Hello all, Just to update, Instruction's can be found at www.omniis.com/asttapi, including where to download it from. This is update 0.02, this now includes a little feedback from Asterisk so that when click to dial has occurred then it is indicated at the start and the end of the call. Now working on inbound calls. Any question, please send to me. Regards Nick
Hi,> -----Original Message----- > Instruction's can be found at www.omniis.com/asttapi, > including where to > download it from. This is update 0.02, this now includes a little > feedback from Asterisk so that when click to dial has occurred then it > is indicated at the start and the end of the call. > > Now working on inbound calls. > > Any question, please send to me.Cool program! What I would like is feedback on this: You are now using the manager interface to initiate calls. This means there can be no valid callerid to either side of the call, and billing (accountcodes) might break. Would it make more sense to signal to a small daemon that can write spool files ? It might be more flexible. Also, we now set the channel to dial with, and it gets the number appended. However, if you were to use chan_local, a suffix would be desirable (to set context other than default). Best regards, Florian
Hello all, Just to inform you all, next version released, please try it and let me know about any bugs you find (or any further features). This release now includes 1/ Inbound calls 2/ Call origination 3/ Call dialling from phone detected 4/ Call origination using contexts 5/ Can set the caller ID 6/ Priority field added to the origination Some bugs fixed also. Known bug at the moment - Caller ID number not presented correctly to TAPI - next to work on. Download from http://www.sf.net/projects/asttapi documentation at www.omniis.com/asttapi Regards Nick
Hi Nick, When I click on dialling options and then line properties asterisk appears 3 times? Is this normal, does it matter which one I choose? If I try to use this outlook replies with the service is busy on another call with nothing actually appearing in the asterisk console. Also one other question do I need to set dial by application or dial by context? Thanks again if this works it will save a heap of time. Cheers, Dean -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Nick Knight Sent: Friday, 7 May 2004 6:32 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asttapi Hello all, Just to inform you all, next version released, please try it and let me know about any bugs you find (or any further features). This release now includes 1/ Inbound calls 2/ Call origination 3/ Call dialling from phone detected 4/ Call origination using contexts 5/ Can set the caller ID 6/ Priority field added to the origination Some bugs fixed also. Known bug at the moment - Caller ID number not presented correctly to TAPI - next to work on. Download from http://www.sf.net/projects/asttapi documentation at www.omniis.com/asttapi Regards Nick _______________________________________________ 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
Hi List, I work with asterisk and two isdn s0 card in my server. All works fine. To the dial, I work with outlook an asttapi. Wenn I click a contact an say dial, my telephone rings. I draft the listener an he dial a number. But I will, wenn I say dial in outlook, that my softphone an my hardphone ring. Dose not matter where I draft, this sip client dial the number. Example: I will dial a call to 0123 456799 in outlook. My softclient ring and my grandstream ring. I draft on softclient an he dial this number. On next time I draft on grandstream and he dial this number. OK. Thank you for your help. Best Regards, Robert Siedl (Gesch?ftsf?hrer) Siedl International Networks A-3500 Krems, Magnesitstra?e 1 Tel: +43 (2732) 71545 DW11 Fax: +43 (2732) 71545 DW91 Web: http://www.sin.co.at Eine gute EDV macht SIN(n): - Linuxl?sungen - Kommunikationssysteme - Groupware- und Workflowsolution - VoIP Telefonie - Security- und VPN Systeme - IT Consulting & Projektmanagement PS: Der SuSE Linux Open Exchange Server ist die openSource Alternative f?r Groupware und Messaging in Kombination mit Outlook oder ein Webinterface! ++ Virenanalyse ++ Ikarus GuardNT hat dieses eMail auf Viren und Trojaner untersucht. Nichts Verd?chtiges gefunden. keine Anlagen gefunden -------------------------------------------------------
Has anyone actually gotten ASTTAPI to work? I can't seem to get it to work, yet I have other TAPI setups (SNAP and xtelsio) working fine. I have noticed that SNAP and Xtelsio act differently. Etelescript is the application that will be calling TAPI. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060927/e6adec2b/attachment.htm
In article <094401c6e251$4fbbbd90$1401010a@ICS.local>, asterisk-users@ics-il.net says...> Has anyone actually gotten ASTTAPI to work? I can't seem to get it to work, yet I have other TAPI setups (SNAP and xtelsio) working fine. I have noticed that SNAP and Xtelsio act differently. Etelescript is the application that will be calling TAPI.Hi Mike! I have been using ASTTAPI, but it takes time to configure it and I'm not sure it's developing any more. Now I'm using SNAP for several days but it seams that it has some bugs. I'm using Snap's forum to check with developer about this, but it's going slowly. I don't think that Snap is for business production yet. If developer doesn't solve those problems with Snap, I'll try Etelescript. Is Etelescript free? Is it open source? -- Tomislav Par?ina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: tomo@pbx.lama.hr e-mail: tparcina#lama.hr http://www.lama.hr
> -----Urspr?ngliche Nachricht----- > Von: Tomislav Parcina [mailto:tparcina@lama.hr] > Gesendet: Donnerstag, 28. September 2006 09:10 > An: Asterisk Users Mailing List - Non-Commercial Discussion > Betreff: [asterisk-users] Re: ASTTAPI > > In article <094401c6e251$4fbbbd90$1401010a@ICS.local>,asterisk-users@ics-il.net> says... > > Has anyone actually gotten ASTTAPI to work? I can't seem to get it towork, yet I> have other TAPI setups (SNAP and xtelsio) working fine. I have noticedthat SNAP> and Xtelsio act differently. Etelescript is the application that will becalling TAPI.> > Hi Mike! > > I have been using ASTTAPI, but it takes time to configure it and I'm notsure it's> developing any more. Now I'm using SNAP for several days but it seams thatit has> some bugs. I'm using Snap's forum to check with developer about this, butit's going> slowly. I don't think that Snap is for business production yet. > > If developer doesn't solve those problems with Snap, I'll try Etelescript.Is Etelescript> free? Is it open source?After spending many many hours on asttapi and other tapisolutions, we found Tapi for Asterisk here: http://www.phonesuite.de/de/produkte/ast_tsp/phonesuite_tapi_for_asterisk.ht m It works like a charm and the licensefee with 25.-?/10 Clients is really fair. We couldn't find any bugs in the software and in combination with tapicall www.tapicall.de it's our preferred link-up to Outlook/Exchange in all of our asterisk installations. Since it's language is in german, you might have a closer look on some german dictionaries, but after configuration is done (5 minutes) you can forget about ever installed it. ;-) Hope, these informations saved you some time, money and nerves Guido