[SR-Users] kamailio realtime integration
Hello all,
I want to use realtime architecture kamailio 3.1 + asterisk 1.6.2
described at
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb,
but use it with another softswitch (it also use mysql db for configuration).
If it possible please answer for several questions:
1.What is "#!define WITH_ASTERISK" option doing ?
Does it only use for:
- load uac module
- read login\passw table info
- some routing cases
or there are some else jobs ?
Maybe there is some description for "#!define WITH_ASTERISK" ?
2.What else fields of asterisk realtime table (except
username\sippasswd) kamailio uses for integration ?
3.From integration manual:
"
CREATE TABLE version (
table_name VARCHAR(32) NOT NULL,
table_version INT UNSIGNED DEFAULT 0 NOT NULL
);
INSERT INTO version (table_name, table_version) VALUES
('sipusers','6');
"
For what purposes this table used ? How to change it for using with
another softswitch ?
--
vf
___
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Re: [SR-Users] kamailio realtime integration
On 06/21/2011 12:55 PM, Daniel-Constantin Mierla wrote:
3.From integration manual:
"
CREATE TABLE version (
table_name VARCHAR(32) NOT NULL,
table_version INT UNSIGNED DEFAULT 0 NOT NULL
);
INSERT INTO version (table_name, table_version) VALUES
('sipusers','6');
"
For what purposes this table used ? How to change it for using with
another softswitch ?
This table is required by kamailio to check the version of tables used
internally - default one for this case would be subscriber table, but
since it connects to asterisk database and uses instead the sipusers
table, it has to be able to see that the version for sipusers table
structure is 6.
Cheers,
Daniel
Daniel, thanx for your help and wonderful software!
I want to use different table structure (non asterisk) for integration,
and the questions are:
- what sipusers table fields it uses (only username\sippasswd or some
else)
- where kamailio save registration information (in sipusers or in
another own table)
--
vf
___
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Re: [SR-Users] kamailio realtime integration
Hi all, Still trying to run kamailio 3.1.x + asterisk 1.6.2.X realtime architecture. As i mentioned earlier, instead of asterisk i use another softswitch with db, table "retailroute" with "login" and "password" fields as user auth data. Of course, troubles... user can't register to kamailio :( Logs( without users attempts to register, only kamailio internal logic: Jun 24 11:23:26 sbc kamailio[17768]: INFO:
Re: [SR-Users] kamailio realtime integration
Yes, NOTIFY was generated by ipphone. Thx, again :) On 06/24/2011 12:00 PM, Daniel-Constantin Mierla wrote: Hello, based on the logs, the authentication fails due to wrong password. Check that you have the right password on client side. Kamailio does not generate any NOTIFY unless the presence server modules are loaded. If not, then the notify comes from somewhere else and it is just looped back due to config file routing logic. Cheers, Daniel On 6/24/11 10:01 AM, vaad.f...@gmail.com wrote: Hi all, Still trying to run kamailio 3.1.x + asterisk 1.6.2.X realtime architecture. As i mentioned earlier, instead of asterisk i use another softswitch with db, table "retailroute" with "login" and "password" fields as user auth data. Of course, troubles... user can't register to kamailio :( Logs( without users attempts to register, only kamailio internal logic: Jun 24 11:23:26 sbc kamailio[17768]: INFO:
Re: [SR-Users] kamailio realtime integration
Hi All and Daniel, My interesting "realtime" story still continues :) All configuration seems to be ok. It's default but some DB fields changes as Daniel advised. Terminals registrations to kamailio is ok (but kamailio don't register as a endpoint on remote softswitch, but calls run :) is it ok? ) Situation: one logic + two different terminals = two different results !!! Kamailio logic haven't changes (exclude DB fields) and it's not a softswitch part. Is it something around two different requests with Proxy-Authenticate/realm (one case 193.58.XXX.XXX, second case VoipSwitch) and why kamailio generates two different realms ? First case (ok). Sip terminal - softphone twinkle Dump from kamailio: http://pastebin.com/uddxwf6F Second case (not ok). Sip terminal - cisco iphone Dump from kamailio: terminal->kamailio INVITE sip:9...@193.58.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-1050babb From: "100" ;tag=dfe3d8151250ffabo0 To: Call-ID: 8760c7dd-41ef9283@192.168.0.180 CSeq: 101 INVITE Max-Forwards: 70 Contact: "100" Expires: 240 User-Agent: Linksys/SPA922-5.1.15(a) Content-Length: 259 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp kamailio->terminal SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-1050babb;rport=5060;received=93.116.XXX.XXX From: "100" ;tag=dfe3d8151250ffabo0 To: ;tag=636f77b6676bf8d50aad9baa13cd3627.9e28 Call-ID: 8760c7dd-41ef9283@192.168.0.180 CSeq: 101 INVITE Proxy-Authenticate: Digest realm="193.58.XXX.XXX", nonce="TgR+dE4EfUj/W4ocJ0iFmM93MMo5SgvE" Server: kamailio (3.1.4 (i386/linux)) Content-Length: 0 terminal->kamailio ACK sip:9...@193.58.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-1050babb From: "100" ;tag=dfe3d8151250ffabo0 To: ;tag=636f77b6676bf8d50aad9baa13cd3627.9e28 Call-ID: 8760c7dd-41ef9283@192.168.0.180 CSeq: 101 ACK Max-Forwards: 70 Contact: "100" User-Agent: Linksys/SPA922-5.1.15(a) Content-Length: 0 INVITE sip:9...@193.58.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-beb5344f From: "100" ;tag=dfe3d8151250ffabo0 To: Call-ID: 8760c7dd-41ef9283@192.168.0.180 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="100",realm="193.58.XXX.XXX",nonce="TgR+dE4EfUj/W4ocJ0iFmM93MMo5SgvE",uri="sip:9...@193.58.xxx.xxx",algorithm=MD5,response="69dcc7441ad6f0a70c259e8948d8fcf0" Contact: "100" Expires: 240 User-Agent: Linksys/SPA922-5.1.15(a) Content-Length: 259 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp kamailio->terminal SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-beb5344f;rport=5060;received=93.116.XXX.XXX From: "100" ;tag=dfe3d8151250ffabo0 To: Call-ID: 8760c7dd-41ef9283@192.168.0.180 CSeq: 102 INVITE Server: kamailio (3.1.4 (i386/linux)) Content-Length: 0 SIP/2.0 407 Proxy Authentication Required CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.0.180:5060;rport=5060;received=93.116.XXX.XXX;branch=z9hG4bK-beb5344f From: "100" ;tag=dfe3d8151250ffabo0 Call-ID: 8760c7dd-41ef9283@192.168.0.180 To: ;tag=05050699909937 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="05f481310df9ef29-24050517506050625525" Content-Length: 0 Record-Route: terminal->kamailio ACK sip:9...@193.58.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-beb5344f From: "100" ;tag=dfe3d8151250ffabo0 To: ;tag=05050699909937 Call-ID: 8760c7dd-41ef9283@192.168.0.180 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="100",realm="193.58.XXX.XXX",nonce="TgR+dE4EfUj/W4ocJ0iFmM93MMo5SgvE",uri="sip:9...@193.58.xxx.xxx",algorithm=MD5,response="69dcc7441ad6f0a70c259e8948d8fcf0" Contact: "100" User-Agent: Linksys/SPA922-5.1.15(a) Content-Length: 0 INVITE sip:9...@193.58.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 192.168.0.180:5060;branch=z9hG4bK-303c9613 From: "100" ;tag=dfe3d8151250ffabo0 To: Call-ID: 8760c7dd-41ef9283@192.168.0.180 CSeq: 103 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="100",realm="VoipSwitch",nonce="05f481310df9ef29-24050517506050625525",uri="sip:9...@193.58.xxx.xxx",algorithm=MD5,response="70b633a0ca9ecf0bfebb7792f843d732" Contact: "100" Expires: 240 User-Agent: Linksys/SPA922-5.1.15(a) Content-Length: 259 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp kamailio->terminal SIP/2.0 40
Re: [SR-Users] kamailio realtime integration
Hi All and Daniel, My interesting "realtime" story still continues :) All configuration seems to be ok. It's default but some DB fields changes as Daniel advised. Terminals registrations to kamailio is ok (but kamailio don't register as a endpoint on remote softswitch, but calls run :) is it ok? ) Situation: one logic + two different terminals = two different results !!! Kamailio logic haven't changes (exclude DB fields) and it's not a softswitch part. Is it something around two different requests with Proxy-Authenticate/realm (one case 193.58.XXX.XXX, second case VoipSwitch) and why kamailio generates two different realms ? First case (ok). Sip terminal - softphone twinkle Dump from kamailio: terminal->kamailio INVITE sip:9...@193.58.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 192.168.0.174:5065;rport;branch=z9hG4bKeweuqqbi Max-Forwards: 70 To: From: "kamailio" ;tag=wvgqh Call-ID: lutmonzrdvxboxk@SipuraSPA CSeq: 117 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.4.2 + SDP kamailio->terminal SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.174:5065;rport=5065;branch=z9hG4bKeweuqqbi;received=93.116.XXX.XXX To: ;tag=636f77b6676bf8d50aad9baa13cd3627.b6c5 From: "kamailio" ;tag=wvgqh Call-ID: lutmonzrdvxboxk@SipuraSPA CSeq: 117 INVITE Proxy-Authenticate: Digest realm="193.58.XXX.XXX", nonce="TgR+704EfcMLI76IHmIgL4DU9ED7DulQ" Server: kamailio (3.1.4 (i386/linux)) Content-Length: 0 terminal->kamailio ACK sip:9...@193.58.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 192.168.0.174:5065;rport;branch=z9hG4bKeweuqqbi Max-Forwards: 70 To: ;tag=636f77b6676bf8d50aad9baa13cd3627.b6c5 From: "kamailio" ;tag=wvgqh Call-ID: lutmonzrdvxboxk@SipuraSPA CSeq: 117 ACK User-Agent: Twinkle/1.4.2 Content-Length: 0 INVITE sip:9...@193.58.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 192.168.0.174:5065;rport;branch=z9hG4bKtepuspep Max-Forwards: 70 Proxy-Authorization: Digest username="200",realm="193.58.XXX.XXX",nonce="TgR+704EfcMLI76IHmIgL4DU9ED7DulQ",uri="sip:9...@193.58.xxx.xxx",response="e584c40d869c851f631c74c2a874cc02",algorithm=MD5 To: From: "kamailio" ;tag=wvgqh Call-ID: lutmonzrdvxboxk@SipuraSPA CSeq: 118 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.4.2 Content-Length: 311 +SDP kamailio->terminal SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.0.174:5065;rport=5065;branch=z9hG4bKtepuspep;received=93.116.XXX.XXX To: From: "kamailio" ;tag=wvgqh Call-ID: lutmonzrdvxboxk@SipuraSPA CSeq: 118 INVITE Server: kamailio (3.1.4 (i386/linux)) Content-Length: 0 SIP/2.0 407 Proxy Authentication Required CSeq: 118 INVITE Via: SIP/2.0/UDP 192.168.0.174:5065;received=93.116.XXX.XXX;rport=5065;branch=z9hG4bKtepuspep From: "kamailio" ;tag=wvgqh Call-ID: lutmonzrdvxboxk@SipuraSPA To: ;tag=050709100033375 Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="05f6635f0e211fb3-24050517506070925525" Content-Length: 0 Record-Route: terminal->kamailio ACK sip:9...@193.58.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 192.168.0.174:5065;rport;branch=z9hG4bKtepuspep Max-Forwards: 70 To: ;tag=050709100033375 From: "kamailio" ;tag=wvgqh Call-ID: lutmonzrdvxboxk@SipuraSPA CSeq: 118 ACK User-Agent: Twinkle/1.4.2 Content-Length: 0 INVITE sip:9...@193.58.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 192.168.0.174:5065;rport;branch=z9hG4bKjgqxpmir Max-Forwards: 70 Proxy-Authorization: Digest username="200",realm="193.58.XXX.XXX",nonce="TgR+704EfcMLI76IHmIgL4DU9ED7DulQ",uri="sip:9...@193.58.xxx.xxx",response="e584c40d869c851f631c74c2a874cc02",algorithm=MD5 Proxy-Authorization: Digest username="200",realm="VoipSwitch",nonce="05f6635f0e211fb3-24050517506070925525",uri="sip:9...@193.58.xxx.xxx",response="c28d02e014adc119f75a839a7ebfb171",algorithm=MD5 To: From: "kamailio" ;tag=wvgqh Call-ID: lutmonzrdvxboxk@SipuraSPA CSeq: 119 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.4.2 Content-Length: 311 +SDP kamailio->terminal SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.0.174:5065;rport=5065;branch=z9hG4bKjgqxpmir;received=93.116.XXX.XXX To: From: "kamailio" ;tag=wvgqh Call-ID: lutmonzrdvxboxk@SipuraSPA CSeq: 119 INVITE
[SR-Users] realtime asterisk double-authentication
Hi all, > By default, Asterisk uses the column *secret* for SIP user password, but if that is filled in, > Asterisk will ask for authentication again, resulting in double-authentication which we want to avoid. What does "double-authentication" mean ? Is it standart "INVITE, 407+nonce, ACK, INVITE+nonce+responce" scheme ? -- Vadim F. ___ 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
[SR-Users] dispatcher gw state autochange
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