-------- Original Message -------- Subject: Re: [Asterisk-Users] Troubles with Kphone Date: Tue, 25 May 2004 15:44:15 +0530 From: Murali Krishnan <murali@bksys.co.in> Reply-To: ismk@myrealbox.com Organization: bk SYSTEMS (P) LTD., To: asterisk-users@lists.digium.com References: <200405250652.46370.klky3@fibertel.com.ar> 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:500@192.168.0.3 SIP/2.0 >Via: SIP/2.0/UDP 192.168.0.2;rport >CSeq: 6121 ACK >To: <sip:500@192.168.0.3>;tag=as12aab0bf >From: "ivan2" <sip:ivan2@192.168.0.3>;tag=7F6911ED >Call-ID: 155660827@192.168.0.2 >Content-Length: 0 >User-Agent: kphone/4.0.2 >Contact: "ivan2" <sip:ivan2@192.168.0.2;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 >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 > >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.
klky3@fibertel.com.ar
2004-May-26 01:05 UTC
Asunto: Re: [Asterisk-Users] Troubles with Kphone]
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 <ismk@myrealbox.com> >To: asterisk-users@lists.digium.com >Subject: Re: [Asterisk-Users] Troubles with Kphone] >Reply-To: asterisk-users@lists.digium.com >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 <murali@bksys.co.in> >Reply-To: ismk@myrealbox.com >Organization: bk SYSTEMS (P) LTD., >To: asterisk-users@lists.digium.com >References: <200405250652.46370.klky3@fibertel.com.ar> > >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:500@192.168.0.3 SIP/2.0 >>Via: SIP/2.0/UDP 192.168.0.2;rport >>CSeq: 6121 ACK >>To: <sip:500@192.168.0.3>;tag=as12aab0bf >>From: "ivan2" <sip:ivan2@192.168.0.3>;tag=7F6911ED >>Call-ID: 155660827@192.168.0.2 >>Content-Length: 0 >>User-Agent: kphone/4.0.2 >>Contact: "ivan2" <sip:ivan2@192.168.0.2;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 >>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 >> >> >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. Compileand> 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, andyou>configured your destination extension routing properly in extension.conf >everything should work fine. > >Get back with success. > >Regards >Murali Krishnan. > > >_______________________________________________ >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________________________________________ FiberTel, el nombre de la banda ancha http://www.fibertel.com.ar