changed drivers, but still no go.

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changed drivers, but still no go.

Doug Smith-6

To whom it may concern:

This is Doug Smith again.  Recently, I sent in a message to the list
requesting help with a linphone problem with receiving but not
transmitting.  I was instructed to change drivers from oss to alsa, so
I did.  

I am now able to send in several pieces of information that might help
one of you solve the problem, but it looks like it is in hardware to
me.  If you reach a different conclusion, please let me know.  

The first piece of information I can give you is my mixer settings for
the system.  They are as follows.  

vol:87:87:P
pcm:87:87:P
speaker:67:67:P
line:67:67:P
mic:100:100:P
cd:67:67:P
igain:100:100:R
ogain:67:67:P
line1:67:67:P
dig1:67:67:P
phin:67:67:P
phout:67:67:P
video:67:67:P

Believe it or not, the thing sounds better with this microphone
setting set to P rather than R.   Go figure that one out.  

Now comes the ~/.linphonec file which has been created in my home
directory.  

[net]
con_type=3
use_nat=0

[sip]
sip_port=5060
guess_hostname=1
contact=sip:knoppix@unknown-host
use_info=0
use_ipv6=0
default_proxy=-1

[rtp]
audio_rtp_port=7078
video_rtp_port=9078
audio_jitt_comp=128
video_jitt_comp=128

[sound]
playback_dev_id=0
capture_dev_id=0
rec_lev=100
play_lev=87
source=m
local_ring=/usr/share/sounds/linphone/rings/oldphone.wav
remote_ring=/usr/share/sounds/linphone/ringback.wav

[video]
enabled=0
show_local=0

[audio_codec_0]
mime=PCMU
rate=8000
enabled=1

[audio_codec_1]
mime=GSM
rate=8000
enabled=1

[audio_codec_2]
mime=PCMA
rate=8000
enabled=1

[audio_codec_3]
mime=speex
rate=8000
enabled=1

[audio_codec_4]
mime=speex
rate=16000
enabled=1

[audio_codec_5]
mime=1015
rate=8000
enabled=1



Now, I can give you two sets of complete debugging information for an
attempted call to sip:[hidden email], which is supposed to be an
echo test for latency.  It also, unfortunately, tells me that,
regardless of the driver set I use, there is nothing going out the
door here.  


First of all, with the oss drivers.
 oRTP:(GLogLevel=32)** oRTP-0.7.1initialized.
MediaStreamer:(GLogLevel=32)** Found /dev/dsp.
MediaStreamer:(GLogLevel=16)** dsp block size set to 2048.
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <eXutils.c: 416> Outgoing interface to reach 15.128.128.93 is 66.169.86.132.

| INFO1 | <eXutils.c: 416> Outgoing interface to reach fwd.pulver.com is 66.169.86.132.

| INFO1 | <eXutils.c: 416> Outgoing interface to reach fwd.pulver.com is 66.169.86.132.

| INFO2 | <osip_transaction.c: 129> allocating transaction ressource 1 1739435548
| INFO2 | <ict.c: 35> allocating ICT context
| INFO2 | <eXutils.c: 511> Not an IPv4 or IPv6 address: fwd.pulver.com
| INFO2 | <eXutils.c: 541> DNS resolution with fwd.pulver.com:5060
| INFO1 | <jcallback.c: 148> Message sent:
INVITE sip:[hidden email] SIP/2.0
Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK211929910
From: <sip:knoppix@66.169.86.132>;tag=143564218
To: <sip:[hidden email]>
Call-ID: 1739435548@66.169.86.132
CSeq: 20 INVITE
Contact: <sip:knoppix@66.169.86.132:5060>
Max-Forwards: 5
User-Agent: Linphone-1.1.0/eXosip
Subject: Phone call
Expires: 120
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
Content-Type: application/sdp
Content-Length:   357

v=0
o=knoppix 123456 654321 IN IP4 66.169.86.132
s=A conversation
c=IN IP4 66.169.86.132
t=0 0
m=audio 7078 RTP/AVP 0 3 8 110 111 115 101
b=AS:20
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
 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 515> cb_sndinvite (id=1)
| INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:457403!
| INFO1 | <udp.c: 2193> Received message:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK211929910
From: <sip:knoppix@66.169.86.132>;tag=143564218
To: <sip:[hidden email]>
Call-ID: 1739435548@66.169.86.132
CSeq: 20 INVITE
Server: Sip EXpress router (0.8.14-6 (i386/linux))
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:1739435548
| INFO1 | <jcallback.c: 601> cb_rcv1xx (id=1)
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <udp.c: 2193> Received message:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK211929910
From: <sip:knoppix@66.169.86.132>;tag=143564218
To: <sip:[hidden email]>;tag=as21924a5a
Call-ID: 1739435548@66.169.86.132
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:613@69.90.168.13:5028>
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:1739435548
| INFO1 | <jcallback.c: 601> cb_rcv1xx (id=1)
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
LinphoneCore:(GLogLevel=32)** CALL_RINGING

LinphoneCore:(GLogLevel=32)** Remote ringing...
MediaStreamer:(GLogLevel=16)** dsp block size set to 2048.
MediaStreamer:(GLogLevel=32)** ms_filter_add_link: ringplay,0 -> OssWrite,0
MediaStreamer:(GLogLevel=32)** Opening sound card [/dev/dsp (Open Sound System)] in playback mode with stereo=0,rate=8000,bits=16
MediaStreamer:(GLogLevel=16)** dsp block size set to 2048.
MediaStreamer:(GLogLevel=32)** dsp blocksize is 2048.
| INFO1 | <udp.c: 2193> Received message:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK211929910
Record-Route: <sip:613@69.90.155.70;ftag=143564218;lr=on>
From: <sip:knoppix@66.169.86.132>;tag=143564218
To: <sip:[hidden email]>;tag=as21924a5a
Call-ID: 1739435548@66.169.86.132
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:613@69.90.168.13:5028>
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 11782 11782 IN IP4 69.90.168.13
s=session
c=IN IP4 69.90.168.13
t=0 0
m=audio 17896 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
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:1739435548
| INFO1 | <jcallback.c: 1213> cb_rcv2xx (id=1)
| INFO1 | <eXutils.c: 416> Outgoing interface to reach 69.90.168.13 is 66.169.86.132.

| INFO2 | <eXutils.c: 492> IPv4 address detected: 69.90.155.70
| INFO2 | <eXutils.c: 541> DNS resolution with 69.90.155.70:5060
| INFO1 | <jcallback.c: 148> Message sent:
ACK sip:613@69.90.168.13:5028 SIP/2.0
Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK1382328653
Route: <sip:613@69.90.155.70;ftag=143564218;lr=on>
From: <sip:knoppix@66.169.86.132>;tag=143564218
To: <sip:[hidden email]>;tag=as21924a5a
Call-ID: 1739435548@66.169.86.132
CSeq: 20 ACK
Contact: <sip:knoppix@66.169.86.132:5060>
Max-Forwards: 5
User-Agent: Linphone-1.1.0/eXosip
Content-Length: 0

 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 189> cb_ict_kill_transaction (id=1)
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
LinphoneCore:(GLogLevel=32)** CALL_ANSWERED

MediaStreamer:(GLogLevel=32)** Mediastreamer processing thread is exiting.
MediaStreamer:(GLogLevel=32)** Closing writing channel of soundcard.
MediaStreamer:(GLogLevel=32)** ms_filter_add_link: OssRead,0 -> MULAWEncoder,0
MediaStreamer:(GLogLevel=32)** ms_filter_add_link: MULAWEncoder,0 -> RTPSend,0
MediaStreamer:(GLogLevel=32)** ms_filter_add_link: RTPRecv,0 -> MULAWDecoder,0
MediaStreamer:(GLogLevel=32)** ms_filter_add_link: MULAWDecoder,0 -> OssWrite,0
MediaStreamer:(GLogLevel=32)** Opening sound card [/dev/dsp (Open Sound System)] in capture mode with stereo=0,rate=8000,bits=16
MediaStreamer:(GLogLevel=16)** dsp block size set to 2048.
MediaStreamer:(GLogLevel=32)** dsp blocksize is 2048.
MediaStreamer:(GLogLevel=32)** Opening sound card [/dev/dsp (Open Sound System)] in playback mode with stereo=0,rate=8000,bits=16
LinphoneCore:(GLogLevel=32)** CALL_STARTAUDIO

| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <eXutils.c: 416> Outgoing interface to reach 69.90.168.13 is 66.169.86.132.

| INFO2 | <osip_transaction.c: 129> allocating transaction ressource 2 1739435548
| INFO2 | <nict.c: 36> allocating NICT context
| INFO2 | <eXutils.c: 492> IPv4 address detected: 69.90.155.70
| INFO2 | <eXutils.c: 541> DNS resolution with 69.90.155.70:5060
| INFO1 | <jcallback.c: 148> Message sent:
BYE sip:613@69.90.168.13:5028 SIP/2.0
Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK6617564
Route: <sip:613@69.90.155.70;ftag=143564218;lr=on>
From: <sip:knoppix@66.169.86.132>;tag=143564218
To: <sip:[hidden email]>;tag=as21924a5a
Call-ID: 1739435548@66.169.86.132
CSeq: 21 BYE
Contact: <sip:knoppix@66.169.86.132:5060>
Max-Forwards: 5
User-Agent: Linphone-1.1.0/eXosip
Content-Length: 0

 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 530> cb_sndbye (id=2)
| INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:495101!
MediaStreamer:(GLogLevel=32)** Mediastreamer processing thread is exiting.
MediaStreamer:(GLogLevel=32)** Closing reading channel of soundcard.
MediaStreamer:(GLogLevel=32)** Closing writing channel of soundcard.
oRTP-stats:(GLogLevel=32)**
   Global statistics :
 number of rtp packet sent=0
 number of rtp bytes sent=0 bytes
 number of rtp packet received=784
 number of rtp bytes received=134848 bytes
 number of incoming rtp bytes successfully delivered to the application=134676
 number of times the application queried a packet that didn't exist=1424
 number of rtp packets received too late=0
 number of rtp packets skipped=1
 number of bad formatted rtp packets=0
 number of packet discarded because of queue overflow=0

| INFO1 | <udp.c: 2193> Received message:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK6617564
Record-Route: <sip:613@69.90.155.70;ftag=143564218;lr=on>
From: <sip:knoppix@66.169.86.132>;tag=143564218
To: <sip:[hidden email]>;tag=as21924a5a
Call-ID: 1739435548@66.169.86.132
CSeq: 21 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:613@69.90.168.13:5028>
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:1739435548
| INFO1 | <jcallback.c: 1213> cb_rcv2xx (id=2)
| INFO1 | <eXosip.c: 339> eXosip: timer sec:4 usec:999923!
| INFO1 | <eXosip.c: 281> Release a terminated transaction
|  BUG  | <osip_transaction.c: 263> transaction already removed from list 1!
| INFO2 | <osip_transaction.c: 286> free transaction ressource 1 1739435548
| INFO2 | <ict.c: 112> free ict ressource
| INFO2 | <osip_transaction.c: 286> free transaction ressource 2 1739435548
| INFO2 | <nict.c: 110> free nict ressource

This is what I previously sent in so that you could tell me what was
wrong was that I was using the incorrect sound drivers.  I changed
them from oss or kernel to the alsa drivers and, for the same call,
got these results.  

Next, with the alsa drivers.

Ready.
linphonec> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
linphonec> linphonec> linphonec> linphonec> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
Contacting  sip:[hidden email]
| INFO1 | <eXutils.c: 416> Outgoing interface to reach 15.128.128.93 is 66.169.86.132.

| INFO1 | <eXutils.c: 416> Outgoing interface to reach fwd.pulver.com is 66.169.86.132.

| INFO1 | <eXutils.c: 416> Outgoing interface to reach fwd.pulver.com is 66.169.86.132.

| INFO2 | <osip_transaction.c: 129> allocating transaction ressource 1 688034463
| INFO2 | <ict.c: 35> allocating ICT context
| INFO2 | <eXutils.c: 511> Not an IPv4 or IPv6 address: fwd.pulver.com
linphonec> | INFO2 | <eXutils.c: 541> DNS resolution with fwd.pulver.com:5060
| INFO1 | <jcallback.c: 148> Message sent:
INVITE sip:[hidden email] SIP/2.0
Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK1708706045
From: <sip:knoppix@66.169.86.132>;tag=1625521769
To: <sip:[hidden email]>
Call-ID: 688034463@66.169.86.132
CSeq: 20 INVITE
Contact: <sip:knoppix@66.169.86.132:5060>
Max-Forwards: 5
User-Agent: Linphone-1.1.0/eXosip
Subject: Phone call
Expires: 120
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
Content-Type: application/sdp
Content-Length:   357

v=0
o=knoppix 123456 654321 IN IP4 66.169.86.132
s=A conversation
c=IN IP4 66.169.86.132
t=0 0
m=audio 7078 RTP/AVP 0 3 8 110 111 115 101
b=AS:20
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
 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 515> cb_sndinvite (id=1)
| INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:473151!
| INFO1 | <udp.c: 2193> Received message:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK1708706045
From: <sip:knoppix@66.169.86.132>;tag=1625521769
To: <sip:[hidden email]>
Call-ID: 688034463@66.169.86.132
CSeq: 20 INVITE
Server: Sip EXpress router (0.8.14-6 (i386/linux))
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:688034463
| INFO1 | <jcallback.c: 601> cb_rcv1xx (id=1)
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <udp.c: 2193> Received message:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK1708706045
From: <sip:knoppix@66.169.86.132>;tag=1625521769
To: <sip:[hidden email]>;tag=as079da4b6
Call-ID: 688034463@66.169.86.132
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:613@69.90.168.13:5028>
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:688034463
| INFO1 | <jcallback.c: 601> cb_rcv1xx (id=1)
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <udp.c: 2193> Received message:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK1708706045
Record-Route: <sip:613@69.90.155.70;ftag=1625521769;lr=on>
From: <sip:knoppix@66.169.86.132>;tag=1625521769
To: <sip:[hidden email]>;tag=as079da4b6
Call-ID: 688034463@66.169.86.132
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:613@69.90.168.13:5028>
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 22055 22055 IN IP4 69.90.168.13
s=session
c=IN IP4 69.90.168.13
t=0 0
m=audio 10144 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
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:688034463
| INFO1 | <jcallback.c: 1213> cb_rcv2xx (id=1)
| INFO1 | <eXutils.c: 416> Outgoing interface to reach 69.90.168.13 is 66.169.86.132.

| INFO2 | <eXutils.c: 492> IPv4 address detected: 69.90.155.70
| INFO2 | <eXutils.c: 541> DNS resolution with 69.90.155.70:5060
| INFO1 | <jcallback.c: 148> Message sent:
ACK sip:613@69.90.168.13:5028 SIP/2.0
Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK1011056904
Route: <sip:613@69.90.155.70;ftag=1625521769;lr=on>
From: <sip:knoppix@66.169.86.132>;tag=1625521769
To: <sip:[hidden email]>;tag=as079da4b6
Call-ID: 688034463@66.169.86.132
CSeq: 20 ACK
Contact: <sip:knoppix@66.169.86.132:5060>
Max-Forwards: 5
User-Agent: Linphone-1.1.0/eXosip
Content-Length: 0

 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 189> cb_ict_kill_transaction (id=1)
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
Connected.
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <eXutils.c: 416> Outgoing interface to reach 69.90.168.13 is 66.169.86.132.

| INFO2 | <osip_transaction.c: 129> allocating transaction ressource 2 688034463
| INFO2 | <nict.c: 36> allocating NICT context
| INFO2 | <eXutils.c: 492> IPv4 address detected: 69.90.155.70
| INFO2 | <eXutils.c: 541> DNS resolution with 69.90.155.70:5060
| INFO1 | <jcallback.c: 148> Message sent:
BYE sip:613@69.90.168.13:5028 SIP/2.0
Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK1417467515
Route: <sip:613@69.90.155.70;ftag=1625521769;lr=on>
From: <sip:knoppix@66.169.86.132>;tag=1625521769
To: <sip:[hidden email]>;tag=as079da4b6
Call-ID: 688034463@66.169.86.132
CSeq: 21 BYE
Contact: <sip:knoppix@66.169.86.132:5060>
Max-Forwards: 5
User-Agent: Linphone-1.1.0/eXosip
Content-Length: 0

 (len=16 sizeof(addr)=128 28)
| INFO1 | <jcallback.c: 530> cb_sndbye (id=2)
| INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:494971!
Communication ended.
linphonec> | INFO1 | <udp.c: 2193> Received message:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK1417467515
Record-Route: <sip:613@69.90.155.70;ftag=1625521769;lr=on>
From: <sip:knoppix@66.169.86.132>;tag=1625521769
To: <sip:[hidden email]>;tag=as079da4b6
Call-ID: 688034463@66.169.86.132
CSeq: 21 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:613@69.90.168.13:5028>
Content-Length: 0


| INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:688034463
| INFO1 | <jcallback.c: 1213> cb_rcv2xx (id=2)
| INFO1 | <eXosip.c: 339> eXosip: timer sec:4 usec:999923!
| INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:1122!
| INFO1 | <jcallback.c: 217> cb_nict_kill_transaction (id=2)
| INFO2 | <udp.c: 2369> eXosip: eXosip_release_finished_calls remove a dialog
| INFO1 | <udp.c: 2604> eXosip: remove a call
| INFO1 | <udp.c: 2620> Release a terminated transaction
|  BUG  | <osip_transaction.c: 263> transaction already removed from list 1!
| INFO2 | <osip_transaction.c: 286> free transaction ressource 1 688034463
| INFO2 | <ict.c: 112> free ict ressource
| INFO1 | <udp.c: 2620> Release a terminated transaction
|  BUG  | <osip_transaction.c: 263> transaction already removed from list 2!
| INFO2 | <osip_transaction.c: 286> free transaction ressource 2 688034463
| INFO2 | <nict.c: 110> free nict ressource
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
| INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
linphonec> linphonec>

When I tried this, I got no better results than with the original oss
drivers.  I could still hear the incoming information from the other
machine, but the server could not hear me.  Nothing was going out from
here.  

Now, when I got my freeworld-dialup account, a friend of mine, in
fact, a fellow member of the Oralux development team gave me this
little snippit to put into my ~/.linphonec file.  I haven't inserted
it as of yet, but I need to ask if this could be of any help.  The
user names and passwords are left out so that I can send this across
the internet.  In fact, he did not put in his own passwords and
usernames for the same security concerns.  Is this necessary to make
things work properly?

[sip]
sip_port=5060
guess_hostname=1
contact=sip:knoppix@unknown-host
use_info=0
use_ipv6=0
default_proxy=-1
username=123456
hostname=fwd.pulver.com
sip_port=5060
use_registrar=1
as_proxy=1
expires=900
registrar=sip:fwdnat.pulver.com:5082
passwd=pass
addr_of_rec=sip:[hidden email]

Well, I hope you can help with some of this with the information
supplied here.  I hope to include linphone in a future release of the
operating system I and my acquaintance are working on.  It is Oralux
which is Debian based.  This means that I am using the linphone Debian
package from the unstable distribution.  The version of linphone is:

version: 1.1.0

I do not know if this can help or not, but I will include it here.  

I would like to get Linphone working so that I can help others with
their computing situations, build my web site and include it as a
communications option, and include it in the new release of Oralux,
the distribution for the Blind.  Please visit:

http://oralux.org



Thank you in advance for helping me.  

--
Doug Smith: C.S.F.C.
Computer Scientist For CHRIST!
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Re: changed drivers, but still no go.

Simon Morlat
Hello,

Unfortunately the logs of your second attempt with alsa drivers is missing the
important part of the logs (stderr), while the logs of your first attempt is
correct.
Please also check that the old oss drivers are blacklisted so that they are
not loaded into your running kernel.

Simon

Le Jeudi 10 Novembre 2005 20:06, Doug Smith a écrit :

> To whom it may concern:
>
> This is Doug Smith again.  Recently, I sent in a message to the list
> requesting help with a linphone problem with receiving but not
> transmitting.  I was instructed to change drivers from oss to alsa, so
> I did.
>
> I am now able to send in several pieces of information that might help
> one of you solve the problem, but it looks like it is in hardware to
> me.  If you reach a different conclusion, please let me know.
>
> The first piece of information I can give you is my mixer settings for
> the system.  They are as follows.
>
> vol:87:87:P
> pcm:87:87:P
> speaker:67:67:P
> line:67:67:P
> mic:100:100:P
> cd:67:67:P
> igain:100:100:R
> ogain:67:67:P
> line1:67:67:P
> dig1:67:67:P
> phin:67:67:P
> phout:67:67:P
> video:67:67:P
>
> Believe it or not, the thing sounds better with this microphone
> setting set to P rather than R.   Go figure that one out.
>
> Now comes the ~/.linphonec file which has been created in my home
> directory.
>
> [net]
> con_type=3
> use_nat=0
>
> [sip]
> sip_port=5060
> guess_hostname=1
> contact=sip:knoppix@unknown-host
> use_info=0
> use_ipv6=0
> default_proxy=-1
>
> [rtp]
> audio_rtp_port=7078
> video_rtp_port=9078
> audio_jitt_comp=128
> video_jitt_comp=128
>
> [sound]
> playback_dev_id=0
> capture_dev_id=0
> rec_lev=100
> play_lev=87
> source=m
> local_ring=/usr/share/sounds/linphone/rings/oldphone.wav
> remote_ring=/usr/share/sounds/linphone/ringback.wav
>
> [video]
> enabled=0
> show_local=0
>
> [audio_codec_0]
> mime=PCMU
> rate=8000
> enabled=1
>
> [audio_codec_1]
> mime=GSM
> rate=8000
> enabled=1
>
> [audio_codec_2]
> mime=PCMA
> rate=8000
> enabled=1
>
> [audio_codec_3]
> mime=speex
> rate=8000
> enabled=1
>
> [audio_codec_4]
> mime=speex
> rate=16000
> enabled=1
>
> [audio_codec_5]
> mime=1015
> rate=8000
> enabled=1
>
>
>
> Now, I can give you two sets of complete debugging information for an
> attempted call to sip:[hidden email], which is supposed to be an
> echo test for latency.  It also, unfortunately, tells me that,
> regardless of the driver set I use, there is nothing going out the
> door here.
>
>
> First of all, with the oss drivers.
>  oRTP:(GLogLevel=32)** oRTP-0.7.1initialized.
> MediaStreamer:(GLogLevel=32)** Found /dev/dsp.
> MediaStreamer:(GLogLevel=16)** dsp block size set to 2048.
>
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <eXutils.c: 416> Outgoing interface to reach 15.128.128.93 is
> | 66.169.86.132.
> |
> | INFO1 | <eXutils.c: 416> Outgoing interface to reach fwd.pulver.com is
> | 66.169.86.132.
> |
> | INFO1 | <eXutils.c: 416> Outgoing interface to reach fwd.pulver.com is
> | 66.169.86.132.
> |
> | INFO2 | <osip_transaction.c: 129> allocating transaction ressource 1
> | 1739435548 INFO2 | <ict.c: 35> allocating ICT context
> | INFO2 | <eXutils.c: 511> Not an IPv4 or IPv6 address: fwd.pulver.com
> | INFO2 | <eXutils.c: 541> DNS resolution with fwd.pulver.com:5060
> | INFO1 | <jcallback.c: 148> Message sent:
>
> INVITE sip:[hidden email] SIP/2.0
> Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK211929910
> From: <sip:knoppix@66.169.86.132>;tag=143564218
> To: <sip:[hidden email]>
> Call-ID: 1739435548@66.169.86.132
> CSeq: 20 INVITE
> Contact: <sip:knoppix@66.169.86.132:5060>
> Max-Forwards: 5
> User-Agent: Linphone-1.1.0/eXosip
> Subject: Phone call
> Expires: 120
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY,
> MESSAGE Content-Type: application/sdp
> Content-Length:   357
>
> v=0
> o=knoppix 123456 654321 IN IP4 66.169.86.132
> s=A conversation
> c=IN IP4 66.169.86.132
> t=0 0
> m=audio 7078 RTP/AVP 0 3 8 110 111 115 101
> b=AS:20
> 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
>  (len=16 sizeof(addr)=128 28)
>
> | INFO1 | <jcallback.c: 515> cb_sndinvite (id=1)
> | INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:457403!
> | INFO1 | <udp.c: 2193> Received message:
>
> SIP/2.0 100 trying -- your call is important to us
> Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK211929910
> From: <sip:knoppix@66.169.86.132>;tag=143564218
> To: <sip:[hidden email]>
> Call-ID: 1739435548@66.169.86.132
> CSeq: 20 INVITE
> Server: Sip EXpress router (0.8.14-6 (i386/linux))
> Content-Length: 0
>
>
> | INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:1739435548
> | INFO1 | <jcallback.c: 601> cb_rcv1xx (id=1)
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <udp.c: 2193> Received message:
>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK211929910
> From: <sip:knoppix@66.169.86.132>;tag=143564218
> To: <sip:[hidden email]>;tag=as21924a5a
> Call-ID: 1739435548@66.169.86.132
> CSeq: 20 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:613@69.90.168.13:5028>
> Content-Length: 0
>
>
> | INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:1739435548
> | INFO1 | <jcallback.c: 601> cb_rcv1xx (id=1)
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
>
> LinphoneCore:(GLogLevel=32)** CALL_RINGING
>
> LinphoneCore:(GLogLevel=32)** Remote ringing...
> MediaStreamer:(GLogLevel=16)** dsp block size set to 2048.
> MediaStreamer:(GLogLevel=32)** ms_filter_add_link: ringplay,0 -> OssWrite,0
> MediaStreamer:(GLogLevel=32)** Opening sound card [/dev/dsp (Open Sound
> System)] in playback mode with stereo=0,rate=8000,bits=16
> MediaStreamer:(GLogLevel=16)** dsp block size set to 2048.
> MediaStreamer:(GLogLevel=32)** dsp blocksize is 2048.
>
> | INFO1 | <udp.c: 2193> Received message:
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK211929910
> Record-Route: <sip:613@69.90.155.70;ftag=143564218;lr=on>
> From: <sip:knoppix@66.169.86.132>;tag=143564218
> To: <sip:[hidden email]>;tag=as21924a5a
> Call-ID: 1739435548@66.169.86.132
> CSeq: 20 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:613@69.90.168.13:5028>
> Content-Type: application/sdp
> Content-Length: 239
>
> v=0
> o=root 11782 11782 IN IP4 69.90.168.13
> s=session
> c=IN IP4 69.90.168.13
> t=0 0
> m=audio 17896 RTP/AVP 0 3 101
> a=rtpmap:0 PCMU/8000
> 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:1739435548
> | INFO1 | <jcallback.c: 1213> cb_rcv2xx (id=1)
> | INFO1 | <eXutils.c: 416> Outgoing interface to reach 69.90.168.13 is
> | 66.169.86.132.
> |
> | INFO2 | <eXutils.c: 492> IPv4 address detected: 69.90.155.70
> | INFO2 | <eXutils.c: 541> DNS resolution with 69.90.155.70:5060
> | INFO1 | <jcallback.c: 148> Message sent:
>
> ACK sip:613@69.90.168.13:5028 SIP/2.0
> Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK1382328653
> Route: <sip:613@69.90.155.70;ftag=143564218;lr=on>
> From: <sip:knoppix@66.169.86.132>;tag=143564218
> To: <sip:[hidden email]>;tag=as21924a5a
> Call-ID: 1739435548@66.169.86.132
> CSeq: 20 ACK
> Contact: <sip:knoppix@66.169.86.132:5060>
> Max-Forwards: 5
> User-Agent: Linphone-1.1.0/eXosip
> Content-Length: 0
>
>  (len=16 sizeof(addr)=128 28)
>
> | INFO1 | <jcallback.c: 189> cb_ict_kill_transaction (id=1)
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
>
> LinphoneCore:(GLogLevel=32)** CALL_ANSWERED
>
> MediaStreamer:(GLogLevel=32)** Mediastreamer processing thread is exiting.
> MediaStreamer:(GLogLevel=32)** Closing writing channel of soundcard.
> MediaStreamer:(GLogLevel=32)** ms_filter_add_link: OssRead,0 ->
> MULAWEncoder,0 MediaStreamer:(GLogLevel=32)** ms_filter_add_link:
> MULAWEncoder,0 -> RTPSend,0 MediaStreamer:(GLogLevel=32)**
> ms_filter_add_link: RTPRecv,0 -> MULAWDecoder,0
> MediaStreamer:(GLogLevel=32)** ms_filter_add_link: MULAWDecoder,0 ->
> OssWrite,0 MediaStreamer:(GLogLevel=32)** Opening sound card [/dev/dsp
> (Open Sound System)] in capture mode with stereo=0,rate=8000,bits=16
> MediaStreamer:(GLogLevel=16)** dsp block size set to 2048.
> MediaStreamer:(GLogLevel=32)** dsp blocksize is 2048.
> MediaStreamer:(GLogLevel=32)** Opening sound card [/dev/dsp (Open Sound
> System)] in playback mode with stereo=0,rate=8000,bits=16
> LinphoneCore:(GLogLevel=32)** CALL_STARTAUDIO
>
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <eXutils.c: 416> Outgoing interface to reach 69.90.168.13 is
> | 66.169.86.132.
> |
> | INFO2 | <osip_transaction.c: 129> allocating transaction ressource 2
> | 1739435548 INFO2 | <nict.c: 36> allocating NICT context
> | INFO2 | <eXutils.c: 492> IPv4 address detected: 69.90.155.70
> | INFO2 | <eXutils.c: 541> DNS resolution with 69.90.155.70:5060
> | INFO1 | <jcallback.c: 148> Message sent:
>
> BYE sip:613@69.90.168.13:5028 SIP/2.0
> Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK6617564
> Route: <sip:613@69.90.155.70;ftag=143564218;lr=on>
> From: <sip:knoppix@66.169.86.132>;tag=143564218
> To: <sip:[hidden email]>;tag=as21924a5a
> Call-ID: 1739435548@66.169.86.132
> CSeq: 21 BYE
> Contact: <sip:knoppix@66.169.86.132:5060>
> Max-Forwards: 5
> User-Agent: Linphone-1.1.0/eXosip
> Content-Length: 0
>
>  (len=16 sizeof(addr)=128 28)
>
> | INFO1 | <jcallback.c: 530> cb_sndbye (id=2)
> | INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:495101!
>
> MediaStreamer:(GLogLevel=32)** Mediastreamer processing thread is exiting.
> MediaStreamer:(GLogLevel=32)** Closing reading channel of soundcard.
> MediaStreamer:(GLogLevel=32)** Closing writing channel of soundcard.
> oRTP-stats:(GLogLevel=32)**
>    Global statistics :
>  number of rtp packet sent=0
>  number of rtp bytes sent=0 bytes
>  number of rtp packet received=784
>  number of rtp bytes received=134848 bytes
>  number of incoming rtp bytes successfully delivered to the
> application=134676 number of times the application queried a packet that
> didn't exist=1424 number of rtp packets received too late=0
>  number of rtp packets skipped=1
>  number of bad formatted rtp packets=0
>  number of packet discarded because of queue overflow=0
>
> | INFO1 | <udp.c: 2193> Received message:
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK6617564
> Record-Route: <sip:613@69.90.155.70;ftag=143564218;lr=on>
> From: <sip:knoppix@66.169.86.132>;tag=143564218
> To: <sip:[hidden email]>;tag=as21924a5a
> Call-ID: 1739435548@66.169.86.132
> CSeq: 21 BYE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:613@69.90.168.13:5028>
> Content-Length: 0
>
>
> | INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:1739435548
> | INFO1 | <jcallback.c: 1213> cb_rcv2xx (id=2)
> | INFO1 | <eXosip.c: 339> eXosip: timer sec:4 usec:999923!
> | INFO1 | <eXosip.c: 281> Release a terminated transaction
> |  BUG  | <osip_transaction.c: 263> transaction already removed from list
> | 1! INFO2 | <osip_transaction.c: 286> free transaction ressource 1
> | 1739435548 INFO2 | <ict.c: 112> free ict ressource
> | INFO2 | <osip_transaction.c: 286> free transaction ressource 2 1739435548
> | INFO2 | <nict.c: 110> free nict ressource
>
> This is what I previously sent in so that you could tell me what was
> wrong was that I was using the incorrect sound drivers.  I changed
> them from oss or kernel to the alsa drivers and, for the same call,
> got these results.
>
> Next, with the alsa drivers.
>
> Ready.
> linphonec> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before
> waking up! linphonec> linphonec> linphonec> linphonec> | INFO1 | <eXosip.c:
> 332> eXosip: Reseting timer to 15s before waking up! Contacting
> sip:[hidden email]
>
> | INFO1 | <eXutils.c: 416> Outgoing interface to reach 15.128.128.93 is
> | 66.169.86.132.
> |
> | INFO1 | <eXutils.c: 416> Outgoing interface to reach fwd.pulver.com is
> | 66.169.86.132.
> |
> | INFO1 | <eXutils.c: 416> Outgoing interface to reach fwd.pulver.com is
> | 66.169.86.132.
> |
> | INFO2 | <osip_transaction.c: 129> allocating transaction ressource 1
> | 688034463 INFO2 | <ict.c: 35> allocating ICT context
> | INFO2 | <eXutils.c: 511> Not an IPv4 or IPv6 address: fwd.pulver.com
>
> linphonec> | INFO2 | <eXutils.c: 541> DNS resolution with
> fwd.pulver.com:5060
>
> | INFO1 | <jcallback.c: 148> Message sent:
>
> INVITE sip:[hidden email] SIP/2.0
> Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK1708706045
> From: <sip:knoppix@66.169.86.132>;tag=1625521769
> To: <sip:[hidden email]>
> Call-ID: 688034463@66.169.86.132
> CSeq: 20 INVITE
> Contact: <sip:knoppix@66.169.86.132:5060>
> Max-Forwards: 5
> User-Agent: Linphone-1.1.0/eXosip
> Subject: Phone call
> Expires: 120
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY,
> MESSAGE Content-Type: application/sdp
> Content-Length:   357
>
> v=0
> o=knoppix 123456 654321 IN IP4 66.169.86.132
> s=A conversation
> c=IN IP4 66.169.86.132
> t=0 0
> m=audio 7078 RTP/AVP 0 3 8 110 111 115 101
> b=AS:20
> 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
>  (len=16 sizeof(addr)=128 28)
>
> | INFO1 | <jcallback.c: 515> cb_sndinvite (id=1)
> | INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:473151!
> | INFO1 | <udp.c: 2193> Received message:
>
> SIP/2.0 100 trying -- your call is important to us
> Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK1708706045
> From: <sip:knoppix@66.169.86.132>;tag=1625521769
> To: <sip:[hidden email]>
> Call-ID: 688034463@66.169.86.132
> CSeq: 20 INVITE
> Server: Sip EXpress router (0.8.14-6 (i386/linux))
> Content-Length: 0
>
>
> | INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:688034463
> | INFO1 | <jcallback.c: 601> cb_rcv1xx (id=1)
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <udp.c: 2193> Received message:
>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK1708706045
> From: <sip:knoppix@66.169.86.132>;tag=1625521769
> To: <sip:[hidden email]>;tag=as079da4b6
> Call-ID: 688034463@66.169.86.132
> CSeq: 20 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:613@69.90.168.13:5028>
> Content-Length: 0
>
>
> | INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:688034463
> | INFO1 | <jcallback.c: 601> cb_rcv1xx (id=1)
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <udp.c: 2193> Received message:
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK1708706045
> Record-Route: <sip:613@69.90.155.70;ftag=1625521769;lr=on>
> From: <sip:knoppix@66.169.86.132>;tag=1625521769
> To: <sip:[hidden email]>;tag=as079da4b6
> Call-ID: 688034463@66.169.86.132
> CSeq: 20 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:613@69.90.168.13:5028>
> Content-Type: application/sdp
> Content-Length: 239
>
> v=0
> o=root 22055 22055 IN IP4 69.90.168.13
> s=session
> c=IN IP4 69.90.168.13
> t=0 0
> m=audio 10144 RTP/AVP 0 3 101
> a=rtpmap:0 PCMU/8000
> 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:688034463
> | INFO1 | <jcallback.c: 1213> cb_rcv2xx (id=1)
> | INFO1 | <eXutils.c: 416> Outgoing interface to reach 69.90.168.13 is
> | 66.169.86.132.
> |
> | INFO2 | <eXutils.c: 492> IPv4 address detected: 69.90.155.70
> | INFO2 | <eXutils.c: 541> DNS resolution with 69.90.155.70:5060
> | INFO1 | <jcallback.c: 148> Message sent:
>
> ACK sip:613@69.90.168.13:5028 SIP/2.0
> Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK1011056904
> Route: <sip:613@69.90.155.70;ftag=1625521769;lr=on>
> From: <sip:knoppix@66.169.86.132>;tag=1625521769
> To: <sip:[hidden email]>;tag=as079da4b6
> Call-ID: 688034463@66.169.86.132
> CSeq: 20 ACK
> Contact: <sip:knoppix@66.169.86.132:5060>
> Max-Forwards: 5
> User-Agent: Linphone-1.1.0/eXosip
> Content-Length: 0
>
>  (len=16 sizeof(addr)=128 28)
>
> | INFO1 | <jcallback.c: 189> cb_ict_kill_transaction (id=1)
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
>
> Connected.
>
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <eXutils.c: 416> Outgoing interface to reach 69.90.168.13 is
> | 66.169.86.132.
> |
> | INFO2 | <osip_transaction.c: 129> allocating transaction ressource 2
> | 688034463 INFO2 | <nict.c: 36> allocating NICT context
> | INFO2 | <eXutils.c: 492> IPv4 address detected: 69.90.155.70
> | INFO2 | <eXutils.c: 541> DNS resolution with 69.90.155.70:5060
> | INFO1 | <jcallback.c: 148> Message sent:
>
> BYE sip:613@69.90.168.13:5028 SIP/2.0
> Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK1417467515
> Route: <sip:613@69.90.155.70;ftag=1625521769;lr=on>
> From: <sip:knoppix@66.169.86.132>;tag=1625521769
> To: <sip:[hidden email]>;tag=as079da4b6
> Call-ID: 688034463@66.169.86.132
> CSeq: 21 BYE
> Contact: <sip:knoppix@66.169.86.132:5060>
> Max-Forwards: 5
> User-Agent: Linphone-1.1.0/eXosip
> Content-Length: 0
>
>  (len=16 sizeof(addr)=128 28)
>
> | INFO1 | <jcallback.c: 530> cb_sndbye (id=2)
> | INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:494971!
>
> Communication ended.
> linphonec> | INFO1 | <udp.c: 2193> Received message:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 66.169.86.132:5060;rport=5060;branch=z9hG4bK1417467515
> Record-Route: <sip:613@69.90.155.70;ftag=1625521769;lr=on>
> From: <sip:knoppix@66.169.86.132>;tag=1625521769
> To: <sip:[hidden email]>;tag=as079da4b6
> Call-ID: 688034463@66.169.86.132
> CSeq: 21 BYE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:613@69.90.168.13:5028>
> Content-Length: 0
>
>
> | INFO3 | <osip_event.c: 84> MESSAGE REC. CALLID:688034463
> | INFO1 | <jcallback.c: 1213> cb_rcv2xx (id=2)
> | INFO1 | <eXosip.c: 339> eXosip: timer sec:4 usec:999923!
> | INFO1 | <eXosip.c: 339> eXosip: timer sec:0 usec:1122!
> | INFO1 | <jcallback.c: 217> cb_nict_kill_transaction (id=2)
> | INFO2 | <udp.c: 2369> eXosip: eXosip_release_finished_calls remove a
> | dialog INFO1 | <udp.c: 2604> eXosip: remove a call
> | INFO1 | <udp.c: 2620> Release a terminated transaction
> |  BUG  | <osip_transaction.c: 263> transaction already removed from list
> | 1! INFO2 | <osip_transaction.c: 286> free transaction ressource 1
> | 688034463 INFO2 | <ict.c: 112> free ict ressource
> | INFO1 | <udp.c: 2620> Release a terminated transaction
> |  BUG  | <osip_transaction.c: 263> transaction already removed from list
> | 2! INFO2 | <osip_transaction.c: 286> free transaction ressource 2
> | 688034463 INFO2 | <nict.c: 110> free nict ressource
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
> | INFO1 | <eXosip.c: 332> eXosip: Reseting timer to 15s before waking up!
>
> linphonec> linphonec>
>
> When I tried this, I got no better results than with the original oss
> drivers.  I could still hear the incoming information from the other
> machine, but the server could not hear me.  Nothing was going out from
> here.
>
> Now, when I got my freeworld-dialup account, a friend of mine, in
> fact, a fellow member of the Oralux development team gave me this
> little snippit to put into my ~/.linphonec file.  I haven't inserted
> it as of yet, but I need to ask if this could be of any help.  The
> user names and passwords are left out so that I can send this across
> the internet.  In fact, he did not put in his own passwords and
> usernames for the same security concerns.  Is this necessary to make
> things work properly?
>
> [sip]
> sip_port=5060
> guess_hostname=1
> contact=sip:knoppix@unknown-host
> use_info=0
> use_ipv6=0
> default_proxy=-1
> username=123456
> hostname=fwd.pulver.com
> sip_port=5060
> use_registrar=1
> as_proxy=1
> expires=900
> registrar=sip:fwdnat.pulver.com:5082
> passwd=pass
> addr_of_rec=sip:[hidden email]
>
> Well, I hope you can help with some of this with the information
> supplied here.  I hope to include linphone in a future release of the
> operating system I and my acquaintance are working on.  It is Oralux
> which is Debian based.  This means that I am using the linphone Debian
> package from the unstable distribution.  The version of linphone is:
>
> version: 1.1.0
>
> I do not know if this can help or not, but I will include it here.
>
> I would like to get Linphone working so that I can help others with
> their computing situations, build my web site and include it as a
> communications option, and include it in the new release of Oralux,
> the distribution for the Blind.  Please visit:
>
> http://oralux.org
>
>
>
> Thank you in advance for helping me.

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