China too wide, but regardless! How is asterisk take care such situation?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk
Sent: Tuesday, October 06, 2009 2:36 AM
To: Asterisk Users Mailing List -
- Original Message -
From: Tony Mountifield t...@softins.clara.co.uk
To: asterisk-users@lists.digium.com
Sent: Monday, July 13, 2009 11:13
Subject: Re: [asterisk-users] ooh323 and h323
In article f43e89e805474ecdbde144af654e8...@benderd,
Dovid Bender asteriskus...@dovid.net wrote:
Hello,
I corrected a bug and did some little optimizations in app_jack.c.
It works great now.
I propose this new file based on revision 140568.
Fabien
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2007 - 2008, Russell Bryant
*
* Russell Bryant russ...@digium.com
*
Hi,
I try to use asterisk as softphone with alsa.
I search how to answer to an incoming sip call (from wan).
Does anyone did it (extensions.conf exemple) ?
Thanks,
Fabien
___
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- Original Message -
From: B.Masoud @ SH
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Sent: Tuesday, October 06, 2009 1:14 AM
Subject: [asterisk-users] Networking Concept
Hello,
I would like to know how Asterisk deal in this case:
Assume I
hi:
in our country callerid is sent with fsk. but it will sent DTAS(dual
tone alerting signal) first, then fsk callerid, then first ring.
I search google, but didn't find the configuration method or patch for this.
any good suggestion for asterisk to detect this? I have trid
asterisk 1.4 and
How they can?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Neland
Sent: Tuesday, October 06, 2009 12:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Networking Concept
In pakistan they have protocol scanners mounted on all the 4 fibers that
enter and leave pakistan. They can detect VOIP usage by upload/download
patterns / etc. They have invaded and arrested many illegal 'calling
card' operators in the past. Im sure the same is true for china. However
I dont know
Hello,
i have a big problem...
i want to connect my asterisk server to a lancom 1722 device (ISDN/SIP) Gateway.
sip.conf:
[general]
context=default
allowguest=yes
realm=10.1.1.209
bindport=5060
bindaddr=0.0.0.0
tos_sip=cs3 ; für SIP-Pakete (Kommunikationsaufbau)
tos_audio=ef ; für
Hi!
Since 1.6, when using AEL, macros are implemented using Gosub(). Is
there workaround to have MACRO_EXTEN also in this case?
regards
Klaus
PS: I know I could use something like
context fromSip {
11 = myMacro(${EXTEN})
}
macro myMacro(MACRO_EXTEN) {
}
but isn't there some
Hello,
Is it possible that asterisk can parse ISUP parameters from SIP messages.
For example:
INVITE
sip:13039263...@den1.level3.comsip%3a13039263...@den1.level3.comSIP/2.0
Via: SIP/2.0/UDP den3.level3.com
From:
Hi All,
I'm trying to test video calls on asterisk, my issue is i have two
asterisk servers linked via an IAX peer and users register on the
asterisk via SIP.
if both video phones are registered on the same asterisk server, video
call works, but call is via SIP. but,if one video phone 1 is
I doubt AGI is the issue. You can verify this by setting up a context to
record a call without AGI. I'd try using the WAV49 format instead of
regular WAV.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kim Delevati
Sent:
Hi. I'm using the Monitor command to record outgoing calls through SIP, but
the quality is very poor, MUCH worse than the call itself who many times is
perfect. I can barely understand anything thats said.
Im using Asterisk 1.4.21.2 on a Ubuntu Server 8.10, on a Pentium R 2ghz with
1gb DDR2.
Doesnt WAV49 have worse quality than WAV? Or am I wrong?
On more info, I've tested it on calls using alaw, gsm, g729, and in all the
recording has poor quality.
Tzafrir answered me talking about a debian patch for this issue, does anyone
know where to find it?
2009/10/6 Danny Nicholas
Hello,
Help need to solve Problem...kindly let me know how to solve the below
problem now i have CentOS 5.3 (Final)
I am new to Asterisk Linux, I installed Cent OS and Updated all Packages.
When i try MAKE ALL in dahdi-linux-complete-2.2.0.2+2.2.0 Folder it is
showing the following error:
- David Home davidhome1...@gmail.com wrote:
Hello,
Help need to solve Problem...kindly let me know how to solve the
below
problem now i have CentOS 5.3 (Final)
I am new to Asterisk Linux, I installed Cent OS and Updated all
Packages.
When i try MAKE ALL in
This may help:
http://www.centos.org/modules/newbb/viewtopic.php?viewmode=flattopic_id=117
72forum=37
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Home
Sent: Tuesday, October 06, 2009 10:41 AM
To:
You need to yum install kernel-devel'
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Oct 6, 2009, at 8:40 AM, David Home wrote:
Hello,
Help need to solve Problem...kindly let me know how to solve the below
problem now i have CentOS 5.3 (Final)
I am new to
Hello,
We need Asterisk to register with a variable and changing number
(hundreds) of VoIP providers, is there a way to do this in a database
and without reloading the entire sip config?
Where Asterisk needs to register is determined by downstream users, so
we need to do it real time and
On Tue, Oct 06, 2009 at 09:10:36PM +0530, David Home wrote:
Hello,
Help need to solve Problem...kindly let me know how to solve the below
problem now i have CentOS 5.3 (Final)
I am new to Asterisk Linux, I installed Cent OS and Updated all Packages.
When i try MAKE ALL in
Fabien Comte schrieb:
I try to use asterisk as softphone with alsa.
I search how to answer to an incoming sip call (from wan).
Does anyone did it (extensions.conf exemple) ?
Maybe something like
Dial(ALSA/hw:0,0);
(untested)
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566
On Tue, 6 Oct 2009, Faraz Khan wrote:
In pakistan they have protocol scanners mounted on all the 4 fibers that
enter and leave pakistan. They can detect VOIP usage by upload/download
patterns / etc. They have invaded and arrested many illegal 'calling
card' operators in the past. Im sure the
Hi,
I am working on Trixbox. I want to create my own dial() function (named
specificdial()) and I want to know how I can create a module and integrate
the module in the trixbox plateform.
thanks a lot
Mickael
___
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WAV49 is by definition lesser quality but Louder sound, which might help the
perceived quality.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kim Delevati
Sent: Tuesday, October 06, 2009 10:23 AM
To: Asterisk Users
On Tue, Oct 06, 2009 at 06:18:58PM +0200, mickael ropars wrote:
Hi,
I am working on Trixbox. I want to create my own dial() function (named
specificdial()) and I want to know how I can create a module and integrate
the module in the trixbox plateform.
The source for Dial() is in
Anyone for this ?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo
Sent: Monday, October 05, 2009 11:02 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] T38 REINVITe issue
Hi
My call flow is
T38
Did you publish it somewhere ?
On 10/05/2009 09:19 PM, Danny Nicholas wrote:
There are plenty of good products out there, but I use my own
PERL/Apache/AMI interface for this
*From:*
Danny Nicholas schrieb:
WAV49 is by definition lesser quality but Louder sound
http://www.vloud.com/
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk:
Hi Tzahir,
thanks a lot for your quick answer. What do you think about creating an
addon ? is it easier to integrate it on trixbox ?
regards
Mickael
2009/10/6 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Oct 06, 2009 at 06:18:58PM +0200, mickael ropars wrote:
Hi,
I am working on
Hi!
I tested with wav49 and now the quality is MUCH better. Almost the same as
the call. I dont know the reason, probably it was some problem on the
decoding of the codec I used to wav, strangely that doesnt happen with
wav49.
The quality now is very acceptable, whereas with wav I couldnt
Hi,
Does anyone know of a module that integrates Asterisk with the Resort Data
Processing property management software? I have a potential VOIP client but
they need PBX integrated with the RDP software for call billing.
Thanks,
David Wathen
Wathen Consulting
Senior Consultant
Cell:
Adittional doubt: can I use WAV49 on the Record() command? Voip-info doesnt
say anything about it (or the WAV listed there is equivalent to WAV49, being
wav = WAV).
2009/10/6 Kim Delevati kim.delev...@gmail.com
Hi!
I tested with wav49 and now the quality is MUCH better. Almost the same as
If you do Record(x.WAV) instead record(x.wav) the output will be in WAV49
format.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kim Delevati
Sent: Tuesday, October 06, 2009 1:08 PM
To: Asterisk Users Mailing List -
Ok, I used Record with wav49 and, instead of recording in a good quality
like Monitor, it was almost as bad as regular wav.
These things crack my head, now I'll have to switch all my Record for
Monitor, somehow limiting time and in a way that # stops the recording.
Should be fun.
2009/10/6 Danny
On Tue, 2009-10-06 at 17:03 +0100, Gordon Henderson wrote:
On Tue, 6 Oct 2009, Faraz Khan wrote:
In pakistan they have protocol scanners mounted on all the 4 fibers that
enter and leave pakistan. They can detect VOIP usage by upload/download
patterns / etc. They have invaded and arrested
On Monday 05 October 2009 10:06:55 Thomas Kenyon wrote:
If I connect to the USB modem with minicom and issue the ATDxxx;
command with a semicolon at the end to signify a voice call I get the same
error response.
Could someone else with this type of USB modem tell me if that
Haven't published b/c some folks can be critical of that kind of thing.
It's about 45 lines of dialplan and 200 lines of PERL if you'd like it.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Xavier
Sent: Tuesday, October
Hello,
I thought to post this here before my manager starts his own coding
project to give us a workaround. My situation I'm running into is as
follows:
1. A call comes into our Asterisk system, it's trunked from one office
to another via IAX.
2. Call enters a queue and is picked up by one of
Google for some of the How Tos built around Elastix and Trixbox. Both of these
are CentOS based as well.
good luck.
- Original Message -
From: James Hankins j...@allpointsmediaworks.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Checkout nerdvittles.com. They have good stuff when they are up.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
astgro...@comcast.net
Sent: Tuesday, October 06, 2009 2:49 PM
To: Asterisk Users Mailing List -
Please how do I stop the following ???
Asterisk ended with exit status 127
Asterisk died with code 127.
Automatically restarting Asterisk.
mpg123: no process killed
Best Regards,
--
This message has been scanned for viruses and
dangerous content and is believed to be clean.
SplatNIX IT
On Tue, Oct 06, 2009 at 09:01:38PM +0100, --[ UxBoD ]-- wrote:
Please how do I stop the following ???
Asterisk ended with exit status 127
Asterisk died with code 127.
Automatically restarting Asterisk.
Asterisk died and was restarted.
mpg123: no process killed
No left-over mpg123
Just a guess. You need to modify safe_asterisk to kill the mpg123 process.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ UxBoD ]--
Sent: Tuesday, October 06, 2009 3:02 PM
To:
Hi,
In dev-list, some people reported Asterisk 1.6.2-rc2 would suddenly restart.
Here, a platform running 1.6.1.6 is also suddenly restarting (once or twice
a day with moderate load (40 users)).
I don't have much details to report here at the moment.
Has someone met something similar ?
Thoughts
On Tue, Oct 06, 2009 at 07:29:17PM +0200, mickael ropars wrote:
Hi Tzahir,
thanks a lot for your quick answer. What do you think about creating an
addon?
Why? In most cases such special things can be immplemented in the
dialplan. Please convince us that this cannot be implemented using plain
Try:
killall -9 safe_asterisk
killall -9 asterisk
/etc/init.d/asterisk start
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
The Asterisk Development Team has announced the next set of Asterisk release
candidates for versions 1.4.27, 1.6.0.16, 1.6.1.7, and 1.6.2.0. These release
candidates are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release candidates address issues
--[ UxBoD ]-- wrote:
Please how do I stop the following ???
Asterisk ended with exit status 127
Asterisk died with code 127.
Automatically restarting Asterisk.
mpg123: no process killed
You figure out why asterisk is crashing. :)
This has nothing to do with mpg123, which is just an
_
From: Danny Nicholas [mailto:da...@debsinc.com]
Sent: Tuesday, October 06, 2009 4:07 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Receptionist GUI?
Here's the part I can publish:
here is the dialplan snippet and the public
Hello, how are you?
I need to be able to send a DTMF to an existing channel remotely. So I made
a php script to do such with the Manager command PlayDTMF. I need it for
example to start a transfer.
isb177*CLI features show
Builtin Feature Default Current
---
Please test some of the newly released RCs, such as 1.6.1.7-rc2 or 1.6.2.0-rc3
(depending on the branch you're using).
Leif.
Olivier wrote:
Hi,
In dev-list, some people reported Asterisk 1.6.2-rc2 would suddenly restart.
Here, a platform running 1.6.1.6 is also suddenly restarting (once or
On 091001 0406, Mindaugas Kezys wrote:
We had many problems with IAX2, changing to SIP solved them all.
Let me paste link to wise-words which clearly illustrates our experience:
http://wiki.kolmisoft.com/index.php/Why_we_do_not_suggest_to_use_IAX2
Thanks for the wise-words. From the
On 091006 1249, Darrin Henshaw wrote:
1. A call comes into our Asterisk system, it's trunked from one office
to another via IAX.
2. Call enters a queue and is picked up by one of the agents.
3. That agent has to transfer the call, could be for a number of
reasons the client wanted someone in
Ujjval Karihaloo wrote:
Her eis my users.conf entry for Asterisk registration to the Sip
Provider. (I know I don’t have T38 as allowed codecs, not sure what to
add for T38)
[trunk_66]
;register
allow = ulaw
dialformat = ${EXTEN:1}
canreinvite = no
hasexten = no
hasiax = no
Already have it...
If provider does not challenge re- invite
Fax works fine!
Ujjval
On Oct 6, 2009, at 11:33 PM, Trevor Peirce tpei...@digitalcon.ca
wrote:
Ujjval Karihaloo wrote:
Her eis my users.conf entry for Asterisk registration to the Sip
Provider. (I know I don’t have T38 as
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