Hello all!
I've Queue Log Analyzer installed
[http://www.micpc.com/qloganalyzer] to check Asterisk main stats.
There's an option of CALLS COMPLETED [ALL] where I can see the
completed calls that entered any of the queues. There, there's a
column that displays the extention number of the
Hello all,
I have an Asterisk PBX with the Queue Log Analyzer installed
[http://www.micpc.com/qloganalyzer].
On the main menu, there's an option of CALLS COMPLETED [ALL] where
I can see the completed calls that entered any of the queues and my
question is:
There's a column that states
with iax2 users?
Your opinions will be very
appreciated!
Regards,
Lan.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk
it.
I think we need to create a knowledge base database, so members can come in
to search and share the information that we need?
Thanks!
Lan.
- Original Message -
From: snacktime [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Hi, When i place a call on hold, and
then return to it, the caller then hears my voice in a delay usually equal
to the amount of time i put them on hold. I have the problem only with G729
codec and with my voip provider (i live in france
and i use wengo)
My
configuration : - Pentium III
Hello Eduardo,
Wednesday, December 17, 2003, 1:08:00 AM, you wrote:
EG Hi list,
EG I'm with a little problem on codec negotiation between a cisco827 and
EG asterisk.
EG My sip.conf is like that:
EG [general]
EG port = 5060
EG bindaddr = 0.0.0.0
EG context = default
EG
Hi Andrew,
Tuesday, November 4, 2003, 5:04:04 AM, you wrote:
AJ I have used G723.1 (although unlicensed) with Asterisk. The info is even
AJ in the Makefile, just drop in a few files in your source directoy,
AJ uncomment something in the Makefile and instant G723.1 support...
Thanks for the
Hello guys,
I have searched high and low, but not found any information about
rules of using DTMF in SIP INFO method. Cisco has described something with
Signal=, but it look like this feature is dependent on implementors?
The problem is chan_sip.c cannot correctly translate received DTMF
digits,
Hello Brancaleoni,
Sunday, October 12, 2003, 4:39:32 PM, you wrote:
BM Hi.
BM The implementation is correct, I can use sip info
BM method to get all the DMTF, *,# included (eg voicemail
BM works great with sip info dtmf)
BM the line atoi(buf) is needed 'cause buf is a char, and
BM we need a
