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Job Description: Asterisk MySQL Support Engineer Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD

Hi, I am using asterisk 1.4.18. I am using it for inbound only call center. The SIP phones are X-Lite. Right now when a call is proxied by Asterisk to X-Lite the agent only sees asterisk written on its CLI screen. I want the agents to be able to view the callees number. Is there any way to make this happen. Regards Syed Nasruddin -------------- next part -------------- An HTML

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List users, Did anyone else recently receive spam from DIDForSale with the subject "DIDForSale 2012 achievements"? I suspect that they are using this list to harvest email addresses and think they should be called out on this poor business practice if that is the case. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer

Hello all Asterisk Users, This is my first post here. We are in a process of moving Dialogic 240/JCT-T1 from old voicemail server to Asterisk box. Which card drivers do we need? Please share experience if anyone have successfully configured Dialogic JCT-T1 card with asterisk? Only source proves that this card work with * http://lists.digium.com/pipermail/asterisk-dev/2003-April/000244.html

We are in the process of preparing to move our Asterisk server to a Digital T1 interface card instead of a analog card (via an Adtran which is now connected to the T1). I did a preliminary test the other day and hooked the T1 line up to the T1 card, bypassing the Adtran. This worked rather well I must say. The two issues I ran into are: 1) Caller ID is not working even though I enabled

...someone hijacks my auth credentials it may be a billing cycle or 2 before I figure it out. ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of BJ Weschke Sent: Wednesday, September 14, 2005 12:50 AM To: C. Savinovich; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anyone knows how to receive a SIP call withoutregistering gateway? What they're asking you to do is quite insecure to be doing over public IP. At the very least, you should confirm that there is a static...

Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don't want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n

Hello everyone, Even though a lot of it was a bit last minute, several of us from the commnunity made it to Baltimore to help Digium with their booth at ISPCon. It was a great time. Gregory Boehnlein, Brian Capouch, Christian Savinovich, Kristian Kielhofner (me), and John Todd (not pictured) were there (as well as others), and some pictures were taken (the up close ones of me were very unfortunate) :). I was there demoing AstLinux on the Soekris Net4801, and Brian Capouch was showing off the WRT54GS running Asterisk (as well...

...ied to increase udp buffer (sysctl -w net.core.rmem_max=8388608) , but seems the problem still persists. Also , here a screenshot of a typical dump from network interface, you can clearly see what's going on. http://img7.imageshack.us/img7/6578/sip.png Thank in advanced , Nir. *C. Savinovich*** did you isolated the issue? , checked firewall , interface errors , routing , sniffed the interface?. also , why using h323 and not IAX2 ? *From:* asterisk-users-bounces at lists.digium.com [mailto: asterisk-users-bounces at lists.digium.com] *On Behalf Of *C. Savinovich *Sent:*...

Hello everyone, I am pulling my hair here because a carrier threw me curve early today. They want to send calls to my asterisk server using SIP. Then they said that their gateways don't have to register with my server, that all they have to do is send a prefix for validation. Whereas I can think of several ways to authenticate their incoming number string, I am only used to the orthodox

Hi, What does retransmitting NAT means? I have a client that uses SPA 942, and his phone sometimes cannot be called. i did a sip sebug and i keep on seeing retransmitting NAT. on the realtime it shows that it is registered, so when i try to call it , asterisk thinks it is still online so it tries to reach it instead of saying it's unavailable, [Oct 9 11:10:33] -- Called 103100 it

Howdy, I am proud to announce that after a good bit of development, community feedback, testing, and code review, the brand new ConfBridge application has been officially merged into Asterisk Trunk!!! http://svnview.digium.com/svn/asterisk?view=revision&revision=314598 If you are already familiar with ConfBridge from Asterisk 1.6.X and 1.8, forget everything you know. This is a completely

Hello Everyone I just bough my Xten Eyebeam but i don figure out how to make the video works i only see a black screen where de remote video suposse to appear, Any help regarding this matter will be very preciated Thank You

I have an Asterisknow.org CD. When I boot up, it seems ready for me to choose update, console, etc. I'm assuming I need to do something at the CLI prompt. Is there a tutorial that would take me from loading CD to making first test call? Computer is Dell Optiplex GX260 50GB free disk space 1.5GB RAM P4 processor external mic speakers Skype is on board, and would be good to use it, if

..."src" field set (Reported by Gregory Massel) * ASTERISK-25592 - chan_unistim: Clang Warning: variable sized type not at end of a struct (Reported by Alexander Traud) * ASTERISK-28488 - pjsip mwi: n+1 sip notify's sent on re-register (Reported by Chris Savinovich) * ASTERISK-28509 - PJSIP cnonce generated on Linux contains 36 characters, NEC only supports up to 32 characters (Reported by Dan Cropp) * ASTERISK-28505 - app_voicemail/IMAP: segfault in leave_voicemail because not checking mailstream (Reported by Alexei Grad...

Dear all, I've implemented an Asterisk 1.4 with SIP service for voip and video. So I can establish a voip + video connection *one-to-one* only....it works OK. But I'd like to implement a videoconference *one-to-many* in order to intercommunicate many clients, is it possible with Asterisk 1.4 ??? (multicast is better than brodcast in this situation of course) Thanks a lot, Alejandro

..."src" field set (Reported by Gregory Massel) * ASTERISK-25592 - chan_unistim: Clang Warning: variable sized type not at end of a struct (Reported by Alexander Traud) * ASTERISK-28488 - pjsip mwi: n+1 sip notify's sent on re-register (Reported by Chris Savinovich) * ASTERISK-28509 - PJSIP cnonce generated on Linux contains 36 characters, NEC only supports up to 32 characters (Reported by Dan Cropp) * ASTERISK-28505 - app_voicemail/IMAP: segfault in leave_voicemail because not checking mailstream (Reported by Alexei Grad...

Hello list, I am glad to announce that XC-AST version 0.9.0 is out today. New functionalities include: * Though not yet available to the end user, this release inclued the basis of the Outbounds Call Manager that will be released for 1.0. If you update from a previous version, have a look at the UPDATING.txt to understand how to upgrade your database schema. * The realtime visualization

Hello, I've been looking at some SIP packet dumps captured with tcpdump on the PBX itself, and analyzed with Wireshark 0.99.4. I'm noticing something strange, at least to me. All of the SIP packets going out from our Asterisk PBX to either of our 2 VoIP providers are consistently 50% out of order. In addition, if I use Wireshark's voip call player, the outgoing side of the call