sr-users

sr-users@lists.kamailio.org

4 participants
37456 discussions

[Users] TLS and asterisk ?
by Bodin Bruno 12 Apr '07

12 Apr '07

Hi, I know asterisk doesn't support tls but that something strange in SIP protocol : TLS UA -> Openser -> no tls UA : that work TLS UA -> Openser -> asterisk : that don't work any one know something about this ? to use asterisk I make rewritehostport. thank for help

3 8

[Users] How to change the expire parameter in a reply message of REGISTER?
by 杨冬 12 Apr '07

12 Apr '07

hi, all: I want to make a client to send the REGISTER update message in a short time, which is used to prevent the NAT device to close the port for the client. So I want to rewrite the reply message for REGISTER and change the expire parameter to a value of my setting. How can I achieve this? Thanks. andy.yang 2007/04/10

4 6

[Users] Redirecting number of ISUP-IAM to Diversion field of SIP-INVITE translation
by Kettmann, Peter 12 Apr '07

12 Apr '07

Hi, we are using Asterisk (1.2.12.1), chan_ss7 library (0.8.4) and a Digium E1 card to perform calls between IP and PSTN/GSM networks. In the next step we need to translate Redirecting number from ISUP-IAM message to Diversion field of SIP-INVITE message. Do you have an idea to make it on Asterisk ? Is it perhaps possible via the OpenSer Diversion module? Thanks for your help, Peter

2 1

[Users] Possible to use subst() to alter first line of INVITE?
by Leif Madsen 12 Apr '07

12 Apr '07

Evening all! I'm trying to use subst() to alter the first line of an INVITE I'm sending to a remote Asterisk box, although I have been unable to get this working. Perhaps I need to use a different function? The INVITE I'm working with looks like (IPs have been changed to protect the innocent): INVITE sip:s@192.168.1.250 SIP/2.0 Record-Route: <sip:192.168.1.190;lr;ftag=as62d28c09> Via: SIP/2.0/UDP 192.168.1.190;branch=z9hG4bKcce2.b907e716.0 Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK5a287301;rport=5060 From: "lmadsen2#lmentinc" <sip:lmadsen2#lmentinc@192.168.1.147>;tag=as62d28c09 To: <4166289921@switch1> Contact: <sip:lmadsen2#lmentinc@192.168.1.147> Call-ID: 791d9cc101c5cbdd2896c0d357cf644c(a)192.168.1.147 CSeq: 102 INVITE User-Agent: Asterisk Max-Forwards: 69 Remote-Party-ID: "lmadsen2#lmentinc" <sip:lmadsen2#lmentinc@192.168.1.147>;privacy=off;screen=no Date: Thu, 12 Apr 2007 05:00:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 240 I have two subst() I've been playing with. This one works which alters the To: line in the INVITE: subst('/^To: (.*)sip:[^@]*+(.*)$/To: \1$hdr(X-DID)\2/g'); This one does not, which I would like to alter the INVITE sip:s@192.168.1.250 SIP/2.0 of the header. This is the very first line in the INVITE. subst('/^INVITE (.*)sip:[^@]*+(.*)$/INVITE sip:$hdr(X-DID)\2/ig'); I tested the regex online as well, and it seems to work. Anything I'm missing? Is there another function I can use to alter the line I want, or am I up poop creek? Thanks! -- Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk

3 3

re:[Users] Sip Tools
by yanlin 12 Apr '07

12 Apr '07

hi, raviprakash sunkara, http://www.voip-info.org/wiki/view/Open+Source+VOIP+Software search here, maybe you can find something useful. i use SIPp to test performance, it can generate calls in certain sequence. regards, yan lin *** Disclaimer: This message may contain privileged and/or confidential information. If you have received this e-mail in error or are not the intended recipient, you may not use, copy, disseminate or distribute it; do not open any attachments, delete it immediately from your system and notify the sender promptly by e-mail that you have done so. Thank you. ***

2 2

[Serusers] re: Malformed CONTACT field in 302 response
by Jignesh Gandhi 11 Apr '07

11 Apr '07

Hello, I am using SER as a redirect server. I recently posted almost the same question. This time the format of the CONTACT is different. It contains call information. Recently I came across an issue where the 302 sent back by SER is not liked by a soft switch. Particularly , the format of the CONTACT field according to the softswitch.. Here is an excerpt of the 302 reply send back ... Session Initiation Protocol Status-Line: SIP/2.0 302 MovedTemporarily Status-Code: 302 Resent Packet: False Message Header Via:SIP/2.0/UDP 70.167.132.2;branch= z9hG4bK-BroadWorks.70.167.132.2-172.20.20.46V5060-0-476027561-38657848-1175625610577 - From:"7000 Arris-EMTA-SIP"<sip:6782347000@70.167.132.2 ;user=phone>;tag=38657848-1175625610577- SIP Display info: "7000 Arris-EMTA-SIP" SIP from address: sip:6782347000@70.167.132.2 SIP tag: 38657848-1175625610577- To:<sip:+16782349999;csel=noind@172.20.20.46:5060;user=phone>;tag= b27e1a1d33761e85846fc98f5f3a7e58.4baf SIP to address: sip:+16782349999 SIP tag: b27e1a1d33761e85846fc98f5f3a7e58.4baf Call-ID:BW144010577030407-1243907316@70.167.132.2 CSeq:476027561 INVITE Contact: sip:+16782349999;csel=noind@172.20.20.37 :5060;user=phone;transport=udp Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 the softswitch is wanting the contact field with <sip:xxxx@xxx.xxx.xxx.xxx:5060> like a FROM or TO URI. It is claiming that since it contains call information within the sip URI, it should be enclosed withing < >. Here is ser.cfg part that does the sl_send_reply(); # do a stateless redirect, if return code is correct if (method=="INVITE") { xlog("L_INFO", "SourceIP <%is> \n"); xlog("L_INFO", "From-uri<%fu>, r-uri <%ru> \n"); if (!gl_redirect()) << custom function that just gets a Server to redirect to { sl_send_reply("480", "TemporarilyUnavailable"); xlog("L_WARN", "Sending a 480 response with r-uri <%ru>\n"); break; } else { sl_send_reply("302", "MovedTemporarily"); xlog("L_DBG", "Sending a 302 response with r-uri <%ru>\n"); break; } }; Any help is appreciated.** thanks, -- Jignesh Gandhi jigpgandhi(a)gmail.com

2 2

[Users] mysql database accounting problems
by Tanguy Bouzéloc 11 Apr '07

11 Apr '07

Hello, I've some issues with my mysql accounting, I can only log successful calls in the acc table and I've no informations about missed calls. In mysql accounting I've only sip 200 ok and no errors. I've not this behavior in my syslog. It's very important for us for evaluate the performance of our providers and we can only do it with a database accouting. For example, If I pass a call and I let it ring til operator timeout, I've no trace of this call in the accounting table. I've also another question, the lcr and gw tables have very small fixed limits (256 for lcr and 32 for gw), since we have much more gateways and sdas I've updated theses limits to a value which fit our needs, do you thinks it can be a problem ? Thanks in advance, Tanguy Bouzéloc Init-Sys # ----------- global configuration parameters ------------------------ #debug=9 fork=yes check_via=no dns=no rev_dns=no port=5060 children=4 sip_warning=0 # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/nathelper.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/lcr.so" loadmodule "/usr/lib/openser/modules/acc.so" loadmodule "/usr/lib/openser/modules/xlog.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/permissions.so" # ----------------- setting module-specific parameters --------------- modparam("lcr", "db_url", "mysql://???") modparam("lcr", "db_mode", 0) modparam("rr","enable_full_lr",1) modparam("acc", "db_url", "mysql://???") modparam("acc", "db_flag", 1) modparam("acc", "db_localtime", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1 ) modparam("permissions", "db_url", "mysql://???") route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); return; }; if (!is_method("REGISTER")) { record_route(); }; if (is_method("BYE") || is_method("CANCEL")) { xlog("L_NOTICE", "End call($ci | $Ri): $fU -> $tU"); if (is_method("BYE")) setflag(1); unforce_rtp_proxy(); }; if (loose_route()) { route(1); return; }; if (!allow_trusted()) { xlog("L_WARN", "Host not trusted ($fu/$si/$rP)"); sl_send_reply("403", "Forbidden"); exit; }; if (!load_gws()) { sl_send_reply("500", "Internal server error, unable to load gateways"); return; } if (!next_gw()) { xlog("L_ERR", "No gw found ($ru/$fu)"); sl_send_reply("503", "Service not available, no gateways found"); return; }; t_on_failure("2"); route(1); } route[1] { setflag(1); if (is_method("INVITE") || is_method("ACK")){ if (is_method("INVITE")) xlog("L_NOTICE", "Incoming call($ci | $Ri): $fU -> $tU"); if (!search("^Content-Length:[ ]*0")) force_rtp_proxy("foc"); # if(src_ip == 217.15.85.178) { # subst('/^From:.*<sip:(.*)@(.*)$/From: "\1" <sip:\1/@\2'); # } t_on_reply("1"); }; if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { setflag(1); if (status=~"[12][0-9][0-9]"){ if (!search("^Content-Length:[ ]*0")) force_rtp_proxy("foc"); }; } failure_route[2] { if (t_check_status("400")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 400 Bad Request"); } else if (t_check_status("401")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 401 Unauthorized"); } else if (t_check_status("403")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 403 Forbidden"); } else if (t_check_status("404")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 404 Not Found"); } else if (t_check_status("405")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 405 Method Not Allowed"); } else if (t_check_status("406")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 406 Not Acceptable"); } else if (t_check_status("407")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 407 Proxy Authentication Required"); } else if (t_check_status("408")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 408 Request Timeout"); } else if (t_check_status("410")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 410 Gone"); } else if (t_check_status("413")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 413 Request Entity Too Large"); } else if (t_check_status("414")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 414 Request-URI Too Long"); } else if (t_check_status("415")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 415 Unsupported Media Type"); } else if (t_check_status("416")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 416 Unsupported URI Scheme"); } else if (t_check_status("420")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 420 Bad Extension"); } else if (t_check_status("421")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 421 Extension Required"); } else if (t_check_status("423")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 423 Interval Too Brief"); } else if (t_check_status("480")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 480 Temporarily Unavailable"); } else if (t_check_status("481")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 481 Call/Transaction Does Not Exist"); } else if (t_check_status("482")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 482 Loop Detected"); } else if (t_check_status("483")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 483 Too Many Hops"); } else if (t_check_status("484")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 484 Address Incomplete"); } else if (t_check_status("485")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 485 Ambiguous"); } else if (t_check_status("486")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 486 Busy Here"); } else if (t_check_status("487")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 487 Request Terminated"); } else if (t_check_status("488")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 488 Not Acceptable Here"); } else if (t_check_status("491")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 491 Request Pending"); } else if (t_check_status("493")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 493 Undecipherable"); } else if (t_check_status("500")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 500 Server Internal Error"); } else if (t_check_status("501")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 501 Not Implemented"); } else if (t_check_status("502")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 502 Bad Gateway"); } else if (t_check_status("503")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 503 Service Unavailable"); } else if (t_check_status("504")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 504 Server Time-out"); } else if (t_check_status("505")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 505 Version Not Supported"); } else if (t_check_status("513")) { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU 513 Message Too Large"); } else { xlog("L_NOTICE", "Failure call($ci | $Ri): $fU -> $tU Unknown code"); } setflag(1); if (is_method("INVITE") && t_check_status("408|500|503")) { if (!next_gw()) { xlog("L_ERR", "No more gw ($ru/$fu)"); t_reply("486", "Busy"); # t_reply("503", "Service not available"); exit; } t_on_failure("2"); force_rtp_proxy("foc"); t_relay(); } }

4 7

Re: [Users] Unable to dial with openser and UAs (Softphone)
by Tsitsi 11 Apr '07

11 Apr '07

Hi Andreas, Thank for your reply. Yes, this is a sipwise config. Openser : I use Debian Sarge apt repository : deb http://www.openser.org/debian sarge main Mysql 4.1 I have set up all tables building with openser_mysql Softphone: Windows XP and X-lite 3.0 build 34025 Regards, Tsitsi --------------------------------- Découvrez une nouvelle façon d'obtenir des réponses à toutes vos questions ! Profitez des connaissances, des opinions et des expériences des internautes sur Yahoo! Questions/Réponses.

2 3

[Users] enabling ipv6
by shruu thii 11 Apr '07

11 Apr '07

hi all, i'm using fedora 4.i've installed openser 1.0.1.how do i enable this for ipv6? i want to use linphone clients.this is for pc-pc testing on lan. --------------------------------- Heres a new way to find what you're looking for - Yahoo! Answers

3 5

[Users] Sip Tools
by raviprakash sunkara 11 Apr '07

11 Apr '07

Hello Urs, Anybody can tell , is there to Test load balance in OpenSER and Asterisk , Not Ethereal and ngrep or sniffers. -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

3 4

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