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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)*>* exten => _XXXX,n,noop(returned from meetme) ;After user hangs up should*>* come here*>* exten => _XXXX,n,SoftHangup(${ORIG_CALLER})*>* exten => _XXXX,n,SoftHangup(${CONF_CALLER})*>* exten => _XXXX,n,Hangup*>* exten => h,1,noop(default-end)*>* exten => h,n,SoftHangup(${ORIG_CALLER})*>* exten => h,n,SoftHangup(${CONF_CALLER})*>* exten => h,n,Hangup*
That’s not how Asterisk works. When the caller hangs up, execution of
the current dialplan extension stops, and control passes to the ‘h’
extension, if one exists in the current context.

Any processing you want to do when the caller hangs up must be done
in the ‘h’ extension. Cheers

Thanks Tony for the quick response. As you would see I have the h
extension defined but execution doesn’t go to that either.

Tony

custom automated meeting

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?

Calls from PSTN on SPA3102

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
include => entradas_pstn
include => sips

[saliente_pstn]
exten => _9ZXXXXXXX,1,Dial(SIP/${EXTEN}@pstn,60,)
exten => _9ZXXXXXXX,n,Hangup

[entradas_pstn]
exten => s,1,Dial(SIP/103,20,tm)
exten => s,2,VoiceMail(103)
exten => s,3,Hangup

[sips]
exten => 100,1,Dial(SIP/100,20,Ttm) ; extensión 100
exten => 100,2,Voicemail(100)
exten => 100,3,Hangup
exten => 101,1,Dial(SIP/101,20,Ttm) ; extensión 101
exten => 101,2,Voicemail(101)
exten => 101,3,Hangup
exten => 102,1,Dial(SIP/102,20,Ttm) ; extensión 102
exten => 102,2,Voicemail(102)
exten => 102,3,Hangup
exten => 103,1,Dial(SIP/103,20,Ttm) ; extensión 103
exten => 103,2,Voicemail(103)
exten => 103,3,Hangup

When I receive a call from outside this is the asterisk console log:

== Using SIP RTP CoS mark 5

Nat Phone in Asterisk 10

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 and iax2 errors

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

Starting asterisk turns bash console text white in rxvt