sr-users March 2005

sr-users@lists.kamailio.org

250 participants
627 discussions

[Serusers] Turn 3xx into new Invite
by Martin Koenig 01 Mar '05

01 Mar '05

Hello, is it possible to take 3xx-responses in a reply_route and turn them into a new INVITE on Ser using append_branch or simliar? Regards, Martin

3 3

[Serusers] syn_branch directive
by Java Rockx 01 Mar '05

01 Mar '05

Hi All. Can anyone point me to information on what the syn_branch directive in ser.cfg does? Regards, Paul

1 0

[Serusers] HELP: SER-0.9 Record Route Problem Between SER and Asterisk
by Java Rockx 01 Mar '05

01 Mar '05

Hi All. I'm using ser-0.9 Can anyone take a quick look at this short SIP conversation and tell me if they think the problem is with my ser.cfg or a bug in Asterisk 1.0.2. We use a 3rd party for PSTN gateway functionality. This 3rd party uses a Sonus box behind a SIP proxy. Our SER proxy talks directly to their SIP proxy as needed to complete PSTN calls. The problem is that when a PSTN caller dials a SIP phone and gets sent to voice mail (Asterisk) because of a no answer or busy condition, Asterisk hangs up after the caller leaves a message. When Asterisk hangs up, the BYE from Asterisk is sent to SER, however, SER incorrectly forwards the BYE directly to their Sonus gateway, rather than the their SIP proxy. This causes our PSTN gateway provider to have "open" billing records in their system. If you look at the BYE message from Asterisk to SER you can see that route headers are missing (I think). The final BYE should have been sent to 68.86.100.20, but it was sent to 68.86.100.30 instead. I am record_route()ing all messages except for REGISTER and I have the mhomed=1 parameter set. Can anyone help me put the blame on either my ser.cfg or Asterisk? Regards, Paul IP LEGEND ----------- 68.86.100.30 - 3rd Party Sonus PSTN Gateway 68.86.100.20 - 3rd Party SIP Proxy 24.11.12.24 - Sip Express Router (eth0) 10.255.255.1 - Sip Express Router (eth1) 10.255.255.2 - Asterisk PBX NOTE: I have Asterisk connected to the SER server with a crossover cable. U 2005/02/23 22:24:18.848582 68.86.100.20:5060 -> 24.11.12.24:5060 INVITE sip:4075551212@24.11.12.24:5060 SIP/2.0. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0. To: 4075551212 <sip:4075551212@68.86.100.30:5060>. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 INVITE. Max-Forwards: 4. Contact: sip:3211231234@68.86.100.30:5060. Record-Route: <sip:68.86.100.20:5060;lr>. Content-Type: application/sdp. Content-Length: 312. . v=0. o=NexTone-MSW 1234 187 IN IP4 66.236.245.98. s=sip call. c=IN IP4 66.236.245.98. t=0 0. m=audio 16814 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=fmtp:4 annexa=yes. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. # U 2005/02/23 22:24:18.860022 24.11.12.24:5060 -> 68.86.100.20:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0. To: 4075551212 <sip:4075551212@68.86.100.30:5060>. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 INVITE. Content-Length: 0. . # U 2005/02/23 22:24:18.860259 10.255.255.1:1033 -> 10.255.255.2:5060 INVITE sip:699@10.255.255.2:5060 SIP/2.0. Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>. Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>. Via: SIP/2.0/UDP 10.255.255.1;branch=z9hG4bKb929.21080974.0. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0. To: 4075551212 <sip:4075551212@68.86.100.30:5060>. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 INVITE. Max-Forwards: 3. Contact: sip:3211231234@68.86.100.30:5060. Record-Route: <sip:68.86.100.20:5060;lr>. Content-Type: application/sdp. Content-Length: 312. . v=0. o=NexTone-MSW 1234 187 IN IP4 66.236.245.98. s=sip call. c=IN IP4 24.11.12.24. t=0 0. m=audio 36574 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=fmtp:4 annexa=yes. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. # U 2005/02/23 22:24:18.871131 10.255.255.2:5060 -> 10.255.255.1:1033 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.255.255.1;branch=z9hG4bKb929.21080974.0;received=10.255.255.1;rport=1033. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: <sip:699@10.255.255.2>. Content-Length: 0. . # U 2005/02/23 22:24:18.879160 10.255.255.2:5060 -> 10.255.255.1:1033 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.255.255.1;branch=z9hG4bKb929.21080974.0;received=10.255.255.1;rport=1033. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0. Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>. Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>. Record-Route: <sip:68.86.100.20:5060;lr>. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: <sip:699@10.255.255.2>. Content-Type: application/sdp. Content-Length: 362. . v=0. o=root 550 550 IN IP4 10.255.255.2. s=session. c=IN IP4 10.255.255.2. t=0 0. m=audio 17900 RTP/AVP 97 18 3 4 2 0 8 101. a=rtpmap:97 iLBC/8000. a=rtpmap:18 G729/8000. a=rtpmap:3 GSM/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U 2005/02/23 22:24:18.883882 24.11.12.24:5060 -> 68.86.100.20:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0. Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>. Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>. Record-Route: <sip:68.86.100.20:5060;lr>. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: <sip:699@10.255.255.2>. Content-Type: application/sdp. Content-Length: 363. . v=0. o=root 550 550 IN IP4 10.255.255.2. s=session. c=IN IP4 24.11.12.24. t=0 0. m=audio 36574 RTP/AVP 97 18 3 4 2 0 8 101. a=rtpmap:97 iLBC/8000. a=rtpmap:18 G729/8000. a=rtpmap:3 GSM/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U 2005/02/23 22:24:19.097436 68.86.100.20:5060 -> 24.11.12.24:5060 ACK sip:699@10.255.255.2 SIP/2.0. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=ef5ce1f8a400adab242436d0564cb045. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=ef5ce1f8a400adab242436d0564cb045. To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 ACK. Max-Forwards: 4. Contact: sip:3211231234@68.86.100.30:5060. Record-Route: <sip:68.86.100.20:5060;lr>. Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>. Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>. Content-Length: 0. . # U 2005/02/23 22:24:19.098087 10.255.255.1:1033 -> 10.255.255.2:5060 ACK sip:699@10.255.255.2 SIP/2.0. Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>. Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>. Via: SIP/2.0/UDP 10.255.255.1;branch=0. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=ef5ce1f8a400adab242436d0564cb045. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=ef5ce1f8a400adab242436d0564cb045. To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 ACK. Max-Forwards: 3. Contact: sip:3211231234@68.86.100.30:5060. Record-Route: <sip:68.86.100.20:5060;lr>. Content-Length: 0. . ### U 2005/02/23 22:24:25.104860 10.255.255.2:5060 -> 10.255.255.1:1033 BYE sip:3211231234@68.86.100.30:5060 SIP/2.0. Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport. Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>,<sip:68.86.100.20:5060;lr>,<sip:3211231234@68.86.100.30:5060>. From: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. To: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Contact: <sip:699@10.255.255.2>. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 102 BYE. User-Agent: Asterisk PBX. Content-Length: 0. . # U 2005/02/23 22:24:25.108961 24.11.12.24:5060 -> 68.86.100.30:5060 BYE sip:3211231234@68.86.100.30:5060 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:24.11.12.24;r2=on;ftag=as588114d9;lr=on>. Record-Route: <sip:10.255.255.1;r2=on;ftag=as588114d9;lr=on>. Via: SIP/2.0/UDP 24.11.12.24;branch=z9hG4bK231b.07949ea5.0. Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport=5060. From: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. To: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Contact: <sip:699@10.255.255.2>. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 102 BYE. User-Agent: Asterisk PBX. Content-Length: 0. Route: <sip:3211231234@68.86.100.30:5060>. . # U 2005/02/23 22:24:25.175832 68.86.100.30:5060 -> 24.11.12.24:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 24.11.12.24;branch=z9hG4bK231b.07949ea5.0. Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport=5060. From: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. To: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 102 BYE. Content-Length: 0. . # U 2005/02/23 22:24:25.176182 10.255.255.1:1033 -> 10.255.255.2:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport=5060. From: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. To: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 102 BYE. Content-Length: 0. .

4 6

[Serusers] Which SIP messages types use lookup("location")?
by Java Rockx 01 Mar '05

01 Mar '05

Hi All, Can anyone tell me for which message types I should call lookup("location") before t_relay()? ACK BYE CANCEL INVITE NOTIFY REFER SUBSCRIBE

7 10

[Serusers] sip_msg->buf does not contain modifications
by José Luis Peñalver Paret 01 Mar '05

01 Mar '05

Hello, according to the guide, changes done on a message while server processing are kept in sip_msg->buf. But if I do something like (in ser.cfg): [...] append_hf("Anyheader: hello"); showrequest(); [...] being 'showrequest' a function in a module that just does: static int showrequest_f(struct sip_msg* msg, char *foo, char *foo2) { parse_headers(msg, HDR_EOH, 0); fprintf(stderr, "MESSAGE:\n"); fprintf(stderr, msg->buf); return 1; } then I can't see the new header! What am I doing wrong? Thank you.

1 0

[Serusers] kphone
by johnny Laura 01 Mar '05

01 Mar '05

Hi again, I'm trying to install Kphone in Mandrake 10.0 but I have the next error: checking for Qt... configure: error: Qt (>= Qt 2.2.2) (libraries) not found. Please check your installation! For more details about this problem, look at the end of config.log. But I have instaled the libqt3 and libqt3-devel; and when I compile ; Icompile this way ./configure --with-qt-dir=/usr/lib/qt3 --with-qt-includes=/usr/lib/qt3/include --with-qt-raries=/usr/lib/qt3/lib And even so Ihave tha same error ..Can anyone help me???

2 1

[Serusers] Can ser Presence Agent support windows messenger 5.0 or above?
by Jeffrey Zhou 01 Mar '05

01 Mar '05

Hi, Can ser Presence Agent support windows messenger 5.0 or above? It seems i can not see the on line contact with windows messenger 5.0 with ser

2 1

[Serusers] parse error
by Tina 01 Mar '05

01 Mar '05

Hello, I added custom module (mailbox.so) to SER to add some DB queries (just like in DB example). The module exports the function mbx_lookupmailbox. The function is available for other modules; unfortunately I obtain the following when trying to call it from ser.cfg: >ser -c >0(20596) parse error (137,24-25): unknown command, missing loadmodule? (Though the module is loaded, the function is registered and found by other functions). Have I missed something? Thank you, Tina ============================================================= # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/mailbox.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication #mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/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) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "fr_timer", 10 ) # ------------------------- 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"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # 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("localhost", "subscriber")) { www_challenge("localhost", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; # if we do not get a positive reply, continue at reply_route[1] # forward the request to all destinations in destination set now t_on_failure("1"); t_relay(); }; } failure_route[1]{ log(1, "failure_route[1]\n"); if (t_check_status("408|486|487")){ log(1, "t_check_status(408|486|487)\n"); mbx_lookupmailbox(); #here will be some processing t_relay(); break; } } --------------------------------- Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard.

1 0

[Serusers] fifo db
by AJ Grinnell 01 Mar '05

01 Mar '05

Is the command fifo_db_url meant to enable access to all tables in the ser database or just certaian tables such as grp? Any changes that I make to the grp table take effect immediately, but if I make a change to aliases or domain, SER requires a restart.

2 1

[Serusers] Can I modify a transaction already in progress?
by Paul Belanger 01 Mar '05

01 Mar '05

I need some help understanding transactions, I know how to start transactions, are create a simple stateful proxy. But I'm having some troubles understanding how I can modify a current transaction. Everytime I try I always get the error: 0(10257) ERROR: t_newtran: transaction already in process Which leads me to believe once the transaction is created, I can't modify it. Is this true? ------------------------------- Paul Belanger (mailto:paul.belanger@pronexus.com) Technical Support Specialist Cisco Certified Network Associate Pronexus Inc. - A Powerful Voice in Communication Solutions ------------------------------- Tel: 613.271.8989 ext. 516 Fax: 613.271.8388 http://support.pronexus.com -------------------------------

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sr-users March 2005