similar to: How is this feature called ?

Hi to all, I was looking in google and also in this mailing list, but I dont find the solution to my problem, so I subscribe me to the list in order to post this e-mail and find the solution. This is the scenario: GSM Phone ----- GSM Network ---- TDM2406E --- ASterisk 1.4.0 (*) -------- VoIP Provider ------- Sip Phone or H323 Phone The problem is that I am generating calls from SIP and also

Hi the new asterisk 1.4 supports to store queue log information directly into a database? (like CDR) ? thanks

Does Asterisk supports 302 SIP message - "Moved temporarely"? I have found mail from Olle E. Johansson (April 2006) that Asterisk doesn't support redirects of registrations. Does it support now? -- Tomislav Parcina firstname.lastname@email.t-com.hr

How to execute some system command from AMI? -- Tomislav Parcina firstname.lastname@email.t-com.hr

I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P switch (with PoE functionality). I have tested three phone's, one is working (7905) and two aren't (7905 and 7940). I have plugged all three phones on same switch port with same cable! Do I need to change anything in phone configuration? Is there something wrong with Linksys switch? How can I troubleshoot

Where can I find alaw, ulaw, gsm, g729 formats for native music on hold? I have some mp3 files and I have tried to transcode them to above, but it seams that SOX can't do that. Please, tell me where to download some MOH files (in above formats) or how to transcode mp3? Thank you for your time! -- Tomislav Parcina tparcina#lama.hr

Hi there! I just try to play with die wake-up function described in http://www.voip-info.org/wiki-Asterisk+tips+wake-up Everything looks fine but there seem to be missing some soundfiles like "wakeup-menu". Where can I get these files in order to make this feature usable? Regards Julian Pawlowski

When agent tries to transfer a phone call (*2 - att transfer) he hangs up. Why? When a phone call isn't from queue then att transfer works fine. In features conf I have *1 for recording, *2 for att transfer and #1 for blind. In queue blind transfer works. For disconnect I have #0. I guess that * is somewhere defined as for hang-up the call, but where? I can't find it anywhere. Any help

Hi list! How to send a call directly to voicemail recording? When I put this exten => 313,n,VoiceMail,u221 Or this exten => 313,n,VoiceMail,b221 In my dial plan, calling person first hears greeting message (busy or unviable). I would like to avoid greeting message (I would play something with Playback application). Is it possible? -- Tomislav Parcina tparcina#lama.hr

The Asterisk and Zaptel development teams have released Zaptel version 1.2.16. In addition to minor bug fixes, this release fixes a build-time problem on systems where the default language is not English, and also corrects a regression in the driver for the Digium dual- and quad-span cards with hardware echo cancelers that could result in kernel panics. Thanks for your support of Asterisk and

The Asterisk and Zaptel development teams have released Zaptel version 1.2.16. In addition to minor bug fixes, this release fixes a build-time problem on systems where the default language is not English, and also corrects a regression in the driver for the Digium dual- and quad-span cards with hardware echo cancelers that could result in kernel panics. Thanks for your support of Asterisk and

Hi all, I've just been told from an ex workmate that my VSP (who I used to work for) has put an anti flooding limit of 80 SIP messages per IP per minute in place. I run the phone system for a facility that has a lot of extensions, but would rarely have more than 4 or 5 simultaneous external calls. Am I in danger of tripping over this limit? It sounds dangerously low to me.

I am use Fedora 3, and run into a 1.4 compile issue. When 'make install' I got this message. [root@asterix asterisk-1.4.1]# make install make: expand.c:489: allocated_variable_append: Assertion `current_variable_set_list->next != 0' failed. make: *** [utils] Aborted [root@asterix asterisk-1.4.1]#

Greetings to everybody. My question is that it?s impossible to pick up a call from ZAP, IAX or mISDN with my Ext Key of my GrandStream. It always give me a Spawn Message on CLI and a ?603? error on my LCD GrandStream. Exactly from my CLI screen i get this message -- Executing NoOp("SIP/11-096c2ac0", "Probando 1 ") in new stack -- Executing

Hello Im having strange asterisk ring. I'm dialing PSTN network, then I get my call answered and I hear a person talking but the same time remote person can't hear me. They get a ring tone. What can be the problem? Where do I need to look for it? have no clue. Running Asterisk 1.2.14 svn rev 48468 Voip gateway is Cisco5300 with IOS 12.3(9) Scheme: 192.168.1.201

please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP and try compile&run clients with srtp (linksys,gxp-2000, minisip, twikle, ...) --------------------------------------- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA - http://lcna.slu.cz =======================================

Hi, Does anyone know of a Video Camera that is based on SIP? There are lots of Video Phones out there, but I can't seem to find a Video Camera. Thanks in advance. -kokmeng.

Hello, is there someone who knows if I can use AOC for billing in Asterisk? I mean: let's say I have an external SIP device that produces AOC data. This device connects me to the telco network. Can Asterisk, if connected via SIP with this device, collect AOC data and put it in the CDR records? If not, which is the right way to use AOC for billing? Thanks a lot Stefano Corsi -- Stefano

Hi, I know this question doesn't exactly relate to the core of this list but I thought it does relate to its "hacker spirit". Is it possible to install a Cisco Call Manager 5.X on a non-Cisco appliance ? A friend of mine working for a Cisco VAR told me his colleagues couldn't make it, even for testing purpose. Do you agree ? Regards -------------- next part -------------- An

hello I want to install Asterisk just to use in my LAN, without a analog or digital devices. I need to install all this packages??? Asterisk 1.2.17 Zaptel 1.2.16 Libpri 1.2.4 Addons 1.2.5 Sounds 1.2.1 thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070328/86e4ad95/attachment.htm