sr-users December 2006

sr-users@lists.kamailio.org

152 participants
257 discussions

[Serusers] Presence support: PUBLISH/SUBSCRIBE
by raj gopal 28 Dec '06

28 Dec '06

Hi, I have installed ser-0.10.99-dev35-pa-4.1 and following up presence handbook. i have configured XCAP using apache. i am getting following errors with ser.cfg of ser package. For PUBLISH request : 0(13411) get_pres_uri(): Error while extracting plain URI 0(13411) handle_publish(): Error while extracting presentity URI For SUBSCRIBE request: 0(13411) ERROR: rls_handler.c:484: XCAP query problems 0(13411) XCAP query failed 0(13411) ERROR: rls_handler.c:484: XCAP query problems 0(13411) XCAP query failed have any one succeed with presence? help me... *-Venkat *

4 4

[Serusers] AVPops and Textops to chang username of request uri
by chungyu 28 Dec '06

28 Dec '06

Hi: I use module AVPops and Textops do chang username of request uri. example: john(a)10.10.16.23 I hope chang => 4762(a)10.10.10.22 my configuration sample: if (uri=~"john@") { avp_write("4762", "s:phone"); subst_user(/(.*)/$avp(phone)/'); rewritehostport("10.10.10.22:5060"); forward(uri:host, uri:port); avp_delete("s:phone"); }; but reply is "Call failed:Device not found" I try this sample is OK if (uri=~"john@") { avp_write("4762", "s:phone"); subst_user(/(.*)/4762/'); rewritehostport("10.10.10.22:5060"); forward(uri:host, uri:port); avp_delete("s:phone"); }; OS: Freebsd 5.4 SIP server: SER-0.9.6 please tell my problem is on? Thank you Chungyu This is my Configuration: # ----------- 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 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" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ 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/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" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avp.so" #loadmodule "/usr/local/lib/ser/modules/uri.so" #loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/avpops.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) # ------------------------- 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 >= 2048 ) { 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 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); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); 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("xx.xx.xx.xx", "subscriber")) { www_challenge("xx.xx.xx.xx", "0"); break; }; save("location"); break; }; if (uri=~"john@") { avp_write("4762","phone"); subst_user('/(.*)/$avp(phone)/'); rewritehostport("10.10.10.22:5060"); forward(uri:host, uri:port); avp_delete("phone"); }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; 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(); }; }

2 1

[Users] MP-104 FXO issues
by Nick De Cristofaro 27 Dec '06

27 Dec '06

Can't make outbound calls, though not related directly to OpenSER.. Maybe someone here knows how to configure these guys. I have inbound calls being automatically dialed into a * extension but If I make any outbound calls I get a syslog output: 16:45:21.609 : 192.168.1.211 : WARNING : ( lgr_call)(458 ) !! [ERROR] Call::GetEndPoint- Can't find endpoint for phone number xxxxxx2018 16:45:21.593 : 192.168.1.211 : WARNING : ( lgr_psbrdif)(457 ) !! [ERROR] AcBoard::GetEndPoint- Can't find EndPoint for Dest:xxxxxxx2018 Source:xxxxxxx11526 SourceIp:c0a80188 16:45:21.593 : 192.168.1.211 : WARNING : ( lgr_TrnkGrp)(456 ) !! [ERROR] #0:TrunkGroup::AllocateEndPoint- Can't find EndPoint for phone number xxxxxx2018 16:45:21.593 : 192.168.1.211 : NOTICE : ( lgr_stk_ses)(455 ) SIPStackSession::HandleStackSetupEV - NEWCALL: SrcPN=0 Any help would be great. Also, caller ID doesn't work - only shows the caller ID of whatever is set at the endpoint phone number settings and not what is actually coming on the line :( Nick

1 0

[Users] TLS problem, not able to start openSER
by Ncheeku Baranov 27 Dec '06

27 Dec '06

Hi, I just compiled openSER with TLS support. I checked that TLS = 1 in the Makefile when I compiled openSER. Now when I try to uncomment the parameters in the openser.cfg to enable the TLS support and restart openSER it does not start (I am using openserctl start command to start openser). It gives an error saying ERROR:PID file /var/run/openser.pid does not exist -- OpenSER start failed. I am using the following parameters in the openser.cfg file for the TLS support: disable_tls = 0 listen = tls:10.30.100.41:5061 tls_verify = 1 tls_require_certificate = 0 tls_method = TLSv1 tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" tls_ca_list = "usr/local/etc/openser/tls/user/user-calist.pem" I have checked that all the paths are correct in defining the tls_certificate, tls_private_key and tls_ca_list. I used the source tarball openser-1.1.0-tls_src.tar.gz<http://www.openser.org/pub/openser/latest/src/openser-1.1.0-tls_src.tar.gz> for installing the openser. Your help is much appreciated. Thanks NCheeku

2 2

[Users] SER vs openSER
by pankaj.munjal＠wipro.com 27 Dec '06

27 Dec '06

Hi all, I am a SER/openSER newbie and was looking at an option of using these SIP stacks. Anyone having any doc/link that compares these two SIP servers. The major criteria that we are looking for are: - Simplicity of using the SIP stack (Most IMP) - Perfomance - Support for Presence Any pointers in this direction will be highly appreciated. Thanks, Pankaj The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

2 1

[Users] subst_body with nated client problem
by Oleg A Chernov 27 Dec '06

27 Dec '06

Hello everyone! My setup is as follow: for local PSTN calls UA (with G.711 and G.729) - my Openser - my Asterisk B2BUA for prepaid - PSTN gateway (G.711 codec only) for international calls UA (with G.711 and G.729) - my Openser - my Asterisk B2BUA for prepaid - ITSP gateway (G.729 codec only) B2BUA on the second leg (B2BUA - gateway) always negotiate the right codec, so no problem, but on the first leg (UA - B2BUA) it always negotiate codec that is more prefered by the client UA side. So we can get situation, when the codecs are different on the both legs. Inspired by the http://lists.iptel.org/pipermail/serusers/2006-August/030129.html I decided to normalize codec preference. Sorry for such a long introduction, at last here goes my problem. I have the following code in my openser.cfg in INVITE section # INVITE section route[3] { ... if ( route to G.711 capable gateway) { subst_body("/^m=audio(.*)RTP\/AVP(.*) 8 (.*)/m=audio\1RTP\/AVP 8 \2 \3/ig"); .... } ... } and it works like a charm for non-nated clients with real IPs. But when we should fire use_media_proxy() for a nated one client we get the problem. SDP body before our substitution (UA-Openser): v=0 o=20000 12893 12893 IN IP4 192.168.2.134 s=ATA186 Call c=IN IP4 192.168.2.134 t=0 0 m=audio 16384 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 After (Openser - B2BUA): v=0 o=20000 12893 12893 IN IP4 192.168.2.134 s=ATA186 Call c=IN IP4 217.77.208.184 t=0 0 m=audio 16384 RTP/AVP 8 0 18 101\01560136\012a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 So our subst_body() rearrange codecs as expected, but use_media_proxy() (it her fault, I think) inserted the mediaproxy fixed port number 60136 between symbols with ASCII codes \015 CR and \012 LF. This behavior was accepted by stable Openser 1.1 and latest cvs 1.2. So the question is : What to do and who is guilty ? :-) -- Best regards, Oleg A. Chernov OAC4-RIPE WildPark ISP, Nikolaev, Ukraine +380512 470555 +380512 500314

2 1

[Serusers] SER server as a client
by jk 26 Dec '06

26 Dec '06

Hello List, I am working on an assignment to develop a load balancing asterisk cluster. I started with google and found out that SER+ASTERISK is the best way to do it. Here is what I am trying to do. I have a sip account / DID (18xx xxx xxxx ) from an abc company. Now the company is gateway for between my asterisk server and PSTN network. My Asterisk server registers with abc company's gateway to receive and make calls. How can I do load balancing in this case. I understand that my SER server should register with the gateway, the gateway should forward call to SER server and then the server should forward call to one of the asterisk servers. Is it possible have the SER server register with another SER or Asterisk server and act like a client to receive and make calls? Can some one please point me to some documentation or case study? Thank you, -JK

2 2

[Users] LCR db_only mode problem
by Jayesh Nambiar 26 Dec '06

26 Dec '06

Hi all, A merry christmas and a great new year to all of you. I am trying to implement LCR in my openser 1.1. My problem is that I have around 40,000 prefix patterns which i need to enter into the LCR table. When I populate my LCR table and try to start my openser it dies. I can see the following messages in my log: convert_row: No memory left convert_rows: Error while converting row #9055 convert_result: Error while converting rows store_result: Error while converting result lcr_reload_gws(): Failed to query lcr data ERROR: lcr:mod_init(): failed to reload gateways and routes init_mod(): Error while initializing module lcr The openser starts properly when I tested with 1000 rows in the LCR table but with 40,000 rows it just dies. Can anyone please help me overcome this issue. Is there any parameter available to make openser properly run with these many rows in the LCR table. I am using the LCR in db_only mode. I have not tried this in caching mode. Will the caching mode work with these many rows?? w/regards, jayesh. Send free SMS to your Friends on Mobile from your Yahoo! Messenger. Download Now! http://messenger.yahoo.com/download.php

2 1

[Serusers] Flex and Bison
by Digital Tuffy 26 Dec '06

26 Dec '06

Could anybody pls brief the use of Flex and Bison with SER. _________________________________________________________________ Christmas is drawing near and the excitement is in the air! Check out this cool photofeature on Christmas celebrations the world over http://www.msnspecials.in/photogallery/photogallery.asp?id=a200611814137906

1 0

[Users] Sending REGISTER adn INVITE--plz help
by imran ahmed 26 Dec '06

26 Dec '06

Hi Users; I am new to use openser so having some problems. I have installed openser-1.0.0.I have the task to do load test of openser through SIPp with AUTHENTICATION. I have added some users in the subscriber table of openser. Now the problem is how to set the openser.cfg so that it can response for INVITE and REGISTER request both like: The flow goes like: 1)SIPp UAC send REGISTER -------> OPENSER 2)407 Proxy Authentication required <------ OPENSER 3)SIPp send REGISTERwith credential------>OPENSER(check from subscrible tables and on the bases of that allow or disallow) hope someone help me; Thanks and Regards; Imran __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

← Newer
1
2
3
4
5
6
7
...
26
Older →

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users December 2006