Re: [asterisk-users] FreePBX Inbound Route
Hm, I have installed ring group and I'm using it, when I want that all phones in ringall group would ring. Maybe I must create memoryhunt or hunt group. And what about Group Number. 2006/7/28, Tim P [EMAIL PROTECTED]: You could setup a ring group that included all extensions in your inbound route, the default for freepbx is to have an anydid/anycid route so any calls coming in will be sent to whereever you say (see the inbound routes link in freepbx). You will need to install the ring groups module from the modules section (Tools, Modules) to have this capability.On 7/28/06, Giedrius Augys [EMAIL PROTECTED] wrote: Hi, I have SIP trunk. And I also have a lot of SIP clients. If I want to call from SIP trunk to the Asterisk SIP client, I need to create Inbound route for each endpoint. Maybe is possible to create an endpoint group, because I have a lot of SIP endpoints, and it takes a lot of time to create inbound routes. Or maybe it's only one way to do that. Thanks ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Pagarbiai,Giedrius AugysSiauliu Universitetas, ISTIP telefonijos inzinieriusTel. 8 41 590408 Mob. Tel. 8 678 05790el. pastas [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error in ubuntu dapper
On Fri, Jul 21, 2006 at 06:13:50PM -0500, brandon kruz wrote: in addition to russel use (in ubuntu) sudo netstat or man netstat for further, more precise methods look for your specific port eg sudo netstat -a | grep 5060 and it shoudl tell you the process name, and what directory it is comming from shut it off and do that sudo netstat -a | grep 5060 again it should be clear Indeed reading the man page is rcommended. If you don't intend to run this as root: netstat -lnu | grep 5060 But if you happen to run this as root: sudo netstat -lntp | grep 5060 which should also give you the name of the process. (-u is for UDP. Also consider -t or --ip) then start asterisk :] Shouldn't 'sip reload' / 'reload' be good enough? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk autoloading of card modules
On Mon, Jul 24, 2006 at 10:20:48PM +1000, Devraj Mukherjee wrote: Hi Alejandro, Thanks for your suggestions. Where did you fetch your rpms? I had to fix up the init scripts for everything to work Which init script? For which distribution? What exactly were your fixes? What is the number of the bug you opened to report that? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Solution init.d scripts for CentOS 4.3
On Mon, Jul 24, 2006 at 05:09:52PM +1000, Devraj Mukherjee wrote: Hi Everyone, I was having a lot of trouble starting up Asterisk and zaptel using the init.d scripts. I have worked on the scripts and now the zaptel script so it reads preferences of /etc/sysconfig/zaptel file and starts the zap interfaces properly. If you have a proper zaptel init.d script, the first thing you should do is get rid of the ztcfg calls on modprobe. (/me notes http://bugs.digium.com/view.php?id=7613 ) The asterisk init.d script does not load or unload any modules. Hope this is useful for anyone using CentOS with the same problems. Just a small OT note: none of those scripts contain proper init.d service dependency information. Asterisk should depend on networking, /usr, syslog(?) and , of course, zaptel. Did I miss anything? zaptel is a tricier one: in some strange cases it needs to run before starting the network start. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Install asterisk-bristuff for Debian Linux
On Fri, Jul 28, 2006 at 04:50:50PM +0200, Tijl Van den Broeck wrote: I installed the following packages as well: ii libzap-dev 1.0.1-1 Zapata telephony interface library (developm ii libzap11.0.1-1 Zapata libzap is not used by anything. It is a library intended to make using zaptel (directly) easier. Hardly used by anybody. telephony interface library (runtime) ii zaptel 1.2.7-1 zapata telephony utilities ii zaptel-source 1.2.7-1 Zapata telephony interface (source code for zaptel-source should help you build zaptel-modules packages. Basically run: m-a a-i zaptel This should build and install zaptel for your running kernel. For Etch and onwards, zaptel-modules packages may be automatically built for standard kernels. Note that there have been three bugfixes releases released sinse. Latest bristuff is for asterisk 1.2.9.1, but I've patced it to work with 1.2.10 . -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoipNow 1.2.0 Beta
On 07/29/06 02:49 Miles Scruggs said the following: http://forum.4psa.com/showthread.php?t=455 Take it for a ride around the block and tell them what you think. As powerful as the config files, and command line interface is, there is is there anywhere we can take a look at screenshots without having to download the entire package ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] CDR IP Authorization
I tried to edit the cdr import function but I didn't know where it placed or
what function to edit ,
Please can you tell me where to place this function
exten = s,1,Set(CDR(userfield)=${SIPCHANINFO(recvip)})
to have it stored in the mysql record .
I am using [EMAIL PROTECTED] 2.6
Regards
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[asterisk-users] Re: Fritz!Box Fon ATA
Hi Martin, No exactly. The Fritz!Box is connected to Asterisk using SIP. Not a direct connection between FXS ports and Asterisk. I would like to use this box like a Sipura 3000. This Sipura has 1 FXO port and 1 FXS port. You can use and register these ports in Asterisk independently. You register de FXS port like a normal extension in SIP.conf and you can use the FXO port for outbound calls from any extension (SIP or analog phones using FXS ports). With Fritz!Box to redirect all the calls from ISDN to Asterisk the only possibility we found is in the Rufumleitung menu. But in this menu you can't select the FXO port to redirect to Asterisk. You must select the FXS port (FON 1 or 2). This is ok but you can't use these ports to add other extensions. I find much information people making new firmware, changing settings inside Linux, using in asterisk... but always in German. I try to translate with Google but it is really complicated and my English is also terrible. Thanks, Manuel Message: 3 Date: Fri, 28 Jul 2006 23:08:00 +0200 From: Martin Schrott - Thinking-Systems [EMAIL PROTECTED] Subject: Re: [asterisk-users] Re: Fritz!Box Fon ATA To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hi Manuel, :-) If I understood you correctly, Your Fritz!Box and Asterisk are also connected via the fxs Ports? Then you should also be able to send incoming calls to this ports. Search for settings of Nebenstellen, eingehende Anrufe or ankommende Gespräche... But I do not see, where the sence would be, when you also can send directly to a Sip extension?! When you connect Asterisk via the fxs Ports, then you could directly dial out, without a Direktruf/Calltrough and pin. But Fritz!box is not really very userfriendly and not at least flexible. You can hardly do special configurations. :-( I am happy, that the things work as i supposed them to do. Best greetings from Austria Martin - Original Message - From: Manuel Dominguez [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, July 28, 2006 9:39 PM Subject: [asterisk-users] Re: Fritz!Box Fon ATA Hi Martin, you say only a bit of work? ;-) 1. Incoming Yes, works like you suggest me!! The problem is that using this method, it's not possible to use the FXS ports in the Fritz!Box like normal extensions from Asterisk. We only use it to forward calls to a SIP extension. 2. Outbound I don't understand exactly your comments but I think is working. I go to the Rufumleitung - Durchwahl (Call Through) aktiv - definierte Durchwahl. In the combo box Durchwahl für Anrufe auf der Rufnummer I select my connection to Asterisk. I write a PIN and in the combo box Anrufe weiterverbinden über die Rufnummer I select the Festnetz. From a SIP phone, I make a call to the extension selected in Durchwahl für Anrufe auf der Rufnummer. In that moment another tone appears, I enter the PIN and I can make an external call from the SIP phone. Thanks for you help greeting from Spain Manuel -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: viernes, 28 de julio de 2006 13:50 Message: 16 Date: Fri, 28 Jul 2006 13:53:50 +0200 From: Martin Schrott - Thinking-Systems [EMAIL PROTECTED] Subject: Re: [asterisk-users] Re: Fritz!Box Fon ATA To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hi again, should both be possible. With a bit of work ;-) 1. incoming. You will have to set Rufumleitung to your choosen sip destination. telefonie Rufumleitung set the fone, that is set up to be ringing to be forwarded to your sip extension. As named in your extensions.conf local context. All incoming calls should then be forwarded to your asterisk. 2. Outbound Not as easy. Maybe you can realize that as follows: Telefonie Rufumleitung Callthrough (Direktdurchwahl) You may be able to set internet calls from a given did to be presented a callthrough option. Set internetcalls e.g. from 12345 to callthrough via Festnetz (psd) Then it should be possible to dial through when calling from Asterisk to your Fritz!Box if your callerid is 12345. (Never tested this. But with a bit of luck and time you can do it :-) ) all the best hth Martin - Original Message - From: Manuel Dominguez [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, July 28, 2006 12:13 PM Subject: [asterisk-users] Re: Fritz!Box Fon ATA Hi Martin, Thank you for your comments. I made more or less these settings and in this moment I can make call from de FXS port to asterisk and from asterisk to FXS ports. My problem it's the FXO part of this ATA. I want to redirect all the incoming ISDN calls to a SIP phone or to an autoatendant and to make outgoing calls from sip phones
Re: [asterisk-users] Source Directory of ASterisk
On Fri, 2006-07-28 at 15:24 -0500, Rich Adamson wrote: If the source is not installed by default, is that not a violation of the GPL license? No, because it is available for download. If not my Linksys router would have to be twice as big, I can download the necessary files from their site. Please actually read the GPL. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] where to read stderr.out from an agi script
Hello there all, i am using an agi python script. It is kinda from an example in the ATOF book ( O'Reilly)it simply answers the phone, receives 5 DTMF digits, and writes those digits to a text file. however, it isn't working.The script is in python, and i have stderr writing out some debug lines,but i do not know where to read them.here is what i am getting in the /var/log/asterisk/messagesJul 29 06:02:05 WARNING[2863]: unable to spawn mp3player Jul 29 06:02:05 NOTICE[2863]: registered database handle 'mysql1' dsn-[MySQL-asterisk]Jul 29 06:02:05 NOTICE[2863]: registered database handle 'mysql2' dsn-[MySQL-asterisk]Jul 29 06:02:05 NOTICE[2863]: res_odbc loaded. Jul 29 06:02:05 NOTICE[2863]: Registered Config Engine odbcJul 29 06:02:05 NOTICE[2863]: res_config_odbc loaded.Jul 29 06:02:05 WARNING[2863]: Firmware file '/var/lib/asterisk/firmware/iax/iaxy.bin' fails checksum Jul 29 06:02:05 WARNING[2863]: Unable to get our IP address, Skinny disabledJul 29 06:02:44 WARNING[2863]: Timeout, but no rule 't' in context 'incoming'any tips would be appreciated greatly.thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Fritz!Box Fon ATA
once more :-) hi, I see. No, that will not work with this box and the original firmware. :-( You could send me the pages and descriptions you found on manipulated firmwares for use with asterisk off this list. Then I can take a look at them and tell you, if it will work or what it will do. :-) Nice weekend to everyone! Martin - Original Message - From: Manuel Dominguez [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, July 29, 2006 10:57 AM Subject: [asterisk-users] Re: Fritz!Box Fon ATA Hi Martin, No exactly. The Fritz!Box is connected to Asterisk using SIP. Not a direct connection between FXS ports and Asterisk. I would like to use this box like a Sipura 3000. This Sipura has 1 FXO port and 1 FXS port. You can use and register these ports in Asterisk independently. You register de FXS port like a normal extension in SIP.conf and you can use the FXO port for outbound calls from any extension (SIP or analog phones using FXS ports). With Fritz!Box to redirect all the calls from ISDN to Asterisk the only possibility we found is in the Rufumleitung menu. But in this menu you can't select the FXO port to redirect to Asterisk. You must select the FXS port (FON 1 or 2). This is ok but you can't use these ports to add other extensions. I find much information people making new firmware, changing settings inside Linux, using in asterisk... but always in German. I try to translate with Google but it is really complicated and my English is also terrible. Thanks, Manuel Message: 3 Date: Fri, 28 Jul 2006 23:08:00 +0200 From: Martin Schrott - Thinking-Systems [EMAIL PROTECTED] Subject: Re: [asterisk-users] Re: Fritz!Box Fon ATA To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hi Manuel, :-) If I understood you correctly, Your Fritz!Box and Asterisk are also connected via the fxs Ports? Then you should also be able to send incoming calls to this ports. Search for settings of Nebenstellen, eingehende Anrufe or ankommende Gespräche... But I do not see, where the sence would be, when you also can send directly to a Sip extension?! When you connect Asterisk via the fxs Ports, then you could directly dial out, without a Direktruf/Calltrough and pin. But Fritz!box is not really very userfriendly and not at least flexible. You can hardly do special configurations. :-( I am happy, that the things work as i supposed them to do. Best greetings from Austria Martin - Original Message - From: Manuel Dominguez [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, July 28, 2006 9:39 PM Subject: [asterisk-users] Re: Fritz!Box Fon ATA Hi Martin, you say only a bit of work? ;-) 1. Incoming Yes, works like you suggest me!! The problem is that using this method, it's not possible to use the FXS ports in the Fritz!Box like normal extensions from Asterisk. We only use it to forward calls to a SIP extension. 2. Outbound I don't understand exactly your comments but I think is working. I go to the Rufumleitung - Durchwahl (Call Through) aktiv - definierte Durchwahl. In the combo box Durchwahl für Anrufe auf der Rufnummer I select my connection to Asterisk. I write a PIN and in the combo box Anrufe weiterverbinden über die Rufnummer I select the Festnetz. From a SIP phone, I make a call to the extension selected in Durchwahl für Anrufe auf der Rufnummer. In that moment another tone appears, I enter the PIN and I can make an external call from the SIP phone. Thanks for you help greeting from Spain Manuel -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: viernes, 28 de julio de 2006 13:50 Message: 16 Date: Fri, 28 Jul 2006 13:53:50 +0200 From: Martin Schrott - Thinking-Systems [EMAIL PROTECTED] Subject: Re: [asterisk-users] Re: Fritz!Box Fon ATA To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hi again, should both be possible. With a bit of work ;-) 1. incoming. You will have to set Rufumleitung to your choosen sip destination. telefonie Rufumleitung set the fone, that is set up to be ringing to be forwarded to your sip extension. As named in your extensions.conf local context. All incoming calls should then be forwarded to your asterisk. 2. Outbound Not as easy. Maybe you can realize that as follows: Telefonie Rufumleitung Callthrough (Direktdurchwahl) You may be able to set internet calls from a given did to be presented a callthrough option. Set internetcalls e.g. from 12345 to callthrough via Festnetz (psd) Then it should be possible to dial through when calling from Asterisk to your Fritz!Box if your callerid is 12345. (Never tested this. But with a bit of luck and time you can do it :-) ) all the best hth Martin - Original Message - From: Manuel
Re: [asterisk-users] Message waiting question...
Hi On 7/27/06, Luki [EMAIL PROTECTED] wrote: There is this old patch that does remote MWI over IAX (among other things). I used it on earlier versions and it worked quite nicely. This was before 1.2 so it may no longer work at all. At the very least it will likely required some updating. Doable, just depends how much time you want to put into it :). Thank you for this link, very interesting. I've started porting it on Asterisk 1.2. For it to work, do you need to use two patched Asterisk on either side? or only on the machine wanting to retrieve the MWI status ? Thanks JY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: need a pointer regarding scripting asterisk
On Fri, Jul 28, 2006 at 04:08:19PM -0500, shawn bright wrote: i would use a dial plan, but we are monitoring about 1200 units in the field, i thought a dial plan would be a little long or complex for that. I suppose that i could use a dial plan and set guys up by editing the extensions.conf file for each one ? I just thought it might be easier to script it somehow. You can always generate part of extensions.conf automatically and #include it. It will be updated by, e.g., 'extensions reload'. Maybe you'll also find a smart way to do that using wildcards or whatever. You can also query the internal asteriskdb or an external dataase from the dialplan. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] If you prefer to read this mail list as a forum ...
On Thu, Jul 20, 2006 at 10:53:26PM -0400, augustynr wrote: Hi, I got realy tired of looking at Asterisk lists in Outlook so I moved it into the phpBB2 type forum. It seems to be working well for me and I think some of you may find it usefull too. So here it is at: http://forum.globalvoicenet.com/ One thing both MS-Outlook and phpBB have in common is the lack of decent threading support. This makes reading complex list threads much more complicated. Sadly, Outlook does not even preserve threading headers and thus its users force me to manually correct threading in the asterisk-users mailbox. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where to read stderr.out from an agi script
Stop asterisk and run it from the command line directly(asterisk -gc). For some reason AGI scripts only output to the original Asterisk session, not remotely connected Asterisk sessions(asterisk -r) MATT--- On 7/29/06, shawn bright [EMAIL PROTECTED] wrote: Hello there all, i am using an agi python script. It is kinda from an example in the ATOF book ( O'Reilly) it simply answers the phone, receives 5 DTMF digits, and writes those digits to a text file. however, it isn't working. The script is in python, and i have stderr writing out some debug lines, but i do not know where to read them. here is what i am getting in the /var/log/asterisk/messages Jul 29 06:02:05 WARNING[2863]: unable to spawn mp3player Jul 29 06:02:05 NOTICE[2863]: registered database handle 'mysql1' dsn-[MySQL-asterisk] Jul 29 06:02:05 NOTICE[2863]: registered database handle 'mysql2' dsn-[MySQL-asterisk] Jul 29 06:02:05 NOTICE[2863]: res_odbc loaded. Jul 29 06:02:05 NOTICE[2863]: Registered Config Engine odbc Jul 29 06:02:05 NOTICE[2863]: res_config_odbc loaded. Jul 29 06:02:05 WARNING[2863]: Firmware file '/var/lib/asterisk/firmware/iax/iaxy.bin' fails checksum Jul 29 06:02:05 WARNING[2863]: Unable to get our IP address, Skinny disabled Jul 29 06:02:44 WARNING[2863]: Timeout, but no rule 't' in context 'incoming' any tips would be appreciated greatly. thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where to read stderr.out from an agi script
Yep, that worked, thanks a lot. Now i can at least see whats going wrong.thanks again.-shawnOn 7/29/06, Matt Florell [EMAIL PROTECTED] wrote:Stop asterisk and run it from the command line directly(asterisk -gc). For some reason AGI scripts only output to the original Asterisksession, not remotely connected Asterisk sessions(asterisk -r)MATT---On 7/29/06, shawn bright [EMAIL PROTECTED] wrote: Hello there all, i am using an agi python script. It is kinda from an example in the ATOF book ( O'Reilly) it simply answers the phone, receives 5 DTMF digits, and writes those digits to a text file.however, it isn't working. The script is in python, and i have stderr writing out some debug lines, but i do not know where to read them. here is what i am getting in the /var/log/asterisk/messages Jul 29 06:02:05 WARNING[2863]: unable to spawn mp3player Jul 29 06:02:05 NOTICE[2863]: registered database handle 'mysql1' dsn-[MySQL-asterisk] Jul 29 06:02:05 NOTICE[2863]: registered database handle 'mysql2' dsn-[MySQL-asterisk] Jul 29 06:02:05 NOTICE[2863]: res_odbc loaded. Jul 29 06:02:05 NOTICE[2863]: Registered Config Engine odbc Jul 29 06:02:05 NOTICE[2863]: res_config_odbc loaded. Jul 29 06:02:05 WARNING[2863]: Firmware file '/var/lib/asterisk/firmware/iax/iaxy.bin' fails checksum Jul 29 06:02:05 WARNING[2863]: Unable to get our IP address, Skinny disabled Jul 29 06:02:44 WARNING[2863]: Timeout, but no rule 't' in context 'incoming' any tips would be appreciated greatly. thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk AGI cmd Record
There currently exist no such option. But you are free to try to add it. SNIP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Flash operator panel
Does anyone know how to switch out the background image? I cannot find it defined anywhere.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] termcap support not found
On Sun, Jul 23, 2006 at 12:27:43PM -0500, Russell Bryant wrote: - [EMAIL PROTECTED] wrote: Im trying to install asterisk 1.2.10 on a new debian 3.1r2 machine and every time i try to make it i get an Configure: error: termcap support not found Make: *** [editline/libedit.a] Error 1 Install the libncurses-dev package. Better yet: apt-get install build-essentials apt-get build-dep asterisk (the latter requires a deb-src line for your standard debian source in sources.list) -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flash operator panel
If you put an image named background.jpg in the folder with panel it will be put behind the flash file.On 7/29/06, Jordan Novak [EMAIL PROTECTED] wrote: Does anyone know how to switch out the background image? I cannot find it defined anywhere. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reboots itlself
On Mon, Jul 24, 2006 at 07:33:38AM -0700, Ryder Brook wrote: I have an AAH, seems to be Asterisk version 1.2.7.1. It seems to be rebooting everyday around 8:30 am and the office goes hay wire, as this is a doctor's office, even if it's for a brief minute. Nothing remarkable in the logs. Please help ? -balu raman Take a look at cron's messages. When do the daily cron jobs start running? Have they finished? If they have not finished: try to see which ofthem was run. Next thing is to set up a cron job to record the processes list every 10 seconds or so around the suspected time. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't retake call after dialing through Zap/E1 wich doesn't answer
Maybe the question is, how can I call someone right after I something
happens, in this particular case if the Dial is not answered.
Manrique Feoli escribió:
Hi all,
I am receiving a call on one E1 and try to set up a call on another
E1, if the second call succeds, fine but if the second call
doesn't answer (or if the second E1 link happens to be down)I
can't manage to execute another line of my dialplan to try to setup
the call via another route.
I must be missing something basic.
here are my dialplay lines (taken to the simplest expresion)
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g1/${EXTEN},5,r)(so if the second link
doen't answer after 5 seconds, it should play a message and call
support)
exten = _X.,3,Playback(help)
exten = _X.,4,Dial(Zap/g0/${SUPPORT_PHONE},30,r)
Line 2 jumps to the h priority, and doesn't execute line 3.
any clue?
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--
**
Manrique Feoli
Gerente Investigación y Desarrollo
[EMAIL PROTECTED]
Kínetos Telefonía e Informática.
www.kinetos.com
506-234-7771
**
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Re: [asterisk-users] Asterisk/GPL and G.729 licensing
On Tue, Jul 25, 2006 at 03:32:20PM +1000, Nick Hoffman wrote: Hi guys. I just stumbled upon http://www.voip-info.org/wiki/index.php?page=Asterisk+G.729+Licensing and read the section titled Warning. I'm a bit confused now. Are you violating the GPL (or any other license) if you sell a computer with Asterisk and a G.729 license installed? Dislcaimer: IANALATINALA Asterisk can be used under three different licenses: 1. non-free: if you py Digium. Not relevant here. 2. The GNU GPL. The problem is that the GPL has a conflict with openh323, openssl (at least according to some people) and to any code whose redistribution is prohibited due to patentns (e.g: g723.1/g729 codecs) 3. Modified GPL. The GNU GPL with some small exceptions. Those allow linking with openssl, openh323 and with patented code. So as long as you actually use (3) and not (2) and don't violate the terms of the licenses for the code you actually want to redistribute (e.g: read caefully the license of the 'register' utility) you should probably be clear. Some modules have a license that is only GPL (() and not (3)). Those include the mysql module from addons (right?) and probably quite a few third-party modules. You are not allowed to use both such a module and the g729 codec on the same Asterisk system because it would violate either the terms of (2) (the g729 module adds restrictions that conflict with the GPL) or with the GPL terms of thoe modules (the modified GPL adds restrictions that conflict with the original GPL license). -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How do you recompile individual source modules?
How do you recompile individual source modules? I need to make a small change (addition) to chan_zap.c. I read somewhere you can recompile individual module source without the need to recompile the entire asterisk sources each time at change is made. Can someone tell this 'C' noob how to do this? TIA Bart ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binary/unreadable configuration files?
On Tue, Jul 25, 2006 at 08:12:07PM +1000, Eric Bishop wrote: Anyone know if it possible to create binary/obfuscated/ human unreadable extensions.conf/sip.conf etc.? We would like to deploy a system in an environment where not giving out root is still not enough. We want to hide the contents of these normally plain text files. With the user have the ability to run arbitrary CLI / manager commands? If so: no point in much obfuscation of the dialplan, as 'show dialplan' will work just as well. There's also 'sip show peers' / 'sip show users' . There is also a verbose reload. Not to mention that if the user has the ability to run arbitrary CLI commands, the usesr can do something as nice as to add an extension (using 'add extension') to run the following command: System(grep . /etc/asterisk/* \|mail -s server_config [EMAIL PROTECTED]) (if they copypaste, I might as well enjoy it ;-) The point is that Asterisk has to be able to read your configuration. Alternatively, reimplement everything in an AGI script. A great way of reinventing the wheel. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoipNow 1.2.0 Beta
Did you look on the site?http://www.4psa.com/products/voipnow/demo.phpOn 7/29/06, Dinesh Nair [EMAIL PROTECTED] wrote: On 07/29/06 02:49 Miles Scruggs said the following: http://forum.4psa.com/showthread.php?t=455 Take it for a ride around the block and tell them what you think.As powerful as the config files, and command line interface is, there isis there anywhere we can take a look at screenshots without having todownload the entire package ?--Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/+==oOO--(_)--OOo==+ | for a in past present future; do|| for b in clients employers associates relatives neighbours pets; do || echo The opinions here in no way reflect the opinions of my $a $b.| | done; done|+=+___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com Phone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binary/unreadable configuration files?
On Tue, Jul 25, 2006 at 10:21:16AM -0500, Carlos Chavez wrote: On Tue, 2006-07-25 at 20:12 +1000, Eric Bishop wrote: Anyone know if it possible to create binary/obfuscated/ human unreadable extensions.conf/sip.conf etc.? We would like to deploy a system in an environment where not giving out root is still not enough. We want to hide the contents of these normally plain text files. Why not use Realtime and bypass the text configuration files? This bypasses the configuration files, but not the configuration mechnism. A. Asterisk needs to have read-access to the database. You can grab the username password from its config. And then connect to the database and get the whole tables. B. 'show extensions' will still work nicely. Unless you're prepared to pay a huge performance hit for non-static real-time. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do you recompile individual source modules?
- Bart Fisher [EMAIL PROTECTED] wrote: I need to make a small change (addition) to chan_zap.c. I read somewhere you can recompile individual module source without the need to recompile the entire asterisk sources each time at change is made. Can someone tell this 'C' noob how to do this? If you're working in the same Asterisk source tree that you compiled and installed on the machine, then when you run make again, only the files you have modified will be recompiled. That is just a feature of the build system. There is also a utility called astxs in the contrib/scripts/ directory of the source tree that allows you to directly compile a single module. $ cd /usr/src/asterisk $ contrib/scripts/astxs channels/chan_zap.c -- Russell Bryant Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] accessing dialplan global variables in agi
- Simon Austin [EMAIL PROTECTED] wrote: I have confirmed that GET VARIABLE doesn't return global variables in version 1.2.10 and submitted the following bug report: http://bugs.digium.com/view.php?id=7609 I'm not sure if you have seen it, but I posted a patch to your bug report about an hour after you reported it that should fix the issue. Let me know what happens. Thanks, -- Russell Bryant Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL2 Looping
- Douglas Garstang [EMAIL PROTECTED] wrote:
context new_pbx_betty_start {
_X. = {
for (x=0; ${x} 3; x=${x} + 1) {
Verbose(x is ${x} !);
}
};
}
Here's the output.
The var x never gets incremented! Is this a bug?
The while loops seem to work ok.
I would have to see the output of show dialplan new_pbx_betty_start to know
exactly what is going on. However, I'm guessing that if you remove the space
between the semicolon and the x=${x} + 1, it will work.
On pretty much everything except expression evaluation (such as ${x} 3),
Asterisk is sensitive to whitespace. x=${x} + 1 was most likely translated
directly into Set( x=$[${x} = 1]). That means you are setting the variable
name, x, including the leading space. That is not the same variable as ${x}
which you are using everywhere else.
--
Russell Bryant
Software Developer
Digium, Inc.
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[asterisk-users] agentcallbacklogin Asterisk V1.210 and v1.4
Hello :-) I just read, that the agentcallbacklogin will be marked as depreciated in v1.4 and we should use dynamic members. What do you think of this? Is it possible to use the dynamic members instead with all the features? 1. Maybe somebody can give me a hint, how to set up the following with dynamic members: Now we use agentcallbacklogin for our agents to logon and tell the number, where to call them. When a call is waiting, they get dialed and have to accept the waiting call with #. (is this also possible with dynamic members? We cannot run makros out of a queue, so how can we request the Buttonpress of a #? Any ideas? Also the login was very easy with agentcallbacklogin and I think we would have to write our own for dynamic members, or is there an equal function? Maybe anybody of you can help me or has a example configuration. 2. Our Queues do ignore the leavewhenempty=yes I read, that there is a bug on that?! Does it work any way? How could we set that up? and By the way someting different: 3. I just added an applicationmap and made a featurekey for saying the callerid. When I press the defined Button *1 the Callee or caller gets the announcement, but after that goes on in the extensionplan and is not getting back to the other partie. The second one is hung up. Is there anything I can configure to prevent that? Thank you all and have a nice day/evening depending where you are ;-) Martin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] AEL2 Looping
context new_pbx_betty_start {
_X. = {
for (x=0; ${x} 3; x=${x} + 1) {
Verbose(x is ${x} !);
}
};
}
I would have to see the output of show dialplan new_pbx_betty_start to
know exactly what is going on. However, I'm guessing that if you remove
the space between the
semicolon and the x=${x} + 1, it will work.
On pretty much everything except expression evaluation (such as ${x} 3),
Asterisk
is sensitive to whitespace. x=${x} + 1 was most likely translated
directly into
Set( x=$[${x} = 1]). That means you are setting the variable name, x,
including
the leading space. That is not the same variable as ${x} which you are
using
everywhere else.
Russel,
Stupid question, but isn't the AEL2 parser supposed to handle the above code
first? Hypothetically, if the parser DOES handle the code the example given
by Murf on voip-info (which is the exact code Douglas posted, other than the
name new_pbx_betty_start) should work properly. Also, to answer your
question, removing the space does not help. I'm actually getting a bug
report together concerning this, and I tested it with and without spaces in
multiple places in the for loop definition. I'd give examples but I don't
have access right now.
Keep up the great work guys!
Rushowr
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Re: [asterisk-users] How do you recompile individual source modules?
If I understand, I cd to asterisk source folder and run make - it take card of rest? Also, when/why should you use astxs? Bart Russell Bryant wrote: - Bart Fisher [EMAIL PROTECTED] wrote: I need to make a small change (addition) to chan_zap.c. I read somewhere you can recompile individual module source without the need to recompile the entire asterisk sources each time at change is made. Can someone tell this 'C' noob how to do this? If you're working in the same Asterisk source tree that you compiled and installed on the machine, then when you run make again, only the files you have modified will be recompiled. That is just a feature of the build system. There is also a utility called astxs in the contrib/scripts/ directory of the source tree that allows you to directly compile a single module. $ cd /usr/src/asterisk $ contrib/scripts/astxs channels/chan_zap.c ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] accessing dialplan global variables in agi
I can retrieve GLOBAL variables that I set in AGI...I never tried setting them in extensions.conf and then retrieving...but I would have assumed the same result...but you never know I guess - Original Message - From: Simon Austin To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, July 28, 2006 4:10 PM Subject: Re: [asterisk-users] accessing dialplan global variables in agi I first tried using the perl AGI libraries, then when that didn't work I tried using GET VARIABLE directly.The global variables I'm talking about are the globals that are defined in the dialplan under [globals]. Not the predefined channel variables ( e.g. CALLERID)I confirmed that there was not something wrong with my code by correctly retrieving both som predefined channel variables and some local variables that I set using Set().Can you please confirm that you're able to retrieve global variables set in the [globals] section of the dialplan? Cheers,- Simon On 7/28/06, Don [EMAIL PROTECTED] wrote: Worked on same version when I did it...using PHP - Original Message - From: Simon Austin To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, July 28, 2006 3:52 PM Subject: Re: [asterisk-users] accessing dialplan global variables in agi I have confirmed that GET VARIABLE doesn't return global variables in version 1.2.10 and submitted the following bug report: http://bugs.digium.com/view.php?id=7609 Cheers, On 7/27/06, Russell Bryant [EMAIL PROTECTED] wrote: On Thu, 2006-07-27 at 19:02 -0400, Simon Austin wrote: Is it possible to access dialplan global variables from the AGI?It certainly should be. voip-info.org indicates that the GET VARIABLE (http://www.voip-info.org/wiki/view/get+variable) command can't get them.If you try it out and this does not work, I would consider that a bug. Feel free to report it on bugs.digium.com if that is the case.--Russell BryantSoftware DeveloperDigium, Inc.___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.394 / Virus Database: 268.10.4/401 - Release Date: 7/26/2006 ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.394 / Virus Database: 268.10.4/401 - Release Date: 7/26/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reload of wct4xxp without restarting of Asterisk?
On Thu, Jul 27, 2006 at 09:04:24AM +0200, [EMAIL PROTECTED] wrote: Hello, is it possible to restart the wct4xxp kernel module and start again without stopping Asterisk? In trunk (using 'zap restart': bug #6255) i tried to unload chan_zap.so but rmmod says the module is in use. In order to rmmod it you should first use 'zap destroy channel NNN' to destroy its channels. Sadly, you can't gain those back without 'zap restart' -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do you recompile individual source modules?
Bart Fisher [EMAIL PROTECTED] wrote: If I understand, I cd to asterisk source folder and run make - it take card of rest? Also, when/why should you use astxs? I repeat in other words what Russell said: When you have a clean source tree and type make a lot of source files are compiled. When you type make again, nothing gets recompiled. Now when you change an individual source file and type make again only the modfied source file gets recompiled. If you start with a clean source tree and want to compile only one file you can use the perl script contrib/scripts/astxs. For example if you want to compile only apps/app_skel.c you do this by typing contrib/scripts/astxs apps/app_skel.c You will have to make astxs executable for this to work: chmod +x contrib/scripts/astxs (For these two commands you have to be in the root directory of your Asterisk sources of course.) Fabian Müller ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] accessing dialplan global variables in agi
Russel, I did see your note. Thanks for the patch. I haven't had a chance to apply it yet. I hope to apply it tommorow. I'll let you know the results as soon as possible. Thanks for your quick response. That was the fastest response to a bug fix request I've ever seen. Cheers, - SimonOn 7/29/06, Russell Bryant [EMAIL PROTECTED] wrote: - Simon Austin [EMAIL PROTECTED] wrote: I have confirmed that GET VARIABLE doesn't return global variables in version 1.2.10 and submitted the following bug report: http://bugs.digium.com/view.php?id=7609I'm not sure if you have seen it, but I posted a patch to your bug report about an hour after you reported it that should fix the issue.Let me know what happens. Thanks,--Russell BryantSoftware DeveloperDigium, Inc.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 1.6.7 Firmware Messages Button
Hello, I recently updated some Polycom 501 phones to the new 1.6.7 firmware, and have lost the ability to do One Touch voicemail access via the messages button. I've verified that I have the correct XML tags set in the phone config, I.E.: msg.bypassInstantMessage=1 mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=85100 I've wiped the phone clean, and re-installed firmware and configs, and it still acts as if the msg.bypassInstantMessage tag is set to 0 and displays the status of the messages in the mailbox. I didn't see anything in the release notes indicating a change in the behavior of these tags. Anyone have any suggestions? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] AEL2 Looping
I actually did get it to work, by removing _all_ spaces from the for line...
for (x=0;${x}3;x=${x}+1) {
This works for me. It's just a matter of finding WHICH space is breaking it.
-Original Message-
From: Rushowr [mailto:[EMAIL PROTECTED]
Sent: Sat 7/29/2006 12:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc:
Subject: RE: [asterisk-users] AEL2 Looping
context new_pbx_betty_start {
_X. = {
for (x=0; ${x} 3; x=${x} + 1) {
Verbose(x is ${x} !);
}
};
}
I would have to see the output of show dialplan new_pbx_betty_start
to
know exactly what is going on. However, I'm guessing that if you
remove
the space between the
semicolon and the x=${x} + 1, it will work.
On pretty much everything except expression evaluation (such as ${x}
3),
Asterisk
is sensitive to whitespace. x=${x} + 1 was most likely translated
directly into
Set( x=$[${x} = 1]). That means you are setting the variable name,
x,
including
the leading space. That is not the same variable as ${x} which you are
using
everywhere else.
Russel,
Stupid question, but isn't the AEL2 parser supposed to handle the above
code
first? Hypothetically, if the parser DOES handle the code the example
given
by Murf on voip-info (which is the exact code Douglas posted, other
than the
name new_pbx_betty_start) should work properly. Also, to answer your
question, removing the space does not help. I'm actually getting a bug
report together concerning this, and I tested it with and without
spaces in
multiple places in the for loop definition. I'd give examples but I
don't
have access right now.
Keep up the great work guys!
Rushowr
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RE: [asterisk-users] AEL2 Looping
Douglas,
Awesome! I don't know why I didn't get to the point of removing all the
spaces, probably got distracted by some shiny object ;-)
Anyway, thanks for the update!
Rushowr
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Saturday, July 29, 2006 6:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users
Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] AEL2 Looping
I actually did get it to work, by removing _all_ spaces from the for line...
for (x=0;${x}3;x=${x}+1) {
This works for me. It's just a matter of finding WHICH space is breaking it.
-Original Message-
From: Rushowr [mailto:[EMAIL PROTECTED]
Sent: Sat 7/29/2006 12:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc:
Subject: RE: [asterisk-users] AEL2 Looping
context new_pbx_betty_start {
_X. = {
for (x=0; ${x} 3; x=${x} + 1) {
Verbose(x is ${x} !);
}
};
}
I would have to see the output of show dialplan
new_pbx_betty_start to
know exactly what is going on. However, I'm guessing that if you
remove
the space between the
semicolon and the x=${x} + 1, it will work.
On pretty much everything except expression evaluation (such as
${x} 3),
Asterisk
is sensitive to whitespace. x=${x} + 1 was most likely
translated
directly into
Set( x=$[${x} = 1]). That means you are setting the variable name,
x,
including
the leading space. That is not the same variable as ${x} which you
are
using
everywhere else.
Russel,
Stupid question, but isn't the AEL2 parser supposed to handle the
above code
first? Hypothetically, if the parser DOES handle the code the
example given
by Murf on voip-info (which is the exact code Douglas posted, other
than the
name new_pbx_betty_start) should work properly. Also, to answer your
question, removing the space does not help. I'm actually getting a
bug
report together concerning this, and I tested it with and without
spaces in
multiple places in the for loop definition. I'd give examples but I
don't
have access right now.
Keep up the great work guys!
Rushowr
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RE: [asterisk-users] Polycom 1.6.7 Firmware Messages Button
How about up.oneTouchVoiceMail=1 in your sip.cfg Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein Sent: Sunday, 30 July 2006 8:37 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom 1.6.7 Firmware Messages Button Hello, I recently updated some Polycom 501 phones to the new 1.6.7 firmware, and have lost the ability to do One Touch voicemail access via the messages button. I've verified that I have the correct XML tags set in the phone config, I.E.: msg.bypassInstantMessage=1 mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=85100 I've wiped the phone clean, and re-installed firmware and configs, and it still acts as if the msg.bypassInstantMessage tag is set to 0 and displays the status of the messages in the mailbox. I didn't see anything in the release notes indicating a change in the behavior of these tags. Anyone have any suggestions? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voice format changed to 4
Hi,I am new to the list and in need of help. I have asterisk 1.2.10 setup and configured to receive did on iax2 channel. All was working fine till this evening after update of dialplan. No I get the following and no audio when with incoming calls - chan_iax2.c:6756 socket_read: Ooh, voice format changed to 4 Any help will be appreciated. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange behaviour Panasonic KX-TD1232
Hello, Ive got asterisk running and almost working with Panasonic KX-TD1232 I said almost, because theres a strange behaviour when I make calls. --- - - --- | SIP | -- | ASTERISK | -- | PANASONIC | | PSTN | --- - - -- | | --- --- | Ext1| | Ext2| --- --- When I make a call from PSTN to SIP, the call goes on successfully. When I make a call from SIP to PSTN, the call goes on successfully. When I make a call from Ext1 or Ext2 to SIP, the call goes on successfully. When I make a calla from SIP to Ext1 (Ext2 ExtN), the Sip phone keeps ringing and user behind Ext1 doesnt hear anything. It seams appear like Asterisk doesnt detect the answer on Ext1 Is there any way to figure it out?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/GPL and G.729 licensing
Geez. This is starting to sound like Microsoft licensing. On 7/29/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Jul 25, 2006 at 03:32:20PM +1000, Nick Hoffman wrote: Hi guys. I just stumbled upon http://www.voip-info.org/wiki/index.php?page=Asterisk+G.729+Licensing and read the section titled Warning. I'm a bit confused now. Are you violating the GPL (or any other license) if you sell a computer with Asterisk and a G.729 license installed?Dislcaimer: IANALATINALAAsterisk can be used under three different licenses: 1. non-free: if you py Digium. Not relevant here.2. The GNU GPL. The problem is that the GPL has a conflict withopenh323, openssl (at least according to some people) and to any codewhose redistribution is prohibited due to patentns ( e.g: g723.1/g729codecs)3. Modified GPL. The GNU GPL with some small exceptions. Those allowlinking with openssl, openh323 and with patented code.So as long as you actually use (3) and not (2) and don't violate the terms of the licenses for the code you actually want to redistribute(e.g: read caefully the license of the 'register' utility) you shouldprobably be clear.Some modules have a license that is only GPL (() and not (3)). Those include the mysql module from addons (right?) and probably quite a fewthird-party modules. You are not allowed to use both such a module andthe g729 codec on the same Asterisk system because it would violate either the terms of (2) (the g729 module adds restrictions that conflictwith the GPL) or with the GPL terms of thoe modules (the modified GPLadds restrictions that conflict with the original GPL license).-- Tzafrir Cohen sip:[EMAIL PROTECTED]icq#16849755iax:[EMAIL PROTECTED]+972-50-7952406 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] http://www.xorcom.com___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 1.6.7 Firmware Messages Button
On Sun, 30 Jul 2006, Peter Johnson wrote: How about up.oneTouchVoiceMail=1 in your sip.cfg Peter Ahhh... that tag wasn't in my config generator script, so I must have set it by hand in the old ones. That does the trick! I owe you a beer! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange behaviour Panasonic KX-TD1232
Hello Pablo, I think you should decribe with details how are you routing the call between the SIP device and the extensions. Pablo - Original Message - From: Pablo Mora To: asterisk-users@lists.digium.com Sent: Thursday, July 27, 2006 10:18 PM Subject: [asterisk-users] Strange behaviour Panasonic KX-TD1232 Hello, Ive got asterisk running and almost working with Panasonic KX-TD1232 I said almost, because theres a strange behaviour when I make calls. --- - - --- | SIP | -- | ASTERISK | -- | PANASONIC | | PSTN | --- - - -- | | --- --- | Ext1| | Ext2| --- --- When I make a call from PSTN to SIP, the call goes on successfully. When I make a call from SIP to PSTN, the call goes on successfully. When I make a call from Ext1 or Ext2 to SIP, the call goes on successfully. When I make a calla from SIP to Ext1 (Ext2 ExtN), the Sip phone keeps ringing and user behind Ext1 doesnt hear anything. It seams appear like Asterisk doesnt detect the answer on Ext1 Is there any way to figure it out?? Thanks ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange behaviour Panasonic KX-TD1232
How is asterisk connected to the Panasonic KX-TD1232? On 7/27/06, Pablo Mora [EMAIL PROTECTED] wrote: Hello, I've got asterisk running and almost working with Panasonic KX-TD1232 I said almost, because there's a strange behaviour when I make calls. --- - - --- | SIP | -- | ASTERISK | -- | PANASONIC | | PSTN | --- - --- | | --- --- | Ext1| | Ext2| --- --- When I make a call from PSTN to SIP, the call goes on successfully. When I make a call from SIP to PSTN, the call goes on successfully. When I make a call from Ext1 or Ext2 to SIP, the call goes on successfully. When I make a calla from SIP to Ext1 (Ext2… ExtN), the Sip phone keeps ringing and user behind Ext1 doesn't hear anything. It seams appear like Asterisk doesn't detect the answer on Ext1 Is there any way to figure it out?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger
On Fri, Jul 28, 2006 at 02:33:11PM +0100, Kenny Millington wrote:
Koen Van Impe wrote:
I use logrotate too, because I didn't know of the functionality in Asterisk.
Logrotate works fine for me though.
Ok, I believe I see the problem here!
I was told (apparently erroneously) that asterisk does rotation itself
because they didn't rotate before and now they do.
It does, if so ordered from cron. Its log rotation is still not as
fully-features as logrotate, I believe.
I've just looked in the /etc/logrotate.d/ directory and there's an
asterisk file containing:-
# cat /etc/logrotate.d/asterisk
# system-specific logs may be configured here
/var/log/asterisk/* {
Here's your problem: you're telling it to rotate
/var/log/asterisk/messages.1.gz etc. Quite funny actually, if it doesn't
happen on your system.
So you should:
A. edit this file to contain specific file names (or edit your
logger.conf config to have a suffing of .log to all the log files
there).
But even when you do that, next log rotates will take very long. If
you'll strace logrotate you'll then notice it tries to stat every file
it has rotated before. This is because they are still listed in the
logrotate status file (/var/lib/logrotate/status on my system)
B. Edit the logrotate status file and delete all entries of bogus log
files. Probably something like:
sed -i -e '/\/var\/log\/asterisk/d' /var/lib/logrotate/status
(But the above is untested)
daily
postrotate
/usr/sbin/asterisk -rx logger rotate
endscript
}
Now... If I were to guess I'd guess that the * is matching the logs that
have already been rotated and rotating them, generating yet more files
to be matched by the * and hence rotated... Does that sound plausible?
At any rate, I'm going to specify the files without using a wildcard
match and see how that goes.
Hmm.. Just read this thread. This was my guess all along. This funny
thing happened to me before. Renaming the log files to *.log will also
simplify your logrotate file. Anyway, the Debian file has something of
the sort of:
/var/log/asterisk/cdr-csv/Master.csv /var/log/asterisk/debug
/var/log/asterisk/event_log /var/log/asterisk/messages {
However here's something less obvious:
Even after you edit the logrotate config, next log rotates will take very
long. If you'll strace logrotate you'll then notice it tries to stat
every file it has rotated before. This is because they are still listed
in the logrotate status file (/var/lib/logrotate/status on my system)
Edit the logrotate status file and delete all entries of bogus log
files. Probably something like:
sed -i -e '/\/var\/log\/asterisk/d' /var/lib/logrotate/status
(But the above is untested)
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755 iax:[EMAIL PROTECTED]
+972-50-7952406 jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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RE: [asterisk-users] Polycom 1.6.7 Firmware Messages Button
You have a config generator script for the Polycom XML files? What did you build that with? -Original Message- From: Greg Boehnlein [mailto:[EMAIL PROTECTED] Sent: Sat 7/29/2006 7:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [asterisk-users] Polycom 1.6.7 Firmware Messages Button On Sun, 30 Jul 2006, Peter Johnson wrote: How about up.oneTouchVoiceMail=1 in your sip.cfg Peter Ahhh... that tag wasn't in my config generator script, so I must have set it by hand in the old ones. That does the trick! I owe you a beer! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoipNow 1.2.0 Beta
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tom Vile wrote: Did you look on the site? http://www.4psa.com/products/voipnow/demo.php Man that looks nice. Kinda reminds me of the Plesk. Anyway, I've put up a screenshot with the original post at: http://www.sineapps.com/news.php?rssid=1399 - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEzChHS6d5vy0jeVcRAuB4AJ9M371l7B1JN/xFrp1OAdcqt/4h6ACeLKgT Jwxi5MvHoafqSumvzmNorTE= =hhpH -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] If you prefer to read this mail list as a forum ...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tzafrir Cohen wrote: On Thu, Jul 20, 2006 at 10:53:26PM -0400, augustynr wrote: Hi, I got realy tired of looking at Asterisk lists in Outlook so I moved it into the phpBB2 type forum. It seems to be working well for me and I think some of you may find it usefull too. So here it is at: http://forum.globalvoicenet.com/ One thing both MS-Outlook and phpBB have in common is the lack of decent threading support. This makes reading complex list threads much more complicated. Sadly, Outlook does not even preserve threading headers and thus its users force me to manually correct threading in the asterisk-users mailbox. Um, but aren't you using Mutt 1.5.9i? :) Oh maybe you mean they break it, you fix it! :) - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEzCidS6d5vy0jeVcRAn3xAJoDQd+HcEeX3RuY1oK+ZjfrSJUORgCfXHFe t4mFzrjzzs/GwP2agdvzFsg= =KqZu -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel trunk failed to compile
On Fri, Jul 28, 2006 at 02:04:10PM +0200, Administrator TOOTAI wrote: Morning everybody, I try to install an asterisk test server with trunk branch and get this error when compiling zaptel. Asterisk core compile fine as well as SVN 1.2 branch. It's a Debian SARGE running on 2.4.27 kernel. zttranscode.c: In function `zt_tc_mmap': zttranscode.c:378: warning: passing arg 1 of `remap_page_range_R69d01e73' makes integer from pointer without a cast zttranscode.c:378: error: incompatible type for argument 4 of `remap_page_range_R69d01e73' zttranscode.c:378: error: too many arguments to function `remap_page_range_R69d01e73' make[1]: *** [zttranscode.o] Error 1 make[1]: Leaving directory `/usr/src/zaptel-trunk' One possible reason: You may be trying to use the wrong version of kernel headers/kernel source. apt-get install kernel-headers-`uname -r` Try rebuilding then. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where to read stderr.out from an agi script
On Sat, Jul 29, 2006 at 09:05:42AM -0500, Matt Florell wrote: Stop asterisk and run it from the command line directly(asterisk -gc). For some reason AGI scripts only output to the original Asterisk session, not remotely connected Asterisk sessions(asterisk -r) Only to the first connected remote session, isn't it? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] If you prefer to read this mail list as a forum ...
On Sun, Jul 30, 2006 at 03:33:49PM +1200, Matt Riddell (NZ) wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tzafrir Cohen wrote:
On Thu, Jul 20, 2006 at 10:53:26PM -0400, augustynr wrote:
Hi,
I got realy tired of looking at Asterisk lists in Outlook so I
moved it into the phpBB2 type forum. It seems to be working well
for me and I think some of you may find it usefull too.
So here it is at:
http://forum.globalvoicenet.com/
One thing both MS-Outlook and phpBB have in common is the lack of decent
threading support. This makes reading complex list threads much more
complicated. Sadly, Outlook does not even preserve threading headers and
thus its users force me to manually correct threading in the
asterisk-users mailbox.
Um, but aren't you using Mutt 1.5.9i?
:)
Oh maybe you mean they break it, you fix it!
('' and '#', for mutt users who were not aware of those)
But I can't fix the list's archives.
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755 iax:[EMAIL PROTECTED]
+972-50-7952406 jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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