Displaying 20 results from an estimated 200 matches similar to: "InterPBX communication using SIP"
2010 Apr 14
1
Interpbx connection
Hi Guys,
i've connecting two pbx server successfully for several times using the
following config:
register => USPBX:mypass at 122.11.176.35 <USPBX%3Amypass at 122.11.176.35>
[PBX1]
type=friend
host=122.11.176.35
trunk=yes
sercret=mypass
context=external
deny=0.0.0.0/0.0.0.0
permit=122.11.176.35/255.255.255.240
insecure=very
allow=all
nat=yes
qualify=yes
canreinvite=no
in the other
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
2003 Aug 17
2
no incoming packets & Sound: Recording overrun
On Sun, Aug 17, 2003 at 03:44:21AM -0500, Gnophone Support wrote:
> Hello, and thank you for registering at gnophone.com. Your login
> information is listed below:
>
> Username: miernik
> Password: *******
> IAX Phone Number: 17002916107
>
> Please login as soon as possible to
> http://x.linux-support.net/directory/ to complete the
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 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
2006 Mar 24
14
IAX Incoming/Outgoing
I'vce got three Asterisk systems here that I'd like to be able to place calls between with IAX. As usual, I've spent several hours playing with it, really getting nowhere. Asterisk is so mentally draining. Each system, pbx1, pbx2, pbx3, should be able to connect to every other. Do I need separate user/peers or can I combine them into a single user=friend for each system? if I place a
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
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
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
2016 Sep 19
3
Asterisk 14.0.0-rc1 Now Available
Marcelo Terres wrote:
> I noticed another different behaviour.
>
> In older versions, when I call rasterisk, I receive some informations
> about it. Fox example:
>
> [root at pbx2 ~]# rasterisk
> Asterisk 11.22.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.
> Created by Mark Spencer<markster at digium.com>
> Asterisk comes with ABSOLUTELY NO WARRANTY; type
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
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
2006 Apr 28
2
Random 1-way audio on IAX2 Connections
I have 2 Asterisk servers connected via IAX2 connections.
PBX1 is on the internet with a public IP Address
- with PRI
PBX 2 is behind a NAT router with IAX2 Ports forwarded
1-way audio is an issue with incoming and outgoing calls using the PRI.
However whenever 1-way audio occurs, PBX2 can call PBX1 extensions and there
are no issues. As well as a restart of asterisk on PBX2
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
2016 Jun 30
2
problem with DTMF detection on calls created with Originate AMI command
Dear all
i'm creating an outgoing call to number xxx with this command:
http://host:port/mxml?action=Originate&Channel=Local/xxx at to-external
&Exten=testDTMF&Context=cRETEUNICA&Priority=1
wich points correctly to this portion of dialplan:
[cRETEUNICA]
exten => testDTMF,1,Answer
exten => testDTMF,n,Read(digito,,1)
exten => testDTMF,n,SayDigits(${digito})
The
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
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:
2016 Sep 19
2
Asterisk 14.0.0-rc1 Now Available
One more thing about my last email: I think that you forgot to update
the configs/samples/res_odbc.conf.sample file, because it still
contains idlecheck and limit parameters.
Regards,
Marcelo H. Terres <mhterres at gmail.com>
IM: mhterres at jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On Mon, Sep 19,
2016 Jun 30
3
how to join 2 channels using AGI/AMI
sorry for top-posting, the two topics started with 2 different reason
subject, but then we finished on the same problem.
btw,the 2 show channel are reported above:
the channel with DTMF working
kcenter*CLI> core show channel SIP/pbx2-000004b9
-- General --
Name: SIP/pbx2-000004b9
Type: SIP
UniqueID: 1467323106.1275
Caller ID: xxxx
Caller ID Name: xxxx
2016 Jun 30
4
how to join 2 channels using AGI/AMI
Dear all
i'm using an "old" Asterisk 1.6.2.9-2+squeeze12, and want to know if is
possible to configure a scenario like this:
1) receive a call and put it on-hold in a queue (OK)
2) monitor the queue and trigger an outbound call to a remote number using
AMI, setting the channel of the on-hold on a specific var named
channel2Link (OK)
3) when the remote number answer, trigger an