Re: [asterisk-users] Inquiry:How to hide Caller Id
That doesn't happen on all phones. Either find a way to block that feature on the phone, or change phones for that location. I assume you don't want the user to know that phone's local number. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi Sent: Monday, August 31, 2009 1:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Inquiry:How to hide Caller Id Thank you for your reply . Yes , he is seeing his own number on his phone upon going off hook and before dialing any number . Can you please do me favor and confirm if it is not a feature of Asterisk that I can disable it ? Regards H.Motamedi On Mon, Aug 31, 2009 at 6:53 AM, Matt Riddell li...@venturevoip.com wrote: On 31/08/09 5:49 PM, hadi motamedi wrote: Sorry for lack of enough information . I mean my subscriber when goes off hook he will see his own number displayed on his phone . I need to disable this feature on my Asterisk .The phone type is ANABELL phone . Please do me favor and let me know how can I disable this feature on my Asterisk ? Hi, If he is seeing his own number on his display before he has dialed any numbers then it is probably a feature of the phone - in which case you need to disable it there. If you're talking about an incoming call then it's different. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to hide Caller Id
I couldn't find any information on this brand of phone on the internet at all. PaulH hadi motamedi wrote: Sorry for lack of enough information . I mean my subscriber when goes off hook he will see his own number displayed on his phone . I need to disable this feature on my Asterisk .The phone type is ANABELL phone . Please do me favor and let me know how can I disable this feature on my Asterisk ? Looking forward your reply Regards H.Motamedi On Mon, Aug 31, 2009 at 6:28 AM, Matt Riddell li...@venturevoip.com mailto:li...@venturevoip.com wrote: On 31/08/09 5:24 PM, hadi motamedi wrote: Dear All Can you please do me favor and let me know how I can hide the subs number being displayed on his phone when he goes off hook ? I mean when the subs goes off hook he sees his assigned number on his phone and I need to disable this feature . I don't know from which configuration file this feature is coming so please let me know how can I disable it . You're not really giving enough information. Who sees the number? Where do they see it? What type of phone? What is a subs? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/app_rpt and bandwidth
On Monday 31 August 2009 12:56:32 pm Steve Totaro wrote: On Sun, Aug 30, 2009 at 5:57 AM, Michael Maxwell metalmic...@gmail.comwrote: When a signal is *not* being received and or transmitted by the radio system attached to Asterisk/app_rpt via its interface, is the incoming and or outgoing data suppressed (silence suppression)? Silence suppression is not supported in Asterisk as far as I know. It is basically PTT or COR that triggers the traffic call AFAIK. Maybe silence suppression was the wrong wording. I ask this, because i run a small network of Asterisk boxes (called PRAIL) which peer to link PMR/CB radio and want to mix both app_rpt and RoA based gateways/nodes with an eye on bandwidth usage. Thanks for the reply, Steve :o) On Monday 31 August 2009 09:34:36 am Eric Fort wrote: the quick answer is I don't know. but here's where the answer can be found: if you don't get a reply by posting to the list specific to app_rpt then drop an email to: hws...@rodgers.sdcoxmail.com he knows. app_rpt has a list all it's own - here's the address to post. app-...@qrvc.com Thanks Eric :o) -- Thanks, Michael Maxwell eMail: metalm...@gmail.com Phone: +61 (03) 8680 4946 Web: mikey.webhop.org Powered By: PCBSD.org | FreeBSD.org | OpenSource.org PRAIL.org - Australia-Wide radio communications for free Sponsor: Hightek Hosting - A New Wave in IT and Hosting Technology Hosting, IT services, sales and onsite support! 1300 85 34 30 - www.hightekhosting.com.au ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR to Postgres Centos
I am following this procedure ou have to compile asterisk with the cdr_pgsql.o module, for this follow the steps: Configure asterisk with postgresql support: ./configure --with-postgres=dir where postgresql is installed Then issue the command: make menuconfig in the menu select 2.Call Detail Recording - then check cdr_pgsql build asterisk make Install it sudo make install Then add, in the file modules.conf, the line: load = cdr_pgsql.so but when i execute this ./configure --with-postgres=dir where postgresql is installed it gives an error for missing an pg_config file . i searched the PC but it really dont exists. but database server is fine and working OK! what to do in this situation On Mon, Aug 31, 2009 at 10:17 AM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Hello I have installed postgres and asterisk on centos. I confused which module i have to install to store CDR info to postgres DB. I have visited some links but mostly are for configuration file in database. I only want CDR to Database not the configuration files for time being Please help -- Best Regards Shakeel Abbas -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to hide Caller Id
Thank you for your reply . Yes , he is seeing his own number on his phone upon going off hook and before dialing any number . Can you please do me favor and confirm if it is not a feature of Asterisk that I can disable it ? Regards H.Motamedi On Mon, Aug 31, 2009 at 6:53 AM, Matt Riddell li...@venturevoip.com wrote: On 31/08/09 5:49 PM, hadi motamedi wrote: Sorry for lack of enough information . I mean my subscriber when goes off hook he will see his own number displayed on his phone . I need to disable this feature on my Asterisk .The phone type is ANABELL phone . Please do me favor and let me know how can I disable this feature on my Asterisk ? Hi, If he is seeing his own number on his display before he has dialed any numbers then it is probably a feature of the phone - in which case you need to disable it there. If you're talking about an incoming call then it's different. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help me testing this webphone atwww.VisionVoIP.com
I just tried it on 3 different numbers. Dialed as 10 digits NPANXX I was told I am sorry but you can only dial within North America..etc. C Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Sunday, August 30, 2009 9:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help me testing this webphone atwww.VisionVoIP.com Thank you everyone who tested the webphone, but I haven't got input from anybody. Most of the calls made were unfortunately unsuccessful, but I would like to know what error you got? Did the webphone stayed in Loading... state and never completed its loading, or your local firewall blocked, it, or something else happened? -- Zeeshan A Zakaria On Sun, Aug 30, 2009 at 1:52 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Greetings everyone, I've been trying to make this java based webphone work for everybody visiting my website, but seems like for many users it doesn't work. In order to get a better idea what is the success rate of this webphone, I would appreciate help from anybody who could make a few calls from it within North America and if it doesn't work, send me what error you get, or if it works, tell me it sounds right, no echoing etc. I am keeping calling free for now for testing purposes. The webphone is located at http://www.visionvoip.com Thanks, -- Zeeshan A Zakaria -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIPP how can we give delays between 2 calls
hello, i am using following SIPP command to test My meetme conference ./sipp -sn uac -d 30 -s 8600 127.0.0.1 -l 20 which generates 20 call to my server but i need to give delay between each call once 1 st call is placed then second call should be placed after few seconds and is there any method to play some file file or data while SIPP call is placed i got very bad sound while sipp calls connect to my meetme room can any body have idea regarding this , regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to hide Caller Id
Thank you for your reply . I really don't want the user to know the phone's local number . Can you please do me favor and propose one of the available phones that doesn't have this feature ? Regards H.Motamedi On Mon, Aug 31, 2009 at 7:12 AM, Cary Fitch ca...@usawide.net wrote: That doesn’t happen on all phones. Either find a way to block that “feature” on the phone, or change phones for that location. I assume you don’t want the user to know that phone’s local number. Cary Fitch -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *hadi motamedi *Sent:* Monday, August 31, 2009 1:09 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Inquiry:How to hide Caller Id Thank you for your reply . Yes , he is seeing his own number on his phone upon going off hook and before dialing any number . Can you please do me favor and confirm if it is not a feature of Asterisk that I can disable it ? Regards H.Motamedi On Mon, Aug 31, 2009 at 6:53 AM, Matt Riddell li...@venturevoip.com wrote: On 31/08/09 5:49 PM, hadi motamedi wrote: Sorry for lack of enough information . I mean my subscriber when goes off hook he will see his own number displayed on his phone . I need to disable this feature on my Asterisk .The phone type is ANABELL phone . Please do me favor and let me know how can I disable this feature on my Asterisk ? Hi, If he is seeing his own number on his display before he has dialed any numbers then it is probably a feature of the phone - in which case you need to disable it there. If you're talking about an incoming call then it's different. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to hide Caller Id
Sorry for mis-typing in phone type . Please be informed that the current phone type our subscribers are using is TP6000 ones . On Mon, Aug 31, 2009 at 7:24 AM, Paul Hales pdha...@optusnet.com.au wrote: I couldn't find any information on this brand of phone on the internet at all. PaulH hadi motamedi wrote: Sorry for lack of enough information . I mean my subscriber when goes off hook he will see his own number displayed on his phone . I need to disable this feature on my Asterisk .The phone type is ANABELL phone . Please do me favor and let me know how can I disable this feature on my Asterisk ? Looking forward your reply Regards H.Motamedi On Mon, Aug 31, 2009 at 6:28 AM, Matt Riddell li...@venturevoip.com mailto:li...@venturevoip.com wrote: On 31/08/09 5:24 PM, hadi motamedi wrote: Dear All Can you please do me favor and let me know how I can hide the subs number being displayed on his phone when he goes off hook ? I mean when the subs goes off hook he sees his assigned number on his phone and I need to disable this feature . I don't know from which configuration file this feature is coming so please let me know how can I disable it . You're not really giving enough information. Who sees the number? Where do they see it? What type of phone? What is a subs? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPP how can we give delays between 2 calls
-r is a flag that regulates the call setup rate per second. DHAVAL INDRODIYA wrote: hello, i am using following SIPP command to test My meetme conference ./sipp -sn uac -d 30 -s 8600 127.0.0.1 -l 20 which generates 20 call to my server but i need to give delay between each call once 1 st call is placed then second call should be placed after few seconds and is there any method to play some file file or data while SIPP call is placed i got very bad sound while sipp calls connect to my meetme room can any body have idea regarding this , regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to hide Caller Id
A Google of that model showed a discontinued Telstra corded phone. But in any case SNOM and Grandstream phones Do show the number before you pick up the handset. I would suggest you use a Grandstream 286 voip adapter and a standard corded or wireless phone so that the caller doesn't have a display to see. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi Sent: Monday, August 31, 2009 1:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Inquiry:How to hide Caller Id Sorry for mis-typing in phone type . Please be informed that the current phone type our subscribers are using is TP6000 ones . On Mon, Aug 31, 2009 at 7:24 AM, Paul Hales pdha...@optusnet.com.au wrote: I couldn't find any information on this brand of phone on the internet at all. PaulH hadi motamedi wrote: Sorry for lack of enough information . I mean my subscriber when goes off hook he will see his own number displayed on his phone . I need to disable this feature on my Asterisk .The phone type is ANABELL phone . Please do me favor and let me know how can I disable this feature on my Asterisk ? Looking forward your reply Regards H.Motamedi On Mon, Aug 31, 2009 at 6:28 AM, Matt Riddell li...@venturevoip.com mailto:li...@venturevoip.com wrote: On 31/08/09 5:24 PM, hadi motamedi wrote: Dear All Can you please do me favor and let me know how I can hide the subs number being displayed on his phone when he goes off hook ? I mean when the subs goes off hook he sees his assigned number on his phone and I need to disable this feature . I don't know from which configuration file this feature is coming so please let me know how can I disable it . You're not really giving enough information. Who sees the number? Where do they see it? What type of phone? What is a subs? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ http://www.astricon.net/ 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 http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ 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 http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to hide Caller Id
Hi, SNOM dosent show the number, it shows user realname. http://wiki.snom.com/wiki/index.php/Settings/user_realname // Magnus Från: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] För Cary Fitch Skickat: den 31 augusti 2009 09:06 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' Ämne: Re: [asterisk-users] Inquiry:How to hide Caller Id A Google of that model showed a discontinued Telstra corded phone. But in any case SNOM and Grandstream phones Do show the number before you pick up the handset. I would suggest you use a Grandstream 286 voip adapter and a standard corded or wireless phone so that the caller doesn't have a display to see. Cary Fitch From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi Sent: Monday, August 31, 2009 1:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Inquiry:How to hide Caller Id Sorry for mis-typing in phone type . Please be informed that the current phone type our subscribers are using is TP6000 ones . On Mon, Aug 31, 2009 at 7:24 AM, Paul Hales pdha...@optusnet.com.aumailto:pdha...@optusnet.com.au wrote: I couldn't find any information on this brand of phone on the internet at all. PaulH hadi motamedi wrote: Sorry for lack of enough information . I mean my subscriber when goes off hook he will see his own number displayed on his phone . I need to disable this feature on my Asterisk .The phone type is ANABELL phone . Please do me favor and let me know how can I disable this feature on my Asterisk ? Looking forward your reply Regards H.Motamedi On Mon, Aug 31, 2009 at 6:28 AM, Matt Riddell li...@venturevoip.commailto:li...@venturevoip.com mailto:li...@venturevoip.commailto:li...@venturevoip.com wrote: On 31/08/09 5:24 PM, hadi motamedi wrote: Dear All Can you please do me favor and let me know how I can hide the subs number being displayed on his phone when he goes off hook ? I mean when the subs goes off hook he sees his assigned number on his phone and I need to disable this feature . I don't know from which configuration file this feature is coming so please let me know how can I disable it . You're not really giving enough information. Who sees the number? Where do they see it? What type of phone? What is a subs? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.comhttp://www.api-digital.com/ http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.nethttp://www.astricon.net/ http://www.astricon.net/ 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.comhttp://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.nethttp://www.astricon.net/ 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.comhttp://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.nethttp://www.astricon.net/ 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flite module for asterisk 1.6.x
Lefteris Zafiris schrieb: I have written a simple application for asterisk 1.6 that uses the Flite tts engine to render text to speech. Source is available here: http://zaf.github.com/Asterisk-Flite/ It works more or less like the festival app, can use cache etc. Its only tested against asterisk 1.6.1 on X86 linux but i guess it works for other 1.6 branches too. Comments, fixes and suggestion are welcome. Hi Zaf! I wonder what is the benefit of using Flite over Festival? thanks klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPP how can we give delays between 2 calls
thanks Alex, it works but can you tell me about any sound playing on SIPP means , once SIPP channels connect in conference room then there is lots of noise , is there any way to reduce it. regards Dhaval On Mon, Aug 31, 2009 at 12:38 PM, Alex Balashov abalas...@evaristesys.comwrote: -r is a flag that regulates the call setup rate per second. DHAVAL INDRODIYA wrote: hello, i am using following SIPP command to test My meetme conference ./sipp -sn uac -d 30 -s 8600 127.0.0.1 -l 20 which generates 20 call to my server but i need to give delay between each call once 1 st call is placed then second call should be placed after few seconds and is there any method to play some file file or data while SIPP call is placed i got very bad sound while sipp calls connect to my meetme room can any body have idea regarding this , regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Measuring voice quality with Asterisk
Olle E. Johansson schrieb: 27 aug 2009 kl. 11.24 skrev Klaus Darilion: Hi! I want to use Asterisk as load generator to test quality degradation with increased load (e.g. testing other SIP equipment or IP-links). Is anybody aware of such a setup with Asterisk - is it possible to get RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)? Check the RTPAUDIOQOS variable documented in channelvariables.txt :-) You propably want to catch it in the h extension and put it in the CDRs or something. That is good for after-call statistics. I am currently trying to get real-time stats. regards klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Measuring voice quality with Asterisk
On 31/08/09 8:47 PM, Klaus Darilion wrote: Olle E. Johansson schrieb: 27 aug 2009 kl. 11.24 skrev Klaus Darilion: Hi! I want to use Asterisk as load generator to test quality degradation with increased load (e.g. testing other SIP equipment or IP-links). Is anybody aware of such a setup with Asterisk - is it possible to get RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)? Check the RTPAUDIOQOS variable documented in channelvariables.txt :-) You propably want to catch it in the h extension and put it in the CDRs or something. That is good for after-call statistics. I am currently trying to get real-time stats. Do the rtcp stats give out info over the manager? Maybe you could use those? They definitely come up in the console if enabled. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flite module for asterisk 1.6.x
Klaus Darilion wrote: Lefteris Zafiris schrieb: I have written a simple application for asterisk 1.6 that uses the Flite tts engine to render text to speech. Source is available here: http://zaf.github.com/Asterisk-Flite/ It works more or less like the festival app, can use cache etc. Its only tested against asterisk 1.6.1 on X86 linux but i guess it works for other 1.6 branches too. Comments, fixes and suggestion are welcome. Hi Zaf! I wonder what is the benefit of using Flite over Festival? thanks klaus Flite is lightweight, simple and easy to install, thers no need for configuration to get it running no deamon etc etc and can be ran even in embedded systems. For more info check flite home page: http://www.speech.cs.cmu.edu/flite/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crystal Recording Interface
Hi, Is there anyone there that installed successfully the CRI package and manages to play the calls listed in the call monitor page? Regards. Original Message Subject: Re: [asterisk-users] Crystal Recording Interface From: Cyprus VoIP voi...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Monday, 31 August, 2009 02:11:21 I manage to see the calls in the monitor now, but unlike the example on Tikal's site, I don't have the Play Call button next to each call, so I can't listen to it. Could it be linked to the file name that I used for storing the recorded calls? Original Message Subject: Re: [asterisk-users] Crystal Recording Interface From: Danny Nicholas da...@debsinc.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Friday, 28 August, 2009 18:56:46 The key as far as I can see is what is in your CDR database. Once it is correct, everything will be as expected. This might not be a fun hill to climb. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cyprus VoIP Sent: Friday, August 28, 2009 10:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Crystal Recording Interface I installed the asterisk add-ons (mysql,cdr), Apache MySQL and php are installed and running and I have the CRI web interface available, too. Seems to me I just need the extensions.conf and maybe something to do with CDRs to work out. Original Message Subject: Re: [asterisk-users] Crystal Recording Interface From: Danny Nicholas da...@debsinc.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Friday, 28 August, 2009 18:22:49 As far as I can see, the main requirements are these: 1. An Apache install to install the software into 2. PHP is active 3. Your Asterisk uses a MYSQL or Postgres CDR. 4. You have access to the database password/id. I'm guessing the documentation is pretty scarce since it's a one-trick-pony GPL offering. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cyprus VoIP Sent: Friday, August 28, 2009 9:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Crystal Recording Interface Hello all, I download from Tikal's site the Crystal Recording Interface and installed it on my Asterisk server, but there's no reference in the installation instructions there regarding the necessary settings on the Asterisk itself. Is anyone using it? Any detailed explanation on the implementation of that solution anywhere? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Regular expression to validate any phonenumber
Hi I am using asterisk version 1.6.0.5 I have build up one utility that will fire Originate Action on Manager... In which, i have define number to call eg. 919912312345 (MobileNumber) How can i know that this number format is true for Indian Number... In originate action, user can enter any international number.. How can I came to know this number format is right for that country...?? IS there any regular expression to validate this number . regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Regular expression to validate any phonenumber
Use the pattern matching P137 in Asterisk: the future of telephony For example Exten = 919X,n, Cordialement, P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de DHAVAL INDRODIYA Envoyé : lundi 31 août 2009 11:53 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Asterisk Regular expression to validate any phonenumber Hi I am using asterisk version 1.6.0.5 I have build up one utility that will fire Originate Action on Manager... In which, i have define number to call eg. 919912312345 (MobileNumber) How can i know that this number format is true for Indian Number... In originate action, user can enter any international number.. How can I came to know this number format is right for that country...?? IS there any regular expression to validate this number . regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Regular expression to validate any phonenumber
Use the pattern matching P137 in Asterisk: the future of telephony For example Exten = _919X,n, Cordialement, P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de DHAVAL INDRODIYA Envoyé : lundi 31 août 2009 11:53 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Asterisk Regular expression to validate any phonenumber Hi I am using asterisk version 1.6.0.5 I have build up one utility that will fire Originate Action on Manager... In which, i have define number to call eg. 919912312345 (MobileNumber) How can i know that this number format is true for Indian Number... In originate action, user can enter any international number.. How can I came to know this number format is right for that country...?? IS there any regular expression to validate this number . regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 and GUI Configuration Help
Hello, I am trying to setup an asterisk box for a small office that has 4 phone lines and a fax. The fax will not be going through the box. I have Digium TDM410P to take 4 analog lines and I will be using grandstream gxp2000 for our setup. I have read the docs just do not understand the dialplan, incoming calls, routing process. I setup the trunks which is the 4 phone lines so the first two numbers go to a IVR and the other two will be direct lines to gxp 2000. How can I configure a dialplan for this. The gui seems to be messing everything up it seems to not want to update or allow you to make changes. It says it made the change but then when you click apply it does do anything. Can someone share what they've done. I know this works as I used Asterisk 1.2. I just want something real simple incoming calls to ivr, with exception to two lines directly to phone. BTW I don't want to done for me just some example code an experiences if possible. Thanks in advance! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to deal with PayPal frauds?
Thanks Matt, and everybody else, very useful information. I guess I'll have to sit again and spend time coding delays, small amount payments for new accounts and paypal=signup email match. -- Zeeshan A Zakaria On Mon, Aug 31, 2009 at 12:07 AM, Matt Riddell li...@venturevoip.comwrote: On 31/08/09 2:45 PM, Zeeshan Zakaria wrote: Those who are more experienced in this business, please advise how to avoid this type of fraud, and which service to use in place of PayPal, because PayPal doesn't seem the right payment solution for a prepaid VoIP service. Also now that they have all the payments put on hold and asking for a resolution, their resolution center is good only for shipped merchendise, not for online services. How would I prove to them that the buyer who is asking his money back has already utilized my service by making lot of international calls, which I now have to pay for to the carrier. I've used CDR for that and don't automatically accept payments. When we receive a payment we compare: 1. IP Address of user (whois normally gives approximate location) 2. Paypal account holder email (should match sign up email) 3. Countries for emails and ip address should match. 4. Initial payment should be $1-$2 (i.e. noone is going to sign up for a service and in order to test it put down $500 via paypal) If any of the above look suspect I ask the paypal account holder to email me and start looking at email headers to see how sus it looks. If it's a large amount then they have to have already been doing business with us successfully with small amounts - most scammers can't be bothered doing this. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help me testing this webphone atwww.VisionVoIP.com
Cary, thanks for your feedback. You tried dialing directory assistance numbers which cost dollar a minute. They can't be free. But you got the voice messages, which means the webphone worked for you. -- Zeeshan A Zakaria On Mon, Aug 31, 2009 at 2:18 AM, Cary Fitch ca...@usawide.net wrote: I just tried it on 3 different numbers. Dialed as 10 digits NPANXX I was told “I am sorry but you can only dial within North America….etc. C Fitch -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria *Sent:* Sunday, August 30, 2009 9:23 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Help me testing this webphone atwww.VisionVoIP.com Thank you everyone who tested the webphone, but I haven't got input from anybody. Most of the calls made were unfortunately unsuccessful, but I would like to know what error you got? Did the webphone stayed in Loading... state and never completed its loading, or your local firewall blocked, it, or something else happened? -- Zeeshan A Zakaria On Sun, Aug 30, 2009 at 1:52 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Greetings everyone, I've been trying to make this java based webphone work for everybody visiting my website, but seems like for many users it doesn't work. In order to get a better idea what is the success rate of this webphone, I would appreciate help from anybody who could make a few calls from it within North America and if it doesn't work, send me what error you get, or if it works, tell me it sounds right, no echoing etc. I am keeping calling free for now for testing purposes. The webphone is located at http://www.visionvoip.com Thanks, -- Zeeshan A Zakaria -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 61, Issue 85
Topic 6: RE:unable to execute command hi there i tried to execute the command as suggest like exten = 1987,1,Playback(posix-restarting) exten = 1987,2,wait(1) exten = 1987,3,System(/usr/bin/python /home/docas/Desktop/mess1.py) exten= 1987,4,Hangup it still doesn't work,now it does it give unable to execute command but it doesn't reach the system command it just execute 1 and 2 priority and then hangup,i checked if the script is executed and i found out its not.i did change permissions so the script is executable; i would apreciate if someone help docas asterisk-users-requ...@lists.digium.com 08/30/09 6:24 PM Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-requ...@lists.digium.com You can reach the person managing the list at asterisk-users-ow...@lists.digium.com When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. Re: asterisk-users Digest, Vol 61, Issue 84 (Gomtesh Jain) 2. GoToIfTime : how to define sep 25th till oct 10th ? (jonas kellens) 3. Re: Asterisk 1.6.0.14 and 1.6.1.5 Now Available - pbx/lua.c changes (Bob Gustafson) 4. Re: GoToIfTime : how to define sep 25th till oct 10th ? (David Backeberg) 5. Re: GoToIfTime : how to define sep 25th till oct 10th ? (MeetMeCall) 6. unable to execute command from (DOCAS DUDU ZULU) 7. Re: unable to execute command from (Alex Balashov) 8. Help me testing this webphone at www.VisionVoIP.com (Zeeshan Zakaria) 9. Re: Asterisk 1.6.0.14 and 1.6.1.5 Now Available - pbx/lua.c changes (Tilghman Lesher) 10. Asterisk/app_rpt and bandwidth (Michael Maxwell) 11. Re: PRI worked fine for months, nowit stopps working after 2-3 hours (Loic Didelot) 12. MySQL syntax error : I really don't see where... (jonas kellens) 13. Re: MySQL syntax error : I really don't see where... (Doug Lytle) 14. Re: MySQL syntax error : I really don't see where... (Per Jessen) 15. Need help - CDR MySQL (Cyprus VoIP) 16. Re: Need help - CDR MySQL (Tilghman Lesher) 17. Re: Need help - CDR MySQL (Cyprus VoIP) 18. Re: Need help - CDR MySQL (Pascal Bruno) 19. Re: Need help - CDR MySQL (Cyprus VoIP) 20. Re: Need help - CDR MySQL (hh174) 21. Re: Need help - CDR MySQL (Cyprus VoIP) -- Message: 1 Date: Sat, 29 Aug 2009 22:48:18 +0530 From: Gomtesh Jain gomt...@gmail.com Subject: Re: [asterisk-users] asterisk-users Digest, Vol 61, Issue 84 To: asterisk-users@lists.digium.com Message-ID: a22146bc0908291018u71500102pf7e976aada0ec...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 On Sat, Aug 29, 2009 at 10:30 PM, asterisk-users-requ...@lists.digium.comwrote: Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-requ...@lists.digium.com You can reach the person managing the list at asterisk-users-ow...@lists.digium.com When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. Re: Accessing to ekiga.net through Asterisk (Daniel Bareiro) 2. Re: cannot run agi scripts (Michael Connors) 3. Re: cannot run agi scripts (Steve Edwards) 4. Re: cannot run agi scripts (Steve Edwards) 5. Re: cannot run agi scripts (Steve Edwards) 6. Asterisk 1.6.0.14 and 1.6.1.5 Now Available (Asterisk Development Team) -- Message: 1 Date: Sat, 29 Aug 2009 12:06:08 -0300 From: Daniel Bareiro daniel-lis...@gmx.net Subject: Re: [asterisk-users] Accessing to ekiga.net through Asterisk To: asterisk-users@lists.digium.com Message-ID: slrnh9igvv.4ko.daniel-lis...@marian.freesoftware.org Content-Type: text/plain; charset=UTF-8 -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 El mi?rcoles 19 de agosto del 2009 a las 08:04:17 -0300, SIP escribi?: Daniel, Hi SIP. I'm a little confused as to what I'm seeing here. You're bounding through two RFC1918 address networks -- 10.1.0.X and 192.168.2.X. Is this some sort of dual NAT scenario? Perhaps if you can explain a little more about your network setup. This it is a scheme of my network configuration: +--+
Re: [asterisk-users] Measuring voice quality with Asterisk
Matt Riddell schrieb: On 31/08/09 8:47 PM, Klaus Darilion wrote: Olle E. Johansson schrieb: 27 aug 2009 kl. 11.24 skrev Klaus Darilion: Hi! I want to use Asterisk as load generator to test quality degradation with increased load (e.g. testing other SIP equipment or IP-links). Is anybody aware of such a setup with Asterisk - is it possible to get RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)? Check the RTPAUDIOQOS variable documented in channelvariables.txt :-) You propably want to catch it in the h extension and put it in the CDRs or something. That is good for after-call statistics. I am currently trying to get real-time stats. Do the rtcp stats give out info over the manager? Maybe you could use those? They definitely come up in the console if enabled. Yes, those events are reported over AMI. In my first tests I connected via AMI and fetched a channel list, and then for each channel I fetched CHANNEL(qos,audio,all). Of course that did not scaled. Thus, I wonder if reading RTP events scale. Further I found out that the RTCP events do not have a reference to a channel. Thus, it is not possible to find out which channel received/sent this RTCP packet. As the QoS settings are also stored within the rtp channel structure, I was thinking of adding some functionality to rtp handler to export QoS stats every few seconds and when the call ends. regards klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Clarifying RX and TX gains
I've done gain tuning as per the info I've found online. I've got my RXGain set so my volumes list as about 14,800 (using a milliwatt test number and ztmonitor -vv). However listening to the line now, this sounds too loud to me. The person speaking sounds fine, but I've now got a large amount of background hiss coming thru. In order to get the recommended levels, most of my lines are in the range of +12.5, so I'm wondering if I'm just exceeding the ability of my Sangoma card to amplify the audio. My TXGain's are all still at 0.0 as the Sangoma card didn't seem to want to let me adjust the gains. My levels are listing in the 5,000 range, but it seemed no matter what I changed the values to, there was virtually no difference made to the audio levels. Listening to the transmit levels everything sounds fine. What I am still getting is occasional static on the lines. Pops, squeaks, that kind of stuff. Sounds exactly like a handset cord is going bad, but it happens on any phone (and all have brand new handset cords). The person on the far end says they hear nothing wrong with the call. However, when I have asterisk record the call, the static is present on the recording. I also get the occasional robotic voice echoing back on anything said by the local person. That sounds like it should be typical echo on the line (I'm using a Sangoma A400DE with hardware echo cancellation). Everything I can find on echo and static tells me I should adjust my RX and TX gains. So that brings me to my confusion/clarification. I've got my RXGain set to the recommended level, and my TXGain doesn't seem to be very adjustable upwards as it would appear to need based on the recommended levels. Could my static and echo be caused by my TXGain being too low, or possibly RX or TX actually being too high (since listening to the RX reveals lots of background hiss)? I'm trying to work out this last little bit on my system as I'm getting grief that these calls have audio issues. I'm running out of things to check, so I'm looking for clarification on where exactly my RX and TX gains should be and which directions they will impact static and echo. Thanks! -chris www.mythtech.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to deal with PayPal frauds?
When you start taking credit card payments (assuming you will), be careful with small payment amounts. You'll become a fraud haven. A lot of CC thieves or people who've just bought a CC number will use a small amount charge to check and see if the card is any good. Check out some of the MaxMind stuff for fraud prevention. They will do a lot of the IP geolocation checks and such for you for an exceptionally SMALL fee per transaction (fraction of a cent). It is absolutely worth it. N. Zeeshan Zakaria wrote: Thanks Matt, and everybody else, very useful information. I guess I'll have to sit again and spend time coding delays, small amount payments for new accounts and paypal=signup email match. -- Zeeshan A Zakaria On Mon, Aug 31, 2009 at 12:07 AM, Matt Riddell li...@venturevoip.com mailto:li...@venturevoip.com wrote: On 31/08/09 2:45 PM, Zeeshan Zakaria wrote: Those who are more experienced in this business, please advise how to avoid this type of fraud, and which service to use in place of PayPal, because PayPal doesn't seem the right payment solution for a prepaid VoIP service. Also now that they have all the payments put on hold and asking for a resolution, their resolution center is good only for shipped merchendise, not for online services. How would I prove to them that the buyer who is asking his money back has already utilized my service by making lot of international calls, which I now have to pay for to the carrier. I've used CDR for that and don't automatically accept payments. When we receive a payment we compare: 1. IP Address of user (whois normally gives approximate location) 2. Paypal account holder email (should match sign up email) 3. Countries for emails and ip address should match. 4. Initial payment should be $1-$2 (i.e. noone is going to sign up for a service and in order to test it put down $500 via paypal) If any of the above look suspect I ask the paypal account holder to email me and start looking at email headers to see how sus it looks. If it's a large amount then they have to have already been doing business with us successfully with small amounts - most scammers can't be bothered doing this. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR to Postgres Centos
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 ABBAS SHAKEEL wrote: but when i execute this ./configure --with-postgres=dir where postgresql is installed it gives an error for missing an pg_config file . i searched the PC but it really dont exists. but database server is fine and working OK! what to do in this situation You should have the following packages installed on your Asterisk system postgresql-libs postgresql-devel postgresql If the database is on the same box, include: postgresql-server If you want to hit the database from the dialplan for any reason, include: postgresql-odbc Once you install these, be sure to rerun ./configure Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKm8rOCFu3bIiwtTARAijbAJ4vt0DVZJYUPRhPrNpXm2KEngRmxACgn24T aHtpBzyGhPBmw8a4veqdLhQ= =TI+m -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to stop IVR once system receives DTMF?
Hi, We are trying to implement a complex business logic in Asterisk. Executing Wait_For_Digit command after playing IVR. We want to stop the IVR once we receive the digit. It is not recognizing the Digit until it completes the IVR. How can we stop the IVR once we receive the digit? Thanks BB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to deal with PayPal frauds?
I didn't know about MaxMind, but seems like a great service. Seems like exactly what I needed. Thanks for the reference. -- Zeeshan A Zakaria On Mon, Aug 31, 2009 at 8:59 AM, SIP s...@arcdiv.com wrote: When you start taking credit card payments (assuming you will), be careful with small payment amounts. You'll become a fraud haven. A lot of CC thieves or people who've just bought a CC number will use a small amount charge to check and see if the card is any good. Check out some of the MaxMind stuff for fraud prevention. They will do a lot of the IP geolocation checks and such for you for an exceptionally SMALL fee per transaction (fraction of a cent). It is absolutely worth it. N. Zeeshan Zakaria wrote: Thanks Matt, and everybody else, very useful information. I guess I'll have to sit again and spend time coding delays, small amount payments for new accounts and paypal=signup email match. -- Zeeshan A Zakaria On Mon, Aug 31, 2009 at 12:07 AM, Matt Riddell li...@venturevoip.com mailto:li...@venturevoip.com wrote: On 31/08/09 2:45 PM, Zeeshan Zakaria wrote: Those who are more experienced in this business, please advise how to avoid this type of fraud, and which service to use in place of PayPal, because PayPal doesn't seem the right payment solution for a prepaid VoIP service. Also now that they have all the payments put on hold and asking for a resolution, their resolution center is good only for shipped merchendise, not for online services. How would I prove to them that the buyer who is asking his money back has already utilized my service by making lot of international calls, which I now have to pay for to the carrier. I've used CDR for that and don't automatically accept payments. When we receive a payment we compare: 1. IP Address of user (whois normally gives approximate location) 2. Paypal account holder email (should match sign up email) 3. Countries for emails and ip address should match. 4. Initial payment should be $1-$2 (i.e. noone is going to sign up for a service and in order to test it put down $500 via paypal) If any of the above look suspect I ask the paypal account holder to email me and start looking at email headers to see how sus it looks. If it's a large amount then they have to have already been doing business with us successfully with small amounts - most scammers can't be bothered doing this. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to stop IVR once system receives DTMF?
Are you using Background(SomeSoundFile) ? or PlayBack(SomeSoundFile) ? Normally Background() will stop if the pressed digit(s) match any dialplan entry. Bharath B. Reddy Bynagari wrote: Hi, We are trying to implement a complex business logic in Asterisk. Executing “Wait_For_Digit” command after playing IVR. We want to stop the IVR once we receive the digit. It is not recognizing the Digit until it completes the IVR. How can we stop the IVR once we receive the digit? Thanks BB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to stop IVR once system receives DTMF?
Use read with one digit. If you want a specific digit, test for it like
this
exten = s,1(readacct),Read(digitacc,record/enteracct,1,skip,1,5])
exten = s,n,Gotoif($[${LEN(${digitacc})} 1]?readacct)
exten = s,n,Gotoif($[${digitacc} != 5]?readacct)
This instance loops until 5 is pressed.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bharath B.
Reddy Bynagari
Sent: Monday, August 31, 2009 8:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to stop IVR once system receives DTMF?
Hi,
We are trying to implement a complex business logic in Asterisk. Executing
Wait_For_Digit command after playing IVR. We want to stop the IVR once we
receive the digit. It is not recognizing the Digit until it completes the
IVR. How can we stop the IVR once we receive the digit?
Thanks
BB
___
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Register Now: http://www.astricon.net
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Re: [asterisk-users] How to stop IVR once system receives DTMF?
I think the IVR audio must be playing in Background mode, not Play Mode. Try that. Background means play the sound and move on to the next instruction. Play means to play the sound and after it is over, move to the next instruction. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bharath B. Reddy Bynagari Sent: Monday, August 31, 2009 8:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to stop IVR once system receives DTMF? Hi, We are trying to implement a complex business logic in Asterisk. Executing Wait_For_Digit command after playing IVR. We want to stop the IVR once we receive the digit. It is not recognizing the Digit until it completes the IVR. How can we stop the IVR once we receive the digit? Thanks BB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to stop IVR once system receives DTMF?
This is correct. You can build a pretty large voice menu using Background(file1file2file3file4) stacked up with the knowledge that the user can skip all of it by pressing a key. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Monday, August 31, 2009 8:49 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How to stop IVR once system receives DTMF? I think the IVR audio must be playing in Background mode, not Play Mode. Try that. Background means play the sound and move on to the next instruction. Play means to play the sound and after it is over, move to the next instruction. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bharath B. Reddy Bynagari Sent: Monday, August 31, 2009 8:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to stop IVR once system receives DTMF? Hi, We are trying to implement a complex business logic in Asterisk. Executing Wait_For_Digit command after playing IVR. We want to stop the IVR once we receive the digit. It is not recognizing the Digit until it completes the IVR. How can we stop the IVR once we receive the digit? Thanks BB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Report
How to extract that CDR from asterisk ? On Sat, Aug 29, 2009 at 2:11 AM, David @ULC ucoms2...@gmail.com wrote: From Asterisk, I need a List of Numbers , asterisk dialed out. I am looking for status of each number dialed out. Whether its failed or successful . Any way ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Report
Depends on your setup. It's either a table in a database or /var/log/asterisk/cdr-csv/Master.csv (or both). _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC Sent: Monday, August 31, 2009 9:49 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Report How to extract that CDR from asterisk ? On Sat, Aug 29, 2009 at 2:11 AM, David @ULC ucoms2...@gmail.com wrote: From Asterisk, I need a List of Numbers , asterisk dialed out. I am looking for status of each number dialed out. Whether its failed or successful . Any way ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange problem
Hi folks! I have an asterisk, version 1.4.22.2, with dahdi 2.1.0.4 and OpenR2 2.1.1.0. Sometimes twice a day, sometimes after 3 days, all sip devices looses registry, but asterisk doesn't show nothing strange, no error log, and all calls in E1 trunk continue running, but sending to voicemail. What could be this problem? What should I research do find a solution? Thanks a lot! Carlos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange problem
It sounds like your SIP devices aren't set up to periodically(frequently) re-register themselves. You can resolve this on the device level or have asterisk poll them for re-registration. It could also be some sort of firewall/NAT problem that chops the connections at some interval. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Eduardo Langoni Sent: Monday, August 31, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Strange problem Hi folks! I have an asterisk, version 1.4.22.2, with dahdi 2.1.0.4 and OpenR2 2.1.1.0. Sometimes twice a day, sometimes after 3 days, all sip devices looses registry, but asterisk doesn't show nothing strange, no error log, and all calls in E1 trunk continue running, but sending to voicemail. What could be this problem? What should I research do find a solution? Thanks a lot! Carlos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to stop IVR once system receives DTMF?
On Monday 31 August 2009 08:48:32 am Cary Fitch wrote: I think the IVR audio must be playing in Background mode, not Play Mode. Try that. Background means play the sound and move on to the next instruction. Play means to play the sound and after it is over, move to the next instruction. No, it does not. Background means to play the sound in the background, and wait for DTMF in the foreground. The only method in Asterisk for playing a soundfile and moving onto the next instruction is MOH. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to hide Caller Id
At 11:56 PM 8/30/2009, you wrote: Sorry for mis-typing in phone type . Please be informed that the current phone type our subscribers are using is TP6000 ones . The phone only knows the number to display if you tell it the number, so tell it to display something else. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Versions of Asterisk 1.6
Hello, I see that there's 1.6.0.x and 1.6.1.y versions of Asterisk. Is there a clear table that describes the features and/or differences between them? Are both stable enough? Is T.38 Fax supported on both? If yes, which spandsp is supported? I saw on voip-info.org that version 6 is not supported, but this information might be outdated. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange problem
I set mine at 300 (5 minutes). You might want a higher value if you have lots of phones, but since I only have 8 at my shop, this causes no noticeable downside. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Eduardo Langoni Sent: Monday, August 31, 2009 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Strange problem Danny, Thank for your reply. I'm sure that is not firewall/nat because all sip devices are using a private class of IP and asterisk has a network adapter with an IP from the same class/network. How muchi is a good value for re-register? Thanks a lot 2009/8/31 Danny Nicholas da...@debsinc.com: It sounds like your SIP devices aren't set up to periodically(frequently) re-register themselves. You can resolve this on the device level or have asterisk poll them for re-registration. It could also be some sort of firewall/NAT problem that chops the connections at some interval. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Eduardo Langoni Sent: Monday, August 31, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Strange problem Hi folks! I have an asterisk, version 1.4.22.2, with dahdi 2.1.0.4 and OpenR2 2.1.1.0. Sometimes twice a day, sometimes after 3 days, all sip devices looses registry, but asterisk doesn't show nothing strange, no error log, and all calls in E1 trunk continue running, but sending to voicemail. What could be this problem? What should I research do find a solution? Thanks a lot! Carlos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Versions of Asterisk 1.6
On Mon, Aug 31, 2009 at 12:37 PM, Cyprus VoIPvoi...@gmail.com wrote: Are both stable enough? Stable enough for whom? You are the only one who can answer that question. Is T.38 Fax supported on both? Yes, with caveats. There continue to be a number of T.38 patches going into various releases. I would call T.38 a moving target. Your particular arrangement may work great with a version from more than a year ago, or you may have problems. Only you can answer whether T.38 will work with your configuration. I can tell you that I've been having problems with various version of Cisco IOS and T.38 on asterisk. I had a stable configuration fax-wise, but I had to upgrade the IOS because of a Cisco bug, and my T.38 has never been the same since. It's hard to blame asterisk for that problem. In fact, if you read through the T.38 bugs in Cisco IOS release notes it makes asterisk T.38 look solid by comparison. If downgrading didn't make my router freeze I'd downgrade the IOS. If yes, which spandsp is supported? You are best off reading app_fax.c for details to that answer. I'm using spandsp-0.0.6pre7 with 1.6.0.13 I saw on voip-info.org that version 6 is not supported, but this information might be outdated. Do you want to link to where it says that? Version 0.0.6 patch-level support has been around since at least early 2009 in SVN, but whether it was in a particular release branch, well. You know. If voip-info categorically says spandsp-0.0-6 is NOT supported under any release, that is incorrect, but it is correct that previous versions required 0.0.5 series. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Report
Ok Got it. Any 3rd party Interface which can get me all these result in a front end ? On Mon, Aug 31, 2009 at 8:19 PM, David @ULC ucoms2...@gmail.com wrote: How to extract that CDR from asterisk ? On Sat, Aug 29, 2009 at 2:11 AM, David @ULC ucoms2...@gmail.com wrote: From Asterisk, I need a List of Numbers , asterisk dialed out. I am looking for status of each number dialed out. Whether its failed or successful . Any way ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange problem
Danny, Thank for your reply. I'm sure that is not firewall/nat because all sip devices are using a private class of IP and asterisk has a network adapter with an IP from the same class/network. How muchi is a good value for re-register? Thanks a lot 2009/8/31 Danny Nicholas da...@debsinc.com: It sounds like your SIP devices aren't set up to periodically(frequently) re-register themselves. You can resolve this on the device level or have asterisk poll them for re-registration. It could also be some sort of firewall/NAT problem that chops the connections at some interval. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Eduardo Langoni Sent: Monday, August 31, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Strange problem Hi folks! I have an asterisk, version 1.4.22.2, with dahdi 2.1.0.4 and OpenR2 2.1.1.0. Sometimes twice a day, sometimes after 3 days, all sip devices looses registry, but asterisk doesn't show nothing strange, no error log, and all calls in E1 trunk continue running, but sending to voicemail. What could be this problem? What should I research do find a solution? Thanks a lot! Carlos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to deal with PayPal frauds?
On Aug 30, 2009, at 7:45 PM, Zeeshan Zakaria wrote: I charge my customers through PayPal, but recently faced a fraud which previously had only heard about. Somebody registered a few accounts, paid online with paypal (as my service is only prepaid) and started making expensive long distance calls. In fact the IP registering the accounts was from Florida, and IPs making calls were from Africa. After about 20 minutes the first payment was reversed. Then a few times more payments were made, and every payment was reversed almost as soon as it was made. Payments were made from different PayPal accounts. And then I started getting emails from PayPal resolution center that some payments were made by users who didn't authorize them. Obviously either somebody was using stolen paypal accounts, or somebody knows that he can pay and reverse the payment and in the meanwhile make enough long distance calls. What is really fishy that reversals were made almost as soon as the payments were made, one after another. Those who are more experienced in this business, please advise how to avoid this type of fraud, and which service to use in place of PayPal, because PayPal doesn't seem the right payment solution for a prepaid VoIP service. Also now that they have all the payments put on hold and asking for a resolution, their resolution center is good only for shipped merchendise, not for online services. How would I prove to them that the buyer who is asking his money back has already utilized my service by making lot of international calls, which I now have to pay for to the carrier. Despite what PayPal and any of the other processors tell you in their marketing material, there is very little protection for online merchants. The only way to be mostly sure, is to accept cash or wire transfers. Having said that, you might want to look into MasterCard's SecureCard program (http://www.mastercard.com/us/merchant/solutions/mastercard_securecode.html ). I don't remember the exact details when a physical product is not involved, but the general idea is that if you enroll in the securecard program, you will be covered from cardholder unauthorized chargebacks, Visa has something similar. AmEx has a number you can call and they will verify transactions over $250 with the card holder. You might also want to consider shipping a welcome packet to the customer, that may cover you under PayPal's physical goods terms. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk MWI issue
Hi, I am using Asterisk personally at home. My SIP client (SPA 3000) supports MWI with SIP NOTIFY messages. With a previous version of Asterisk I had no problems with MWI. But now I am using the following version which comes with Trixbox 2.8.0.1, and I have problems with MWI. Asterisk 1.6.0.9-samy-r27 Problem description: When a voicemail is left on the extension, a SIP NOTIFY message is sent to my SIP client and the MWI is received ok. This is good. But when I delete all Voicemail through AMPortal, SIP NOTIFY message notifying that there is no voicemail left is not sent to the client. Normally , I expect to receie a SIP NOTIFY message as soon as my inbox is empty. However, this does not happen when I delete voicemail through GUI. If I delete voicemail through phone, I receive the SIP NOTIFY message. How can I fix this problem ? Is this a misconfiguration or a bug ? If it is a bug, is there any fix for it ? Thanks, ilker___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to deal with PayPal frauds?
Good idea Eric regarding welcome package. -- Zeeshan A Zakaria On Mon, Aug 31, 2009 at 1:07 PM, Eric Chamberlain e...@rf.com wrote: On Aug 30, 2009, at 7:45 PM, Zeeshan Zakaria wrote: I charge my customers through PayPal, but recently faced a fraud which previously had only heard about. Somebody registered a few accounts, paid online with paypal (as my service is only prepaid) and started making expensive long distance calls. In fact the IP registering the accounts was from Florida, and IPs making calls were from Africa. After about 20 minutes the first payment was reversed. Then a few times more payments were made, and every payment was reversed almost as soon as it was made. Payments were made from different PayPal accounts. And then I started getting emails from PayPal resolution center that some payments were made by users who didn't authorize them. Obviously either somebody was using stolen paypal accounts, or somebody knows that he can pay and reverse the payment and in the meanwhile make enough long distance calls. What is really fishy that reversals were made almost as soon as the payments were made, one after another. Those who are more experienced in this business, please advise how to avoid this type of fraud, and which service to use in place of PayPal, because PayPal doesn't seem the right payment solution for a prepaid VoIP service. Also now that they have all the payments put on hold and asking for a resolution, their resolution center is good only for shipped merchendise, not for online services. How would I prove to them that the buyer who is asking his money back has already utilized my service by making lot of international calls, which I now have to pay for to the carrier. Despite what PayPal and any of the other processors tell you in their marketing material, there is very little protection for online merchants. The only way to be mostly sure, is to accept cash or wire transfers. Having said that, you might want to look into MasterCard's SecureCard program ( http://www.mastercard.com/us/merchant/solutions/mastercard_securecode.html ). I don't remember the exact details when a physical product is not involved, but the general idea is that if you enroll in the securecard program, you will be covered from cardholder unauthorized chargebacks, Visa has something similar. AmEx has a number you can call and they will verify transactions over $250 with the card holder. You might also want to consider shipping a welcome packet to the customer, that may cover you under PayPal's physical goods terms. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Versions of Asterisk 1.6
Cyprus VoIP wrote: Hello, I see that there's 1.6.0.x and 1.6.1.y versions of Asterisk. Is there a clear table that describes the features and/or differences between them? Are both stable enough? Is T.38 Fax supported on both? If yes, which spandsp is supported? I saw on voip-info.org that version 6 is not supported, but this information might be outdated. For information about the difference between 1.6.x releases, see http://www.asterisk.org/node/48602 (About the new Asterisk versioning method). For specific changes between the versions, see the ChangeLog file. For changes between the two versions, see the CHANGES file in the Asterisk source. Also read the UPGRADE.txt file if moving between major versions (1.4 - 1.6.0 - 1.6.1). Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gizmo Dial Out No CALLED PARTY AUDIO??
Rob a écrit : Yes ... as a matter of fact here is the sip.conf ... obviously private info removed [...] Did you try to call Gizmo numbers to see if you have success with them? ** Hear your Gizmo5 number repeated back to you. *0 Test your router's SIP compatibility. 411 The voice-activated Tellme information service. 1-747-474-ECHO 1-747-474-3246 Echo Test - Repeats back whatever you say. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] List Access
To view the post and reply , I always to use below link.. http://lists.digium.com/pipermail/asterisk-users/2009-August/thread.htmlhttp://lists.digium.com/pipermail/asterisk-users/2009-February/thread.html Any better way to access the forum ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Report
David @ULC wrote: Ok Got it. Any 3rd party Interface which can get me all these result in a front end ? http://www.areski.net/asterisk-stat-v2/about.php 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Upgrading Asterisk in Trixbox installation
Hi, My Trixbox 2.8.0.1 installation includes the following Asterik version: 1.6.0.9-samy-r27 I am having some problems with it and I think they might be solved if I use the latest Asterisk version. Is it a good idea to update Asterisk in Trixbox externally ? Is it safe ? If so, which version should I prefer ? 1.6.1.5 or 1.6.0.14 ? Thanks, ilker___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrading Asterisk in Trixbox installation
On Mon, 31 Aug 2009, ilker Aktuna wrote: Hi, My Trixbox 2.8.0.1 installation includes the following Asterik version: 1.6.0.9-samy-r27 I am having some problems with it and I think they might be solved if I use the latest Asterisk version. Is it a good idea to update Asterisk in Trixbox externally ? I've done it in the 1.4 branch. Is it safe ? Should be, as long as you stay within the same branch. That being the case, I would stick with 1.6.0.14 if I were you. Make sure you don't make samples :) j If so, which version should I prefer ? 1.6.1.5 or 1.6.0.14 ? Thanks, ilker ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrading Asterisk in Trixbox installation
Thank you. That was quick and helpful :) Then I'll just make and make install What should I backup, in case of rollback requirement ? Thanks. - Original Message - From: Jeff LaCoursiere j...@jeff.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 31, 2009 11:15 PM Subject: Re: [asterisk-users] Upgrading Asterisk in Trixbox installation On Mon, 31 Aug 2009, ilker Aktuna wrote: Hi, My Trixbox 2.8.0.1 installation includes the following Asterik version: 1.6.0.9-samy-r27 I am having some problems with it and I think they might be solved if I use the latest Asterisk version. Is it a good idea to update Asterisk in Trixbox externally ? I've done it in the 1.4 branch. Is it safe ? Should be, as long as you stay within the same branch. That being the case, I would stick with 1.6.0.14 if I were you. Make sure you don't make samples :) j If so, which version should I prefer ? 1.6.1.5 or 1.6.0.14 ? Thanks, ilker ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrading Asterisk in Trixbox installation
On Mon, 31 Aug 2009, ilker Aktuna wrote: Thank you. That was quick and helpful :) Then I'll just make and make install What should I backup, in case of rollback requirement ? That's a bit tougher. At the least /usr/lib/asterisk/modules, /etc/asterisk, and /usr/sbin/asterisk... someone else may need to chime in here... I've always been a fan of trixbox, and I have done a lot of installations, but when it comes down to it all I really want it for is for a quick installations of asterisk and FreePBX. I don't think I actually use any of the trixbox-only features. I've also been enamored with Ubuntu of late, and have dumped CentOS. YMMV, but you might consider starting over with a clean build of the linux of your choice, and doing asterisk + addons + FreePBX from source. j Thanks. - Original Message - From: Jeff LaCoursiere j...@jeff.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 31, 2009 11:15 PM Subject: Re: [asterisk-users] Upgrading Asterisk in Trixbox installation On Mon, 31 Aug 2009, ilker Aktuna wrote: Hi, My Trixbox 2.8.0.1 installation includes the following Asterik version: 1.6.0.9-samy-r27 I am having some problems with it and I think they might be solved if I use the latest Asterisk version. Is it a good idea to update Asterisk in Trixbox externally ? I've done it in the 1.4 branch. Is it safe ? Should be, as long as you stay within the same branch. That being the case, I would stick with 1.6.0.14 if I were you. Make sure you don't make samples :) j If so, which version should I prefer ? 1.6.1.5 or 1.6.0.14 ? Thanks, ilker ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrading Asterisk in Trixbox installation
One thing I kind of like that Trixbox does is their endpoint manager. That's about the only feature I haven't been able to replace. Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Monday, August 31, 2009 4:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Upgrading Asterisk in Trixbox installation On Mon, 31 Aug 2009, ilker Aktuna wrote: Thank you. That was quick and helpful :) Then I'll just make and make install What should I backup, in case of rollback requirement ? That's a bit tougher. At the least /usr/lib/asterisk/modules, /etc/asterisk, and /usr/sbin/asterisk... someone else may need to chime in here... I've always been a fan of trixbox, and I have done a lot of installations, but when it comes down to it all I really want it for is for a quick installations of asterisk and FreePBX. I don't think I actually use any of the trixbox-only features. I've also been enamored with Ubuntu of late, and have dumped CentOS. YMMV, but you might consider starting over with a clean build of the linux of your choice, and doing asterisk + addons + FreePBX from source. j Thanks. - Original Message - From: Jeff LaCoursiere j...@jeff.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 31, 2009 11:15 PM Subject: Re: [asterisk-users] Upgrading Asterisk in Trixbox installation On Mon, 31 Aug 2009, ilker Aktuna wrote: Hi, My Trixbox 2.8.0.1 installation includes the following Asterik version: 1.6.0.9-samy-r27 I am having some problems with it and I think they might be solved if I use the latest Asterisk version. Is it a good idea to update Asterisk in Trixbox externally ? I've done it in the 1.4 branch. Is it safe ? Should be, as long as you stay within the same branch. That being the case, I would stick with 1.6.0.14 if I were you. Make sure you don't make samples :) j If so, which version should I prefer ? 1.6.1.5 or 1.6.0.14 ? Thanks, ilker ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrading Asterisk in Trixbox installation
I am a big fan of ubuntu LTS and freepbx and recently I saw mention of a custom module to add auto configuring endpoints for linksys (but i cna't find it again right now) Trixbox had too much stuff whereas the source install of just what you want is nice and clean Cheers Duncan Jeff LaCoursiere wrote: On Mon, 31 Aug 2009, ilker Aktuna wrote: Thank you. That was quick and helpful :) Then I'll just make and make install What should I backup, in case of rollback requirement ? That's a bit tougher. At the least /usr/lib/asterisk/modules, /etc/asterisk, and /usr/sbin/asterisk... someone else may need to chime in here... I've always been a fan of trixbox, and I have done a lot of installations, but when it comes down to it all I really want it for is for a quick installations of asterisk and FreePBX. I don't think I actually use any of the trixbox-only features. I've also been enamored with Ubuntu of late, and have dumped CentOS. YMMV, but you might consider starting over with a clean build of the linux of your choice, and doing asterisk + addons + FreePBX from source. j Thanks. - Original Message - From: Jeff LaCoursiere j...@jeff.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 31, 2009 11:15 PM Subject: Re: [asterisk-users] Upgrading Asterisk in Trixbox installation On Mon, 31 Aug 2009, ilker Aktuna wrote: Hi, My Trixbox 2.8.0.1 installation includes the following Asterik version: 1.6.0.9-samy-r27 I am having some problems with it and I think they might be solved if I use the latest Asterisk version. Is it a good idea to update Asterisk in Trixbox externally ? I've done it in the 1.4 branch. Is it safe ? Should be, as long as you stay within the same branch. That being the case, I would stick with 1.6.0.14 if I were you. Make sure you don't make samples :) j If so, which version should I prefer ? 1.6.1.5 or 1.6.0.14 ? Thanks, ilker ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue issue
I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.6.1 + TDM840 FSK MWI problem
Hi, Using 1.4.26.1 DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? Thanks, --Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue issue
Paul Hales escribió: I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi, Maybe maxlen = 1? Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue issue
Miguel Molina wrote: Paul Hales escribió: I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH Hi, Maybe maxlen = 1? Cheers, Hmmm - almost. Maxlen limits the amounts of calls waiting for the queue, not the amount of callers talking to queue members. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium PRI cards for data usage?
Greetings- I'm wondering if the Digium PRI cards can be used for data (Cisco HDLC, PPP, etc) or if they're for voice circuits only. I haven't been able to find any information on this. All documentation direct from Digium seems to indicate their hardware is for voice applications only. Sangoma's cards work in either voice or data mode but of course this is configured in their Wanpipe software. Thanks for any pointers. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Selective canreinvite in multi-tenant environment
On Thu, 2009-08-27 at 14:23 -0400, John A. Sullivan III wrote: Hello, all. In our multi-tenant environment, we would like to be able to use the reinvite media redirection within Asterisk for calls within a tenant but not between tenants. We would like inter-tenant calls to be fully proxied by the Asterisk server. I think the answer is, we can't, but I thought I'd ask anyway. I'd dearly like to remove the substantial traffic associated with intra-tenant traffic from the Asterisk server and reduce the intra-tenant latency by doing so. However, I am very, very hesitant to allow our VPN connections to tenants to function as a router between tenants allowing one tenant to directly access phones on another tenant (that's not as wild as it sounds because of our use of the ISCS project - iscs.sourceforge.net). Since the tenants are all connecting via VPN, we are using RFC1918 addresses and no NAT is involved thus the canreinvite=nonat option does not help us. If we set canreinvite=nonat, that will allow for intra-tenant direct media but, if one tenant tries to call another via SIP, it will redirect the media at the Asterisk level but the packets will be dropped at the firewall / router level (or sooner as there may be no route to the destination) and the call will connect but with no sound. Any guidance would be greatly appreciated. Thanks - John As mentioned in another post, we were able to solve this by setting a w dial option to all inbound SIP calls from the Internet. Thus, all internal calls could reinvite but external calls could not. However, just when we thought this was working splendidly well, we turned up another roadblock - transfers. We find that once we transfer a call using our Snom phones, the call between the new call partners does not seem bound by the w constraint, Asterisk tries to reinvite the call, and the audio breaks because the firewall cannot associate the new RTP stream with a SIP session. How have others gotten around the problem of transfers causing reinvites on calls which should not allow reinvites? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium PRI cards for data usage?
On Monday 31 August 2009 21:59:28 Tim Nelson wrote: Greetings- I'm wondering if the Digium PRI cards can be used for data (Cisco HDLC, PPP, etc) or if they're for voice circuits only. I haven't been able to find any information on this. All documentation direct from Digium seems to indicate their hardware is for voice applications only. Sangoma's cards work in either voice or data mode but of course this is configured in their Wanpipe software. Thanks for any pointers. You can. The keyword is nethdlc in /etc/dahdi/system.conf, although to enable it, you need to uncomment CONFIG_DAHDI_NET in include/dahdi/dahdi_config.h and recompile the dahdi drivers. Once the active spans are configured with nethdlc, use the sethdlc command line utility to set up the bonded channels into the various network interfaces (hdlc0 through hdlcN). Depending upon your configuration, you may or may not also need to then configure the corresponding pvcN devices. Here is an article on the old Zaptel interface. While the name of the driver may have changed, the procedures remain the same: http://www.softwink.com/papers/Installation_Securing_VoIP_With_Linux/ By the way, the method for determining which channels are bonded are as simple as the number of channels you configure together (on a single line) in /etc/dahdi/system.conf. For example, you can do as little as nethdlc=1 (for a single 64k channel) up to nethdlc=1-192 (for 8 T1s bonded into a single data device). Each nethdlc line in the config becomes a separate hdlcN device. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Web Meetme module not loading
Matt Riddell wrote: On 31/08/09 2:33 PM, Glen wrote: I have asterisk 1.4.21 and web meetme (latest release 3.1) I have also installed the latest versions of mysql and php. I followed the readme file that came with the web meetme app and everything seemed to go fine up until I realised the module wasnt being loaded. When I stop asterisk and try to start it, it errors out and does not load and I get the following message: Parsing '/etc/asterisk/cbmysql.conf': Found asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_cbmysql.so: undefined symbol: mysql_init Likely you don't have mysql-devel libraries installed - though I wonder how it would have compiled. mysql_init is a function provided by the libmysqlclient library - if you didn't compile app_cbmysql.so yourself, you could type ldd app_cbmysql.so to see what it links to then check your lib directory to see if you have the same - you might have 64 bit when it was compiled for 32 bit or something \ Hi Matt, I have the following mysql packages installed MySQL-client-community-5.1.37 MySQL-devel-community-5.1.37 MySQL-server-community-5.1.37 MySQL-shared-community-5.1.37 Also I get no errors when compiling app_cbmysql.so (I do compile this from source) Any idea's? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Web Meetme module not loading
I meant /usr/lib not /var/lib sorry -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Web Meetme module not loading
On 1/09/09 4:31 PM, Glen wrote: Matt Riddell wrote: On 31/08/09 2:33 PM, Glen wrote: I have asterisk 1.4.21 and web meetme (latest release 3.1) I have also installed the latest versions of mysql and php. I followed the readme file that came with the web meetme app and everything seemed to go fine up until I realised the module wasnt being loaded. When I stop asterisk and try to start it, it errors out and does not load and I get the following message: Parsing '/etc/asterisk/cbmysql.conf': Found asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_cbmysql.so: undefined symbol: mysql_init Likely you don't have mysql-devel libraries installed - though I wonder how it would have compiled. mysql_init is a function provided by the libmysqlclient library - if you didn't compile app_cbmysql.so yourself, you could type ldd app_cbmysql.so to see what it links to then check your lib directory to see if you have the same - you might have 64 bit when it was compiled for 32 bit or something \ Hi Matt, I have the following mysql packages installed MySQL-client-community-5.1.37 MySQL-devel-community-5.1.37 MySQL-server-community-5.1.37 MySQL-shared-community-5.1.37 Also I get no errors when compiling app_cbmysql.so (I do compile this from source) What do you get if you type: ldd /var/lib/asterisk/modules/app_cbmysql.so -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Problem with Call Parking
Dear All Can you please do me favor and let me know what is the problem with my Asterisk call parking as it is not functioning correctly on my Asterisk ? Please find attached my features.conf . According to my configuration , the subscriber needs to press hash (pound) key and dial 700 to initiate the transfer . We tried but it didn't get through on our Asterisk . Can you please let me know what extra config needs to be done for putting it into operation ? Regards H.Motamedi features.conf Description: Binary data ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Problem with Call Parking
Just a quick guess - is it because you did not program your Polycom digit plan properly in sip.cfg? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi Sent: Tuesday, 1 September 2009 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Inquiry:Problem with Call Parking Dear All Can you please do me favor and let me know what is the problem with my Asterisk call parking as it is not functioning correctly on my Asterisk ? Please find attached my features.conf . According to my configuration , the subscriber needs to press hash (pound) key and dial 700 to initiate the transfer . We tried but it didn't get through on our Asterisk . Can you please let me know what extra config needs to be done for putting it into operation ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Web Meetme module not loading
Matt Riddell wrote: On 1/09/09 4:31 PM, Glen wrote: Matt Riddell wrote: On 31/08/09 2:33 PM, Glen wrote: I have asterisk 1.4.21 and web meetme (latest release 3.1) I have also installed the latest versions of mysql and php. I followed the readme file that came with the web meetme app and everything seemed to go fine up until I realised the module wasnt being loaded. When I stop asterisk and try to start it, it errors out and does not load and I get the following message: Parsing '/etc/asterisk/cbmysql.conf': Found asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_cbmysql.so: undefined symbol: mysql_init Likely you don't have mysql-devel libraries installed - though I wonder how it would have compiled. mysql_init is a function provided by the libmysqlclient library - if you didn't compile app_cbmysql.so yourself, you could type ldd app_cbmysql.so to see what it links to then check your lib directory to see if you have the same - you might have 64 bit when it was compiled for 32 bit or something \ Hi Matt, I have the following mysql packages installed MySQL-client-community-5.1.37 MySQL-devel-community-5.1.37 MySQL-server-community-5.1.37 MySQL-shared-community-5.1.37 Also I get no errors when compiling app_cbmysql.so (I do compile this from source) What do you get if you type: ldd /usr/lib/asterisk/modules/app_cbmysql.so This is the output linux-gate.so.1 = (0xe000) libpthread.so.0 = /lib/libpthread.so.0 (0xb7f6) libc.so.6 = /lib/libc.so.6 (0xb7e2d000) /lib/ld-linux.so.2 (0x8000) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Web Meetme module not loading
On 1/09/09 4:54 PM, Glen wrote: Parsing '/etc/asterisk/cbmysql.conf': Found asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_cbmysql.so: undefined symbol: mysql_init ldd /usr/lib/asterisk/modules/app_cbmysql.so This is the output linux-gate.so.1 = (0xe000) libpthread.so.0 = /lib/libpthread.so.0 (0xb7f6) libc.so.6 = /lib/libc.so.6 (0xb7e2d000) /lib/ld-linux.so.2 (0x8000) Er weird - so it's not even requesting a link to the mysql library. Looks like the linking somehow went wrong - if it can't find mysql_init and it doesn't look for it then surely something went wrong at the linking stage. Can you post me the output of the compilation? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Problem with Call Parking
Thank you for your reply . Please find attached my Asterisk sip.conf . Can you please let me know what modifications are needed ? Regards H.Motamedi On Tue, Sep 1, 2009 at 5:55 AM, Lee, John (Sydney) john@compuware.comwrote: Just a quick guess - is it because you did not program your Polycom digit plan properly in sip.cfg? From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi Sent: Tuesday, 1 September 2009 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Inquiry:Problem with Call Parking Dear All Can you please do me favor and let me know what is the problem with my Asterisk call parking as it is not functioning correctly on my Asterisk ? Please find attached my features.conf . According to my configuration , the subscriber needs to press hash (pound) key and dial 700 to initiate the transfer . We tried but it didn't get through on our Asterisk . Can you please let me know what extra config needs to be done for putting it into operation ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users sip.conf Description: Binary data ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Problem with Call Parking
On 1/09/09 5:08 PM, hadi motamedi wrote: Thank you for your reply . Please find attached my Asterisk sip.conf . Can you please let me know what modifications are needed ? Regards H.Motamedi He actually asked for the sip.cfg (i.e. the config for the polycom rather than for Asterisk) -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Problem with Call Parking
Polycom sip.cfg is not the same as the Asterisk sip.conf file hadi motamedi wrote: Thank you for your reply . Please find attached my Asterisk sip.conf . Can you please let me know what modifications are needed ? Regards H.Motamedi On Tue, Sep 1, 2009 at 5:55 AM, Lee, John (Sydney) john@compuware.com mailto:john@compuware.com wrote: Just a quick guess - is it because you did not program your Polycom digit plan properly in sip.cfg? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Web Meetme module not loading
On Tue, Sep 1, 2009 at 3:06 PM, Matt Riddell li...@venturevoip.com wrote: On 1/09/09 4:54 PM, Glen wrote: Parsing '/etc/asterisk/cbmysql.conf': Found asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_cbmysql.so: undefined symbol: mysql_init ldd /usr/lib/asterisk/modules/app_cbmysql.so This is the output linux-gate.so.1 = (0xe000) libpthread.so.0 = /lib/libpthread.so.0 (0xb7f6) libc.so.6 = /lib/libc.so.6 (0xb7e2d000) /lib/ld-linux.so.2 (0x8000) Er weird - so it's not even requesting a link to the mysql library. Looks like the linking somehow went wrong - if it can't find mysql_init and it doesn't look for it then surely something went wrong at the linking stage. Can you post me the output of the compilation? -- Cheers, Matt Riddell Director When compiling the module I simply recompiled asterisk (I was told this is the best way), below is the output of that. remote:/usr/src/asterisk-1.4.20.1 # make make install Generating input for menuselect ... menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. [CC] app_cbmysql.c - app_cbmysql.o app_cbmysql.c:37:1: warning: AST_MODULE redefined command-line: warning: this is the location of the previous definition app_cbmysql.c: In function âcheckMaxâ: app_cbmysql.c:116: warning: implicit declaration of function âast_say_numberâ app_cbmysql.c: In function âroomQueryâ: app_cbmysql.c:181: warning: unused variable âeatimeâ app_cbmysql.c:337: warning: control reaches end of non-void function [LD] app_cbmysql.o - app_cbmysql.so make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. +- Asterisk Build Complete -+ + Asterisk has successfully been built, and + + can be installed by running: + + + + make install+ +---+ menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. if [ x`/usr/bin/id -un` = xroot ]; then CFLAGS= -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -include /usr/src/asterisk-1.4.20.1/include/asterisk/autoconfig.h -march=i686 sh build_tools/mkpkgconfig /usr/lib/pkgconfig; fi mkdir -p /var/lib/asterisk/static-http for x in static-http/*; do \ /usr/bin/install -c -m 644 $x /var/lib/asterisk/static-http ; \ done mkdir -p /var/lib/asterisk/images for x in images/*.jpg; do \ /usr/bin/install -c -m 644 $x /var/lib/asterisk/images ; \ done mkdir -p /var/lib/asterisk/agi-bin make -C sounds install make[1]: Entering directory `/usr/src/asterisk-1.4.20.1/sounds' make[1]: Leaving directory `/usr/src/asterisk-1.4.20.1/sounds' mkdir -p /usr/lib/asterisk/modules mkdir -p /usr/sbin mkdir -p /etc/asterisk mkdir -p /usr/bin mkdir -p /var/run mkdir -p /var/spool/asterisk/voicemail mkdir -p /var/spool/asterisk/dictate mkdir -p /var/spool/asterisk/system mkdir -p /var/spool/asterisk/tmp mkdir -p /var/spool/asterisk/meetme mkdir -p /var/spool/asterisk/monitor make[1]: Entering directory `/usr/src/asterisk-1.4.20.1/utils' for x in astman stereorize streamplayer aelparse muted; do \ if [ $x != none ]; then \ /usr/bin/install -c -m 755 $x /usr/sbin/$x; \ fi; \ done make[1]: Leaving directory `/usr/src/asterisk-1.4.20.1/utils' make[1]: Entering directory `/usr/src/asterisk-1.4.20.1/agi' mkdir -p /var/lib/asterisk/agi-bin for x in agi-test.agi eagi-test eagi-sphinx-test jukebox.agi; do /usr/bin/install -c -m 755 $x /var/lib/asterisk/agi-bin ; done make[1]: Leaving directory `/usr/src/asterisk-1.4.20.1/agi' make[1]: Entering directory `/usr/src/asterisk-1.4.20.1/res' for x in res_adsi.so res_agi.so res_clioriginate.so res_convert.so res_features.so res_indications.so res_monitor.so res_musiconhold.so res_smdi.so res_speech.so; do /usr/bin/install -c -m 755 $x /usr/lib/asterisk/modules ; done make[1]: Leaving directory `/usr/src/asterisk-1.4.20.1/res' make[1]: Entering directory `/usr/src/asterisk-1.4.20.1/channels' for x in chan_agent.so chan_iax2.so chan_local.so chan_mgcp.so chan_oss.so chan_phone.so
Re: [asterisk-users] Inquiry:Problem with Call Parking
Please find attached my Asterisk sip.conf . Can you please let me know what modifications are needed ? Yes, I am referring to the Polycom sip.cfg and not the sip.cfg in Asterisk. Somethere down in sip.cfg, there is a line that looks like this: digitmap dialplan.digitmap=#700| ... Basically, Polycom will scan your input to see when it will pass all the keystrokes to Asterisk. In above, if it detects that you have entered #700, it will automatically send it to Asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Web Meetme module not loading
On 1/09/09 5:19 PM, Glen Ganderton wrote: app_cbmysql.c:37:1: warning: AST_MODULE redefined command-line: warning: this is the location of the previous definition app_cbmysql.c: In function âcheckMaxâ: app_cbmysql.c:116: warning: implicit declaration of function âast_say_numberâ app_cbmysql.c: In function âroomQueryâ: app_cbmysql.c:181: warning: unused variable âeatimeâ app_cbmysql.c:337: warning: control reaches end of non-void function I'm not sure how Asterisk is supposed to know that this requires a link to MySQL without being told. Are you using the latest version of the app_cbmysql? It looks like it needs to be updated for the latest version. Alternatively it may say somewhere on their website which version of Asterisk this works with? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Problem with Call Parking
But they do taste similar. PaulH Darrick Hartman wrote: Polycom sip.cfg is not the same as the Asterisk sip.conf file hadi motamedi wrote: Thank you for your reply . Please find attached my Asterisk sip.conf . Can you please let me know what modifications are needed ? Regards H.Motamedi On Tue, Sep 1, 2009 at 5:55 AM, Lee, John (Sydney) john@compuware.com mailto:john@compuware.com wrote: Just a quick guess - is it because you did not program your Polycom digit plan properly in sip.cfg? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Web Meetme module not loading
Hmm, it looks like it has a makefile in the cb_mysql directory which is supposed to do the linking. Have you tried running make from there? It also has a copyright of Mark Spencer, but I know 100% he didn't write it. The person you're probably looking for is Dan Austin, but I can't track him down. Yeah, the lines in that makefile which do it are: app_cbmysql.o: app_cbmysql.c $(CC) -pipe -I/usr/include/mysql -L/usr/lib/mysql $(CFLAGS) -c -o app_cbmysql.o app_cbmysql.c app_cbmysql.so: app_cbmysql.o $(CC) -shared -Xlinker -x -o $@ $ -I/usr/include/mysql -L/usr/lib/mysql -lmysqlclient -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Web Meetme module not loading
I've sent you Dan Austin's email address off list just in case he is able to help out :D -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Web Meetme module not loading
In the latest readme for WebMeetMe (3.1.0) it states: * Compile and install CBMySQL App_cbmysql is now included in the web-meetme package, located in ./cbmysql. To install just run make; make install Copy the sample cbmysql.conf to /etc/asterisk and create a dialplan similar to the one in cb-extensions.conf.sample Modify the settings to suit your system. The location of the mysql.sock file is likely not correct, check /etc/my.conf for the correct location. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Web Meetme module not loading
Matt Riddell wrote: In the latest readme for WebMeetMe (3.1.0) it states: * Compile and install CBMySQL App_cbmysql is now included in the web-meetme package, located in ./cbmysql. To install just run make; make install Copy the sample cbmysql.conf to /etc/asterisk and create a dialplan similar to the one in cb-extensions.conf.sample Modify the settings to suit your system. The location of the mysql.sock file is likely not correct, check /etc/my.conf for the correct location. That fixed it Matt, just compiling in the wrong directory. Thanks for all your help. -Glen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
