success! just replaced MeetMe to Bridge in softkey.xml and conf works now with
the latest fw!
On Feb 26, 2015, at 9:00 AM, Nick Awesome jl...@me.com wrote:
I have not working 3way conference, when I trying to connect second call,
phone says “unable to set up conference”
and sending some
On Thursday 26 Feb 2015, ricky gutierrez wrote:
Hi A J , I have a sangoma gsm gateway 4channels , not use chan dahdi
O.K. So what does your existing Dial() statement in extensions.conf look
like?
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list,
Hi Nick,
maybe this will help?
exten = _XXX,n,Dial(SIP/${EXTEN})
exten = _XXX,n,NoOp(SIP return code :
${HASH(SIP_CAUSE,${CDR(dstchannel)})})
(http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause)
Markus
Am 27.02.2015 um 18:56 schrieb Nick Olsen:
Hello Everyone.
In my outbound
Hello Everyone.
In my outbound contexts, I'm using ${DIALSTATUS} to fail over to other
routes if the chosen route rejects the call.
Now, My current scenario is if I get BUSY back from the first provider,
I send a busy back to my customer. If I get something like CHANUNAVAIL
(Like a SIP
In Version 1.8 asterisk introduced this parameter preferred_codec_only, when
set to yes the 200 OK to the INVITE contains 1 codec only from the available
ones in the user sip profile.
But in version 13.1 (I think version 11.2 also) is not working like that , it
keeps sending all the codecs and
hello list,
i have created a queue with and i have a question related to musiconhold
f there is any way to set the musiconhold just for caller not for agent
logged in the queue
thanks and regards.
--
_
-- Bandwidth and
2015-02-27 10:25 GMT-06:00 A J Stiles asterisk_l...@earthshod.co.uk:
O.K. So what does your existing Dial() statement in extensions.conf look
like?
apology, put the gateway was sangoma but is a openvox ,
all my outgoing calls out for this context:
[my-mobile-out]
exten =
