Re: [Asterisk-Dev] INFO and Duration=250

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,

RE: [Asterisk-Dev] INFO and Duration=250

[EMAIL PROTECTED] asterisk]# grep Duration= channels/*c channels/chan_sip.c:snprintf(tmp, sizeof(tmp), "Signal=%c\r\nDuration=250\r\n", digit); [EMAIL PROTECTED] asterisk]# > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-dev- > [EMAIL PROTECTED] On Behalf Of James Siz

[Asterisk-Dev] INFO and Duration=250

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, howev

Re: [Asterisk-Dev] which part of the code to receive and breakdownof SIP 401

Raymond Chen wrote: > I understand it's not a standard way of sending the sip header, but we what > to work with this provider nonetheless. So please point me to the part > where I can read the nonce from the sip 401 header. I don't think you can do that right now, because we don't even fold m

RE: [Asterisk-Dev] which part of the code to receive and breakdownof SIP 401

Kevin, I understand it's not a standard way of sending the sip header, but we what to work with this provider nonetheless. So please point me to the part where I can read the nonce from the sip 401 header. Thanks Ray -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED

RE: [Asterisk-Dev] which part of the code to receive and breakdownof SIP 401

Kevin, I understand it's not a standard way of sending the sip header, but we what to work with this provider nonetheless. So please point me to the part where I can read the nonce from the sip 401 header. Thanks Ray -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED

Re: [Asterisk-Dev] res_sqlite and sqliteInt.h

Tzafrir Cohen wrote: The makefile in the subdirectory res_sqlite grabs sqlite 3.2.1 . Is that sqlite 2? cdr_sqlite works with the libsqlite0 packages from Debian. I can't venture a guess about res_sqlite, since it's not part of Asterisk I've never looked at it :-) You'd be better off to post

Re: [Asterisk-Dev] which part of the code to receive and breakdown of SIP 401

Raymond Chen wrote: WWW-Authenticate: Digest realm="", nonce="c39ec9664fd22a67b9cd79c70877c24f" That is broken. Blank lines are not allowed in the headers; the first blank line signals the end of the headers and the start of the body. Whatever device is sending you this is very broken. ___

Re: [Asterisk-Dev] Voicemail -> new feature request

well adding this to voicemailmain will be silly IMHO you can do this with, say, something like this, giving the user an option of enabling voicemail by typing *91# and waiting for the tone, and turning it off with #91#. More error checking would be appropriate, though.. exten => *91#

[Asterisk-Dev] Statuses for IAX

Dear Team, I was looking for some help for IAX2 Callback functions using C#, iam using iaxclient.dll library and it is working fine but i need to get the statuses and the events but don't know how to make it using C#. Can anyone help please? Thank you, Tim