Re: [asterisk-users] Trying to configure xorcom on Suse 11
On 27 Mar 2010, at 21:48, JD Austin wrote: Xorcom hardware uses three layers; you must resolve issues in the following order: 1. USB 2. Dahdi 3. Asterisk I suspect you're having trouble with the usb layer. Run lsusb It will display a line like this if the firmware isn't loaded: Bus 001 Device 004: ID e4e4:1161 If it is e4e4:1162 then the firmware is loaded. You can manually load the firmware like this: /usr/share/dahdi/xpp_fxloader load or /usr/share/dahdi/xpp_fxloader usb It seems to load (some) usb firmware ok, as you can see from the syslog, but I suspect it is loading the wrong version. I got e4e4:1164 (I think - I've lost contact with the box for the moment). Thanks for the explanation too. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- 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] Attended transfer and callerID updates forSiemens Openstage phones
Hi Kevin, is this feature implemented in the Business Version of Asterisk? Best reards, Loïc. On Thu, 2010-03-25 at 07:20 -0500, Kevin P. Fleming wrote: Loic Didelot wrote: I am testing the Openstage phones from Siemens but I can not find a solution on how to update the caller-id after a successful attended transfer. Of course, I mean an attended transfer by using the phones functionality, not something defined in asterisks features.conf. This is called 'Connected Party ID', and it isn't supported in any released version of Asterisk... but it is supported in SVN trunk and will be part of Asterisk 1.8. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying to configure xorcom on Suse 11
On Sat, Mar 27, 2010 at 04:48:41PM -0500, Tim Panton wrote: I'm having trouble getting a xorcom set up. A large part of the problem is that the box is a _long_ way away and I can't get to/at it easily, so while I could probably fix this in a few hours if the machine were in front of me, I'm struggling over a slow unreliable laggy link. Ok, enough whining from me. I have a new Xorcom plugged into the usb of a Suse 11 machine I built Dahdi from trunk (last thursday) # svn checkout http://svn.digium.com/svn/dahdi/linux/trunk dahdi-linux # svn checkout http://svn.digium.com/svn/dahdi/tools/trunk dahdi-tools dahdihardware -v sees the box but no spans. Generally '/etc/init.d/dahdi start' . Or more specifically, 'dahdi_registration on' . See also: http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_installation_scenarios -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Updating Asterisk and its use with MySQL
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! I'm using Asterisk 1.4.24.1 with dahdi-linux-2.1.0.4 and dahdi-tools-2.1.0.2 compiled by myself with the source code of the official site of the project. I would like to update to one more newer version. I suppose that the recommendable thing is to maintain me in branch 1.4, reason why in this case it would be 1.4.30 that I suppose that it will have several bugs fixed. Also I see that there are new versions of DADHI Linux and DAHDI Tools; 2.2.1.1 for both cases. I image DAHDI Complete package include both DAHDI Linux an DAHDI tools. For this package it is necessary to continue making the compilation separately? But going to the question to that I make mention in subject, which would be the procedure to update the versions of these software maintaining the configurations? It is correct to think that the procedure would be to stop the Asterisk server and DAHDI, and to follow the same steps for the compilation and installation but without doing make config? On the other hand, at this moment I'm testing with few extensiones on low scale, but my idea is to raise the test a little more 50 extensions. For this case I suppose that it is more efficient to work with a database management system (MySQL, for example) for the configurations instead of files. There is some procedure that can recommend to me to migrate the configurations in files to a DBMS? My idea is to continue making the configurations by hand at the moment, that it is the way that I used until now, to familiarize to me with the handling of Asterisk at lower level, without using a graphical interface, and in a later stage of the tests to take these configurations through something like FreePBX. What think of this form to think? Thanks in advance for your reply and recommendations. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkuvcg0ACgkQZpa/GxTmHTdPNQCeL5oBGnuhcvqj8Sw8cuvUOBA8 DIoAn03AkmpGKN0XY1lMrLZ87RA2fhj4 =EKwq -END PGP SIGNATURE- -- _ -- 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] Libtonezone
Trying to find out what the libtonezone shared object built with dahdi-tools is for, the default dahdi package installation from the Digium repo's pull it in, so when is it needed? Thanks, jlc -- _ -- 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] Updating Asterisk and its use with MySQL
Make sure not to do make samples or you will overwrite your .conf file. This is the important one to watch out for. You can save off your .conf files and then restore them or compare your files with the new ones to see if there are any important new settings. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Mar 28, 2010, at 8:19 AM, Daniel Bareiro wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! I'm using Asterisk 1.4.24.1 with dahdi-linux-2.1.0.4 and dahdi-tools-2.1.0.2 compiled by myself with the source code of the official site of the project. I would like to update to one more newer version. I suppose that the recommendable thing is to maintain me in branch 1.4, reason why in this case it would be 1.4.30 that I suppose that it will have several bugs fixed. Also I see that there are new versions of DADHI Linux and DAHDI Tools; 2.2.1.1 for both cases. I image DAHDI Complete package include both DAHDI Linux an DAHDI tools. For this package it is necessary to continue making the compilation separately? But going to the question to that I make mention in subject, which would be the procedure to update the versions of these software maintaining the configurations? It is correct to think that the procedure would be to stop the Asterisk server and DAHDI, and to follow the same steps for the compilation and installation but without doing make config? On the other hand, at this moment I'm testing with few extensiones on low scale, but my idea is to raise the test a little more 50 extensions. For this case I suppose that it is more efficient to work with a database management system (MySQL, for example) for the configurations instead of files. There is some procedure that can recommend to me to migrate the configurations in files to a DBMS? My idea is to continue making the configurations by hand at the moment, that it is the way that I used until now, to familiarize to me with the handling of Asterisk at lower level, without using a graphical interface, and in a later stage of the tests to take these configurations through something like FreePBX. What think of this form to think? Thanks in advance for your reply and recommendations. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkuvcg0ACgkQZpa/GxTmHTdPNQCeL5oBGnuhcvqj8Sw8cuvUOBA8 DIoAn03AkmpGKN0XY1lMrLZ87RA2fhj4 =EKwq -END PGP SIGNATURE- -- _ -- 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] Updating Asterisk and its use with MySQL
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Jim. On Sun, 28 Mar 2010, Jim Dickenson wrote: Make sure not to do make samples or you will overwrite your .conf file. This is the important one to watch out for. You can save off your .conf files and then restore them or compare your files with the new ones to see if there are any important new settings. I had thought that make config was what I would have to avoid. Which is the difference? does make config create the init scripts and make samples the example configuration files? Do these two makes have the same behavior for Asterisk and DAHDI? I have understood that make config in DAHDI Tools is the one that creates both the configuration files and init scripts. When I compiled the version that I'm using at the moment of DAHDI Linux only I used make and make install without using make samples or make config. Are also generated configuration files with DAHDI Linux? Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkuvjioACgkQZpa/GxTmHTcNIgCfZ1PEUqz/3kjGRTa0ECO97jSH 53YAni5ICLJGEL2U1Hcwc2hKsDUMYH6V =Dp0r -END PGP SIGNATURE- -- _ -- 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] Updating Asterisk and its use with MySQL
-- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Mar 28, 2010, at 10:19 AM, Daniel Bareiro wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Jim. On Sun, 28 Mar 2010, Jim Dickenson wrote: Make sure not to do make samples or you will overwrite your .conf file. This is the important one to watch out for. You can save off your .conf files and then restore them or compare your files with the new ones to see if there are any important new settings. I had thought that make config was what I would have to avoid. Which is the difference? does make config create the init scripts and make samples the example configuration files? Yes, make config installs /etc/init.d/asterisk on Linux systems and does the appropriate chkconfig steps so will start on boot while make samples installs the .conf files in, by default, /etc/asterisk. Do these two makes have the same behavior for Asterisk and DAHDI? I have understood that make config in DAHDI Tools is the one that creates both the configuration files and init scripts. There is no make config for dahdi. I think /etc/dahdi files do not get overwritten if they are there already. When I compiled the version that I'm using at the moment of DAHDI Linux only I used make and make install without using make samples or make config. Are also generated configuration files with DAHDI Linux? I think if you are installing dahdi complete from source you do make all and make install and make config Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkuvjioACgkQZpa/GxTmHTcNIgCfZ1PEUqz/3kjGRTa0ECO97jSH 53YAni5ICLJGEL2U1Hcwc2hKsDUMYH6V =Dp0r -END PGP SIGNATURE- -- _ -- 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] problem with polarity reverse
On Sat, Mar 27, 2010 at 08:02:02PM +0200, Justas Gulbinskas wrote: Hi, I have a problem with polarity reverse I use asterisk 1.4.30 linux kernel version 2.6.27 dahdi version 2.2.1 and analog card is Sangoma a400 with fxo ports this is my config [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no answeronpolarityswitch=yes and then i call from sip to mobile over gsm gw (nokia 32) which have a polarity reverse i pick up the mobile phone in sip phone i hear that polarity revers was but at the in asterisk core show channels verbose Channel Context ExtensionPrio State Application
[asterisk-users] Trying to configure xorcom on Suse 11
I've never seen e4e4:1164 before.
What does this output?:
lsusb|sed -e 's/:/ /g'| grep e4e4| awk '{print astribank_tool -n -D
/proc/bus/usb/$2/$4}'| bash
reset the astribank:
#(if you use freepbx)
amportal stop
#(if you start asterisk that way)
/etc/init.d/asterisk stop
/etc/init.d/dahdi stop
/usr/share/dahdi/xpp_fxloader reset
#give it time
sleep 30
/usr/share/dhadi/xpp_fxloader load
#(you should see e4e4:1162)
lsusb
#(you should see the hardware here)
dahdi_hardware -v
#presuming you have /etc/dahdi/system.conf right this will work
/etc/init.d/dahdi start
#if you use freepbx
amportal start
#or
/etc/init.d/asterisk start
--
JD Austin
Twin Geckos Technology Services LLC
j...@twingeckos.com
Voice: 480.288.8195x201
Fax: 480.406.6753
http://www.twingeckos.com
--
_
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Re: [asterisk-users] Updating Asterisk and its use with MySQL
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 - -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Jim. On Sun, 28 Mar 2010, Jim Dickenson wrote: Make sure not to do make samples or you will overwrite your .conf file. This is the important one to watch out for. You can save off your .conf files and then restore them or compare your files with the new ones to see if there are any important new settings. I had thought that make config was what I would have to avoid. Which is the difference? does make config create the init scripts and make samples the example configuration files? Yes, make config installs /etc/init.d/asterisk on Linux systems and does the appropriate chkconfig steps so will start on boot while make samples installs the .conf files in, by default, /etc/asterisk. Perfect. Do these two makes have the same behavior for Asterisk and DAHDI? I have understood that make config in DAHDI Tools is the one that creates both the configuration files and init scripts. There is no make config for dahdi. I think /etc/dahdi files do not get overwritten if they are there already. Hmmm... nevertheless I have documented this procedure in my Dokuwiki of the time that I made the installation and compilation: # tar xvzf dahdi-linux-2.1.0.4.tar.gz # tar xvzf dahdi-tools-2.1.0.2.tar.gz ~/Asterisk/dahdi-linux-2.1.0.4# make ~/Asterisk/dahdi-linux-2.1.0.4# make install ~/Asterisk/dahdi-tools-2.1.0.2# ./configure ~/Asterisk/dahdi-tools-2.1.0.2# make menuselect # In order to select a customized configuration ~/Asterisk/dahdi-tools-2.1.0.2# make ~/Asterisk/dahdi-tools-2.1.0.2# make install ~/Asterisk/dahdi-tools-2.1.0.2# make config # In order to install scripts and config files When I compiled the version that I'm using at the moment of DAHDI Linux only I used make and make install without using make samples or make config. Are also generated configuration files with DAHDI Linux? I think if you are installing dahdi complete from source you do make all and make install and make config Thanks. I will consider it if I install this package of DAHDI. Thanks for your reply. Regards, Daniel - -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkuvqhAACgkQZpa/GxTmHTfqMQCfT2V7RR4JMFp/EpH4J0F8Tfk9 3SYAoJJLhKfdznWoYddRNhmmyN1ygzJm =Q6s+ - -END PGP SIGNATURE- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkuvqh8ACgkQZpa/GxTmHTfoygCfZtRoPj8ieJjWVtsIqPFIk5Q/ 4QQAnjWRKkOJls9dFVwVM0IQORkmDIPd =YxoR -END PGP SIGNATURE- -- _ -- 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] Updating Asterisk and its use with MySQL
My idea is to continue making the configurations by hand at the moment, that it is the way that I used until now, to familiarize to me with the handling of Asterisk at lower level, without using a graphical interface, and in a later stage of the tests to take these configurations through something like FreePBX. What think of this form to think? I would suggest trying Digium's GUI first and then FreePBX since the first one I find it more readable. You'll find out eventually that there's no easy way to migrate from pure command line to a GUI, but you'll learn a lot in the meantime. Have Fun! Alyed 2010/3/28 Daniel Bareiro daniel-lis...@gmx.net -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 - -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Jim. On Sun, 28 Mar 2010, Jim Dickenson wrote: Make sure not to do make samples or you will overwrite your .conf file. This is the important one to watch out for. You can save off your .conf files and then restore them or compare your files with the new ones to see if there are any important new settings. I had thought that make config was what I would have to avoid. Which is the difference? does make config create the init scripts and make samples the example configuration files? Yes, make config installs /etc/init.d/asterisk on Linux systems and does the appropriate chkconfig steps so will start on boot while make samples installs the .conf files in, by default, /etc/asterisk. Perfect. Do these two makes have the same behavior for Asterisk and DAHDI? I have understood that make config in DAHDI Tools is the one that creates both the configuration files and init scripts. There is no make config for dahdi. I think /etc/dahdi files do not get overwritten if they are there already. Hmmm... nevertheless I have documented this procedure in my Dokuwiki of the time that I made the installation and compilation: # tar xvzf dahdi-linux-2.1.0.4.tar.gz # tar xvzf dahdi-tools-2.1.0.2.tar.gz ~/Asterisk/dahdi-linux-2.1.0.4# make ~/Asterisk/dahdi-linux-2.1.0.4# make install ~/Asterisk/dahdi-tools-2.1.0.2# ./configure ~/Asterisk/dahdi-tools-2.1.0.2# make menuselect # In order to select a customized configuration ~/Asterisk/dahdi-tools-2.1.0.2# make ~/Asterisk/dahdi-tools-2.1.0.2# make install ~/Asterisk/dahdi-tools-2.1.0.2# make config # In order to install scripts and config files When I compiled the version that I'm using at the moment of DAHDI Linux only I used make and make install without using make samples or make config. Are also generated configuration files with DAHDI Linux? I think if you are installing dahdi complete from source you do make all and make install and make config Thanks. I will consider it if I install this package of DAHDI. Thanks for your reply. Regards, Daniel - -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkuvqhAACgkQZpa/GxTmHTfqMQCfT2V7RR4JMFp/EpH4J0F8Tfk9 3SYAoJJLhKfdznWoYddRNhmmyN1ygzJm =Q6s+ - -END PGP SIGNATURE- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkuvqh8ACgkQZpa/GxTmHTfoygCfZtRoPj8ieJjWVtsIqPFIk5Q/ 4QQAnjWRKkOJls9dFVwVM0IQORkmDIPd =YxoR -END PGP SIGNATURE- -- _ -- 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] Updating Asterisk and its use with MySQL
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Sun, 28 Mar 2010, Alyed wrote: My idea is to continue making the configurations by hand at the moment, that it is the way that I used until now, to familiarize to me with the handling of Asterisk at lower level, without using a graphical interface, and in a later stage of the tests to take these configurations through something like FreePBX. What think of this form to think? I would suggest trying Digium's GUI first and then FreePBX since the first one I find it more readable. You'll find out eventually that there's no easy way to migrate from pure command line to a GUI, but you'll learn a lot in the meantime. I didn't know that there was Digium's GUI. It is FLOSS? I was looking for in the site of Digium in the download section, but the unique thing that I saw that it speaks of a GUI is AsteriskNow, that in fact it is a complete distribution of GNU/Linux. You talked about to the GUI provided by AsteriskNow? Because if is this case, I don't believe that it is very practical. When I spoke of GUI was referring to a separated component to install over which already one had running. As far as the use of Asterisk with a DBMS (MySQL, for example), do you know some document or reference where indicate the steps to follow to migrate from config files? Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkuvu8oACgkQZpa/GxTmHTdFVACePM0WaIfeHQmM+w8cpLuGGt/5 XSAAoI+YrC+9Y91ElRhFBrxAG6XVxEyh =e4iN -END PGP SIGNATURE- -- _ -- 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] Updating Asterisk and its use with MySQL
Yes I'm talking about Asterisk Now's GUI and yes, you can just install this component. google for Asterisk Gui 2.0 and you'll find plenty of info. Regarding the DB I can't help you here, maybe someone else can. Alyed 2010/3/28 Daniel Bareiro daniel-lis...@gmx.net -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Sun, 28 Mar 2010, Alyed wrote: My idea is to continue making the configurations by hand at the moment, that it is the way that I used until now, to familiarize to me with the handling of Asterisk at lower level, without using a graphical interface, and in a later stage of the tests to take these configurations through something like FreePBX. What think of this form to think? I would suggest trying Digium's GUI first and then FreePBX since the first one I find it more readable. You'll find out eventually that there's no easy way to migrate from pure command line to a GUI, but you'll learn a lot in the meantime. I didn't know that there was Digium's GUI. It is FLOSS? I was looking for in the site of Digium in the download section, but the unique thing that I saw that it speaks of a GUI is AsteriskNow, that in fact it is a complete distribution of GNU/Linux. You talked about to the GUI provided by AsteriskNow? Because if is this case, I don't believe that it is very practical. When I spoke of GUI was referring to a separated component to install over which already one had running. As far as the use of Asterisk with a DBMS (MySQL, for example), do you know some document or reference where indicate the steps to follow to migrate from config files? Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkuvu8oACgkQZpa/GxTmHTdFVACePM0WaIfeHQmM+w8cpLuGGt/5 XSAAoI+YrC+9Y91ElRhFBrxAG6XVxEyh =e4iN -END PGP SIGNATURE- -- _ -- 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] Libtonezone
You could read the source code, but based on it's name I would say it is a library responsible for zone specific tone generation. Many parts of the world have different tone patterns than the U.S. and Asterisk is used worldwide. A better question is, why are you concerned by it? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] on behalf of Joseph L. Casale [jcas...@activenetwerx.com] Sent: Sunday, March 28, 2010 9:13 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Libtonezone Trying to find out what the libtonezone shared object built with dahdi-tools is for, the default dahdi package installation from the Digium repo's pull it in, so when is it needed? Thanks, jlc -- _ -- 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] SIP/2.0 403 Forbidden
You need to ask your carrier what you are not sending them that they would like. It's usually a fromdomain or authname. - Original Message - From: Aaron chen To: Asterisk Users Mailing List - Non-Commercial Discussion ; Asterisk Developers Mailing List Sent: Friday, March 26, 2010 09:22 Subject: [asterisk-users] SIP/2.0 403 Forbidden hi,all when i send a call to other server use SIP trunk, i got this below, --- SIP read from 222.46.18.52:5060 --- SIP/2.0 403 Forbidden what's rong with is? Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 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. = == Parsing '/etc/asterisk/asterisk.conf': Found Connected to Asterisk 1.4.21.2 currently running on gd-branch (pid = 3145) Verbosity is at least 3 -- Executing [015921256...@from-internal:1] Set(SIP/75002-b7705298, MOHCLASS=none) in new stack -- Executing [015921256...@from-internal:2] Macro(SIP/75002-b7705298, user-callerid|SKIPTTL|) in new stack -- Executing [...@macro-user-callerid:1] Set(SIP/75002-b7705298, AMPUSER=75002) in new stack -- Executing [...@macro-user-callerid:2] GotoIf(SIP/75002-b7705298, 0?report) in new stack -- Executing [...@macro-user-callerid:3] ExecIf(SIP/75002-b7705298, 1|Set|REALCALLERIDNUM=75002) in new stack -- Executing [...@macro-user-callerid:4] Set(SIP/75002-b7705298, AMPUSER=75002) in new stack -- Executing [...@macro-user-callerid:5] Set(SIP/75002-b7705298, AMPUSERCIDNAME=75002) in new stack -- Executing [...@macro-user-callerid:6] GotoIf(SIP/75002-b7705298, 0?report) in new stack -- Executing [...@macro-user-callerid:7] Set(SIP/75002-b7705298, AMPUSERCID=75002) in new stack -- Executing [...@macro-user-callerid:8] Set(SIP/75002-b7705298, CALLERID(all)=75002 75002) in new stack -- Executing [...@macro-user-callerid:9] ExecIf(SIP/75002-b7705298, 0|Set|CHANNEL(language)=) in new stack -- Executing [...@macro-user-callerid:10] GotoIf(SIP/75002-b7705298, 1?continue) in new stack -- Goto (macro-user-callerid,s,19) -- Executing [...@macro-user-callerid:19] NoOp(SIP/75002-b7705298, Using CallerID 75002 75002) in new stack -- Executing [015921256...@from-internal:3] Set(SIP/75002-b7705298, _NODEST=) in new stack -- Executing [015921256...@from-internal:4] Macro(SIP/75002-b7705298, record-enable|75002|OUT|) in new stack -- Executing [...@macro-record-enable:1] GotoIf(SIP/75002-b7705298, 1?check) in new stack -- Goto (macro-record-enable,s,4) -- Executing [...@macro-record-enable:4] AGI(SIP/75002-b7705298, recordingcheck|20100326-141638|1269584198.62) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20100326-141638|1269584198.62: Outbound recording enabled. recordingcheck|20100326-141638|1269584198.62: CALLFILENAME=OUT75002-20100326-141638-1269584198.62 -- AGI Script recordingcheck completed, returning 0 -- Executing [...@macro-record-enable:999] MixMonitor(SIP/75002-b7705298, /var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav||) in new stack -- Executing [...@macro-record-enable:1000] Set(SIP/75002-b7705298, RecordingFileName=/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav) in new stack -- Executing [...@macro-record-enable:1001] NoOp(SIP/75002-b7705298, /var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav) in new stack -- Executing [...@macro-record-enable:1002] Set(SIP/75002-b7705298, CDR(userfield)=/var/spool/asterisk/monitor/gd-branch/gdbranchOUT75002-20100326-141638-1269584198.62.wav) in new stack -- Executing [015921256...@from-internal:5] Macro(SIP/75002-b7705298, dialout-trunk|7|015921256331||) in new stack -- Executing [...@macro-dialout-trunk:1] Set(SIP/75002-b7705298, DIAL_TRUNK=7) in new stack -- Executing [...@macro-dialout-trunk:2] GosubIf(SIP/75002-b7705298, 0?sub-pincheck|s|1) in new stack -- Executing [...@macro-dialout-trunk:3] GotoIf(SIP/75002-b7705298, 0?disabletrunk|1) in new stack -- Executing [...@macro-dialout-trunk:4] Set(SIP/75002-b7705298, DIAL_NUMBER=015921256331) in new stack -- Executing [...@macro-dialout-trunk:5] Set(SIP/75002-b7705298, DIAL_TRUNK_OPTIONS=Ttr) in new stack -- Executing [...@macro-dialout-trunk:6] Set(SIP/75002-b7705298,
[asterisk-users] 24 FXS Port Voip Gateway and Asterisk
Hi, Can anyone recommend a 24 fxs port voip gateway that has worked well with asterisk? I have a couple of analog handsets that I want to hookup to my asterisk server? Any tested and tried product recommendations are welcome. Thanks. Best Regards, Joseph -- _ -- 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] 24 FXS Port Voip Gateway and Asterisk
I have used ones from Rhino and no complains. On 2010-03-28 9:00 PM, Joseph Begumisa j.begum...@gmail.com wrote: Hi, Can anyone recommend a 24 fxs port voip gateway that has worked well with asterisk? I have a couple of analog handsets that I want to hookup to my asterisk server? Any tested and tried product recommendations are welcome. Thanks. Best Regards, Joseph -- _ -- 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] Failed to play transfer sound! during attended transfer
Dear sir, Thanks for your reply. We have tested in another phone like Bria(2.4.3 buid 50906) with same phenomenon. But we are getting same error Failed to play transfer sound! during attended transfer. Is there anything which causes this problem? And we are not facing this problem first time. Before we faced in this problem occasionally. But recently, this problem occurs frequently. Is there any other problem or any other prerequisite for this problem? Or is it the problem of asterisk? How we can overcome this problem ? Please give us solution. Thanks in advance Nahar On Sat, Mar 27, 2010 at 1:33 AM, Alyed al...@vivoxie.com wrote: so doesn't looks like overload Could it be a problem with the firmware of your softphones? Have you been using some new phones lately? someone else in a different thread pointed on attended transfer bugs with SNOM phones. We are eagerly waiting for your solution. Hope we can help but don't so much pressure on me or the listers :) Alyed 2010/3/26 kamrun nahar bina bina...@gmail.com Dear sir, Thanks for your reply. our memory size is 4GB. concurrent calls no : 30. Our memory condition is below : Cpu(s): 0.3%us, 0.7%sy, 0.0%ni, 98.5%id, 0.0%wa, 0.1%hi, 0.3%si, 0.0%st Mem: 4147888k total, 3986540k used, 161348k free,76852k buffers Swap: 2031608k total, 56k used, 2031552k free, 3170396k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 23160 root 15 0 440m 415m 5688 S 4.3 10.3 398:13.93 asterisk Our disk space condition is below: FilesystemSize Used Avail Use% Mounted on /dev/mapper/VolGroup00-LogVol00 901G 245G 610G 29% / /dev/sda1 99M 18M 77M 19% /boot tmpfs 2.0G 0 2.0G 0% /dev/shm We are eagerly waiting for your solution. Thanks in advance. Nahar On Fri, Mar 26, 2010 at 2:32 PM, Alyed al...@vivoxie.com wrote: If you didn't have this problem before I'll check up for any changes lately (i suppose you have done so, but ask this just to be safe) I see you have lots of agents and also lots of hard disk space, so I guess disk space is not an issue. Please check it anyway. how many concurrent calls you have? 2 GB in RAM seems little against 600 registered agents. Alyed 2010/3/25 kamrun nahar bina bina...@gmail.com Dear sir, We have been using asterisk for 4 years. Now we have got problems which occurs during the attended transfer. But we are not always getting this problem. Sometimes it happens. But now we cannot understand why this is happening? problem is:Failed to play transfer sound! The log of asterisk is as like as followings: [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Invalid SIP message - rejected , no callid, len 366 [Mar 25 17:58:40] DEBUG[13617] audiohook.c: Write factory 0x1bfa000c was pretty quick last time, waiting for them. [Mar 25 17:58:40] DEBUG[13579] audiohook.c: Read factory 0x1bd61fe0 was pretty quick last time, waiting for them. [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Setting SIP_ALREADYGONE on dialog 5bd1acee539e699b4f5e79c94a348...@113.34.235.8 [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Received bye, issuing owner hangup [Mar 25 17:58:40] DEBUG[13617] audiohook.c: Write factory 0x1bfa000c was pretty quick last time, waiting for them. [Mar 25 17:58:40] WARNING[13591] res_features.c: Failed to play transfer sound! Our system is as like as: The number of User agent is: 1650 The number of Actual registered user agent is: 600 Our System configuration is : IBM X3550 CPU: Intel(R) Xeon(R) CPU X5460 @ 3.16GHz Memory: 2GB HDD: 3.5 SATA 1TB x 2 version of asterisk: 1.4.23.1 Asterisk and the User-Agent is connected through the Internet. ..And Is there any solution to solve this problem? We have investigated in several places but we cannot find out the reason? We need this solution very urgently. We are eagerly waiting for reply. Thanks in advance Nahar -- _ -- 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
Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk
At 19:55 3/28/2010, Joseph Begumisa wrote: Hi, Can anyone recommend a 24 fxs port voip gateway that has worked well with asterisk? Â I have a couple of analog handsets that I want to hookup to my asterisk server? Â Any tested and tried product recommendations are welcome. Â Thanks. You may want to buy 12 Linksys PAP2s. Have used 8 port, and 24 port boxes in the past. The problem with this is if just one port goes bad, you need to replace the whole unit. Also, you need to keep spares on hand. From getting burned in the past, I would recommend using multiples of these 2 port ATAs, and keeping a few spares on hand. Best Regards, Joseph -- _ -- 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] Cisco 7960 become UNREACHABLE behind pix firewall
Alyed wrote: From: http://www.voip-info.org/wiki/view/Asterisk+sip+qualify If you turn on *qualify* in the configuration of a SIP device in sip.confhttp://www.voip-info.org/wiki/view/Asterisk+config+sip.conf, asterisk will send a SIP OPTIONShttp://www.voip-info.org/wiki/view/SIP+method+optionscommand regularly to check that the device is still online. If the device does not answer within the configured (or default) period (in ms) Asterisk considers the device off-line for future calls. This status can be checked by the SIPPEER functionhttp://www.voip-info.org/wiki/view/Asterisk+func+sippeer, and inversely this function will only provide status information for peers which have *qualify=yes*. My guess is that your Nat/firewall is closing the connection after some time the phone is idle, so this way Asterisk will make sure to always have communication going trhough that connection so your NAT/firewall won't just close it. Sorry, should have mentioned that all these phones have qualify=yes and nat=yes in sip.conf. Thanks. -- James On Sat, Mar 27, 2010 at 8:17 AM, James Lamanna jlama...@gmail.com wrote: Hi, I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall. After some period of time, asterisk says that some of them are unreachable, and the phones lose their registration. The only way to make the phones recover is to clear the NAT translation tables for the phones on the PIX (clear xlate...) Does anyone know how to fix this? As you can imagine, it is quite annoying. And it does not happen to all the phones either. sip fixup is enabled on the PIX phone config parts: nat_enable : 1 nat_received_processing : 0 nat_address: [public ip of PIX] -- _ -- 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] Trying to configure xorcom on Suse 11
On 28 Mar 2010, at 10:13, Tzafrir Cohen wrote: On Sat, Mar 27, 2010 at 04:48:41PM -0500, Tim Panton wrote: I'm having trouble getting a xorcom set up. A large part of the problem is that the box is a _long_ way away and I can't get to/at it easily, so while I could probably fix this in a few hours if the machine were in front of me, I'm struggling over a slow unreliable laggy link. Ok, enough whining from me. I have a new Xorcom plugged into the usb of a Suse 11 machine I built Dahdi from trunk (last thursday) # svn checkout http://svn.digium.com/svn/dahdi/linux/trunk dahdi-linux # svn checkout http://svn.digium.com/svn/dahdi/tools/trunk dahdi-tools dahdihardware -v sees the box but no spans. Generally '/etc/init.d/dahdi start' . Or more specifically, 'dahdi_registration on' . See also: http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_installation_scenarios I've must be missing something here - this is what I see now. sh-4.0# dahdi_hardware -v usb:001/020 xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware LABEL=[usb:X1037246] connect...@usb-:00:1a.7-4 Shouldn't I see spans ??? I think the box (I've never seen it, but I know what I asked for) has 8fxs+8fxo+2E1 . Thanks, Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- 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] 24 FXS Port Voip Gateway and Asterisk
On Sun, 28 Mar 2010, Joseph Begumisa wrote: Can anyone recommend a 24 fxs port voip gateway that has worked well with asterisk? I have a couple of analog handsets that I want to hookup to my asterisk server? Any tested and tried product recommendations are welcome. Thanks. Adtran channel banks are a great trailing edge technology. You can get them off Ebay for pennies on the original dollar and they are built like a tank. (voip gateway is not very specific. If you meant SIP or IAX, you might want to specify which.) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying to configure xorcom on Suse 11
This is what I get on my unit but my is working: dahdi_hardware -v Failed running '/usr/sbin/astribank_tool': at /usr/lib/perl5/site_perl/5.8.8/Dahdi/Xpp/Mpp.pm line 181. usb:001/003 xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware LABEL=[usb:X1036340] connect...@usb-:00:1a.0-1.4 XBUS-00/XPD-00: T1 (24) Span 1 DAHDI-SYNC XBUS-00/XPD-10: FXS (8) Span 2 XBUS-00/XPD-20: FXO (8) Span 3 -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Mar 28, 2010, at 7:16 PM, Tim Panton wrote: On 28 Mar 2010, at 10:13, Tzafrir Cohen wrote: On Sat, Mar 27, 2010 at 04:48:41PM -0500, Tim Panton wrote: I'm having trouble getting a xorcom set up. A large part of the problem is that the box is a _long_ way away and I can't get to/at it easily, so while I could probably fix this in a few hours if the machine were in front of me, I'm struggling over a slow unreliable laggy link. Ok, enough whining from me. I have a new Xorcom plugged into the usb of a Suse 11 machine I built Dahdi from trunk (last thursday) # svn checkout http://svn.digium.com/svn/dahdi/linux/trunk dahdi-linux # svn checkout http://svn.digium.com/svn/dahdi/tools/trunk dahdi-tools dahdihardware -v sees the box but no spans. Generally '/etc/init.d/dahdi start' . Or more specifically, 'dahdi_registration on' . See also: http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_installation_scenarios I've must be missing something here - this is what I see now. sh-4.0# dahdi_hardware -v usb:001/020 xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware LABEL=[usb:X1037246] connect...@usb-:00:1a.7-4 Shouldn't I see spans ??? I think the box (I've never seen it, but I know what I asked for) has 8fxs+8fxo+2E1 . Thanks, Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- 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] 24 FXS Port Voip Gateway and Asterisk
On Sun, Mar 28, 2010 at 8:28 PM, Steve Edwards asterisk@sedwards.com wrote: On Sun, 28 Mar 2010, Joseph Begumisa wrote: Can anyone recommend a 24 fxs port voip gateway that has worked well with asterisk? I have a couple of analog handsets that I want to hookup to my asterisk server? Any tested and tried product recommendations are welcome. Thanks. Adtran channel banks are a great trailing edge technology. You can get them off Ebay for pennies on the original dollar and they are built like a tank. (voip gateway is not very specific. If you meant SIP or IAX, you might want to specify which.) I've actually had decent success with the GXW-4024 (FXS - SIP) from Grandstream which is probably one of the cheapest 24 FXS port boxes you'll find out there. -- James -- _ -- 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] Cisco 7960 become UNREACHABLE behind pix firewall
I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall. After some period of time, asterisk says that some of them are unreachable, and the phones lose their registration. The only way to make the phones recover is to clear the NAT translation tables for the phones on the PIX (clear xlate...) Does anyone know how to fix this? As you can imagine, it is quite annoying. And it does not happen to all the phones either. sip fixup is enabled on the PIX Are you able to TFTP new phone configs? Assuming so, and it's for only 10 phones, try decreasing the registration time. I've got a 7960 on my desk and documented it with a TFTP-ready config: http://help.cloudvox.com/faqs/sip-phones/cisco-7900-ip-phone It's at the end, commented out. I don't think that config's been used much - most Cloudvox folks are just using SIP to test their AGI apps, not as primary phones. If you want another data point that still crosses your NAT boundary, feel free to sign up for and register with Cloudvox and see whether your registration lasts, using that same config. We switched to pay-as-you-go pricing, so even the free accounts include SIP. If your registrations to Cloudvox also time out, it's probably the PIX. Troy -- Cloudvox -- http://cloudvox.com/ Asterisk in the cloud -- AGI, HTTP/JSON, SIP, REST, live in minutes -- _ -- 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] dnd not working correctly
I would say that from what I know DND function in FreePBX will not automatically configure the phone DND function but it set a flag into Asterisk DB: -- Executing [...@from-internal:5] Set(SIP/117-01f6, DB(DND/117)=YES) in new stack You report that you do not hear nothing but in the log we see: -- Executing [...@from-internal:8] Playback(SIP/117-01f6, do-not-disturbactivated) in new stack I advice to investigate why you are not hearing those prompts before any other steps. HTH, Ioan. On Sat, Mar 27, 2010 at 12:41 AM, Ott Rose sixfourimp...@hotmail.com wrote: i have posted this question couple of times and never really got any hits i wasn't able to provide any debug info -- _ -- 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] Cisco 7960 become UNREACHABLE behind pix firewall
On Mon, Mar 29, 2010 at 12:25 AM, Troy Davis t...@yort.com wrote: sip fixup is enabled on the PIX Try disabling the sip fixup on the PIX and see if that helps. You may have to adjust the configs on the phones themselves when you do this. -- Thanks, --Warren Selby http://www.selbytech.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
