--- On Tue, 2/23/10, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Look at qualify= for sip.conf, and consider to extend your
diaplan for a
better routing decision with a snippet like this
Actually, I noticed that setting qualify= alone solves my issue. I apparently
Tilghman Lesher wrote:
On Tuesday 23 February 2010 05:27:55 Per Jessen wrote:
To be honest I don't remember any more, I just know my queueing
doesn't work unless I reload. I think it's a timing issue at
startup - that app_queue gets loaded too early or something. ah,
here is my question
24 feb 2010 kl. 01.22 skrev Kristian Kielhofner:
On Tue, Feb 23, 2010 at 8:22 AM, Michelle Dupuis supp...@ocg.ca wrote:
We're creating a SIP gateway for a client that will take one leg of a call
in via SIP, and out the other side via H.323. To minimize load on the
gateway, we would like to
Hi!
What I want is, if a call coming from a trunk 100 rings, and if the
caller wants to be transfered to 101, the transfer is denied. In other
words, 101 can't get transfered calls.
WHat about using featuresmap to replace the usual transfer application
with code that tests to see the
Hi!
Look at qualify= for sip.conf, and consider to extend your
diaplan for a
better routing decision with a snippet like this
Actually, I noticed that setting qualify= alone solves my issue. I
apparently don't require extra dialplan logic because if the peer is
unreachable (according
Hello list,
anybody has handy an example of how to loop over an ASTDB family by
getting all the keys in the dialplan?
Like I have the AstDB set as:
/test/102 : 205
/test/106 : 203
/test/113 : 209
I would like to get (in any order) the 102, 106 and 113 as members of
the family test.
TIA,
l.
Hello,
I´m running Asterisk 1.6.1.11 .
Ive got 2 classes in my musiconhold.conf:
[general]
[default]
mode=files
directory=/var/lib/asterisk/moh
[signal]
mode=files
directory=/var/lib/asterisk/moh_signal
I use this for 2 different queues and it works fine.
When I
Hi People, I don't know if my problem should be reported in this forum, but
maybe somebody knows about it.
I'm using the tool .NET WebService Studio to test the web service which is
working with asterisk by AMI.
It is working fine, the dialplan is executed correctly... the problem is when
the
On Tue, 23 Feb 2010, Tzafrir Cohen wrote:
On Tue, Feb 23, 2010 at 12:19:31PM +, Gordon Henderson wrote:
On Tue, 23 Feb 2010, Tzafrir Cohen wrote:
But then again, lxc uses much of the work on containers done also by and
for OpenVZ. Sort of like the VMWare/Xen/KVM story all over again,
hi,
do u know a good, inexpensive hosting company in israel, that will host voip?
i want to have my asterisk server here, in the u.s., to hook up to a voip host
in israel. most traffic would be to israel. would prefer one base rate to any
landline location in israel. and, something
Hi Guys
We are using asterisk 1.4 on all of our platforms for a while now.
Some of our partners recommended to use asterisk 1.6 in order to improve
overall stability and performance.
Can someone please let me know if you have a such experience?
Also, do you have any other negative or
On Tue, Feb 23, 2010 at 09:28:23PM +0200, Rudi Oosthuizen wrote:
Hi All,
We have encountering issue that IAX enable voice gateways not
registering with asterisk after upgrade from asterisk 1.4.18.1 - 1.4.29
Before that IAX works very well.
If any one have similar issue and solution
Tzafrir Cohen wrote:
http://downloads.asterisk.org/pub/security/AST-2009-006.html
http://downloads.asterisk.org/pub/security/IAX2-security.html
And more importantly, the UPGRADE files included in the source code that
the OP downloaded pointed to all of this stuff.
--
Kevin P. Fleming
Digium,
On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro juan.san...@hotmail.com wrote:
Hi Guys
We are using asterisk 1.4 on all of our platforms for a while now.
Some of our partners recommended to use asterisk 1.6 in order to improve
overall stability and performance.
Can someone please let me know
Well.. we do from time to time have SIP attacks, Core dumps and lately very
weird issues with Cisco phone becoming unreachable.
Anyone had issues with Cisco 7940 where by ALL of the phones will for 30-90
seconds become unreachable?
All phones are on T1 MPLS network using Cisco 26xx routers..
Wednesday, February 24, 2010, 3:56:50 PM, David wrote:
On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro juan.san...@hotmail.com wrote:
Hi Guys
We are using asterisk 1.4 on all of our platforms for a while now.
Some of our partners recommended to use asterisk 1.6 in order to improve
overall
Hello,
Asterisk Real time database worked on astersik 1.6.2.0 but now i am working
on Asterisk to latest version which is 1.6.2.2 ,there is a a warning
[Feb 24 16:26:14] WARNING[4053]: config.c:2025 find_engine: Realtime mapping
for 'sippeers' found to engine 'mysql', but the engine is not
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
Hi
It might seem that you have installed 1.6.2.2 over 1.6.2.0, but not
updated the correct modules.
If you - from the CLI - do a mode show like sql. What is your result?
- - Tommy
ahmed magdy skrev:
Hello,
Asterisk Real time database worked
Hi gurus,
In need of a little help here. I¹m trying to do the Asterisk media release
by using canreinvite=yes. But I found weird behaviour when comes to BYE.
Below are my current setup:
Client A is registered to Opensips
Client B is registered to Asterisk
A Opensips Asterisk B
On hangup
Have you check if MySql is already running?
Have you check HD space?
regards.
2010/2/24 ahmed magdy amagdy.ibra...@gmail.com
Hello,
Asterisk Real time database worked on astersik 1.6.2.0 but now i am working
on Asterisk to latest version which is 1.6.2.2 ,there is a a warning
[Feb 24
Gergo Csibra escribió:
Wednesday, February 24, 2010, 3:56:50 PM, David wrote:
On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro juan.san...@hotmail.com wrote:
Hi Guys
We are using asterisk 1.4 on all of our platforms for a while now.
Some of our partners recommended to use asterisk 1.6
Lawrence Na (my-...@vyke) wrote:
On hangup below are the SIP flow which I’ve notice from the Asterisk
server itself:
1. Opensips forward the BYE to Asterisk
2. Asterisk response with 200 OK
3. Asterisk send INVITE to B
4. B response with 200 OK with SDP
5. Asterisk reply
Hi People,
I work in a company that are using asterisk as pbx.
I need a way to identify what client my employees are calling. For example:
- For each call that an employee of my company make to a customer, must
identify the client name in the CDR table.
- Is there a way of my employee enter a
Douglas Pasqua wrote:
Hi People,
I work in a company that are using asterisk as pbx.
I need a way to identify what client my employees are calling. For
example:
- For each call that an employee of my company make to a customer,
must identify the client name in the CDR table.
- Is there
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
Douglas Pasqua skrev:
I need a way to identify what client my employees are calling. For example:
- For each call that an employee of my company make to a customer, must
identify the client name in the CDR table.
- Is there a way of my
Why not set your clients up as extensions so your employee's call them
with an extension code instead of dialing a number? For example
Exten = 1001,1,Dial(DAHDI/g1/18005551212)
Exten = 1002,1,Dial(DAHDI/g1/18005551213)
Exten = 1003,1,Dial(DAHDI/g1/18005551214)
Or more efficiently
Exten =
Op 24-02-10 17:35, Douglas Pasqua schreef:
Hi People,
I work in a company that are using asterisk as pbx.
I need a way to identify what client my employees are calling. For example:
- For each call that an employee of my company make to a customer, must
identify the client name in the CDR
FWIW, we recently moved a 1.4.29 Asterisk system onto a VMWare guest machine
and with 40+ call legs (20+ calls), it isn't even breaking a sweat. We have
had no complaints from users nor have we noticed any degradation in voice
quality, be it live, voicemail or conference bridge (with six
*Code:*
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback(Local/91441425477...@default-b9f2,1,
sip-silence) in new stack
-- Playing 'sip-silence' (language 'en')
-- Executing AGI(Local/91441425477...@default-b9f2,1, agi://
127.0.0.1:4577/call_log) in new stack
On Wednesday 24 February 2010 10:16:25 Miguel Molina wrote:
Gergo Csibra escribió:
Wednesday, February 24, 2010, 3:56:50 PM, David wrote:
On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro juan.san...@hotmail.com
wrote:
Hi Guys
We are using asterisk 1.4 on all of our platforms for a while
It looks like your channel has been hungup during the AMD application,
not that the AMD application is hanging up the call. The source is your
friend (http://www.asterisk.org/doxygen/asterisk1.4/app__amd_8c.html):
00205 /* If we fail to read in a frame, that means they hung up */
00206
I have always used ooh323 between Avaya and Asterisk.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Tuesday, February 23, 2010 2:24 PM
To: 'Asterisk Users List'
Subject: [asterisk-users] Which H.323 to use in Ast
Hi All,
are you aware of any solution which can encrypt calls between a mobile gsm and
isdn (asterisk) ?
Thanks for your attention,
have a nice day.
Mike
--
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-- Bandwidth and Colocation Provided by
Ok so a while back I found an example for having a number dial multiple
numbers and then whoever answers and confirms gets the call. (don't
recall who the example was from, but thank you!)
But Now today I've been playing with TTS and STT and came across the
BackgroundDetect command. Now If I
Hello,
I have a setup that includes a cellphone a proxy running Kamailio and
rtpproxy and a SIP server (VoipSwitch/Asterisk). Call flow works well
while using Asterisk, however when VoipSwitch is used i find that the BYE
message from VoipSwitch has an RURI = acco...@voipswitch, so the proxy
ends
Could you share your config for the Asterisk and Avaya side too? Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie A.
Stapleton
Sent: Wednesday, February 24, 2010 3:37 PM
To: Asterisk Users List
I changed my VOIP, and now things are ok.
But didnt understand, how can VOIP can affect it ?
On Wed, Feb 24, 2010 at 11:53 PM, David @ULC ucoms2...@gmail.com wrote:
*Code:*
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback(Local/91441425477...@default-b9f2,1,
I'm having a problem with conferences both meetme and app_conference,
though I've done most of the testing with meetme.
Essentially, I get little gaps in the audio - usually fewer than a dozen
or so samples, though it does vary. They seem to occur at random, but I
usually get one ever few
Hi people,
I'm having problems of connection with a Linksys SPA IP PHONE 942 when I use
the WAN port, most of the time when I try to connect to the network or restart
the IP Phone I can't get internet connectivity . I tried using both static IP
and DHCP, but the problem is the same. Some have
How did you connect your phone to the network ? Please describe your connection.
Jimmy
-Original Message-From: renato...@yahoo.com.brSent: Wed, 24 Feb 2010 15:51:35 -0800 (PST)To: asterisk-users@lists.digium.comSubject: [asterisk-users] Problems with Linksys IP Phone SPA 942
Hi
Jonathan-
I'm having a problem with conferences both meetme and app_conference,
though I've done most of the testing with meetme.
Essentially, I get little gaps in the audio - usually fewer than a dozen
or so samples, though it does vary. They seem to occur at random, but I
usually get one
Sent: Wednesday, February 24, 2010 5:52 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problems with Linksys IP Phone SPA 942
Hi people,
I'm having problems of connection with a Linksys SPA IP PHONE 942 when I
use the WAN port, most of the time when I try to connect to the
hello,all
there is a AudioCodes Mediant 2000 out there. i want to realise ip to
PSTN and PSTN to ip connection.
after some configuration on AudioCodes Mediant 2000, PSTN to ip
connecttion works.
next ,i want to dial from asterisk to PSTN now. i have see the sample
in the extensions.conf
On Wednesday 24 February 2010 19:23:01 Shanon Swafford wrote:
One thing made be feel dumb the other day. We boot these phones 10 at a
time and a new guy had accidentally plugged in one of them using the LAN
port. This caused all sorts of problems in the network for some reason.
After chasing
Hi Kevin,
Thx for your kind response. Is there any options/steps that I could trigger
to skip from redirecting the media back to Asterisk?
Regards,
Lawrence
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I'm having a problem with conferences both meetme and app_conference,
though I've done most of the testing with meetme.
What version of DAHDI are you running?
--
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-- Bandwidth and Colocation Provided by
Lawrence Na (my-...@vyke) wrote:
Thx for your kind response. Is there any options/steps that I could
trigger to skip from redirecting the media back to Asterisk?
If your mail client allows, please *reply* to messages in a thread,
rather than starting a new thread with the same subject. This
Hello,
I have setup the n-way conferencing with Asterisk and it's working when I use
with my budgetone 100 phone but it doesn't work for any of the voip software or
other ATA that I have. When I turned the debug on, I see that the correct keys
(*0) were entered but asterisk doesn't detect the
Hello,
Is it possible use asterisk curl function with ssl sertificate?
Thanks
--
Best Regards,
Giedrius
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