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Asterisk Users Mailing List - Non-Commercial Discussion
June 2010 - page 1
Mike — 127535615501 Jun 2010*
Hi, I have a test server with 2 NICs, each with it own IP address. Let`s say 192.168.1.2 and 192.168.1.3. I would like some phones to register by usin...
Best way to limit outgoing calls per trunk (13 Replies)
bruce — 127535936301 Jun 2010*
Hi Guys, I am looking to use System() function along with some bash scripting to determine if a Trunk is being used during certain time of the day or ...
Gareth Blades — 127538866101 Jun 2010
Does anyone know if ANI is supported in the standard version of libpri? We are currently running the latest asterisk 1.4 but with an older version of ...
Benny Amorsen — 127539975801 Jun 2010
Is it possible to disable silence suppression by adding silenceSupp:off to the SDP Asterisk transmits even when Asterisk is using internal timing? As ...
Jeff LaCoursiere — 127540602201 Jun 2010
Is this the same thread about having multiple ISP's, and you have external phones hitting the asterisk server on one or the other, and you want th...
Alexandre Rodrigues — 127541471201 Jun 2010
Hello all, My pbx server is connected to a sip gateway, when I call an originate command from the asterisk console, to establish a sip connection, the...
Mike — 127541635101 Jun 2010*
Hi, Reward offered: 50$ (paypal), and I am sure this is a ridiculous thing I have missing. My goal: On a 2 NIC Asterisk box, to send packets that came...
Asterisk and gtalk part 2 (1 Reply)
Julien Claassen — 127541717701 Jun 2010*
Hello everyone! So I've just scanned through the debug log, defined like this in logger.conf: full => notice,warning,error,debug,verbose I coul...
Asterisk Development Team — 127542470501 Jun 2010
The Asterisk Development Team has announced the release of Asterisk 1.4.32. This release is available for immediate download at http://downloads.aster...
Voicemail bug(?) with Asterisk 1.6.2.8-rc1 (4 Replies)
Julien Claassen — 127543040101 Jun 2010*
Hello everyone! I have a problem with my voicemail. When someone leaves a message - using googletalk at least - the message file starts silnet, stays ...
Definite app_jack trouble - unsolvable (4 Replies)
Julien Claassen — 127546457502 Jun 2010*
Greetings! I now found someone to test gtalk with and found out, that app_jack has a problem here. My voice gets transmitted fine, but I only get whit...
HElP me I am a beginner (2 Replies)
prashant shrestha — 127547023002 Jun 2010*
HY all,I am completely new to the asterisk so can any one help me with it as I have some questions queries 1. first n for most what are the tools/equi...
hugolivude — 127547659402 Jun 2010
Asterisk 1.6 CentOS 5.0 All - I'd like to offer my users the ability to hangup a call by pressing **. I'm using an attendant, so when ** is di...
James Puckett — 127549394802 Jun 2010
I've been struggling with a Trixbox running Asterisk 1.6 for one of our customers as of late. The service provider in question is using BroadWorks...
Asterisk 1.6.2.8 Now Available (2 Replies)
Asterisk Development Team — 127549612102 Jun 2010*
The Asterisk Development Team has announced the release of Asterisk 1.6.2.8. This release is available for immediate download at http://downloads.aste...
no sound between extensions (8 Replies)
Gary Baribault — 127549737202 Jun 2010*
Hello all, I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a Digium 8 port FXO card. The local network is 100Mbps Ethernet and my pho...
Time variables in system application (13 Replies)
khalid touati — 127550451202 Jun 2010*
Hi Guys, i have a weird thing here: when using time variables (%F & %T) in a shell script, out of dial plan (particularly system() app); it displa...
timeout problem with basic conf (1 Reply)
iscario at free.fr — 127550464102 Jun 2010*
Hi, I set up an asterisk server that i use with iax accounts. Everything is working fine, but, for personnal reason i need to insert a home made proxy...
Asterisk Development Team — 127551455202 Jun 2010
The Asterisk Development Team has announced the release of version 1.4.11.1 of libpri. This release is available for immediate download at http://down...
Jim Dickenson — 127552185102 Jun 2010
I have an asterisk system in Costa Rica that connects to a SIP provider in Atlanta. Sometimes SIP packets seem get dropped or retransmitted too quickl...
Necati Demir — 127556356103 Jun 2010
I am using DeadAGI script and using this context. exten => 10,1,Dial(SIP/${EXTEN}) exten => 10,n,Wait(1) exten => 10,n,Playback(${PLAYFILE}) ...
Sebastian Milioto — 127558637703 Jun 2010
Hi all, do you know any firmware release which fixes that issue for cisco ATA186? ATA 186 3.x.x Cisco ATA 186 v3.* CANCEL requests can be sent with a ...
Kenny Gryp — 127559040403 Jun 2010
Hi, I'm trying to get the match_auth_username=yes sip configuration working. It's mentioned as an experimental new feature of 1.6.2.x. (I...
Error compiling DAHDI... (1 Reply)
Warren Selby — 127560152603 Jun 2010*
The resolution [1] to this issue was to uninstall and reinstall [2] the kernel headers on the machine...just in case anyone else runs into this issue ...
James Lamanna — 127560601803 Jun 2010
Hi, I work for a small VoIP provider in Southern California. We are looking for someone very knowledgeable in Asterisk to help work on the following: ...
Codec G.129 A vs A/B (2 Replies)
Alejandro Cabrera Obed — 127561497804 Jun 2010*
Dear all, I've read that Asterisk supports only the G.729 A audio codec. I have several Grandstream IP phones with G.729 A/B codec implementation....
Richard Kenner — 127562184904 Jun 2010*
I did a usual "svn update", "./configure" and "make" and got [CC] chan_oss.c -> chan_oss.o gcc: @SDL_INCLUDE@: No suc...
DAHDI volume (7 Replies)
Greg Woods — 127564822304 Jun 2010*
Is there a reasonably easy way to increase the volume on a DAHDI channel? The VOIP phones in the house work OK, but for the phones connected to DAHDI ...
how to get call duration (7 Replies)
Necati Demir — 127564884104 Jun 2010*
Hello, I want to ask how to get call duration....
giovanni_re — 127564909404 Jun 2010
You're invited to join in with the friendly people at the BerkeleyTIP global meeting - newbie to Ph.D. - everyone is invited. Get a headset ...
Asterisk on Ubuntu (4 Replies)
Danny Dias — 127565620204 Jun 2010*
Hello Asterisk users, I'm having a little problem with an Asterisk installation on Ubuntu, i had installed many asterisks on CentOS but never in U...
Persuing the gtalk issue - not only jack-related (7 Replies)
Julien Claassen — 127567244004 Jun 2010*
Hello everyone! So I hacked app_jack.c today, as best I could. Whic came mostly down to inserting ast_log() messages. I discovered the following with ...
<UsingWaitorPlaybackinhextension@gmail.com>, (8 Replies)
Zeeshan Zakaria — 127567588104 Jun 2010*
Can somebody please confirm that Wait or Playback commands can't be used in h extension. This is for asterisk 1.4. Is there a way to delay the han...
1.6 issues (3 Replies)
Sascha Ferley — 127571592405 Jun 2010*
Hi, I am starting to notice some weird errors with DAHDI. dahdi: Disabled echo canceller NLP because of CED tx detected on channel 2 ... Anyone ever s...
originating a sip call from the CLI (1 Reply)
Julien Claassen — 127573442705 Jun 2010*
Hello again! I just got a SIP account and it seems - from a config on the net -, that I've configured it correctly. But I get no call to the outsi...
Press twice * (4 Replies)
Anahi Ludueña — 127574089105 Jun 2010*
Hi people, I need to detect when the user presses twice *... In the dialplan I added the following, but it doesn't work. Could you help me with th...
hugolivude — 127574104205 Jun 2010
Asterisk 1.6 CentOS 5.0 All - I'd like to offer my users the ability to hangup a call by pressing **. I'm using an attendant, so when ** is di...
hugolivude — 127574593405 Jun 2010
Asterisk 1.6 CentOS 5.0 All - I'd like to offer my users the ability to hangup a call by pressing **. I'm using an attendant, so when ** is di...
other codecs (1 Reply)
Hans Witvliet — 127574823205 Jun 2010*
Just curious, Any chance of using amr for asterisk? http://en.wikipedia.org/wiki/Adaptive_Multi-R... The codecs (both wb and nb) seems to be available...
problem with inserting records into cdr (6 Replies)
covici at ccs.covici.com — 127575472905 Jun 2010*
Hi. For several months now asterisk will mysteriously stop inserting records into cdr database. I am using mysql and the asterisk addons 1.6.2 to acco...
Richard Kenner — 127575502605 Jun 2010
I'm getting a crash relating to this field and I'm missing something. It seems to be initialized to zero, then used in memmove, then DECREMENT...
Edwin Quijada — 127575609105 Jun 2010
I installed a queue for a client with 10 officers so far so good. Now the client wants an agent when making a call to this will leave any customer inf...
Håkon Nessjøen — 127577509205 Jun 2010*
Hi, I'm now thinking of always dialing out to Local/xxx at outbound/n on all my queue members. The reason for this, is both to be able to limit th...
Problem with GROUP() (4 Replies)
Jonas Kellens — 127578685006 Jun 2010*
Hello list, using asterisk 1.4.30 and trying GROUP() and GROUP_COUNT() for the first time... Having some troubles. This the dialplan (using a sub) : e...
Tim Uckun — 127578800906 Jun 2010
I am trying to help a guy out with his Atcom IP04. He has set it up like this. He has a handful of IP phones all connecting via SIP. He has two phone ...
Assign dhadi channel to several groups (2 Replies)
Adolphe Cher-aime — 127584333506 Jun 2010*
Hello guys, I was wondering if it's possible to assign a dahdi channel to two diferent groups. Thanks Adolphe Cher-aime From my Iphone...
problem with port 5090 registration (3 Replies)
bruce — 127586285106 Jun 2010*
Hi Guys, I have tried every single rule I could into iptables but I can't register this VPS to a provider Spikko. Finally I did an iptable accept ...
How to play Floating point numbers? (4 Replies)
Faheem — 127591660407 Jun 2010*
Hi all,?Is there any way to play floating number using asterisk dialplan? Thanks,Faheem...
Still sipping frustration - only getting state ACK (13 Replies)
Julien Claassen — 127592396707 Jun 2010*
Hello everyone! I still am not much further along with my sip calling. I changed my sip.conf taking suggestions from the net (voip-info.org in particu...
Julien Claassen — 127592414107 Jun 2010
Hello everyone! So now I'm testing with chan_sip and I discovered, that I can make calls, even if they're only listed as active channels. But ...