I have a call recording (audio) requirement that isn't addressed by
local Monitor/Record features.
All signalling and media currently pass through the Asterisk servers,
so that won't be an issue.
Instead of locally recording audio, for certain calls I need to add
what is effectively a 3rd leg to
I think it's 'divert.noanswer', found in site.cfg, or at least that's where I
have it. It's set to enabled and it still doesn't work. Out of curiosity, do
you have reg.1.fwd.noanswer.status set anywhere?
From: asterisk-users-boun...@lists.digium.com
Tom Browning wrote:
I have a call recording (audio) requirement that isn't addressed by
local Monitor/Record features.
All signalling and media currently pass through the Asterisk servers,
so that won't be an issue.
Instead of locally recording audio, for certain calls I need to add
what is
I am trying to get a digital accoustics talkmaster to register to
asterisk 1.4.43
I am getting the 401 unauthorized.
I have
host=dynamic
I have verified the passwords match
What else is there?
I dont see any further clues in sip set debug.
all it says is using request as basis request
What
Please post the sip.conf entry with any confidential data xxx'ed out.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, December 13, 2012 2:09 PM
To: Asterisk Users Mailing List -
[5001]
type=friend
username=5001
secret=XXX
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
context=incoming
host=dynamic
canreinvite=no
qualify=no
trustrpid=yes
sendrpid=no
nat=no
I did notice one more thing:
chan_sip.c:17045 handle_request_register: Registration from
The two things I would try are changing type from friend to peer and
sendrpid from no to yes. The no matching peer usually means the device
username isn't matching the sip.conf username=.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
The two things I would try are changing type from friend to peer and
sendrpid from no to yes. The no matching peer usually means the device
username isn't matching the sip.conf username=.
I have tried both friend and peer. I changed the sendrpid to yes
and made no difference either. Still get
This animal might be like the OBI110 box where you set it up in users.conf
instead of sip.conf.
Something like this:
[5001]
transfer=yes
call-limit=5
registersip=no
host = 1.2.3.4
context=default
hasvoicemail=no
dtmfmode=inband
threewaycalling=no
hasdirectory=no
callwaiting=no
This animal might be like the OBI110 box where you set it up in users.conf
instead of sip.conf.
Something like this:
[5001]
transfer=yes
call-limit=5
registersip=no
host = 1.2.3.4
context=default
hasvoicemail=no
dtmfmode=inband
threewaycalling=no
hasdirectory=no
callwaiting=no
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, December 13, 2012 3:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Digital accoustics trying to register to
On Thu, Dec 13, 2012 at 9:45 AM, Joshua Colp jc...@digium.com wrote:
If you don't want to incur the overhead of a full blown conference bridge
you can use ChanSpy to spy on a channel. It will provide a mixed stream of
the incoming and outgoing part of the channel. So essentially use Originate
Hi all,
I'm caught up in a struggle between people how can not make up their
mind... Half way implementing a asterisk farm and they come up with
another feature they've seen in kamaillo.
What he showed me was this: three registered sip users,
a) one changes his presence status on his softphone,
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