[Linphone-users] problems with asterisk

Discussion:

Jon Lawrence

2004-01-14 12:37:37 UTC

Hi,
I'm trying to use linphone to make sip calls through my asterisk server.
Asterisk version is 0.5.0
linphone version is 0.12.1
linux distro is gentoo
kernel is 2.4.23

I can register to the * server without problems but whenever I attempt to make
a call I get a '407 proxy authentication required' message.

Anyone any ideas what I can do ?

I don't know if this is a problem with my asterisk setup or with linphone.

TIA
Jon Lawrence

Jason A. Pattie

2004-01-14 16:08:27 UTC

Permalink

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Jon Lawrence wrote:
| Hi,
| I'm trying to use linphone to make sip calls through my asterisk server.
| Asterisk version is 0.5.0
| linphone version is 0.12.1
| linux distro is gentoo
| kernel is 2.4.23
|
| I can register to the * server without problems but whenever I attempt
to make
| a call I get a '407 proxy authentication required' message.
|
| Anyone any ideas what I can do ?
|
| I don't know if this is a problem with my asterisk setup or with linphone.

I get the same thing when attempting to use linphone on an iPAQ, but I
also get a 401 Unauthorized error message before the 407 error message.
~ I've tried (in sip.conf) using 'host=dynamic' and 'host=<ip address>' +
'defaultip=<ip address>' with the same results. I'm also receiving a
415 error message (sometimes) saying that the media type is not
recognized/supported.

- --
Jason A. Pattie
***@xperienceinc.com
Xperience, Inc. (http://www.xperienceinc.com)
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Tom Poindexter

2004-01-14 20:09:01 UTC

Permalink

Note that the Asterisk folks have just released 0.7.0, and there are a number
of SIP related fixes. You probably want to download and try out the newer
version.

-tp

--
Tom Poindexter
***@nyx.net
http://www.nyx.net/~tpoindex/

Jason A. Pattie

2004-01-14 23:03:36 UTC

Permalink

Tom Poindexter

2004-01-15 05:11:00 UTC

Permalink

| Note that the Asterisk folks have just released 0.7.0, and there are a
number
| of SIP related fixes. You probably want to download and try out the newer
| version.
That's exactly what I just did (actually, upgraded to the CVS version).
~ For the first time in a long time, I was actually able to hear scratchy
sounds again from linphone-gpe! It worked the first time I ever tried
it, about a month ago, and then didn't ever work after that. Now that I
upgraded Asterisk, it worked again. Here's what happened when I
attempted to check voicemail (ext. 8500) from linphone-gpe (hope it's

Excellent. Hehe - asterisk 0.7.1 was released just after I posted here,
although I don't think there are SIP related changes. Probaby still worth
downloading again.

I got linphone to work one-way with asterisk, I haven't yet installed the
ALSA drivers, but audio from asterisk sounds great. I'm also playing with
Gnophone, and have two way audio after some hacks. There's also a couple
other IAX clients, iaxclient.sf.net and tel.sf.net. tel (tkPhone) needed some
hacking too, and an improved client seems to be under development, see
http://www.linuxjournal.com/article.php?sid=6769

-tp

--
Tom Poindexter
***@nyx.net
http://www.nyx.net/~tpoindex/

Jason A. Pattie

2004-01-15 15:52:27 UTC

Permalink

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Tom Poindexter wrote:
| Excellent. Hehe - asterisk 0.7.1 was released just after I posted here,
| although I don't think there are SIP related changes. Probaby still worth
| downloading again.

Guess I'll upgrade again, then. I only saw that 0.7.0 was out in the
Debian feeds (although I built Asterisk from CVS).

| I got linphone to work one-way with asterisk, I haven't yet installed the
| ALSA drivers, but audio from asterisk sounds great. I'm also playing with
| Gnophone, and have two way audio after some hacks. There's also a couple
| other IAX clients, iaxclient.sf.net and tel.sf.net. tel (tkPhone)
needed some
| hacking too, and an improved client seems to be under development, see
| http://www.linuxjournal.com/article.php?sid=6769

I can only use ALSA (since I bought a Plantronics DSP 400 USB headset)
and now it seems that the default sound system for the familiar
distribution is ALSA as well. I've tried all sorts of other software
phones and none work properly with the USB headset except (almost)
linphone. linphone seems to work the best as far as audio is concerned.
~ All the rest give completely garbled sound that pulsates for the
duration of the sound prompts, etc. and is completely illegible (I know
that's the word for not being able to understand writing; don't know
what the word is for not being able to understand sound or garbled
sound, but this is more than just garbled. Nothing is able to be made
out). However, sound being transmitted from the microphone of the USB
headset reaches the other end flawlessly and sounds excellent.

On another note, would it be possible to add IAX/IAX2 support to
linphone? I know linphone is a SIP phone, but it would be nice to be
able to choose IAX as well as SIP since IAX would be more efficient in
the ability to traverse NAT'd networks and is supposed to take less
overhead, etc. I am aware that your choices of services and
connectivity would currently be limited to only gateways and devices
that support IAX, but having a software phone that could do both would
be really neat (and linphone is the only one that supports ALSA right
now that I know of, except kphone, but it doesn't have DTMF digit
support or a keypad, yet, that I know of).

- --
Jason A. Pattie
***@xperienceinc.com
Xperience, Inc. (http://www.xperienceinc.com)
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Tom Poindexter

2004-01-15 17:29:28 UTC

Permalink

Post by Jason A. Pattie
I can only use ALSA (since I bought a Plantronics DSP 400 USB headset)
and now it seems that the default sound system for the familiar
distribution is ALSA as well. I've tried all sorts of other software
phones and none work properly with the USB headset except (almost)
linphone. linphone seems to work the best as far as audio is concerned.

Re: USB headsets - I just bought a Logitech 300 USB headset, it seems to work
for me, but not with linphone since linphone doesn't play well with OSS.
The in-line mute and volume controls don't work, but I can live with that.
I'm using the stock OSS & usb drivers with RedHat 9. I had to add a /dev
node for /dev/dsp2 (14,35), my internal sound hardware takes /dev/dsp and
/dev/dsp1. When I have more free cycles, I'll finally get the ALSA
drivers installed and try the various combinations. I'd rather have an *
softphone solution that doesn't require extra drivers.

I've had the best luck with Gnophone as a softphone for *, still looking for
a better solution.

--
Tom Poindexter
***@nyx.net
http://www.nyx.net/~tpoindex/

Jason A. Pattie

2004-01-16 16:46:39 UTC

Permalink

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Tom Poindexter wrote:
| Re: USB headsets - I just bought a Logitech 300 USB headset, it seems
to work

I tried the Logitech 200 and it was terrible, under both Linux and
Win2K, and I thought it would work great under Win2K. The microphone
was not sensitive at all unless you literally stuck it inside your
mouth. But who's going to do that? And it kept rebooting my laptop
(even under Win2K). I don't know if that's because of the load or what
or if my laptop has USB/hardware problems. I didn't attempt to use it
with ALSA at that time.

| for me, but not with linphone since linphone doesn't play well with OSS.
| The in-line mute and volume controls don't work, but I can live with that.
| I'm using the stock OSS & usb drivers with RedHat 9. I had to add a /dev
| node for /dev/dsp2 (14,35), my internal sound hardware takes /dev/dsp and
| /dev/dsp1. When I have more free cycles, I'll finally get the ALSA
| drivers installed and try the various combinations. I'd rather have an *
| softphone solution that doesn't require extra drivers.
|
|
| I've had the best luck with Gnophone as a softphone for *, still
looking for
| a better solution.

Me too. Let me know if Gnophone works with your Logitech USB headset
under ALSA. I've noticed that Gnophone works the best with OSS drivers,
not ALSA. Something to do (possibly) with on-the-fly bitrate
conversions or something in the OSS emulation support for ALSA.

If you like the IAX protocol, you might try iaxComm (iaxclient.sf.net)
or under Windows DIAX. They both use the iaxClient libraries (I think).

- --
Jason A. Pattie
***@xperienceinc.com
Xperience, Inc. (http://www.xperienceinc.com)
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Martin List-Petersen

2004-01-19 10:19:07 UTC

Permalink

Post by Tom Poindexter

Re: USB headsets - I just bought a Logitech 300 USB headset, it seems to
work for me, but not with linphone since linphone doesn't play well with OSS.

I'm not sure, what headset you are talking about. The headset's that i know of
and which Logitech advertises on their website are the "Premium Stereo USB
Headset 30" and the "Stereo USB Headset 20". I've got the Premium Stereo USB
Headset 30, which works very well with Linux and OSS drivers.

Post by Tom Poindexter
The in-line mute and volume controls don't work, but I can live with that.

If you've got the Premium 30 the in-line mute should work, since that is done
inside the headset. For the volume buttons you need a little application, that
reads the input events from usb and applies those to your mixer.

Post by Tom Poindexter
I've had the best luck with Gnophone as a softphone for *, still looking
for a better solution.

kphone does also work, however i would love to get linphone up and running with
it.

Regards,
Martin List-Petersen
martin at list-petersen dot se
--
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-- William Shakespeare, "The Merchant of Venice"

Simon MORLAT

2004-01-16 13:13:02 UTC

Permalink

That's strange. People reported that authentifed INVITE worked with
Asterisk server.
Can you attach a log file so that we see the sip messages ?
use the -d 5 option if you are running linphonec.
Simon

Post by Jon Lawrence
Hi,
I'm trying to use linphone to make sip calls through my asterisk server.
Asterisk version is 0.5.0
linphone version is 0.12.1
linux distro is gentoo
kernel is 2.4.23
I can register to the * server without problems but whenever I attempt to make
a call I get a '407 proxy authentication required' message.
Anyone any ideas what I can do ?
I don't know if this is a problem with my asterisk setup or with linphone.
TIA
Jon Lawrence
_______________________________________________
Linphone-users mailing list
http://mail.nongnu.org/mailman/listinfo/linphone-users

Simon MORLAT

2004-01-16 14:00:22 UTC

Permalink

Not sure if this what you need or not ?

Yes it is......

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Proxy-Authenticate: Digest realm="asterisk", nonce="0e7bcddd"
Content-Length: 0

The "Contact:" field given by Asterisk is empty. Asterisk server should
put a valid Contact header in its response, as far as I know.
Does anybody talk about that with Asterisk's maintainers ?
Simon

Jon Lawrence

2004-01-16 15:24:40 UTC

Permalink

Post by Simon MORLAT

Not sure if this what you need or not ?

Yes it is......

The "Contact:" field given by Asterisk is empty. Asterisk server should
put a valid Contact header in its response, as far as I know.
Does anybody talk about that with Asterisk's maintainers ?
Simon

I'll ask on the Astrerisk IRC.
I assume that this isn't sometihng that needs to go in a config file somewhere
on the * server.

Jon

Jon Lawrence

2004-01-16 13:24:15 UTC

Permalink

Post by Simon MORLAT

Post by Jon Lawrence
I'm trying to use linphone to make sip calls through my asterisk server.
Asterisk version is 0.5.0
linphone version is 0.12.1
linux distro is gentoo
kernel is 2.4.23
I can register to the * server without problems but whenever I attempt to
make a call I get a '407 proxy authentication required' message.
Anyone any ideas what I can do ?
I don't know if this is a problem with my asterisk setup or with linphone.
TIA
Jon Lawrence

That's strange. People reported that authentifed INVITE worked with
Asterisk server.
Can you attach a log file so that we see the sip messages ?
use the -d 5 option if you are running linphonec.

Not sure if this what you need or not ?

Jon

bash-2.05b$ linphone -d 5
| INFO1 | <osipua.c: 65> Starting osip stack and osipua layer.

| INFO1 | <udp.c: 112> Entering osipua thread.

MediaStreamer-Message: Found /dev/dsp.

(linphone:1699): MediaStreamer-WARNING **: dsp block size set to 16384.
| INFO1 | <osipmanager.c: 148> port already listened

| INFO1 | <utils.c: 409> Outgoing interface to reach 10.168.4.1 is
10.168.4.100.

| INFO1 | <osipmanager.c: 148> port already listened

| INFO1 | <udp.c: 295> Sending message:
REGISTER sip:10.168.4.1 SIP/2.0
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK858617539
From: <sip:***@10.168.4.1>;tag=181171502
To: <sip:***@10.168.4.1>;tag=181171502
Call-ID: ***@10.168.4.100
CSeq: 0 REGISTER
Contact: <sip:***@10.168.4.100>
max-forwards: 10
expires: 900
user-agent: oSIP/Linphone-0.12.1
Content-Length: 0

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK858617539
From: <sip:***@10.168.4.1>;tag=181171502
To: <sip:***@10.168.4.1>;tag=181171502
Call-ID: ***@10.168.4.100
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:***@10.168.4.1>
Content-Length: 0

| INFO1 | <nict_callbacks.c: 42> OnEvent_New_Incoming1xxResponse!

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK858617539
From: <sip:***@10.168.4.1>;tag=181171502
To: <sip:***@10.168.4.1>;tag=181171502
Call-ID: ***@10.168.4.100
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:***@10.168.4.1>
Proxy-Authenticate: Digest realm="asterisk", nonce="64304b19"
Content-Length: 0

| INFO1 | <nict_callbacks.c: 107> OnEvent_New_Incoming4xxResponse!

| INFO1 | <nict_callbacks.c: 127> User need to authenticate to REGISTER!

| INFO1 | <utils.c: 409> Outgoing interface to reach 10.168.4.1 is
10.168.4.100.

| INFO1 | <authentication.c: 377> Response in proxy_authorization
|1bee09b9b25b49002bd59a4659f20ace|

| INFO1 | <udp.c: 295> Sending message:
REGISTER sip:10.168.4.1 SIP/2.0
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK844368701
From: <sip:***@10.168.4.1>;tag=1675459016
To: <sip:***@10.168.4.1>;tag=1675459016
Call-ID: ***@10.168.4.100
CSeq: 1 REGISTER
Contact: <sip:***@10.168.4.100>
Proxy-Authorization: Digest username="2001", realm="asterisk",
nonce="64304b19", uri="sip:10.168.4.1",
response="1bee09b9b25b49002bd59a4659f20ace", algorithm=MD5
max-forwards: 10
expires: 900
user-agent: oSIP/Linphone-0.12.1
Content-Length: 0

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK844368701
From: <sip:***@10.168.4.1>;tag=1675459016
To: <sip:***@10.168.4.1>;tag=1675459016
Call-ID: ***@10.168.4.100
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:***@10.168.4.1>
Content-Length: 0

| INFO1 | <nict_callbacks.c: 42> OnEvent_New_Incoming1xxResponse!

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK844368701
From: <sip:***@10.168.4.1>;tag=1675459016
To: <sip:***@10.168.4.1>;tag=1675459016
Call-ID: ***@10.168.4.100
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 900
Contact: <sip:***@10.168.4.1>;expires=900
Date: Fri, 16 Jan 2004 13:20:52 GMT
Content-Length: 0

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
NOTIFY sip:***@10.168.4.100 SIP/2.0
Via: SIP/2.0/UDP 10.168.4.1:5060;branch=z9hG4bK56c5fa23
From: "asterisk" <sip:***@10.168.4.1>;tag=as6e21ec3c
To: <sip:***@10.168.4.100>
Contact: <sip:***@10.168.4.1>
Call-ID: ***@10.168.4.1
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 36

Messages-Waiting: no
Voicemail: 0/0

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
NOTIFY sip:***@10.168.4.100 SIP/2.0
Via: SIP/2.0/UDP 10.168.4.1:5060;branch=z9hG4bK56c5fa23
From: "asterisk" <sip:***@10.168.4.1>;tag=as6e21ec3c
To: <sip:***@10.168.4.100>
Contact: <sip:***@10.168.4.1>
Call-ID: ***@10.168.4.1
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 36

Messages-Waiting: no
Voicemail: 0/0

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
NOTIFY sip:***@10.168.4.100 SIP/2.0
Via: SIP/2.0/UDP 10.168.4.1:5060;branch=z9hG4bK56c5fa23
From: "asterisk" <sip:***@10.168.4.1>;tag=as6e21ec3c
To: <sip:***@10.168.4.100>
Contact: <sip:***@10.168.4.1>
Call-ID: ***@10.168.4.1
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 36

Messages-Waiting: no
Voicemail: 0/0

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
NOTIFY sip:***@10.168.4.100 SIP/2.0
Via: SIP/2.0/UDP 10.168.4.1:5060;branch=z9hG4bK56c5fa23
From: "asterisk" <sip:***@10.168.4.1>;tag=as6e21ec3c
To: <sip:***@10.168.4.100>
Contact: <sip:***@10.168.4.1>
Call-ID: ***@10.168.4.1
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 36

Messages-Waiting: no
Voicemail: 0/0

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
NOTIFY sip:***@10.168.4.100 SIP/2.0
Via: SIP/2.0/UDP 10.168.4.1:5060;branch=z9hG4bK56c5fa23
From: "asterisk" <sip:***@10.168.4.1>;tag=as6e21ec3c
To: <sip:***@10.168.4.100>
Contact: <sip:***@10.168.4.1>
Call-ID: ***@10.168.4.1
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 36

Messages-Waiting: no
Voicemail: 0/0

| INFO1 | <nict_callbacks.c: 30> Transaction 1 killed.

| INFO1 | <nict_callbacks.c: 30> Transaction 2 killed.

| INFO1 | <osipdialog.c: 1915> Dialog is removed. It remains 0 dialog(s) in
the ua list.

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
NOTIFY sip:***@10.168.4.100 SIP/2.0
Via: SIP/2.0/UDP 10.168.4.1:5060;branch=z9hG4bK56c5fa23
From: "asterisk" <sip:***@10.168.4.1>;tag=as6e21ec3c
To: <sip:***@10.168.4.100>
Contact: <sip:***@10.168.4.1>
Call-ID: ***@10.168.4.1
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 36

Messages-Waiting: no
Voicemail: 0/0

| INFO1 | <utils.c: 409> Outgoing interface to reach 10.168.4.1 is
10.168.4.100.

| INFO1 | <udp.c: 295> Sending message:
INVITE sip:***@10.168.4.1 SIP/2.0
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
Contact: <sip:***@10.168.4.100>
max-forwards: 10
user-agent: oSIP/Linphone-0.12.1
Content-Type: application/sdp
Content-Length: 369

v=0
o=2001 123456 654321 IN IP4 10.168.4.100
s=A conversation
c=IN IP4 10.168.4.100
t=0 0
m=audio 7078 RTP/AVP 0 3 8 110 111 115 101
b=AS:110 20
b=AS:111 28
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:110 speex/8000/1
a=rtpmap:111 speex/16000/1
a=rtpmap:115 1015/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="0e7bcddd"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="0e7bcddd"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 295> Sending message:
INVITE sip:***@10.168.4.1 SIP/2.0
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
Contact: <sip:***@10.168.4.100>
max-forwards: 10
user-agent: oSIP/Linphone-0.12.1
Content-Type: application/sdp
Content-Length: 369

v=0
o=2001 123456 654321 IN IP4 10.168.4.100
s=A conversation
c=IN IP4 10.168.4.100
t=0 0
m=audio 7078 RTP/AVP 0 3 8 110 111 115 101
b=AS:110 20
b=AS:111 28
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:110 speex/8000/1
a=rtpmap:111 speex/16000/1
a=rtpmap:115 1015/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="5f594b31"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="0e7bcddd"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="5f594b31"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="0e7bcddd"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="5f594b31"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="0e7bcddd"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 295> Sending message:
INVITE sip:***@10.168.4.1 SIP/2.0
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
Contact: <sip:***@10.168.4.100>
max-forwards: 10
user-agent: oSIP/Linphone-0.12.1
Content-Type: application/sdp
Content-Length: 369

v=0
o=2001 123456 654321 IN IP4 10.168.4.100
s=A conversation
c=IN IP4 10.168.4.100
t=0 0
m=audio 7078 RTP/AVP 0 3 8 110 111 115 101
b=AS:110 20
b=AS:111 28
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:110 speex/8000/1
a=rtpmap:111 speex/16000/1
a=rtpmap:115 1015/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="5f594b31"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="4d70df5d"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="0e7bcddd"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="5f594b31"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="4d70df5d"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="5f594b31"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="4d70df5d"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="4d70df5d"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="4d70df5d"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 295> Sending message:
INVITE sip:***@10.168.4.1 SIP/2.0
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
Contact: <sip:***@10.168.4.100>
max-forwards: 10
user-agent: oSIP/Linphone-0.12.1
Content-Type: application/sdp
Content-Length: 369

v=0
o=2001 123456 654321 IN IP4 10.168.4.100
s=A conversation
c=IN IP4 10.168.4.100
t=0 0
m=audio 7078 RTP/AVP 0 3 8 110 111 115 101
b=AS:110 20
b=AS:111 28
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:110 speex/8000/1
a=rtpmap:111 speex/16000/1
a=rtpmap:115 1015/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="75bebf1c"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="4d70df5d"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="75bebf1c"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <utils.c: 409> Outgoing interface to reach 10.168.4.1 is
10.168.4.100.

| INFO1 | <udp.c: 295> Sending message:
REGISTER sip:10.168.4.1 SIP/2.0
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK1005722411
From: <sip:***@10.168.4.1>;tag=231416342
To: <sip:***@10.168.4.1>;tag=231416342
Call-ID: ***@10.168.4.100
CSeq: 0 REGISTER
Contact: <sip:***@10.168.4.100>
max-forwards: 10
expires: 0
user-agent: oSIP/Linphone-0.12.1
Content-Length: 0

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK1005722411
From: <sip:***@10.168.4.1>;tag=231416342
To: <sip:***@10.168.4.1>;tag=231416342
Call-ID: ***@10.168.4.100
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:***@10.168.4.1>
Content-Length: 0

| INFO1 | <nict_callbacks.c: 42> OnEvent_New_Incoming1xxResponse!

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK1005722411
From: <sip:***@10.168.4.1>;tag=231416342
To: <sip:***@10.168.4.1>;tag=231416342
Call-ID: ***@10.168.4.100
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:***@10.168.4.1>
Proxy-Authenticate: Digest realm="asterisk", nonce="06cf980a"
Content-Length: 0

| INFO1 | <nict_callbacks.c: 107> OnEvent_New_Incoming4xxResponse!

| INFO1 | <nict_callbacks.c: 127> User need to authenticate to REGISTER!

| INFO1 | <utils.c: 409> Outgoing interface to reach 10.168.4.1 is
10.168.4.100.

| INFO1 | <authentication.c: 377> Response in proxy_authorization
|f7562e97f0420eefde30321d21f11a53|

| INFO1 | <udp.c: 295> Sending message:
REGISTER sip:10.168.4.1 SIP/2.0
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK603461285
From: <sip:***@10.168.4.1>;tag=4157259184
To: <sip:***@10.168.4.1>;tag=4157259184
Call-ID: ***@10.168.4.100
CSeq: 1 REGISTER
Contact: <sip:***@10.168.4.100>
Proxy-Authorization: Digest username="2001", realm="asterisk",
nonce="06cf980a", uri="sip:10.168.4.1",
response="f7562e97f0420eefde30321d21f11a53", algorithm=MD5
max-forwards: 10
expires: 0
user-agent: oSIP/Linphone-0.12.1
Content-Length: 0

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK603461285
From: <sip:***@10.168.4.1>;tag=4157259184
To: <sip:***@10.168.4.1>;tag=4157259184
Call-ID: ***@10.168.4.100
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:***@10.168.4.1>
Content-Length: 0

| INFO1 | <nict_callbacks.c: 42> OnEvent_New_Incoming1xxResponse!

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK603461285
From: <sip:***@10.168.4.1>;tag=4157259184
To: <sip:***@10.168.4.1>;tag=4157259184
Call-ID: ***@10.168.4.100
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:***@10.168.4.1>
Date: Fri, 16 Jan 2004 13:21:20 GMT
Content-Length: 0

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.4.100:5060;branch=z9hG4bK2587076668
From: <sip:***@10.168.4.1>;tag=1675459016;tag=304916514
To: <sip:***@10.168.4.1>;tag=as6d423d80
Call-ID: ***@10.168.4.100
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="75bebf1c"
Content-Length: 0

| ERROR | <msg_parser.c: 316> Could not set header: "contact"
| ERROR | <msg_parser.c: 532> End of header Not found
| ERROR | <msg_parser.c: 712> error in msg_headers_parse()
| ERROR | <sipevent.c: 76> could not parse message
| INFO1 | <osipua.c: 89> Shutting down osip stack and osipua layer.

| INFO1 | <udp.c: 233> Exiting osipua thread.

closing transaction : 81563b8closing transaction : 8095760closing transaction
: 8095760closing transaction : 0