Re: [asterisk-users] Polycom and NAT
That should do it, it tells Asterisk to override the contact field which includes the private IP, and use the public IP and port it received the packet from instead. Try a 'sip debug peer peer' and see what it is coming in as. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Wednesday, August 22, 2007 05:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom and NAT I have both of those command lines for my natted sip device. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darryl Dunkin Sent: Wednesday, 22 August 2007 1:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom and NAT In your sip.conf, for the user: nat=yes To send keepalives for the UDP connection (depending on how flimsy the device handling NAT is): qualify=yes From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Tuesday, August 21, 2007 17:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom and NAT Hi All, I have a Polycom 501 that is behind a NAT. When it registers to the Asterisk server it is using the IP address on the private network and not the public IP of the NAT address. Can someone tell me what I need to do so the phone registerers using an internet address rather than the remote network NAT address. ___ --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] ASTCC and IVR
Hi list; ASTCC supports IVR or there is a separate module for IVR? Can someone advise me a link to start download and ready about ASTCC to do the configuration? Regards, - ITS IP Telephony and Contact Center Engineer Bilal Ghayad Mobile: 00865 9849460 Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ --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] contact header is missing in 200OK for SUBSCRIBE
Hi, I am trying to SUBSCRIBE for message waiting indications to asterisk, it sends 200 OK but contact header is missing(it is mandatory since subscribe is dialog establishing method), due to which parsing fails, any body knows about this issue...? Regards, Subramanya ___ --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] B410P and echo
Hi, Where can I find some tools, such as ztmonitor for zaptel devices, to adjust rxgain and txgain correctly on this card? I've some troubles with finding the optimal configuration for the echocancellator. Thanks in advance, Stefano Arata ___ --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] Multiple servers using realtime
That's a good note about MySQL replication. You can use it to remove point-of-failure which currently is your DB server. Regards/Pagarbiai, Mindaugas Kezys http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Collier Sent: Thursday, August 23, 2007 12:21 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Multiple servers using realtime I use a centralized database (with replication) for several servers, and it works very well. I keep all the mysql traffic on a separate network from the SIP traffic. It makes it easy to add capacity. If you are doing all the mySQL on one box anyway, I don?t see any adavantage to using separate databases. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Peder @ NetworkOblivion Enviado el: miercoles, 22 de agosto de 2007 19:06 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] Multiple servers using realtime I am in the process of setting up several * servers using realtime and connecting to mysql. I am trying to figure out if I should just use one database and one set of tables for all of the user data. Or if I should have separate databases for each * box. The boxes are independent of each other in that customerA only connects to box A. They will never fail over to box B or anything like that. I want to use realtime for queues,voicemail, sippeers and extensions. The only issue that I have come up with so far is that a common voicemail table would cause each box to try and send out mwi indicators since it appears each * box pulls all of the voicemail boxes from the DB every 10 seconds. Any thoughts on whether I should go with one DB, or separate per box DB's? There is one mysql box, I am not referring to mysql on each box, I am referring to whether I should use separate DB's within the one mysql box for each * box. Thanks. Peder ___ --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
[asterisk-users] asterisk configurator with E120P E1 card
Dear all I want to configure 2 port E1 card on my asterisk so which version is best 1.2.x or 1.4.x can anyone suggest me which one is best right now for asterisk and anyone have configuration file to configure E1 card and zaptel.conf so i can configure it - Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.___ --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] Cisco firmwares 3.6.3 vs 3.8.6
Thanks for that Arnaud, saw it myself this morning, but the download link takes me to a page not found cisco page :( I've reported it on their broken links page... Adrian Marsh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Arnaud Ligot Sent: 22 August 2007 13:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco firmwares 3.6.3 vs 3.8.6 FYI about cisco firmware: http://www.cisco.com/warp/public/707/cisco-sr-20070821-sip.shtml A. On Wed, 2007-08-22 at 12:26 +0100, Adrian Marsh wrote: Hi All, A question for those with Cisco 7940/60 SIP phones. I used to load POS3-06-03-00 Firmware to the cisco phones. A month or so ago, I ran some tests and found that latest 3.8.6 firmware worked well, and solved an issue or two on the phones. I've a number of users who work outside of the LAN. Our phones use DNS names to talk to A*k, so in theory, just enabling NAT makes the phone work outside the LAN (home users, remote users, etc). However, when we loaded the 3.8.6 firmware to these phones, we've found the phones no longer work outside of the LAN. Using Etherreal, we've found that the communication between the Phone and A*k breaks (A*k never sees the Register packets, but the phone does seem to send them. I'll post more detail if its needed, but I wondered if anyone else has seen this ? The size of the IP packet for register is different (larger on the 3.8.6), but the important content of the Register message seems the same. I've ruled out ISP/firewall interference, as its happened on a number of users. Obviously there are fixes in 3.8.6, so I don't want to downgrade the phones again, but I can't see why they'd fail... Adrian Marsh ___ --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
Re: [asterisk-users] [Serusers] why combine ser with asterisk
Asterisk is an excellent PBX system, and makes a very good endpoint in the SIP chain for all sorts of things -- IVR systems, voicemail applications, automated messages, etc. It has an extremely well-written CDR engine, so many people mesh it with billing applications to produce accurate accounting information. It also is fully aware of the media stream, which means it's capable of cutting off a call mid-stream, injecting audio into the call, etc, etc. Programming for Asterisk addons can be easily done in just about any language, and it meshes well with the overall structure. Programming for SER is... not so simple. As for running them both on the same box, the biggest problem would be resources. Unlike SER, Asterisk is not designed to be a carrier-grade SIP proxy. If you're actually proxying the media stream, you'd be hard-pressed to squeeze more than 150 simultaneous calls out of Asterisk on even the beefiest of hardware. Add SER to the same box, and you will quickly run into resource problems in medium-sized environments. It also doesn't have a lot of the SIP proxy functionality that SER has. If you're careful, you can configure Asterisk NOT to handle the media stream and still use it for prepaid solutions (using astcc or asterisk-b2bua), and this will save you bandwidth (but you'll still likely run into NAT issues that need to be dealt with somehow) and still let you use Asterisk as an in-between point. Together, Asterisk and SER make a very powerful combination for providing a full suite of services to clientele, and each plays well off the other's strengths. N. Nhadie wrote: Hi All, What's the advantage of combining ser with asterisk? I always see comments like using ser with asterisk is a very good solution etc. etc. the thing i liked with ser is that it does not do codec translation, which saves me cpu usage and also bandwidth. if i combine it with asterisk, would it not use codec translation? i also read that there is also a problem when ser and asterisk is run on the same machine, why is it so? if use prepaid billing solution for asterisk like astcc, would i then be able to provide prepaid service? soryy for asking too much, i'd just like to really understand it. Thank You in advanced. Regards, Nhadie ___ Serusers mailing list [EMAIL PROTECTED] http://lists.iptel.org/mailman/listinfo/serusers ___ --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] contact header is missing in 200OK for SUBSCRIBE
Hi, Hi, I am trying to SUBSCRIBE for message waiting indications to asterisk, it sends 200 OK but contact header is missing(it is mandatory since subscribe is dialog establishing method), due to which parsing fails and also expires is 0 in the 200 OK any body knows about these issue...? On 8/23/07, sumanth achar [EMAIL PROTECTED] wrote: Hi, I am trying to SUBSCRIBE for message waiting indications to asterisk, it sends 200 OK but contact header is missing(it is mandatory since subscribe is dialog establishing method), due to which parsing fails, any body knows about this issue...? Regards, Subramanya ___ --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] [Serusers] why combine ser with asterisk
Offers them? Yes. Offers them in a clean, friendly, usable package? Not so much yet. SEMS has raw capability, but if you want it to do many of the things Asterisk can do, you need to know how to code that yourself, or you're going to be digging about the code for documentation on features (since the current docs are not the world's greatest). Don't get me wrong, SEMS has its place, and is a constantly evolving work of art (we use SEMS for several things in our environment), but comparing SEMS to Asterisk is a bit like comparing a bunch of car parts to a Porsche. N. Fredrik Lundmark wrote: I'm still learning myself, but SEMS (iptel.org/sems) seems to offer many of the media- and/or b2bua-functions that Asterisk do. ///Fredrik - Original Message - From: SIP [EMAIL PROTECTED] To: Nhadie [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Thursday, August 23, 2007 1:38 PM Subject: Re: [Serusers] why combine ser with asterisk Asterisk is an excellent PBX system, and makes a very good endpoint in the SIP chain for all sorts of things -- IVR systems, voicemail applications, automated messages, etc. It has an extremely well-written CDR engine, so many people mesh it with billing applications to produce accurate accounting information. It also is fully aware of the media stream, which means it's capable of cutting off a call mid-stream, injecting audio into the call, etc, etc. Programming for Asterisk addons can be easily done in just about any language, and it meshes well with the overall structure. Programming for SER is... not so simple. As for running them both on the same box, the biggest problem would be resources. Unlike SER, Asterisk is not designed to be a carrier-grade SIP proxy. If you're actually proxying the media stream, you'd be hard-pressed to squeeze more than 150 simultaneous calls out of Asterisk on even the beefiest of hardware. Add SER to the same box, and you will quickly run into resource problems in medium-sized environments. It also doesn't have a lot of the SIP proxy functionality that SER has. If you're careful, you can configure Asterisk NOT to handle the media stream and still use it for prepaid solutions (using astcc or asterisk-b2bua), and this will save you bandwidth (but you'll still likely run into NAT issues that need to be dealt with somehow) and still let you use Asterisk as an in-between point. Together, Asterisk and SER make a very powerful combination for providing a full suite of services to clientele, and each plays well off the other's strengths. N. Nhadie wrote: Hi All, What's the advantage of combining ser with asterisk? I always see comments like using ser with asterisk is a very good solution etc. etc. the thing i liked with ser is that it does not do codec translation, which saves me cpu usage and also bandwidth. if i combine it with asterisk, would it not use codec translation? i also read that there is also a problem when ser and asterisk is run on the same machine, why is it so? if use prepaid billing solution for asterisk like astcc, would i then be able to provide prepaid service? soryy for asking too much, i'd just like to really understand it. Thank You in advanced. Regards, Nhadie ___ Serusers mailing list [EMAIL PROTECTED] http://lists.iptel.org/mailman/listinfo/serusers ___ Serusers mailing list [EMAIL PROTECTED] http://lists.iptel.org/mailman/listinfo/serusers ___ --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] unable to load chan_unicall.so
Hi, I am using debian 4.0 with version 2.6.18-4-686 I have downloaded the required files form site asterisk-1.2.24.tar.gz libmfcr2-0.0.3-1.4.tar.bz2 libsupertone-0.0.2.tar.gz libunicall-0.0.3-1.4.tar.bz2 spandsp-20060903.tar.gz I downloaded and installed the files in the follwing sequence spandsp libsupertone libunicall libmfcr2-0.0.3 is giving a lot of definition error I converted .src.rpm file of libmfcr2 to .deb file and installed it. the copying the chn_unicall.c and channels_Makefile.patch to channels subdirectory of asterisk-1.2.24 but when I run ,asterisk -vvgc' on command line it gives error unable to load chan_unicall.so, but it is present in /usr/lib/asterisk/modules. Can anybody tell me how to trobleshoot it. Can anybody tell me what to do how to remove this. Thanka and regards sanchal ___ --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] libmfcr2 is giving definition error while compiling
Hi, I am using debian 4.0 with version 2.6.18-4-686 I have downloaded the required files form site asterisk-1.2.24.tar.gz libmfcr2-0.0.3-1.4.tar.bz2 libsupertone-0.0.2.tar.gz libunicall-0.0.3-1.4.tar.bz2 spandsp-20060903.tar.gz I downloaded and installed the files in the follwing sequence spandsp libsupertone libunicall Till here it is compiling and copying .so library to /usr/local/lib/ libmfcr2-0.0.3 is giving a lot of definition error I converted .src.rpm file of libmfcr2 to .deb file and installed so .so files are not their in /usr/local/lib Can anybody tell me what to do how to remove this. Thanka and regards sanchal ___ --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] libmfcr2 is giving definition error while compiling
Hi, I am using debian 4.0 with version 2.6.18-4-686 I have downloaded the required files form site asterisk-1.2.24.tar.gz libmfcr2-0.0.3-1.4.tar.bz2 libsupertone-0.0.2.tar.gz libunicall-0.0.3-1.4.tar.bz2 spandsp-20060903.tar.gz I downloaded and installed the files in the follwing sequence spandsp libsupertone libunicall Till here it is compiling and copying .so library to /usr/local/lib/ libmfcr2-0.0.3 is giving a lot of definition error I converted .src.rpm file of libmfcr2 to .deb file and installed so .so files are not their in /usr/local/lib With some minor changes the libmfcr2 get compiled successfully but some rpath error was coming. Can anybody tell me what to do how to remove this. Thanka and regards sanchal ___ --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] [PHP-AGI] Problem executing script
Hello,I have succeded in compiling and configuring My TDM Card and asterisk, all works fine. But I have a problem using the PHP Agi.The CLI tells me that when I call my number :-- Starting simple switch on 'Zap/4-1'-- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack-- Executing [EMAIL PROTECTED]:2] AGI(Zap/4-1, rabot.agi) in new stack-- Launched AGI Script /var/lib/asterisk/agi-bin/rabot.agi == rabot.agi: Failed to execute '/var/lib/asterisk/agi-bin/rabot.agi': No such file or directory-- AGI Script rabot.agi completed, returning 0 == Auto fallthrough, channel 'Zap/4-1' status is 'UNKNOWN'-- Hungup 'Zap/4-1'I tought first that it was a problem of chmod so I change the chmod of all the directory agi-bin TO 777But it changed nothing. I have verify that php was well indicate at the beginning of the script :#!/usr/bin/php-qAnd there is a php exec at /usr/binAny ideas about this problem ?Thank for you helpKheraud _ Windows Live Messenger vous offre 30 nouvelles émoticônes gratuites, installées directement dans votre Messenger ! http://www.emoticones-messenger.fr/___ --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 Message Logs
Hello, Is it possible to print the Asterisk message logs to a file, or is this already done? By message logs I mean the display that shows up on the asterisk server when a call is made from one user to another. I believe if the verbosity is high, it can show what parts of the extension.conf file that it uses when making the call. I am trying to use two Jain-sip-applet-phones, connected through an Asterisk server. I can't seem to get communication between the two phones. Does anyone have any experience using these open-source Jain-sip-applet-phones? Thanks, Denis ___ --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] [PHP-AGI] Problem executing script
Hey there, The file must be accessible by the process calling upon it. Try a chown using the Asterisk process user name. Cheers, Louis-Eric At 09:51 AM 8/23/2007, Karim H wrote: Hello, I have succeded in compiling and configuring My TDM Card and asterisk, all works fine. But I have a problem using the PHP Agi. The CLI tells me that when I call my number : -- Starting simple switch on 'Zap/4-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] AGI(Zap/4-1, rabot.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/rabot.agi == rabot.agi: Failed to execute '/var/lib/asterisk/agi-bin/rabot.agi': No such file or directory -- AGI Script rabot.agi completed, returning 0 == Auto fallthrough, channel 'Zap/4-1' status is 'UNKNOWN' -- Hungup 'Zap/4-1' I tought first that it was a problem of chmod so I change the chmod of all the directory agi-bin TO 777 But it changed nothing. I have verify that php was well indicate at the beginning of the script : #!/usr/bin/php-q And there is a php exec at /usr/bin Any ideas about this problem ? Thank for you help Kheraud -- Besoin d'un e-mail ? Créez gratuitement un compte Windows Live Hotmail et bénéficiez de 2 Go de stockage ! http://www.windowslive.fr/hotmail/default.aspWindows Live Hotmail ___ --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] [PHP-AGI] Problem executing script
Did you confirm that the file exists? /var/lib/asterisk/agi-bin/rabot.agi Also, in your script (wherever it actually is), put a space between php and -q #!/usr/bin/php -q Karim H wrote: Hello, I have succeded in compiling and configuring My TDM Card and asterisk, all works fine. But I have a problem using the PHP Agi. The CLI tells me that when I call my number : -- Starting simple switch on 'Zap/4-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] AGI(Zap/4-1, rabot.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/rabot.agi == rabot.agi: Failed to execute '/var/lib/asterisk/agi-bin/rabot.agi': No such file or directory -- AGI Script rabot.agi completed, returning 0 == Auto fallthrough, channel 'Zap/4-1' status is 'UNKNOWN' -- Hungup 'Zap/4-1' I tought first that it was a problem of chmod so I change the chmod of all the directory agi-bin TO 777 But it changed nothing. I have verify that php was well indicate at the beginning of the script : #!/usr/bin/php-q And there is a php exec at /usr/bin Any ideas about this problem ? Thank for you help Kheraud Besoin d'un e-mail ? Créez gratuitement un compte Windows Live Hotmail et bénéficiez de 2 Go de stockage ! Windows Live Hotmail http://www.windowslive.fr/hotmail/default.asp ___ --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] Zaptel 1.2.20.1 and 1.4.5.1 released
Steve Kennedy wrote: On Wed, Aug 22, 2007 at 01:19:14PM -0500, Asterisk Development Team wrote: The Asterisk.org development team has announced the release of Zaptel versions 1.2.20.1 and 1.4.5.1. These releases are to correct an error in the install target in the Makefile of the 1.4.5 and 1.2.20 Zaptel releases, as well as a handful of other issues. See the respective Changelogs for more details. Both releases are available as a tarball as well as a patch against the previous release. They are available for download from downloads.digium.com. Don't seem to be on www.asterisk.org (1.2.19 and 1.4.4) Sorry, I still have to get the powers that be to update the home page :-) -- Matthew Fredrickson Software/Firmware Engineer 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] libmfcr2 is giving definition error while compiling
On Thu, 2007-08-23 at 20:16 +0530, [EMAIL PROTECTED] wrote: Hi, I am using debian 4.0 with version 2.6.18-4-686 I have downloaded the required files form site asterisk-1.2.24.tar.gz libmfcr2-0.0.3-1.4.tar.bz2 libsupertone-0.0.2.tar.gz libunicall-0.0.3-1.4.tar.bz2 spandsp-20060903.tar.gz I use spandsp-0.0.3. Try that and see if it solves your problem. I downloaded and installed the files in the follwing sequence spandsp libsupertone libunicall Till here it is compiling and copying .so library to /usr/local/lib/ libmfcr2-0.0.3 is giving a lot of definition error I converted .src.rpm file of libmfcr2 to .deb file and installed so .so files are not their in /usr/local/lib Not sure if I understand you correctly but can't you just manually copy the .so files to the right directory where libmfcr2 can find them. Or maybe you just need to run ldconfig so the libs in /usr/local/lib are picked up correctly (check with ldconfig -v). With some minor changes the libmfcr2 get compiled successfully but some rpath error was coming. I don't know what is causing the rpath problems. I don't get rpath errors when the rpath check is done during the rpmbuild on a Fedora 7 box. Regards, Patrick ___ --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] [PHP-AGI] Problem executing script
Problem resolved :Two things :1 - php agi instead of php-cli (I have apache running on my server too and it edits himself all the php.ini it finds)2 - Error using php-q it is amistake : php -q (of course =) )Their is a real problem with asterisk concerning errors in agi script.If there is an error in the script itself asterisk give back : No such file or directory even if the error is just that ; is missing...Thanks for the helpFrom: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Thu, 23 Aug 2007 14:51:27 +Subject: [asterisk-users] [PHP-AGI] Problem executing script Hello,I have succeded in compiling and configuring My TDM Card and asterisk, all works fine. But I have a problem using the PHP Agi.The CLI tells me that when I call my number :-- Starting simple switch on 'Zap/4-1'-- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack-- Executing [EMAIL PROTECTED]:2] AGI(Zap/4-1, rabot.agi) in new stack-- Launched AGI Script /var/lib/asterisk/agi-bin/rabot.agi == rabot.agi: Failed to execute '/var/lib/asterisk/agi-bin/rabot.agi': No such file or directory-- AGI Script rabot.agi completed, returning 0 == Auto fallthrough, channel 'Zap/4-1' status is 'UNKNOWN'-- Hungup 'Zap/4-1'I tought first that it was a problem of chmod so I change the chmod of all the directory agi-bin TO 777But it changed nothing. I have verify that php was well indicate at the beginning of the script :#!/usr/bin/php-qAnd there is a php exec at /usr/binAny ideas about this problem ?Thank for you helpKheraudBesoin d'un e-mail ? Créez gratuitement un compte Windows Live Hotmail et bénéficiez de 2 Go de stockage ! Windows Live Hotmail _ Windows Live Messenger vous offre 30 nouvelles émoticônes gratuites, installées directement dans votre Messenger ! http://www.emoticones-messenger.fr/___ --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 Message Logs
On Thu, 2007-08-23 at 11:14 -0400, [EMAIL PROTECTED] wrote: Is it possible to print the Asterisk message logs to a file, or is this already done? You want to look at the logger.conf configuration file, and see how your Asterisk system is set to log the various types of information (such as debug messages, verbose messages, DTMF messages, etc.) are logged. After changing logger.conf, you can type logger reload at the Asterisk CLI to make the changes take effect. -- Jared Smith Community Relations 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
[asterisk-users] meetme conference problem
Hi, im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running meetme conference, when i try to call meetme i get this from the asterisk console Aug 24 00:14:12 WARNING[15466]: pbx.c:1720 pbx_extension_helper: No application 'MeetMe' for extension (sample, 65000, 1) i recompiled my zaptel and asterisk, but the app_meetme file still didn't install, what am i missing here? ___ --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] How to configure and use GCE4019VOIP phone using asterisk
Hi, I have GCE4019VOIP IP phone with me. Can anybody tell me the steps how to use it for communication in the LAN with other sip phones. I want help from the IP phone side as I have already done it with SIP soft phone... Thanks and Regards, Sanchal ___ --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] 1.4 Branch -- which revision
I'm finally migrating from the 6/7/05 CVS version to 1.4. Had quite a run, I have to admit. Asterisk itself only segfaulted once or twice, but the dns issues have been bothering me. And the box just needs to go. Everything is going on a Ubuntu 6.06TLS server, that's been perfectly stable. I had 1.4.1 installed and running, but not configured. Yesterday I upgraded to 1.4.11, which went smoothly... alas, I really wanted chan_mobile. I attempted to use asterisk 1.4.11 with addons-branch 1.4, but that didn't do it. So it looks like I'd have to use the 1.4 branch for both asterisk and addons. What's the recommended revision here? I don't need bleeding edge (obviously), I just need it stable with chan_mobile and not too much else. 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
Re: [asterisk-users] 1.4 Branch -- which revision
Jay Milk wrote: I'm finally migrating from the 6/7/05 CVS version to 1.4. Had quite a run, I have to admit. Asterisk itself only segfaulted once or twice, but the dns issues have been bothering me. And the box just needs to go. Everything is going on a Ubuntu 6.06TLS server, that's been perfectly stable. I had 1.4.1 installed and running, but not configured. Yesterday I upgraded to 1.4.11, which went smoothly... alas, I really wanted chan_mobile. I attempted to use asterisk 1.4.11 with addons-branch 1.4, but that didn't do it. So it looks like I'd have to use the 1.4 branch for both asterisk and addons. What's the recommended revision here? I don't need bleeding edge (obviously), I just need it stable with chan_mobile and not too much else. Thanks! chan_mobile isn't in asterisk-addons in 1.4 - only trunk. You'll likely have to backport it... (it was developed against 1.4, so the diff from trunk is probably trivial) -- Jason Parker Digium ___ --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 Message Logs
Thanks for your reply. I have previously looked at the logger.conf file. I see that the various types of information can be logged in different ways. After setting the various information types with whatever I want logged, is it possible to save the actual logs to a file (ie: As the messages are bring printed, save them all to a file to be viewed later). Thanks, Denis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jared Smith Sent: Thursday, August 23, 2007 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Message Logs On Thu, 2007-08-23 at 11:14 -0400, [EMAIL PROTECTED] wrote: Is it possible to print the Asterisk message logs to a file, or is this already done? You want to look at the logger.conf configuration file, and see how your Asterisk system is set to log the various types of information (such as debug messages, verbose messages, DTMF messages, etc.) are logged. After changing logger.conf, you can type logger reload at the Asterisk CLI to make the changes take effect. -- Jared Smith Community Relations 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 ___ --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 Message Logs
On Thursday 23 August 2007 12:46, [EMAIL PROTECTED] wrote: Thanks for your reply. I have previously looked at the logger.conf file. I see that the various types of information can be logged in different ways. After setting the various information types with whatever I want logged, is it possible to save the actual logs to a file (ie: As the messages are bring printed, save them all to a file to be viewed later). I utilize this command: nohup script -f -c asterisk -vvvTn /tmp/asterisk.log To start up my apps. This will log everything to a log file. Ron ___ --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 Message Logs
[EMAIL PROTECTED] wrote: Hello, Is it possible to print the Asterisk message logs to a file, or is this already done? By message logs I mean the display that shows up on the asterisk server when a call is made from one user to another. I believe if the verbosity is high, it can show what parts of the extension.conf file that it uses when making the call. I am trying to use two Jain-sip-applet-phones, connected through an Asterisk server. I can't seem to get communication between the two phones. Does anyone have any experience using these open-source Jain-sip-applet-phones? Thanks, Denis Add this to logger.conf: full = notice,warning,error,debug,verbose and you should have most of the output stored in /var/log/asterisk/full Brian. ___ --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 Message Logs
[EMAIL PROTECTED] wrote: Thanks for your reply. I have previously looked at the logger.conf file. I see that the various types of information can be logged in different ways. After setting the various information types with whatever I want logged, is it possible to save the actual logs to a file (ie: As the messages are bring printed, save them all to a file to be viewed later). What do you mean by actual logs? Console (CLI) output? Brian. Thanks, Denis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jared Smith Sent: Thursday, August 23, 2007 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Message Logs On Thu, 2007-08-23 at 11:14 -0400, [EMAIL PROTECTED] wrote: Is it possible to print the Asterisk message logs to a file, or is this already done? You want to look at the logger.conf configuration file, and see how your Asterisk system is set to log the various types of information (such as debug messages, verbose messages, DTMF messages, etc.) are logged. After changing logger.conf, you can type logger reload at the Asterisk CLI to make the changes take effect. ___ --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] 1.4 Branch -- which revision
Jason Parker wrote: Jay Milk wrote: I'm finally migrating from the 6/7/05 CVS version to 1.4. Had quite a run, I have to admit. Asterisk itself only segfaulted once or twice, but the dns issues have been bothering me. And the box just needs to go. Everything is going on a Ubuntu 6.06TLS server, that's been perfectly stable. I had 1.4.1 installed and running, but not configured. Yesterday I upgraded to 1.4.11, which went smoothly... alas, I really wanted chan_mobile. I attempted to use asterisk 1.4.11 with addons-branch 1.4, but that didn't do it. So it looks like I'd have to use the 1.4 branch for both asterisk and addons. What's the recommended revision here? I don't need bleeding edge (obviously), I just need it stable with chan_mobile and not too much else. Thanks! chan_mobile isn't in asterisk-addons in 1.4 - only trunk. You'll likely have to backport it... (it was developed against 1.4, so the diff from trunk is probably trivial) Hmm, I got myself confused into thinking I checked out the 1.4 branch somehow. Or maybe that 22-1.4.4.patch file had partial success. So, to restate the question -- Which trunk revision are folks using successfully? (and no, a diff isn't trivial to someone who barely keeps a command prompt ahead of certain disaster ;-) ___ --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] 1.4 Branch -- which revision
Jay Milk wrote: I'm finally migrating from the 6/7/05 CVS version to 1.4. Had quite a run, I have to admit. Asterisk itself only segfaulted once or twice, but the dns issues have been bothering me. And the box just needs to go. Everything is going on a Ubuntu 6.06TLS server, that's been perfectly stable. I had 1.4.1 installed and running, but not configured. Yesterday I upgraded to 1.4.11, which went smoothly... alas, I really wanted chan_mobile. I attempted to use asterisk 1.4.11 with addons-branch 1.4, but that didn't do it. So it looks like I'd have to use the 1.4 branch for both asterisk and addons. What's the recommended revision here? I don't need bleeding edge (obviously), I just need it stable with chan_mobile and not too much else. Thanks! If you just want chan_mobile, there was a message just yesterday that covered this: Thomas Kenyon [EMAIL PROTECTED] wrote: Try checking out r421 of asterisk-addons, and replacing ast_debug(1, with ast_log(LOG_DEBUG, in all instances in chan_mobile.c. (Still only compile chan_mobile.c. This appears to work with 421, but not 423. I've done this myself with asterisk 1.4.10 and I was able to compile it and install it. I haven't been able to test it until I can borrow a phone with bluetooth. -Dave ___ --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 Message Logs
On Thu, Aug 23, 2007 at 01:07:36PM -0400, Ron Joffe wrote: On Thursday 23 August 2007 12:46, [EMAIL PROTECTED] wrote: Thanks for your reply. I have previously looked at the logger.conf file. I see that the various types of information can be logged in different ways. After setting the various information types with whatever I want logged, is it possible to save the actual logs to a file (ie: As the messages are bring printed, save them all to a file to be viewed later). I utilize this command: nohup script -f -c asterisk -vvvTn /tmp/asterisk.log To start up my apps. This will log everything to a log file. Why nohup? And if you have nohup, why script? It will log everything until the cotrolling terminal is lost, right? I think what you're actually looking for is screen. If you want asterisk daemonized but still want it verbose, use -F -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by 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 Message Logs
Yes, any output from the console logs. I tried viewing the full file and it looks like it's what I was looking for. Thanks for the help. Denis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian Jones Sent: Thursday, August 23, 2007 1:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Message Logs [EMAIL PROTECTED] wrote: Thanks for your reply. I have previously looked at the logger.conf file. I see that the various types of information can be logged in different ways. After setting the various information types with whatever I want logged, is it possible to save the actual logs to a file (ie: As the messages are bring printed, save them all to a file to be viewed later). What do you mean by actual logs? Console (CLI) output? Brian. Thanks, Denis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jared Smith Sent: Thursday, August 23, 2007 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Message Logs On Thu, 2007-08-23 at 11:14 -0400, [EMAIL PROTECTED] wrote: Is it possible to print the Asterisk message logs to a file, or is this already done? You want to look at the logger.conf configuration file, and see how your Asterisk system is set to log the various types of information (such as debug messages, verbose messages, DTMF messages, etc.) are logged. After changing logger.conf, you can type logger reload at the Asterisk CLI to make the changes take effect. ___ --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] meetme conference problem
On Fri, Aug 24, 2007 at 12:16:24AM +0800, Mark Quitoriano wrote: Hi, im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running meetme conference, when i try to call meetme i get this from the asterisk console Aug 24 00:14:12 WARNING[15466]: pbx.c:1720 pbx_extension_helper: No application 'MeetMe' for extension (sample, 65000, 1) i recompiled my zaptel and asterisk, but the app_meetme file still didn't install, what am i missing here? Do you have a zaptel timing source? if head -c 0 /dev/zap/pseudo; then echo I have a working timing source; fi -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by 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] channel not hungup (zombie?) so call limit not reset to zero
im having a strange problem related to call-limit for peers. well im not sure if its related to call-limmit or not. Bottom line is: I call a user A, from user B. user B hears silence, untill it goes to voicemail. when user B hangsup. user B's call limit is reset to 0 but user A's call limit is not reset.strange thing is user A's status on cli is shown as NOANSWER, while user B did not even hear ringing. when i use sip show channels command, it shows me a channel for user A like below: crunch 30d926c1055 00102/0 unkn No Init: INVITE It stays in INVITE state unless i restart my asterisk server. when i restart the channel is clear (ofcorse) So my guess is, its a zombiee channel which asterisk forgot to hangup. WHY? i dont know, maybe there is a problem in sip signalling due to which asterisk didnt recieve the bye signal in the first place or maybe its asterisk fault totally. So because it is not hungup by asterisk thats why its call limit is not reset to zero. I dont have sip debug for this problem yet, i'll post it later when i have it. meanwhile if somebody has experienced a similar problem and has successfully fixed it, then plz share my burden and help me. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --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 Message Logs
On Thursday 23 August 2007 14:00, Tzafrir Cohen wrote: I utilize this command: nohup script -f -c asterisk -vvvTn /tmp/asterisk.log To start up my apps. This will log everything to a log file. Why nohup? And if you have nohup, why script? It will log everything until the cotrolling terminal is lost, right? I think what you're actually looking for is screen. If you want asterisk daemonized but still want it verbose, use -F I call asterisk startup from a shell script. nohup will guarantee that the process will not die if the calling process (whatever started the shell script) dies. script is what I use to make sure that everything that would otherwise go to the asterisk cli output makes it into that file. We spawn our own extensions from asterisk which the asterisk logging facility does not capture. This way we get everything that would be seen on the cli. I'm not looking for screen functionality. -F is not an option on my version of asterisk. Ron ___ --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 Message Logs
The problem with any of these choices is that they do not address logfile rotation. Try this: http://cr.yp.to/daemontools.html Ron Joffe wrote: On Thursday 23 August 2007 14:00, Tzafrir Cohen wrote: I utilize this command: nohup script -f -c asterisk -vvvTn /tmp/asterisk.log To start up my apps. This will log everything to a log file. Why nohup? And if you have nohup, why script? It will log everything until the cotrolling terminal is lost, right? I think what you're actually looking for is screen. If you want asterisk daemonized but still want it verbose, use -F I call asterisk startup from a shell script. nohup will guarantee that the process will not die if the calling process (whatever started the shell script) dies. script is what I use to make sure that everything that would otherwise go to the asterisk cli output makes it into that file. We spawn our own extensions from asterisk which the asterisk logging facility does not capture. This way we get everything that would be seen on the cli. I'm not looking for screen functionality. -F is not an option on my version of asterisk. Ron ___ --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 -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Message Logs
You can configure logger.conf so that it will log just about everything you could want. http://www.voip-info.org/wiki/index.php?page=Asterisk+config+logger.conf -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de [EMAIL PROTECTED] Enviado el: jueves, 23 de agosto de 2007 17:15 Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] Asterisk Message Logs Hello, Is it possible to print the Asterisk message logs to a file, or is this already done? By message logs I mean the display that shows up on the asterisk server when a call is made from one user to another. I believe if the verbosity is high, it can show what parts of the extension.conf file that it uses when making the call. I am trying to use two Jain-sip-applet-phones, connected through an Asterisk server. I can't seem to get communication between the two phones. Does anyone have any experience using these open-source Jain-sip-applet-phones? Thanks, Denis ___ --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 attachment: winmail.dat___ --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] What is this?
Hi List;
I saw this is written in that link:
http://www.voip-info.org/wiki/view/Asterisk+options
And really I was not able to understand for what is
that and where I can learn about it and how to write
such thing? Can some one advise me?
!/bin/bash
asterisk Startup script for the asterisk PBX Server
chkconfig: - 87 15
description: Asterisk is a PBX server.
processname: asterisk
config: /etc/asterisk/
pidfile: /var/run/asterisk.pid
Source function library.
. /etc/rc.d/init.d/functions
asterisk=/usr/sbin/asterisk
prog=Asterisk
pidfile=/var/run/asterisk.pid
lockfile=/var/lock/subsys/asterisk
RETVAL=0
start() {
echo -n $Starting $prog:
daemon $asterisk $OPTIONS
RETVAL=$?
echo
$RETVAL = 0 touch ${lockfile}
return $RETVAL
}
stop() {
echo -n $Stopping $prog:
killproc $asterisk
RETVAL=$?
echo
$RETVAL = 0 rm -f ${lockfile} ${pidfile}
}
reload() {
echo -n $Reloading $prog config files
$asterisk -rx reload
RETVAL=$?
echo
}
See how we were called.
case $1 in
start)
start
;;
stop)
stop
;;
restart)
stop
start
;;
reload)
reload
;;
*)
echo $Usage: $prog
{start|stop|restart|reload}
exit 1
esac
exit $RETVAL
Regards,
Bilal Ghayad
Need a vacation? Get great deals
to amazing places on Yahoo! Travel.
http://travel.yahoo.com/
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[asterisk-users] Asterisk Prompt
Hi List; I read the following sentence: The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable In the following link: http://www.voip-info.org/wiki/index.php page=Asterisk+CLI+prompt The question is: what is the ASTERISK_PROMPT UNIX environment variable and where I can access it to change it? Also where I can find information about it? Regards Bilal Ghayad Park yourself in front of a world of choices in alternative vehicles. Visit the Yahoo! Auto Green Center. http://autos.yahoo.com/green_center/ ___ --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 Message Logs
On Thu, Aug 23, 2007 at 12:43:15PM -0600, Anthony Francis wrote: The problem with any of these choices is that they do not address logfile rotation. Because this can be done with the standard system logrotate, or even by asterisk (if you trust it to that). Decently-packaged Asterisk comes with log rotation configuration. Indeed the output of the CLI should not be simply logged. Asterisk has a good enough logging facility that need not be replicated. There is no need to start asterisk vebosely by default and spend useless CPU time on useless messages. Use verbose messages when trying to debug a problem. Let errors stand out when they come. Try this: http://cr.yp.to/daemontools.html and endure the voodoo. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by 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] meetme conference problem
On 8/23/07, Mark Quitoriano [EMAIL PROTECTED] wrote: Hi, im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running meetme conference, when i try to call meetme i get this from the asterisk console Aug 24 00:14:12 WARNING[15466]: pbx.c:1720 pbx_extension_helper: No application 'MeetMe' for extension (sample, 65000, 1) i recompiled my zaptel and asterisk, but the app_meetme file still didn't install, what am i missing here? check meetme.conf ram ___ --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 Prompt
I'm not sure what features/variables you can use, or where to find information about that, but what this basically means is you can change your CLI prompt by this: export ASTERISK_PROMPT=new prompt then, what you access the CLI, instead of: hostname*CLI you get new prompt Moj bilal ghayyad wrote: Hi List; I read the following sentence: The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable In the following link: http://www.voip-info.org/wiki/index.php page=Asterisk+CLI+prompt The question is: what is the ASTERISK_PROMPT UNIX environment variable and where I can access it to change it? Also where I can find information about it? Regards Bilal Ghayad Park yourself in front of a world of choices in alternative vehicles. Visit the Yahoo! Auto Green Center. http://autos.yahoo.com/green_center/ ___ --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] TC400B and show transcoder
Ben Dinnerville wrote: The problem occurs when we have external (pstn) calls coming into / out of the system (via an iax trunk), in which case we have no control over frame size, as well as occurring with handsets directly connected to the system. Please contact Digium Support to work through these problems, as you have unlimited installation support with the purchase of the product. No, the TC400B does not provide any Zaptel 'spans', so it does not provide a timing source. What documentation is referring to the 'show transcoder' command? That command is not in either Asterisk 1.2 or 1.4, so we need to get that documentation fixed... -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --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] Someone please explain FXO/FXS module/channel relationships in Zaptel configuration to me
From memory, your zaptel and zapata files look ok. signalling for an FXO module would be FXS, and vice versa. As far as I can tell, you're ok there. Now, it's the FXO card that plugs into the phone line. The FXS card gets a phone hooked up to it. Dialing the phone would be Dial(Zap/1... and dialing out the phone LINE would be Dial(Zap/2/18005551212... for example, to dial 1 800 555 1212 Moj Robert La Ferla wrote: Please explain the relationship between modules from the driver (wctdm), the /etc/zaptel.conf file and zapata.conf. Specifically, if I have a FXS module 0 and FXO module 1, what should be used in zaptel.conf and what should be used in zapata.conf? Then finally, in extensions.conf, what is the Zap channel for dialing out? Zap/? % dmesg Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Not installed Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules) % cat /etc/zaptel.conf fxoks=1 fxsks=2 % cat zapata.conf ... signalling=fxo_ks context=outgoing-analog echocancel=yes callerid=asreceived channel = 1 signalling=fxs_ks context=incoming-analog echocancel=yes callerid=asreceived channel = 2 ___ --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] Stable-Stable Asterisk
Hi, folks. I've been on the Asterisk Announce list for a while now, and it seems to me that the release versions of Asterisk are a bit bleeding-edge. They qualify as stable, but I wouldn't call them production stable since half the time a new one comes out, a fix for it comes out the next day. So... that said, what's a good version to linger on? I don't *need* anything particularly fancy, feature-wise, but would like to keep it as secure and stable as possible. And I certainly don't mind fancy features. :) Also (please forgive a newbie), how can I tell what version of Asterisk I'm running? My current install was set up by a vendor and I'm still learning the ropes. Where's the best place to look to find the build number? I do know that I'm running some version of 1.2, and am also not sure if I should stay there, or move up to 1.4. 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
[asterisk-users] Speech Rec on Voicemail
I've had requests to processes incoming voicemails with voice recognition routine and add the output text to the body of the email message from * with the attached .wav file. Has anyone implemented this type of feature and willing to share some notes? Thanks! Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax http://www.rissman.com/ ___ --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] Stable-Stable Asterisk
Show version from the CLI. Ed Pastore wrote: Hi, folks. I've been on the Asterisk Announce list for a while now, and it seems to me that the release versions of Asterisk are a bit bleeding-edge. They qualify as stable, but I wouldn't call them production stable since half the time a new one comes out, a fix for it comes out the next day. So... that said, what's a good version to linger on? I don't *need* anything particularly fancy, feature-wise, but would like to keep it as secure and stable as possible. And I certainly don't mind fancy features. :) Also (please forgive a newbie), how can I tell what version of Asterisk I'm running? My current install was set up by a vendor and I'm still learning the ropes. Where's the best place to look to find the build number? I do know that I'm running some version of 1.2, and am also not sure if I should stay there, or move up to 1.4. 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 -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stable-Stable Asterisk
Ed Pastore wrote: since half the time a new one comes out, a fix for it comes out the next day. So... that said, what's a good version to linger on? I don't *need* Until 1.4 improves, I'm staying with 1.2 I do know that I'm running some version of 1.2, and am also not sure if I should stay there, or move up to 1.4. Show version Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Stable-Stable Asterisk
On 8/23/07, Ed Pastore [EMAIL PROTECTED] wrote: Hi, folks. I've been on the Asterisk Announce list for a while now, and it seems to me that the release versions of Asterisk are a bit bleeding-edge. They qualify as stable, but I wouldn't call them production stable since half the time a new one comes out, a fix for it comes out the next day. That's the niche that ABE is supposed to fill. I personally don't use it, though. I just test the features I plan to use, disable everything else, and seem to do OK. ___ --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] Speech Rec on Voicemail
On 8/23/07, Ryan M. Colbert [EMAIL PROTECTED] wrote: I've had requests to processes incoming voicemails with voice recognition routine and add the output text to the body of the email message from * with the attached .wav file. Has anyone implemented this type of feature and willing to share some notes? That would be very interesting to see, if you get it working. Last I checked, though, speech-to-text didn't work very well without a very small language to choose from, far smaller than English. ___ --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] Is it posible for an incoming to ring to Polycom and cell at the same time?
If it is posible for a imcoming call to ring both the Polycom desk phone and my cell phone at the same time, if I dont answer fall back to my voice mail box. I would like to hire someone to cofigure that for me. Bob -- We've Got Your Name at http://www.mail.com ! Get a FREE E-mail Account Today - Choose From 100+ Domains ___ --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] TDM400P FXO click sounds
1- I've tried running fxotune 2- I've tried turning off all un-necessary hardware in the BIOS 3- I've tried on a different PCI slot. 4- I've tried these suggestions: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting 5- How I check if it the clicking and popping correlates to hard drive activity ? 6- I've not tried installing this board in another PC to test my FXOs 7- I've an MSI motherboard and AMD athlon 64 x2 Dual core processor 8- I've Turning off echotraining. Thanks for any suggest to solve this issue. ___ --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] Is it posible for an incoming to ring to Polycom and cell at the same time?
dial(SIP/polycom-on-my-deskLocal/5551212,15,tr) as an example. Bob Gibson wrote: If it is posible for a imcoming call to ring both the Polycom desk phone and my cell phone at the same time, if I dont answer fall back to my voice mail box. I would like to hire someone to cofigure that for me. Bob -- We've Got Your Name at Mail.com http://www.mail.com/?utm_source=mail_sent_footerutm_medium=emailutm_term=070621utm_content=textlinkutm_campaign=we_got_your_name Get a *FREE* E-mail Account Today - Choose From 100+ Domains ___ --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 -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Prompt
Dear Mojo; Thanks for your help. Why you said export ASTERISK_PROMPT=new prompt ? Regards Bilal I'm not sure what features/variables you can use, or where to find information about that, but what this basically means is you can change your CLI prompt by this: export ASTERISK_PROMPT=new prompt then, what you access the CLI, instead of: hostname*CLI you get new prompt Moj bilal ghayyad wrote: Hi List; I read the following sentence: The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable In the following link: http://www.voip-info.org/wiki/index.php page=Asterisk+CLI+prompt The question is: what is the ASTERISK_PROMPT UNIX environment variable and where I can access it to change it? Also where I can find information about it? Regards Bilal Ghayad Park yourself in front of a world of choices in alternative vehicles. Visit the Yahoo! Auto Green Center. http://autos.yahoo.com/green_center/ ___ --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] Stable-Stable Asterisk
David Gomillion wrote: On 8/23/07, *Ed Pastore* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, folks. I've been on the Asterisk Announce list for a while now, and it seems to me that the release versions of Asterisk are a bit bleeding-edge. They qualify as stable, but I wouldn't call them production stable since half the time a new one comes out, a fix for it comes out the next day. That's the niche that ABE is supposed to fill. I personally don't use it, though. I just test the features I plan to use, disable everything else, and seem to do OK. I stay with 1.2.12 or somewhere around there. End Of Life but seems to have a better ticker than 1.4. Thanks, Steve ___ --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] meetme conference problem
On 8/24/07, ram [EMAIL PROTECTED] wrote: On 8/23/07, Mark Quitoriano [EMAIL PROTECTED] wrote: Hi, im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running meetme conference, when i try to call meetme i get this from the asterisk console Aug 24 00:14:12 WARNING[15466]: pbx.c:1720 pbx_extension_helper: No application 'MeetMe' for extension (sample, 65000, 1) i recompiled my zaptel and asterisk, but the app_meetme file still didn't install, what am i missing here? check meetme.conf i don't know what's the problem, when i installed 1.2.20.1 zaptel everything works. ___ --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 as a softswitch
Can i use asterisk as a softswitch? ___ --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] Linksys (PAP2) delay time between hung up and line release
I have a PAP2 with 2 phone ports. When I call them everything works fine until I hung up the call. There is about 30-40 seconds until I can call to that extension again. Before that it gives me busy messages. Extension config: exten = 199,1,Dial(SIP/199,30) exten = 199,102,Hangup Any suggestions? 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
Re: [asterisk-users] Stable-Stable Asterisk
Steve Totaro wrote: David Gomillion wrote: On 8/23/07, *Ed Pastore* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, folks. I've been on the Asterisk Announce list for a while now, and it seems to me that the release versions of Asterisk are a bit bleeding-edge. They qualify as stable, but I wouldn't call them production stable since half the time a new one comes out, a fix for it comes out the next day. That's the niche that ABE is supposed to fill. I personally don't use it, though. I just test the features I plan to use, disable everything else, and seem to do OK. I stay with 1.2.12 or somewhere around there. End Of Life but seems to have a better ticker than 1.4. Thanks, Steve 1.2.12/14/17 all have seemed very stable to me so far. -- Warm Regards, Lee ___ --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] xPL and Asterisk?
I tried asking in another thread this week, but I'm not sure people saw the actual subject of the question. Does anyone know where to find documentation of xPL, the home automation interface? Specifically for integrating it with Asterisk. xPL is part of Trixbox, so it's being used, but where is some expertise for using it without Trixbox? -- (C) Matthew Rubenstein ___ --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 as a softswitch
Probably. PaulH On Fri, 2007-08-24 at 06:55 +0800, Mark Quitoriano wrote: Can i use asterisk as a softswitch? ___ --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 as a softswitch
Probably. PaulH On Fri, 2007-08-24 at 06:55 +0800, Mark Quitoriano wrote: Can i use asterisk as a softswitch? It is really a soft switch. Not a good one for high carrier class telco usage I'd say but just fine for office PBX replacement, which is what it was designed for. What it is missing AFAIK that a carrier class switch has to have is a SS7 stack. The world's TDM exchanges use SS7 to route calls and most carrier class soft switches can do SS7 over ip. Here's a link that defines a soft switch according to the International Softswitch Consortium: http://www.pcmag.com/encyclopedia_term/0,2542,t=softswitchi=51659,00.asp -Clay ___ --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 as a softswitch
Mark Quitoriano wrote: Can i use asterisk as a softswitch? This thread has been discussed over and over. Search the archives, there are more thoughts and opinions there than you probably have time or desire to read. Thanks, Steve Totaro ___ --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 as a softswitch
Yes you could, but asterisk was designed to be a PBX. I would not use it as soft switch due its limitations. It really depends on how much traffic you are going to be passing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 24 August 2007 1:11 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk as a softswitch Mark Quitoriano wrote: Can i use asterisk as a softswitch? This thread has been discussed over and over. Search the archives, there are more thoughts and opinions there than you probably have time or desire to read. Thanks, Steve Totaro ___ --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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.4/969 - Release Date: 23/08/2007 4:04 p.m. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.484 / Virus Database: 269.12.4/969 - Release Date: 23/08/2007 4:04 p.m. ___ --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] unable to load chan_unicall.so
Edit logger.conf and learn how to enable debugging,verbose and all kind of messages. Enable all levels of messages, try again and tell us what is the error message exactly. Regards, On 8/23/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I am using debian 4.0 with version 2.6.18-4-686 I have downloaded the required files form site asterisk-1.2.24.tar.gz libmfcr2-0.0.3-1.4.tar.bz2 libsupertone-0.0.2.tar.gz libunicall-0.0.3-1.4.tar.bz2 spandsp-20060903.tar.gz I downloaded and installed the files in the follwing sequence spandsp libsupertone libunicall libmfcr2-0.0.3 is giving a lot of definition error I converted .src.rpm file of libmfcr2 to .deb file and installed it. the copying the chn_unicall.c and channels_Makefile.patch to channels subdirectory of asterisk-1.2.24 but when I run ,asterisk -vvgc' on command line it gives error unable to load chan_unicall.so, but it is present in /usr/lib/asterisk/modules. Can anybody tell me how to trobleshoot it. Can anybody tell me what to do how to remove this. Thanka and regards sanchal ___ --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 -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --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] Speech Rec on Voicemail
Ryan M. Colbert wrote: I’ve had requests to processes incoming voicemails with voice recognition routine and add the output text to the body of the email message from * with the attached .wav file. Has anyone implemented this type of feature and willing to share some notes? I seem to recall that Lt. Cmdr Jordi Laforge did some stuff like this not too long ago. I get requests like this all the time -- but the technology is very far from being there. -Stephen- ___ --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] Is it posible for an incoming to ring to Polycom and cell at the same time?
Anthony Francis wrote: dial(SIP/polycom-on-my-deskLocal/5551212,15,tr) Will this work even if the Local is pointing to a Zap channel? As far as I know, this only works with SIP or IAX outgoing. -Stephen- ___ --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 as a softswitch
Clayton Milos wrote: It is really a soft switch. Not a good one for high carrier class telco usage I'd say but just fine for office PBX replacement, which is what it was designed for. What it is missing AFAIK that a carrier class switch has to have is a SS7 stack. The world's TDM exchanges use SS7 to route calls and most carrier class soft switches can do SS7 over ip. chan_zap in Asterisk trunk has SS7 support using libss7, a library written by Matthew Fredrickson, who works at Digium. Some people are already using it by running trunk. It will be in Asterisk 1.6. -- Russell Bryant Software Engineer 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] Is it posible for an incoming to ring to Polycom and cell at the same time?
On Thursday 23 August 2007 11:22:23 pm Stephen Bosch wrote: dial(SIP/polycom-on-my-deskLocal/5551212,15,tr) Will this work even if the Local is pointing to a Zap channel? As far as I know, this only works with SIP or IAX outgoing. I'm not sure where you are getting that assumption from, as I have been Dialing Zap/fooZap/bar, SIP/fooSIP/bar, IAX/fooIAX/bar and combinations of all three for the past several years. The only trick, as Anthony already showed, is to use 'r' when dialing a cell phone so that the caller hears the expected ringback, and not the carrier's The cell phone you are trying to reach is out of the area messages if the cell is out of range or off. -A. ___ --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] xPL and Asterisk?
Matthew Rubenstein wrote: I tried asking in another thread this week, but I'm not sure people saw the actual subject of the question. Does anyone know where to find documentation of xPL, the home automation interface? Specifically for integrating it with Asterisk. xPL is part of Trixbox, so it's being used, but where is some expertise for using it without Trixbox? http://www.google.com/search?q=xpl+home+automation 1st and 3rd results. ___ --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] Saftware RAID1 or Hardware RAID1 with Asterisk
On Tue, 21 Aug 2007 09:50:57 -0400 Dave Fullerton [EMAIL PROTECTED] wrote: Zane C.B. wrote: On Tue, 21 Aug 2007 07:33:23 +0530 Vidura Senadeera [EMAIL PROTECTED] wrote: Dear All, I would like to get community's feedback with regard to RAID1 ( Software or Hardware) implementations with asterisk. This is my setup Motherboard with SATA RAID1 support CENT OS 4.4 Asterisk 1.2.19 Libpri/zaptel latest release 2.8 Ghz Intel processor 2 80 GB SATA Hard disks 256 MB RAM digium PRI/E1 card Following are the concerns I am having I'm planing to put this asterisk server in production enviorment which is having E1 connection to the asterisk server, approximately 20 con-current calls, Music on hold, voice mail boxes. 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) 2. If I use Hardware based RAID 1, what would be the impact to the system? 3. According to your practical experiance what is the ideal solution among both options? I will be highly appreciate your feedback on this regard. 1: Software RAID on Linux is way less than impressive. Plus last a I checked Linux can't handle mirroring a entire disk. Last I looked at it around a year ago you were limited to only mirroring partitions, which is a joke from a administrative standpoint. 2: No real impact other than a bad disk won't mean a reinstall. 3: On Linux, go hardware. On FreeBSD it is personal choice. You can (sort of) run raid on an entire disk, but you have to use LVM. You basically create a single partition on the disk, run raid on that partition and then use LVM with the /dev/md? device as a physical volume that you can then partition with LVM. Yeah, still a ugly solution though. ___ --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] Saftware RAID1 or Hardware RAID1 with Asterisk
On Wed, 22 Aug 2007 12:37:26 -0600 Stephen Bosch [EMAIL PROTECTED] wrote: Zane C.B. wrote: 1: Software RAID on Linux is way less than impressive. Plus last a I checked Linux can't handle mirroring a entire disk. Last I looked at it around a year ago you were limited to only mirroring partitions, which is a joke from a administrative standpoint. How is this any different in FreeBSD? Could you explain to me how else you are going to mirror an entire disk in software when your boot partition is on the disk? The raid info is done the same as on other decent system, it is stored at the in the last sector of the provider. making a mirrored freebsd system is like this... 1: install freebsd 2: dd if=current drive of=2nd drive for mirror 3: gmirror label some name 2nd drive 4: mount 2nd drive and edit fstab to boot using /dev/gmirror/whatever 5: boot from 2nd drive 6: gmirror insert name original drive /me loves GEOM, the goddess of all disk subsystems or whatever. http://www.freebsd.org/cgi/man.cgi?query=gmirrorapropos=0sektion=0manpath=FreeBSD+6.2-RELEASEformat=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] Speech Rec on Voicemail
Nuance offers an SDK to do something similar, I think they say you can only expect between 45-60% accuracy using it though. Total cost is about $6K to $8K for one server license. If there are enough people interested in pooling money I'd be willing to help set up a system to process voicemails and provide the Nuance converted transcript. However, I figure the low accuracy would be the biggest turn off from using Nuance. On 8/23/07, Stephen Bosch [EMAIL PROTECTED] wrote: Ryan M. Colbert wrote: I've had requests to processes incoming voicemails with voice recognition routine and add the output text to the body of the email message from * with the attached .wav file. Has anyone implemented this type of feature and willing to share some notes? I seem to recall that Lt. Cmdr Jordi Laforge did some stuff like this not too long ago. I get requests like this all the time -- but the technology is very far from being there. -Stephen- ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --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
