Jan Fricke
2012-Feb-17 13:04 UTC
[asterisk-users] Presence subscription from other pbx systems
Hi members, I have a question regarding presence in asterisk. I have two PBX systems and would like to connect them. After configuring each other as sip providers calls between users of the pbx systems are possible. Now I'm trying to implement presence between the systems. PBX1 sends dialog-event SUBSCRIBE messages to PBX2. Asterisk just answers 404 not found although user 410 exists. I think this is for security reasons. Is there an option to allow presence subscription from configured providers? Sincerely Jan PS: Here are sample sip messages: <--- SIP read from UDP:10.99.10.2:5060 ---> SUBSCRIBE sip:410 at 10.99.10.14;sipx-noroute=VoiceMail;sipx-userforward=false SIP/2.0 Record-Route: <sip:10.99.10.2:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EWEhLMUd5%21f063cbdfa e9d680ffaa83f6db4234704> From: <sip:sipXrls at 10.99.10.1:51829>;tag=XHK1Gy To: <sip:410 at 10.99.10.14;sipx-noroute=VoiceMail;sipx-userforward=false> Call-Id: eZwhSebwLCc187 Cseq: 2 SUBSCRIBE Contact: <sip:10.99.10.1:51829;transport=udp;x-sipX-nonat> Event: dialog Accept: application/dialog-info+xml Expires: 3153 Date: Mon, 13 Feb 2012 09:45:50 GMT Max-Forwards: 19 User-Agent: sipXecs/4.4.0 sipXecs/rls (Linux) Accept-Language: en Proxy-Authorization: Digest username="~~id~sipXrls", realm="voip.mydomain.local", nonce="3998fbca7da46e21895d383a16356f424f38dbce", uri="sip:410 at 10.99.10.14;sipx-noroute=VoiceMail;sipx-userforward=false", response="53ae73a9ce6a3a6acbe35deda3f731be", cnonce="a42sMg", qop=auth, nc=00000001 Via: SIP/2.0/UDP 10.99.10.2;branch=z9hG4bK-XX-18ddpkccVUQr6IO02D7a9Q5x0A Via: SIP/2.0/UDP 10.99.10.1:51829;branch=z9hG4bK-XX-f75bT2Zlly8RPJBMDcOw5dyOxw Content-Length: 0 <-------------> --- (18 headers 0 lines) --- Creating new subscription Sending to 10.99.10.2:5060 (no NAT) list_route: hop: <sip:10.99.10.2:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EWEhLMUd5%21f063cbdfa e9d680ffaa83f6db4234704> No matching peer for 'sipXrls' from '10.99.10.2:5060' Looking for 410 in public-direct-dial (domain 10.99.10.14) <--- Transmitting (no NAT) to 10.99.10.2:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.99.10.2;branch=z9hG4bK-XX-18ddpkccVUQr6IO02D7a9Q5x0A;received=10.99.10.2 Via: SIP/2.0/UDP 10.99.10.1:51829;branch=z9hG4bK-XX-f75bT2Zlly8RPJBMDcOw5dyOxw From: <sip:sipXrls at 10.99.10.1:51829>;tag=XHK1Gy To: <sip:410 at 10.99.10.14;sipx-noroute=VoiceMail;sipx-userforward=false>;tag=as73 d6e628 Call-ID: eZwhSebwLCc187 CSeq: 2 SUBSCRIBE Server: AskoziaPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120217/afe98ab3/attachment-0001.htm>