Re: [asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture.
On Sun, 6 May 2012, Nunya Biznatch wrote: Thanks for the info. It got me digging deeper. I definitely don't want to screw this one up, but I've got to pinch pennies to get this done, so don't want to buy anything that would just be nice to have. ...but if I have to get it, that's what I'll do. Aside from capacity, think about maintenance. If you 'front' your Asterisk servers with Kamailio running on 2 servers (even if these servers are also your Asterisk servers) you have the ability to take an Asterisk server out of production just by reconfiguring Kamailio and waiting the calls in progress to finish. Then you can install patches, replace failing disks, etc, etc, etc. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mysql identifier not found
Hello; is it normal that connid and resutid have values of 73 and 74 ?? How come this value increases ? What does this mean for values 1 to 70 ?? [May 7 08:52:41] -- Executing [h@sub:10] NoOp(SIP/kal3-024f, ) in new stack [May 7 08:52:41] -- Executing [h@sub:11] NoOp(SIP/kal3-024f, clear MySQL-connections) in new stack [May 7 08:52:41] -- Executing [h@sub:12] MYSQL(SIP/kal3-024f, Clear 74) in new stack [May 7 08:52:41] WARNING[15003]: app_mysql.c:194 find_identifier: Identifier 74, identifier_type 2 not found in identifier list [May 7 08:52:41] WARNING[15003]: app_mysql.c:510 aMYSQL_clear: Invalid result identifier 74 passed in aMYSQL_clear [May 7 08:52:41] -- Executing [h@sub:13] MYSQL(SIP/kal3-024f, Disconnect 73) in new stack [May 7 08:52:41] WARNING[15003]: app_mysql.c:194 find_identifier: Identifier 73, identifier_type 1 not found in identifier list [May 7 08:52:41] WARNING[15003]: app_mysql.c:527 aMYSQL_disconnect: Invalid connection identifier 73 passed in aMYSQL_disconnect [May 7 08:52:41] -- Executing [h@sub:14] NoOp(SIP/kal3-024f, clear MySQL-connections) in new stack [May 7 08:52:41] -- Executing [h@sub:15] NoOp(SIP/kal3-024f, end) in new stack There are currently only 8 calls going on... Kind regards, Jonas. Original Message Subject:Re: [asterisk-users] Mysql identifier not found Date: Sat, 05 May 2012 12:06:38 +0200 From: Jonas Kellens jonas.kell...@telenet.be To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com I ask this because I find the MySQL status information a bit alarming (2946 connections) : mysql status -- mysql Ver 14.12 Distrib 5.0.95, for redhat-linux-gnu (x86_64) using readline 5.1 Connection id:2922 Current database: Current user:root@localhost SSL:Not in use Current pager:stdout Using outfile:'' Using delimiter:; Server version:5.0.95 Source distribution Protocol version:10 Connection:Localhost via UNIX socket Server characterset:latin1 Db characterset:latin1 Client characterset:latin1 Conn. characterset:latin1 UNIX socket:/var/lib/mysql/mysql.sock Uptime:6 hours 54 min 31 sec Threads: 3 Questions: 4919496 Slow queries: 0 Opens: 47 Flush tables: 1 Open tables: 41 Queries per second avg: 197.800 -- mysql show status like 'Conn%'; +---+---+ | Variable_name | Value | +---+---+ | Connections | 2946 | +---+---+ 1 row in set (0.00 sec) Jonas. On 05/05/2012 11:53 AM, Jonas Kellens wrote: Hello, notice in the console output beneath that there is a resultid 6 but it can not be cleared : [May 5 11:46:27] -- Executing [s@sub:3] MYSQL(SIP/vart-0336, Connect connid localhost dialplan host Asterisk) in new stack [May 5 11:46:27] -- Executing [s@sub:4] MYSQL(SIP/vart-0336, Query resultid 4 DELETE FROM pickuptbl WHERE pickmark LIKE %SIP/vart2-0336%) in new stack [May 5 11:46:27] -- Executing [s@sub:5] MYSQL(SIP/vart-0336, Clear 6) in new stack [May 5 11:46:27] WARNING[17803]: app_mysql.c:194 find_identifier: Identifier 6, identifier_type 2 not found in identifier list [May 5 11:46:27] WARNING[17803]: app_mysql.c:510 aMYSQL_clear: Invalid result identifier 6 passed in aMYSQL_clear [May 5 11:46:27] -- Executing [s@sub:6] MYSQL(SIP/vart-0336, Disconnect 4) in new stack [May 5 11:46:27] -- Executing [s@sub:7] Return(SIP/vart-0336, ) in new stack How come ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] using Wifi smartphones as SIP clients
All, has anyone any experience in using Wifi smartphones as SIP clients? Does this work properly? What models/brands are optimal for this (in terms of ease of use, battery life etc)? Thx!! B. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Wifi smartphones as SIP clients
Used the Snom M9 Wifi DECT phones, they work like charm. SIPDroid on Android phones work good too, however latency is going to be nightmare for u in softphone n wifi kinda scenario. Use good quality Access Points like Ruckus Wireless. Mitul On May 7, 2012 1:55 PM, Bart Coninckx bart.conin...@telenet.be wrote: All, has anyone any experience in using Wifi smartphones as SIP clients? Does this work properly? What models/brands are optimal for this (in terms of ease of use, battery life etc)? Thx!! B. -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Wifi smartphones as SIP clients
Hi, thx! The Snom M9 does not look like a Wifi phone however, nor is it a smarthpone. In order to use that I would have to use access points that can handle both Wifi and DECT. Astraa seems to have (an expensive) one. SIPDroid crossed my mind. You say they work OK but the latency is problematic? Is the experience as a whole then not poblematic? thx! B. On 05/07/12 10:29, Mitul Limbani wrote: Used the Snom M9 Wifi DECT phones, they work like charm. SIPDroid on Android phones work good too, however latency is going to be nightmare for u in softphone n wifi kinda scenario. Use good quality Access Points like Ruckus Wireless. Mitul On May 7, 2012 1:55 PM, Bart Coninckx bart.conin...@telenet.be mailto:bart.conin...@telenet.be wrote: All, has anyone any experience in using Wifi smartphones as SIP clients? Does this work properly? What models/brands are optimal for this (in terms of ease of use, battery life etc)? Thx!! B. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Wifi smartphones as SIP clients
Yeah well Snom M9 is a Wifi DECT Phone (it creates its own 2.4Ghz Frequency which is understood only between Base Station n M9 Units, It works just like your standard Chordless phones) difference being that the Base station is SIP enabled. SIP Droid is great to work with, it solves most of the issues. You should definitely give it a shot. Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Mon, May 7, 2012 at 2:12 PM, Bart Coninckx bart.conin...@telenet.bewrote: Hi, thx! The Snom M9 does not look like a Wifi phone however, nor is it a smarthpone. In order to use that I would have to use access points that can handle both Wifi and DECT. Astraa seems to have (an expensive) one. SIPDroid crossed my mind. You say they work OK but the latency is problematic? Is the experience as a whole then not poblematic? thx! B. On 05/07/12 10:29, Mitul Limbani wrote: Used the Snom M9 Wifi DECT phones, they work like charm. SIPDroid on Android phones work good too, however latency is going to be nightmare for u in softphone n wifi kinda scenario. Use good quality Access Points like Ruckus Wireless. Mitul On May 7, 2012 1:55 PM, Bart Coninckx bart.conin...@telenet.be wrote: All, has anyone any experience in using Wifi smartphones as SIP clients? Does this work properly? What models/brands are optimal for this (in terms of ease of use, battery life etc)? Thx!! B. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Wifi smartphones as SIP clients
Bart Coninckx bart.conin...@telenet.be writes: has anyone any experience in using Wifi smartphones as SIP clients? Yes... Does this work properly? It works nicely for home use for power users who can accept the odd lost call and know how to restart the app or the phone when something goes wrong. Unfortunately I haven't found anything so far which works for business use. The largest problem is that smartphones can't afford (battery-wise) to check for wifi connectivity all the time. If the phone loses connection to the wifi, it often takes more than a minute before it is ready to receive calls again. What models/brands are optimal for this (in terms of ease of use, battery life etc)? iPhone, Android, and Symbian are about equally troublesome. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Wifi smartphones as SIP clients
Benny, very useful, thank you. So, in short, at this stage it's best to go DECT for wireless and if DECT and Wifi need to be combined (because both types of devices exist in the organization), it's preferable to go to access points that offer both networks. Correct? thx, BC On 05/07/12 10:57, Benny Amorsen wrote: Bart Coninckxbart.conin...@telenet.be writes: has anyone any experience in using Wifi smartphones as SIP clients? Yes... Does this work properly? It works nicely for home use for power users who can accept the odd lost call and know how to restart the app or the phone when something goes wrong. Unfortunately I haven't found anything so far which works for business use. The largest problem is that smartphones can't afford (battery-wise) to check for wifi connectivity all the time. If the phone loses connection to the wifi, it often takes more than a minute before it is ready to receive calls again. What models/brands are optimal for this (in terms of ease of use, battery life etc)? iPhone, Android, and Symbian are about equally troublesome. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_dahdi with asterisk 1.4 and new Linux versions
Dear Tzafrir; For sure I re run ./configure Actually, I formatted the machine and reinstalled .. also I removed the asterisk 1.4 and extracted again and I ran ./configure, make and make install. All of these I tried ! As long the Linux OS is new, then it is not possible to get chan_dahdi. And if the OS is new, then we can not compile old DAHDI (versions before 2.4 and maybe 2.4 it self can not be compiled if the OS is new). I tried Fedora and Ubuntu. But with asterisk 1.8, things are running well. Any help? Regards Bilal -- On Fri, May 04, 2012 at 09:24:56AM -0700, bilal ghayyad wrote: What is happening with me that when I used fedora core 16, I compiled and installed dahdi 2.6 and then compiled and installed asterisk 1.4 and it did not create chan_dahdi. I tried to select it by running make menuselect and I discover that it is not possible !! By the way: this problem is not existed with old linux versions .. Have you re-run ./configure ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Wifi smartphones as SIP clients
What about phones like the Unidata WPU-7800 ( http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have experience with those? Would these also suffer from connection losses? thx, BC On 05/07/12 10:57, Benny Amorsen wrote: Bart Coninckxbart.conin...@telenet.be writes: has anyone any experience in using Wifi smartphones as SIP clients? Yes... Does this work properly? It works nicely for home use for power users who can accept the odd lost call and know how to restart the app or the phone when something goes wrong. Unfortunately I haven't found anything so far which works for business use. The largest problem is that smartphones can't afford (battery-wise) to check for wifi connectivity all the time. If the phone loses connection to the wifi, it often takes more than a minute before it is ready to receive calls again. What models/brands are optimal for this (in terms of ease of use, battery life etc)? iPhone, Android, and Symbian are about equally troublesome. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Wifi smartphones as SIP clients
Well in that case, you might seriously want to look @ M9 they are cost effective n definitely work. Most of these cell phone type looking phones have a serious battery drainage problem. Smartphones really have a long way to understand how to preserve battery and deliver one thing (i.e. calls) very effectively, infact a Multi Function device (MFD) like smart phone has this issue. Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Mon, May 7, 2012 at 3:33 PM, Bart Coninckx bart.conin...@telenet.bewrote: What about phones like the Unidata WPU-7800 ( http://www.udcsystems.com/product/wpu7800.php ) ? Does anyone have experience with those? Would these also suffer from connection losses? thx, BC On 05/07/12 10:57, Benny Amorsen wrote: Bart Coninckx bart.conin...@telenet.be bart.conin...@telenet.be writes: has anyone any experience in using Wifi smartphones as SIP clients? Yes... Does this work properly? It works nicely for home use for power users who can accept the odd lost call and know how to restart the app or the phone when something goes wrong. Unfortunately I haven't found anything so far which works for business use. The largest problem is that smartphones can't afford (battery-wise) to check for wifi connectivity all the time. If the phone loses connection to the wifi, it often takes more than a minute before it is ready to receive calls again. What models/brands are optimal for this (in terms of ease of use, battery life etc)? iPhone, Android, and Symbian are about equally troublesome. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Wifi smartphones as SIP clients
man, 07 05 2012 kl. 12:03 +0200, skrev Bart Coninckx: What about phones like the Unidata WPU-7800 ( http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have experience with those? Would these also suffer from connection losses? I don't know that particular phone, but dedicated wifi phones definitely CAN work for professional use. E.g. ASCOM phones work absolutely great, they are just expensive. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Wifi smartphones as SIP clients
The phone I pointed to is a WiFi phone, the M9 is a DECT phone. Different animal. The question is regarding using WiFi as the WiFi infrastructure is already in place. I understand smartphones is not a good option, but what about these WiFi SIP phones? thx! B. On 05/07/12 12:21, Mitul Limbani wrote: Well in that case, you might seriously want to look @ M9 they are cost effective n definitely work. Most of these cell phone type looking phones have a serious battery drainage problem. Smartphones really have a long way to understand how to preserve battery and deliver one thing (i.e. calls) very effectively, infact a Multi Function device (MFD) like smart phone has this issue. Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in mailto:mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Mon, May 7, 2012 at 3:33 PM, Bart Coninckx bart.conin...@telenet.be mailto:bart.conin...@telenet.be wrote: What about phones like the Unidata WPU-7800 ( http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have experience with those? Would these also suffer from connection losses? thx, BC On 05/07/12 10:57, Benny Amorsen wrote: Bart Coninckxbart.conin...@telenet.be mailto:bart.conin...@telenet.be writes: has anyone any experience in using Wifi smartphones as SIP clients? Yes... Does this work properly? It works nicely for home use for power users who can accept the odd lost call and know how to restart the app or the phone when something goes wrong. Unfortunately I haven't found anything so far which works for business use. The largest problem is that smartphones can't afford (battery-wise) to check for wifi connectivity all the time. If the phone loses connection to the wifi, it often takes more than a minute before it is ready to receive calls again. What models/brands are optimal for this (in terms of ease of use, battery life etc)? iPhone, Android, and Symbian are about equally troublesome. /Benny -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Wifi smartphones as SIP clients
Il 07/05/2012 10.46, Mitul Limbani ha scritto: Yeah well Snom M9 is a Wifi DECT Phone (it creates its own 2.4Ghz Frequency which is understood only between Base Station n M9 Units Certainly nothing to do with wifi, neither the frequency/phisical layer nor the protocols involved matches. - http://en.wikipedia.org/wiki/DECT - http://wiki.snom.com/FAQ/What_is_the_channel_diagram_for_DECT_6.0_between_EU_and_US -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Wifi smartphones as SIP clients
On 5/7/2012 4:24 AM, Bart Coninckx wrote: has anyone any experience in using Wifi smartphones as SIP clients? Does this work properly? What models/brands are optimal for this (in terms of ease of use, battery life etc)? www.acrobits.cz has Acrobits and Groundwire, which are both great on iPhone. They also ahve software for Android, but I cant attest. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Wifi smartphones as SIP clients
On 05/07/12 13:04, Benny Amorsen wrote: man, 07 05 2012 kl. 12:03 +0200, skrev Bart Coninckx: What about phones like the Unidata WPU-7800 ( http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have experience with those? Would these also suffer from connection losses? I don't know that particular phone, but dedicated wifi phones definitely CAN work for professional use. E.g. ASCOM phones work absolutely great, they are just expensive. /Benny Does anyone know how things have evolved regarding sessions/call handover in wifi? I remember reading this document in 2008: http://www.abpsec.com/blog/the-dect-versus-wlan-wifi-debate/ http://www.abpsec.com/blog/the-dect-versus-wlan-wifi-debate/ which basically states that it's a bad idea as interruptions of 70 ms are involved. Is this still a challenge? Rgds, BC http://www.abpsec.com/blog/the-dect-versus-wlan-wifi-debate/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Wifi smartphones as SIP clients
Bart Coninckx wrote: I understand smartphones is not a good option, but what about these WiFi SIP phones? I thing smartfones simply lack a business grade softhone implementation. WiFi SIP phones share some problem with smartphones: battery runtime and, most important, roaming and handover. In WiFi network handling roaming/handover is up to the client, this is the same kind of problem that arises in mobile WiFi POS systems. Compared to DECT standard which implements roaming/handover this is a major drawback. Implementing a proper roaming/handover in WiFi networks, using wireless controllers and suitable access points, is very expensive. Where such expensive WiFi infrastructure have to be built only to properly serve wireless phones a DECT multicell system is definitely a winner considering coverage, features and battery runtime. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Wifi smartphones as SIP clients
On 05/07/12 13:50, giovanni.v wrote: I thing smartfones simply lack a business grade softhone implementation. WiFi SIP phones share some problem with smartphones: battery runtime and, most important, roaming and handover. In WiFi network handling roaming/handover is up to the client, this is the same kind of problem that arises in mobile WiFi POS systems. Compared to DECT standard which implements roaming/handover this is a major drawback. Implementing a proper roaming/handover in WiFi networks, using wireless controllers and suitable access points, is very expensive. Where such expensive WiFi infrastructure have to be built only to properly serve wireless phones a DECT multicell system is definitely a winner considering coverage, features and battery runtime. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Giovanni, I think you're completely right. Even to this day, it seems that Wifi is not ready for voice, except while investing a lot of time/money. DECT it is, BC -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to hang up a call after sending SendDTMF() ?
When SendDTMF() finish the process then I want to hang up the call after 20 seconds.. What is the solution to do this? I know there is S(x) option for Dial() application but it still count during SendDTMF() process. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture.
Finally, I'll have a Windows-based workstation that will be used to remote into all the services, for administration, etc... Why? Unfortunately, the existing PBX Administration Software only works on WinBloze. I'm stuck with it until I can decommission it. I need to plan to use FreePBX on all Asterisk Servers, but I don't intend to install it until I'm in regular MAC maintenance mode. It is ashame you are going this far with your setup to rely on FreePBX. For something this complex, you are setting your self up for some heartache. It is my intention to do everything from the command line. However, there will be times when I'll have Interns coming in and doing some of the MAC activities, and I thought this might be an easier way for the day to day to get done. I've never seen it myself either, so am curious. Finally, there's the glitter factor. When my bosses come in and want a dog and pony show on the new phone system, they want to see fluffy bunnies and kittens, not the Ox that's doing the pulling. CLI = old in the minds of those that don't comprehend. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_dahdi with asterisk 1.4 and new Linux versions
On Mon, May 07, 2012 at 02:59:17AM -0700, bilal ghayyad wrote: Dear Tzafrir; For sure I re run ./configure Actually, I formatted the machine and reinstalled .. also I removed the asterisk 1.4 and extracted again and I ran ./configure, make and make install. All of these I tried ! As long the Linux OS is new, then it is not possible to get chan_dahdi. And if the OS is new, then we can not compile old DAHDI (versions before 2.4 and maybe 2.4 it self can not be compiled if the OS is new). I tried Fedora and Ubuntu. But with asterisk 1.8, things are running well. What is chan_dahdi missing? ./menuselect/contrib/dummy-select -c ./menuselect/contrib/dummy-select -m chan_dahdi -v -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P: Lifetime Replacement
On Sun, May 06, 2012 at 10:42:16AM -0600, Greg Woods wrote: I have a Digium TDM400P card that appears to have died. The first noted symptoms were that dahdi would fail to reload on boot. On closer inspection, the card looks totally dead; no lights on at all. Does the device show up in lspci? -- Russ Meyerriecks Digium, Inc. | Brogrammer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P: Lifetime Replacement
On Mon, 2012-05-07 at 10:26 -0500, Russ Meyerriecks wrote: On Sun, May 06, 2012 at 10:42:16AM -0600, Greg Woods wrote: I have a Digium TDM400P card that appears to have died. The first noted symptoms were that dahdi would fail to reload on boot. On closer inspection, the card looks totally dead; no lights on at all. Does the device show up in lspci? I'll check that when I get home tonight; I didn't try lspci. But it does NOT show up in dmesg at boot time, which means it probably won't be there for lspci either. --Greg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.8 busypatterns
Hi, is it possible to detect 4 length pattern busy cadence detection on FXO lines in 1.8?? Here the tones are: 425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off) in asterisk 1.4 busy detect worked in asterisk 1.6 didn´t work and i was told that 1.6 can´t handle 4 length patterns, but what about 1.8?? for now I can only hangup by asking the provider polarity switch. Thanks best regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Wifi smartphones as SIP clients
On Mon, May 07, 2012 at 12:03:17PM +0200, Bart Coninckx wrote: What about phones like the Unidata WPU-7800 ( http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have experience with those? Would these also suffer from connection losses? I've been using a UTStarcom GF-210 for the last year and more as my personal phone - dual-mode 2G GSM and SIP/802.11. Sound quality on SIP is slightly better than 2G, getting it to talk to Asterisk is no problem at all, but certainly if you're moving from one wifi device to another you will get dropped calls. If that's your use case, it's going to be that way whatever hardware you use - I haven't seen any implementations of 802.11F or 802.11r in the field. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Wifi smartphones as SIP clients
On Mon, 2012-05-07 at 19:03 +0100, Roger Burton West wrote: On Mon, May 07, 2012 at 12:03:17PM +0200, Bart Coninckx wrote: What about phones like the Unidata WPU-7800 ( http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have experience with those? Would these also suffer from connection losses? I've been using a UTStarcom GF-210 for the last year and more as my personal phone - dual-mode 2G GSM and SIP/802.11. Sound quality on SIP is slightly better than 2G, getting it to talk to Asterisk is no problem at all, but certainly if you're moving from one wifi device to another you will get dropped calls. If that's your use case, it's going to be that way whatever hardware you use - I haven't seen any implementations of 802.11F or 802.11r in the field. Hope that these are better that the utstar F1000: Keep on re-chargibg as battery is empty in no-time, and security is lousy; just wep, no wpa. hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Wifi smartphones as SIP clients
On Mon, May 07, 2012 at 09:14:36PM +0200, Hans Witvliet wrote: Hope that these are better that the utstar F1000: Keep on re-chargibg as battery is empty in no-time, and security is lousy; just wep, no wpa. WPA and WPA2. Battery lasts about a day in dual mode, much longer in 2G-only of course. And at UKP30 they may be worth a punt even if you end up upgrading to something else. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.10.1 ring tone in thebackground after call sucessfully answered.
motty.cruz motty.cruz at gmail.com writes: Hello I apologize for not being specific. I'm using SIP. Our provider is call Wiline. I do not have Dahdi install on this server. The phone I used is Polycom soundpoint ip 450. Rining continues in the background after successfully answered, is a ramdom not constantly. Thanks, -Motty -Original Message- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, March 19, 2012 8:54 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.8.10.1 ring tone in thebackground after call sucessfully answered. Which Technology are you using for the call (DAHDI/SIP/other)? -Original Message- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of motty.cruz Sent: Monday, March 19, 2012 10:42 AM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Asterisk 1.8.10.1 ring tone in the background after call sucessfully answered. Hello All, I upgraded Asterisk from 1.8.4 to Asterisk 1.8.10.1 last week. After the upgrade when I make a call and the other party answers the call a ringing tone is heard in the background even thought the call was successfully answered. This issue is ramdom and is not consistent. I do think is a Asterisk issue can some one point me a solution. Thanks, Motty -- _ Hello Motty, I have the exact same issue with several Polycom Soundpoint IP 450's. The phones are using the Broadsoft platform. Seeing your problem on an Asterisk platform makes me think this is a Polycom issue. We have two other types of phones: Polycom Soundpoint IP 331 and 650's. These types do not have these problems at all. I'm going to contact Polycom to see if they have any record of this issue. Please contact me if you happen to stumble upon a solution! Regards, Maarten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8 busypatterns
- Original Message - From: Sebastian Gutierrez scg...@gmail.com To: asterisk-users@lists.digium.com Sent: Monday, May 7, 2012 10:38:03 AM Subject: [asterisk-users] 1.8 busypatterns Hi, is it possible to detect 4 length pattern busy cadence detection on FXO lines in 1.8?? Here the tones are: 425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off) in asterisk 1.4 busy detect worked in asterisk 1.6 didn´t work and i was told that 1.6 can´t handle 4 length patterns, but what about 1.8?? for now I can only hangup by asking the provider polarity switch. Thanks best regards. No. I implemented this in Asterisk 10. From the CHANGES file: from CHANGES in the trunk: -- --- Functionality changes from Asterisk 1.8 to Asterisk 10 --- -- ... chan_dahdi -- * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used with busydetect. usage example: busypattern=200,200,200,600 So you'll need to upgrade to Asterisk 10 if you want to use that feature. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.10.1 ring tone in thebackground after call sucessfully answered.
Hello Maarten, Our issue was with our VOIP provider or so I think, because after I called my provider the issued went away, I have not heard any complaints since, it happened a few times two days in a row then it went away. Hopefully you documented time and phone numbers when it happened and take it to your provider. Good luck finding the solution for your problem. Thanks, Motty -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Maarten Sent: Monday, May 07, 2012 2:37 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users]Asterisk 1.8.10.1 ring tone in thebackground after call sucessfully answered. motty.cruz motty.cruz at gmail.com writes: Hello I apologize for not being specific. I'm using SIP. Our provider is call Wiline. I do not have Dahdi install on this server. The phone I used is Polycom soundpoint ip 450. Rining continues in the background after successfully answered, is a ramdom not constantly. Thanks, -Motty -Original Message- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, March 19, 2012 8:54 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.8.10.1 ring tone in thebackground after call sucessfully answered. Which Technology are you using for the call (DAHDI/SIP/other)? -Original Message- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of motty.cruz Sent: Monday, March 19, 2012 10:42 AM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Asterisk 1.8.10.1 ring tone in the background after call sucessfully answered. Hello All, I upgraded Asterisk from 1.8.4 to Asterisk 1.8.10.1 last week. After the upgrade when I make a call and the other party answers the call a ringing tone is heard in the background even thought the call was successfully answered. This issue is ramdom and is not consistent. I do think is a Asterisk issue can some one point me a solution. Thanks, Motty -- _ Hello Motty, I have the exact same issue with several Polycom Soundpoint IP 450's. The phones are using the Broadsoft platform. Seeing your problem on an Asterisk platform makes me think this is a Polycom issue. We have two other types of phones: Polycom Soundpoint IP 331 and 650's. These types do not have these problems at all. I'm going to contact Polycom to see if they have any record of this issue. Please contact me if you happen to stumble upon a solution! Regards, Maarten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1913 / Virus Database: 2425/4983 - Release Date: 05/07/12 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8 busypatterns
Can you point me to the commit to see if i can backport it? Thanks El 07/05/2012 18:50, Jonathan Rose jr...@digium.com escribió: - Original Message - From: Sebastian Gutierrez scg...@gmail.com To: asterisk-users@lists.digium.com Sent: Monday, May 7, 2012 10:38:03 AM Subject: [asterisk-users] 1.8 busypatterns Hi, is it possible to detect 4 length pattern busy cadence detection on FXO lines in 1.8?? Here the tones are: 425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off) in asterisk 1.4 busy detect worked in asterisk 1.6 didn´t work and i was told that 1.6 can´t handle 4 length patterns, but what about 1.8?? for now I can only hangup by asking the provider polarity switch. Thanks best regards. No. I implemented this in Asterisk 10. From the CHANGES file: from CHANGES in the trunk: -- --- Functionality changes from Asterisk 1.8 to Asterisk 10 --- -- ... chan_dahdi -- * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used with busydetect. usage example: busypattern=200,200,200,600 So you'll need to upgrade to Asterisk 10 if you want to use that feature. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P: Lifetime Replacement
On Mon, 2012-05-07 at 10:26 -0500, Russ Meyerriecks wrote: On Sun, May 06, 2012 at 10:42:16AM -0600, Greg Woods wrote: I have a Digium TDM400P card that appears to have died. The first noted symptoms were that dahdi would fail to reload on boot. On closer inspection, the card looks totally dead; no lights on at all. Does the device show up in lspci? I just checked, and as expected, it does not show up with lspci (all of the other PCI devices do). --Greg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture.
On Mon, 7 May 2012, Nunya Biznatch wrote: It is my intention to do everything from the command line. However, there will be times when I'll have Interns coming in and doing some of the MAC activities, and I thought this might be an easier way for the day to day to get done. Sounds like a recipe for hard to find problems to me. You'll change something, they'll change something, something will be broken. I'd suggest one or the other. Personally, I like plain text configuration files because I can annotate them with a modification history showing what I* changed, when, and why. I can also use tools like 'diff' to compare working to broken configurations. I back up all of the configuration files for all of the hosts** for all of the clients I administer every day. Each host runs a script to stuff everything I think is important into a tarball and email it to a 'backup' email address at my office. If I really trash something on a client host, I can always get the last known good files. I have the tarballs going back 5 years so if a client every said 'hey, remember when you did...' I can pull a rabbit out of my hat. *) I usually work alone or in small (2-4) teams. **) I skip hosts that are supposed to be exact clones of other hosts. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
