similar to: Native vs Intl calls

How long does it take to get a vanity number? I signed up for an account, pre-paid some money, and then placed a vanity number order. I did all of that around Dec. 31st 2004. They said it would take 2-10 business days. It is now Feb. 15th and still no vanity number. I've called them about a dozen times and every time they tell me to keep calling the number to check it and just wait. I'm

Hi; As you probably know, SER style of handling an incoming call is : 1) try to look-up it from registrar DB 2) if not found there, try to do some thing else Is there any possibility of doing the above at "Asterisk Dial-plan"? Regards Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm Andrew. (Hi Andrew) I'm a toll-free number junkie. I've had an account with iax.cc/sixtel for about a week, and every few days, I find myself sitting at the DID menu clicking the link that reads "Click here to get a random toll free number". I have three toll-free numbers now, and I don't know if it will stop... Is there any hope for me? -- Andrew

HI ALL; Anybody can explain the difference between "call parking " vs "call transfer" Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040720/2975991b/attachment.htm

HI All; I want to use Asterisk for just Voicemail Server and I need a Dynamic creation of Mailboxes. My users 's Mailboxes are same as "Extensions" but I donot want to add mailboxes in "Voicemail.conf" Is there any way to create mailbox from Asterisk dial-plan ? Appreciate any suggestions Mohammad Mirzaee -------------- next part -------------- An HTML attachment was

Dear All I want to use asterisk for the following Senario and Need help to find a SAMPLE extension.conf Incoming call >>>>>>>>>>>Asterisk >>>>>>>>>>>>>>GSM Termination Gw first leg second leg What I want to do is putting first call leg on

Stefan Gofferje wrote: > Hi Folks, > > on my home asterisk, I have a "huntgroup" for incoming calls on the > private line which first let ring my phones in my office and living > room, after a while then office, living room and bedroom. > I do this by simply putting two dial statements in sequence: > > > [private_huntgroup_day] > exten =>

HI ALL HI MICHAEL; My name is mohammad and I am iranian.I have been trying to install oh323 channel but I come up with dead end. In fact it makes me crazy. plz help me michael. I saw mailing list and I trid serevel CVS headers such as , 2004-06-07( seven of june) 0r 2004-07-02( second of july) besides I use: 1-openh323 v1.12.2 2-pwlib v1.5.2 3- asterisk CVS (2004-06-07,

Greetings, I'd like to thank everyone that has responded to my original email. I have received information from several companies, and will be testing several of them. I also would like to update a statement from my original message to clarify it: >My strikelist: nufone, voicepulse, iax/sixtel The strikelist is just a list of carriers that didn't meet the needs a resonable

HI ALL; I have problem converting a windows .wav file to .gsm format by Sox. Could anyone help. Cheers, Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050706/3408bfd5/attachment.htm

HI ALL; Is there anybody who use app_radius(astersik radius module)??????????? is it stable? Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040803/8a096bfe/attachment.htm

HI; Thanks for your reply. The reason for why I am going through asterisk in such case is just "using asterisk voicemail service" I mean: ATA1 calls ATA2, suppose ATA2 is unreachable or he is not at the office, then the call reroute (my GK is able to reroute calls if the first route is not valid) to atersik for voicemail service. Do you think I can handle it with asterisk native

Does any one know what happened with voice conduits? I have been trying to reach them for nearly three weeks now. Their voice mail boxes are full and writing email to them does not get any returns. Thoughts or sightings are appreciated. -- R.J.

Hi; Any idea of how to have different ringing tone on called party for different caller-id by means of "Alert-Info" header. Regards Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041226/3611eeae/attachment.htm

HI ALL; Is astersik enable to translate between different codecs. I have couple of SIP-UA , one with (a-law) and the other with (g729), registered with my astersik box.Can astersik translate between alaw-g729 and vice varsa. Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi; I use the following asterisk, openh323, pwlib: asterisk = cvs-head-03/09/05 openh323 = 1.13.5 pwlib = 1.6.6 asterisk-oh323= 0.7.1 Asterisk, openh323, pwlib were compiled successfully but when I try to compile Asterisk-Oh323-0.7.1 , I got the following error: chan_oh323.o chan_oh323.c chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory ......... ...........

OK, I'm trying to send an email to the list the contiune a thread which describes a problem I'm having. This particualy email I wish to send contains an ls -l describing my problem (too many open files) and is apparently too large to be considered a "normal" post, so I get a message that it's being held until a moderator can view it. Fine. So now I get an autoresponder

Hello, When I dial out there is a long delay in dialing. Is this normal? Thanks, David

I have a Quicknet Linejack in /dev/phone0. My phone.conf is: [interfaces] mode=dialtone format=slinear echocancel=medium context=mayores device => /dev/phone0 Only I can mark 7 digits, soon asterisk tries dial automatically. I cannot mark 8 or more digits. 6 or less digits work ok. -------------- next part -------------- An HTML attachment was

Hi I have a simple question but I cannot find the answer. I have a line with 2 different phone numbers I want to redirect each phone number called to a different IP phone Example Someone calls 5551234 and the call is redirected to IP phone 192.168.0.2 Someone calls 5551235 and the call is redirected to IP phone 192.168.0.3 Thanks Patrick