[Asterisk-Users] No sound with kphone 4.05 on SuSE 8.2 and asterisk
Volker Jahns
volker at thalreit.de
Wed Nov 10 13:57:54 MST 2004
Problem to get kphone 4.05 working w/ SuSE 8.2
----------------------------------------------
I amtrying to get asterisk running but I do get stuck at the first steps.
asterisk installation OK.
UA kphone 3.13 (on SuSE 9.1 system)
UA kphone 4.05 (on SuSE 8.2 system)
When connecting to the asteriask server with kphone both systems show up
on the CLI :
--
*CLI> sip show peers
Name/username Host Dyn Nat ACL Mask Port Status
2000/marlies 10.XX.YY.ZZ D 255.255.255.255 5060 OK (18 ms)
1000/1000 10.XX.YY.PP D 255.255.255.255 5060 OK (21 ms)
--
When trying to make connection between the 2 systems ringing is OK, but
once the connection is coming up, there is _no_ sound.
--
*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Format
10.XX.YY.ZZ marlies 6ee8c36c665 00104/00000 ULAW
10.XX.YY.PP 1000 885236207 at 1 00103/01321 ULAW
2 active SIP channel(s)
--
n the SuSE 8.2 machine kphone (4.05) murmurs the following stuff:
--
t=0 0
m=audio 32772 RTP/AVP 0 97 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
CallAudio: Using G711u for output
CallAudio: Sending to remote site 10.XX.YY.ZZ:32772
UDPMessageSocket::SetTOS: Operation not permitted
CallAudio: Opening OSS device /dev/dsp for Input
CallAudio: Creating OSS->RTP Diverter
CallAudio: Sending to remote site 10.XX.YY.ZZ:32772
UDPMessageSocket::SetTOS: Operation not permitted
CallAudio: Opening OSS device /dev/dsp for Input
CallAudio: Creating OSS->RTP Diverter
Attempt to start a thread already running
--
+ the application evidently hangs: Thw windows don't get redrawn,
asterisk claims this client to be UNREACHABLE.
( The kphone rpm which is presented by SuSE will not come up as there
is a symbol from a qt library which is missing. So I had to compile
from src rpm. )
What can I do?
--
Volker Jahns, Volker.Jahns at thalreit.de, http://thalreit.de, DG7PM
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