similar to: Directory App() is running for a while, like blocked/freeze? in the same name...

[macro-hangupcall] exten => s,1,ResetCDR(w) exten => s,2,NoCDR() exten => s,3,Wait(5) exten => s,4,Hangup Hi all, currently i've been getting troubles with SIpphone Sjphone and Xlite seems also to get delay but no crash on hanging. I found that Ast@home is executing this Wait(5) and it seems to me that Sjphone is giving timeout error because of it... Why is this 5 seconnds? any

Hi guys, I've been looking on wiki, but i could find it only for chan_capi: http://www.voip-info.org/wiki/view/Asterisk+PBX+functions In the CAPI channel See Asterisk CAPI channels * Call Deflection (CD) (redirect without answering): Implemented by chan_capi How can i do it with my Softphone Xlite? Any one can help me? I want to redirect a call without answering it. Best regards,

Hi all, I've asterisk 1.2.5 , and what is happening is this: Sip user agent "A" calls a pstn "phone B" Sip User agent Activates MOH. "B" starts listening. "A" doesn't hangup and just Disconnect Sipoftphone XLITE (exit) "B" stills listenning Music on Hold and "A" has left Asterisk, who hangs the call? only when B hangs...

Hi all Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: http://bugs.digium.com/view.php?id=5884 Haven't tried it out yet. Benoit Panizzon -- I m p r o W a r e A G - System Services ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz

Hi, Is there any softphone or TAPI plug-in that allows one to dial from a web page? As you may know, Skype has a mechanism that converts phone numbers on a web page to a click-to-dial application. I'd like to use this but on a normal softphone (Bria, Xlite, other). Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I use the latest Codeweavers WINE and several DOS programs wouldn't run correctly. Where can I find the best DOS emulator for Linux? Thanks!

Hi there; I have samba-2.2.7a installed as PDC for Windows2K, XP, 98 (have not tested yet. It appears to work with 2K and XP. However, I have a few problems and I'd appreciate your comments. 1) net time \\linux /yes takes 30-40second on both 2K and XP 2) logging in on the server gives me the following errors: unix_clean_name [/wshENU.DLL] unix_clean_name [wshENU.DLL]

---------- Forwarded message ---------- From: Marco Mouta <marco.mouta@gmail.com> Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: asterisk-users-request@lists.digium.com Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope that Asterisk experient users could give me

Hi, All, I am implementing real-time Rsync on Windows 2008 system. I set up Rsync server and Rsync client on two machines. An windows service is watching all the Windows file events with FileSystemWatcher. However, the service cannot tell the exactly what happened to folders such as create, delete, or modified. So, I ignored folder event, and only catch file changing events. After I catch

Dear all, I have tried to run an asterisk instance together with XLite on a single machine (a PowerBook). The intent is to take advantage of IAX connections to easily cross NATs while traveling. While the IAX setup proved 'easy', just having to fiddle a little with working configs at both sides, I did not succeed so far in getting XLite to connect to the local Asterisk server, AND be

Hi, I have an xlite registered with asterisk server. When i dial a number AGI is invoked. and in this we are running to threads one to record files and one to play files. So i dialed the extension and i started recording and playing at the same time. On the xlite i m getting an indication when recording my voice and at the same time i could see playing the other file too. But in the directory

My wife and daughter, and to lesser extent myself and my daughters boyfriend would like a communications system which allowed us to talk to each other, both on a one on one basis, but also occassionally in conference. My wife and I live in a house with an internal LAN with each of us with a desktop machine (hers in Windows XP, mine runs Linux) and a Linux server acting as firewall and NAT

Hi guys, This should be has an easy answer for you, my users are complaining that when they press # and then ear gorgeous Allison "Transfer" the timeout is very small, they must enter immediatly the extension to transfer the call. Is it possible to change this? ;transferdigittimeout => 3 ; Number of seconds to wait between digits when transfering a call This is timeout

Boa tarde, Ap?s alguma experi?ncia com o Asterisk, e com muito ainda para aprender, gostaria de saber se h? algu?m nesta mailing list que pretenda criar um Asterisk Users Group para Portugal. Visto que acaba sempre por ser uma enorme aprendizagem ( valor acrescentado) a partilha de experi?ncias/problemas e solu??es nas implementa??es Asterisk. H? spre detalhes que variam entre os Telco's de

Hi all, I've the current scenario: User "A" - Zaptel call incoming in my Asterisk to my SIP user "B". "B" gets the Call. "A" says : "B" i would like to call PSTN user "C" "B" places a call to user "C" and asks if "C" wants the call from "A". "C" says yes i want, then B needs to

Hi guys, I've setup on box with a TE110P and time to time I need to access remote equipment outside of our office and use a data channel. I'm wondering if do I need to buy a POTS line only for this time to time acess or what's the easiest way to do that via my TE110P on asterisk box. I know that is possible data transmission with this Digium Card, I'm wondering how... Any tip any

Hi all, I've a Planet VIP-150T VoIP Hardphone connected to Asterisk. When I'm in a call and i press Hold button, the other party starts listening Music on Hold but then when i press the button again to get the call back it doesn't work! I've checked asterisk CLI: -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 --

Hi all I'm deploying a VoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions. On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application. I've several SIP extensions on this Asterisk box, and calls between Asterisk extensions and legacy PBX

Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine;

Hi, I've got an Asterisk box with grandstream and xlite clients on it. No here's the thing: - I grey out all the codecs on the Xlite except for GSM - I call the Grandstream from the Xlite, the Xlite uses the GSM codec and the Grandstream uses ulaw, with Asterisk doing the conversion, everything fine - I call the Xlite from the Grandstrea, the Xlite ends up using the ulaw codec as