similar to: zaptel.conf multiple devices

Hi, all I am still battling to connect * and voipbuster. What protocol does it use? Ethereal capture shows UDP traffic, but no SIP or IAX traffic when using their client. VoipBuster client connects to connectionserver.voipbuster.com on port 11112 for authentication. Call itself is placed on different server. I have tried to connect using SIP and IAX and it seems that no authentication is

Hi, I have two asterisk servers with the same version of 1.4.29.1. The first server named it as MYE1. MYE1 is an incoming server that can accept incoming calls from PSTN(ZAP E1). The second server is a pbx functions server and named it as MYPBX(SIP). The sip.conf of MYE1 likes below: [MYPBX] type=peer host=mypbx.abc.com nat=no disallow=all allow=g729 canreinvite=yes qualify=no context=default

Hello, I have been trying to get a TrippLite Internet Office 750 UPS to talk to my Linux PBX for a couple of evenings now and I'm getting nowhere... I tried searching the list archives before posting here (I'm sure I'm not the first one to try to get this going) but they seem to be offline... The UPS is unfortunately USB based but I thought I'd give it a try anyway. Here is

Hi, I am looking into advising a client on the pro's and cons of using Installing asterisk on a server vs appliance(e.g digium mypbx). the appliance seems cheaper initially.

Hi All, I am using asterisk 1.4 branch on server. Here is a my dialplan. i have set the incoming route to incoming context, and then i have set dial with local channel, The call comes to my server and the call is routed to matched case, so my phone 1001 ring for 30 seconds. If i got the NOANSWER then the channel is not passing to next priority. I need to pass that channel to the next priority of

Hello my friends, I'm very worry about a problem i'm having...my asterisk got freez some times, every 5 or 6 days with NO trace in /var/log/asterisk/messages What i want to know is if safe_asterisk has something to be with this? This is what i have on my server: [root at mypbx ~]# ps -A | grep asterisk 9118 ? 00:01:30 asterisk [root at dreampbx ~]# ps aux | grep asterisk root

Hello my friends... Currently we are using the following firmware versions on ours aastra 55i: Firmware Information Attribute Value Firmware Version 2.1.0.2145 Firmware Release Code SIP Boot Version 2.0.1.1055 Date/Time Jun 20 2007 06:20:29 Can we make a firmware upgrade to the latest one: 6755i (55i) SIP, V2.5.3.18, January 2010 , English , ZIP , 2,849 KB on the site:

Why do attacks from the Internet get shown in the Asterisk logs with myAsteriskServerIP instead of the attacker's IP?! Really useful for blocking them, that is... Example: [Mar 6 00:00:00] NOTICE[1926] chan_sip.c: Failed to authenticate user 5550000<sip:5550000 at myAsteriskServerIP>;tag=ab8537ae (I replaced our IP address with myAsteriskServerIP. The attacks are not coming from

Ronald Wiplinger wrote: > Steve Totaro wrote: >> Ronald Wiplinger wrote: >>> kevin ling wrote: >>>> Hi, >>>> >>>> Check the vm_general.inc file >>>> >>>> >>> Where should this file be? >>> >>> >>> bye >>> >>> Ronald Wiplinger >>> >>> >> You

Hello I'm trying for several days now to get ICE support for my Asterisk 11.23 on CentOS 6. My call setup : sipml5_webRTC (nat) --> public Asterisk on 178.18.90.230 --> softphone Zoiper (problem : no audio) Reverse does not work either. (problem : failed get local SDP) I followed this guide : https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5

Hi, I have a list myList (see below) that consists of id's $'7',....,$'10' and each id has an individual array, the length of which can vary from id to id. How can I convert this list into an array, ie. stacking the individual arrays into one large array? Thanks Zava myList: $`7` [1] 20050201 20050301 20050401 20050501 $`8` [1] 20050201 20050301 20050401

Hello Using Asterisk 12.8.2. I now have the "via ICE" messages in the RTP debug (see below). If you look in the SIP debug (see below), you also now see the "ice-ufrag" and "ice-pwd" in the 200 OK SIP-message from Asterisk to the webRTC client. But still no audio ! None at all ! In both directions. You can see in the SIP debug that the IP-address in de

Hello thank you for your answer. I don't understand how there are many tutorials and examples on the web where every time the outcome is a working setup. Very strange I feel now after my personal experience with Asterisk 11 and webRTC. You also say Asterisk 13. How about Asterisk 12 then ?? Kind regards. On 10-08-16 21:53, Matt Fredrickson wrote: > I don't see an ice-ufrag or

Hi, I'm banging my head on compiling bristuff modules for Suse 10.0 with kernel : Linux laps1 2.6.13-15.7-smp #1 SMP Tue Nov 29 14:32:29 UTC 2005 x86_64 x86_64 x86_64 GNU/Linux and Asterisk 1.2.1-BRIstuffed-0.3.0-PRE-1f . I get this : laps1:~/Voipy/1.2.1/bristuff-0.3.0-PRE-1f/zaphfc # make load make -C /usr/src/linux-2.6 SUBDIRS=/root/Voipy/1.2.1/bristuff-0.3.0-PRE-1f/zaphfc

I'm using the default mp3 files that ship with Asterisk: fpm-calm-river.mp3 fpm-world-mix.mp3 If they were variable bit rate, I think I would see a warning about 'varibel' or similar...is anyone else able to get these files to work? -Mike- ________________________________ From: Sander [mailto:crombeen@rommelweb.nl] Sent: Tuesday, May 17, 2005 6:34 PM To: 'Asterisk

Thanks Zeeshan, SAngoma told me that the asterisk problem is unrelated to wanpipe drivers, they told me to reinstall asterisk again But, i still having doubts about the problem :( Thanks in advance > > Message: 10 > Date: Thu, 18 Mar 2010 00:21:06 -0400 > From: Zeeshan Zakaria <zishanov at gmail.com> > Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE >

I'm trying to load ztdummy on my Asterisk, running in a XEN domain. I've modified the code to disable the use of an RTC. I can load the zaptel module just fine, the ztdummy also loads without problem. But when running ztcfg I get this error. ----- s n i p ----- graham:~# ztcfg -vvvv Zaptel Version: 1.4.6 Echo Canceller: MG2 Configuration ====================== SPAN 1: CCS/ AMI

Hello all. Since half december we are trying to implement * as our primary PBX. We had a test machine running with 2 sip phones and a single ISDN card. Since this was working ok, we installed * on our main Debian server. Some specs: P4 2.8 Ghz HyperThreading 512 MB RAM Debian Sid updated every week. Linux 2.6.9 vanilla kernel (not the debian package) 1 Intel Pro 100 nic for internal network 2

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-15" http-equiv="Content-Type"> <title></title> </head> <body bgcolor="#ffffff" text="#000000"> <span class="postbody">Hello! <br> <br> I have a

Hi, We are looking towards a 8+ CO line setup (20 extensions) in our office but we do not want an IVR(auto-attendant) feature. All incoming will be answered by a receptionist. I have read the multi-line configuration for cisco 7960 thread in this list but that way I believe we could only display 6 incoming lines. What will happen to the rest? Does the expansion module for the cisco 7960 work