Thank you all for your suggestions.
1. [macro-age] is a macro and not an extension badly named.
2. I am able to use Read to fulfill the purpose but we can't use Read()
after Background(). To use read we need Playback() [ am I right?]. But
Playback do not provide barge-in facility i.e. user have
We have a plain vanilla installation of AsteriskNOW using Digium D40/50 phones.
All transfers are failing from any source to any extension with the message
"that is not a valid extension". Does anyone have any ideas about where to
begin looking for the source of that error?
Phil Ledon
--
On Sun, 7 Sep 2014, Steve Edwards wrote:
In specific, your ordering of '_xx' in the middle of 's' is odd. This would
disrupt the value of the priority in older versions of Asterisk, but it
appears that it does work in modern (I'm using 11) versions.
Disregard that. I can't even follow my own
Please don't top-post.
On Sun, Sep 7, 2014 at 1:41 PM, Anurag Rana wrote:
I created a dummy dialplan where I ask the user to enter the age.
[macro-age]
exten => s,1,Background(my/age) ;;Play recorded message to enter age
exten => s,n,WaitExten(10)
exten => _XX,1,Set(
Hi,
upto asterisk 1.8 you used to get this error if there were more than 1
m= line in an invite... Asterisk was just telling you it was declining
the second. I belive from 10.0 onwards asterisk now just replies back
with port 0 to the stream it isn't interested in...
You can ignore it - if its bo
The first issue I see is you are attempting to insert your pattern match in
the middle of your 's' extension, That's going to break your 's' extension.
The second issue is that you are matching on XX which will match two
digits, You need to match on _X instead if you are attempting to match on
the
Hi,
I created a dummy dialplan where I ask the user to enter the age.
[macro-age]
exten => s,1,Background(my/age) ;;Play recorded message to enter age
exten => s,n,WaitExten(10)
exten => _XX,1,Set(AGE=${EXTEN});; this line is not executing, instead
dialplan is terminating with error giv
I had some confusion here. The endpoint needs a transport in order to
carry calls out. But the transports are also used by the application
PJSIP at large, in order to listen for incoming connections. In order
to just receive calls, I think you only need a transport, but no need
to assign that trans
I am having the issue described in this question:
http://lists.digium.com/pipermail/asterisk-users/2005-May/099075.html
Does anybody has an insight? I guess Asterisk is trying to match the
combination IP:Port, but in H223 this changes call by call. There is
no way to add "insecure=port" like in ch
CDR wrote:
I have a multihomed machine. How can I assign multiple IPs to and
endpoint, not all of them, just two, for instance, out of many?
Suppose the machine as 30 IPs, but my asterisk needs listen on two,
and one single endpoint needs to be associated with those two IPs. I
tried to add a seco
I have a multihomed machine. How can I assign multiple IPs to and
endpoint, not all of them, just two, for instance, out of many?
Suppose the machine as 30 IPs, but my asterisk needs listen on two,
and one single endpoint needs to be associated with those two IPs. I
tried to add a second "bind" lin
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