[asterisk-users] SIP with Qualify and NAT

[David Bath]

Hi guys,

 

I'm having a really strange problem, which I'm pretty sure has only
appeared since my last upgrade (1.2.12.1) .

 

It's about NAT and Qualify.  I'm using Asterisk to register with some
external SIP providers.  However, they're always marked as UNREACHABLE,
when they weren't before! 

 

A typical debug looks like this: 

 

hera*CLI> sip reload

 Reloading SIP

  == Parsing '/etc/asterisk/sip.conf': Found

  == Parsing '/etc/asterisk/sip_notify.conf': Found

 

<registration>

 

Reliably Transmitting (no NAT) to 195.189.173.10:5060:

OPTIONS sip:sip.voipfone.co.uk SIP/2.0

Via: SIP/2.0/UDP 87.194.194.249:5060;branch=z9hG4bK07c29ff6;rport

From: "asterisk" <sip:asterisk at 87.194.194.249>;tag=as38a9e906

To: <sip:sip.voipfone.co.uk>

Contact: <sip:asterisk at 87.194.194.249>

Call-ID: 7dd0587b016684785b7bda1e6f1b2478 at 87.194.194.249

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 31 Oct 2006 23:22:19 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0

 

---

hera*CLI>

<-- SIP read from 195.189.173.10:5060:

SIP/2.0 200 OK

Via: SIP/2.0/UDP
87.194.194.249:5060;branch=z9hG4bK07c29ff6;received=10.0.0.8;rport=65509

Record-Route: <sip:195.189.173.10:5060;lr=on>

From: "asterisk" <sip:asterisk at 10.0.0.8>;tag=as38a9e906

To: <sip:sip.voipfone.co.uk>;tag=as7165a192

Call-ID: 7dd0587b016684785b7bda1e6f1b2478 at 10.0.0.8

CSeq: 102 OPTIONS

User-Agent: Voipfone Sip Network

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:195.189.173.16>

Accept: application/sdp

Content-Length: 0

 

<repeats three times...... >

 

--- (12 headers 0 lines)---

Destroying call '7dd0587b016684785b7bda1e6f1b2478 at 10.0.0.8'

Retransmitting #4 (no NAT) to 195.189.173.10:5060:

OPTIONS sip:sip.voipfone.co.uk SIP/2.0

Via: SIP/2.0/UDP 87.194.194.249:5060;branch=z9hG4bK07c29ff6;rport

From: "asterisk" <sip:asterisk at 87.194.194.249>;tag=as38a9e906

To: <sip:sip.voipfone.co.uk>

Contact: <sip:asterisk at 87.194.194.249>

Call-ID: 7dd0587b016684785b7bda1e6f1b2478 at 87.194.194.249

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 31 Oct 2006 23:22:19 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0

 

 

---

Oct 31 23:22:23 NOTICE[30434]: chan_sip.c:11613 sip_poke_noanswer: Peer
'duncVF_proxy-out' is now UNREACHABLE!  Last qualify: 0

Destroying call '7dd0587b016684785b7bda1e6f1b2478 at 87.194.194.249'

hera*CLI>

<-- SIP read from 195.189.173.10:5060:

SIP/2.0 200 OK

Via: SIP/2.0/UDP
87.194.194.249:5060;branch=z9hG4bK07c29ff6;received=10.0.0.8;rport=65509

Record-Route: <sip:195.189.173.10:5060;lr=on>

From: "asterisk" <sip:asterisk at 10.0.0.8>;tag=as38a9e906

To: <sip:sip.voipfone.co.uk>;tag=as300cbe8d

Call-ID: 7dd0587b016684785b7bda1e6f1b2478 at 10.0.0.8

CSeq: 102 OPTIONS

User-Agent: Voipfone Sip Network

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:195.189.173.16>

Accept: application/sdp

Content-Length: 0

 

 

--- (12 headers 0 lines)---

Destroying call '7dd0587b016684785b7bda1e6f1b2478 at 10.0.0.8'

hera*CLI> sip no debug

 

 

As you can see, the 200 OK's appear to be being ignored... and no amount
of fiddling seems to fix it...

 

The SIP config is as follows:

 

type=peer

username=******

fromuser==******

secret==******

fromdomain=sip.voipfone.co.uk

host=sip.voipfone.co.uk

call-limit=5

insecure=very

dtmfmode=rfc2833

nat=yes

qualify=yes

canreinvite=no

context=voipfone-in

disallow=all

allow=g729

allow=ulaw

 

 

Any insight would be very much appreciated.

 

Cheers,

 

Dave

 

 

 

 

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