Displaying 20 results from an estimated 2000 matches similar to: "Make the call finish after executing Dial(G())"
2010 Jul 12
0
ResetCDR not working after forced hangup
Hello, Asterisk party,
If block the call before dialing (Hangup()), CDR's don't write to
MySQL or CSV. Otherwise, if Hangup() is after Dial(), CDRs write
normally.
Here is the dialplan:
; we skipped dial, because the number is "blocked"
exten => _X.,n(Finish),Hangup()
exten => h,1,NoOP("hangup")
exten => h,2,ResetCDR(w)
exten => h,n,NoCDR()
exten =>
2010 May 05
1
Getting calee audio in Asterisk (real time)
Hello,
I need to capture calee's audio in real-time in order to capture operator
messages (I've written sound recognition software that works with Jack:
http://github.com/Motiejus/SoundPatty/).
Jack does the following:
Incoming call audio -> audio in to jack, audio out from jack ->
current Asterisk application
Outgoing call audio <- current Asterisk application
However, I need
2010 May 26
1
Jack in /usr/local/ means failure for asterisk
Hi,
I have been headbanging with asterisk and Jack for a while, decited to
ask other linuxists for an advice.
The problem is that Jack is compiled from source (0.118) in /usr/local/, but
menuselect says "XXX" for it (cannot enable it). I need jack...
Otherwise I will inotify Monitor WAVs, what is bad :`(
After installing jack from sources:
Add system-wide
2005 Mar 01
1
Connecting Asterisks via SIP
Hi.
It is propbably a really naive problem, but I really couldn't find
answer how to connect two Astrisks via SIP. I managed to do it via IAX
without any problem. But this is a test installation and I would like to
connect them via SIP.
So I have two computers:
pbx1 - 10.1.3.207
pbx2 - 10.1.3.204
pbx1 handles extensions 1xx and pbx2 for extensions 2xx. I would like to
call user from pbx2 to
2010 Apr 29
2
Code in extensions.conf to leave a voice mail in another PBX ?!
Hi Guys,
i spent some time to figure this out (since i love how dialplan is written)
but i decided to ask for your help guys.
i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
just hang up.
in pbx2 extensions.conf:
i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
in pbx1, i
2010 Mar 04
1
InterPBX communication using SIP
Hi Guys,
i am using the following config in pbx1:
register => pbx1:endopass at 172.16.200.175 <pbx1%3Aendopass at 172.16.200.175>
[pbx2]
type=friend
host=dynamic
trunk=yes
sercret=password
context=[default]
deny=0.0.0.0/0.0.0.0
permit=172.16.200.175/255.255.255.128
in pbx2:
register => pbx2:endopass at 172.16.200.176 <pbx2%3Aendopass at 172.16.200.176>
[pbx1]
type=friend
2006 Dec 28
0
Re: asterisk-users Digest, Vol 29, Issue 114
> Can someone tell me how Asterisk handles music-on-hold between servers?
> Documentation for this is non-existent.
>
> Lets say user A, who is registered on pbx1, calls user B, who is registered on pbx2.
>
> 1. User A puts user B on hold. The moh that is played to user B should be specified according to user A. Which pbx box should this be set on? pbx1? pbx2? Both?
>
> 2.
2007 May 13
2
TC400B load problem
Hi
Im trying to install my TC400B trans coder card when I do:
modprobe wctc4xxp
tail -f /var/log/messages says:
May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with
92 transcoders (srcs=0000000c, dsts=00000101)
May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Decoder' with
92 transcoders (srcs=00000101, dsts=0000000c)
May 13 14:56:36 pbx2
2010 Jan 28
3
Dial cellphone from one PBX1 to PBX2? is it possible?
Hi Guys,
i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way:
1) use a phone in PBX1
2) call extension in PBX2
3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a
cellphone)
my questions now is : am i gonna be able to dial from an IPphone registered
within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2?
anybody know
2009 Mar 16
1
Transfers on an inter-PBX PRI link
Hi,
I am trying to understand why some of my call transfers fail.
My scenario is as follows:
Legacy PBX1 ---PRI (EuroISDN) Zaptel--- Asterisk PBX2
Step1: PBX1 extension 101 calls PBX2 extension 102
Step2: PBX2 extension 102 answers the call and transfers it to PBX1 extension 103
Step3: PBX1 extension 103 answers the call and transfers it to PBX2 extension 104
Step3 fails and extension 103
2004 Jun 23
1
Iax unable to transfer
Dear List
I have notice this kind of problem between my two * box.
My scenario is :
Iax GSM
IaxClient----->PBX1------------>PBX2-->TDM
today CVS Stable V1
I use as Client FireFly with IAX2/GSM and try to call my PBX1 this server call
PBX2 to terminate the call trought a TDM line (TE410P) but after PBX2 join
the two call i can see the log below from my PBX1, i can speak for
2006 Dec 28
1
Music On Hold Between Servers
Can someone tell me how Asterisk handles music-on-hold between servers?
Documentation for this is non-existent.
Lets say user A, who is registered on pbx1, calls user B, who is registered on pbx2.
1. User A puts user B on hold. The moh that is played to user B should be specified according to user A. Which pbx box should this be set on? pbx1? pbx2? Both?
2. Is the situation any different if the
2006 May 23
0
Wacky Failover Situation w/SIP - Bug?
I have a weird situation. A polycom phone is configured to use system pbx1 as the primary outgoing 'proxy', followed by systems pbx3 and pbx2. All three systems have identical sip.conf files. The phone is registered on pbx1.
I shut down the Asterisk application on pbx1. I make a call. The phone sends an INVITE to pbx2. Pbx2 sends back a 407 Proxy Auth message to the phone. The phone sends
2006 May 25
4
Failover Problem
I have a weird situation. A polycom phone is configured to use system pbx1 as the primary outgoing 'proxy', followed by systems pbx3 and pbx2. All three systems have identical sip.conf files. The phone is registered on pbx1.
I shut down the Asterisk application on pbx1. I make a call. The phone sends an INVITE to pbx2. Pbx2 sends back a 407 Proxy Auth message to the phone. The phone sends
2016 Jun 30
2
how to join 2 channels using AGI/AMI
this is the point, and the strange thing:
DTMF is set to rfc2833, but is working both on incoming and outgoing calls,
it is not working only on calls generated with the Originate AMI command,
or with the queue member that point to Local dialplan, as you suggested
2016-06-30 22:53 GMT+02:00 John Kiniston <johnkiniston at gmail.com>:
> Looking at your logs it looks like you may need to
2016 Jun 30
2
how to join 2 channels using AGI/AMI
thanks John
yeah, your approach is much siple, i've tried it but i'm not able do detect
DTMF tones.
it seems that on calls that i receive DTMF tones are handled correctly, but
on calls generated from Asterisk to the world when the called side sends
some DTMF digits they are not detected:
-- Executing [s at macro-myconnector:1] NoOp("SIP/pbx2-000004b2", "") in
new
2013 Apr 21
1
Strange problem with Asterisk 1.8.9.3
Hello List.
Last month i started to face a strange issue on an asterisk server
1.8.9.3 built on Centos 5.3 x86_64 dedicated server.
out of the blue UDP stops responding .. and keep getting the following output:
---------------------------------- Opening message for the problem
--------------------------------------------------
[Mar 21 09:57:04] ERROR[6748] netsock2.c:
2010 Jan 26
2
[inter-pbx commnication] trying to make PBX1 talk to PBX2
Hi All,
i want to make an extension from pbx1 able to tlak to another extension from
pbx2 or use pbx2's trunk to dial outside calls.
so i edited in both servers accordinally the iax.conf:
register => pbx1:pass at 172.16.200.175 <pbx1%3Apass at 172.16.200.175>
[pbx2]
type=friend
host=dynamic
trunk=yes
sercret=pass
context=[default] ; i used the biggest context to avoid confusion as
2008 May 06
3
asterisk queue cluster
I setup two asterisk servers with identical settings
(same extensions, same queues, etc). Each one is
connected to the same amount of incoming/outgoing
links (1 PRI, 4 BRI, 1 IAX friend, etc, on each box).
Most extensions are sip and they register via DNS SRV
and other methods so that the two servers are load
balanced. Incoming PSTN calls (BRI) reach 50% each
server so that's load balanced
2003 May 13
4
app_transfer
I've added an important new application: app_transfer. This application
is designed to allow Asterisk to request the transfer of an incoming call
to a different extension. Consider the following diagram:
Caller -> [ PBX1 ] -> SIP or IAX2 -> [PBX2] -> Transfer App
A caller calls an extension on PBX1 which forwards to PBX2. PBX2 executes
app_transfer, which requests that hte