Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2
Hi, Again I stuck up with OpenSuse 12.1 also, Installed OpenSuse 12.1 (32bit) version in virtualbox. Downloaded Asterisk 1.8.15. Installed, installation went fine. While starting Asterisk, it hangs here, *Asterisk Dynamic Loader Starting:* * == Parsing '/etc/asterisk/modules.conf': == Found* *[Aug 23 14:56:14] NOTICE[19340]: loader.c:1133 load_modules: 186 modules will be loaded.* any my linux machine uname -a output is below, *Linux linux-w6le.site 3.1.0-1.2-default #1 SMP Thu Nov 3 14:45:45 UTC 2011 (187dde0) i686 i686 i386 GNU/Linux* * * Any suggestion would be much appreciated. Regards, Gopal. On Tue, Aug 21, 2012 at 11:24 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Ok Thanks Bryant, let me try with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 7:46 PM, Bryant Zimmerman brya...@zktech.comwrote: I have the current version of 8.x and 10.x on systems. I am using OpenSuse 12.1, We are working on getting a 12.2 boxs up just running out of time. Asterisk on all of our boxes are complied from source. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- *From*: Gopalakrishnan N gopalakrishnan...@gmail.com *Sent*: Monday, August 20, 2012 10:11 AM *To*: Bryant Zimmerman brya...@zktech.com *Subject*: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 It's really glad that asterisk is installed at your machine in open suse. Can you let me know which version you are using and the architecture. Regards. On Aug 20, 2012 6:22 PM, Bryant Zimmerman brya...@zktech.com wrote: I compile from source.. Sent from my Verizon Wireless Phone - Reply message - From: Gopalakrishnan N gopalakrishnan...@gmail.com Date: Mon, Aug 20, 2012 8:15 am Subject: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com From the forum I understand OpenSuse 12.2 is pre-relase and better to use OpenSuse 12.1. Lets check with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Its really weird working with OpenSuse. I am not sure how others are using with OpenSuse. Through Yast also I tried to install Asterisk package, it didn't find. Now I am clueless to work with OpenSuse. Regards. On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi Patrick, Thanks for your suggestion, even though I added my hostname in the /etc/hosts, still the problem persists. Also I tried to install in OpenSuse 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like hanging at modules while starting Asterisk. Regards, Gopal. Please do not top post and properly trim your replies. Have you made sure that on the OpenSuse box your DNS is configured properly? You should be able to lookup your IP address/FQDN both ways. So for example 192.168.1.1 (replace with your IP adres) should resolve in your.box.com (replace with your FQDN) and vice versa your.box.comshould resolve into 192.168.1.1. See man dig or man nslookup for commands that can do DNS lookups. Regards, Patrick -- _ -- 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] CDR default table specification?
Andrew, thank you very much, I like especially https://wiki.asterisk.org/wiki/display/AST/MySQL+CDR+Backend as it's quite official :-) Dumb question, AST = asterisk or is it by chance just a part of asterisk or sth. like this? 2012/8/20 Andrew White and...@computersforall.com.au Hey Stefan, ** ** Have you had a look at http://www.voip-info.org/wiki/view/Asterisk+cdr+odbc? Voip-info.org isn’t official, but it’s a pretty good site. ** ** Have a look under “Setting up the CDR Database/Table”. ** ** There’s also the AST: ** ** https://wiki.asterisk.org/wiki/display/AST/MSSQL+CDR+Backend https://wiki.asterisk.org/wiki/display/AST/MySQL+CDR+Backend ** ** Good luck! ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Stefan at WPF *Sent:* Monday, 20 August 2012 6:51 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] CDR default table specification? ** ** I am using adaptive ODBC and I would like to create a table for the default CDR fields build into Asterisk. I managed to find several third party resources, but no official resource including the table specification for the default CDR fields. Where can I officially find this table specification, therefore data types and lengths etc. for each field? ** ** Thank you very much :-) -- _ -- 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] Asterisk 1.8 and 11
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan Sent: Wednesday, August 22, 2012 8:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 and 11 - Original Message - From: Danny Nicholas da...@debsinc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, August 22, 2012 2:47:00 PM Subject: Re: [asterisk-users] Asterisk 1.8 and 11 That's the theory. 11 is supposed to be 1.8 EOL version with some new tweaks. Keep in mind that 11 is officially a beta product, so if you're going to eat your own dog food 1.8 is probably the best option for now. So, that's not exactly true. Each year, a new major version of Asterisk is made from the Asterisk trunk. During that year, a focus is put onto different topics for major project/new feature development. For Asterisk 10, a focus was put on architectural changes (media architecture overhaul, utilization of the Bridging API in ConfBridge, etc.) Asterisk 11 focused more on stability and end user experience, and stayed away from major architectural refactorings. So, its not really accurate to say that 11 is 1.8 EOL version. Each version of Asterisk builds on the previous, with a different focus on the major projects for that version. For example, the media architecture of Asterisk 10 exists in Asterisk 11, as does ConfBridge, T.38 Gateway, etc. Asterisk 11 is in beta. The more people that help us test it out the better it will be when it is released! Note that you can find more information on Asterisk versions and their lifetimes on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions You can find more information about upgrading to Asterisk 11 on the wiki as well: https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 And you can also find out more about the new features in Asterisk 11 here: https://wiki.asterisk.org/wiki/display/AST/New+in+11 Note that we're still working on the documentation for the new features, so expect so more pages to show up there in the future. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org Too busy to say checked out from trunk so I said EOL. Since I use 10.X, 1.8 (and 1.6) were EOL for me out of the box. -- _ -- 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] TE110P Wildcard does not work with Ubuntu 12.04 server
Op 22-08-12 12:09, Shitian Long schreef: I am trying to setup TE110P wildcard on a PBX running ubuntu 12.04 server edition. I followed the procedure from http://docs.digium.com/misc/ADL_quickstart.pdf step by step. During the process of installing dahdi-linux-complete I got following warnings: root@ubuntu:/usr/local/src/dahdi-linux-complete-2.6.1+2.6.1# make perl: warning: Setting locale failed. perl: warning: Please check that your locale settings: LANGUAGE = en_US:en, LC_ALL = (unset), LC_CTYPE = UTF-8, LANG = en_US.UTF-8 are supported and installed on your system. perl: warning: Falling back to the standard locale (C). Frist of, I am wondering if this error matters? Second question, after installation process complete, and reboot the machine I got the following error, when machine boot up: Loading DAHDI hardware modules: wcte11xp: error I think the TE110P card is no properly loaded. I try to confirm my thought by using root@ubuntu:~# dahdi_tool There is no interface listed on the table. I am wondering if anyone got idea about this issue. Thanks. longst -- _ -- 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 Please try the command lspci and see if the card is mentioned in the results. Regards. Michel. -- _ -- 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] Asterisk 1.6 / voicemail / final voice auth-thankyou
Hi, voicemail plays after hitting # as final file auth-thankyou. Is there any possibility to change this behaviour? Custom soundfile or disable it perhaps? Thanks for your answer(s)! -Thorsten- -- _ -- 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 hangs while starting in OpenSuse 12.2
Have you tried vmware or hyper-v as your host. I have had issues with OpenSuse 12.x with Virtual Box. Asterisk not starting was one of them. Also in a virtual env I found that I had to alwyas build asterisk from source to make things work don't know why but that was the mix that worked for me. I moved to Hyper-V. OpenSuse 12.x as a VM is kind of a black art with asterisk for some reason. Once you get it working it works great. You have to watch how your virtual nic's are setup that can really mess with you as well. But virtual box was a no go for me never spent the time to figure out why. I took the path of least resistance. Thanks Bryant From: Gopalakrishnan N gopalakrishnan...@gmail.com Sent: Thursday, August 23, 2012 5:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi, Again I stuck up with OpenSuse 12.1 also, Installed OpenSuse 12.1 (32bit) version in virtualbox. Downloaded Asterisk 1.8.15. Installed, installation went fine. While starting Asterisk, it hangs here, Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': == Found [Aug 23 14:56:14] NOTICE[19340]: loader.c:1133 load_modules: 186 modules will be loaded. any my linux machine uname -a output is below, Linux linux-w6le.site 3.1.0-1.2-default #1 SMP Thu Nov 3 14:45:45 UTC 2011 (187dde0) i686 i686 i386 GNU/Linux Any suggestion would be much appreciated. Regards, Gopal. On Tue, Aug 21, 2012 at 11:24 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Ok Thanks Bryant, let me try with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 7:46 PM, Bryant Zimmerman brya...@zktech.com wrote: I have the current version of 8.x and 10.x on systems. I am using OpenSuse 12.1, We are working on getting a 12.2 boxs up just running out of time. Asterisk on all of our boxes are complied from source. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Gopalakrishnan N gopalakrishnan...@gmail.com Sent: Monday, August 20, 2012 10:11 AM To: Bryant Zimmerman brya...@zktech.com Subject: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 It's really glad that asterisk is installed at your machine in open suse. Can you let me know which version you are using and the architecture. Regards. On Aug 20, 2012 6:22 PM, Bryant Zimmerman brya...@zktech.com wrote: I compile from source.. Sent from my Verizon Wireless Phone - Reply message - From: Gopalakrishnan N gopalakrishnan...@gmail.com Date: Mon, Aug 20, 2012 8:15 am Subject: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com From the forum I understand OpenSuse 12.2 is pre-relase and better to use OpenSuse 12.1. Lets check with OpenSuse 12.1. Regards. On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Its really weird working with OpenSuse. I am not sure how others are using with OpenSuse. Through Yast also I tried to install Asterisk package, it didn't find. Now I am clueless to work with OpenSuse. Regards. On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi Patrick, Thanks for your suggestion, even though I added my hostname in the /etc/hosts, still the problem persists. Also I tried to install in OpenSuse 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like hanging at modules while starting Asterisk. Regards, Gopal. Please do not top post and properly trim your replies. Have you made sure that on the OpenSuse box your DNS is configured properly? You should be able to lookup your IP address/FQDN both ways. So for example 192.168.1.1 (replace with your IP adres) should resolve in your.box.com (replace with your FQDN) and vice versa your.box.com should resolve into 192.168.1.1. See man dig or man nslookup for commands that can do DNS lookups. Regards, Patrick -- _ -- 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
[asterisk-users] RemoveQueueMember and realtime queues
Hello, using asterisk 1.8.11.1 using realtime queues When trying to remove a queue member, I get the following : -- Executing [122@from-TESTCORP:2] RemoveQueueMember(SIP/testcorp5-000c, testcorpq1,SIP/testcorp7) in new stack WARNING[18788]: app_queue.c:5653 rqm_exec: Unable to remove interface from queue 'testcorpq1': 'SIP/testcorp7' is not a dynamic member How can one remove a queue member when using realtime queues ? Extra question : adding a queue member to a queue, will AddQueueMember work ? -- Executing [122@from-TESTCORP:5] AddQueueMember(SIP/testcorp5-000c, testcorpq1,SIP/testcorp7) in new stack WARNING[18788]: app_queue.c:5708 aqm_exec: Unable to add interface 'SIP/testcorp7' to queue 'testcorpq1': Already there -- Executing [122@from-TESTCORP:6] NoOp(SIP/testcorp5-000c, AQMSTATUS = MEMBERALREADY) in new stack Kind regards, Jonas. -- _ -- 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] RemoveQueueMember and realtime queues
On 08/23/2012 10:05 AM, Jonas Kellens wrote: Hello, using asterisk 1.8.11.1 using realtime queues When trying to remove a queue member, I get the following : -- Executing [122@from-TESTCORP:2] RemoveQueueMember(SIP/testcorp5-000c, testcorpq1,SIP/testcorp7) in new stack WARNING[18788]: app_queue.c:5653 rqm_exec: Unable to remove interface from queue 'testcorpq1': 'SIP/testcorp7' is not a dynamic member The answer is right there: you can't remove the member because it's not a dynamic member. A dynamic member is one that was added with AddQueueMember. If you want to remove a static member (that is, one that was defined in queues.conf, or in your realtime database) then you have to remove it by removing it from that configuration. RemoveQueueMember can't do it. You can verify that a member is dynamic by observing (dynamic) next to the agent's name in queue show. Perhaps you can set persistentmembers=yes in queues.conf [general] section (or your realtime database, if you like), and then dynamic members will persist across asterisk restarts. Then get rid of your static members. Then implement extensions that agents can call to log in and out, that call AddQueueMember and RemoveQueueMember. -- _ -- 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] quick questions on version 10
With MeetMe there was a MeetMeAdmin() that could KILL a conference. How do I do that with the new ConfBridge. I dont see a way to kill the conference. There also used to be a core show channels concise, this is deprecated. What is the correct way to do this now and get all the information? Jerry -- _ -- 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] sip trunk failing to register causes sip phones to become unreachable
Hi, I have only seen this problem when using sipgate SIP trunks which actually register. If the ADSL connection goes down that the sip trunk uses, the sip phones registered locally become unreachable. This happens on any 1.6.x or 1.8 version of asterisk I've tried. Is there a work around that doesn't involve putting an opensips server between the asterisk server and the sip trunk? Thanks. Regards, John -- _ -- 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] quick questions on version 10
[snip] There also used to be a core show channels concise, this is deprecated. What is the correct way to do this now and get all the information? The AMI action CoreShowChannels deprecated the CLI concise command because the output of the AMI action is extensible without breaking existing systems. The CLI command is not extensible without breaking existing systems. Richard -- _ -- 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 analog phones
On Monday, August 20, 2012, Ikka.vertika ikka.vert...@mitrakreasindo.com wrote: u need an analog telephone adaptor to connect your analog phone with switch hub. but the device is rather expensive. GrandStream 8 port ATA is about $200.00 which is much less than a single new SIP phone. Much much less expensive than new. The Cisco ATA is about $400.00 but may only give 4 analog telephone ports (FXS). Sent from Samsung Mobile Noam Birnbaum n...@maccentricsolutions.com wrote: Hi folks, A client wants to keep their old Inter-Tel KTS analog phones for budget reasons. Two questions: 1. How could they use these with FreePBX? 2. Would they be losing any features that they currently have with their analog PBX? Thanks! Noam Birnbaum Mac Daddy http://www.maccentricsolutions.com 877.luv.macs x666 tweet @noamb Tech support — 877.luv.macs or supp...@maccentricsolutions.com -- _ -- 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] Japanese voicefiles
Hi Guys, I've a few questions around languages I'm on 1.4.18 (old yes I know, but upgradings not an option just yet). I've downloaded the gsm Japanese files from ftp://ftp.voip-info.jp/asterisk/sounds/ and put them in place I've found that when I switch to jp, and play some of my own voicefiles in Background and Playback, that it chooses the /var/lib/asterisk/sounds/jp folder files and plays them, but, voicemail doesn't seem to do this, instead it picks the English files (although the debug output says its using 'jp'). I've seen references to a patch for this, but any idea where the patch is ? Secondly, I'm trying to open .gsm files in Audacity (in particular these japanese ones, so I can confirm they are Japanese), but I just can't get the audio format right (audacity 2) Open RAW: Encoding ? Byte Order ? Channels: mono, Sample rate 8000hz. I've set my Pref Quality defaults to 8000hz and 16-bit, but I think that's only for recording. Anyone know the correct setting? I've been able to play them in Quicktime so I think they're ok, I just want to see them in Audacity. Thanks, Adrian -- _ -- 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] Japanese voicefiles
On 23/8/12 5:26 pm, Adrian Marsh wrote: I've a few questions around languages I'm on 1.4.18 (old yes I know, but upgradings not an option just yet). I've downloaded the gsm Japanese files from ftp://ftp.voip-info.jp/asterisk/sounds/ and put them in place I've found that when I switch to jp, and play some of my own voicefiles in Background and Playback, that it chooses the /var/lib/asterisk/sounds/jp folder files and plays them, but, voicemail doesn't seem to do this, instead it picks the English files (although the debug output says its using 'jp'). We use a similar method to play British English sounds - they're in /var/lib/asterisk/sounds/britishfemale - and voicemail seems to pick them up correctly. Have you made sure to specify language = jp in the relevant places? You need to do it in whichever module is originating the call to Voicemail - so if it's a SIP client, you'd probably do it in sip.conf, but if it's an incoming call, it's probably easier to do it in extensions.conf. FWIW, this is also using an old version - 1.4.21, so unless something's changed between .18 and .21, it should work with your setup. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Easy to install CDR-Viewer?
Hello, just wondering if there is any easy to install CDR viewer? Easy meaning install some package (debian system) and that's it. Had some problems installing CDR-Stats, FreePBX also seems to be a longer task for setting up. Isn't there a simple (productive :p) solution? Thanks :-) Best regards Stefan -- _ -- 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] Easy to install CDR-Viewer?
- Original Message - just wondering if there is any easy to install CDR viewer? Easy meaning install some package (debian system) and that's it. Had some problems installing CDR-Stats, FreePBX also seems to be a longer task for setting up. Isn't there a simple (productive :p) solution? CDR-stat is about as easy as it gets, assuming you can setup a basic LAMP stack, and edit a config file or two (database parameters for CDRs). What issues are you having with that installation? --Tim -- _ -- 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] Easy to install CDR-Viewer?
- Original Message - - Original Message - just wondering if there is any easy to install CDR viewer? Easy meaning install some package (debian system) and that's it. Had some problems installing CDR-Stats, FreePBX also seems to be a longer task for setting up. Isn't there a simple (productive :p) solution? CDR-stat is about as easy as it gets, assuming you can setup a basic LAMP stack, and edit a config file or two (database parameters for CDRs). Caveat... I'm referring to the 'old' CDR-stats which was simple PHP based, not the 'new fangled' CDR-stats these young punks are using... :D --Tim -- _ -- 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] quick questions on version 10
The AMI action CoreShowChannels deprecated the CLI concise command because the output of the AMI action is extensible without breaking existing systems. The CLI command is not extensible without breaking existing systems. Richard, Thanks - I tried the CoreShowChannels AMI and it says: Response: Follows Privilege: Command No such command 'CoreShowChannels' (type 'core show help CoreShowChannels' for other possible commands) --END COMMAND-- In my manager.conf I have read = system,call,command,agent,user,reporting write = system,call,command,agent,user,originate,reporting Did I miss something? Jerry -- _ -- 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] quick questions on version 10
I thought this was discussed and it was going to be left in? Sent from my iPhone On 23/08/2012, at 2:30 PM, Jerry Geis ge...@pagestation.com wrote: The AMI action CoreShowChannels deprecated the CLI concise command because the output of the AMI action is extensible without breaking existing systems. The CLI command is not extensible without breaking existing systems. Richard, Thanks - I tried the CoreShowChannels AMI and it says: Response: Follows Privilege: Command No such command 'CoreShowChannels' (type 'core show help CoreShowChannels' for other possible commands) --END COMMAND-- In my manager.conf I have read = system,call,command,agent,user,reporting write = system,call,command,agent,user,originate,reporting Did I miss something? Jerry -- _ -- 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] quick questions on version 10
The AMI action CoreShowChannels deprecated the CLI concise command because the output of the AMI action is extensible without breaking existing systems. The CLI command is not extensible without breaking existing systems. Richard, Thanks - I tried the CoreShowChannels AMI and it says: Response: Follows Privilege: Command No such command 'CoreShowChannels' (type 'core show help CoreShowChannels' for other possible commands) --END COMMAND-- In my manager.conf I have read = system,call,command,agent,user,reporting write = system,call,command,agent,user,originate,reporting Did I miss something? Yes. It is: Action: CoreShowChannels not Action: Command Command: CoreShowChannels It is an actual AMI action not a CLI command. Richard -- _ -- 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] GotoIf redirection to label not working correctly
I run a hotdesking system based on the example from Asterisk: The Definitive
Guide. Calls come into the [hotdesk] context, which verifies the phone has a
logged in user and sends the call to users,${EXTEN},1 if there is a user logged
in. The [users] context then includes several other contexts for
internal/external call handling, as follows:
[users]
include = internal
include = dummyextensions
switch = DUNDi/dundi-peer
Internal office calls should get caught by the include = internal, and run
through the [internal] context as follows:
[internal]
exten = _3XX,1,NoOp()
same = n,Set(E=${EXTEN})
same = n,Set(${E}_VMCONTEXT=${HOTDESK_INFO(vmcontext,${E})})
same = n,Set(USER_LOCATION=${HOTDESK_USER_STATUS(${E})})
same = n,GotoIf($[${ODBCROWS} 1]?notloggedin)
same =
n,Dial(SIP/${USER_LOCATION},20,wWU(blf-begincall^${E}^INUSE)b(blf-begincall^s^1(${E}^RINGING)))
same = n,Voicemail(${E}@${${E}_VMCONTEXT},b)
same = n,Hangup()
same = n(notloggedin),Set(LOGGED_OFF=1)
same = n,Voicemail(${E}@${${E}_VMCONTEXT},u)
same = n,Hangup()
In both Asterisk 10 and Asterisk 11, the GotoIf does not work under the
circumstances above, giving me the following error and hanging up the call:
[Aug 23 15:17:35] WARNING[3558][C-0565]: pbx.c:11799 pbx_parseable_goto:
Priority 'notloggedin' must be a number 0, or valid label
I can work around the issue with any of the following:
- Change the GotoIf to point to internal,${EXTEN},notloggedin
- Change the GotoIf to point to 9
- Comment out the DUNDi switch in [users]
- Unload the pbx_dundi.so module
In the latter two cases, the call redirects to the notloggedin priority label
within [internal],${EXTEN} without me changing the GotoIf - as long as the
DUNDi switch is not active in [users]. With the DUNDi switch active, I get the
warning message above and the call hangs up.
Is there something I'm doing wrong in my config? Is this basically expected
behavior that I need to adjust around?
Thank you,
Noah Engelberth
MetaLINK Technologies
--
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Re: [asterisk-users] sip trunk failing to register causes sip phones to become unreachable
On Aug 23, 2012, at 10:30 AM, John Cahill j...@dmcip.com wrote: I have only seen this problem when using sipgate SIP trunks which actually register. If the ADSL connection goes down that the sip trunk uses, the sip phones registered locally become unreachable. This happens on any 1.6.x or 1.8 version of asterisk I've tried. Is there a work around that doesn't involve putting an opensips server between the asterisk server and the sip trunk? This is a common issue that I've seen many times. The problem has to do with DNS cache look-ups and timeouts. What typically solves it for me is to install a local cacheing-only DNS server on the asterisk box and point the resolvers on the asterisk box to itself. This will only solve the issue of an internet outage causing the sip phones to stop working, and only for as long as the local cache stays relevant. If there is a power outage that takes out both the asterisk server and the internet, and your asterisk box comes up but your internet doesn't, this won't work. -- Thanks, Warren Selby, dCAP -- _ -- 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] sip trunk failing to register causes sip phones to become unreachable
Adding the IPs of ALL local interfaces to /etc/hosts has helped solve this issue for me in the past. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Thursday, August 23, 2012 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip trunk failing to register causes sip phones to become unreachable On Aug 23, 2012, at 10:30 AM, John Cahill j...@dmcip.com wrote: I have only seen this problem when using sipgate SIP trunks which actually register. If the ADSL connection goes down that the sip trunk uses, the sip phones registered locally become unreachable. This happens on any 1.6.x or 1.8 version of asterisk I've tried. Is there a work around that doesn't involve putting an opensips server between the asterisk server and the sip trunk? This is a common issue that I've seen many times. The problem has to do with DNS cache look-ups and timeouts. What typically solves it for me is to install a local cacheing-only DNS server on the asterisk box and point the resolvers on the asterisk box to itself. This will only solve the issue of an internet outage causing the sip phones to stop working, and only for as long as the local cache stays relevant. If there is a power outage that takes out both the asterisk server and the internet, and your asterisk box comes up but your internet doesn't, this won't work. -- Thanks, Warren Selby, dCAP -- _ -- 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] GotoIf redirection to label not working correctly
I run a hotdesking system based on the example from Asterisk: The
Definitive Guide. Calls come into the [hotdesk] context, which
verifies the phone has a logged in user and sends the call to
users,${EXTEN},1 if there is a user logged in. The [users] context
then includes several other contexts for internal/external call
handling, as follows:
[users]
include = internal
include = dummyextensions
switch = DUNDi/dundi-peer
Internal office calls should get caught by the include = internal,
and run through the [internal] context as follows:
[internal]
exten = _3XX,1,NoOp()
same = n,Set(E=${EXTEN})
same = n,Set(${E}_VMCONTEXT=${HOTDESK_INFO(vmcontext,${E})})
same = n,Set(USER_LOCATION=${HOTDESK_USER_STATUS(${E})})
same = n,GotoIf($[${ODBCROWS} 1]?notloggedin)
same =
n,Dial(SIP/${USER_LOCATION},20,wWU(blf-begincall^${E}^INUSE)b(blf-begincall^s^1(${E}^RINGING)))
same = n,Voicemail(${E}@${${E}_VMCONTEXT},b)
same = n,Hangup()
same = n(notloggedin),Set(LOGGED_OFF=1)
same = n,Voicemail(${E}@${${E}_VMCONTEXT},u)
same = n,Hangup()
In both Asterisk 10 and Asterisk 11, the GotoIf does not work under
the circumstances above, giving me the following error and hanging
up the call:
[Aug 23 15:17:35] WARNING [3558][C-0565]: pbx.c : 11799
pbx_parseable_goto : Priority 'notloggedin' must be a number 0, or
valid label
I can work around the issue with any of the following:
- Change the GotoIf to point to internal,${EXTEN},notloggedin
- Change the GotoIf to point to 9
- Comment out the DUNDi switch in [users]
- Unload the pbx_dundi.so module
In the latter two cases, the call redirects to the notloggedin
priority label within [internal],${EXTEN} without me changing the
GotoIf – as long as the DUNDi switch is not active in [users]. With
the DUNDi switch active, I get the warning message above and the
call hangs up.
Is there something I’m doing wrong in my config? Is this basically
expected behavior that I need to adjust around?
I am wondering if it will work if you changed the label name from
notloggedin to something else. There might be a name conflict
when using the DUNDi switch. Otherwise, it looks like a bug.
Richard
--
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Re: [asterisk-users] GotoIf redirection to label not working correctly
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Thursday, August 23, 2012 5:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] GotoIf redirection to label not working correctly
I run a hotdesking system based on the example from Asterisk: The
Definitive Guide. Calls come into the [hotdesk] context, which
verifies the phone has a logged in user and sends the call to
users,${EXTEN},1 if there is a user logged in. The [users] context
then includes several other contexts for internal/external call
handling, as follows:
[users]
include = internal
include = dummyextensions
switch = DUNDi/dundi-peer
Internal office calls should get caught by the include = internal,
and run through the [internal] context as follows:
[internal]
exten = _3XX,1,NoOp()
same = n,Set(E=${EXTEN})
same = n,Set(${E}_VMCONTEXT=${HOTDESK_INFO(vmcontext,${E})})
same = n,Set(USER_LOCATION=${HOTDESK_USER_STATUS(${E})})
same = n,GotoIf($[${ODBCROWS} 1]?notloggedin) same =
n,Dial(SIP/${USER_LOCATION},20,wWU(blf-begincall^${E}^INUSE)b(blf-begi
ncall^s^1(${E}^RINGING))) same =
n,Voicemail(${E}@${${E}_VMCONTEXT},b)
same = n,Hangup()
same = n(notloggedin),Set(LOGGED_OFF=1) same =
n,Voicemail(${E}@${${E}_VMCONTEXT},u)
same = n,Hangup()
In both Asterisk 10 and Asterisk 11, the GotoIf does not work under
the circumstances above, giving me the following error and hanging up
the call:
[Aug 23 15:17:35] WARNING [3558][C-0565]: pbx.c : 11799
pbx_parseable_goto : Priority 'notloggedin' must be a number 0, or
valid label
I can work around the issue with any of the following:
- Change the GotoIf to point to internal,${EXTEN},notloggedin
- Change the GotoIf to point to 9
- Comment out the DUNDi switch in [users]
- Unload the pbx_dundi.so module
In the latter two cases, the call redirects to the notloggedin
priority label within [internal],${EXTEN} without me changing the
GotoIf – as long as the DUNDi switch is not active in [users]. With
the DUNDi switch active, I get the warning message above and the call
hangs up.
Is there something I’m doing wrong in my config? Is this basically
expected behavior that I need to adjust around?
I am wondering if it will work if you changed the label name from notloggedin
to something else. There might be a name conflict when using the DUNDi switch.
Otherwise, it looks like a bug.
Richard
--
I've tried not_logged_in, notlogged in, and a few other short labels. The
dundi switch only has numeric extensions it's announcing from the other side.
Thank you,
Noah Engelberth
MetaLINK Technologies
--
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Re: [asterisk-users] GotoIf redirection to label not working correctly
- Original Message - From: Noah Engelberth n...@directlinkcomputers.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, August 23, 2012 4:30:21 PM Subject: Re: [asterisk-users] GotoIf redirection to label not working correctly I am wondering if it will work if you changed the label name from notloggedin to something else. There might be a name conflict when using the DUNDi switch. Otherwise, it looks like a bug. Richard -- I've tried not_logged_in, notlogged in, and a few other short labels. The dundi switch only has numeric extensions it's announcing from the other side. quote - Change the GotoIf to point to 9 /quote If this worked when the notloggedin label was at position 9 but not when it is at position 10, then it almost has to be a bug. Not being able to find a named label when you have more than 10 priorities in an extension seems sub-optimal. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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] quick questions on version 10
- Original Message - From: Matt Riddell (lists) li...@venturevoip.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, August 23, 2012 1:52:00 PM Subject: Re: [asterisk-users] quick questions on version 10 I thought this was discussed and it was going to be left in? It hasn't been removed. There are no current plans to remove it. It is deprecated, as basing a script on it is probably not a wise idea, and there are better mechanisms available for third parties to get at that information. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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] GotoIf redirection to label not working correctly
- Original Message - From: Matthew Jordan mjor...@digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, August 23, 2012 4:37:53 PM Subject: Re: [asterisk-users] GotoIf redirection to label not working correctly - Original Message - From: Noah Engelberth n...@directlinkcomputers.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, August 23, 2012 4:30:21 PM Subject: Re: [asterisk-users] GotoIf redirection to label not working correctly I am wondering if it will work if you changed the label name from notloggedin to something else. There might be a name conflict when using the DUNDi switch. Otherwise, it looks like a bug. Richard -- I've tried not_logged_in, notlogged in, and a few other short labels. The dundi switch only has numeric extensions it's announcing from the other side. If this worked when the notloggedin label was at position 9 but not when it is at position 10, then it almost has to be a bug. Not being able to find a named label when you have more than 10 priorities in an extension seems sub-optimal. Disregard! Richard pointed out what you actually had done as testing. (Its still a bug however) -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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] Asterisk hangs while starting in OpenSuse 12.2
On Thu, 2012-08-23 at 15:01 +0530, Gopalakrishnan N wrote: Hi, Again I stuck up with OpenSuse 12.1 also, Installed OpenSuse 12.1 (32bit) version in virtualbox. Downloaded Asterisk 1.8.15. Installed, installation went fine. Have you tried the versions from the OBS? Or perhaps a virtualbox issue? Its notorious for vapourizing cpu-cycles... 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
[asterisk-users] Bug or Not
I think I found another bug, but please let me know if there is a workaround, since my bugs never get fixed. In safe_asterisk, there is a section where the script executes some startup scripts, located in /etc/asterisk/startup.d However, when you restart asterisk with core restart now or you go ahead and kill the asterisk process, these scripts that are so important do not get executed. The question is: where in the safe_asterisk script can I copy the whole loop so in any event, if Asterisk gets restarted, these scripts get properly executed. Otherwise there is no way to ensure that the finite-state machine starts from an known start point. -- _ -- 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] CDR default table specification?
No worries - took me quite a while to find myself ages ago! Whoops, I just meant to say the wiki, not the AST. I assume the AST in the url refers to Asterisk. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan at WPF Sent: Thursday, 23 August 2012 9:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CDR default table specification? Andrew, thank you very much, I like especially https://wiki.asterisk.org/wiki/display/AST/MySQL+CDR+Backend as it's quite official :-) Dumb question, AST = asterisk or is it by chance just a part of asterisk or sth. like this? 2012/8/20 Andrew White and...@computersforall.com.aumailto:and...@computersforall.com.au Hey Stefan, Have you had a look at http://www.voip-info.org/wiki/view/Asterisk+cdr+odbc? Voip-info.org isn't official, but it's a pretty good site. Have a look under Setting up the CDR Database/Table. There's also the AST: https://wiki.asterisk.org/wiki/display/AST/MSSQL+CDR+Backend https://wiki.asterisk.org/wiki/display/AST/MySQL+CDR+Backend Good luck! From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan at WPF Sent: Monday, 20 August 2012 6:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] CDR default table specification? I am using adaptive ODBC and I would like to create a table for the default CDR fields build into Asterisk. I managed to find several third party resources, but no official resource including the table specification for the default CDR fields. Where can I officially find this table specification, therefore data types and lengths etc. for each field? Thank you very much :-) -- _ -- 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
