Well .. I'm now using Kphone 3.11 and alsa and everithing looks good.. but when i dial an extension i only hear and horrible ticking sound ... like a burned dial up modem ... i can see how the call initiates, and finishes in the console ..
thanks for all Ivan >-- Mensaje original -- >From: Murali Krishnan <[EMAIL PROTECTED]> >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] Troubles with Kphone] >Reply-To: [EMAIL PROTECTED] >Date: Tue, 25 May 2004 16:14:11 +0530 > > > > >-------- 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 >> >> >> >> >>_______________________________________________ >>Asterisk-Users mailing list >>[EMAIL PROTECTED] >>http://lists.digium.com/mailman/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >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. > > >_______________________________________________ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ________________________________________ FiberTel, el nombre de la banda ancha http://www.fibertel.com.ar _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
