Re: [asterisk-users] Context documentation for the newbie!
Bsumrall, Take a look on this document, http://bef.eventphone.de/a/Ast.%20C.%20I._files/ast-ci-draft1.pdf /Mats On 6/1/07, C F [EMAIL PROTECTED] wrote: I can give the following example, let me know if it helps. Mr 1 has a child Mr 10 and another child Mr 11, now Mr 10 has Mr 100 and Mr 11 has Mr 111. Mr 10 adopts Mr 111. Also Mr 88 adopts Mr 10. Which brings us to the family tree, if you are a child of one, you are a grandchild of that ones parent, and as such included in that tree. Now one of the children could be adopted by some other parent as well, which makes that child a child of another parent hence a grandchild of that parents parent. Subistute child and adopt for include =, and Mr for context so you got: [1] include = 10 include = 11 [10] include = 100 include = 111 [11] include = 111 [88] include = 10 Within each context you got the instruction code, which is an extension (exten) prioritized with numbers (or n for next number). The instructions are executed one after the other, unless a jump is encountered. Each extension is a pointer within that context that starts the instruction set. In Asterisk one starts in a context, when an extension is called (by dialing, or s when the extension number wasn't given) Asterisk looks for that extension in that context, if it can't find it there it searches in that contexts family tree, if still no match it searches in default context, if still no match it searches for the i extension in the same order, if still no match then 404 is given. Hope this helps. On 5/31/07, BSumrall [EMAIL PROTECTED] wrote: Does anyone know where there is better documentation on understanding context relations and priorities with examples? http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Introduction Does tell me anything other than they point to each other. Not how or who comes first or even how to get them to work with each other! ___ --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] Context documentation for the newbie!
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 On 5/31/07, BSumrall [EMAIL PROTECTED] wrote: Does anyone know where there is better documentation on understanding context relations and priorities with examples? http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Introduction Does tell me anything other than they point to each other. Not how or who comes first or even how to get them to work with each other! ___ --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] Zaptel linux26
1. Read the error No such file or... Do a yum install autoconf 2. And the other error No rule to make target `linux26' just use make NOT make linux26 since it seems that that is removed! 3. Get rid of that disclaimer ! This will help you for 99.9% of your problems: echo '16i[q]sa[ln0=aln100%Pln100/snlbx]sbA0D4D465452snlbxq' | dc On 5/29/07, Khaled Chehab [EMAIL PROTECTED] wrote: I am using centos 4.4 ,when I am compiling zapltel using l make linux26 ,error accrued ,what s missing [EMAIL PROTECTED] zaptel]# make linux26 grep: /include/linux/autoconf.h: No such file or directory make: *** No rule to make target `linux26'. Stop. Regards -- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] test tools of Asterisk server
HP's tool can be found at sipp.sf.net. Im unshure if you have to use unstable to get rtp support or if they hasve released it as stable. /M Andrew Joakimsen wrote: HP has a tool that is a free Open Source test tool / traffic generator for the SIP protocol. On 5/26/07, khawla khawla [EMAIL PROTECTED] wrote: I am using Aserisk as a SIP server to interconnect differents PBX in differents sites. I am now looking for a tool that can test the performance of this solution: I mean is there a tool that enables me to test the capacity of this SIP server in terms of simultaneous calls that could be treated, the comsuption of bandwidth.. or any thing like this? I am in urgent need to such a tool, If anyone could help, I would be geatful. This will help you for 99.9% of your problems: echo '16i[q]sa[ln0=aln100%Pln100/snlbx]sbA0D4D465452snlbxq' | dc ___ --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] cpu usuage
Does such formula exist ? And do you have other functions/apps that demands cpu power that needs to be taken into the formula. And please skip that disclaimer you have in the bottom ! /Mats On 5/15/07, Khaled Chehab [EMAIL PROTECTED] wrote: Do any one knows the formula to calculate memory and cpu usuage for channel on g729 codec,to know the hardware required for 100 concurrent call. Regards ___ --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] Upgrading my office - Need help
Gary Guthary wrote: Hi - If I'm posting this in the wrong place, pease forgive me Folks, I need help... One company I consult for is upgrading their office and will need a PBX replacement in the next two months. I'm seriously thinking of offering them an 'Asterisk' solution versus them getting locked in with some PBX vendor. This means I've got to come up with some sort of demo system to show them. I've got the hardware. - Dedicated Linux box I can re-config and/or re-load as needed. - Have one FXS/FXO card to demo intfc to telco. - And a bunch of Cisco 79xx phones I can use for demos. If I can get this rolling, I'll be more than happy to pay ***MONEY*** to anybody who can help. - Also, I'm not afraid to pay for already-developed admin software I can use to manage my system. - I just don't know 'which is which'. To prevent cluttering up this board, please send all responses to: [EMAIL PROTECTED] Thanks in advance again apologize if this is not the right place to post this. Gary Guthary Take a look at trixbox.org. /M -- Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) ___ --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] zaptel errors
Ariel Batista wrote: Justin Johnson wrote: Hi All, I have centOS 4.3 installed and have attempted to install asterisk separately. I have installed all the modules as suggested on Asterisk downloads, more (via SVN) However, on the zaptel install I am getting the following errors. centosbug is, like, a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h This doesn't work if you use a smp kernel ! So I use : sed -i s/rw_lock/rwlock/ /usr/src/kernels/2.6*/include/linux/spinlock.h But ensure that there is only one kernel-devel version, uninstall the one that isn't in use ! make[3]: *** [/usr/src/zaptel/torisa.o] Error 1 make[2]: *** [_module_/usr/src/zaptel] Error 2 make[2]: Leaving directory `/usr/src/kernels/2.6.9-34.0.1.EL-i686' make[1]: *** [linux26] Error 2 make[1]: Leaving directory `/usr/src/zaptel' make: *** [all] Error 2 Any one have any ideas how I can solve this? Thanks in advance, Justin /Mats -- Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) ___ --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] Backup Question?
Tzafrir Cohen wrote: On Thu, Jun 15, 2006 at 08:20:29AM -0400, BerkHolz, Steven wrote: This may be slightly off topic. I am using FreePBX, and using it's backup feature. Here is the question part: I would like to copy my backup off the asterisk server. Please define what is it exactly that you want to back up. FreePBX is just creating a tar file of asterisk config files and a mysql dump and some more config files. And I asume that it's this file you wan't to transport to the windows machine mentioned later in your original mail. From your experiences, which approach seems more resilient to failure: Push the backup from asterisk to another server using STFP or FTP? Pull the backups from asterisk from another server using SFTP or other? The destination would most likely be going to a Windows Server. I also have the option of installing Veritas Netbackup Client on the asterisk server, but I assume that this would not be good for the PBX's performance. This may be useful handy for data backup. For a system backup I found mondorescue very handy: http://www.mondorescue.org/ And you could use one of the addons in the putty package, PSCP (an SCP client, i.e. command-line secure file copy). SCP is using SSH as transport so it's quite safe. /Mats -- Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk server
Andrew Nowrot wrote: Hi, I have to build Asterisk server for about 30 user (30 concurrent calls). I decided to buy this box: -- motherboard Intel E7210 + Hence Rapids -- processor P4 3.0 GHz -- RAM 2x512 MB DDR ECC -- network interface Intel 82541 GI Is this configuration enough to handle 30 users at the same time. I am not planning to use any transcoding (everything will be alaw). Cheers Andrew Yes. /M -- Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Nokai E60 and E61 , working fine with Asterisk , with new access points
What about using a search engine like google before ... http://www.google.se/search?client=firefox-arls=org.mozilla%3Aen-GB%3Aofficial_shl=svq=nokia+e60+asteriskmeta=lr%3Dlang_en%7Clang_no%7Clang_svbtnG=Google-s%C3%B6kning and you will find several examples that is working for you! /Mats [EMAIL PROTECTED] wrote: Hi, Im an unsuccessful user of E60. Please post the configs on the phone in detail Thanks Dan On 11/06/06, Markus Schuster [EMAIL PROTECTED] wrote: John Joseph wrote: Was able to communicate clearly with e60 and E61 with asterisk with new access point [..] Could you please post some details (or even better: write them in some sort of Wiki) on the configuration you did on the Nokia? I'm thinking about buying a Nokia E60 but after a short web search there seem to be some problems about the correct configuration of the phone. Greetings, Markus -- Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) ___ --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] JAMAICA DID'S - 1-876
Post this to asterisk-biz, NOT asterisk-users /M -- Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) ___ --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] Transcoding g.711 - g.729
This tool can convert from and to any codec loaded in asterisk: http://redice.krisk.org/res_conv-0.1.tgz Except for some small errors in README regarding versions ++ have it been a nice tool and helped me a lot. I have a small script that I have used to convert many of my prompts, mail me if you are intressted. /Mats Douglas Garstang wrote: I don't know about g.729, but this will work for wav - g711. sox file.wav file.ul Doug. -Original Message- From: Matthew Crocker [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 06, 2006 12:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Transcoding g.711 - g.729 Hello, I have an asterisk server running with 23 g.729 licenses. I have also purchased a sound file from thevoice.digium.com. I need to covert this file (uLaw, PCM I think) to g.711, g.729 g.723 for use with an IVR system. Is there a way I can convert the files using the g.729 digium codec? sox? Thanks -Matt -- Matthew S. Crocker Vice President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.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
