similar to: Long duration calls with Asterisk out to VoIP telco

Anyone tried the digium TC400B transcoding card? What are your opinions? Thnx

Hi all, I need the zap channels going, but got the following error. What do I need to change in my configuration? Thnx. chan_zap.c: In function `zap_send_keypad_facility_exec': chan_zap.c:2309: warning: implicit declaration of function `pri_keypad_facility' chan_zap.c: In function `pri_dchannel': chan_zap.c:9292: structure has no member named `call' make[1]: *** [chan_zap.o]

My TE411p does not seem to like to have two PRIs from different telcos (span 1 and span 2). I can get one working, but not the other. However, if I use vpmsupport=0 when loading the wct4xxp module, they both work. But here is the problem, vpmsupport=0 disables the on board echo cancellation. Any ideas? BTW, here is zaptel.conf span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs bchan=1-23 dchan=24

Hi all, Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for "asterisk1/700" Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in

NVWhisper. Justin ------------------------------ Date: Thu, 08 Mar 2007 16:25:28 -0500 From: Wai Wu <wkwu@calltrol.com> Subject: [asterisk-users] Coaching in asterisk Is there a way to setup a conference where party A can coach another Party B, at the same time, all other parties cannot hear party A? In order words, partis A and B can hear every one, and party A can only be heard by

Hi everyone, I am running into wall today with simultaneous call limits. I have two Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a lot of sip calls from one machine to the other by issuing AMI Originate commands to one machine. The machine that makes calls plays a message (demo-intruct) upon the other machine answer. The machine receives the calls just waits for 40 seconds

Hi all, I am handed a project to setup *. The requirement is that it can handle 8 T1s. Half of the calls coming into the system will be routed to SIP extensions (with transcoding). The machine we have in our disposal is a new dual Xeon 3.2gHz server with 2g of ram and an dual 1000mb nic. Voice will be coming in from the PSTN (through 2 quad digium cards) in g711ulaw, and most of the time will

Hi all, Is there a way to connect two live calls through the manager api without directing them to a meeting room? Currently, I can connect them by sending them to a meeting room. However, I don't know what the overhead is, and I kind of think that if I can connect them or link them up, the overhead would be minimum. -------------- next part -------------- An HTML attachment was scrubbed...

Hello everybody, I am running 1.2.10-BRIstuffed-0.3.0-PRE-1s with florz-patches on Linux 2.6.16 with 4 HFC-Cards in TE-mode connected to 4 PtP ISDN-"Anlagenanschluesse". There are about 40 SIP-clients connected (mostly Sipura/Linksys PAP2, and some SNOMs and softclients) to this server. Everything works fine, except that my CDR reports some very long _concurrent_ calls from one sip

Can someone share pointers to Asterisk's Dialogic support? Which boards are supported, driver status, and etc. Thnx

Hi, To check telco billing, I'm usinfg Asterisk-Stats from http://www.areski.net/asterisk-stat-v2/about.php . How can you tweak this application to display graphics and data that use Billsec instead of Duration ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I am setting up a script to record all the call. There are two app for recording. "Monitor" and "Mixmonitor", one mixing the audio on the fly and one mixing it at the end but also allow a option not to mixing the audio at all. If mixing the audio on the fly is not that taxing on the CPU, I would like to use 'mixmonitor' app. My question is, what is penalty on

Hi all, Anyone know how to dynamically create meetme conference rooms on the flight? I remembered a while ago there was a switch that tell meetme to create the conference room is the room is not defined in the meetme.conf. It doen't seem to be working for me anymore. Thnx

Hi I'm having a problem with Asterisk 1.6.2.9 with the MySQL cdr addon. With some calls, the value in the `billsec` field in the CDR is exceeding the value in the `duration` field. I didn't think this was supposed to happen? Our old installation (some ancient version, sorry not available) always had `billsec` <= `duration`. The problem does not seem to be evenly distributed

\On 6-Jun-08, at 1:21 PM, Jon Kristensen wrote: > Do you know a way that I could get around this somehow? Maybe some > header data that I have missed? No, you didn't miss anything. Ogg is designed to always be streamable, so the only way to get the duration is to look at the end of the stream. > If not, what is the easiest way to check the numbers of samples? It's a bit

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Hello. I would like to know if somebody did a wireles voip with Asterisk PBX. I think to deploy a wireless for about 500 potential customers, it's a 3 km radius maximum coverage with houses without phone lines, I work for public places telephony small enterprises ( a common bussines in Spain) so I can get good rates from 4 telcos and do LCR at my asterisk PBX. Is anybody did this before

yeah that's what came up before when I asked the list about this a couple months ago. The concensus was that in the case of a lightning strike or what have you the 24 awg copper would immediately fry and would not transmit too much of the current sustained. of course my neat little trick here would be fried all to hell, but... -----Original Message----- From: James Harper

Hi, Anyone know how to compile asterisk for a hyperthreaded processor? Thnx