-------- Original Message -------- Subject: Re: [Asterisk-Users] Troubles with Kphone Date: Tue, 25 May 2004 15:44:15 +0530 From: Murali Krishnan <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] Organization: bk SYSTEMS (P) LTD., To: [EMAIL PROTECTED] References: <[EMAIL PROTECTED]>
enano wrote:
Hi ,
I'm triying to use kphone 4.02, but when i'm make a call the programs doesn't respond any command, so i can't hear any sound ..
in sip.conf that's my codec config:
disallow=all allow=gsm
allow=ulaw allow=ilbc
and the kphone give the follow : SipClient: Sending: 06:46:28.116
--------------------------------
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2;rport
CSeq: 6121 ACK
To: <sip:[EMAIL PROTECTED]>;tag=as12aab0bf
From: "ivan2" <sip:[EMAIL PROTECTED]>;tag=7F6911ED
Call-ID: [EMAIL PROTECTED]
Content-Length: 0
User-Agent: kphone/4.0.2
Contact: "ivan2" <sip:[EMAIL PROTECTED];transport=udp>
res_search: NO result ! res_search: NO result ! SipClient: Sending to '192.168.0.3:5060' SipCallMember: localStatusUpdated: 200 CallAudio: Using GSM for output CallAudio: Sending to remote site 192.168.0.3:19696 UDPMessageSocket::SetTOS: Operation not permitted CallAudio: OSS device already open (readwrite)
anyone can help me ??
thanks
Ivan
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Check the following things.
1. Make sure your sound card is configured properly for record/playback - if not, do it with either kmix and test with gnome-sound-recorder 2. Make sure your identity is configured in sip.conf and extension.conf correctly 3. Make sure kphone is registered with Asterisk File->Identity - see whether 'Unregister' is there, (means you are registered ) 4. Watch for Asterisk Messages for any clue. ( asterisk -vvvvvc ) 5. Make sure the destination extension you are dialing from kphone has proper dialplan sequence in extension.conf 6. If you have OSS sound configuration, immediately switch to ALSA. - visit alsa-project.org and search docs for your card type. Compile and install the packages. ( this OSS would be the major headache if you are not getting sound ).
If you are registered with Asterisk and your sound card is proper, and you configured your destination extension routing properly in extension.conf everything should work fine.
Get back with success.
Regards Murali Krishnan.
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