Dropping outgoing calls and log errors

I have an Asterisk 13 server running with Ubuntu 18.04 in AWS server in that the client is behind a NAT, we have many customers and endpoints working well with different networks and ISP providers, in an specific network for a customers the endpoints is logging ERROR in my Asterisk server and probably it is dropping the outgoing calls with 1 minute, only outgoings.

This is the error log:

[Jul 16 14:48:08] ERROR[32508]: pjproject:0 <?>:                  sip_parser.c Error parsing '517272': String value was greater than the maximum allowed value.
[Jul 16 14:48:08] ERROR[32508]: pjproject:0 <?>:               sip_transport.c Error processing 705 bytes packet from UDP 201.X.Y.33:2107 : PJSIP invalid value error exception when parsing 'Via' header on line 2 col 39:
REGISTER sip:52.X.Y.22:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 102.X.Y.43:517272;branch=z9hG4bKPjXZjgPYPazqiw96uC02zPOImj2ZQmlmrGMYwl
Max-Forwards: 70
From: <sip:21023002@52.X.Y.22>;tag=L2RiJJDfBrrjBSTAxLQ03u8nROLPNlogmEO5
To: <sip:21023002@52.X.Y.22>
Call-ID: 7oQluUMc2OJtVMseEZLbkzWsI3kUAmXFYx1Q
CSeq: 24348 REGISTER
Contact: <sip:21023002@102.X.Y.43:5172;transport=udp>;+sip.instance="<urn:uuid:44:3b:32:7b:8e:9c>";+sip.model="Telephone_TIP125";+sip.version="2.0"
Expires: 90
User-Agent: Telephone_TIP125
Allow-Events: check-sync
Content-Length:  0

I don’t know what exactly is causing this issue, but the router is from the ISP. I supposed that was the SIP ALG or something like that, but this modem doesn’t has this option to setup in the web interface, the brand of the equipment is DOCSIS 3.0 HW 3.0.

Can anyone help?

Add point that I forget to say is that in this network is not using the default port 5060 to connect with the endpoints as other customers, because it doesn’t register, we are using a different port as default, the 4444 that is a mirrored port of 5060 (double listen).

Then basically we have 2 issues,

1 → 5060 port doesn’t connect;
2 → 4444 port is dropping the calls after 1 minute and logging errors;

You need to be more specific about the setup. Is this a REGISTER request by Asterisk?

The VIA header lists the port 517272, which is not possible. The Contact header reports port 5172, which is probably what was meant. If the impossible port was not added manually, there could be a couple of reasons including a bogus configuration in the first place. Maybe it is possible to fiddle with the reply route in Kamailio, but actually I don’t know how this can be done.

The equipment is sending a REGISTER but in the truth it was already registered…
And this port 517272 is not the correct, is a wrong port that probably the modem/router changed it.

I think the only solution is change the modem/router equipment, because I suppose is a SIP ALG option mode actived but I didn’t find it in the settings to turn off it.

I wouldn’t insist on it being the router. If so, then the device would have to parse the SIP headers with considerable effort. That doesn’t seem obvious to me.

Can you generate PCAP traces before and behind the Router? Phones usually allow that too.

I already change the modem/router, then the issue was solved.

Thanks all.

Could you explain what you changed?

I replaced the equipment (modem/router).

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