[asterisk-users] chan_sip.c:1968 create_addr: No such host:
I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application. Added the following [callingcard] ; CallingCard application exten = 777,1,Answer exten = 777,2,Wait,2 exten = 777,3,DeadAGI,a2billing.php exten = 777,4,Wait,2 exten = 777,5,Hangup I am using 777 as the calling card application. when I call that extension, instead of getting please enter you pin number it fails and this is the output from the cli: -- Executing Dial(SIP/9614-e7ba, SIP/777|200|rt) in new stack Feb 18 05:03:38 WARNING[11725]: chan_sip.c:1968 create_addr: No such host: 777 Feb 18 05:03:38 NOTICE[11725]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/9614-e7ba' status is 'CHANUNAVAIL' Any Help will be greatly appreciated.___ --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] chan_sip.c:1968 create_addr: No such host:
I guess the obvious question would be whether the callingcard context is included into the context that the call is coming from. That's the usual reason for a failure like this. [EMAIL PROTECTED] wrote: I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application. Added the following [callingcard] ; CallingCard application exten = 777,1,Answer exten = 777,2,Wait,2 exten = 777,3,DeadAGI,a2billing.php exten = 777,4,Wait,2 exten = 777,5,Hangup I am using 777 as the calling card application. when I call that extension, instead of getting please enter you pin number it fails and this is the output from the cli: -- Executing Dial(SIP/9614-e7ba, SIP/777|200|rt) in new stack Feb 18 05:03:38 WARNING[11725]: chan_sip.c:1968 create_addr: No such host: 777 Feb 18 05:03:38 NOTICE[11725]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/9614-e7ba' status is 'CHANUNAVAIL' Any Help will be greatly appreciated. ___ --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] chan_sip.c:1968 create_addr: No such host:
Thanks Rob, that helped a little bit but now getting a different kind of error: -- Executing Dial(SIP/9614-3896, SIP/777|200|rt) in new stack Feb 18 06:00:30 NOTICE[12236]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/9614-3896' status is 'CHANUNAVAIL' -- Original message -- From: Rob Hillis [EMAIL PROTECTED] I guess the obvious question would be whether the callingcard context is included into the context that the call is coming from. That's the usual reason for a failure like this. [EMAIL PROTECTED] wrote: I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application. Added the following [callingcard] ; CallingCard application exten = 777,1,Answer exten = 777,2,Wait,2 exten = 777,3,DeadAGI,a2billing.php exten = 777,4,Wait,2 exten = 777,5,Hangup I am using 777 as the calling card application. when I call that extension, instead of getting please enter you pin number it fails and this is the output from the cli: -- Executing Dial(SIP/9614-e7ba, SIP/777|200|rt) in new stack Feb 18 05:03:38 WARNING[11725]: chan_sip.c:1968 create_addr: No such host: 777 Feb 18 05:03:38 NOTICE[11725]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/9614-e7ba' status is 'CHANUNAVAIL' Any Help will be greatly appreciated. ___ --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 ---BeginMessage--- ___ --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 ---End Message--- ___ --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] Strange behaviour with Dial cmd
On 15 Feb 2007, at 09:55, Yuan LIU wrote: From: Il Neofita [EMAIL PROTECTED] Date: Thu, 15 Feb 2007 03:37:14 -0500 But I tought that hangup was suppose to close the call, however, is not the case and a really did not catch why. Now I see where the confusion comes from. Asterisk doesn't really speak English - or Chinese for that matter:-) In telephony, there is no way for the callee to tell the caller to stop ringing - unless you answer it first. Once you answer, you can do a number of things, the rudest being to immediately hang up. (I saw live people doing this intentionally.) Your only other option really is to ignore. That isn't exactly true - ISDN and IAX (SIP?) support the concept of rejecting a call without answering it. The asterisk dial plan only supports this indirectly. If there is no extension that maps to the called number in the relevant context then asterisk will reject the call without answering. I don't think this helps the OP's situation, but for the sake of the archives I think its worth clarifying Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] chan_sip.c:1968 create_addr: No such host:
Its not right. I am using a2billing calling card and it works fine Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email If you need to contract Customer Service Please use our IAX2 WebPhone at the link below http://www.bochterservices.com/voip/iaxphone.php [EMAIL PROTECTED] wrote: I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application. Added the following [callingcard] ; CallingCard application exten = 777,1,Answer exten = 777,2,Wait,2 exten = 777,3,DeadAGI,a2billing.php exten = 777,4,Wait,2 exten = 777,5,Hangup I am using 777 as the calling card application. when I call that extension, instead of getting please enter you pin number it fails and this is the output from the cli: -- Executing Dial(SIP/9614-e7ba, SIP/777|200|rt) in new stack Feb 18 05:03:38 WARNING[11725]: chan_sip.c:1968 create_addr: No such host: 777 Feb 18 05:03:38 NOTICE[11725]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/9614-e7ba' status is 'CHANUNAVAIL' Any Help will be greatly appreciated. ___ --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 Inbound (clean). Database: 000714-3, 02/18/2007 - 2/18/2007 10:42:53 AM ___ --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] chan_sip.c:1968 create_addr: No such host:
nope Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email If you need to contract Customer Service Please use our IAX2 WebPhone at the link below http://www.bochterservices.com/voip/iaxphone.php Rob Hillis wrote: I guess the obvious question would be whether the callingcard context is included into the context that the call is coming from. That's the usual reason for a failure like this. [EMAIL PROTECTED] wrote: I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application. Added the following [callingcard] ; CallingCard application exten = 777,1,Answer exten = 777,2,Wait,2 exten = 777,3,DeadAGI,a2billing.php exten = 777,4,Wait,2 exten = 777,5,Hangup I am using 777 as the calling card application. when I call that extension, instead of getting please enter you pin number it fails and this is the output from the cli: -- Executing Dial(SIP/9614-e7ba, SIP/777|200|rt) in new stack Feb 18 05:03:38 WARNING[11725]: chan_sip.c:1968 create_addr: No such host: 777 Feb 18 05:03:38 NOTICE[11725]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/9614-e7ba' status is 'CHANUNAVAIL' Any Help will be greatly appreciated. ___ --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 Inbound (clean). Database: 000714-3, 02/18/2007 - 2/18/2007 10:43:30 AM ___ --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] Native format prompts
I am trying to implement native format (ulaw) voice prompts and music on hold. Different documentation has different file extensions. Does Asterisk recognise them all? So far I have .ulaw .ul .pcm . Which should I use so Asterisk recognises them as native uLaw files From what I know, .ulaw hth ___ --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] Native format prompts
On Fri, Feb 16, 2007 at 08:19:27AM +1100, Eric Bishop wrote: Hi all, I am trying to implement native format (ulaw) voice prompts and music on hold. Different documentation has different file extensions. Does Asterisk recognise them all? So far I have .ulaw .ul .pcm . Which should I use so Asterisk recognises them as native uLaw files What do you see on 'show file formats' ? -- 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
[asterisk-users] Asterisk consultant needed in Paris
Hi list, We are looking for an Asterisk consultant for a 3 months mission in Paris. If you are interested, please contact us at [EMAIL PROTECTED] Best regards, Brian ___ --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] Strange behaviour with Dial cmd
Asterisk supports this directly by issuing the hangup command before the answer command. However, when using an analog interface like FXO the line has no way of knowing you just hung up and will continue to ring, which asterisk will see as a new call. in my experience even when using a PRI if i dont give the pri cause the provider re initiates the call. On 2/18/07, Tim Panton [EMAIL PROTECTED] wrote: On 15 Feb 2007, at 09:55, Yuan LIU wrote: From: Il Neofita [EMAIL PROTECTED] Date: Thu, 15 Feb 2007 03:37:14 -0500 But I tought that hangup was suppose to close the call, however, is not the case and a really did not catch why. Now I see where the confusion comes from. Asterisk doesn't really speak English - or Chinese for that matter:-) In telephony, there is no way for the callee to tell the caller to stop ringing - unless you answer it first. Once you answer, you can do a number of things, the rudest being to immediately hang up. (I saw live people doing this intentionally.) Your only other option really is to ignore. That isn't exactly true - ISDN and IAX (SIP?) support the concept of rejecting a call without answering it. The asterisk dial plan only supports this indirectly. If there is no extension that maps to the called number in the relevant context then asterisk will reject the call without answering. I don't think this helps the OP's situation, but for the sake of the archives I think its worth clarifying Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] chan_sip.c:1968 create_addr: No such host:
Hello, I'm not familiar with A2billing but for me it is strange that you dial SIP/777 - 777 should be an extension. Could you post your user context - or at least the one which direct you to: Dial(SIP/9614-3896, SIP/777|200|rt) Best regards, ## nini @ www.modulo.ro ## [EMAIL PROTECTED] wrote: Thanks Rob, that helped a little bit but now getting a different kind of error: -- Executing Dial(SIP/9614-3896, SIP/777|200|rt) in new stack Feb 18 06:00:30 NOTICE[12236]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/9614-3896' status is 'CHANUNAVAIL' -- Original message -- From: Rob Hillis [EMAIL PROTECTED] I guess the obvious question would be whether the callingcard context is included into the context that the call is coming from. That's the usual reason for a failure like this. [EMAIL PROTECTED] wrote: I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application. Added the following [callingcard] ; CallingCard application exten = 777,1,Answer exten = 777,2,Wait,2 exten = 777,3,DeadAGI,a2billing.php exten = 777,4,Wait,2 exten = 777,5,Hangup I am using 777 as the calling card application. when I call that extension, instead of getting please enter you pin number it fails and this is the output from the cli: -- Executing Dial(SIP/9614-e7ba, SIP/777|200|rt) in new stack Feb 18 05:03:38 WARNING[11725]: chan_sip.c:1968 create_addr: No such host: 777 Feb 18 05:03:38 NOTICE[11725]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/9614-e7ba' status is 'CHANUNAVAIL' Any Help will be greatly appreciated. ___ --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 Subject: Re: [asterisk-users] chan_sip.c:1968 create_addr: No such host: From: Rob Hillis [EMAIL PROTECTED] Date: Sun, 18 Feb 2007 12:43:59 + To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.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 ___ --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] chan_sip.c:1968 create_addr: No such host:
a2billing is your default context in the a2billing.conf file from setup, if you have changed this, and also changed your context for your card/sip entry to callingcard you should, when you dial from your phone, be using the callingcard context. Please check your SIP entry for your phone in additional_a2billing_sip.conf to ensure that your context is set to callingcard, not a2billing (default), and then reload you sip config from the CLI. Your phone that is registered sounds like it is not assigned to that context by default. J- On Sun, 2007-02-18 at 20:38 +0200, Ioan Indreias wrote: Hello, I'm not familiar with A2billing but for me it is strange that you dial SIP/777 - 777 should be an extension. Could you post your user context - or at least the one which direct you to: Dial(SIP/9614-3896, SIP/777|200|rt) Best regards, ## nini @ www.modulo.ro ## [EMAIL PROTECTED] wrote: Thanks Rob, that helped a little bit but now getting a different kind of error: -- Executing Dial(SIP/9614-3896, SIP/777|200|rt) in new stack Feb 18 06:00:30 NOTICE[12236]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/9614-3896' status is 'CHANUNAVAIL' -- Original message -- From: Rob Hillis [EMAIL PROTECTED] I guess the obvious question would be whether the callingcard context is included into the context that the call is coming from. That's the usual reason for a failure like this. [EMAIL PROTECTED] wrote: I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application. Added the following [callingcard] ; CallingCard application exten = 777,1,Answer exten = 777,2,Wait,2 exten = 777,3,DeadAGI,a2billing.php exten = 777,4,Wait,2 exten = 777,5,Hangup I am using 777 as the calling card application. when I call that extension, instead of getting please enter you pin number it fails and this is the output from the cli: -- Executing Dial(SIP/9614-e7ba, SIP/777|200|rt) in new stack Feb 18 05:03:38 WARNING[11725]: chan_sip.c:1968 create_addr: No such host: 777 Feb 18 05:03:38 NOTICE[11725]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/9614-e7ba' status is 'CHANUNAVAIL' Any Help will be greatly appreciated. ___ --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 Subject: Re: [asterisk-users] chan_sip.c:1968 create_addr: No such host: From: Rob Hillis [EMAIL PROTECTED] Date: Sun, 18 Feb 2007 12:43:59 + To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.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 ___ --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] Looking for starting point?
Hi, I am a retired telephone tech/manager who recently had a bad experience with a local company offering digital phone service (VoIP). I have spent the last thirty years in the PSTN network, switching, PBX and key system field and am interested in learning more about VoIP. My background also includes programming, mostly specialized applications to interact with the PSTN network. Most of my experience in this field have been with Borland products, specifically Delphi. I also have been involved with database programming, same platform as the communications. My computer experience started with the operating system CPM (I'm not really that old, only 56). The best platform now seems to be Linux so now since I am retired now, it seems a good time to learn something new. I also have been looking at Asterisk which most companies seem to be using for a PBX platform. I found out by accident that the local company I had the problem with uses this PBX software. Could someone steer me in the right direction as to where to start? I spent most of my career in the telephone industry in a 'bush' area of Alaska so pretty much had to teach myself what I needed to know about computers but I can learn almost anything from a book and by asking questions when I get stuck. Most of my experience was before the Internet so I plan on using this avenue to advance my knowledge. I understand what a broad scope I am asking about so would appreciate any tips to help me get started. Since there are many 'brands' of Linux what is the best one to start with? Which Linux will be better when I get to the point of working with Asterisk? Any tips or ideas on books, online tutors, discussions or anything of this nature would be much appreciated. I hope to add to this group if I can be any assistance from the 'other side', the PSTN network. Thank You, Gary H. Thompson ___ --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] Problem with busydetect and cell phones
Ryan McDaniel wrote: I have a very strange problem I'm hoping someone has encountered already. I've scoured the internet for an answer to this one. My phone company provides no disconnect supervision. Hence I'm forced to use the busydetect feature. I have a TDM400 with two FXO ports. If I call from an internal extension to a landline and then hangup the landline Asterisk detects the busy signal correctly and clears the line. If I call from an internal extension to a cell phone and then hangup the cell phone Asterisk will never detect the busy signal though it is clearly there. Asterisk will happily sit there listening to the busy signal. I suspect that the busy signal styles are slightly different though it is undetectable to me. How can I fix this??? It causes severe issues when a call is forwarded to a cell phone via the Zap interfaces as once you hangup the cell phone Asterisk never releases the channel. The landlines are with ATT. The cell phones I'm testing with are Cingular (ATT subsidiary). There must be a subtle difference in the busy signals. How can I make it catch busy signals from both carriers? Have you tried calling ATT and asking for call disconnect supervision? I realise that this can be a thankless and tedious endeavour, but it IS worth trying. There are almost no commercial switches that don't support this; it's a matter of activating it for the specific circuit in software. Particularly if you have a business line -- you can demand it. All PBXs need it if they use analog lines (and plenty still do) so I'm sure this is not an alien concept to ATT. It's just a matter of finding the right Earthling there who can help you. This might be one of those times where a beer with the technician will get you some joy, if calling Repair doesn't give you any joy. -Stephen- ___ --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] Summary of Trixbox vs. custom install
I also include a consideration from mine: I would happily use Trixbox, because I did FreePBX setup once and it was a real pain, but I'm very frightened by a few issues: 1) Trixbox Macho installation that installs everything without asking. I, for example, would like to use software RAID (maybe it's wrong with Asterisk, but I want to do it!). I wouldn't like doing it manually after Trixbox installation. I would like to have an installer doing it for me. Centos (ex redhat) installer does it, so why Trixbox choose to install everything without prompting? You can just install CentOS with RAID and whatever you want, then use the Trixbox tar package instead of the ISO. Still, why on earth did the Trixbox team didn't leave the option of doing a custom install with the ISO ? 2) How easy it is to find Trixbox SRPMS? Is it possible to compile new software (i.e. Asterisk or FreePBX, etc..) on Trixbox without having to rewrite all the configuration files, changing all paths, all permissions, and so on... You can update Asterisk/zaptel/whatever by just downloading the source and compiling it. My home system was installed with [EMAIL PROTECTED] on version 0.8 (if memory serves me right) and I upgraded Asterisk and Zaptel to version 1.0.10 by downloading and compiling. I know, this is a really old version and I should upgrade, but hey, it is doing everything I need and it is stable (uptime of 315 days). IMHO, Trixbox can me customized alot, but you need to know where and what to modify. I believe that if you know enough about how Asterisk work, you can get around Trixbox limitations. One thing to remember is that the files you can modify are the _custom.conf files. Never touch the _additional.conf files, they will get overwritten next time you click Apply changes in the GUI. The normal base files (sip.conf. iax.conf, etc) can be modified since the GUI doesn't touch them. But I also think that there is nothing that can beat a plain install as far as customization go. YMMV ___ --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] Looking for starting point?
Go to your book store and get the Fedora/Linux reference. Get yourself a PC with 20GB drives, a CD burner, and decent ram. The PC should have either an i386 or x86_64 processor. If you'll be purchasing a PC, go to the computer store, purchase the piece parts, and assemble it yourself (I like Athlon CPUs). Since you have a PC and were able to post a message, go to http://fedora.redhat.com and follow the links to Documentation and then to Download. Follow the link for your PC's architecture (i386 or x86_64) and then download the six ISO images. Burn each image to a CD. Install Linux. Take all the defaults. Load all packages. When it's running, login as root and open the browser. Go to http://asterisk.org . Take the download tab and download the five Asterisk 1.2 tar files into directory /usr/src. 'cd' to /usr/src. Use 'tar xzf file' on each of the downloaded files. Enter the zaptel directory and execute 'make', check for errors, and then 'make install' and 'make config'. Enter the libpri directory and execute 'make;make install'. Enter the asterisk directory and execute 'make', 'make install', and 'make samples'. Enter the asterisk-addons directory and execute 'make' and then 'make install' Enter the asterisk-sounds directory and execute 'make install'. Execute 'service zaptel start'--this will load the zap drivers. These will also load on reboot. Execute 'asterisk -c'. This will start Asterisk. N.B. People spend years between step #1 and a running Asterisk system. For help with asterisk, google on 'site:lists.digium.com search words'. For the wiki, google on 'site:voip-info.org search words'. The wiki is most helpful. Keep a blog of your experience and let other newbies learn from you. :=) Cheers, Gary H. Thompson wrote: Hi, I am a retired telephone tech/manager who recently had a bad experience with a local company offering digital phone service (VoIP). I have spent the last thirty years in the PSTN network, switching, PBX and key system field and am interested in learning more about VoIP. My background also includes programming, mostly specialized applications to interact with the PSTN network. Most of my experience in this field have been with Borland products, specifically Delphi. I also have been involved with database programming, same platform as the communications. My computer experience started with the operating system CPM (I’m not really that old, only 56). The best platform now seems to be ƒ so now since I am retired now, it seems a good time to learn something new. I also have been looking at Asterisk which most companies seem to be using for a PBX platform. I found out by accident that the local company I had the problem with uses this PBX software. Could someone steer me in the right direction as to where to start? I spent most of my career in the telephone industry in a ‘bush’ area of Alaska so pretty much had to teach myself what I needed to know about computers but I can learn almost anything from a book and by asking questions when I get stuck. Most of my experience was before the Internet so I plan on using this avenue to advance my knowledge. I understand what a broad scope I am asking about so would appreciate any tips to help me get started. Since there are many ‘brands’ of Linux what is the best one to start with? Which Linux will be better when I get to the point of working with Asterisk? Any tips or ideas on book s, online tutors, discussions or anything of this nature would be much appreciated. I hope to add to this group if I can be any assistance from the ‘other side’, the PSTN network. Thank You, Gary H. Thompson ___ --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] Problem with busydetect and cell phones
Original Message- Ryan McDaniel wrote: I have a very strange problem I'm hoping someone has encountered already. I've scoured the internet for an answer to this one. My phone company provides no disconnect supervision. Hence I'm forced to use the busydetect feature. I have a TDM400 with two FXO ports. If I call from an internal extension to a landline and then hangup the landline Asterisk detects the busy signal correctly and clears the line. If I call from an internal extension to a cell phone and then hangup the cell phone Asterisk will never detect the busy signal though it is clearly there. Asterisk will happily sit there listening to the busy signal. I suspect that the busy signal styles are slightly different though it is undetectable to me. How can I fix this??? It causes severe issues when a call is forwarded to a cell phone via the Zap interfaces as once you hangup the cell phone Asterisk never releases the channel. The landlines are with ATT. The cell phones I'm testing with are Cingular (ATT subsidiary). There must be a subtle difference in the busy signals. How can I make it catch busy signals from both carriers? Have you tried calling ATT and asking for call disconnect supervision? I realise that this can be a thankless and tedious endeavour, but it IS worth trying. There are almost no commercial switches that don't support this; it's a matter of activating it for the specific circuit in software. Particularly if you have a business line -- you can demand it. All PBXs need it if they use analog lines (and plenty still do) so I'm sure this is not an alien concept to ATT. It's just a matter of finding the right Earthling there who can help you. This might be one of those times where a beer with the technician will get you some joy, if calling Repair doesn't give you any joy. -Stephen- Unfortunately I tried that. Apparently my lines are on one of the last really ancient junction boxes in Southern California. When using busydetect is it looking for any on / off repetitive sound to identify the busy signal, or for a specific length sound as defined in the indications.conf region? I'd really like to avoid using callprogress if possible. Is there a way to tweak it so it will accept a wider variety of busy patterns? - Ryan ___ --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] Looking for starting point?
Michael Welter wrote: Go to your book store and get the Fedora/Linux reference. Not to start a religious argument here, but it seems from reading the list that the CentOS flavors have fewer problems Install Linux. Take all the defaults. Load all packages. When it's running, login as root and open the browser. Go to http://asterisk.org . Take the download tab and download the five Asterisk 1.2 tar files into directory /usr/src. Also get the book, Asterisk the future of telephony free for the taking. John Novack ___ --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] Looking for starting point?
Another perspective: Fedora is not the only linux distro. If I did use it, I would be building RPM rather than doing local builds on the target asterisk server. I use debian so I build deb files instead of rpm files. Here is a good way to get started using debian linux and some cheap old hardware: http://www.xorcom.com/rapid/index.html Install the above just to play around a bit. Why buy any new hardware before you need it? The longer you wait, the more performance you get for your money. Michael Welter wrote: Go to your book store and get the Fedora/Linux reference. Get yourself a PC with 20GB drives, a CD burner, and decent ram. The PC should have either an i386 or x86_64 processor. If you'll be purchasing a PC, go to the computer store, purchase the piece parts, and assemble it yourself (I like Athlon CPUs). Since you have a PC and were able to post a message, go to http://fedora.redhat.com and follow the links to Documentation and then to Download. Follow the link for your PC's architecture (i386 or x86_64) and then download the six ISO images. Burn each image to a CD. Install Linux. Take all the defaults. Load all packages. When it's running, login as root and open the browser. Go to http://asterisk.org . Take the download tab and download the five Asterisk 1.2 tar files into directory /usr/src. 'cd' to /usr/src. Use 'tar xzf file' on each of the downloaded files. Enter the zaptel directory and execute 'make', check for errors, and then 'make install' and 'make config'. Enter the libpri directory and execute 'make;make install'. Enter the asterisk directory and execute 'make', 'make install', and 'make samples'. Enter the asterisk-addons directory and execute 'make' and then 'make install' Enter the asterisk-sounds directory and execute 'make install'. Execute 'service zaptel start'--this will load the zap drivers. These will also load on reboot. Execute 'asterisk -c'. This will start Asterisk. N.B. People spend years between step #1 and a running Asterisk system. For help with asterisk, google on 'site:lists.digium.com search words'. For the wiki, google on 'site:voip-info.org search words'. The wiki is most helpful. Keep a blog of your experience and let other newbies learn from you. :=) Cheers, Gary H. Thompson wrote: Hi, I am a retired telephone tech/manager who recently had a bad experience with a local company offering digital phone service (VoIP). I have spent the last thirty years in the PSTN network, switching, PBX and key system field and am interested in learning more about VoIP. My background also includes programming, mostly specialized applications to interact with the PSTN network. Most of my experience in this field have been with Borland products, specifically Delphi. I also have been involved with database programming, same platform as the communications. My computer experience started with the operating system CPM (I’m not really that old, only 56). The best platform now seems to be ƒ so now since I am retired now, it seems a good time to learn something new. I also have been looking at Asterisk which most companies seem to be using for a PBX platform. I found out by accident that the local company I had the problem with uses this PBX software. Could someone steer me in the right direction as to where to start? I spent most of my career in the telephone industry in a ‘bush’ area of Alaska so pretty much had to teach myself what I needed to know about computers but I can learn almost anything from a book and by asking questions when I get stuck. Most of my experience was before the Internet so I plan on using this avenue to advance my knowledge. I understand what a broad scope I am asking about so would appreciate any tips to help me get started. Since there are many ‘brands’ of Linux what is the best one to start with? Which Linux will be better when I get to the point of working with Asterisk? Any tips or ideas on book s, online tutors, discussions or anything of this nature would be much appreciated. I hope to add to this group if I can be any assistance from the ‘other side’, the PSTN network. Thank You, Gary H. Thompson ___ --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:
Re: [asterisk-users] Disconnection supervision: what about PBX
Hi, Trevor: Trevor G. Hammonds wrote: Stephen Bosch wrote: Are BRI circuits what phone companies call digital lines for use with digital sets, such as with digital Centrex? I'm not aware that Telus even offers BRI. Sorry -- BRI is ISDN, not digital Centrex. I'm still not aware that Telus even offers ISDN anymore :) ...and by that I mean ISDN BRI ;) -Stephen- Throughout most of the United States, Digital Centrex or CentrexIS is ISDN as part of a Centrex group. If the circuit is meant for a single device, it would be a BRI. If the circuit is Hi-Cap or meant to be hooked up to a PBX or the like, it would be a PRI. I am not that familiar with Telus, but what Bell is calling Digital Voice service is merely VoIP over one of their DSL connections. While I know that both companies offer Centrex over PRI, I am unsure if either company supports BRI widely anymore. I know BRI service is available, and most of their switches are capable of offering BRI circuits. For example, digital secretarial enhanced key telephone sets are ISDN phones that work via a BRI. In my experience, most telcos in the US and Canada will not tell you about BRI unless you specifically ask. And if you do, they shuffle you off to another department where they may or may not know how to properly provision the circuit. Somehow, all the LECs in North American look at BRI as a data-only service and never really saw the advantages of offering it to voice-only customers. As such, now that 128k (or 144k) is too slow of a data connection for most, BRI has just been passed by. Such a shame... I can still find information pages on BRI on the Telus website (buried, but there); as you point out, though, they refer to data connections only. I am going to give it a try and see what I come up with. There's every possibility they'll offer it but at a ridiculous price, just to discourage adoption enough to let them phase it out. I'll bet that this stuff will disappear when the switching equipment is upgraded. -Stephen- ___ --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] Does Asterisk support DNIS?
It will do so automatically if it is working. Asterisk will stuff those
digits into ${EXTEN}, therefore you need an exten = _XXX,1,Whatever if
you are expecting 3 digits.
Until recently we had DID service from our telco on an EM Wink
channelized voice T-1. The above is what we did.
David Ruggles wrote:
Yuan (and Matt),
Thanks for the reply, I'm sorry I kind of vented, I just got very frustrated
with trying to configure Asterisk for what (in a proprietary PBX) is
normally one of the easiest parts of configuration.
With a wink start T1 the DNIS digits are transmitted in-band. The Network
goes off hook, the PBX winks (goes off hook for 200ms) and then the network
sends DNIS (and ANI if used) as DTMF tones, after the PBX gets the tones it
answers the call (goes off hook). So you would tell the PBX to look for x
number of digits and then after getting that number of digits it will answer
the call. I have the Sangoma A101 configured for wink start, but I can't
find anything that says how to specify the number DNIS digits to expect. If
the PBX answers the call instead of just winking, the DTMF tones will be
transmitted during the call which is what seems to be happening here.
For more specific information a good overview of the wink start process can
be found here:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a0080
1123bb.shtml#topic2a
Can anyone tell me how to configure Asterisk to pickup the DNIS digits off a
wink start T1?
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU
Sent: Friday, February 16, 2007 5:57 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] Does Asterisk support DNIS?
Matt already replied to your other posting of similar content. I'm also a
bit confused. Do you mean you have observed that Asterisk is brought into
the intended context, but start to react to digits in DNIS one after
another? If so, can you estimate the interval Asterisk stays in each
extension?
If this is true, it seems to suggest that your provider is sending DNIS as a
DTMF string after Asterisk has answered the call. Isn't this a bit weird?
What does the card's manual say about DNIS (with wink start)?
Yuan Liu
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Re: [asterisk-users] Does Asterisk support DNIS?
Michael Winstead wrote: A PRI connection is required to pass DNIS digits. Just a 24 channel wink start T-1 with no D channel will not pass DNIS. That would depend on how you define DNIS. Dialed Number Identification Service. Most of the time the DNIS is the same as EXTEN, since the dialed number is the dialed extension. The only time this might not be the case is of someone dialed a number that was forwarded to a number on your Asterisk. If you don't care about such situations then a Channelized Voice T-1 will work just fine for DID service. ___ --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] Strange behaviour with Dial cmd
C F wrote: Asterisk supports this directly by issuing the hangup command before the answer command. However, when using an analog interface like FXO the line has no way of knowing you just hung up and will continue to ring, which asterisk will see as a new call. in my experience even when using a PRI if i dont give the pri cause the provider re initiates the call. This would only happen if you blindly run two Dial lines in sequence in your dialplan. Don't do this. ___ --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] Looking for starting point?
Gary H. Thompson wrote: Hi, Most of my experience in this field have been with Borland products, specifically Delphi. I also have been involved with database programming, same platform as the communications. I can't tell you anything more helpful than Michael has. I purchased a tomb called Fedora 5 and Red Hat Enterprise Bible form Wiley press, which has been very helpful. Also I would suggest purchasing or downloading Asterisk The Future of Telephony. See the Asterisk Doc project: http://www.asteriskdocs.org I personally like a paper book to have around. I am a Delphi man myself. As you become more familiar with and comfortable with Asterisk, you'll inevitably want to customize Asterisk's behavior through one of the interfaces available. One of the most common is the AGI interface which allows Asterisk to communicate with external programs through standard input and standard output (ie: Writeln(), Readln()). You can call these external programs directly from the dialplan and most people seem to prefer PHP, bash or other integrated scripting. Being a Delphi programmer as well, I wanted to let you know that I have had great success in using open source FreePascal and Lazarus IDE for developing linux based executables for the AGI interface which are basically just console type programs. I personally prefer using binaries because I think they execute faster than interpreted languages. I wrote a Cepstral Text to Speech wrapper using those tools not too long ago. Info and source is available on the wiki: http://www.voip-info.org/wiki/view/DTSwift+Cepstral+AGI+Wrapper When you're at the point where you want to start writing AGI and you if you decide to try freepascal/lazarus, that may help you get an idea of writing an AGI using FP/Laz. Welcome to the community. -- Warm Regards, Lee ___ --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] Strange behaviour with Dial cmd
Yes, but I would like to try a number and after to try a second one. Any Idea how to avoid this. On 2/18/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: C F wrote: Asterisk supports this directly by issuing the hangup command before the answer command. However, when using an analog interface like FXO the line has no way of knowing you just hung up and will continue to ring, which asterisk will see as a new call. in my experience even when using a PRI if i dont give the pri cause the provider re initiates the call. This would only happen if you blindly run two Dial lines in sequence in your dialplan. Don't do this. ___ --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] Looking for starting point?
hi, hi, i did wrote (assembler) programs for cp/m! if your experience is more on telephony', i think you will find trixbox easier. in one cd you will have a ready system. if your hardware is fully recognized, great ! do not use a too old machine nor a too new one. mind that the install will erase your hd. so buy a new cd if you would want to go back to windows. a 'normal' distribution will allow for dual-boot (not trixbox). mandrake, suse, fedora, red-hat centeos are good candidate. (k)ubuntu, debian, ... are also nice one but then you would have to download, compile, setup all. this could be harder to learn. but you learn a deeper way maybe go to a linux group in your neighborhood most people there are happy to welcome a newbie. buy some books about linux and stick with one distribution (you can change later) my 2c from a young chap of nearly 48 t. jacobson -- Initial header --- From : [EMAIL PROTECTED] To : asterisk-users@lists.digium.com CC : Date : Sun, 18 Feb 2007 14:05:15 -0500 Subject : [asterisk-users] Looking for starting point? Hi, I am a retired telephone tech/manager who recently had a bad experience with a local company offering digital phone service (VoIP). I have spent the last thirty years in the PSTN network, switching, PBX and key system field and am interested in learning more about VoIP. My background also includes programming, mostly specialized applications to interact with the PSTN network. Most of my experience in this field have been with Borland products, specifically Delphi. I also have been involved with database programming, same platform as the communications. My computer experience started with the operating system CPM (I'm not really that old, only 56). The best platform now seems to be Linux so now since I am retired now, it seems a good time to learn something new. I also have been looking at Asterisk which most companies seem to be using for a PBX platform. I found out by accident that the local company I had the problem with uses this PBX software. Could someone steer me in the right direction as to where to start? I spent most of my career in the telephone industry in a 'bush' area of Alaska so pretty much had to teach myself what I needed to know about computers but I can learn almost anything from a book and by asking questions when I get stuck. Most of my experience was before the Internet so I plan on using this avenue to advance my knowledge. I understand what a broad scope I am asking about so would appreciate any tips to help me get started. Since there are many 'brands' of Linux what is the best one to start with? Which Linux will be better when I get to the point of working with Asterisk? Any tips or ideas on books, online tutors, discussions or anything of this nature would be much appreciated. I hope to add to this group if I can be any assistance from the 'other side', the PSTN network. Thank You, Gary H. Thompson --- Scarlet One Unlimited Free national calls, surf up to 6 Mbit/s, 50 GB download volume For only EUR 49,95 per month. No Belgacom subscription needed. All in! http://www.scarlet.be ___ --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] HT488 doesn't disconnect FXO
Hi, I have HT488 with it's FXO connected to Israeli PSTN (bezeq) when dialing to that PSTN line asterisk see gets the call and direct it to the right extension but if the extension doesn't answer and the dialer is hanging the call the extension will keep on ringing. I'm not an expert but it seems like my asterisk doesn't recognize the hangup signal from the HT488 -or it's the HT88 which doesn't hangup upon the signal. this is my HT488 FXO config: PSTN Disconnect Tone: Frequency: f1 420 f2 420 PSTN Disconnect Tone Cadence:Choice 1: On 500ms Off 500ms Asterisk SVN-trunk-r48967 could someone please help? Thanks, Itamar. ___ --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] HT488 doesn't disconnect FXO
On Sun, Feb 18, 2007 at 10:55:20PM +0200, Itamar Lavender wrote: Hi, I have HT488 with it's FXO connected to Israeli PSTN (bezeq) when dialing to that PSTN line asterisk see gets the call and direct it to the right extension but if the extension doesn't answer and the dialer is hanging the call the extension will keep on ringing. I'm not an expert but it seems like my asterisk doesn't recognize the hangup signal from the HT488 -or it's the HT88 which doesn't hangup upon the signal. Chances are that the unit does not have proper busy detection. Perhaps it does not know how the Israeli busy tone sounds like. And no: Bezeq does not provide any better disconnect supervision. this is my HT488 FXO config: PSTN Disconnect Tone: Frequency: f1 420 f2 420 Try 400? PSTN Disconnect Tone Cadence:Choice 1: On 500ms Off 500ms Asterisk SVN-trunk-r48967 -- 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
Re: [asterisk-users] Does Asterisk support DNIS?
Arriving late to this discussion, sorry if this has already been mentioned
but DNIS and ANI can be variable length without confusion if the sender
uses the * DTMF tone as a delimiter. Thus sending *ANI*DNIS* (pronounced
Star ANI Star DNIS Star allows the receiver to identify the two values
unambiguously and to find the trailing boundary (when the 3rd * has been
received).
We have a Channelized Voice T1 from a long distance provider that is set
up this way into our non-Asterisk PBX where the provider sends us ANI as
the full originating phone number and DNIS as the last 4 digits. So the
DTMF string seen by our PBX for someone calling one of our toll-free
numbers, say 800-123-4567, from a local phone in Hawaii, say 808-555-1313,
would be *8085551313*4567*. The PBX parses this string and uses the
last 4 digits DNIS to route the call from the T1 trunk group to the proper
internal extension or hunt group.
Do Asterisk and Digium or Sangoma T1/E1 cards know about delimited ANI and
DNIS?
--Ron
On Sun, 18 Feb 2007, Eric ManxPower Wieling wrote:
It will do so automatically if it is working. Asterisk will stuff those
digits into ${EXTEN}, therefore you need an exten = _XXX,1,Whatever if
you are expecting 3 digits.
Until recently we had DID service from our telco on an EM Wink
channelized voice T-1. The above is what we did.
David Ruggles wrote:
Yuan (and Matt),
Thanks for the reply, I'm sorry I kind of vented, I just got very frustrated
with trying to configure Asterisk for what (in a proprietary PBX) is
normally one of the easiest parts of configuration.
With a wink start T1 the DNIS digits are transmitted in-band. The Network
goes off hook, the PBX winks (goes off hook for 200ms) and then the network
sends DNIS (and ANI if used) as DTMF tones, after the PBX gets the tones it
answers the call (goes off hook). So you would tell the PBX to look for x
number of digits and then after getting that number of digits it will answer
the call. I have the Sangoma A101 configured for wink start, but I can't
find anything that says how to specify the number DNIS digits to expect. If
the PBX answers the call instead of just winking, the DTMF tones will be
transmitted during the call which is what seems to be happening here.
For more specific information a good overview of the wink start process can
be found here:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a0080
1123bb.shtml#topic2a
Can anyone tell me how to configure Asterisk to pickup the DNIS digits off a
wink start T1?
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU
Sent: Friday, February 16, 2007 5:57 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] Does Asterisk support DNIS?
Matt already replied to your other posting of similar content. I'm also a
bit confused. Do you mean you have observed that Asterisk is brought into
the intended context, but start to react to digits in DNIS one after
another? If so, can you estimate the interval Asterisk stays in each
extension?
If this is true, it seems to suggest that your provider is sending DNIS as a
DTMF string after Asterisk has answered the call. Isn't this a bit weird?
What does the card's manual say about DNIS (with wink start)?
Yuan Liu
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Re: [asterisk-users] HT488 doesn't disconnect FXO
when directing the calls to the HT488 FXS instead of asterisk it does disconnect! that's why I believe this is has to do with asterisk in it's busy/hangup tone detection. Itamar. On Sun, 2007-02-18 at 23:09 +0200, Tzafrir Cohen wrote: Chances are that the unit does not have proper busy detection. Perhaps it does not know how the Israeli busy tone sounds like. And no: Bezeq does not provide any better disconnect supervision. this is my HT488 FXO config: PSTN Disconnect Tone: Frequency: f1 420 f2 420 Try 400? PSTN Disconnect Tone Cadence:Choice 1: On 500ms Off 500ms Asterisk SVN-trunk-r48967 ___ --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] Does Asterisk support DNIS?
Why would the card care? This would be something you'd take care of in your
dialplan.
On 2/18/07, Ron Fox [EMAIL PROTECTED] wrote:
Arriving late to this discussion, sorry if this has already been mentioned
but DNIS and ANI can be variable length without confusion if the sender
uses the * DTMF tone as a delimiter. Thus sending *ANI*DNIS* (pronounced
Star ANI Star DNIS Star allows the receiver to identify the two values
unambiguously and to find the trailing boundary (when the 3rd * has been
received).
We have a Channelized Voice T1 from a long distance provider that is set
up this way into our non-Asterisk PBX where the provider sends us ANI as
the full originating phone number and DNIS as the last 4 digits. So the
DTMF string seen by our PBX for someone calling one of our toll-free
numbers, say 800-123-4567, from a local phone in Hawaii, say 808-555-1313,
would be *8085551313*4567*. The PBX parses this string and uses the
last 4 digits DNIS to route the call from the T1 trunk group to the proper
internal extension or hunt group.
Do Asterisk and Digium or Sangoma T1/E1 cards know about delimited ANI and
DNIS?
--Ron
On Sun, 18 Feb 2007, Eric ManxPower Wieling wrote:
It will do so automatically if it is working. Asterisk will stuff those
digits into ${EXTEN}, therefore you need an exten = _XXX,1,Whatever if
you are expecting 3 digits.
Until recently we had DID service from our telco on an EM Wink
channelized voice T-1. The above is what we did.
David Ruggles wrote:
Yuan (and Matt),
Thanks for the reply, I'm sorry I kind of vented, I just got very
frustrated
with trying to configure Asterisk for what (in a proprietary PBX) is
normally one of the easiest parts of configuration.
With a wink start T1 the DNIS digits are transmitted in-band. The
Network
goes off hook, the PBX winks (goes off hook for 200ms) and then the
network
sends DNIS (and ANI if used) as DTMF tones, after the PBX gets the
tones it
answers the call (goes off hook). So you would tell the PBX to look
for x
number of digits and then after getting that number of digits it will
answer
the call. I have the Sangoma A101 configured for wink start, but I
can't
find anything that says how to specify the number DNIS digits to
expect. If
the PBX answers the call instead of just winking, the DTMF tones will
be
transmitted during the call which is what seems to be happening here.
For more specific information a good overview of the wink start
process can
be found here:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a0080
1123bb.shtml#topic2a
Can anyone tell me how to configure Asterisk to pickup the DNIS digits
off a
wink start T1?
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU
Sent: Friday, February 16, 2007 5:57 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] Does Asterisk support DNIS?
Matt already replied to your other posting of similar content. I'm
also a
bit confused. Do you mean you have observed that Asterisk is brought
into
the intended context, but start to react to digits in DNIS one after
another? If so, can you estimate the interval Asterisk stays in each
extension?
If this is true, it seems to suggest that your provider is sending
DNIS as a
DTMF string after Asterisk has answered the call. Isn't this a bit
weird?
What does the card's manual say about DNIS (with wink start)?
Yuan Liu
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Re: [asterisk-users] Does Asterisk support DNIS?
BTW. This seems kinda backwards. Why not just get a PRI. PRIs have all
the intelligence you need to do it right.
On 2/18/07, Matt [EMAIL PROTECTED] wrote:
Why would the card care? This would be something you'd take care of in
your dialplan.
On 2/18/07, Ron Fox [EMAIL PROTECTED] wrote:
Arriving late to this discussion, sorry if this has already been
mentioned
but DNIS and ANI can be variable length without confusion if the sender
uses the * DTMF tone as a delimiter. Thus sending *ANI*DNIS*
(pronounced
Star ANI Star DNIS Star allows the receiver to identify the two values
unambiguously and to find the trailing boundary (when the 3rd * has
been
received).
We have a Channelized Voice T1 from a long distance provider that is set
up this way into our non-Asterisk PBX where the provider sends us ANI as
the full originating phone number and DNIS as the last 4 digits. So
the
DTMF string seen by our PBX for someone calling one of our toll-free
numbers, say 800-123-4567, from a local phone in Hawaii, say
808-555-1313,
would be *8085551313*4567*. The PBX parses this string and uses the
last 4 digits DNIS to route the call from the T1 trunk group to the
proper
internal extension or hunt group.
Do Asterisk and Digium or Sangoma T1/E1 cards know about delimited ANI
and
DNIS?
--Ron
On Sun, 18 Feb 2007, Eric ManxPower Wieling wrote:
It will do so automatically if it is working. Asterisk will stuff
those
digits into ${EXTEN}, therefore you need an exten = _XXX,1,Whatever
if
you are expecting 3 digits.
Until recently we had DID service from our telco on an EM Wink
channelized voice T-1. The above is what we did.
David Ruggles wrote:
Yuan (and Matt),
Thanks for the reply, I'm sorry I kind of vented, I just got very
frustrated
with trying to configure Asterisk for what (in a proprietary PBX) is
normally one of the easiest parts of configuration.
With a wink start T1 the DNIS digits are transmitted in-band. The
Network
goes off hook, the PBX winks (goes off hook for 200ms) and then the
network
sends DNIS (and ANI if used) as DTMF tones, after the PBX gets the
tones it
answers the call (goes off hook). So you would tell the PBX to look
for x
number of digits and then after getting that number of digits it
will answer
the call. I have the Sangoma A101 configured for wink start, but I
can't
find anything that says how to specify the number DNIS digits to
expect. If
the PBX answers the call instead of just winking, the DTMF tones
will be
transmitted during the call which is what seems to be happening
here.
For more specific information a good overview of the wink start
process can
be found here:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a0080
1123bb.shtml#topic2a
Can anyone tell me how to configure Asterisk to pickup the DNIS
digits off a
wink start T1?
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yuan
LIU
Sent: Friday, February 16, 2007 5:57 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] Does Asterisk support DNIS?
Matt already replied to your other posting of similar content. I'm
also a
bit confused. Do you mean you have observed that Asterisk is
brought into
the intended context, but start to react to digits in DNIS one after
another? If so, can you estimate the interval Asterisk stays in
each
extension?
If this is true, it seems to suggest that your provider is sending
DNIS as a
DTMF string after Asterisk has answered the call. Isn't this a bit
weird?
What does the card's manual say about DNIS (with wink start)?
Yuan Liu
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Re: [asterisk-users] Strange behaviour with Dial cmd
Check the value of DIALSTATUS then decide of you want to dial the 2nd number. See [macro-std-exten] in extensions.conf for an example of checking the value of DIALSTATUS. The only time you might want two Dial lines in a row is if you always, not matter what, want to dial the 2nd number. Il Neofita wrote: Yes, but I would like to try a number and after to try a second one. Any Idea how to avoid this. On 2/18/07, *Eric ManxPower Wieling* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: C F wrote: Asterisk supports this directly by issuing the hangup command before the answer command. However, when using an analog interface like FXO the line has no way of knowing you just hung up and will continue to ring, which asterisk will see as a new call. in my experience even when using a PRI if i dont give the pri cause the provider re initiates the call. This would only happen if you blindly run two Dial lines in sequence in your dialplan. Don't do this. ___ --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] Does Asterisk support DNIS?
On Sun, 18 Feb 2007, Matt wrote:
Why would the card care? This would be something you'd take care of in your
dialplan.
Right, the card wouldn't care. So does Asterisk know about how to send
and receive delimited ANI and DNIS through a channelized voice T1?
--Ron
On 2/18/07, Ron Fox [EMAIL PROTECTED] wrote:
Arriving late to this discussion, sorry if this has already been mentioned
but DNIS and ANI can be variable length without confusion if the sender
uses the * DTMF tone as a delimiter. Thus sending *ANI*DNIS* (pronounced
Star ANI Star DNIS Star allows the receiver to identify the two values
unambiguously and to find the trailing boundary (when the 3rd * has been
received).
We have a Channelized Voice T1 from a long distance provider that is set
up this way into our non-Asterisk PBX where the provider sends us ANI as
the full originating phone number and DNIS as the last 4 digits. So the
DTMF string seen by our PBX for someone calling one of our toll-free
numbers, say 800-123-4567, from a local phone in Hawaii, say 808-555-1313,
would be *8085551313*4567*. The PBX parses this string and uses the
last 4 digits DNIS to route the call from the T1 trunk group to the proper
internal extension or hunt group.
Do Asterisk and Digium or Sangoma T1/E1 cards know about delimited ANI and
DNIS?
--Ron
On Sun, 18 Feb 2007, Eric ManxPower Wieling wrote:
It will do so automatically if it is working. Asterisk will stuff those
digits into ${EXTEN}, therefore you need an exten = _XXX,1,Whatever if
you are expecting 3 digits.
Until recently we had DID service from our telco on an EM Wink
channelized voice T-1. The above is what we did.
David Ruggles wrote:
Yuan (and Matt),
Thanks for the reply, I'm sorry I kind of vented, I just got very
frustrated
with trying to configure Asterisk for what (in a proprietary PBX) is
normally one of the easiest parts of configuration.
With a wink start T1 the DNIS digits are transmitted in-band. The
Network
goes off hook, the PBX winks (goes off hook for 200ms) and then the
network
sends DNIS (and ANI if used) as DTMF tones, after the PBX gets the
tones it
answers the call (goes off hook). So you would tell the PBX to look
for x
number of digits and then after getting that number of digits it will
answer
the call. I have the Sangoma A101 configured for wink start, but I
can't
find anything that says how to specify the number DNIS digits to
expect. If
the PBX answers the call instead of just winking, the DTMF tones will
be
transmitted during the call which is what seems to be happening here.
For more specific information a good overview of the wink start
process can
be found here:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a0080
1123bb.shtml#topic2a
Can anyone tell me how to configure Asterisk to pickup the DNIS digits
off a
wink start T1?
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU
Sent: Friday, February 16, 2007 5:57 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] Does Asterisk support DNIS?
Matt already replied to your other posting of similar content. I'm
also a
bit confused. Do you mean you have observed that Asterisk is brought
into
the intended context, but start to react to digits in DNIS one after
another? If so, can you estimate the interval Asterisk stays in each
extension?
If this is true, it seems to suggest that your provider is sending
DNIS as a
DTMF string after Asterisk has answered the call. Isn't this a bit
weird?
What does the card's manual say about DNIS (with wink start)?
Yuan Liu
___
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asterisk-users mailing list
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Re: [asterisk-users] Does Asterisk support DNIS?
Ron Fox wrote:
Arriving late to this discussion, sorry if this has already been mentioned
but DNIS and ANI can be variable length without confusion if the sender
uses the * DTMF tone as a delimiter. Thus sending *ANI*DNIS* (pronounced
Star ANI Star DNIS Star allows the receiver to identify the two values
unambiguously and to find the trailing boundary (when the 3rd * has been
received).
We have a Channelized Voice T1 from a long distance provider that is set
up this way into our non-Asterisk PBX where the provider sends us ANI as
the full originating phone number and DNIS as the last 4 digits. So the
DTMF string seen by our PBX for someone calling one of our toll-free
numbers, say 800-123-4567, from a local phone in Hawaii, say 808-555-1313,
would be *8085551313*4567*. The PBX parses this string and uses the
last 4 digits DNIS to route the call from the T1 trunk group to the proper
internal extension or hunt group.
Do Asterisk and Digium or Sangoma T1/E1 cards know about delimited ANI and
DNIS?
Not really, but you can emulate this by something like this (not tested,
but the idea and method is sound):
exten = _*NXXNXX**,1,Set(MY_ANI=${EXTEN:1:10})
exten = _*NXXNXX**,1,Goto(${EXTEN:12:4},1)
If you really have variable length of ANI and DID/DNIS then you would
need to use the Cut() function and specify the delimiter as *. Doing it
using Cut() is slightly more complicated, and requires more thought for
the design, but would be more reliable.
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Re: [asterisk-users] Does Asterisk support DNIS?
Matt wrote: BTW. This seems kinda backwards. Why not just get a PRI. PRIs have all the intelligence you need to do it right. There can be many reasons not to get a PRI. Most of them have to do with cost. Depending on the location and telco, a PRI can be MUCH more expensive than a CT1. PRI is, of course, the best solution, but it is not always possible to get a PRI. ___ --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] Does Asterisk support DNIS?
On Sun, 18 Feb 2007, Matt wrote:
BTW. This seems kinda backwards. Why not just get a PRI. PRIs have all
the intelligence you need to do it right.
You may not have that option. For example, you want to split a T1 from a
legacy PBX to 12 channels to a proprietary IVR system and 12 channels to
an Asterisk box. Can't do that with with PRI and a single T1 because you
only have one control channel.
--Ron
On 2/18/07, Matt [EMAIL PROTECTED] wrote:
Why would the card care? This would be something you'd take care of in
your dialplan.
On 2/18/07, Ron Fox [EMAIL PROTECTED] wrote:
Arriving late to this discussion, sorry if this has already been
mentioned
but DNIS and ANI can be variable length without confusion if the sender
uses the * DTMF tone as a delimiter. Thus sending *ANI*DNIS*
(pronounced
Star ANI Star DNIS Star allows the receiver to identify the two values
unambiguously and to find the trailing boundary (when the 3rd * has
been
received).
We have a Channelized Voice T1 from a long distance provider that is set
up this way into our non-Asterisk PBX where the provider sends us ANI as
the full originating phone number and DNIS as the last 4 digits. So
the
DTMF string seen by our PBX for someone calling one of our toll-free
numbers, say 800-123-4567, from a local phone in Hawaii, say
808-555-1313,
would be *8085551313*4567*. The PBX parses this string and uses the
last 4 digits DNIS to route the call from the T1 trunk group to the
proper
internal extension or hunt group.
Do Asterisk and Digium or Sangoma T1/E1 cards know about delimited ANI
and
DNIS?
--Ron
On Sun, 18 Feb 2007, Eric ManxPower Wieling wrote:
It will do so automatically if it is working. Asterisk will stuff
those
digits into ${EXTEN}, therefore you need an exten = _XXX,1,Whatever
if
you are expecting 3 digits.
Until recently we had DID service from our telco on an EM Wink
channelized voice T-1. The above is what we did.
David Ruggles wrote:
Yuan (and Matt),
Thanks for the reply, I'm sorry I kind of vented, I just got very
frustrated
with trying to configure Asterisk for what (in a proprietary PBX) is
normally one of the easiest parts of configuration.
With a wink start T1 the DNIS digits are transmitted in-band. The
Network
goes off hook, the PBX winks (goes off hook for 200ms) and then the
network
sends DNIS (and ANI if used) as DTMF tones, after the PBX gets the
tones it
answers the call (goes off hook). So you would tell the PBX to look
for x
number of digits and then after getting that number of digits it
will answer
the call. I have the Sangoma A101 configured for wink start, but I
can't
find anything that says how to specify the number DNIS digits to
expect. If
the PBX answers the call instead of just winking, the DTMF tones
will be
transmitted during the call which is what seems to be happening
here.
For more specific information a good overview of the wink start
process can
be found here:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a0080
1123bb.shtml#topic2a
Can anyone tell me how to configure Asterisk to pickup the DNIS
digits off a
wink start T1?
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yuan
LIU
Sent: Friday, February 16, 2007 5:57 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] Does Asterisk support DNIS?
Matt already replied to your other posting of similar content. I'm
also a
bit confused. Do you mean you have observed that Asterisk is
brought into
the intended context, but start to react to digits in DNIS one after
another? If so, can you estimate the interval Asterisk stays in
each
extension?
If this is true, it seems to suggest that your provider is sending
DNIS as a
DTMF string after Asterisk has answered the call. Isn't this a bit
weird?
What does the card's manual say about DNIS (with wink start)?
Yuan Liu
___
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Re: [asterisk-users] Does Asterisk support DNIS?
Again,
I think this would be something you'd have to do in your dial-plan. If you
want Asterisk to SEND the info. You need to code your dial plan to send it
when the call starts up.
On 2/18/07, Ron Fox [EMAIL PROTECTED] wrote:
On Sun, 18 Feb 2007, Matt wrote:
Why would the card care? This would be something you'd take care of in
your
dialplan.
Right, the card wouldn't care. So does Asterisk know about how to send
and receive delimited ANI and DNIS through a channelized voice T1?
--Ron
On 2/18/07, Ron Fox [EMAIL PROTECTED] wrote:
Arriving late to this discussion, sorry if this has already been
mentioned
but DNIS and ANI can be variable length without confusion if the
sender
uses the * DTMF tone as a delimiter. Thus sending *ANI*DNIS*
(pronounced
Star ANI Star DNIS Star allows the receiver to identify the two
values
unambiguously and to find the trailing boundary (when the 3rd * has
been
received).
We have a Channelized Voice T1 from a long distance provider that is
set
up this way into our non-Asterisk PBX where the provider sends us ANI
as
the full originating phone number and DNIS as the last 4 digits. So
the
DTMF string seen by our PBX for someone calling one of our toll-free
numbers, say 800-123-4567, from a local phone in Hawaii, say
808-555-1313,
would be *8085551313*4567*. The PBX parses this string and uses the
last 4 digits DNIS to route the call from the T1 trunk group to the
proper
internal extension or hunt group.
Do Asterisk and Digium or Sangoma T1/E1 cards know about delimited ANI
and
DNIS?
--Ron
On Sun, 18 Feb 2007, Eric ManxPower Wieling wrote:
It will do so automatically if it is working. Asterisk will stuff
those
digits into ${EXTEN}, therefore you need an exten = _XXX,1,Whatever
if
you are expecting 3 digits.
Until recently we had DID service from our telco on an EM Wink
channelized voice T-1. The above is what we did.
David Ruggles wrote:
Yuan (and Matt),
Thanks for the reply, I'm sorry I kind of vented, I just got very
frustrated
with trying to configure Asterisk for what (in a proprietary PBX)
is
normally one of the easiest parts of configuration.
With a wink start T1 the DNIS digits are transmitted in-band. The
Network
goes off hook, the PBX winks (goes off hook for 200ms) and then
the
network
sends DNIS (and ANI if used) as DTMF tones, after the PBX gets the
tones it
answers the call (goes off hook). So you would tell the PBX to
look
for x
number of digits and then after getting that number of digits it
will
answer
the call. I have the Sangoma A101 configured for wink start, but I
can't
find anything that says how to specify the number DNIS digits to
expect. If
the PBX answers the call instead of just winking, the DTMF tones
will
be
transmitted during the call which is what seems to be happening
here.
For more specific information a good overview of the wink start
process can
be found here:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a0080
1123bb.shtml#topic2a
Can anyone tell me how to configure Asterisk to pickup the DNIS
digits
off a
wink start T1?
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yuan
LIU
Sent: Friday, February 16, 2007 5:57 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] Does Asterisk support DNIS?
Matt already replied to your other posting of similar
content. I'm
also a
bit confused. Do you mean you have observed that Asterisk is
brought
into
the intended context, but start to react to digits in DNIS one
after
another? If so, can you estimate the interval Asterisk stays in
each
extension?
If this is true, it seems to suggest that your provider is sending
DNIS as a
DTMF string after Asterisk has answered the call. Isn't this a
bit
weird?
What does the card's manual say about DNIS (with wink start)?
Yuan Liu
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RE: [asterisk-users] Summary of Trixbox vs. custom install
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: Sunday, February 18, 2007 11:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Summary of Trixbox vs. custom install snip I also include a consideration from mine: I would happily use Trixbox, because I did FreePBX setup once and it was a real pain, but I'm very frightened by a few issues: 1) Trixbox Macho installation that installs everything without asking. I, for example, would like to use software RAID (maybe it's wrong with Asterisk, but I want to do it!). I wouldn't like doing it manually after Trixbox installation. I would like to have an installer doing it for me. Centos (ex redhat) installer does it, so why Trixbox choose to install everything without prompting? You can just install CentOS with RAID and whatever you want, then use the Trixbox tar package instead of the ISO. Still, why on earth did the Trixbox team didn't leave the option of doing a custom install with the ISO ? /snip Please note that the recent (2.x) releases of trixbox allow you to select which modules to install, including raid. This message is confidential. It may also be privileged or otherwise protected by work product immunity or other legal rules. If you have received it by mistake, please let us know by e-mail reply and delete it from your system; you may not copy this message or disclose its contents to anyone. Please send us by fax any message containing deadlines as incoming e-mails are not screened for response deadlines. The integrity and security of this message cannot be guaranteed on the Internet. ___ --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] moving WiFi phone
I, too, have heard about that best practice of using different channels for different AP's on the same SSID. As far as I can tell, This is standard textbook stuff. Read Cisco press's 'Deploying License Free Wireless Wide-Area Networks' by Jack Unger. it's BS. I don't know who started it, but it has never worked in any of the situations I've encountered. In fact, I know of at least one AP manufacturer (Apple) that has a utility to auto-configure WDS networks, and it auto-configures to use the same channel. That's Using the same channel is bad, because the APs will interfere with each other and your throughput will be reduced. Imagine if you have a total of 2 APs with 10 clients each, the bandwidth will have to be shared amongst the 22 devices. So, if you're able to get 54Mbps on that channel, the net result is everybody gets 54/22 = 2.45Mbps each. Not a very pretty sight. Roaming with multiple APs on the same channel is OK for small set ups. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Does Asterisk support DNIS?
I'm sending 12345 as DNIS on a Wink Start T1. In case it makes a difference, I'm using a Sangoma A101 card. Asterisk sees each digit as a separate extension number so most of the dialplain suggestions offered so far won't work. I did try the Wait() function as was suggested. I tried it first in an s extension but this didn't work, it still gave the error: Unknown extension '1' in context '1st-T1' requested I then changed it to extension 1 and while it does seem to work (it doesn't try the other extensions) it seems like the DNIS is completely lost. As I said in my first post (although it may have been a little too abrasive) this configuration is very standard and so I find it hard to believe that Asterisk can't handle it. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[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] Does Asterisk support DNIS?
David Ruggles wrote: I'm sending 12345 as DNIS on a Wink Start T1. In case it makes a difference, I'm using a Sangoma A101 card. Asterisk sees each digit as a separate extension number so most of the dialplain suggestions offered so far won't work. I did try the Wait() function as was suggested. I tried it first in an s extension but this didn't work, it still gave the error: Unknown extension '1' in context '1st-T1' requested I then changed it to extension 1 and while it does seem to work (it doesn't try the other extensions) it seems like the DNIS is completely lost. As I said in my first post (although it may have been a little too abrasive) this configuration is very standard and so I find it hard to believe that Asterisk can't handle it. We had to add this to the /etc/asterisk/zapata.conf to make Asterisk work with the EM Wink start T-1 from our telco. ___ --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] Does Asterisk support DNIS?
Eric ManxPower Wieling wrote: David Ruggles wrote: I'm sending 12345 as DNIS on a Wink Start T1. In case it makes a difference, I'm using a Sangoma A101 card. Asterisk sees each digit as a separate extension number so most of the dialplain suggestions offered so far won't work. I did try the Wait() function as was suggested. I tried it first in an s extension but this didn't work, it still gave the error: Unknown extension '1' in context '1st-T1' requested I then changed it to extension 1 and while it does seem to work (it doesn't try the other extensions) it seems like the DNIS is completely lost. As I said in my first post (although it may have been a little too abrasive) this configuration is very standard and so I find it hard to believe that Asterisk can't handle it. We had to add this to the /etc/asterisk/zapata.conf to make Asterisk work with the EM Wink start T-1 from our telco. I guess I could paste the settings this time. wink=270 rxwink=270 You might want to play with those settings. ___ --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] Problem with busydetect and cell phones
Stephen Bosch wrote: Have you tried calling ATT and asking for call disconnect supervision? I realise that this can be a thankless and tedious endeavour, but it IS worth trying. There are almost no commercial switches that don't support this; it's a matter of activating it for the specific circuit in software. Particularly if you have a business line -- you can demand it. All PBXs need it if they use analog lines (and plenty still do) so I'm sure this is not an alien concept to ATT. It's just a matter of finding the right Earthling there who can help you. This might be one of those times where a beer with the technician will get you some joy, if calling Repair doesn't give you any joy. Better luck with ATT than I had with the Monon Telephone Company. They have a switch that's fairly new, so I called them--I'm a loyal but tightly captive customer of the last 25 years. Their chief technician told me, Sure, our switch is new. There's nothing to it more than a setting on a software screen. But we don't have to do it because it's not in our tariffs. So forget it. And then he hung up on me. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Does Asterisk support DNIS?
Also check out immediate=no On 2/18/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Eric ManxPower Wieling wrote: David Ruggles wrote: I'm sending 12345 as DNIS on a Wink Start T1. In case it makes a difference, I'm using a Sangoma A101 card. Asterisk sees each digit as a separate extension number so most of the dialplain suggestions offered so far won't work. I did try the Wait() function as was suggested. I tried it first in an s extension but this didn't work, it still gave the error: Unknown extension '1' in context '1st-T1' requested I then changed it to extension 1 and while it does seem to work (it doesn't try the other extensions) it seems like the DNIS is completely lost. As I said in my first post (although it may have been a little too abrasive) this configuration is very standard and so I find it hard to believe that Asterisk can't handle it. We had to add this to the /etc/asterisk/zapata.conf to make Asterisk work with the EM Wink start T-1 from our telco. I guess I could paste the settings this time. wink=270 rxwink=270 You might want to play with those settings. ___ --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] Problem with busydetect and cell phones
Brian Capouch wrote: Better luck with ATT than I had with the Monon Telephone Company. They have a switch that's fairly new, so I called them--I'm a loyal but tightly captive customer of the last 25 years. Their chief technician told me, Sure, our switch is new. There's nothing to it more than a setting on a software screen. But we don't have to do it because it's not in our tariffs. So forget it. And then he hung up on me. ...wow. This society is doomed. -s ___ --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] Does Asterisk support DNIS?
Would you attach your whole zaptel.conf and zapata.conf? --- C F [EMAIL PROTECTED] wrote: Also check out immediate=no On 2/18/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Eric ManxPower Wieling wrote: David Ruggles wrote: I'm sending 12345 as DNIS on a Wink Start T1. In case it makes a difference, I'm using a Sangoma A101 card. Asterisk sees each digit as a separate extension number so most of the dialplain suggestions offered so far won't work. I did try the Wait() function as was suggested. I tried it first in an s extension but this didn't work, it still gave the error: Unknown extension '1' in context '1st-T1' requested I then changed it to extension 1 and while it does seem to work (it doesn't try the other extensions) it seems like the DNIS is completely lost. As I said in my first post (although it may have been a little too abrasive) this configuration is very standard and so I find it hard to believe that Asterisk can't handle it. We had to add this to the /etc/asterisk/zapata.conf to make Asterisk work with the EM Wink start T-1 from our telco. I guess I could paste the settings this time. wink=270 rxwink=270 You might want to play with those settings. ___ --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 Expecting? Get great news right away with email Auto-Check. Try the Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/newmail_tools.html ___ --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] Problem with busydetect and cell phones
Stephen Bosch wrote: And then he hung up on me. ...wow. This society is doomed. Actually, it isn't so much society as the legacy telcos. But unfortunately, they've been pretty smart about using the billions that they've stolen from us over the years: they use a lot of it to line the pockets of our legislators, and then have them write laws (such as the recent SBC Benficiency Law in Indiana) that stifle competition in the local loop and put their competitors at a disadvantage. Martin at the FCC has been a disaster for competition; SBC now has all the old ATT properties, and they're just a few new regulatory laws away from having their monopoly back. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
