Re: [asterisk-users] Problem with DAHDI and OSLEC integration.
Hi Joseph and Tzafrir. Thank you for your suggestions, feedbacks. For Joseph: yes. I had the same warning messages and I solved with the trick I suggested. Now oslec seems working (or at least and I can set it through the dahdi_cfg command ;-) ). For Tzafrir: here are the steps I did: 1. Taken the svn revision 5366 into my temporary folder /home/marco/Install/dahdi-linux 2. Taken the linux-2.6.27 kernel sources baseline and placed in my temporary folder /home/marco/install/linux-2.6.27 3. Taken the Linux kernel patch-2.6.28-rc6.gz, unzipped and applied to the baseline kernel 2.6.27. This generates the folder ...linux-2.6.27/drivers/staging/echo 4. Copied the folder /staging/echo into /home/marco/Install/dahdi-linux/drivers 5. Uncommented the oslec related two lines in the file Kbuild 6. From the folder /home/marco/install/dahdi-linx I've issued the command make The compiler starts and seems not able to compile what's present in the folder /home/marco/Install/dahdi-linux/drivers/staging/echo. This produces the warning already reported by Joseph and the inability to run the oslec module. I've had better results modifying the line: obj-m += ../staging/echo/ with obj-m += ../staging/echo/echo.o in the Kbuild file. I don't know if could be helpful, but I'm running these stuffs on OpenSuse 11. Thank you and best regards, Marco Signorini. Joseph L. Casale wrote: Have you copied there the files from the directory drivers/staging/echo in a recent (that is: = 2.6.28-rc1) kernel tree? Tzafrir, Thank you for following up on this. I don't have a quick command for only the three files, I just grabbed the tar ball. But like the OP, the only difference was that he used 2.6.28-rc6 and I used 2.6.28-rc5. I am pretty sure we had the same errors which I posted: http://lists.digium.com/pipermail/asterisk-users/2008-November/222063.html Thanks for any pointers! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Return-Path: [EMAIL PROTECTED] Received: from mailrelay07.libero.it (192.168.32.94) by ims67c.libero.it (8.0.019) id 489AF14B0071194C for [EMAIL PROTECTED]; Mon, 24 Nov 2008 00:45:09 +0100 X-IronPort-Anti-Spam-Filtered: true X-IronPort-Anti-Spam-Result: AkgAABB6KUnYz/URlGdsb2JhbACBbZFvAQEBAQkLCAkRBLlNgnyBVA X-IronPort-AV: E=Sophos;i=4.33,655,1220227200; d=scan'208;a=576251938 Received: from lists.digium.com ([216.207.245.17]) by mailrelay07.libero.it with ESMTP; 23 Nov 2008 23:45:08 + Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by lists.digium.com with esmtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1L4OY6-0007qt-UX; Sun, 23 Nov 2008 17:39:31 -0600 Received: from idcmail-mo2no.shaw.ca ([64.59.134.9]) by lists.digium.com with esmtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1L4OXy-0007qj-L3 for asterisk-users@lists.digium.com; Sun, 23 Nov 2008 17:39:22 -0600 Received: from pd6ml1no-ssvc.prod.shaw.ca ([10.0.153.160]) by pd7mo1no-svcs.prod.shaw.ca with ESMTP; 23 Nov 2008 16:39:17 -0700 X-Cloudmark-SP-Filtered: true X-Cloudmark-SP-Result: v=1.0 c=0 a=Kd8dHRva:8 a=OvmmjL8WYY_iC51gqwYA:9 a=pC6ppZV0J0nBJUWzJgg687EU0CoA:4 a=YPYbZooERpMA:10 a=AKRigw6aElYA:10 Received: from s0106001e8c610de2.cg.shawcable.net (HELO mail.activenetwerx.com) ([68.144.97.215]) by pd6ml1no-dmz.prod.shaw.ca with ESMTP; 23 Nov 2008 16:39:16 -0700 Received: from exchange.activenetwerx.com (mail.activenetwerx.com [127.0.0.1]) by mail.activenetwerx.com (Postfix) with ESMTP id 6EC8768136 for asterisk-users@lists.digium.com; Sun, 23 Nov 2008 16:39:24 -0700 (MST) Received: from exchange.activenetwerx.com ([192.168.0.3] helo=exchange.activenetwerx.com) by mail.activenetwerx.com; 23 Nov 2008 16:39:24 -0700 Received: from Mail.activenetwerx.int ([::1]) by Mail.activenetwerx.int ([::1]) with mapi; Sun, 23 Nov 2008 16:39:15 -0700 From: Joseph L. Casale [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Sun, 23 Nov 2008 16:39:14 -0700 Thread-Topic: [asterisk-users] Problem with DAHDI and OSLEC integration. Thread-Index: AclNwtn/mA/EFuLiQUqEGjb+9HhEWQAABJlg Message-ID: [EMAIL PROTECTED] References: [EMAIL PROTECTED] [EMAIL PROTECTED] In-Reply-To: [EMAIL PROTECTED] Accept-Language: en-US Content-Language: en-US X-MS-Has-Attach: X-MS-TNEF-Correlator: acceptlanguage: en-US MIME-Version: 1.0 Subject: Re: [asterisk-users] Problem with DAHDI and OSLEC integration. X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.9 Precedence: list Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
[asterisk-users] SendText and non-ASCII characters
Hi, Is is possible to translate non-english text into ASCII text so that SIP phones would correctly display non-ASCII characters received from SendText() ? I think SIP MESSAGE (rfc3428) on which SendText() currently relies, defines text/plain Content-type but googling, I can't find a source describing what text/plain can or cannot be. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Server Fine Tuning for Best Performance
Hi, Is there some asterisk fine tunning documentation related with System Hardware Optimizations, Operating System Tuning, Network Stack Tuning, Asterisk Settings, Network Hardware Settings, etc to get the best performance possible? Regards, Fernando ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and v4l
On Sun, Nov 23, 2008 at 09:49:53PM -0500, Jerry Geis wrote: Again - I am looking at using a USB web camera (v4l) and connecting that to a video phone with asterisk. Is there anything like that out there? chan_oss should include some v4l support. Not sure about other console drivers. Alternatively you could use a software voip phone that supports a camera (ekiga, linphone, and probably some others). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: SPA2100 transfer to ASTERISK CID
Are my e-mails arriving to the list? can somebody confirm? Sebastian -- Forwarded message -- Date: Fri, Nov 21, 2008 at 11:06 AM Subject: SPA2100 transfer to ASTERISK CID To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi all, I have around 100 SPA2100 registered in my provider openSER. I'd like to add an Asterisk registered into openSER, to the network, to deploy voicemail service for those SPAs. Due to administration access levels, I have no access to SER box, so I'm wondering if that possible: - Some foreign user (say A) calls one of my SPA (say B). - B don't answer. So.. B SPA is setted up to transfer in 20 seconds on no answer to the number in Asterisk. All ok so far, however, in Asterisk I receive de caller ID = A, but I need B CID. Having B caller ID I could let A leave the message into B mailbox. Can anybody helpme with that please? Thanks very much in advance Sebastian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: SPA2100 transfer to ASTERISK CID
Yes Sent via BlackBerry from T-Mobile -Original Message- From: Sebastian Milioto [EMAIL PROTECTED] Date: Mon, 24 Nov 2008 08:52:04 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [asterisk-users] Fwd: SPA2100 transfer to ASTERISK CID ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] play sound while executing agi script
Hello, Is it possible to do like this: play a sound file (if needed play in loop) while php agi script finishes work ? And how to do this? When on my server is huge load , I don't want that client hears silent , but hears music. Thanks -- Pagarbiai / Best Regards, Giedrius Augys ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR Design
I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal (link ommitted to see if email gets accepted). After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation that is already overly so. I think it's a mistake to try and think about all the different call scenarios and come up with little tricks for the more complicated ones. There will always be something missed; app_shotgun initiates calls to 100 random numbers and as soon as three or more calls are answered it will start randonly transferring them amongst each other at 2 second intervals. I think it's important to clarify at the outset what a CDR should be. The most fundamental requirement for CDRs is that they accurately record the following pieces of information for EVERY call entering or leaving the system (note every means every and not; channel calls but not peer calls). 1. Destination (aka as A Number) 2. AccountCode (aka as B Number) 3. Call Start Time (answer time), 4. Duration. Of course adding extra information can be very useful and I'm not proposing any fields be removed from the current implementation (although for pity's sake one change that should be made it to use a GUID/UUID for the CDR's uniqueid and save endless confusion). People that really do need verbose or enhanced CDRs to do things like tracking a call's flow as it travels in and out of queues, parking lots etc. would be better off using AMI or the new CEL and not CDRs. At the very least if problems arise with their call flow tracking they will still be able to rely on the accuracy of the CDRs to piece it altogether to work out what's going wrong. My proposal of creating a 1-to-1 relationship between CDRs and Asterisk channels already exsits but somewhere along the line it's going awry. As an experiment, and to actually do something instead of continually moaning about it, I started commenting out the blocks of code in res_featrures.c and sip_channel.c that muck around with the channel CDRs when a transfer occurs. The results of that were that the CDRs for blind and attended transfers actually got better! They're still not quite right but are pretty close with only one CDR on each having a wrong destination. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play sound while executing agi script
On Monday 24 November 2008 06:51:23 Giedrius Augys wrote: Is it possible to do like this: play a sound file (if needed play in loop) while php agi script finishes work ? And how to do this? When on my server is huge load , I don't want that client hears silent , but hears music. Thanks Please see StartMusicOnHold and StopMusicOnHold. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Design
I think that the custom cdr back-end can be successfully used to maximize or minimize the CDRs detailing on a per-needs basis. Furthermore, the CDR() function gives plenty of room for even more detailing. In my opinion the detail level (fields) is not the issue with the CDRs generation nor is the lack of backends (asterisk gives a lot of different backends to store your CDRs). I find the issue with asterisk CDRs to be in the lack of proper CDRs generation for the B-leg of every call. If we want to really track what happens during a call through the CDRs one has to have all the details not only for the incoming channel but for the outgoing one as well. Furthermore, one needs to be able to tweak the B-leg CDRs like he does with the incoming legs. So what needs to be done in my opinion is record every B-leg CDR when such an event occurs and give the user access to the CDR info by extending the CDR() function (so that one can specify the channel of the CDR that is being tweaked) or creating a seperate one for the outgoing channels. Grey Man wrote: I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal (link ommitted to see if email gets accepted). After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation that is already overly so. I think it's a mistake to try and think about all the different call scenarios and come up with little tricks for the more complicated ones. There will always be something missed; app_shotgun initiates calls to 100 random numbers and as soon as three or more calls are answered it will start randonly transferring them amongst each other at 2 second intervals. I think it's important to clarify at the outset what a CDR should be. The most fundamental requirement for CDRs is that they accurately record the following pieces of information for EVERY call entering or leaving the system (note every means every and not; channel calls but not peer calls). 1. Destination (aka as A Number) 2. AccountCode (aka as B Number) 3. Call Start Time (answer time), 4. Duration. Of course adding extra information can be very useful and I'm not proposing any fields be removed from the current implementation (although for pity's sake one change that should be made it to use a GUID/UUID for the CDR's uniqueid and save endless confusion). People that really do need verbose or enhanced CDRs to do things like tracking a call's flow as it travels in and out of queues, parking lots etc. would be better off using AMI or the new CEL and not CDRs. At the very least if problems arise with their call flow tracking they will still be able to rely on the accuracy of the CDRs to piece it altogether to work out what's going wrong. My proposal of creating a 1-to-1 relationship between CDRs and Asterisk channels already exsits but somewhere along the line it's going awry. As an experiment, and to actually do something instead of continually moaning about it, I started commenting out the blocks of code in res_featrures.c and sip_channel.c that muck around with the channel CDRs when a transfer occurs. The results of that were that the CDRs for blind and attended transfers actually got better! They're still not quite right but are pretty close with only one CDR on each having a wrong destination. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN Cause codes
Hi Robert all, Maybe someone else can speak to using Progress(), but I don't know if it is required or not. On our system, we didn't need it, and these settings below (plus a call to the telco to tell them to turn on operator messages, don't eat them) did the trick. Good luck, Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Boardman Sent: Saturday, November 22, 2008 11:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ISDN Cause codes I have found that the messages are not played as the hangup cause clears down the channel and passed hangup to the other end should I have progress() before the dial command? Robb Martin Smith wrote: Hi Robert, I'd recommend the following options for Dial() so that you corroborate operator messages w/ cause codes: 1. remove R and r - we've found this can supress operator recordings on early audio 2. likewise, remove m to disable MOH Also, check the values of DIALSTATUS to compare to HANGUPCAUSE. Good luck, Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Boardman Sent: Friday, November 21, 2008 3:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ISDN Cause codes Thanks for the reply Could you be a little more specific? Thanks Robb Martin Smith wrote: Hi Robert, I'd suggest tweaking the Dial() arguments so that you (1) allow early audio, (2) don't force it play ringing to the calling party, and (3) modify any other options to be as relaxed as possible. if you make those changes, you'll start hearing the operator message recordings and those are sometimes easier to reference against the cause codes. Cheers, Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Boardman Sent: Thursday, November 20, 2008 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ISDN Cause codes Hi All Just been looking at stats for one of my sites, and I'm conserned about the number of error cause codes being returned from the telco for example 12000 calls processed 131 are cause code 31* normal. unspecified.* 139 are cause code 28 * invalid number format (address incomplete).* 112 are cause code 1 *Unallocated (unassigned) number. *this adds up to about 3% of calls not completing. there are various other codes including 17 busy 34 channel unavaliable and 44 requested channel unavaliable, which add up to another 1%.* * the telco says there is no problem with the line, I'm trying to understand what the problem could be now alot of calls complete OK so I don't think is my configs Any advice would be appriciated Versions asterisk 1.4.21.1 zaptel 1.4.12.1 Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and
Re: [asterisk-users] database queries from extensions.conf
On Sun, 2008-11-23 at 00:42 -0500, Al Baker wrote: func_odb only allows a SINGLE database statement Ergo you cannot do Transactions or Multi-statement SQL It's my understanding that one of the Digium developers is working on adding transaction support to func_odbc. Digium should support and ADD to this rather than non putting a SINGLE mention of it in the last book and making no mention of it at Astricon. I'm not sure to which book you're referring, as I'm not aware of any book that Digium has published. If you're referring to the O'Reilly book, then the blame lands squarely on my shoulders and the shoulders of the other two authors. But just to be clear -- the O'Reilly book was done completely independently of Digium, and both editions were written before I was ever hired by Digium. In short, please don't blame Digium for my own personal shortcomings :-) As for AstriCon, please remember that the speakers themselves set their topics, not Digium. In my own case, I was asked to fill in for another speaker who couldn't make it to the conference, and I chose my own topic. I'm pretty sure that I mentioned in my presentation on func_odbc that it's not the only way (or even the best way) to query a relational database from the dialplan, it just happens to be my preferred method. (I'd be happy to articulate offline why I feel that way, but I don't think this is the proper venue for me to do that) And remember, I don't claim to speak for Digium or anyone else in this regard. I'm speaking as to my *personal* preferences here. -Jared Smith ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP voicemail with Exchange (was: A way to run extenrnotify when IMAP events take place...)
Hi Noah, Yes, there is a way with Exchange 2003 to use a master user. After doing lots of IMAP hacking and testing on Exchange 2003, I found that there IS A WAY!!! I am using Asterisk 1.6.1-Beta2, but this should also work in 1.4.x as it is Exchange specific, not Asterisk specific. I'm sure this is the long awaited for secret that many IT Professionals have been looking for and here is how it works... In your voicemail.conf: ext_num = vm_pass,user_name,user_email,user_pager_email|imapuser=domain.com\admin_ user_name\mailbox_name|imappassword=apmin_user_password The admin username is just the username, and the mailbox name is just the prefix (before the @ symbol) of the e-mail address. Example: 1688 = 2604,1688,[EMAIL PROTECTED],,tz=central|imapuser=domain.com\vmadmin\user| imappassword=Asterisk123 It works for me, let me know if it works for the rest of you!!! Thanks, Jeff Phelps IT Support Specialist Hi Jeff - I have IMAP voicemail working with Exchange 2003 using a single username and password for multiple mailboxes. Sorry to hijack this thread (at least I changed the Subject), but this really caught my eye. I was under the impression that Exchange's IMAP doesn't have the master user feature and therefore can't do single username authentication for multiple mailboxes. Care to share how you accomplished this? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users from my recent research Exchange2003 does have a master user that can be given write access to all mailboxes. Exchange2007, though removes the MasterUser capability. * Asterisk/Exchange Voicemail http://blog.lithiumblue.com/2007/07/asterixexchange-voicemail.html * Asterisk 1.6.0 + Exchange 2007 SP1 Unified Messaging http://blogs.technet.com/gclark/archive/2008/10/22/asterisk-1-6-0-excha nge-2007-sp1-unified-messaging.aspx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText and non-ASCII characters
Olivier schrieb: Is is possible to translate non-english text into ASCII text It is. Unicode decomposition (NFD or NFKD) is what you're looking for. Many programming languages can do that out of the box or there are extensions or libraries available. so that SIP phones would correctly display non-ASCII characters received from SendText() ? I think SIP MESSAGE (rfc3428) on which SendText() currently relies, defines text/plain Content-type but googling, I can't find a source describing what text/plain can or cannot be. You could try to add a charset attribute like so: Content-Type: text/plain; charset=utf-8 but it's unlikely that any phones pay attention. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP voicemail with Exchange (was: A way torun extenrnotify when IMAP events take place...)
BTW... I have only tested this on Exchange 2003, I have not yet had the chance to check it out on Exchange 2007, but I'm guessing that it works... I will update when I know... Thanks, Jeff Phelps IT Support Specialist Hi Noah, Yes, there is a way with Exchange 2003 to use a master user. After doing lots of IMAP hacking and testing on Exchange 2003, I found that there IS A WAY!!! I am using Asterisk 1.6.1-Beta2, but this should also work in 1.4.x as it is Exchange specific, not Asterisk specific. I'm sure this is the long awaited for secret that many IT Professionals have been looking for and here is how it works... In your voicemail.conf: ext_num = vm_pass,user_name,user_email,user_pager_email|imapuser=domain.com\admin_ user_name\mailbox_name|imappassword=apmin_user_password The admin username is just the username, and the mailbox name is just the prefix (before the @ symbol) of the e-mail address. Example: 1688 = 2604,1688,[EMAIL PROTECTED],,tz=central|imapuser=domain.com\vmadmin\user| imappassword=Asterisk123 It works for me, let me know if it works for the rest of you!!! Thanks, Jeff Phelps IT Support Specialist Hi Jeff - I have IMAP voicemail working with Exchange 2003 using a single username and password for multiple mailboxes. Sorry to hijack this thread (at least I changed the Subject), but this really caught my eye. I was under the impression that Exchange's IMAP doesn't have the master user feature and therefore can't do single username authentication for multiple mailboxes. Care to share how you accomplished this? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users from my recent research Exchange2003 does have a master user that can be given write access to all mailboxes. Exchange2007, though removes the MasterUser capability. * Asterisk/Exchange Voicemail http://blog.lithiumblue.com/2007/07/asterixexchange-voicemail.html * Asterisk 1.6.0 + Exchange 2007 SP1 Unified Messaging http://blogs.technet.com/gclark/archive/2008/10/22/asterisk-1-6-0-excha nge-2007-sp1-unified-messaging.aspx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText and non-ASCII characters
Philipp Kempgen schrieb: Olivier schrieb: Is is possible to translate non-english text into ASCII text It is. Unicode decomposition (NFD or NFKD) is what you're looking for. Forgot to add some pointers. http://en.wikipedia.org/wiki/Unicode_normalization http://www.unicode.org/unicode/faq/normalization.html Many programming languages can do that out of the box or there are extensions or libraries available. http://www.php.net/manual/en/book.unicode.php http://www.php.net/manual/en/book.recode.php http://www.php.net/manual/en/book.iconv.php http://www.icu-project.org/ ... Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText and non-ASCII characters
Philipp Kempgen schrieb: Olivier schrieb: Is is possible to translate non-english text into ASCII text It is. Unicode decomposition (NFD or NFKD) is what you're looking for. Many programming languages can do that out of the box or there are extensions or libraries available. https://svn.amooma.com/gemeinschaft/trunk/opt/gemeinschaft/lib/utf8-normalize/ If your non-english text is in UTF-8 encoding the gs_utf8_decompose_to_ascii() function in gs_utf_normal.php does what you need. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6, IMAP Voicemail and externnotify
I too am looking for a way to get the externnotify= script to run on poll events. Right now, I have a script that runs as a cron job every 60 seconds, but with 150 voicemail boxes, I constantly have at least 40 or 50 instances of the script running at a time because it takes so long to run it through all the mailboxes... Thanks, Jeff Phelps IT Support Specialist McConnell Jones Lanier and Murphy, LLP 3040 Post Oak Blvd., Suite 1600, Houston, TX 77056 (713) 968-1600 (phone) (713) 968-1688 (direct phone) (713) 968-1601 (main fax) http://www.mjlm.com/ IRS Circular 230 Disclosure: To ensure compliance with requirements imposed by the IRS, McConnell Jones, LLP informs you that any U.S. federal tax advice contained in this communication (including any attachments, enclosures, or other accompanying material) is not intended or written to be used, and cannot be used, for the purpose of (i) avoiding penalties under the Internal Revenue Code or (ii) promoting, marketing, or recommending to another party any transaction or matter addressed herein; for IRS audit, tax disputes or other purposes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry L. Kline Sent: Sunday, 23 November, 2008 14:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.6, IMAP Voicemail and externnotify -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have Asterisk sitting between the PSTN and a legacy PBX. Asterisk is doing some IVR work prior to forwarding calls to the PBX and it also acts as the voice mail server for the PBX, with Asterisk configured for IMAP storage. When a call comes in and the caller leaves a voice mail, the VoiceMail application calls the program configured in voicemail.conf (externnotify=). I use that program to create a call file which then turns the MWI on the PBX's phones on or off. Turning the MWI on is fine when voicemail is left and turning the MWI off works great when the user checks his/her voicemail using the handset. My problem is that I want the MWI to be turned off is the user checks his voicemail via an email client. I'm aware of the new IMAP polling* parameters in voicemail.conf, and I have them set. It has become apparent to me that the only time the externnotify script is called is when the VoiceMail[Main] application is accessed. It appears that the script is not called when Asterisk polls the IMAP server to check voicemail. Is that correct? Thanks. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJKbz0CFu3bIiwtTARAlIIAJ9MIcoB53xzW/R7/1BJfe6P3PmsLACfUILL 5x61VCRvoFcPuQudQlt+Qlg= =7KfO -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] database queries from extensions.conf
On Sun, 2008-11-23 at 00:47 -0500, Al Baker wrote: Quote The preferred method is to use func_odbc, which takes SQL queries and builds custom dialplan functions from them. I've used it quite a bit, and am very happy with it. How can you be VERY HappY with something that allows ONLY single statemts of SQL My intention here is not to start a flamewar over which one is *best*, or worse to start arguing about who is right instead of what is right. You're absolutely correct in your assertion that func_odbc doesn't currently support multi-statement or transactional statements, which is obviously a limitation to some people. As I pointed out in my other response to this thread this morning, Tilghman Lesher is working on that. Feel free to look at his odbc_tx_support branch on the web at http://svn.digium.com/view/asterisk/team/tilghman/odbc_tx_support/, or to check it out via Subversion at http://svn.digium.com/svn/asterisk/team/tilghman/odbc_tx_support/ One other way of working around the problem is to use stored procedures in the database. That being said, I guess I'll articulate my own personal reasons for preferring func_odbc, and leave it at that. 1) I like that my dialplan isn't tied to one particular database. I've done a *lot* of database work in my short career, including being a sysadmin for one of the largest MySQL database installations in the world. I *love* the fact that the ODBC abstraction layer means I can easily change my backend database from MySQL to PostgreSQL (or Oracle or SQL Server, heaven forbid!) at the drop of a hat. I realize that might not be a big attraction for some, but for me it's a big plus. 2) I don't like the licensing mess associated with linking MySQL directly to Asterisk. I'm sure there are a few people on the list that really enjoy the convoluted logic of tip-toeing the licensing minefield of linking (dual-licensed) Asterisk with (dial-licensed) MySQL, but I prefer to avoid the minefield altogether and use ODBC. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip trunking and call transfer
On Mon, Nov 24, 2008 at 12:14 AM, Raj Jain [EMAIL PROTECTED] wrote: Yes, by using the SIP REFER method. Caller 2 will send a SIP REFER to Caller 1 asking it to talk to Caller 3. This will cause Caller 1 to drop it's session with Caller 2 and send a new INVITE to Caller 3. So, this is how you do it from a SIP protocol perspective. I'm not sure to what extent Asterisk supports this capability. -- Raj Jain ok, thanks for your reply! I'll search about Asterisk SIP referer implementation. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6, IMAP Voicemail and externnotify
On Mon, Nov 24, 2008 at 10:28 AM, Jeffrey Phelps [EMAIL PROTECTED] wrote: I too am looking for a way to get the externnotify= script to run on poll events. Right now, I have a script that runs as a cron job every 60 seconds, but with 150 voicemail boxes, I constantly have at least 40 or 50 instances of the script running at a time because it takes so long to run it through all the mailboxes... Thanks, Jeff Phelps IT Support Specialist McConnell Jones Lanier and Murphy, LLP 3040 Post Oak Blvd., Suite 1600, Houston, TX 77056 (713) 968-1600 (phone) (713) 968-1688 (direct phone) (713) 968-1601 (main fax) http://www.mjlm.com/ IRS Circular 230 Disclosure: To ensure compliance with requirements imposed by the IRS, McConnell Jones, LLP informs you that any U.S. federal tax advice contained in this communication (including any attachments, enclosures, or other accompanying material) is not intended or written to be used, and cannot be used, for the purpose of (i) avoiding penalties under the Internal Revenue Code or (ii) promoting, marketing, or recommending to another party any transaction or matter addressed herein; for IRS audit, tax disputes or other purposes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry L. Kline Sent: Sunday, 23 November, 2008 14:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.6, IMAP Voicemail and externnotify -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have Asterisk sitting between the PSTN and a legacy PBX. Asterisk is doing some IVR work prior to forwarding calls to the PBX and it also acts as the voice mail server for the PBX, with Asterisk configured for IMAP storage. When a call comes in and the caller leaves a voice mail, the VoiceMail application calls the program configured in voicemail.conf (externnotify=). I use that program to create a call file which then turns the MWI on the PBX's phones on or off. Turning the MWI on is fine when voicemail is left and turning the MWI off works great when the user checks his/her voicemail using the handset. My problem is that I want the MWI to be turned off is the user checks his voicemail via an email client. I'm aware of the new IMAP polling* parameters in voicemail.conf, and I have them set. It has become apparent to me that the only time the externnotify script is called is when the VoiceMail[Main] application is accessed. It appears that the script is not called when Asterisk polls the IMAP server to check voicemail. Is that correct? Thanks. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJKbz0CFu3bIiwtTARAlIIAJ9MIcoB53xzW/R7/1BJfe6P3PmsLACfUILL 5x61VCRvoFcPuQudQlt+Qlg= =7KfO -END PGP SIGNATURE- Can someone share or at least point me in the direction of these scripts? Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6, IMAP Voicemail and externnotify
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jeffrey Phelps wrote: I too am looking for a way to get the externnotify= script to run on poll events. Right now, I have a script that runs as a cron job every 60 seconds, but with 150 voicemail boxes, I constantly have at least 40 or 50 instances of the script running at a time because it takes so long to run it through all the mailboxes... I'm going to do some investigation with the AMI to see if it throws any interesting information up regarding email status. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJKs6SCFu3bIiwtTARAnpNAKCKTwSqt4c5cDVSwXC1DK+sHvs2zwCeKGkl d9FG/zypUC8uijoMjliQRlY= =XNQ3 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play sound while executing agi script
On Mon, 24 Nov 2008, Tilghman Lesher wrote: On Monday 24 November 2008 06:51:23 Giedrius Augys wrote: Is it possible to do like this: play a sound file (if needed play in loop) while php agi script finishes work ? And how to do this? When on my server is huge load , I don't want that client hears silent , but hears music. Thanks Please see StartMusicOnHold and StopMusicOnHold. You can create a thread in your AGI to STREAM FILE. I do this to play a message while waiting for a credit card authorization. Usually, I get the response before the message finishes so the auth appears to be instantaneous to the customer. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Design
It is my belief that before redesigning the CDR engine some time should be spent looking at current PSTN CDR formats and what information is kept in them. The main problem that I feel we face is that calls can be complicated, but we want the record of it to not be. In reality a CDR that incorporates all information about a call would have at least two dimensions. In the first you would have the base call record as we do now, in the second we would have the event list. The event list would be a time indexed list of event names and attributes, just as you would currently store event information. The event list would be your glue (with a bit of front end logic of course.) that would relate a call that dialed X external numbers to the X different new CDR's that generated. That would allow you all the call path info you could ever want. The most important thing would be a new config file that allows an administrator granular control over what information is important to them. And of course a keep it simple stupid mode that just writes the top level cdr as it does now. [EMAIL PROTECTED] wrote: I think that the custom cdr back-end can be successfully used to maximize or minimize the CDRs detailing on a per-needs basis. Furthermore, the CDR() function gives plenty of room for even more detailing. In my opinion the detail level (fields) is not the issue with the CDRs generation nor is the lack of backends (asterisk gives a lot of different backends to store your CDRs). I find the issue with asterisk CDRs to be in the lack of proper CDRs generation for the B-leg of every call. If we want to really track what happens during a call through the CDRs one has to have all the details not only for the incoming channel but for the outgoing one as well. Furthermore, one needs to be able to tweak the B-leg CDRs like he does with the incoming legs. So what needs to be done in my opinion is record every B-leg CDR when such an event occurs and give the user access to the CDR info by extending the CDR() function (so that one can specify the channel of the CDR that is being tweaked) or creating a seperate one for the outgoing channels. Grey Man wrote: I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal (link ommitted to see if email gets accepted). After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation that is already overly so. I think it's a mistake to try and think about all the different call scenarios and come up with little tricks for the more complicated ones. There will always be something missed; app_shotgun initiates calls to 100 random numbers and as soon as three or more calls are answered it will start randonly transferring them amongst each other at 2 second intervals. I think it's important to clarify at the outset what a CDR should be. The most fundamental requirement for CDRs is that they accurately record the following pieces of information for EVERY call entering or leaving the system (note every means every and not; channel calls but not peer calls). 1. Destination (aka as A Number) 2. AccountCode (aka as B Number) 3. Call Start Time (answer time), 4. Duration. Of course adding extra information can be very useful and I'm not proposing any fields be removed from the current implementation (although for pity's sake one change that should be made it to use a GUID/UUID for the CDR's uniqueid and save endless confusion). People that really do need verbose or enhanced CDRs to do things like tracking a call's flow as it travels in and out of queues, parking lots etc. would be better off using AMI or the new CEL and not CDRs. At the very least if problems arise with their call flow tracking they will still be able to rely on the accuracy of the CDRs to piece it altogether to work out what's going wrong. My proposal of creating a 1-to-1 relationship between CDRs and Asterisk channels already exsits but somewhere along the line it's going awry. As an experiment, and to actually do something instead of continually moaning about it, I started commenting out the blocks of code in res_featrures.c and sip_channel.c that muck around with the channel CDRs when a transfer occurs. The results of that were that the CDRs for blind and attended transfers actually got better! They're still not quite right but are pretty close with only one CDR on each having a wrong destination. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth
Re: [asterisk-users] SendText and non-ASCII characters
Hi, At the moment, I'm trying to send Unicoded text to a SIP phone using dialplan application SendText. SendText(Hello World) works. How can I insert letter 00E9 (from http://www.unicode.org/charts/PDF/U0080.pdf) which can be written eacute; in HTML ? regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RFC 3428 (was: Re: SendText and non-ASCII characters)
Philipp Kempgen schrieb: Olivier schrieb: I think SIP MESSAGE (rfc3428) on which SendText() currently relies, defines text/plain Content-type but googling, I can't find a source describing what text/plain can or cannot be. You could try to add a charset attribute like so: Content-Type: text/plain; charset=utf-8 but it's unlikely that any phones pay attention. And BTW that's why RFCs shouldn't be written by people who have never left their limited 7-bit ASCII world. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH Realtime Problem
Hi, 1) MOH in realtime is not working, I have configured it but never go to look at the database, no warning or error found and I can do a query using realtime and the family from the cli. I didn't tested about MOH in realtime, but could you please share your table structure to me? I'd like to see if it works. Thanks Nguyễn Đình Trung --- QiS Technologies, ltd. Tel: 0168 528 7522 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC 3428 (was: Re: SendText and non-ASCII characters)
2008/11/24 Philipp Kempgen [EMAIL PROTECTED] Philipp Kempgen schrieb: Olivier schrieb: I think SIP MESSAGE (rfc3428) on which SendText() currently relies, defines text/plain Content-type but googling, I can't find a source describing what text/plain can or cannot be. You could try to add a charset attribute like so: Content-Type: text/plain; charset=utf-8 but it's unlikely that any phones pay attention. And BTW that's why RFCs shouldn't be written by people who have never left their limited 7-bit ASCII world. So text/plain means anything that can be written in UTF-8 or do you other charsets are allowed ? Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText and non-ASCII characters
Olivier schrieb: At the moment, I'm trying to send Unicoded text Unicode is not an encoding. It's just a list or table of characters (glyphs). http://en.wikipedia.org/wiki/Unicode Unicode is typically represented in encodings (misleadingly called charsets) such as UTF-8, UTF-16 ... http://en.wikipedia.org/wiki/UTF-8 http://en.wikipedia.org/wiki/UTF-16 to a SIP phone using dialplan application SendText. SendText(Hello World) works. How can I insert letter 00E9 (from http://www.unicode.org/charts/PDF/U0080.pdf) which can be written eacute; in HTML ? Interesting. Maybe an Asterisk developer can comment on that. I'd try to type the character (latin small letter e with acute) in the text editor of your choice and either save the file in ISO-8859-1 encoding or in UTF-8 encoding so when viewed in a hexdump (hd) it has 2 bytes: C3 A9 http://www.utf8-chartable.de/unicode-utf8-table.pl?start=233 But I thought you were trying to avoid non-english characters because the phone doesn't display them anyway. If that's what you want then just send one of the decompositioned forms, namely e´ or just e (easy to type). Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RFC 3428
Olivier schrieb: 2008/11/24 Philipp Kempgen [EMAIL PROTECTED] Philipp Kempgen schrieb: Olivier schrieb: I think SIP MESSAGE (rfc3428) on which SendText() currently relies, defines text/plain Content-type but googling, I can't find a source describing what text/plain can or cannot be. You could try to add a charset attribute like so: Content-Type: text/plain; charset=utf-8 but it's unlikely that any phones pay attention. And BTW that's why RFCs shouldn't be written by people who have never left their limited 7-bit ASCII world. So text/plain means anything that can be written in UTF-8 or do you other charsets are allowed ? The text/plain MIME type basically just says This is plain text without any markup, processing instructions etc. http://en.wikipedia.org/wiki/Plain_text http://en.wikipedia.org/wiki/Internet_media_type http://www.iana.org/assignments/media-types/text/ It's defined in http://tools.ietf.org/html/rfc2046#section-4.1.3 text/plain does not say anything about the character encoding. ; charset=us-ascii is (sort of) the default if not specified but you are free to send UTF-8-encoded plain text if declared as ; charset=utf-8. Of course that would require you to modify the source code of app_sendtext.c. But as RFC 3428 doesn't talk about charsets nobody is required to support the charset you send. :-/ Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pick up IAX2 calls
Bruno Castelo Branco wrote: Somebody knows if pickup call works with IAX2? I enable *8 in features.conf, but doesn't works with IAX2 extensions. As I understand it, IAX2 does not support callgroup= and pickupgroup= and *8. This link might be helpful: http://www.freepbx.org/trac/ticket/1568 -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pick up IAX2 calls
On Mon, Nov 24, 2008 at 12:12 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Bruno Castelo Branco wrote: Somebody knows if pickup call works with IAX2? I enable *8 in features.conf, but doesn't works with IAX2 extensions. As I understand it, IAX2 does not support callgroup= and pickupgroup= and *8. This link might be helpful: http://www.freepbx.org/trac/ticket/1568 -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html Ditch IAX2 and go to SIP if at all possible, and where there is a will there is a way -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Design
In order to avoid a multidimensional schema we could have 1 cdr per call leg. So , for instance, one call that had 3 different dial() commands as outgoing attempts would be described by 4 CDRs (1 for the incoming leg that has all the originating channel data and 3 for the outgoing legs that hold all the terminating channel's data). Those CDRs would be bound by a unique identifier field (the same for all 4). The terminating CDRs could be also identified by a increment field that indicates the order that the channels were called. Another issue is that failed attempts should also be logged because this is valuable info for many (or at least have the option to choose the desired behavior - which is available in asterisk as we speak). Anthony Francis wrote: It is my belief that before redesigning the CDR engine some time should be spent looking at current PSTN CDR formats and what information is kept in them. The main problem that I feel we face is that calls can be complicated, but we want the record of it to not be. In reality a CDR that incorporates all information about a call would have at least two dimensions. In the first you would have the base call record as we do now, in the second we would have the event list. The event list would be a time indexed list of event names and attributes, just as you would currently store event information. The event list would be your glue (with a bit of front end logic of course.) that would relate a call that dialed X external numbers to the X different new CDR's that generated. That would allow you all the call path info you could ever want. The most important thing would be a new config file that allows an administrator granular control over what information is important to them. And of course a keep it simple stupid mode that just writes the top level cdr as it does now. [EMAIL PROTECTED] wrote: I think that the custom cdr back-end can be successfully used to maximize or minimize the CDRs detailing on a per-needs basis. Furthermore, the CDR() function gives plenty of room for even more detailing. In my opinion the detail level (fields) is not the issue with the CDRs generation nor is the lack of backends (asterisk gives a lot of different backends to store your CDRs). I find the issue with asterisk CDRs to be in the lack of proper CDRs generation for the B-leg of every call. If we want to really track what happens during a call through the CDRs one has to have all the details not only for the incoming channel but for the outgoing one as well. Furthermore, one needs to be able to tweak the B-leg CDRs like he does with the incoming legs. So what needs to be done in my opinion is record every B-leg CDR when such an event occurs and give the user access to the CDR info by extending the CDR() function (so that one can specify the channel of the CDR that is being tweaked) or creating a seperate one for the outgoing channels. Grey Man wrote: I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal (link ommitted to see if email gets accepted). After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation that is already overly so. I think it's a mistake to try and think about all the different call scenarios and come up with little tricks for the more complicated ones. There will always be something missed; app_shotgun initiates calls to 100 random numbers and as soon as three or more calls are answered it will start randonly transferring them amongst each other at 2 second intervals. I think it's important to clarify at the outset what a CDR should be. The most fundamental requirement for CDRs is that they accurately record the following pieces of information for EVERY call entering or leaving the system (note every means every and not; channel calls but not peer calls). 1. Destination (aka as A Number) 2. AccountCode (aka as B Number) 3. Call Start Time (answer time), 4. Duration. Of course adding extra information can be very useful and I'm not proposing any fields be removed from the current implementation (although for pity's sake one change that should be made it to use a GUID/UUID for the CDR's uniqueid and save endless confusion). People that really do need verbose or enhanced CDRs to do things like tracking a call's flow as it travels in and out of queues, parking lots etc. would be better off using AMI or the new CEL and not CDRs. At the very least if problems arise with their call flow tracking they will still be able to rely on the accuracy of the CDRs to piece it altogether to work out what's going wrong. My proposal of creating a 1-to-1 relationship between CDRs and Asterisk channels already exsits but somewhere along the line
Re: [asterisk-users] HPEC performance
On Wed, Nov 19, 2008 at 03:46:35PM -0700, Joseph L. Casale wrote: Not trivial but not as voodoo as before: http://docs.tzafrir.org.il/dahdi-linux/#_oslec Tzafrir, I pulled down linux-2.6.28-rc5.tar.bz2 and followed the doc, now when compiling I get the following: WARNING: oslec_create [/.../dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.ko] undefined! WARNING: oslec_free [/.../dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.ko] undefined! WARNING: oslec_update [/.../dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.ko] undefined! It seems that the hack that I thought that worked didn't actually work. I think that for the moment just copy the echo directory below dahdi and fix the Kbuild file. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendImage()
On Sun, Nov 23, 2008 at 10:00 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Saturday 22 November 2008 22:18:05 Rob Hillis wrote: Philipp Kempgen wrote: SendImage() in 1.4: ---cut--- SendImage(filename): Sends an image on a channel. If the channel supports image transport but the image send fails, the channel will be hung up. Otherwise, the dialplan continues execution. The option string may contain the following character: 'j' -- jump to priority n+101 if the channel doesn't support image transport This application sets the following channel variable upon completion: SENDIMAGESTATUSThe status is the result of the attempt as a text string, one of OK | NOSUPPORT ---cut--- in 1.6: ---cut--- SendImage(filename): Sends an image on a channel. Result of transmission will be stored in SENDIMAGESTATUS channel variable: SUCCESS Transmission succeeded FAILURE Transmission failed UNSUPPORTED Image transmission not supported by channel ---cut--- Is there any reason to break backwards compatibility? Why is SUCCESS better than OK and UNSUPPORTED better than NOSUPPORT? IMHO there was no need to change anything except for adding the FAILURE return status. This is a case of damned if you do, damned if you don't. That is a perfect complaint, and I understand it completely. On the other side, we are criticized for inconsistent behavior, inconsistent status names, etc. So we've chosen to make Asterisk more consistent going forward, with the one-time problem of a slight change in behavior. Current users see an issue either way, and future users won't see a problem at all. Perhaps somebody from -dev team can be delegated to check naming consistency of new features? So, whenever a feature is added (perhaps at code review), he checks naming to match best of he's opinion. I know that original developers might be stubborn to keep their own names, however that leads to inconsistencies and such changes later. So, if one person is responsible of that, even if change is insignificant, nobody should be offended.. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 mysql cdr log problem
On Sun, Nov 23, 2008 at 2:19 PM, Artifex Maximus [EMAIL PROTECTED] wrote: Hi all! I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers and tools but my calls aren't logged. I'd enabled mysql log and noticed that asterisk send a 'DESC cdr' so connection is working between asterisk and mysql and I am able to call other phones so Asterisk is working as well. No error messages on startup though. Any idea why is it happen? As I realized there is some differences between 1.2 (my previous system) and 1.6 log system. You should also check Asterisk log for warnings. 1.6 should detect table structure and warn about missing fields. If it's so, perhaps you can change asterisk - mysql (res_cdr_addon_mysql if i remember correctly) to do an alter on your table - then it will automagically create missing fields. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] reducing iax packet size
Dear, is any way to change the iax packets? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Log level of 500 Server Internal Error.
On Fri, Nov 21, 2008 at 7:48 PM, Alex Balashov [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Fri, Nov 21, 2008 at 7:32 PM, Alex Balashov [EMAIL PROTECTED] wrote: Atis Lezdins wrote: Hi, VERBOSE[6120] logger.c: -- Got SIP response 500 Server Internal Error I just noticed that i sometimes get those back from provider. They are currently general SIP message log entries with verbose level 3. I wonder if such SIP fails could generate at least WARNING in log? Currently i'm checking logs for warnings and errors, so i probably have missed those.. It would be great indication that something is not ok - either outgoing trunk or local phone is bad. That would generate a lot of debate about what sorts of signaling error classes are useful to include in the fixed logs and which aren't. Best thing to do is just to run your own packet capture and grep for things of interest to you. Yes, that's what i would like to start. If a call fails, i think it's reasonable enough to log a warning message. If i haven't seen this before, how would i know that it's bad and search for it? IMHO it's a good indication for network problem (as was midget packet warning recently) Define fails. There are many different scenarios applicable to many different people's situations, and I doubt Asterisk can be set up to log them all. SIP also has a complicated state machine; sometimes call failures can occur further up the setup flow and not as an immediate failure response. That, I think, is what I was trying to put forth as a possible reason why Asterisk doesn't do what you're asking, which is otherwise a fairly obvious thing to do. Well, of course there are different scenarios. Asterisk shouldn't warn if device sends REGISTER and it replies with UNAUTHORIZED, however it shold warn when device sends wrong authorization. There are lot of cases, perhaps not all can be implemented right now, as some would need complete state information to determine correct/wrong behavior. Of course there are people who do handling of unsuccessful Dial() and send outgoing call trough other provider or incoming - to voicemail. However if SIP device is registered or set as peer, and replays with 500 Internal server error or something similar - that would give pretty much useful info to newbies about what's going wrong. As there's currently no complete way how to react to SIP responses, and DIALSTATUS=CONGESTION isn't much useful. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText and non-ASCII characters
2008/11/24 Philipp Kempgen [EMAIL PROTECTED] Olivier schrieb: At the moment, I'm trying to send Unicoded text Unicode is not an encoding. It's just a list or table of characters (glyphs). http://en.wikipedia.org/wiki/Unicode Unicode is typically represented in encodings (misleadingly called charsets) such as UTF-8, UTF-16 ... http://en.wikipedia.org/wiki/UTF-8 http://en.wikipedia.org/wiki/UTF-16 to a SIP phone using dialplan application SendText. SendText(Hello World) works. How can I insert letter 00E9 (from http://www.unicode.org/charts/PDF/U0080.pdf) which can be written eacute; in HTML ? Interesting. Maybe an Asterisk developer can comment on that. I'd try to type the character (latin small letter e with acute) in the text editor of your choice and either save the file in ISO-8859-1 encoding or in UTF-8 encoding so when viewed in a hexdump (hd) it has 2 bytes: C3 A9 http://www.utf8-chartable.de/unicode-utf8-table.pl?start=233 But I thought you were trying to avoid non-english characters because the phone doesn't display them anyway. Obviously, the phone (Thomson st2030s) displays several latin charsets but the media to use for that is to use SIP MESSAGE. Thanks to your (crystal clear) explaination, I suppose I can't tailor SendText to use UTF-8 encoding so I typed the decompositioned form (ie e´). It doesn't display the way I wanted to. If I could simply use non-ascii in dialplay functions ... I also tried URIENCODE ... If that's what you want then just send one of the decompositioned forms, namely e´ or just e (easy to type). Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] database queries from extensions.conf
For me, the best is the curl function, along with res_config_curl. Best of all worlds - pass a web query to *whatever* backend system you want to implement. No messy ODBC drivers. It's really, really good stuff ;) Julian. Jared Smith wrote: On Sun, 2008-11-23 at 00:47 -0500, Al Baker wrote: Quote The preferred method is to use func_odbc, which takes SQL queries and builds custom dialplan functions from them. I've used it quite a bit, and am very happy with it. How can you be VERY HappY with something that allows ONLY single statemts of SQL My intention here is not to start a flamewar over which one is *best*, or worse to start arguing about who is right instead of what is right. You're absolutely correct in your assertion that func_odbc doesn't currently support multi-statement or transactional statements, which is obviously a limitation to some people. As I pointed out in my other response to this thread this morning, Tilghman Lesher is working on that. Feel free to look at his odbc_tx_support branch on the web at http://svn.digium.com/view/asterisk/team/tilghman/odbc_tx_support/, or to check it out via Subversion at http://svn.digium.com/svn/asterisk/team/tilghman/odbc_tx_support/ One other way of working around the problem is to use stored procedures in the database. That being said, I guess I'll articulate my own personal reasons for preferring func_odbc, and leave it at that. 1) I like that my dialplan isn't tied to one particular database. I've done a *lot* of database work in my short career, including being a sysadmin for one of the largest MySQL database installations in the world. I *love* the fact that the ODBC abstraction layer means I can easily change my backend database from MySQL to PostgreSQL (or Oracle or SQL Server, heaven forbid!) at the drop of a hat. I realize that might not be a big attraction for some, but for me it's a big plus. 2) I don't like the licensing mess associated with linking MySQL directly to Asterisk. I'm sure there are a few people on the list that really enjoy the convoluted logic of tip-toeing the licensing minefield of linking (dual-licensed) Asterisk with (dial-licensed) MySQL, but I prefer to avoid the minefield altogether and use ODBC. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] database queries from extensions.conf
On Mon, Nov 24, 2008 at 8:01 PM, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: For me, the best is the curl function, along with res_config_curl. Best of all worlds - pass a web query to *whatever* backend system you want to implement. No messy ODBC drivers. It's really, really good stuff ;) However you probably can't use it for transactions within call workflow. For example: Customer calls in Start transaction Do query 1 Play prompt A Do query 2 Play prompt B Do query 3 End transaction So, if customer hangs up in middle, you don't execute transaction. That's the thing how it should be done with ODBC or whatever :) Regards, Atis Julian. Jared Smith wrote: On Sun, 2008-11-23 at 00:47 -0500, Al Baker wrote: Quote The preferred method is to use func_odbc, which takes SQL queries and builds custom dialplan functions from them. I've used it quite a bit, and am very happy with it. How can you be VERY HappY with something that allows ONLY single statemts of SQL My intention here is not to start a flamewar over which one is *best*, or worse to start arguing about who is right instead of what is right. You're absolutely correct in your assertion that func_odbc doesn't currently support multi-statement or transactional statements, which is obviously a limitation to some people. As I pointed out in my other response to this thread this morning, Tilghman Lesher is working on that. Feel free to look at his odbc_tx_support branch on the web at http://svn.digium.com/view/asterisk/team/tilghman/odbc_tx_support/, or to check it out via Subversion at http://svn.digium.com/svn/asterisk/team/tilghman/odbc_tx_support/ One other way of working around the problem is to use stored procedures in the database. That being said, I guess I'll articulate my own personal reasons for preferring func_odbc, and leave it at that. 1) I like that my dialplan isn't tied to one particular database. I've done a *lot* of database work in my short career, including being a sysadmin for one of the largest MySQL database installations in the world. I *love* the fact that the ODBC abstraction layer means I can easily change my backend database from MySQL to PostgreSQL (or Oracle or SQL Server, heaven forbid!) at the drop of a hat. I realize that might not be a big attraction for some, but for me it's a big plus. 2) I don't like the licensing mess associated with linking MySQL directly to Asterisk. I'm sure there are a few people on the list that really enjoy the convoluted logic of tip-toeing the licensing minefield of linking (dual-licensed) Asterisk with (dial-licensed) MySQL, but I prefer to avoid the minefield altogether and use ODBC. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] database queries from extensions.conf
Atis Lezdins wrote: On Mon, Nov 24, 2008 at 8:01 PM, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: For me, the best is the curl function, along with res_config_curl. Best of all worlds - pass a web query to *whatever* backend system you want to implement. No messy ODBC drivers. It's really, really good stuff ;) However you probably can't use it for transactions within call workflow. For example: Yeah, you are right. However, I would never want a transaction to span user interaction. Yeuch. Gather, verify, process. Done. Customer Calls in Record inbound call details. Play prompt A Record further details Play prompt B Record further details We tie these three discrete transactions together by a guid. Julian Customer calls in Start transaction Do query 1 Play prompt A Do query 2 Play prompt B Do query 3 End transaction So, if customer hangs up in middle, you don't execute transaction. That's the thing how it should be done with ODBC or whatever :) Regards, Atis Julian. Jared Smith wrote: On Sun, 2008-11-23 at 00:47 -0500, Al Baker wrote: Quote The preferred method is to use func_odbc, which takes SQL queries and builds custom dialplan functions from them. I've used it quite a bit, and am very happy with it. How can you be VERY HappY with something that allows ONLY single statemts of SQL My intention here is not to start a flamewar over which one is *best*, or worse to start arguing about who is right instead of what is right. You're absolutely correct in your assertion that func_odbc doesn't currently support multi-statement or transactional statements, which is obviously a limitation to some people. As I pointed out in my other response to this thread this morning, Tilghman Lesher is working on that. Feel free to look at his odbc_tx_support branch on the web at http://svn.digium.com/view/asterisk/team/tilghman/odbc_tx_support/, or to check it out via Subversion at http://svn.digium.com/svn/asterisk/team/tilghman/odbc_tx_support/ One other way of working around the problem is to use stored procedures in the database. That being said, I guess I'll articulate my own personal reasons for preferring func_odbc, and leave it at that. 1) I like that my dialplan isn't tied to one particular database. I've done a *lot* of database work in my short career, including being a sysadmin for one of the largest MySQL database installations in the world. I *love* the fact that the ODBC abstraction layer means I can easily change my backend database from MySQL to PostgreSQL (or Oracle or SQL Server, heaven forbid!) at the drop of a hat. I realize that might not be a big attraction for some, but for me it's a big plus. 2) I don't like the licensing mess associated with linking MySQL directly to Asterisk. I'm sure there are a few people on the list that really enjoy the convoluted logic of tip-toeing the licensing minefield of linking (dual-licensed) Asterisk with (dial-licensed) MySQL, but I prefer to avoid the minefield altogether and use ODBC. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Desgin
On Sat, 2008-11-22 at 04:02 +, Grey Man wrote: I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation that is already overly so. I think it's a mistake to try and think about all the different call scenarios and come up with little tricks for the more complicated ones. There will always be something missed; app_shotgun initiates calls to 100 random numbers and as soon as three or more calls are answered it will start randonly transferring them amongst each other at 2 second intervals. I think it's important to clarify at the outset what a CDR should be. The most fundamental requirement for CDRs is that they accurately record the following pieces of information for EVERY call entering or leaving the system (note every means every and not; channel calls but not peer calls). 1. Destination (aka as A Number) 2. AccountCode (aka as B Number) 3. Call Start Time (answer time), 4. Duration. Of course adding extra information can be very useful and I'm not proposing any fields be removed from the current implementation (although for pity's sake one change that should be made it to use a GUID/UUID for the CDR's uniqueid and save endless confusion). People that really do need verbose or enhanced CDRs to do things like tracking a call's flow as it travels in and out of queues, parking lots etc. would be better off using AMI or the new CEL and not CDRs. At the very least if problems arise with their call flow tracking they will still be able to rely on the accuracy of the CDRs to piece it altogether to work out what's going wrong. My proposal of creating a 1-to-1 relationship between CDRs and Asterisk channels already exsits but somewhere along the line it's going awry. As an experiment, and to actually do something instead of continually moaning about it, I started commenting out the blocks of code in res_featrures.c and sip_channel.c that muck around with the channel CDRs when a transfer occurs. The results of that were that the CDRs for blind and attended transfers actually got better! They're still not quite right but are pretty close with only one CDR on each having a wrong detstination. Regards, Greyman. Greyman-- For the moment, let's not worry about the implementation. Let's get consensus on the spec first. In the scenario, where A calls B, B xfers A to C, C xfers A to D, or some such similar scenario, half the world wants a single CDR for A, from the time it started, to the time it hung up with D. The other half wants A-B's dial and bridge, a cdr for A C's bridge, a CDR for A D's bridge, and mayhaps some CDRs to describe the xfers, where B xfers A to C and C xfers A to D. My document is pointing in the former direction, and either we need to spec both, or pick one. murf -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendText and non-ASCII characters
Olivier schrieb:
2008/11/24 Philipp Kempgen [EMAIL PROTECTED]
Olivier schrieb:
At the moment, I'm trying to send Unicoded text
Unicode is not an encoding. It's just a list or table of characters
(glyphs).
http://en.wikipedia.org/wiki/Unicode
Unicode is typically represented in encodings (misleadingly called
charsets) such as UTF-8, UTF-16 ...
http://en.wikipedia.org/wiki/UTF-8
http://en.wikipedia.org/wiki/UTF-16
to a SIP phone using
dialplan application SendText.
SendText(Hello World) works.
How can I insert letter 00E9 (from
http://www.unicode.org/charts/PDF/U0080.pdf) which can be written
eacute;
in HTML ?
Interesting. Maybe an Asterisk developer can comment on that.
I'd try to type the character (latin small letter e with acute) in
the text editor of your choice and either save the file in
ISO-8859-1 encoding or in UTF-8 encoding so when viewed in a
hexdump (hd) it has 2 bytes: C3 A9
http://www.utf8-chartable.de/unicode-utf8-table.pl?start=233
But I thought you were trying to avoid non-english characters
because the phone doesn't display them anyway.
Obviously, the phone (Thomson st2030s) displays several latin charsets but
the media to use for that is to use SIP MESSAGE.
Thanks to your (crystal clear) explaination, I suppose I can't tailor
SendText to use UTF-8 encoding so I typed the decompositioned form (ie
e´).
It doesn't display the way I wanted to.
If I could simply use non-ascii in dialplay functions ...
The required modification to add ;charset=UTF-8 to the Content-
Type header is simple and has already been done in Asterisk 1.6.1
(not in 1.6.0). It's in the add_text() function in chan_sip.c:
http://svn.digium.com/view/asterisk/tags/1.4.22/channels/chan_sip.c?view=markup#l_6229
http://svn.digium.com/view/asterisk/tags/1.6.0.1/channels/chan_sip.c?view=markup#l_7747
http://svn.digium.com/view/asterisk/tags/1.6.1-beta1/channels/chan_sip.c?view=markup#l_8022
/*! \brief Add text body to SIP message */
static int add_text(struct sip_request *req, const char *text)
{
/* XXX Convert \n's to \r\n's XXX */
- add_header(req, Content-Type, text/plain);
+ add_header(req, Content-Type, text/plain;charset=UTF-8);
add_header_contentLength(req, strlen(text));
add_line(req, text);
return 0;
}
You could easily make the same modification in 1.4 or 1.6.0.
It may help or it may not. Depends on the phone.
I also tried URIENCODE ...
Not the way to go here.
If that's what you want then just send one of the decompositioned
forms, namely e´ or just e (easy to type).
Philipp Kempgen
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Re: [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities
Yet another option is a commercial system with in-house staff. I used to maintain a NEC (NEAX 2400) for many years. I went to factory training and had total responsibility for it. Some manufacturers discourage or prevent this, but others are open to it. There are also 3rd party organizations (such as Source) that can supply parts and even expertise for those going that direction. Whether the result would be higher availability than Asterisk, I don't know. Given I'm both a telco guy and a computer guru (CS degree) I'd probably go the Asterisk route myself, because its open and I would have more control. Wilton and bug fixes than any commercial product sold in the intra-industrial channel ... and they won't charge you a $30,000 license fee for the upgrade.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities
Fronting with OpenSER or FS, you should have no problems providing you plan to use SIP extensions. What is critical are the max simultaneous trunks you are going to use. I would go TDM although universities have good bandwidth, and SUPERIOR bandwidth between others. I would think a TDM DS3 or two just to be safe. It should be pretty trivial besides gotchas, like cat3 to the rooms, although channel banks may be an even better solution if phones are already in place. Then you just use SIP when needed or wanted, and Asterisk is simple, although more costly. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Fri, Nov 21, 2008 at 6:24 PM, Wilton Helm [EMAIL PROTECTED] wrote: Yet another option is a commercial system with in-house staff. I used to maintain a NEC (NEAX 2400) for many years. I went to factory training and had total responsibility for it. Some manufacturers discourage or prevent this, but others are open to it. There are also 3rd party organizations (such as Source) that can supply parts and even expertise for those going that direction. Whether the result would be higher availability than Asterisk, I don't know. Given I'm both a telco guy and a computer guru (CS degree) I'd probably go the Asterisk route myself, because its open and I would have more control. Wilton and bug fixes than any commercial product sold in the intra-industrial channel ... and they won't charge you a $30,000 license fee for the upgrade. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need Recording Solution in Asterisk
- Original Message - From: Kashif Naeem To: [EMAIL PROTECTED] Sent: Saturday, November 22, 2008 7:08 AM Subject: [asterisk-users] Need Recording Solution in Asterisk Hello All One of our client Bank has 900 employees working in different locations. They need to record all internal and external calls. Can any body suggest Call Recording Solution for this requirement. We need to know the Hardware / Bandwidth and all requirements and costing. Regards, -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Kashif www.bicomsystems.com and pbxware, we often find the customer has unique search requirements and a little custom care adds greatly to the turnkey solution. A few questions that need reply offline though, are: 1. number of concurrent calls est. 2. codecs, or PRI interfacing 3. Any search criteria e.g. timestamp, cli ... to be put into filename of recording 4. Do you need search fields available through an interface for your client ? 5. Will you failover require redundancy or is RAID enough ? Regards Steve 'at~ bicomsystems . c0m___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 mysql cdr log problem
On Monday 24 November 2008 11:38:09 am Atis Lezdins wrote: On Sun, Nov 23, 2008 at 2:19 PM, Artifex Maximus [EMAIL PROTECTED] wrote: I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers and tools but my calls aren't logged. I'd enabled mysql log and noticed that asterisk send a 'DESC cdr' so connection is working between asterisk and mysql and I am able to call other phones so Asterisk is working as well. No error messages on startup though. Any idea why is it happen? As I realized there is some differences between 1.2 (my previous system) and 1.6 log system. I suspect that you have some unique index on the table which is conflicting with the inserted fields. Once you figure out which field is causing the conflict, it should be easier to figure out where the problem actually lies. You should also check Asterisk log for warnings. 1.6 should detect table structure and warn about missing fields. If it's so, perhaps you can change asterisk - mysql (res_cdr_addon_mysql if i remember correctly) to do an alter on your table - then it will automagically create missing fields. You remember incorrectly. None of the CDR drivers currently have the capability to alter tables. What they will do is to adapt to the table structure and insert only the required fields. Only realtime table drivers have the capability of altering tables and then, only if you turn that behavior on. By default, Asterisk does not alter table structures. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pick up IAX2 calls
hi thanks Luis , but doesn't work. For SIP extensions works well *8, but for IAX a tried *8 and ** + iax extension and didn't works Luis Morales wrote: Try with ** + iax extension Regards, Luis Morales On Mon, Nov 24, 2008 at 10:20 PM, Bruno Castelo Branco [EMAIL PROTECTED] wrote: Hi Somebody knows if pickup call works with IAX2? I enable *8 in features.conf, but doesn't works with IAX2 extensions. Any idea? thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 mysql cdr log problem
On Monday 24 November 2008 06:19:44 pm Tilghman Lesher wrote: On Monday 24 November 2008 11:38:09 am Atis Lezdins wrote: On Sun, Nov 23, 2008 at 2:19 PM, Artifex Maximus [EMAIL PROTECTED] wrote: I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers and tools but my calls aren't logged. I'd enabled mysql log and noticed that asterisk send a 'DESC cdr' so connection is working between asterisk and mysql and I am able to call other phones so Asterisk is working as well. No error messages on startup though. Any idea why is it happen? As I realized there is some differences between 1.2 (my previous system) and 1.6 log system. I suspect that you have some unique index on the table which is conflicting with the inserted fields. Once you figure out which field is causing the conflict, it should be easier to figure out where the problem actually lies. BTW, if you 'core set debug 2 cdr_addon_mysql.c' (and make sure debug is enabled to the console, via /etc/asterisk/logger.conf), then the SQL will be printed to the console. That should help you find where the problem lies. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cdr mysql error
Hi, Need help on mysql cdr, i keep on seeing this log on the console. but my db is up and i see the calls being logged on the cdr table. is there a timeout when there is no activity? can i remove the timeout if there is any? thanks [Nov 25 13:22:37] ERROR[21026]: cdr_addon_mysql.c:171 mysql_log: cdr_mysql: Server has gone away. Attempting to reconnect. [Nov 25 14:20:32] ERROR[21061]: cdr_addon_mysql.c:171 mysql_log: cdr_mysql: Server has gone away. Attempting to reconnect. regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cdr mysql error
Increase the timeout in my.cnf in mysql. -Jai Buy unmetered VoIP DIDs www.didforsale.com Free Trail On Mon, Nov 24, 2008 at 11:10 PM, Nhadie [EMAIL PROTECTED] wrote: Hi, Need help on mysql cdr, i keep on seeing this log on the console. but my db is up and i see the calls being logged on the cdr table. is there a timeout when there is no activity? can i remove the timeout if there is any? thanks [Nov 25 13:22:37] ERROR[21026]: cdr_addon_mysql.c:171 mysql_log: cdr_mysql: Server has gone away. Attempting to reconnect. [Nov 25 14:20:32] ERROR[21061]: cdr_addon_mysql.c:171 mysql_log: cdr_mysql: Server has gone away. Attempting to reconnect. regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Desgin
If we only implement A-D cdr we lose information. On the other hand, if we implement all 3 CDRs for one call we can either use this info or ignore it and act like its not there. The first way is prohibiting for some users. The second one can match any scenario with none to little effort. Steve Murphy wrote: On Sat, 2008-11-22 at 04:02 +, Grey Man wrote: I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation that is already overly so. I think it's a mistake to try and think about all the different call scenarios and come up with little tricks for the more complicated ones. There will always be something missed; app_shotgun initiates calls to 100 random numbers and as soon as three or more calls are answered it will start randonly transferring them amongst each other at 2 second intervals. I think it's important to clarify at the outset what a CDR should be. The most fundamental requirement for CDRs is that they accurately record the following pieces of information for EVERY call entering or leaving the system (note every means every and not; channel calls but not peer calls). 1. Destination (aka as A Number) 2. AccountCode (aka as B Number) 3. Call Start Time (answer time), 4. Duration. Of course adding extra information can be very useful and I'm not proposing any fields be removed from the current implementation (although for pity's sake one change that should be made it to use a GUID/UUID for the CDR's uniqueid and save endless confusion). People that really do need verbose or enhanced CDRs to do things like tracking a call's flow as it travels in and out of queues, parking lots etc. would be better off using AMI or the new CEL and not CDRs. At the very least if problems arise with their call flow tracking they will still be able to rely on the accuracy of the CDRs to piece it altogether to work out what's going wrong. My proposal of creating a 1-to-1 relationship between CDRs and Asterisk channels already exsits but somewhere along the line it's going awry. As an experiment, and to actually do something instead of continually moaning about it, I started commenting out the blocks of code in res_featrures.c and sip_channel.c that muck around with the channel CDRs when a transfer occurs. The results of that were that the CDRs for blind and attended transfers actually got better! They're still not quite right but are pretty close with only one CDR on each having a wrong detstination. Regards, Greyman. Greyman-- For the moment, let's not worry about the implementation. Let's get consensus on the spec first. In the scenario, where A calls B, B xfers A to C, C xfers A to D, or some such similar scenario, half the world wants a single CDR for A, from the time it started, to the time it hung up with D. The other half wants A-B's dial and bridge, a cdr for A C's bridge, a CDR for A D's bridge, and mayhaps some CDRs to describe the xfers, where B xfers A to C and C xfers A to D. My document is pointing in the former direction, and either we need to spec both, or pick one. murf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
