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Re: [Linphone-users] No sound under 0.12.1?
From: |
Micah Anderson |
Subject: |
Re: [Linphone-users] No sound under 0.12.1? |
Date: |
Wed, 10 Dec 2003 19:35:02 -0600 |
Never got this resolved, does anyone have any suggestions?
Micah
Micah Anderson schrieb am Saturday, den 22. November 2003:
> I've just downloaded and compiled the newest release, everything went
> fine, but when I try and call somewhere, I get no sound at all. I
> attach a log of the terminal from the start of linphone, thanks for
> any help!
>
>
> Script started on Sat Nov 22 00:05:32 2003
> | INFO1 | <osipua.c: 65> Starting osip stack and osipua layer.
>
> MediaStreamer-Message: Found /dev/dsp.
> MediaStreamer-Message: Found ALSA device: Sound Blaster 16
> | INFO1 | <osipmanager.c: 148> port already listened
>
> | INFO1 | <udp.c: 112> Entering osipua thread.
>
> | INFO1 | <utils.c: 409> Outgoing interface to reach fwd.pulver.com is
> 10.0.0.10.
>
> | ERROR | <osipdialog.c: 1681> generating_request_out_of_dialog: setting
> ua->ua_family=2 from localip 10.0.0.10
>
> | INFO1 | <utils.c: 92> info: Name resolution requested.
>
> | INFO1 | <resolver.c: 109> Entering name resolution thread.
>
> | INFO1 | <osipmanager.c: 148> port already listened
>
> | INFO1 | <resolver.c: 115> Doing asynchronous name resolution.
>
> | INFO1 | <udp.c: 295> Sending message:
> REGISTER sip:fwd.pulver.com SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.10:5060;branch=z9hG4bK2505696190
> From: <sip:address@hidden>;tag=4161034811
> To: <sip:address@hidden>;tag=4161034811
> Call-ID: address@hidden
> CSeq: 0 REGISTER
> Contact: <sip:address@hidden>
> max-forwards: 10
> expires: 900
> user-agent: oSIP/Linphone-0.12.1
> Content-Length: 0
>
>
> | INFO1 | <udp.c: 186> info: Message from 192.246.69.223:5060
>
> | INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> 10.0.0.10:5060;branch=z9hG4bK2505696190;received=68.73.117.216
> From: <sip:address@hidden>;tag=4161034811
> To: <sip:address@hidden>;tag=4161034811
> Call-ID: address@hidden
> CSeq: 0 REGISTER
> WWW-Authenticate: Digest realm="fwd.pulver.com",
> nonce="3fbeff319fa77fc4249d09f3687eadb0b360078a"
> Server: Free World Dialup (0.8.11rc3 (i386/linux))
> Content-Length: 0
>
>
>
> | INFO1 | <nict_callbacks.c: 107> OnEvent_New_Incoming4xxResponse!
>
> | INFO1 | <nict_callbacks.c: 127> User need to authenticate to REGISTER!
>
> | INFO1 | <osipdialog.c: 48> ua needs specific binding on 10.0.0.1
>
> | ERROR | <osipdialog.c: 1681> generating_request_out_of_dialog: setting
> ua->ua_family=2 from localip 10.0.0.1
>
> | INFO1 | <authentication.c: 269> Response in authorization
> |0d14592605b6daaadb50d1dcc58cdd27|
>
> | INFO1 | <utils.c: 92> info: Name resolution requested.
>
> | INFO1 | <resolver.c: 115> Doing asynchronous name resolution.
>
> | INFO1 | <udp.c: 295> Sending message:
> REGISTER sip:fwd.pulver.com SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK2663927980
> From: <sip:address@hidden>;tag=2006411513
> To: <sip:address@hidden>;tag=2006411513
> Call-ID: address@hidden
> CSeq: 1 REGISTER
> Contact: <sip:address@hidden>
> Authorization: Digest username="46333", realm="fwd.pulver.com",
> nonce="3fbeff319fa77fc4249d09f3687eadb0b360078a", uri="sip:fwd.pulver.com",
> response="0d14592605b6daaadb50d1dcc58cdd27", algorithm=MD5
> max-forwards: 10
> expires: 900
> user-agent: oSIP/Linphone-0.12.1
> Content-Length: 0
>
>
> | INFO1 | <udp.c: 186> info: Message from 192.246.69.223:5060
>
> | INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
> SIP/2.0 200 That is ugly -- real IP next time
> Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK2663927980;received=68.73.117.216
> From: <sip:address@hidden>;tag=2006411513
> To: <sip:address@hidden>;tag=2006411513
> Call-ID: address@hidden
> CSeq: 1 REGISTER
> Server: Free World Dialup (0.8.11rc3 (i386/linux))
> Content-Length: 0
>
>
>
> | INFO1 | <nict_callbacks.c: 30> Transaction 1 killed.
>
> | INFO1 | <nict_callbacks.c: 30> Transaction 2 killed.
>
> | INFO1 | <osipdialog.c: 1915> Dialog is removed. It remains 0 dialog(s) in
> the ua list.
>
> | INFO1 | <osipdialog.c: 48> ua needs specific binding on 10.0.0.1
>
> | ERROR | <osipdialog.c: 1681> generating_request_out_of_dialog: setting
> ua->ua_family=2 from localip 10.0.0.1
>
> | INFO1 | <utils.c: 92> info: Name resolution requested.
>
> | INFO1 | <resolver.c: 115> Doing asynchronous name resolution.
>
> | INFO1 | <udp.c: 295> Sending message:
> INVITE sip:address@hidden SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK1111456789
> From: <sip:address@hidden>;tag=2006411513;tag=3739115415
> To: <sip:address@hidden>
> Call-ID: address@hidden
> CSeq: 20 INVITE
> Contact: <sip:address@hidden>
> max-forwards: 10
> user-agent: oSIP/Linphone-0.12.1
> Content-Type: application/sdp
> Content-Length: 362
>
> v=0
> o=46333 123456 654321 IN IP4 10.0.0.1
> s=A conversation
> c=IN IP4 10.0.0.1
> t=0 0
> m=audio 7078 RTP/AVP 0 3 8 110 111 115 101
> b=AS:110 20
> b=AS:111 28
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:3 GSM/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:110 speex/8000/1
> a=rtpmap:111 speex/16000/1
> a=rtpmap:115 1015/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
>
> | INFO1 | <udp.c: 186> info: Message from 192.246.69.223:5060
>
> | INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK1111456789;received=68.73.117.216
> From: <sip:address@hidden>;tag=2006411513;tag=3739115415
> To: <sip:address@hidden>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6db
> Call-ID: address@hidden
> CSeq: 20 INVITE
> Proxy-Authenticate: Digest realm="fwd.pulver.com",
> nonce="3fbeff40bd69d38b84587ea4f62d2fe26418e8ca"
> Server: Free World Dialup (0.8.11rc3 (i386/linux))
> Content-Length: 0
>
>
>
> | INFO1 | <udp.c: 295> Sending message:
> ACK sip:address@hidden SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK1111456789
> From: <sip:address@hidden>;tag=2006411513;tag=3739115415
> To: <sip:address@hidden>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6db
> Call-ID: address@hidden
> CSeq: 20 ACK
> Content-Length: 0
>
>
> | INFO1 | <ict_callbacks.c: 233> OnEvent_New_Incoming4xxResponse!
>
> | INFO1 | <ict_callbacks.c: 261> User need to authenticate to INVITE!
>
> | INFO1 | <authentication.c: 377> Response in proxy_authorization
> |e8ec2e2cd7ca2d0ce9c4ae9e61d0322b|
>
> | INFO1 | <utils.c: 92> info: Name resolution requested.
>
> | INFO1 | <resolver.c: 115> Doing asynchronous name resolution.
>
> | INFO1 | <udp.c: 295> Sending message:
> INVITE sip:address@hidden SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK3642497320
> From: <sip:address@hidden>;tag=2006411513;tag=3739115415
> To: <sip:address@hidden>
> Call-ID: address@hidden
> CSeq: 21 INVITE
> Contact: <sip:address@hidden>
> Proxy-Authorization: Digest username="46333", realm="fwd.pulver.com",
> nonce="3fbeff40bd69d38b84587ea4f62d2fe26418e8ca", uri="sip:address@hidden",
> response="e8ec2e2cd7ca2d0ce9c4ae9e61d0322b", algorithm=MD5
> max-forwards: 10
> user-agent: oSIP/Linphone-0.12.1
> Content-Type: application/sdp
> Content-Length: 362
>
> v=0
> o=46333 123456 654321 IN IP4 10.0.0.1
> s=A conversation
> c=IN IP4 10.0.0.1
> t=0 0
> m=audio 7078 RTP/AVP 0 3 8 110 111 115 101
> b=AS:110 20
> b=AS:111 28
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:3 GSM/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:110 speex/8000/1
> a=rtpmap:111 speex/16000/1
> a=rtpmap:115 1015/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
>
> | INFO1 | <udp.c: 186> info: Message from 192.246.69.223:5060
>
> | INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
> SIP/2.0 100 trying -- your call is important to us
> Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK3642497320;received=68.73.117.216
> From: <sip:address@hidden>;tag=2006411513;tag=3739115415
> To: <sip:address@hidden>
> Call-ID: address@hidden
> CSeq: 21 INVITE
> Server: Free World Dialup (0.8.11rc3 (i386/linux))
> Content-Length: 0
>
>
>
> | INFO1 | <udp.c: 295> Sending message:
> INVITE sip:address@hidden SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK3642497320
> From: <sip:address@hidden>;tag=2006411513;tag=3739115415
> To: <sip:address@hidden>
> Call-ID: address@hidden
> CSeq: 21 INVITE
> Contact: <sip:address@hidden>
> Proxy-Authorization: Digest username="46333", realm="fwd.pulver.com",
> nonce="3fbeff40bd69d38b84587ea4f62d2fe26418e8ca", uri="sip:address@hidden",
> response="e8ec2e2cd7ca2d0ce9c4ae9e61d0322b", algorithm=MD5
> max-forwards: 10
> user-agent: oSIP/Linphone-0.12.1
> Content-Type: application/sdp
> Content-Length: 362
>
> v=0
> o=46333 123456 654321 IN IP4 10.0.0.1
> s=A conversation
> c=IN IP4 10.0.0.1
> t=0 0
> m=audio 7078 RTP/AVP 0 3 8 110 111 115 101
> b=AS:110 20
> b=AS:111 28
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:3 GSM/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:110 speex/8000/1
> a=rtpmap:111 speex/16000/1
> a=rtpmap:115 1015/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
>
> | INFO1 | <ict_callbacks.c: 41> OnEvent_New_Incoming1xxResponse!
>
> | INFO1 | <udp.c: 186> info: Message from 192.246.69.223:5060
>
> | INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.0.0.1:5060;received=68.73.117.216;branch=z9hG4bK3642497320
> Record-Route: <sip:address@hidden;ftag=3739115415;lr=on>
> From: <sip:address@hidden>;tag=2006411513;tag=3739115415
> To: <sip:address@hidden>;tag=as415b5758
> Call-ID: address@hidden
> CSeq: 21 INVITE
> User-Agent: Asterisk PBX
> Contact: <sip:address@hidden:5028>
> Content-Type: application/sdp
> Content-Length: 238
>
> v=0
> o=root 23668 23668 IN IP4 65.39.205.112
> s=session
> c=IN IP4 65.39.205.112
> t=0 0
> m=audio 10122 RTP/AVP 3 0 8 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
>
>
> | INFO1 | <ict_callbacks.c: 71> OnEvent_New_Incoming2xxResponse!
>
> | INFO1 | <ict_callbacks.c: 122> Found body application/sdp
>
> MediaStreamer-Message: ms_filter_add_link: OssRead,0 -> GSMEncoder,0
> MediaStreamer-Message: ms_filter_add_link: GSMEncoder,0 -> RTPSend,0
> MediaStreamer-Message: ms_filter_add_link: RTPRecv,0 -> GSMDecoder,0
> MediaStreamer-Message: ms_filter_add_link: GSMDecoder,0 -> OssWrite,0
> MediaStreamer-Message: Opening sound card in capture mode with
> stereo=0,rate=8000,bits=16
> MediaStreamer-Message: dsp blocksize is 512.
> MediaStreamer-Message: Opening sound card in playback mode with
> stereo=0,rate=8000,bits=16
> | INFO1 | <udp.c: 295> Sending message:
> ACK sip:address@hidden:5028 SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK2390809715
> Route: <sip:address@hidden;ftag=3739115415;lr=on>
> From: <sip:address@hidden>;tag=2006411513;tag=3739115415
> To: <sip:address@hidden>;tag=as415b5758
> Call-ID: address@hidden
> CSeq: 21 ACK
> max-forwards: 10
> user-agent: oSIP/Linphone-0.12.1
> Content-Length: 0
>
>
> | INFO1 | <ict_callbacks.c: 30> Transaction 4 killed.
>
> | INFO1 | <udp.c: 186> info: Message from 192.246.69.223:5060
>
> | INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.0.0.1:5060;received=68.73.117.216;branch=z9hG4bK3642497320
> Record-Route: <sip:address@hidden;ftag=3739115415;lr=on>
> From: <sip:address@hidden>;tag=2006411513;tag=3739115415
> To: <sip:address@hidden>;tag=as415b5758
> Call-ID: address@hidden
> CSeq: 21 INVITE
> User-Agent: Asterisk PBX
> Contact: <sip:address@hidden:5028>
> Content-Type: application/sdp
> Content-Length: 238
>
> v=0
> o=root 23668 23668 IN IP4 65.39.205.112
> s=session
> c=IN IP4 65.39.205.112
> t=0 0
> m=audio 10122 RTP/AVP 3 0 8 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
>
>
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
> | INFO1 | <ict_callbacks.c: 30> Transaction 3 killed.
>
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
>
> (linphone:16026): oRTP-WARNING **: Error sending rtp packet: Connection
> refused.
> MediaStreamer-Message: Mediastreamer processing thread is exiting.
> oRTP-stats-Message:
> Global statistics :
> packet_sent=2875
> sent=129375 bytes
> packet_recv=0
> hw_recv=0 bytes
> recv=0 bytes
> unavaillable=2876 bytes
> outoftime=0
> bad=0
> discarded=0
>
> | INFO1 | <utils.c: 92> info: Name resolution requested.
>
> | INFO1 | <resolver.c: 115> Doing asynchronous name resolution.
>
> | INFO1 | <udp.c: 295> Sending message:
> BYE sip:address@hidden:5028 SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK1686450358
> Route: <sip:address@hidden;ftag=3739115415;lr=on>
> From: <sip:address@hidden>;tag=2006411513;tag=3739115415
> To: <sip:address@hidden>;tag=as415b5758
> Call-ID: address@hidden
> CSeq: 22 BYE
> max-forwards: 10
> user-agent: oSIP/Linphone-0.12.1
> Content-Length: 0
>
>
> | INFO1 | <udp.c: 186> info: Message from 192.246.69.223:5060
>
> | INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.0.0.1:5060;received=68.73.117.216;branch=z9hG4bK1686450358
> Record-Route: <sip:address@hidden;ftag=3739115415;lr=on>
> From: <sip:address@hidden>;tag=2006411513;tag=3739115415
> To: <sip:address@hidden>;tag=as415b5758
> Call-ID: address@hidden
> CSeq: 22 BYE
> User-Agent: Asterisk PBX
> Contact:
> Content-Length: 0
>
>
>
> | ERROR | <msg_parser.c: 316> Could not set header: "contact"
> | ERROR | <msg_parser.c: 532> End of header Not found
> | ERROR | <msg_parser.c: 712> error in msg_headers_parse()
> | ERROR | <sipevent.c: 76> could not parse message
> | INFO1 | <udp.c: 295> Sending message:
> BYE sip:address@hidden:5028 SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK1686450358
> Route: <sip:address@hidden;ftag=3739115415;lr=on>
> From: <sip:address@hidden>;tag=2006411513;tag=3739115415
> To: <sip:address@hidden>;tag=as415b5758
> Call-ID: address@hidden
> CSeq: 22 BYE
> max-forwards: 10
> user-agent: oSIP/Linphone-0.12.1
> Content-Length: 0
>
>
> | INFO1 | <udp.c: 186> info: Message from 192.246.69.223:5060
>
> | INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.0.0.1:5060;received=68.73.117.216;branch=z9hG4bK1686450358
> Record-Route: <sip:address@hidden;ftag=3739115415;lr=on>
> From: <sip:address@hidden>;tag=2006411513;tag=3739115415
> To: <sip:address@hidden>;tag=as415b5758
> Call-ID: address@hidden
> CSeq: 22 BYE
> User-Agent: Asterisk PBX
> Contact:
> Content-Length: 0
>
>
>
> | ERROR | <msg_parser.c: 316> Could not set header: "contact"
> | ERROR | <msg_parser.c: 532> End of header Not found
> | ERROR | <msg_parser.c: 712> error in msg_headers_parse()
> | ERROR | <sipevent.c: 76> could not parse message
> | INFO1 | <udp.c: 295> Sending message:
> BYE sip:address@hidden:5028 SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK1686450358
> Route: <sip:address@hidden;ftag=3739115415;lr=on>
> From: <sip:address@hidden>;tag=2006411513;tag=3739115415
> To: <sip:address@hidden>;tag=as415b5758
> Call-ID: address@hidden
> CSeq: 22 BYE
> max-forwards: 10
> user-agent: oSIP/Linphone-0.12.1
> Content-Length: 0
>
>
> | INFO1 | <udp.c: 186> info: Message from 192.246.69.223:5060
>
> | INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.0.0.1:5060;received=68.73.117.216;branch=z9hG4bK1686450358
> Record-Route: <sip:address@hidden;ftag=3739115415;lr=on>
> From: <sip:address@hidden>;tag=2006411513;tag=3739115415
> To: <sip:address@hidden>;tag=as415b5758
> Call-ID: address@hidden
> CSeq: 22 BYE
> User-Agent: Asterisk PBX
> Contact:
> Content-Length: 0
>
>
>
> | ERROR | <msg_parser.c: 316> Could not set header: "contact"
> | ERROR | <msg_parser.c: 532> End of header Not found
> | ERROR | <msg_parser.c: 712> error in msg_headers_parse()
> | ERROR | <sipevent.c: 76> could not parse message
> | INFO1 | <udp.c: 295> Sending message:
> BYE sip:address@hidden:5028 SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK1686450358
> Route: <sip:address@hidden;ftag=3739115415;lr=on>
> From: <sip:address@hidden>;tag=2006411513;tag=3739115415
> To: <sip:address@hidden>;tag=as415b5758
> Call-ID: address@hidden
> CSeq: 22 BYE
> max-forwards: 10
> user-agent: oSIP/Linphone-0.12.1
> Content-Length: 0
>
>
> | INFO1 | <udp.c: 186> info: Message from 192.246.69.223:5060
>
> | INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.0.0.1:5060;received=68.73.117.216;branch=z9hG4bK1686450358
> Record-Route: <sip:address@hidden;ftag=3739115415;lr=on>
> From: <sip:address@hidden>;tag=2006411513;tag=3739115415
> To: <sip:address@hidden>;tag=as415b5758
> Call-ID: address@hidden
> CSeq: 22 BYE
> User-Agent: Asterisk PBX
> Contact:
> Content-Length: 0
>
>
>
> | ERROR | <msg_parser.c: 316> Could not set header: "contact"
> | ERROR | <msg_parser.c: 532> End of header Not found
> | ERROR | <msg_parser.c: 712> error in msg_headers_parse()
> | ERROR | <sipevent.c: 76> could not parse message
>
> ]0;[00:07 Sat 22/11/2003] address@hidden/9] (~/working)
[0m[27m[24m[J[1mpond[0m:00:07 > [K[57C[32m~/working[37m[66D
> Script done on Sat Nov 22 00:07:10 2003
>
>
> _______________________________________________
> Linphone-users mailing list
> address@hidden
> http://mail.nongnu.org/mailman/listinfo/linphone-users
- Re: [Linphone-users] No sound under 0.12.1?,
Micah Anderson <=