OK. Great Regards
2015-07-17 20:38 GMT+02:00 Ben Fitzgerald <b...@letscorp.us>: > I think I have fixed the authentication issue yet the SIP dialog has > completely changed. Now the dialog involves Asterisk sending SIP NOTIFY to > Kamailio, which is then forwarded to the client. Kamailio.cfg has no routes > to handle NOTIFY and there are no SUBSCRIBE messages preceding the NOTIFY. > Only REGISTER and 200 OK. Is this expected behavior? The sipregs database > is now correctly updated when a peer registers so that's good. > > Benjamin Fitzgerald > LETS Corporation > (925) 235-1154 > b...@letscorp.us > > > > > *******Confidential Notice: > This message is intended only for the use of the individual or entity to > which it is addressed and may contain information that is privileged, > confidential and exempt from disclosure under applicable law. If the reader > of this message is not the intended recipient, you are hereby notified that > any dissemination, distribution or copying of this communication is > strictly prohibited. If you have received this message in error, please > delete this message from all computers and contact Orion Systems/LETS Corp > immediately by return e-mail and/or telephone at (925) 566-5600 > > On Thu, Jul 16, 2015 at 2:59 PM, Ben Fitzgerald <b...@letscorp.us> wrote: > >> Thank you for the qualify solution, that worked. >> >> However, on the KB by asipto, they only create a `sipreg` and `sipusers` >> table and then in extconfig.conf for asterisk, sipusers and sippeers are >> both using the `sipusers` table in MySQL. >> >> I included a sip trace in the original email but I will include a more >> detailed sip debug here. It looks like Asterisk and Kamailio can exchange >> messages but for some reason, the SIP dialog stops after Asterisk sends >> back a SIP 401 Unauthorized to Kamailio. Any ideas? >> >> *1. Kamailio using sipgrep* >> >> T 2015/07/16 14:50:52.393582 UserAgentIP:64521 -> KamailioIP:5060 >> [AP] >> REGISTER sip:opvpnx.ulets.us SIP/2.0. >> Via: SIP/2.0/TCP 192.168.0.179:64521 >> ;alias;branch=z9hG4bK.j~V~btADL;rport. >> From: <sip:1...@opvpnx.ulets.us>;tag=QZ7de-7u5. >> To: sip:1...@opvpnx.ulets.us. >> CSeq: 29 REGISTER. >> Call-ID: puXkrkIICT. >> Max-Forwards: 70. >> Supported: outbound. >> Accept: application/sdp, text/plain, application/vnd.gsma.rcs-ft-http+xml. >> Contact: <sip:102@ >> >> UserAgentIP:64521;transport=tcp>;+sip.instance="<urn:uuid:f8f0aa7c-5b20-4ff2-ac5a-d7b4004afb50>". >> Expires: 3600. >> User-Agent: Alpha TalkIphone/2.2.5-80-g783bf67 (belle-sip/1.4.0). >> Content-Length: 0. >> Authorization: Digest realm="opvpnx.ulets.us", >> nonce="VagoaFWoJzylK0MxoOAIPTRhtZBlmVmr", username="102", uri="sip: >> opvpnx.ulets.us", response="24b8f292fca38e72fbcf36417dcecd24". >> . >> >> >> T 2015/07/16 14:50:52.440789 KamailioIP:5060 -> UserAgentIP:64521 >> [AP] >> SIP/2.0 200 OK. >> Via: SIP/2.0/TCP 192.168.0.179:64521 >> ;alias;branch=z9hG4bK.j~V~btADL;rport=64521;received= UserAgentIP. >> From: <sip:1...@opvpnx.ulets.us>;tag=QZ7de-7u5. >> To: sip:1...@opvpnx.ulets.us;tag=723cfa83f1495d1e63c1f1bb20bde818.a56d. >> CSeq: 29 REGISTER. >> Call-ID: puXkrkIICT. >> Contact: <sip:102@ >> UserAgentIP:64521;transport=tcp>;expires=3600;received="sip: >> UserAgentIP:64521;transport=tcp";+sip.instance="<urn:uuid:f8f0aa7c-5b20-4ff2-ac5a-d7b4004afb50>". >> LETSSBC. >> Content-Length: 0. >> . >> >> *#* >> *# These next two messages when Kamailio forwards REGISTER to Asterisk* >> *#* >> >> T 2015/07/16 14:50:52.466461 KamailioIP:43488 -> AsteriskIP:5060 >> [AP] >> REGISTER sip: AsteriskIP:5060;transport=tcp SIP/2.0. >> Via: >> SIP/2.0/TCP KamailioIP;branch=z9hG4bK328c.29246e24000000000000000000000000.0. >> To: <sip:102@ AsteriskIP >. >> From: <sip:102@ AsteriskIP >;tag=32fda68bf54efeeb04e3edc67b53c63d-3497. >> CSeq: 10 REGISTER. >> Call-ID: 2ee5ec48557bba33-31464@ KamailioIP. >> Max-Forwards: 70. >> Content-Length: 0. >> User-Agent: kamailio (4.3.0 (x86_64/linux)). >> Contact: <sip:102@ KamailioIP:5060>. >> Expires: 3600. >> . >> >> >> T 2015/07/16 14:50:52.494578 AsteriskIP:5060 -> KamailioIP:43488 >> [AP] >> SIP/2.0 401 Unauthorized. >> Via: >> SIP/2.0/TCP >> KamailioIP;branch=z9hG4bK328c.29246e24000000000000000000000000.0;received= >> KamailioIP. >> From: <sip:102@ AsteriskIP >;tag=32fda68bf54efeeb04e3edc67b53c63d-3497. >> To: <sip:102@ AsteriskIP >;tag=as0eb2442e. >> Call-ID: 2ee5ec48557bba33-31464@ KamailioIP. >> CSeq: 10 REGISTER. >> Server: Asterisk PBX 11.6-cert2. >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH. >> Supported: replaces, timer. >> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", >> nonce="5b30f8aa". >> Content-Length: 0. >> >> *2. Asterisk using sip set debug on* >> >> t91*CLI> >> >> <--- SIP read from TCP: KamailioIP:43488 ---> >> REGISTER sip: AsteriskIP:5060;transport=tcp SIP/2.0 >> Via: >> SIP/2.0/TCP KamailioIP;branch=z9hG4bK328c.29246e24000000000000000000000000.0 >> To: <sip:102@ AsteriskIP > >> From: <sip:102@ AsteriskIP >;tag=32fda68bf54efeeb04e3edc67b53c63d-3497 >> CSeq: 10 REGISTER >> Call-ID: 2ee5ec48557bba33-31464@ KamailioIP >> Max-Forwards: 70 >> Content-Length: 0 >> User-Agent: kamailio (4.3.0 (x86_64/linux)) >> Contact: <sip:102@ KamailioIP:5060> >> Expires: 3600 >> >> <-------------> >> --- (11 headers 0 lines) --- >> Sending to KamailioIP:5060 (no NAT) >> Sending to KamailioIP:5060 (no NAT) >> >> <--- Transmitting (no NAT) to KamailioIP:5060 ---> >> SIP/2.0 401 Unauthorized >> Via: >> SIP/2.0/TCP >> KamailioIP;branch=z9hG4bK328c.29246e24000000000000000000000000.0;received= >> KamailioIP >> From: <sip:102@ AsteriskIP >;tag=32fda68bf54efeeb04e3edc67b53c63d-3497 >> To: <sip:102@ AsteriskIP >;tag=as0eb2442e >> Call-ID: 2ee5ec48557bba33-31464@ KamailioIP >> CSeq: 10 REGISTER >> Server: Asterisk PBX 11.6-cert2 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH >> Supported: replaces, timer >> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b30f8aa" >> Content-Length: 0 >> >> >> <------------> >> Scheduling destruction of SIP dialog '2ee5ec48557bba33-31464@ KamailioIP' >> in 32000 ms (Method: REGISTER) >> Scheduling destruction of SIP dialog '2ee5ec48557bba33-31464@ KamailioIP' >> in 32000 ms (Method: REGISTER) >> >> Benjamin Fitzgerald >> LETS Corporation >> (925) 235-1154 >> b...@letscorp.us >> >> >> >> >> *******Confidential Notice: >> This message is intended only for the use of the individual or entity to >> which it is addressed and may contain information that is privileged, >> confidential and exempt from disclosure under applicable law. If the reader >> of this message is not the intended recipient, you are hereby notified that >> any dissemination, distribution or copying of this communication is >> strictly prohibited. If you have received this message in error, please >> delete this message from all computers and contact Orion Systems/LETS Corp >> immediately by return e-mail and/or telephone at (925) 566-5600 >> >> On Thu, Jul 16, 2015 at 11:48 AM, Alberto Sagredo < >> alberto.sagr...@avanzada7.com> wrote: >> >>> Maybe you got to get some traces with sip set debug on on asterisk or >>> ngrep in kamailio to check whereis the problem. >>> >>> I think you are not authenticating correctly >>> >>> Check if you insert on sipusers and sipppers table what is commented on >>> KB by asipto. >>> >>> Maybe your Kamailio is not responding to OPTIONS (qualify=yes) >>> >>> add at the beginning of your kamailio.cfg file >>> request_route { >>> >>> if(is_method("OPTIONS") ) { >>> >>> sl_send_reply("200","Keepalive"); >>> >>> exit; >>> >>> } >>> >>> ..... >>> >>> >>> To solve qualify problem >>> >>> >>> BR >>> >>> 2015-07-16 19:31 GMT+02:00 Ben Fitzgerald <b...@letscorp.us>: >>> >>>> Thanks for your response. >>>> >>>> I did read the section about the secret in the kb url. I followed the >>>> example and inserted the test users on tFe url (101, 102, 103) and they >>>> have secret set to NULL. I have tried both secret=NULL and secret="" and >>>> Asterisk still asks for authentication. Also when I do "sip show peers" I >>>> get: >>>> >>>> Name/username Host Dyn >>>> Forcerport ACL Port Status Description >>>> Realtime >>>> kamailio-inbound kamailioIP a >>>> 5060 Unmonitored >>>> >>>> I added qualify=yes and now: >>>> >>>> Name/username Host Dyn >>>> Forcerport ACL Port Status Description >>>> Realtime >>>> kamailio-inbound kamailioIP a >>>> 5060 UNREACHABLE >>>> >>>> Could this be the issue? I have verified that Kamailio receives the >>>> responses by doing ngrep and I can see the SIP 401 from Asterisk. >>>> >>>> Maybe I am missing something else? I'm not sure I understand how >>>> Asterisk's peer selection affects this. When I received the registration >>>> request from Kamailio, the From: address and domain are the same as the To: >>>> address and domain, which are the values I have set in the sipusers table. >>>> >>>> Another thing, even though the client handset says registered, the >>>> table 'sipregs' is not updated with fullcontact, regseconds, or any data at >>>> all. Yet I can still make a call. So maybe Asterisk is not authenticating >>>> INVITES (whether or not it's registered) and that's why I can call. >>>> >>>> Any further help or things I should try? >>>> >>>> Benjamin Fitzgerald >>>> LETS Corporation >>>> (925) 235-1154 >>>> b...@letscorp.us >>>> >>>> >>>> >>>> >>>> *******Confidential Notice: >>>> This message is intended only for the use of the individual or entity >>>> to which it is addressed and may contain information that is privileged, >>>> confidential and exempt from disclosure under applicable law. If the reader >>>> of this message is not the intended recipient, you are hereby notified that >>>> any dissemination, distribution or copying of this communication is >>>> strictly prohibited. If you have received this message in error, please >>>> delete this message from all computers and contact Orion Systems/LETS Corp >>>> immediately by return e-mail and/or telephone at (925) 566-5600 >>>> >>>> On Thu, Jul 16, 2015 at 3:40 AM, Alberto Sagredo < >>>> alberto.sagr...@avanzada7.com> wrote: >>>> >>>>> You could remove secret= on extensiones to check if its related to >>>>> authentication or not >>>>> >>>>> You must not request authentication to kamailio in order to work >>>>> properly in front of Asterisk >>>>> >>>>> As Daniel mention check if Kamailio peer is created and extensiones >>>>> have no secret.. you would need to add alternate sippasswd table for >>>>> kamailio authentication >>>>> >>>>> BR >>>>> >>>>> 2015-07-16 1:42 GMT+02:00 Ben Fitzgerald <b...@letscorp.us>: >>>>> >>>>>> Hi, I've been following this integration tutorial >>>>>> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb >>>>>> and have a successful registration and I can even make calls through my >>>>>> asterisk box. >>>>>> >>>>>> However what is unusual to me is that every time a phone registers >>>>>> with Kamailio, that is forwarded to Asterisk (as expected), yet Asterisk >>>>>> replies with 401 Unauthorized. Oddly enough the phone registers and can >>>>>> still make calls. What worries me is that as we scale to 100's of cps, >>>>>> this >>>>>> seemingly erroneous message may slow down Asterisk because it's trying to >>>>>> handle authentication for users which have already been authenticated by >>>>>> Kamailio. If this behavior is expected, then that would be good to know >>>>>> as >>>>>> well. >>>>>> >>>>>> This is the sip debug from ASTERISK (I have replaced IP's with the >>>>>> names of the servers): >>>>>> >>>>>> >>>>>> <--- SIP read from TCP:kamailio:41205 ---> >>>>>> REGISTER sip:asteriskIP:5060;transport=tcp SIP/2.0 >>>>>> Via: SIP/2.0/TCP >>>>>> kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0 >>>>>> To: <sip:40081@asteriskIP> >>>>>> From: <sip:40081@asteriskIP >>>>>> >;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 >>>>>> CSeq: 10 REGISTER >>>>>> Call-ID: 0005ce130bcee5c4-26538@kamailio >>>>>> Max-Forwards: 70 >>>>>> Content-Length: 0 >>>>>> User-Agent: kamailio (4.3.0 (x86_64/linux)) >>>>>> Contact: <sip:40081@kamailio:5060> >>>>>> Expires: 3600 >>>>>> >>>>>> <-------------> >>>>>> --- (11 headers 0 lines) --- >>>>>> Sending to kamailio:5060 (no NAT) >>>>>> Sending to kamailio:5060 (no NAT) >>>>>> >>>>>> <--- Transmitting (no NAT) to kamailio:5060 ---> >>>>>> SIP/2.0 401 Unauthorized >>>>>> Via: >>>>>> SIP/2.0/TCP >>>>>> kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0;received= >>>>>> kamailio >>>>>> From: <sip:40081@asteriskIP >>>>>> >;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 >>>>>> To: <sip:40081@asteriskIP>;tag=as404bac9a >>>>>> Call-ID: 0005ce130bcee5c4-26538@ kamailio >>>>>> CSeq: 10 REGISTER >>>>>> Server: Asterisk PBX 11.6-cert2 >>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, >>>>>> INFO, PUBLISH >>>>>> Supported: replaces, timer >>>>>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", >>>>>> nonce="262b338e" >>>>>> Content-Length: 0 >>>>>> >>>>>> >>>>>> <------------> >>>>>> Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' >>>>>> in 32000 ms (Method: REGISTER) >>>>>> Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' >>>>>> in 32000 ms (Method: REGISTER) >>>>>> Really destroying SIP dialog '0005ce130bcee5c1-26536@ kamailio' >>>>>> Method: REGISTER >>>>>> >>>>>> ========================= >>>>>> >>>>>> sip.conf for kamailio trunk: >>>>>> >>>>>> [kamailio-inbound] >>>>>> type=friend >>>>>> dtmfmode=auto >>>>>> host=kamailioIP >>>>>> allow=all >>>>>> context=sipout >>>>>> insecure=port,invite >>>>>> canreinvite=no >>>>>> >>>>>> ======================== >>>>>> >>>>>> Asterisk version: 11.6-cert2 >>>>>> Kamailio version: 4.3 >>>>>> >>>>>> Benjamin Fitzgerald >>>>>> LETS Corporation >>>>>> (925) 235-1154 >>>>>> b...@letscorp.us >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> *******Confidential Notice: >>>>>> This message is intended only for the use of the individual or entity >>>>>> to which it is addressed and may contain information that is privileged, >>>>>> confidential and exempt from disclosure under applicable law. If the >>>>>> reader >>>>>> of this message is not the intended recipient, you are hereby notified >>>>>> that >>>>>> any dissemination, distribution or copying of this communication is >>>>>> strictly prohibited. If you have received this message in error, please >>>>>> delete this message from all computers and contact Orion Systems/LETS >>>>>> Corp >>>>>> immediately by return e-mail and/or telephone at (925) 566-5600 >>>>>> >>>>>> _______________________________________________ >>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>>>>> list >>>>>> sr-users@lists.sip-router.org >>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>>> sr-users@lists.sip-router.org >>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users@lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users@lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >> > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
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