similar to: SIP Debugging to separate log file

Hi All; Anyone can advise for a good IP Phone that has the ability to support SIP firmware and IAX firmware? Ofcourse, SIP there is a lot, but we need also the ability to use IAX (as it is good for NAT). Any advise. Regards Bilal ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo!

Hi List; What the following mean: CONSOLE=Phone/phone0 CONSOLE=Console/dsp TRUNK=Zap/g2 I know SIP/John and Zap/1 but I do not know above (I do not know also the difference between Zap/2 and Zap/g2)? Any advise? Regards Bilal ____________________________________________________________________________________ Got a little couch potato? Check out fun summer activities for kids.

Hi List; I have this example for Macro and I am not able to understand some line, if any one can help me plz :)- [macro-voicemail] exten => s,1,Dial(${ARG1},20) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) exten => s-NOANSWER,2,Goto(incoming,s,1) exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN}) exten => s-BUSY,2,Goto(incoming,s,1) exten

Hi List; How can I configure asterisk to receive a call from SIP end point without being registered at asterisk and its IP address is dynamic, and authentication to be based on the username and password or any other string? I know that if I place the host with static IP then no need to register, but what if the voip gateway was having dynamic IP and I do not need to register on asterisk, but I

Hi List; [incoming] include => parkedcalls exten=103,1,Dial(SIP/Bob,,tT) exten=104,1,Dial(SIP/Charlie,,tT) When we use tT and when we use t alone or T alone, I know this for call parking, but I do not know what the tT does? Regards Bilal ____________________________________________________________________________________ Sucker-punch spam with award-winning protection. Try the free

Hello good ppl, Any way of stopping asterisk writing into syslogs or any other file, if I set verbose 6 on the console? All I want is the verbose output only on the console, nowhere else. My logger.conf says : console=> notice,error ;messages => notice,warning,error Thanks in advance. - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential

Hi List; If I am configuring Diguim Analoge card, then I need to run #modprobe wctdm, but the question why I need to run #modprobe zaptel also? What #modprobe zaptel does a things that #modprobe wctdm does not do? Any help? Regards Bilal ____________________________________________________________________________________ Looking for a deal? Find great prices on flights and hotels

Hi List; My Asterisk version is 1.4 and I am trying to use the ResponseTimeOut() application to control the response time of the Background function, but when the running arrive for the ResponseTimeOut() then the call drop and my debuging says: No Application 'ResponseTimeout' for extension (Test_Bilal,s,3) Spawn extension (Test_Bilal,s,3) exited non-zero on 'Zap/1-1' Hangup

Dear List; I am trying to create a link between Asterisk and My softswitch, the link to be SIP Trunk. I did the below configuration and I do not know if any one can help me and advise me to have better configuration to be sure that link is fine. But I do not know how to determine the SIP username to be sent for my softswitch as sometimes the softswitch need to check it. Also, does asterisk

Hi List; I am trying to use wiki via the link (http://www.voip-info.org/wiki/index.php?page=Asterisk) in effective way to find the needed resource for me, but still it is hard to arrive for the needed information. For example: what is the best (shortest) way to search for information related to the command playbak()? Using the backlines, it make the eyes feel hard by keep reading without

Hi All; How can I test manager.conf? Can I telnet to the asterisk machine at the port 5038 and send and receive commands to test if the manager is working fine? How? Regards Bilal

Hi All, I'm interested in how people are "clustering" Asterisk, if that's possible, or how you might be achieving a redundant solution. I've a single Asterisk server driving the company. Its well backed-up, and I've a cloned machine that (in theory) with a DNS change could take over operations. However I'd like to achieve something more automated if possible. I

Hi all, I have a problem with some asterisk boxes. I have a standard installation with 1.4.14 (I also test with 1.4.4) in core duo Mac Mini on Debian Etch. I use SJphone softphone, Linksys SPA921 or Thomson 2030 for phones. All my phones are in a LAN with good status of 2ms max. Randomly I have dropped calls during communication. No absolutetimeout or other calling limitation options. Any

I'm trying to see what users are sending for login. I added the rawlog binary in front of the imap-login in the config file. I'm not seeing any rawlogs. I added the dovecot.rawlog in the dovecot user directory (since that's what the imap-login process seems to run as) and it's permission are wide open. Still nothing. Help?

Hello Guys, After adding money into my sysmaster phone account I am able to make calls outside.thnx _____ From: Mani Nair [mailto:mnair at nvloisp.com] Sent: Friday, September 07, 2007 9:16 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Sysmaster and Asterisk Hello Guys, I am unable to make calls to outside number from some of my extensions.

Hi: I want to have conference call service and I use A102d sangoma's card.Do I should install ztdummy or app-conference? Best regards. --------------------------------- Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have an account with FWD and I have configured my SIP.conf with [fwd] type=friend secret=password username=901835 host=fwd.pulver.com But when I am trying to dial out my own DID , I dont see any call landing in asterisk. In extension.conf (vicidial) file I have exten => 2062036895 ,1,Ringing() exten => 2062036895 ,2,Wait(1) exten => 2062036895 ,3,Answer() exten => 2062036895

Hi all, I've found that the FLAC__FrameHeader struct's blocksize member has values limited to the range (0, FLAC__MAX_BLOCK_SIZE] where FLAC__MAX_BLOCK_SIZE is 65535. In the encoder, what determines the size of a block? Cheers, Erik -- ----------------------------------------------------------------- Erik de Castro Lopo -----------------------------------------------------------------

hi has the flac encoder multi-core support (so I mean using 2 or more cpu's simultaneously to encode 1 track at the time, not encoding 2 or more tracks in parallel )? And if yes, how to enable this? thx

On 9/26/07, Josh Coalson <xflac@yahoo.com> wrote: > --- Martin Leese <martin.leese@stanfordalumni.org> wrote: ... > > Where can I find more detail on what is a > > "non-audio" RIFF chunk? > > it is any riff chunk that is not "fmt " or "data" > > > Ambisonic ".amb" files are WAVE-EX files with > > a