search for: mirzaee

...erisk 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 scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070429/c5c1b59c/attachment.htm

...putting first call leg on HOLD(My own pre-recorded IVR message) And meanwhile dialing out the second leg.When Asterisk detect "Ringing" on the second leg ( from GSM provider) , the first call leg should be unhold and then Bridging will be occured. Appreciate any help, Mohammad Mirzaee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080518/39f2d21b/attachment.htm

Hi, I wonder if it is possible to perform the "find_by_contents" on a subset of fields indexed in acts_as_ferret.If so, how? In my code I have: acts_as_ferret (:fields => [''title'', ''focus'', ''purpose'']) However, I like to have two search options one on all fields and one only on the title. Any help is most appreciated.

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

...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 h323 channel???????/ Regards; mohammad ----- Original Message ----- From: "administrator tootai" <admin@tootai.net> To: "mohammad mirzaee" <agent@rasatech.com> Sent: Wednesday, July 21, 2004 5:59 PM Subject: Re: h323ep----gnugk-----astersik------h323ext mohammad mirzaee a ?crit : > HI DANIEL; > > > I found you expert on asterisk call routing, will you plz help me for > the following problem:: > > &...

...erisk 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 Mohammad Mirzaee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070430/df54bb14/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 list, I need to represent a relationship between three tables: Tags (id, name) Users (id, name, email, ... ) Documents (id, title, ... ) I created a forth table called Assignments(id, tag-id, user-id, document-id, date). I have couple of questions: Should I use belongs-to and has-many to capture this? If so, How can I do that? should I have the id as the primary key in Assignment table or

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,

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 ALL; I found in the following page: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing 1-If I could record all IVR promts in G729 format 2-If I could record voicemail in g279 format with """format_g729.c""""" then I donot need any g729 license (I suppose all my clients have g729 ip phones) My question is, how

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

Hello. LCR means least cost routing, and it's billing system problem where to route a call, not b2bua's. But currently I dunno any free billing system that support it, so i moved this logic to b2bua. On Sat, 12 Feb 2005 07:05:39 +0330, mohammad <mohammad@mirzaee.net> wrote: > Hi Mike; > Thanks for your new application, but I think it would be better if you put > everything under the radius. > I mean for example LCR (radius based call routing). have you any plan to do > that? > > Warmest Regards > Mohammad > >

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; 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 ......... ...........

Hi ALL: I was compiled oh323 successfully on redhat 8 with gcc 3.2-7. But the same asterisk-oh323 on another machine with redhat 9 has problem.i shall say redhat 9 uses gcc 3.2.2 as its default compiler. any suggestion? mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL:

HI ALL: I am newbie to IAX, my iax.conf is as follows: [general] port=5036 ..... but I donot why it doesnot listen on UDP PROT 5035, instead it listens on 4569 Asterisk CLI debug says: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found Nov 30 11:52:12 WARNING[1076220544]:

Hi ALL; I got the latest Asterisk-addons for Mysql-Cdr, but I have problem compiling that.It says: # make ..... ........ res_config_mysql.c: In function `realtime_mysql': res_config_mysql.c:143: warning: passing arg 1 of `ast_strlen_zero' makes pointer from integer without a cast res_config_mysql.c: In function `realtime_multi_mysql': res_config_mysql.c:242: warning: passing arg

Hi; I have problem inserting cdr to mysql,Asterisk has logged to asteriskcdrdb successfully.It says: Dec 6 13:36:26 DEBUG[1221985728]: cdr_addon_mysql.c:178 mysql_log: cdr_mysql: inserting a CDR record. Dec 6 13:36:26 DEBUG[1221985728]: cdr_addon_mysql.c:197 mysql_log: cdr_mysql: SQL command as follows: INSERT INTO cdr

HI; I have an Asterisk with 10 "SIP" ip-phones, our pbx features are now: Voicemail and Call Transfer. How can I serve both "Call Waiting / 3 way calling" for our SIP Phones.?/ Appreciate Any Help Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: