Well, I uploaded the files to be built, but it takes forever to get results. And the build failed. I see why, because asterisk (by default), during the build grabs the sound files from Digium's servers, and the build machine croaked on that (perhaps they don't allow the build machine to go out to 'net, which makes sense). Hrumph.

i think if you could attach the IP dump file that could help to analyze why the sofia-sip stack back "500" to Asterisk. in my opinion, some incorrect parameter of SDP or other reason in INVITE could cause "500" response.

funpig,

I did a tethereal and fired it off to the Maemo developers. They confirm that there is nothing strange in the SIP communication going on. Asterisk sends a REGISTER with a media description, N900 answers TRYING, RINGING and then issues the Internal Server Error.

I did debug the sofia-sip stack. It complains "nua_set_params(): failed: Error Setting SOA Parameters" before it issues the Internal Server Error. Looking in the code, this seems to be right after it has figured out all the media stuff. I have tried on Asterisk 1.4 as well, but it is the same deal there.

It could very well be that there is one media option in the SIP negotiation that i causing this. I don't know what it could be though. For sure there is never a single RTP package sent

i assumption you had a mistake for REGISTER. normal, the REGISTER didn't with any media description. and TRYING, RINGING will be sent after N900 received the INVITE.

i'm not familiar with sofia-sip, but as you said, "SOA" means SDP Offer/Answer Engine Module. you could check http://sofia-sip.sourceforge.net/ref...dex.html#index . then i agree with you that some parameter is incorrect.

i also hope you could post the detail of INVITE. so, we're could see what's parameters in that INVITE

That would be bug 1860.

funpig:

Session Initiation Protocol
Request-Line: INVITE sip:3450@192.168.2.94:64550 SIP/2.0
Method: INVITE
Request-URI: sip:3450@192.168.2.94:64550
Request-URI User Part: 3450
Request-URI Host Part: 192.168.2.94
Request-URI Host Port: 64550
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 192.168.2.51:5060;branch=z9hG4bK678ab923;rport
Transport: UDP
Sent-by Address: 192.168.2.51
Sent-by port: 5060
Branch: z9hG4bK678ab923
RPort: rport
From: "asterisk" <sip:asterisk@192.168.2.51>;tag=as0396138a
SIP Display info: "asterisk"
SIP from address: sip:asterisk@192.168.2.51
SIP from address User Part: asterisk
SIP from address Host Part: 192.168.2.51
SIP tag: as0396138a
To: <sip:3450@192.168.2.94:64550>
SIP to address: sip:3450@192.168.2.94:64550
SIP to address User Part: 3450
SIP to address Host Part: 192.168.2.94
SIP to address Host Port: 64550
Contact: <sip:asterisk@192.168.2.51>
Contact Binding: <sip:asterisk@192.168.2.51>
URI: <sip:asterisk@192.168.2.51>
SIP contact address: sip:asterisk@192.168.2.51
Call-ID: 67fe5a3f41b5fd6978a96b81452a8ec0@192.168.2.51
CSeq: 102 INVITE
Sequence Number: 102
Method: INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 07 Dec 2009 23:22:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 240
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 18095 18095 IN IP4 192.168.2.51
Owner Username: root
Session ID: 18095
Session Version: 18095
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 192.168.2.51
Session Name (s): session
Connection Information (c): IN IP4 192.168.2.51
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: 192.168.2.51
Time Description, active time (t): 0 0
Session Start Time: 0
Session Stop Time: 0
Media Description, name and address (m): audio 10004 RTP/AVP 0 101
Media Type: audio
Media Port: 10004
Media Protocol: RTP/AVP
Media Format: ITU-T G.711 PCMU
Media Format: DynamicRTP-Type-101
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute Fieldname: rtpmap
Media Format: 0
MIME Type: PCMU
Sample Rate: 8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute Fieldname: rtpmap
Media Format: 101
MIME Type: telephone-event
Sample Rate: 8000
Media Attribute (a): fmtp:101 0-16
Media Attribute Fieldname: fmtp
Media Format: 101 [telephone-event]
Media format specific parameters: 0-16
Media Attribute (a): silenceSuppff - - - -
Media Attribute Fieldname: silenceSupp
Media Attribute Value: off - - - -
Media Attribute (a): ptime:20
Media Attribute Fieldname: ptime
Media Attribute Value: 20
Media Attribute (a): sendrecv

funpig:

http://pastebin.com/m171b9345

Also, another update on this problem. I installed sofia-sip, gstreamer and empathy on a regular desktop today. I successfully connected to my asterisk server and received a phone call. The only version difference between the N900 was a newer gstreamer. The INVITE is identical and sofia-sip responds, like it should, with an OK + Session Description

Just to make this thread complete. I got this working by adding a stun server to my internal network (thanks to the maemo guys). After I had added the stun server I went to the advanced settings (for the SIP configuration of the N900) and entered the stun server's address. Now everything is working great.

Obviously you shouldn't need a stun server on an internal network where there is no NAT. Anyhow, it is a work-around that is acceptable to me.

hi, can you adise about the stun sever name and stun server name?

because i cannot sign in in gtalk on my n900.

thx