Asunto: Re: [Asterisk-Users] Troubles with Kphone]

[klky3 at fibertel.com.ar]

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 at myrealbox.com>
>To: asterisk-users at lists.digium.com
>Subject: Re: [Asterisk-Users] Troubles with Kphone]
>Reply-To: asterisk-users at 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 at bksys.co.in>
>Reply-To: ismk at myrealbox.com
>Organization: bk SYSTEMS (P) LTD.,
>To: asterisk-users at lists.digium.com
>References: <200405250652.46370.klky3 at 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 at 192.168.0.3 SIP/2.0
>>Via: SIP/2.0/UDP 192.168.0.2;rport
>>CSeq: 6121 ACK
>>To: <sip:500 at 192.168.0.3>;tag=as12aab0bf
>>From: "ivan2" <sip:ivan2 at 192.168.0.3>;tag=7F6911ED
>>Call-ID: 155660827 at 192.168.0.2
>>Content-Length: 0
>>User-Agent: kphone/4.0.2
>>Contact: "ivan2" <sip:ivan2 at 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 at 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.
>
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
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>   http://lists.digium.com/mailman/listinfo/asterisk-users


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