[Asterisk-Users] CAPI call transfer

[Roland Zagler]

Here's the post i used to get this thing going, maybe it helps:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg41324.htm
l


Roland Zagler
mailto:laureen at laureen.at

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 12:16 PM
To: asterisk-users at lists.digium.com
Subject: Re: RE: RE: [Asterisk-Users] CAPI call transfer


No I'm on kernal 2.4.22 Fedora core 1.

Thanks, Jonathan

-------- Original Message --------

==> From: "Roland Zagler" <laureen at laureen.at> ==> Date: Tue, 10 Aug
2004 11:48:17  0200

Are you using kernel 2.6.x ?

 Roland Zagler mailto:laureen at laureen.at

-----Original Message----- From:
asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:36 AM To:
asterisk-users at lists.digium.com Subject: Re: RE: [Asterisk-Users] CAPI
call transfer

 Thanks for your reply Roland, unfortunately adding the 'b' didn't make
any diference.

Regards. Jonathan



-------- Original Message --------

==> From: "Roland Zagler" <laureen at laureen.at> ==> Date: Tue, 10 Aug
2004 11:18:32  0200

Try specifying your number you want to dial with "b" in front of, e.g.
"Dial(CAPI/01824708169:b01824708752,60)" in your extensions.conf!

Regards, roland 

 Roland Zagler mailto:laureen at laureen.at -----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:12 AM To:
asterisk-users at lists.digium.com Subject: [Asterisk-Users] CAPI call
transfer

Hi,

I am having trouble configuring CAPI so that call transfers work. I make
a SIP call to asterisk which goes out on ISDn via CAPI.  Then I try to
do a transfer from the SIP phone which doesn't work and results in the
call being disconnected.

The error message given by asterisk is that it chan_capi can't find an
entry for the outgoing msn for the transfer however the outgoing msn is
the same as that used to make the original call.

Has anyone got any ideas please?

The asterisk trace and my capi.conf are below:

Thank you. Best regards. Jonathan 

    -- Executing Dial("SIP/192.168.1.162-08186af8",
"CAPI/01824708169:01824708752") in new stack -- creating pipe for
PLCI=-1 > sent CONNECT_REQ MN =0x9ee -- Called
01824708169:01824708752
-- CAPI[contr1/01824708169]/11 is making progress passing it to
SIP/192.168.1.162-08186af8 -- CAPI[contr1/01824708169]/11 is ringing
-- CAPI[contr1/01824708169]/11 answered SIP/192.168.1.162-08186af8 ==
Spawn extension (SIP, 01824708752, 1) exited non-zero on
'SIP/192.168.1.162-08186af8' -- Executing
Dial("SIP/192.168.1.162-08186af8", "CAPI/01824708169:h") in new stack
-- creating pipe for PLCI=-1 > sent CONNECT_REQ MN =0xa5c -- Called
01824708169:h -- Executing Dial("CAPI[contr1/01824708169]/11",
"CAPI/01824708169:170") in new stack Aug 10 09:55:29 NOTICE[442386]:
chan_capi.c:1172 capi_request: didn't find capi device with outgoing msn
= 01824708169. you should check your config! Aug 10 09:55:29
NOTICE[442386]: app_dial.c:706 dial_exec: Unable to create channel of
type 'CAPI' == Everyone is busy/congested at this time -- Executing
Congestion("CAPI[contr1/01824708169]/11", "") in new stack -- CAPI
Hangingup > sent DISCONNECT_REQ PLCI=0x201 -- removed pipe for PLCI =
0x201 == Spawn extension (SIP, h, 1) exited non-zero on
'SIP/192.168.1.162-08186af8'

capi.conf [general] nationalprefix=0 internationalprefix=44
rxgain=0.8
txgain=0.8

[interfaces]

incomingmsn=* softdtmf=1

mode=immediate isdnmode=ptp msn=01824708,01824708169 controller=1
devices=2

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