similar to: OT Astricon/Digium Beach Ball Mailing

Hello all, Anyone expecialized with call center monitoring and reporting solution based on asterisk. A client of us, want to install a call center reporting solution for an asterisk server but I do not know which could be the best tool for that. I need a tool for reporting queue calls, agent calls, and disconnect cause. Any clue will be appreciated. Thanks in advance. VoipCrazy

Hi, I am requesting for a E1 connection from my telco. They are asking if I want DSS1 or SS7, and I am stuck here. Could someone tell me the difference between the two? How should I decide which one to use? Thanks in advance for your help. Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

Any know what Digium hasn't released the DS3 card? It was supposed to be out a while ago. -Matt

Hi All; I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile: Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server? Regards Bilal

Hi, I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk 1.4 + CDRTool with freeradius telephony system. Asterisk is used only for voice mail and redirectioning calls. Every calls should pass through mediaproxy so that i can account them. The goal was to create a simple prototype of what could be a VoIP provider. Now i need to dimensioning this system to work

Hi all, I am trying to send an extra header in SIP INVITE Message , i.e (email="me at me.com") but when I check the Message at the target that header is not there So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message? Thank u

I have a customer installation with an Adtran DS3 mux. The DS1's go into my Asterisk servers that run IVR/Call recorders. The DS3 provider is Qwest, and they tell me that they routinely drop the DS3 service to redundant back-up's and that this is a common practice that happens thousands of times to DS3 lines daily across the US without any service interruptions. They say that the

hello all, i have 2 asterisk boxes on that 1 have public IP Address and another is only have local IP address now on public IP there are some 7 DID forwarded , now i want to forward 3 DID out of 7 DID to local machine we called server B , I know there are DIal , and Switch statement in asterisk , but is there any other convenient way to do this. because if call ratio is high then my call legs

This is a global message to all to announce our callcenter / dialer / predictive dialer / vicidial program is now open. Codecs: G711, GSM, G729, G723 Protocols: SIP Duration Rate : 30/6 (6/6 with monthly minutes over 100,000) Channels : 100 to start with , more on demand. We are predictive dialer friendly , your account will not be shut off. Contact us to do a test run. Mike

http://www.theregister.co.uk/2008/12/01/richard_bennett_utorrent_udp/ FUD? Interesting? Boring? New news? Old news? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

Really? Me? Oh Pavel! I would be inestimably honoured. On 04/27/2012 01:55 AM, Pavel Ismailov wrote: > Hello! > > My name is Pavel Ismailov > and I`m CEO of www.flashphoner.com project. > > We noticed that you quite active in Asterisk-user > mail list, and would like to offer you buy signature > in your messages for some monthly price. > > Is it interested for

I have a Vitelity DID which generally works, but calls from a particular caller do not reach it. Vitelity has thus far disavowed any responsibility for working through this problem. I recognize that some action might be required by another provider which is outside Vitelity's control, but it seems that they should at least be trying to help resolve the problem by helping me determine

Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm

Is there a way to limit simultaneous calls. I like to limit simultaneous outgoing calls as more than few simulataneous calls are charged by my voip providers. However, I do not want to have any such restriction for internal calls. Thanks Jim

Hi, I have read about SS7 recently and learnt that it is a signalling protocol used in PSTN for call management, setup, etc. The thing that I don't understand is how SS7 plays a role in VOIP. When I make calls between landline and Asterisk via PSTN, I don't need to do anything with SS7. Is it because the SS7 signalling is already done by Asterisk already? From the prespective of

I made a little write-up that attempts to synthesise a lot of the information out there about how to get HylaFAX working with Asterisk by way of IAXmodem for inbound faxing: http://blog.evaristesys.com/?p=24 Of course, there are bound to be some things I've left out or are grossly in need of correction. So, before I link it off the voip-wiki I am extremely eager to solicit the input of

hi! First of all i want to tell i have a dedicated server on layeredtech with direct internet connection and i currently dont use iptables, so this is not about network configuration =). well so, i install asterisk-1.4.2 on my server, and next install asterisk-gui from the digium repository. next i go to: http://pbxa.com:8088/asterisk/static/config/cfgbasic.html and install a default

In reference to an old post from 2002: http://www.marko.net/asterisk/archives/0203/0103.html How does one go about doing this? Also, what is the present status of the OpenSS7 stack in Asterisk? What can it do now? And is there any possibility in the future of developing a DS3 card for it, if only for the purpose of mostly DACSing? Which is still a level of intelligent call control on the

Hi all, I am planning to implement LCR routing on my already running asterisk server. Uptill now i have found out that asterisk has no support for lcr, i have to do something about it myself, for example using the AGI. Im looking for ideas here. Whats the best way to start implementing lcr in asterisk. Should i use agi and start implementing my own lcr script or is there any plugin available which

Anyone have an idea how to implement a phone number that can only be called once? The first time it will process normally and any subsequent calls will be rejected.