search for: hakak

...060 ; Port to bind to disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=gsm [yairphone] type=friend insecure=no username=yairphone secret=yairphone host=dynamic dtmfmode=inband callerID = "Yair Hakak" nat=true extensions.conf [general] ; static=yes writeprotect=no [default] exten => _1NXXNXXXXXX,1,Dial(IAX2/myUserName@voicepulse/${EXTEN}@VPWS,20) exten => 8665,1,Dial(SIP/yairphone,20) iax.conf [general] port=5036 disallow=all allow=ulaw jitterbuffer=no [voicepulse] context = V...

hello all, i need some help and after trying the wiki i'm even more confused than i was. i'm trying to set up call forwarding and running into problems... i want the most basic call forwarding imaginable. 1. caller dials extension (say, 154) 2. dialplan is updated to forward caller's extension (based on CALLERIDNUM) to voicemail, instead of ringing his endpoint. 3. caller is

Hello all, I'm having a bit for a problem with the dial command limit option. I have the following dial command (executed from inside the a2billing agi) AGI Script Executing Application: (Dial) Options: ( IAX2/username@voipjet/18005551212|30|HL(60000:20000:00000)0) Now, from what i read in the wiki, this is supposed to limit me to one minute (60000 ms), and warn me when there are 20

We are using a platform from AmarFone Inc. It great full featured , everything you want to run a calling card and does not cost your a lot of money. Their support is awesome. You can contact them at sales@amarfone.com. Ehsanul Karim

hi all, you'll have to excuse the ignorance (i'm a software guy, not a telcom guy..) Is there any way to know if a channel has been answered by an automatic system (like voicemail) rather than a human being? Specifically, I want to use a .call to make a call on a channel and only do something if a person answers, not a machine of any kind. Is this even possible, or is an answered

Hi all, i'm trying to integrate cepstral and asterisk, and i have a problem i'd appreciate any help with (i know it's a bit tangential, but i figure this is the place with the most knowledge of app_swift and asterisk). I've installed swift from cepstral.com with alison's voice, and it works fine, from the command line i can do swift "hello there" -o test.wav and then

hi all, I keep asking the question and getting no replies, so i'll keep asking :-) In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request from SER, specifically rewritehostport("myIP:5070"); (asterisk running on port 5070) asterisk picks up the request and matches it to the dialplan, i.e. if in ser i was sending to 151@myServer, it will make it

Hi all I have a couple of questions maybe you guys can help me with them I have sip phones , SER server , Asterisk. what is the best way to do that (also with accounting and authentication). which one of those options 1) sipphone -> SER -> ASTERISK -> SER -> PSTN 2) sipphone -> SER ->ASTERISK ->PSTN on the first option i am trying to return the call to the ser

Hi guys, I have a weird thing happening sometimes with users calling from a GrandStream phone through Asterisk onto a PSTN. Sometimes after a user hangs up a call on a GrandStream phone the phone starts ringing after a couple seconds. When the call is answered there is no one there. Anyone had this before ? Kindest regards David Wilson _______________________________ D c D a t a Tel +27 33 342

*spews coffee over keyboard* - FUNNIEST - THREAD - EVER - Also one of the most insightful. Teddy, your gmail invite is on the way.

...8. Re: Out of 5 Grandstream BudgeTone 101 THREE are defect !!! (from Pulverstore) (el Flynn) 9. fax over tdm400p (Sergio) 10. Best Grandstream firmware to use? (Paul Fielding) 11. RE: Best Grandstream firmware to use? (David Norton) 12. Re: Best Grandstream firmware to use? (Yair Hakak) 13. Re: Wait(n) -v- Background(silence/n) ? (Tony Mountifield) 14. Re: France has their (first?) SIP carrier with "unlimited" calls for 6eu/mo (Remco Barende) ---------------------------------------------------------------------- Message: 1 Date: Tue, 18 Jan 2005 15:46:24 +0...

...s: the router is opening the SIP message and > translating the Call-Id header IP, but I don't believe in that. > > Any clue? > > Thanks? > > Renato > > > ------------------------------ > > Message: 17 > Date: Fri, 11 Mar 2005 14:11:12 +0200 > From: Yair Hakak <yhakak@gmail.com> > Subject: Re: [Asterisk-Users] CDR database > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <43ff3394050311041125ebfca3@mail.gmail.com> > Content-Type: text/plain; charset=US-ASCII &...

So it looks like Livevoip went Bankrupt ------------------------------------------- There is a Federal Court Order in place and has been since Friday early a.m. ALL Suppliers are now under a Court Order that prevents them from terminating any and all services to LiveVoip LLC. If they take such any action they will be in direct violation of a U.S. Federal Court Order. If you have any questions

hello, if anyone is using asterisk as a voicemail system for SER I would be grateful if i could see a working ser.cfg and extensions.conf of such a setup. I am having some issues with rollover to voicemail when busy, and in setting up a VM extension for users to retrieve their mail without having to enter their own extension. When i get this working i'll write it up clearly for the wiki

(sorry, posted without subject) hello, if anyone is using asterisk as a voicemail system for SER I would be grateful if i could see a working ser.cfg and extensions.conf of such a setup. I am having some issues with rollover to voicemail when busy, and in setting up a VM extension for users to retrieve their mail without having to enter their own extension. When i get this working i'll write

Hi all, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of course) does this adequately protect the server from unauthorized users or is there

Hi all, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure because anyone can bypass the SER and register themselves as a peer with the asterisk. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of

hi list, is anyone succesfully using asterisk with libretel port-of-call (www.libretel.com)? If so, i would be grateful for configs..i set up libretel to forward to mynumber@myserver.com:5070 (asterisk is running on 5070 and SER on 5060) and when i call the number i see SIP messages with ngrep but the asterisk CLI doesn't seem to catch them. I assume i need to register...is this even possible

hi all, does anyone know what time variables are fed to to the "calldate" field in cdr_mysql? I have my system time set to israel time zone, have restarted mysql and a show variables shows timzone as "IST" which means now() should return israel time, but the calldate field keeps getting the system clock. I don't have the source for asterisk-addons handy so i can't

Hi asterisk list, this is a bit off topic, but can anyone explain the point of the commercial far-end solutions floating around (jasomi, for example)? or are the far-end things just hyped up media proxies? They claim to be b2bua devices but that's a very wide category and only implies that the media stream passes through it - exactly what can be done with fairly simple OSS stuff. In short,