search for: evaristesys

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

...both of these conceptual and product universes. Toward that end, I am hosting a new mailing list with this succinct purpose, if slightly unwieldy name, and encourage all interested to join. It is called 'SER-Asterisk-Interwork' and can be accessed for subscription here: http://lists.evaristesys.com/mailman/listinfo/ser-asterisk-interwork The archives are available here: http://lists.evaristesys.com/pipermail/ser-asterisk-interwork/ You can post to the list at: ser-asterisk-interwork at lists.evaristesys.com It's the same GNU Mailman stuff you are already used to. While it could...

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 the community. If you get a chance and take a look, I would appreciate it. Thanks! -- Alex Balasho...

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

...ome monthly price. > > Is it interested for you? > > -- > Thanks, > Pavel Ismailov > skype: pavel.ismailov > www.flashphoner.com > -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

...s in place, but Asterisk never seems to compile with MeetMe() application support enabled, nor does there appear to be a module I am failing to load that would contain this application. Is there something really obvious I am missing? Thanks, -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

Just an FYI for Digium. I received a mailing today from you guys which was nice. The price of mailing was ~$1.60 and inside was an inflatable beach ball. Cool, but I tried to blow up the beach ball and the the seam where the part opens to inflate the ball was not connected to the ball whatsoever, so it went right in the trash. I wonder if the sick heat had anything to do with it, was mine just

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

...--- > An HTML attachment was scrubbed... > URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081121/9a8636b6/attachment-0001.htm > > ------------------------------ > > Message: 9 > Date: Fri, 21 Nov 2008 09:46:13 -0500 > From: Alex Balashov <abalashov at evaristesys.com> > Subject: Re: [asterisk-users] Large Asterisk installations (~10, 000 > extensions), preferably at universities > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <4926C9B5.8080902 at evar...

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

...To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <BANLkTimxrxPaA+A1ocdWm8yP-6x+HHEcSg at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" 2011/6/20 Alex Balashov <abalashov at evaristesys.com> > On 06/20/2011 04:20 AM, Olivier wrote: > > What about incoming calls ? >> Do you have a way to have calls that normally comes from ITPS1 to >> comes from ITSP2 ? >> > > No, there is no BGP for the PSTN. > Yes, that's what I thou...

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

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

...ul so if there is some question that I can ask them to be able to figure out what is going on here let me know and I will ask the technicians from the ISP and post the responses here. Thanks once again for taking time out to help me. On Fri, Feb 13, 2009 at 3:30 AM, Alex Balashov <abalashov at evaristesys.com> wrote: > Oh--you mentioned in an earlier post that the Cisco switch was installed by > the ISP, so presumably that is something they consider their CPE as well. > > You can't rate-limit IP bandwidth on Layer 2 switches, and a Catalyst 2950 > does not have a Layer 3 featur...

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

So, is there anyone out there that provides rather generic but comprehensive CNAM-style directory services via SIP, to end-users? So I can put names to my calling numbers? Thanks! -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

...hey ever did before. I've been running 1.4.x for a long time. A packet capture reveals that no SIP INVITE goes to the junction_networks peer at all, even though it is available and qualified as reachable. Anyone know what gives? Cheers, -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671