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: pjproject:0 <?>: sip_parser.c Error parsing '517272': String value was greater than the maximum allowed value. [Jul 16 14:48:08] ERROR: 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 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, REFER, OPTIONS, INFO, PUBLISH, REGISTER, SUBSCRIBE, NOTIFY, REFER 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?