We're using CVS-HEAD-04/21/05-11:34:04 and seem to be having problems with
Remote Agents receiving calls. When someone's in the queue, they get
dialed, but the AckCall '#' isn't being accepted and the call is simply
stuck in the queue while the remote agents are constantly dialed until
someone
On 02/16/2011 09:43 PM, ERIC HERRON wrote:
On IP430s
cat sip.ver
VVX-1500 3.2.2.0481
All others 3.2.2.0477
2345-11402-001.bootrom.ld sip.ld
Phone1.cfg
msg msg.bypassInstantMessage=1
mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact
msg.mwi.1.callBack=*97 msg.mwi.2.subscribe=
Sip.cfg
On 02/17/2011 12:10 AM, Ryan Wagoner wrote:
[snip]
Backlight works fine on a IP550 with 3.3.1 . I have mine set to off
when idle. I like that the 3.3.x series doesn't required the default
sip.cfg and phone1.cfg files. The structure of the XML seems cleaner
and more consistent.
up
On 02/17/2011 12:17 AM, ERIC HERRON wrote:
I haven’t played with the backlights yet.
One annoyance at a time.
Agreed :)
To disabled the mwi chirp can be set to silence.
MESSAGE_WAITING se.pat.misc.1.name=message waiting
se.pat.misc.1.inst.1.type=silence ….
Thanks for the tip Eric. The
On 02/17/2011 12:17 AM, ERIC HERRON wrote:
[snip]
I am trying the different firmwares now to see if it makes any difference.
In the admin guide I just came across: up.oneTouchVoiceMail default 0
If set to 1, the voice mail summary display is bypassed and voice mail
is dialed directly (if
On 02/17/2011 03:20 AM, Ryan Wagoner wrote:
[snip]
whichsection it is under. My 3.2.x config file worked except for
alert info, ringer, and feature settings, which was outlined in
Simplified_Configuration_Improvements_in_UC_Software3_3_0_TB60519.pdf
Just went through that doc. Interesting
On 02/17/2011 12:17 AM, ERIC HERRON wrote:
To disabled the mwi chirp can be set to silence.
MESSAGE_WAITING se.pat.misc.1.name=message waiting
se.pat.misc.1.inst.1.type=silence ….
This did not work but looking at the example files in the 3.3.1 firmware
the snippet below did work (mind the
On 02/17/2011 02:28 PM, ERIC HERRON wrote:
I have ind.pattern.
In firmware version 3.3.1? I have only found a reference to ind.pattern
in the Simplify Configuration Improvements Guide which you mentioned
yesterday. Afaict there is no ind.pattern section in the 3.3.0 Admin
Guide and the
On 02/17/2011 02:26 PM, ERIC HERRON wrote:
Yeah it’s the same thing; I think.
I think we have different config files…are you using the split?
Unfortunately I have no idea what the split means. Can you please explain?
Regards,
Patrick
--
On 02/17/2011 02:23 PM, Ryan Wagoner wrote:
[snip]
The color screen must be different or it is a firmware bug. Was it any
different on 3.2.x vs 3.3.x? On my IP550 you can still read the screen
I have not tried firmware 3.2.x. I'll give that a try once I figure out
the old config system.
On 02/17/2011 06:06 PM, viswavardhanreddy karna wrote:
Hi all,
I have a few doubts regarding asterisk...
1. Is asterisk can be used as stateless and stateful proxy? If yes i
Afaik Asterisk is a B2BUA (google B2BUA if you don't know what it is)
and does not have proxy capabilities
On 03/14/2011 04:36 PM, Bruce B wrote:
Thanks for the input.
I can see the module loaded but yet the command core show channel
types doesn't show H323 in channel list. Maybe ooh323 is not supposed
to show in that list?
server55667*CLI module show like 323
Module
On 05/24/2011 09:56 AM, Kristijan Vrban wrote:
http://code.google.com/p/zaphfc/
Thanks! For the archives I found another site where the zaphfc code from
that google site is integrated into the DAHDI source:
http://sourceforge.net/projects/dahdi-zaphfc/
Regards,
Patrick
--
On 05/24/2011 10:50 AM, Olivier wrote:
2011/5/24 Kristijan Vrban vrban.l...@googlemail.com
mailto:vrban.l...@googlemail.com
http://code.google.com/p/zaphfc/
Hi,
Which hardware would be recommended to try this code ?
As far as I know you can use any ISDN card with a Cologne HFC-S chip
On 05/24/2011 11:02 AM, Steve Davies wrote:
[snip]
I use astmanproxy with Asterisk 1.6.2.18 - It works fine. The most
recent version is on Github, and is not that old. In fact that reminds
me that I really must upload my latest changes to my github fork of
the project!
The last update to Dave
On 05/24/2011 03:43 PM, Steve Davies wrote:
[snip]
Repo updated. I have tried to merge all of the other changes that are
out there also.
Thanks Steve.
Regards,
Patrick
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On 05/25/2011 01:20 AM, bilal ghayyad wrote:
Hi All;
To enable the compilation for the addon that is coming with Asterisk 1.8 when
doing compilation for the Asterisk, what should I do?
make menuselect and enable what you would like to be build.
Regards,
Patrick
--
On 05/27/2011 05:10 PM, Michelle Dupuis wrote:
I'm looking for recommendations for standalond PRI to SIP converters. (Needs
to be outside the asterisk box - so a PCIe card won't do)
I've used redfone but this project doesn't need the redundancy features...
Have a look at Patton or
Hi Rajib,
Comments inline.
On 05/30/2011 10:03 AM, Deka, Rajib IN MAA SL wrote:
Hello List,
What version of DAHDi should be installed for CentOS Kernel version
2.16.18–194.el5.
I would use the latest DAHDI version which is currently:
DAHDI tools: 2.4.1
DAHDI linux: 2.4.1.2
You can find
On 06/06/2011 08:07 PM, Andrew Joakimsen wrote:
Anyone have an update as to when Digium will ship a working package?
According to https://issues.asterisk.org/view.php?id=18748 new packages
should already have been pushed. If not perhaps you could join #asterisk
or #asterisk-dev on
On 06/22/2011 03:32 PM, Steve Davies wrote:
Does anybody know if the updated licence on iLBC makes it safe to
include in Asterisk when used in a commercial environment again?
https://sites.google.com/site/webrtc/ilbc-freeware
It seems to require that the Google iLBC licence document is on
On 07/08/2011 05:01 PM, Mark Rosedale wrote:
This is not working the source code of 1.8.4.4. I assume that the patch
is for a different version. Any ideas about how to apply this patch to
1.8.4.4 so that I can avoid using the svn branch?
It's a 2 line patch. If you look at the source it's
On 07/08/2011 08:46 PM, Michael L. Young wrote:
Patrick,
The patch was merged in to the 1.8 branch on 5/13/2011 as revision 318783
(http://svnview.digium.com/svn/asterisk?revision=318783view=revision).
1.8.5-rc1 was tagged on 06/29/2011
On 07/09/2011 01:28 AM, Doug Lytle wrote:
Can you say a Virtualized Asterisk with a PRI card!
http://www.phoronix.com/scan.php?page=news_itempx=OTY0OQ
With virtualized environments prone to timing issues does this make
sense at all?
Regards,
Patrick
--
On 07/09/2011 02:59 AM, Krishna Sumanth Chava wrote:
Hi Doug,
May be you can try a PCI-E card that has a PRI port and asterisk on
itself and eliminating the need to install asterisk.
www.positrontelecom.com http://www.positrontelecom.com have these cards.
Interesting, never seen those
On 07/10/2011 05:02 PM, Matiss Jekabsons wrote:
Is there some detailed documentation for 1.8.5? I am tryin to make
Asterisk 1.8.5 with MySQL backend, TLS transport and SRTP encryption.
For now with no success :-(
https://wiki.asterisk.org/wiki/display/AST/Asterisk+1.8+Documentation
Regards,
Hi Tony,
On 07/22/2011 09:44 PM, Tony Mountifield wrote:
I read Kevin's piece in asterisk-announce about the new numbering scheme,
and saw in svn-commits some tagging of 10.0.0-beta1.
Totally missed that one. Just did a quick browse.
Perhaps I'm thick (I hope not!), but I really can't see
On 07/23/2011 04:00 PM, Paul Belanger wrote:
A UAS rejecting an offer contained in an INVITE SHOULD return a 488
(Not Acceptable Here) response. Such a response SHOULD include a
Warning header field value explaining why the offer was rejected.
If the choice is to get hacked/DDOS'ed/etc or
On 07/26/2011 09:21 PM, Bryant Zimmerman wrote:
I want to add an entry to a database every time a brute force
registration attempt is done.
from this database we are updating cisco routers with our ban list so
our entire network is protected.
The database side of things is working and has been
On 07/31/2011 07:22 AM, Pezhman Lali wrote:
Dear,
with asterisk 1.6.2.18 and sccp-bv3stable on two servers, we tried to
register about 1200 cisco phones, for a company.
in out of official hours, all 1200 phones registered and the cpu and ram
was below 5%.
In my experience, registering a Cisco
On 08/10/2011 12:00 AM, Steve Totaro wrote:
Try iaxmodem and hylafax. Alex B did a very nice writeup on how to
set that up so that it works very well.
Could not agree more. I have been using a box with an Eicon Diva Server
card, asterisk 1.4, chan_capi, iaxmodem hylafax for many years and
On 08/14/2011 10:31 PM, Shaun Ruffell wrote:
[snip]
While I can't say I've run against that particular CENTOS kernel version, I
would be very surprised (and interested to know) if you had any problems.
Iirc that kernel (2.6.32-71.el6.i686) is a stock Red Hat Enterprise
Linux 6 (or CentOS 6)
On 08/15/2011 06:04 AM, Shaun Ruffell wrote:
On Mon, Aug 15, 2011 at 05:21:36AM +0200, Patrick Lists wrote:
On 08/14/2011 10:31 PM, Shaun Ruffell wrote:
[snip]
While I can't say I've run against that particular CENTOS kernel version,
I would be very surprised (and interested to know) if you
On 08/30/2011 06:32 PM, Tamer Higazi wrote:
Hi people!
I have managed to set up asterisk 1.8.5. with my 2 ISDN HFC boards on
asterisk. On which DAHDI tells me also properly that both of my boards
are registered, one in TE and the other on in NT mode.
Calls do successfully come inside, but I
On 08/30/2011 08:37 PM, Tamer Higazi wrote:
Hi Patrick!
Now i got it.
I am using Gentoo Linux, Asterisk 1.8.5 and Dahdi 2.4.1.
The patches are automatically integrated at Gentoo. I didn't have to
patch anything. That did the community.
Thanks for the info.
Another question, I really
On 09/03/2011 05:33 PM, Paul Belanger wrote:
This is a bug with netsock2.c unable to resolve a hostname or SIP peer
into an IP address.
https://issues.asterisk.org/jira/browse/ASTERISK-17146
Is this bug still present in 1.8.6.0 or 10.0.0-beta1? I am planning on
migrating from 1.4 so it's
On 09/06/2011 09:08 AM, Arjan Kroon | Mobillion wrote:
Hi,
I'm looking for a PCIe card with 1 ISDN2 connection which works with Asterisk
Could anybody give me an advise which card I can use?
Both Sangoma and Digium have PCIe ISDN cards although a single BRI port
might be a bit of a
On 09/07/2011 02:17 AM, A Dunor wrote:
Hello list, I am a beginner at asterisk. I want to access my asterisk
box from my laptop, on a different network (mobile hotspot). The
asterisk box doesn't have a static ip, how do I connect with it using
ssh or other such programs?
Thanks for your
Hi,
Does anyone know which exact version of mISDN is required for chan_misdn
in 1.8 10?
TIA,
Patrick
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On 10/13/2011 06:21 PM, Mike Diehl wrote:
I'm thinking it's a firewall/NAT timeout issue. Has anyone seen this? Has
anyone fixed it? Any ideas, otherwise?
Did you try turning off the SIP ALG on the Sonicwall?
Regards,
Patrick
--
Hi,
After a make menuselect I now have menuselect.makeopts,
menuselect.makedeps and menuselect-tree.
How do I get the buildsystem to use the settings in those files? Thus
far they just seem to get overwritten if I do:
$ cd asterisk-1.8.8.0-rc1
$ cp -v /tmp/menuselect* .
$ ./configure
$
On 10/19/2011 04:52 AM, Luke Hamburg wrote:
I think this might actually be a bug.
https://issues.asterisk.org/jira/browse/ASTERISK-18137
Thank you very much for pointing that out Luke. Seems I bumped into the
same bug.
Regards,
Patrick
--
On 10/19/2011 03:08 PM, Jason Parker wrote:
On 10/18/2011 09:52 PM, Luke Hamburg wrote:
I think this might actually be a bug.
https://issues.asterisk.org/jira/browse/ASTERISK-18137
It is indeed a bug, but it's not the bug you referenced. This issue only
exists in 1.8.8.0-rc1. It has been fixed
On 10/19/2011 03:08 PM, Jason Parker wrote:
On 10/18/2011 09:52 PM, Luke Hamburg wrote:
I think this might actually be a bug.
https://issues.asterisk.org/jira/browse/ASTERISK-18137
It is indeed a bug, but it's not the bug you referenced. This issue only
exists in 1.8.8.0-rc1. It has been fixed
On 10/21/2011 04:33 AM, Luke Hamburg wrote:
Patrick:
would you mind sharing how you're using the menuselect.makeopts exactly? I
first had this problem with 1.8.7.0 and again with 1.8.8.0-rc1. I've since
updated to 1.8.8.0-rc2 but for some reason I am still unable to get
menuselect to use the
On 10/27/2011 08:57 PM, Vinod Dharashive wrote:
Hi Richard,
Thanks for reply, how can I identify clocking issues on card. Manually calls
goes successfully on both the card. After 10 min signaling links goes down.
Since you have a Sangoma card why don't you ask Sangoma support instead
of
On 10/28/2011 01:02 AM, motty.cruz wrote:
Hello,
I noticed Asterisk 1.8.4.1 execute number dial twice
Log
== Using SIP RTP CoS mark 5
-- Executing [912066604@sipphones:1] Set(SIP/4773-0003e920,
CALLERID(num)=2066604) in new stack
== Extension Changed 4773[sipphones] new state InUse
On 10/31/2011 10:39 PM, bilal ghayyad wrote:
Dear;
It look like AGI is not suiting this, maybe I need AMI?
Because no need to do a call to collect the data, I need to collect data
without doing a call ... so what do u suggest?
What I need, is to know how many concurrent calls in the queue,
On 11/15/2011 02:45 AM, jordan pan wrote:
Recently，I met a very strange phenomenon。I found that my asterisk bin
file had changed when running。I checked a lot of machines ， and the
result is almost all of the bin files have taken place。
[snip]
Maybe prelink? See man prelink.
Regards,
Patrick
On 11/22/2011 08:14 AM, Jai Rangi wrote:
[removed commercial offer]
You posted to the wrong list. The correct list for commercial business
related discussion is asterisk-biz. Please do not spam the
asterisk-users list again with your commercial offers.
Regards,
Patrick
--
On 29-11-11 19:54, bilal ghayyad wrote:
Can u help me to determine this for Cisco IP Phones model 7942G with SIP image
and Polycom IP Phones?
Why don't you just download the admin manuals of these phones and look
it up?
Regards,
Patrick
--
On 14-12-11 10:18, Brynjolfur Thorvardsson wrote:
Hi all
I’ve been saddled with recreating a running Asterisk PBX setup (with
Ruby on Rails). Due to some wrangling between my client and the original
developers I am not able to talk to the developers themselves but have
been given full SSH
On 14-12-11 13:56, Paulo Santos wrote:
Hello list,
An Asterisk installation that was doing fine suddenly stared segfaulting
a couple of times per day. I enabled all the logging and debugging to
try to find a pattern but there was too much information to see exactly
where it broke. So I enabled
Hi,
In the thread Interesting attack tonight fail2ban them Bruce B
mentioned it would be nice to have input from the Community to come up
with the best set of fail2ban filters. That's a great idea. So let's
start with Bruce's filters (thanks!) and take it from there. Anyone have
any
On 03-01-12 11:22, Marco Mooijekind wrote:
Dear all,
I have the following challenge using Asterisk 1.8, using a Digium B410P
card on BRI (The Netherlands, KPN ISDN) .
DAHDI is running, dahdi_tools indicates OK on my span and light on back
of card is green.
However, in Asterisk i get the
On 03-01-12 14:13, Faraj Khasib wrote:
anyone?
what should x-lite account be for guest user ?I tried guest but didnt work
A guest does not need an account on your asterisk server so you do not
need to configure an account on xlite. Instead on xlite you just dial
On 03-01-12 14:47, Kaushal Shriyan wrote:
Hi,
Please help me understand the following applications and what are its
advantages if we compare between each of them.
Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
How about Googling first? Google for these two terms:
-
On 03-01-12 16:24, Danny Nicholas wrote:
Hello List,
I work in an environment where I have to request IPTABLES changes rather
than doing them myself. Is there a way to utilize my SSH (port 22) to
get a functional (and with good sound) Asterisk installation with
multiple channels up without
On 16-01-12 19:47, Russ Meyerriecks wrote:
[snip]
2- Do I even need Dahdi, if the server doesn't connect to PSTN at all and
it's all SIP? If yes, what do I need it for?
Dahdi is a set of drivers for telephony hardware. You won't need it for pure
sip Asterisk implementations.
Unless things
On 17-01-12 18:23, Mike Diehl wrote:
I've got some users reporting an odd problem.
Once in a while, their Polycom phones ring and they are unable to answer
them... any of them.
When they pick up the handset, all of the phones continue to ring. Same thing
happens if they grab a different
On 18-01-12 04:57, Roi Stork wrote:
Hi,
I wasn't aware of the difference in quality between landline and
mobile phones, or that cellphones use a low bit rate.
I did a test again, landline voice quality is better.
From what you observed, how much drop in quality do I expect when
switching from
On 19-01-12 20:32, Vinod Dharashive wrote:
Hi Kevin,
Is there any possibility of asterisk supported for dialogic
cards in future. does digium has any plan for supporting it?
The Dialogic Diva boards (formerly known as Eicon Diva Server boards)
are supported by Asterisk via chan_capi
On 26-01-12 18:08, Jeff LaCoursiere wrote:
[snip]
I'm also very interested in working examples, especially if someone has
set it up for SIP termination trunks rather than Dahdi.
Maybe I am missing something here but why would you want to emulate a
keysystem with analog (thus single call)
On 31-01-12 22:47, Michelle Konzack wrote:
Hello *,
is here someone with an experience of the Eicon Diva PRO 3.0?
I need 2 or 3 cards to connect an Anlagenanschluß (I do not know the
name in englisch, but is 2 or more lines with the same numbers) to my
intranet servers (IBM x335/x345 -
On 07-02-12 16:30, Shaun Ruffell wrote:
[snip]
It looks like your development box is having problems with
interrupts from the card. Once you run dahdi_cfg for the span you
should be getting 1 interrupts/sec and above I can see you only
got 28.
I've seen this recently with some risers
On 07-02-12 17:54, Shaun Ruffell wrote:
[snip]
I hear what you're saying although I don't think it's practical in
this case.
[snip]
Thank you for your elaborate feedback. It's clear why you are the one
developing and I send my 0.02 to the mailinglist :) I see your point.
Digging in
On 07-02-12 18:41, Josh wrote:
[snip]
Thanks, another mystery solved then - Asterisk does rely on the
Linux/Unix routing, in which case I would definitely need to take care
of the SNAT/DNAT and proper routing/forwarding of packets between
interfaces using core Linux/Unix tools. Am I correct in
On 09-02-12 14:52, Stefan Schmidt wrote:
Am 09.02.12 14:19, schrieb Bryant Zimmerman:
Stefan
This is on target with my configuration I am working on. What kind of
dialplan were you using when running the tests.
Were you doing database lookups or just answering the calls and playing
hold
On 15-02-12 14:31, Danny Nicholas wrote:
You could register the agent to a SIP extension with followme. When the
queue went to ring the SIP extension, followme would send the call on to the
mobile/land line.
It's been decades since I last bought an ISDN phone and I'm not even
sure you can
On 16-02-12 20:18, Luke Hamburg wrote:
https://reviewboard.asterisk.org/r/1599/
I so wish that this patch would be backported to the 1.8 branch! I am
considering switching to trunk just for this alone.
I know it's a stretch but, given the popularity of running Fail2Ban
alongside Asterisk,
On 25-02-12 19:47, Jason Parker wrote:
yum and rpm do not support downgrades.
Incorrect. There is yum downgrade. See man yum.
Regards,
Patrick
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On 02-03-12 07:57, Anita Hall wrote:
[snip]
This module has not been updated for the last 2 years during which the
linux kernel has changed (I am told).
Have you asked Atcom?
Is there any other manufacturer of torrent card who would be using the
same architecture and keeping his drivers
On 05-03-12 17:41, Royce Souther wrote:
Last week I got an email from Link2VoIP saying that they are shutting it
down in a few months. Sighting competition and the unethical changes
being made to the Internet by special interest groups.
I use Link2VoIP for termination, connecting my Asterisk
On 06-03-12 23:03, Jason Parker wrote:
[snip]
It should only set them if the directory does not exist. If it's changing them,
something is very seriously broken.
An RPM which updates a previous version will change the user/group
permissions of any existing directory or file as it is
On 06-03-12 23:07, Karl Fife wrote:
Yep. That's what's happening. I'll file a bug.
AFAICT it's not a bug but the way RPM works.
Regards,
Patrick
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On 06-03-12 23:36, Jason Parker wrote:
On 03/06/2012 04:24 PM, Patrick Lists wrote:
On 06-03-12 23:07, Karl Fife wrote:
Yep. That's what's happening. I'll file a bug.
AFAICT it's not a bug but the way RPM works.
Regards,
Patrick
He didn't suggest that he was talking about RPMs
On 07-03-12 01:28, Mike Diehl wrote:
I've been logging sip registrations from this IP address for 2 days now. I've
emailed the domain's admin, but nothing seems to come of it.
I've routed him into oblivion, but still, I think 50 requests a second for 2
days is a bit much.
Any ideas?
Did you
On 07-03-12 11:44, David Klaverstyn wrote:
Hi All,
Can someone please tell me if it is possible and if so how do I go about
streaming a live conference to the internet for internet users to listen
to? I’d hope to be able to do thus dynamically as conferences are
created and internet users can
On 15-03-12 01:54, Jan Blom wrote:
Hello,
Is anyone aware of an AMR-NB codec for Asterisk 10? The old patch
floating around for older versions of Asterisk doesn’t seem to work anymore.
The only patch I have seen is the one for 1.8 which is on sourceforge
(search for asterisk-amr). I did a
On 22-03-12 16:47, Danny Nicholas wrote:
So is Asterisk 10 supposed to be 32 or 64 bit? P.S. a pox on sqlite3!
Depends on how you compile it or what version you downloaded. Both are
possible. Asterisk 10 32bit requires a 32bit OS and Asterisk 10 64bit
requires a 64bit OS. I didn't
On 22-03-12 17:26, Danny Nicholas wrote:
Confusion? I'm looking at the Asterisk Release download directory and only
see asterisk-10.1.3.tar.gz, not asterisk-10.1.3-32.tar.gz and
asterisk-10.1.3.64.tar.gz. is this a configuration or make option? As for
the pox on sqlite3, I have had varying
On 03/29/2012 03:47 PM, Markus wrote:
I'm pretty sure there are a bunch of people who would be happy to pay
money for a better a2billing. Including myself :)
Have you looked at jbilling.com ? It's F/OSS with commercial support.
Regards,
Patrick
--
Hi Olivier,
On 04/03/2012 10:31 AM, Olivier wrote:
For training sessions, I'm evaluating the possibility to use a single
physical server to host 5 virtual servers, each with its own Dahdi
PCIe card, instead of using 5 physical machines, hoping a single
physical server would easier to transport,
On 04/11/2012 01:39 PM, upendra wrote:
Hi,
can anyone tell me what does that 2.4.0+2.4.0 means in dahdi release
numbering ??? 2.4.0?
A combination of dahdi-linux 2.4.0 and dahdi-tools 2.4.0.
See http://www.asterisk.org/downloads
I recommend you use the latest version of both (2.6.0).
On 04/12/2012 06:11 AM, upendra wrote:
Hi,
thanks for reply , i want to know the 2.4.0 or 2.6.0 means what , how
they naming it , by the kernel version or its just a official release
number of digum...??
It's a Digium created release number.
Regards,
Patrick
--
On 04/12/2012 09:09 PM, Brent Davidson wrote:
I'm trying to get app_swift (app_swift-2.1-b1-ast10) working with
Asterisk 10.2.1 on Ubuntu Server 11.10. Everything appears to compile
correctly, but when I go to load the module I get the following:
server*CLI module load app_swift.so
Unable to
On 04/15/2012 01:15 PM, Gustavo Garcia Bernardo wrote:
Is it a good idea to use asterisk transcoding from G711 to iLBC or
should I find out any other solution not involving transcoding (f.e.
using G.729 that is supported in both sides). I'm worried about voice
quality and trying to avoid paying
On 04/16/2012 08:06 AM, Olivier wrote:
Hi,
Which free or non-free (as beer) Sugarcrm plugin would you recommend
to add click to dial feature with asterisk ?
I can see a quite long list of such plugins but not all of them seem
up-to-date (judging by comparing with latest Sugarcrm version
On 09-05-12 18:46, khalid touati wrote:
Should I understand that no Asterisk user has issues with ISDN system
access configuration from UK? or maybe no one is using Asterisk In UK :) ?
I have no idea. But other than the error you have given very little
information to go on. Which card are you
On 09-05-12 19:54, khalid touati wrote:
Thank you for your answer, I think I posted dhadi version and so but
let me add more details and recap them below:
We are using asterisk 1.8.12.0 with dahdi 2.6.0, on CentOS 6.2. it's a
digium card 1HA8-0400BLF
output of dahdi_hardware:
On 09-05-12 20:57, khalid touati wrote:
Yeah sorry for that, I realized something is missing after I sent the
email, but it is exactly what I have (other than order here, which
doesn't really matter: you posted ami,te,term, I have ami,term,te).
Actually I had couple technicians from digium
On 10-05-12 21:10, khalid touati wrote:
Hi All,
I did downgrade to asterisk 1.6.2.6 and dahdi 2.3 which is working with
other BRI line (in our NL office), but I get this type of errores:
-- Called G1/0788744550
[May 10 20:07:20] ERROR[27479]: chan_dahdi.c:12280 dahdi_pri_error: ACK
On 10-05-12 23:47, Kevin P. Fleming wrote:
On 05/10/2012 03:20 PM, khalid touati wrote:
Thank you Patrick for the detailed info, it does make perfect sense to
me, I never expected that Digium cards have such an problem!
There are patches in the works already (being tested by users in Europe)
Hi Khalid,
Judging from that bug report I *think*:
On 11-05-12 03:39, khalid touati wrote:
Patrick,
I got confused though is this true:
Any Asterisk soft+digium hdw = it doesn't work
There seem to be combinations that do work. It is my understanding from
that bugreport that an older libpri
On 11-05-12 05:44, khalid touati wrote:
Thank you for your reply Patrick!
for the first situation, I did try asterisk 1.6.2.6 and dahdi 2.3 but
with no success.
Can anyone suggest a combination that works till a patch is released?
Unfortunately not and I don't have a Digium BRI card to test
On 05/13/2012 03:07 AM, khalid touati wrote:
My Issie is finally fixed and I can make calls, I received actually from
digium the fix, I'll try to give as much details as I can to make sure
people who find this thread understand pb and solution.
Problem: not able to dial calls using BRI from
On 16-05-12 17:10, gincantalupo wrote:
Hi Larry,
thank you for your answer.
This is same test I did. After this I lowered again to 4800...result:
iaxmodem receives at 9600 b/s.that's why I cannot solve the puzzle.
I put that line (Class1RMQueryCmd: !24,48,72) in config.IAXtty
Hi Khalid,
On 18-05-12 20:50, khalid touati wrote:
Hi Patrick,
it seems like you have the magic ball, I think what you described is
exactly what happened:
After we tested the server+ link and we were able to have simultaneous
calls (as expected), and knowing that this server was not touched
Hi zoa,
On 31-05-12 17:39, joachim wrote:
Ellow,
We released zoiper for Android today, available for free here:
https://play.google.com/store/apps/details?id=com.zoiper.android.app
SIP and IAX is supported, should work quite well, unfortunately it is
really hard to test all android and
On 06-06-12 11:41, Thorsten Göllner wrote:
Where can I find such ip-lists, please?
http://www.ipdeny.com/
Regards,
Patrick
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