Hi everyone,
I'm having an odd problem with one way RTP on SIP to SIP calls.
I have two SIP servers, one is an Asterisk and the remote SIP server
is a Nortel SIP server.
When a call comes to the Nortel server through the PSTN and is routed
to the Asterisk, audio is fine. Two way RTP and no
=ANSI88, don't
remember seeing in plain SIP calls, so that is why I suspect is
configured as a SIP-T.
Örn Arnarson wrote:
Hi everyone,
I'm having an odd problem with one way RTP on SIP to SIP calls.
I have two SIP servers, one is an Asterisk and the remote SIP server
is a Nortel SIP
Julio,
It seems you had something going there; I disallowed ISUP messages on
the SIP-T server and now I have two way audio.
Thanks a lot for your help!
Best regards,
Örn
On 10/1/07, Örn Arnarson [EMAIL PROTECTED] wrote:
You are right, the remote server is a SIP-T.
I haven't had any problems
there.. :-)
Örn Arnarson wrote:
Julio,
It seems you had something going there; I disallowed ISUP messages on
the SIP-T server and now I have two way audio.
Thanks a lot for your help!
Best regards,
Örn
On 10/1/07, Örn Arnarson [EMAIL PROTECTED] wrote:
You are right, the remote
Sorry for the spam, but there was a typo. I was running ISN09, but the
upgrade was to ISN09u, which I am currently running. That was the
upgrade that caused the interoperability problem with Asterisk that I
mentioned.
On 10/1/07, Örn Arnarson [EMAIL PROTECTED] wrote:
Good point. Here goes.
I
We have a CS2K and don't have problems with Asterisk communicating at all.
The NGSS/SSTK on the CS2K doesn't have (as far as I know) support for
user and password, but IP authentication is working fine. I don't know
about the CS15K.
We did run into some issues with getting phone calls to work
This isn't the official manual, but it's quite good. Helped me out when I
started anyway.
http://dumbme.voipeye.com.au/trixbox/trixbox_without_tears.pdf
Best regards,
Örn
On Sun, May 4, 2008 at 9:07 AM, Sam Tam [EMAIL PROTECTED] wrote:
I have been trying to source a trixbox ce manual in pdf
There's not any direct way of which I am aware in a single command, but from
the shell you could do the following (and yes, this is a bit of a hack):
for i in `rasterisk -x queue show |grep wait |awk -F '{print $2}'`; do
rasterisk -x core show channel $i | grep Caller ID;done
That will return
Hello,
When I bridge an incoming and outgoing call (attempting to simulate
call-forwarding) I'm only getting one CDR -- that of the outgoing call.
A (PSTN) calls B (residing on Asterisk) and the Asterisk calls C (cell phone
on PSTN) and bridges the call.
The only CDR created is from B to C. I
Hi,
Not sure where to submit this to so I'll try here. Below is the toneset for
Iceland. Hopefully this can be added into the asterisk package.
[is]
description = Iceland
ringcadence = 1000,4000
dial = 425
busy = 425/250,0/250
ring = 425/1000,0/5000
congestion = 425+250/250,0/250
callwaiting =
Hi,
I'm having problem with call parking.
When I park call, either via transfer to xten or park digit sequence from
features.conf, I hear the parking lot number read to me and the user gets
transferred.
However, MOH stops for the caller the moment user is transferred.
The user can be retrieved
all. I am starting to think that this must be an asterisk bug...
version is 1.6.0.1.
Regards,
Örn
On Thu, Feb 12, 2009 at 3:05 PM, Örn Arnarson o...@arnarson.net wrote:
Hi,
I'm having problem with call parking.
When I park call, either via transfer to xten or park digit sequence from
I am seeing this problem on 1.6.0.1 when dialing a busy DAHDI channel...
On Fri, Feb 13, 2009 at 8:40 AM, Rilawich Ango maillist...@gmail.comwrote:
I also experience that problem. Is it a bug?
On Wed, Feb 4, 2009 at 5:53 AM, Mark Michelson mmichel...@digium.com
wrote:
Remco Barendse
could answer or explain.
Best regards,
Örn Arnarson
On Nov 21, 2007 2:15 PM, Kyriakos [EMAIL PROTECTED] wrote:
Guys can someone answer how the ACD works when it needs to decide which
call
to take next from queues with equal weights? Does it take the call with
the
longest period of watiting
Hi,
I've noticed that the UNIQUEID for a call is not the same in the
Dialplan (when executed e.g. exten = s,n,NoOp(${UNIQUEID}) as it is
when passed via STDIN to an AGI script.
Is this normal, and is this supposed to behave this way?
The UNIQUEID received in the AGI is usually .001 higher than
] On Behalf Of Örn Arnarson
Sent: Wednesday, September 09, 2009 7:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] UNIQUEID not the same in Dialplan as passed to AGI
Hi,
I've noticed that the UNIQUEID for a call is not the same in the
Dialplan (when
, does anyone have a suggestion as to how I
can find out the UNIQUEID of the new leg? (Asterisk-Endpoint) in the
middle of the call? Should I be able to find it somehow through the
other call-leg, via the channel id or something?
Best regards,
Örn
2009/9/9 Örn Arnarson o...@arnarson.net:
Thanks
instead of directly
executing the AGI command.
This may or may not work, but it should IMO.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn Arnarson
Sent: Wednesday, September 09, 2009 8:34 AM
...@lists.digium.com] On Behalf Of Örn Arnarson
Sent: Wednesday, September 09, 2009 8:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] UNIQUEID not the same in Dialplan as
passedtoAGI
If only features.conf accepted the normal syntax of running
applications
That selects which channel is active for the call. I should have
realized this earlier.
Thanks again for your help.
Örn
2009/9/9 Örn Arnarson o...@arnarson.net:
Hi Danny,
Thanks. Yes, that's where I'm getting the UNIQUEID. The problem is
that it is not for the same leg as the UNIQUEID in the Dialplan
-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn Arnarson
Sent: Wednesday, September 09, 2009 9:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] UNIQUEID not the same in Dialplan
...@lists.digium.com] On Behalf Of Örn Arnarson
Sent: Wednesday, September 09, 2009 9:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] UNIQUEID not the same in Dialplan
aspassedtoAGI
Yes, exactly.
I'm curious as to what would happen if I were
I'm seeing the same behavior in 1.6.1.6.
Any info on this?
On Wed, Apr 29, 2009 at 12:49 PM, Oguzhan Kayhan
oguzh...@bilkent.edu.tr wrote:
Hello,
I just tried to upgrade to 1.6.1.0 from 1.6.0.9 and i had problems in
registering users.
As i see from debug it successfully reads from users.conf
/users). A
downgrade from 1.6.1.6 to 1.6.0.9 promptly fixed it, as with Oguzhan.
Regards,
Örn
2009/9/18 Benny Amorsen benny+use...@amorsen.dk:
Örn Arnarson o...@arnarson.net writes:
I'm seeing the same behavior in 1.6.1.6.
Any info on this?
It would be helpful if you copied the exact error
as peers (using sip show peers/users). A
downgrade from 1.6.1.6 to 1.6.0.9 promptly fixed it, as with Oguzhan.
Regards,
Örn
2009/9/18 Benny Amorsen benny+use...@amorsen.dk:
Örn Arnarson o...@arnarson.net writes:
I'm seeing the same behavior in 1.6.1.6.
Any info on this?
It would
, and then the No such host
error.
Any ideas whatsoever?
Best regards,
Örn Arnarson
___
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing
Hello.
I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On Hold
functionality has changed (or is bugged?).
I have Aastra 6757i and Aastra 6731i phones, and now when i press the
MusicOnHold button / change lines on the phone, MOH no longer starts. It did
this in v 1.6.0.9.
Hello again,
I just tried version 1.6.1.9, and the MOH works well there. It seems to be a
bug introduced in 1.6.1.10.
Best regards,
Örn
2009/11/23 Örn Arnarson o...@arnarson.net
Hello.
I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On
Hold functionality has changed
Brilliant, thanks a lot.
Best regards,
Örn
On Tue, Nov 24, 2009 at 1:39 PM, Santiago Gimeno
santiago.gim...@gmail.comwrote:
Hi,
I think it can be related to https://issues.asterisk.org/view.php?id=16268
Best regards,
Santi
2009/11/24 Örn Arnarson o...@arnarson.net
Hello again,
I
Hello,
What are the current methods for playing digits on different languages? I
presume the big ones like German have been dealt with, saying 2 and 20 to
announce 22. How is this currently decided? What about languages that say 20
and 2?
Is there a way of configuring via config files or
;
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Örn Arnarson
*Sent:* Thursday, January 14, 2010 9:33 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Languages
is sent).
Can anyone think of a way to get the call back to the transferrer after this
timeout?
Best regards,
Örn Arnarson
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
Thanks a lot guys. Exactly what I needed.
Best regards,
Örn
On Tue, Jan 26, 2010 at 8:48 PM, Olle E. Johansson o...@edvina.net wrote:
26 jan 2010 kl. 16.48 skrev Örn Arnarson:
Hi guys,
I am wondering (and have been unable to find out thus far) whether
Asterisk sets some special
at what the output looks like, but
it would be nice if someone could point me to a document going through
the changes so I don't have to re-invent the wheel.
Anyone have any info on either one?
Best regards,
Örn Arnarson
Hello again,
Here's the header as it appears in 1.6.2.11 CLI output:
INVITE sip:1...@192.168.10.169:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.10.3:5060;branch=z9hG4bK73713002;rport
Max-Forwards: 70
From: SIP ehf/Örn Arnarson sip:7712...@192.168.10.3;tag=as2813a8fe
To: sip:1
Hi,
I'm seeing weird behavior with AMI where no events are output until
some input is detected (can be an empty line), at which time all the
buffered output is spewed out at once.
I am maintaining multiple Asterisk installations, and with one
installation I have run into a weird buffering
No, because the same is happening with telnet. If I telnet to AMI, I
observe exactly the same behavior. Otherwise I would put it down to
PHP.
On Tue, May 31, 2011 at 5:41 PM, Alex Balashov
abalas...@evaristesys.com wrote:
On 05/31/2011 01:38 PM, Örn Arnarson wrote:
Hi,
I'm seeing weird
: Expired
Event: ExtensionStatus
Privilege: call,all
Exten: 2170
Context: default
Hint: SIP/2170
Status: 4
Response: Error
Message: Missing action in request
On Tue, May 31, 2011 at 10:56 PM, Matt Riddell li...@venturevoip.com wrote:
On 1/06/11 5:38 AM, Örn Arnarson wrote:
The problem presents
. The problem only presents itself when
telnetting to this particular AMI.
Regards,
Örn
On Wed, Jun 1, 2011 at 11:02 AM, Alex Balashov
abalas...@evaristesys.com wrote:
Are you using the same telnet client in both cases?
On 06/01/2011 06:54 AM, Örn Arnarson wrote:
No, because the same is happening
the external IP address rather than
127.0.0.1 I observe the same results.
--
Örn
On Thu, Jun 2, 2011 at 3:19 AM, Matt Riddell li...@venturevoip.com wrote:
On 1/06/11 11:03 PM, Örn Arnarson wrote:
Hi Matt,
Yes, passing two carriage returns. I login successfully. Here's
example output (with my
en_US.UTF-8 in all cases.
On Thu, Jun 2, 2011 at 3:33 PM, Mark Deneen mden...@gmail.com wrote:
2011/6/2 Örn Arnarson o...@arnarson.net:
To clarify; I observe the exact same results no matter how I connect
to the AMI on this particular server. I tried connecting FROM this
server to an AMI
Just wanted to say that the issue fixed itself with a linux kernel
upgrade, in case anyone encounters this in the future and finds this
thread. Did not have to recompile Asterisk to get it working on the
new kernel.
Regards,
Örn
2011/6/2 Örn Arnarson o...@arnarson.net:
en_US.UTF-8 in all cases
Hi Brian,
Did you ever figure out what's causing this, and how to deal with it?
I'm seeing the same behavior with call-pickups (it's rare, but it's
happened a few times) on Asterisk 1.6.1.11
Did you figure out a way to get rid of the channel without restarting?
Regards,
Örn
On Wed, Jul 28,
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