SIP Trunk 1

[h382]

I have 3 SIP trunks with Broadvox, connected to a Linksys router. I am able to get the SIP trunks to register after a reboot, and when I make a call in, it rings and is picked up by voicemail.

However, after about a minute and 20 seconds, all 3 channels fall out of service. I can't get them back unless I reboot again, and then they fall back out of service. But it works for the first minute or so.

Here is part of the log while the call doesn't come in after a minute.


120565mS Sip: Cannot accept call: reason=503 (Service Unavailable)
121560mS SIP Rx: UDP 209.249.3.59:5060 -> 192.168.1.41:5060
INVITE sip:7042002769@192.168.1.41 SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Supported: timer, 100rel
To: <sip:7042002769@209.249.3.56:5060>
From: <sip:7046545312@209.249.3.59>;tag=3478051898-929947
P-Asserted-Identity:<sip:7046545312@64.158.162.74:5060>
Call-ID: 2558459-3478051898-929941@NXT02.broadvox.net
CSeq: 1 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Via: SIP/2.0/UDP 209.249.3.59:5060;branch=z9hG4bK6cdaeb9795fc22cff6ff514d64a7e479
Contact: <sip:7046545312@209.249.3.59:5060>
Call-Info: <sip:209.249.3.59>;method="NOTIFY;Event=telephone-event;Duration=1000"
Content-Type: application/sdp
Content-Length: 250

v=0
o=NXT02 17234 19987 IN IP4 209.249.3.59
s=sip call
c=IN IP4 209.249.3.60
t=0 0
m=audio 10224 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

 

[IPOfficeIreland]

What version are you on?

 

[jeffthephoneguy]

I just sent u an email. We'll get it working. I'm no expert with SIP, but I can get it to work.

Jeff
"the phone guy"

http://www.bluestartelephone.com

 

[h382]

@TheTaker

6.0.8

 

[IPOfficeIreland]

I had some oddvissues with 508 and sip upgarde to 5018

 

[IPOfficeIreland]

Sorry just noticed rhe 608, Reading on an iPhone! Did you have binding refresh set to start with? Turn it off.

 

[hairlessupportmonkey]

are you certian the ITSP uses 5060 for registered trunks?

our practice is to use 5060 for trusted IP trunks and 5062 for registed trunks. this could round the other way for your ITSP, or something to this effect.

I am also thinking that your NAT keep alives are keeping the ports open and with no traffic closing, therefore possibly, de-resistering your trunk.

Another thought - Cisco TCP timeout options and inpect maps can and will de-register an IP handset from the switch.

Does the Linksys you are using have any SIP / H323 ALG's? if so, turn it off.

 

[h382]

At long last.. I have the trunk working..

It all started with a firmware upgrade on the Linksys router. That didn't fix it, it actually bricked the router. Next morning one of the IT guys goes out and buys what I thought was going to be a POS router, some little 4 port Dynex router. The web interface looks a lot like Netgear though.

Anyway.. So after I install the new router, bam, trunks become idle. Still can't make calls though. I noticed I was getting a 403 Forbidden in my SIP trace, and I realized that my public had changed when I installed the new router, and I wasn't using password authentication with the Provider, I was using registration by IP (mistake).

I ended up getting that changed later that day, and after that I was able to successfully make calls inbound/outbound.

In the end, my breaking of the router ended up leading me to the resolution. Though I will hand a star out to HSM as he mentioned it could be a router issue. I kind of bypassed that thought at first since they were initially coming into service..

Thanks guys for all the input..

 

[hairlessupportmonkey]

YAY - glad I could help(sort of). just seen this too many times now for it not to be a router issue.

Obviously the SIP stuff in v6 is now even more sensitive to any form of packet manipulation by routers / firewall.

there just isnt any real need to inspect h323 traffic.