Re: [asterisk-users] CallerID shows wrong values in manager interface
Thanks all :) Appreciate it. On Feb 1, 2008 12:04 PM, Ex Vito [EMAIL PROTECTED] wrote: I've struggled with this recently. In short: - Observed behaviour is expected as of asterisk 1.2 and later, as previously described by Mojo - If you want to get the caller id for the channel calling (dialling) into that channel for that specific Newstate: Ringing event, you can use the 'o' flag to the Dial command; in this case you'll get old asterisk 1.0 behaviour -- do you really want to depend on such an old behaviour ? well I decided I didn't... - Otherwise, you'll need to track other events (IIRC, at least, Dial, AgentCalled, Newstate, etc) in the AMI so as to know who is calling who at a given instant - BEWARE: if memory serves me right (search the list archives in the Nov/Dec timeframe), the behaviour is not 100% homogeneous for different channel types SIP, ZAP, mISDN, IAX, etc. What this means for a simple Dial() from one channel to the other is that a) at times you get the Dial event first then the Newstate: Ringing event; and that b) with other/different orig/dest channel types you'll get the events in the reverse order... Nothing much but: i) you'll have to track them either way and ii) it reveals that the AMI events aren't 100% clean!!! :/ -- exvito On Feb 1, 2008 12:08 AM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: The snippet is asterisk telling you I'm just letting you know that the correct caller id for Channel: SIP/103-098500d8 is CallerID: 103 This is absolutely correct, it's just not a piece of information you expected to be receiving at that point. You probably also received a packet like that with the following: Channel: SIP/101- CallerID: 101 telling you, again, the caller id for only that channel. Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallerID shows wrong values in manager interface
Hi everyone, My manager interface seems to be producing wrong CallerIDs when internal extensions call each other. Can anyone see anything wrong in the configuration snippets pasted below? The following instance has extension 101 call 103. The phone does show the right caller ID, but notice that the manager interface has the CallerID as the target number (103). Thanks a lot for your time. Manager interface output: CallerIDName: unknown State: Ringing Event: Newstate Privilege: call,all Uniqueid: 1201748091.843 Channel: SIP/103-098500d8 CallerID: 103 SIP.conf snippets: [101] type=friend callerid=(Devraj Mukherjee 101) username=101 secret=password context=default host=dynamic allow=alaw [EMAIL PROTECTED] [103] type=friend callerid=(System admin Den 103) username=103 secret=password context=default host=dynamic allow=all [EMAIL PROTECTED] Extension.conf looks like: ; Standard POTS line configuration to pickup calls exten = _s,1,Wait(2) exten = _s,2,Queue(wagga-office-phones,90) exten = _s,3,VoiceMail([EMAIL PROTECTED]) exten = _s,4,Hangup exten = 101,1,Wait(1) exten = 101,2,SetCIDNum(101) exten = 101,3,Dial(SIP/101,30,trw) exten = 101,4,Voicemail(s101) exten = 101,5,Hangup exten = 103,1,Wait(1) exten = 103,2,Dial(SIP/103,30,trw) exten = 103,3,Voicemail(s103) exten = 103,4,Hangup -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I am having a problem connecting my X-Lite to my Asterix box
Does Asterisk give you any feedback on the console? On Jan 21, 2008 12:14 PM, Andrew Ladanowski [EMAIL PROTECTED] wrote: I have added two extentsions. I am try to test connecting X-lite to the server. I have two extension one 1000 with password 1234 and one 2000 with password 2000. I have the softphone on the same network so I do not have to worry about ports being open. So I have in the properties of Account Display Name: Andrew Username :1000 Password: 1234 Authorization name :1000 Domain:192.168.3.128 I have tried domain Proxy on and off When on I enter proxy as 192.168.3.128 I even added port 5060 to both IP's any one have some clue on this one. Andrew Ladanowski AddInSolutions Inc. www.addinsol.com [EMAIL PROTECTED] Phone: 954-815-2402 Fax: 954-414-8432 CONFIDENTIAL : The information in this email (including any attachments) is confidential and may be privileged. If you are not the intended recipient, you may not and must not read, print, forward, use or disseminate the information contained herein. Although this email (and any attachments) are believed to be free of any virus or other defect that might affect any computer system into which it is received and opened, it is the responsibility of the recipient to ensure that it is free of viruses or defects and no responsibility is accepted by the sender for any loss or damage arising or resulting in any way from its receipt or use. If you are not the intended recipient of this message, please reply to the sender and include this message and then delete this message from your inbox and your archive and/or discarded messages files. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I am having a problem connecting my X-Litetomy Asterix box
Which Linux distribution are you using? SSH for root might be denied in your setup On Jan 21, 2008 1:29 PM, Andrew Ladanowski [EMAIL PROTECTED] wrote: I can not exit out of my Asterisk set up it. When I try to login to my server using ssh in denies the username and password. I assume the default name was root when I set up the Asterisk. I remember the password. Andrew Ladanowski AddInSolutions Inc. www.addinsol.com [EMAIL PROTECTED] Phone: 954-815-2402 Fax: 954-414-8432 CONFIDENTIAL : The information in this email (including any attachments) is confidential and may be privileged. If you are not the intended recipient, you may not and must not read, print, forward, use or disseminate the information contained herein. Although this email (and any attachments) are believed to be free of any virus or other defect that might affect any computer system into which it is received and opened, it is the responsibility of the recipient to ensure that it is free of viruses or defects and no responsibility is accepted by the sender for any loss or damage arising or resulting in any way from its receipt or use. If you are not the intended recipient of this message, please reply to the sender and include this message and then delete this message from your inbox and your archive and/or discarded messages files. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson Sent: Sunday, January 20, 2008 9:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] I am having a problem connecting my X-Litetomy Asterix box On Jan 20, 2008 8:06 PM, Andrew Ladanowski [EMAIL PROTECTED] wrote: Windows XP. Andrew - you're going to need to get us your sip.conf before we can really assist you any further. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk desktop tools for OS X
Hi everyone, I have been long working on a project (http://asterisktools.org, to be released under GPL) that aims to provide desktop tools for Macs. I am finally getting to the release stages of this application and hope to have an early BETA available next weekend. If there is anybody who is interested in this tool, please send me an email as I am looking for people who can test the application for me before we make a final release. The code is already available via SVN and there are some really cool and thoughtful features. Thanks a lot. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk desktop tools for OS X
Thanks for your response guys. There are still some issues with the code (Svn on SourceForge). I am working on getting these fixed up and will post a message when its ready for download. I will yell out if I need some Asterisk/Cocoa help. Thanks a lot. On Jan 18, 2008 7:19 AM, Adrià Vidal [EMAIL PROTECTED] wrote: I'm interested too Devraj, please send a copy of if possible to try it. Thanks. On Jan 17, 2008 12:25 PM, Devraj Mukherjee [EMAIL PROTECTED] wrote: Hi everyone, I have been long working on a project ( http://asterisktools.org, to be released under GPL) that aims to provide desktop tools for Macs. I am finally getting to the release stages of this application and hope to have an early BETA available next weekend. If there is anybody who is interested in this tool, please send me an email as I am looking for people who can test the application for me before we make a final release. The code is already available via SVN and there are some really cool and thoughtful features. Thanks a lot. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Adrià Vidal [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk desktop tools for OS X
Hi Tzafrir, Yes it does use the Manager Interface. It account does require call level access. That may then result in umlimited access to Asterisk (well to originate calls anyway). However I have made real conscious efforts to filter messages that are being transmitted over the socket so the application doesn't listen or talk on behalf of a single extension. If this is a concern, is every desktop application that integrates using the Manager Interface a problem for Asterisk administrators? Also, what is a way around it then? I see desktop tools for Asterisk being one of the biggest advantages over traditional PBXes. On Jan 18, 2008 7:19 AM, Adrià Vidal [EMAIL PROTECTED] wrote: I'm interested too Devraj, please send a copy of if possible to try it. Thanks. On Jan 17, 2008 12:25 PM, Devraj Mukherjee [EMAIL PROTECTED] wrote: Hi everyone, I have been long working on a project ( http://asterisktools.org, to be released under GPL) that aims to provide desktop tools for Macs. I am finally getting to the release stages of this application and hope to have an early BETA available next weekend. If there is anybody who is interested in this tool, please send me an email as I am looking for people who can test the application for me before we make a final release. The code is already available via SVN and there are some really cool and thoughtful features. Thanks a lot. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Adrià Vidal [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Asterisk on to CentOS 4
I installed it using yum from the atrpms repo and it all seems to work. Did you compile from source? On 9/11/07, Abdul [EMAIL PROTECTED] wrote: Hi expets, I have installed Asterisk 1.4.11 on CentOS4 successfully without any error. But when i am trying to start asterisk with following cmd i am getting unknown command. [EMAIL PROTECTED] ~]$ asterisk -vvc -bash: asterisk: command not found [EMAIL PROTECTED] ~]$ I checked modules and other configuration files which are installed correctly. Please help me to locate this problem. Thank You Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make call from asterisk?
Hi Neoh, All you have to do is configure your VoIp provider as another SIP extension on your Asterisk server and then use extensions.conf to set dialout rules, so when you do dial a number your asterisk server forwards it to the VoIp provider. Examples of extensions.conf can be found at http://www.asteriskguru.com/tutorials/extensions_conf.html http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf I can send you my extensions.conf if you want a working example. I do something very similar with a VoIP provider that provides an SIP interface. Hope this helps. On 9/5/07, neoh kumyee [EMAIL PROTECTED] wrote: Hi, Thanks for your reply.. I am intend to dial using a VOIP provider.(developed by us) Software: x-Lite (SIP softphone) Registration of account number is fine, but for the case when i dial a number, it prompt out a message that the number not found. From my understanding, asterisk can be SIP server? or we need to implement a SIP server to integrate with Asterisk in order to provide full picture of VOIP system? Thanks. Date: Wed, 5 Sep 2007 13:30:21 +1000 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How to make call from asterisk? Helps us help you further, what do you intend to do? - Dial using a normal telephone line - Dial using a VoIP provider? What hardware do you have, etc On 9/5/07, neoh kumyee [EMAIL PROTECTED] wrote: Hi, I'm new to asterisk, in order to enable X-lite to make a call, what should i do before making a call? Current stage, 1. i have create a few accounts in sip.conf. 2. Registration are successful. Pls advice me how to continue then... Thanks Call and stay connected with your friends and family for free. Seen and be heard with high-definition video calls on Windows Live Messenger. Try it! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Live Search: Better results, fast Try it now! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Manager Interface, reliably monitor NewCall for an extension
Hi Everyone, I am writing an open source application that brings desktops widgets to OS X (http://sourceforge.net/projects/astrxtools4osx/), for which I am trying to get my head around the Asterisk Manager Interface. I had been using the Event: NewCallerid to detect a new call which my Asterisk server doesn't seem to send to the socket anymore, because of which I have reverted to using Event: Newexten. Which is the most efficient way of monitoring if a new phone call is coming my way? Also my application will only monitor a single extension, should I filter the requests on the client side or can a manager interface user be restricted to a single extensions events. Thanks for your time. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager Interface, reliably monitor NewCall for an extension
Hi Atis, Is your code open source, or are you willing to share your PHP code snippets with me? And thanks for the information on Asterisk's stability. Do you think there is an issue in the implementation or just network/traffic issues? Thanks for your time. On 9/4/07, Atis [EMAIL PROTECTED] wrote: On 9/4/07, Devraj Mukherjee [EMAIL PROTECTED] wrote: Hi Everyone, I am writing an open source application that brings desktops widgets to OS X (http://sourceforge.net/projects/astrxtools4osx/), for which I am trying to get my head around the Asterisk Manager Interface. I had been using the Event: NewCallerid to detect a new call which my Asterisk server doesn't seem to send to the socket anymore, because of which I have reverted to using Event: Newexten. Which is the most efficient way of monitoring if a new phone call is coming my way? Also my application will only monitor a single extension, should I filter the requests on the client side or can a manager interface user be restricted to a single extensions events. I don't know about manager, but i've done the same using PHP script that executes from dialplan before dial + ActiveMQ (message queue) + custom app. I just didn't wanted to do filtering with manager, and so on.. Additionally, from my experience, creating a bunch of manager connections isn't quite good for asterisk stability.. Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make call from asterisk?
Helps us help you further, what do you intend to do? - Dial using a normal telephone line - Dial using a VoIP provider? What hardware do you have, etc On 9/5/07, neoh kumyee [EMAIL PROTECTED] wrote: Hi, I'm new to asterisk, in order to enable X-lite to make a call, what should i do before making a call? Current stage, 1. i have create a few accounts in sip.conf. 2. Registration are successful. Pls advice me how to continue then... Thanks Call and stay connected with your friends and family for free. Seen and be heard with high-definition video calls on Windows Live Messenger. Try it! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Manager Interface, response types
Hi everyone, I am writing a project that uses the Asterisk Manager Interface to monitor events. I just wanted to confirm if the types of messages sent back by the AMI are - Event - Response - Status If there are any other can anyone please point them to me or point me to some documentation where I could read about this Thanks. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Programming with libiax2
Hi everyone, I am considering writing some code using libiax2. Are there any good resources to get started with this? Books? Sites? Thanks -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Programming with libiax2
Thanks. Got the source, will explore further :) On 7/30/07, Sylvain Boily [EMAIL PROTECTED] wrote: Hello, Le lundi 30 juillet 2007 à 14:19 +1000, Devraj Mukherjee a écrit : Hi everyone, I am considering writing some code using libiax2. Are there any good resources to get started with this? Books? Sites? There is a small example here : http://proformatique.org/spip.php?article101 Sorry it's in french because it was writing for a french magasine. Sylvain ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] extension.conf doesn't reload?
Hi everyone, I have just installed Asterisk 1.4.6 on CentOS 5. When I issues the reload command in the asterisk command prompt, it doesn't seem to read my configuration files. Any suggestions? pbx*CLI reload The 'reload' command is deprecated and will be removed in a future release. Please use 'module reload' instead. == Parsing '/etc/asterisk/cdr.conf': Found [Jul 23 14:14:54] NOTICE[28392]: cdr.c:1359 do_reload: CDR simple logging enabled. == Parsing '/etc/asterisk/dnsmgr.conf': Found == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 == Parsing '/etc/asterisk/http.conf': Found -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extension.conf doesn't reload?
Hey Bruce, Thanks for your prompt response. Your suggestion lead to me finding out that the dialplan module was not loaded. I investigated this further and found out that /etc/asterisk/asterisk.conf was looking in /usr/lib/asterisk for modules. My machine is running 64bit CentOS and has all the modules in /usr/lib64/asterisk I modified /etc/asterisk/asterisk.conf to look for modules in that directory and everything works just fine. Thanks again. On 7/23/07, Bruce Ferrell [EMAIL PROTECTED] wrote: Devraj Mukherjee wrote: Hi everyone, I have just installed Asterisk 1.4.6 on CentOS 5. When I issues the reload command in the asterisk command prompt, it doesn't seem to read my configuration files. Any suggestions? pbx*CLI reload The 'reload' command is deprecated and will be removed in a future release. Please use 'module reload' instead. == Parsing '/etc/asterisk/cdr.conf': Found [Jul 23 14:14:54] NOTICE[28392]: cdr.c:1359 do_reload: CDR simple logging enabled. == Parsing '/etc/asterisk/dnsmgr.conf': Found == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 == Parsing '/etc/asterisk/http.conf': Found Try dialplan reload ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAP device reference in Zaptel 1.4
monk*CLI zap show channels No such command 'zap show' (type 'help' for help) Does that mean I dont have ZAP support in Asterisk? On 4/4/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Apr 04, 2007 at 02:55:02PM +1000, Devraj Mukherjee wrote: Also I can cat /dev/zap/3 and /dev/zap/4 and they respond to the various changes in signals You're looking at the kernel level . Maybe it's fine there, but asterisk does not know about it. What is the output of: zap show channels What is the contents of /etc/asterisk/zapata.conf ? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] ZAP device reference in Zaptel 1.4
No I don't. So that will be my problem. Thanks. On 4/4/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: Hi On Wed, Apr 04, 2007 at 04:59:41PM +1000, Devraj Mukherjee wrote: monk*CLI zap show channels No such command 'zap show' (type 'help' for help) Does that mean I dont have ZAP support in Asterisk? Maybe. ls -l /usr/lib/asterisk/modules/chan_zap.so I also repeat my second question: What is the contents of /etc/asterisk/zapata.conf ? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] ZAP device reference in Zaptel 1.4
Hi Everyone, I am using Zaptel and Asterisk 1.4 and have a Digium card with two FXS modules. The card works and ztcfg reports that it finds the two modules. Howevery when I try and place a call through the gateway I get the following error message. I have tried to refer to the ZAP device as ZAP/g2 etc Any suggestions? Anything that's different about Zaptel 1.4? -- Executing [EMAIL PROTECTED]:1] SetCDRUserField(SIP/103-b7802230, Telstra) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/103-b7802230, ZAP/4/69223139) in new stack [Apr 4 10:47:43] WARNING[5659]: channel.c:3024 ast_request: No channel type registered for 'ZAP' [Apr 4 10:47:43] WARNING[5659]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/103-b7802230' status is 'CHANUNAVAIL' Zaptel Version: 1.4.1 Echo Canceller: MG2 Configuration == Channel map: Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 2 channels configured. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] ZAP device reference in Zaptel 1.4
Hi Eric, Thanks for your suggestion I just reinstalled Asterisk, it still doesn't seem to know anything about Zaptel. I am using CentOS and installed Asterisk using yum from ATrpms. Anything else I can try? On 4/4/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Devraj Mukherjee wrote: Hi Everyone, I am using Zaptel and Asterisk 1.4 and have a Digium card with two FXS modules. The card works and ztcfg reports that it finds the two modules. Howevery when I try and place a call through the gateway I get the following error message. I have tried to refer to the ZAP device as ZAP/g2 etc Any suggestions? Anything that's different about Zaptel 1.4? -- Executing [EMAIL PROTECTED]:1] SetCDRUserField(SIP/103-b7802230, Telstra) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/103-b7802230, ZAP/4/69223139) in new stack [Apr 4 10:47:43] WARNING[5659]: channel.c:3024 ast_request: No channel type registered for 'ZAP' [Apr 4 10:47:43] WARNING[5659]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented) You need to reinstall Asterisk. You installed Asterisk before installing Zaptel so Asterisk did not build anything that requires Zaptel. ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] ZAP device reference in Zaptel 1.4
Hi Yuan, zaptel is in fact loaded. Wonder what's going wrong? [EMAIL PROTECTED] ~]# lsmod Module Size Used by wcusb 18976 0 wctdm 34752 0 wcfxo 13472 0 wctdm24xxp 69696 0 wcte11xp 24608 0 wct1xxp15904 0 wct4xxp 229312 0 tor2 89760 0 zaptel184100 10 wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2 parport_pc 28033 0 lp 15661 0 parport38025 2 parport_pc,lp autofs423109 0 sunrpc144037 1 md5 8129 1 ipv6 242657 28 crc_ccitt 6209 1 zaptel dm_mirror 31901 0 dm_mod 60741 1 dm_mirror button 10705 0 battery12997 0 ac 8901 0 joydev 14337 0 uhci_hcd 32857 0 ehci_hcd 32325 0 tg3 101189 0 e1000 109369 0 floppy 58193 0 ext3 119113 2 jbd59609 1 ext3 raid1 19649 3 ata_piix 15557 6 libata 67613 1 ata_piix sd_mod 20545 8 scsi_mod 117709 2 libata,sd_mod On 4/4/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Devraj Mukherjee [EMAIL PROTECTED] Date: Wed, 4 Apr 2007 11:46:11 +1000 Hi Eric, Thanks for your suggestion I just reinstalled Asterisk, it still doesn't seem to know anything about Zaptel. I am using CentOS and installed Asterisk using yum from ATrpms. Anything else I can try? Try lsmod to confirm that zaptel is indeed installed. I'm not familiar with CentOS or yum, but I assume you installed a binary package, so chan_zap.so is probably included. Hope this helps. Yuan Liu On 4/4/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Devraj Mukherjee wrote: Hi Everyone, I am using Zaptel and Asterisk 1.4 and have a Digium card with two FXS modules. The card works and ztcfg reports that it finds the two modules. Howevery when I try and place a call through the gateway I get the following error message. I have tried to refer to the ZAP device as ZAP/g2 etc Any suggestions? Anything that's different about Zaptel 1.4? -- Executing [EMAIL PROTECTED]:1] SetCDRUserField(SIP/103-b7802230, Telstra) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/103-b7802230, ZAP/4/69223139) in new stack [Apr 4 10:47:43] WARNING[5659]: channel.c:3024 ast_request: No channel type registered for 'ZAP' [Apr 4 10:47:43] WARNING[5659]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented) You need to reinstall Asterisk. You installed Asterisk before installing Zaptel so Asterisk did not build anything that requires Zaptel. ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] ZAP device reference in Zaptel 1.4
Also I can cat /dev/zap/3 and /dev/zap/4 and they respond to the various changes in signals On 4/4/07, Devraj Mukherjee [EMAIL PROTECTED] wrote: Hi Yuan, zaptel is in fact loaded. Wonder what's going wrong? [EMAIL PROTECTED] ~]# lsmod Module Size Used by wcusb 18976 0 wctdm 34752 0 wcfxo 13472 0 wctdm24xxp 69696 0 wcte11xp 24608 0 wct1xxp15904 0 wct4xxp 229312 0 tor2 89760 0 zaptel184100 10 wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2 parport_pc 28033 0 lp 15661 0 parport38025 2 parport_pc,lp autofs423109 0 sunrpc144037 1 md5 8129 1 ipv6 242657 28 crc_ccitt 6209 1 zaptel dm_mirror 31901 0 dm_mod 60741 1 dm_mirror button 10705 0 battery12997 0 ac 8901 0 joydev 14337 0 uhci_hcd 32857 0 ehci_hcd 32325 0 tg3 101189 0 e1000 109369 0 floppy 58193 0 ext3 119113 2 jbd59609 1 ext3 raid1 19649 3 ata_piix 15557 6 libata 67613 1 ata_piix sd_mod 20545 8 scsi_mod 117709 2 libata,sd_mod On 4/4/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Devraj Mukherjee [EMAIL PROTECTED] Date: Wed, 4 Apr 2007 11:46:11 +1000 Hi Eric, Thanks for your suggestion I just reinstalled Asterisk, it still doesn't seem to know anything about Zaptel. I am using CentOS and installed Asterisk using yum from ATrpms. Anything else I can try? Try lsmod to confirm that zaptel is indeed installed. I'm not familiar with CentOS or yum, but I assume you installed a binary package, so chan_zap.so is probably included. Hope this helps. Yuan Liu On 4/4/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Devraj Mukherjee wrote: Hi Everyone, I am using Zaptel and Asterisk 1.4 and have a Digium card with two FXS modules. The card works and ztcfg reports that it finds the two modules. Howevery when I try and place a call through the gateway I get the following error message. I have tried to refer to the ZAP device as ZAP/g2 etc Any suggestions? Anything that's different about Zaptel 1.4? -- Executing [EMAIL PROTECTED]:1] SetCDRUserField(SIP/103-b7802230, Telstra) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/103-b7802230, ZAP/4/69223139) in new stack [Apr 4 10:47:43] WARNING[5659]: channel.c:3024 ast_request: No channel type registered for 'ZAP' [Apr 4 10:47:43] WARNING[5659]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented) You need to reinstall Asterisk. You installed Asterisk before installing Zaptel so Asterisk did not build anything that requires Zaptel. ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] build rpm fails
Hi everyone, I am trying to get Asterisk 1.4 running on CentOS 4.4 (Kernel 2.6.9-42.0.10.ELsmp) and am having a lot of trouble getting asterisk running on it. I had a fair bit of success with the ATrpms binaries (Zaptel worked but asterisk failed to startup because it couldn't find the speex modules). I am trying to thus recompile the asterisk rpm for CentOS 4.4 with the least amounts of external dependencies. make rpm gives me an error saying astman could not be found. How do I build astman? Has anyone succeeded making rpm on CentOS? Any feedback is appreciated. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] Re: build rpm fails
Thanks for saving me the time. I will try and yum from ATrpms. On 3/2/07, Axel Thimm [EMAIL PROTECTED] wrote: On Fri, Mar 02, 2007 at 10:05:54AM +1100, Devraj Mukherjee wrote: Hi everyone, I am trying to get Asterisk 1.4 running on CentOS 4.4 (Kernel 2.6.9-42.0.10.ELsmp) and am having a lot of trouble getting asterisk running on it. I had a fair bit of success with the ATrpms binaries (Zaptel worked but asterisk failed to startup because it couldn't find the speex modules). Get it from here: http://atrpms.net/dist/el4/speex/, or since your using a yum based distribution, point yum to atrpms and let it do the work. I am trying to thus recompile the asterisk rpm for CentOS 4.4 with the least amounts of external dependencies. Well, you'll probably find out at the end that you need to upgrade speex to the version above. make rpm gives me an error saying astman could not be found. How do I build astman? Has anyone succeeded making rpm on CentOS? The above rpms are effectively on CentOS: They were built on RHEL, but CentOS is a clone from RHEL. -- Axel.Thimm at ATrpms.net ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] Re: build rpm fails
Hi Axel, Everything installed and working well. Thanks very much. Quick question, do you have MySQL support compiled into the rpms? On 3/2/07, Axel Thimm [EMAIL PROTECTED] wrote: On Fri, Mar 02, 2007 at 10:05:54AM +1100, Devraj Mukherjee wrote: Hi everyone, I am trying to get Asterisk 1.4 running on CentOS 4.4 (Kernel 2.6.9-42.0.10.ELsmp) and am having a lot of trouble getting asterisk running on it. I had a fair bit of success with the ATrpms binaries (Zaptel worked but asterisk failed to startup because it couldn't find the speex modules). Get it from here: http://atrpms.net/dist/el4/speex/, or since your using a yum based distribution, point yum to atrpms and let it do the work. I am trying to thus recompile the asterisk rpm for CentOS 4.4 with the least amounts of external dependencies. Well, you'll probably find out at the end that you need to upgrade speex to the version above. make rpm gives me an error saying astman could not be found. How do I build astman? Has anyone succeeded making rpm on CentOS? The above rpms are effectively on CentOS: They were built on RHEL, but CentOS is a clone from RHEL. -- Axel.Thimm at ATrpms.net ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] Unable to start Asterisk 1.4 on CentOS 4.4 (installed from ATrpms)
Hi Everyone, I am still unable to start Asterisk 1.4 that I installed using ATrpms. I was initially suspecting some permissions issues but it seems to me that its more to do with a speex codec not loading properly. Here is the message I get if I run asterisk -cvv app_userevent.so = (Custom User Event Application) == Parsing '/etc/asterisk/codecs.conf': Found -- CODEC SPEEX: Setting Quality to 3 -- CODEC SPEEX: Setting Complexity to 2 -- CODEC SPEEX: Perceptual Enhancement Mode. [on] -- CODEC SPEEX: VAD Mode. [on] -- CODEC SPEEX: VBR Mode. [on] -- CODEC SPEEX: Disabling ABR -- CODEC SPEEX: Setting VBR Quality to 4.00 -- CODEC SPEEX: DTX Mode. [off] -- CODEC SPEEX: Preprocessing. [off] -- CODEC SPEEX: Preprocessor VAD. [off] -- CODEC SPEEX: Preprocessor AGC. [off] -- CODEC SPEEX: Setting preprocessor AGC Level to 8000.00 -- CODEC SPEEX: Preprocessor Denoise. [off] -- CODEC SPEEX: Preprocessor Dereverb. [off] -- CODEC SPEEX: Setting preprocessor Dereverb Decay to 0.40 -- CODEC SPEEX: Setting preprocessor Dereverb Level to 0.30 asterisk: symbol lookup error: /usr/lib/asterisk/modules/codec_speex.so: undefined symbol: speex_decode_int I can confirm that I have speex and speex-devel installed [EMAIL PROTECTED] ~]# rpm -qa | grep speex speex-1.0.4-4 speex-devel-1.0.4-4 [EMAIL PROTECTED] ~]# Thanks for any pointers. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] Re: Can't find asterisk.ctl under CentOS installation
Thanks AT. On 1/24/07, Axel Thimm [EMAIL PROTECTED] wrote: On Tue, Jan 23, 2007 at 06:20:08PM +0100, Axel Thimm wrote: On Tue, Jan 23, 2007 at 03:54:49PM +0200, Tzafrir Cohen wrote: On Tue, Jan 23, 2007 at 02:48:07PM +0100, Axel Thimm wrote: If I call asterisk -r as root it succeeds, if as another user it will give Devraj's error message. That's probably how it is supposed to work, or not? Just a thought: shouldn't the asterisk user be allowed write access to that control socket? Or maybe the asterisk group? The asterisk user is allowed, too, of course, the group not (yet). (for quickdirty shell scripts) I think that makes very much sense. The socket is created by asterisk, is there a parameter to specify permissions/umask of that socket? Looks like all there is needed is to uncomment the following line in the default config file: [files] astctlpermissions = 0660 But since upstream defaults to not do so and only have this done by the user, I wouldn't like to change this policy on the package level. -- Axel.Thimm at ATrpms.net ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] Can't find asterisk.ctl under CentOS installation
Hi Everyone, I recently upgraded to Asterisk 1.4 using the RPMS at ATrpms.net on CentOS 4.4, Asterisk starts up but when I start the console it reports this error and drops out. Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist)? I have checked to see that the file asterisk.ctl actually exists. Any suggestions? -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] ZAP chanel doesn't reset if external caller hangs up in menu
Hello world,
My asterisk server doesnt seem to disconnect the call if someone
hangsup say while they are listening to the menu as a result of which
my phone is engaged forever. Any pointers on fixing this issue?
Thanks
my extensions.conf
[incoming]
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,NoOp(${CALLERIDNUM})
exten = s,4,Wait(1)
exten = s,5,Playback(eternity_welcome)
exten = s,6,Background(eternity_mainmenu)
exten = s,7,Wait(4)
exten = s,8,Playback(eternity_loop)
exten = s,9,Goto(incoming,s,6)
; Support
exten = 1,1,Goto(internal,202,1)
; Server and support
exten = 2,1,Goto(internal,102,1)
; LiveCD support
exten = 3,1,Goto(internal,101,1)
; Reception
exten = 0,1,Goto(internal,102,1)
exten = s,10,Hangup()
--
I never look back darling, it distracts from the now, Edna Mode (The
Incredibles)
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Re: [asterisk-users] IP Phones
Nokia E series with proper firmware upgrade :) On 10/5/06, Steve Glaus [EMAIL PROTECTED] wrote: bilal ghayyad wrote: Hi List; I would like to know where I can find the IP Phones that can be used with Asterisk? Is there any link? Regards Bilal Ghayad Mobile: 00965 9849460 Office: 00965 2623174 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 Any phone supporting SIP or IAX are good choices for asterisk. ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] SkypeOut with Asterisk?
Thanks Sharon. On 9/20/06, Sharon Lim [EMAIL PROTECTED] wrote: I have successful link skype with asterisk with http://www.nch.com.au/skypetosip/index.html but not sure whether you need this. here is another link http://www.voip-info.org/wiki/index.php?page=Skype%20Gateways. Good luck! On 9/20/06, Devraj Mukherjee [EMAIL PROTECTED] wrote: Has anyone managed to use SkypeOut as your VoIP provider? -- I never look back, it distracts from the now, Edna Mode (The Incredibles) ___ --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 -- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] SkypeOut with Asterisk?
Has anyone managed to use SkypeOut as your VoIP provider? -- I never look back, it distracts from the now, Edna Mode (The Incredibles) ___ --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] Selecting outbound trunk
Hi Iain,
Thank you for that. That should work well for me.
On 8/29/06, Iain Young [EMAIL PROTECTED] wrote:
On Tue, Aug 29, 2006 at 02:18:32PM +1000, Devraj Mukherjee wrote:
The simplest way I can think of solving this is using prefixes, so
someone appends a 0 or 1 and the dialplan puts the call through the
selected trunk, where 0 being voip and 1 being PSTN.
Whats wrong with something like this :
exten = _91X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _92X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _93X.,1,Dial(IAX2/iaxprov/${EXTEN:2})
Users would dial 91 to dial outbound on SIP Provider 1, 92 for
outbound on SIP Provider 2, and 93 for outbound on IAX. Personally
I use 9X for automatic routing (along with some sane forced routing,
ie local, emerg calls etc), and am planning on using 8X for manual
forced routing.
I have figured out how to use a Substring like function to extract the
number out of the dialed extension. My question is how do I make a
decision in the dialplan to dynamically select a trunk for the call?
Is there a SetIf function or an If function by itself?
Checkout the command GotoIf()
Heres an example that I use to in my exten Macro, that does
slightly different things depending on the number range the
extension dialed is from:
[macro-exten]
exten = s,1,GotoIf($[${ARG1:0:1} = 1]?11:21) ; Did we call a real ext ?
exten = s,11,SetVar(TODIAL=${ARG2}/${ARG1}); Yes so we have the ext
exten = s,12,Goto(91) ; Jump to Dial() routint
exten = s,21,GotoIf($[${ARG1:0:2} = 20]?31:41) ; Did we call a virt or soft ?
exten = s,31,SetVar(VMBOX=${ARG1}) ; Virt, So vm is the same
exten = s,32,SetVar(TODIAL=${VIRT[${ARG1}]}) ; Grab the list of real exts
exten = s,33,Goto(91) ; Jump to the dial routine
exten = s,41,SetVar(VMBOX=20${ARG1:1:1}) ; Soft, So vm is the virt
exten = s,42,SetVar(TODIAL=${ARG2}/${ARG1}); But it is a real ext
exten = s,43,Goto(91)
exten = s,91,Dial(${TODIAL},25,Tt)
exten = s,92,GotoIf($[${ARG1:0:1} = 2]?93:94) ; Do we need to handle vm ?
exten = s,93,GoSub(s-${DIALSTATUS},1)
exten = s,94,Hangup()
exten = s-NOANSWER,1,Voicemail(u${VMBOX}) ; Virtual extensions have
exten = s-BUSY,1,Voicemail(b${VMBOX}) ; VM, so transfer caller
exten = s-CHANUNAVAIL,1,Voicemail(u${VMBOX}) ; Offline, so transfer call
I have a dialplan where 1xx are real extensions, with no voicemail,
20x are virtual extensions, identified with an induvidual, with voicemail, and
2xy are extensions assoiated with the same induvidual as the virtual
number (ie 21x are all linked to 201 etc..)
HTH
Iain
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Re: [asterisk-users] SIP ATA Channels for outbound calls - How to select in dialplan
I am not sure if you have solved this already, but this may be
something you are interested in
[outbound-local]
exten = _9NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9NXX,2,Congestion( )
exten = _9NXX,102,Congestion( )
exten = 911,1,Dial(${OUTBOUNDTRUNK}/911)
exten = 9911,1,Dial(${OUTBOUNDTRUNK}/911)
On 7/18/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote:
I have setup 3 Linksys SPA-3000 devices to pass/send our analog voice calls
into/out of asterisk. The inbound calls work fine as I have set the
spa-3000's to forward all calls to an extension. I have added them to the
sip.conf as spa-3k1, spa-3k2, and spa-3k3. Is there a way for when some
picks up a phone to dial, it starts at 3k1, if congestion, move onto the
sk2, and so on. I'm looking for it to find the first available line to use.
Is this possible in the dialplan?
Thanks,
Dean.
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[asterisk-users] Selecting outbound trunk
Hi Everyone,
I am trying to implement a process where by people can select their
outbound trunk for calls since VoIP is not always the best option to
place the call with.
The simplest way I can think of solving this is using prefixes, so
someone appends a 0 or 1 and the dialplan puts the call through the
selected trunk, where 0 being voip and 1 being PSTN.
I have figured out how to use a Substring like function to extract the
number out of the dialed extension. My question is how do I make a
decision in the dialplan to dynamically select a trunk for the call?
Is there a SetIf function or an If function by itself?
Or is there a better way of doing this? Thank you for your input.
[outbound-statewide]
exten = _X,1,Set(CARRIER=IAX2/nehos)
exten = _X,2,SetCDRUserField(${CARRIER})
exten = _X,3,Dial(${CARRIER}/${EXTEN:2:8})
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[asterisk-users] SIP response 400 Bad request
Hi everyone, This is a message I am getting on the Asterisk console, 192.168.1.80 refers to my Nokia E61, any ideas what this means? -- Got SIP response 400 Bad Request back from 192.168.1.80 Thanks. ___ --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] Asterisk SIP problems with Nokia E61
Hi everyone, I am trying to get my Nokia E61 working with Asterisk, all is well it registers but every now and then the registration drops out. I can still ping the Wi-fi adapter, and qualify in the sip.conf is set to yes. Also the phone claims that it is still registered with the SIP server. This is the first Wifi phone I am trying to get working. Any pointers will be really appreciated. SIP conf for E61: [111] type=friend callerid=(Devraj Mukherjee 111) username=111 host=dynamic secret=password regcontext=default regexten=111 dtmfmode=rfc2833 insecure=very canreinvite=yes nat=yes context=default ;pickupgroup=1 ;callgroup=1 [EMAIL PROTECTED] allow=ulaw qualify=yes Output from Asterisk console: monk*CLI sip show peers Name/username HostDyn Nat ACL Port Status 111/111192.168.1.80 D N 5060 OK (181 ms) 1 sip peers [1 online , 0 offline] -- Got SIP response 400 Bad Request back from 192.168.1.80 monk*CLI sip show peers Name/username HostDyn Nat ACL Port Status 111/111192.168.1.80 D N 5060 OK (306 ms) 1 sip peers [1 online , 0 offline] Aug 1 02:05:41 NOTICE[30603]: chan_sip.c:11396 sip_poke_noanswer: Peer '111' is now UNREACHABLE! Last qualify: 203 Aug 1 02:06:19 NOTICE[30603]: chan_sip.c:9721 handle_response_peerpoke: Peer '111' is now REACHABLE! (233ms / 2000ms) monk*CLI sip show peers Name/username HostDyn Nat ACL Port Status 111/111192.168.1.80 D N 5060 OK (233 ms) 1 sip peers [1 online , 0 offline] Finally ping report: --- 192.168.1.80 ping statistics --- 2 packets transmitted, 2 received, 0% packet loss, time 1002ms rtt min/avg/max/mdev = 97.654/156.043/214.433/58.390 ms, pipe 2 [EMAIL PROTECTED] asterisk]# ping 192.168.1.80 PING 192.168.1.80 (192.168.1.80) 56(84) bytes of data. 64 bytes from 192.168.1.80: icmp_seq=0 ttl=69 time=221 ms 64 bytes from 192.168.1.80: icmp_seq=1 ttl=69 time=140 ms --- 192.168.1.80 ping statistics --- 2 packets transmitted, 2 received, 0% packet loss, time 1001ms rtt min/avg/max/mdev = 140.624/181.153/221.682/40.529 ms, pipe 2 [EMAIL PROTECTED] asterisk]# ping 192.168.1.80 PING 192.168.1.80 (192.168.1.80) 56(84) bytes of data. 64 bytes from 192.168.1.80: icmp_seq=0 ttl=69 time=32.6 ms 64 bytes from 192.168.1.80: icmp_seq=1 ttl=69 time=157 ms --- 192.168.1.80 ping statistics --- 2 packets transmitted, 2 received, 0% packet loss, time 1001ms rtt min/avg/max/mdev = 32.646/95.308/157.970/62.662 ms, pipe 2 ___ --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] Re: Asterisk SIP problems with Nokia E61
Further to the problem I reported earlier, I changed my sip.conf to have the value 1000 for the property qualify and the phone seems to be connected longer monk*CLI sip show peers Name/username HostDyn Nat ACL Port Status 111/111192.168.1.80 D N 5060 OK (162 ms) 1 sip peers [1 online , 0 offline] -- Got SIP response 400 Bad Request back from 192.168.1.80 monk*CLI sip show peers Name/username HostDyn Nat ACL Port Status 111/111192.168.1.80 D N 5060 OK (305 ms) 1 sip peers [1 online , 0 offline] monk*CLI sip show peers Name/username HostDyn Nat ACL Port Status 111/111192.168.1.80 D N 5060 OK (209 ms) 1 sip peers [1 online , 0 offline] monk*CLI sip show peers Name/username HostDyn Nat ACL Port Status 111/111192.168.1.80 D N 5060 OK (314 ms) 1 sip peers [1 online , 0 offline] monk*CLI sip show peers Name/username HostDyn Nat ACL Port Status 111/111192.168.1.80 D N 5060 OK (307 ms) 1 sip peers [1 online , 0 offline] Aug 1 05:12:53 NOTICE[30603]: chan_sip.c:9727 handle_response_peerpoke: Peer '111' is now TOO LAGGED! (1330ms / 1000ms) Aug 1 05:13:03 NOTICE[30603]: chan_sip.c:9721 handle_response_peerpoke: Peer '111' is now REACHABLE! (240ms / 1000ms) monk*CLI sip show peers Name/username HostDyn Nat ACL Port Status 111/111192.168.1.80 D N 5060 OK (303 ms) 1 sip peers [1 online , 0 offline] monk*CLI sip show peers Name/username HostDyn Nat ACL Port Status 111/111192.168.1.80 D N 5060 OK (334 ms) 1 sip peers [1 online , 0 offline] monk*CLI sip show peers Name/username HostDyn Nat ACL Port Status 111/111192.168.1.80 D N 5060 OK (305 ms) 1 sip peers [1 online , 0 offline] monk*CLI sip show peers Name/username HostDyn Nat ACL Port Status 111/111192.168.1.80 D N 5060 OK (305 ms) 1 sip peers [1 online , 0 offline] -- Got SIP response 400 Bad Request back from 192.168.1.80 Aug 1 05:46:56 NOTICE[30605]: chan_iax2.c:7787 iax2_poke_noanswer: Peer 'ifone' is now UNREACHABLE! Time: 39 Aug 1 05:47:06 NOTICE[30605]: chan_iax2.c:7119 socket_read: Peer 'ifone' is now REACHABLE! Time: 39 monk*CLI sip show peers Name/username HostDyn Nat ACL Port Status 111/111192.168.1.80 D N 5060 OK (304 ms) 1 sip peers [1 online , 0 offline] On 8/1/06, Devraj Mukherjee [EMAIL PROTECTED] wrote: Hi everyone, I am trying to get my Nokia E61 working with Asterisk, all is well it registers but every now and then the registration drops out. I can still ping the Wi-fi adapter, and qualify in the sip.conf is set to yes. Also the phone claims that it is still registered with the SIP server. This is the first Wifi phone I am trying to get working. Any pointers will be really appreciated. SIP conf for E61: [111] type=friend callerid=(Devraj Mukherjee 111) username=111 host=dynamic secret=password regcontext=default regexten=111 dtmfmode=rfc2833 insecure=very canreinvite=yes nat=yes context=default ;pickupgroup=1 ;callgroup=1 [EMAIL PROTECTED] allow=ulaw qualify=yes Output from Asterisk console: monk*CLI sip show peers Name/username HostDyn Nat ACL Port Status 111/111192.168.1.80 D N 5060 OK (181 ms) 1 sip peers [1 online , 0 offline] -- Got SIP response 400 Bad Request back from 192.168.1.80 monk*CLI sip show peers Name/username HostDyn Nat ACL Port Status 111/111192.168.1.80 D N 5060 OK (306 ms) 1 sip peers [1 online , 0 offline] Aug 1 02:05:41 NOTICE[30603]: chan_sip.c:11396 sip_poke_noanswer: Peer '111' is now UNREACHABLE! Last qualify: 203 Aug 1 02:06:19 NOTICE[30603]: chan_sip.c:9721 handle_response_peerpoke: Peer '111' is now REACHABLE! (233ms / 2000ms) monk*CLI sip show peers Name/username HostDyn Nat ACL Port Status 111/111192.168.1.80 D N 5060 OK (233 ms) 1 sip peers [1 online , 0 offline] Finally ping report: --- 192.168.1.80 ping statistics --- 2 packets transmitted, 2 received, 0% packet loss, time 1002ms rtt min/avg/max/mdev = 97.654/156.043/214.433/58.390 ms, pipe 2 [EMAIL PROTECTED] asterisk]# ping 192.168.1.80 PING 192.168.1.80 (192.168.1.80) 56(84) bytes of data. 64 bytes from 192.168.1.80: icmp_seq=0 ttl=69 time=221 ms 64 bytes from 192.168.1.80: icmp_seq=1 ttl=69 time=140 ms --- 192.168.1.80 ping statistics --- 2 packets transmitted, 2 received, 0% packet loss, time 1001ms rtt min/avg
[asterisk-users] Solution init.d scripts for CentOS 4.3
Hi Everyone, I was having a lot of trouble starting up Asterisk and zaptel using the init.d scripts. I have worked on the scripts and now the zaptel script so it reads preferences of /etc/sysconfig/zaptel file and starts the zap interfaces properly. The asterisk init.d script does not load or unload any modules. Hope this is useful for anyone using CentOS with the same problems. asterisk Description: Binary data zaptel Description: Binary data ___ --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] Asterisk autoloading of card modules
Hi Alejandro, Thanks for your suggestions. Where did you fetch your rpms? I had to fix up the init scripts for everything to work On 7/24/06, Alejandro Kauffmann [EMAIL PROTECTED] wrote: My /etc/sysconfig/zaptel configuration has only one MODULES directive enabled MODULES=$MODULES wctdm However when I start asterisk it loads the wct1xxp module. Which configuration file controls the loading of card modules? Check /etc/modprobe.conf I clear that out and just leave the module I want enabled in /etc/sysconfig/zaptel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trouble configuring TDM400P on Dell SC420
Hi Everyone, I am running Asterisk 1.2.7 Zaptel 1.2.5 on CentOS 4.3 on a Dell PowerEdge SC420. I was running an older version of Asterisk (can't remember what, but was using the wcfxs kernel module) under Gentoo Linux and succsessfully had Asterisk talking to my TDM400P card. However on my CentOS installation and Asterisk upgrade the TDM400P has stopped responding, /dev/zap channels dont get created. I have gathered that there is an incompatibility with SC 420 server hardware and TDM400P (http://www.digium.com/en/docs/misc/compatibility_notes.php) Has anyone had similar issues, is there a work around? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk autoloading of card modules
Hi everyone, I am using Asterisk on CentOS 4.3 with a TDM400P and have managed to get things up and running except this one part. My /etc/sysconfig/zaptel configuration has only one MODULES directive enabled MODULES=$MODULES wctdm However when I start asterisk it loads the wct1xxp module. Which configuration file controls the loading of card modules? Thanks. ___ --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: SV: [Asterisk-Users] Nokia E61
Does the GSM and Wi-Fi phone feature work at the same time? :) Thanks for your time On 7/5/06, Amund Nygaard [EMAIL PROTECTED] wrote: Hello I done some more testing, i have no problems connection behind natted networks. It even connected with 3G, but as you can imagine G711 is not very suited for that :P BR Amund Nygaard -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Antonio Rabena Sendt: 5. juli 2006 10:26 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] Nokia E61 Hi, I had no issues connecting/calling to my asterisk box (on public ip), even my phone is behind a hotspot. Its just that i need to use G711 codec. At 03:34 PM 7/5/2006, you wrote: Hello Has anyone tried a Nokia E6x phone when it is natted? Like behind a hotspot or similar? BR Amund Nygaard -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Devraj Mukherjee Sendt: 4. juli 2006 12:49 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Nokia E61 Thanks guys. How about the quality of the call etc? Are you happy with the phone, do you recommend them? ___ --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 ___ --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 ___ --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] Nokia E61
Thanks guys. How about the quality of the call etc? Are you happy with the phone, do you recommend them? On 7/4/06, Antonio Rabena [EMAIL PROTECTED] wrote: Hi, configuration for E61 is the same as E60. As for the codec, G729 works between E60/61 phones (G729 passthru). At 03:44 PM 7/4/2006, you wrote: Devraj Mukherjee wrote: Hello world, Any success stories of getting a Nokia E61 to work with Asterisk server? Interested to hear before we buy them for work :) I don't know about e61, but I connected an e60 up yesterday that wasn't any hassle. Even the stories about poor quality with WPA + G.729 seemed to be false. ___ --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 ___ --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 ___ --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] Nokia E61
Hello world, Any success stories of getting a Nokia E61 to work with Asterisk server? Interested to hear before we buy them for work :) Thanks for your time ___ --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] Asterisk Manager interface
I am currently writing some tools that work with the Asterisk Manager interface. Part of the issue is number of socket connections that the client opens back to the manager itnerface. Most of these connections are short lived. Is this is a problem from a design perspective? Or is the management interface designed to handle this. Devraj ___ --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] Asterisk Tools for OSX
Hello Asterisk Users, I am an Objective-C enthusiast and have been writing some clever tools to integrate Asterisk functionality with Mac OS X applications. Please find my project on http://www.sf.net/projects/astrxtools4osx/ The objectives of my project are as follows 1. Implement an Objective-C framework to communicate effectively with the Asterisk Management Interface 2. Address Book plugin to enable call back functionality 3. A System Preferences pane to allow administrators to easily configure Asterisk options on a Mac 4. Dashboard Widget that allows users to quickly call arbitary numbers 5. iTunes integration to stop and star iTunes to play when the phone rings etc. The source code is in pre-Alpha stage at the moment but I am hoping to release a Beta at the end of next week. Please feel free to download and use these extensions. I hope they turn out to be useful and would appreciate any feedback. Devraj ___ --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] Call Monitoring / Call Takeover with Asterisk
http://www.voip-info.org/wiki-Asterisk+manager+API I have been doing some work with the Asterisk Management API and there is a commadn where you can transfer a call. This is what you may be looking for Not sure, trying to be as helpful as I can On 3/29/06, Steve Totaro [EMAIL PROTECTED] wrote: I do not think so but it would be a great feature. -Original Message- From: Cory Andrews [mailto:[EMAIL PROTECTED] Sent: Tue 3/28/2006 9:59 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] Call Monitoring / Call Takeover with Asterisk Does Asterisk support, in a call center type environment, the ability for a supervisor to monitor a call between a system user and a 3rd party, and allow them to physically take over the call. For instance if a call center supervisor is randomlay monitoring agent calls, and for some reason need to intervene on a call without first having been conferenced into the call? Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY ___ --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 -- Devraj Mukherjee Eternity Technologies Pty Limited ___ --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
