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Re: [Linphone-users] linphone on a specific IP address


From: Jason A. Pattie
Subject: Re: [Linphone-users] linphone on a specific IP address
Date: Mon, 17 Nov 2003 11:07:48 -0600
User-agent: Mozilla/5.0 (X11; U; Linux i586; en-US; rv:1.3) Gecko/20030327 Debian/1.3-4

Simon MORLAT wrote:

Hello,

Did you try with pre-release availlable at http://simon.morlat.free.fr/download/unstable/source ?
It probably solves your problems. If not come back here.
Simon


:( Still the same problem. I downloaded and installed the debian package for 'unstable' (dated Nov. 6, 2003). Maybe it doesn't have the fixes you think it should (i.e., it seemed to be for pre3 and the source tarball is for pre4). I will attempt to compile pre4, but I don't think I will be able to get the dependencies for the compile worked out, just like last time.

Here are the error messages I received. After the INFO1 <udp.c: 292> Sending message line, the '(null)' you see is right after modifying the configuration preferences screen for SIP where you put in the SIP registrar address and click 'Apply'. Some part of linphone then crashes and GNOME throws up an error with a Segmentation Fault, but the GUI still runs, although it doesn't seem to work properly after that.

| INFO1 | <osipua.c: 65> Starting osip stack and osipua layer.

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

MediaStreamer-Message: Found /dev/dsp.

(linphone:4687): MediaStreamer-WARNING **: oss_card_probe: can't open /dev/dsp: No such device.

(linphone:4687): LinphoneCore-WARNING **: General level is quite low (0). Linphone rises it up for you.
| INFO1 | <osipmanager.c: 148> port already listened

| INFO1 | <utils.c: 408> Outgoing interface to reach 192.168.1.28 is 192.168.20.1.

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

| INFO1 | <udp.c: 292> Sending message:
REGISTER sip:192.168.1.28 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK1337614312
From: <sip:address@hidden>;tag=4172440369
To: <sip:address@hidden>;tag=4172440369
Call-ID: address@hidden
CSeq: 0 REGISTER
Contact: <sip:address@hidden>
max-forwards: 10
expires: 900
user-agent: oSIP/Linphone-0.12.1
Content-Length: 0


| INFO1 | <udp.c: 186> info: Message from 192.168.1.28:1024

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK1337614312
From: <sip:address@hidden>;tag=4172440369
To: <sip:address@hidden>;tag=4172440369
Call-ID: address@hidden
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:address@hidden>
Content-Length: 0



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

| INFO1 | <udp.c: 186> info: Message from 192.168.1.28:1024

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK1337614312
From: <sip:address@hidden>;tag=4172440369
To: <sip:address@hidden>;tag=4172440369
Call-ID: address@hidden
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:address@hidden>
Proxy-Authenticate: Digest realm="asterisk", nonce="0e4b0c84"
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: 408> Outgoing interface to reach 192.168.1.28 is 192.168.20.1.

| INFO1 | <authentication.c: 374> Response in proxy_authorization |39b3153f46b925c6d04b02616298de12|

| INFO1 | <udp.c: 292> Sending message:
(null)



Jason A. Pattie wrote:

Hello,

I have a specific setup where I connect to my work's network using a VPN with a Virtual IP inside the office network's IP range. The firewall is setup to proxy ARP requests for the Virtual IP so traffic originating from the internal network gets routed to the firewall and encrypted and sent to my workstation at home.

The reason I'm writing all this is because it doesn't seem that linphone has the ability to specify the IP address of the 'Contact:' portion of a SIP REGISTER event. I've watched the traffic go from my workstation at home to the SIP endpoint (using Asterisk PBX in the office), but when it attempts to send messages "back", they originate on the Asterisk machine wanting to go to my workstation's local IP address, not my workstation's Virtual IP address.

So, I put an iptables rule on the PBX system to destination NAT packets destined for my workstation's local IP address to my workstation's Virtual IP address. This caused the packets to flow properly, but then linphone crashed, and a GNOME dialog popped up telling me that linphone had crashed and to click to visit the GNOME bug report page, or something like that. Currently, linphone does not crash if I do not direct it to use the Asterisk PBX system as a registrar. But, then I cannot make calls without registering.

I did download the tarball of linphone 0.12 and attempted to find a way to cause linphone to use my Virtual IP instead of the local IP. I found the code that gets the IP address to send in the REGISTER message and replaced the calls with direct assignments to my Virtual IP address. This caused the packets to flow properly (I only compiled the linphonec console app since I couldn't find .deb packages for libgnomeui-2.0, etc.), but linphonec crashed with a Segmentation Fault error.

Thanks for your help.



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