Setting up SIP trunk in GUI 2.0

When I should set up an trunk I can do so in SIP.CONF with a line looking like…

register => user:secret:authuser@host/extension

…but I can’t figure out a good way to do this in the GUI, the authuser parameter is not available there. Tried define the trunk in the GUI and then edit SIP.CONF and EXTENSIONS.CONF, but I could not really figure out how to make this work.

//Mattias

Has tried a littel bit more with this, and I’m still not sure if this is me not understanding the configuration or a bug in the 1.6.x releases. I’ve tried some diffrent version but is now running on…

OS Version:
Linux server1 2.6.24-19-server #1 SMP Wed Aug 20 23:54:28 UTC 2008 i686 GNU/Linux

Asterisk Build:
Asterisk/1.6.1-rc1
Asterisk GUI-version : SVN-branch-2.0-r4493

After updating to the latest GUI, it has a new option for getting the authuser param into the configfiles, I’ve tried to add it myself earlier, and has ended up with the same problem as now.

Here is a block from USERS.CONF

[Digisip.net]
context = DID_Digisip.net
type=peer
host = proxy.digisip.net
username = MYPHONENR
secret = MYPASSWORD
trunkname = Digisip.net ; GUI metadata
hasiax = no
registeriax = no
hassip = yes
registersip = yes
trunkstyle = voip
hasexten = no
fromdomain = proxy.digisip.net
fromuser = MYPHONENR
authuser = MYUSERNAME
insecure = port,invite
disallow = all
allow = alaw,gsm,ulaw

Enabling SIP DEBUG show the the following register string…

REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 82.209.165.194:5060:
REGISTER sip:proxy.digisip.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK24f188c9;rport
Max-Forwards: 70
From: sip:MYPHONENR@proxy.digisip.net;tag=as709e3f08
To: sip:MYPHONENR@proxy.digisip.net
Call-ID: 6bba555c1acb3acc7c8af2214cf64589@192.168.1.11
CSeq: 102 REGISTER
User-Agent: Asterisk PBX 1.6.1-rc1
Expires: 120
Contact: sip:s@192.168.1.11
Content-Length: 0

However if i remove the trunk from the USERS.CONF and adds the register line to SIP.CONF then

register => MYPHONENR:MYPASSWORD:MYUSERNAME@proxy.digisip.net

…gives…

REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 82.209.165.194:5060:
REGISTER sip:proxy.digisip.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK448a20ec;rport
Max-Forwards: 70
From: sip:MYPHONENR@proxy.digisip.net;tag=as638db923
To: sip:MYPHONENR@proxy.digisip.net
Call-ID: 0532aafa66ffe9b51b9c89ed6b1f50d3@192.168.1.11
CSeq: 103 REGISTER
User-Agent: Asterisk PBX 1.6.1-rc1
Authorization: Digest username=“MYUSERNAME”, realm=“proxy.digisip.net”, algorithm=MD5, uri=“sip:proxy.digisip.net”, nonce=“xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx”, response="xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx"
Expires: 120
Contact: sip:s@192.168.1.11
Content-Length: 0

Anyone that has seen the same problem? As I understand there is maybe not so many sip-providers that require you to send the authuser name when registering?