Hi,
I use CEL or Call Event Logging in 1.8 to get a more concise picture of what
happened in a call. We use it for a bunch of stuff including billing attended
and unattended transfers differently.
If you are thinking of upgrading, it's worth a try.
Nic.
-Original Message-
From:
On blind transfers I believe the two cdr's have the same unique id . On
attended transfers there is no real way I have found to address this issue.
CDR's with transfers really suck the way they are right now. On blind transfers
you can do some flagging of the second CDR by checking in your
Nic
How stable is 1.8 really? It sounds like you are running it in production is
this the case? CDR Transfer issues and rfc2833 DTMF issues are hitting us hard
with 1.6.2.x. We want to move as soon as 1.8 is stable enough.
Thanks
Bryant
From: Nic
Can I generate SIP registration and call from Asterisk without a SIP client? I
need to initiate a call from asterisk and play a recorded message.
Gautam
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Yes. Send your code. Consider using call files.
Here is a part of what works for me.
[-system]
exten = s,1,Answer
exten = s,n,Wait(2)
exten = s,n,Playback(pa-welcome) please record your broadcast
after the beep
;exten = s,n,Playback(beep)
exten = s,n,Wait(1)
exten =
Am 06.09.2010 00:20, schrieb Gautam Desai:
Can I generate SIP registration and call from Asterisk without a SIP client?
I
need to initiate a call from asterisk and play a recorded message.
Gautam
hello,
have a look at the sip.conf.sample file how to register asterisk as
It can be done either using a call file and a clever dial plan or via
the manager interface, again with a clever dialplan
On 09/05/2010 03:20 PM, Gautam Desai wrote:
Can I generate SIP registration and call from Asterisk without a SIP
client? I need to initiate a call from asterisk and play a
On 9/4/2010 1:31 AM, Jeremy Kister thought:
On 8/29/2010 3:25 AM, Jeremy Kister wrote:
whenever a call goes through the 1760's FXO or FXS (in or out) there is
a 915 second maximum call time due to asterisk hanging up the call
because of a critical packet being missed.
hmm, either no one
Bryant,
We have been using a pre-1.8 trunk version of asterisk that has been pretty
stable for us. We have a fairly small user base currently and decided to take
the risk with a trunk version after some testing basically because of the
availability of CEL as it lets us do a bunch of things we
