search for: 1001,20

...XTEN},60) ;voipjet NANPA exten => _011.,1,Dial(IAX2/402@voipjet/${EXTEN},60) ;voipjet WORLD [bpns-external] exten => s,1,Playback,bpnsmenu exten => 1,1,Dial(SIP/1003,20,tr) exten => 1,2,Voicemail,u1003 exten => 1,102,Voicemail,b1003 exten => 2,1,Dial(SIP/1001,20,tr) exten => 2,2,Voicemail,u1001 exten => 2,102,Voicemail,b1001 exten => 3,1,Dial(SIP/1002,20,tr) exten => 3,2,VOicemail,u1002 exten => 3,102,Voicemail,b1002 exten => 1001,1,Dial(SIP/1001,20,tr) exten => 1001,2,Voicemail,u1001 exten => 1001,102,VOicemail,b1002 ex...

Hi, I am attempting to change my dialplan to use 'n' priorities and labels for easier reading, and less re-numbering :) but how do you handle the plus 101 ? In my extensions.conf I have a simple plan for testing :- [inbound-sip] exten => uxbod,1,Dial(sip/1001,20,t) exten => uxbod,n,PlayBack(uxbod) exten => uxbod,n,VoiceMail(1001@voicemail,s) exten => uxbod,n,Hangup() exten => uxbod,103,PlayBack(uxbod) exten => uxbod,104,VoiceMail(1001@voicemail,s) exten => uxbod,105,Hangup() So when the extension has to add 101 do I just do n+101 ? T...

...ls to work. So I have gone back to a very basic FWD config, with one phone which as far as I am aware should work, but doesn't. I cannot find info on how to fix this. Below is my sip.conf [general] port = 5060 bindaddr = xxx.xxx.xxx.xxx context = sip register => 2xxxx:xxxx@fwd.pulver.com/1001 [fwd] type=friend secret=xxxxxx username=xxxxxx host=fwd.pulver.com ; ; [1001] type=friend username=xxxxxx host=dynamic secret=xxxxxxx callerid=Home <1001> dtmfmode=RFC2833 mailbox=1001 context=sip and here is my extensions.conf: [general] static=yes writeprotect=no ; [globals] HOME=S...

Hello all. Now I aleady installed the Asterisk. I could make communication between 2 XLite client through Asterisk. I tryed to test the voicemail function as follow. 1, I make a call to 1001 from 1002 2, Start ringing 3, Wait untill time out for ringing If no problem, 1001 go to voicemail and unavailable message will be played. But 1001 receive a 403 forbidden massage and connection go down. And Icould not leave a messages. Please teach me how to resolve this problem. Here is c...

Hi, Is there a way to add another extension to a live dial, for example Dial(PJSIP/1000,,) and after 20 secondes change it to Dial(PJSIP/1000&PJSIP/1001,,) I am open to suggestions such as using manager or stasis. Thanks in advance. Best regards, Kkh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180819/6f288f6a/attachment.html>

...l to an available local rep in another state. I thought this was possible .... until I realized the "transfer" only works on xPRO, which isn't available for linux. So I cant rely on SIP to handle this, I set up my extensions.conf have transfers, ie: [sip-exten] exten => 1001,1,Dial(SIP/1001,20,Trt) exten => 1001,2,Hangup And features.conf is : [featuremap] blindxfer => *1 ; Blind transfer ;disconnect => *0 ; Disconnect ;automon => *1 ; One Touch Record atxfer => *2 ; Attended transfer OK...

When working with call queues, if an agent is logged in via AgentCallbackLogin and the extension they are registered at becomes "unavailable" (from a bad connection, or something of the like), calls routed to that extension seemed to be accepted by it, so if the next action for that extension is to go to voicemail, the caller in the queue is sent to the extensions voicemail. Even worse,

...fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=2 txgain=2 group=1 channel => 1-7 extensions.conf ... [from-sip] ignorepat => 9 exten => _9NXXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) exten => _91XXXNXXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) ; generic phone extension exten => 1001,1,Dial(SIP/1001,20) exten => 1001,2,VoiceMail(u1001) exten => 1001,102,VoiceMail(b1001) exten => 1001,103,Hangu ... sip.conf ... [1001] type=friend username=1001 fromuser=1001 callerid=User Name <1001> host=dynamic nat=yes canreinvite=yes dtmfmode=info mailbox=1001@default disall...

Hello, I'm trying to set up a conference room. When I dial it's extension, I get an audible error saying "Not a valid conference room, please try again" followed by a disconnect. I've got debug sip peer 1001 (my X-Lite client) and I see this in the logs: (I'm pretty sure it has something to do with ztdummy, but I dunno... I have the port installed, but I haven't been able to find anything indicating whether or not I need to configure it somewhere or if there are any special FreeBSD kernel...

...oks in the > directory called "default", or is there a way to make MWI look in another VM > directory. > > thanks > > lee goodman > > > voicemail.conf > [general] > format=wav > maxmessage=180 > [sip] > 1000 => 1000,LG,xxxx@comcast.net > 1001 => 1001,TG,yyyyy@comcast.net > 1002 => 1002,BG,zzzzz@comcast.net > > extensions.conf > > [incoming] > exten => s,1,Background(goodmanmenu) > exten => s,2,DigitTimeout,5 > exten => s,3,responsetimeout,10 > exten => 1000,1,Goto(sip,1000,1) > exten =&g...

...ello, everyone. Right now, we are trying launch our own PBX system based on Asterisk(Fedora) with Cisco 2611. Cisco has 2 port FXO card installed on it. For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx fine. (I'll call it F). Using softphone, I can dial in extension 1001 on asterisk, which should talk to cisco. After initial connection to Asterisk, I have try to call F, and it will ring. Voice from softphone to F carries over and I can hear it; however, no voice from F to softphone will carry. I have been experimenting with different codec and other cisco/asterisk...

...um.com/pipermail/asterisk-users/attachments/20040903/aeac906a/attachment.htm -------------- next part -------------- Hi all, Im using asterisk. I have one doubt. Im running asterisk in one machine(RedHat9.0) running firefly softphone in 3 windows machine I hv 3 users in sip.conf like 1001, 2001 & 3001 appropriate entry for those users are also include in extensions.conf like -------------- [mainmenu] exten => 1001,1,Dial(SIP/1001,20,r) exten => 1001,2,Congestion exten => 1001,103,Busy exten => 2001,1,Dial(SIP/2001,20,r) exten => 2001,2,...

...constantly. I have the following configurations: zaptel.conf fxsks=4 loadzone=uk defaultzone=uk zapata.conf ; Zapata telephony interface ; Configuration file ; [channels] language=uk group=1 context=from-pstn signalling=fxs_ks channel => 4 extensions.conf [from-pstn] exten => s,1,Dial(SIP/1001,20) exten => s,2,Hangup The SIP elements of my system are working well, I just need to get this incoming call on a POTS line working. I have tried to keep things as simple as possible. Does anyone know why my call is not being handed to my sip phone? What is CID timed out waiting for ring? Is...

...l to an available local rep in another state. I thought this was possible .... until I realized the "transfer" only works on xPRO, which isn't available for linux. So I cant rely on SIP to handle this, I set up my extensions.conf have transfers, ie: [sip-exten] exten => 1001,1,Dial(SIP/1001,20,Trt) exten => 1001,2,Hangup And features.conf is : [featuremap] blindxfer => *1 ; Blind transfer ;disconnect => *0 ; Disconnect ;automon => *1 ; One Touch Record atxfer => *2 ; Attended transfer OK...

In http://www.voip-info.org/wiki-Asterisk+agents as followings, what type of channel of 28 and 29 is? agents.conf [agents] agent => 1001,4321,Wayne Kerr queues.conf [queue1] member => Agent/1001 extensions.conf exten => 28,1,AgentLogin(1001) exten => 29,1,Queue(queue1) I use the following in extension.conf with Vonage softphone account, it works well to call SIP extension 1001. exten => 180xxxxxx,1,...

Hi list, I experiencing a strange behaviour when transferring a call. The use case is like this: - Incoming call from Zap/1-1 - Routed to SIP phone SIP/1001 - The called user (SIP/1001) wants to redirect the call and presses "#" - IVR (default setup) says "Transfer" and user gets dial tone - User dials 1002 - IVR says "No such extension - please try again" ??? It seems that the 1st digit gets canceled out? Debugging the s...

...'Zap/1-1' -- Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on Zap/1-1 -- Unable to find extension '' in context 'local_extensions' ... extensions.conf ... [local_extensions] include => outgoing ; Local extensions exten => 1001,1,Dial(SIP/1001,20,rtT) exten => 1002,1,Dial(SIP/1002,20,rtT) exten => 1003,1,Dial(SIP/1003,20,rtT) ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070421/ca23cae0/attachment.htm

...vision an option to the Dial cmd's option 'r', where you could specify a ringtone to play if not the default, i.e. In indications.conf: [us] ... ... ring = 400+450/400,0/200,400+450/400,0/2000 intercom = 400+450/400,0/200,400+450/400,0/2000 ;FRESHLY ADDED AND STOLEN FROM [uk] section. 1001,1,Dial(SIP/1001,20,r{intercom}) For what its worth, I'm trying to use the standard UK ringtones for an internal extension. This behavior mimics several different PBXs and KSUs on the market. Does anyone have something like this working? Chris Coulthurst chris@shuksan.com -------------- nex...

Howdy, folks. I'm having a problem finding a way to auto-record calls (both incoming and outgoing). I know how to make it so either party can initiate recording, but I want it done as soon as both ends are connected (or prior to that if that's what it takes). It's probably right in front of me and I'm just missing it. Any help would be much appreciated. Thanks, Jay