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Asterisk Users Mailing List - Non-Commercial Discussion
October 2010 - page 1
Alex — 128592942501 Oct 2010
Hello, The harddisk of my etch/bristuffed asterisk1.2 box finally died. I moved the cheap (1397:2bd0) HFC-S card to a squeeze host (i686) and built da...
Phibee Network Operation Center — 128593511901 Oct 2010
Hi Do you know if the number of extension are limited into Asterisk 1.6.1 whe we use Realtime and a Database MySQL ? Asterisk stop when a context with...
didier.cuffaut at neuf.fr — 128593788101 Oct 2010
Hello, Perhaps i'm wrong but i don't find a real documentation for cmd Bridge (i've take a look to the source code but i'm not a guru ...
sean darcy — 128595603701 Oct 2010
I'm having problems with DTMF on outgoing DAHDI. See https://issues.asterisk.org/view.php?id=18084 I've tested that the dtmf comes into * corr...
AMI Originate (3 Replies)
Dan Cropp — 128596942401 Oct 2010*
When calling Originate Action, it rings my phone. If I do not answer, I receive a Channel Event: Hangup, followed by receiving an OriginateResponse ev...
Danny Dias — 128598164702 Oct 2010*
Hello, We are having issues with a NEW Sangoma A108D: -- Executing [691918892 at pbx1:1] Dial("SIP/xtravoip200-009d24b0", "DAHDI/g0/691...
Asterisk/Realtime and MySQL (7 Replies)
Phibee Network Operation Center — 128605952802 Oct 2010*
Hi i am not a expert on Asterisk and search a lot of small information : I use Asterisk 1.6.1.4 with MySQL. That's work and in my extension.conf, ...
Flash WAV Player (1 Reply)
Dan Journo — 128611805703 Oct 2010*
Hello, Does anyone know of a Flash or Java player that can play WAV files created by Asterisk? Thanks Dan...
Andrew Thomas — 128611953703 Oct 2010*
What happens if you change to: signalling=bri_cpe_ptp -Original Message- From: Andrew Thomas [mailto:asterisk-users-bounces at lists.digium.com] On Be...
bruce — 128612325303 Oct 2010*
Hi Everyone I think PAP2T supports DynDNS and other Dynamic DNS providers. I have a box that needs to be secured at all times. Currently it's not ...
bruce — 128613918003 Oct 2010*
Hi Everyone, Like always, here are IPs from China that try to hack an Asterisk server. Can someone please explain what is happening or what the hacker...
Inter pbx communication via BRI (1 Reply)
Zakir Mahomedy — 128619349604 Oct 2010*
hi I am setting up a test lab using BRI to connect two pbxs together. I am new to BRI / ISDN. For now I am just need some advice on what hardware whic...
RE : Re: differential billing (1 Reply)
Grygoriy Dobrovolskyy — 128619945204 Oct 2010*
Stop advertising. Le 26 sept. 2010 09:46, "Faisal Hanif" a ?crit : Hi Abdul-Basit, If you need only different intervals of billing you can e...
Module reload (3 Replies)
Flavio Miranda — 128620141904 Oct 2010*
Hi all, Every time I reload my asterisk it fall down and the following message appear on log: parse error: No category context for line 7 of /etc/aste...
Zakir Mahomedy — 128621810704 Oct 2010
Hi ? I am playing around the idea of setting up an asterisk box on the public domain. This box will then be connected to various other sip providers f...
Maxim Dolgikh — 128622864104 Oct 2010
Hello, I have the same problem, you said that was a configuration problem. I play with configuration but can not get it working. Can you please show y...
other end hangup (1 Reply)
jagan thoutam — 128623058004 Oct 2010*
how can i disable other end hangup when i recive incomming call tfrom asterisk...
Phibee Network Operation Center — 128625048705 Oct 2010
Hi A question, i have upgraded a beta serveur from Asterisk 1.6.1 to 1.6.2 and now all SIP Relatime user are rejeted : [Oct 5 05:39:22] DEBUG[15081]: ...
DHAVAL INDRODIYA — 128627873005 Oct 2010*
hello List, i am in a situation where i cannot get cdr records for call originated from CLI , i am not able to get when i used application or extensio...
Ujjval Karihaloo — 128629185505 Oct 2010*
We have multiple entries like the one below in our users.conf file... where the username. Contact and secret changes for different customers and we re...
SIP flood attacK (6 Replies)
Greg Saunders — 128629601705 Oct 2010*
Hello all. I was recently the victim of a SIP flood attack. I'm wondering what is the best method to prevent such things in the future. Many thank...
Daniel Knoll — 128630905305 Oct 2010
Has anyone a solution for me - with "Meetme(,Ms)" asterisk plays "conf-invalid" if a room not exist - with "Meetme(123,Ms)...
minimum card for dahdi timing source ? (5 Replies)
mancyborg at gmail.com — 128631043505 Oct 2010*
Hi All, for a vicidial server which uses only voip, which is the minimum telephony card which would provide the required clock timing source for confe...
Flavio Miranda — 128632209405 Oct 2010
Hi there! I am trying to configure Web-meetme on Asterisk 1.6. I have followed the README and everything looks ok,therefore, when I try to open the we...
more condition check for gotoif (1 Reply)
Daniel Knoll — 128632760906 Oct 2010*
Hello, is it possible to check more than one condition for GOTOIF in the dialplan? Or is the normal way to cascade the diaplan each GOTOIF? The Backgr...
take input and store in variable (4 Replies)
Tom Lohmuller — 128633771006 Oct 2010*
I am using a context to change values in a DB. Currently in my context, I am passing it to exten => s,1,WaitExten(7) ; 7 seconds to input exten =...
Asterisk CDR Radius error (3 Replies)
bakko — 128634369306 Oct 2010*
Hello, I'm trying to configure Asterisk with Radius cdr support. Asterisk version 1.6.2.13 Server Radius: Freeradius version 1.X Radius client: ra...
Olivier — 128635110006 Oct 2010*
Hello, I'm looking for a virtualization technique with which I could easily assign PCI/PCIe boards to virtual machines. If this matters, I don...
MYSQL ADDON INSTALLATION ERROR (4 Replies)
Rizwan Hisham — 128636687806 Oct 2010*
Hi All, Please refresh my memory. I am trying to install asterisk after 2 years. I hav'nt used it since 2008 (version 1.4.2). Now I am trying to i...
Matteo Fortini — 128636797906 Oct 2010
Hi, if I Page more than one extension, then the MeetMe conference stays up even if all the called extensions aren't available or are hung up. Is t...
Cisco SIP 8.5 and 9.0 Issues (2 Replies)
Gerard — 128636979106 Oct 2010*
Hi list, I was wondering if anyone had any solution to either one of two issues I'm having: I have a cisco 7962G with the latest (from cisco) 8.5(...
Cisco 7941 localized menus with SIP firmware (11 Replies)
Olivier — 128637217706 Oct 2010*
Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to...
Adrian Marsh — 128637696206 Oct 2010*
Hi All, I've two asterisk servers on the same LAN, both 1.4, and I keep getting "Got SIP response 489 "Bad event" back from 192.168...
Zarko Zivanovic — 128638296006 Oct 2010*
Hello, I would need a little help about using 16 bit wav or mp3 files for moh on asterisk 1.2.x When i try to use these files as moh, the caller gets ...
Roger Burton West — 128638378406 Oct 2010*
I now have an OpenVox A400P and it is working well. Thanks to Ade Vickers for the recommendation, which I second. However, I need to make a slow trans...
Andy Graybeal — 128639631806 Oct 2010*
Greetings, I need a 2 way intercom for separate kitchens to communicate without having to walk back and forth. The speaker has to be loud but clear, n...
AMI connection limit (1 Reply)
Jerry Geis — 128639692706 Oct 2010*
Is there a limit to the AMI connections? I have upto 3 process that might connect and originate calls, 2 of those processes are very infrequent - howe...
Zeeshan Zakaria — 128639826806 Oct 2010
Hi list, A few times I have been asked if I could do encrypted VoIP development, for embedded systems, and in C++. And my answer has been in negative....
Alejandro Recarey — 128642693207 Oct 2010*
I just upgraded my asterisk box from 1.4 + Zaptel to 1.6 + DAHDI and services I was using perfectly before are suddenly broken. I have a DISA access c...
How to test BRI lines energy saving mode ? (2 Replies)
Olivier — 128643604107 Oct 2010*
Hello, If my understanding is correct, these days it seems that many ISDN BRI lines are configured in energy saving mode in which signalling D-channel...
Damian Turburville — 128644107507 Oct 2010
Anyone? - Forwarded Message Hi, I am hoping someone can help me with a problem I am having. I am trying to setup a connection from an Elastix 2 server...
Andrew Thomas — 128644600907 Oct 2010*
The D-channel isn't actually 'dropped' - it is put in to a 'power-save' state. See http://www.isdn-test.de/ihhe12.htm#Heading37 an...
Andrew Thomas — 128644904207 Oct 2010
Well, to go slightly O/T: If you read the issue tracker for 17270 - it appears to be a LibPri 'fault'. So I would say that the main work would...
Difference (9 Replies)
Rizwan Hisham — 128645479207 Oct 2010*
Hi All, Please can anyone tell me the difference between 1.4, 1.6 and 1.8 asterisk versions. Thanks...
Polycom: full caller ID? (1 Reply)
Ken D'Ambrosio — 128646259607 Oct 2010*
Hi, all. When I get calls on my SoundPoints, I only see the number -- is there a way to get the alpha portion of the CID, as well? Thanks! -Ken...
Harel Cohen — 128646470807 Oct 2010
Hi all. Is there a free, or at least non-expensive, solution that can convert g729A <-->g729B (with VAD)? The no-support for g729B on Asterisk g...
Olivier — 128646549407 Oct 2010
Hi, I'm facing the following request : "When someone is starting an assisted transfer using Asterisk's features codes, he will ear a prom...
SIP authentication - Thoughts please (2 Replies)
Steve Davies — 128646630407 Oct 2010*
Hi, We have a scenario where we need multiple discrete SIP trunks (peers) from/to a single endpoint. Because the authentication system starts by match...
RTP Read too short (1 Reply)
Bryant Zimmerman — 128646706807 Oct 2010*
Hi All In the console I am seeing warring rtp.c:1635 ast_rtp_read: RTP Read too short I get these all of the time things seem to be working fine but I...
AMI getting related channels in Ringing state (2 Replies)
Daniel Tryba — 128646983707 Oct 2010*
Issuing the AMI Status command results in a list of active channels. But how to figure out which channels are related before the call is answered? 2 c...