[Linphone-users] changed drivers, but still no go.

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[Linphone-users] changed drivers, but still no go.

From:	Doug Smith
Subject:	[Linphone-users] changed drivers, but still no go.
Date:	Thu, 10 Nov 2005 14:06:52 -0500
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:address@hidden
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:address@hidden, 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:address@hidden SIP/2.0

Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK211929910

From: <sip:address@hidden>;tag=143564218

To: <sip:address@hidden>

Call-ID: address@hidden

CSeq: 20 INVITE

Contact: <sip:address@hidden: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:address@hidden>;tag=143564218

To: <sip:address@hidden>

Call-ID: address@hidden

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:address@hidden>;tag=143564218

To: <sip:address@hidden>;tag=as21924a5a

Call-ID: address@hidden

CSeq: 20 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:address@hidden: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:address@hidden;ftag=143564218;lr=on>

From: <sip:address@hidden>;tag=143564218

To: <sip:address@hidden>;tag=as21924a5a

Call-ID: address@hidden

CSeq: 20 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:address@hidden: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:address@hidden:5028 SIP/2.0

Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK1382328653

Route: <sip:address@hidden;ftag=143564218;lr=on>

From: <sip:address@hidden>;tag=143564218

To: <sip:address@hidden>;tag=as21924a5a

Call-ID: address@hidden

CSeq: 20 ACK

Contact: <sip:address@hidden: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:address@hidden:5028 SIP/2.0

Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK6617564

Route: <sip:address@hidden;ftag=143564218;lr=on>

From: <sip:address@hidden>;tag=143564218

To: <sip:address@hidden>;tag=as21924a5a

Call-ID: address@hidden

CSeq: 21 BYE

Contact: <sip:address@hidden: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:address@hidden;ftag=143564218;lr=on>

From: <sip:address@hidden>;tag=143564218

To: <sip:address@hidden>;tag=as21924a5a

Call-ID: address@hidden

CSeq: 21 BYE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:address@hidden: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:address@hidden
| 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:address@hidden SIP/2.0

Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK1708706045

From: <sip:address@hidden>;tag=1625521769

To: <sip:address@hidden>

Call-ID: address@hidden

CSeq: 20 INVITE

Contact: <sip:address@hidden: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:address@hidden>;tag=1625521769

To: <sip:address@hidden>

Call-ID: address@hidden

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:address@hidden>;tag=1625521769

To: <sip:address@hidden>;tag=as079da4b6

Call-ID: address@hidden

CSeq: 20 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:address@hidden: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:address@hidden;ftag=1625521769;lr=on>

From: <sip:address@hidden>;tag=1625521769

To: <sip:address@hidden>;tag=as079da4b6

Call-ID: address@hidden

CSeq: 20 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:address@hidden: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:address@hidden:5028 SIP/2.0

Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK1011056904

Route: <sip:address@hidden;ftag=1625521769;lr=on>

From: <sip:address@hidden>;tag=1625521769

To: <sip:address@hidden>;tag=as079da4b6

Call-ID: address@hidden

CSeq: 20 ACK

Contact: <sip:address@hidden: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:address@hidden:5028 SIP/2.0

Via: SIP/2.0/UDP 66.169.86.132:5060;rport;branch=z9hG4bK1417467515

Route: <sip:address@hidden;ftag=1625521769;lr=on>

From: <sip:address@hidden>;tag=1625521769

To: <sip:address@hidden>;tag=as079da4b6

Call-ID: address@hidden

CSeq: 21 BYE

Contact: <sip:address@hidden: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:address@hidden;ftag=1625521769;lr=on>

From: <sip:address@hidden>;tag=1625521769

To: <sip:address@hidden>;tag=as079da4b6

Call-ID: address@hidden

CSeq: 21 BYE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:address@hidden: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:address@hidden
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:address@hidden

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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