RE: [Linphone-users] Incoming sound problem

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RE: [Linphone-users] Incoming sound problem

From:	Zloty
Subject:	RE: [Linphone-users] Incoming sound problem
Date:	Thu, 1 Dec 2005 09:24:16 +0100
Hello,

I check on version 1.2.0-pre5 and 1.2.0-pre6 and problem still exists,
Actual I have win xp with sjphone and redhat with asterisk+linphone 
On redhat I don't have any firewall and ipforward is enabled.
I don't have only incoming sound, but I hear ringing on linphone.
Tcpdump show that packets are send from xp to redhat by eth0, and on
ifconfig I can see how counter is incrementing( for both interfaces eth and
lo). Then I think that xp communicate with asterisk by eth0 and it sent
packets to linphone by lo. But linphone doesn't got it.

I look into   Global statistics :
 number of rtp packet received=0
 number of rtp bytes received=0 bytes
Why, any sugestions? Maybe linphone don't listen on lo interface.

I also include log from asterisk.

Why in logs from linphone I see something like that
Warning:alsa_set_params: The rate 8000 Hz is not supported by your hardware.
 ==> Using 8000 Hz instead.
Is it the same, I think.

How I can reduce latency on linphone, when I check echo(asterisk) on the
same computer I got echo about 0,5s, ( I've PIII450, isn't enough).

Best regards
Robert


address@hidden linphone-1.2.0pre6]# linphonec -d 6
INFO: no logfile, logging to stdout
oRTP-message:oRTP-0.7.2pre3initialized.
Message:Found /dev/dsp.
Message:Found ALSA device: Ensoniq AudioPCI
Message:Found ALSA device: Ensoniq AudioPCI
Warning:Cannot open directory /usr/lib/linphone/plugins/mediastreamer: No
such file or directory
Warning:alsa_set_params: The rate 8000 Hz is not supported by your hardware.
 ==> Using 8000 Hz instead.

Warning:alsa_set_params: The period size 256 is not supported by your
hardware.
 ==> Using 256 instead.

Message:alsa_set_params:  blocksize=512.
Warning:alsa_set_params: The rate 8000 Hz is not supported by your hardware.
 ==> Using 8000 Hz instead.

Warning:alsa_set_params: The period size 256 is not supported by your
hardware.
 ==> Using 256 instead.

Message:alsa_set_params:  blocksize=512.
| INFO1 | <eXosip.c: 333> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <eXutils.c: 416> Outgoing interface to reach 10.1.1.230 is
10.1.1.230.

| INFO1 | <eXutils.c: 416> Outgoing interface to reach 10.1.1.230 is
10.1.1.230.

| INFO2 | <osip_transaction.c: 130> allocating transaction ressource 1
985656913
| INFO2 | <nict.c: 36> allocating NICT context
Ready.
linphonec> | INFO2 | <eXutils.c: 492> IPv4 address detected: 10.1.1.230
| INFO2 | <eXutils.c: 541> DNS resolution with 10.1.1.230:5060
| INFO1 | <jcallback.c: 148> Message sent: 
REGISTER sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:25060;rport;branch=z9hG4bK142363793
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 1 REGISTER
Contact: <sip:address@hidden:25060>
Max-Forwards: 5
User-Agent: Linphone-1.2.0pre6/eXosip
Expires: 600
Content-Length: 0

 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 525> cb_sndregister (id=1)
| INFO1 | <eXosip.c: 340> eXosip: timer sec:0 usec:495123!
| INFO1 | <udp.c: 2193> Received message: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.1.1.230:25060;rport;branch=z9hG4bK142363793;received=10.1.1.230
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:985656913
| INFO1 | <jcallback.c: 601> cb_rcv1xx (id=1)
| INFO1 | <eXosip.c: 340> eXosip: timer sec:0 usec:493211!
| INFO1 | <udp.c: 2193> Received message: 
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
10.1.1.230:25060;rport;branch=z9hG4bK142363793;received=10.1.1.230
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>;tag=as4c87be7c
Call-ID: address@hidden
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
WWW-Authenticate: Digest realm="asterisk", nonce="175ae16f"
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:985656913
| INFO1 | <jcallback.c: 1437> cb_rcv4xx (id=1)
| INFO1 | <eXosip.c: 340> eXosip: timer sec:4 usec:999985!
Message:REGISTRATION_FAILURE

Message:cfg= sip:address@hidden, cfg->rid=1, rid=1
| INFO2 | <nict.c: 129> free nict ressource
| INFO1 | <eXutils.c: 416> Outgoing interface to reach 10.1.1.230 is
10.1.1.230.

| INFO2 | <eXosip.c: 2123> INFO: authinfo: "asterisk" "asterisk"
| INFO1 | <eXosip.c: 2214> authinfo: zloty
| INFO2 | <osip_transaction.c: 130> allocating transaction ressource 2
985656913
| INFO2 | <nict.c: 36> allocating NICT context
| INFO2 | <eXutils.c: 492> IPv4 address detected: 10.1.1.230
| INFO2 | <eXutils.c: 541> DNS resolution with 10.1.1.230:5060
| INFO1 | <jcallback.c: 148> Message sent: 
REGISTER sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:25060;rport;branch=z9hG4bK420678613
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 2 REGISTER
Contact: <sip:address@hidden:25060>
Authorization: Digest username="zloty", realm="asterisk", nonce="175ae16f",
uri="sip:address@hidden", response="5748d16b12a6082cf135b41b90c442ea",
algorithm=MD5
Max-Forwards: 5
User-Agent: Linphone-1.2.0pre6/eXosip
Expires: 600
Content-Length: 0

 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 525> cb_sndregister (id=2)
| INFO1 | <eXosip.c: 340> eXosip: timer sec:0 usec:496645!
| INFO1 | <udp.c: 2193> Received message: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.1.1.230:25060;rport;branch=z9hG4bK420678613;received=10.1.1.230
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:985656913
| INFO1 | <jcallback.c: 601> cb_rcv1xx (id=2)
| INFO1 | <eXosip.c: 340> eXosip: timer sec:0 usec:494973!
| INFO1 | <udp.c: 2193> Received message: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.1.1.230:25060;rport;branch=z9hG4bK420678613;received=10.1.1.230
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>;tag=as4c87be7c
Call-ID: address@hidden
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Expires: 600
Contact: <sip:address@hidden:25060>;expires=600
Date: Thu, 01 Dec 2005 03:52:11 GMT
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:985656913
| INFO1 | <jcallback.c: 1213> cb_rcv2xx (id=2)
| INFO1 | <eXosip.c: 340> eXosip: timer sec:4 usec:999985!
Registration on sip:address@hidden sucessful.
Message:cfg= sip:address@hidden, cfg->rid=1, rid=1
| INFO1 | <udp.c: 2193> Received message: 
INVITE sip:address@hidden:25060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK1ca8d7c5;rport
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>
Contact: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 01 Dec 2005 03:52:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 209

v=0
o=root 7285 7285 IN IP4 10.1.1.230
s=session
c=IN IP4 10.1.1.230
t=0 0
m=audio 17546 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

| INFO3 | <osip_event.c: 84> MESSAGE REC.
CALLID:60a5eeb93fc191a337ea2fdf4547d60a
| INFO1 | <udp.c: 2230> This is a request
| INFO2 | <osip_transaction.c: 130> allocating transaction ressource 3
60a5eeb93fc191a337ea2fdf4547d60a
| INFO2 | <ist.c: 32> allocating IST context
| INFO1 | <eXutils.c: 416> Outgoing interface to reach 10.1.1.230 is
10.1.1.230.

| INFO1 | <jcallback.c: 332> cb_rcvinvite (id=3)
| INFO2 | <eXutils.c: 492> IPv4 address detected: 10.1.1.230
| INFO2 | <eXutils.c: 541> DNS resolution with 10.1.1.230:5060
| INFO1 | <jcallback.c: 148> Message sent: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK1ca8d7c5;rport=5060
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>
Call-ID: address@hidden
CSeq: 102 INVITE
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Length: 0

 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 1669> cb_snd1xx (id=3)
| INFO2 | <eXutils.c: 492> IPv4 address detected: 10.1.1.230
| INFO2 | <eXutils.c: 541> DNS resolution with 10.1.1.230:5060
| INFO1 | <jcallback.c: 148> Message sent: 
SIP/2.0 101 Dialog Establishement
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK1ca8d7c5;rport=5060
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>;tag=2046923003
Call-ID: address@hidden
CSeq: 102 INVITE
Contact: <sip:address@hidden:25060>
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Length: 0

 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 1669> cb_snd1xx (id=3)
| INFO1 | <eXosip.c: 340> eXosip: timer sec:0 usec:738207!
| INFO1 | <eXosip.c: 340> eXosip: timer sec:0 usec:737896!
Message:CALL_NEW

| INFO1 | <eXutils.c: 416> Outgoing interface to reach 15.128.128.93 is
10.1.1.230.

| INFO1 | <eXutils.c: 416> Outgoing interface to reach 10.1.1.230 is
10.1.1.230.

"Robert" <sip:address@hidden> is calling you.
| INFO1 | <eXutils.c: 416> Outgoing interface to reach 10.1.1.230 is
10.1.1.230.

Warning:alsa_set_params: The rate 8000 Hz is not supported by your hardware.
 ==> Using 8000 Hz instead.

Warning:alsa_set_params: The period size 256 is not supported by your
hardware.
 ==> Using 256 instead.

Message:alsa_set_params:  blocksize=512.
Message:Starting local ring...
Message:ms_filter_add_link: ringplay,0 -> OssWrite,0
Message:Opening sound card [Ensoniq AudioPCI (Advanced Linux Sound
Architecture)] in playback mode with stereo=1,rate=44100,bits=16
Warning:alsa_set_params: The rate 44100 Hz is not supported by your
hardware.
 ==> Using 44100 Hz instead.

Message:alsa_set_params:  blocksize=5120.
| INFO2 | <eXutils.c: 492> IPv4 address detected: 10.1.1.230
| INFO2 | <eXutils.c: 541> DNS resolution with 10.1.1.230:5060
| INFO1 | <jcallback.c: 148> Message sent: 
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK1ca8d7c5;rport=5060
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>;tag=2046923003
Call-ID: address@hidden
CSeq: 102 INVITE
Contact: <sip:address@hidden:25060>
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Length: 0

 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 1669> cb_snd1xx (id=3)
| INFO1 | <eXosip.c: 340> eXosip: timer sec:0 usec:710797!
| INFO1 | <jcallback.c: 217> cb_nict_kill_transaction (id=2)
| INFO1 | <eXosip.c: 333> eXosip: Reseting timer to 15s before waking up!
answer
Message:Mediastreamer processing thread is exiting.
Message:Closing writing channel of soundcard.
| INFO1 | <eXutils.c: 416> Outgoing interface to reach 10.1.1.230 is
10.1.1.230.

Connected.
oRTP-warning:rtp_session_set_jitter_compensation: cannot set because the
payload type (0) is unknown
Message:ms_filter_add_link: OssRead,0 -> GSMEncoder,0
Message:ms_filter_add_link: GSMEncoder,0 -> RTPSend,0
Message:ms_filter_add_link: RTPRecv,0 -> GSMDecoder,0
Message:ms_filter_add_link: GSMDecoder,0 -> OssWrite,0
Message:Opening sound card [Ensoniq AudioPCI (Advanced Linux Sound
Architecture)] in capture mode with stereo=0,rate=8000,bits=16
Message:alsa_set_params:  blocksize=512.
Message:Opening sound card [Ensoniq AudioPCI (Advanced Linux Sound
Architecture)] in playback mode with stereo=0,rate=8000,bits=16
Warning:alsa_set_params: The rate 8000 Hz is not supported by your hardware.
 ==> Using 8000 Hz instead.

Warning:alsa_set_params: The period size 256 is not supported by your
hardware.
 ==> Using 256 instead.

Message:alsa_set_params:  blocksize=512.
linphonec> | INFO2 | <eXutils.c: 492> IPv4 address detected: 10.1.1.230
| INFO2 | <eXutils.c: 541> DNS resolution with 10.1.1.230:5060
| INFO1 | <jcallback.c: 148> Message sent: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK1ca8d7c5;rport=5060
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>;tag=2046923003
Call-ID: address@hidden
CSeq: 102 INVITE
Contact: <sip:address@hidden:25060>
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Type: application/sdp
Content-Length:   194

v=0
o=zloty 123456 654321 IN IP4 10.1.1.230
s=A conversation
c=IN IP4 10.1.1.230
t=0 0
m=audio 27078 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 1684> cb_snd2xx (id=3)
| INFO1 | <eXosip.c: 333> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <udp.c: 2193> Received message: 
ACK sip:address@hidden:25060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK490cfd46;rport
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>;tag=2046923003
Contact: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC.
CALLID:60a5eeb93fc191a337ea2fdf4547d60a
| INFO1 | <udp.c: 2230> This is a request
| INFO1 | <eXosip.c: 333> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <udp.c: 2193> Received message: 
INVITE sip:address@hidden:25060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK2748d6c5;rport
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>;tag=2046923003
Contact: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 7285 7286 IN IP4 10.1.1.239
s=session
c=IN IP4 10.1.1.239
t=0 0
m=audio 49160 RTP/AVP 0 3 8 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

| INFO3 | <osip_event.c: 84> MESSAGE REC.
CALLID:60a5eeb93fc191a337ea2fdf4547d60a
| INFO1 | <udp.c: 2230> This is a request
| INFO2 | <osip_transaction.c: 130> allocating transaction ressource 4
60a5eeb93fc191a337ea2fdf4547d60a
| INFO2 | <ist.c: 32> allocating IST context
| INFO1 | <eXutils.c: 416> Outgoing interface to reach 10.1.1.230 is
10.1.1.230.

| INFO1 | <jcallback.c: 332> cb_rcvinvite (id=4)
| INFO2 | <eXutils.c: 492> IPv4 address detected: 10.1.1.230
| INFO2 | <eXutils.c: 541> DNS resolution with 10.1.1.230:5060
| INFO1 | <jcallback.c: 148> Message sent: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK2748d6c5;rport=5060
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>;tag=2046923003
Call-ID: address@hidden
CSeq: 103 INVITE
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Length: 0

 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 1669> cb_snd1xx (id=4)
| INFO2 | <eXutils.c: 492> IPv4 address detected: 10.1.1.230
| INFO2 | <eXutils.c: 541> DNS resolution with 10.1.1.230:5060
| INFO1 | <jcallback.c: 148> Message sent: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK2748d6c5;rport=5060
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>;tag=2046923003
Call-ID: address@hidden
CSeq: 103 INVITE
Contact: <sip:address@hidden:25060>
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Type: application/sdp
Content-Length:   160

v=0
o=userX 20000001 20000001 IN IP4 10.1.1.230
s=session
c=IN IP4 10.1.1.230
t=0 0
m=audio 10500 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 1684> cb_snd2xx (id=4)
| INFO1 | <eXosip.c: 333> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <eXosip.c: 333> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <udp.c: 2193> Received message: 
ACK sip:address@hidden:25060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK4dabaaf1;rport
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>;tag=2046923003
Contact: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC.
CALLID:60a5eeb93fc191a337ea2fdf4547d60a
| INFO1 | <udp.c: 2230> This is a request
| INFO1 | <eXosip.c: 333> eXosip: Reseting timer to 15s before waking up!
Message:CALL_ACK

Message:CALL_HOLD ou OFFHOLD

Message:CALL_ACK

terminate| INFO1 | <eXosip.c: 333> eXosip: Reseting timer to 15s before
waking up!

| INFO1 | <eXutils.c: 416> Outgoing interface to reach 10.1.1.230 is
10.1.1.230.

| INFO2 | <osip_transaction.c: 130> allocating transaction ressource 5
60a5eeb93fc191a337ea2fdf4547d60a
| INFO2 | <nict.c: 36> allocating NICT context
Message:Mediastreamer processing thread is exiting.
| INFO2 | <eXutils.c: 492> IPv4 address detected: 10.1.1.230
| INFO2 | <eXutils.c: 541> DNS resolution with 10.1.1.230:5060
| INFO1 | <jcallback.c: 148> Message sent: 
BYE sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:25060;rport;branch=z9hG4bK1248993035
From: <sip:address@hidden:25060>;tag=2046923003
To: "Robert" <sip:address@hidden>;tag=as013f2e80
Call-ID: address@hidden
CSeq: 103 BYE
Contact: <sip:address@hidden:25060>
Max-Forwards: 5
User-Agent: Linphone-1.2.0pre6/eXosip
Content-Length: 0

 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 530> cb_sndbye (id=5)
| INFO1 | <eXosip.c: 340> eXosip: timer sec:0 usec:498998!
Message:Closing reading channel of soundcard.
Message:Closing writing channel of soundcard.
oRTP-stats-message:
   Global statistics :
 number of rtp packet sent=788
 number of rtp bytes sent=35460 bytes
 number of rtp packet received=0
 number of rtp bytes received=0 bytes
 number of incoming rtp bytes successfully delivered to the application=0 
 number of times the application queried a packet that didn't exist=1580 
 number of rtp packets received too late=0
 number of rtp packets skipped=0
 number of bad formatted rtp packets=0
 number of packet discarded because of queue overflow=0

Communication ended.
linphonec> | INFO1 | <udp.c: 2193> Received message: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.1.1.230:25060;rport;branch=z9hG4bK1248993035;received=10.1.1.230
From: <sip:address@hidden:25060>;tag=2046923003
To: "Robert" <sip:address@hidden>;tag=as013f2e80
Call-ID: address@hidden
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


| INFO3 | <osip_event.c: 84> MESSAGE REC.
CALLID:60a5eeb93fc191a337ea2fdf4547d60a
| INFO1 | <jcallback.c: 1213> cb_rcv2xx (id=5)
| INFO1 | <eXosip.c: 340> eXosip: timer sec:4 usec:999978!
| INFO1 | <eXosip.c: 340> eXosip: timer sec:0 usec:6535!
| INFO1 | <jcallback.c: 217> cb_nict_kill_transaction (id=5)
| INFO2 | <udp.c: 2371> eXosip: eXosip_release_finished_calls remove a
dialog
| INFO1 | <udp.c: 2606> eXosip: remove a call
| INFO1 | <udp.c: 2622> Release a terminated transaction
| INFO2 | <osip_transaction.c: 288> free transaction ressource 3
60a5eeb93fc191a337ea2fdf4547d60a
| INFO2 | <ist.c: 85> free ist ressource
| INFO1 | <udp.c: 2622> Release a terminated transaction
|  BUG  | <osip_transaction.c: 265> transaction already removed from list 5!
| INFO2 | <osip_transaction.c: 288> free transaction ressource 5
60a5eeb93fc191a337ea2fdf4547d60a
| INFO2 | <nict.c: 129> free nict ressource
| INFO1 | <udp.c: 2622> Release a terminated transaction
| INFO2 | <osip_transaction.c: 288> free transaction ressource 4
60a5eeb93fc191a337ea2fdf4547d60a
| INFO2 | <ist.c: 85> free ist ressource
| INFO1 | <eXosip.c: 333> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <eXosip.c: 333> eXosip: Reseting timer to 15s before waking up!
Message:CALL_RELEASED

quit
linphonec> 
|  BUG  | <osip_transaction.c: 265> transaction already removed from list 2!
| INFO2 | <nict.c: 129> free nict ressource
| INFO1 | <eXutils.c: 416> Outgoing interface to reach 10.1.1.230 is
10.1.1.230.

| INFO2 | <osip_transaction.c: 130> allocating transaction ressource 6
985656913
| INFO2 | <nict.c: 36> allocating NICT context
| INFO2 | <eXutils.c: 492> IPv4 address detected: 10.1.1.230
| INFO2 | <eXutils.c: 541> DNS resolution with 10.1.1.230:5060
| INFO1 | <jcallback.c: 148> Message sent: 
REGISTER sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:25060;rport;branch=z9hG4bK588408792
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 3 REGISTER
Contact: <sip:address@hidden:25060>
Max-Forwards: 5
User-Agent: Linphone-1.2.0pre6/eXosip
Expires: 0
Content-Length: 0

 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 525> cb_sndregister (id=6)
| INFO1 | <eXosip.c: 340> eXosip: timer sec:0 usec:499232!
| INFO1 | <udp.c: 2193> Received message: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.1.1.230:25060;rport;branch=z9hG4bK588408792;received=10.1.1.230
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:985656913
| INFO1 | <jcallback.c: 601> cb_rcv1xx (id=6)
| INFO1 | <eXosip.c: 340> eXosip: timer sec:0 usec:496732!
| INFO1 | <udp.c: 2193> Received message: 
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
10.1.1.230:25060;rport;branch=z9hG4bK588408792;received=10.1.1.230
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>;tag=as4f0d0700
Call-ID: address@hidden
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
WWW-Authenticate: Digest realm="asterisk", nonce="6eeb357a"
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:985656913
| INFO1 | <jcallback.c: 1437> cb_rcv4xx (id=6)
| INFO1 | <eXosip.c: 340> eXosip: timer sec:4 usec:999987!
Message:REGISTRATION_FAILURE

Message:cfg= sip:address@hidden, cfg->rid=1, rid=1
| INFO2 | <nict.c: 129> free nict ressource
| INFO1 | <eXutils.c: 416> Outgoing interface to reach 10.1.1.230 is
10.1.1.230.

| INFO2 | <eXosip.c: 2123> INFO: authinfo: "asterisk" "asterisk"
| INFO1 | <eXosip.c: 2214> authinfo: zloty
| INFO2 | <osip_transaction.c: 130> allocating transaction ressource 7
985656913
| INFO2 | <nict.c: 36> allocating NICT context
| INFO2 | <eXutils.c: 492> IPv4 address detected: 10.1.1.230
| INFO2 | <eXutils.c: 541> DNS resolution with 10.1.1.230:5060
| INFO1 | <jcallback.c: 148> Message sent: 
REGISTER sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:25060;rport;branch=z9hG4bK1016139707
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 4 REGISTER
Contact: <sip:address@hidden:25060>
Authorization: Digest username="zloty", realm="asterisk", nonce="6eeb357a",
uri="sip:address@hidden", response="c192cf97e697cdcc103d033969c23ba1",
algorithm=MD5
Max-Forwards: 5
User-Agent: Linphone-1.2.0pre6/eXosip
Expires: 0
Content-Length: 0

 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 525> cb_sndregister (id=7)
| INFO1 | <eXosip.c: 340> eXosip: timer sec:0 usec:499240!
| INFO1 | <udp.c: 2193> Received message: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.1.1.230:25060;rport;branch=z9hG4bK1016139707;received=10.1.1.230
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:985656913
| INFO1 | <jcallback.c: 601> cb_rcv1xx (id=7)
| INFO1 | <eXosip.c: 340> eXosip: timer sec:0 usec:496991!
| INFO1 | <udp.c: 2193> Received message: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.1.1.230:25060;rport;branch=z9hG4bK1016139707;received=10.1.1.230
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>;tag=as4f0d0700
Call-ID: address@hidden
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Expires: 0
Date: Thu, 01 Dec 2005 03:52:53 GMT
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:985656913
| INFO1 | <jcallback.c: 1213> cb_rcv2xx (id=7)
| INFO1 | <eXosip.c: 340> eXosip: timer sec:4 usec:999984!
Registration on sip:address@hidden sucessful.
Message:cfg= sip:address@hidden, cfg->rid=1, rid=1
| INFO1 | <jreg.c: 80> Release a non-terminated transaction
| INFO2 | <osip_transaction.c: 288> free transaction ressource 7 985656913
| INFO2 | <nict.c: 129> free nict ressource
address@hidden linphone-1.2.0pre6]#
-----------------------------------------
Asterisk log
*CLI> sip debug
SIP Debugging enabled
*CLI> 
<-- SIP read from 10.1.1.230:25060: 
REGISTER sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:25060;rport;branch=z9hG4bK142363793
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 1 REGISTER
Contact: <sip:address@hidden:25060>
Max-Forwards: 5
User-Agent: Linphone-1.2.0pre6/eXosip
Expires: 600
Content-Length: 0


--- (11 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 10.1.1.230 : 25060 (non-NAT)
Transmitting (no NAT) to 10.1.1.230:25060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.1.1.230:25060;rport;branch=z9hG4bK142363793;received=10.1.1.230
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
Content-Length: 0


---
Transmitting (no NAT) to 10.1.1.230:25060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
10.1.1.230:25060;rport;branch=z9hG4bK142363793;received=10.1.1.230
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>;tag=as4c87be7c
Call-ID: address@hidden
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
WWW-Authenticate: Digest realm="asterisk", nonce="175ae16f"
Content-Length: 0


---
Scheduling destruction of call 'address@hidden' in 15000 ms

<-- SIP read from 10.1.1.230:25060: 
REGISTER sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:25060;rport;branch=z9hG4bK420678613
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 2 REGISTER
Contact: <sip:address@hidden:25060>
Authorization: Digest username="zloty", realm="asterisk", nonce="175ae16f",
uri="sip:address@hidden", response="5748d16b12a6082cf135b41b90c442ea",
algorithm=MD5
Max-Forwards: 5
User-Agent: Linphone-1.2.0pre6/eXosip
Expires: 600
Content-Length: 0


--- (12 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 10.1.1.230 : 25060 (non-NAT)
Transmitting (no NAT) to 10.1.1.230:25060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.1.1.230:25060;rport;branch=z9hG4bK420678613;received=10.1.1.230
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
Content-Length: 0


---
    -- Registered SIP 'zloty' at 10.1.1.230 port 25060 expires 600
    -- Saved useragent "Linphone-1.2.0pre6/eXosip" for peer zloty
Transmitting (no NAT) to 10.1.1.230:25060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.1.1.230:25060;rport;branch=z9hG4bK420678613;received=10.1.1.230
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>;tag=as4c87be7c
Call-ID: address@hidden
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Expires: 600
Contact: <sip:address@hidden:25060>;expires=600
Date: Thu, 01 Dec 2005 03:52:11 GMT
Content-Length: 0


---
Scheduling destruction of call 'address@hidden' in 15000 ms

<-- SIP read from 10.1.1.239:5060: 
INVITE sip:address@hidden SIP/2.0
l: 333
m: <sip:address@hidden:5060>
i: address@hidden
c: application/sdp
f: "Robert"<sip:address@hidden>;tag=1078107825399
CSeq: 1 INVITE
Max-Forwards: 70
t: <sip:address@hidden>
v: SIP/2.0/UDP
10.1.1.239;rport;branch=z9hG4bK0a0101ef0131c9b1438e254d00006c4d0000008d
User-Agent: SJphone/1.50.271d (SJ Labs)

v=0
o=- 3342377933 3342377933 IN IP4 10.1.1.239
s=SJphone
c=IN IP4 10.1.1.239
t=0 0
a=direction:active
m=audio 49160 RTP/AVP 3 97 98 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16

--- (11 headers 15 lines)---
Using INVITE request as basis request -
address@hidden
Sending to 10.1.1.239 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 10.1.1.239:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
10.1.1.239;rport;branch=z9hG4bK0a0101ef0131c9b1438e254d00006c4d0000008d;rece
ived=10.1.1.239
From: "Robert"<sip:address@hidden>;tag=1078107825399
To: <sip:address@hidden>;tag=as22b2ec2f
Call-ID: address@hidden
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
Proxy-Authenticate: Digest realm="asterisk", nonce="3a890a79"
Content-Length: 0


---
Scheduling destruction of call
'address@hidden' in 15000 ms
Found user 'zloty2'

<-- SIP read from 10.1.1.239:5060: 
ACK sip:address@hidden SIP/2.0
l: 0
i: address@hidden
Max-Forwards: 70
CSeq: 1 ACK
f: "Robert"<sip:address@hidden>;tag=1078107825399
t: <sip:address@hidden>;tag=as22b2ec2f
v: SIP/2.0/UDP
10.1.1.239;branch=z9hG4bK0a0101ef0131c9b1438e254d00006c4d0000008d


--- (8 headers 0 lines)---

<-- SIP read from 10.1.1.239:5060: 
INVITE sip:address@hidden SIP/2.0
l: 333
m: <sip:address@hidden:5060>
i: address@hidden
c: application/sdp
f: "Robert"<sip:address@hidden>;tag=1078107825399
CSeq: 2 INVITE
Max-Forwards: 70
t: <sip:address@hidden>
v: SIP/2.0/UDP
10.1.1.239;rport;branch=z9hG4bK0a0101ef0000003b438e254d0000047c0000008f
User-Agent: SJphone/1.50.271d (SJ Labs)
Proxy-Authorization: Digest
username="zloty2",realm="asterisk",nonce="3a890a79",uri="sip:address@hidden"
,response="2eeae9679a7b5703aa2b1c8a47fc515f"

v=0
o=- 3342377933 3342377933 IN IP4 10.1.1.239
s=SJphone
c=IN IP4 10.1.1.239
t=0 0
a=direction:active
m=audio 49160 RTP/AVP 3 97 98 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16

--- (12 headers 15 lines)---
Using INVITE request as basis request -
address@hidden
Sending to 10.1.1.239 : 5060 (non-NAT)
Found user 'zloty2'
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.1.1.239:49160
Found description format GSM
Found description format iLBC
Found description format iLBC
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x40e
(gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 601 in default (domain 10.1.1.230)
list_route: hop: <sip:address@hidden:5060>
Transmitting (no NAT) to 10.1.1.239:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.1.1.239;rport;branch=z9hG4bK0a0101ef0000003b438e254d0000047c0000008f;rece
ived=10.1.1.239
From: "Robert"<sip:address@hidden>;tag=1078107825399
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
Content-Length: 0


---
    -- Executing Dial("SIP/zloty2-926c", "SIP/zloty") in new stack
We're at 10.1.1.230 port 17546
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 10.1.1.230:25060:
INVITE sip:address@hidden:25060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK1ca8d7c5;rport
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>
Contact: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 01 Dec 2005 03:52:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 209

v=0
o=root 7285 7285 IN IP4 10.1.1.230
s=session
c=IN IP4 10.1.1.230
t=0 0
m=audio 17546 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called zloty

<-- SIP read from 10.1.1.230:25060: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK1ca8d7c5;rport=5060
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>
Call-ID: address@hidden
CSeq: 102 INVITE
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Length: 0


--- (8 headers 0 lines)---

<-- SIP read from 10.1.1.230:25060: 
SIP/2.0 101 Dialog Establishement
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK1ca8d7c5;rport=5060
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>;tag=2046923003
Call-ID: address@hidden
CSeq: 102 INVITE
Contact: <sip:address@hidden:25060>
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Length: 0


--- (9 headers 0 lines)---
    -- SIP/zloty-5196 is making progress passing it to SIP/zloty2-926c
We're at 10.1.1.230 port 12096
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (no NAT) to 10.1.1.239:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
10.1.1.239;rport;branch=z9hG4bK0a0101ef0000003b438e254d0000047c0000008f;rece
ived=10.1.1.239
From: "Robert"<sip:address@hidden>;tag=1078107825399
To: <sip:address@hidden>;tag=as7dfbccad
Call-ID: address@hidden
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 7285 7285 IN IP4 10.1.1.230
s=session
c=IN IP4 10.1.1.230
t=0 0
m=audio 12096 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
a=silenceSupp:off - - - -

---

<-- SIP read from 10.1.1.230:25060: 
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK1ca8d7c5;rport=5060
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>;tag=2046923003
Call-ID: address@hidden
CSeq: 102 INVITE
Contact: <sip:address@hidden:25060>
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Length: 0


--- (9 headers 0 lines)---
    -- SIP/zloty-5196 is ringing
Transmitting (no NAT) to 10.1.1.239:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
10.1.1.239;rport;branch=z9hG4bK0a0101ef0000003b438e254d0000047c0000008f;rece
ived=10.1.1.239
From: "Robert"<sip:address@hidden>;tag=1078107825399
To: <sip:address@hidden>;tag=as7dfbccad
Call-ID: address@hidden
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
Content-Length: 0


---

<-- SIP read from 10.1.1.239:5060: 
OPTIONS sip:10.1.1.230:5060 SIP/2.0
l: 0
i: address@hidden
f: <sip:address@hidden>;tag=107903436928
CSeq: 28 OPTIONS
Max-Forwards: 70
t: <sip:10.1.1.230:5060>
v: SIP/2.0/UDP
10.1.1.239;rport;branch=z9hG4bK0a0101ef0131c9b1438e255600000da100000093


--- (8 headers 0 lines)---
Looking for 10.1.1.230:5060 in default (domain )
Transmitting (no NAT) to 10.1.1.239:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
10.1.1.239;rport;branch=z9hG4bK0a0101ef0131c9b1438e255600000da100000093;rece
ived=10.1.1.239
From: <sip:address@hidden>;tag=107903436928
To: <sip:10.1.1.230:5060>;tag=as071384dc
Call-ID: address@hidden
CSeq: 28 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:10.1.1.230>
Accept: application/sdp
Content-Length: 0


---
Destroying call 'address@hidden'
Destroying call 'address@hidden'

<-- SIP read from 10.1.1.230:25060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK1ca8d7c5;rport=5060
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>;tag=2046923003
Call-ID: address@hidden
CSeq: 102 INVITE
Contact: <sip:address@hidden:25060>
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Type: application/sdp
Content-Length:   194

v=0
o=zloty 123456 654321 IN IP4 10.1.1.230
s=A conversation
c=IN IP4 10.1.1.230
t=0 0
m=audio 27078 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

--- (10 headers 9 lines)---
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 10.1.1.230:27078
Found description format GSM
Found description format telephone-event
Capabilities: us - 0x2 (gsm), peer - audio=0x2 (gsm)/video=0x0 (nothing),
combined - 0x2 (gsm)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:address@hidden:25060>
set_destination: Parsing <sip:address@hidden:25060> for address/port to
send to
set_destination: set destination to 10.1.1.230, port 25060
Transmitting (no NAT) to 10.1.1.230:25060:
ACK sip:address@hidden:25060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK490cfd46;rport
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>;tag=2046923003
Contact: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/zloty-5196 answered SIP/zloty2-926c
We're at 10.1.1.230 port 12096
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.1.1.239:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.1.1.239;rport;branch=z9hG4bK0a0101ef0000003b438e254d0000047c0000008f;rece
ived=10.1.1.239
From: "Robert"<sip:address@hidden>;tag=1078107825399
To: <sip:address@hidden>;tag=as7dfbccad
Call-ID: address@hidden
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 7285 7286 IN IP4 10.1.1.230
s=session
c=IN IP4 10.1.1.230
t=0 0
m=audio 12096 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
a=silenceSupp:off - - - -

---
    -- Attempting native bridge of SIP/zloty2-926c and SIP/zloty-5196
set_destination: Parsing <sip:address@hidden:25060> for address/port to
send to
set_destination: set destination to 10.1.1.230, port 25060
We're at 10.1.1.230 port 17546
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 13 lines
Reliably Transmitting (no NAT) to 10.1.1.230:25060:
INVITE sip:address@hidden:25060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK2748d6c5;rport
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>;tag=2046923003
Contact: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 7285 7286 IN IP4 10.1.1.239
s=session
c=IN IP4 10.1.1.239
t=0 0
m=audio 49160 RTP/AVP 0 3 8 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---

<-- SIP read from 10.1.1.230:25060: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK2748d6c5;rport=5060
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>;tag=2046923003
Call-ID: address@hidden
CSeq: 103 INVITE
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Length: 0


--- (8 headers 0 lines)---

<-- SIP read from 10.1.1.230:25060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK2748d6c5;rport=5060
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>;tag=2046923003
Call-ID: address@hidden
CSeq: 103 INVITE
Contact: <sip:address@hidden:25060>
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO
Content-Type: application/sdp
Content-Length:   160

v=0
o=userX 20000001 20000001 IN IP4 10.1.1.230
s=session
c=IN IP4 10.1.1.230
t=0 0
m=audio 10500 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

--- (10 headers 8 lines)---
Found RTP audio format 0
Found RTP audio format 8
Peer audio RTP is at port 10.1.1.230:10500
Found description format PCMU
Found description format PCMA
Capabilities: us - 0x2 (gsm), peer - audio=0xc (ulaw|alaw)/video=0x0
(nothing), combined - 0x0 (nothing)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Nov 30 22:52:27 NOTICE[7293]: chan_sip.c:3588 process_sdp: No compatible
codecs!
set_destination: Parsing <sip:address@hidden:25060> for address/port to
send to
set_destination: set destination to 10.1.1.230, port 25060
Transmitting (no NAT) to 10.1.1.230:25060:
ACK sip:address@hidden:25060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK4dabaaf1;rport
From: "Robert" <sip:address@hidden>;tag=as013f2e80
To: <sip:address@hidden:25060>;tag=2046923003
Contact: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

<-- SIP read from 10.1.1.239:5060: 
ACK sip:address@hidden SIP/2.0
l: 0
m: <sip:address@hidden:5060>
i: address@hidden
Max-Forwards: 70
CSeq: 2 ACK
f: <sip:address@hidden>;tag=1078107825399
t: <sip:address@hidden>;tag=as7dfbccad
User-Agent: SJphone/1.50.271d (SJ Labs)
v: SIP/2.0/UDP
10.1.1.239;rport;branch=z9hG4bK0a0101ef0131c9b1438e2559000057f800000096


--- (10 headers 0 lines)---
set_destination: Parsing <sip:address@hidden:5060> for address/port to
send to
set_destination: set destination to 10.1.1.239, port 5060
We're at 10.1.1.230 port 12096
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 10.1.1.239:5060:
INVITE sip:address@hidden:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK3640230b;rport
From: <sip:address@hidden>;tag=as7dfbccad
To: "Robert"<sip:address@hidden>;tag=1078107825399
Contact: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 234

v=0
o=root 7285 7287 IN IP4 10.1.1.230
s=session
c=IN IP4 10.1.1.230
t=0 0
m=audio 10500 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---

<-- SIP read from 10.1.1.239:5060: 
SIP/2.0 200 OK
l: 215
m: <sip:address@hidden:5060>
i: address@hidden
c: application/sdp
f: <sip:address@hidden>;tag=as7dfbccad
CSeq: 102 INVITE
Server: SJphone/1.50.271d (SJ Labs)
t: "Robert"<sip:address@hidden>;tag=1078107825399
v: SIP/2.0/UDP
10.1.1.230:5060;rport=5060;received=10.1.1.230;branch=z9hG4bK3640230b

v=0
o=- 3342377933 3342377934 IN IP4 10.1.1.239
s=SJphone
c=IN IP4 10.1.1.239
t=0 0
a=direction:active
m=audio 49160 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16

--- (10 headers 10 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 10.1.1.239:49160
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:address@hidden:5060>
set_destination: Parsing <sip:address@hidden:5060> for address/port to
send to
set_destination: set destination to 10.1.1.239, port 5060
Transmitting (no NAT) to 10.1.1.239:5060:
ACK sip:address@hidden:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK62b1cfea;rport
From: <sip:address@hidden>;tag=as7dfbccad
To: "Robert"<sip:address@hidden>;tag=1078107825399
Contact: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

<-- SIP read from 10.1.1.230:25060: 
BYE sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:25060;rport;branch=z9hG4bK1248993035
From: <sip:address@hidden:25060>;tag=2046923003
To: "Robert" <sip:address@hidden>;tag=as013f2e80
Call-ID: address@hidden
CSeq: 103 BYE
Contact: <sip:address@hidden:25060>
Max-Forwards: 5
User-Agent: Linphone-1.2.0pre6/eXosip
Content-Length: 0


--- (10 headers 0 lines)---
Sending to 10.1.1.230 : 25060 (non-NAT)
Transmitting (no NAT) to 10.1.1.230:25060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.1.1.230:25060;rport;branch=z9hG4bK1248993035;received=10.1.1.230
From: <sip:address@hidden:25060>;tag=2046923003
To: "Robert" <sip:address@hidden>;tag=as013f2e80
Call-ID: address@hidden
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
set_destination: Parsing <sip:address@hidden:5060> for address/port to
send to
set_destination: set destination to 10.1.1.239, port 5060
We're at 10.1.1.230 port 12096
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (no NAT) to 10.1.1.239:5060:
INVITE sip:address@hidden:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK50be62d2;rport
From: <sip:address@hidden>;tag=as7dfbccad
To: "Robert"<sip:address@hidden>;tag=1078107825399
Contact: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 7285 7288 IN IP4 10.1.1.230
s=session
c=IN IP4 10.1.1.230
t=0 0
m=audio 12096 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
a=silenceSupp:off - - - -

---
  == Spawn extension (default, 601, 1) exited non-zero on 'SIP/zloty2-926c'

<-- SIP read from 10.1.1.239:5060: 
SIP/2.0 200 OK
l: 214
m: <sip:address@hidden:5060>
i: address@hidden
c: application/sdp
f: <sip:address@hidden>;tag=as7dfbccad
CSeq: 103 INVITE
Server: SJphone/1.50.271d (SJ Labs)
t: "Robert"<sip:address@hidden>;tag=1078107825399
v: SIP/2.0/UDP
10.1.1.230:5060;rport=5060;received=10.1.1.230;branch=z9hG4bK50be62d2

v=0
o=- 3342377933 3342377935 IN IP4 10.1.1.239
s=SJphone
c=IN IP4 10.1.1.239
t=0 0
a=direction:active
m=audio 49160 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16

--- (10 headers 10 lines)---
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 10.1.1.239:49160
Found description format GSM
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x2
(gsm)/video=0x0 (nothing), combined - 0x2 (gsm)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:address@hidden:5060> for address/port to
send to
set_destination: set destination to 10.1.1.239, port 5060
Transmitting (no NAT) to 10.1.1.239:5060:
ACK sip:address@hidden:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK3e1dfee9;rport
From: <sip:address@hidden>;tag=as7dfbccad
To: "Robert"<sip:address@hidden>;tag=1078107825399
Contact: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:address@hidden:5060> for address/port to
send to
set_destination: set destination to 10.1.1.239, port 5060
Reliably Transmitting (no NAT) to 10.1.1.239:5060:
BYE sip:address@hidden:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:5060;branch=z9hG4bK57b1582e;rport
From: <sip:address@hidden>;tag=as7dfbccad
To: "Robert"<sip:address@hidden>;tag=1078107825399
Contact: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Destroying call 'address@hidden'

<-- SIP read from 10.1.1.239:5060: 
SIP/2.0 200 OK
l: 0
m: <sip:address@hidden:5060>
i: address@hidden
f: <sip:address@hidden>;tag=as7dfbccad
CSeq: 104 BYE
Server: SJphone/1.50.271d (SJ Labs)
t: "Robert"<sip:address@hidden>;tag=1078107825399
v: SIP/2.0/UDP
10.1.1.230:5060;rport=5060;received=10.1.1.230;branch=z9hG4bK57b1582e


--- (9 headers 0 lines)---
Destroying call 'address@hidden'

<-- SIP read from 10.1.1.239:5060: 
OPTIONS sip:10.1.1.230:5060 SIP/2.0
l: 0
i: address@hidden
f: <sip:address@hidden>;tag=108103431026
CSeq: 29 OPTIONS
Max-Forwards: 70
t: <sip:10.1.1.230:5060>
v: SIP/2.0/UDP
10.1.1.239;rport;branch=z9hG4bK0a0101ef0131c9b1438e256a00004b230000009c


--- (8 headers 0 lines)---
Looking for 10.1.1.230:5060 in default (domain )
Transmitting (no NAT) to 10.1.1.239:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
10.1.1.239;rport;branch=z9hG4bK0a0101ef0131c9b1438e256a00004b230000009c;rece
ived=10.1.1.239
From: <sip:address@hidden>;tag=108103431026
To: <sip:10.1.1.230:5060>;tag=as5d2be53a
Call-ID: address@hidden
CSeq: 29 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:10.1.1.230>
Accept: application/sdp
Content-Length: 0


---
Destroying call 'address@hidden'

<-- SIP read from 10.1.1.239:5060: 
REGISTER sip:10.1.1.230 SIP/2.0
l: 0
m: <sip:address@hidden:5060>;events="message-summary"
i: address@hidden
Max-Forwards: 70
f: <sip:address@hidden>;tag=108183283764
CSeq: 11 REGISTER
t: <sip:address@hidden>
v: SIP/2.0/UDP
10.1.1.239;rport;branch=z9hG4bK0a0101ef0131c9b1438e2572000049170000009e
User-Agent: SJphone/1.50.271d (SJ Labs)


--- (10 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 10.1.1.239 : 5060 (non-NAT)
Transmitting (no NAT) to 10.1.1.239:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.1.1.239;rport;branch=z9hG4bK0a0101ef0131c9b1438e2572000049170000009e;rece
ived=10.1.1.239
From: <sip:address@hidden>;tag=108183283764
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 11 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
Content-Length: 0


---
Transmitting (no NAT) to 10.1.1.239:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
10.1.1.239;rport;branch=z9hG4bK0a0101ef0131c9b1438e2572000049170000009e;rece
ived=10.1.1.239
From: <sip:address@hidden>;tag=108183283764
To: <sip:address@hidden>;tag=as6c5d7dd1
Call-ID: address@hidden
CSeq: 11 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
WWW-Authenticate: Digest realm="asterisk", nonce="6ddebf27"
Content-Length: 0


---
Scheduling destruction of call
'address@hidden' in 15000 ms

<-- SIP read from 10.1.1.239:5060: 
REGISTER sip:10.1.1.230 SIP/2.0
l: 0
m: <sip:address@hidden:5060>;events="message-summary"
i: address@hidden
Max-Forwards: 70
f: <sip:address@hidden>;tag=1081834323257
CSeq: 12 REGISTER
t: <sip:address@hidden>
v: SIP/2.0/UDP
10.1.1.239;rport;branch=z9hG4bK0a0101ef0131c9b1438e25720000323c000000a1
User-Agent: SJphone/1.50.271d (SJ Labs)
Authorization: Digest
username="zloty2",realm="asterisk",nonce="6ddebf27",uri="sip:10.1.1.230",res
ponse="ea615b9747679809896508732b919bf6"


--- (11 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 10.1.1.239 : 5060 (non-NAT)
Transmitting (no NAT) to 10.1.1.239:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.1.1.239;rport;branch=z9hG4bK0a0101ef0131c9b1438e25720000323c000000a1;rece
ived=10.1.1.239
From: <sip:address@hidden>;tag=1081834323257
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 12 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
Content-Length: 0


---
Transmitting (no NAT) to 10.1.1.239:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.1.1.239;rport;branch=z9hG4bK0a0101ef0131c9b1438e25720000323c000000a1;rece
ived=10.1.1.239
From: <sip:address@hidden>;tag=1081834323257
To: <sip:address@hidden>;tag=as6c5d7dd1
Call-ID: address@hidden
CSeq: 12 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Expires: 120
Contact: <sip:address@hidden:5060>;expires=120
Date: Thu, 01 Dec 2005 03:52:53 GMT
Content-Length: 0


---
Scheduling destruction of call
'address@hidden' in 15000 ms

<-- SIP read from 10.1.1.230:25060: 
REGISTER sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:25060;rport;branch=z9hG4bK588408792
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 3 REGISTER
Contact: <sip:address@hidden:25060>
Max-Forwards: 5
User-Agent: Linphone-1.2.0pre6/eXosip
Expires: 0
Content-Length: 0


--- (11 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 10.1.1.230 : 25060 (non-NAT)
Transmitting (no NAT) to 10.1.1.230:25060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.1.1.230:25060;rport;branch=z9hG4bK588408792;received=10.1.1.230
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
Content-Length: 0


---
Transmitting (no NAT) to 10.1.1.230:25060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
10.1.1.230:25060;rport;branch=z9hG4bK588408792;received=10.1.1.230
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>;tag=as4f0d0700
Call-ID: address@hidden
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
WWW-Authenticate: Digest realm="asterisk", nonce="6eeb357a"
Content-Length: 0


---
Scheduling destruction of call 'address@hidden' in 15000 ms

<-- SIP read from 10.1.1.230:25060: 
REGISTER sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 10.1.1.230:25060;rport;branch=z9hG4bK1016139707
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 4 REGISTER
Contact: <sip:address@hidden:25060>
Authorization: Digest username="zloty", realm="asterisk", nonce="6eeb357a",
uri="sip:address@hidden", response="c192cf97e697cdcc103d033969c23ba1",
algorithm=MD5
Max-Forwards: 5
User-Agent: Linphone-1.2.0pre6/eXosip
Expires: 0
Content-Length: 0


--- (12 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 10.1.1.230 : 25060 (non-NAT)
Transmitting (no NAT) to 10.1.1.230:25060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.1.1.230:25060;rport;branch=z9hG4bK1016139707;received=10.1.1.230
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:address@hidden>
Content-Length: 0


---
    -- Unregistered SIP 'zloty'
Transmitting (no NAT) to 10.1.1.230:25060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.1.1.230:25060;rport;branch=z9hG4bK1016139707;received=10.1.1.230
From: <sip:address@hidden>;tag=1017599984
To: <sip:address@hidden>;tag=as4f0d0700
Call-ID: address@hidden
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Expires: 0
Date: Thu, 01 Dec 2005 03:52:53 GMT
Content-Length: 0


---
Scheduling destruction of call 'address@hidden' in 15000 ms
sip no debug
SIP Debugging Disabled
*CLI>

-----Original Message-----
From: Simon Morlat [mailto:address@hidden 
Sent: Wednesday, November 30, 2005 6:16 PM
To: address@hidden
Cc: Zloty
Subject: Re: [Linphone-users] Incoming sound problem

Hello,

First please try with linphone-1.2.0pre6 available at 
http://simon.morlat.free.fr/download/unstable
then, if the problem persists, send me a full call log using "linphone 
--verbose"
Thanks

Simon

Le Lundi 28 Novembre 2005 01:42, Zloty a écrit :
> Hi,
>
> I just installed environment to test linphone, and I have problem with
> sound.
> I have two computers :
> 1. win xp with sjphone
> 2. red hat 9 (two sound cards), (asterisk 1.0.7 and linphone 1.0.1 (1.1.0)
> )
>
>
> Both clients linphone and sjphone register in asterisk. I can make call
> from sjphone to asterisk - ok
> linphone to asterisk - ok
> linphone to asterisk - I don't hear this which I tell on sjphone,
>
> Previous I'm think that it's problem with config file, I ad if_name=eth0,
> but it doesn't help.
>
> I check both versions 1.0.1 and 1.1.0 and result are the same.
>
> Any suggestions?
>
> Anyway I've problem with libspeex and 1.1.0 configuration, the solution
> were very simply, install libspeex from rpm not from source.
>
> Regards
> Robert
>
>
>
>
> _______________________________________________
> Linphone-users mailing list
> address@hidden
> http://lists.nongnu.org/mailman/listinfo/linphone-users
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