similar to: AMP: Losing backslash characters in config files

Hi Fran, you could do it using Adobe/Macromedia Flash Media Server 2, but I guess that's not the answer you are looking for. If you manage to do this and release it under GPL I'll kick in $50 for a bounty. Regards, Dean Collins dean@collins.net.pr +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -----Original Message----- From: asterisk-users-bounces@lists.digium.com

What are the alternatives to FWD with IAX2 registration capability. FWD is great, but their IAX2 is not the priority and if it goes down it takes days to restore it. I want to use IAX2 protocol but the end point (Sipura unit) need to be able to register over SIP behind firewall. Line1 is registered with FWD PSTN need to be registered with somebody else. What are my alternatives? -- #Joseph

should I use mpg123 with asterisk 1.2.7 or should i use the native player asterisk has? the target machine will receive heavy load. also, has anyone succedded in compiling mpg123 in a dual core pentium with centos 4.3 ? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

Hi friends, I am using Real Time Asterisk Architecture where I have put the Sip users/peers and extensions defining the dialplan in tables in a mysql database. Currently, asterisk points to my single database server as configured: ------------------------------------------ /etc/asterisk/res_mysql.conf ------------------------------------------ [general] dbhost = xxx dbname =

Someone can confirm the new release is out? Haven's seen any post about it!!!!! -- Adri? Vidal adriavidal@gmail.com

Hi all i want to know where i can find french sounds for asterisk. I don't have any studio to register good sounds. Bests regards Serge ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et

Does anyone know of a way to make an originate request coming over the management interface (e.g. AstTapi click-to-dial) include the relevant Alert-Info SIP headers to enable the originating phone to auto-answer? I've tried setting up a custom context (see below), but the dial plan is only entered AFTER the originating call is answered, so the SIP header is added to the terminating call leg,

I am having trouble understanding how to use this. I want to transfer certain incoming calls from an IAX ITSP based on caller ID. From what I can make of the docs, I thought I need to do something like this... exten => _NXXNXXXXXX,n(nocid),transfer(1000) exten => _NXXNXXXXXX,n,noop(boo,${TRANSFERSTATUS}) exten => _NXXNXXXXXX,n,hangup exten =>

I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Since both machines are on public ip, what other problem can it be ? There's one configuration working : lynksys pap -sip-> asterisk server -sip-> quescom this way both sides can hear voice but with : lynksys pap connected to a switch -sip->

Is there any way to have more specific disposition codes in the CDR? Currently there are only 3 values: ANSWER, NO ANSWER, BUSY. In this way, when i call a cell phone that is switched off i get "NO ANSWER", while i would like to be able to log that the call is not answered because "The customer you have dialed is unavailable at the moment". The same for "non

Got the answer on the chan_sccp list. Thanks John On 1/25/06, John Reynolds <reynoldsjc@gmail.com> wrote: > > Anyone care to post the format of this file? I've been looking all over, > couldn't find it on the Cisco website. I'm open to correction. > > Thanks for you assistance, > > John > -------------- next part -------------- An HTML attachment was

Hi all, I am new to this list. I have been looking for a Manager API mailing = list for a while, but could'nt find any. Is there a such list? Thnx. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060126/bf3b67e2/attachment.htm

Besides the codecs that * supports. Is there any ISAC implementation for asterisk available? This is to be used mainly with softphones, i haven't seen any hardphones that support this codec. Thanks, -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

Hi, We have a task to reduce voice call bandwidth. IP+UDP+RTP are using 40 bytes per packet and for voice GSM FR 33 bytes. We are trying to reduce this bandwidth accommodating multiple GSM frames in one packet. If we want to use per packet 10 GSM frames how to do this using asterisk? Assume the sip client is able to split these packets in to individual GSM frames. Any help will be sincerely

Hi Group, I am sending my question again why I don't have answer yet: I am developing a application, this use "Manager API" to connect with Asterisk. But when I call to an external number (over a zap channel), I don't receive any event when the target answer, Who can help me?, Which event notify me that the phone call was answered? Thank you. Ezequiel -------------- next

Hi I'm browsing around the internet looking for signs that the IAX client library and app_meetme support video. I stumbled across this post by SteveK on the 27th of Feb 2006. "My company is looking to hire a full-time developer, who will be working about 25-50% of the time on iaxclient; in particular to finally integrate, build, polish and enhance video in iaxclient, add video

I have ztdummy installed: Module Size Used by ztdummy 3464 0 zaptel 218756 1 ztdummy crc_ccitt 2176 1 zaptel ohci_hcd 16388 0 floppy 49028 0 pcspkr 2180 0 piix 8580 0 [permanent] ehci_hcd 24456 0 uhci_hcd 26256 0 rtc

Hi, What port does mpg123 uses to play music on when it starts MoH after asterisk has put called on hold? Zach A

I'm very new to Asterisk, I'm tracing the Asterisk code, i'm feeling difficulty in understanding the code, so please tell me where i can get the documentation of the code and, design and architecture of the code. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060309/0751f7e2/attachment.htm

Hello, I am using Asterisk 1.2.5 and the voice mail application. I don't actually want to keep the voicemails on the server so I am emailing them. However if I have the line in my voicemail.conf 1234 => 4242,Temp_User,voicemail,,delete=1 Then my voicemails are all sent with the same subject line and Message id eg: [PBX]: New message 1 in mailbox 1234 Is there anyway, to delete a