sr-users July 2008

sr-users@lists.kamailio.org

147 participants
207 discussions

[OpenSER-Users] XMPP/SIP-SIMPLE Problem
by Cesar Troncoso 05 Aug '08

05 Aug '08

Hi everyone, I have a problem with xmpp module. Messages going from xmpp to sip-simple work fine. The problem I have is in the opposite direction (from sip-sipmle to xmpp). If I set fork=no or children=1 in my config file then all messages are sent and received in both directions. This is what I see in the logs: xmpp: invalid content-type 0x300ff xmpp: unable to write to command pipe: Bad file descriptor Any Idea how to overcome this situation? Thanks! Cesar

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[Kamailio-Users] Debugging Openser v1.4 scripts
by Jim Burke 02 Aug '08

02 Aug '08

Hi All, We are working to get our script working on Version 1.4 and are having problems debugging the script. Previously if I saw the following error in the logs it would highlight the line and colomb numbers were the error existed. Can someone help out? Jul 29 22:29:18 [19064] CRITICAL:core:yyerror: parse error in config file, line 9459656, column 1-1: syntax error Jul 29 22:29:18 [19064] CRITICAL:core:yyerror: parse error in config file, line 9459656, column 1-1: bad command!) Jul 29 22:29:18 [19064] ERROR:core:main: bad config file (2 errors) Thanks, Jim

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[OpenSER-Users] Need help on Openser + freeradius
by ha do 01 Aug '08

01 Aug '08

Hi i am trying to config Openser to talk to Freeradius i install openser on first PC : 192.168.1.4 i install freeradius on second PC : 192.168.1.5 on the Openser, i change: edit /etc/radiusclient-ng/radiusclients.conf : authserver & accserver 192.168.1.5 edit etc/radiusclient-ng/servers : 192.168.1.5 openser on the FreeRadius, i change: edit clients.conf : listen { 192.168.1.4 ....} but when i make call Openser doesnt talk to FreeRadius does any ideal on it my openser config: modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("registrar", "received_avp", "$avp(i:42)") modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 4) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("auth_db|permissions|uri_db|usrloc","db_url","mysql://openser:openserrw@localhost/openser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("avpops", "avp_url", "mysql://openser:openserrw@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("nathelper","rtpproxy_disable", 1) modparam("nathelper","natping_interval", 0) modparam("nathelper","received_avp", "$avp(i:42)") modparam("mediaproxy","natping_interval",20) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients") modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 2) modparam("acc", "radius_missed_flag", 3) modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$ruri; \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Billing-Party=$avp(s:billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") oute[3] { ## Requests handler if (method=="INVITE") { # Account INVITE packets setflag(2); # Account Missed calls setflag(3); # Radius Extra $avp(s:sip_proxy_ip)="192.168.1.5"; }; Thanks

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[Kamailio-Users] CDRTool: no price showing
by Adrian Georgescu 01 Aug '08

01 Aug '08

Central Time is not s valid timezone Further than this your rates look ok at first glance, if it does not work maybe other can help with their experiences. Adrian

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[Kamailio-Users] Patch for mi_xmlrpc module Makefile
by Sergio Gutierrez 31 Jul '08

31 Jul '08

Hello everybody. A patch has been published on tracker, which fixes a problem related to not define a macro which makes fail the module compilation in Solaris. It has been registered with Request ID: 2034302* * Hope it be useful. Best regards. Sergio Gutiérrez.* *

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[Kamailio-Users] Kamailio Logo
by Darren Sessions 31 Jul '08

31 Jul '08

Off technical topic, I know, but has the Kamailio logo already been designed? - Darren _____________________________ dmsessions(a)gmail.com http://www.darrensessions.com http://www.linkedin.com/in/dsessions _____________________________

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[Kamailio-Users] CDRTool: no price showing
by Adrian Georgescu 31 Jul '08

31 Jul '08

You need to add at list one customer entry, associated profiles <----

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[Kamailio-Users] CDRTool: no price showing
by Adrian Georgescu 31 Jul '08

31 Jul '08

I am afraid that rating all calls the same regardless of destination will not work unless you define some destination to match all your traffic. The rating engine must find a destination prefix from which it derives the price. Read the RATING.txt to understand the logic behind it. Adrian >> I'm attempting to understand how call rating works within CDRTool. I am able to link destinations to the calls, but unable to generate prices. I'm using CDRTool 6.4.1 with FreeRadius/MySQL and a Cisco gateway. What I'm trying to setup seems simple: I have one set rate of $.08/min for any call beginning with a 1 (I'll exclude 800 numbers later). This rate should be applied globally with no differentiation between caller party or destination...

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Re: [Kamailio-Users] CDRTool: no price showing
by Adrian Georgescu 31 Jul '08

31 Jul '08

Yes You need to add at list one customer entry, associated profiles and finally the rates linked with the profile which contain the actual prices for each destination. You have example rating files that you can modify. Adrian On Jul 31, 2008, at 3:28 PM, Brian Del Shasta wrote: > That's not a problem. I am currently able to successfully match > destinations by looking for anything that starts with a '1'. I have > a destination setup called "Billable" which applies to any > destination beginning with 1. I also have destinations setup for > 1800, 1877, etc to filter out the 800 numbers. > > What I cannot figure out is how to link a rate to a destination. Do > I have to setup a profile and customer as well? Or should I be able > to price calls using only destination and rates? > > On Thu, Jul 31, 2008 at 8:42 AM, Adrian Georgescu <ag@ag- > projects.com> wrote: > I am afraid that rating all calls the same regardless of destination > will not work unless you define some destination to match all your > traffic. > The rating engine must find a destination prefix from which it derives > the price. Read the RATING.txt to understand the logic behind it. > Adrian > >> > I'm attempting to understand how call rating works within CDRTool. > I am > able to link destinations to the calls, but unable to generate > prices. I'm > using CDRTool 6.4.1 with FreeRadius/MySQL and a Cisco gateway. What > I'm > trying to setup seems simple: I have one set rate of $.08/min for any > call > beginning with a 1 (I'll exclude 800 numbers later). This rate should > be > applied globally with no differentiation between caller party or > destination... > > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

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Re: [Kamailio-Users] NAT Problems
by Ross Beer 31 Jul '08

31 Jul '08

Hi Neill, The outgoing audio works fine. The audio that is not passed is from PSTN -> Phone. The Snom does have symmetrical RTP set. I did have this working about a week ago then it stopped working! How does asterisk deal with NAT, as if I connect the phone directly to Asterisk everything works perfectly! Both Asterisk & Kamailio are on publically accessible IP addresses. Thank you for your help so far. Ross From: neill.wilkinson(a)btinternet.comTo: users(a)lists.kamailio.orgDate: Thu, 31 Jul 2008 13:32:05 +0100Subject: Re: [Kamailio-Users] NAT Problems Which direction is the one way audio in – from the Phone to the PSTN or Vice-Versa – I assume audio outbound is OK and it’s the audio returning to the phone from the PSTN you’re missing? Do you have symmetrical RTP set in the Line settings (under NAT) on the SNOM? Is Kamailio inside or outside the NAT? Neill...;o) Neill WilkinsonPrincipal Consultant Aeonvista Ltd - opening up new ideas From: users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] On Behalf Of Ross BeerSent: 31 July 2008 12:50To: users(a)lists.kamailio.orgSubject: [Kamailio-Users] NAT Problems I am having problems with a NAT device. I have an asterisk server that is hosted on a public IP and when I connect my Snom phones to it directly audio passes correctly both ways. When I introduce Kamailio 1.3 into the mix I get one way audio. I have fixed the SDP and contacts etc and everything appears to be ok with the sip packet. I would be very grateful for any ideas what the problem could be. Outgoing calls from Kamailio -> asterisk -> PSTN work fine. Thank you for your help in advance, it is much appreciated. =================== config ======================= 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;};# NAT detectionforce_rport();if(nat_uac_test("3")){fix_nated_contact();}if (!method=="REGISTER"){record_route();}if (loose_route()) {append_hf("P-hint: rr-enforced\r\n"); route(1);};if (!uri==myself) {append_hf("P-hint: outbound\r\n"); route(1);};if (uri==myself) {if (method=="REGISTER") {if (!www_authorize("", "subscriber")) {www_challenge("", "0");exit;};if (isflagset(5)) {# set branch flag -- when someone will call this user# the INVITE will have branch flag 6 set after lookup("location")setbflag(6); };fix_nated_contact();fix_nated_register();consume_credentials();save("location");exit;};if (!lookup("location")) {sl_send_reply("404", "Not Found");exit;};append_hf("P-hint: usrloc applied\r\n"); };if(method=="MESSAGE"){if (!lookup("location")){sl_send_reply("404", "User Offline");exit;}route(4);}if (method=="REGISTER"){save("location");} route(1);} route[1] {if (subst_uri('/(sip:.*);nat=yes/\1/')){setbflag(6);};if (isflagset(5)||isbflagset(6)) {route(3);}t_on_reply("1");if (!t_relay()) {sl_reply_error();};exit;} route[3] {if (is_method("BYE")) {unforce_rtp_proxy();} else if (is_method("INVITE")){fix_nated_sdp("3"); };if (isflagset(5)){search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');}t_on_reply("1");}route[4]{if (!t_relay()) {sl_reply_error();};exit;} failure_route[2] {if (isbflagset(6) || isflagset(5)) {unforce_rtp_proxy();}}onreply_route[1] {fix_nated_contact();if (((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") || is_method("INVITE")) {fix_nated_sdp("2");force_rport(); }search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');} ==================== SIP PACKETS ===========================INVITE sip:10001*202@**** IP ****;line=e25l7qyi SIP/2.0Record-Route: <sip:*** IP ***;lr=on;ftag=as0a2e0c2e>Via: SIP/2.0/UDP *** IP ***;branch=z9hG4bKb0a1.cf08f082.0v: SIP/2.0/UDP *** IP ***:5060;branch=z9hG4bK2bc03835;rport=5060f: "Name" <sip:*** DOMAIN ***>;tag=as0a2e0c2et: <sip:10001*202@** DOMAIN **>m: <sip:number@**IP**>i: 74391cce38d7c7f7549c863a651b8f81@*** DOMAIN ****CSeq: 102 INVITEUser-Agent: asteriskMax-Forwards: 69Date: Thu, 31 Jul 2008 11:38:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYk: replacesc: application/sdpl: 422P-hint: usrloc appliedv=0o=root 5390 5390 IN IP4 194.xxx.xxx.xxxs=sessionc=IN IP4 194.xxx.xxx.xxxb=CT:384t=0 0m=audio 12558 RTP/AVP 0 8 3 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecvm=video 17918 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=sendrecv ////////////// PHONE REPLY ////////////////////SIP/2.0 200 OkVia: SIP/2.0/UDP **** IP ****:5060;branch=z9hG4bK19d7ec13;rport=5060Record-Route: <sip:*** IP ***;lr=on;ftag=as486f79a5>From: "*** NUMBER *****" <sip:*** DOMAIN *****>;tag=as486f79a5To: <sip:10001*202@*** DOMAIN *****>;tag=jgdy33ee6nCall-ID: *** ID ***CSeq: 102 INVITEContact: <sip:*** DOMAIN ****:54686;line=e25l7qyi;nat=yes>;reg-id=1User-Agent: snom370/7.3.7Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFOAllow-Events: talk, hold, refer, call-infoSupported: timer, replaces, from-changeContent-Type: application/sdpContent-Length: 508v=0o=root 1569142945 1569142946 IN IP4 192.168.1.20s=callc=IN IP4 ** EXTERNAL IP ***t=0 0m=audio 10014 RTP/AVP 0 8 3 101a=rtpmap:0 pcmu/8000a=rtpmap:8 pcma/8000a=rtpmap:3 gsm/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014a=sendrecvm=video 0 RTP/AVP 34 103 99a=rtpmap:34 H263/90000a=rtpmap:103 h263-1998/90000a=rtpmap:99 H264/90000a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014a=sendrecva=oldmediaip:192.168.1.20 Find out how to make Messenger your very own TV! 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sr-users July 2008