Re: [asterisk-users] Voicemail ODBC
Yes, with isql it's working fine, I can see the database and all fields. On Sun, Jun 20, 2010 at 9:11 PM, Tilghman Lesher tles...@digium.com wrote: On Sunday 20 June 2010 13:15:11 Andraž wrote: If I use MySQL with the same fields it's working fine. I think that is something wrong with FreeTDS drivers. Could also be that you're specifying the database name incorrectly. Are you able to see the tables when using the 'isql' command line tool? -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call dialing
Hi listusers, I am using call file to dial out the sip on a different machine. The problem is whenever i dial the call lands up on the softphone but i have to pick it up 2 times, for both line 1 and line 2. If i reject it in the 1st time only then both are rejected. channel: SIP/2001 CallerId: 2001 MaxRetries: 0 RetryTime: 10 WaitTime: 60 Account: test Context: phones Extension: 2001 Priority: 1 Archive: Yes Can any of you suggest anything, i have one more problem, after i have mage the call I want to play some file to the user. Is there any way in aserisk through which i can do this, i mean while the call is running. -- Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] features.conf - parkedcalls - transfer
Hello dear list. I am having issues on parkedcalls. I am using a Cisco SPA525G as a test phone, and I have the transfer button there when I am in a call, But when I want to transfer the current call I am in, I push the transfer button, and onscreen I se Enter Number, and if I enter ex sip 200, I have to wait Almost 10 seconds, before the transfer to sip 200 is made, can I reduce that timer? And I can't see any button on the Cisco phone which will function like a direct transfer now, do I have to wait...? And, secondly, is there a another way to do transfer/send to another sip phone? Ex. Push *200 and the SIP phone will directly call SIP/200. Or push *401 and the Sip phone will directly call SIP401? Default features.conf context. Thank you. Med vennlig hilsen / Best regards Abacus IT AS - din Visma Software Partner - your Visma Software Partner Tor Aksel Celasun Mobilnummer/cell phone: (+47) 900 15 103 Sentralbord/Support 4000 1850 ak...@abacus-it.nomailto:ak...@abacus-it.no -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compiling H323
On Sun, Jun 20, 2010 at 07:55:07PM -0400, Michelle Dupuis wrote: And after cleaning everything 323/pwlib/ptlib related, and reinstalling ptlib and h323plus, I can't even get asterisk to compile chan_h323 anymore. Perhaps something old was left over. My .configure run shows: checking /usr/src/openh323plus/h323plus/../pwlib/version.h usability... no checking /usr/src/openh323plus/h323plus/../pwlib/version.h presence... no checking for /usr/src/openh323plus/h323plus/../pwlib/version.h... no checking /root/pwlib/include/ptlib.h usability... yes checking /root/pwlib/include/ptlib.h presence... yes checking for /root/pwlib/include/ptlib.h... yes checking if PWLib version 2.4.5 is compatible with chan_h323... yes and checking h323.h usability... no checking h323.h presence... no checking for h323.h... no Anyone got good advice? I can make ooh323 work but a bug with faststart (in h323) is forcing me to another h323 stack. Is there a way to make opal+pwlib from centos packages work? (trick asterisk .configure to accept them)? Not really. My experience is different: H.323plus: http://bugs.debian.org/586541 (Mostly a Debian packaging issue, but I'm still not sure where this BOOL is supposed to come from) And then there is the other fork of OpenH323 - OPAL. Unlike H323Plus it maintains the full bloat of openh323 (everything you need for Ekiga). But it seems to have a saner build system. Sadly, Asterisk already has to jump through hoops to build chan_h323. So using the checking for the saner alternative seems to make things even more complex: https://issues.asterisk.org/view.php?id=17226 And yet, my first impression is that those libraries are not compatible enough (see https://issues.asterisk.org/view.php?id=17226#123634 ). Or am I missing some compatibility layer? BTW: Please check the following as well. While I strongly suspect it is Debian-specific, it should be simple to check (ldd channels/chan_h323.so) https://issues.asterisk.org/view.php?id=17162 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get asterisk to playback personal greetings using grandstream gxp-2000
On 18/06/10 20:22, Eddie Mikell wrote: All: I am using the standard voicemail in asterisk. Everything works well, except, if a users wants to record their own personal greeting, it doesn't playback. I can see the soundfile being created. I suspect it is a setting in the voicemail.conf, or an option I am over-looking on the grandstream, but if anyone can point me in the write direction, I would certainly appreciate the help. Also, I would like for the user to be able to set up their own password. I set the initial password the same as the extension, so that forces voicemail to prompt for a new password. The problem is, I can see where asterisk is trying to write the password in the voicemail.conf file, but it is denied because the user doesn't have permission. I hate to open /etc/asterisk directory to the incorrect permissions. What would be the best way to enable the user to be able to change their password? Thanks, Eddie Hi Are you using the u option in the Voicemail step? As for the password, all the files in /etc/asterisk should be owned by asterisk.asterisk Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk appache issue
thanks for reply. how can i give the root permission to apache ? sudo. i also tried sudo . However, without careful configuration you will probably be giving root access to any process that runs as your apache user. I've never done it, but I'm guessing you could create a group, make your asterisk user and your apache user members of that group and protect resources appropriately. What are you trying to accomplish that you can't using AMI, querying a database, creating a call file or parsing a log file? Alternatively, as of 1.6.1 (or is it 1.6.2) you have CLI permissions. You can allow anybody to write to the socket, but only a limited set of commands to the user 'apache' or whatever. See /etc/asterisk/cli_permissions.conf . i am using asterisk 1.6.2 but did't find /etc/asterisk/cli_permissions.conf. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail ODBC
On Monday 21 June 2010 01:16:30 Andraž wrote: Yes, with isql it's working fine, I can see the database and all fields. On Sun, Jun 20, 2010 at 9:11 PM, Tilghman Lesher tles...@digium.com wrote: On Sunday 20 June 2010 13:15:11 Andraž wrote: If I use MySQL with the same fields it's working fine. I think that is something wrong with FreeTDS drivers. Could also be that you're specifying the database name incorrectly. Are you able to see the tables when using the 'isql' command line tool? Did you remember to turn on pooling? TDS databases only support a single query per connection, thus connections may not be shared. Let's see your res_odbc.conf. Also, what version of Asterisk are you using? -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail ODBC
Now it's workin fine. It was problem with drivers, because it doesn't support all kind of fields. I just changed from varblog to picture data type and now it's working fine. Tnx for help. On Mon, Jun 21, 2010 at 2:08 PM, Tilghman Lesher tles...@digium.com wrote: On Monday 21 June 2010 01:16:30 Andraž wrote: Yes, with isql it's working fine, I can see the database and all fields. On Sun, Jun 20, 2010 at 9:11 PM, Tilghman Lesher tles...@digium.com wrote: On Sunday 20 June 2010 13:15:11 Andraž wrote: If I use MySQL with the same fields it's working fine. I think that is something wrong with FreeTDS drivers. Could also be that you're specifying the database name incorrectly. Are you able to see the tables when using the 'isql' command line tool? Did you remember to turn on pooling? TDS databases only support a single query per connection, thus connections may not be shared. Let's see your res_odbc.conf. Also, what version of Asterisk are you using? -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISP down internal phones become unavailable
I saw the following lines in the log this morning. From my router logs I see that the connection went down as my ISP was doing maintenance for a few minutes last night. I can understand the external registrations timing out, but why do the phones become unreachable. They are on the internal lan within the same subnet as the Asterisk server. Internal DHCP and DNS was functional. If I had a PRI card in this system as well that would mean I couldn't make phone calls because the Internet is down. Ryan [Jun 21 01:51:26] NOTICE[13657]: chan_sip.c:11569 sip_reg_timeout: -- Registration for '...@newyork.voip.ms' timed out, trying again (Attempt #1) [Jun 21 01:51:46] NOTICE[13657]: chan_sip.c:22943 sip_poke_noanswer: Peer '1850' is now UNREACHABLE! Last qualify: 15 [Jun 21 01:51:46] NOTICE[13657]: chan_sip.c:11569 sip_reg_timeout: -- Registration for '...@sip.flowroute.com' timed out, trying again (Attempt #1) [Jun 21 01:52:07] NOTICE[13657]: chan_sip.c:22943 sip_poke_noanswer: Peer '1800' is now UNREACHABLE! Last qualify: 7 [Jun 21 01:52:07] NOTICE[13657]: chan_sip.c:22943 sip_poke_noanswer: Peer '1801' is now UNREACHABLE! Last qualify: 11 [Jun 21 01:52:07] NOTICE[13657]: chan_sip.c:11569 sip_reg_timeout: -- Registration for '...@newyork.voip.ms' timed out, trying again (Attempt #2) == Extension Changed 2028[ext-local] new state Unavailable for Notify User 1850 [Jun 21 01:52:17] NOTICE[13657]: chan_sip.c:18314 handle_response_peerpoke: Peer '1800' is now Reachable. (10ms / 2000ms) == Extension Changed 2028[ext-local] new state Idle for Notify User 1850 [Jun 21 01:52:17] NOTICE[13657]: chan_sip.c:18314 handle_response_peerpoke: Peer '1801' is now Reachable. (14ms / 2000ms) [Jun 21 01:52:22] NOTICE[13657]: chan_sip.c:18314 handle_response_peerpoke: Peer '1850' is now Reachable. (16ms / 2000ms) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [AGI] What scripting language for embedded hardware?
Hello I'm learning how to work with Asterisk on an embedded system (MMU-less Blackfin processor, 64MB RAM and 256MB NAND), and was wondering what people use as scripting language to handle calls through the dialplan and AGI, considering the hardware limitations? Ideally, I'd rather use a rich language like PHP or Python, but can those be fit with even their common modules into such small hardware? I'm also thinking of Lua and modules, provided they can be included in the buildroot. Thank you for any feedback. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] using call file
HI list-users, Greetings!! I have been using call file, i playback my file using * application:playback* and once the playback is over the call is disconnected. Is there any way it can wait and also record the dtmf inputs once the playback is over. Thanks in advace Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using call file
Use a context instead of the playback command. Like this
[playit]
exten = s,1,NoOp(Answer)
exten = s,n,SetMusicOnHold(default)
exten = s,n,Waitexten(5,m)
exten = s,n,Verbose(play ${ARG1})
exten = s,n,Playback(${ARG1})
So you replace playback(file) with playit(file).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of nikhil
singhania
Sent: Monday, June 21, 2010 7:55 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] using call file
HI list-users,
Greetings!!
I have been using call file, i playback my file using application:playback
and once the playback is over the call is disconnected. Is there any way it
can wait and also record the dtmf inputs once the playback is over.
Thanks in advace
Nikhil Kumar
summer intern:simmortel voice technologies
rit2007033
b.tech IT 6th sem
IIIT Allahabad
cont...@9793905858
email: rit2007...@iiita.ac.in
niksingha...@gmail.com
http://profile.iiita.ac.in/RIT2007033/
--
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Re: [asterisk-users] [AGI] What scripting language for embedded hardware?
On Mon, 21 Jun 2010, Gilles wrote: Hello I'm learning how to work with Asterisk on an embedded system (MMU-less Blackfin processor, 64MB RAM and 256MB NAND), and was wondering what people use as scripting language to handle calls through the dialplan and AGI, considering the hardware limitations? Ideally, I'd rather use a rich language like PHP or Python, but can those be fit with even their common modules into such small hardware? I'm also thinking of Lua and modules, provided they can be included in the buildroot. Thank you for any feedback. You could always type asterisk blackfin into google and see what it suggests. Here, I'll save you the effort: http://www.rowetel.com/ucasterisk/ They use micro Perl. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [AGI] What scripting language for embedded hardware?
On Mon, 21 Jun 2010, Gilles wrote: I'm learning how to work with Asterisk on an embedded system (MMU-less Blackfin processor, 64MB RAM and 256MB NAND), and was wondering what people use as scripting language to handle calls through the dialplan and AGI, considering the hardware limitations? I'm a big fan of compiled languages like C. You can execute XXX AGIs written in C in the time it takes to load an interpreter and parse a script. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [AGI] What scripting language for embedded hardware?
On Mon, 21 Jun 2010 14:06:22 +0100 (BST), Gordon Henderson gordon+aster...@drogon.net wrote: You could always type asterisk blackfin into google and see what it suggests. Here, I'll save you the effort: Thanks but I already know this (uCasterisk is deprecated). And can't stand Perl ;-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [AGI] What scripting language for embedded hardware?
Even though I'm a PERL Weenie, I'll second this suggestion because you have to have gcc present for PERL or Micro PERL. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Monday, June 21, 2010 8:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [AGI] What scripting language for embedded hardware? On Mon, 21 Jun 2010, Gilles wrote: I'm learning how to work with Asterisk on an embedded system (MMU-less Blackfin processor, 64MB RAM and 256MB NAND), and was wondering what people use as scripting language to handle calls through the dialplan and AGI, considering the hardware limitations? I'm a big fan of compiled languages like C. You can execute XXX AGIs written in C in the time it takes to load an interpreter and parse a script. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [AGI] What scripting language for embedded hardware?
If you can install python or PHP in that machine (in means of storage), you are free to run it there. 64 RAM is really enough to run python, so you have to just try if it suits in the application. If it takes too slow to initialize - try to find some embedded versions. openwrt, for instance, has one, that means it's possible to run python on wrt54gl (16 MB RAM, 200MHz MIPS), so your platform should be really possible. On Mon, Jun 21, 2010 at 3:48 PM, Gilles codecompl...@free.fr wrote: Hello I'm learning how to work with Asterisk on an embedded system (MMU-less Blackfin processor, 64MB RAM and 256MB NAND), and was wondering what people use as scripting language to handle calls through the dialplan and AGI, considering the hardware limitations? Ideally, I'd rather use a rich language like PHP or Python, but can those be fit with even their common modules into such small hardware? I'm also thinking of Lua and modules, provided they can be included in the buildroot. Thank you for any feedback. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compiling H323
Yuk! I did manage to get it compiling again, but same error. I found an environment variable which makes the loader tell you what it's doing, and when I load chan_h323.so I see that it is running the init code when it segfaults. So I'm ok up to that point... What a mess the entire H.323 is! I found 5 different h323 stakes (some ancestors of others)...none work. Only chan_ooh323 builds and runs cleanly, but has a pointer bug (double release) causing module to crash. Anyone know if the chan_ooh323 programmers are available for fee/contract to fix bugs? Michelle From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen [tzafrir.co...@xorcom.com] Sent: Monday, June 21, 2010 3:39 AM To: Asterisk Users List Subject: Re: [asterisk-users] Compiling H323 On Sun, Jun 20, 2010 at 07:55:07PM -0400, Michelle Dupuis wrote: And after cleaning everything 323/pwlib/ptlib related, and reinstalling ptlib and h323plus, I can't even get asterisk to compile chan_h323 anymore. Perhaps something old was left over. My .configure run shows: checking /usr/src/openh323plus/h323plus/../pwlib/version.h usability... no checking /usr/src/openh323plus/h323plus/../pwlib/version.h presence... no checking for /usr/src/openh323plus/h323plus/../pwlib/version.h... no checking /root/pwlib/include/ptlib.h usability... yes checking /root/pwlib/include/ptlib.h presence... yes checking for /root/pwlib/include/ptlib.h... yes checking if PWLib version 2.4.5 is compatible with chan_h323... yes and checking h323.h usability... no checking h323.h presence... no checking for h323.h... no Anyone got good advice? I can make ooh323 work but a bug with faststart (in h323) is forcing me to another h323 stack. Is there a way to make opal+pwlib from centos packages work? (trick asterisk .configure to accept them)? Not really. My experience is different: H.323plus: http://bugs.debian.org/586541 (Mostly a Debian packaging issue, but I'm still not sure where this BOOL is supposed to come from) And then there is the other fork of OpenH323 - OPAL. Unlike H323Plus it maintains the full bloat of openh323 (everything you need for Ekiga). But it seems to have a saner build system. Sadly, Asterisk already has to jump through hoops to build chan_h323. So using the checking for the saner alternative seems to make things even more complex: https://issues.asterisk.org/view.php?id=17226 And yet, my first impression is that those libraries are not compatible enough (see https://issues.asterisk.org/view.php?id=17226#123634 ). Or am I missing some compatibility layer? BTW: Please check the following as well. While I strongly suspect it is Debian-specific, it should be simple to check (ldd channels/chan_h323.so) https://issues.asterisk.org/view.php?id=17162 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [AGI] What scripting language for embedded hardware?
On Mon, 21 Jun 2010 17:25:09 +0300, Motiejus Jaktys desired@gmail.com wrote: If you can install python or PHP in that machine (in means of storage), you are free to run it there. 64 RAM is really enough to run python, so you have to just try if it suits in the application. If it takes too slow to initialize - try to find some embedded versions. openwrt, for instance, has one, that means it's possible to run python on wrt54gl (16 MB RAM, 200MHz MIPS), so your platform should be really possible. Thanks for the tip. I'll check how it's done in OpenWrt. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [AGI] What scripting language for embedded hardware?
Hey Gilles, for whatever reason your messages appear twice twice on this list. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?
Hi Everyone,
I want to know if a specific codec type is used at least one. For example, I
want to know if out of the 100 calls on the system if there is a 1 channel
that is running G.729 codec right now. If using dial-plan and I dial in, I
can use this to obtain information about CURRENT channel. But it won't allow
me to obtain information about OTHER channels and that is what I want to do.
I want a search for all channels and an output spit out as g729 or TRUE or
FALSE if there is a g729 channel.
exten = s,1,Answer()
exten = s,n,Set(foo=${CHANNEL(audioreadformat)})
exten = s,n,NoOp(${foo})
Above NoOp spits out g729 if I call in with a g729 codec. But I
want that to be about other channels and not the one I am calling
into.
Thanks,
Bruce
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Re: [asterisk-users] features.conf - parkedcalls - transfer
On Mon, Jun 21, 2010 at 2:27 AM, Aksel Celasun ak...@abacus-it.no wrote: I am using a Cisco SPA525G as a test phone, and I have the transfer button there when I am in a call, But when I want to transfer the current call I am in, I push the transfer button, and onscreen I se “Enter Number”, and if I enter ex sip 200, I have to wait Almost 10 seconds, before the transfer to sip 200 is made, can I reduce that timer? And I can’t see any button on the Cisco phone which will function like a “direct transfer now”, do I have to wait…? On the Cisco 79xx series phones, you would modify a config file called dialplan.xml. I think on the SPA5xx series you can configure this parameter either in the main config file or from the web interface. You need to look for something like dialplan or dial plans, etc. This controls the timeout when entering digits. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Switchboad like application
Hello all, Anybody could point me any clue about an Open Source or licensed switchboard for my users? ARI or FOP is not enought for my users. Thanks in advance. VoipCrazy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What is the voicemail u option
All: Still trying to get Grandstream to play personal greetings recorded by user - no luck. Someone mentioned the u option. What is that? Something in voicemail.conf? Eddie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the voicemail u option
RTFM - B = busy, U = unavailable, S = Silent. User can record custom busy and unavailable messages; if you use S you can just playback any kind of message you want. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eddie Mikell Sent: Monday, June 21, 2010 10:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] What is the voicemail u option All: Still trying to get Grandstream to play personal greetings recorded by user - no luck. Someone mentioned the u option. What is that? Something in voicemail.conf? Eddie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the voicemail u option
Eddie Mikell wrote: user - no luck. Someone mentioned the u option. What is that? core show application voicemail Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI: Inbound BRI call, DDI not presented
Hello- I have a system with one D410P and one B200P (both OpenVox). All is well with the D410P, inbound and outbound, and I can initiate calls on the B200P BRI span, but there may be something wrong with my inbound BRI setup: there is no indication of an inbound call when I dial in to it from the PSTN. When I run pri intense debug and make a call to the BRI span, I can see a message containing the DDI that I'm dialing, in this case 336027 (BT supplies only the last 6 digits of a delivered number). See debug output below... Have I neglected to set up some needed parameter? This all worked on older boards when using bristuff, but now I want to use dahdi. My client is in the UK, connected to BT, and I have specified euroisdn as the switch type. many thanks - (snippet during inbound call to 336027) Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 008 P/F: 1 0 bytes of data -- ACKing all packets from 8 to (but not including) 8 -- Stopping T200 timer -- Starting T203 timer Handling message for SAPI/TEI=0/0 TEI: 0 State 7 V(S) 8 V(A) 8 V(R) 8 K 1, RC 0, l3initiated 0, reject_except 0 ack_pend 0 T200 0, N200 3, T203 1 [ 02 ff 03 08 01 01 05 a1 04 03 80 90 a3 18 01 89 1e 02 84 83 6c 02 21 a3 70 07 81 33 33 36 30 32 37 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 30 bytes of data Handling message for SAPI/TEI=0/127 TEI: 0 State 7 V(S) 8 V(A) 8 V(R) 8 K 1, RC 0, l3initiated 0, reject_except 0 ack_pend 0 T200 0, N200 3, T203 1 [ 02 ff 03 08 01 01 05 a1 04 03 80 90 a3 18 01 89 1e 02 84 83 6c 02 21 a3 70 07 81 33 33 36 30 32 37 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 30 bytes of data Handling message for SAPI/TEI=0/127 - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [AGI] What scripting language for embedded hardware?
Uhmmm.. remember for each channel you run perl or php interpreter so with that amount of memory maybe this can be a problem. For that kind of project I'd use C or java as fastagi protocol From: desired@gmail.com Date: Mon, 21 Jun 2010 17:25:09 +0300 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] [AGI] What scripting language for embedded hardware? If you can install python or PHP in that machine (in means of storage), you are free to run it there. 64 RAM is really enough to run python, so you have to just try if it suits in the application. If it takes too slow to initialize - try to find some embedded versions. openwrt, for instance, has one, that means it's possible to run python on wrt54gl (16 MB RAM, 200MHz MIPS), so your platform should be really possible. On Mon, Jun 21, 2010 at 3:48 PM, Gilles codecompl...@free.fr wrote: Hello I'm learning how to work with Asterisk on an embedded system (MMU-less Blackfin processor, 64MB RAM and 256MB NAND), and was wondering what people use as scripting language to handle calls through the dialplan and AGI, considering the hardware limitations? Ideally, I'd rather use a rich language like PHP or Python, but can those be fit with even their common modules into such small hardware? I'm also thinking of Lua and modules, provided they can be included in the buildroot. Thank you for any feedback. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Create Conference and exit myself
Hi, I am using Trixbox trixbox CE 2.6.2.3 (Stable) using Asterisk 1.4.22-4 I am looking for the following functionality: `` I receive a call from Mr. A. I put Mr. A on hold. I dial Mr. B I connect Mr. A's call (which was on hold) to Mr. B and I get out of the call. Mr. A Mr. B are in conversation, while my line is free to accept a new calls. What is the simplest way to achieve this ?? Thx Sans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to find a single call in logs
Hello everyone. I am wondering whether there is a certain technique I should use to identify all log lines in the asterisk/full logfile that are related to a single call. If a user reports that something strange happened with a certain call, I'd like to be able to easily go back and look at the asterisk/full logfile, and look at only the lines that are relevant. I am having some difficulty figuring out the best way to do this. At first I thought that the best way to do it would be to locate the call in the logs, then see what PID was used for that line in the log, and then grep asterisk/full for that PID. I can then find the call again, but only have to look at the lines with that PID (which seem to relate to that specific call during that time period). However, what I'm unclear is: 1. Are there other PIDs that could be acting on the same call at the same time? If so, I'd be missing part of the picture if I am only looking at 1 PID. 2. Can a single call get handed off from one asterisk PID to another? In looking at the asterisk/full logfile, I think this is true. But I am not 100% sure. If either of these are true, what is the best way to go back and look at the asterisk/full logfile in relation to the reported call? I would really appreciate any input on this. Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCP, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom firmware: split vs. combined
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html Howdy, all. What's the difference between split and combined firmware, which can be seen at the above link? I've googled to no avail, I'm afraid. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf - parkedcalls - transfer
At 12:27 AM 6/21/2010, you wrote: Almost 10 seconds, before the transfer to sip 200 is made, can I reduce that timer? And I cant see any button on the Cisco phone which will function like a direct transfer now, do I have to wait ? On my Aastra phones, I press Transfer 101 Transfer. So you might just try hanging up or pressing transfer again. Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom firmware: split vs. combined
On Mon, Jun 21, 2010 at 12:10 PM, Ken D'Ambrosio k...@jots.org wrote: http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.htmlHowdy, all. What's the difference between split and combined firmware, which can be seen at the above link? I've googled to no avail, I'm afraid. The split contains all the firmwares for the different model phones as separate files, the combined combines all of the firmwares into one big firmware file. The combined will cover any supported polycom phone model, but it takes longer to load. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom firmware: split vs. combined
From polycom web site: PLEASE NOTE: Combined download should be used where phones may be running pre-4.0 BootROM. Split download file is recommended, but requires that all phones are running BootROM 4.0 or newer. -Original Message- From: k...@jots.org Sent: Mon, 21 Jun 2010 13:10:37 -0400 (EDT) To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom firmware: split vs. combined http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html Howdy, all. What's the difference between split and combined firmware, which can be seen at the above link? I've googled to no avail, I'm afraid. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Send your photos by email in seconds... TRY FREE IM TOOLPACK at http://www.imtoolpack.com/default.aspx?rc=if3 Works in all emails, instant messengers, blogs, forums and social networks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom firmware: split vs. combined
Read: http://downloads.polycom.com/voice/voip/relnotes/spip_ssip_3_1_2_relnote s.pdf starting page 11/88: 1.4 Distribution Files 1.4.1 Release using individual (split) files 1.4.2 Release using Combined Image HTH, JDB -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio Sent: Monday, June 21, 2010 1:11 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom firmware: split vs. combined http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html Howdy, all. What's the difference between split and combined firmware, which can be seen at the above link? I've googled to no avail, I'm afraid. Thanks! -Ken -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom firmware: split vs. combined
On Mon, Jun 21, 2010 at 10:19 AM, Warren Selby wcse...@selbytech.com wrote: On Mon, Jun 21, 2010 at 12:10 PM, Ken D'Ambrosio k...@jots.org wrote: Howdy, all. What's the difference between split and combined firmware, which can be seen at the above link? I've googled to no avail, I'm afraid. The release notes talk about which one to choose and why in Section 1.4 Distribution Files http://downloads.polycom.com/voice/voip/relnotes/spip_ssip_vvx_release_notes_v3_2_3.pdf The split contains all the firmwares for the different model phones as separate files, the combined combines all of the firmwares into one big firmware file. The combined will cover any supported polycom phone model, but it takes longer to load. You also need to use the combined if you have a BootRom release older than 4.0.0. -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to tell if a dropped call is my fault
:53:28] VERBOSE[21559] logger.c: -- Executing [5053203...@from-internal-ntc-custom:2] Set(SIP/611-b7b9ae38, _NODEST=) in new stack [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [5053203...@from-internal-ntc-custom:3] Macro(SIP/611-b7b9ae38, record-enable|611|OUT|) in new stack [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [...@macro-record-enable:1] GotoIf(SIP/611-b7b9ae38, 1?check) in new stack [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Goto (macro-record-enable,s,4) [Jun 21 08:53:28] DEBUG[21559] app_macro.c: Executed application: GotoIf [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [...@macro-record-enable:4] AGI(SIP/611-b7b9ae38, recordingcheck|20100621-085328|1277132008.6419) in new stack [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck [Jun 21 08:53:29] VERBOSE[21559] logger.c: recordingcheck|20100621-085328|1277132008.6419: Outbound recording not enabled [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- AGI Script recordingcheck completed, returning 0 [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: AGI [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [...@macro-record-enable:5] MacroExit(SIP/611-b7b9ae38, ) in new stack [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [5053203...@from-internal-ntc-custom:4] Macro(SIP/611-b7b9ae38, dialout-trunk|2|5053203372||) in new stack [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [...@macro-dialout-trunk:1] Set(SIP/611-b7b9ae38, DIAL_TRUNK=2) in new stack [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: Set [Jun 21 08:53:29] DEBUG[21559] func_db.c: DB: AMPUSER/611/pinless not found in database. [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [...@macro-dialout-trunk:2] GosubIf(SIP/611-b7b9ae38, 0?sub-pincheck|s|1) in new stack [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: GosubIf [Jun 21 08:53:29] DEBUG[21559] func_db.c: DB: AMPUSER/611/pinless not found in database. [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [...@macro-dialout-trunk:3] GotoIf(SIP/611-b7b9ae38, 0?disabletrunk|1) in new stack [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: GotoIf [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [...@macro-dialout-trunk:4] Set(SIP/611-b7b9ae38, DIAL_NUMBER=5053203372) in new stack [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: Set [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [...@macro-dialout-trunk:5] Set(SIP/611-b7b9ae38, DIAL_TRUNK_OPTIONS=tr) in new stack [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: Set [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [...@macro-dialout-trunk:6] Set(SIP/611-b7b9ae38, OUTBOUND_GROUP=OUT_2) in new stack [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: Set [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [...@macro-dialout-trunk:7] GotoIf(SIP/611-b7b9ae38, 1?nomax) in new stack [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Goto (macro-dialout-trunk,s,9) [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: GotoIf [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [...@macro-dialout-trunk:9] GotoIf(SIP/611-b7b9ae38, 0?skipoutcid) in new stack [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: GotoIf [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [...@macro-dialout-trunk:10] Set(SIP/611-b7b9ae38, DIAL_TRUNK_OPTIONS=) in new stack [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: Set [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [...@macro-dialout-trunk:11] Macro(SIP/611-b7b9ae38, outbound-callerid|2) in new stack [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [...@macro-outbound-callerid:1] ExecIf(SIP/611-b7b9ae38, 0|SetCallerPres|) in new stack [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: ExecIf [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [...@macro-outbound-callerid:2] ExecIf(SIP/611-b7b9ae38, 0|Set|REALCALLERIDNUM=611) in new stack [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: ExecIf [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [...@macro-outbound-callerid:3] GotoIf(SIP/611-b7b9ae38, 1?normcid) in new stack [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Goto (macro-outbound-callerid,s,6) [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: GotoIf [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [...@macro-outbound-callerid:6] Set(SIP/611-b7b9ae38, USEROUTCID=5053251685) in new stack [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: Set [Jun 21 08:53:29] DEBUG[21559] func_db.c: DB: DEVICE/611/emergency_cid not found in database. [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [...@macro-outbound-callerid:7] Set(SIP
Re: [asterisk-users] How to tell if a dropped call is my fault
On Monday, June 21, 2010, Douglas Mortensen d...@impalanetworks.com wrote: I just had a user report that they called out to someone on a cell phone this morning, and after a short conversation, the call was dropped/lost. The person on the cell phone says that this is very rare. Snip I really would suspect the cell phone. /611-b7b9ae38' in macro 'dialout-trunk' [Jun 21 08:53:56] VERBOSE[21559] logger.c: == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/611-b7b9ae38' My experience is that with most programs, if they exit with a non-zero status, that means that there was some kind of error. If that is the case, I cannot tell what that would have been, based on the log. As far a the log/verbose level is concerned, asterisk was started as /usr/sbin/asterisk -vvvg Here's the complete asterisk/full as it pertains to the given time period. I do think that there are a few lines below that are relating to a different call, which we're not concerned with. [Jun 21 08:53:28] VERBOSE[22157] logger.c: Extension Changed 611[ext-local] new state InUse for Notify User 610 [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [5053203...@from-internal-ntc-custom:1] Macro(SIP/611-b7b9ae38, user-callerid|SKIPTTL|) in new stack [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [...@macro-user-callerid:1] Set(SIP/611-b7b9ae38, AMPUSER=611) in new stack [Jun 21 08:53:28] DEBUG[21559] app_macro.c: Executed application: Set [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [...@macro-user-callerid:2] GotoIf(SIP/611-b7b9ae38, 0?report) in new stack [Jun 21 08:53:28] DEBUG[21559] app_macro.c: Executed application: GotoIf [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [...@macro-user-callerid:3] ExecIf(SIP/611-b7b9ae38, 1|Set|REALCALLERIDNUM=611) in new stack [Jun 21 08:53:28] DEBUG[21559] app_macro.c: Executed application: ExecIf [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [...@macro-user-callerid:4] Set(SIP/611-b7b9ae38, AMPUSER=611) in new stack [Jun 21 08:53:28] DEBUG[21559] app_macro.c: Executed application: Set [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [...@macro-user-callerid:5] Set(SIP/611-b7b9ae38, AMPUSERCIDNAME=Kurt) in new stack [Jun 21 08:53:28] DEBUG[21559] app_macro.c: Executed application: Set [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [...@macro-user-callerid:6] GotoIf(SIP/611-b7b9ae38, 0?report) in new stack [Jun 21 08:53:28] DEBUG[21559] app_macro.c: Executed application: GotoIf [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [...@macro-user-callerid:7] Set(SIP/611-b7b9ae38, AMPUSERCID=611) in new stack [Jun 21 08:53:28] DEBUG[21559] app_macro.c: Executed application: Set [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [...@macro-user-callerid:8] Set(SIP/611-b7b9ae38, CALLERID(all)=Kurt 611) in new stack [Jun 21 08:53:28] DEBUG[21559] app_macro.c: Executed application: Set [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [...@macro-user-callerid:9] Set(SIP/611-b7b9ae38, REALCALLERIDNUM=611) in new stack [Jun 21 08:53:28] DEBUG[21559] app_macro.c: Executed application: Set [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [...@macro-user-callerid:10] ExecIf(SIP/611-b7b9ae38, 0|Set|CHANNEL(language)=) in new stack [Jun 21 08:53:28] DEBUG[21559] app_macro.c: Executed application: ExecIf [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [...@macro-user-callerid:11] GotoIf(SIP/611-b7b9ae38, 1?continue) in new stack [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Goto (macro-user-callerid,s,20) [Jun 21 08:53:28] DEBUG[21559] app_macro.c: Executed application: GotoIf [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [...@macro-user-callerid:20] NoOp(SIP/611-b7b9ae38, Using CallerID Kurt 611) in new stack [Jun 21 08:53:28] DEBUG[21559] app_macro.c: Executed application: Noop [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [5053203...@from-internal-ntc-custom:2] Set(SIP/611-b7b9ae38, _NODEST=) in new stack [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [5053203...@from-internal-ntc-custom:3] Macro(SIP/611-b7b9ae38, record-enable|611|OUT|) in new stack [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [...@macro-record-enable:1] GotoIf(SIP/611-b7b9ae38, 1?check) in new stack [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Goto (macro-record-enable,s,4) [Jun 21 08:53:28] DEBUG[21559] app_macro.c: Executed application: GotoIf [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Executing [...@macro-record-enable:4] AGI(SIP/611-b7b9ae38, recordingcheck|20100621-085328|1277132008.6419) in new stack [Jun 21 08:53:28] VERBOSE[21559] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck [Jun 21 08:53:29] VERBOSE[21559] logger.c: recordingcheck
[asterisk-users] when to use e1/t1 card?
This is a really rookie question: when should i use TE110P ISDN PRI Card? -- Necati DEMİR --- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] when to use e1/t1 card?
On Mon, Jun 21, 2010 at 3:04 PM, Necati Demir nde...@demir.web.tr wrote: This is a really rookie question: when should i use TE110P ISDN PRI Card? -- Necati DEMİR When you have a single PRI / BRI line you wish to terminate into an asterisk system. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] when to use e1/t1 card?
On Mon, Jun 21, 2010 at 03:12:40PM -0400, David Backeberg wrote: On Mon, Jun 21, 2010 at 3:04 PM, Necati Demir nde...@demir.web.tr wrote: This is a really rookie question: when should i use TE110P ISDN PRI Card? When you have a single PRI / BRI line you wish to terminate into an asterisk system. It's not going to help you terminate a BRI line :-) I should also note that this card can also be used for non-ISDN E1/T1. E.g. MFC/R2. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] when to use e1/t1 card?
El 21/06/10 14:04, Necati Demir escribió: This is a really rookie question: when should i use TE110P ISDN PRI Card? -- Necati DEMİR --- When you need to... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] when to use e1/t1 card?
On Mon, 21 Jun 2010, Necati Demir wrote: This is a really rookie question: when should i use TE110P ISDN PRI Card? From an economic standpoint? When you have more than x POTS lines where x depends on where you are in the world. Generally speaking, somewhere around 8 to 12. There are many advantages to PRI over POTS: ) Instantaneous call setup. ) Higher reliability. ) Less cabling behind the server. ) Better support from your provider. ) More features. ) Better audio quality. IMO, you should always use a PRI unless you can't afford it. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISP down internal phones become unavailable
On Mon, Jun 21, 2010 at 7:32 AM, Ryan Wagoner rswago...@gmail.com wrote: I saw the following lines in the log this morning. From my router logs I see that the connection went down as my ISP was doing maintenance for a few minutes last night. I can understand the external registrations timing out, but why do the phones become unreachable. They are on the internal lan within the same subnet as the Asterisk server. Internal DHCP and DNS was functional. If I had a PRI card in this system as well that would mean I couldn't make phone calls because the Internet is down. I have a PRI, and when the Internet connection goes out so do my phones. I suspect it is some type of DNS issue. I do have a SIP trunk, and it appears that if I lose DNS to the SIP trunk, the entire PBX is offline. I have no actual proof of any of this, and have not done any extensive testing to prove or disprove this. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf - parkedcalls - transfer
And I can't see any button on the Cisco phone which will function like a direct transfer now, do I have to wait...? Thank you for your reply. In my Dialplan menu on the SPA525g, I have a field where the input are, and I must say, I don't know if this is the right one, but the field contains this: Dial plan: (*xx|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.) What should I edit, if this is the right code On the Cisco 79xx series phones, you would modify a config file called dialplan.xml. I think on the SPA5xx series you can configure this parameter either in the main config file or from the web interface. You need to look for something like dialplan or dial plans, etc. This controls the timeout when entering digits. -- Thanks, --Warren Selby http://www.selbytech.com Best regards Aksel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] features.conf - parkedcalls - transfer
Hello, and thank you for your response. When I push transfer, the buttons with the function transfer disappears, and then I enter the sip number, Wait 10 seconds and then it transfers with the MOH in the background, when the connection/channel is made, Then transfer button is revealed again suddenly, and then I can push transfer again, and it transfers... =). I'm gonna try with callback function when no-answer, and the hangup option which you mentioned. Best regards Aksel -Opprinnelig melding- Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne av Ira Sendt: 21. juni 2010 19:16 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] features.conf - parkedcalls - transfer At 12:27 AM 6/21/2010, you wrote: Almost 10 seconds, before the transfer to sip 200 is made, can I reduce that timer? And I can't see any button on the Cisco phone which will function like a direct transfer now, do I have to wait.? On my Aastra phones, I press Transfer 101 Transfer. So you might just try hanging up or pressing transfer again. Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How do I access the Dialstatus numeric code received?
I need to access number received after a I dial a SIP or H323 call? suppose I get one of these: *404 Not found **486 Busy here **408 Request Timeout **480 Temporarily unavailable **480 Temporarily unavailable **403 Forbidden (+) ** 410 Gone **301 Moved Permanently **410 Gone ** 404 Not Found (=) **502 Bad Gateway **484 Address incomplete* How do I get the 404, 486, etc. F.A. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Create Conference and exit myself
I am sure you'll have to write your own dialplan for it in asterisk. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-21 12:31 PM, RSCL Mumbai rscl.mum...@gmail.com wrote: Hi, I am using Trixbox trixbox CE 2.6.2.3 (Stable) using Asterisk 1.4.22-4 I am looking for the following functionality: `` I receive a call from Mr. A. I put Mr. A on hold. I dial Mr. B I connect Mr. A's call (which was on hold) to Mr. B and I get out of the call. Mr. A Mr. B are in conversation, while my line is free to accept a new calls. What is the simplest way to achieve this ?? Thx Sans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to find a single call in logs
Every call is assigned a unique SIP channel id. I usually look for this id and then grep the log file by this id. It looks something like SIP/201-a08rfr7... if I remember correctly. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-21 1:13 PM, Douglas Mortensen d...@impalanetworks.com wrote: Hello everyone. I am wondering whether there is a certain technique I should use to identify all log lines in the asterisk/full logfile that are related to a single call. If a user reports that something strange happened with a certain call, I'd like to be able to easily go back and look at the asterisk/full logfile, and look at only the lines that are relevant. I am having some difficulty figuring out the best way to do this. At first I thought that the best way to do it would be to locate the call in the logs, then see what PID was used for that line in the log, and then grep asterisk/full for that PID. I can then find the call again, but only have to look at the lines with that PID (which seem to relate to that specific call during that time period). However, what I'm unclear is: 1. Are there other PIDs that could be acting on the same call at the same time? If so, I'd be missing part of the picture if I am only looking at 1 PID. 2. Can a single call get handed off from one asterisk PID to another? In looking at the asterisk/full logfile, I think this is true. But I am not 100% sure. If either of these are true, what is the best way to go back and look at the asterisk/full logfile in relation to the reported call? I would really appreciate any input on this. Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCP, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMD
Hi I am using the AMD application in a power dialing. All works well when I use an internal extension but when I try to use an external number, the AMD every times returns non human status. Also the AMDCAUSE returns Total-Time-5500. I am using the default parameters in AMD.CONF. Anybody has some idea? Thanks Sergio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Create Conference and exit myself
On Mon, Jun 21, 2010 at 12:25 PM, RSCL Mumbai rscl.mum...@gmail.com wrote: What is the simplest way to achieve this ?? Use the transfer button on your phone? -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I access the Dialstatus numeric code received?
On Monday 21 June 2010 16:09:22 CDR wrote:
I need to access number received after a I dial a SIP or H323 call?
suppose I get one of these:
*404 Not found
**486 Busy here
**408 Request Timeout
**480 Temporarily unavailable
**480 Temporarily unavailable
**403 Forbidden (+) **
410 Gone
**301 Moved Permanently
**410 Gone **
404 Not Found (=)
**502 Bad Gateway
**484 Address incomplete*
How do I get the 404, 486, etc.
F.A.
Not available in anything other than trunk (to be 1.8). It depends upon a new
feature, so it's not something you can easily backport. After dialling, the
SIP code is available in ${HASH(SIP_CAUSE,channel-name)}
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] Is this failed Asterisk setup typical?
On Thu, Jun 3, 2010 at 6:16 AM, Gilles codecompl...@free.fr wrote: Hello I just read this article and would like some feedback from experienced Asterisk users: === Failed open source VoIP deployment leads to hosted VoIP strategy By Jessica Scarpati snip http://searchunifiedcommunications.techtarget.com/news/article/0,289142,sid186_gci1508323_mem1,00.html (free registration required) === So it looks like this company had the following issues: * No in-house technical expertise to set up and maintain Asterisk * Not enough bandwidth * DID module apparently not reliable Based on your experience, are those problems typical? Thank you. Not in our experience as a 500-phone, 20-site install for a municipal government. We are just migrating from our first generation install to replacement hardware (to new blades from servers that are now 5 years old) and are still committed to Asterisk for it. Done right, Asterisk saved us over three quarters of a million dollars over a big-C install. CP -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is this failed Asterisk setup typical?
On Mon, 21 Jun 2010 16:47:08 -0700, CunningPike cunningp...@gmail.com wrote: Not in our experience as a 500-phone, 20-site install for a municipal government. We are just migrating from our first generation install to replacement hardware (to new blades from servers that are now 5 years old) and are still committed to Asterisk for it. Thanks for the feedback. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD
Sometimes you have to play some audio before calling AMD or other audio functions for whatever reason... Like play 100ms of silence in a .wav file immediately after answer. This causes RTP to be sent out to the carrier. John From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tetra Informatica Sent: Monday, June 21, 2010 3:39 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] AMD Hi I am using the AMD application in a power dialing. All works well when I use an internal extension but when I try to use an external number, the AMD every times returns non human status. Also the AMDCAUSE returns Total-Time-5500. I am using the default parameters in AMD.CONF. Anybody has some idea? Thanks Sergio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Update to chan_ooh323 wrapper
I see that objective systems has updated their ooh323 stack, but it is not compatible with the latest chan_ooh323 wrapper available on their site. Has anyone update the chan_ooh323 wrapper for Asterisk 1.6.2.x ? Michelle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can't make calls out of Astribank - show regdump doesn't show voltage - Getting warning due to old driver
Hi Guys, An 8 channel Astribank is connected to Trixbox 2.8 and I ran freepbx-module-zapauto but I get the following when running these commands and can't make calls out: [Trixbox]# dahdi_genconf xpporder /usr/sbin/dahdi_genconf: warning - OLD DRIVER: missing '/sys/bus/xpds/devices/00:0:0/timing_priority'. Fall back to /proc pbx*CLI dahdi show channels Chan Extension Context Language MOH Interpret Blocked State pseudo default default In Service 1 from-pstn default In Service 2 from-pstn default In Service 3 from-pstn default In Service 4 from-pstn default In Service 5 from-pstn default In Service 6 from-pstn default In Service 7 from-pstn default In Service 8 from-pstn default In Service pbx*CLI dahdi show status Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO Xorcom XPD #00/00: FXO OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) pbx*CLI dahdi show regdump 1 Unable to get registers on channel 1 Unable to get stats on channel 1 [Trixbox]# dahdi_hardware -v /usr/sbin/dahdi_hardware: warning - OLD DRIVER: missing '/sys/bus/xpds/devices/00:0:0/timing_priority'. Fall back to /proc usb:002/002 xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware LABEL=[usb:X1044207] connect...@usb-:00:1d.7-4 XBUS-00/XPD-00: FXO (8) Span 1 DAHDI-SYNC Note: This Astribank is deployed in United Arab Emirates and I am not sure what the line type is in terms of Ground or Loop start and wondering if that makes a difference with the Astribank and the fact that it can't how the voltage using show regdump And I am definitly not sure what that warning of OLD DRIVER is about. Any help is appreciated. Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
