On 2020-01-16 02:16, Administrator wrote: > > Le 15/01/2020 à 19:50, C.Maj a écrit : >> On 2020-01-15 11:24, Administrator wrote: >> >> 8<'s >> >>> One of the provider took a pcap and told us that expiration was set to 0 >>> that's why they don't accept the registration. We took a pcap on our >>> side when SIP packet goes out of our server and we see that the >>> expiration parameter is setted to 3600 ! >> Howdy, >> >> Maybe the clipping of your SIP packet is occurring on another provider's >> (faulty) node somewhere in between your dualing pcaps at the endpoints ? > > No.tcpdump -nqt -s 0 -i enp0s31f6 -A "dst xxx.yyy.78.36 and dst port > 5060" where xxx.yyy.78.36 is the provider Kamailio IP > > Capture being: > > IP zzz.xyz.174.138.58738 > xxx.yyy.78.36.5060: UDP, length 570 > E..V.T@.?...X....2N$.r...B..REGISTER sip:sip.myprovider.net SIP/2.0 > Via: SIP/2.0/UDP > zzz.xyz.174.138:5060;rport;branch=z9hG4bKPj673a37a2-da52-4f8f-b460-17a93005bc98 > > From: > <sip:[email protected]>;tag=d0be9b76-6363-4ce8-b747-7d75f222eef7 > To: <sip:[email protected]> > Call-ID: e906c156-a23f-4cff-b099-43c61a4447c5 > CSeq: 47982 REGISTER > Contact: <sip:[email protected]:5060> > Expires: 3600 > Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, > CANCEL, UPDATE, PRACK, MESSAGE, REFER > Max-Forwards: 70 > User-Agent: TOOTAiAudio
That "User-Agent" might be getting filtered by the provider as a basic security measure. Can you try the default string for your version of Asterisk ? > Content-Length: 0 > > Please notice the > > E..V.T@.?...X....2N$.r...B.. > > in front of REGISTER, could this create the problem ? Probably not. Those dots are hiding some details about the packet. The output is ASCII due to "-A" flag to tcpdump. Try changing to "-X" for hexadecimal and ASCII. Or, write the packets to a file, and then open the file in Wireshark (there are many helpful SIP analysis tools built in to Wireshark.) >> As for what you can control, first, you might try reducing the >> expiration from 3600 to 999, or maybe something in the 30-60 range is >> better for you. If that works, then raise it from there, but I think an >> hour is more than enough. > We tried with 99, 60, 986, without setting expiration leting Asterisk > using his default value, no changes :( >> Or, change network paths; by adding new outbound SIP connection to the >> provider from alternate port and/or IP on the PBX/firewall, use VPN, etc. > Not a solution, to risky. How about giving it a try from a hosted/cloud virtual machine running somewhere else on the internet ie. not from behind your firewall ? Regards, -- 🤠 C. Maj, Technology Captain @ Penguin PBX Solutions 📞 USA Toll Free 1-833-PNGNPBX (1-833-764-6729) 🤙 International & SMS Texting +1.720.32.42.72.9 🐧 Visit on the World Wide Web at PENGUINPBX.COM -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
