[Asterisk-Users] How to read ISDN messages - URGENT!!!!
Hi, Were using Asterisk with Digium TE110P card for the PSTN E1 interface. Our PRI is enabled with detecting the connected-party-number feature. When an OUTBOUND call is made to a phone, the PRI will send back an ISDN messages containing the connected-number and we can use that information to validate the extension user is calling the party that he/she is authorized to. This is to avoid the user letting know the receiver about the call and getting the receiver to divert the phone to some other number. How can we read this ISDN messages from Asterisk? Your help would be VERY much appreciated. Lilantha ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk supports VXML?
Hi Foong, Thats a good question youve put out there. Yes, Asterisk supports VXML and heres how its done; Firstly in the SIP.conf, you need to have your VXML application/browser defined; sip.conf: [vxmlapp] type=friend insecure=yes username=777 reinvite=no host=123.45.67.8 Then in the EXTENSIONS.conf it will look like this; extensions.conf: exten = 777,1,Setvar,VXML_URL=voicexml=http%3A%2F%2F123.45.67.20%3A6969%2Fhellovxml%2Fhellovxml exten = 777,2,Dial,sip/vxmlapp|10 exten = 777,3,HangUp Hope thisll clear your thoughts. Cheers! Lilantha Karunaratne MSCS Tel: (65) 90403497 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chee Foong Sent: Friday, February 25, 2005 10:17 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk supports VXML? Hello, Does asterisk supports VXML? Couldn't find much resource on that on google and wiki. Thanks Foong ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Comparison of Business Edition VS Open Source
Hi, I would like to find out; Anyone using Asterisk Business Edition? If yes, could you give us a brief description of what type of solution you use it for? What made you to choose the Business Edition over the Open Source? What advantage/disadvantage you have found? Whats the level of stability in terms of number of calls, dropped calls in conversation, echo cancellation, poor voice quality, etc? Were faced with a major project to use Asterisk and wed like to find out more information statistics in real-life before we decide which version to use. Anyone can help us would be greatly appreciated. Cheers! Lilantha ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Goiax.com DID not working anymore?
Guess we all have the incoming problem! When I make a call out form the 1st to GoIAX and into my 2nd box, I see this Oct 21 13:46:39 NOTICE[4948]: Rejected connect attempt from 204.13.233.114 which in other words mean that the call comes in but nothing happens to it. In my extensions_additional.conf [ext-did] include = ext-did-custom exten = 1258,1,Goto(from-pstn,s,1) ; exten = s,1,Goto(from-pstn,s,1) ; In iax_additional.conf register=878201001258:[EMAIL PROTECTED] [g6] ;outbound username=878201001258 type=peer secret= xxx host=server1.goiax.com context=ext-did auth=md5 [1258] ;inbound auth=md5 context=ext-did host=server1.goiax.com secret= xxx type=friend username=878201001258 Outgoing calls to US numbers work like a charm but I cannot get the calls coming into the box at all. Anyone can give me some guide as to where Im doing something wrong would be really appreciated. Cheers! Lilantha From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas Mavrides Sent: Friday, October 21, 2005 1:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Goiax.com DID not working anymore? mine is 339 - Original Message - From: Blake Krone To: [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, October 21, 2005 1:38 AM Subject: Re: [Asterisk-Users] Goiax.com DID not working anymore? What is your prefix? Mine is 978, maybe only certain ones are having problems? On 10/20/05, Robert Webb [EMAIL PROTECTED] wrote: Just tested mine and it is working fine. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Blake Krone Sent: Thursday, October 20, 2005 5:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Goiax.com DID not working anymore? I've been using my goiax.com DID for a few days now and it is no longer working. I get the number or code you dialed can not be found. I haven't touched any configs or anything on the asterisk box since it was working last night. Anyone else having problems using the DID from goiax? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Goiax.com DID not working anymore?
Thanks for the update. I can make calls out to 1-800 numbers in the US but IAX to IAX between the 2 offices doesnt work which runs purely on IP. Thats the problem were having. Cheers! Lilantha From: Blake Krone [mailto:[EMAIL PROTECTED] Sent: Friday, October 21, 2005 5:32 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Goiax.com DID not working anymore? I can't even call out anymore! That stopped working probably 3 or 4 days ago. My iax looks pretty much the same as yours. I always get circuits busy. :( On 10/20/05, Lilantha Karunaratne [EMAIL PROTECTED] wrote: Guess we all have the incoming problem! When I make a call out form the 1st to GoIAX and into my 2nd box, I see this Oct 21 13:46:39 NOTICE[4948]: Rejected connect attempt from 204.13.233.114 which in other words mean that the call comes in but nothing happens to it. In my extensions_additional.conf [ext-did] include = ext-did-custom exten = 1258,1,Goto(from-pstn,s,1) ; exten = s,1,Goto(from-pstn,s,1) ; In iax_additional.conf register= 878201001258:[EMAIL PROTECTED] [g6] ;outbound username=878201001258 type=peer secret= xxx host=server1.goiax.com context=ext-did auth=md5 [1258] ;inbound auth=md5 context=ext-did host=server1.goiax.com secret= xxx type=friend username=878201001258 Outgoing calls to US numbers work like a charm but I cannot get the calls coming into the box at all. Anyone can give me some guide as to where I'm doing something wrong would be really appreciated. Cheers! Lilantha From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Andreas Mavrides Sent: Friday, October 21, 2005 1:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Goiax.com DID not working anymore? mine is 339 - Original Message - From: Blake Krone To: [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, October 21, 2005 1:38 AM Subject: Re: [Asterisk-Users] Goiax.com DID not working anymore? What is your prefix? Mine is 978, maybe only certain ones are having problems? On 10/20/05, Robert Webb [EMAIL PROTECTED] wrote: Just tested mine and it is working fine. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Blake Krone Sent: Thursday, October 20, 2005 5:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Goiax.com DID not working anymore? I've been using my goiax.com DID for a few days now and it is no longer working. I get the number or code you dialed can not be found. I haven't touched any configs or anything on the asterisk box since it was working last night. Anyone else having problems using the DID from goiax? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Fax support using T.38
Hi, Just wondering whether anyone has done fax relaying or pass-through using Asterisk T.38 Please let me know your thoughts as I need to come up with a fax server using Asterisk with T.38 possible? Cheers! Lilantha ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Fax support using T.38
Yes while we agree on your point, if the requirement is to relay faxes over a VoIP network by an operator, then we need to use FoIP methods via T.37 / 38. Do you know of anyone using * with T.38 on a commercial implementation? Cheers! Lilantha From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel Sent: Thursday, November 10, 2005 6:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Fax support using T.38 T.37 actually isnt that bad. When its 'fax' (ie analog data passed to represent data) its TDM, however to cross the internet and reap cost savings its effectively mime encoded and sent via SMTP. This gives you the TDM capabilities (no jitter, low latency, all the things faxes like). The combination of the two is appealing from a technical standpoint vs analog fax over a network because of bandwidth savings, potentially easier to distribute to multiple destinations, allows for TCP connections on a lower QoS to be sent, etc. At least appealing to me. This also makes it easier to integrate into a unified messaging solution because you would have the fax as an email at some point, delivery does not have to be to another fax machine. C F wrote: The same way the best roads can handle landing and takeoffs of 747s but weren't meant for it, a runway is what's needed, VoIP could have faxing with it, but TDM is really whats needed. Please search the archives for this question, it has been asked over and over and over, again and again. On 11/9/05, Lilantha Karunaratne [EMAIL PROTECTED] wrote: Hi, Just wondering whether anyone has done fax relaying or pass-through using Asterisk T.38 Please let me know your thoughts as I need to come up with a fax server using Asterisk with T.38 possible? Cheers! Lilantha ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Fax support using T.38
Thanks for the URL pointer. Apparently this is live project and we do not think we could do any testing on that but will do these tests internally I suppose. Anyone using * with T.38 on a commercial platform? Cheers! Lilantha From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Thursday, November 10, 2005 7:53 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Fax support using T.38 Andy Kuo wrote: Lilantha I've been looking for fax solutions with Asterisk too. Unfortunately, it seems like there's no T.38 support for Asterisk so far. In fact, I think there's only fax-to-email solution for * now. I'm getting some SIP ATA's with T.38 support next week, but I am not sure if I can somehow get it to work with Asterisk. I wonder if there's anyone in the Asterisk community able to do fax between 2 SIP/IAX ATA's, or it's just simply not possible. There is a patch set on the bugtracker at the moment to add T.38 support to Asterisk. I am currently in the process of testing it. http://bugs.digium.com/view.php?id=5090 -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel tone description
Hi, Were trying to use TDM04B with a few analog switches and weve noticed that it works with the tones from USA only. As its documented saying that those tones are hard-coded in the source for analog cards. Wed like to know if theres anyone who could tell us under which file these settings would be hard-coded as we could like to do some experiments which will benefit to all Zaptel / Asterisk users in Asian part of the world. Would appreciate an early reply. Cheers! Lilantha Karunaratne MSCS Tel: (65) 90403497 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel tone description
Hi Paul, Thanks for the update. Weve tried all these conf files but to no success as its stated, the tone descriptions are hard-coded into the Zaptel drivers. A better place to work in would be zonedata.c Cheers! Lilantha Karunaratne MSCS Tel: (65) 90403497 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hewlett Sent: Thursday, October 06, 2005 2:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zaptel tone description On Wednesday 05 October 2005 17:46, Ricardo Poppi wrote: Lilantha, the tones are supposed to be switched using the loadzone and defaultzone lines in /etc/zaptel.conf , and, progzone in /etc/asterisk/zapata.conf. Also look at /etc/asterisk/indications.conf Paul -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel tone description
Thanks Ricardo for your suggestion. Well take a look at the zonedata.c and see how we could make some changes. Cheers! Lilantha Karunaratne MSCS Tel: (65) 90403497 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Poppi Sent: Wednesday, October 05, 2005 11:47 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Zaptel tone description Lilantha, the tones are supposed to be switched using the loadzone and defaultzone lines in /etc/zaptel.conf , and, progzone in /etc/asterisk/zapata.conf. The information about countries and frequencies/times are at zonedata.c located in the sourcecode of zaptel. As you may know, changing zonedata.c information requires a re-compilation of the zaptel module. Hope it helps, Ricardo Poppi. Date: Wed, 5 Oct 2005 19:02:23 +0800 From: Lilantha Karunaratne [EMAIL PROTECTED] Subject: [Asterisk-Users] Zaptel tone description To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: !~!UENERkVCMDkAAQACABgA2xL2SF4ybEylW69jV2juZcKQJ0Zvf9/YaUil/[EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi, We're trying to use TDM04B with a few analog switches and we've noticed that it works with the tones from USA only. As it's documented saying that those tones are hard-coded in the source for analog cards. We'd like to know if there's anyone who could tell us under which file these settings would be hard-coded as we could like to do some experiments which will benefit to all Zaptel / Asterisk users in Asian part of the world. Would appreciate an early reply. Cheers! Lilantha Karunaratne MSCS Tel: (65) 90403497 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hold on with Asterisk Manager
Hi, We have the same exact problem as you. We want to put the caller on hold or do a supervised transfer through the Manager API but it doesnt seem to work. We can re-direct the call to a different context for example [Hold] but the moment we do that one party in the call leg gets disconnected. When we try the transfer, the same thing happens. It accepts the context re-direction but it disconnects the call-leg. Were using Java and asterisk-java API to communicate with the Manager API. Anyone who has managed to get over this issue? Lilantha From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of amaury BOSSE Sent: Tuesday, January 17, 2006 4:54 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hold on with Asterisk Manager Hello, I am writing a program based on Astersik Manager which needs to put calls on hold and to redirect them to others extensions. I haven't funded any action able to do this. Is there a way to put calls on hold using Asterisk Manager Actions? Amaury ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
