Sofia-SIP throws "Internal Server Error"

Phone -> Accounts -> SIP
Username: 3450@192.168.2.92
Password: xxxxxxxxx
Advanced Settings:
User for telephone numbers: Checked
User name: 3450
Transport: Auto
Outbound proxy: 192.168.2.92
Port: 5060
The rest: Default

The phone registers fine with asterisk. However, when trying to make a call from another SIP phone to the N900 I get:

SIP Phone N900
------------------------
Invite with media description =>
100 Trying <=
180 Ringing <=
500 Internal Server Error <=

I have played with the settings like crazy, but nothing seems to help. Any other SIP phone works fine with my Asterisk (Cisco 9640, X-lite ....)

Did anybody else succeed in using the N900 with Asterisk?

Cheers,

// Jonas

Could be related to this:
https://bugs.maemo.org/show_bug.cgi?id=5342
https://bugs.maemo.org/show_bug.cgi?id=4259

https://sourceforge.net/tracker/?fun...roup_id=143636



I have built asterisk-1.6.1.11 .debs and have asterisk *running* on the phone fine.

Good luck,

-Jeff

Thanks Jeff,

I haven't tried to actually run Asterisk on the N900. Seems like a cool thing to do though (I use a SheevaPlug for Asterisk 1.6).

The links you have provided seems to be related to the register phase, which actually works for me. My problem is that the N900 doesn't return an OK with session description like is should after the RINGING. Instead it returns the 500 "Internal Server Error". I am thinking this has to do with some bug in the Sofia-SIP stack. However, if somebody else has successfully connected to Asterisk I guess the problem lies elsewhere.

Anyhow, thanks a lot for trying to help out!

// Jonas

Could the package be made available please?

Im using my N900 with an asterisk box. Can make and receive calls. Only difference in my config is my proxy is blank.

Dumb question but you have the phone setup in extensions.conf right? Im sure you do but had to ask.

bbrindle,

Great news that you can use it with Asterisk. Then there is some hope. To answer your question, yes the extension is well set up in extensions.conf. In fact, I have tested the same extension with a Cisco 7940 and X-Lite and they both work.

My problem is really visible in the SIP communication where you can see that the sip-sofia stack is sending an Internal Server Error. I know that this can be because of a number of reasons though.

However, with your set up working I want to focus on the asterisk side. Do you mind posting your sip.conf for your N900 working extension?

Mine non-working is
[3450]
username=3450
type=friend
secret=xxxxxxxxx
qualify=500
nat=no
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
context=internal
canreinvite=no
callerid=Jonas HP <3450>

Also my general section is
[general]
canreinvite=no
; Include registers
#include "sip_register.conf"
dtmfmode=rfc2833
disallow=all
allow=ulaw
context=trunk ; Default context for incoming calls
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
limitonpeers = yes
externip = xx.xxx.xxx.xxx
localnet=192.168.0.0/255.255.0.0
nat=yes

Thanks,

// Jonas

Here's my config:

Settings on the N900:

User name: 5005@192.168.0.82
Password: [secret]

Enabled is checked

Advanced:
Use for telephone numbers is checked
User Name: Not set
Transport: Auto
Outbound Prody: Not set
Port 5060
Discover public address disabled
Autodected STUN disabled


Asterisk configuration:
SIP.CONF
[5005]
type=friend ; This device takes and makes calls
host=dynamic ; This host is not on the same IP addr every time
username=Nokia ; Not really used
secret=*****
nat=no ; nat=yes if this phone is behind a NAT box or firewall
context=home ; My internal context for phones
insecure=very ;bypass authentication
qualify=30000

EXTENSIONS.CONF:

exten => 7005,1,Dial(SIP/7005,25) ;
exten => 7005,2,Hangup ; and then hangup.

Left off my general. Here it is.

[general]
useragent=SIP
externip=xxx.xxx.xxx.xxx
localnet=192.168.0.0/255.255.255.0
context = ********
bindport=5060
callerid=No CallID
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=iLBC
maxexpirey=180
defaultexpirey=160
tos=reliability
nat=no ; Rely on the fixup protocols to handle the sip xlateion
qualify=yes
dtmfmode=rfc2833
canreinvite=no
realm=asterisk

Okay,

I made sure I had the same parameters in my configuration (commented out the ones that I had extra).

The following is visible in the Asterisk cli when the call from the other SIP phone comes in to the N900 (3450)

-- Called 3450
-- SIP/3450-00462198 is ringing
-- Got SIP response 500 "Internal Server Error" back from 192.168.2.94
-- SIP/3450-00462198 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)


I still get the "Internal Server Error". I can't help to thing that it could be a SIP option in the general section that is messing it up (something that sip-sofia doesn't understand). What does your general section look like?

Also, what version of Asterisk are you running?

Cheers,

// Jonas

Okay, will try. Thanks!