Displaying 20 results from an estimated 600 matches similar to: "iax2 trunks between asterisk servers"
2006 Jun 08
0
Problems with IAX
Hi,
Here's my setup:
(PSTN)--[ASTERISK1]--(IAX)--[ASTERISK2]--(IAX)--[ASTERISK3]
I don't run asterisk 1, but I do run asterisk 2 and asterisk 3. I
have a DID via PSTN on asterisk 1 that is directed at asterisk 2 via
IAX. On asterisk 2 I want to direct that DID at asterisk 3. I have
done so, I have the IAX stuff setup between asterisk 2 and asterisk
3, but when I try to call
2007 May 30
2
(no subject)
Need some help with IAX trunking.
I've got six systems:
AsteriskM (main)
___________________|____________________
| | | | |
Asterisk1 Asterisk2 Asterisk3 Asterisk4 Asterisk5
AsteriskM has two Sangoma A102 2 Port T1 cards in it, the other Asterisk
boxes are using ztdummy for timing, they are all using IAX trunking.
My calls come in
2004 Jul 21
0
extensions.conf variable declaration
Hi,
I'm setting up multiple asterisk servers and trying to do the classic
DIAL(IAX2/asterisk1/${EXTEN}&IAX2/asterisk2/${EXTEN}&IAX2/asterisk3/${EXTEN},15)
After googling a bit, I fell on a discussion about putting this in a
variable so that adding additionnal servers would be easy. I can't seem to
find the link anymore, but it went something like this:
extensions.conf:
2006 Jun 16
5
asterisk load balance
Hi,
I am designing a asterisk load balancing model as follow. There are
3 asterisks connected to a single DB and a single server storing all
the configuration file and voicemail. Round Robin DNS will distribute
the request to asterisks.
DNS round robin ---+ asterisk1--------------------------+ DB and file server
+---asterisk2-----------------------+
2007 May 16
0
Passing dialstatus back through an IAX chain ..
I feel I'm doing something obviously wrong here and will kick myself when I see
the answer!!!
The scenario:
SIP phone -> Asterisk1 -> IAX -> Asterisk2 -> IAX -> Asterisk3 -> PSTN
So I place a call from the SIP phone. A1 picks it up and forwards it to A2
which forwards it to A3. A3 sends the call to the PSTN. I control A1 and A2,
but not A3.
When a call fails (for
2014 Nov 09
0
taskprocessor fails to allocate memory
I keep getting this error
[Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:614
__allocate_taskprocessor: Unable to start taskprocessor listener for
taskprocessor bbe08c34-9d1c-4e5f-8ae0-0cc75289caca
[Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:245
default_listener_shutdown: pthread_join(): Cannot allocate memory
[Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:614
__allocate_taskprocessor: Unable to
2006 Jun 20
8
fail to make call
Hi
I have the following configuration
|
UA1 --|------ asterisk1 -----------------------+
UA2 --|------ asterisk2 -----------------------+ DB
UA3 --|------ asterisk3 -----------------------+
UA4 --|------ asterisk4 -----------------------+
|
All UA is located in the same area. A seperated PC is used as a
centralized DB for storing a common dial plan, user account and
register
2005 May 13
0
Problem with IAX trunking
Hi all,
I'm trying to get IAX2 trunking between two * boxes and am having
extreme difficulty :) What happens is when the sending * server (the one
initiating the call) receives the ACCEPT back from the receiving server
it immediately replies with INVAL. I've checked the code and it seems to
be not matching the accept packet with the relevant item in the iaxs
array due to the following
2010 Aug 07
0
shrinkcallerid
Am I really the only one having problems with this new "shrinkcallerid"? I
can't find anything on Google about it.
Was happening on 1.6.2.10 and now on 1.8.0-beta2
In sip.conf shrinkcallerid=no, yet a name like "Joe Smith" ends up being
"JoeSmith"
Whoever though this up anyway is stupid. Why would you want to strip spaces
out of a caller ID?
Is there a fix?
2004 Aug 05
0
problems with asterisk and the IAX protocol
Hello group,
I wanted to try out the asterisk iax protocol between two asterisk
machines but have several problems with it.
My scenario looks like follows. I am using asterisk 0.9.0 on both machines.
SER1 <-> asterisk1 <-> IAX <-> asterisk2 <-> SER2
Both SER and asterisk run on a machine with a public IP address. When
the telephone on one side makes a call the telephone
2004 Aug 09
0
FW: problems with asterisk and the IAX protocol
Hi Kevin,
no you didn't miss the reply and I've not resolved it yet.
Have you got similar problems?
Pamela
Kevin Fjelsted wrote:
>Pamela,
>Did you resolve the problems you described?
>I didn't see a reply on the list but I may have missed it.
>
>-Kevin
>
>-----Original Message-----
>From: Pamela Weis [mailto:peawy@gmx.at]
>Sent: Thursday, August 05, 2004
2015 Jan 05
0
Asterisk removes a charachter from sip peer name
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Olli Heiskanen
Sent: 03 January 2015 08:04
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk removes a charachter from sip peer name
Hello all,
Just wondering on a behavior I noticed while testing with realtime sip peers
with names
2011 Apr 20
1
[IAX] Everyone is busy/congested at this time (1:0/0/1)
Hi,
I have a problem with IAX accounts...
I set up a few months ago an Asterisk server, with mysql support to load iax
accounts.
Settings seems fine because apparently the system works as expected.
Yesterday I tried to add another iax account in the iax.conf directly. And I
have a problem with this new account (named 444).
I can authenticate from 444 to the server, and I can receive calls from
2012 May 29
2
Fax Server for Asterisk
Hello,
For those customers with only analog lines, who ask for fax2email and
email2fax, whats the most reliable solution available and tested with
Asterisk?
Thanks
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2011 Jul 25
0
Registration problems, Linksys SPA 3102 on Asterisk 1.4.20
Sorry, I am resending this, I tried earlier, but I
couldn't see it appear on the archives -
apologogies if it appears double!
--------------------------------------------------
My Sipura 3000 ATA died on me this morning. I had
a Linksys SPA 3102 available which I would like to
use as a replacement. Unfortunately, the SPA3102
is not able to register with the asterisk server -
I am
2008 Dec 18
1
Ghost in the Channel-Banks
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
I've been struggling with an ongoing problem the last month.
Here is the layout of the wiring:
T1 from ISP > DiTech Echo Cancel device > Voice Channel-Bank
(same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1 server
zap card > fax channel bank
(same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1
2013 Mar 29
0
Getting Unknown Error while configuring Asterisk with Linux HA
Hi,
I recently configured Linux HA for Asterisk service (using Asterisk
resource agent downloaded from link:
https://github.com/ClusterLabs/resource-agents/blob/master/heartbeat/asterisk
).
As per configuration it is working good but when I include "monitor_sipuri="
sip:42 at 10.3.152.103" " parameter in primitive section it is giving me an
errors like listed below;
root at
2008 Dec 03
0
chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE to
Hello,
I need help for that error message:
?chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE
to?
My network is:
Client1--
-----------asterisk1------asterisk2
Client2--
? With client1, I do a call
? Asterisk1 forward the call to asterisk2
? Asterisk2 forward the call to asterisk1
? Asterisk1 forward the call to
2008 Dec 03
0
problem with RTP
Hello,
My network is:
Client_SS7_1--
-----------asterisk1------asterisk2
Client_SS7_2--
? I receive a fax from Client_SS7_1
? Asterisk1 forward the call to asterisk2
? Asterisk2 forward the call to asterisk1
? Then, asterisk2 forward the fax to Client_SS7_2
I want that the SIP signaling go to asterisk2,
But, I need that the RTP don?t go
2005 Oct 06
0
Issue with trunking
Hi all.
Ive recently setup two Asterisk boxes (running Asterisk@Home to be specific), and Im trying to get a trunk going between them.
So far I have tried a combination of IAX and SIP configuring them through AMP and writing the config files manually, but I cant seem to get calls going between the two.
I have named each box asterisk1 and asterisk2.
Does anyone have some working SIP and/or IAX