similar to: Call Transfer Function in *

I'm running Asterisk 1.8 from Debian. I have some analog phones connected via a TDM400P. I'm testing them with these simple extensions: exten => 600,1,Answer() same => n,Festival(This is an echo test) same => n,Festival(Hang up or press pound when you are done) same => n,Echo() same => n,Festival(Good-bye) same => n,Hangup() exten

Ok, I've seen discussion before on doing transfers (attended and unattended), there seems to be much confusion over it. As things sit, I've been trying (unsuccessfully) to do transfers with a zap channel (analog phone attached to TDM400). I have no idea what I'm missing. My current understanding is that I need to have transfer=yes in zapata.conf, do a flash hook, dial the 2nd

Hi i was trying to call 17009978275 which is my Fwd line on my notebook from Asterisk and i keep getting this message on the console. -- Executing Dial("Zap/2-1", "IAX2/@iaxtel.com/17009978275@iaxtel") in new stack -- Called @iaxtel.com/17009978275@iaxtel WARNING[1150495040]: File chan_iax2.c, Line 4547 (socket_read): I don't know how to authenticate rob to

I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse

I recently plugged in Phone to my TDM400P Card to test out something I mostly use sip phones to interface with *. All of sudden I'm getting lot of line static noise coming of the card is there any settings I should look at or anything I need to do on the command line at this point I'm open to any ideas I'm running 0.7.1 on Redhat 9.0 machine. Any insight would be greatly appreciated.

Hi there, Maybe this has been cared about before but I could not find any solution to this problem either in the wikis or in the list archived. If someone has found a solution before please just tell me where I can find that info..thanks :) I am using iaxComm to call other people who are either using SIP or also IAX clients, like me. All of us are connected to the same asterisk server.

With the 1U Apple G5 server at a good reliability/storage/pricepoint level... got to thinking about compiling Asterisk on OS X.. at least for SIP phone call switching, voicemail, etc. Has anybody attempted this? Email me off list if this is too dev-heavy for the user list. Thanks, Ted W -----Original Message----- From: asterisk-users-request@lists.digium.com

Is there a way to initiate a transfer using an analog handset? For instance I'm looking for a way to do something like the following: External call comes in and is answered by user A. After talking to the caller they determine that the caller really needs to speak to user B. Is there any way for user A to initiate a transfer to user B, using only their analog handset? Now to make

Hello, I''m experiencing a strange, but very bad behavior with Mongrel 0.3.13.4 and Pound 1.8. Every 6 hours or so one of our nine (not consistent on which one) application servers ( each one running several mongrel processes ) will start leaving lots of socket connections with pound open. This leads to "Too many open files" errors. I''ve set pound to close connections

I try to explain..On wine 1.0 i was able to run programms ipning them with wine and then they all running in a box (wine desktop) taht was apart from my Unguntu system.Instead now i'm useing wine 1.0.1 and olso if i open the windows program with wine i have the running out tje "wine personal desktop" i will get all program running integrated in the ubuntu system taht i'm afraid

Hello. I am a spanish student, so excuse my English. I have this HW: - 2 X100P PCI with two analog lines plugged in. These lines are two extensions of a panasonic PBX. Zap/1 = X100P <-- analog line --> extension #237 PBX Panasonic Zap/2 = X100P <-- analog line --> extension #245 PBX Panasonic - 1 TDM20B with two analog telephones plugged in. Zap/3 = TDM20B

I set up a context to allow me to call in to my * server (via Teliax in this case using IAX2) from my cellphone, and let me do a number of things, including dial other extensions, AND dial outbound again so callers could see my proper work CallerID when I use this service. Is there a way to be able to transfer calls to other extensions of my asterisk server FROM the cell phone/ This isn't a

Brian, Great job fixing up the list server... postings are happening very quickly now (within minutes)! Thanks, Rich

Hi, Anyone know if Asterisk and Digium Hardware supports Centos-3.1 which is clone of Redhat Enterprise 3.1 server.? -- Best regards, Frankie (fgravato@cfsdigital.com) mailto:nanog@cfsdigital.com

Hi, I'm having problems with transfer from an analog line via a X100p and Cisco 7960's running SIP. With an attended transfer the a call comes in, I transfer it to another 7960, they answer I announce the call, press transfer again, the two parties talk for 1-2 seconds then the analog line drops, though the Cisco phone is not aware of this, i.e. nothing on the screen changes. The

Hi, i want to to this thing with php AGI: #!/usr/local/bin/php -q <?php set_time_limit(30); require('phpagi.php'); error_reporting(E_ALL); $agi = new AGI(); $agi->answer(); $cid = $agi->parse_callerid(); $agi->text2wav("Hello, {$cid['name']}."); $agi->text2wav('Enter some numbers and then press the pound key. Press 1 1 1 followed by the pound key

So I have read and read and read... google is my friend and the wiki is by brother... However, I am still unclear on what the preferred method of using the pound sign is. If the Pound sign is set aside for Transfer.. Then when I make an outbound call to my bank I can not "Enter my PIN followed by Pound" Likewise if I turn off the ability to transfer when initiating a call, my bank pin

Hi, is it possible to use Asteriks for translating SIP to H323 and vice versa? I am looking to implement the following Setup SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC Basicly i want SIP fones to talk to H323 fones and and SIP Fones to access PSTN Gateway(s) in a H323 network. Anyone got something similiar running? Any ideas? best regards,

I am using Festival to synthesize some menu Interaction with a caller and am having a problem. What I am working on is a remote callback where I can remotely call in to an extension, and enter a callback number (or use the CALLERID info) and a second outbound dialing number to connect to. Things work O.K. until I set usecache=yes in festival.conf. After doing this, things run well for

Hi List, I hope this setup must be done by our astersik users.. I am using Sipura 3000 to receive PSTN calls and forward those calls to asterisk for voice processing and after that, I am transferring call to extension through FXS port on SPA 3000. Currently, media of call is trombone through asterisk. i.e achieving blind transfers on asterisk with SPA 3000. Is it possible to stop trombone