[asterisk-users] Putting incoming sip call leg on MOH while dialing out other party**********NEED HELP************

your Help Mohammad Mirzaee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Bridging a call on hold with an active call

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

[asterisk-users] Call Initiation with Asterisk

Hi List; I need help for the following senario: Initiating a call from Asterisk to an extension and after it answers, IVR prompts will be played Mohammad Mirzaee +989121750530___ --Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] voicemail + Dynamic mailbox

Mohammad Mirzaee Mohammad Mirzaee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Voicemail Creation

Mohammad Mirzaee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk+Nat+sipura (Help)

Hi ALL; I have users with Sipura/Linksysphones regsitered behind Nat( useing STUNat phonenot portforwarding) in my Asterisk box, when I try to call them with another phone i got: Got SIP response 404 "Not Found" back from 217.6.190.4 SIP/217.6.190.4:5060-666d is circuit-busy Isabove

[Asterisk-Users] Asterisk+Nat+Sipura/Linksys

Hi ALL; I have users with Sipura/Linksysphones regsitered behind Nat( useing STUNat phonenot portforwarding) in my Asterisk box, when I try to call them with another phone i got: Got SIP response 404 "Not Found" back from 217.6.190.4 SIP/217.6.190.4:5060-666d is circuit-busy Isabove

[Asterisk-Users] redhat 9 and oh323

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

[Asterisk-Users] asterisk+radius

HI ALL; Is there anybody who use app_radius(astersik radius module)??? is it stable? Regards mohammad

[Asterisk-Users] converting gsm file to g729 format

hi all; Anybodey know how to convert astersik *.gsm files (like voicemail ivr prompts) to .g729 format? Regards mohammad

[Asterisk-Users] sip ua---------asterisk-------h323gw

HI ALL; I have a sip ua (ATA) registered in my asterisk box.I want my astersik box to route all incoming calls from sip ua(A-Z prefix)to h323 GW. what syntax should I use in extension.conf for routing all prefixes to h323 GW. Regards mohammad

[Asterisk-Users] h323 call flow fails

HI ALL; I have an ATA phone registered with GUNGK.Iwant to send a call to another ATA with has an extention in my * box. my network looks like the following: (h323 registration) ATA1(h323 ep)gungkasteriskATA2(h323 ext) But when I try to send a call from ATA1 to ATA2, it fails. I

[Asterisk-Users] Re: h323ep----gnugk-----astersik------h323ext

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 [EMAIL PROTECTED] To: mohammad mirzaee [EMAIL PROTECTED] Sent: Wednesday, July 21, 2004 5:59 PM Subject: Re

[Asterisk-Users] callparking vs calltransfer

HI ALL; Anybody can explain the difference between "call parking " vs "call transfer" Regards mohammad

[Asterisk-Users] codec translate

HI ALL; Is astersik enable to translate between different codecs. I have couple ofSIP-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

[Asterisk-Users] sip-oh323

HI ALL; I have couple of ip phones connected to my asterisk box 1-cisco ata with sip protocol 2-sjphone with h323 protocol as I understand, asterisk isable to translate siph323 and vice versa ( am I right)???/ but when I try to connectfrom ATA toSJPHONE and vice versa it fails.

[Asterisk-Users] sip-h323

hi all; hi DANIEL; I setup asterisk as a translator between sip-h323(I used oh323 not native). But there is a problem and it is as follows: whenI try to dial FIRST from sip UA to h323 client, or h323 client to sip UA , it is ok BUT the second try from any of them to another have no

[Asterisk-Users] 0h323/ h323-registration

HI ALL HI MICHAEL; Thanx Michael for your help, oh323 compiled successfully. As I understand, oh323 channel cannot act as h323-gatekeeper and it just role as a simpleh323-GW ( am i right micheal). If this is the case, how can I register my h323EP ? shall I user a 3rd partyGK like

[Asterisk-Users] OH323-COMPILE

HI ALL HI MICHAEL; My name is mohammad and I am iranian.I have been trying to install oh323 channel but Icome up with dead end. In factit makes mecrazy. 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

[Asterisk-Users] voicemail

HI ALL; Is asterisk voicemail service can be run under H323 or it just run under SIP. mohammad