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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 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

hello, after playing with an asterisk configuration for voip for a few weeks i'm trying to get outbound dialing with voicepulse going - i've cut down the asterisk to a very minimal install (1 SIP client) to try to localize the problem. The SIP client works fine (SIP and * on the same NAT) and could access the demo from samples before i removed it, and can call itself - so i am

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

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

...ter 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 > > ht...

...d to be clean. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/4f9fe0 67/attachment-0001.htm ------------------------------ Message: 12 Date: Tue, 18 Jan 2005 10:55:01 +0200 From: Yair Hakak <yhakak@gmail.com> Subject: Re: [Asterisk-Users] Best Grandstream firmware to use? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <43ff339405011800552275aff@mail.gmail.com> Content-Type: text/plain; charset=US-ASCII i've actual...

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,

hello all, another desperate request for help debugging my dialplan... from a certain extension i do the following: DBput(CFIM/${CALLERIDNUM}=${CALLERIDNUM}) a NoOp to the console says DBput: family=CFIM, key=2122022001, value=2122022001 and database show says /CFIM/2122022001 : 2122022001 so far, so good. but in a macro, when i try to get the data, exten

Hello, I have the following setup: sip phones <->SER <-> asterisk <-> voip provider1 <-> voip provider2 i got a toll-free DID from voipprovider1 to allow people from outside to call into asterisk, get authenticated, and use voipprovider2 to call out (kind of a primitive calling card app). anyway, voiprovider is giving my

hello all, I have a DID from nufone, transported via SIP to my * box, and even though i'm using rfc2833 DTMF i'm still getting double digits and all sorts of other stuff... sip.conf is as follows: [general] port = 5070 ; Port to bind to disallow=all ; Disallow all codecs allow=ulaw allow=alaw allow=ilbc allow=gsm dtmfmode=rfc2833 register =>

hello all, i have a strange problem....i am running SER in front of asterisk, and am testing softphones. x-lite works fine...i can dial, hang up, DTMF, all good. Firefly looks really cool and i'm very impressed with the IM-like interface and the skinning ability, but something strange is happening...when i call from the firefly and run something on the server and press hangup on the client,

hi all, is there a switch statement in the dialplan? or do i have to daisy-chain GoToIf statements? i don't see a switch statement on the wiki, but you never know... thanks yair

hello, please 'scuse the slightly offtopic question, but i see a lot of posts about the adit600, used as a channel bank, but from what i understand it can be used as a PRI interface as well. If anyone who is using the adit600 to interface to 4 T1/E1's has feedback, i would appreciate it, specifically involving the asterisk interface, echo, and DTMF. thanks, yair