28th February
I am working with Asterisk 1.2.15. I have configured sip.conf for two soft
phones (I am using Xlite).I have installed the Bluez stack and so far, i
manage to make a phone call from a soft phone to a GSM network. However, i
have an audio problem. The soft phone can be heart by the GSM costumer but
the voice in Xlite is not transmitted to the GSM. In asterisk all i got is
the next lines:
-- Executing Dial("SIP/Jack-081e39b0", "BLT/nokia/07863342772") in new stack
[AG] nokia < ATD07863342772;
-- Called nokia
[AG] nokia > OK
[AG] nokia > +CIEV: 3,2
[AG] nokia > +CIEV: 4,2
[AG] nokia > +CIEV: 3,3
-- BLT/nokia is ringing
[AG] nokia > +CIEV: 4,3
[AG] nokia > +CIEV: 1,1
-- BLT/nokia answered SIP/Jack-081e39b0
Feb 22 14:48:10 WARNING[5473]: /usr/src/bt/chan_bluetooth.c:622 sco_thread:
SCO
thread started on fd 38, pid 5445
[AG] nokia > +CIEV: 3,0
[AG] nokia > +CIEV: 4,0
Feb 22 14:48:20 NOTICE[5470]: rtp.c:576 ast_rtp_read: Unknown RTP codec 126
rece ived
Feb 22 14:48:31 NOTICE[5470]: rtp.c:576 ast_rtp_read: Unknown RTP codec 126
rece ived
Feb 22 14:48:41 NOTICE[5470]: rtp.c:576 ast_rtp_read: Unknown RTP codec 126
rece ived
Feb 22 14:48:52 NOTICE[5470]: rtp.c:576 ast_rtp_read: Unknown RTP codec 126
rece ived
[AG] nokia < ATH
[AG] nokia < AT+CHUP
== Spawn extension (internal, 007863342772, 1) exited non-zero on
'SIP/Jack-08
1e39b0'
Do you know where could be the problem?
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