Hello,
Do your verify, the codecs, of both clients, in your sip.conf?
What codec do you use?
Best Regards
On Thu, Mar 20, 2008 at 12:13 AM, Pete Kay <[EMAIL PROTECTED]> wrote:
> Hi,
> I am sorry my questinos are too fundamental. I am new to Asterisk, and
> hope to catch up as fast as I can.
>
> Problem 1:
>
> I have my SIP client ( in one PC .102) and SIP server ( in another PC
> .101) within the same land. They can make SIP connection, but when the SIP
> client makes call to play an audio file, I can only hear a "beat" sounds,
> and then nothing else. In the console, I can see:
> *CLI> -- Executing [EMAIL PROTECTED]:1] Answer("SIP/2001-081dd6e0", "")
> in new stack
> -- Executing [EMAIL PROTECTED]:2] VoiceMail("SIP/2001-081dd6e0", "2000")
> in new stack
> Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037718, ts
> 000160, len 000160)
> -- <SIP/2001-081dd6e0> Playing 'vm-intro' (language 'en')
> Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037719, ts
> 000320, len 000160)
> Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037720, ts
> 000480, len 000160)
> Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037721, ts
> 000640, len 000160)
> Got RTP packet from 192.168.1.102:8000 (type 00, seq 062222, ts
> 1373137124, len 000160)
> Sent RTP packet to 192.168.1.102:8000 (type 00, seq 037722, ts
> 000800, len 000160)
> Sent RTP packet to 192.168.1.102:8000 (type 00, seq 037723, ts
> 000960, len 000160)
>
> Is it the prolem? First it sends to the public address of the the router,
> then it sends to the virtual IP. Is this the problem that causing my to
> hear just one "beat" sound and then no audio?
>
> Problem 2:
>
> The problem is isolated from Problem 1, cuz I run the SIP client on the
> same machine as the server, so there should not be network problem. I
> recorded some voice mails and they are stored as .wav files ok. When I
> tried to hear back the message, It does not work. Is there any
> configuration that I have to go through to have Asterisk to play .wav file?
>
>
> Thank you very much in advance for all your kind help.
>
> Pete
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users