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Karim Mardhani http://lists.digium.com/mailman/listinfo/asterisk-users> wrote:
>* Hi everyone,*>* *>* I am trying to get Meetme to return back to the context from where it*>* joined the meetme. For example a user uses the following context to join a*>* conference, once user hangs up I would like to continue executing the rest*>* of the dialplan. But when caller hangs up from the conference I see on CLI*>* that meetme exited with non-zero status but none of the rest of the*>* dialplan is executed. Please help. I am using asterisk 1.6.2.20*>* *>* [default]*>* exten => _XXXX,1,MeetMe(1000,1pdMX)*>*…

Asterisk Users 3.9 years ago 0 Answers

custom automated meeting

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I need your help in implementing the following scenario: A certain extension will ring two sip phones simultaneously and when one
of them answers, the other keeps ringing until it answers too, and then
all three (the caller and the other two) are immediately placed in a
conference room (same room for all three). Can we do it?

Asterisk Users 3.9 years ago 10 Answers

Calls from PSTN on SPA3102

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Hello list, this is my first post on this list. I have a server with Asterisk and a Linksys SPA3102 with 3 SIP phones.
I have configured the SPA PSTN line as trunk to receive and send
calls. I can call outside from SIP phone throw the PSTN line and all is OK,
the problem is when I receive a call from the PSTN, on the out caller
phone there is a demo playback. I want to redirect the call to a
internal SIP phone. This is the extensions.conf: [spa]
include => saliente_pstn

Asterisk Users 3.9 years ago 2 Answers

Nat Phone in Asterisk 10

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Hello listers, Another opportunity presents itself in my 1.4 to
10.0 conversion. My asterisk is set up for 192.168.23.xx and most of my
phones are 192.168.23.yy peers. I work on two subnets so I have one phone
defined as 192.168.33.xx. This phone comes up and registers and accepts
calls and calls out in 1.4.41 but shows unreachable in 10.0. What changed
that is killing my "off-network" phone? Thanks in Advance Danny Nicholas

Asterisk Users 3.9 years ago 4 Answers

asterisk and iax2 errors

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Hi
I am having intermittent iax2 errors on 2 asterisk systems connected with iax2. Below are the errors, and everything I have read online says this was a bug that has been fixed.
Both sides are set for trunking with dahdi complete installed. No dahdi hardware just dahdi dummy for timing. Any help would be greatly appreciated. chan_iax2.c: Received trunked frame before first full voice frame server 1
centos 5.6
asterisk 1.8.4.3
dahdi-2.5.0 server 2
centos 5.6
asterisk-1.4.42
dahdi-2.5.0.1 Thanks Kelly

Asterisk Users 3.9 years ago 0 Answers

Temporarily disabling voicemail recordings (but not greetings)

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If you are using the "silent" option of voicemail (b - busy, u -
unavailable, s - silent) you could set up a context to play the "normal
silent" message, then goodbye. [normal-voicemail] Exten => start,1,playback(unavail-msg) Exten => start,n,voicemail(${ARG1}@default) Exten => start,n,playback(vm-goodbye) Exten => start,n,hangup() [full-voicemail] Exten => start,1,playback(unavail-msg) Exten => start,n,playback(vm-goodbye) Exten => start,n,hangup() From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Olivier
Sent: Monday, October 31, 2011 5:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Temporarily disabling voicemail recordings (but
not greetings) Hi, Googling, I couldn't find any…

Asterisk Users 3.9 years ago 2 Answers

sip issue

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hello list i have installed asterisk 1.8.7.1 and i have configured 2 account for sip
in order to do internal call when i use x-lite and eyebeam1.5 i can call from 222 to 223 ,and alson from
223 to 222 but when i use my snom 320 i can call from my x-lite or eyebeam1.5 to
snom320 but the issue i can not call from my snom i have this issue just Asterisk 1.8 when i tested with asterisk 1.4 theres
is no problem see the sip.conf and extenssions.conf below and also the cli when i…

Asterisk Users 3.9 years ago 2 Answers

Meetme does not return back to the dialplan

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Hi everyone, I am trying to get Meetme to return back to the context from where it
joined the meetme. For example a user uses the following context to join a
conference, once user hangs up I would like to continue executing the rest
of the dialplan. But when caller hangs up from the conference I see on CLI
that meetme exited with non-zero status but none of the rest of the
dialplan is executed. Please help. I am using asterisk 1.6.2.20 [default]
exten => _XXXX,1,MeetMe(1000,1pdMX)
exten => _XXXX,n,noop(returned from meetme) ;After user…

Asterisk Users 3.9 years ago 0 Answers

raju@linux-delhi.org

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On Sunday 30 Oct 2011, bilal ghayyad wrote:
> Actually I need to do a dash board for reporting, so I beleive the
> only way is to use the AGI, correct? But where I can find documents
> or link that can help me to do this?
>
> About ur sentence:
>
> "some ready-made packages (both FOSS and proprietary) that will
> display this information nicely formatted".
>
> What is the FOSS and proprietary? Any link for it?
> And this ready-made packages can work…

Asterisk Users 3.9 years ago 0 Answers