Back to the drawing board for me.
Anyone know what this value is supposed to effect?
James Sizemore wrote:
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
==
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Re: [Asterisk-Dev] INFO and Duration=250
On 10/17/2005, James Sizemore [EMAIL PROTECTED] wrote: Doubling the value to 500 did not seem to effect the length of the tone played at allhm. Back to the drawing board for me. Anyone know what this value is supposed to effect? 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? Since it is the duration of the DTMF that goes INTO the PRI, I would be looking at where that happens - the Cisco config or the Digium config. Don't know a thing about the Cisco - but there is a toneduration in the zapata.conf. Since you say you get a call from the Digium card - SIP - Cisco - PRI I really think you need to look at the Cisco. Brett ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
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Re: [Asterisk-Dev] Bug 4301 - ztdummy accuracy problem
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