Re: [asterisk-users] Polycom SoundStation IP 6000 does not register
On Wed, Dec 21, 2016 at 7:50 AM, Yveswrote: > Hi Mark, > > yes, you are right... these are different VLANs > I configured the other phone to use the same IP (192.168.1.13)... and it > worked flawlessly... on the SAME Networkcable in the same plug... > so it must have something to do with the polycom phone config... remember... > when I use tcp the phone tries to register, but does not even try with > udp... > > thank you, > yves > I am a bit confused: is your problematic phone's IP 192.168.0.13 (what the error log is reporting below) or 192.168.1.13? > > Am 21.12.2016 um 13:34 schrieb Mark Wiater: > > Yves, > > Didn't you say that > > AsteriskServer: 192.168.1.211 > SIP-user: 165 > > ? > > On 12/21/2016 4:24 AM, Yves wrote: > > . It is sure for 100% that there is no firewall or something else mangeling > in between... another Hardphone works as expected using the same > Netzworkcable on the same Networkplug with UDP on Port 5060... > > > This other hardphone, what IP does it have? > > > 50.848|cfg |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask > 255.255.255.0 > > The line above suggests to me that your phone and your asterisk server are > on a different network, there has to be something that routes between those > two networks. Often what routes, can firewall. > > 000122.941|sip |4|03|Registration failed User: 165, Error Code:480 > Temporarily not available > > > > Mark > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Corrupted sqlite3 astdb back end
So I found out my astdb database is boink (asterisk 1.8.something): [Jul 15 06:42:28] ERROR[18769] res_config_sqlite.c: database disk image is malformed # first stop asterisk before doing this, and do it on a copy: sqlite3 ./sqlite.db .dump |less PRAGMA foreign_keys=OFF; BEGIN TRANSACTION; / ERROR: (26) file is encrypted or is not a database */ ROLLBACK; -- due to errors (END) Anything I can do to repair it? If not, how can I recreate it? What is stored in it (https://wiki.asterisk.org/wiki/display/AST/SQLite3+astdb+back-end was not particularly helpful)? -- _ -- 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 on OSX
On Fri, Apr 11, 2014 at 8:53 AM, Manu et...@manu-dpk.net wrote: Hi, Thank you all for your replies... 1. No sound for playback() dial plan function... Did you try Matt's suggestion? I have this error in my cli : execution error for playback I take that means you had the asterisk cli open on one terminal while doing tests. Maybe bump the debugging level up a bit? I set the path of audio files correctly. 2. No sound when I call for my GSM to asterisk. 3. No sound when I call from asterisk to other numbers... Thank you for your help... AMICALEMENT... __ Manu - E-Mail : et...@manu-dpk.net - TimeLine Twitter : @manudpk - Skype : manu-dpk Découvrez mes compositions musicales personnelles ainsi que mes podcasts : http://www.manu-dpk.net ___ Respectons l'environnement ! N'imprimer ce mail, uniquement si cela vous est nécessaire !!! Le 11 avr. 2014 à 12:33, Matt Behrens m...@zigg.com a écrit : On Apr 11, 2014, at 3:01 AM, Manu et...@manu-dpk.net wrote: Hi, I used asterisk on Debian7 and it was good experience. Now, i'm using osx on mac mini. I'd like to install asterisk 12. I tried to compile it and after lot of searches, I got it. All sip accounts log in. I can call but I haven't any sounds. - for IVR, - for voicemail, - for out and in calling... Please help me... Thank you in advance Last time I tried building Asterisk 11 on OS X, the res_timing_kqueue module wasn't working properly, IIRC. Switching to res_timing_pthread worked. I don't know how well that setup performs under load, though; I only run Asterisk on my MacBook for development (and have switched to doing so inside a Vagrant virtual machine anyway, which definitely stutters.) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6
On Fri, Apr 4, 2014 at 10:05 AM, Daniel Taylor dtay...@vocalabs.com wrote: I don't know what platform you are on, but if you are on Linux (and possibly BSD) you could use fail2ban to block them at the network interface. I second fail2ban. If you need some ideas to configure it, you can steal them from the freepbx setup. How many sip phones do you have outside your network? If few and in well-known IPs, consider limiting access to only those (and the sip provider you are using). On 04/04/2014 09:00 AM, motty cruz wrote: Hello All, my asterisk server is constantly under attack [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password is there a way to reject their registration after a three consecutive tries? Thanks, Call Send SMS Add to Skype You'll need Skype CreditFree via Skype -- Daniel Taylor VP OperationsVocal Laboratories, inc.dtay...@vocalabs.com http://www.vocalabs.com/(612)235-5711 -- _ -- 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] dahdi show channels no such command
On Thu, Dec 5, 2013 at 8:41 PM, Joseph Towery tech...@bellsouth.net wrote: I have followed the instructions in Asterisk The Definitive Guide 4th edition. Once I load DAHDI I run the dahdi show channels command and get no such command. I have setup all the conf files. I compiled DADHI prior to Asterisk. Any ideas? I am not sure. Could you provide more details? I just tried in my setup and it seems to work (though it tells me nothing exciting): [root@voip ~]# asterisk -r Asterisk 1.8.11-cert1, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Connected to Asterisk 1.8.11-cert1 currently running on voip (pid = 2575) Verbosity is at least 3 voip*CLI da dahdi data database voip*CLI dahdi show channels Chan Extension Context Language MOH Interpret BlockedState pseudodefaultdefault In Service voip*CLI quit [root@voip ~]# -- _ -- 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] What linux distro most popular for Asterisk
On Wed, Oct 16, 2013 at 9:48 AM, John Doe boogieman2...@gmail.com wrote: I've had great success with centos. Stand alone, VM, or asterisk Distro. On Oct 16, 2013 12:58 AM, Michelle Dupuis mdup...@ocg.ca wrote: Is there a recent survey of that Linux distro and version people are using for the Asterisk installations? I recall seeing a pie chart over a year ago (I think on a wiki but I can't find it again)also hoping for something more current. I suspect RH5 and RH6 are most popular...but I'm looking for facts I do believe AsteriskNow uses Centos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems sending log to rsyslog
On Thu, Sep 26, 2013 at 11:16 AM, Gareth Blades mailinglist+aster...@dns99.co.uk wrote: On 26/09/13 15:25, Mauricio Tavares wrote: So I have asterisk 1.8.23 and want to send my logs to rsyslog. I tell asterisk to use syslog in addition to messages: root@voip:~# tail -10 /etc/asterisk/logger.conf ;debug = debug console = notice,warning,error ;console = notice,warning,error,debug messages = notice,warning,error ;full = notice,warning,error,debug,verbose,dtmf,fax ;syslog keyword : This special keyword logs to syslog facility ; syslog.local0 = notice,warning,error ; root@voip:~# After reloading (asterisk -rx 'logger reload') the logger, it seems that Asterisk is happy: root@voip:~# asterisk -rx 'logger show channels' Channel Type StatusConfiguration --- --- syslog.local0 Syslog Enabled- NOTICE WARNING ERROR /var/log/asterisk/messages File Enabled- NOTICE WARNING ERROR Console Enabled- NOTICE WARNING ERROR root@voip:~# So I set rsyslog: root@voip:~# fgrep asterisk /etc/rsyslog.d/50-default.conf local0.* /var/log/asterisk/messages.log root@voip:~# and restart it. And then check the asterisk log directory: root@voip:~# ls -lh /var/log/asterisk/ total 3.7M drwxr-xr-x 2 asterisk asterisk 4.0K Jul 22 20:57 cdr-csv drwxr-xr-x 2 asterisk asterisk 4.0K Jun 28 14:16 cdr-custom -rw-rw 1 asterisk asterisk 252K Sep 26 09:37 messages -rw-rw 1 asterisk asterisk 248K Sep 22 05:14 messages.1 -rw-r- 1 syslog adm 0 Sep 26 06:47 messages.log -rw-rw 1 asterisk asterisk 118 Sep 26 10:07 queue_log root@voip:~# It does not seem like much is being written to messages.log compared to messages. Anything I missed? Have you checked the /var/log/asterisk directory permissions? I dont know how rsyslog is setup on your system but its possible it gets started as root, sees the destination file doesnt exist so creates it and sets the file permissions, and then drops down to running as the syslog user. At this point it doesnt have write permission to the /var/log/asterisk directory so cannot append to the file. And you were absolutely right: root@voip:~# sudo -u syslog touch /var/log/asterisk/my_nose touch: cannot touch `/var/log/asterisk/my_nose': Permission denied root@voip:~# ls -lhd /var/log/asterisk drwxr-xr-x 4 asterisk asterisk 4.0K Sep 26 10:10 /var/log/asterisk root@voip:~# getent group asterisk asterisk:x:114:www-data root@voip:~# So, I decided to be lazy and add syslog to the asterisk group: root@voip:~# id syslog uid=101(syslog) gid=103(syslog) groups=114(asterisk),103(syslog) root@voip:~# chmod g+w /var/log/asterisk root@voip:~# sudo -u syslog touch /var/log/asterisk/my_nose root@voip:~# Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Problems sending log to rsyslog
So I have asterisk 1.8.23 and want to send my logs to rsyslog. I tell asterisk to use syslog in addition to messages: root@voip:~# tail -10 /etc/asterisk/logger.conf ;debug = debug console = notice,warning,error ;console = notice,warning,error,debug messages = notice,warning,error ;full = notice,warning,error,debug,verbose,dtmf,fax ;syslog keyword : This special keyword logs to syslog facility ; syslog.local0 = notice,warning,error ; root@voip:~# After reloading (asterisk -rx 'logger reload') the logger, it seems that Asterisk is happy: root@voip:~# asterisk -rx 'logger show channels' Channel Type StatusConfiguration --- --- syslog.local0 Syslog Enabled- NOTICE WARNING ERROR /var/log/asterisk/messages File Enabled- NOTICE WARNING ERROR Console Enabled- NOTICE WARNING ERROR root@voip:~# So I set rsyslog: root@voip:~# fgrep asterisk /etc/rsyslog.d/50-default.conf local0.* /var/log/asterisk/messages.log root@voip:~# and restart it. And then check the asterisk log directory: root@voip:~# ls -lh /var/log/asterisk/ total 3.7M drwxr-xr-x 2 asterisk asterisk 4.0K Jul 22 20:57 cdr-csv drwxr-xr-x 2 asterisk asterisk 4.0K Jun 28 14:16 cdr-custom -rw-rw 1 asterisk asterisk 252K Sep 26 09:37 messages -rw-rw 1 asterisk asterisk 248K Sep 22 05:14 messages.1 -rw-r- 1 syslog adm 0 Sep 26 06:47 messages.log -rw-rw 1 asterisk asterisk 118 Sep 26 10:07 queue_log root@voip:~# It does not seem like much is being written to messages.log compared to messages. Anything I missed? -- _ -- 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] LUA
On Thu, Jul 18, 2013 at 10:29 AM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 07/18/2013 03:56 PM, jacob.e.mi...@l-3com.com wrote: I am attempting to setup my server to use Lua for the dialplan (extentions.lua), but I am unable to get the asterisk configure script to find the installation of Lua on my box. I have downloaded the Lua sources from the www.lua.org site, and I have installed via the “make linux install” command. I can execute lua scripts via the command line, but asterisk configure script is unable to find the installation of Lua. That's probably because Asterisk is not looking in /usr/local. I am on a closed network, so no access to the internet so I am not able to just install Lua using yum. You should have downloaded the lua RPMs to e.g. your laptop, then copy them to your Asterisk box with e.g. a USB stick and then install the Lua RPMs on your Asterisk box with: $ sudo yum install ./lua* You can find the CentOS 6.4 x86_64 Lua RPMs here: http://mirror.stanford.edu/yum/pub/centos/6.4/os/x86_64/Packages/ Is lua self-sufficient or does it require additional packages? 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] PCI Passthrough of T1 cards
On Wed, Jun 19, 2013 at 11:52 AM, Nick Khamis sym...@gmail.com wrote: Hello James, Thank you so much for your response. I should have chose my words carefully. PCI pass-through in terms of virtualization of devices and it's draw back are well know. I was leaning more towards near host performance virtualization using SR-IOV. I know I am late in the show, but what are the drawbacks as far as using Asterisk is concerned? This moves emphasis back to the production drivers of the interface card using virtual functions etc., and can provide near host performance. Rephrasing my question, are any of the T1 pci manufactures providing support for virtualization using SR-IOV and virutal functions? Kind Regards, Nick -- _ -- 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] app.c: No audio available on SIP
What is the log file trying to tell me? [Dec 3 04:09:26] WARNING[17902] app.c: No audio available on SIP/83.169.5.208-0108?? [Dec 3 04:09:28] WARNING[17907] app.c: No audio available on SIP/83.169.5.208-0109?? [Dec 3 04:09:30] WARNING[17912] app.c: No audio available on SIP/83.169.5.208-010a?? [Dec 3 04:09:32] WARNING[17918] app.c: No audio available on SIP/83.169.5.208-010b?? [Dec 3 04:09:34] WARNING[17923] app.c: No audio available on SIP/83.169.5.208-010c?? [Dec 3 04:09:36] WARNING[17928] app.c: No audio available on SIP/83.169.5.208-010d?? [Dec 3 04:09:38] WARNING[17933] app.c: No audio available on SIP/83.169.5.208-010e?? [Dec 3 04:09:54] WARNING[17992] app.c: No audio available on SIP/83.169.5.208-010f?? [Dec 3 04:09:56] WARNING[17997] app.c: No audio available on SIP/83.169.5.208-0110?? [Dec 3 04:09:58] WARNING[18002] app.c: No audio available on SIP/83.169.5.208-0111?? -- _ -- 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