[Asterisk-Users] CAPI call transfer
Roland Zagler
laureen at laureen.at
Tue Aug 10 05:13:04 MST 2004
Have you tried removing "${CALLERIDNUM}" from your 1st line in context
[SIP] in extensions.conf?
Is your ISDN Line configured to transfer the Extensions to you
(Provider-dependent)?
And try to put "Answer" before calling to CAPI!
I do it like this:
[MyContext1]
exten => _.,1,Answer
exten => _.,2,Dial,CAPI/50:b${EXTEN},60
exten => _.,100,Hangup
Roland Zagler
mailto:laureen at laureen.at
mobile:4369910713694
-----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 2:03 PM
To: asterisk-users at lists.digium.com
Subject: Re: RE: RE: RE: RE: [Asterisk-Users] CAPI call transfer
My extensions.conf is:
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface
for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel
username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
TRUNK=capi
;TRUNK=IAX2/user:pass at provider
[SIP]
exten => _.,1,Dial,CAPI/01824708${CALLERIDNUM}:b${EXTEN}
exten => _.,2,congestion
exten => _.,3,hangup
My sip.conf is:
[general]
context=default
autocreatepeer=yes
localnet=192.168.1.162
port=5062
bindaddr=0.0.0.0
rtptimeout=60
rtpholdtimeout=300
useragent=PBX Gateway
[sip_proxy]
context=SIP
type=peer
Host=192.168.1.162
Thanks and best regards.
Jonathan
-------- Original Message --------
==> From: "Roland Zagler" <laureen at laureen.at> ==> Date: Tue, 10 Aug
2004 13:35:32 0200
Can you post your extensions.conf, maybe i can find something!
Roland Zagler mailto:laureen at laureen.at mobile:4369910713694
-----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 1:30 PM To:
asterisk-users at lists.digium.com Subject: Re: RE: RE: RE:
[Asterisk-Users] CAPI call transfer
Hi Roland,
Nothing on that message helps me unfortunately. I can make calls from
SIP to ISDN I just can't get call transfer to work.
Regards, Jonathan
-------- Original Message --------
==> From: "Roland Zagler" <laureen at laureen.at> ==> Date: Tue, 10 Aug
2004 12:21:29 0200
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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