similar to: RE : Re: differential billing

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

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>

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

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

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

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.

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!

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 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

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

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:

Hi All, I have a quick question with regards the pbx_lua module. Would the lua dialplan have direct access to the odbc configuration within Asterisk, those odbc connections/dsn's that are defined in res_odbc.conf/extconfig.conf/cdr.conf? Thanks Bruce

Hi, The subject says it all. Is it possible to put the IP address of the peer in the CDR records? Using AGI maybe? -- Kind regards, Signet bv Remco Bressers T 040 - 707 4 907 F 040 - 707 4 909 E rbressers at signet.nl

Hi all, Trying to find where i got wrong in my config.... Is the "realm" parameter in sip.conf only used for possible autentication? The thing is, i got my box more-or-less working as i wanted, but i can only reach internal functions (like echo-test and so on) and other sip-clients if i dial "1234 at fqdn", while i was expected to be able to just dial "1234" I

Hello everyone. I need a quick help on how to capture who answered the call with agi. Here is an example: -- Zap/32-1 is ringing -- Zap/33-1 is ringing -- Zap/34-1 is ringing -- Zap/35-1 is ringing -- SIP/operator1-e77f answered Zap/23-1 So how can I capture this value and write it to mysql? I already have this:

I need graph the utilization of my t1s. Does anyone know of a plug-in, code, or web interface I can use to help do this. I am currently using Asterisk 1.4 Tony -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100726/c1463f31/attachment.htm

How should I configure my asterisk server so that I can receive calls from an unregistered peer from whom I also receive registrations of sip phones? I'm asking you this, because with my actual configuration, when I register a contact from that peer's IP, no more inbound calls are accepted from that peer, as my asterisk rejects those INVITEs with "407 Proxy Authentication

Hello guys, I need to be able to throw cdrs on more than one servers at a time. Please let me know how this can be done. Thanks

Hello, trying to load ael module in asterisk ver 1.6.2 got the following: asterisk*CLI> module load pbx_ael.so Unable to load module pbx_ael.so Command 'module load pbx_ael.so' failed. [Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined symbol: ast_compile_ael2 [Feb 18 11:25:47]

1.8 or higher. Att, H?lvio Junior SafeId - Gest?o de identidades e Acessos +55 41 | 9893-2694, single-sign-on.com.br helvio.junior at safetrend.com.br On 29/06/2015 14:43, Abdul Basit wrote: > Hi Helviom > > I am interested to evaluate your product. > > What asterisk version you build this product around? > > -- > regards, > > abdul basit | p: +92 32 1416 4196 | o: