[asterisk-users] chan-mobile bug: bluetooth connection is disconnected immediately when the call hangup
Hi all, I have a problem with chan_mobile: I installed chan_mobile, the phone (6230i) can connect to Asterisk through a bluetooth dongle (Cambridge silicon radio), Asterisk can receive calls from the mobile phone but when the call is hangup, the bluetooth connection of the mobile phone is disconnected imtermmediately ?. What is the problem ?. Thans in advance Giang -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ss7_channel or ss7lib
Hi all, Are there anyone use ss7_lib or ss7_channel in production ?. What about its quality and reliablity ?. Can an Asterisk servce with ss7_lib or ss7_channel can processs 480 conccurent call (8 E1 line) ? Many thanks, Giang -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN SS7
Are these solutions reliable and stable ?. Have you used these solutions in production ? What about its quality ? From: Tzafrir Cohen tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Sent: Tue, October 26, 2010 3:12:21 AM Subject: Re: [asterisk-users] ISDN SS7 On Sun, Oct 24, 2010 at 11:33:28AM -0500, Cary Fitch wrote: SS7 is an inter-telco system using a separate network for all signaling. You must have an SS7 network connection before anything will work. Then the T1 Spans run 24 64k audio paths. The SS7 net exchanges the call data and connection info between the switches. Asterisk doesn't support SS7 natively although I believe there are one or more add-on vendors. The vendors of addons such as http://svn.asterisk.org/svn/libss7 and http://svn.asterisk.org/svn/asterisk/trunk/channels/sig_ss7.c ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN SS7
I'm planning to use SGM with Asterisk, it is a commercial product. What is the different between SGM and libs77 and chan_ss7 ? Should I use SGM ? From: Tzafrir Cohen tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Sent: Tue, October 26, 2010 3:12:21 AM Subject: Re: [asterisk-users] ISDN SS7 On Sun, Oct 24, 2010 at 11:33:28AM -0500, Cary Fitch wrote: SS7 is an inter-telco system using a separate network for all signaling. You must have an SS7 network connection before anything will work. Then the T1 Spans run 24 64k audio paths. The SS7 net exchanges the call data and connection info between the switches. Asterisk doesn't support SS7 natively although I believe there are one or more add-on vendors. The vendors of addons such as http://svn.asterisk.org/svn/libss7 and http://svn.asterisk.org/svn/asterisk/trunk/channels/sig_ss7.c ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN SS7
Hi all, I'm being requested to deploy an IVR service using SS7. I've deployed Asterisk before using ISDN connection, but never with SS7. Can anyone explain me the different between using ISDN and SS7 ? What need I do now to change to use SS7 ?. Many thanks, Giang -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN SS7
Hi cary, Can you recommend me what add-on vendors I should use ? Can a open source solution such as chan_ss7 or libss7 support many conncurrent calls (for example 240 calls) ? Thanks From: Cary Fitch ca...@usawide.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sun, October 24, 2010 9:33:28 AM Subject: Re: [asterisk-users] ISDN SS7 SS7 is an inter-telco system using a separate network for all signaling. You must have an SS7 network connection before anything will work. Then the T1 Spans run 24 64k audio paths. The SS7 net exchanges the call data and connection info between the switches. Asterisk doesn’t support SS7 natively although I believe there are one or more add-on vendors. Cary Fitch From:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of huu giang Sent: Sunday, October 24, 2010 10:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ISDN SS7 Hi all, I'm being requested to deploy an IVR service using SS7. I've deployed Asterisk before using ISDN connection, but never with SS7. Can anyone explain me the different between using ISDN and SS7 ? What need I do now to change to use SS7 ?. Many thanks, Giang -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astersik can not detect DTMF key
Hi all, I'm building a karaoke service. Asterisk will play a music file, people can detect the point when they want to sing and record by press * key during the music is playing, and press # key to stop recording. I use 2 functions: ast_streamfile and ast_seekstream to play audio file, and function ast_waitstream_fr to detect whenever people press DTMF key. The problems is that, Asterisk can detect * key when I press, but to detect # key, I have to press two times. What is the problem ? Sorry for my English. Very thanks, Giang -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astersik can not detect DTMF key
Hi Almost phones I used meet this problem, they are - Nokia 1200 - Nokia 6210 - Nokia E72. When I used Softphone to test on IP (SIP), some softphones meet similar problem (twinkle), some don't meet (Xlite, Kapanga). Very Thanks Giang From: Zeeshan Zakaria zisha...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thu, June 24, 2010 5:48:39 PM Subject: Re: [asterisk-users] Astersik can not detect DTMF key What type of phone you are using? It is possible that # is used by this phone as one of its internal functions. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-24 6:41 AM, huu giang huugiang...@yahoo.com wrote: Hi all, I'm building a karaoke service. Asterisk will play a music file, people can detect the point when they want to sing and record by press * key during the music is playing, and press # key to stop recording. I use 2 functions: ast_streamfile and ast_seekstream to play audio file, and function ast_waitstream_fr to detect whenever people press DTMF key. The problems is that, Asterisk can detect * key when I press, but to detect # key, I have to press two times. What is the problem ? Sorry for my English. Very thanks, Giang -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astersik can not detect DTMF key
Hi, When I require user enter a code and end wich # key, for example 1234#, Asterisk can detect # key and detect the code people just enter. Thanks From: Zeeshan Zakaria zisha...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thu, June 24, 2010 5:48:39 PM Subject: Re: [asterisk-users] Astersik can not detect DTMF key What type of phone you are using? It is possible that # is used by this phone as one of its internal functions. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-24 6:41 AM, huu giang huugiang...@yahoo.com wrote: Hi all, I'm building a karaoke service. Asterisk will play a music file, people can detect the point when they want to sing and record by press * key during the music is playing, and press # key to stop recording. I use 2 functions: ast_streamfile and ast_seekstream to play audio file, and function ast_waitstream_fr to detect whenever people press DTMF key. The problems is that, Asterisk can detect * key when I press, but to detect # key, I have to press two times. What is the problem ? Sorry for my English. Very thanks, Giang -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)
Dear Goe. Do you mean, I just need request Telco provide me a E1 line, ask them to configure MSC support SS7/ISUP, so my Asterisk can receive calls. What is the benefits if I use ISDN instead of ISUP/SS7 and vice versa. Thanks. --- On Wed, 4/14/10, Goke M Aruna gok...@gmail.com wrote: From: Goke M Aruna gok...@gmail.com Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, April 14, 2010, 9:39 PM Hi Huu, Asterisk support ss7. Check chan_ss7 and libss7, both project are active and working like charm. Thanks On 4/15/10, huu giang huugiang...@yahoo.com wrote: Dear Goke, I don't use ISDN to connect to MSC, it connect to ISDN network. There are other people deploy IVR using this protocol. About ISUP/SS7, supporting technical from the vendor I bought Sangoma Card, they said that If I want to connect to MSC, I have to use ISUP/SS7 protocl, but now Asterisk doesn't support SS7 protocol and I have to buy a SS7 package to install on Asterisk Server so Astersik can work with SS7. Is it right ?, It is the first time I deploy Asterisk, so please consult me. Thanks Hiện tại, nếu anh dùng luồng ISDN thì không cần báo hiệu SS7 nhưng anh muốn kết nối trực tiếp vào MSC của Telco(SS7) là anh cần phải có giao thức SS7 để chúng bắt tay làm việc, nhưng hiện tại trên tổng đài soft không hổ trợ, anh cần mua gói phần mền SS7 để cài đặt vào tổng đài để chúng làm việc với giao thức SS7. --- On Wed, 4/14/10, Goke M Aruna gok...@gmail.com wrote: From: Goke M Aruna gok...@gmail.com Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, April 14, 2010, 4:24 AM hello Huu, Can you share their explanation with me at least, I can gain from it too. Thanks On Wed, Apr 14, 2010 at 10:01 AM, huu giang huugiang...@yahoo.com wrote: Hi Goke, Some experienced people said me to use ISDN to connect to MSC. Thanks very much. --- On Wed, 4/14/10, Goke M Aruna gok...@gmail.com wrote: From: Goke M Aruna gok...@gmail.com Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, April 14, 2010, 1:50 AM Hello Huu, use E1/SS7 signaling or if you MSC speak SIP, then use SIP. Thanks On Tue, Apr 13, 2010 at 11:46 AM, huu giang huugiang...@yahoo.com wrote: Hi all, My Asterisk connect to GSM core network (connect directly to MSC) through E1 lines. What the kind of protocol is used ?. It is ISUP/SS7 protocol ? Thanks in advance -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sent from my mobile device -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)
Hi, I have a question about point code. When I install a sangoma card, there weren't any step require me to configure point code. I thin when telco provide us E1 lines, MSC will configure to forward calles through the E1 lines to my Asterisk ?. Please point me how can I configure point code for Asterisk. Do I have to config Sangoma Card ? Thanks. --- On Thu, 4/15/10, Ngo-Vi Hoai-Anh hoai...@gmx.de wrote: From: Ngo-Vi Hoai-Anh hoai...@gmx.de Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, April 15, 2010, 1:18 AM Hi there, ISDN is not a protocol. See http://en.wikipedia.org/wiki/Integrated_Services_Digital_Network#Internationale_Verbreitung for more details. As far as I know Vietnam uses DSS1 (Euro-ISDN). The benefit of SS7 is that you can use a single signaling channel (D-Chan) for more E1s. I have 8 E1s with a single D-Chan at our site. Using DSS1 you have one D-Chan for one E1. From signaling point of view there is no gain if you just need one E1. If you want to have a interconnection with that GSM telco you need a point code from them and then configure your Asterisk box properly to receive calls. I will fly to Vietnam next month. If you want to just feel free to email me. Maybe I can give you a helping hand there. huu giang schrieb: Dear Goe. Do you mean, I just need request Telco provide me a E1 line, ask them to configure MSC support SS7/ISUP, so my Asterisk can receive calls. What is the benefits if I use ISDN instead of ISUP/SS7 and vice versa. Thanks. --- On *Wed, 4/14/10, Goke M Aruna /gok...@gmail.com/* wrote: From: Goke M Aruna gok...@gmail.com Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, April 14, 2010, 9:39 PM Hi Huu, Asterisk support ss7. Check chan_ss7 and libss7, both project are active and working like charm. Thanks On 4/15/10, huu giang huugiang...@yahoo.com /mc/compose?to=huugiang...@yahoo.com wrote: Dear Goke, I don't use ISDN to connect to MSC, it connect to ISDN network. There are other people deploy IVR using this protocol. About ISUP/SS7, supporting technical from the vendor I bought Sangoma Card, they said that If I want to connect to MSC, I have to use ISUP/SS7 protocl, but now Asterisk doesn't support SS7 protocol and I have to buy a SS7 package to install on Asterisk Server so Astersik can work with SS7. Is it right ?, It is the first time I deploy Asterisk, so please consult me. Thanks Hiện tại, nếu anh dùng luồng ISDN thì không cần báo hiệu SS7 nhưng anh muốn kết nối trực tiếp vào MSC của Telco(SS7) là anh cần phải có giao thức SS7 để chúng bắt tay làm việc, nhưng hiện tại trên tổng đài soft không hổ trợ, anh cần mua gói phần mền SS7 để cài đặt vào tổng đài để chúng làm việc với giao thức SS7. --- On Wed, 4/14/10, Goke M Aruna gok...@gmail.com /mc/compose?to=gok...@gmail.com wrote: From: Goke M Aruna gok...@gmail.com /mc/compose?to=gok...@gmail.com Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com /mc/compose?to=asterisk-us...@lists.digium.com Date: Wednesday, April 14, 2010, 4:24 AM hello Huu, Can you share their explanation with me at least, I can gain from it too. Thanks On Wed, Apr 14, 2010 at 10:01 AM, huu giang huugiang...@yahoo.com /mc/compose?to=huugiang...@yahoo.com wrote: Hi Goke, Some experienced people said me to use ISDN to connect to MSC. Thanks very much. --- On Wed, 4/14/10, Goke M Aruna gok...@gmail.com /mc/compose?to=gok...@gmail.com wrote: From: Goke M Aruna gok...@gmail.com /mc/compose?to=gok...@gmail.com Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com /mc/compose?to=asterisk-us...@lists.digium.com Date: Wednesday, April 14, 2010, 1:50 AM Hello Huu, use E1/SS7 signaling or if you MSC speak SIP, then use SIP. Thanks On Tue, Apr 13, 2010 at 11:46 AM, huu giang huugiang...@yahoo.com /mc/compose?to=huugiang...@yahoo.com wrote: Hi all, My Asterisk connect
Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)
Hi Hoanh Anh, Thanks for your answer. You are very kind. With Asterisk, I'm just a newbie, so I really need your help. I'll very happy if I can have a meeting with you next month in Vietnam. Thanks --- On Thu, 4/15/10, Ngo-Vi Hoai-Anh hoai...@gmx.de wrote: From: Ngo-Vi Hoai-Anh hoai...@gmx.de Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, April 15, 2010, 1:18 AM Hi there, ISDN is not a protocol. See http://en.wikipedia.org/wiki/Integrated_Services_Digital_Network#Internationale_Verbreitung for more details. As far as I know Vietnam uses DSS1 (Euro-ISDN). The benefit of SS7 is that you can use a single signaling channel (D-Chan) for more E1s. I have 8 E1s with a single D-Chan at our site. Using DSS1 you have one D-Chan for one E1. From signaling point of view there is no gain if you just need one E1. If you want to have a interconnection with that GSM telco you need a point code from them and then configure your Asterisk box properly to receive calls. I will fly to Vietnam next month. If you want to just feel free to email me. Maybe I can give you a helping hand there. huu giang schrieb: Dear Goe. Do you mean, I just need request Telco provide me a E1 line, ask them to configure MSC support SS7/ISUP, so my Asterisk can receive calls. What is the benefits if I use ISDN instead of ISUP/SS7 and vice versa. Thanks. --- On *Wed, 4/14/10, Goke M Aruna /gok...@gmail.com/* wrote: From: Goke M Aruna gok...@gmail.com Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, April 14, 2010, 9:39 PM Hi Huu, Asterisk support ss7. Check chan_ss7 and libss7, both project are active and working like charm. Thanks On 4/15/10, huu giang huugiang...@yahoo.com /mc/compose?to=huugiang...@yahoo.com wrote: Dear Goke, I don't use ISDN to connect to MSC, it connect to ISDN network. There are other people deploy IVR using this protocol. About ISUP/SS7, supporting technical from the vendor I bought Sangoma Card, they said that If I want to connect to MSC, I have to use ISUP/SS7 protocl, but now Asterisk doesn't support SS7 protocol and I have to buy a SS7 package to install on Asterisk Server so Astersik can work with SS7. Is it right ?, It is the first time I deploy Asterisk, so please consult me. Thanks Hiện tại, nếu anh dùng luồng ISDN thì không cần báo hiệu SS7 nhưng anh muốn kết nối trực tiếp vào MSC của Telco(SS7) là anh cần phải có giao thức SS7 để chúng bắt tay làm việc, nhưng hiện tại trên tổng đài soft không hổ trợ, anh cần mua gói phần mền SS7 để cài đặt vào tổng đài để chúng làm việc với giao thức SS7. --- On Wed, 4/14/10, Goke M Aruna gok...@gmail.com /mc/compose?to=gok...@gmail.com wrote: From: Goke M Aruna gok...@gmail.com /mc/compose?to=gok...@gmail.com Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com /mc/compose?to=asterisk-us...@lists.digium.com Date: Wednesday, April 14, 2010, 4:24 AM hello Huu, Can you share their explanation with me at least, I can gain from it too. Thanks On Wed, Apr 14, 2010 at 10:01 AM, huu giang huugiang...@yahoo.com /mc/compose?to=huugiang...@yahoo.com wrote: Hi Goke, Some experienced people said me to use ISDN to connect to MSC. Thanks very much. --- On Wed, 4/14/10, Goke M Aruna gok...@gmail.com /mc/compose?to=gok...@gmail.com wrote: From: Goke M Aruna gok...@gmail.com /mc/compose?to=gok...@gmail.com Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com /mc/compose?to=asterisk-us...@lists.digium.com Date: Wednesday, April 14, 2010, 1:50 AM Hello Huu, use E1/SS7 signaling or if you MSC speak SIP, then use SIP. Thanks On Tue, Apr 13, 2010 at 11:46 AM, huu giang huugiang...@yahoo.com /mc/compose?to=huugiang...@yahoo.com wrote: Hi all, My Asterisk connect to GSM core network (connect directly to MSC) through E1 lines. What the kind of protocol is used ?. It is ISUP/SS7 protocol
Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)
Hi Goke, Some experienced people said me to use ISDN to connect to MSC. Thanks very much. --- On Wed, 4/14/10, Goke M Aruna gok...@gmail.com wrote: From: Goke M Aruna gok...@gmail.com Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, April 14, 2010, 1:50 AM Hello Huu, use E1/SS7 signaling or if you MSC speak SIP, then use SIP. Thanks On Tue, Apr 13, 2010 at 11:46 AM, huu giang huugiang...@yahoo.com wrote: Hi all, My Asterisk connect to GSM core network (connect directly to MSC) through E1 lines. What the kind of protocol is used ?. It is ISUP/SS7 protocol ? Thanks in advance -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)
Dear Goke, I don't use ISDN to connect to MSC, it connect to ISDN network. There are other people deploy IVR using this protocol. About ISUP/SS7, supporting technical from the vendor I bought Sangoma Card, they said that If I want to connect to MSC, I have to use ISUP/SS7 protocl, but now Asterisk doesn't support SS7 protocol and I have to buy a SS7 package to install on Asterisk Server so Astersik can work with SS7. Is it right ?, It is the first time I deploy Asterisk, so please consult me. Thanks Hiện tại, nếu anh dùng luồng ISDN thì không cần báo hiệu SS7 nhưng anh muốn kết nối trực tiếp vào MSC của Telco(SS7) là anh cần phải có giao thức SS7 để chúng bắt tay làm việc, nhưng hiện tại trên tổng đài soft không hổ trợ, anh cần mua gói phần mền SS7 để cài đặt vào tổng đài để chúng làm việc với giao thức SS7. --- On Wed, 4/14/10, Goke M Aruna gok...@gmail.com wrote: From: Goke M Aruna gok...@gmail.com Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, April 14, 2010, 4:24 AM hello Huu, Can you share their explanation with me at least, I can gain from it too. Thanks On Wed, Apr 14, 2010 at 10:01 AM, huu giang huugiang...@yahoo.com wrote: Hi Goke, Some experienced people said me to use ISDN to connect to MSC. Thanks very much. --- On Wed, 4/14/10, Goke M Aruna gok...@gmail.com wrote: From: Goke M Aruna gok...@gmail.com Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, April 14, 2010, 1:50 AM Hello Huu, use E1/SS7 signaling or if you MSC speak SIP, then use SIP. Thanks On Tue, Apr 13, 2010 at 11:46 AM, huu giang huugiang...@yahoo.com wrote: Hi all, My Asterisk connect to GSM core network (connect directly to MSC) through E1 lines. What the kind of protocol is used ?. It is ISUP/SS7 protocol ? Thanks in advance -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)
Hi all, My Asterisk connect to GSM core network (connect directly to MSC) through E1 lines. What the kind of protocol is used ?. It is ISUP/SS7 protocol ? Thanks in advance -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RES: Cache sound files for faster processing
I haven't ever try any ram disk before. --- On Tue, 4/6/10, Flavio E. Goncalves fla...@voffice.com.br wrote: From: Flavio E. Goncalves fla...@voffice.com.br Subject: [asterisk-users] RES: Cache sound files for faster processing To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Tuesday, April 6, 2010, 9:56 AM Did you tried the good old ram disk? Flavio E. Goncalves www.asteriskguide.com -Mensagem original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Em nome de David Backeberg Enviada em: Tuesday, April 06, 2010 12:50 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] Cache sound files for faster processing On Tue, Apr 6, 2010 at 12:36 AM, huu giang huugiang...@yahoo.com wrote: Dear List, Are there any way of configuring of Asterisk so it'll cache sound files in memory, and when Asterisk receive a call, instead of loading sound files from the disk, it will load from the memory and so Asterisk can process much more call at a time than with faster speed it is not caching. Thanks, Aside from the suggestions, you could try out an SSD drive, which is both expensive compared to a traditional hard drive and very fast. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cache sound files for faster processing
Thanks Steve for your information. As you said, I don't need care for caching sound files ?, Linux is responsible for the job ?, So at the first time, Asterisk will load sound files from hard disk, and after that, it will load from RAM. Thanks. --- On Tue, 4/6/10, Steve Edwards asterisk@sedwards.com wrote: From: Steve Edwards asterisk@sedwards.com Subject: Re: [asterisk-users] Cache sound files for faster processing To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, April 6, 2010, 7:15 AM Are there any way of configuring of Asterisk so it'll cache sound files in memory, and when Asterisk receive a call, instead of loading sound files from the disk On Mon, 5 Apr 2010, Luki wrote: Not directly, but it's not really needed. A long as the machine has enough RAM, the files will be served from RAM by the operating system. Sure there is the overhead of opening/closing files and reading them, but on modern OS this overhead is negligible if the files are cached (asterisk may even use mmap, but I'm not sure). You can also make a ram disk (say via tmpfs), copy the sounds there and symlink the sound directory to that location. However, I don't think you will gain much. A bit off topic, but recently I was trying to improve the performance of a MythTV frontend (a Linux home theater application). I tried tmpfs and /dev/ramx and neither yielded noticeable improvement. My informal conclusion is that Linux does a good enough job at managing memory that tweaking is probably not worth it. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cache sound files for faster processing
Dear List, Are there any way of configuring of Asterisk so it'll cache sound files in memory, and when Asterisk receive a call, instead of loading sound files from the disk, it will load from the memory and so Asterisk can process much more call at a time than with faster speed it is not caching. Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk load balancing and failover
As I Known, when a call coming to a MSC, the MSC will request the HLR for subscriber information. HLR response will include the address of the Asterisk Server, so the MSC will forward the call to Asterisk Server. Is it right ? If I have three Asterisk Server (each has a point code), is there any way to configure HLR so when MSC ask HLR about the Asterisk Server, HLR will response the address in a round robin way. So the call will be balance to three Asterisk Server ?. Do you mean above is the way MSC do load balancing ?. --- On Thu, 4/1/10, Ngo-Vi Hoai-Anh hoai...@gmx.de wrote: From: Ngo-Vi Hoai-Anh hoai...@gmx.de Subject: Re: [asterisk-users] Asterisk load balancing and failover To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, April 1, 2010, 3:01 AM I'm not quite sure what do you mean with MSC. Anyway, I assume your environment is like [PSTN (Public Switched Telephone Network)]--[DTM Switch]---SS7 (PRI line)[Asterisk Box]VoIP (SIP/IAX etc...)- IP net If you mean MSC Mobile Switching Center it could look like [GSM Network]-[MSC]- SS7 [Asterisk Box]-VoIP--IP net Normally, the DTM Switch or MSC should be configurable for load-balancing and failover. Point code is for SS7 networking like IP address for IP networking. huu giang schrieb: Do you mean that SS7 switch is a MSC and do all MSC support load balancing without any hardware between it and my Server. Sorry for my English, what do you mean two point codes for my servers ?. I have at least two servers. --- On *Wed, 3/31/10, Tobias Wolf /tobias.w...@evision.de/* wrote: From: Tobias Wolf tobias.w...@evision.de Subject: Re: [asterisk-users] Asterisk load balancing and failover To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, March 31, 2010, 4:27 AM huu giang schrieb: Hi Zeeshan I know a solution using DRBD, Heartbeat and RedFone hardware to provide failover ability to Asterisk. If I have two Asterisk Servers, and each server has a TDM card and a PRI line connect to each card, how your solution can provide failover ability to Asterisk ? Do you need any other hardware? The calles to my IVR System don't just come from IP network (SIP) but can come from SS7 network. Well, if that case the SS7 Switch to which you are connected should be able to load balance the call to both of your servers. I guess you have two point codes for you servers? If one server goes down, the ss7 switch received the red alarms and stops to route calls to it. Once the server is up again it will get new calls. So, we only thing you have to worry about is to keep state information between the two servers consistent if people record messages or access databases. Regards, Tobias Thanks. --- On *Fri, 3/26/10, Zeeshan Zakaria /zisha...@gmail.com /mc/compose?to=zisha...@gmail.com/* wrote: From: Zeeshan Zakaria zisha...@gmail.com /mc/compose?to=zisha...@gmail.com Subject: Re: [asterisk-users] Asterisk load balancing and failover To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com /mc/compose?to=asterisk-us...@lists.digium.com Date: Friday, March 26, 2010, 1:51 AM About two years ago I setup two high availability solutions using DRBD and Heartbeat. The worked great and shutting down or unplugging one server stayed transparent for the callers, as IVRs stayed available. Having said this, it was not very straight forward to set it up, but not very difficut either. So Heartbeat and DRBD can be a good starting point for you. -- Zeeshan A Zakaria On 2010-03-26 4:40 AM, huu giang huugiang...@yahoo.com /mc/compose?to=huugiang...@yahoo.com /mc/compose?to=huugiang...@yahoo.com /mc/compose?to=huugiang...@yahoo.com wrote: Hi List, I'm finding a solution to provide failover and load balancing features to my IVR system. Anyone suggest me what is the best solution please?. what the hardware I should use ?. I heard about RedFone, but someone on the mail list said that it is not good because *TDMoE* module in asterisk is not so *stable* and TDMoE is stale. And It seems that RedFone doesn't not support load balancing ability (I can't find any document about
Re: [asterisk-users] Asterisk load balancing and failover
Hi Hoai Anh, I've asked a telecommunication engineer, and he said me that MSC support load balancing. Thanks for your answer. --- On Thu, 4/1/10, Ngo-Vi Hoai-Anh hoai...@gmx.de wrote: From: Ngo-Vi Hoai-Anh hoai...@gmx.de Subject: Re: [asterisk-users] Asterisk load balancing and failover To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, April 1, 2010, 3:01 AM I'm not quite sure what do you mean with MSC. Anyway, I assume your environment is like [PSTN (Public Switched Telephone Network)]--[DTM Switch]---SS7 (PRI line)[Asterisk Box]VoIP (SIP/IAX etc...)- IP net If you mean MSC Mobile Switching Center it could look like [GSM Network]-[MSC]- SS7 [Asterisk Box]-VoIP--IP net Normally, the DTM Switch or MSC should be configurable for load-balancing and failover. Point code is for SS7 networking like IP address for IP networking. huu giang schrieb: Do you mean that SS7 switch is a MSC and do all MSC support load balancing without any hardware between it and my Server. Sorry for my English, what do you mean two point codes for my servers ?. I have at least two servers. --- On *Wed, 3/31/10, Tobias Wolf /tobias.w...@evision.de/* wrote: From: Tobias Wolf tobias.w...@evision.de Subject: Re: [asterisk-users] Asterisk load balancing and failover To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, March 31, 2010, 4:27 AM huu giang schrieb: Hi Zeeshan I know a solution using DRBD, Heartbeat and RedFone hardware to provide failover ability to Asterisk. If I have two Asterisk Servers, and each server has a TDM card and a PRI line connect to each card, how your solution can provide failover ability to Asterisk ? Do you need any other hardware? The calles to my IVR System don't just come from IP network (SIP) but can come from SS7 network. Well, if that case the SS7 Switch to which you are connected should be able to load balance the call to both of your servers. I guess you have two point codes for you servers? If one server goes down, the ss7 switch received the red alarms and stops to route calls to it. Once the server is up again it will get new calls. So, we only thing you have to worry about is to keep state information between the two servers consistent if people record messages or access databases. Regards, Tobias Thanks. --- On *Fri, 3/26/10, Zeeshan Zakaria /zisha...@gmail.com /mc/compose?to=zisha...@gmail.com/* wrote: From: Zeeshan Zakaria zisha...@gmail.com /mc/compose?to=zisha...@gmail.com Subject: Re: [asterisk-users] Asterisk load balancing and failover To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com /mc/compose?to=asterisk-us...@lists.digium.com Date: Friday, March 26, 2010, 1:51 AM About two years ago I setup two high availability solutions using DRBD and Heartbeat. The worked great and shutting down or unplugging one server stayed transparent for the callers, as IVRs stayed available. Having said this, it was not very straight forward to set it up, but not very difficut either. So Heartbeat and DRBD can be a good starting point for you. -- Zeeshan A Zakaria On 2010-03-26 4:40 AM, huu giang huugiang...@yahoo.com /mc/compose?to=huugiang...@yahoo.com /mc/compose?to=huugiang...@yahoo.com /mc/compose?to=huugiang...@yahoo.com wrote: Hi List, I'm finding a solution to provide failover and load balancing features to my IVR system. Anyone suggest me what is the best solution please?. what the hardware I should use ?. I heard about RedFone, but someone on the mail list said that it is not good because *TDMoE* module in asterisk is not so *stable* and TDMoE is stale. And It seems that RedFone doesn't not support load balancing ability (I can't find any document about this feature). Best Regards, Giang Huu. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello
Re: [asterisk-users] Asterisk load balancing and failover
Do you mean that SS7 switch is a MSC and do all MSC support load balancing without any hardware between it and my Server. Sorry for my English, what do you mean two point codes for my servers ?. I have at least two servers. --- On Wed, 3/31/10, Tobias Wolf tobias.w...@evision.de wrote: From: Tobias Wolf tobias.w...@evision.de Subject: Re: [asterisk-users] Asterisk load balancing and failover To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, March 31, 2010, 4:27 AM huu giang schrieb: Hi Zeeshan I know a solution using DRBD, Heartbeat and RedFone hardware to provide failover ability to Asterisk. If I have two Asterisk Servers, and each server has a TDM card and a PRI line connect to each card, how your solution can provide failover ability to Asterisk ? Do you need any other hardware? The calles to my IVR System don't just come from IP network (SIP) but can come from SS7 network. Well, if that case the SS7 Switch to which you are connected should be able to load balance the call to both of your servers. I guess you have two point codes for you servers? If one server goes down, the ss7 switch received the red alarms and stops to route calls to it. Once the server is up again it will get new calls. So, we only thing you have to worry about is to keep state information between the two servers consistent if people record messages or access databases. Regards, Tobias Thanks. --- On *Fri, 3/26/10, Zeeshan Zakaria /zisha...@gmail.com/* wrote: From: Zeeshan Zakaria zisha...@gmail.com Subject: Re: [asterisk-users] Asterisk load balancing and failover To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, March 26, 2010, 1:51 AM About two years ago I setup two high availability solutions using DRBD and Heartbeat. The worked great and shutting down or unplugging one server stayed transparent for the callers, as IVRs stayed available. Having said this, it was not very straight forward to set it up, but not very difficut either. So Heartbeat and DRBD can be a good starting point for you. -- Zeeshan A Zakaria On 2010-03-26 4:40 AM, huu giang huugiang...@yahoo.com /mc/compose?to=huugiang...@yahoo.com wrote: Hi List, I'm finding a solution to provide failover and load balancing features to my IVR system. Anyone suggest me what is the best solution please?. what the hardware I should use ?. I heard about RedFone, but someone on the mail list said that it is not good because *TDMoE* module in asterisk is not so *stable* and TDMoE is stale. And It seems that RedFone doesn't not support load balancing ability (I can't find any document about this feature). Best Regards, Giang Huu. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk load balancing and failover
Anyone has experience in configuring Redfone to support failover, please share with me?. --- On Fri, 3/26/10, Eric Wheeler aster...@ew.ewheeler.org wrote: From: Eric Wheeler aster...@ew.ewheeler.org Subject: Re: [asterisk-users] Asterisk load balancing and failover To: huugiang...@yahoo.com Cc: asterisk-users asterisk-users@lists.digium.com Date: Friday, March 26, 2010, 9:12 AM If I have two Asterisk Servers, and each server has a TDM card and a PRI line connect to each card, how your solution can provide failover ability to Asterisk ? Do you need any other hardware? Have a look at this article and how they shared a single T1 line across two servers for failover: http://www.linuxjournal.com/article/7661 (sorry about the missing in-reply-to header. I'm not sure how to get Evolution to add in-reply-to manually and I'm receiving messages in digest form.) -- Eric Wheeler President Portland Linux Support www.PortlandLinuxSupport.com 503-330-4277 PO Box 86710 Portland, OR 97286 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk load balancing and failover
I'm sorry, Anyone has experience in configuring Redfone to support load-balancing, please share with me? I can't find any guide about this feature from RedFone. --- On Mon, 3/29/10, huu giang huugiang...@yahoo.com wrote: From: huu giang huugiang...@yahoo.com Subject: Re: [asterisk-users] Asterisk load balancing and failover To: Eric Wheeler aster...@ew.ewheeler.org Cc: asterisk-users asterisk-users@lists.digium.com Date: Monday, March 29, 2010, 12:37 AM Anyone has experience in configuring Redfone to support failover, please share with me?. --- On Fri, 3/26/10, Eric Wheeler aster...@ew.ewheeler.org wrote: From: Eric Wheeler aster...@ew.ewheeler.org Subject: Re: [asterisk-users] Asterisk load balancing and failover To: huugiang...@yahoo.com Cc: asterisk-users asterisk-users@lists.digium.com Date: Friday, March 26, 2010, 9:12 AM If I have two Asterisk Servers, and each server has a TDM card and a PRI line connect to each card, how your solution can provide failover ability to Asterisk ? Do you need any other hardware? Have a look at this article and how they shared a single T1 line across two servers for failover: http://www.linuxjournal.com/article/7661 (sorry about the missing in-reply-to header. I'm not sure how to get Evolution to add in-reply-to manually and I'm receiving messages in digest form.) -- Eric Wheeler President Portland Linux Support www.PortlandLinuxSupport.com 503-330-4277 PO Box 86710 Portland, OR 97286 -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Continue a dialplan when the client hang up the call
Hi all, When a user make a call to Asterisk, and when user hang up the call at any point of the conversation, Asterisk will stop Diaplan intermediately. At this situation, Are there any way to make Asterisk continue execute the Diaplan ?, so Asterisk can do something like that delete temporary file, .. etc. Thanks in advance, Giang -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Continue a dialplan when the client hang up the call
Hi Ishfaq When Asterisk continue the dialplan, can it discover that the client has hang up the call ?. Is there any way ?. --- On Mon, 3/29/10, Ishfaq Malik i...@pack-net.co.uk wrote: From: Ishfaq Malik i...@pack-net.co.uk Subject: Re: [asterisk-users] Continue a dialplan when the client hang up the call To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Monday, March 29, 2010, 2:34 AM There is the h exten to deal with exactly what you want http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension huu giang wrote: Hi all, When a user make a call to Asterisk, and when user hang up the call at any point of the conversation, Asterisk will stop Diaplan intermediately. At this situation, Are there any way to make Asterisk continue execute the Diaplan ?, so Asterisk can do something like that delete temporary file, .. etc. Thanks in advance, Giang -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Continue a dialplan when the client hang up the call
Thanks Ishfaq, h extension is the answer for my question :). --- On Mon, 3/29/10, huu giang huugiang...@yahoo.com wrote: From: huu giang huugiang...@yahoo.com Subject: Re: [asterisk-users] Continue a dialplan when the client hang up the call To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Monday, March 29, 2010, 2:52 AM Hi Ishfaq When Asterisk continue the dialplan, can it discover that the client has hang up the call ?. Is there any way ?. --- On Mon, 3/29/10, Ishfaq Malik i...@pack-net.co.uk wrote: From: Ishfaq Malik i...@pack-net.co.uk Subject: Re: [asterisk-users] Continue a dialplan when the client hang up the call To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Monday, March 29, 2010, 2:34 AM There is the h exten to deal with exactly what you want http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension huu giang wrote: Hi all, When a user make a call to Asterisk, and when user hang up the call at any point of the conversation, Asterisk will stop Diaplan intermediately. At this situation, Are there any way to make Asterisk continue execute the Diaplan ?, so Asterisk can do something like that delete temporary file, .. etc. Thanks in advance, Giang -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk load balancing and failover
Hi List, I'm finding a solution to provide failover and load balancing features to my IVR system. Anyone suggest me what is the best solution please?. what the hardware I should use ?. I heard about RedFone, but someone on the mail list said that it is not good because TDMoE module in asterisk is not so stable and TDMoE is stale. And It seems that RedFone doesn't not support load balancing ability (I can't find any document about this feature). Best Regards, Giang Huu. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk load balancing and failover
Hi Zeeshan I know a solution using DRBD, Heartbeat and RedFone hardware to provide failover ability to Asterisk. If I have two Asterisk Servers, and each server has a TDM card and a PRI line connect to each card, how your solution can provide failover ability to Asterisk ? Do you need any other hardware? The calles to my IVR System don't just come from IP network (SIP) but can come from SS7 network. Thanks. --- On Fri, 3/26/10, Zeeshan Zakaria zisha...@gmail.com wrote: From: Zeeshan Zakaria zisha...@gmail.com Subject: Re: [asterisk-users] Asterisk load balancing and failover To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, March 26, 2010, 1:51 AM About two years ago I setup two high availability solutions using DRBD and Heartbeat. The worked great and shutting down or unplugging one server stayed transparent for the callers, as IVRs stayed available. Having said this, it was not very straight forward to set it up, but not very difficut either. So Heartbeat and DRBD can be a good starting point for you. -- Zeeshan A Zakaria On 2010-03-26 4:40 AM, huu giang huugiang...@yahoo.com wrote: Hi List, I'm finding a solution to provide failover and load balancing features to my IVR system. Anyone suggest me what is the best solution please?. what the hardware I should use ?. I heard about RedFone, but someone on the mail list said that it is not good because TDMoE module in asterisk is not so stable and TDMoE is stale. And It seems that RedFone doesn't not support load balancing ability (I can't find any document about this feature). Best Regards, Giang Huu. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hook playback or ControlPlayBack cmd
Dear all, I want playback or ControlPlayback cmd to trigger me when a DTMF key is pressed, so I can execute Monitor cmd or any thing I want. Anyone did this job before?. Please help me. Thanks in advance, Giang -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Define an array of sip number in sip.conf
Hi List, How can I define an array of sip number in sip.conf ? I want to define an array of sip number from 1000 to 2000, so I can make a performance test on Asterisk using sipp. Thanks in Advance, Giangnh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] record a user call while playing a background music
Hi all. I want to write a diaplan which can make asterisk act as a karaoke serivce. It mean that A user can call to Asterisk, and while the user singing a song, the asterisk play a background music. Is it possible to do that ? please help me. Thanks in Advance, Giangnh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
