Re: [asterisk-users] Call Center
what about astercc? Regards On 1 Aug 2015 18:57, Murthy Gandikota murth...@hotmail.com wrote: Hi All Has anyone used Asterisk for a Call Center operation? What I mean is: given a list of phone numbers, can Asterisk dial each number, play a message and accept some DTMF? I ask because I am an employee of a non-profit company based in San Diego, CA. I already evaluated Voicent and Voxeo. The former has expensive licensing terms and the latter is not best suited for a call center. I would appreciate your kind comments. Thanking you murthy -- _ -- 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] screen capture for asterisk call center solution
Hi all, Can someone suggest a good solution on agent screen capture in asterisk call center? open source preferred but cheap offer is welcome too. Regards -- _ -- 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] screen capture for asterisk call center solution
Thank you AJ, Just want to know from people who uses asterisk as call center solution, how and what screen capture solution / applications are in use. Regards On 20 Dec 2013 11:06, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Friday 20 December 2013, Goke M Aruna wrote: Hi all, Can someone suggest a good solution on agent screen capture in asterisk call center? open source preferred but cheap offer is welcome too. ??? What are you trying to do? Asterisk is *just* a telephony construction kit. It doesn't know any more about call centre agents than that they are someone or something on the end of a telephone line, and certainly has no interest in what may or may not be on their screens (if they even have screens). -- AJS Answers come *after* questions. -- _ -- 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] screen capture for asterisk call center solution
Thanks AJ, The capturing of agent activities on their desktop by the supervisor. Regards On 20 Dec 2013 12:18, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Friday 20 December 2013, Goke M Aruna wrote: Thank you AJ, Just want to know from people who uses asterisk as call center solution, how and what screen capture solution / applications are in use. What do you mean by screen capture ? -- AJS Answers come *after* questions. -- _ -- 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] screen capture for asterisk call center solution
Thanks Dardini. On Fri, Dec 20, 2013 at 1:39 PM, Leandro Dardini ldard...@gmail.com wrote: Just use VNC... 2013/12/20 Goke M Aruna gok...@gmail.com Thanks AJ, The capturing of agent activities on their desktop by the supervisor. Regards On 20 Dec 2013 12:18, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Friday 20 December 2013, Goke M Aruna wrote: Thank you AJ, Just want to know from people who uses asterisk as call center solution, how and what screen capture solution / applications are in use. What do you mean by screen capture ? -- AJS Answers come *after* questions. -- _ -- 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 -- _ -- 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] screen capture for asterisk call center solution
hello AJ, Can I benefit from that code of yours? regards On Fri, Dec 20, 2013 at 4:56 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: * THIS IS NOT WHERE YOUR REPLY GOES * On Friday 20 December 2013, Goke M Aruna wrote: Thanks AJ, The capturing of agent activities on their desktop by the supervisor. Regards Ah. That is really nothing to do with Asterisk. We run our business on a custom in-house app (written by me) which runs in a web browser -- and there is only a very minimal set of apps on anyone's desktop. But this minimal set does include openssh-server; so if we wanted to, we could ssh into any machine and run `ps aux`. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call in the queue to listen to the Channel
hello all, I have call queue management system where all call comes in, put in the queue while the caller speak with the online support team / teacher. However, my major concern is those under MOH (in the queue) will not be able to listen to the teacher until their turns and this is required. my Request, I want to make the teachers audio as the MOH and every new calls that comes in listen to the life channels with the teachers. Any help will be highly welcome. Regards -- _ -- 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] Call in the queue to listen to the Channel
hello Carlos, Thanks. I will try that and give back later. Regards On Tue, Aug 14, 2012 at 6:49 PM, Carlos Chavez cur...@telecomabmex.comwrote: On Tue, 2012-08-14 at 18:43 +0100, Goke M Aruna wrote: hello all, I have call queue management system where all call comes in, put in the queue while the caller speak with the online support team / teacher. However, my major concern is those under MOH (in the queue) will not be able to listen to the teacher until their turns and this is required. my Request, I want to make the teachers audio as the MOH and every new calls that comes in listen to the life channels with the teachers. Any help will be highly welcome. This would work better using a MeetMe/Confbridge room. Insert users muted so they can only listen to the conference and then the instructor can unmute them when it is their turn to speak. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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] any working calling card solution open source
hi all, Can someone give me information on any open source asterisk calling card solution? I have laid my hands on astpp, astcc, asterisk-prepaid-0.3.1, agi-ccard.agi without luck. I guess my problem is Asterisk-perl I will be glad for a quick response. Regards -- _ -- 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] any working calling card solution open source
thank Carlos, Thanks but too big for a demo interms of setup no demo data. I got the astcc working but still looking for alternative Thanks On Mon, Jul 16, 2012 at 8:32 PM, Carlos Rojas crt.ro...@gmail.com wrote: Hello a2billing works fine Regards On Mon, Jul 16, 2012 at 1:47 PM, Goke M Aruna gok...@gmail.com wrote: hi all, Can someone give me information on any open source asterisk calling card solution? I have laid my hands on astpp, astcc, asterisk-prepaid-0.3.1, agi-ccard.agi without luck. I guess my problem is Asterisk-perl I will be glad for a quick response. Regards -- _ -- 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] Best Scripting Language
Can you give me a reference on how to compile perl? Thanks On 4/1/11, Danny Nicholas da...@debsinc.com wrote: I'm going to vote for PERL as well. C is not a scripting language. Also keep in mind that you can compile PERL into C for your hundreds of calls per second box. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel Sent: Friday, April 01, 2011 8:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Best Scripting Language Do you think C is a scripting language? -- Sent from my iPhone On Apr 1, 2011, at 8:27 AM, Roger Burton West ro...@firedrake.org wrote: On Fri, Apr 01, 2011 at 05:27:20PM +0530, Gopalakrishnan A.N wrote: Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Depends on the other parameters. Perl is great for rapid development, but I wouldn't run it per-call on a box taking hundreds of calls per second. (Ditto Ruby and Python.) C will be much faster, but it's more effort to write and debug. -- _ -- 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 -- 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
[asterisk-users] Using asterisk and icecast for live audio streaming.
Hi all, Can someone give me a direction on how to use asterisk and icecast or any other apps for a live audio cast? The audio feed is external to the asterisk server. Voip-info.org is not detailed on this. Thank you -- _ -- 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 Exchange - a waste of money?
can someone give more eduaction to me about what the asterisk exchange is all about? thanks On Thu, Dec 9, 2010 at 5:43 AM, Sevana Oy sa...@sevana.fi wrote: Hi, A couple of months ago we registered our product AQuA at Asterisk Exchange. We were told that it collects like 14K visitors per month and knowing interest to our product from Asterisk community we have calculated a certain super-mini-minimal % of visitors coming from Asterisk Exchange to our web site... Here comes the funny thing - there was no traffic increase since then, there were no referral visist from Asterisk Exchange... We are getting something a bit less than 100 product inquiries a month and NONE has ever mentioned that learnt about us from Asterisk Exchange... My question is: have we just wasted $2500 for being listed there? Is Asterisk Exchange some kind of bubble? Unfortunately we never got response from the people who sold us this service :-) Thanks and cheers, Vallu Sevana Oy -- _ -- 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] Implementing Asterisk on a WAN or an Intranet
Ay, Run make sample in the asterisk src dir3. And check the etc asterisk dir3 for sample config files. If you are not clear after reading contact me goksie at gmail. Thanks On 5/30/10, ayodele abejide ayodeleabej...@hotmail.com wrote: Hi everyone, I want to Implement asterisk on my company's intranet comprising of 100 local area networks I don't know how to configure my dialplan and my sip.conf files Thanks Date: Sun, 30 May 2010 10:06:12 -0400 From: vene...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to use one single IP as origination I have an Asterisk with multiple IP's, on the same subnet. When a call comes in, I need to send it back out via SIP, but need that only one IP is used as originating IP for all calls. For example machines has 192.168.50.3 192.168.50.4 192.168.50.5 but when I originate the second leg of a call, the IP address that is supposed to be read as source IP must be 192.168.50.5, regardless of how the call arrived. How do I do that? _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969 -- 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 huu, ss7 is preffered for me. tested on huawei msc / siemens ewsd / alcatel s12 just to mention few and its working fine. On Thu, Apr 15, 2010 at 8:04 AM, huu giang huugiang...@yahoo.com wrote: 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.comhttp://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.comhttp://mc/compose?to=gok...@gmail.com wrote: From: Goke M Aruna gok...@gmail.comhttp://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.comhttp://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.comhttp://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.comhttp://mc/compose?to=gok...@gmail.com wrote: From: Goke M Aruna gok...@gmail.comhttp://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.comhttp://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.comhttp://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 ? 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
Re: [asterisk-users] protocol used to connect Asterisk and GSM core network (MSC)
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 -- _ -- 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)
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.comhttp://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 ? 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 -- _ -- 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 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] how to start callerid for india
Hi, Edit your logger.conf, set messages in debug mode, make test incoming and outgoing calls. Copy the log in message dirz3* and post. Goke On 3/20/10, Zeeshan Zakaria zisha...@gmail.com wrote: Does your regular phone shows callerid on this line. If the service provider is sending the callerid, asterisk doesn't have to do anything special to retrieve it. -- Zeeshan On 2010-03-20 1:25 PM, cool dude cool_dudeof...@yahoo.co.in wrote: i belong to india. i am making pbx using sangoma fxo card. i want that when ever call comes to my PSTN line i should see the no from where call is coming. so i have to configures chan_dahdi.conf according to my region. i checked dahdi.conf and in that they have mentioned for india ## ; Hide the name part and leave just the number part of the caller ID ; string. Only applies to PRI channels. ;hidecalleridname=yes ; ; Type of caller ID signalling in use ; bell = bell202 as used in US (default) ; v23 = v23 as used in the UK ; v23_jp = v23 as used in Japan ; dtmf = DTMF as used in Denmark, Sweden and Netherlands ; smdi = Use SMDI for caller ID. Requires SMDI to be enabled (usesmdi). ; ;cidsignalling=v23 ; ; What signals the start of caller ID ; ring= a ring signals the start (default) ; polarity= polarity reversal signals the start ; polarity_IN = polarity reversal signals the start, for India, ; for dtmf dialtone detection; using DTMF. ; (see doc/India-CID.txt) ; ;cidstart=polarity so i edited chan_dahdi.conf according to my region. ### vi chan_dahdi.conf ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2010-03-18 ;Dahdi Channels Configurations ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;cidstart=ring ;cidstart=polarity ;callerid=asreceived cidsignalling=polarity_IN sendcalleridafter=2 ;Sangoma AU100 [slot:0 bus: span:1] wanpipe1 context=from-zaptel group=0 echocancel=yes signalling = fxs_ks channel = 1 context=from-zaptel group=0 echocancel=yes signalling = fxs_ks channel = 2 now when call comes to PSTN line i am not able to see the no. here is cli log *CLI -- Starting simple switch on 'DAHDI/1-1' [Mar 20 20:12:51] NOTICE[6659]: chan_dahdi.c:8692 ss_thread: Got event 18 (Ring Begin)... [Mar 20 20:12:51] NOTICE[6659]: chan_dahdi.c:8692 ss_thread: Got event 17 (Polarity Reversal)... [Mar 20 20:12:51] NOTICE[6659]: chan_dahdi.c:8692 ss_thread: Got event 17 (Polarity Reversal)... -- Executing [...@from-zaptel:1] Wait(DAHDI/1-1, 2) in new stack -- Executing [...@from-zaptel:2] GotoIfTime(DAHDI/1-1, 23:59-7:59|mon-sun|*|*?closed,s,1) in new stack -- Executing [...@from-zaptel:3] Dial(DAHDI/1-1, SIP/112,60,tT) in new stack == Using SIP RTP CoS mark 5 -- Called 112 -- SIP/112- is ringing == Spawn extension (from-zaptel, s, 3) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' # plz help me out. -- Your Mail works best with the New Yahoo Optimized IE8. Get it NOW!http://in.rd.yahoo.com/tagline_ie8_new/*http://downloads.yahoo.com/in/internetexplorer/ . -- _ -- 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] Dial Plan Application(main-menu)
Do you have the main-menu sound file in the correct format? Goksie On 11/20/09, Steve Edwards asterisk@sedwards.com wrote: On Fri, 20 Nov 2009, aster...@opensourcesolution.in wrote: the problem is that when call comes it answers but backgroup main menu dosent play,for test purpose i had done The problem is that you do not have (or have not provided) sufficient information to solve today's problem. You should bump up logging (logger.conf, console = debug,dtmf,error,event,notice,verbose,warning) and contemplate (for a very long time) the meaning of the messages. There are resources available on the Internet (google.com, voip-info.org) where you can find answers faster and without annoying the hell out of the list as you attempt to have others write your dialplan line-by-line, day-by-day. -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
