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Hi all, Yes, this is not the right list for such a question and I am using Asterisk myself its for a friend who isn't used to Linux. You can write me off list if you want. He is looking for a Windows based PBX with same functionality as Asterisk. Any tips? Many thanks!

Hi, I have 8 E1 PRI Lines and i have 200 phone numbers and 200 channels (25 channels per PRI). is there a utility available in Asterisk to dial out 200 numbers and run a campaign for 200 numbers concurrently and play a mp3 file ? Please suggest/guide Regards Kaushal

Stop advertising. Le 26 sept. 2010 09:46, "Faisal Hanif" <faisal at vopium.com> a ?crit : Hi Abdul-Basit, If you need only different intervals of billing you can easily do it using any AGI as we are doing it in Perl AGIs using post call billing. But if you need realtime billing then the most stable and flexible option is to use FastAGI+ AMI. I have tested it in JAVA...

Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -------------- next

Hi guys, is there a way to see how many channels of an specific tecnology are being used? Like, i have a zap card, e1 (30 channels), and there are 10 channels being used at this moment. When the E1 reaches 15 busy channels I need to receive a call or something like this, telling me that 15 of 30 channels are busy. How can I do this? Thanks! -------------- next part -------------- An HTML

Please could someone advise good manual for using lua for asterisk dialplan. There is not much docu about it. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110218/42642787/attachment.htm>

Hello, I was wondering: What does Dial(Local/...) offer that a Goto() doesn't? For instance: ======== ;exten => h,n,Goto(callback,start) exten => h,n,Dial(Local/start at callback) [callback] exten => start,1,Verbose(In callback) ======== Thank you.

Anyone here who has configured zaptel/dahdi for a singtel E1 line? What are the settings for coding, framing, line type and switchtype? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110210/4288ed84/attachment.htm>

Hi Everyone Asterisk 1.4.33 is running with Sangoma/Dahdi for analogue lines to Bell Canada. User claims that call hangup without any interferance to the phone set. Is there ANYWAY to find out which party hang-up the call or if the call was cut-off due to other reasons? I checked the *"asteriskcdrb"* table and it's pretty much useless in this case as it only logs the duration and

Hi, I seem to be facing an intrusion issue, inspite of firewall (script attached). What am I missing ?? Any suggestions / recommendation are welcome pls. Best regards, Sans -------------- next part -------------- #!/bin/bash echo 0 > /proc/sys/net/ipv4/ip_forward # Clear any existing firewall stuff before we start /sbin/iptables --flush # As the default policies, drop all incoming

Hi, I have followed steps which were mentioned on forum and given below. Still couldn't get speex working. On test calls getting error "chan_sip.c: sip_call: No audio format found to offer." # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not

We just finished an upgrade of our Asterisk system to an HA environment on a pair of servers using Linux-HA. As part of the upgrade, we also moved to Asterisk version 1.8.4.3 Most things are working quite nicely on the new system. However, I?m having trouble getting a paging feature to work. In Asterisk 1.4, we simply used the Page() application like this:

Hello list, I need to know about Asterisk's friendly nature with RTCP. I've phones which support RTCP and they connect to the outer world via multiple carriers. In one of my recent packet traces I've observed that the caller initiated a call with rtcp string in SDP while for the same call dialling our from Asterisk to the carrier has no RTCP string in SDP ! Can anyone please tell why

Hello Team, I was looking for audio conferencing solution where i got Web-meetme. I had installed Asterisk 1.6.2.9 on Centos 5.4. Its perfecting working fine. I tried using Meetme even meetme app is working perfectly fine. I installed Webmeetme 4.0 and integrated with my asterisk. When i try to dial the conference number it take me to an IVR wherein it asks for the conference number. The time i

Hello, I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it in wav-audio at the Asterisk server. I found the loudness level of the recorded audio was too high comparing with the orginal audio. How can I ajust it, so that there will be no amplifier used for recording. Thanks a lot. best regards Felix -------------- next part -------------- An HTML attachment was

Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before

Hi, We are using Asterisk 1.6.2 and it is continually failing to resolve Verizon SRV and sending following message, WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com' DNS settings on OS level is working fine. Can anyone have an idea about it? Regards, Faisal Hanif