I am trying to use a the SIP tapi from www.enum.at <http://www.enum.at/> . This works fine from all kinds of applications which support TAPI, like outlook and Dialer Pro. However when making tapi controlled calls, the signaling to and from PSTN seems to fail. I have used the digium hardware ISDN PRI boards, but also a SIP gateway. Both result in a audio message from asterisk saying that the number is unavailable. But, what I need is to have the original PSTN status transferred to the SIP phone( xten eyebeam in this case) so I can see whether the end point was just busy, or that the number dialed was just plain wrong. Any help would be very very much appreciated. Joash Maanlander 14a/b m: +31 6 53 80 28 20 3824 MP Amersfoort e: joash.herbrink@kahuna.nl t: +31 33 4500370 ext 1006 URL: www.kahuna.nl <file:///\\www.kahuna.nl\> f: +31 33 4500371 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051128/55a5bb1d/attachment.htm
Has anyone successfully implemented SIPTAPI with asterisk? It would appear to require a true proxy. I assume it will need a seperate user account to register and place calls, but I have been unable to get it to attempt to register with asterisk. If you have it working, example configuration would be appreciated. Thanks
users@cingerr.com
2006-May-11 05:39 UTC
[Asterisk-Users] I killed my install, help me restore :(
Try removing /usr/lib/asterisk/modules/* that would help. check if you have extra modules in /usr/lib/asterisk/modules and backup them. after that do a make install in asterisk-1.2.7.1 and that`s all.> > Never try upgrades half-asleep and 1/4-knowledgable! > > Got a link from a friend about the FLITE TTS that was rewritten to work > really well with Asterisk. So I downloaded and installed it on my 1.0.9 > server - oops. So, I downloaded Asterisk 1.2.7.1 did the proper install > process, got all kinds of warnings about incompatible modules. Forget > what > all I did but I eventually got it to compile and install, but now when I > run > asterisk -vvvvc it dies at chan_oss. > > What all directories/files do I need to remove (I have backups at least) > to > completely remove Asterisk so I can start over with 1.2.7.1? > > thanks > > _______________________________________________ > --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 >
Jerry Jones wrote:> Has anyone successfully implemented SIPTAPI with asterisk? It wouldi use it with Asterisk without problems.> appear to require a true proxy. I assume it will need a seperate useryou need either a separate user or use the user of the SIP phone.> account to register and place calls, but I have been unable to get it to > attempt to register with asterisk.It does not REGISTER by design - it will not receive calls, just make them.> If you have it working, example configuration would be appreciated.Configure the siptapi exactly like your sip phone. make sure the call limit for this user >= 3 (maybe 2 works too) make sure in your extensions.conf to have a routing to the sipphone using the username used in the SIP configuration. read the docs and use ethereal and debugview to watch the log messages. klaus> > Thanks > _______________________________________________ > --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
Shawn Porter
2006-May-11 06:30 UTC
[Asterisk-Users] I killed my install, help me restore :(
Never try upgrades half-asleep and 1/4-knowledgable! Got a link from a friend about the FLITE TTS that was rewritten to work really well with Asterisk. So I downloaded and installed it on my 1.0.9 server - oops. So, I downloaded Asterisk 1.2.7.1 did the proper install process, got all kinds of warnings about incompatible modules. Forget what all I did but I eventually got it to compile and install, but now when I run asterisk -vvvvc it dies at chan_oss. What all directories/files do I need to remove (I have backups at least) to completely remove Asterisk so I can start over with 1.2.7.1? thanks
Gareth Blades
2006-May-11 06:38 UTC
[Asterisk-Users] I killed my install, help me restore :(
It could be an old module still left behind from the previous version. I would delete everything in /usr/lib/asterisk/modules and then reinstall (make install) and see if it will start. On Thu, 2006-05-11 at 14:30, Shawn Porter wrote:> Never try upgrades half-asleep and 1/4-knowledgable! > > Got a link from a friend about the FLITE TTS that was rewritten to work > really well with Asterisk. So I downloaded and installed it on my 1.0.9 > server - oops. So, I downloaded Asterisk 1.2.7.1 did the proper install > process, got all kinds of warnings about incompatible modules. Forget what > all I did but I eventually got it to compile and install, but now when I run > asterisk -vvvvc it dies at chan_oss. > > What all directories/files do I need to remove (I have backups at least) to > completely remove Asterisk so I can start over with 1.2.7.1? > > thanks > > _______________________________________________ > --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
Shawn Porter
2006-May-11 06:52 UTC
[Asterisk-Users] I killed my install, help me restore :(
I did have some extra modules (mysql_cdr, cepstral tts) but I can start-over. Based on your suggestion, I went one step further. I have gone through and deleted (rm -Rf just to make sure :) ) /etc/asterisk /var/lib/asterisk /usr/lib/asterisk /usr/include/asterisk /usr/sbin/asterisk I am just running the install process again. make clean make make install Will post results as soon as my poor machine finishes the compiling. -----Original Message----- From: Gareth Blades [mailto:list-asterisk@linguaphone.co.uk] Sent: Thursday, May 11, 2006 9:38 AM To: ivr_solutions@rogers.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Bulk] Re: [Asterisk-Users] I killed my install, help me restore :( It could be an old module still left behind from the previous version. I would delete everything in /usr/lib/asterisk/modules and then reinstall (make install) and see if it will start. On Thu, 2006-05-11 at 14:30, Shawn Porter wrote:> Never try upgrades half-asleep and 1/4-knowledgable! > > Got a link from a friend about the FLITE TTS that was rewritten to work > really well with Asterisk. So I downloaded and installed it on my 1.0.9 > server - oops. So, I downloaded Asterisk 1.2.7.1 did the proper install > process, got all kinds of warnings about incompatible modules. Forgetwhat> all I did but I eventually got it to compile and install, but now when Irun> asterisk -vvvvc it dies at chan_oss. > > What all directories/files do I need to remove (I have backups at least)to> completely remove Asterisk so I can start over with 1.2.7.1? > > thanks > > _______________________________________________ > --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
Hello, Anyone try to use SIP TAPI (http://www.enum.at/index.php?id=479) with Asterisk? Pretty nice, pretty simple. I am hung up on something, though, and google doesn't specifically address my issue. The program seems to go to the s extension in the default context of the sip user it is configured for. Is there a way to set it to go to an extension of the default context? I couldn't figure out how... Assuming you can't specify an extension within the default context, then that leads me to believe that a SIP user needs to be created specifically for each instance of SIP TAPI. So I tried that. I made a context just for this, with the s extension set to immediately dial my real sip phone. This works, and Asterisk bridges the call, but that's it. The docs are sparse, and I don't think this program was specifically written for Asterisk - what am I missing? Sincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 www.torrenga.com
FYI, I've got a working version of asttapi that will work with Asterisk 1.2 up on my site at http://www.kirkhamsystems.com/asttapi . It's the debug build, so it contains some extra code, but that's merely to help me out if anyone sends in a bug report (which so far out of apparently 80 something downloads, no bug reports yet, I guess it's working well). Only reason I mention it is that I can't imagine trying to drop down to SIP level support in asterisk when the asterisk management interface works so well with asttapi. Clint -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brent Torrenga Sent: Wednesday, May 24, 2006 10:32 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP TAPI Hello, Anyone try to use SIP TAPI (http://www.enum.at/index.php?id=479) with Asterisk? Pretty nice, pretty simple. I am hung up on something, though, and google doesn't specifically address my issue. The program seems to go to the s extension in the default context of the sip user it is configured for. Is there a way to set it to go to an extension of the default context? I couldn't figure out how... Assuming you can't specify an extension within the default context, then that leads me to believe that a SIP user needs to be created specifically for each instance of SIP TAPI. So I tried that. I made a context just for this, with the s extension set to immediately dial my real sip phone. This works, and Asterisk bridges the call, but that's it. The docs are sparse, and I don't think this program was specifically written for Asterisk - what am I missing? Sincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 www.torrenga.com _______________________________________________ --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
Clint, Crap. Wish I would have seen your setup first. I played with asttapi for a few days, and gave up. My problems were manager related, and you cover those points well enough on your page. I was able to get SIP TAPI to work this way: - each install of SIP TAPI needs a SIP user in sip.conf. - each SIP user made for SIP TAPI needs a context in extensions.conf. - each context made for SIP TAPI looks like: [blah-tapi] exten => s,1,Dial(SIP/blah) Include => blah-internal-context It seems to work great this way. The software is taken out of the loop immediately after connecting to SIP/blah, thus does not have call state like ast tapi does. However, I think this also means that you can have an unlimited number of simultaneous calls, unlike ast tapi. Also, this does not provide for pop-ups on incoming calls or call progress, whereas ast tapi does. What I really don't like about my setup is the lack of "outbound" caller-id on your phone - no way to use the redial button. I guess a plus for SIP TAPI here is that it doesn't require manager events to be put into the dial plan - yay! Clint, in your opinion, do I have the differences between the two programs summarized correctly?>FYI, I've got a working version of asttapi that will work with Asterisk >1.2 up on my site at http://www.kirkhamsystems.com/asttapi . It's the >debug build, so it contains some extra code, but that's merely to help >me out if anyone sends in a bug report (which so far out of apparently >80 something downloads, no bug reports yet, I guess it's working well). > >Only reason I mention it is that I can't imagine trying to drop down to >SIP level support in asterisk when the asterisk management interface >works so well with asttapi. > >ClintSincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 www.torrenga.com
Yeah, that sounds about right. I can see advantages and disadvantages to both. The main advantage I see to AstTapi besides signaling incoming calls (which I haven't tested on my modified code, I guess I should work on that) is that once you've setup a user in the Asterisk Management interface and modified your dial plan accordingly, you're done, you don't have to add new entries for every instance of AstTapi. That would be a burden I'd think in a larger installation of SIPTapi with Asterisk. The nice advantage also to AstTapi is that signaling is ongoing while the call is in progress, so you can end the call from the TAPI application. This is a real boon in real CTI setups for callcenters where the phones might be set to autoanswer incoming calls on a headset, display information, and the user ends the call. Seems like there should be a simpler way to do an TAPI interface with the Asterisk management interface w/o a bunch of UserEvents though. I think I'll look into that, because it'd be nice if all you had to do was add the user to the manager.conf and be done. I know I could probably do that on outbound calls, incoming calls might be a little more difficult. It could probably be done with some assumptions about extension length, etc. Sorry, just thinking aloud, but that's probably where it should go from here. Clint -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brent Torrenga Sent: Wednesday, May 24, 2006 12:44 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: SIP TAPI Clint, Crap. Wish I would have seen your setup first. I played with asttapi for a few days, and gave up. My problems were manager related, and you cover those points well enough on your page. I was able to get SIP TAPI to work this way: - each install of SIP TAPI needs a SIP user in sip.conf. - each SIP user made for SIP TAPI needs a context in extensions.conf. - each context made for SIP TAPI looks like: [blah-tapi] exten => s,1,Dial(SIP/blah) Include => blah-internal-context It seems to work great this way. The software is taken out of the loop immediately after connecting to SIP/blah, thus does not have call state like ast tapi does. However, I think this also means that you can have an unlimited number of simultaneous calls, unlike ast tapi. Also, this does not provide for pop-ups on incoming calls or call progress, whereas ast tapi does. What I really don't like about my setup is the lack of "outbound" caller-id on your phone - no way to use the redial button. I guess a plus for SIP TAPI here is that it doesn't require manager events to be put into the dial plan - yay! Clint, in your opinion, do I have the differences between the two programs summarized correctly?>FYI, I've got a working version of asttapi that will work with Asterisk >1.2 up on my site at http://www.kirkhamsystems.com/asttapi . It's the >debug build, so it contains some extra code, but that's merely to help >me out if anyone sends in a bug report (which so far out of apparently >80 something downloads, no bug reports yet, I guess it's working well). > >Only reason I mention it is that I can't imagine trying to drop down to >SIP level support in asterisk when the asterisk management interface >works so well with asttapi. > >ClintSincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 www.torrenga.com _______________________________________________ --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
Clint, thanks for your comments and documentation on asttapi, great work! Some weeks ago after hours of reverseengineering we gave up on asttapi :( Provided with your informations, things seem to become clearer now and we'll try again. Guido