SIP Woes

[DaveHope]

Hello all,

I’ve been trying to play with asterisk (after a two month break) and am
having some problems getting my SIP connection to a third party provider
to work. In the asterisk console I notice:

debian*CLI> set verbose 999
Verbosity was 0 and is now 999
Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:2355 sip_alloc: Allocating new
SIP call for 88e1fcfa-c618-db11-91fa-000fea3f84d4@localhost
Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:5441 check_user_full: Setting
NAT on RTP to 4
Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:840 __sip_ack: Stopping
retransmission on ‘88e1fcfa-c618-db11-91fa-000fea3f84d4@localhost’ of
Response 1: Found
Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:5441 check_user_full: Setting
NAT on RTP to 4
Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:7329 handle_request: Check for
res for 200
Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:1620 update_user_counter: Call
from user ‘200’ is 1 out of 0
Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:840 __sip_ack: Stopping
retransmission on ‘88e1fcfa-c618-db11-91fa-000fea3f84d4@localhost’ of
Response 2: Found

I believe that’s some sort of SIP routing issue related to ReInvite’s ?

• Is there a workaround for this? In the attempt that someone may be
  able to shed some light on the matter, I’ve uploaded my current
  configuration to:

  files.davehope.co.uk/asterisk-problem/

I’ve also uploaded the output of ‘sip debug’. The interesting bit in
that (to me at least) is the message:

Looking for 100000 in Outgoing
Reliably Transmitting (NAT):
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP

Is it so simple that I’ve missed something out in my outgoing bit on my
dialplan ? Anyway, the complete log can be found here:

[files.davehope.co.uk/asterisk-problem/debug.log](http://files.davehope.co.uk/asterisk-problem/debug.log)

Ohh. And:

root@debian:/etc/asterisk# asterisk -V
Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k

If anyone would be so kind as to shed some insight into the matter it’d
be greatly appreciated!,

[zmanea]

There does not appear to be any files at the link you posted, I get a 404. Are you having problems registering, making, or receiving calls? Post your related sip.conf, extensions.conf settings, and information about your network setup (IE: is the asterisk box behind NAT?).