[asterisk-users] Asterisk on NGINX Server?
Hi. Was the AGI Server to write dialplans in any programming language in Asterisk assumed to be configured for the apache web server? Or should it not matter what web server you have (in my case NGINX)? - Dominic The ability to simplify means to eliminate the unnecessary so that the necessary may speak. -Hofstadter's Law ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.11, res_features.so, SegFault
Hi All,
I have a really strange issue occuring where if I run show dialplan or
dialplan show or dialplan show parkedcalls, then asterisk dumps core.
It only appears to happen with contexts that are created within
res_features. I am able to display all my other dialplans, but, every
time I try to just do a normal dialplan show asterisk core dumps
(Segmentation Fault).
My environment is as follows:
Asterisk v 1.4.11
Solaris 10 update 3 (11/06), patched current
gcc v3.4.3
example console output
--
*CLI dialplan show
[ Context 'default' created by 'pbx_config' ]
Include ='demo'
[pbx_config]
[ Context 'page' created by 'pbx_config' ]
'_X.' = 1. Macro(page|SIP/${EXTEN})
[pbx_config]
[ Context 'demo' created by 'pbx_config' ]
SNIP
[ Context 'ael-dundi-e164-local' created by 'pbx_ael' ]
Include ='ael-dundi-e164-canonical'[pbx_ael]
Include ='ael-dundi-e164-customers'[pbx_ael]
Include ='ael-dundi-e164-via-pstn' [pbx_ael]
[ Context 'parkedcalls' created by 'res_features' ]
Segmentation Fault (core dumped)
--
Here are the traces:
--
(gdb) bt
#0 0xfebe4d0c in strlen () from /lib/libc.so.1
#1 0xfec3a386 in _ndoprnt () from /lib/libc.so.1
#2 0xfec3d144 in snprintf () from /lib/libc.so.1
#3 0x080babea in show_dialplan_helper (fd=1, context=0x0, exten=0x0,
dpc=0x8047840, rinclude=0x0, includecount=0,
includes=0x8047640) at pbx.c:6156
#4 0x080bb1b7 in handle_show_dialplan (fd=1, argc=2, argv=0x80478e0) at
pbx.c:3663
#5 0x0808e1f0 in ast_cli_command (fd=1, s=0x0) at cli.c:1979
#6 0x08074167 in main (argc=135703622, argv=0x8047a5c) at asterisk.c:1388
--
--
(gdb) bt full
#0 0xfebe4d0c in strlen () from /lib/libc.so.1
No symbol table info available.
#1 0xfec3a386 in _ndoprnt () from /lib/libc.so.1
No symbol table info available.
#2 0xfec3d144 in snprintf () from /lib/libc.so.1
No symbol table info available.
#3 0x080babea in show_dialplan_helper (fd=1, context=0x0, exten=0x0,
dpc=0x8047840, rinclude=0x0, includecount=0,
includes=0x8047640) at pbx.c:6156
p = (struct ast_exten *) 0x81865b9
c = (struct ast_context *) 0x8186808
old_total_exten = 0
__PRETTY_FUNCTION__ = show_dialplan_helper
#4 0x080bb1b7 in handle_show_dialplan (fd=1, argc=2, argv=0x80478e0) at
pbx.c:3663
exten = 0x0
context = 0x0
counters = {total_context = 40, total_exten = 67, total_prio =
134, context_existence = 1, extension_existence = 1}
incstack = {0x1 Address 0x1 out of bounds, 0x0, 0x0, 0x80a8537
\215eô[^_ÉÃÇ\003\200, 0x0, 0x8120b95 logger.c,
0x37c Address 0x37c out of bounds, 0x811596c ast_verbose, 0x0,
0x0, 0x80476e8 `\025\025\b\001v\004\b\210\026\b,
0x80e3ee6 \203Ä0\215eô[^\211ø_ÉÃ\220©\200, 0x8047890
çrÄþp¯\027\b\002, 0x100 Address 0x100 out of bounds, 0x81a8830 Ò,
0x1b Address 0x1b out of bounds, 0xfec8c640 , 0x0, 0xfec8c640 ,
0xfeba2000 , 0xfec88000 \034\213\f, 0x0,
0x811d611 *CLI , 0x8151566 , 0x80476b4
[EMAIL PROTECTED]@\206µþøv\004\bDÑÃþ\027Ö\021\b\fw\004\bàv\004\b,
0xfec5a75a \203Ä\004\205Àt,Pè9], 0xfec8c640 , 0xfeba2000 ,
0xfec88000 \034\213\f, 0x80476d4 àv\004\b,
0xfec504ae \203Ä\0043É\213Eü\211\b_^[\213å]Ãj, 0xfec8c640 ,
0xfeb58640 , 0x80476f8 \230x\004\bÎ\006\a\b`\025\025\bÈ,
0xfec3d144 \203Ä\020\213L$\bÆ\001, 0x811d617 , 0x804770c ,
0x80476e0 Á, 0x0, 0x8163e88 ´\005\a\b\006,
0xc1 Address 0xc1 out of bounds, 0x8151566 , 0x8151560 *CLI ,
0x8047601 , 0x8163e88 ´\005\a\b\006, 0x0,
0x8047898 \002, 0x80706ce
\203Ä\020\215eô[^¸`\025\025\b_ÉÃPh\030Ö\021\bëÙ\211ö\213µ\204þÿÿ\205ötßj\024j\036j%\215uÈV1öèu8\a,
0x8151560 *CLI , 0xc8 Address 0xc8 out of bounds, 0x811d611
*CLI , 0x0, 0xfeba2000 , 0xfec88000 \034\213\f,
0x8047958 øy\004\b, 0x0, 0xfec8b800 , 0xfda18200 @\202¡ý, 0x0,
0xfec88000 \034\213\f, 0x4 Address 0x4 out of bounds,
0x0, 0x0, 0x804788c t\035\025\bçrÄþp¯\027\b\002, 0x8047888
àx\004\bt\035\025\bçrÄþp¯\027\b\002,
0x2f Address 0x2f out of bounds, 0x1 Address 0x1 out of bounds,
0x0, 0x1 Address 0x1 out of bounds, 0x0, 0x0,
0x43 Address 0x43 out of bounds, 0x0, 0x0, 0x0, 0xfbebdff8 , 0x0,
0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
0x5 Address 0x5 out of bounds, 0x5 Address 0x5 out of bounds, 0x0,
0x0,
0x812a83c No such command '%s' (type 'help' for help)\n, 0x81bedc3
s' ]\n, 0xfeba2000 , 0xfec88000 \034\213\f,
0x5f00796f Address 0x5f00796f out of bounds, 0xfec8c800 ,
0xfec8c800 , 0x80477d8 »ÔÃþ\020\225Èþ,
0xfec5a75a \203Ä\004\205Àt,Pè9], 0xfec8c800 , 0xfec8c800 ,
0x8047808 x\004\b2/Àþ\020\225Èþ,
0xfec3d4bb \203Ä\020\213L$\bÆ\001, 0xfec89510 , 0x0, 0xfec02e74
Re: [asterisk-users] Siemans SIP/PSTN phone S450
Adrian Marsh wrote: Hi All, Just added a Siemens DECT SIP/PSTN S450 phone to login to my A*k server, and I see Got SIP response 405 Method Not Allowed back from 192.168.3.64 but the phone seems to work ok. Any ideas where it falls over in the SIP protocol? I've included this in the debug below. It is in response to Notify packets because the Siemens phone doesn't support presence at the moment. It wont effect the operation of the phone or Asterisk at all. cheers, Paul. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Flash IDE
Hi We have a number offices accommodating 4-6 people each hence it is very important for PBX to be fanless and silent. We have been looking at using IDE flash disks also called DOM. The performance tests we have done so far satisfy our requirements, however we are concerned with DOM durability. We have installed debian and vanilla asterisk on 1GB DOM. All seems to work fine at the moment however will DOM last? How long it will last? Is anyone able to share similar experience? Any other information/tips? Regards, Juan _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Installing Asterisk on to CentOS 4
Hi expets, I have installed Asterisk 1.4.11 on CentOS4 successfully without any error. But when i am trying to start asterisk with following cmd i am getting unknown command. [EMAIL PROTECTED] ~]$ asterisk -vvc -bash: asterisk: command not found [EMAIL PROTECTED] ~]$ I checked modules and other configuration files which are installed correctly. Please help me to locate this problem. Thank You - Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out.___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Asterisk on to CentOS 4
Add /usr/sbin to your PATH, or run /usr/sbin/asterisk. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Asterisk on to CentOS 4
I installed it using yum from the atrpms repo and it all seems to work. Did you compile from source? On 9/11/07, Abdul [EMAIL PROTECTED] wrote: Hi expets, I have installed Asterisk 1.4.11 on CentOS4 successfully without any error. But when i am trying to start asterisk with following cmd i am getting unknown command. [EMAIL PROTECTED] ~]$ asterisk -vvc -bash: asterisk: command not found [EMAIL PROTECTED] ~]$ I checked modules and other configuration files which are installed correctly. Please help me to locate this problem. Thank You Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.
Sir, I want to dedicate two or three Panasonic port to communicate with Asterisk and vise-versa. I am having Panasonic pbx 1232. Rajeev. Hello, 2007/9/10, C F [EMAIL PROTECTED]: Which Panasonic PBX? On 9/10/07, Sanspareils Greenlans [EMAIL PROTECTED] wrote: Sir, I am having Asterisk pbx which is running without any problem now i want to connect this with Panasonic pbx with FXS port so, if any body want to call panasonic users than he will call or vise-versa. How ? Do you plan to dedicate Panasonic PBX FXS ports to act as a trunk or would dedicate one port for each Asterisk user ? -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070910/1b2b2 6a8/attachment-0001.htm -- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Prevent multiple sip registrations
Hi all, Is there anyway i can prevent multiple sip registrations from different IPs using single username in asterisk. Does asterisk provide any aid in this respect? As far as my knowledge is concerned i dont think there is any support for this in asterisk, so i think i'll have to makeup a script which sniffs sip packets coming for asterisk and detect for multiple register requests coming from different IPs for the same username. Can anybody help? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent multiple sip registrations
I believe you can use the host= to configure the allowed IP in sip.conf From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rizwan Hisham Sent: 11 September 2007 11:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Prevent multiple sip registrations Hi all, Is there anyway i can prevent multiple sip registrations from different IPs using single username in asterisk. Does asterisk provide any aid in this respect? As far as my knowledge is concerned i dont think there is any support for this in asterisk, so i think i'll have to makeup a script which sniffs sip packets coming for asterisk and detect for multiple register requests coming from different IPs for the same username. Can anybody help? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent multiple sip registrations
We cant do that. Thats becoz the original user may change his/her location which will result in change of ip address. We have to set host=dynamic for allownig the original user to register from anywhere. So any other ideas? On 9/11/07, Adrian Marsh [EMAIL PROTECTED] wrote: I believe you can use the host= to configure the allowed IP in sip.conf From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rizwan Hisham Sent: 11 September 2007 11:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Prevent multiple sip registrations Hi all, Is there anyway i can prevent multiple sip registrations from different IPs using single username in asterisk. Does asterisk provide any aid in this respect? As far as my knowledge is concerned i dont think there is any support for this in asterisk, so i think i'll have to makeup a script which sniffs sip packets coming for asterisk and detect for multiple register requests coming from different IPs for the same username. Can anybody help? -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent multiple sip registrations
On 9/11/07, Rizwan Hisham [EMAIL PROTECTED] wrote: We cant do that. Thats becoz the original user may change his/her location which will result in change of ip address. We have to set host=dynamic for allownig the original user to register from anywhere. So any other ideas? So, if he can change IP, then he probably is changing? Why do you get multiple registrations? SIP secret should be known only for that user (or if he has HW device - just for device). Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flash IDE
On Tue, 11 Sep 2007, Juan Sandro wrote: Hi We have a number offices accommodating 4-6 people each hence it is very important for PBX to be fanless and silent. We have been looking at using IDE flash disks also called DOM. The performance tests we have done so far satisfy our requirements, however we are concerned with DOM durability. We have installed debian and vanilla asterisk on 1GB DOM. All seems to work fine at the moment however will DOM last? How long it will last? Is anyone able to share similar experience? Any other information/tips? You could read the archives from a week or 2 ago under the heading: Build your own appliance I use these deices, but I unload them entirely into RAM. I have seen devices (eary mikrotik routers?) with them as live (and ext3 no less!) filesystems, but I would be very concerend about their lifespan. One thing to note and this might well shaft you is that they use POI mode rather than DMA (or at least the ones I'm using do) so they will really crowbar the bus cpu when doing transfers to/from them, however with only 4-6 people and not doing much like writing voicemail, etc. you may not notice it. If you're sticking a normal disctibution on it, I'd suggest dumping the DOM and getting a laptop type IDE/SATA drive and using that instead. It's not silent, but will be very quiet. Gordon ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent multiple sip registrations
well he does not have access to hi sip settings, so he cant edit the host=differentIP every time he moves or registers from anyother place. Actually he should be able to register from anywhere in the world but once he has registered with us, i dont want anyone else to register with my asterisk using his credentials. On 9/11/07, Atis [EMAIL PROTECTED] wrote: On 9/11/07, Rizwan Hisham [EMAIL PROTECTED] wrote: We cant do that. Thats becoz the original user may change his/her location which will result in change of ip address. We have to set host=dynamic for allownig the original user to register from anywhere. So any other ideas? So, if he can change IP, then he probably is changing? Why do you get multiple registrations? SIP secret should be known only for that user (or if he has HW device - just for device). Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent multiple sip registrations
On 9/11/07, Rizwan Hisham [EMAIL PROTECTED] wrote: well he does not have access to hi sip settings, so he cant edit the host=differentIP every time he moves or registers from anyother place. Actually he should be able to register from anywhere in the world but once he has registered with us, i dont want anyone else to register with my asterisk using his credentials. So, why it is a problem to give really secure password? Well, actually some idea popped out of my mind just now.. You can have SIP users in realtime, and create DB trigger.. As soon as ipaddr field is changed (SIP registration happened), SET host=ipaddr. Maybe you also want set back host=dynamic when regseconds (time before re-registration) ends. Anyway - creating custom scenarios should be possible. Regards, Atis -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent multiple sip registrations
Rizwan Hisham wrote: well he does not have access to hi sip settings, so he cant edit the host=differentIP every time he moves or registers from anyother place. Actually he should be able to register from anywhere in the world but once he has registered with us, i dont want anyone else to register with my asterisk using his credentials. Then make sure nobody else knows his credentials. This isn't rocket science. How exactly do you propose to determine of the user moved the device to a new location .vs. a 2nd device trying to register with the same credentials. In any case, Asterisk does not have any facilities to do what you want to do. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flash IDE
On Tue, 11 Sep 2007 04:04:27 -0500, Juan Sandro wrote: Hi We have a number offices accommodating 4-6 people each hence it is very important for PBX to be fanless and silent. We have been looking at using IDE flash disks also called DOM. The performance tests we have done so far satisfy our requirements, however we are concerned with DOM durability. We have installed debian and vanilla asterisk on 1GB DOM. All seems to work fine at the moment however will DOM last? How long it will last? Is anyone able to share similar experience? Any other information/tips? Why reinvent the wheel...try Astlinux. It's built to be run from flash devices and includes both Asterisk and edge routing capability for small PCs or embeded hardware. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 c713-201-1262 skype mjgraves fwd 54245 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Another State Of The Punctuation Mark question - Vonage
There was a flurry of Vonage is going to unlock SIP activity last year; did anything productive ever come of it? Are *you* using your Vonage lines directly into Asterisk? In lieu of that, for a 4 line small business that doesn't need to pay Vonage $150 a month, who? Broadvoice? Someone else? I'm a touch unimpressed with the fact that BV's website *won't quote you BYOD pricing* until you actually place the damn order -- or so it appears to my eyes. 727. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Asterisk on to CentOS 4
Abdul wrote: Hi expets, I have installed Asterisk 1.4.11 on CentOS4 successfully without any error. But when i am trying to start asterisk with following cmd i am getting unknown command. [EMAIL PROTECTED] ~]$ asterisk -vvc -bash: asterisk: command not found [EMAIL PROTECTED] ~]$ I checked modules and other configuration files which are installed correctly. Please help me to locate this problem. Thank You Try the command as root: [EMAIL PROTECTED] ~]$ su - *enter password* [EMAIL PROTECTED] ~]# asterisk -cvv Rgds, Ove Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.485 / Virus Database: 269.13.14/999 - Release Date: 10.09.2007 17:43 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent multiple sip registrations
On Sep 11, 2007, at 7:29 AM, Eric ManxPower Wieling wrote: Rizwan Hisham wrote: well he does not have access to hi sip settings, so he cant edit the host=differentIP every time he moves or registers from anyother place. Actually he should be able to register from anywhere in the world but once he has registered with us, i dont want anyone else to register with my asterisk using his credentials. Then make sure nobody else knows his credentials. This isn't rocket science. How exactly do you propose to determine of the user moved the device to a new location .vs. a 2nd device trying to register with the same credentials. In any case, Asterisk does not have any facilities to do what you want to do. How about some method of checking for a current registration when a new one is received. If presently registered at a different IP then disallow new attempt. No I dont know any existing tool within * to accomplish. Anyone else? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.
You can do that. Just make sure you have a proper dialplan in asterisk, among others make sure you teach your users and configure properly how to transfer back to the Panasonic, you will need to use app_flash with features.conf On 9/11/07, Sanspareils Greenlans [EMAIL PROTECTED] wrote: Sir, I want to dedicate two or three Panasonic port to communicate with Asterisk and vise-versa. I am having Panasonic pbx 1232. Rajeev. Hello, 2007/9/10, C F [EMAIL PROTECTED]: Which Panasonic PBX? On 9/10/07, Sanspareils Greenlans [EMAIL PROTECTED] wrote: Sir, I am having Asterisk pbx which is running without any problem now i want to connect this with Panasonic pbx with FXS port so, if any body want to call panasonic users than he will call or vise-versa. How ? Do you plan to dedicate Panasonic PBX FXS ports to act as a trunk or would dedicate one port for each Asterisk user ? -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070910/1b2b2 6a8/attachment-0001.htm -- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent multiple sip registrations
But then how do you know which is the correct user? This is where the whole point of secrets/passwords should come into play. If no-one else knows his details, then no-one else can register. In the land of IP, you can't even guarantee that a remote ends IP will be the same from minute to minute.. (eg user connects by both wifi and LAN, initially connects via LAN then gets up and moves to wifi - as many of my users do daily). ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.
On 9/11/07, Olivier [EMAIL PROTECTED] wrote: Hi, So, if you dedicate PBX ports to serve as a trunk, you're likely to loose the abilty to forward DID calls : when a call for an Asterisk user comes into Panasonic PBX, it will be forwarded to Panasonic FXS trunk ports. Then, Asterisk should have no mean to decode to which extension, the call has to be forwarded, has it comes from an FXO port which won't carry any data such as CallerID. I'm not sure exactly what you are saying but if you meant to say that if the panasonic side uses station (FXS) ports and on asterisk FXO ports then read on, otherwise you are right. In general this is not true, CallerID will be passed on with the right cards in the system (in particular panasonic TD1232). DID/extension can be passed with almost every semi-decent system. I have done it with the Avaya Partner, Panasonic and ohters. In most cases you setup a VoiceMail system on the host (legacy) machine, then put the FXS ports on the legacy PBX in a group as a DTMF integrated voicemail system, what happens next is that the host PBX sends you DTMF for the DID/Extension after asterisk picked up the phone before it is bridged. Works for me. For some documentation on it: http://www.voip-info.org/wiki/view/Asterisk-Partner+ACS I'm not 100% sure of that but that's the way analog ports works here, on some legacy PBX : analog port means no service. regards ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] nat=yes
Dear Benjamin; So in that case, when we set nat = yes? For what we do this? C F, I have nat=yes set by default for all my extensions(with canreinvite=no). And things work fine. Bilal, about Asterisk sending packets to public/private : Asterisk will send packets to the public IP advertised by the msg/recv from address. It is the NAT's headache on the endpoints network periphery to send the response from Asterisk to the endpoint. C F wrote: If you set yes then asterisk assumes that the address its coming from is not the same as the UA thinks it is. most devices will not operate properly if set to yes when they are in fact local. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. http://smallbusiness.yahoo.com/webhosting ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent multiple sip registrations
The whole point of doing this is because if the user gives away his username/password to his friends or relative and allows them to use his account, that way we r gona have a lot more traffic in our asterisk server. Also we charge our users a fix amount of money every month for their account so if any user gives out his username and password then his account is more likely to do 2 to 3 times the calls as compared to aan account which is used by only one user. So ultimately we lose money. On 9/11/07, Adrian Marsh [EMAIL PROTECTED] wrote: But then how do you know which is the correct user? This is where the whole point of secrets/passwords should come into play. If no-one else knows his details, then no-one else can register. In the land of IP, you can't even guarantee that a remote ends IP will be the same from minute to minute.. (eg user connects by both wifi and LAN, initially connects via LAN then gets up and moves to wifi - as many of my users do daily). ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent multiple sip registrations
My requirement is to prevent registrations for aan account if that account is already registered with a user. On 9/11/07, Adrian Marsh [EMAIL PROTECTED] wrote: But then how do you know which is the correct user? This is where the whole point of secrets/passwords should come into play. If no-one else knows his details, then no-one else can register. In the land of IP, you can't even guarantee that a remote ends IP will be the same from minute to minute.. (eg user connects by both wifi and LAN, initially connects via LAN then gets up and moves to wifi - as many of my users do daily). ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite
Dear C F; So in that case, if I placed canrenvite=yes for both endpoint, it is not condition that traffic will be directly via the endpoint while signaling via Asterisk as still Asterisk should detect whethor it is necessary to stay in the path or not? Please advise. How can I know that the traffic went directly between the endpoints and did not go via the asterisk? Regards Bilal Ghayad Mobile: 009659849460 - By default assuming you have no global setting otherwise, if asterisk doesnt see a need to stay in the path then it wont. hence if it has to transcode between different codecs, capture DTMF or different protocols it will stay in the path. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send traffic to each other directly or it will be via Asterisk? Regards Bilal Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.11, res_features.so, SegFault
Open a bug in http://bugs.digium.com/ including all the information
you provided here.
Also remember to read the bugs guidelines before openning the bug,
this might be already reported.
Regards
On 9/11/07, Bruce McAlister [EMAIL PROTECTED] wrote:
Hi All,
I have a really strange issue occuring where if I run show dialplan or
dialplan show or dialplan show parkedcalls, then asterisk dumps core.
It only appears to happen with contexts that are created within
res_features. I am able to display all my other dialplans, but, every
time I try to just do a normal dialplan show asterisk core dumps
(Segmentation Fault).
My environment is as follows:
Asterisk v 1.4.11
Solaris 10 update 3 (11/06), patched current
gcc v3.4.3
example console output
--
*CLI dialplan show
[ Context 'default' created by 'pbx_config' ]
Include ='demo'
[pbx_config]
[ Context 'page' created by 'pbx_config' ]
'_X.' = 1. Macro(page|SIP/${EXTEN})
[pbx_config]
[ Context 'demo' created by 'pbx_config' ]
SNIP
[ Context 'ael-dundi-e164-local' created by 'pbx_ael' ]
Include ='ael-dundi-e164-canonical'[pbx_ael]
Include ='ael-dundi-e164-customers'[pbx_ael]
Include ='ael-dundi-e164-via-pstn' [pbx_ael]
[ Context 'parkedcalls' created by 'res_features' ]
Segmentation Fault (core dumped)
--
Here are the traces:
--
(gdb) bt
#0 0xfebe4d0c in strlen () from /lib/libc.so.1
#1 0xfec3a386 in _ndoprnt () from /lib/libc.so.1
#2 0xfec3d144 in snprintf () from /lib/libc.so.1
#3 0x080babea in show_dialplan_helper (fd=1, context=0x0, exten=0x0,
dpc=0x8047840, rinclude=0x0, includecount=0,
includes=0x8047640) at pbx.c:6156
#4 0x080bb1b7 in handle_show_dialplan (fd=1, argc=2, argv=0x80478e0) at
pbx.c:3663
#5 0x0808e1f0 in ast_cli_command (fd=1, s=0x0) at cli.c:1979
#6 0x08074167 in main (argc=135703622, argv=0x8047a5c) at asterisk.c:1388
--
--
(gdb) bt full
#0 0xfebe4d0c in strlen () from /lib/libc.so.1
No symbol table info available.
#1 0xfec3a386 in _ndoprnt () from /lib/libc.so.1
No symbol table info available.
#2 0xfec3d144 in snprintf () from /lib/libc.so.1
No symbol table info available.
#3 0x080babea in show_dialplan_helper (fd=1, context=0x0, exten=0x0,
dpc=0x8047840, rinclude=0x0, includecount=0,
includes=0x8047640) at pbx.c:6156
p = (struct ast_exten *) 0x81865b9
c = (struct ast_context *) 0x8186808
old_total_exten = 0
__PRETTY_FUNCTION__ = show_dialplan_helper
#4 0x080bb1b7 in handle_show_dialplan (fd=1, argc=2, argv=0x80478e0) at
pbx.c:3663
exten = 0x0
context = 0x0
counters = {total_context = 40, total_exten = 67, total_prio =
134, context_existence = 1, extension_existence = 1}
incstack = {0x1 Address 0x1 out of bounds, 0x0, 0x0, 0x80a8537
\215eô[^_ÉÃÇ\003\200, 0x0, 0x8120b95 logger.c,
0x37c Address 0x37c out of bounds, 0x811596c ast_verbose, 0x0,
0x0, 0x80476e8 `\025\025\b\001v\004\b\210\026\b,
0x80e3ee6 \203Ä0\215eô[^\211ø_ÉÃ\220(c)\200, 0x8047890
çrÄþp¯\027\b\002, 0x100 Address 0x100 out of bounds, 0x81a8830 Ò,
0x1b Address 0x1b out of bounds, 0xfec8c640 , 0x0, 0xfec8c640 ,
0xfeba2000 , 0xfec88000 \034\213\f, 0x0,
0x811d611 *CLI , 0x8151566 , 0x80476b4
Ôv\004\b(r)[EMAIL PROTECTED]@\206µþøv\004\bDÑÃþ\027Ö\021\b\fw\004\bàv\004\b,
0xfec5a75a \203Ä\004\205Àt,Pè9], 0xfec8c640 , 0xfeba2000 ,
0xfec88000 \034\213\f, 0x80476d4 àv\004\b,
0xfec504ae \203Ä\0043É\213Eü\211\b_^[\213å]Ãj, 0xfec8c640 ,
0xfeb58640 , 0x80476f8 \230x\004\bÎ\006\a\b`\025\025\bÈ,
0xfec3d144 \203Ä\020\213L$\bÆ\001, 0x811d617 , 0x804770c ,
0x80476e0 Á, 0x0, 0x8163e88 ´\005\a\b\006,
0xc1 Address 0xc1 out of bounds, 0x8151566 , 0x8151560 *CLI ,
0x8047601 , 0x8163e88 ´\005\a\b\006, 0x0,
0x8047898 \002, 0x80706ce
\203Ä\020\215eô[^¸`\025\025\b_ÉÃPh\030Ö\021\bëÙ\211ö\213µ\204þÿÿ\205ötßj\024j\036j%\215uÈV1öèu8\a,
0x8151560 *CLI , 0xc8 Address 0xc8 out of bounds, 0x811d611
*CLI , 0x0, 0xfeba2000 , 0xfec88000 \034\213\f,
0x8047958 øy\004\b, 0x0, 0xfec8b800 , 0xfda18200 @\202¡ý, 0x0,
0xfec88000 \034\213\f, 0x4 Address 0x4 out of bounds,
0x0, 0x0, 0x804788c t\035\025\bçrÄþp¯\027\b\002, 0x8047888
àx\004\bt\035\025\bçrÄþp¯\027\b\002,
0x2f Address 0x2f out of bounds, 0x1 Address 0x1 out of bounds,
0x0, 0x1 Address 0x1 out of bounds, 0x0, 0x0,
0x43 Address 0x43 out of bounds, 0x0, 0x0, 0x0, 0xfbebdff8 , 0x0,
0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
0x5 Address 0x5 out of bounds, 0x5 Address 0x5 out of bounds, 0x0,
0x0,
0x812a83c No
Re: [asterisk-users] Prevent multiple sip registrations
Hmmm. Then SIP is not your solution. SIP servers have no ability to tell one user from another if they share secrets. Strongly suggest that you change the ethos behind how you're manage you're users. Unfortunately, with the business plan as-is, you're using end-user trust not to abuse the system. The *ONLY* way I can see a setup working for you, is if you do use a backend DB to log when the user last registered, and set a time limit (eg if 1hour since last change then reject) - but its principally flawed (what if a genuine user does move from wifi to fixed and back again), and would probably only work well for home-users who aren't mobile at all. Not sure how you'd implement this into Asterisk though. Adrian Marsh ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linux-HA and Asterisk
We have gotten stuck trying to get a highly available Asterisk cluster fully functional. We used Linux-HA with Asterisk 1.4.11 on privtae IP's behind the virtual public IP. I got as far as getting phones registered and being able to place calls that rang and you could answer, but there was no audio. So, I enabled RTP debugging and discovered Asterisk was still attempting to send the audio packets to the phones private address, even though the device was set up as NAT=yes. externip was set to the virtual public IP. Any thoughts on clearing the final hurdle? Also, we are using Polycom phones. Thanks, Mike Clark ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent multiple sip registrations
On 9/11/07, Rizwan Hisham [EMAIL PROTECTED] wrote: The whole point of doing this is because if the user gives away his username/password to his friends or relative and allows them to use his account, that way we r gona have a lot more traffic in our asterisk server. Also we charge our users a fix amount of money every month for their account so if any user gives out his username and password then his account is more likely to do 2 to 3 times the calls as compared to aan account which is used by only one user. So ultimately we lose money. Ok, i think the way i described few mails ago would certainly do that. You can customize DB triggers as much as you want.. So, just look into realtime and mysql triggers. Just probably you would need to reset them once per certain interval (if you want to allow your users to change IP once per day or something like that) Regards, Atis On 9/11/07, Adrian Marsh [EMAIL PROTECTED] wrote: But then how do you know which is the correct user? This is where the whole point of secrets/passwords should come into play. If no-one else knows his details, then no-one else can register. In the land of IP, you can't even guarantee that a remote ends IP will be the same from minute to minute.. (eg user connects by both wifi and LAN, initially connects via LAN then gets up and moves to wifi - as many of my users do daily). ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, IT Responsible of BEST Riga, [EMAIL PROTECTED] ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone: +1 800 7502835 [Toll free, USA] ?BEST? - www.BEST.eu.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite
Don't know about IAX. As for SIP, You will know what ip address and port the audios should be transmitted to by looking at the sdp session. Just goto the * console and enable sip debug. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Tuesday, September 11, 2007 10:14 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] canreinvite Dear C F; So in that case, if I placed canrenvite=yes for both endpoint, it is not condition that traffic will be directly via the endpoint while signaling via Asterisk as still Asterisk should detect whethor it is necessary to stay in the path or not? Please advise. How can I know that the traffic went directly between the endpoints and did not go via the asterisk? Regards Bilal Ghayad Mobile: 009659849460 - By default assuming you have no global setting otherwise, if asterisk doesnt see a need to stay in the path then it wont. hence if it has to transcode between different codecs, capture DTMF or different protocols it will stay in the path. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send traffic to each other directly or it will be via Asterisk? Regards Bilal Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite
How can I know that the traffic went directly between the endpoints and did not go via the asterisk? I'm sure there are many ways to do this one way would be to do rtp debug on the cli and watch for media packets another would be to do tcpdump on the command line and watch for packets there. Regards Bilal Ghayad Mobile: 009659849460 - By default assuming you have no global setting otherwise, if asterisk doesnt see a need to stay in the path then it wont. hence if it has to transcode between different codecs, capture DTMF or different protocols it will stay in the path. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send traffic to each other directly or it will be via Asterisk? Regards Bilal Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite
The others answered correctly personal I like using rtp debug. As for making sure in the DialPlan that the RTP goes end to end without asterisk. 1. Make sure they both use the same codec and protocol. 2. Don't put any options in app_dial, like tTwW or anything else that will force asterisk to stay in the stream to listen for DTMF. On 9/11/07, bilal ghayyad [EMAIL PROTECTED] wrote: Dear C F; So in that case, if I placed canrenvite=yes for both endpoint, it is not condition that traffic will be directly via the endpoint while signaling via Asterisk as still Asterisk should detect whethor it is necessary to stay in the path or not? Please advise. How can I know that the traffic went directly between the endpoints and did not go via the asterisk? Regards Bilal Ghayad Mobile: 009659849460 - By default assuming you have no global setting otherwise, if asterisk doesnt see a need to stay in the path then it wont. hence if it has to transcode between different codecs, capture DTMF or different protocols it will stay in the path. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send traffic to each other directly or it will be via Asterisk? Regards Bilal Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] exit ChanSpy with DTMF
Part of a supervisor menu I'm writing requires that I allow the supervisor to choose to ChanSpy a channel from the main menu then return back to the menu (dialplan) to choose other options when she's done. Is there a way to 'exit' ChanSpy and continue down the dialplan? Or is a caller stuck in ChanSpy until they hangup the phone? Thanks. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent multiple sip registrations
On 9/11/07, Rizwan Hisham [EMAIL PROTECTED] wrote: My requirement is to prevent registrations for aan account if that account is already registered with a user. That is a perfectly valid requirement. This is not a SIP protocol issue. This is a SIP Registrar implementation/policy issue. If a SIP Registrar implementation could limit the number of Contact bindings per AoR then this goal can be accomplished. Asterisk's SIP Registrar does not support this today. However, it should be a relatively minor enhancement to add this in the Asterisk SIP Registrar itself (as opposed to implementing this through back-end database hooks). As an example, the OpenSER's SIP Registrar supports a parameter called max_contact to accomplish the same goal: http://www.openser.org/docs/modules/1.2.x/registrar.html#AEN199 Raj ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dtmfmode rfc2833 and info
I have two asterisk machines setup. M1 is asterisk 1.4.11 connected with a PRI M2 is asterisk 1.2.23 connected to M1 over sip. When M2 calls out through M1 and tries to use SendDTMF() in an agi I get varied results. 1) In sip.conf if dtmfmode=rfc2833 I do not hear the sendDTMF() 2) In sip.conf if dtmfmode=info I do hear the sendDTMF() Why is there a difference? Jerry ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on NGINX Server?
Was the AGI Server to write dialplans in any programming language in Asterisk assumed to be configured for the apache web server? No, it's not assumed to be for any web server at all... AGI scripts can be written in any language that reads from STDIN and writes to STDOUT, or can listen on a network socket (in the case of FastAGI). --- Jared Smith Community Relations Manager Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Another State Of The Punctuation Mark question - Vonage
Vonage has a business offering, but they aren't really structured to provide business quality support. I wouldn't use them for a business. For several years now, we've used VoicePulse Connect http://connect.voicepulse.com/ for our Asterisk IAX and SIP trunks. Ravi and KP are both technical guys and know Asterisk extremely well. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth Sent: Tuesday, September 11, 2007 5:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Another State Of The Punctuation Mark question - Vonage There was a flurry of Vonage is going to unlock SIP activity last year; did anything productive ever come of it? Are *you* using your Vonage lines directly into Asterisk? In lieu of that, for a 4 line small business that doesn't need to pay Vonage $150 a month, who? Broadvoice? Someone else? I'm a touch unimpressed with the fact that BV's website *won't quote you BYOD pricing* until you actually place the damn order -- or so it appears to my eyes. 727. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)
http://vonmag.com/editorial/web-exclusives/mark-spencer-digium-is-growing-up Seems the Adtran relationship goes way back... Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIPAddHeader cmd from Realtime MySQL, not getting all the 'appdata' field
Hi All, I'm doing some simple paging functions and using the SIPAddHeader cmd. * 1.2 branch. Using it in the extensions.conf file, it works fine: exten = _*2XX,1,SIPAddHeader(Call-Info: sip:\;answer-after=0) in * console: lab2*CLI -- Executing SIPAddHeader(SIP/204-0818dcd0, Call-Info: sip:;answer-after=0) in new stack When i put the same cmd in Realtime MySQL, I get this: -- Executing SIPAddHeader(SIP/53683-7ca5, ;answer-after=0) in new stack Pulling the command from MySQL I loos the 'Call-Info: sip:' part of the appdata field Here is a direct query form MySQL: mysql select * from extensions where id like '5801'\G; *** 1. row *** id: 5801 context: demo exten: *107 priority: 1 app: SIPAddHeader appdata: Call-Info: sip:\;answer-after=0 accountcode: notes: 1 row in set (0.01 sec) I'm wondering if the colons or the back slash is affecting this coming into asterisk? Thanks. JR -- JR Richardson Engineering for the Masses ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flash IDE
Juan Sandro wrote: Hi We have a number offices accommodating 4-6 people each hence it is very important for PBX to be fanless and silent. We have been looking at using IDE flash disks also called DOM. The performance tests we have done so far satisfy our requirements, however we are concerned with DOM durability. We have installed debian and vanilla asterisk on 1GB DOM. All seems to work fine at the moment however will DOM last? How long it will last? Is anyone able to share similar experience? Any other information/tips? I worried a lot about the same, in the end I went for a small laptop drive for safety (it's inaudible) However, this came up on slashdot recently and if you search around the logic seems to be that: - Flash rewrites quite a few times - The good stuff has wear levelling so that most roughly speaking the whole thing should work until it suddenly all fails - Given a big enough drive with a fair bit of free space then you should find it hard to wear it out in less than quite a few years even if you are hitting it quite hard (probably multiples of this). Simply do the maths to get the rough life So basically it seems that given a large enough flash drive with decent wear levelling the lifetime should be completely ample... ...Thats the theory anyway. I feel quite bullish about the whole thing, but I think I would avoid the *really* discounted cheapo flash drives since they may not have the correct wear levelling. Decent brand names should be fine though (and you can google for details on their specs) Ed W ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Installed X100p
Steve Totaro wrote: I have had Digium tech support tell me to do the same thing I'm hoping that wasn't the final conclusion in the tech support debugging process. If it was, than I am very sorry to hear that, and will make note of it. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. Thanks, Steve G B wrote: Hi, I appreciate the help. I called the vendor of the card and they recommended removing all of the PCI cards on the system (including the video card), and moving the card to a new PCI slot. I did all of them together, ran the system headless, and ssh'ed in remotely. It worked! haha... This must be proof that I have purchased a real piece of @#$. Thanks for all of your help. Date: Sat, 8 Sep 2007 02:41:50 +0300 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] New Installed X100p On Fri, Sep 07, 2007 at 08:45:11AM -0700, G B wrote: Hi Tzafrir, I am not sure what to look for, so I haveattached both the contents of /var/log/kern.log as well as the outputof dmesg. If you are looking for something specific, I simply asked for a few lines around that message. Anyway, the relevant lines are: Relevant lines: [ 39.337207] Failed to initailize DAA, giving up... [ 39.337283] wcfxo: probe of :00:0c.0 failed with error -5 No more details. This may be a defective card. I have also seen some cases where some voodoo at the PCI layer was required (e.g: passing the boot option pci=noacpi). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Can you find the hidden words? Take a break and play Seekadoo! Play now! http://club.live.com/seekadoo.aspx?icid=seek_wlmailtextlink ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 56k modem configuration
Andrea Spadaccini wrote: Hello everybody, I've got a 56k usb modem, lsusb says: Bus 002 Device 002: ID 0572:130 Conexant Systems (Rockwell), Inc. I'd like to let it work with Asterisk. I think that I should use chan_modem and/or chan_modem_bestdata, but I found little or no documentation. Can anybody please post some instructions? I would be very surprised if chan_modem actually works... I don't think I've *ever* seen it setup before. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Installed X100p
Matthew Fredrickson wrote: Steve Totaro wrote: I have had Digium tech support tell me to do the same thing I'm hoping that wasn't the final conclusion in the tech support debugging process. If it was, than I am very sorry to hear that, and will make note of it. Thanks, yes, that was the final resolution. I have also heard That motherboard or that server is not supported. Again, this was quite some time ago and Digium has changed as a company as well as the product line, the whole entity has matured. Might be a non-issue now. Let me ask you this, is using a T1 card for ISDN data supported now? That one irked me since it was a selling point, but when calling for support I was told, It can do it but it is not supported. and info on the net was VERY sparse for accomplishing this (circa 2003) Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flash IDE
CF flash deviced work fine provided that a) The CF has a wear leveling controller inside (not all do, especially the cheap ones) so even a ext2 filesystem wan't create problems b) You use a distro with read only (or partial write) filesystem .i.e logs to ram or remote server etc Other than that we have deployed a very large number of devices with embedded linux in a CF (not all of them asterisk) with minimal problems Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Juan Sandro Sent: Tuesday, September 11, 2007 12:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Flash IDE Hi We have a number offices accommodating 4-6 people each hence it is very important for PBX to be fanless and silent. We have been looking at using IDE flash disks also called DOM. The performance tests we have done so far satisfy our requirements, however we are concerned with DOM durability. We have installed debian and vanilla asterisk on 1GB DOM. All seems to work fine at the moment however will DOM last? How long it will last? Is anyone able to share similar experience? Any other information/tips? Regards, Juan _ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] exit ChanSpy with DTMF
On 9/11/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Part of a supervisor menu I'm writing requires that I allow the supervisor to choose to ChanSpy a channel from the main menu then return back to the menu (dialplan) to choose other options when she's done. Is there a way to 'exit' ChanSpy and continue down the dialplan? Or is a caller stuck in ChanSpy until they hangup the phone? In 1.4, they are stuck. -trunk has an option to allow them to escape out to a context using a DTMF digit; check the changelog in SVN for details. I'm not sure how portable it might be back to 1.4/1.2 if you want to attempt that. -- j. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 56k modem configuration
Ciao Matthew, I would be very surprised if chan_modem actually works... I don't think I've *ever* seen it setup before. Well.. So there's no hope to make that modem work with Asterisk, right? Thanks, -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.it - Tel: +39 (0) 95 7224945 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stop log/debug messages into /var/log/messages
Hi Benjamin; I am also interested in the same issue, but I would like to know how you can know where these logs are stored (in which file and path)? I readed that syslog, can you please help me about that? Regards Bilal Ghayad Mobile: 00965 9849460 --- When you access the A*k console, is this via a tty connection (ssh/telnet), or actually on the physical console of the server? I don't think it's A*k that's directly logging to the console - the config doesn't show that... I'm guessing, that you're accessing A*k via the local terminal, and that your syslog config for the server is configured to log this to messsages Maybe.. hmmm. interesting. need to investigate syslog now. Even me thinks, as far as I've read(abt logger and the existing configuration), it shouldn't be writing to any syslogs. btw, am accessing the * console via ssh. thanks for ur help. - Benjamin Jacob. Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flash IDE
Gordon Henderson wrote: One thing to note and this might well shaft you is that they use POI mode rather than DMA (or at least the ones I'm using do) so they will really crowbar the bus cpu when doing transfers to/from them, however with only 4-6 people and not doing much like writing voicemail, etc. you may not notice it. For real! I see the BIOS, then I see GRUB Loading stage 1.5 and then a good 60 seconds go by before the kernel and initrd have been loaded and control switches over to them. Stock kernel on CentOS 4.4 Moj ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400P not answering or making calls
Hello, I have recently purchased a TDM400P card with one FXO expansion card, and I'm having problems. The card does not pick up incoming calls. Asterisk detects the ringing line and rings various SIP phones as required. When a sip phone answers, the sip user hears nothing and the PSTN user continues to hear ringing. Here is the asterisk output for an incoming call: - == Starting post polarity CID detection on channel 3 -- Starting simple switch on 'Zap/3-1' -- Executing Set(Zap/3-1, CALLERID(all)=call to 322817) in new stack -- Executing Dial(Zap/3-1, Local/[EMAIL PROTECTED]|45) in new stack -- Called [EMAIL PROTECTED] -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/203|30) in new stack -- Called 201 -- SIP/203-0814c448 is ringing -- Local/[EMAIL PROTECTED],1 is ringing -- SIP/201-081445e8 is ringing -- SIP/201-081445e8 answered Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 stopped sounds -- Local/[EMAIL PROTECTED],1 answered Zap/3-1 == Spawn extension (special, 601, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' == Spawn extension (special, 601, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' Sep 11 19:25:55 WARNING[3073]: chan_zap.c:3934 zt_handle_event: Ring/Off-hook in strange state 6 on channel 3 == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' - I have set opermode to 'UK' (as I'm in the UK), and dmesg confirms the setting. Outgoing calls also fail, the SIP user hears nothing, yet asterisk claims that the call has been picked up. Here is the asterisk log, looks perfectly normal: - -- Executing Dial(SIP/201-08154a38, Zap/3/0800800800) in new stack -- Called 3/0800800800 -- Zap/3-1 answered SIP/201-08154a38 - I am running Debian Etch with kernel 2.6.18 and asterisk version 1.2.13. I'm beginning to think it's a fault with the expansion card... anyone else got any ideas? Oh and the BT line is fine, it works (as well as can be expected) with a X100P card I have. Thanks, Tom Attached: dmesg output: apata Telephony Interface Registered on major 196 Zaptel Version: 1.2.16 Zaptel Echo Canceller: MG2 ACPI: PCI Interrupt :00:0b.0[A] - Link [LNKD] - GSI 12 (level, low) - IRQ 12 Freshmaker version: 73 Freshmaker passed register test Module 0: Not installed Module 1: Not installed Module 2: Installed -- AUTO FXO (UK mode) Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (1 modules) Registered tone zone 4 (United Kingdom) - zapata.conf: [channels] language=en context=incoming signalling=fxs_ks busydetect=yes busycount=4 callprogress=no relaxdtmf=yes callwaiting=no callwaitingcallerid=no threewaycalling=no transfer=yes cancallforward=yes usecallerid=no cidsignalling=v23 cidstart=polarity callerid=no hidecallerid=no echotraining=yes echocancel=yes echocancelwhenbridged=yes musiconhold=default immediate=no - zaptel.conf: --- fxsks=3 loadzone= uk defaultzone = uk --- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] bug in 1.2.24
GUys.. I dont know if this is a known bug or not but I just tested and replicated this one over and over again. It involves call transfer from calls that entered the pbx via a queue.. say a call comes in and its thrown in a queue, somebody answers the call but then wants to transfer the call to somebody else outside the queue, of course... the bug comes in here.. Im using mixmonitor to record calls and when this scenario happens, the recording of the call coming in is OK, the call when in the queue and taking to the agent is OK, but then, when the agent transfers the call using attended transfer, mixmonitor stops recording... this doesn't happen if the call is transfer using BLIND transfer, just when using ATTENDED. Anybody seen this? Any bug fix or patch for 1.2.24 for this? Thx guys ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Another State Of The Punctuation Mark question - Vonage
On Tue, Sep 11, 2007 at 08:56:53AM -0400, Jay R. Ashworth wrote: There was a flurry of Vonage is going to unlock SIP activity last year; did anything productive ever come of it? Are *you* using your Vonage lines directly into Asterisk? In lieu of that, for a 4 line small business that doesn't need to pay Vonage $150 a month, who? Broadvoice? Someone else? I'm a touch unimpressed with the fact that BV's website *won't quote you BYOD pricing* until you actually place the damn order -- or so it appears to my eyes. Broadvoice can't handle multiple lines being billed to the same account and using the same SIP credentials, which is probably not too large a deal for a 4 line install, but would quickly become unmanageable for anything larger. jeff 727. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 -- Jeff Bachtel ([EMAIL PROTECTED],TAMU) http://www.cepheid.org/~jeff The sciences, each straining in [finger [EMAIL PROTECTED] for PGP key] its own direction, have hitherto harmed us little; - HPL, TCoC ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Another State Of The Punctuation Mark question - Vonage
On Tue, Sep 11, 2007 at 09:32:06AM -0700, Eric Chamberlain wrote: For several years now, we've used VoicePulse Connect http://connect.voicepulse.com/ for our Asterisk IAX and SIP trunks. Ravi and KP are both technical guys and know Asterisk extremely well. They'd better be good; their business price is twice everyone elses. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux-HA and Asterisk
On Tue, Sep 11, 2007 at 10:32:14AM -0400, Mike Clark wrote: We have gotten stuck trying to get a highly available Asterisk cluster fully functional. We used Linux-HA with Asterisk 1.4.11 on privtae IP's behind the virtual public IP. I got as far as getting phones registered and being able to place calls that rang and you could answer, but there was no audio. So, I enabled RTP debugging and discovered Asterisk was still attempting to send the audio packets to the phones private address, even though the device was set up as NAT=yes. externip was set to the virtual public IP. Is that correct that Asterisk was sending to a RFC1918 address? From your description of your setup, it seemed that the backend Asterisk servers themselves had 1918 addresses, and your Polycoms would have public addresses. In that case, the Asterisk instances might be using their 1918 addresses in packets to the Polycoms. To correct that, the Asterisk instances would need to use something like STUN or a preset public IP address in its SIP configuration. jeff Any thoughts on clearing the final hurdle? Also, we are using Polycom phones. Thanks, Mike Clark -- Jeff Bachtel ([EMAIL PROTECTED],TAMU) http://www.cepheid.org/~jeff The sciences, each straining in [finger [EMAIL PROTECTED] for PGP key] its own direction, have hitherto harmed us little; - HPL, TCoC ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Another State Of The Punctuation Mark question - Vonage
On Tue, 11 Sep 2007, Jeff Bachtel wrote: Broadvoice can't handle multiple lines being billed to the same account and using the same SIP credentials, which is probably not too large a deal for a 4 line install, but would quickly become unmanageable for anything larger. So it is not easy to provision with them, say, a PRI worth of call appearances off a single SIP contactable? How does one manage this relationship when you need to order large amounts of end-user trunks? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Asterisk on to CentOS 4
Ove Aursand wrote: Abdul wrote: Hi expets, I have installed Asterisk 1.4.11 on CentOS4 successfully without any error. But when i am trying to start asterisk with following cmd i am getting unknown command. [EMAIL PROTECTED] ~]$ asterisk -vvc -bash: asterisk: command not found [EMAIL PROTECTED] ~]$ I checked modules and other configuration files which are installed correctly. Please help me to locate this problem. Thank You Try the command as root: [EMAIL PROTECTED] ~]$ su - *enter password* [EMAIL PROTECTED] ~]# asterisk -cvv Rgds, Ove Be a better Globetrotter. Get better travel answers http://us.rd.yahoo.com/evt=48254/*http://answers.yahoo.com/dir/_ylc=X3oDMTI5MGx2aThyBF9TAzIxMTU1MDAzNTIEX3MDMzk2NTQ1MTAzBHNlYwNCQUJwaWxsYXJfTklfMzYwBHNsawNQcm9kdWN0X3F1ZXN0aW9uX3BhZ2U-?link=listsid=396545469from someone who knows. Yahoo! Answers - Check it out. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.485 / Virus Database: 269.13.14/999 - Release Date: 10.09.2007 17:43 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ove, You should seriously reconsider showing your IP in posts. You have open 22, 25, 53, 110, 111, 80, 143, 443, 3306 MySQL open to the world? Seriously? Yikes Anthony ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P not answering or making calls
Heloo, I think that your error is: zaptel.conf: --- fxsks=1 loadzone= uk defaultzone = uk zapata.conf: [channels] language=en context=incoming signalling=fxs_ks busydetect=yes busycount=4 callprogress=no relaxdtmf=yes callwaiting=no callwaitingcallerid=no threewaycalling=no transfer=yes cancallforward=yes usecallerid=no cidsignalling=v23 cidstart=polarity callerid=no hidecallerid=no echotraining=yes echocancel=yes echocancelwhenbridged=yes musiconhold=default immediate=no channel = 3 Best Regards On 9/11/07, Tom Playford [EMAIL PROTECTED] wrote: Hello, I have recently purchased a TDM400P card with one FXO expansion card, and I'm having problems. The card does not pick up incoming calls. Asterisk detects the ringing line and rings various SIP phones as required. When a sip phone answers, the sip user hears nothing and the PSTN user continues to hear ringing. Here is the asterisk output for an incoming call: - == Starting post polarity CID detection on channel 3 -- Starting simple switch on 'Zap/3-1' -- Executing Set(Zap/3-1, CALLERID(all)=call to 322817) in new stack -- Executing Dial(Zap/3-1, Local/[EMAIL PROTECTED]|45) in new stack -- Called [EMAIL PROTECTED] -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/203|30) in new stack -- Called 201 -- SIP/203-0814c448 is ringing -- Local/[EMAIL PROTECTED],1 is ringing -- SIP/201-081445e8 is ringing -- SIP/201-081445e8 answered Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 stopped sounds -- Local/[EMAIL PROTECTED],1 answered Zap/3-1 == Spawn extension (special, 601, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' == Spawn extension (special, 601, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' Sep 11 19:25:55 WARNING[3073]: chan_zap.c:3934 zt_handle_event: Ring/Off-hook in strange state 6 on channel 3 == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' - I have set opermode to 'UK' (as I'm in the UK), and dmesg confirms the setting. Outgoing calls also fail, the SIP user hears nothing, yet asterisk claims that the call has been picked up. Here is the asterisk log, looks perfectly normal: - -- Executing Dial(SIP/201-08154a38, Zap/3/0800800800) in new stack -- Called 3/0800800800 -- Zap/3-1 answered SIP/201-08154a38 - I am running Debian Etch with kernel 2.6.18 and asterisk version 1.2.13. I'm beginning to think it's a fault with the expansion card... anyone else got any ideas? Oh and the BT line is fine, it works (as well as can be expected) with a X100P card I have. Thanks, Tom Attached: dmesg output: apata Telephony Interface Registered on major 196 Zaptel Version: 1.2.16 Zaptel Echo Canceller: MG2 ACPI: PCI Interrupt :00:0b.0[A] - Link [LNKD] - GSI 12 (level, low) - IRQ 12 Freshmaker version: 73 Freshmaker passed register test Module 0: Not installed Module 1: Not installed Module 2: Installed -- AUTO FXO (UK mode) Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (1 modules) Registered tone zone 4 (United Kingdom) - zapata.conf: [channels] language=en context=incoming signalling=fxs_ks busydetect=yes busycount=4 callprogress=no relaxdtmf=yes callwaiting=no callwaitingcallerid=no threewaycalling=no transfer=yes cancallforward=yes usecallerid=no cidsignalling=v23 cidstart=polarity callerid=no hidecallerid=no echotraining=yes echocancel=yes echocancelwhenbridged=yes musiconhold=default immediate=no - zaptel.conf: --- fxsks=3 loadzone= uk defaultzone = uk --- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400P periodic sound clicks on FXS
Hi, I am having periodic sound clicks (2-3 per second) on all FXS of a TDM400P when the remote end is my VoIP provider. However: - recording the conversation on the asterisk, does not have the glitches, although I can hear them on a real phone. - My VoIP provider to my VoIP phones through the same asterisk is OK. - TDM to TDM through the same asterisk is OK. I tried with and without echocancel and different values of echotrain (including 'no'), without luck. The card is not sharing interrupts. Any ideas? Kernel is 2.6.9 asterisk is 1.2.19-BRIstuffed-0.3.0-PRE-1y-h # lspci 00:00.0 Host bridge: Intel Corporation 82945G/GZ/P/PL Memory Controller Hub (rev 02) 00:02.0 VGA compatible controller: Intel Corporation 82945G/GZ Integrated Graphics Controller (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev e1) 00:1f.0 ISA bridge: Intel Corporation 82801GB/GR (ICH7 Family) LPC Interface Bridge (rev 01) 00:1f.1 IDE interface: Intel Corporation 82801G (ICH7 Family) IDE Controller (rev 01) 00:1f.2 IDE interface: Intel Corporation 82801GB/GR/GH (ICH7 Family) SATA IDE Controller (rev 01) 00:1f.3 SMBus: Intel Corporation 82801G (ICH7 Family) SMBus Controller (rev 01) 01:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) 01:02.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 01:07.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ (rev 10) # cat /proc/interrupts CPU0 0:4262013 XT-PIC timer 1: 8 XT-PIC i8042 2: 0 XT-PIC cascade 3:4217220 XT-PIC qozap 5: 11979 XT-PIC eth0 8: 1 XT-PIC rtc 11: 29016 XT-PIC libata 15:4211433 XT-PIC wctdm NMI: 0 ERR: 0 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux-HA and Asterisk
Jeff Bachtel wrote: On Tue, Sep 11, 2007 at 10:32:14AM -0400, Mike Clark wrote: We have gotten stuck trying to get a highly available Asterisk cluster fully functional. We used Linux-HA with Asterisk 1.4.11 on privtae IP's behind the virtual public IP. I got as far as getting phones registered and being able to place calls that rang and you could answer, but there was no audio. So, I enabled RTP debugging and discovered Asterisk was still attempting to send the audio packets to the phones private address, even though the device was set up as NAT=yes. externip was set to the virtual public IP. Is that correct that Asterisk was sending to a RFC1918 address? From your description of your setup, it seemed that the backend Asterisk servers themselves had 1918 addresses, and your Polycoms would have public addresses. In that case, the Asterisk instances might be using their 1918 addresses in packets to the Polycoms. To correct that, the Asterisk instances would need to use something like STUN or a preset public IP address in its SIP configuration. jeff Yes, the Asterisk boxes were on private addresses. The Polycoms are also behind a NAT. Yes, I tried using externip in sip.conf and this allowed registration, and calls to be placed, but no audio. Unfortunately, Polycom does not support STUN. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Installed X100p
Steve Totaro wrote: Matthew Fredrickson wrote: Steve Totaro wrote: I have had Digium tech support tell me to do the same thing I'm hoping that wasn't the final conclusion in the tech support debugging process. If it was, than I am very sorry to hear that, and will make note of it. Thanks, yes, that was the final resolution. I have also heard That motherboard or that server is not supported. Again, this was quite some time ago and Digium has changed as a company as well as the product line, the whole entity has matured. Might be a non-issue now. I would hope so too as well. We're working to change a lot of things that have caused us problems in the past. Part of that problem was learning to deal with a tremendous amount of growth in a short period of time, which I would imagine is difficult for any small company. Let me ask you this, is using a T1 card for ISDN data supported now? I believe it should be working well now, a while ago I spent a bit of time making sure the zaptel portion of it functionally didn't have any problems across a range of kernels. I know that one of the reasons why that support did not support that was (IIRC) it sometimes involved recompiling a systems kernel, or upgrading a systems kernel, which is not an insignificant thing to do for a customer. Though I have not had to look at it in a while, I believe that at the very least it could be easier now, with the packaging of some of the hdlc utils in zaptel so that it works correctly across kernel versions. That one irked me since it was a selling point, but when calling for support I was told, It can do it but it is not supported. and info on the net was VERY sparse for accomplishing this (circa 2003) Sorry again about you trouble with that. I hope that somehow we can win you back :-) Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 56k modem configuration
Andrea Spadaccini wrote: Ciao Matthew, I would be very surprised if chan_modem actually works... I don't think I've *ever* seen it setup before. Well.. So there's no hope to make that modem work with Asterisk, right? Unless someone speaks otherwise, I would say that the most accurate answer is, your mileage may vary, but don't hope for a lot :-) -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P periodic sound clicks on FXS
Costa Tsaousis wrote: Hi, I am having periodic sound clicks (2-3 per second) on all FXS of a TDM400P when the remote end is my VoIP provider. However: - recording the conversation on the asterisk, does not have the glitches, although I can hear them on a real phone. - My VoIP provider to my VoIP phones through the same asterisk is OK. - TDM to TDM through the same asterisk is OK. If TDM to TDM is ok, then it would strongly point towards a problem with perhaps the VoIP provider. This is just shooting off of my hip, but maybe a jitterbuffer issue, like with the phones? I think when Asterisk bridges SIP-SIP calls, it doesn't do any jitter buffering. Matthew Fredrickson I tried with and without echocancel and different values of echotrain (including 'no'), without luck. The card is not sharing interrupts. Any ideas? Kernel is 2.6.9 asterisk is 1.2.19-BRIstuffed-0.3.0-PRE-1y-h # lspci 00:00.0 Host bridge: Intel Corporation 82945G/GZ/P/PL Memory Controller Hub (rev 02) 00:02.0 VGA compatible controller: Intel Corporation 82945G/GZ Integrated Graphics Controller (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev e1) 00:1f.0 ISA bridge: Intel Corporation 82801GB/GR (ICH7 Family) LPC Interface Bridge (rev 01) 00:1f.1 IDE interface: Intel Corporation 82801G (ICH7 Family) IDE Controller (rev 01) 00:1f.2 IDE interface: Intel Corporation 82801GB/GR/GH (ICH7 Family) SATA IDE Controller (rev 01) 00:1f.3 SMBus: Intel Corporation 82801G (ICH7 Family) SMBus Controller (rev 01) 01:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) 01:02.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 01:07.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ (rev 10) # cat /proc/interrupts CPU0 0:4262013 XT-PIC timer 1: 8 XT-PIC i8042 2: 0 XT-PIC cascade 3:4217220 XT-PIC qozap 5: 11979 XT-PIC eth0 8: 1 XT-PIC rtc 11: 29016 XT-PIC libata 15:4211433 XT-PIC wctdm NMI: 0 ERR: 0 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Chan_sip Entry
Hello, I am trying to get to Jain Sip softphones to call one another via an Asterisk server. When I call from phone 1 to phone 2 there is audio transmission both ways, but when I call from phone 2 to phone 1 I don't get audio transmission and reception both ways. When I look at the asterisk log file it has an entry which says: Oooh, format changed to 2. Would anyone know why this is occuring one way and not the other, and more importantly, how would I fix this. After some examination I see that when I send the OK to the INVITE, this SDP body should have a 0 for the codec which is ulaw. When this Ok message gets to the other pc after going through asterisk it seems like asterisk adds a codec because the SDP body now contains the codecs 0 and 3. I believe the problem has something to do with this but I am not sure why it would work one way but not the other. Any help would be greatly appreciated. Thanks very much, Denis Kutman ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P periodic sound clicks on FXS
Matthew Fredrickson wrote: Costa Tsaousis wrote: Hi, I am having periodic sound clicks (2-3 per second) on all FXS of a TDM400P when the remote end is my VoIP provider. However: - recording the conversation on the asterisk, does not have the glitches, although I can hear them on a real phone. - My VoIP provider to my VoIP phones through the same asterisk is OK. - TDM to TDM through the same asterisk is OK. If TDM to TDM is ok, then it would strongly point towards a problem with perhaps the VoIP provider. This is just shooting off of my hip, but maybe a jitterbuffer issue, like with the phones? I think when Asterisk bridges SIP-SIP calls, it doesn't do any jitter buffering. If it is a jitterbuffer, then why the recordings (of the same calls I hear the clicks, not other calls) do not have them? Also, I believe it cannot be an issue of the provider since all ATAs I tested do not have the issue (same provider, same account). ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P periodic sound clicks on FXS
Costa Tsaousis wrote: Matthew Fredrickson wrote: Costa Tsaousis wrote: Hi, I am having periodic sound clicks (2-3 per second) on all FXS of a TDM400P when the remote end is my VoIP provider. However: - recording the conversation on the asterisk, does not have the glitches, although I can hear them on a real phone. - My VoIP provider to my VoIP phones through the same asterisk is OK. - TDM to TDM through the same asterisk is OK. If TDM to TDM is ok, then it would strongly point towards a problem with perhaps the VoIP provider. This is just shooting off of my hip, but maybe a jitterbuffer issue, like with the phones? I think when Asterisk bridges SIP-SIP calls, it doesn't do any jitter buffering. If it is a jitterbuffer, then why the recordings (of the same calls I hear the clicks, not other calls) do not have them? Well, I could be wrong since I haven't checked the code, but I believe that asterisk only enables jitterbuffering on a call if it terminates either at a non-rtp endpoint, such as a zaptel TDM interface or perhaps a recording to a file. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P periodic sound clicks on FXS
On Wed, Sep 12, 2007 at 12:33:02AM +0300, Costa Tsaousis wrote: Matthew Fredrickson wrote: Costa Tsaousis wrote: Hi, I am having periodic sound clicks (2-3 per second) on all FXS of a TDM400P when the remote end is my VoIP provider. However: - recording the conversation on the asterisk, does not have the glitches, although I can hear them on a real phone. - My VoIP provider to my VoIP phones through the same asterisk is OK. - TDM to TDM through the same asterisk is OK. If TDM to TDM is ok, then it would strongly point towards a problem with perhaps the VoIP provider. This is just shooting off of my hip, but maybe a jitterbuffer issue, like with the phones? I think when Asterisk bridges SIP-SIP calls, it doesn't do any jitter buffering. If it is a jitterbuffer, then why the recordings (of the same calls I hear the clicks, not other calls) do not have them? Also, I believe it cannot be an issue of the provider since all ATAs I tested do not have the issue (same provider, same account). How about calls from either the card or the trunk to an echo test extension? to a local SIP/IAX phone? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 NAT issues
Hello, I am playing around with IAX2 and I have encountered a problem trying to setup an asterisk box through NAT using IAX2. This is the problem: Asterisk box = Advanced Firewall = Internet = User's router = User The user can register, the server can answer, calls can be made. Asterisk box = Very simple router = Internet = User's router = User User's packet reach the server, the server cannot reply because the udp connection is lost, several RX retry TX retry, user is unable to call. In both cases the firewall and the router are forwarding the port 4569 to Asterisk, user's router is not forwarding anything, the user has qualify=yes to maintain the connection open but the very simple router will drop the connection before Asterisk can reply to the packet. So I ask the list: Is there a way to overcome this problem? Udp connection timeout in Asterisk? Should I get a new router? Thanks, PLL. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flash IDE
Ed W wrote: I worried a lot about the same, in the end I went for a small laptop drive for safety (it's inaudible) However, this came up on slashdot recently and if you search around the logic seems to be that: - Flash rewrites quite a few times - The good stuff has wear levelling so that most roughly speaking the whole thing should work until it suddenly all fails - Given a big enough drive with a fair bit of free space then you should find it hard to wear it out in less than quite a few years even if you are hitting it quite hard (probably multiples of this). Simply do the maths to get the rough life So basically it seems that given a large enough flash drive with decent wear levelling the lifetime should be completely ample... ...Thats the theory anyway. I feel quite bullish about the whole thing, but I think I would avoid the *really* discounted cheapo flash drives since they may not have the correct wear levelling. Decent brand names should be fine though (and you can google for details on their specs) I've had CF units fail in service, but it's true that reliability is increasing, especially as they get bigger. I would recommend going with the largest CF you can afford. -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P not answering or making calls
Hi, Thanks for spotting that, however that was a copy-and-paste mistake, 'channel = 3' was is in my zapata.conf. I wish it were that simple! Thanks again, Tom On 11/09/2007, Carlos Rojas [EMAIL PROTECTED] wrote: Heloo, I think that your error is: zaptel.conf: --- fxsks=1 loadzone= uk defaultzone = uk zapata.conf: [channels] language=en context=incoming signalling=fxs_ks busydetect=yes busycount=4 callprogress=no relaxdtmf=yes callwaiting=no callwaitingcallerid=no threewaycalling=no transfer=yes cancallforward=yes usecallerid=no cidsignalling=v23 cidstart=polarity callerid=no hidecallerid=no echotraining=yes echocancel=yes echocancelwhenbridged=yes musiconhold=default immediate=no channel = 3 Best Regards On 9/11/07, Tom Playford [EMAIL PROTECTED] wrote: Hello, I have recently purchased a TDM400P card with one FXO expansion card, and I'm having problems. The card does not pick up incoming calls. Asterisk detects the ringing line and rings various SIP phones as required. When a sip phone answers, the sip user hears nothing and the PSTN user continues to hear ringing. Here is the asterisk output for an incoming call: - == Starting post polarity CID detection on channel 3 -- Starting simple switch on 'Zap/3-1' -- Executing Set(Zap/3-1, CALLERID(all)=call to 322817) in new stack -- Executing Dial(Zap/3-1, Local/[EMAIL PROTECTED] |45) in new stack -- Called [EMAIL PROTECTED] -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/203|30) in new stack -- Called 201 -- SIP/203-0814c448 is ringing -- Local/[EMAIL PROTECTED],1 is ringing -- SIP/201-081445e8 is ringing -- SIP/201-081445e8 answered Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 stopped sounds -- Local/[EMAIL PROTECTED],1 answered Zap/3-1 == Spawn extension (special, 601, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' == Spawn extension (special, 601, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' Sep 11 19:25:55 WARNING[3073]: chan_zap.c:3934 zt_handle_event: Ring/Off-hook in strange state 6 on channel 3 == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' - I have set opermode to 'UK' (as I'm in the UK), and dmesg confirms the setting. Outgoing calls also fail, the SIP user hears nothing, yet asterisk claims that the call has been picked up. Here is the asterisk log, looks perfectly normal: - -- Executing Dial(SIP/201-08154a38, Zap/3/0800800800) in new stack -- Called 3/0800800800 -- Zap/3-1 answered SIP/201-08154a38 - I am running Debian Etch with kernel 2.6.18 and asterisk version 1.2.13. I'm beginning to think it's a fault with the expansion card... anyone else got any ideas? Oh and the BT line is fine, it works (as well as can be expected) with a X100P card I have. Thanks, Tom Attached: dmesg output: apata Telephony Interface Registered on major 196 Zaptel Version: 1.2.16 Zaptel Echo Canceller: MG2 ACPI: PCI Interrupt :00:0b.0[A] - Link [LNKD] - GSI 12 (level, low) - IRQ 12 Freshmaker version: 73 Freshmaker passed register test Module 0: Not installed Module 1: Not installed Module 2: Installed -- AUTO FXO (UK mode) Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (1 modules) Registered tone zone 4 (United Kingdom) - zapata.conf: [channels] language=en context=incoming signalling=fxs_ks busydetect=yes busycount=4 callprogress=no relaxdtmf=yes callwaiting=no callwaitingcallerid=no threewaycalling=no transfer=yes cancallforward=yes usecallerid=no cidsignalling=v23 cidstart=polarity callerid=no hidecallerid=no echotraining=yes echocancel=yes echocancelwhenbridged=yes musiconhold=default immediate=no - zaptel.conf: --- fxsks=3 loadzone= uk defaultzone = uk --- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] bug in 1.2.24
It is not a bug. attended Transfer is using Local channel, if you have a look the debug log from CLI, you will see why it fails. To solve this problem, enable recording before the calls go into the queue. Exten = ,1,MixMonitor(...) Exten = ,2,Goto(ext-queue, , 1) This will ensure you to record the customer/caller's channel instead of exten's channel. So no matter where you transfer the call and as long as the caller not hangup the call, it will be always recorded. By the way, 1.2.24 stable, we got problem with 1.2.21. 1.2.17 seems stable. Good luck, Isaac Xiao WARNING - This e-mail and any attachments may be CONFIDENTIAL and are for the intended addressee only. If received in error, please delete and inform us by returning an email. Any unauthorized copying, disclosure or distribution of the material in this email is strictly prohibited. E-mail transmission cannot be guaranteed to be secure, error-free or virus-free. The sender therefore does not accept liability for any errors, omissions or consequences which arise as a result of e-mail transmission. This e-mail and its attachments are not intended to constitute financial advice or recommendation of, or an offer to buy or sell, any securities or other financial products. We recommend that you seek your own independent legal or financial advice before proceeding with any investment decision. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Project: AskoziaPBX
On Mon, Sep 10, 2007 at 10:04:31AM +0200, Michael Iedema wrote: This is not a live-cd but rather an image that must initially be written to a disk, so a dedicated machine is needed. After that, the entire system is upgradeable through the webGUI. You might want to note: http://www.webgui.org Yeah, it's not quite trademarkable, but why confuse matters, right? ;-) Cheers, -- jr 'stddisclaimer.h' a -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Another State Of The Punctuation Mark question - Vonage
On Tue, Sep 11, 2007 at 03:53:56PM -0400, Alex Balashov wrote: On Tue, 11 Sep 2007, Jeff Bachtel wrote: Broadvoice can't handle multiple lines being billed to the same account and using the same SIP credentials, which is probably not too large a deal for a 4 line install, but would quickly become unmanageable for anything larger. So it is not easy to provision with them, say, a PRI worth of call appearances off a single SIP contactable? How does one manage this relationship when you need to order large amounts of end-user trunks? Well, it sounds like you go somewhere else. I'm investigating Voicepulse, as someone else suggested. I don't have back CDR to feed them for comparative pricing, so I'm going to have to go disassemble a years worth of Vonage bills. Luckily, I *have* a years worth, right there on line. I don't see TBCT or network-outage forwarding though, in my as yet limited investigation. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux-HA and Asterisk
On Tue, Sep 11, 2007 at 04:30:03PM -0400, Mike Clark wrote: Yes, the Asterisk boxes were on private addresses. The Polycoms are also behind a NAT. Yes, I tried using externip in sip.conf and this allowed registration, and calls to be placed, but no audio. Unfortunately, Polycom does not support STUN. So you were trying to connect phones behind one NAT to a server behind another NAT, without a VPN. Sounds like the audio paths are trying to be set up from the server end. I'm not a SIP mechanic (yet :-), but yeah, that ain't gonna work. Only one phone per location? Routers got port triggering? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)
I liked the queue game concept! although it could be cruel! On 9/11/07, Steve Totaro [EMAIL PROTECTED] wrote: http://vonmag.com/editorial/web-exclusives/mark-spencer-digium-is-growing-up Seems the Adtran relationship goes way back... Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco UC 500
I'm trying to get some more information on this myself as its a new product from Cisco. What i know, Cisco attendant console works with skinny,Cisco page and SLA also works wiht skinny and not SIP. So its either having these or SIP. On 9/10/07, Drew Gibson [EMAIL PROTECTED] wrote: Jeremy Mann wrote: Is the Cisco UC 500 able to integrate with Asterisk? Specifically does it work via SIP? Just curious, as the Cold Call Cisco sales rep had no clue what SIP even was, and this device looks interesting. Google cisco UC500, hit #2 = http://www.cisco.com/en/US/products/ps7293/products_data_sheet0900aecd8061fb06.html Quotes: Core components of the Cisco Unified Communications 500 Series include: Cisco Unified IP phones, including wireless handsets and Session Initiation Protocol (SIP) phones PSTN interfaces and features: SIP trunks and RFC 2833 support Does that help? I'll bet Asterisk is cheaper though. :-) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Partitioning DSL input
Although you can find a router with QOS or dedicated bandwidth feature, I would suggest a QOS enabled Switch. Any IEEE802.1p enables switch,(these days less than $100 for 16 port) can do the job. you cant do alot when your traffic reaches internet, thats why most you can do is up to your modem. cos bit works best at layer 2 , and pretty much TOS is useless if you dont own your wlan line. On 9/10/07, David Gomillion [EMAIL PROTECTED] wrote: On 9/10/07, Ira [EMAIL PROTECTED] wrote: At 02:11 PM 9/10/2007, you wrote: Can people on this list share their experiences on how they partition a DSL for small business internet service with a router so that a portion is dedicated to VOIP and another portion to computers. Of course, the idea is to do this with a low cost router (under $100). dd-wrt or Sveasoft on a Linksys router though I understand there are better choices in routers today. Don't expect too much out of traffic shaping. While it should work nearly perfectly upstream, there's only so much you can do to control the downstream (from your ISP to you). ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flash IDE
Disable DMA on that drive. Thee HD/DOM/CF-card does not support DMA and linux tries to DMA it. On 9/11/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: For real! I see the BIOS, then I see GRUB Loading stage 1.5 and then a good 60 seconds go by before the kernel and initrd have been loaded and control switches over to them. Stock kernel on CentOS 4.4 (gmail quoting stinks) Ed W, wrote - Flash rewrites quite a few times - The good stuff has wear levelling so that most roughly speaking the whole thing should work until it suddenly all fails - Given a big enough drive with a fair bit of free space then you should find it hard to wear it out in less than quite a few years even if you are hitting it quite hard (probably multiples of this). Simply do the maths to get the rough life In the last 2 years I have personally killed 2 DOMs of 512 MB. They where running Debian Sarge, and were set up to run on TMPFS. The reason why they died is because I tested the installation on those systems: this means zero out HDA and then copy it all over again from a backup. In real life, in real usage, I think those will last quite more, since the disk is not been written all that much. But still, be warned. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
