search for: udpmessagesocket

...sks? Thank you a lot for any help! -- Best regards, Timur Elzhov -------------- next part -------------- $ kphone & [1] 29730 $ Found 1 interfaces. SipClient: Listening UDP on port: 5062 SipClient: Our address: 127.0.0.1 KCallWidget: Switching calls... CallAudio: listening for incomming RTP UDPMessageSocket: Listening on 32809 UDPMessageSocket: Retrying... UDPMessageSocket: Listening on 32810 CallAudio: Opening OSS device /dev/dsp for Input and Output ERROR: Open Failed ** audioOut: openDevice Failed. CallAudio: Creating RTP->OSS Diverter SipClient: Sending: 11:22:24.494 --------------------------...

.../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

...t;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.c...

Hi all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem