similar to: dialplan tips

Hi, How can I match an extension "ending with 3" (just an example but applicable to any other digit, including * or #)? exten => _ZX.3,n,... exten => _ZX.#,n,... (the above does not work) Can regular expressions be used in the standard dialplan (end with: "$")? Thanks, Vieri

I cant seem to get outgoing calls to be placed properly .. No matter what I try I get an error from the PSTN company saying that the "call can not be completed as dialed" or "you need to dial a one..." The asterisk debugging seems to show the correct number being dialed out of the zap interface... the "9" is being stripped and there is a "1" where it is

Morning, Have some questions regarding receiving and sending faxes... 1:st example: exten => 101,1,Answer() exten => 101,2,Wait(3) exten => 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) exten => 101,4,System(tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff > /var/spool/asterisk/tmp/fax.pdf) exten => 101,5,System(mutt -s 'New FAX for you sir' -a

Hi all, I have a voice menu system ( Outlined below ), and I'd like to give the user some feedback when they dial an extension ( ringing, music, SOMETHING ). As it stands, when a user enters an extension from the menu system, they hear silence while the line rings. I even tried including the Ringing application before calling my macro to dial the phones, with no luck. Any help is

Thanks to help on this list and Sangoma's support we have incoming and outgoing calls passing through asterisk. However both incoming and outgoing calls are greeted by silence. I've noted our existing config below with our test extensions.conf. Help much appreciated Rory Zaptel ----------------------------------------------------------------------- loadzone=uk defaultzone=uk #Sangoma

Hello, i?ve a problem i dont find the reason for. An incoming call coming over iax is connected to a Sip phone. Until the phone picks up the call i could hear moh without problems. Then the phone sets the call on hold and opens another call to another extension. The incoming call hears the Hold music and also the call to the other extension hears another moh. Everything works so far as it

hello, I have instaled a analog line, and when I call on the console apears that: I want to redirect the call to 101 extension. *CLI> -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at default,s,1 still failed so falling back to context 'default' Apr 27 08:15:53 WARNING[3494]:

Priorityjumping is totally ignored by my asterisk (tested 1.4.4 and 1.4.7.1 on FreeBSD 6.2) [general] priorityjumping=yes With n+101: exten => 1337,1,Dial(SIP/zytek,5,Ttj) exten => 1337,102,Dial(SIP/zytek,${RINGTIME},${OPTIONS}) exten => 1337,n,Hangup -- Executing [1337 at firma:1] Dial("SIP/113-087a3000", "SIP/zytek|5|Ttj") in new stack -- Called zytek

Hi all, This issue is giving me a lot of grief with my customers. I have 5 asterisk servers running in production, each one with almost 70 simultaneous calls at peak hour. Most of my customers complain that their calls drop after 20 seconds or so. After running through my cdr's, I see that the number of 20 second calls is MUCH larger than any other number. (see below) billsec count(*) 1 924

Hi All I am trying to simplify a dialplan for a few thousand users. Would what I have below work? If someone dials exten 710001 would it go through answer and then to the macro to try dialing the SIP phone thats registered on 710001 and then onto voicemail if no answer or not signed on? exten => 71XXXX,1,Answer() exten => 71XXXX,2,Macro(71macro,${EXTEN}) exten => 71XXXX,3,Hangup()

Hi List; Can someone advise me why in the below context, it does not run the Background step? Once I dial 1000, then it hangup and give congestion signal? If I comment the ResponseTimeOut, then it run the Background but it does not wait till caller enter the digits, once the sound file finish, then it hangup (congestion signal), also in all the situation, it does not go for the t extension, why?

hi, I am not registered my SIP Phone with Asterisk i spend almost one day but find no luck.I know very well this is not kind a problem discussed in this group but i try my best and all in vein so finally i am here hoping you ppl helping me out.I discussed this problem in asterisk's-users group and adding feedback from asterisk-users group my configs are sip.conf [general] port=5060

Setup is: Asterisk --->NAT--> SIP Proxy I have following entry for SIP Proxy in sip.conf [Proxy] type=peer host=Static IP (NAT Firewalls public IP) username=xxxx secret=xxxxx nat=yes???????????????? canreinvite=no???????? qualify=yes Proxy sends a call and I get this error Found no matching peer or user for <NAT's Public IP:70001 NAT is using 70001 as the source port in the

Hi, I'm currently programming an interface for my Asterisk service. I've noticed an issue if someone sets up call forwarding on their sip phone. Asterisk receives a 302 "Moved Temporarily" message, and forwards the call successfully. However, the CDR is not correct. If I set up call forwarding to a mobile on extension 201, and then place a call from extension 202, the call

Hello all, I need that Asterisk Always use Reason in a CANCEL. How to do? thank you *Fran?ois * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100521/f3a91f36/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: francois.vcf Type: text/x-vcard Size: 400

I searched the 22490 messages I have in my own personal asterisk-users archive and have not found the answer, and it also does not appear on the wiki. I have a SIP phone and a regular phone on a TDM400P FXS interface. Extensions are 100 and 101, respectively. On the SIP phone I can hit #, get the "Transfer" prompt and enter an extension I want to transfer to. No problem. I can do

Hi everyone, when i use the automated dial out,I found that once the zap answerd,the contex will be exectued, but i don't hope do it ,i hope when extern phone answered ,then ,the context will be exectued. Anyone can help me solve the problem! the call file is: Channel: Zap/g0/15015895665 Context: myivr RetryTime: 60 MaxRetries: 2 Waittime: 60 Extension: 808 Priority: 1 Callerid:

Hi Everyone, I am using OutCall 1.6 (latest) with Asterisk 1.6 and Windows Vista. I can originate calls see the program login nicely but when a call comes in it only shows the Name portion of the CLID and not the number hence it pulls up a new contact on Outlook. The new contact only show name and last name and no CLID Number again. So, this repeats every-time I call even if I manually enter a

Hello everybody i am Ashish here. i am new to this mailing list. so dont know rules and regulation, just trying to post my problem of voicemail.conf Actuallt right now i am using Asterisk 1.2 on my LAN environment. i am able to call all my extension very nicely. Right now i am trying to deploying voicemail facility for all extensions, so if anybody is not present, then he/she can leave message,

Hi, Does anyone have a suggestion how I can analyze the concurrent usage of ISDN channels? A client complains about their clients sometimes getting a busy tone when trying to call them. I suspect they just need to add an additional ISDN2 line but I need some conclusive information that they are indeed maxing out their ISDN channels. Thanks, Patrick