search for: _7xxx

Hellos, My voip carrier has assigned me a IP based account...where they only give me the IP to call through. I have setup the dial plan exten => _7XXX.,1,Answer() exten => _7XXX.,2,vmauthenticate(${CALLERID(number)}) exten => _7XXX.,3,Dial(SIP/${EXTEN:1}@Y.Y.Y.Y) exten => _7XXX.,4,Hungup() Where Y.Y.Y.Y is the assigned IP. After Dialing I asterisk logs the error SIP/Y.Y.Y.Y-35dc is circuit-busy Are there any settings I am leaving out...

...ternal] exten => _2XXX,1,Dial(${TELX-MICS2}/${EXTEN:${TELX-MICS2-MSD}}) exten => _2XXX,2,Congestion exten => _3XXX,1,Dial(${TELX-MICS1}/${EXTEN:${TELX-MICS1-MSD}}) exten => _3XXX,2,Congestion exten => _71XX,1,Dial(IAX2/telx-NY17S/${EXTEN}) exten => _71XX,2,Congestion exten => _7XXX,1,Dial(IAX2/telx-atl/${EXTEN}) exten => _7XXX,2,Congestion ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 2. iax.conf (telx-nyc) [general] allow=all jitterbuffer=no tos=lowdelay bindaddr=0.0.0.0 ; Registration to Junction Networks ;register => telx:E3fiw84j...

Ok so I added to my Freepbx config running Asterisk 1.2.4 in extensions_custom.conf ; intercom exten => _7XXX,1,SIPAddHeader(Alert-Info: Auto Answer) exten => _7XXX,2,Dial(SIP/${EXTEN:1},12,Tt) and configured my Polycoms via this page http://www.voip-info.org/wiki/view/Polycom+auto-answer+config for auto answer and that works fine if I dial 7 then the 3 digit extension. No problems, the receivi...

...ely transferred to an Asterisk menu/IVR. If they select the option to call a SIP phone directly (eg. entering the operator's SIP extension) then the callee/operator can transfer the call to a phone within the Bosch system. What Asterisk does is execute the following code sequence: exten => _7XXX,1,Flash() exten => _7XXX,2,SendDTMF(${EXTEN}) exten => _7XXX,3,HangUp() where _7XXX is a phone within the Bosch system. This frees the zap channel and the caller will communicate with the transferred destination directly through the Bosch system without passing through Asterisk. Howev...

...ce quality although there is no ringing tones(either ring back or ringing tone),but i can live without right now. I find that for each user i want registered i have to add his name and his ip address.I have been using "host = dynamic".Isnt there any way that i can define a dialmap such as _7XXX and all users can then be registered with the server and get allocated the their individual numbers by the server.(till now i define the numbers with callerid field). i need to do this so that i can add 15+ users without having to add each individually. what would be the entry inthe iax.conf for th...

Hi list, I set up two asterisk servers , 1001 is the first asterisk server's sip user, and 2001 is the second asterisk server's sip user. Each of them work well, but I don't konw how to connect them. I want to let the user in 1th Asterisk can call the user in 2nd Asterisk. First asterisk server ip : 192.168.3.101 Second asterisk server ip : 192.168.3.102 can someone

...siptrunk.conf siptrunk.conf: [box2] ; All box2 extensions; see extensions.conf type=peer context=local_SIP host=172.17.145.145 ; IP for new system disallow=all allow=g729 canreinvite=yes qualify=no extensions.conf snippet: [local_SIP] include => aggregate include => passthrough exten => _7XXX,1,Dial(SIP/box2/${EXTEN}) exten => _7XXX,2,Hangup() ----------------------------------------------------------------------- When I dial, all I get is (I'll attach the full dialog up to that point from SIP debug, below.) -- Executing [7444 at local_SIP:1] Dial("SIP/6110-08291cb0&q...

Greetings, I'm having a miserable time getting Asterisk working with FWD. All the samples show something like... exten => _7., .... How do I get Asterisk to wait until the user is finished dialing instead of trying as soon as it gets the second digit? I can use _7XXX, and dial the FWD 3-digit test numbers fine, but I'd like to be able to dial others... Same problem for outside analog line...how do I convince Asterisk to send anything that starts with a "9" to it? If it makes a difference, I'm playing with some QuickNet cards to learn the sys...

...group = 0,24 echocancel = yes signalling = pri_cpe channel => 1-23 Following is my extensions.conf stuff on both machine (extension number could be change) [from-pstn] exten => s,1,Answer() same => n,Playback(hello-world) same => n,Hangup() [from-sip] exten => _7XXX,1,Answer() same => n,Dial(SIP/${EXTEN}) same => n,Hangup() exten => 7527,1,Dial(DAHDI/G0/7527) But i am getting following error when i am calling from A to B satish-desktop*CLI> [Mar 25 15:40:19] WARNING[4519]: app_dial.c:2039 dial_exec_full: Unable to create channe...

I know that Asterisk supports DID, but does anyone have documentation on how to write the configuration for it? I'll be trying to setup a hybrid system where some incoming numbers will be DID enabled and others won't, so I'll need to be able to sort between the two, i.e. directly connect the DID dialed numbers and route the others to an autoattendant for extension dialing.

Since the 3000 has been out for a little while, has anyone done a review of the product? (couldn't find anything on google for wiki). Can the fxo and fxs ports be used as two independent channels? Is it really read for prime time? Etc. Rich