similar to: no source on calllogs

I am looking for Video Conference Software (Open Source) , But but not for free Trial.. please give reference about it. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090427/cc9690ee/attachment.htm

Hello, a local prison contacted us regarding some calling card solution. they need 4 E1s to serve 120 rooms in that prison. we are planning on using 4 servers to serve the calls and one for the database servers' specifications are: 2.8 Dual Core Proccessors 2 GB Ram 160 Sata Drive each server will be provided with 1 E1 card Questions are: 1- will those servers be able to handle that ammount

Hi, I use realtime and I found that changing accountcode needed to restart asterisk to activate that code and shown in CDR. Does it has a way to update accountcode without restart asterisk? ango

Is it possible to make use of queues for incoming calls but to have agents that do not need to log in ? If I create a queue and make certain SIP-users member of the queue, do these SIP-users always need to log in to the queue to be able to receive calls that are in the queue ?? Can't a member be just available when the phone is registered to the Asterisk-server ? In stead of also having to

Hi I have On func_odbc [EXEC] readhandle=ressqlserver writehandle=ressqlserver readsql=${ARG1} writesql=${ARG1} I'm trying an update on dialplan: exten=> 141,3,Set(dummy=${ODBC_EXEC(UPDATE Tabla set campo = ${EXTEN})}) On Cli: WARNING[3579]: func_odbc.c:353 acf_odbc_read: Error -1 in FETCH [UPDATE Tabla set campo = 4356] Any idea why is this?? The query

I am testing Asterisk 1.4.22, zaptel 1.4.10.1 and libpri 1.4.4 on RHEL5 on DELL PE2950 and using ISDN-10. I thought about cutting over to production tonight when I noticed a serious problem. SIP calls are fine but if I dialed to outside (Dial(Zap/g1)) a few times or someone called in a few times, Asterisk just froze (cannot enter anything on the CLI console) and then even the machine had to be

Hi all, I have configured asterisk and a2billing.for inbound i have also configured did and its forwarded to sip extensions. But i want to enable queues with inbound numbers(DID).But i could not find a way to do this in a2billing. I want enable that if some did comes to asterisk/a2billing it should be forwarded to queues not sip extensions and their i want to enable hunting so if one

I recently did a set up where I replaced a simple D-link home router that was having trouble processing a T1's worth of bandwidth with a linux machine running iptables. the kernel was 2.6.29-r5 and I chose the SIP connection tracking modules from the menuconfig. Router worked fine for normal traffic, but I was unable to get the SIP phones to work. Using ngrep it was plain to see

Hello, I just tried to upgrade to 1.6.1.0 from 1.6.0.9 and i had problems in registering users. As i see from debug it successfully reads from users.conf but later,when a user tries to logon it say peer not found.... And there were an error msg about mysql about the username field..Smthing changed in mysql tables??? Now i downgraded to 1.6.0.9 again and everything is working..

Hi all, I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension on the other * I get a "Failed to

Hello, I've been tasked to look for ways to resell to others the service that one of a trunk provides.. In other words, i want to configure my current Asterisk (Ver. 1.4.26.1) with Freepbx 2.6.0 so i can act as a trunk to others.. I would provide an IP to them from one of my servers and they will use that IP to connect to me and i will connect them to my trunk/provider. If

Hello, for a long time i am using asterisk 1.6 with astgui. but for production system i intend to use asterisk 1.4 which i think might be more robust. And for a more developed service options i preferd to install with freepbx. But still there are big plusses and minusses for both system. My complain about astgui+1.6 was.. For example there were no backup trunk config running on that version.Even

Hi, i managed to connect to Ericsson MD110 with PRI at last. And made a successful call thru asterisk to ericsson. But when i try to call from ericsson to asterisk i got an error on asterisk side. And i couldnt figure out why. Here's my extensions.conf about incoming calls. [DID_span_1] include = DID_span_1_timeinterval_all,${timeinterval_all} DID_span_1_timeinterval_all] exten =

Hi, i know i am asking a lot of questions lately in this forum..sorry about that first of all. :) Ok, here is the deal.. I am trying to make a hybrid system with an ericsson MD110 and asterisk. Internally we have 4 digit phone extensions on ericsson.. and so in asterisk. So, what i want to do is to call pbx side without adding 9 or etc to the begining of the number from asterisk clients.. For

Hello, I have a default installation of asterisk 1.6.1.9-2 When i create a user in users.conf via asterisk-gui, calls, voicemail etc works. But if i create a user realtime (and my realtime caching is available too) i can see the realtime user with sip show peers. But, my local dial rules does not work. I can call from realtime user to static users(the ones in users.conf) and if they are not

Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Regards -------------- next part

Hi, In our office, we're slowly migrating from a cisco call manager set up to asterisk. Problem is management doesn't want to buy any other hardware ?as they had already invested a lot in cisco. The main cause of this is asterisk's added features like unique FAX number for everyone in the company (which will be the same as phone DID), Voice mail, Auto Answer etc yet we need thousands

Greetings List. we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following. 1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server. 2- Internet

Greetings, Because of the heavy load and the high expectations of an asterisk server offered as a solution integrated with our CRM software.. we were looking into other possibilities than software Licenses for G729 and G723 codecs.. to lower the pressure on the processor giving it more space to do more work. We heard of a hardware (PCI CARDS) can be used with Asterisk that does the work. And we

Hi, Looking for some reliable and quality providers of USA DIDs. Any pointers ? Thx Sans -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100623/816aecdd/attachment.htm