Discussion:
SIP TAPI
(too old to reply)
Brent Torrenga
2006-05-24 15:31:41 UTC
Permalink
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
***@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
Clint Sharp
2006-05-24 16:08:58 UTC
Permalink
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-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of Brent
Torrenga
Sent: Wednesday, May 24, 2006 10:32 AM
To: asterisk-***@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
***@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

_______________________________________________
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
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Brent Torrenga
2006-05-24 17:43:32 UTC
Permalink
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?
Post by Clint Sharp
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
Sincerely,

Brent A. 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
Clint Sharp
2006-05-24 18:36:25 UTC
Permalink
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-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of Brent
Torrenga
Sent: Wednesday, May 24, 2006 12:44 PM
To: asterisk-***@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?
Post by Clint Sharp
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
Sincerely,

Brent A. 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
Jerry Jones
2006-05-24 21:57:47 UTC
Permalink
A big disadvantage for a hosted provider is creating manager accounts
for end users. for a single company installation it may not be a big
deal, but SIP TAPI looks much cleaner to me as a service provider.

Of course I have yet to get a good configuration for it working
properly.
Post by Clint Sharp
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-----
Torrenga
Sent: Wednesday, May 24, 2006 12:44 PM
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.
- 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.
[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?
Post by Clint Sharp
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
Sincerely,
Brent A. Torrenga
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
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users
Guido Hecken
2006-05-24 19:39:00 UTC
Permalink
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
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