[Users] Asterisk OpenSER and Exchange 2007 Unified Messaging
Daniel-Constantin Mierla
daniel at voice-system.ro
Mon Feb 5 10:49:50 CET 2007
Hello,
usually the proxy should not change the contact header in any
request/response. Special cases are with natted clients, where the nat
traversal is done by the proxy (for that you have to use
nathelper/rttproxy or mediaproxy/mediaproxy).
What is the error given by asterisk (just run it in sip debug mode and
see printed messages)? Isn't able to deal with maddr?
Cheers,
Daniel
On 02/03/07 21:54, Jon Webster wrote:
> Greetings list,
>
> I'm in the process of configuring Asterisk <-> OpenSER <-> Microsoft
> Exchange 2007 Unified Messaging.
>
> I have a windows based proxy that works great, and want to accomplish
> the same feat using OpenSER. Below are network captures from the working
> proxy and OpenSER. From comparing the working call and the OpenSER call
> it looks like OpenSER needs to modify the CONTACT header in its last
> "OK" response. I'm not sure that's right, but more importantly I'm not
> sure how to do this in the config. Any help or direction is greatly
> appreciated.
>
> Thanks,
> -jon
>
> p.s. Please cc me on any replies, I'm not currently subscribed to the
> list.
>
> openSER unsuccessfull call negotiation
> ===========
> #
> U +9.215930 asterisk:5060 -> openser:5060
> INVITE sip:8885555 at openser SIP/2.0.
> Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK629bdb92;rport.
> From: "Jon Webster" <sip:3149 at asterisk>;tag=as43593b57.
> To: <sip:8885555 at openser>.
> Contact: <sip:3149 at asterisk>.
> Call-ID: 46109877480231241e3b8e9d57036e69 at asterisk.
> CSeq: 102 INVITE.
> User-Agent: Asterisk PBX.
> Max-Forwards: 70.
> Date: Sat, 03 Feb 2007 18:01:48 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Content-Type: application/sdp.
> Content-Length: 216.
> .
> v=0.
> o=root 381 381 IN IP4 asterisk.
> s=session.
> c=IN IP4 asterisk.
> t=0 0.
> m=audio 13614 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
>
> #
> U +0.000305 openser:5060 -> asterisk:5060
> SIP/2.0 100 trying -- your call is important to us.
> Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK629bdb92;rport=5060.
> From: "Jon Webster" <sip:3149 at asterisk>;tag=as43593b57.
> To: <sip:8885555 at openser>.
> Call-ID: 46109877480231241e3b8e9d57036e69 at asterisk.
> CSeq: 102 INVITE.
> Server: OpenSer (1.1.1-tls (i386/linux)).
> Content-Length: 0.
> Warning: 392 openser:5060 "Noisy feedback tells: pid=6890
> req_src_ip=asterisk req_src_port=5060 in_uri=sip:8885555 at openser
> out_uri=sip:5555 at openser via_cnt==1".
> .
>
> #
> U +0.025313 openser:5060 -> asterisk:5060
> SIP/2.0 180 Ringing.
> FROM: "Jon Webster"<sip:3149 at asterisk>;tag=as43593b57.
> TO: <sip:8885555 at openser>;epid=BD-70-82-06-F9;tag=e38b6cdb9b.
> CSEQ: 102 INVITE.
> CALL-ID: 46109877480231241e3b8e9d57036e69 at asterisk.
> MAX-FORWARDS: 70.
> VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK629bdb92;rport=5060.
> CONTENT-LENGTH: 0.
> SERVER: RTCC/2.0.6017.0.
> .
>
> #
> U +0.016314 openser:5060 -> asterisk:5060
> SIP/2.0 200 OK.
> FROM: "Jon Webster"<sip:3149 at asterisk>;tag=as43593b57.
> TO: <sip:8885555 at openser>;epid=BD-70-82-06-F9;tag=e38b6cdb9b.
> CSEQ: 102 INVITE.
> CALL-ID: 46109877480231241e3b8e9d57036e69 at asterisk.
> MAX-FORWARDS: 70.
> VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK629bdb92;rport=5060.
> CONTACT: <sip:exchangeUM:5065;transport=Tcp;maddr=exchangeUM>.
> CONTENT-LENGTH: 197.
> CONTENT-TYPE: application/sdp.
> SERVER: RTCC/2.0.6017.0.
> .
> v=0.
> o=- 0 0 IN IP4 exchangeUM.
> s=Microsoft Exchange Speech Engine.
> c=IN IP4 exchangeUM.
> t=0 0.
> m=audio 59544 RTP/AVP 0 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
>
>
>
> working proxy
> ===================
> #
> U +8.036801 asterisk:5060 -> goodproxy:5060
> INVITE sip:5555 at goodproxy SIP/2.0.
> Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK42383fff;rport.
> From: "Jon Webster" <sip:3149 at asterisk>;tag=as07e96b4e.
> To: <sip:5555 at goodproxy>.
> Contact: <sip:3149 at asterisk>.
> Call-ID: 7d0450601533753042ce2d835a3f98d2 at asterisk.
> CSeq: 102 INVITE.
> User-Agent: Asterisk PBX.
> Max-Forwards: 70.
> Date: Sat, 03 Feb 2007 18:02:40 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Content-Type: application/sdp.
> Content-Length: 216.
> .
> v=0.
> o=root 381 381 IN IP4 asterisk.
> s=session.
> c=IN IP4 asterisk.
> t=0 0.
> m=audio 17160 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
>
> #
> U +0.003872 goodproxy:5060 -> asterisk:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP
> asterisk:5060;branch=z9hG4bK42383fff;rport;received=goodproxy.
> From: "Jon Webster" <sip:3149 at asterisk>;tag=as07e96b4e.
> To: <sip:5555 at goodproxy>.
> Call-ID: 7d0450601533753042ce2d835a3f98d2 at asterisk.
> CSeq: 102 INVITE.
> User-Agent: M-Networks USR/1.0.
> Allow: INVITE, INFO, ACK, CANCEL, BYE, NOTIFY, BENOTIFY, SUBSCRIBE.
> Content-Length: 0.
> .
>
> #
> U +0.139238 goodproxy:5060 -> asterisk:5060
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP
> asterisk:5060;received=goodproxy;branch=z9hG4bK42383fff;rport.
> FROM: "Jon Webster"<sip:3149 at asterisk>;tag=as07e96b4e.
> TO: <sip:5555 at exchangeUM>;epid=BD-70-82-06-F9;tag=eb4f3cf30.
> CSEQ: 102 INVITE.
> CALL-ID: 7d0450601533753042ce2d835a3f98d2 at asterisk.
> MAX-FORWARDS: 70.
> CONTENT-LENGTH: 0.
> SERVER: RTCC/2.0.6017.0.
> .
>
> #
> U +0.010224 goodproxy:5060 -> asterisk:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> asterisk:5060;received=goodproxy;branch=z9hG4bK42383fff;rport.
> FROM: "Jon Webster"<sip:3149 at asterisk>;tag=as07e96b4e.
> TO: <sip:5555 at exchangeUM>;epid=BD-70-82-06-F9;tag=eb4f3cf30.
> CSEQ: 102 INVITE.
> CALL-ID: 7d0450601533753042ce2d835a3f98d2 at asterisk.
> MAX-FORWARDS: 70.
> CONTACT: <sip:goodproxy:5060>.
> CONTENT-LENGTH: 197.
> CONTENT-TYPE: application/sdp.
> SERVER: RTCC/2.0.6017.0.
> .
> v=0.
> o=- 0 0 IN IP4 exchangeUM.
> s=Microsoft Exchange Speech Engine.
> c=IN IP4 exchangeUM.
> t=0 0.
> m=audio 32458 RTP/AVP 0 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
> .
>
> #
> U +0.000894 asterisk:5060 -> goodproxy:5060
> ACK sip:goodproxy:5060 SIP/2.0.
> Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK440651a7;rport.
> From: "Jon Webster" <sip:3149 at asterisk>;tag=as07e96b4e.
> To: <sip:5555 at goodproxy>;tag=eb4f3cf30.
> Contact: <sip:3149 at asterisk>.
> Call-ID: 7d0450601533753042ce2d835a3f98d2 at asterisk.
> CSeq: 102 ACK.
> User-Agent: Asterisk PBX.
> Max-Forwards: 70.
> Content-Length: 0.
> .
>
> #
> U +5.475898 asterisk:5060 -> goodproxy:5060
> BYE sip:goodproxy:5060 SIP/2.0.
> Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK5fabb75e;rport.
> From: "Jon Webster" <sip:3149 at asterisk>;tag=as07e96b4e.
> To: <sip:5555 at goodproxy>;tag=eb4f3cf30.
> Call-ID: 7d0450601533753042ce2d835a3f98d2 at asterisk.
> CSeq: 103 BYE.
> User-Agent: Asterisk PBX.
> Max-Forwards: 70.
> Content-Length: 0.
> .
>
> #
> U +0.028752 goodproxy:5060 -> asterisk:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> asterisk:5060;received=goodproxy;branch=z9hG4bK5fabb75e;rport.
> FROM: "Jon Webster"<sip:3149 at asterisk>;tag=as07e96b4e.
> TO: <sip:5555 at exchangeUM>;tag=eb4f3cf30;epid=BD-70-82-06-F9.
> CSEQ: 103 BYE.
> CALL-ID: 7d0450601533753042ce2d835a3f98d2 at asterisk.
> MAX-FORWARDS: 70.
> CONTENT-LENGTH: 0.
> SERVER: RTCC/2.0.6017.0.
> .
>
>
>
>
>
> #
> # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $
> #
> # simple quick-start config script
> #
>
> # ----------- global configuration parameters ------------------------
>
> debug=4 # debug level (cmd line: -dddddddddd)
> fork=yes
> log_stderror=yes # (cmd line: -E)
>
> /* Uncomment these lines to enter debugging mode
> #fork=no
> #log_stderror=yes
> */
>
> check_via=no # (cmd. line: -v)
> dns=no # (cmd. line: -r)
> rev_dns=no # (cmd. line: -R)
> listen=openser
> port=5060
> children=4
> fifo="/tmp/openser_fifo"
>
> #
> # uncomment the following lines for TLS support
> #disable_tls = 0
> #listen = tls:your_IP:5061
> #tls_verify = 1
> #tls_require_certificate = 0
> #tls_method = TLSv1
> #tls_certificate =
> "/home/darilion/software/openser-1.0.1-cvs/sip-server/debian/openser/etc
> /openser/tls/user/user-cert.pem"
> #tls_private_key =
> "/home/darilion/software/openser-1.0.1-cvs/sip-server/debian/openser/etc
> /openser/tls/user/user-privkey.pem"
> #tls_ca_list =
> "/home/darilion/software/openser-1.0.1-cvs/sip-server/debian/openser/etc
> /openser/tls/user/user-calist.pem"
>
> # ------------------ module loading ----------------------------------
>
> # Uncomment this if you want to use SQL database
> #loadmodule "/usr/lib/openser/modules/mysql.so"
>
> loadmodule "/usr/lib/openser/modules/sl.so"
> loadmodule "/usr/lib/openser/modules/tm.so"
> loadmodule "/usr/lib/openser/modules/rr.so"
> loadmodule "/usr/lib/openser/modules/maxfwd.so"
> loadmodule "/usr/lib/openser/modules/usrloc.so"
> loadmodule "/usr/lib/openser/modules/registrar.so"
> loadmodule "/usr/lib/openser/modules/textops.so"
>
> loadmodule "/usr/lib/openser/modules/uri.so"
> loadmodule "/usr/lib/openser/modules/nathelper.so"
>
> # Uncomment this if you want digest authentication
> # mysql.so must be loaded !
> #loadmodule "/usr/lib/openser/modules/auth.so"
> #loadmodule "/usr/lib/openser/modules/auth_db.so"
>
> # ----------------- setting module-specific parameters ---------------
>
> # -- usrloc params --
>
> modparam("usrloc", "db_mode", 0)
>
> # Uncomment this if you want to use SQL database
> # for persistent storage and comment the previous line
> #modparam("usrloc", "db_mode", 2)
>
> # -- auth params --
> # Uncomment if you are using auth module
> #
> #modparam("auth_db", "calculate_ha1", yes)
> #
> # If you set "calculate_ha1" parameter to yes (which true in this
> config),
> # uncomment also the following parameter)
> #
> #modparam("auth_db", "password_column", "password")
>
> # -- rr params --
> # add value to ;lr param to make some broken UAs happy
> modparam("rr", "enable_full_lr", 1)
>
> # ------------------------- request routing logic -------------------
>
> # main routing logic
>
> route{
>
> # initial sanity checks -- messages with
> # max_forwards==0, or excessively long requests
> if (!mf_process_maxfwd_header("10")) {
> sl_send_reply("483","Too Many Hops");
> exit;
> };
>
> if (msg:len >= 2048 ) {
> sl_send_reply("513", "Message too big");
> exit;
> };
>
> if (method=="INVITE" && uri=~"^sip:888[0-9]*@")
> {
> log("\n\nSIP invite received
> from exchange12\n\n");
> strip(3);
> t_relay("tcp:exchangeUM:5065");
> exit;
> };
>
> # we record-route all messages -- to make sure that
> # subsequent messages will go through our proxy; that's
> # particularly good if upstream and downstream entities
> # use different transport protocol
> if (!method=="REGISTER")
> record_route();
>
> # subsequent messages withing a dialog should take the
> # path determined by record-routing
> if (loose_route()) {
> # mark routing logic in request
> append_hf("P-hint: rr-enforced\r\n");
> route(1);
> };
>
> if (uri=="sip:openser:5060;transport=TCP") {
> sl_send_reply("200", "OK");
> exit;
> }
>
> if (!uri==myself) {
> # mark routing logic in request
> append_hf("P-hint: outbound\r\n");
> # if you have some interdomain connections via TLS
> #if(uri=~"@tls_domain1.net") {
> # t_relay_to_tls("IP_domain1","port_domain1");
> # exit;
> #} else if(uri=~"@tls_domain2.net") {
> # t_relay_to_tls("IP_domain2","port_domain2");
> # exit;
> #}
> route(1);
> };
>
> # if the request is for other domain use UsrLoc
> # (in case, it does not work, use the following command
> # with proper names and addresses in it)
> if (uri==myself) {
>
> if (method=="REGISTER") {
>
> # Uncomment this if you want to use digest
> authentication
> #if (!www_authorize("openser.org",
> "subscriber")) {
> # www_challenge("openser.org", "0");
> # exit;
> #};
>
> save("location");
> exit;
> };
>
> lookup("aliases");
> if (!uri==myself) {
> append_hf("P-hint: outbound alias\r\n");
> route(1);
> };
>
> # native SIP destinations are handled using our USRLOC
> DB
> if (!lookup("location")) {
>
> sl_send_reply("404", "Not Found");
> exit;
> };
> append_hf("P-hint: usrloc applied\r\n");
> };
>
> route(1);
> }
>
>
> route[1] {
> # send it out now; use stateful forwarding as it works reliably
> # even for UDP2TCP
> if (!t_relay()) {
> sl_reply_error();
> };
> exit;
> }
>
> route[2] {
> t_on_reply("1");
> }
>
>
>
> ------------------------------------------------------------------------
>
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