Sip debug, please take a look

General Help
1

Hi,
For some reason I can’t call outside anymore.
Here’s my sip debug, can someone who’s familiar with this take a look?
I always get the same error in the logs:
[2012-04-17 18:00:54] WARNING[2859] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6400ms with no response

Pbx = 192.168.2.4
Router = M0n0wall, 192.168.2.1
Ports forwarded: UDP 5060-5069 --> 192.168.2.4

  • UDP 10000-10049–> 192.168.2.4
    rdp.conf contains ports 10000-10049

Is this a firewall problem or a freepbx problem?
Hope someone can help.

SIP DEBUG:

Retransmitting #2 (NAT) to 192.168.2.1:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 81.82.253.38:5060;branch=z9hG4bK67cf5bb8;rport
Max-Forwards: 70
From: “+323956000” sip:[email protected];tag=as6f012ce4
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FPBX-2.10.0(1.8.10.0)
Date: Tue, 17 Apr 2012 16:00:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: “+323956000” sip:[email protected];party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 380

v=0
o=root 1434855353 1434855353 IN IP4 81.82.253.38
s=Asterisk PBX 1.8.10.0
c=IN IP4 81.82.253.38
b=CT:384
t=0 0
m=audio 10030 RTP/AVP 8 3 9 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 10022 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv

Retransmitting #3 (NAT) to 192.168.2.1:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 81.82.253.38:5060;branch=z9hG4bK67cf5bb8;rport
Max-Forwards: 70
From: “+323956000” sip:[email protected];tag=as6f012ce4
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FPBX-2.10.0(1.8.10.0)
Date: Tue, 17 Apr 2012 16:00:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: “+323956000” sip:[email protected];party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 380

v=0
o=root 1434855353 1434855353 IN IP4 81.82.253.38
s=Asterisk PBX 1.8.10.0
c=IN IP4 81.82.253.38
b=CT:384
t=0 0
m=audio 10030 RTP/AVP 8 3 9 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 10022 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv

2

There is nothing in this trace that indicates the call is not working.

3

Yeah, and the strange thing is, it works again since 10 minutes.
I changed firewall, played with config, etc. At the end I restored config backup and put back our m0n0wall and it all started to work again.
Ghost pbx, that’s what it is :slight_smile:

EDIT: What does this mean;
Retransmitting #3 (NAT) to 192.168.2.1:5060 why does it retransmit?
I’m at a 70MBit down/ 5Mbit up internet connection, with no load for now, so it sure ain’t due low bandwidth or network lag.

Thx for your reply scott.
kind regards

4

Speed really does not have anything to do with reliability. I would rather have a T1 with consistent low latency and jitter than some high bandwidth circuit that is all over the place. The 70/3 is a cable modem connection.

The error you are getting means that Asterisk send a SIP message, with NAT correction to 192.168.2.1 and it did not respond in time. What is 192.168.2.1?

5

192.168.2.1 is the gateway / router ip.
During my struggling I did reboot freepbx eventually, I think this is the only valuable explanation why it works again now…even it sounds strange.

On the other hand, my caller id doesn’t show anymore again and fromuser is set.
(config from last night was restored, and then it worked, strange but through)

I go sleep a night over it, i had enough for today. We’re reachable by phone again, that’s the most important.

6

And today I found a stupid typo in the nr behind fromuser=
CallerID working again :slight_smile:

7