Re: [asterisk-users] two questions regarding incoming call
Update,
My first question solved already.
There was an error on my agi script.
But second problem still valid.
On Tuesday, March 01, 2011 11:04:50 am Oguzhan Kayhan wrote:
Hello,
I want to make an agi script to match incoming DIDs with usernames.
I tried to do such entry in incoming trunk.
[DID_diddw]
include = from-didww
[from-didww]
exten = 3130XXX,1,AGI(did.php)
exten = 3130XXX,n,DIAL(SIP/${yup_no},20)
but when i run the rule it says
chan_sip.c:20152 handle_request_invite: Call from '81.85.224.41' to
extension '3130111' rejected because extension not found in context
'from-didww' Cant I use such agi scripts on incoming calls?
PS:
exten = 3130XXX,n,DIAL(SIP/) works alone.
My second question.
I got two incoming trunk sip channels on my server.
One of them is as follows.
[46.19.209.1]
host = 46.19.209.1
type = friend
insecure = invite
context = from-didww
canreinvite=no
The other is as follows:
[62.180.237.73]
host = 62.180.237.73
type = friend
insecure = invite
context = from-btnet2
canreinvite = no
The problem is, i get all calls coming from trunk1(didww) without a problem
but, when i receive a call from trunk2(btnet) it tries to authenticate the
sip call and denies it. It works only if i allow guest calls.
What can be the reason for that?
Thank you.
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Re: [asterisk-users] two questions regarding incoming call
You don't need to put quotes around AGI name.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Oguzhan Kayhan
Sent: Tuesday, March 01, 2011 2:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] two questions regarding incoming call
Hello,
I want to make an agi script to match incoming DIDs with usernames.
I tried to do such entry in incoming trunk.
[DID_diddw]
include = from-didww
[from-didww]
exten = 3130XXX,1,AGI(did.php)
exten = 3130XXX,n,DIAL(SIP/${yup_no},20)
but when i run the rule it says
chan_sip.c:20152 handle_request_invite: Call from '81.85.224.41' to
extension '3130111' rejected because extension not found in context
'from-didww'
Cant I use such agi scripts on incoming calls?
PS:
exten = 3130XXX,n,DIAL(SIP/) works alone.
My second question.
I got two incoming trunk sip channels on my server.
One of them is as follows.
[46.19.209.1]
host = 46.19.209.1
type = friend
insecure = invite
context = from-didww
canreinvite=no
The other is as follows:
[62.180.237.73]
host = 62.180.237.73
type = friend
insecure = invite
context = from-btnet2
canreinvite = no
The problem is, i get all calls coming from trunk1(didww) without a problem
but, when i receive a call from trunk2(btnet) it tries to authenticate the
sip call and denies it. It works only if i allow guest calls.
What can be the reason for that?
Thank you.
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [Asterisk-Users] Two questions about [EMAIL PROTECTED] and backups.
Thanks to all who responded. Jim. Alex Robar wrote: Jim, There are SourceForge.net forums for [EMAIL PROTECTED] where you'll probably find better answers to your AAH questions. They are located here: https://sourceforge.net/forum/?group_id=123387 https://sourceforge.net/forum/?group_id=123387 In terms of backup, AAH has a built-in backup feature as part of the FreePBX GUI. Set your schedule, tell it which components (configs, voicemails, etc) you want to backup, and it'll generate a tarball that you can pull off the system at your leisure. Alex ___ --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] Two questions about [EMAIL PROTECTED] and backups.
[EMAIL PROTECTED] is welcome as long as you are referring to the asterisk portion of it and not the GUI or dialplans that make [EMAIL PROTECTED] different from the typical asterisk. I believe [EMAIL PROTECTED] offers a backup button that will backup all pertinent files for you. I.e. dialplan, modules, sip.conf, etc. bp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Lynch Sent: Friday, May 26, 2006 10:39 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Two questions about [EMAIL PROTECTED] and backups. First question, is there a forum for [EMAIL PROTECTED] specific questions? I've asked what must have been questions about [EMAIL PROTECTED] here and gotten some indication they weren't welcome. Second, does anyone know what files need to be backed up? I don't need to back up the entire system since I can reinstall from the CD in fairly quick order, however, other than the files in /etc, where else does asterisk keep files that need to be backed up? Thanks, Jim. ___ --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 __ NOD32 1.1443 (20060314) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --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] Two questions about [EMAIL PROTECTED] and backups.
Jim,There are SourceForge.net forums for [EMAIL PROTECTED] where you'll probably find better answers to your AAH questions. They are located here: https://sourceforge.net/forum/?group_id=123387 In terms of backup, AAH has a built-in backup feature as part of the FreePBX GUI. Set your schedule, tell it which components (configs, voicemails, etc) you want to backup, and it'll generate a tarball that you can pull off the system at your leisure. AlexOn 5/26/06, Jim Lynch [EMAIL PROTECTED] wrote: First question, is there a forum for [EMAIL PROTECTED] specific questions?I've asked what must have been questions about [EMAIL PROTECTED] here andgotten some indication they weren't welcome.Second, does anyone know what files need to be backed up? I don't need to back up the entire system since I can reinstall from the CD in fairlyquick order, however, other than the files in /etc, where else doesasterisk keep files that need to be backed up?Thanks,Jim. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar [EMAIL PROTECTED] ___ --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] Two questions about [EMAIL PROTECTED] and backups.
You might try these sites: http://sourceforge.net/forum/forum.php?forum_id=420324 Backup has been discussed many times here. Unfortunately, the SF forums suck in terms of searching. http://www.freepbx.org/ http://aussievoip.com.au/wiki/index.php?page=FreePBX https://sourceforge.net/projects/amportal/ http://voipspeak.net/forum/ http://nerdvittles.com/ and sometimes, http://forums.whirlpool.net.au/forum-threads.cfm?f=107 Doug At 09:38 AM 5/26/2006, you wrote: First question, is there a forum for [EMAIL PROTECTED] specific questions? I've asked what must have been questions about [EMAIL PROTECTED] here and gotten some indication they weren't welcome. Second, does anyone know what files need to be backed up? I don't need to back up the entire system since I can reinstall from the CD in fairly quick order, however, other than the files in /etc, where else does asterisk keep files that need to be backed up? Thanks, Jim. ___ --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] Two questions about Asterisk Call Center
Hello, routing based on DNIS is dependant on what your telco sends you. Usually on Robbed-bit T1s(RBS) they will send you ANI and DNIS together separated by stars like this: *7275551212*1234* (where 7275551212 is the ANI[callerID] and 1234 is the DNIS[last 4 digits of the number dialed]) In Asterisk this shows up all as the exten and you need *NXXNXX*1234 in your dialplan. If you have PRI T1s then you can usually receive both the CallerID and the full 10-digit number dialed from the carrier and you will get the full number dialed as the extension, so 8881231234 in your dialplan. Collecting wrapup codes is another thing. This means you need a database for the calls coming in and in case of Asterisk that means tinkering with the code. There are several add-ons that add this functionality to Asterisk and some of them cost money, just do a search for queues and agents in Asterisk on google. Or you could go with a package like Aheeva or VICIDIAL that have GUI interfaces and allow you a great deal more interoperability with other systems and the ability for the agent to enter more info. MATT--- -Original Message- From: Tielin Xu [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 02, 2005 2:26 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Two questions about Asterisk Call Center Hi: I am new at Asterisk. Does anyone know how to define the call routing based on DNIS as our conventional ACD to route a call in Asterisk? Second, how do I collect Wrap-Up code for agents in Asterisk? Many thanks. Tielin ___ 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 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] two questions about the Sipura 841?
What is the purpose of the beeping? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joel Duffield Sent: Saturday, May 07, 2005 12:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] two questions about the Sipura 841? Ok my first question is I have seen messages about a patch for asterisk so that I can do auto answer on these phones. I found the message in the archives but I do not have that message as an email still, so I do not have the attachment. Can anyone tell me where to get it? Also on this phone how can I set the phone to release the line sooner without playing the anoying beeping for 5 seconds, I can change how long until the beeping starts but how do I shorten the beeps? Thanks Joel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.6 - Release Date: 5/6/2005 ___ 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 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] two questions about the Sipura 841?
The beeping is to tell you that the remote end has hungup, im sorry I don't know the technical term for it but it happens on your regular home phone if the other end was to hang up and you did not hang up your receiver. the web interface calls it the Reorder. Thanks Joel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jim Sturtevant Sent: Saturday, May 07, 2005 4:25 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] two questions about the Sipura 841? What is the purpose of the beeping? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joel Duffield Sent: Saturday, May 07, 2005 12:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] two questions about the Sipura 841? Ok my first question is I have seen messages about a patch for asterisk so that I can do auto answer on these phones. I found the message in the archives but I do not have that message as an email still, so I do not have the attachment. Can anyone tell me where to get it? Also on this phone how can I set the phone to release the line sooner without playing the anoying beeping for 5 seconds, I can change how long until the beeping starts but how do I shorten the beeps? Thanks Joel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.6 - Release Date: 5/6/2005 ___ 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 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.6 - Release Date: 5/6/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.6 - Release Date: 5/6/2005 ___ 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] two questions
Christopher L. Wade schrieb: Please understand that Digium (Mark) is the reason * exists. That's clear to me. Selling a $10 winmodem for $80 is the reason Digium exists. But the difference between $10 and $80 is really to much for me for just playing around like I'm doing with asterisk. If I would use it in a productive system - like at work - it would be no problem. But at home? No. I would buy an original Digium card when a X100P card would cost maybe $20 for non commercial usage and in low quantities. (And then without commercial support) Digium's existence is the reason Mark can eat while he's coding for *. Mark being able to code * while he eats is the reason * is such a great project. Way to go Mark, et. al. (even the ones who don't get paid to eat while coding :) I surely understand that. I'm working as a programmer as well. There must be a way to get the money you need. Via software or via hardware. But I wouldn't bought asterisk if it wasn't free. Then I had bought a simple SIP-adapter and had connected it with my phone. Bye! Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] two questions
On Tuesday 07 December 2004 04:36, Erick Perez wrote: Hi people, question one i see that asterisk is now in 1.x release. having tried it in the past i want to know if i can use a voice modem as an outgoing line. i know in the past that was not possible/supported so im just asking in case the option is now available. yes, if that voice modem is a x100p or clone (same chipset). question two im planing to use asterisk as a pure voip solution with sip phones and h323 phones no need for digium/dialogic hardware at this moment (but i will in the near future). however i have not been able to find a documentation (not so complicated for a newbie) that help me to setup asterisk in this mode. suggestion/comments/flames welcomed. see www.voip-info.org Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] two questions
I see that the 100p is a modem with an Ambient chipset. Why does it sell for 80$ in some places? i can get Ambient pci modem down here for 9 dollars. Any difference? On Tue, 7 Dec 2004 10:58:55 +, Jon Lawrence [EMAIL PROTECTED] wrote: On Tuesday 07 December 2004 04:36, Erick Perez wrote: Hi people, question one i see that asterisk is now in 1.x release. having tried it in the past i want to know if i can use a voice modem as an outgoing line. i know in the past that was not possible/supported so im just asking in case the option is now available. yes, if that voice modem is a x100p or clone (same chipset). question two im planing to use asterisk as a pure voip solution with sip phones and h323 phones no need for digium/dialogic hardware at this moment (but i will in the near future). however i have not been able to find a documentation (not so complicated for a newbie) that help me to setup asterisk in this mode. suggestion/comments/flames welcomed. see www.voip-info.org Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] two questions
Erick Perez wrote: I see that the 100p is a modem with an Ambient chipset. Why does it sell for 80$ in some places? i can get Ambient pci modem down here for 9 dollars. Any difference? Because Digium is selling support plus the modem, not just the modem. -Chris -- Christopher L. Wade Unistar-Sparco Computers, Inc. Senior Systems Administratordba Sparco.com Email: [EMAIL PROTECTED] 7089 Ryburn Drive Phone: (901) 872 2272 / (800) 840 8400Millington, TN 38053 Fax: (901) 872 8482 USA ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] two questions
Christopher L. Wade schrieb: Because Digium is selling support plus the modem, not just the modem. But when you don't need the support? Bye! Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] two questions
Michael Vogel wrote: Christopher L. Wade schrieb: Because Digium is selling support plus the modem, not just the modem. But when you don't need the support? Bye! Michael Exactly. Choose the level of support you want from Digium and/or the list. Historically, Digium equipment gets support from both Digium and the list, non-Digium equipment only gets support from the list, and only when a valid reason for not using Digium exists. Please understand that Digium (Mark) is the reason * exists. Selling a $10 winmodem for $80 is the reason Digium exists. Digium's existence is the reason Mark can eat while he's coding for *. Mark being able to code * while he eats is the reason * is such a great project. Way to go Mark, et. al. (even the ones who don't get paid to eat while coding :) My $0.50. -Chris -- Christopher L. Wade Unistar-Sparco Computers, Inc. Senior Systems Administratordba Sparco.com Email: [EMAIL PROTECTED] 7089 Ryburn Drive Phone: (901) 872 2272 / (800) 840 8400Millington, TN 38053 Fax: (901) 872 8482 USA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] two questions
Hello Christopher L. Wade, I must say I 100% agree. I bought a cheap $10 because it is just for fun to test it out. If I put it in my work I will definatley go with the digium true hardware to support the company that put out the software for free. You can pick up their other hardware at reasonable prices if you compare it to others on the market. Best regards, === At 2004-12-07, 09:08:06 you wrote: === Michael Vogel wrote: Christopher L. Wade schrieb: Because Digium is selling support plus the modem, not just the modem. But when you don't need the support? Bye! Michael Exactly. Choose the level of support you want from Digium and/or the list. Historically, Digium equipment gets support from both Digium and the list, non-Digium equipment only gets support from the list, and only when a valid reason for not using Digium exists. Please understand that Digium (Mark) is the reason * exists. Selling a $10 winmodem for $80 is the reason Digium exists. Digium's existence is the reason Mark can eat while he's coding for *. Mark being able to code * while he eats is the reason * is such a great project. Way to go Mark, et. al. (even the ones who don't get paid to eat while coding :) My $0.50. -Chris -- Christopher L. Wade Unistar-Sparco Computers, Inc. Senior Systems Administratordba Sparco.com Email: [EMAIL PROTECTED] 7089 Ryburn Drive Phone: (901) 872 2272 / (800) 840 8400Millington, TN 38053 Fax: (901) 872 8482 USA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = = = = = = = = = = = = = = = = = = = = xpollyspies-koswald [EMAIL PROTECTED] 2004-12-07 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Two * Questions
On Mon, 2 Jun 2003, Jason Smith wrote: We're getting ready to ditch our hosted virtual PBX for an Asterisk YAY! more power to ya... 1. Is it possible to have duplicate extensions between the two PBXs? Eg: 555-x100 and 555-x100 on same * server don't see any reason why not... 2. Is there a way to configure follow-me-calling to have an extension attempt to ring a series of numbers, eventually dropping to voicemail? oh, most definitely... priority levels on a particular extension will make this trivial, examples abound on the list, even in the sample confs, i think.. - wasim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
