Hello,
I have asterisk 1.6.0.1. I'm using realtime asterisk with MS SQL.
Everything is OK, but I have noticed strange thing with queue members. If I
modify just 'membername' - asterisk do not refresh this info. But if I make
changes also in 'interface' column, and after executing command `queue
I think to reslove latency of communication, you should disable media server
option on asterisk, so that RTP packets are exchanged only between the Two SIP
clients.
De : Mark Michelson mmichel...@digium.com
À : Asterisk Users Mailing List - Non-Commercial
Hello,
we are using an EICON/DIALOGIC DIVA Server 4BRI together with asterisk
1.4.21.2 and chan-capi-head (20-11-2008)
From time to time error messages like the following appear several
times in /var/log/asterisk/messages:
ERROR[31152] chan_capi.c: Could not write to pipe for ISDN4#02
Is
Hello,
Is it possible, that during the call one side , for examples clicks the
button on the web, and this call starts recording? It's possible with
asterisk feature automon and DTMF. So it is possible to start recording the
channel using AMI or ... ?
Thanks
--
Pagarbiai / Best Regards,
yes it is posible
you will need to know the channel.
David
2008/12/16 Giedrius Augys voi...@gmail.com
Hello,
Is it possible, that during the call one side , for examples clicks the
button on the web, and this call starts recording? It's possible with
asterisk feature automon and DTMF.
Since when can you segment PRI channels off at the telco end? I know
you could do with DASS - but I'm not aware you can do it with PRI.
Andrew Thomas
Technical Services Manager
DataVox Ltd
Saddleworth Business Centre
Huddersfield Road
Delph, Oldham
OL3 5DF
On Tue, Dec 16, 2008 at 2:43 PM, Michael mich...@networkstuff.co.nz wrote:
Recently i also posted some rough configuration sample of my setup on
http://lists.digium.com/pipermail/asterisk-users/2008-November/222531.html
Please mind, that if you're trying T38modem, you should get versions
2008/12/16 David fire ddf...@gmail.com
yes it is posible
you will need to know the channel.
David
2008/12/16 Giedrius Augys voi...@gmail.com
Hello,
Is it possible, that during the call one side , for examples clicks the
button on the web, and this call starts recording? It's
Hello, I am running asterisk 1.4.22 and Iam recording calls in agents.conf
with the following configuration:
recordagentcalls=yes
recordformat=wav
createlink=yes
The calls are being recorded , but no entry appears in mysql cdr, and, on
the other hand I have other pbx running asterisk 1.2 that
I've worked with many providers who are able to do this. In fact, we're using
such a setup on our office PRI. I'm not sure how they're achieving this on
their end however...
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Andrew Thomas a...@datavox.co.uk wrote:
Simple. A PRI can easily have multiple trunk groups. They just assign
chan 1-22 to trunk group 1. Chan 23 to trunk group 2. D to chan 24. As
an example, adjust to suit your needs.
On Dec 16, 2008, at 9:27 AM, Andrew Thomas wrote:
I can only assume it's a T1 thing - as E1's tend not to have
I have seen bug http://bugs.digium.com/view.php?id=13525 - i think
it is
releated to it
I talked to file (who closed the above bug) and he thinks that it is
probably a different issue with similar symptoms. If you could go
ahead and post a bug on bugs.digium.com for it, that'll give us
I can only assume it's a T1 thing - as E1's tend not to have that
facility. Oh well, you live and learn :)
Andrew Thomas
Technical Services Manager
DataVox Ltd
Saddleworth Business Centre
Huddersfield Road
Delph, Oldham
OL3 5DF
--
Sebastian wrote:
Is this going to be released in any 1.6 version soon??
Your branch (queue-reset) is supouse to be the same as trunk but with this
functionality?
Is this branch updated every time trunk is committed?? I checked the log and
seems to have the latest commits of trunk, but I
2008/12/16 Giedrius Augys voi...@gmail.com
2008/12/16 David fire ddf...@gmail.com
yes it is posible
you will need to know the channel.
David
2008/12/16 Giedrius Augys voi...@gmail.com
Hello,
Is it possible, that during the call one side , for examples clicks the
button on the
In the USA (maybe other T-1 countries) you can have a channelized T-1.
Each channel is assigned signaling just like an analog line, FXS, FXO,
EM, etc. I have worked with a carrier in the past that could put FXO
channels on a T-1 along with a PRI channels on the same T-1.
Andrew Thomas wrote:
Am Montag, den 15.12.2008, 22:35 +0200 schrieb michel freiha:
Dear All,
I run the below tcp dump on my asterisk server
tcpdump -i eth0 -n -s0 -v udp port 5060
I got the following result
20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17,
length: 373)
Recently i also posted some rough configuration sample of my setup on
http://lists.digium.com/pipermail/asterisk-users/2008-November/222531.html
Please mind, that if you're trying T38modem, you should get versions
exactly as specified in voip-info.org, otherwise they might not work
with Opal
On Sun, Dec 14, 2008 at 1:01 PM, Michael mich...@networkstuff.co.nz wrote:
This path will not work. As You mentioned, * supports T38 path through
only. In Your setup there will be a conversion on the * box between T38
via SIP provider and IAX (which uses G711 codec in this case).
To make it
On Monday 15 December 2008 22:03:37 Chris Bagnall wrote:
Greetings list,
Over the last few days I've been gearing up to replace a couple of our
servers with 1.6 as something of a testbed, but I'm encountering a few
problems, and wondering if anyone can help...
In extensions.conf, there are
On Monday 15 December 2008 18:37:05 Barton Fisher wrote:
I don't see a method to detect the success or failure for the Record CMD.
I'd like to know the reason why the recording ended
Am I wrong?
exten = recordmsg,1,Noop()
exten = recordmsg,n,Record(${NEWPHRASEID}:ulaw|4|180)
So you'd be
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Tilghman Lesher
Sent: Tuesday, December 16, 2008 1:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 1.6
Hi all,
I am trying to isntall the v1.6 version of Asterisk on my Intrepid
system, but I get an error after I have typed make:
[CC] manager.c - manager.o
manager.c: In function ‘action_getvar’:
manager.c:1732: error: ‘SENTINEL’ undeclared (first use in this function)
manager.c:1732: error: (Each
On Tuesday 16 December 2008 13:14:06 Christian wrote:
Hi all,
I am trying to isntall the v1.6 version of Asterisk on my Intrepid
system, but I get an error after I have typed make:
[CC] manager.c - manager.o
manager.c: In function ‘action_getvar’:
manager.c:1732: error: ‘SENTINEL’ undeclared
On 20:14, Tue 16 Dec 08, Christian wrote:
Hi all,
I am trying to isntall the v1.6 version of Asterisk on my Intrepid
system, but I get an error after I have typed make:
[CC] manager.c - manager.o
manager.c: In function ‘action_getvar’:
manager.c:1732: error: ‘SENTINEL’ undeclared (first use
Christian wrote:
Hi all,
I am trying to isntall the v1.6 version of Asterisk on my Intrepid
system, but I get an error after I have typed make:
[CC] manager.c - manager.o
manager.c: In function ‘action_getvar’:
manager.c:1732: error: ‘SENTINEL’ undeclared (first use in this function)
There was a problem introduced in 1.6.0.2, as I recall, that is fixed in the
current release candidate for 1.6.0.3.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Christian christia...@runbox.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
- Original Message -
From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December 16, 2008 10:40 AM
Subject: Re: [asterisk-users] Record CMD
On Monday 15 December 2008
Hello all,
Many thanks for all your tips, will give that a try.
I downloaded it from downloads.digium.com/pub/asterisk and the
asterisk-1.6-current.tar.gz file.
Best regards and thanks,
Christian
On 2008-12-16 at 12:00 Jim Dickenson wrote:
There was a problem introduced in 1.6.0.2, as I
Hi list, I have for a year I have an account to call with broadvoice from
about 3 days beginning a not registered problem of, asterisk shows to a
message of error with the DNS, and my dns this working fine
WARNING[5770]: chan_sip.c:7595 transmit_register: Probably a DNS error for
registration to
On Tue, Dec 16, 2008 at 8:36 PM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
On Monday 15 December 2008 22:03:37 Chris Bagnall wrote:
Greetings list,
Over the last few days I've been gearing up to replace a couple of our
servers with 1.6 as something of a testbed, but I'm
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Atis Lezdins
Sent: Tuesday, December 16, 2008 5:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 1.6
On Tue, Dec 16, 2008 at 11:04 PM, troxlinux xserverli...@gmail.com wrote:
Hi list, I have for a year I have an account to call with broadvoice from
about 3 days beginning a not registered problem of, asterisk shows to a
message of error with the DNS, and my dns this working fine
I have a couple of numbers that are diverted to a number that is
conected to an isdn30 card, running asterisk 1.4.
eg.
123456 = 22334455
654321 = 22334455
What I would like to know is the number of the orginal number dialled
(123456 or 654321). I thought that RDNIS was the answer, but it is
http://www.theregister.co.uk/2008/12/16/infonetics_enterprise_telephony_numbers_q3_2008/
--
Drew Gibson
Systems Administrator
OANDA Corporation
www.oanda.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
I’ve got an interesting problem and am wondering if anyone can shed light …
I am running Asterisk on RHEL Server release 5.2 connecting to an Avaya
Definity G3R via a Digium TE220.
Asterisk 1.4.20
Zaptel 1.4.4
Libpri 1.4.4
MySQL 5.0.45
Festival Speech Synthesis System: 1.95
We have about 4200
my service was very well until I have not had behind, for some days I made
any change and my dns it works perfectly..
I check my account and my parameters in broadvoice and look that they
changed the out proxy for my account
if I change in the sip the parameter host = proxy-nyc.broadvoice.com
Hi Tillman,
I am havingthe same problem can you expand on your answer here? I am
not sure I understand what your saying. Are you saying that this is
really not an Asterisk problem? And just another thought. Where is
sentinel coming from? Interesting I wounder if it's something left over
from
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