[asterisk-users] Dail in modem
Hello I am required to do some thing like Dail in modem . User will have to call a modem just like we do in dail up connection now we need to handle that request and retrieve some parameters from that send a HTTp request to a web server and then after getting http response send user a feed back .. this is a requirement .. Is it possible ?? what is the way forward ?? please give me a direction Best Regards Shakeel Abbas ___ -- 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] help setting tone zone
On Fri, Jun 19, 2009 at 06:04:17PM +1000, Alex Samad wrote: Hi I am seeing this in my syslog [235900.797660] dahdi: Registered tone zone 0 (United States / North America) I am in Australia so I would want to set them to AUS zone I have got this though options wctdm24xxp opermode=AUSTRALIA Set loadzone and defaultzone in /etc/dahdi/system.conf . -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] help setting tone zone
Hi I am seeing this in my syslog [235900.797660] dahdi: Registered tone zone 0 (United States / North America) I am in Australia so I would want to set them to AUS zone I have got this though options wctdm24xxp opermode=AUSTRALIA thanks -- See, we love -- we love freedom. That's what they didn't understand. They hate things - we love things. - George W. Bush 08/29/2002 Oklahoma City, OK signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recompiling dahdi-linux after kernel update - To minimize downtime
On Thu, Jun 18, 2009 at 11:24:40PM -0500, Karl Fife wrote: After a kernel update (but before rebooting) Is there a way to recompile Zap/Dahdi against the new kernel? My objective is to eliminate the additional downtime that occurs while recompiling/installing zap/dahdi after booting into the new kernel. Please correct me if I'm wrong: My understanding is that until you reboot (after a kernel update), recompiling zap/dahdi still compiles against the OLD kernel, and that's why zap/dahdi doesn't start after rebooting into the new kernel (even if you recompiled it just before rebooting). So my question is: Is there a method to recompile dahdi/zap against the new kernel such that the only downtime is the actual server bounce itself? OR is the current best practice just to simply to reboot, recompile, restart? Thanks in advance. 1. install new kernel (but don't reboot yet) 2. in the dahdi-linux source directory: make KVERS=new version make KVERS=new version install or: make KSRC=/path/to/kernel/source/tree make KSRC=/path/to/kernel/source/tree install 3. reboot -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7941G Auth
Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018 SIP41.8-0-2SR1S.loads term41.default.loads term61.default.loads XMLDefault.cnf SEPmac_address.cnf.xml ..and in tftp log I have: Connection received from 192.168.1.61 on port 49153 [19/06 10:16:35.968] Read request for file CTLSEPmac_address.tlv. Mode octet [19/06 10:16:35.968] File CTLSEPmac_address.tlv : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:35.968] Connection received from 192.168.1.61 on port 49154 [19/06 10:16:36.109] Read request for file SEPmac_address.cnf.xml. Mode octet [19/06 10:16:36.109] Using local port 3995 [19/06 10:16:36.109] SEPmac_address.cnf.xml: sent 15 blks, 7239 bytes in 0 s. 0 blk resent [19/06 10:16:36.171] Connection received from 192.168.1.61 on port 49155 [19/06 10:16:40.046] Read request for file /mk-sip.jar. Mode octet [19/06 10:16:40.046] File \mk-sip.jar : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:40.046] Connection received from 192.168.1.61 on port 49156 [19/06 10:16:40.984] Read request for file Italy/g3-tones.xml. Mode octet [19/06 10:16:40.999] File Italy\g3-tones.xml : error 3 in system call CreateFile Impossibile trovare il percorso specificato. [19/06 10:16:40.999] Connection received from 192.168.1.61 on port 49164 [19/06 10:16:42.843] Read request for file dialplan.xml. Mode octet [19/06 10:16:42.859] Using local port 3998 [19/06 10:16:42.859] dialplan.xml: sent 1 blk, 104 bytes in 0 s. 0 blk resent [19/06 10:16:42.906] In XMLDefault.cnf I have: loadInformation309 SIP41.8-0-2SR1S/loadInformation309 ..and on 7941G I have: App Load IDjar41sip.8-0-1-18.sbn Boot Load ID7941G_64-02070631Amd64megRel.bin VersionSIP41.8-0-2SR1S Thanks. -- Salvatore. ___ -- 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] agent login status visual clue on Polycom?
Hi, Is there a way on Polycom phones to show an agent whether he is logged in or not? ___ -- 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] help setting tone zone
On Fri, Jun 19, 2009 at 11:08:49AM +0300, Tzafrir Cohen wrote: On Fri, Jun 19, 2009 at 06:04:17PM +1000, Alex Samad wrote: Hi I am seeing this in my syslog [235900.797660] dahdi: Registered tone zone 0 (United States / North America) I am in Australia so I would want to set them to AUS zone I have got this though options wctdm24xxp opermode=AUSTRALIA Set loadzone and defaultzone in /etc/dahdi/system.conf . I have and when i trolled through my syslog, I notice sometimes it sets it to 0 and some times not! -- Well, that's going to be up to the pundits and the people to make up their mind. I'll tell you what is a president for him, for example, talking about my record in the state of Texas. I mean, he's willing to say anything in order to convince people that I haven't had a good record in Texas. - George W. Bush 09/20/2000 on MSNBC signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] agent login status visual clue on Polycom?
From memory, it is doable but this is a feature that Polycom never quite finished writing. PaulH On Fri, 2009-06-19 at 10:58 +0200, Louis-David Mitterrand wrote: Hi, Is there a way on Polycom phones to show an agent whether he is logged in or not? ___ -- 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] How run AsyncAGI commands in background
Hi Moy, I'm using an asterisk 1.4.18 from scratch patched with the last AsyncAGI patch, which fixes a bug about stopping AsyncAGI applications, as may be you can recall from the thread [asterisk-users] async agi question in http://lists.digium.com/pipermail/asterisk-users/2009-April/230488.html. This patched asterisk works fine and it stops the async agi applications launched from the AsyncAGI loop before the Redirect as it's expected. It's for that I don't think stopping the mixmonitor application launched from the AsyncAGI loop would be a bug if I redirect the call. I would be only getting the same behavior than I got with the stream file application as you explained it should be at http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/#comment-365 I'm only asking if there's any way to prevent stopping applications launched on a channel from the AsyncAGI loop if this channel is redirected afterward, with something like a continue_running_in_background flag in the previous AGI invocation from AMI. Of course, it bring us the problem we'll need some kind of identifier and some stop action to be able to stop those applications running in background launched from the AsyncAGI loop Anyway, as you asked me some days ago, I have published at http://docs.google.com/View?id=ahfnfrcrh3rr_4dkcx9dgw a simple configuration and a simple scenario in order you can try to reproduce what I'm saying. I don't need anyone to do anything for me. I'm willing to do the work, I like programming and trying new things as well, but I'll need some guidelines to go straight ahead. Thanks all Jose Moises Silva escribió: On Sun, Jun 7, 2009 at 4:37 PM, Jose Ariascyr2...@gmail.com wrote: Hi Moy, I'll do it so, but for your answer, it seems you are thinking about it as it could be a bug. I don't think so. I mean: the redirect action on a channel in AsyncAGI stops the current agi execution. It's the normal behavior. It's the way to stop a playfile on a channel if it was previously launched from AsyncAGI: making a redirect out of the AsyncAGI loop. Therefore, when I realized the previously launched EXE MixMonitor AsyncAGI execution was stopping after doing a redirect to meetme, I didn't think it was a bug. I though what I was needing it was a way to tell AsyncAGI, hey, don't stop this agi execution on the channel, even it will be redirected out of AGI on an individual basis for each AsyncAGI EXEC command launched. Thanks Jose The way I see it if you make EXEC MixMonitor inside AsyncAGI loop and then redirect to MeetMe and you don't get the audio recorded, then it's not a normal behavior, MixMonitor is an application that should passively monitor the channel audio independently of where the channel is (regardless of whether the command was executed in Async AGI or dial plan or whatever). However you are also using an old asterisk version and is not likely you can report a bug unless you upgrade to the latest Asterisk and reproduce without a patched Asterisk (for example executing EXEC MixMonitor inside a regular AGI script and then redirect to MeetMe). ___ -- 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] gap between Playback and Queue
Well, at least this did not add to the wait time of your callers :) It should be possible to do silence detection/removal automagically using sox as well - see e.g. http://www.justlinux.com/forum/showthread.php?t=136678 2009/6/18 Louis-David Mitterrand vindex+lists-asterisk-us...@apartia.orgvindex%2blists-asterisk-us...@apartia.org Thanks for this suggestion. The problem was indeed a silence at the beginning of my musiconhold tracks. Audacity did a fine job and fixed my problem. -- Loway - home of QueueMetrics - http://queuemetrics.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] snom mass deploy help
I fail to see how this script is useful in order to use Snom's Plug'n'play config. Who said it does? The Topic is snom mass deploy - not Plug'n'play config. It does not use snoms Plug'n'play config, but it still provides for snom mass deploy using the phones' built-in dhcp/http mechanism. Conrad ___ -- 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 and EC2 today at 12 Noon EDT
Nir Simionovich is about to become a father. He will be joining our conference at 12 Noon EDT today from the Maternity Ward to talk about Amazon EC2 cloud computing with Asterisk. Nir gave a very good presentation on this at AMOOCON a few weeks ago (see http://www.amoocon.de for more on that). The advantage here though is that he'll be live with us for your questions. All the details on how to join us are here: http://VUC.me IRC: #voip-users-conference You all have free or cheap dialing: Call (724) 444-7444 and enter 22622# PIN# - Get your PIN at Talkshoe.com or use 1# as a guest sip:7463#22622#...@proxy.ideasip.com (g711 u) or talks...@vuc.onsip.com and enter 22622#PIN# wideband, sip:200...@login.zipdx.com (g722) See you there with your EC2 questions and comments! /r ___ -- 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] snom mass deploy help
Conrad Wood wrote: On Thu, 2009-06-18 at 14:21 +0200, Philipp Kempgen wrote: On Jun 18, 2009, at 7:25 AM, Alex Samad a...@samad.com.au wrote: I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. FWIW I use a home-grown cgi script to configure the mass-deploy. (attached) Conrad I've also been asked to start thinking about doing something similar soon and your solution looks like the sort of thing I'll have to employ. Don't really know perl but I'm sure I'll work out what's going on. Thanks for the head start. Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and openvpn and sip
On Thu, 2009-06-18 at 19:18 -0400, John A. Sullivan III wrote: On Thu, 2009-06-18 at 14:55 +0200, Giorgio Incantalupo wrote: Hi John, I already have the ccd dir with the iroute (mandatory for routing to pc/phone connected to vpn client). During the last test I could register and make a call but voice disappears after 1, 2 seconds. I'm trying to understand if it is a bandwidth problem. At the moment I have my phone connected to the openvpn client (which is its gateway) but I have to use the vpn ip (10.0.0.1) to register the phone, the openvpn server local ip (192.168.1.12) is not working. I suppose it is a sip protocol problem: I had to change the sip.conf setting nat=yes to make the phone dial and domain = 10.0.0.1 to make the voice pass (or at least the first 2 seconds). I keep on working on the vpn since it seems so little is missing to have a clear conversation. Let me know if your tests are successfull. Thank you. Giorgio snip Hi, Giorgio. So far so good. I have twinkle running on my laptop (the VPN client), a Snom 320 and a Snom 360 on the internal network routing through my laptop. I haven't done much more than register and execute a very basic dialplan but it is all working so far. I hit a couple of small bumps but nothing to do with *. I had forgotten to tell my DNS to accept requests from the test network. One of the phones somehow decided the data center firewall was an outbound SIP proxy. Once I removed that setting, it all worked just fine. I am using native addresses across the VPN; there is no NAT. I've not yet had sustained conversations. I'll be doing that in a while hopefully - John Sustained conversations are working fine with reinvite=yes. Take care - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] snom mass deploy help
Alex Samad schrieb: On Thu, Jun 18, 2009 at 02:21:47PM +0200, Philipp Kempgen wrote: Snom supports what they call PnP config. Technically: ---cut--- # SIP Event Notification: # http://tools.ietf.org/html/rfc3265 # SIP UA Profile Event Package: # http://tools.ietf.org/html/draft-ietf-sipping-config-framework-15 # http://tools.ietf.org/html/draft-channabasappa-sipping-app-profile-type-03 # # Snom 3xx: # http://wiki.snom.com/SIP_Traces#PnP_Config # other drafts: # http://tools.ietf.org/html/draft-petrie-sip-config-framework-01 # http://www.cs.columbia.edu/sip/drafts/sip/draft-schulzrinne-sip-config-events-00.txt ---cut--- Gemeinschaft (Asterisk-based open-source PBX) comes with a SIP UA config responder. http://www.amooma.de/gemeinschaft/ https://svn.amooma.com/gemeinschaft/trunk/opt/gemeinschaft/sbin/gs-sip-ua-config-responder/gs-sip-ua-config-responder I saw links to this on the voip-info site, but my german is non existant. but on first glance through this seems to be what I want. Yeah sorry, that page is not available in English. At least the SIP traces, RFCs, drafts and the code are in English (or rather in Perl :-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] Dail in modem
On Fri, 2009-06-19 at 11:45 +0500, ABBAS SHAKEEL wrote: I am required to do some thing like Dail in modem . User will have to call a modem just like we do in dail up connection now we need to handle that request and retrieve some parameters from that send a HTTp request to a web server and then after getting http response send user a feed back .. Why do you need a modem? What will be dialing into the Asterisk system, a human or a machine? ___ -- 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] How run AsyncAGI commands in background
On Fri, Jun 19, 2009 at 5:32 AM, Jose Ariascyr2...@gmail.com wrote: Hi Moy, I'm using an asterisk 1.4.18 from scratch patched with the last AsyncAGI patch, which fixes a bug about stopping AsyncAGI applications, as may be you can recall from the thread [asterisk-users] async agi question in http://lists.digium.com/pipermail/asterisk-users/2009-April/230488.html. This patched asterisk works fine and it stops the async agi applications launched from the AsyncAGI loop before the Redirect as it's expected. It's for that I don't think stopping the mixmonitor application launched from the AsyncAGI loop would be a bug if I redirect the call. I would be only getting the same behavior than I got with the stream file application as you explained it should be at http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/#comment-365 I'm only asking if there's any way to prevent stopping applications launched on a channel from the AsyncAGI loop if this channel is redirected afterward, with something like a continue_running_in_background flag in the previous AGI invocation from AMI. Of course, it bring us the problem we'll need some kind of identifier and some stop action to be able to stop those applications running in background launched from the AsyncAGI loop Anyway, as you asked me some days ago, I have published at http://docs.google.com/View?id=ahfnfrcrh3rr_4dkcx9dgw a simple configuration and a simple scenario in order you can try to reproduce what I'm saying. I don't need anyone to do anything for me. I'm willing to do the work, I like programming and trying new things as well, but I'll need some guidelines to go straight ahead. Jose, the thing is that MixMonitor IS a background application in nature, that's why I say is unexpected that after a redirect the recording no longer works. In fact, that's why StopMixMonitor application is needed, because all MixMonitor does is to launch a background thread that hooks into the channel audio, then the channel continues to execute other applications in the dial plan while this background thread monitors its audio, on a redirect StopMixMonitor thread should continue saving audio until StopMixMonitor is called. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Cisco 7941G Auth
Sasa wrote: Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018 SIP41.8-0-2SR1S.loads term41.default.loads term61.default.loads XMLDefault.cnf SEPmac_address.cnf.xml ..and in tftp log I have: Connection received from 192.168.1.61 on port 49153 [19/06 10:16:35.968] Read request for file CTLSEPmac_address.tlv. Mode octet [19/06 10:16:35.968] File CTLSEPmac_address.tlv : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:35.968] Connection received from 192.168.1.61 on port 49154 [19/06 10:16:36.109] Read request for file SEPmac_address.cnf.xml. Mode octet [19/06 10:16:36.109] Using local port 3995 [19/06 10:16:36.109] SEPmac_address.cnf.xml: sent 15 blks, 7239 bytes in 0 s. 0 blk resent [19/06 10:16:36.171] Connection received from 192.168.1.61 on port 49155 [19/06 10:16:40.046] Read request for file /mk-sip.jar. Mode octet [19/06 10:16:40.046] File \mk-sip.jar : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:40.046] Connection received from 192.168.1.61 on port 49156 [19/06 10:16:40.984] Read request for file Italy/g3-tones.xml. Mode octet [19/06 10:16:40.999] File Italy\g3-tones.xml : error 3 in system call CreateFile Impossibile trovare il percorso specificato. [19/06 10:16:40.999] Connection received from 192.168.1.61 on port 49164 [19/06 10:16:42.843] Read request for file dialplan.xml. Mode octet [19/06 10:16:42.859] Using local port 3998 [19/06 10:16:42.859] dialplan.xml: sent 1 blk, 104 bytes in 0 s. 0 blk resent [19/06 10:16:42.906] In XMLDefault.cnf I have: loadInformation309 SIP41.8-0-2SR1S/loadInformation309 ..and on 7941G I have: App Load IDjar41sip.8-0-1-18.sbn Boot Load ID7941G_64-02070631Amd64megRel.bin VersionSIP41.8-0-2SR1S Thanks. -- Salvatore. I have had sucess with creating a zero length file named CTLSEPmac_address.tlv Or whatever the damn thing wants, and it then seems to be happy. With Cisco 7960's Your results may vary John Novack ___ -- 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 -- Dog is my co-pilot ___ -- 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] agent login status visual clue on Polycom?
Can't tell you the how, but you should be able to do this as a BLF or buddy function perhaps using hints. I know the GUI can tell if an agent is logged in or out, so it can't be that hard. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Louis-David Mitterrand Sent: Friday, June 19, 2009 3:58 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] agent login status visual clue on Polycom? Hi, Is there a way on Polycom phones to show an agent whether he is logged in or not? ___ -- 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] Cisco 7941G Auth
I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Friday, June 19, 2009 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G Auth Sasa wrote: Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018 SIP41.8-0-2SR1S.loads term41.default.loads term61.default.loads XMLDefault.cnf SEPmac_address.cnf.xml ..and in tftp log I have: Connection received from 192.168.1.61 on port 49153 [19/06 10:16:35.968] Read request for file CTLSEPmac_address.tlv. Mode octet [19/06 10:16:35.968] File CTLSEPmac_address.tlv : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:35.968] Connection received from 192.168.1.61 on port 49154 [19/06 10:16:36.109] Read request for file SEPmac_address.cnf.xml. Mode octet [19/06 10:16:36.109] Using local port 3995 [19/06 10:16:36.109] SEPmac_address.cnf.xml: sent 15 blks, 7239 bytes in 0 s. 0 blk resent [19/06 10:16:36.171] Connection received from 192.168.1.61 on port 49155 [19/06 10:16:40.046] Read request for file /mk-sip.jar. Mode octet [19/06 10:16:40.046] File \mk-sip.jar : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:40.046] Connection received from 192.168.1.61 on port 49156 [19/06 10:16:40.984] Read request for file Italy/g3-tones.xml. Mode octet [19/06 10:16:40.999] File Italy\g3-tones.xml : error 3 in system call CreateFile Impossibile trovare il percorso specificato. [19/06 10:16:40.999] Connection received from 192.168.1.61 on port 49164 [19/06 10:16:42.843] Read request for file dialplan.xml. Mode octet [19/06 10:16:42.859] Using local port 3998 [19/06 10:16:42.859] dialplan.xml: sent 1 blk, 104 bytes in 0 s. 0 blk resent [19/06 10:16:42.906] In XMLDefault.cnf I have: loadInformation309 SIP41.8-0-2SR1S/loadInformation309 ..and on 7941G I have: App Load IDjar41sip.8-0-1-18.sbn Boot Load ID7941G_64-02070631Amd64megRel.bin VersionSIP41.8-0-2SR1S Thanks. -- Salvatore. I have had sucess with creating a zero length file named CTLSEPmac_address.tlv Or whatever the damn thing wants, and it then seems to be happy. With Cisco 7960's Your results may vary John Novack ___ -- 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 -- Dog is my co-pilot ___ -- 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] Cisco 7941G Auth
The free solar winds TFTP server worked well for me, as well as the CentOS TFTP server The Solar Winds one produces an on screen log file which is very nice while troubleshooting The Cisco 7960's I have set up want to find the file name, but seem not to care if it is empty. Both with the windows and linux TFTP I feel sure there are some differences unknown to me in the 41/61 and 40/60 Ciscos, as well as the 7970 I can't get my 7960's to work beyond version 7.4, but they work so I leave well enough alone IMO the only phone worse to get working than the Cisco is the Polycom! You may feel differently John Novack David Gibbons wrote: I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Friday, June 19, 2009 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G Auth Sasa wrote: Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018 SIP41.8-0-2SR1S.loads term41.default.loads term61.default.loads XMLDefault.cnf SEPmac_address.cnf.xml ..and in tftp log I have: Connection received from 192.168.1.61 on port 49153 [19/06 10:16:35.968] Read request for file CTLSEPmac_address.tlv. Mode octet [19/06 10:16:35.968] File CTLSEPmac_address.tlv : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:35.968] Connection received from 192.168.1.61 on port 49154 [19/06 10:16:36.109] Read request for file SEPmac_address.cnf.xml. Mode octet [19/06 10:16:36.109] Using local port 3995 [19/06 10:16:36.109] SEPmac_address.cnf.xml: sent 15 blks, 7239 bytes in 0 s. 0 blk resent [19/06 10:16:36.171] Connection received from 192.168.1.61 on port 49155 [19/06 10:16:40.046] Read request for file /mk-sip.jar. Mode octet [19/06 10:16:40.046] File \mk-sip.jar : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:40.046] Connection received from 192.168.1.61 on port 49156 [19/06 10:16:40.984] Read request for file Italy/g3-tones.xml. Mode octet [19/06 10:16:40.999] File Italy\g3-tones.xml : error 3 in system call CreateFile Impossibile trovare il percorso specificato. [19/06 10:16:40.999] Connection received from 192.168.1.61 on port 49164 [19/06 10:16:42.843] Read request for file dialplan.xml. Mode octet [19/06 10:16:42.859] Using local port 3998 [19/06 10:16:42.859] dialplan.xml: sent 1 blk, 104 bytes in 0 s. 0 blk resent [19/06 10:16:42.906] In XMLDefault.cnf I have: loadInformation309 SIP41.8-0-2SR1S/loadInformation309 ..and on 7941G I have: App Load IDjar41sip.8-0-1-18.sbn Boot Load ID7941G_64-02070631Amd64megRel.bin VersionSIP41.8-0-2SR1S Thanks. -- Salvatore. I have had sucess with creating a zero length file named CTLSEPmac_address.tlv Or whatever the damn thing wants, and it then seems to be happy. With Cisco 7960's Your results may vary John Novack ___ -- 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 -- Dog is my co-pilot ___ -- 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 -- Dog is my co-pilot ___ -- 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 7941G Auth
John Novack wrote: I have had sucess with creating a zero length file named CTLSEPmac_address.tlv Or whatever the damn thing wants, and it then seems to be happy. With Cisco 7960's Your results may vary ...with CTLSEPmac_address.tlv in tftp dir in log file I have: Using local port 3131 [19/06 17:14:02.816] CTLSEPmac_address.tlv: sent 1 blk, 0 bytes in 0 s. 0 blk resent [19/06 17:14:02.863] Connection received from 192.168.1.61 on port 49188 [19/06 17:14:06.988] Read request for file CTLSEPmac_address.tlv. Mode octet [19/06 17:14:06.988] ..and the problem isn't resolved. Thanks. -- Salvatore. - Original Message - From: John Novack jnov...@stromberg-carlson.org To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 19, 2009 4:38 PM Subject: Re: [asterisk-users] Cisco 7941G Auth Sasa wrote: Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018 SIP41.8-0-2SR1S.loads term41.default.loads term61.default.loads XMLDefault.cnf SEPmac_address.cnf.xml ..and in tftp log I have: Connection received from 192.168.1.61 on port 49153 [19/06 10:16:35.968] Read request for file CTLSEPmac_address.tlv. Mode octet [19/06 10:16:35.968] File CTLSEPmac_address.tlv : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:35.968] Connection received from 192.168.1.61 on port 49154 [19/06 10:16:36.109] Read request for file SEPmac_address.cnf.xml. Mode octet [19/06 10:16:36.109] Using local port 3995 [19/06 10:16:36.109] SEPmac_address.cnf.xml: sent 15 blks, 7239 bytes in 0 s. 0 blk resent [19/06 10:16:36.171] Connection received from 192.168.1.61 on port 49155 [19/06 10:16:40.046] Read request for file /mk-sip.jar. Mode octet [19/06 10:16:40.046] File \mk-sip.jar : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:40.046] Connection received from 192.168.1.61 on port 49156 [19/06 10:16:40.984] Read request for file Italy/g3-tones.xml. Mode octet [19/06 10:16:40.999] File Italy\g3-tones.xml : error 3 in system call CreateFile Impossibile trovare il percorso specificato. [19/06 10:16:40.999] Connection received from 192.168.1.61 on port 49164 [19/06 10:16:42.843] Read request for file dialplan.xml. Mode octet [19/06 10:16:42.859] Using local port 3998 [19/06 10:16:42.859] dialplan.xml: sent 1 blk, 104 bytes in 0 s. 0 blk resent [19/06 10:16:42.906] In XMLDefault.cnf I have: loadInformation309 SIP41.8-0-2SR1S/loadInformation309 ..and on 7941G I have: App Load IDjar41sip.8-0-1-18.sbn Boot Load ID7941G_64-02070631Amd64megRel.bin VersionSIP41.8-0-2SR1S Thanks. -- Salvatore. I have had sucess with creating a zero length file named CTLSEPmac_address.tlv Or whatever the damn thing wants, and it then seems to be happy. With Cisco 7960's Your results may vary John Novack ___ -- 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 -- Dog is my co-pilot ___ -- 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] Cisco 7941G Auth
David Gibbons wrote: I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. I have tried with tftp on linux machine but the result isn't changed. Thanks. -- Salvatore. - Original Message - From: David Gibbons d...@videon-central.com To: novacks...@gmail.com; 'Asterisk Users MailingList - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, June 19, 2009 4:50 PM Subject: Re: [asterisk-users] Cisco 7941G Auth I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Friday, June 19, 2009 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G Auth Sasa wrote: Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018 SIP41.8-0-2SR1S.loads term41.default.loads term61.default.loads XMLDefault.cnf SEPmac_address.cnf.xml ..and in tftp log I have: Connection received from 192.168.1.61 on port 49153 [19/06 10:16:35.968] Read request for file CTLSEPmac_address.tlv. Mode octet [19/06 10:16:35.968] File CTLSEPmac_address.tlv : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:35.968] Connection received from 192.168.1.61 on port 49154 [19/06 10:16:36.109] Read request for file SEPmac_address.cnf.xml. Mode octet [19/06 10:16:36.109] Using local port 3995 [19/06 10:16:36.109] SEPmac_address.cnf.xml: sent 15 blks, 7239 bytes in 0 s. 0 blk resent [19/06 10:16:36.171] Connection received from 192.168.1.61 on port 49155 [19/06 10:16:40.046] Read request for file /mk-sip.jar. Mode octet [19/06 10:16:40.046] File \mk-sip.jar : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:40.046] Connection received from 192.168.1.61 on port 49156 [19/06 10:16:40.984] Read request for file Italy/g3-tones.xml. Mode octet [19/06 10:16:40.999] File Italy\g3-tones.xml : error 3 in system call CreateFile Impossibile trovare il percorso specificato. [19/06 10:16:40.999] Connection received from 192.168.1.61 on port 49164 [19/06 10:16:42.843] Read request for file dialplan.xml. Mode octet [19/06 10:16:42.859] Using local port 3998 [19/06 10:16:42.859] dialplan.xml: sent 1 blk, 104 bytes in 0 s. 0 blk resent [19/06 10:16:42.906] In XMLDefault.cnf I have: loadInformation309 SIP41.8-0-2SR1S/loadInformation309 ..and on 7941G I have: App Load IDjar41sip.8-0-1-18.sbn Boot Load ID7941G_64-02070631Amd64megRel.bin VersionSIP41.8-0-2SR1S Thanks. -- Salvatore. I have had sucess with creating a zero length file named CTLSEPmac_address.tlv Or whatever the damn thing wants, and it then seems to be happy. With Cisco 7960's Your results may vary John Novack ___ -- 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 -- Dog is my co-pilot ___ -- 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] Strange res_config_odbc error messages in 1.6.1.1
When I try to use 1.6.1.1 with ODBC and MySQL, I get these: [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_...@asterisk: column type (-9) unrecognized for column 'name' [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_...@asterisk: column type (-9) unrecognized for column 'ipaddr' [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_...@asterisk: column type (-9) unrecognized for column 'port' [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_...@asterisk: column 'regseconds' is not long enough to contain realtime data (needs 11) [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_...@asterisk: column type (-9) unrecognized for column 'defaultuser' [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_...@asterisk: column type (-9) unrecognized for column 'fullcontact' [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_...@asterisk: column type (-9) unrecognized for column 'regserver' [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_...@asterisk: column type (-9) unrecognized for column 'useragent' [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_...@asterisk: column type (-9) unrecognized for column 'lastms' I don't understand the column type (-9) messages, and I'm also confused about regseconds, because: `regseconds` int(11) NOT NULL default '0' With 1.6.0.9 everything works fine. It is of course possible that I somehow miscompile 1.6.1.1 but not 1.6.0.9, but I'm trying to keep the configurations the same. Has any of you seen this? /Benny ___ -- 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 7941G Auth
On Fri, Jun 19, 2009 at 05:25:18PM +0200, Sasa wrote: David Gibbons wrote: I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. I have tried with tftp on linux machine but the result isn't changed. Thanks. TFTP is a simple protocol. Linux also comes with a TFTP client. Have you tried using it to grab the file? -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange res_config_odbc error messages in 1.6.1.1
On Friday 19 June 2009 10:25:15 Benny Amorsen wrote: When I try to use 1.6.1.1 with ODBC and MySQL, I get these: [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_...@asterisk: column type (-9) unrecognized for column 'name' [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_...@asterisk: column type (-9) unrecognized for column 'ipaddr' [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_...@asterisk: column type (-9) unrecognized for column 'port' [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_...@asterisk: column 'regseconds' is not long enough to contain realtime data (needs 11) [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_...@asterisk: column type (-9) unrecognized for column 'defaultuser' [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_...@asterisk: column type (-9) unrecognized for column 'fullcontact' [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_...@asterisk: column type (-9) unrecognized for column 'regserver' [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_...@asterisk: column type (-9) unrecognized for column 'useragent' [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_...@asterisk: column type (-9) unrecognized for column 'lastms' I don't understand the column type (-9) messages, and I'm also confused about regseconds, because: `regseconds` int(11) NOT NULL default '0' With 1.6.0.9 everything works fine. It is of course possible that I somehow miscompile 1.6.1.1 but not 1.6.0.9, but I'm trying to keep the configurations the same. Has any of you seen this? Looks like you're using a widevarchar column, which is something I didn't plan for. It should be in SVN shortly, however. On the regseconds, it's a typo in the code (UINTEGER4 should be INTEGER4). -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7941G Auth
What does your SEPMacAddress.cnf.xml file look like? In my experience, the XMLDefault.cnf.xml file is not retrieved from my 79x1 devices and I had to specify the firmware version in each SEP file. I am using 8-4-4S, but for you this would be something like this: device loadInformationSIP41.8-0-2SR1S/loadInformation /device And you shouldn't need the tlv file. -Jonathan On Fri, Jun 19, 2009 at 8:25 AM, Sasa s...@shoponweb.it wrote: David Gibbons wrote: I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. I have tried with tftp on linux machine but the result isn't changed. Thanks. -- Salvatore. - Original Message - From: David Gibbons d...@videon-central.com To: novacks...@gmail.com; 'Asterisk Users MailingList - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, June 19, 2009 4:50 PM Subject: Re: [asterisk-users] Cisco 7941G Auth I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Friday, June 19, 2009 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G Auth Sasa wrote: Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018 SIP41.8-0-2SR1S.loads term41.default.loads term61.default.loads XMLDefault.cnf SEPmac_address.cnf.xml ..and in tftp log I have: Connection received from 192.168.1.61 on port 49153 [19/06 10:16:35.968] Read request for file CTLSEPmac_address.tlv. Mode octet [19/06 10:16:35.968] File CTLSEPmac_address.tlv : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:35.968] Connection received from 192.168.1.61 on port 49154 [19/06 10:16:36.109] Read request for file SEPmac_address.cnf.xml. Mode octet [19/06 10:16:36.109] Using local port 3995 [19/06 10:16:36.109] SEPmac_address.cnf.xml: sent 15 blks, 7239 bytes in 0 s. 0 blk resent [19/06 10:16:36.171] Connection received from 192.168.1.61 on port 49155 [19/06 10:16:40.046] Read request for file /mk-sip.jar. Mode octet [19/06 10:16:40.046] File \mk-sip.jar : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:40.046] Connection received from 192.168.1.61 on port 49156 [19/06 10:16:40.984] Read request for file Italy/g3-tones.xml. Mode octet [19/06 10:16:40.999] File Italy\g3-tones.xml : error 3 in system call CreateFile Impossibile trovare il percorso specificato. [19/06 10:16:40.999] Connection received from 192.168.1.61 on port 49164 [19/06 10:16:42.843] Read request for file dialplan.xml. Mode octet [19/06 10:16:42.859] Using local port 3998 [19/06 10:16:42.859] dialplan.xml: sent 1 blk, 104 bytes in 0 s. 0 blk resent [19/06 10:16:42.906] In XMLDefault.cnf I have: loadInformation309 SIP41.8-0-2SR1S/loadInformation309 ..and on 7941G I have: App Load IDjar41sip.8-0-1-18.sbn Boot Load ID7941G_64-02070631Amd64megRel.bin VersionSIP41.8-0-2SR1S Thanks. -- Salvatore. I have had sucess with creating a zero length file named CTLSEPmac_address.tlv Or whatever the damn thing wants, and it then seems to be happy. With Cisco 7960's Your results may vary John Novack ___ -- 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 -- Dog is my co-pilot ___ -- 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:
Re: [asterisk-users] Dail in modem
Hello Actually i am required to make two application 1) that user use 2) that is deployed on server Application for user will be just like the windows standard connection using dail up modem but user will dail my PSTN number instead of the number we inter provided by ISP. on deployed server side we will get he usename and pass and other parameters of application and then use them in java code is it possible ? (nothing is impossible but for a Asterisk and java developer with limited time frame) Thanks On Fri, Jun 19, 2009 at 7:24 PM, Bob Piercepier...@westmancom.com wrote: On Fri, 2009-06-19 at 11:45 +0500, ABBAS SHAKEEL wrote: I am required to do some thing like Dail in modem . User will have to call a modem just like we do in dail up connection now we need to handle that request and retrieve some parameters from that send a HTTp request to a web server and then after getting http response send user a feed back .. Why do you need a modem? What will be dialing into the Asterisk system, a human or a machine? ___ -- 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] Dail in modem
is it just me or am i right in thinking this has nothing to do with asterisk? 2009/6/19 ABBAS SHAKEEL shakeel.abbas@gmail.com Hello Actually i am required to make two application 1) that user use 2) that is deployed on server Application for user will be just like the windows standard connection using dail up modem but user will dail my PSTN number instead of the number we inter provided by ISP. on deployed server side we will get he usename and pass and other parameters of application and then use them in java code is it possible ? (nothing is impossible but for a Asterisk and java developer with limited time frame) Thanks On Fri, Jun 19, 2009 at 7:24 PM, Bob Piercepier...@westmancom.com wrote: On Fri, 2009-06-19 at 11:45 +0500, ABBAS SHAKEEL wrote: I am required to do some thing like Dail in modem . User will have to call a modem just like we do in dail up connection now we need to handle that request and retrieve some parameters from that send a HTTp request to a web server and then after getting http response send user a feed back .. Why do you need a modem? What will be dialing into the Asterisk system, a human or a machine? ___ -- 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] Dail in modem
Geraint lee I also dont know .what kind of requirements are these :P i am just looking if it can happen On Fri, Jun 19, 2009 at 9:33 PM, Geraint Leegera...@gmail.com wrote: is it just me or am i right in thinking this has nothing to do with asterisk? 2009/6/19 ABBAS SHAKEEL shakeel.abbas@gmail.com Hello Actually i am required to make two application 1) that user use 2) that is deployed on server Application for user will be just like the windows standard connection using dail up modem but user will dail my PSTN number instead of the number we inter provided by ISP. on deployed server side we will get he usename and pass and other parameters of application and then use them in java code is it possible ? (nothing is impossible but for a Asterisk and java developer with limited time frame) Thanks On Fri, Jun 19, 2009 at 7:24 PM, Bob Piercepier...@westmancom.com wrote: On Fri, 2009-06-19 at 11:45 +0500, ABBAS SHAKEEL wrote: I am required to do some thing like Dail in modem . User will have to call a modem just like we do in dail up connection now we need to handle that request and retrieve some parameters from that send a HTTp request to a web server and then after getting http response send user a feed back .. Why do you need a modem? What will be dialing into the Asterisk system, a human or a machine? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dail in modem
On 19 Jun 2009, at 17:33, Geraint Lee wrote: is it just me or am i right in thinking this has nothing to do with asterisk? My thoughts too. Was keeping quiet incase I was misunderstanding. ___ -- 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] Speex problem installing on CentOS 5.3
Steve Totaro wrote: On Thu, Jun 18, 2009 at 12:46 PM, Brent Davidson br...@texascountrytitle.com mailto:br...@texascountrytitle.com wrote: John A. Sullivan III wrote: Hello, all. I am delightfully slogging my way through installing and configuring Asterisk 1.6.1.1 on CentOS 5.3. I'm learning lots and admiring the product but I'm having a problem getting speex to install and I would very much like to use it. It is not available in menuselect and the problem appears to be with speex_preprocess_ctl: [r...@pbx01 asterisk-1.6.1.1]# grep -i speex config.log configure:43813: checking for speex_encode in -lspeex configure:43848: gcc -o conftest -g -O2 conftest.c -lspeex -lm 5 configure:43906: checking speex/speex.h usability configure:43947: checking speex/speex.h presence configure:44015: checking for speex/speex.h configure:44076: checking for speex_preprocess_ctl in -lspeex configure:44111: gcc -o conftest -g -O2 conftest.c -lspeex -lm 5 /home/compuser/Asterisk/asterisk-1.6.1.1/conftest.c:306: undefined reference to `speex_preprocess_ctl' | #define HAVE_SPEEX 1 | #define HAVE_SPEEX_VERSION | char speex_preprocess_ctl (); | return speex_preprocess_ctl (); configure:44341: checking for speex_preprocess_ctl in -lspeexdsp configure:44376: gcc -o conftest -g -O2 conftest.c -lspeexdsp -lm 5 /usr/bin/ld: cannot find -lspeexdsp | #define HAVE_SPEEX 1 | #define HAVE_SPEEX_VERSION | char speex_preprocess_ctl (); | return speex_preprocess_ctl (); Internet searches have only further confused the issue for me. It seems this is part of libspeex which in the RedHat world is provided by the speex-devel package (which I have installed): [r...@pbx01 ~]# rpm -qa | grep speex speex-devel-1.0.5-4.el5_1.1 speex-1.0.5-4.el5_1.1 What is the magic to make speex available to Asterisk on CentOS 5.3? Or am I stuck having to uninstall the speex packages and install speex from source? Thanks - John I ended up having to install from source. There are apparently bits of speex that are not included in the RPM's. It's a farily simple install though. Good luck, -Brent I am curious if a yum -y install speex* would have worked for you? I will give it a try on my next 5.3 box. That was the first thing I tried before trying yum -y install speex-devel There was always some link or library missing or possibly just in a non-standard location. Installing from source I just did a configure, make, and make install then all was good. ___ -- 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] Switchvox HA options
What are the HA options for Switchvox systems? Is it possible to set up redundant systems with DRBD? I know on the digium website they talk about Optional cold spare failover What does this mean? Is this an active spare ready for some sort of automated failover? Thanks for you help, Bob ___ -- 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-gui: http://id_address:8088/asterisk/static/config/cfgadvanced.html
Hi Danny; I found cfgbasic.html under the /var/lib/asterisk/static-http/config and did not find cfgadvanced.html, any advise? About the root directory: do u mean that I have to set my root directoty to be /var/lib/asterisk/ at the httpd server? Because by default the httpd server has another root directory than this, or you are talking about another root directory? Please advise. By the way: what about the port 8088, from where I can set it (in case I need to change that port to be another port)? Looking to hear from you. Regards Bilal /var/lib/asterisk/static-http/config/cfgadvanced.html is the file location. The root directory is /var/lib/asterisk unless you change it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Thursday, June 18, 2009 4:48 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk-gui:http://id_address:8088/asterisk/static/config/cfgadvanced.html Hello List; Actually based on what I read at Guru that after I did the installation and configuration of the asterisk-gui, I can access it using the link: http://id_address:8088/asterisk/static/config/cfgadvanced.html I tried to search for something like /asterisk/static/config/cfgadvanced.html but did not find it at all, where this cfgadvanced.html? Another issue: if we look for the above link, the question is: do I configure the httpd server and determine the root directory, so the root directory should contain the /asterisk/static/config/cfgadvanced.html? Any advise? So how the installation will know the default httpd path and install the asterisk/static/config/cfgadvanced.html under that default? Any advise? Regards Bilal ___ -- 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-gui: http://id_address:8088/asterisk/static/config/cfgadvanced.html
On Fri, Jun 19, 2009 at 10:28:28AM -0700, bilal ghayyad wrote: Hi Danny; I found cfgbasic.html under the /var/lib/asterisk/static-http/config and did not find cfgadvanced.html, any advise? cfgbasic.html is now merely a redirection to index.html . cfgadvanced.html is now gone - the advanced mode of editing sip.conf, iax.conf etc. directly has been removed in ver. 2 of the GUI. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-gui: http://id_address:8088/asterisk/static/config/cfgadvanced.html
On Fri, Jun 19, 2009 at 10:28:28AM -0700, bilal ghayyad wrote: About the root directory: do u mean that I have to set my root directoty to be /var/lib/asterisk/ at the httpd server? Because by default the httpd server has another root directory than this, or you are talking about another root directory? Please advise. By the way: what about the port 8088, from where I can set it (in case I need to change that port to be another port)? As it is served by the Asterisk httpd, you may change it in /etc/asterisk/http.conf . That said, I generally prefer to proxy the file serving through apache (or lightttpd, or whatever) -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-gui: http://id_address:8088/asterisk/static/config/cfgadvanced.html
bilal ghayyad schrieb: what about the port 8088, from where I can set it (in case I need to change that port to be another port)? That would be the bindport parameter in /etc/asterisk/http.conf I guess. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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 on AVR32
why is CROSS_ARCH=Linux? is this something the AVR32 distro is doing, or something you did? it should be something line avr or avr32 On Thu, Jun 18, 2009 at 3:08 AM, Paulo Santos paulo.r.san...@sapo.ptwrote: Greetings everyone, I'm trying to compile asterisk for an AVR32 (Atmel NGW100). Buildroot for AVR32 already has the asterisk package, though it has bugs. Firstly it tries to apply a patch for 1.2 on a 1.6, but deleting the contents of the patch file did the trick. Now, the problem is making asterisk. The first error is because asterisk needed to be ./configure:ed. Trying to just do ./configure, make gives an error [1]. Trying to do ./configure with the same args as make plus --host it can't even configure [2] I don't know much about cross-compiling, or even regular compiling for that matter. Does any one have any idea on how to do this? Thanks in advance, Best regards, Paulo Santos [1] menuselect/menuselect --check-deps menuselect.makeopts /bin/bash: menuselect/menuselect: cannot execute binary file make[1]: *** [menuselect.makeopts] Error 126 make[1]: Leaving directory `/home/psantos/br/buildroot-avr32-v2.3.0/build_avr32/asterisk-1.6.0-beta6' make: *** [/home/psantos/br/buildroot-avr32-v2.3.0/build_avr32/asterisk-1.6.0-beta6/asterisk] Error 2 [2] configure: WARNING: If you wanted to set the --build type, don't use --host. If a cross compiler is detected then cross compile mode will be used. checking build system type... i686-pc-linux-gnu checking host system type... Invalid configuration `CROSS_ARCH=Linux': machine `CROSS_ARCH=Linux' not recognized configure: error: /bin/bash ./config.sub CROSS_ARCH=Linux failed ___ -- 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] Nagios under *
Hi Steve
I tried your script :
STATUS=$(sudo asterisk -rnx pri show span 1\
| awk '/Status/ {print $3}'\
)
if [ Up, == ${STATUS} ]
thenecho PRI UP
exit 0
elseecho PRI DOWN
exit 2
fi
but still i get PRI down in the Nagios web interface while if i execute this
command from command line i get PRI UP...i m really going mad..did a clean
install again but still same problem.. Iv;e also given permission of 777 to the
script and saved it under /usr/local/nagios and given the same path in
commands.cfg under objects folder of /usr/local/nagios/etc ... can anyone
please help me out ?
Thanks Sriram
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Re: [asterisk-users] Nagios under *
Do you need to path sudo (/usr/sbin/sudo)? Try running the script as nobody
and see what happens.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sriram
Sent: Friday, June 19, 2009 1:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Nagios under *
Hi Steve
I tried your script :
STATUS=$(sudo asterisk -rnx pri show span 1\
| awk '/Status/ {print $3}'\
)
if [ Up, == ${STATUS} ]
thenecho PRI UP
exit 0
elseecho PRI DOWN
exit 2
fi
but still i get PRI down in the Nagios web interface while if i execute this
command from command line i get PRI UP...i m really going mad..did a clean
install again but still same problem.. Iv;e also given permission of 777 to
the script and saved it under /usr/local/nagios and given the same path in
commands.cfg under objects folder of /usr/local/nagios/etc ... can anyone
please help me out ?
Thanks Sriram
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Re: [asterisk-users] Asterisk + mySQL
On Thu, 2009-06-18 at 11:52 -0500, Tilghman Lesher wrote: In modules.conf: noload = cdr_csv.so Are there other modules I need to load or unload ?? asterisk*CLI module show like cdr Module Description Use Count cdr_addon_mysql.so MySQL CDR Backend 0 app_setcdruserfield.so CDR user field apps 0 func_cdr.soCDR dialplan function 0 app_cdr.so Tell Asterisk to not maintain a CDR for 0 cdr_manager.so Asterisk Manager Interface CDR Backend 0 app_forkcdr.so Fork The CDR into 2 separate entities 0 cdr_csv.so Comma Separated Values CDR Backend 0 cdr_custom.so Customizable Comma Separated Values CDR 0 8 modules loaded asterisk*CLI module show like odbc Module Description Use Count 0 modules loaded asterisk*CLI module show like sql Module Description Use Count cdr_addon_mysql.so MySQL CDR Backend 0 app_addon_sql_mysql.so Simple Mysql Interface 0 res_config_mysql.soMySQL RealTime Configuration Driver 0 3 modules loaded modules.conf : autoload=yes noload=pbx_gtkconsole.so load=res_musiconhold.so load=cdr_addon_mysql.so noload=chan_alsa.so Why is there a res_mysql.conf and a cdr_mysql.conf ?? They both look alike... ___ -- 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] Incoming Call trouble with new *Now 1.5 setup
For determining security risks, its specific to how your dialplan is set up. If a person connects to your asterisk, what can they do? what happens? did you set the incoming context to one with outgoing dialing rules? Also for filtering calls, you'll probably want to either look at the incoming sip packets or ask your ITSP for info on how the calls come in. I have a DID with les.net and in their web interface I can choose between having the calls addressed to sipu...@did.voip.les.net or d...@did.voip.les.net Or maybe even check the cdr files. Or just look at the error message when theres no catchall, it's an error like incoming call for extension 523523 doesn't match anything in context whatever. On Wed, Jun 17, 2009 at 12:20 PM, Zaheer Master zkml...@aisww.com wrote: Hi All, I’m having a bit of trouble with my new *NOW setup. I’ve downloaded and installed *NOW 1.5. We’re using 1 SIP Trunk from SimpleSignal.com. Outbound calling works great, but I’m having some trouble with inbound calls. First, we would get the “the number you have dialed is not in service” error on inbound calls. After some googling, I found out that I needed to enable anonymous SIP calls in to the system. When I did that, it started to work. I was a little worried about potential security risks so I wanted to filter inbound calls by DID. I tried the formats DID, +DID, and +1DID, but all of them caused the box to hang up or give me the “number not in service” error message. Are there any known security risks by allowing anonymous SIP and having an ANY/ANY inbound route? Thanks in advance for any help! --Zaheer ___ -- 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] IMAP voice mail storage
Hello, all. I am attempting to use IMAP voice mail storage in Asterisk
1.6.1.1 on CentOS 5.3 using Zimbra 5.1.6. I will not be using it as it
has proved terribly unstable - Asterisk segfaults on every voice mail
message although the message is successfully deliver to my email inbox -
but I thought I should report it. Here are the errors from the Asterisk
console:
-- Executing [...@client1-internal:4] VoiceMail(SIP/1001-ac0566e8,
2...@default,u) in new stack
-- SIP/1001-ac0566e8 Playing 'vm-theperson.gsm' (language 'en')
[Jun 19 14:51:38] NOTICE[28930]: channel.c:2860 __ast_read: Dropping
incompatible voice frame on SIP/1001-ac0566e8 of format ulaw since our native
format has changed to 0x2 (gsm)
-- SIP/1001-ac0566e8 Playing 'digits/2.gsm' (language 'en')
-- SIP/1001-ac0566e8 Playing 'digits/1.gsm' (language 'en')
-- SIP/1001-ac0566e8 Playing 'digits/0.gsm' (language 'en')
-- SIP/1001-ac0566e8 Playing 'vm-isunavail.gsm' (language 'en')
-- SIP/1001-ac0566e8 Playing 'vm-intro.gsm' (language 'en')
[Jun 19 14:51:48] WARNING[28930]: app_voicemail.c:2704 check_quota: Mailstream
not available for mailbox: INBOX
-- SIP/1001-ac0566e8 Playing 'beep.gsm' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/210/tmp/umh157
format: wav49, 0x2c05b798
-- x=1, open writing: /var/spool/asterisk/voicemail/default/210/tmp/umh157
format: gsm, 0x2c05c1c8
-- x=2, open writing: /var/spool/asterisk/voicemail/default/210/tmp/umh157
format: wav, 0x2c08d788
-- User ended message by pressing #
-- SIP/1001-ac0566e8 Playing 'auth-thankyou.gsm' (language 'en')
== Parsing '/var/spool/asterisk/voicemail/default/210/INBOX/msg.txt':
== Found
[Jun 19 14:51:54] ERROR[28930]: app_voicemail.c:2309 mm_log: IMAP Error: Server
disables LOGIN, no recognized SASL authenticator
[Jun 19 14:51:54] ERROR[28930]: app_voicemail.c:2068 init_mailstream: Can't
connect to imap server
{zimbra.mycompany.com:143/imap/notls/user...@mycompany.com}inbox
[Jun 19 14:51:54] ERROR[28930]: app_voicemail.c:1819 imap_store_file: Could not
initialize mailstream for
== Parsing '/var/spool/asterisk/voicemail/default/210/INBOX/msg.txt':
== Found
Segmentation fault
voicemail.conf looked like this:
; IMAP voice mail storage
imapserver=zimbra.mycompany.com
;imapport=143
;expungeonhangup=yes
;imapfolder=INBOX
;imapflags=notls
;authuser=...@mycompany.com
;authpassword=didn'tworkanyway
210 = 6370,John Sullivan,,imapuser...@mycompany.com
I also tried without authuser and setting passwords individually.
By the way, how does one disable IMAP storage? Without the IMAP
settings, I keep getting:
[Jun 19 15:04:48] WARNING[29032]: app_voicemail.c:1628 messagecount:
IMAP user not set for mailbox 210
I did not see a module for IMAP storage. It would seem strange that I
would have to recompile. Thanks - John
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
Making Christianity intelligible to secular society
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Re: [asterisk-users] Asterisk + mySQL
jonas kellens escribió: On Thu, 2009-06-18 at 11:52 -0500, Tilghman Lesher wrote: In modules.conf: noload = cdr_csv.so Are there other modules I need to load or unload ?? asterisk*CLI module show like cdr Module Description Use Count cdr_addon_mysql.so MySQL CDR Backend0 app_setcdruserfield.so CDR user field apps 0 func_cdr.soCDR dialplan function0 app_cdr.so Tell Asterisk to not maintain a CDR for 0 cdr_manager.so Asterisk Manager Interface CDR Backend 0 app_forkcdr.so Fork The CDR into 2 separate entities0 cdr_csv.so Comma Separated Values CDR Backend 0 cdr_custom.so Customizable Comma Separated Values CDR 0 8 modules loaded asterisk*CLI module show like odbc Module Description Use Count 0 modules loaded asterisk*CLI module show like sql Module Description Use Count cdr_addon_mysql.so MySQL CDR Backend0 app_addon_sql_mysql.so Simple Mysql Interface 0 res_config_mysql.soMySQL RealTime Configuration Driver 0 3 modules loaded modules.conf : autoload=yes noload=pbx_gtkconsole.so load=res_musiconhold.so load=cdr_addon_mysql.so noload=chan_alsa.so Why is there a /res_mysql.conf/ and a /cdr_mysql.conf/ ?? They both look alike... There's no other modules you need to load/unload. To disable CSV CDR recording just add what Tilghman told you into modules.conf. cdr_mysql.conf is for MySQL CDR backend database settings. res_mysql.conf is for MySQL Asterisk Realtime Architecture (ARA) backend database settings. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone 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] IMAP voice mail storage
John A. Sullivan III schrieb: By the way, how does one disable IMAP storage? I did not see a module for IMAP storage. It would seem strange that I would have to recompile. Sadly you have to recompile. Disable voicemail IMAP storage in `make menuselect`. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] Nagios under *
On Fri, 19 Jun 2009, Sriram wrote:
I tried your script :
STATUS=$(sudo asterisk -rnx pri show span 1\
| awk '/Status/ {print $3}'\
)
if [ Up, == ${STATUS} ]
thenecho PRI UP
exit 0
elseecho PRI DOWN
exit 2
fi
but still i get PRI down in the Nagios web interface while if i execute
this command from command line i get PRI UP...i m really going mad..did
a clean install again but still same problem..
Iv;e also given permission of 777 to the script
Always a bad idea and a clear indication of a newbie -- sorry.
So, let's think about this. It runs when you (probably running as root --
also AABIAACIOAN) run it from a shell, but not when Nagios runs it. So,
what's the difference:
) What username does Nagios run the script as?
) Is that user authorized to run Asterisk as root (or whatever username is
running Asterisk) in /etc/sudoers?
Be sure to use visudo to make changes to /etc/sudoers. Also, at some
point, sudo introduced requiretty which broke a lot of my cron scripts.
If you have requiretty set in sudoers, try commenting it out.
) What PATH does the script have when run by the Nagios process?
) Are there any permissions issues on the directories in the path to the
script?
Not having ever run Nagios, I'm shooting from the hip a bit. I'm guessing
these commands may shed some light:
) Get the username running Nagios.
ps -aef | grep --ignore-case nagios
) What output do you get from a command like:
cd /tmp/
sudo -u username-running-Nagios full-path-of-script
Thanks in advance,
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Switchvox HA options
On Fri, Jun 19, 2009 at 1:02 PM, Bob Pierce pier...@westmancom.com wrote: What are the HA options for Switchvox systems? Is it possible to set up redundant systems with DRBD? I know on the digium website they talk about Optional cold spare failover What does this mean? Is this an active spare ready for some sort of automated failover? Thanks for you help, Bob A Cold Spare generally means you buy two identical boxes and only plug one in, hence the Cold. You keep it in the rack but turned off. It is configured exactly the same way as the Hot box and you would need a daily (or whatever suits you) backup on the network, tape, or some other media to bring everything up to speed. It is not a Hot Spare which would be heartbeat or something that could take a few ms. to take over. I am sure you could do it with SwitchVox if you hosted the DB on a separate server and setup heartbeat and some scripts to change the from IP address to the virtual heartbeat address instead of the real IP of the NIC. There is some trickery when it comes to heartbeat since a machine taking over the active role will most likely by default receive traffic on it's heartbeat virtual IP but reply on it's real IP so the phones get confused and see it as unsolicited traffic, or otherwise reject it, or could manifest in one way audio oddities between boxen. That of course would unsupported since even having remote phones or a Switchvoxen open to the interweb. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.6 and mISDN
Hi on the list, does anyone of you have experience with asterisk 1.6 and mISDN, pri primarily? Thanks in advance Regards, Christophorus ___ -- 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] SIP Silence Suppression?
We're using Asterisk 1.6.1. When our SIP clients have silence suppression turned on, it's a problem for many apps. Is there a workaround for this in Asterisk? Other than turning silence suppression off in the SIP client, is there anything I can do on the Asterisk side to make things work again? Basically, Asterisk will often not send any audio to the client, until it receives an audio packet from the client, which is not going to happen when the client itself is silent (and when it has silence suppression enabled). I know this is an old problem for Asterisk, but I would be surprised to learn that after so long, it's still unresolved. Thank you, Bryan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Silence Suppression?
Just turn CNG on the phone and it should be fine ;-) On Fri, Jun 19, 2009 at 6:08 PM, Bryan Field-Elliot bryan+asterisk-us...@nextalarm.com bryan%2basterisk-us...@nextalarm.comwrote: We're using Asterisk 1.6.1. When our SIP clients have silence suppression turned on, it's a problem for many apps. Is there a workaround for this in Asterisk? Other than turning silence suppression off in the SIP client, is there anything I can do on the Asterisk side to make things work again? Basically, Asterisk will often not send any audio to the client, until it receives an audio packet from the client, which is not going to happen when the client itself is silent (and when it has silence suppression enabled). I know this is an old problem for Asterisk, but I would be surprised to learn that after so long, it's still unresolved. Thank you, Bryan ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] newbie questions
I have an Asterisknow.org CD. When I boot up, it seems ready for me to choose update, console, etc. I'm assuming I need to do something at the CLI prompt. Is there a tutorial that would take me from loading CD to making first test call? Computer is Dell Optiplex GX260 50GB free disk space 1.5GB RAM P4 processor external mic speakers Skype is on board, and would be good to use it, if possible. If I want to use Skype, do I need anything additional? Would it be better to install CD on my hard drive? Any help appreciated. Tom ___ -- 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
