Re: [asterisk-users] Help With dial plan
maybe the user is dialing something other than 3000 and that extension is not registered on your asterisk. just a wild guess. On Tue, Jul 22, 2008 at 10:41 AM, James Mutuku [EMAIL PROTECTED] wrote: Hi list, Have installed trixbox and I am working with a fxo gateway to get fxo calls to trixbox. I am using sip to send the calls from the gateway to trixbox. I have an extension 3000 on trixbox on [from-sip-external] on extensions.conf ,I have put the dial plan below. exten = 3000,1,dial(sip/3000) exten= 3000,2,answer() exten = 3000,3,congestion() exten= 3000,4,hangup() this works fine. But I when I put it in the form exten = _3XXX,1,dial(sip/${EXTEN}) exten= _3XXX,2,answer() exten =_3XXX,3,congestion() exten= _3XXX,4,hangup() the call goes into congestion and I get a busy tone. What could I be doing wrong? James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help With dial plan
James Mutuku wrote: Hi list, Have installed trixbox and I am working with a fxo gateway to get fxo calls to trixbox. I am using sip to send the calls from the gateway to trixbox. I have an extension 3000 on trixbox on [from-sip-external] on extensions.conf ,I have put the dial plan below. exten = 3000,1,dial(sip/3000) exten= 3000,2,answer() exten = 3000,3,congestion() exten= 3000,4,hangup() this works fine. But I when I put it in the form exten = _3XXX,1,dial(sip/${EXTEN}) exten= _3XXX,2,answer() exten =_3XXX,3,congestion() exten= _3XXX,4,hangup() the call goes into congestion and I get a busy tone. What could I be doing wrong? James As a sanity check, you may want to place a NoOp(${EXTEN}) prior to the dial. If you set the verbosity high on the Asterisk console, then you can see what the value of EXTEN is when the NoOp occurs. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help With dial plan
Thanks for the wild guess. But The user(who is myself) is dialing 3000. It only failes to work when I use patterns. So I thought I am making a mistake on the syntax, I have checked all the books I have and the internet and I can't see anything wrong. :-\ Rizwan Hisham wrote: maybe the user is dialing something other than 3000 and that extension is not registered on your asterisk. just a wild guess. On Tue, Jul 22, 2008 at 10:41 AM, James Mutuku [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi list, Have installed trixbox and I am working with a fxo gateway to get fxo calls to trixbox. I am using sip to send the calls from the gateway to trixbox. I have an extension 3000 on trixbox on [from-sip-external] on extensions.conf ,I have put the dial plan below. exten = 3000,1,dial(sip/3000) exten= 3000,2,answer() exten = 3000,3,congestion() exten= 3000,4,hangup() this works fine. But I when I put it in the form exten = _3XXX,1,dial(sip/${EXTEN}) exten= _3XXX,2,answer() exten =_3XXX,3,congestion() exten= _3XXX,4,hangup() the call goes into congestion and I get a busy tone. What could I be doing wrong? James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya email;internet:[EMAIL PROTECTED],[EMAIL PROTECTED] title:Lead Consultant tel;work:+254-722-490994 tel;home:+254-722-490994 tel;cell:+254-722-490994 url:www.agile.co.ke version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help With dial plan
Hi James - Thanks for the wild guess. But The user(who is myself) is dialing 3000. It only failes to work when I use patterns. So I thought I am making a mistake on the syntax, I have checked all the books I have and the internet and I can't see anything wrong. :-\ Sounds like time for some more in depth troubleshooting. What happens when you follow Mark's suggestion of adding a NoOp statement? What happens when you create other pattern-match extensions? Do they work? What messages are you getting on the console? Is the call being rejected by the SIP device? What messages do you get when SIP debugging is turned on? etc, blah, blah, blah... - Noah Rizwan Hisham wrote: maybe the user is dialing something other than 3000 and that extension is not registered on your asterisk. just a wild guess. On Tue, Jul 22, 2008 at 10:41 AM, James Mutuku [EMAIL PROTECTED] wrote: Hi list, Have installed trixbox and I am working with a fxo gateway to get fxo calls to trixbox. I am using sip to send the calls from the gateway to trixbox. I have an extension 3000 on trixbox on [from-sip-external] on extensions.conf ,I have put the dial plan below. exten = 3000,1,dial(sip/3000) exten= 3000,2,answer() exten = 3000,3,congestion() exten= 3000,4,hangup() this works fine. But I when I put it in the form exten = _3XXX,1,dial(sip/${EXTEN}) exten= _3XXX,2,answer() exten =_3XXX,3,congestion() exten= _3XXX,4,hangup() the call goes into congestion and I get a busy tone. What could I be doing wrong? James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with dial plan
Hi list, Have installed trixbox and I am working with a fxo gateway to get fxo calls to trixbox. I am using sip to send the calls from the gateway to trixbox. I have an extension 3000 on trixbox on [from-sip-external] on extensions.conf ,I have put the dial plan below. exten = 3000,1,dial(sip/3000) exten= 3000,2,answer() exten = 3000,3,congestion() exten= 3000,4,hangup() this works fine. But I when I put it in the form exten = _3XXX,1,dial(sip/${EXTEN}) exten= _3XXX,2,answer() exten =_3XXX,3,congestion() exten= _3XXX,4,hangup() the call goes into congestion and I get a busy tone. What could I be doing wrong? James begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya email;internet:[EMAIL PROTECTED],[EMAIL PROTECTED] title:Lead Consultant tel;work:+254-722-490994 tel;home:+254-722-490994 tel;cell:+254-722-490994 url:www.agile.co.ke version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help With dial plan
Hi list, Have installed trixbox and I am working with a fxo gateway to get fxo calls to trixbox. I am using sip to send the calls from the gateway to trixbox. I have an extension 3000 on trixbox on [from-sip-external] on extensions.conf ,I have put the dial plan below. exten = 3000,1,dial(sip/3000) exten= 3000,2,answer() exten = 3000,3,congestion() exten= 3000,4,hangup() this works fine. But I when I put it in the form exten = _3XXX,1,dial(sip/${EXTEN}) exten= _3XXX,2,answer() exten =_3XXX,3,congestion() exten= _3XXX,4,hangup() the call goes into congestion and I get a busy tone. What could I be doing wrong? James begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya email;internet:[EMAIL PROTECTED],[EMAIL PROTECTED] title:Lead Consultant tel;work:+254-722-490994 tel;home:+254-722-490994 tel;cell:+254-722-490994 url:www.agile.co.ke version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with dial plan - two attempts at calling agent before logging agent off?
Hi List, I'm attempting to set up a queue and agents using agent call back. This is all working fine with the queue and the agents login etc However. In my dial plan I a set variable when a call is entered into the queue to identify the origin of the call, then when the agent is called I test to see if the call is from the queue. If it is, the dial plan does not go to VM if the agent does not answer, it gives BUSY and the call is returned to the queue. The call could well be passed to the same agent again from the queue, which I am okay with - BUT I only want it to try twice before logging the agent out (just in case they have gone AWOL and not logged out). The autologoff=xx in agents.conf doesn't seem to work with agentcallback. I have tried setting another variable as a counter with some logic tests to see the number of attempts to call the agent, but this is failing as the variable appears to be lost when the call goes back to the queue. Can anyone suggest an answer to this puzzle for me. Many thanks Chris -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with dial plan - two attempts at calling agent before logging agent off?
snip I have tried setting another variable as a counter with some logic tests to see the number of attempts to call the agent, but this is failing as the variable appears to be lost when the call goes back to the queue. Local variables are destroyed once the call terminates. You'll have to use a global variable (yuck) or use the DB functions. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help on dial plan
The following is my dialplan for outgoing international call. What I want are: - when people dial 9011604xxx , 9011605xxx, 90114411xxx, 90114421xxx, use voipstunt to dial out - otherwise, use my pstn to dial out. What I've found is when i dial 9011604xxx , 9011605xxx, 90114411xxx, 90114421xxx, it always use the pstn to dial out. Anything wrong with my dial plan? Thanks!! [outbound-oversea] exten = _9011604.,1,Macro(outgoingcall3,${VOIPSTUNT},4) ;penang exten = _9011605.,1,Macro(outgoingcall3,${VOIPSTUNT},4) ;perak exten = _90114411.,1,Macro(outgoingcall3,${VOIPSTUNT},4) ;uk landline exten = _90114421.,1,Macro(outgoingcall3,${VOIPSTUNT},4) ;uk landline exten = _9011.,1,Macro(outgoingcall,${OUTBOUNDTRUNK}) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help on dial plan
On Sun, 2006-02-12 at 10:05 -0500, Wooi Koay wrote: The following is my dialplan for outgoing international call. What I want are: - when people dial 9011604xxx , 9011605xxx, 90114411xxx, 90114421xxx, use voipstunt to dial out - otherwise, use my pstn to dial out. What I've found is when i dial 9011604xxx , 9011605xxx, 90114411xxx, 90114421xxx, it always use the pstn to dial out. Anything wrong with my dial plan? Thanks!! [outbound-oversea] exten = _9011604.,1,Macro(outgoingcall3,${VOIPSTUNT},4) ;penang exten = _9011605.,1,Macro(outgoingcall3,${VOIPSTUNT},4) ;perak exten = _90114411.,1,Macro(outgoingcall3,${VOIPSTUNT},4) ;uk landline exten = _90114421.,1,Macro(outgoingcall3,${VOIPSTUNT},4) ;uk landline exten = _9011.,1,Macro(outgoingcall,${OUTBOUNDTRUNK}) Asterisk evalutates your extens from least to most significant. So the _9011. is always selected first. To overcome this you could do: [outbound-oversea] exten = _9011604.,1,Macro(outgoingcall3,${VOIPSTUNT},4) ;penang exten = _9011605.,1,Macro(outgoingcall3,${VOIPSTUNT},4) ;perak exten = _90114411.,1,Macro(outgoingcall3,${VOIPSTUNT},4) ;uk landline exten = _90114421.,1,Macro(outgoingcall3,${VOIPSTUNT},4) ;uk landline include = outbound-oversea-zap-backup [outbound-oversea-zap-backup] exten = _9011.,1,Macro(outgoingcall,${OUTBOUNDTRUNK}) If you do show dialplan outbound-oversea in the Asterisk CLI than you will see the order. For more info see: http://www.voip-info.org/wiki/index.php?page=Asterisk+config +extensions.conf+sorting Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with Dial Plan
Thanks Steve, the 'w's worked great. I managed to tune it down to them only hearing a please wait out of the greeting.. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, October 19, 2005 4:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Help with Dial Plan On Wed, 19 Oct 2005, Dave Morrow wrote: Thanks Steve. It almost works, but never dials the extension. Also, is there a way I could mute the line while the remote attendant comes on? Oops sorry - the dangers of posting without testing. The ,s are wrong - they should be w. Each w is 1/2 second of waiting. So that makes it: exten = _6XXX,1,Dial(Zap/gX/1234567890,60,D(${EXTEN})) As for the muting - bit of a loss about that one. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with Dial Plan
Title: Help with Dial Plan Hi all. So far this list is proving it's worth, even on my first day using it! I hope that someone might know an easy solution to this one. I would like to create a dial plan which will allow me to have all extensions 6XXX cause a dial-out of my T1 interface (TE110P) to a local number, wait for an answer, wait 2 seconds and then enter the extension. Can I do this in a dial plan somehow? This will allow me to pseudo-integrate a legacy telephone switch (whose extensions are all 6XXX) to my Asterisk system for direct extension dialing. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with Dial Plan
On Wed, 19 Oct 2005, Dave Morrow wrote: Hi all. So far this list is proving it's worth, even on my first day using it! I hope that someone might know an easy solution to this one. I would like to create a dial plan which will allow me to have all extensions 6XXX cause a dial-out of my T1 interface (TE110P) to a local number, wait for an answer, wait 2 seconds and then enter the extension. Can I do this in a dial plan somehow? This will allow me to pseudo-integrate a legacy telephone switch (whose extensions are all 6XXX) to my Asterisk system for direct extension dialing. exten = _6XXX,1,Dial(Zap/gX/1234567890,60,D(,,${EXTEN})) where: gX needs to become the group of the channels of your T1, 1234567890 is the number of your legacy system. 60 is the dial timeout You may need to adjust the number of commas to get the right delay. Hope that helps, Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with Dial Plan
Thanks Steve. It almost works, but never dials the extension. Also, is there a way I could mute the line while the remote attendant comes on? David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, October 19, 2005 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Help with Dial Plan On Wed, 19 Oct 2005, Dave Morrow wrote: Hi all. So far this list is proving it's worth, even on my first day using it! I hope that someone might know an easy solution to this one. I would like to create a dial plan which will allow me to have all extensions 6XXX cause a dial-out of my T1 interface (TE110P) to a local number, wait for an answer, wait 2 seconds and then enter the extension. Can I do this in a dial plan somehow? This will allow me to pseudo-integrate a legacy telephone switch (whose extensions are all 6XXX) to my Asterisk system for direct extension dialing. exten = _6XXX,1,Dial(Zap/gX/1234567890,60,D(,,${EXTEN})) where: gX needs to become the group of the channels of your T1, 1234567890 is the number of your legacy system. 60 is the dial timeout You may need to adjust the number of commas to get the right delay. Hope that helps, Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with Dial Plan
On Wed, 19 Oct 2005, Dave Morrow wrote: Thanks Steve. It almost works, but never dials the extension. Also, is there a way I could mute the line while the remote attendant comes on? Oops sorry - the dangers of posting without testing. The ,s are wrong - they should be w. Each w is 1/2 second of waiting. So that makes it: exten = _6XXX,1,Dial(Zap/gX/1234567890,60,D(${EXTEN})) As for the muting - bit of a loss about that one. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users