Hello,When i use mixmonitor() I manage not to detect still the silence. what a local channel. My asterisk is connected to an operator among whom the channel it is for example "SIP/from provider 0048".In fact, I do not want to record the silence.I want that when caller do not speak, I can as
Dear asterisk friends,
Can someone tell me whether asterisk supports PAM authentication or not?
--
Met vriendelijke groeten,
With kind regards,
Mit freundlichen Gruessen,
Will
*
W.K. Offermans
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Hi Everyone,
I am trying to setup an Audio Call from firefox WebRTC to Asterisk. The
Flow is:
PC -> SIPoWS -> KAMAILIO -> SIPoUDP -> ASTERISK
Regular call (no srtp) works fine. However when I setup SRTP the asterisk
replies with 488 Not Acceptable Here.
I followed the Secure Calling Tutorial, but
A recent update to the glibc-headers package (2.22-17) changed the order of
members in the sockaddr_storage structure which will cause an Asterisk
compile failure. We shouldn't have been relying on the order and therefore
patches are up on gerrit for the 11, 13 and master branches.
master: https
Hello,
My ITSP provides me with a SIP trunk which requires a CallerID value for
any outbound call.
Though a CallerID is required, anonymous calls are allowed.
See extracts from a successfull anonymous call:
From: "Anonymous" ;tag=438b284694b5b3de
Privacy: id
P-Asserted-Identity: "FooBar" >
I'm
On Thu, Jun 9, 2016 at 11:40 AM, Olivier wrote:
> Hello,
>
> My ITSP provides me with a SIP trunk which requires a CallerID value for
> any outbound call.
> Though a CallerID is required, anonymous calls are allowed.
> See extracts from a successfull anonymous call:
>
> From: "Anonymous" ;tag=438
Looks like the hookstate is listed as offhook. I don't think
chan_dahdi will attempt to make a call out a device that is offhook.
Hope that helps,
Matthew Fredrickson
On Tue, Jun 7, 2016 at 1:36 PM, Brent Davidson
wrote:
> In trying to troubleshoot the Delay after Answer problem I had before (w
