search for: _2xxx

...ssage button the BT dials the number indicated as the voice mailbox number. Thus in my case a small in the main extension pattern matching gives me what I was looking for (just press the button and you get your voice mails..) This is how it was before: ; EXT. 2XXX ; generic dialer exten => _2XXX,1,Dial(SIP/${EXTEN},20) exten => _2XXX,2,Voicemail(u${EXTEN}) exten => _2XXX,3,Hangup() exten => _2XXX,102,Voicemail(b${EXTEN}) exten => _2XXX,103,Hangup() And this is how I changed it: exten => _2XXX,1,GotoIf(${CALLERIDNUM}=${EXTEN}?5) exten => _2XXX,2,Dial(SIP/${EXTEN},20) ex...

...es the A to B call. When the A to C call is ended, the A to B call clears up. Ending the A to B call first does not improve the A to C call. The dialplans are setup so each server passes all non-local extensions to it's neighbor. Hence, for A, the relevant part of the dialplan is exten => _2XXX,1,Verbose(1|Extension 2xxx) exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN}) exten => _2XXX,n,Hangup() exten => _3XXX,1,Verbose(1|Extension 3xxx) exten => _3xxx,n,Dial(IAX2/asterisk_B/${EXTEN}) exten => _3xxx,n,Hangup() For B: exten => _1XXX,1,NoOp() exten => _1XXX,n,Dial(IAX...

...some light on it. It used to work with HEAD a few weeks ago. -Matt /etc/asterisk/zapata.conf [channels] echocancel=yes echotraining=400 context=default signalling=em_w musiconhold=default group=1 channel=>1-24 Context that handles incoming from extensions.conf [process-from-trunk] exten=>_2XXX,1,noop exten=>_2XXX,2,Dial(SIP/${EXTEN},20,tr) exten=>_2XXX,3,Voicemail(u${EXTEN}) exten=>_2XXX,4,Playback(ss-noservice) exten=>_2XXX,5,Hangup exten=>_2xxx,103,Voicemail(u${EXTEN}) exten=>_2XXX00,1,StripLSD(2)

Hi all, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090722/30afa9ee/attachment.htm

Anyone know if - it is possible to limit 1 agent per extension where the last agent to log in overrides any previous agents or - a Command/application to clear all agents logged in on extension Does this look like it would require a custom mod to do it? J __________________________________ Discover Yahoo! Get on-the-go sports scores, stock quotes, news and more. Check it out!

...dial an extension on another asterisk server. My question is: How do I get asterisk to connect immediately without the annoying 2 ring wait before I can start dialing a number. snippets of extensions.conf [net_incoming] exten => s,1,DISA(no-password,net_outgoing) [net_outgoing] exten => _2XXX,1,Dial(${PYRMONT}/${EXTEN:1}) exten => _2XXX,n,Hangup() logging: Jan 23 07:39:47 NOTICE[16526]: chan_zap.c:6184 ss_thread: Got event 18 (Ring Begin)... Jan 23 07:39:49 NOTICE[16526]: chan_zap.c:6184 ss_thread: Got event 2 (Ring/Answered)... Jan 23 07:39:50 NOTICE[16526]: chan_zap.c:6184 ss_thre...

...Goto(aa-main,s,1) [default] include => aa-main exten => _NXXNXXXXXX,1,NoOp("Context is default") exten => _NXXNXXXXXX,2,Goto(aa-main,s,1) [internal] include => ext-local include => ext-internal [ext-local] exten => 7000,1,Goto(aa-main,s,1) [ext-internal] 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-at...

...[iax1Centos]type=friendsecret=volgausername=iax1Centoscontext=centos-contexthost=dynamictrunk=yes--------------------;Centos server extension.conf[centos-context]exten =>3000,1,Set(TIMEOUT(absolute)=10000) exten =>3000,2,dial(sip/3000/20)exten =>3000,3,voicemail(u3000 at default)exten=> _2XXX,1,Answer()exten=> _2XXX,2,Dial(IAX2/iax1Fedora:volga at 192.168.0.25/${EXTEN}@fedora-context)exten=> _2XXX,3,Hangup()0------------------------------------------; centos sip.conf [general]port=5060bindaddr=0.0.0.0[3000]type=friendsecret=3000qualify=yescall-limit=1 ;limit No of calls this ext...

...returns: "NOTICE[4296]: chan_iax2.c:7398 socket_read: Rejected connect attempt from 10.0.10.20, request '2100 at 2XXX' does not exist" "sip show peers" does register the phone: 2100/2100 10.0.10.237 D 5060 Unmonitored --------------------- Admin(.20) exten => _2XXX,1,Answer() exten => _2XXX,n,Dial(IAX2/2XXX/${EXTEN},20) IAX.conf [2XXX] type=friend username=2XXX secret= auth=plaintext host=10.0.10.21 context=internal qualify=yes trunk=yes --------------------- Call Center(.21) [2XXX] exten => s,1,Answer() exten => _2XXX,n,Dial(SIP/${EXTEN}) IAX.conf...

Hi All; If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal). Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I

Hi, I'm trying to set up a remote office with its own Asterisk Server they'll have a dedicated land line, but we'll still want them connected to the main office via VOIP (IAX2 via VPN). I've tested using IAX2 to bridge between the offices based on extensions, since the extensions we want to share are in isolated blocks of numbers. I'm not sure how to handle voicemail though.

...,n,Hangup() in the unit 2 without card sip.conf [asterisk2] type=freind host=ipadresseserver1 context=internal insecure=invite allow=all [1000] type=friend host=dynamic context=internal allow=all extensions.conf [internal] exten => 1000,1,Dial(SIP/1000) exten => 1000,n,Hangup() exten => _2XXX,1,Dial(SIP/${EXTEN}@asterisk2) exten => _2XXX,n,Hangup() i can call the 2000 from my unit and 1000 from my server without issue my question how i can i do in order to call an external number from my unit(server 2) with my server 1 i can call the external number without issue thanks and regar...

Good day all How do i set a connection between 2 asterisk servers via PRI In Bri I would set one to NT and TE How shoud the zapata.conf and zaptel.conf look And how should the cable be? All I got on the web was to set one to "pri_net"...this cant be all? And the cable > pin1 <--> pin4> pin2 <--> pin5> pin3 <--> pin6> pin4 <--> pin1> pin5 <-->

Hi All, I am in desperate need of this feature. I want to play background music during a call while the 2 parties are having some lovely conversation (or maybe give them a sort of cursing background if they are cursing each other). I found this post which talks about creating a ghost call with the help of queues and putting that queue in a meetme room where queue will play the song/curse and the

Hi all, I'd like to setup a really lean Asterisk installation that essentially has a full ISDN PRI (AT&T, T1, 23 B-chans, 1 D-chan, BZ8S, 5ESS, National dialplan) on a Digium TE207P adapter that all it does is convert the ISDN channels to SIP/IAX channels. Then I would add this Asterisk 'gateway' as a provider on one (or many) Asterisk systems on the back. With such a config I

This may be a stupid question, but I couldn't find anything on the wiki about it or on google. I have about 5 agents in my call center. I want them to login using agentcallbacklogin. The reason being is that I don't get so many inbound calls. We mostly make outbound calls. Therefore, during the times where we don't get calls, I want my agents to manually dial out. When calls

..., send to voicemail w/ busy announce exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain [phones] exten => _2XXX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN}) DID example: 2001 = 5552871701 2002 = 5552871702 2003 = 5552871703 Thanks! -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

...xten => main,9,Background(press-9) exten => h,1,Hangup exten => t,1,Goto(default,main,1) exten => i,1,Playback(invalid) exten => i,2,Playback(goodbye) exten => i,3,Goto(default,main,1) exten => T,1,Playback(goodbye) exten => T,2,Hangup ; Use dialed an extention exten => _2XXX,1,Goto(extentions,${EXTEN},1) [inbound] ; This is the list if inbound lines exten => 2181,1,Answer exten => 2181,2,Playback(silence/1) exten => 2181,3,Goto(default,main,1) exten => 2181,3,Hangup exten => h,1,Hangup exten => t,1,Hangup exten => i,1,Hangup exten => T,1,Hangu...

Hi all, I'm a newbie and I have a configuration problem with Asterisk. Seems that I'm not able to call an outbound number. I'm quite sure that it is a configuration problem, but I'm not able to find out where is the mistake, even reading several docs to www.voip-info.org. I do not have a good knowledge of Asterisk, I'm not very familiar with its configuration and I've a

...rying to test connectivity with the remote system or somthing else to do with the switch command becasue when I comment out the switch command and use wildcard extension mappings the problem is gone.. so effectively what I have done is replaced this.. switch => IAX2/... with this.. exten => _2xxx,1,IAX2/... etc.. Now I am sure many of you are going to respond and tell me that my dialplan is wrong or that you are using switch and its perfect.. Thats all fine and I agree that switch is perfect when the link between the two systems is up.. All I am saying is that one day when you link is down...