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Hi!
I am having problems with the PlayTones application and VoIP softphones.
I have the following in my extensions.conf:
exten => 123,1,Answer
exten => 123,2,PlayTones(Busy)
exten => 123,3,Hangup
But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call
just hangs up immediately.
I get the following on the console:
-- Executing Answer("IAX[mrbk@mrbk]/4", "") in new stack
-- Executing Playtones("IAX[mrbk@mrbk]/4", "Busy") in
new stack
-- Executing Hangup("IAX[mrbk@mrbk]/4", "") in new stack
== Spawn extension (icepage, 123, 3) exited non-zero on
'IAX[mrbk@mrbk]/4'
May 29 20:00:10 NOTICE[21526]: channel.c:1478 ast_set_write_format: Unable to
find a path from UNKN to GSM
-- Hungup 'IAX[mrbk@mrbk]/4'
I don't have any FXS-hardware so I can't try it with a real phone.
I am running the (stable) asterisk version from cvs from 3 days ago.
Sincerely,
Bartek
- --
* GPG-Key: http://evil.gnarf.org/mrbk.pgp
A: Because we read from top to bottom, left to right.
Q: Why should i start my reply below the quoted text?
- -- http://www.i-hate-computers.demon.co.uk/
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On Sat, 29 May 2004, Bartek Kania wrote:> I am having problems with the PlayTones application and VoIP softphones. > I have the following in my extensions.conf: > exten => 123,1,Answer > exten => 123,2,PlayTones(Busy) > exten => 123,3,Hangup > > But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call > just hangs up immediately.My extensions.conf always include a "wait" statement after playtones. Try it: exten => 123,1,Answer exten => 123,2,PlayTones(Busy) exten => 123,3,Wait(2) ; play busy tone for 2 seconds exten => 123,4,Hangup
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Sat, 29 May 2004, Hermann Wecke wrote:> On Sat, 29 May 2004, Bartek Kania wrote: > > I am having problems with the PlayTones application and VoIP softphones. > > But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call > > just hangs up immediately. > My extensions.conf always include a "wait" statement after playtones. Try > it: > exten => 123,1,Answer > exten => 123,2,PlayTones(Busy) > exten => 123,3,Wait(2) ; play busy tone for 2 seconds > exten => 123,4,HangupThanks a bunch! I feel a little stupid for missing something like this though =) But I still get the following message on the console: May 29 21:01:58 NOTICE[22550]: channel.c:1478 ast_set_write_format: Unable to find a path from UNKN to GSM Can I just ignore it? /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFAuN6hWYjaxM2wIe4RAjqKAKCM37PIM+7Bb1OdOJIzOsWfocA0nACgncga dPgH1FX+jHVW8S67fM9jpVU=tFbW -----END PGP SIGNATURE-----
yep use exten => 123,1,Answer exten => 123,2,Busy bkw ----- Original Message ----- From: "Bartek Kania" <mrbk@gnarf.org> To: <asterisk-users@lists.digium.com> Sent: Saturday, May 29, 2004 12:04 PM Subject: [Asterisk-Users] PlayTones problem> -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hi! > I am having problems with the PlayTones application and VoIP softphones. > I have the following in my extensions.conf: > exten => 123,1,Answer > exten => 123,2,PlayTones(Busy) > exten => 123,3,Hangup > > But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call > just hangs up immediately. > I get the following on the console: > -- Executing Answer("IAX[mrbk@mrbk]/4", "") in new stack > -- Executing Playtones("IAX[mrbk@mrbk]/4", "Busy") in new stack > -- Executing Hangup("IAX[mrbk@mrbk]/4", "") in new stack > == Spawn extension (icepage, 123, 3) exited non-zero on'IAX[mrbk@mrbk]/4'> May 29 20:00:10 NOTICE[21526]: channel.c:1478 ast_set_write_format: Unableto find a path from UNKN to GSM> -- Hungup 'IAX[mrbk@mrbk]/4' > > I don't have any FXS-hardware so I can't try it with a real phone. > > I am running the (stable) asterisk version from cvs from 3 days ago. > > Sincerely, > Bartek > - -- > * GPG-Key: http://evil.gnarf.org/mrbk.pgp > > A: Because we read from top to bottom, left to right. > Q: Why should i start my reply below the quoted text? > - -- http://www.i-hate-computers.demon.co.uk/ > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.2.4 (GNU/Linux) > > iD8DBQFAuNDJWYjaxM2wIe4RAtS1AKCwMqmaKILwzLg9ZnKx0+uDEw5drwCdFqqv > vUBt3kLL7jVDsnWVrKYGr9w> =P8PB > -----END PGP SIGNATURE----- > _______________________________________________ > 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:
I have an Asterisk server (version 0.9.0) working with a Digium
E100P card that makes an EuroISDN connection with a Siemens Hicom 300 PBX.
Hicom 300 is pri_net and Asterisk is pri_cpe.
Furthermore, makes an OH323 connection with a GNU gatekeeper
When you make a call between a Hicom 300 extension an IP phone its
generates succefully, but a Hicom 300 extension still hear a ringback when
an IP phone
pick up.
My extensions.conf includes:
exten => _30XX,1,Playtones(ring)
exten => _30XX,2,Dial,OH323/${EXTEN}
exten => _30XX,3,StopPlaytones
but StopPlaytones never executes.
If use Ringing instead Playtones(ring), a Hicom 300 extension can`t hear a
ringback sound
when a call progress.
Could you give me some hints?
Thanks.
Rafael Olivieri.
Mayor Rafael Mario Olivieri
Comando de Comunicaciones e Inform?tica
Dpto Comunicaciones - Jefe Div C4
4346-6137
4346-6100 int 6137
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