- sr-users - kamailio.org

[Users] Reg:NAT Routing
by venkid 08 Feb '07

08 Feb '07

hai, can any body help me to test audio call between two different private IPs. am using X-lite as client .am not able to get voice between two private ips and able to get voice with public IP .i have run media proxy saparatly. please find attached my openser.cfg file too. Thanks and Regards venkatesh.d debug=9 fork=yes log_stderror=yes listen=192.168.0.25 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://openser:openser@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/presence.so" modparam("auth_db|domain|uri_db|usrloc","db_url","mysql://openser:openser@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("presence", "db_url","mysql://openser:openser@localhost/openser") modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "to_tag_pref", 'a') modparam("presence", "totag_avpid", 987) modparam("presence", "lock_set_size", 8) modparam("presence", "expires_offset", 10) modparam("presence", "force_active", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE log("setting nat as yes *****************"); record_route_preset("192.168.0.25:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); exit; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); if (client_nat_test("3") || search("^Route:.*;nat=yes")) { log("setting falg 6 here **********"); setflag(6); use_media_proxy(); }; }; route(1); exit; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); exit; }; if (method=="ACK") { route(1); exit; } else if (method=="CANCEL") { route(1); exit; } else if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { # if (method="INVITE" || method=="ACK") { # end_media_session(); # }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(4); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; }

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Re: [Serusers] Problemswith compiling ser 0.10.99
by Dmitry Isakbayev 07 Feb '07

07 Feb '07

The OSP toolkit project has been moved from sipfoundry to sourceforge - http://sourceforge.net/projects/osp-toolkit Regards, Dmitry > Date: Mon, 08 Jan 2007 10:10:47 -0500 > From: Dmitry Isakbayev <dmitry(a)transnexus.com> > Subject: Re: [Serusers] Problemswith compiling ser 0.10.99 > To: "tzieleniewski" <tzieleniewski(a)o2.pl> > Cc: "'Support@Transnexus. Com'" <support(a)transnexus.com>, > serusers(a)lists.iptel.org > Message-ID: <45A25EF7.7070502(a)transnexus.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi, > > The OSP toolkit project is being moved from sipfoundry to sourceforge. > It should be available shortly. Meanwhile, the latest snapshot of the > toolkit can be downloaded from > > http://www.transnexus.com/OSP%20Toolkit/OSP%20Toolkit.htm > > Regards, > Dmitry > > Date: Mon, 08 Jan 2007 11:25:24 +0100 > > From: "tzieleniewski" <tzieleniewski(a)o2.pl> > > Subject: [Serusers] Problemswith compiling ser 0.10.99 > > To: serusers(a)lists.iptel.org > > Message-ID: <54c54dce.20e2e287.45a21c14.f0624(a)o2.pl> > > Content-Type: text/plain; charset="utf-8" > > > > Hi All!! > > I am trying to compile the lattest cvs version of ser but I get errors > > which I cann't handle myself so i kindly ask for your help:) > > What I want to achive basically is the ser instance with presence > > support. > > I am compiling ser in the following way: > > make group_include="standard standard-dep stable" all > > I read that osp requires osptoolkit i tried to get it from sipfoundry > > but i couldn't find any source of that. > > I also cann't fight problems with tls i didn't includethe experimental > > modules(tls) but still I get some errors.

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[Serusers] SIP Proxy - CAC
by Michel Bensoussan 07 Feb '07

07 Feb '07

Hello, I need to write a CAC (Call Admission Control) module for an 802.11 AP (Access Point). The idea is to use a SIP Proxy to monitoring bandwith utilization, for example according to codec, and allow or disallow new sessions, depending on resources. I'm thinking using SER (or maybe OpenSER) for this purpose. On a first stage I want to install a SIP Proxy on the AP (embedded with OS Linux) and on a second stage adding the CAC feature. There is many modules in SER and I'm not sure which one I should use for a SIP Proxy. I'm learning about SIP but my knowledge is limited (till last week I never heard about it). Thank you for advising. Regards, Michel.

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[Serusers] Newbie ser user, moving from asterisk
by Antonio José dos Santos Brandão 07 Feb '07

07 Feb '07

Hi, I'm starting with ser and would like to know if it's possible no make the following: Today i use asterisk with sip users registered to it and a DeadAgi (a2billing) to manage calls and prepaid/postpaid accounts. Works fine. Since i have too many features and customizations i can't abone it. The problem is that many users are behind nat and have to canreinvite=no, forcing the rtp media to passthru asterisk. I would like to implement media_proxy in this scenario to balance the rtp's bandwith with another hosts. If there were some kind of media_proxy for asterisk would be great, but there isn't. So, i wonder: can I put ser in this, after or before asterisk, making possible to user media_proxy without change so much in asterisk setup? tks, -- Antonio J. S. Brandão

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Re: [Serusers] Cisco AS 5300 can't send BYE for SER... It's bug?
by Sahria Hao 07 Feb '07

07 Feb '07

Hi Edson. I want to know that why my Cisco AS 5300 didn't send BYE for SER...? Maybe... I doubt that maybe my 5300 have only dial-peer 6000 voice "POTS" configure for outgoing PSTN call. In case of PSTN incoming call have no problem about sending BYE for SER, Because it is apply dial-peer voice 5000 "VOIP" confiure as follows: [SER] <- [5300 (VOIP dial-peer)] <- [PSTN] So I'll try to re-configure my 5300 dial-peer, or please give me a hint If anyone have some way to solve this problem. Thanks, Sahria 2007/2/6, Edson <4lists(a)gmail.com>: > > I have this same behaviour, but never give it great importance, since we > didn't bill incomming calls… > > But it would be great to know if it's because of a misconfiguration or a > bug… but we notice that many ports become unavaliable (blocked) over time. > To release we programmed a reboot every day on 3AM…J > > Even with 'ngrep' the BYE, when PSTN side disconnects, didn't show up… > > Edson. > > ------------------------------ > > *From:* serusers-bounces(a)lists.iptel.org [mailto: > serusers-bounces(a)lists.iptel.org] *On Behalf Of *Sahria Hao > *Sent:* segunda-feira, 5 de fevereiro de 2007 08:36 > *To:* serusers(a)lists.iptel.org > *Subject:* Re: [Serusers] Cisco AS 5300 can't send BYE for SER... It's > bug? > > Hi Greger, > > And I'm very sorry for my poor exposition. > > >>Do you get an error on the 5300? > > No, my 5300 works well and there's no error. > > >> Is it sent, but never reaches SER? > > No, when I finished call by PSTN side, 5300 didn't send BYE for SER. > >>Does SER receive, but does not recognize it? > > SER didn't receive a message from 5300 entirely. > > I think that when I finished this call, 5300 must send a BYE message for > SER... but didn't send it. > 2007/2/5, Greger V. Teigre <greger(a)teigre.com>: > > 09. [Cisco] can't send BYE for SER *****why??***** > > What does that mean?! Do you get an error on the 5300? Is it sent, but > never reaches SER? > Does SER receive, but does not recognize it? > g-) > > Sho Aihara wrote: > > Hi all. > > I have a problem for the following scenario. > When I make a call for PSTN and on hook by PSTN side, > Cisco As can't send BYE for SER. > > 01. [UA via Asterisk] dialing "08022223333" -> [SER] > 02. [SER] prefix("0333") and rewritehostport("my.cisco.ip.address:5060") > -> [Cisco] > 03. [Cisco] dial-peer voice 6000 pots, translate-outgoing called from > "033308022223333" to "008022223333" > 04. [Cisco] process an outbound call to "008022223333" -> [e.g. Mobile] > 05. [e.g. Mobile] Catch call > 06. [SER] log CDR start > 07. [Cisco] talking > 08. [e.g. Mobile] On hook and call disconnect > 09. [Cisco] can't send BYE for SER *****why??***** > 10. [UA via Asterisk] On hook > 11. [UA via Asterisk] Send BYE for SER > 12. [SER] log CDR End [Cisco] Call finished > > But another scenario, if make a call from PSTN to Asterisk and > on hook by PSTN side, Cisco As send BYE to SER. > > 01. [e.g. Mobile] dialing "0377771111(Asterisk user number)" > 02. [Cisco] receive "77771111" call number > 03. [Cisco] dial-peer voice 5000 voip, session target ipv4: > my.ser.ip.address -> [SER] > 04. [SER] process an incoming call to "0377771111" -> [UA via Asterisk] > 05. [UA via Asterisk] Catch call > 06. [SER] log CDR start > 07. [UA via Asterisk] talking > 08. [e.g. Mobile] On hook and call disconnect > 09. [Cisco] Send BYE to SER > 10. [SER] log CDR End [Cisco] Call finished > 11. [UA via Asterisk] receive BYE from SER > > And sorry for my diffucult example. > > Why Cisco AS 5300 can't send BYE to SER > When PSTN call is disconnected by PSTN side? > > My ser.cfg as follows: > > # > -------------------------------------------------------------------------- > # global configuration parameters > # > -------------------------------------------------------------------------- > fork=no > log_stderror=yes > check_via=no > dns=no > rev_dns=no > listen=my.ser.ip.address > port=5060 > fifo="/tmp/ser_fifo" > fifo_db_url="mysql://ser:heslo@localhost/ser" > > # > -------------------------------------------------------------------------- > # module loading > # > -------------------------------------------------------------------------- > 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/auth.so" > loadmodule "/usr/local/lib/ser/modules/auth_db.so" > loadmodule "/usr/local/lib/ser/modules/avpops.so" > loadmodule "/usr/local/lib/ser/modules/permissions.so" > loadmodule "/usr/local/lib/ser/modules/acc.so" > loadmodule "/usr/local/lib/ser/modules/exec.so" > > # > -------------------------------------------------------------------------- > # setting module-specific parameters > # > -------------------------------------------------------------------------- > modparam("usrloc", "db_mode", 2) > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > modparam("rr", "enable_full_lr", 1) > modparam("usrloc", "db_url", " mysql://ser:heslo@localhost/ser") > modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") > modparam("permissions", "db_url", "mysql://ser:heslo@localhost /ser") > modparam("tm", "fr_inv_timer", 27) > modparam("tm", "fr_inv_timer_avp", "inv_timeout") > modparam("permissions", "db_mode", 1) > modparam("permissions", "trusted_table", "trusted") > modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") > modparam("acc", "db_flag", 2) > modparam("acc", "db_missed_flag", 3) > > # > -------------------------------------------------------------------------- > # route pattern > # > -------------------------------------------------------------------------- > route { > > 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; > }; > > record_route(); > > if (loose_route()) { > if (method=="ACK") { > acc_db_request("01:CallStart\n", "acc"); > }; > if (method=="BYE" || method=="CANCEL") { > acc_db_request("02:CallEnd\n", "acc"); > }; > t_relay(); > break; > }; > > if (uri==myself) { > if (method=="REGISTER") { > if (!www_authorize("", "subscriber")) { > www_challenge("", "0"); > break; > }; > save("location"); > break; > }; > > if (search("^(f|From): .*(a)(my\.cisco\.ip\.address<.*(a)%28my%5C.cisco%5C.ip%5C.address>)")) > { > #PSTN Incoming call from Cisco AS 5300 e.g. 0377771111 > rewritehost("my.asterisk.ip.address "); > }; > > lookup("aliases"); > > if (!lookup("location")) { > if (method=="INVITE" && !search("^(f|From): > .*(a)(my\.cisco\.ip\.address <.*(a)%28my%5C.cisco%5C.ip%5C.address>)")) { > if (!proxy_authorize("", "subscriber")) { > proxy_challenge("", "0"); > break; > }; > if (uri=~"^sip:0[0-9]{10}@") { > # PSTN Outgoing call to Cisco AS 5300 e.g. 08022223333 > prefix("0333"); > rewritehostport("my.cisco.ip.address:5060"); > avp_write("i:45", "inv_timeout"); > } else { > sl_send_reply("404", "Not Found"); > break; > }; > consume_credentials(); > }; > }; > > }; > > if (!t_relay()) { > sl_reply_error(); > }; > > } > > And my Cisco AS 5300 config as follows: > > voice call send-alert > voice rtp send-recv > > voice service pots > fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco > > voice service voip > fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco > sip > min-se 60 > > translation-rule 50 > Rule 0 0333 0 > Rule 1 ^7777 037777 > > voice class codec 2 > codec preference 1 g711ulaw > codec preference 2 g711alaw > > dial-peer voice 5000 voip > tone ringback alert-no-PI > description ser-asterisk-cisco-test > huntstop > destination-pattern 77771111$ > translate-outgoing called 50 > voice-class codec 2 > session protocol sipv2 > session target ipv4:my.ser.ip.address > dtmf-relay rtp-nte > max-conn 1 > > dial-peer voice 6000 pots > application session > max-conn 2 > destination-pattern 0333T > progress_ind alert enable 8 > translate-outgoing called 50 > port 0:D > > Thanks, > Sahria > > > ------------------------------ > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > -- > ---------- > shosuke > msn : anseie(a)hotmail.co.jp > email : sahria.hao(a)gmail.com >

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[Users] distinguish between fr_timer and fr_inv_timer timeout
by Klaus Darilion 07 Feb '07

07 Feb '07

Hi! If a failure route is executed - how can I distinguish between fr_timer timeout (a broken gateway) and fr_inv_timer timeout (nobdy took up the phone)? thanks klaus

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[Users] 180 Ringing / 183 Session Progress
by Sam Lee 07 Feb '07

07 Feb '07

Hey Guys, Is it mandatory for SIP servers to generate to UA 180 Ringing to inform them that the called party is now ringing ? The reason i'm asking is because I am seeing strange results using different UAs. Using SIP clients to call out , instead of seeing 180 Ringing , i see 183 session progress even before call was picked up. On some sip client , ring back can be heard. On some others , ring back cannot be heard. Is there any reason to suspect that these UAs needs to receive 180 Ringing before generating any ringback ? Regards, Sam

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[Users] OpenSER v1.0.1 to v1.1.1
by Max Gregorian 06 Feb '07

06 Feb '07

Hi, with regards to the release of v.1.1.1 I have a few questions. Firstly, what is to become of v1.0.1? Are you recommending all users with 1.0.1 also upgrade directly to v.1.1.1? Secondly, does 1.0.1 suffer from most (if not all) the same bugs as 1.1.0? I am particularly interested in any documented issues with handling response ACKs for 4xx class errors as I am currently experiencing a few issues with these, and I noticed in the changelog for 1.1.1 that a bug with ACK matching for non-200 class responses has recently been fixed. I ask this I noticed you are gradually phasing out all documentation to do with 1.0.1 on the openser website. As such, is 1.0.1 to be deprecated?

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[Serusers] Has anyone got Jabber module to work w/SER
by SER LIST 06 Feb '07

06 Feb '07

Subject says it all. I am trying to make this work. I have installed JABBERD2 with a PyMSNT gateway. I can use a jabber cliient (exodus) and connect to Jabberd server and communicate w/ other Jabber users and MSN users. I now want to forward IM messages from Eyebeam UA to Jabber users and MSN users. I followed all instructions. When the message gets forwarded to SER and it executes the "xjab" function, I get an error saying "Invalid Content-Type". Any pointers will be highly appreciated. _________________________________________________________________ Buy what you want when you want it on Sympatico / MSN Shopping http://shopping.sympatico.msn.ca/content/shp/?ctId=2,ptnrid=176,ptnrdata=08…

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[Users] avp_db_load question
by Klaus Darilion 06 Feb '07

06 Feb '07

Hi! What are the return values of avp_db_load? E.g. is it FALSE on problems accessing the DB or FALSE if it did not found any AVP to load? I couldn't find this documented somewhere. regards klaus -- Klaus Darilion nic.at

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[Serusers] Anyone using CDRTool ?
by Lokesh Kumar 06 Feb '07

06 Feb '07

Hello All, Is there anyone using CDRTool in conjugation with ser, asterisk, radius setup. Can anyone give me a hand to setup it properly? Lokesh -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.411 / Virus Database: 268.17.27/671 - Release Date: 2/5/2007

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[Serusers] Request for Daniel-Constantin Mierla
by Giovanni Chirico 06 Feb '07

06 Feb '07

Hi Daniel, Im trying to use the dispatcher module. Im using the call forwarding ser.cfg file of the SER Getting Started <http://www.iptel.org/ser/doc/gettingstarted> pack. I have inserted the function ds_select_dst("1", "0"); In the main route() and the function forward(uri:host, uri:port); in the PSTN Handler, route(5). How I can made the failure route, so if a gateway is down another gateway is load? In attachment there are my ser.cfg and dispatcher.list files. Please help me :-( Bye, Giò

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[Serusers] automated response
by Myron Groening 06 Feb '07

06 Feb '07

Vacation Alert !! I will be out of the office, and will not be checking my message till FEB. 26 2007 Thanks Myron Groening

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[Serusers] new version of radiusclient-ng
by Jan Janak 06 Feb '07

06 Feb '07

Hello, I just released a new version of radiusclient-ng library. The new version, 0.5.4, fixes problems on 64-bit architectures. You can download the new release from http://ftp.iptel.org/pub/radiusclient-ng/radiusclient-ng-0.5.4.tar.gz Credits go to Peter Nixon. Jan.

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[Users] ERROR: module version mismatch for,/usr/local/lib/openser/modules/mysql.so; core: openser 1.1.1-tls,(i386/freebsd); module: openser 1.1.1-notls (i386/freebsd)
by Hansi Lao 06 Feb '07

06 Feb '07

Hello All, I have installed openser-1.1.1 with tls on my box. However, when I try to run openser. It returns me this error: ERROR: module version mismatch for /usr/local/lib/openser/modules/mysql.so; core: openser 1.1.1-tls (i386/freebsd); module: openser 1.1.1-notls (i386/freebsd). Would someone be kind enough to help me how to compile a mysql.so which supports tls? Thank you.

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[Users] migrating openser v1.1.x to v1.2.x
by Daniel-Constantin Mierla 05 Feb '07

05 Feb '07

Hello, since we got in testing phase for 1.2.0 release, I think is better to start early a document with migration guidelines. Issues encountered during upgrade should be gathered together in the next page, to share and help each others: http://openser.org/dokuwiki/doku.php/install:1.1.x-to-1.2.x It is just the skeleton and few stuff there. Hopefully the content will grow quickly. Everybody can contribute. Cheers, Daniel

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[Users] How can i make my OpenSER server with presence module work properly with the db_text database?
by Kc Ayuba 05 Feb '07

05 Feb '07

Hi, I'm trying to use db_text database with the presence module of OpenSER(1.2.x ). User agent is X-Lite(3.0). When I try to send a REGISTER message to OpenSER, OpenSER still works. But if i sent a SUBSCRIBE or a PUBLISH message, then openser crashes. If i use the MySQL database, then everything works just fine. But i prefer to use the db_text database. Here is the core file that was generated when OpenSER crashed. Core generated: Program terminated with signal 11, Segmentation fault. #0 0xa7ced659 in msg_active_watchers_clean (ticks=100, param=0x0) at subscribe.c:610 610 subs.event_id.len = strlen(row_vals[event_id_col].val.str_val.s); (gdb) bt #0 0xa7ced659 in msg_active_watchers_clean (ticks=100, param=0x0) at subscribe.c:610 #1 0x080a3f6c in timer_ticker () at timer.c:141 #2 0x0806b51f in main_loop () at main.c:724 #3 0x0806d43b in main (argc=3, argv=0xaff0a2e4) at main.c:1438 (gdb) frame 0 #0 0xa7ced659 in msg_active_watchers_clean (ticks=100, param=0x0) at subscribe.c:610 610 subs.event_id.len = strlen(row_vals[event_id_col].val.str_val.s); (gdb) list 605 606 subs.from_domain.s = row_vals[from_domain_col].val.str_val.s; 607 subs.from_domain.len = strlen(row_vals[from_domain_col].val.str_val.s); 608 609 subs.event_id.s = row_vals[event_id_col].val.str_val.s; 610 subs.event_id.len = strlen(row_vals[event_id_col].val.str_val.s); 611 612 subs.to_tag.s = row_vals[to_tag_col].val.str_val.s; 613 subs.to_tag.len = strlen(row_vals[to_tag_col].val.str_val.s); 614 (gdb) 615 subs.from_tag.s = row_vals[from_tag_col].val.str_val.s; 616 subs.from_tag.len = strlen(row_vals[from_tag_col].val.str_val.s); 617 618 subs.callid.s = row_vals[callid_col].val.str_val.s; 619 subs.callid.len = strlen(row_vals[callid_col].val.str_val.s); 620 621 subs.contact.s = row_vals[contact_col].val.str_val.s; 622 subs.contact.len = strlen(row_vals[contact_col].val.str_val.s); 623 624 subs.cseq = row_vals[cseq_col].val.int_val; I really hope that someone can help me with this problem. Thanks in advance. Kind regards, Kc

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RE: [Users] Asterisk OpenSER and Exchange 2007 Unified Messaging
by Jon Webster 05 Feb '07

05 Feb '07

The significant difference of debug from a good call and openser is next. Below that are the full debug logs of both the openser and goodproxy call with an ngrep of openser's call in between. significant differences ======================== Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:30660 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- SIP/openser-082966e8 answered SIP/jonlaptop-0828e888 -- Attempting native bridge of SIP/jonlaptop-0828e888 and SIP/openser-082966e8 == Spawn extension (local, 8886000, 2) exited non-zero on 'SIP/jonlaptop-0828e888' Feb 5 09:38:58 WARNING[28172]: chan_sip.c:1227 retrans_pkt: Maximum retries exceeded on transmission 59f0033f3761cbf949fd42714b5d2b8f@asterisk for seqno 103 (Non-critical Request) pbx*CLI> VERSES --- (11 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:47688 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: <sip:goodproxy:5060> set_destination: Parsing <sip:goodproxy:5060> for address/port to send to set_destination: set destination to goodproxy, port 5060 Transmitting (no NAT) to goodproxy:5060: asterisk debug of openser call =========== --- (9 headers 0 lines) --- pbx*CLI> <-- SIP read from openser:5060: SIP/2.0 180 Ringing FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078 TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f CSEQ: 102 INVITE CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk MAX-FORWARDS: 70 VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060 CONTENT-LENGTH: 0 SERVER: RTCC/2.0.6017.0 --- (9 headers 0 lines) --- -- Called openser/8886000 -- SIP/openser-082966e8 is ringing pbx*CLI> <-- SIP read from openser:5060: SIP/2.0 200 OK FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078 TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f CSEQ: 102 INVITE CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk MAX-FORWARDS: 70 VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060 CONTACT: <sip:swordfish: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 30660 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --- (11 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:30660 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- SIP/openser-082966e8 answered SIP/jonlaptop-0828e888 -- Attempting native bridge of SIP/jonlaptop-0828e888 and SIP/openser-082966e8 == Spawn extension (local, 8886000, 2) exited non-zero on 'SIP/jonlaptop-0828e888' Feb 5 09:38:58 WARNING[28172]: chan_sip.c:1227 retrans_pkt: Maximum retries exceeded on transmission 59f0033f3761cbf949fd42714b5d2b8f@asterisk for seqno 103 (Non-critical Request) pbx*CLI> <-- SIP read from openser:5060: BYE sip:3149@asterisk SIP/2.0 Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f> Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f> FROM: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f TO: <sip:3149@asterisk>;tag=as4aa53078 CSEQ: 1 BYE CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk MAX-FORWARDS: 69 Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825 VIA: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65 CONTENT-LENGTH: 0 USER-AGENT: RTCC/2.0.6017.0 P-hint: outbound --- (13 headers 0 lines) --- Sending to openser : 5060 (non-NAT) Transmitting (no NAT) to openser:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825;received=openser Via: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65 Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f> Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f> From: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f To: <sip:3149@asterisk>;tag=as4aa53078 Call-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:3149@asterisk> Content-Length: 0 ngrep of above call ====================== # U +0.000616 openser:5060 -> asterisk:5060 SIP/2.0 180 Ringing. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078. TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. CSEQ: 102 INVITE. CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. MAX-FORWARDS: 70. VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060. CONTENT-LENGTH: 0. SERVER: RTCC/2.0.6017.0. . # U +0.067855 openser:5060 -> asterisk:5060 SIP/2.0 200 OK. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as4aa53078. TO: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. CSEQ: 102 INVITE. CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. MAX-FORWARDS: 70. VIA: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK088c0f40;rport=5060. CONTACT: <sip:swordfish: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 30660 RTP/AVP 0 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. # U +32.353338 openser:5060 -> asterisk:5060 BYE sip:3149@asterisk SIP/2.0. Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f>. Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f>. FROM: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. TO: <sip:3149@asterisk>;tag=as4aa53078. CSEQ: 1 BYE. CALL-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. MAX-FORWARDS: 69. Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825. VIA: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65. CONTENT-LENGTH: 0. USER-AGENT: RTCC/2.0.6017.0. P-hint: outbound. . # U +0.000734 asterisk:5060 -> openser:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP openser;branch=z9hG4bK81fa.47210ed5.0;i=825;received=openser. Via: SIP/2.0/TCP exchangeUM:5065;branch=z9hG4bKa55fac65. Record-Route: <sip:openser;r2=on;lr=on;ftag=c5bbe5d85f>. Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=c5bbe5d85f>. From: <sip:8886000@openser>;epid=BD-70-82-06-F9;tag=c5bbe5d85f. To: <sip:3149@asterisk>;tag=as4aa53078. Call-ID: 59f0033f3761cbf949fd42714b5d2b8f@asterisk. CSeq: 1 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Contact: <sip:3149@asterisk>. Content-Length: 0. . good call from working proxy ======== We're at asterisk port 17508 Video is at asterisk port 11130 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to goodproxy:5060: INVITE sip:6000@goodproxy SIP/2.0 Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK4af21ea7;rport From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy> Contact: <sip:3149@asterisk> Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 05 Feb 2007 14:55:02 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 17508 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called exchange12/6000 pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK4af21ea7;rport;received=goodproxy From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy> Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 102 INVITE User-Agent: M-Networks USR/1.0 Allow: INVITE, INFO, ACK, CANCEL, BYE, NOTIFY, BENOTIFY, SUBSCRIBE Content-Length: 0 --- (9 headers 0 lines) --- pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK4af21ea7;rport FROM: "Jon Webster"<sip:3149@asterisk>;tag=as78ded60a TO: <sip:6000@exchangeUM>;epid=BD-70-82-06-F9;tag=f51839ab98 CSEQ: 102 INVITE CALL-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk MAX-FORWARDS: 70 CONTENT-LENGTH: 0 SERVER: RTCC/2.0.6017.0 --- (9 headers 0 lines) --- -- SIP/exchange12-082b8e18 is ringing pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK4af21ea7;rport FROM: "Jon Webster"<sip:3149@asterisk>;tag=as78ded60a TO: <sip:6000@exchangeUM>;epid=BD-70-82-06-F9;tag=f51839ab98 CSEQ: 102 INVITE CALL-ID: 109c82647d6340be2a5ab19e7eb18a14@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 47688 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --- (11 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port exchangeUM:47688 Peer video RTP is at port exchangeUM:65535 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: <sip:goodproxy:5060> set_destination: Parsing <sip:goodproxy:5060> for address/port to send to set_destination: set destination to goodproxy, port 5060 Transmitting (no NAT) to goodproxy:5060: ACK sip:goodproxy:5060 SIP/2.0 Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK6c3f8b53;rport From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy>;tag=f51839ab98 Contact: <sip:3149@asterisk> Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/exchange12-082b8e18 answered SIP/jonlaptop-082b0fb8 -- Attempting native bridge of SIP/jonlaptop-082b0fb8 and SIP/exchange12-082b8e18 Scheduling destruction of call '109c82647d6340be2a5ab19e7eb18a14@asterisk' in 32000 ms set_destination: Parsing <sip:goodproxy:5060> for address/port to send to set_destination: set de5060 SIP/2.0 Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK47d7f855;rport From: "Jon Webster" <sip:3149@asterisk>;tag=as78ded60a To: <sip:6000@goodproxy>;tag=f51839ab98 Call-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (local, 7776000, 3) exited non-zero on 'SIP/jonlaptop-082b0fb8' pbx*CLI> <-- SIP read from goodproxy:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP asterisk:5060;received=goodproxy;branch=z9hG4bK47d7f855;rport FROM: "Jon Webster"<sip:3149@asterisk>;tag=as78ded60a TO: <sip:6000@exchangeUM>;tag=f51839ab98;epid=BD-70-82-06-F9 CSEQ: 103 BYE CALL-ID: 109c82647d6340be2a5ab19e7eb18a14@asterisk MAX-FORWARDS: 70 CONTENT-LENGTH: 0 SERVER: RTCC/2.0.6017.0 --- (9 headers 0 lines) --- Destroying call '109c82647d6340be2a5ab19e7eb18a14@asterisk' pbx*CLI>

1 0

[Users] CVS frozen - testing phase of 1.2.0
by Daniel-Constantin Mierla 05 Feb '07

05 Feb '07

Hello, starting with today the development for 1.2.0 is closed. No new features should be added until the release. The next month will be dedicated to testing, integration of submitted patches and fixing reported bugs. We have a lot of new code to test, your help is very much appreciated. I want to list the major changes you should focus on while testing: - usage of pseudo-variables and transformations is script (http://openser.org/dokuwiki/doku.php/transformations:devel) - TM timers (millisecond accuracy) - IP blacklists - DNS failover - automatic error handling - flags system (script flags, message flags, branch flags) - usrloc enhancements (licking system refurbished, database content optimization via DB fetch support) - new management interface (mi_fifo, mi_xmlrpc) - SIP Servlet Java Application server interoperability (seas module) - PERL AGI (perl module) - SNMP support via snmpstats module - Presence support (modules presence, pua, pua_usrloc, pua_mi) - xmpp IM gateway (xmpp module) - sip session timer support (module sst) - accounting clean up - federation based peering via domainpolicy module If I forgot something, please contribute. In addition, great help would be with creating 1.2.0 variants of documentation and dokuwiki pages (where is the case), testing the tools (openserctl, DB creation scripts) and developing scripts to migrate database from 1.1.x to 1.2.0. To keep track of discovered issues, please register them on the tracker -- it this way you make sure it is not lost in mail threads: http://sourceforge.net/tracker/?group_id=139143&atid=743020 Have nice testing sessions! Your feedback is always helpful! Cheers, Daniel

2 3

[Serusers] DB disconnects with MySQL 5.0.27
by sip 05 Feb '07

05 Feb '07

Something I've never run into before when I was using MySQL 4.x, but now crops up on MySQL 5.0.27... I get these errors: Feb 5 07:30:40 qabalah ser[26765]: submit_query(): MySQL server has gone away Feb 5 07:30:40 qabalah ser[26765]: db_update: Error while submitting query Feb 5 07:30:40 qabalah ser[26765]: db_upd_ucontact(): Error while updating database Feb 5 07:30:40 qabalah ser[26765]: wb_timer(): Error while updating contact in db I assume there's some sort of timeout of the SER connection to mysql? Is there a standard way to rectify this? Is this something new in MySQL 5? N.

1 1

[Users] Socket Errors
by Martin Burns 05 Feb '07

05 Feb '07

We have over 4 proxies running openser-1.1.0-notls. Recently we started to see the following error in the logs of one of the proxies: Feb 4 06:46:21 proxy01 /usr/local/sbin/openser[4670]: ERROR: udp_send: sendto(sock,0xb3c35898,594,0,0xb3c257f4,16): Invalid argument(22) Feb 4 06:46:21 proxy01 /usr/local/sbin/openser[4670]: CRITICAL: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net Feb 4 06:46:21 proxy01 /usr/local/sbin/openser[4670]: msg_send: ERROR: udp_send failed Feb 4 06:46:21 proxy01 /usr/local/sbin/openser[4670]: ERROR:tm:t_forward_nonack: sending request failed Now the errors suggests a binding to localhost, however our openser.cfg has the following: listen=eth0:5060 After a brief search we came across a couple of over people having a similar problem. Once of the suggested fixes was to add a mhomed=1 line to the cfg. Well we tried this (even though we are not multi homed) and the error changed to: Feb 5 02:54:09 proxy01 /usr/local/sbin/openser[2209]: ERROR: get_out_socket: connect failed: Invalid argument Feb 5 02:54:09 proxy01 /usr/local/sbin/openser[2209]: ERROR: get_out_socket: no socket found Feb 5 02:54:09 proxy01 /usr/local/sbin/openser[2209]: ERROR:tm:add_uac: can't fwd to af 2, proto 1 (no corresponding listening socket) Feb 5 02:54:09 proxy01 /usr/local/sbin/openser[2209]: ERROR:tm:t_forward_nonack: failure to add branches Does anyone have any suggestions or ideas as to what is causing this error. Thanks mburns

2 7

[Users] dbtext script
by Ovidiu Sas 05 Feb '07

05 Feb '07

Hi, Since dbtext is part of the standard, my suggestion would be to ship/install the sc.dbtext script along with the openserctl script. We can install it as openser_dbtext_ctl and in openserctl we can make a reference to it if no DBENGINE is provided: see the following patch for openserct (openser-v1.1.1): diff -Naurp sip-server/scripts/openserctl openser/scripts/openserctl --- sip-server/scripts/openserctl 2006-09-15 08:41:42.000000000 -0400 +++ openser/scripts/openserctl 2007-01-17 09:39:09.000000000 -0500 @@ -126,6 +126,8 @@ do $f done echo + echo "Note: Use openser_dbtext_ctl for DBTEXT operations!" + echo } Regards, Ovidiu Sas

2 3

[Users] auth_radius.so - ERROR: load_module: could not open module
by Luca Corti 05 Feb '07

05 Feb '07

Hello, I've built openser 1.1.1 from source with radius support. I installed radiusclient-ng. When I execute openser with to loadmodule auth_radius.so in my configuration I get: # openser 0(20268) ERROR: load_module: could not open module </usr/local/lib/openser/modules/auth_radius.so loadmodule >: /usr/local/lib/openser/modules/auth_radius.so loadmodule : cannot open shared object file: No such file or directory 0(20268) parse error (60,13-14): failed to load module 0(20268) parse error (60,14-15): syntax error 0(20268) parse error (60,14-15): ERROR: bad config file (3 errors) But the module is there. # ls -l /usr/local/lib/openser/modules/auth_radius.so -rwxr-xr-x 1 root staff 61369 Feb 5 10:33 /usr/local/lib/openser/modules/auth_radius.so This is really strange. ciao Luca

2 3

Re: [Users] In NAT, I didn't see the ACK packet after the Invited the Callee
by Bogdan-Andrei Iancu 05 Feb '07

05 Feb '07

have you checked this on the x-lite machine? regards, bogdan raviprakash sunkara wrote: > it didn't see any Ack packet is sending to SIP server, > > > On 2/5/07, *Bogdan-Andrei Iancu* <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > I would say the X-lite does not obey the RR header - please try to get > an trace from the X-lite machine to see where the ACK is sent. > > regards, > bogdan > > raviprakash sunkara wrote: > > > Hi BogDan, > > > > Yes, I see Record-Route header in the 200 ok....... > > Please can u check the ngrepped of the X-lite > > with Hard phones iit is fine ........... > > Below one is ngrep from X-lite users, > > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > M:+91 9985077535 > O:+91 40 23114549 > F:+91 40 40208727 > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > www.hyperion-tech.com <http://www.hyperion-tech.com>

1 0

[Users] default send-socket
by Klaus Darilion 05 Feb '07

05 Feb '07

Hi! What is the default send socket? Is it the first listen=..... statement in openser.cfg? Or can I manually set the default sendsocket? regards klaus

2 1

[Serusers] Re: load mysql.so failed
by Su Kerry 05 Feb '07

05 Feb '07

2 1

[Users] mediaproxy MySQL server has gone away error
by James Holden 05 Feb '07

05 Feb '07

Hi all, I'm having a problem with the mediaproxy rtp proxy. I'm using to do accounting into a MySQL database. After a period of inactivity, it loses it's connection to the MySQL server and the update queries begin to fail. Openser stays connected, however. The error in the logs is: error: accounting failed. MySQL error: (2006, 'MySQL server has gone away') I'm running: mediaproxy 1.8.0 openser-1.1.0 (using the RHEL3 rpm from the openser website) openser-mysql-1.1.0 (again using the openser RHEL3 rpm) Python-2.4.4 MySQL-python-1.2.1_p2 My Python skills aren't up to much. So far, I've tried added the following line into /usr/local/mediaproxy/modules/mysql.py just before the query call, to try and re-establish the connection: def query(self, query): self._query = query for c in range(0, 3): try: self.conn.db.ping() <<<<<<---- added by me return self.cursor.execute(query) except MySQLdb.OperationalError, why: if why[0] not in (CR.SERVER_GONE_ERROR, CR.SERVER_LOST): raise raise ... but it hasn't helped. Has anyone seen this before? Best regards, James -- James Holden AQ Limited Senior Systems Architect 13-15 Hunslet Road james.holden(a)uk.aql.com Leeds, LS10 1JQ 07989 702968 0800 389 5815 ===================================================== http://aql.com/documents/news/about_aql.pdf "The service has been perfect." BSI Inspectorate ltd "really really really useful. easy to use and reduces manpower." Fortnums "without a shadow of a doubt...excellent" Ruella James Plc ----------------------------------------------------- (aq) Limited 13-15 Hunslet Road Leeds LS10 1JQ 194 Harrogate Road Leeds LS7 4NZ http://www.aql.com Sales: 08707 449227 Fax: 08707 449228 PGP public key http://aql.co.uk/A-Q.txt Subject to our standard terms, found at http://aql.com/site/termsofservice.php ----------------------------------------------------- ****************************** CONFIDENTIALITY This e-mail is intended only for the use of the addressees named above and may be confidential or legally privileged. If you are not an addressee you must not read it and must not use any information contained in nor copy it nor inform any person other than (aq) limited or the addressees of its existence or contents. If you have received this email and are not a named addressee, please delete it and notify the (aq) limited customer service department via admin(a)a-q.co.uk. Many thanks. =========================================================

2 3

[Serusers] Dispatcher Module Help
by Giovanni Chirico 05 Feb '07

05 Feb '07

Hi, I would like to use the dispatcher module. I have configured it on my ser.cfg file for the PSTN calls and it works fine. However when the select gateway is unavailable the module don't switch to another gateway. Can you help me to write the "failure_route" so I can use another gateway when the first is unavailable? 'm using the "call forwarding ser.cfg" file of the SER Getting Started <http://www.iptel.org/ser/doc/gettingstarted> pack (in attachment I'm sending you my ser.cfg and dispatcher.list files) Thanks in advance for your help. Bye, Giovanni Chirico

1 0

Re: [Users] strip_tail() string length?
by Daniel-Constantin Mierla 05 Feb '07

05 Feb '07

Just a typo correction On 02/05/07 13:54, Brandon Armstead wrote: > Daniel-Constantin Mierla wrote: >> Stripping is now possible with the new introduced 'transformations': >> http://openser.org/dokuwiki/doku.php/transformations:devel >> >> So if you want to strip last two characters of username in R-URI >> >> $var(x) = $(rU{s.len})-2; >> $rU = $(ru{s.substr,0,$var(x)}); ^^^ this is: $rU = $(rU{s.substr,0,$var(x)}); Cheers, Daniel >> >> Cheers, >> Daniel >> >> >> On 01/30/07 09:08, Juha Heinanen wrote: >>> Brandon Armstead writes: >>> >>> > i.e. strip_tail(strlen($ru) - 2); ... or something like such. >>> >>> sorry, but my suggestion >>> >>> substr($ru, -2, 2) >>> >>> does not work, since it returns the last two chars of $ru not the >>> remaining chars. strlen is indeed needed for the above. using php >>> substr and strlen >>> >>> strip_tail(strlen($ru) - 2) >>> >>> becomes >>> >>> substr($ru, 0, strlen($ru) - 2) >>> >>> -- juha >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> > Hello Daniel, > > Glad to see this implemented, I think it will be a definite asset > to the future and improvements of OpenSER. Keep up the good work! > > Regards, > Brandon >

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Re: [Users] In NAT, I didn't see the ACK packet after the Invited the Callee
by Bogdan-Andrei Iancu 05 Feb '07

05 Feb '07

Ravi, so you see any Record-Route hdrs in the 200 OK reply? regards, bogdan raviprakash sunkara wrote: > Hi BogDan, > Good AfterNoon, and Thanks For replying to me, > > I used the Following code for Record_route(); > > route { > .... > ..... > if(method=="INVITE" && nat_uac_test("3") ) > { > record_route_preset("sip: > 192.168.2.3:5060 <http://192.168.2.3:5060>;nat=yes"); > }else if ( method ! = " REGISTER ") > { > record_route(); > } > } > where 192.168.2.3:5060 <http://192.168.2.3:5060> is the private IP of > the SIP server behind the NAT. > > > On 2/5/07, * Bogdan-Andrei Iancu* <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > Hi Ravi, > > do you do record_route() for the INVITE? if not, the sequential > requests > of the dialog will bypass your server. > > regards, > bogdan > > raviprakash sunkara wrote: > > > Hello Users, > > > > When I tested the OpenSER server with CISCO ATA 186/188 IP phones > > SIP EyeP media (Softphnoes ) is working fine with RTP , when Both > > Users and SIP Server is in Behind the NAT, > > > > But Problem with Softphones ( X-lite ) . > > > > When I ngrep the Calls of X-lite users i didn't see any Ack > Packet to > > OpenSER After the Invite Packets received . > > > > Please Help me in this Part > > > > > > -- > > Thanks and Regards > > Ravi Prakash Sunkara > > ravi.sunkara(a)hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com> > <mailto:ravi.sunkara@hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com>> > > M:+91 9985077535 > > O:+91 40 23114549 > > F:+91 40 40208727 > > ravi.sunkara(a)hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com> <mailto: > ravi.sunkara(a)hyperion-tech.com > <mailto:ravi.sunkara@hyperion-tech.com>> > > www.hyperion-tech.com <http://www.hyperion-tech.com> > <http://www.hyperion-tech.com> > > > >------------------------------------------------------------------------ > > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org <mailto:Users@openser.org> > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > M:+91 9985077535 > O:+91 40 23114549 > F:+91 40 40208727 > ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com> > www.hyperion-tech.com <http://www.hyperion-tech.com>

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Re: [Users] 200 ok not cought in ONREPLY
by Bogdan-Andrei Iancu 05 Feb '07

05 Feb '07

Dan, that is rather strange if you are confident you are setting correctly a reply route (not default one) for the INVITE transaction. add the following debug message in modules/tm/t_reply.c, after line 1312: DBG("---------onreply route is %u\n",t->on_reply); recompile ans see the output on runtime. regards, bogdan Dan-Cristian Bogos wrote: > Salut Bogdan, > > theoretically I have. Saw now that with default onreply I can catch it > (log). It is really strange since I am relying both BYE and INVITES in > the same t_relay block with t_on_reply("1"); > t_on_failure("1"); but the 200 OK comes to default. > > Cheers, > Dan > > On 2/5/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > >> Salut Dan, >> >> are you sure you have an onreply route sent for this transactions? from >> the output I see no problems. >> >> regards, >> bogdan >> >> Dan-Cristian Bogos wrote: >> >> > Hi Guys, >> > >> > By receiving incoming traffic from some other sip server (configured >> > for different domain than mines) I will not be able to catch in the >> > routing, onreply section, the 200 OK for the INVITE , OK which my >> > client, behind my server, is sending to remote side, therefore not >> > beeing able to modify sdp to use mediaproxy. Bellow is a level8 of >> > debug of the OK. >> > Is there any other way of catching this message in the routing? Any >> > help will be appreciated. >> > >> > Thxs in advance, >> > Dan >> > >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: SIP Reply >> > (status): >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: version: >> > <SIP/2.0> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: status: >> <200> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: reason: >> <OK> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bKeaa9.11b641a2.0>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: this is the first via >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: After >> > parse_msg... >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > forward_reply: found module nathelper, passing reply to it >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bKeaa9.2ffc88b3.0>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: this is the second via >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 234, <received> = <87.139.12.167>; state=6 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 235, <rport> = <5096>; state=6 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bK521554023>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > add_param: tag=8b2d7e45 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:parse_to:end of header reached, state=29 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DBUG:parse_to: display={}, ruri={sip:dan@de.babble.net} >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > get_hdr_field: <To> [38]; uri=[sip:dan@de.babble.net] >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: to >> > body [<sip:dan@de.babble.net>] >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > get_hdr_field: cseq <CSeq>: <20> <INVITE> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > forward_reply: found module tm, passing reply to it >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_check: msg id=1 global id=0 T start=0xffffffff >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=22 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=8 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_reply_matching: hash 39598 label 705981201 branch 0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:REF_UNSAFE: after is 1 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_reply_matching: reply matched (T=0x4081f430)! >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_check: msg id=1 global id=1 T end=0x4081f430 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:reply_received: org. status uas=180, uac[0]=180 local=0 >> > is_invite=1) >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:t_should_relay_response: T_code=180, new_code=200 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:relay_reply: branch=0, save=0, relay=0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: old size: >> > 911, new size: 850 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > build_res_from_sip_res: copied size: orig:77, new: 16, rest: 834 msg= >> > SIP/2.0 200 OK^M Via: SIP/2.0/UDP >> > 87.102.50.17;branch=z9hG4bKeaa9.2ffc88b3.0^M Via: SIP/2.0/UDP >> > >> 10.10.10.132:5096;received=87.139.12.167;rport=5096;branch=z9hG4bK521554023^M >> >> > >> > Record-Route: <sip:87.102.50.13;lr=on;ftag=2240571840>^M Record-Route: >> > <sip:87.102.50.17;lr;ftag=2240571840>^M Contact: >> > <sip:dan@87.139.12.167:7472;rinstance=37d0f83d79f3ede2>^M To: >> > <sip:dan@de.babble.net>;tag=8b2d7e45^M From: >> > <sip:dan@babble.net>;tag=2240571840^M Call-ID: >> > 1174762859(a)10.10.10.132^M CSeq: 20 INVITE^M Allow: INVITE, ACK, >> > CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO^M >> > Content-Type: application/sdp^M User-Agent: X-Lite release 1006e stamp >> > 34025^M Content-Length: 185^M ^M v=0^M o=- 5 2 IN IP4 87.139.12.167^M >> > s=CounterPath X-Lite 3.0^M c=IN IP4 87.139.12.167^M t=0 0^M m=audio >> > 34556 RTP/AVP 0 101^M a=fmtp:101 0-15^M a=rtpmap:101 >> > telephone-event/8000^M a=sendrecv^M >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > add_to_tail_of_timer[2]: 0x4081f478 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:relay_reply: sent buf=0x814a048: SIP/2.0 2..., >> > shmem=0x40821528: SIP/2.0 2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > cleanup_uac_timers: RETR/FR timers reset >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:UNREF_UNSAFE: after is 0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:destroy_avp_list: destroying list (nil) >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: receive_msg: >> > cleaning up >> > >> > _______________________________________________ >> > Users mailing list >> > Users(a)openser.org >> > http://openser.org/cgi-bin/mailman/listinfo/users >> > >> >> >

1 0

[Users] Hijack 'Move Temporarily'
by Sam Lee 05 Feb '07

05 Feb '07

Hey Guys, Normally if the SIP-UA is call-forwarded to a different number , it will send a 'Move Temporarily' to the proxy , which in turn relay this message to the originating gateway. The problem is , if they call forward to a number which the originating gateway doesn't know what to do with it , it will just kill the call. Can i hijack 'Move Temporarily' directly in the proxy and sending this call to a different gateway rather than relaying it to the originating gateway ? Please let me know if you can't understand me. Thanks! Regards, Sam

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[Users] database persistent connection
by tele 05 Feb '07

05 Feb '07

Hi, I've an annoying problem with database persistent connection. I'm using mysql database for location and avp stuff. During the day all it's but after a day i retest some call and seams that the proxy lost connection with the database, and avp and location stuff didn't works anymore. I'm using this version of openser: # openser -V version: openser 1.2.0-dev8-notls (i386/freebsd) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, select, kqueue. @(#) $Id: main.c,v 1.23 2006/10/02 15:51:45 bogdan_iancu Exp $ main.c compiled on 12:38:52 Nov 27 2006 with gcc 3.4.4 i've also configure a small ping_interval for the mysql module modparam("mysql", "ping_interval", 60) modparam("mysql", "auto_reconnect", 1) and the usrloc: modparam("usrloc", "db_mode", 3) modparam("usrloc", "matching_mode", 1) I've also check that the NTP is correct on the servers. I don't know why openser lost connection with the database. any hints? thanks :tele

2 2

[Users] Help. in OpenSER in Post-paid and pre-paid
by raviprakash sunkara 05 Feb '07

05 Feb '07

Hello Users, I'm Using The Following software of My Sip productions, 1) OpenSER is a Sip Server,proxy, in front-end 2) Asterisk is PBX server, in Back-end 3) Radius for AAA to OpenSER, 4) Asterisk-java for IVR for Billing Customer care if I did any mistakes in these mail , forgive me . My Question is 1) If Caller calls to Callee , Before the call established to callee. The Caller has to know his Balance Account. if the Balance is there, then only the call has to establish, 2) I want the OpenSER total in RealTime application Server like Java application . Call Authorization ( Question no 1 ). If Asterisk is Front-end , ---> problem is Solved, How can a b2bua implement with OpenSER, please when -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

2 2

[Users] strip_tail() string length?
by Brandon Armstead 05 Feb '07

05 Feb '07

Hello everyone, I'm wondering what the easiest method would be to grab how long the RURI is and strip based on that information... or would I have to actually use regex, or is there an easier STRLEN() or LENGTH() operation? Any ideas, or is this something that could be implemented? Thanks. i.e. strip_tail(strlen($ru) - 2); ... or something like such. -- Brandon Armstead

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[Serusers] Cisco AS 5300 can't send BYE for SER... It's bug?
by Sho Aihara 05 Feb '07

05 Feb '07

Hi all. I have a problem for the following scenario. When I make a call for PSTN and on hook by PSTN side, Cisco As can't send BYE for SER. 01. [UA via Asterisk] dialing "08022223333" -> [SER] 02. [SER] prefix("0333") and rewritehostport("my.cisco.ip.address:5060") -> [Cisco] 03. [Cisco] dial-peer voice 6000 pots, translate-outgoing called from "033308022223333" to "008022223333" 04. [Cisco] process an outbound call to "008022223333" -> [e.g. Mobile] 05. [e.g. Mobile] Catch call 06. [SER] log CDR start 07. [Cisco] talking 08. [e.g. Mobile] On hook and call disconnect 09. [Cisco] can't send BYE for SER *****why??***** 10. [UA via Asterisk] On hook 11. [UA via Asterisk] Send BYE for SER 12. [SER] log CDR End [Cisco] Call finished But another scenario, if make a call from PSTN to Asterisk and on hook by PSTN side, Cisco As send BYE to SER. 01. [e.g. Mobile] dialing "0377771111(Asterisk user number)" 02. [Cisco] receive "77771111" call number 03. [Cisco] dial-peer voice 5000 voip, session target ipv4: my.ser.ip.address -> [SER] 04. [SER] process an incoming call to "0377771111" -> [UA via Asterisk] 05. [UA via Asterisk] Catch call 06. [SER] log CDR start 07. [UA via Asterisk] talking 08. [e.g. Mobile] On hook and call disconnect 09. [Cisco] Send BYE to SER 10. [SER] log CDR End [Cisco] Call finished 11. [UA via Asterisk] receive BYE from SER And sorry for my diffucult example. Why Cisco AS 5300 can't send BYE to SER When PSTN call is disconnected by PSTN side? My ser.cfg as follows: # -------------------------------------------------------------------------- # global configuration parameters # -------------------------------------------------------------------------- fork=no log_stderror=yes check_via=no dns=no rev_dns=no listen=my.ser.ip.address port=5060 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # -------------------------------------------------------------------------- # module loading # -------------------------------------------------------------------------- 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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/exec.so" # -------------------------------------------------------------------------- # setting module-specific parameters # -------------------------------------------------------------------------- modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") modparam("permissions", "db_url", "mysql://ser:heslo@localhost/ser") modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) # -------------------------------------------------------------------------- # route pattern # -------------------------------------------------------------------------- route { 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; }; record_route(); if (loose_route()) { if (method=="ACK") { acc_db_request("01:CallStart\n", "acc"); }; if (method=="BYE" || method=="CANCEL") { acc_db_request("02:CallEnd\n", "acc"); }; t_relay(); break; }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); break; }; if (search("^(f|From): .*(a)(my\.cisco\.ip\.address)")) { #PSTN Incoming call from Cisco AS 5300 e.g. 0377771111 rewritehost("my.asterisk.ip.address"); }; lookup("aliases"); if (!lookup("location")) { if (method=="INVITE" && !search("^(f|From): .*(a)(my\.cisco\.ip\.address)")) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; }; if (uri=~"^sip:0[0-9]{10}@") { # PSTN Outgoing call to Cisco AS 5300 e.g. 08022223333 prefix("0333"); rewritehostport("my.cisco.ip.address:5060"); avp_write("i:45", "inv_timeout"); } else { sl_send_reply("404", "Not Found"); break; }; consume_credentials(); }; }; }; if (!t_relay()) { sl_reply_error(); }; } And my Cisco AS 5300 config as follows: voice call send-alert voice rtp send-recv voice service pots fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco voice service voip fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco sip min-se 60 translation-rule 50 Rule 0 0333 0 Rule 1 ^7777 037777 voice class codec 2 codec preference 1 g711ulaw codec preference 2 g711alaw dial-peer voice 5000 voip tone ringback alert-no-PI description ser-asterisk-cisco-test huntstop destination-pattern 77771111$ translate-outgoing called 50 voice-class codec 2 session protocol sipv2 session target ipv4:my.ser.ip.address dtmf-relay rtp-nte max-conn 1 dial-peer voice 6000 pots application session max-conn 2 destination-pattern 0333T progress_ind alert enable 8 translate-outgoing called 50 port 0:D Thanks, Sahria

3 2

Re: [Users] 200 ok not cought in ONREPLY
by Daniel-Constantin Mierla 05 Feb '07

05 Feb '07

Hello, On 02/05/07 12:30, Dan-Cristian Bogos wrote: > Dear Daniel, > > many thanks for the reply. I can perfectly catch the OK now. ok > > The only thing which I must find out is why for a normal INVITE the OK > was not coming in the same onreply block as the BYE, since both INVITE > and BYE are relied in the same way out and processed theoretically in > the same route block. BYE is usually following the loose routing -- but I do not know if it is the case of your comfig. Cheers, Daniel > > Anyway, thank you again, it was really great help. > > Cheers, > Dan > > > On 2/5/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> Hello, >> >> to get a reply in an onreply_route other than the default on, you have >> to arm it via t_on_reply() when you relay the request. >> ... >> t_on_reply("1"); >> ... >> >> t_on_reply[1] { >> } >> >> Otherwise, all replies are processed in the default onreply_route (the >> one which has no index) >> >> onreply_route { >> .... >> } >> >> Cheers, >> Daniel >> >> On 02/01/07 15:48, Dan-Cristian Bogos wrote: >> > Hi Guys, >> > >> > By receiving incoming traffic from some other sip server (configured >> > for different domain than mines) I will not be able to catch in the >> > routing, onreply section, the 200 OK for the INVITE , OK which my >> > client, behind my server, is sending to remote side, therefore not >> > beeing able to modify sdp to use mediaproxy. Bellow is a level8 of >> > debug of the OK. >> > Is there any other way of catching this message in the routing? Any >> > help will be appreciated. >> > >> > Thxs in advance, >> > Dan >> > >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: SIP Reply >> > (status): >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: version: >> > <SIP/2.0> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: status: >> <200> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: reason: >> <OK> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bKeaa9.11b641a2.0>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: this is the first via >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: After >> > parse_msg... >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > forward_reply: found module nathelper, passing reply to it >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bKeaa9.2ffc88b3.0>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: this is the second via >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 234, <received> = <87.139.12.167>; state=6 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 235, <rport> = <5096>; state=6 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param >> > type 232, <branch> = <z9hG4bK521554023>; state=16 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of >> > header reached, state=5 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: Via found, flags=20 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > add_param: tag=8b2d7e45 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:parse_to:end of header reached, state=29 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DBUG:parse_to: display={}, ruri={sip:dan@de.babble.net} >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > get_hdr_field: <To> [38]; uri=[sip:dan@de.babble.net] >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: to >> > body [<sip:dan@de.babble.net>] >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > get_hdr_field: cseq <CSeq>: <20> <INVITE> >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > forward_reply: found module tm, passing reply to it >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_check: msg id=1 global id=0 T start=0xffffffff >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=22 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > parse_headers: flags=8 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_reply_matching: hash 39598 label 705981201 branch 0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:REF_UNSAFE: after is 1 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_reply_matching: reply matched (T=0x4081f430)! >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > t_check: msg id=1 global id=1 T end=0x4081f430 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:reply_received: org. status uas=180, uac[0]=180 local=0 >> > is_invite=1) >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:t_should_relay_response: T_code=180, new_code=200 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:relay_reply: branch=0, save=0, relay=0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: old size: >> > 911, new size: 850 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > build_res_from_sip_res: copied size: orig:77, new: 16, rest: 834 msg= >> > SIP/2.0 200 OK^M Via: SIP/2.0/UDP >> > 87.102.50.17;branch=z9hG4bKeaa9.2ffc88b3.0^M Via: SIP/2.0/UDP >> > >> 10.10.10.132:5096;received=87.139.12.167;rport=5096;branch=z9hG4bK521554023^M >> >> > >> > Record-Route: <sip:87.102.50.13;lr=on;ftag=2240571840>^M Record-Route: >> > <sip:87.102.50.17;lr;ftag=2240571840>^M Contact: >> > <sip:dan@87.139.12.167:7472;rinstance=37d0f83d79f3ede2>^M To: >> > <sip:dan@de.babble.net>;tag=8b2d7e45^M From: >> > <sip:dan@babble.net>;tag=2240571840^M Call-ID: >> > 1174762859(a)10.10.10.132^M CSeq: 20 INVITE^M Allow: INVITE, ACK, >> > CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO^M >> > Content-Type: application/sdp^M User-Agent: X-Lite release 1006e stamp >> > 34025^M Content-Length: 185^M ^M v=0^M o=- 5 2 IN IP4 87.139.12.167^M >> > s=CounterPath X-Lite 3.0^M c=IN IP4 87.139.12.167^M t=0 0^M m=audio >> > 34556 RTP/AVP 0 101^M a=fmtp:101 0-15^M a=rtpmap:101 >> > telephone-event/8000^M a=sendrecv^M >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > add_to_tail_of_timer[2]: 0x4081f478 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:relay_reply: sent buf=0x814a048: SIP/2.0 2..., >> > shmem=0x40821528: SIP/2.0 2 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: >> > cleanup_uac_timers: RETR/FR timers reset >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:tm:UNREF_UNSAFE: after is 0 >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: >> > DEBUG:destroy_avp_list: destroying list (nil) >> > Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: receive_msg: >> > cleaning up >> > >> > _______________________________________________ >> > Users mailing list >> > Users(a)openser.org >> > http://openser.org/cgi-bin/mailman/listinfo/users >> > >> >

1 0

[Users] OpenSER 1.2.0 + Presence
by Bernard Chan 05 Feb '07

05 Feb '07

Hello, I'm currently experimenting with OpenSER 1.2.0 with Presence module. Earlier I tried OpenSER 1.0/1.1 with PA. I'm using a third-party softphone which for some reasons, I would not like to disclose the name of it. After switching to 1.2.0+Presence, the softphone segfaults on startup. After tracing the source code of the phone and OpenSER and performing hundreds of packet sniffing over the past several days, I have grown to suspect that a certain behaviour with the new presence module may be the cause, but I would like to verify whether that is the case. For some reasons it is infeasible for me to modify the source code of the softphone used (actually I may be able to but that is what I would like to avoid). But it seems to choke on receiving a NOTIFY before the OK for the corresponding SUBSCRIBE is received by the softphone. So, any chance that I can test it by just modifying the config of OpenSER alone without changing the sources, both of OpenSER and the softphone? Thanks in advance for any invaluable insights. Regards, Bernard Chan.

2 1

[Users] In NAT, I didn't see the ACK packet after the Invited the Callee
by raviprakash sunkara 05 Feb '07

05 Feb '07

Hello Users, When I tested the OpenSER server with CISCO ATA 186/188 IP phones SIP EyeP media (Softphnoes ) is working fine with RTP , when Both Users and SIP Server is in Behind the NAT, But Problem with Softphones ( X-lite ) . When I ngrep the Calls of X-lite users i didn't see any Ack Packet to OpenSER After the Invite Packets received . Please Help me in this Part -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

2 1

[Users] 200 ok not cought in ONREPLY
by Dan-Cristian Bogos 05 Feb '07

05 Feb '07

Hi Guys, By receiving incoming traffic from some other sip server (configured for different domain than mines) I will not be able to catch in the routing, onreply section, the 200 OK for the INVITE , OK which my client, behind my server, is sending to remote side, therefore not beeing able to modify sdp to use mediaproxy. Bellow is a level8 of debug of the OK. Is there any other way of catching this message in the routing? Any help will be appreciated. Thxs in advance, Dan Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: SIP Reply (status): Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: version: <SIP/2.0> Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: status: <200> Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: reason: <OK> Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: flags=2 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param type 232, <branch> = <z9hG4bKeaa9.11b641a2.0>; state=16 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of header reached, state=5 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: Via found, flags=2 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: this is the first via Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: After parse_msg... Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: forward_reply: found module nathelper, passing reply to it Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: flags=20 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param type 232, <branch> = <z9hG4bKeaa9.2ffc88b3.0>; state=16 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of header reached, state=5 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: Via found, flags=20 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: this is the second via Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param type 234, <received> = <87.139.12.167>; state=6 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param type 235, <rport> = <5096>; state=6 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: Found param type 232, <branch> = <z9hG4bK521554023>; state=16 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: end of header reached, state=5 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: Via found, flags=20 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: add_param: tag=8b2d7e45 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:parse_to:end of header reached, state=29 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DBUG:parse_to: display={}, ruri={sip:dan@de.babble.net} Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: get_hdr_field: <To> [38]; uri=[sip:dan@de.babble.net] Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: to body [<sip:dan@de.babble.net>] Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: get_hdr_field: cseq <CSeq>: <20> <INVITE> Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: forward_reply: found module tm, passing reply to it Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: t_check: msg id=1 global id=0 T start=0xffffffff Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: flags=22 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: parse_headers: flags=8 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: t_reply_matching: hash 39598 label 705981201 branch 0 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:tm:REF_UNSAFE: after is 1 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: t_reply_matching: reply matched (T=0x4081f430)! Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: t_check: msg id=1 global id=1 T end=0x4081f430 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:tm:reply_received: org. status uas=180, uac[0]=180 local=0 is_invite=1) Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:tm:t_should_relay_response: T_code=180, new_code=200 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:tm:relay_reply: branch=0, save=0, relay=0 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: old size: 911, new size: 850 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: build_res_from_sip_res: copied size: orig:77, new: 16, rest: 834 msg= SIP/2.0 200 OK^M Via: SIP/2.0/UDP 87.102.50.17;branch=z9hG4bKeaa9.2ffc88b3.0^M Via: SIP/2.0/UDP 10.10.10.132:5096;received=87.139.12.167;rport=5096;branch=z9hG4bK521554023^M Record-Route: <sip:87.102.50.13;lr=on;ftag=2240571840>^M Record-Route: <sip:87.102.50.17;lr;ftag=2240571840>^M Contact: <sip:dan@87.139.12.167:7472;rinstance=37d0f83d79f3ede2>^M To: <sip:dan@de.babble.net>;tag=8b2d7e45^M From: <sip:dan@babble.net>;tag=2240571840^M Call-ID: 1174762859(a)10.10.10.132^M CSeq: 20 INVITE^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO^M Content-Type: application/sdp^M User-Agent: X-Lite release 1006e stamp 34025^M Content-Length: 185^M ^M v=0^M o=- 5 2 IN IP4 87.139.12.167^M s=CounterPath X-Lite 3.0^M c=IN IP4 87.139.12.167^M t=0 0^M m=audio 34556 RTP/AVP 0 101^M a=fmtp:101 0-15^M a=rtpmap:101 telephone-event/8000^M a=sendrecv^M Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: add_to_tail_of_timer[2]: 0x4081f478 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:tm:relay_reply: sent buf=0x814a048: SIP/2.0 2..., shmem=0x40821528: SIP/2.0 2 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:tm:UNREF_UNSAFE: after is 0 Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: DEBUG:destroy_avp_list: destroying list (nil) Feb 1 13:33:48 localhost /usr/local/sbin/openser[17816]: receive_msg: cleaning up

3 2

Res: [Users] 407 and algorithm=MD5
by Bruno Machado 05 Feb '07

05 Feb '07

The section 3.2.1 (The WWW-Authenticate Response Header) from RFC 2617 says: "algorithm A string indicating a pair of algorithms used to produce the digest and a checksum. If this is not present it is assumed to be "MD5". " I send a INVITE to Openser but it send back a 407 (Proxy Authentication). The parameter response was OK. But, without the parameter 'algorithm', the proxy didn't allow the authentication. It neither consult at Radius. Looking at the RFC above, the SIP proxy must to use MD5 by default, but it don't accept the message without the parameter. Who needs to assume the algorithm? UAC or SIP Proxy? Can it be a bug of OpenSER? Or my interpretation is wrong? Best regards. Bruno Machado ----- Mensagem original ---- De: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> Para: Bruno Machado <bsmufu(a)yahoo.com.br> Cc: users(a)openser.org Enviadas: Segunda-feira, 29 de Janeiro de 2007 14:21:31 Assunto: Re: [Users] 407 and algorithm=MD5 Hi Bruno, you cannot configure to have the algorithm param added. If I'm not wrong, the digest RFC says this is an optional param and the default value is MD5. regards, bogdan Bruno Machado wrote: > Hi. > > When I use the function proxy_challenge(), the 407 don't have the > parameter 'algorithm=MD5'. Look below: > > ... > Proxy-Authenticate: Digest realm="proxy.com.br", > nonce="45be00f941f137ff037436c77f80a9531a02155f". > ... > > How can I put this parameter in digest informations? > > ... > Proxy-Authenticate: Digest realm="proxy.com.br", > nonce="45be00f941f137ff037436c77f80a9531a02155f", > algorithm="MD5". > ... > > > Thanks in advance. > > Bruno > > __________________________________________________ > Fale com seus amigos de graça com o novo Yahoo! Messenger > http://br.messenger.yahoo.com/ > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > __________________________________________________ Fale com seus amigos de graça com o novo Yahoo! Messenger http://br.messenger.yahoo.com/

3 3

[Users] permission.allow example
by Ion Minzu 05 Feb '07

05 Feb '07

Hello , Can someone describe me a real example how to write rules in permission.allow, permission.deny. I use the example from file permission.* and it doesn't work, permission is set ALL to ALL. Or maybe exist other method to make call restrictions? #openser.cfg .... loadmodule "/usr/lib/openser/modules/permissions.so" .... modparam("permissions", "default_deny_file", "/etc/openser/permissions.deny") modparam("permissions", "default_allow_file", "/etc/openser/permissions.allow") .... -- Best regards, Ion Minzu, mailto:ion.minzu@cts.md

2 1

[Serusers] Cisco AS 5300 can't send BYE for SER
by Sahria Hao 05 Feb '07

05 Feb '07

Hi all. I have a problem for the following scenario. When I make a call for PSTN and on hook by PSTN side, Cisco As can't send BYE for SER. 01. [UA via Asterisk] dialing "08022223333" -> [SER] 02. [SER] prefix("0333") and rewritehostport("my.cisco.ip.address:5060") -> [Cisco] 03. [Cisco] dial-peer voice 6000 pots, translate-outgoing called from "033308022223333" to "008022223333" 04. [Cisco] process an outbound call to "008022223333" -> [e.g. Mobile] 05. [e.g. Mobile] Catch call 06. [SER] log CDR start 07. [Cisco] talking 08. [e.g. Mobile] On hook and call disconnect 09. [Cisco] can't send BYE for SER *****why??***** 10. [UA via Asterisk] On hook 11. [UA via Asterisk] Send BYE for SER 12. [SER] log CDR End [Cisco] Call finished But another scenario, if make a call from PSTN to Asterisk and on hook by PSTN side, Cisco As send BYE to SER. 01. [e.g. Mobile] dialing "0377771111(Asterisk user number)" 02. [Cisco] receive "77771111" call number 03. [Cisco] dial-peer voice 5000 voip, session target ipv4: my.ser.ip.address -> [SER] 04. [SER] process an incoming call to "0377771111" -> [UA via Asterisk] 05. [UA via Asterisk] Catch call 06. [SER] log CDR start 07. [UA via Asterisk] talking 08. [e.g. Mobile] On hook and call disconnect 09. [Cisco] Send BYE to SER 10. [SER] log CDR End [Cisco] Call finished 11. [UA via Asterisk] receive BYE from SER And sorry for my diffucult example. Why Cisco AS 5300 can't send BYE to SER When PSTN call is disconnected by PSTN side? My ser.cfg as follows: # -------------------------------------------------------------------------- # global configuration parameters # -------------------------------------------------------------------------- fork=no log_stderror=yes check_via=no dns=no rev_dns=no listen=my.ser.ip.address port=5060 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # -------------------------------------------------------------------------- # module loading # -------------------------------------------------------------------------- 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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/exec.so" # -------------------------------------------------------------------------- # setting module-specific parameters # -------------------------------------------------------------------------- modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("usrloc", "db_url", " mysql://ser:heslo@localhost/ser") modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") modparam("permissions", "db_url", " mysql://ser:heslo@localhost /ser") modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) # -------------------------------------------------------------------------- # route pattern # -------------------------------------------------------------------------- route { 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; }; record_route(); if (loose_route()) { if (method=="ACK") { acc_db_request("01:CallStart\n", "acc"); }; if (method=="BYE" || method=="CANCEL") { acc_db_request("02:CallEnd\n", "acc"); }; t_relay(); break; }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); break; }; if (search("^(f|From): .*(a)(my\.cisco\.ip\.address<.*(a)(my%5C.cisco%5C.ip%5C.address>)")) { #PSTN Incoming call from Cisco AS 5300 e.g. 0377771111 rewritehost("my.asterisk.ip.address "); }; lookup("aliases"); if (!lookup("location")) { if (method=="INVITE" && !search("^(f|From): .*(a)(my\.cisco\.ip\.address<.*(a)(my%5C.cisco%5C.ip%5C.address>)")) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; }; if (uri=~"^sip:0[0-9]{10}@") { # PSTN Outgoing call to Cisco AS 5300 e.g. 08022223333 prefix("0333"); rewritehostport("my.cisco.ip.address:5060"); avp_write("i:45", "inv_timeout"); } else { sl_send_reply("404", "Not Found"); break; }; consume_credentials(); }; }; }; if (!t_relay()) { sl_reply_error(); }; } And my Cisco AS 5300 config as follows: voice call send-alert voice rtp send-recv voice service pots fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco voice service voip fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco sip min-se 60 translation-rule 50 Rule 0 0333 0 Rule 1 ^7777 037777 voice class codec 2 codec preference 1 g711ulaw codec preference 2 g711alaw dial-peer voice 5000 voip tone ringback alert-no-PI description ser-asterisk-cisco-test huntstop destination-pattern 77771111$ translate-outgoing called 50 voice-class codec 2 session protocol sipv2 session target ipv4:my.ser.ip.address dtmf-relay rtp-nte max-conn 1 dial-peer voice 6000 pots application session max-conn 2 destination-pattern 0333T progress_ind alert enable 8 translate-outgoing called 50 port 0:D Thanks, Sahria

1 0

[Users] OpenSER devel & SNOM 360
by mika.saari＠wipsl.com 05 Feb '07

05 Feb '07

Hi, I do have OpenSER 1.1.0 with TLS support on Debian Sarge working nicely, and my SNOM 360 registers against it nicely. When using OpenSER 1.2 (2 days old CVS build) with TLS on Debian Etch with SNOM 360 and own signed certificates (with tls/tools & own machine names) I get debug info (below) from OpenSER. Any ideas with that "peer did not send sertificate" ? Thank you so many times, -Mika --- clip --- Feb 2 13:15:10 localhost openser[20855]: tcp_receive_timeout: 0xb59e1900 expired (296, 297) lt=0 Feb 2 13:15:10 localhost openser[20855]: DBG: io_watch_del (0x8122980, 18, -1, 0x10) fd_no=2 called Feb 2 13:15:10 localhost openser[20855]: releasing con 0xb59e1900, state 0, fd=18, id=2 Feb 2 13:15:10 localhost openser[20855]: extra_data (nil) Feb 2 13:15:10 localhost openser[20859]: handle_tcp_child: reader response= b59e1900, 0 from 0 Feb 2 13:15:10 localhost openser[20859]: DBG: io_watch_add(0x8122820, 23, 2, 0xb59e1900), fd_no=15 Feb 2 13:15:10 localhost openser[20859]: handle_tcp_child: CONN_RELEASE 0xb59e1900 refcnt= 0 Feb 2 13:15:10 localhost openser[20859]: tcpconn_new: new tcp connection to: 193.65.183.233 Feb 2 13:15:10 localhost openser[20859]: tcpconn_new: on port 2063, type 3 Feb 2 13:15:10 localhost openser[20859]: tls_tcpconn_init: Entered: Creating a whole new ssl connection Feb 2 13:15:10 localhost openser[20859]: tls_tcpconn_init: Looking up socket based TLS server domain [193.65.183.13:5061] Feb 2 13:15:10 localhost openser[20859]: tls_find_server_domain: virtual TLS server domain not found, Using default TLS server domain settings Feb 2 13:15:10 localhost openser[20859]: tls_tcpconn_init: Found socket based TLS server domain [0.0.0.0:0] Feb 2 13:15:10 localhost openser[20859]: tls_tcpconn_init: Setting in ACCEPT mode (server) Feb 2 13:15:10 localhost openser[20859]: tcpconn_add: hashes: 462, 9 Feb 2 13:15:10 localhost openser[20859]: handle_new_connect: new connection: 0xb5a06c88 24 flags: 0002 Feb 2 13:15:10 localhost openser[20859]: send2child: to tcp child 0 6(20855), 0xb5a06c88 Feb 2 13:15:10 localhost openser[20855]: received n=4 con=0xb5a06c88, fd=18 Feb 2 13:15:10 localhost openser[20855]: DBG: io_watch_add(0x8122980, 18, 2, 0xb5a06c88), fd_no=1 Feb 2 13:15:11 localhost openser[20855]: tls_update_fd: New fd is 18 Feb 2 13:15:11 localhost openser[20855]: tls_update_fd: New fd is 18 Feb 2 13:15:11 localhost openser[20855]: tls_accept: Error in SSL: Feb 2 13:15:11 localhost openser[20855]: tls_error: error:140890C7:SSL routines:SSL3_GET_CLIENT_CERTIFICATE:peer did not return a certificate Feb 2 13:15:11 localhost openser[20855]: DBG: io_watch_del (0x8122980, 18, -1, 0x10) fd_no=2 called Feb 2 13:15:11 localhost openser[20855]: releasing con 0xb5a06c88, state -2, fd=18, id=9 Feb 2 13:15:11 localhost openser[20855]: extra_data 0xb59f1a18 Feb 2 13:15:11 localhost openser[20859]: handle_tcp_child: reader response= b5a06c88, -2 from 0 Feb 2 13:15:11 localhost openser[20859]: tcpconn_destroy: destroying connection 0xb5a06c88, flags 0002 Feb 2 13:15:11 localhost openser[20859]: tls_close: Closing SSL connection Feb 2 13:15:11 localhost openser[20859]: tls_update_fd: New fd is 24 Feb 2 13:15:11 localhost openser[20859]: tls_shutdown: Shutdown successful Feb 2 13:15:11 localhost openser[20859]: tls_tcpconn_clean: Entered -- clip ---

2 2

[Users] Asterisk OpenSER and Exchange 2007 Unified Messaging
by Jon Webster 05 Feb '07

05 Feb '07

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@openser SIP/2.0. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK629bdb92;rport. From: "Jon Webster" <sip:3149@asterisk>;tag=as43593b57. To: <sip:8885555@openser>. Contact: <sip:3149@asterisk>. Call-ID: 46109877480231241e3b8e9d57036e69@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@asterisk>;tag=as43593b57. To: <sip:8885555@openser>. Call-ID: 46109877480231241e3b8e9d57036e69@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@openser out_uri=sip:5555@openser via_cnt==1". . # U +0.025313 openser:5060 -> asterisk:5060 SIP/2.0 180 Ringing. FROM: "Jon Webster"<sip:3149@asterisk>;tag=as43593b57. TO: <sip:8885555@openser>;epid=BD-70-82-06-F9;tag=e38b6cdb9b. CSEQ: 102 INVITE. CALL-ID: 46109877480231241e3b8e9d57036e69@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@asterisk>;tag=as43593b57. TO: <sip:8885555@openser>;epid=BD-70-82-06-F9;tag=e38b6cdb9b. CSEQ: 102 INVITE. CALL-ID: 46109877480231241e3b8e9d57036e69@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@goodproxy SIP/2.0. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK42383fff;rport. From: "Jon Webster" <sip:3149@asterisk>;tag=as07e96b4e. To: <sip:5555@goodproxy>. Contact: <sip:3149@asterisk>. Call-ID: 7d0450601533753042ce2d835a3f98d2@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@asterisk>;tag=as07e96b4e. To: <sip:5555@goodproxy>. Call-ID: 7d0450601533753042ce2d835a3f98d2@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@asterisk>;tag=as07e96b4e. TO: <sip:5555@exchangeUM>;epid=BD-70-82-06-F9;tag=eb4f3cf30. CSEQ: 102 INVITE. CALL-ID: 7d0450601533753042ce2d835a3f98d2@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@asterisk>;tag=as07e96b4e. TO: <sip:5555@exchangeUM>;epid=BD-70-82-06-F9;tag=eb4f3cf30. CSEQ: 102 INVITE. CALL-ID: 7d0450601533753042ce2d835a3f98d2@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@asterisk>;tag=as07e96b4e. To: <sip:5555@goodproxy>;tag=eb4f3cf30. Contact: <sip:3149@asterisk>. Call-ID: 7d0450601533753042ce2d835a3f98d2@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@asterisk>;tag=as07e96b4e. To: <sip:5555@goodproxy>;tag=eb4f3cf30. Call-ID: 7d0450601533753042ce2d835a3f98d2@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@asterisk>;tag=as07e96b4e. TO: <sip:5555@exchangeUM>;tag=eb4f3cf30;epid=BD-70-82-06-F9. CSEQ: 103 BYE. CALL-ID: 7d0450601533753042ce2d835a3f98d2@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"); }

2 1

[Users] Change a Status Code From 182 to 180
by Marc Schaefer 05 Feb '07

05 Feb '07

Hello all, I've a small problem that causes a lot of problems and is in some way specific: I've a Openser 1.1.1 Server running on a public rootserver. User are registered with FritzBox ( DSL-Router with Voip-Stack) or a Soft/Hardphone being Nated. For the PSTN connectivity i've access to a PSTN Gateway from the BT. Everything works very nice with just one problem remaining. When one User gets a call from PSTN and picks it up, no problems. But the problem comes when a second call is made to the same User. His phone tells him that somebody is knocking and he can pickup the phone without any problem. The problem is on the callers side because he doesn't get any feedback that he is ringing/knocking, so most of the time he will just cancel the call. I did some research about it and here are my conclusions: The INVITE from the second call isn't answered with a "180 Ringing.." but with a "182 Queued.." which doesn't trigger a Ringing for the PSTN Call. When I talked to the BT support they did not really know about the status code 182 and could not give me an answer how their mediagateway will handle it ( They promised me to get the informtion :-) ). On the other side I can't change the Firmware so there's no way of persuading the DSL-Router to use 180 instead of 182. So my questions are: Can i change the status code at the proxy? If so, in which part of the routing logic do I have to change it? I tried a lot of different subst() and replace() commands but none had any effect on the sip messages it always kept being 182. Below you can see my configuration which is based on the template from ixcall ( http://www.ixcall.net/?q=node/32 ) with a lot of changes and support from the "getting started" guide from onsip.org, of course adapted to the openser 1.1.1 version I have attached my config. Keep in mind that this config will get through a m4 parser before being ready to be used ( mechanism can be seen at ixcall ). I hope my english is good enough to have my problem explained in a understandable manner Have fun Marc /* ==================================================================== Schnuffle Openser Configuration V6.3 Based on IxCall template V2.4 (http://www.ixcall.net/?q=node/32) And OnSip.org "Getting Started" Guid (http://siprouter.onsip.org/doc/gettingstarted/) ==================================================================== list of flags used 1 - Accounting 2 - Missed calls Accouting 3 - NAT SIP ping flag 4 - NAT invite detected 5 - registrar module nat_flag 6 - Enable NAT support 7 - Enable Trunk (pstn, asterisk or yate) 13 - Enable debugging -------------------------------------------------------------------- Routes 1 - Relay with NAT helper and no proxy authentication 2 - Method Register handler 3 - Method Invite handler (call routing) 4 - Trunk and PSTN gateway routing 5 - TLS routing ( not really used but still in config to keep the given message flow intact ) 6 - Relay with no NAT support and no proxy authentication 7 - User verify 10 - Relay with NAT helper and proxy authentication Reply 1 - NAT helper Fail 1 - Voicemail and message store ( stripped from all the voicemail and message storage stuff, so in fact just empty ) -------------------------------------------------------------------- */ #Global settings debug=3 log_facility=LOG_LOCAL7 fork=yes log_stderror=no mhomed=no # Multihomed - set this to no if you set a "listen" addres listen=OPENSER_IP # only want to bind to this ip port=OPENSER_PORT children=4 tcp_children=4 tcp_accept_aliases=yes alias=OPENSER_REALM # your realm alias=OPENSER_HOST # alias domains of your SRV records alias=OPENSER_IP # and for the die hard ip users. # uncomment to override config values for debugging /* debug=7 # debug level (cmd line: -ddd) fork=no port=OPENSER_PORT log_stderror=yes # (cmd line: -E) fifo="OPENSER_FIFO" */ dns=no rev_dns=no fifo="OPENSER_FIFO" fifo_mode=OPENSER_FIFO_MODE #TLS specific settings disable_tls = DISABLE_TLS #listen = tls:OPENSER_TLS_IP:OPENSER_TLS_PORT #tls_verify = 1 #tls_require_certificate = 0 #tls_method = SSLv23 #tls_certificate = "DEFAULT_TLS_CERTIFICATE" #tls_private_key = "DEFAULT_TLS_PRIVATE_KEY" #tls_ca_list = "DEFAULT_TLS_CA_LIST" mpath="OPENSER_MODULE_PATH" #path to openser modules loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "acc.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "alias_db.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "mediaproxy.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "avpops.so" loadmodule "domain.so" loadmodule "permissions.so" loadmodule "group.so" loadmodule "xlog.so" loadmodule "enum.so" loadmodule "lcr.so" fifo_db_url="mysql://OPENSER_DB_USER:OPENSER_DB_PW@OPENSER_DB_HOST/OPENSER_DB" modparam("avpops|auth_db|alias_db|permissions|uri_db|usrloc|acc|msilo|group|domain|lcr|pdt","db_url", "mysql://OPENSER_DB_USER:OPENSER_DB_PW@OPENSER_DB_HOST/OPENSER_DB") modparam("auth_db", "calculate_ha1", CALCULATE_HA1) modparam("auth_db", "password_column", "PASSWORD_COLUMN") modparam("auth_db", "password_column_2", "ha1b") modparam("auth", "secret", "OPENSER_DIGEST_PW") #nathelper, mediaproxy or both can be used #we will only be using mediaproxy modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "MEDIAPROXY_SOCKET") #modparam("mediaproxy","sip_asymmetrics","MEDIAPROXY_SIP_ASYMMETRICS") #modparam("mediaproxy","rtp_asymmetrics","MEDIAPROXY_RTP_ASYMMETRICS") # -- lcr params -- modparam("lcr","gw_table","gw") modparam("lcr","gw_name_column","gw_name") modparam("lcr","ip_addr_column","ip_addr") modparam("lcr","port_column","port") modparam("lcr","uri_scheme_column","uri_scheme") modparam("lcr","transport_column","transport") modparam("lcr","grp_id_column","grp_id") modparam("lcr","db_mode",0) # -- group params -- modparam("group", "table", "grp") modparam("group", "use_domain", 0) # might be usefull for future expansions modparam("group", "user_column", "username") modparam("group", "domain_column", "domain") modparam("group", "group_column", "grp") # -- usrloc params -- modparam("usrloc", "db_mode", 1) modparam("usrloc", "desc_time_order", 1) # -- registrar params -- modparam("registrar", "sip_natping_flag", 3) modparam("registrar", "nat_flag", 5) modparam("registrar", "desc_time_order", 1) # -- rr params -- modparam("rr", "enable_full_lr", 1) # -- tm params -- #modparam("tm", "fr_timer", 20 ) modparam("tm", "fr_inv_timer", 90 ) #modparam("tm", "wt_timer", 20 ) modparam("tm", "via1_matching", 1) modparam("tm", "fr_inv_timer_avp", "s:inv_timeout") # -- permission params -- modparam("permissions", "db_mode", 0) modparam("permissions", "trusted_table", "trusted") # -- acc params -- modparam("acc", "db_flag", 1 ) modparam("acc", "db_missed_flag", 2) modparam("acc", "report_cancels", 1) modparam("acc", "log_flag", 1) # -- alias_db params -- modparam("alias_db", "user_column", "username") modparam("alias_db", "domain_column", "domain") modparam("alias_db", "alias_user_column", "alias_username") modparam("alias_db", "alias_domain_column", "alias_domain") #--enum params -- #modparam("enum", "domain_suffix", "e164.arpa.") # -- domain params -- modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") # -- avpops params -- # avpops is used to store the local prefix from each customer in subscriber table # and append it as soon as a local call is done (meaning a call matching [^0][0-9]*) modparam("avpops","db_scheme","prefix-scheme:username_col=username;domain_col=domain;value_col=prefix;value_type=string;table=subscriber") modparam("avpops","avp_table","subscriber") modparam("avpops","use_domain",0) modparam("avpops","username_column","username") modparam("avpops","domain_column","domain") modparam("avpops","avp_aliases","prefix=s:800") # ------------------------- request routing logic ------------------- # main routing logic route { # ----------------------------------------------------------------- # Configuration flags # ----------------------------------------------------------------- NAT_SUPPORT_FLAG_6 TRUNK_ENABLED_FLAG_7 DEBUG_FLAG_13 # changing return code 182 to 180 so that the calling phone will ring # does not work at all #if (status=="182") { # xlog("Changing Status Code from 182 to 180"); # subst('/182 Queued/180 Ringing/ig'); #} #xlog("Replacing Status 1820 with 180"); #replace_all("182 Queued","180 Ringing"); if (isflagset(13) && !is_method("ACK") && !is_method("REGISTER")) { #debug route route(13); }; # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { if (method!="ACK") { sl_send_reply("483", "Too Many Hops"); }; return; }; if (msg:len > max_len) { if (method!="ACK") { sl_send_reply("513", "Message Overflow"); }; return; }; if (status=="482") { #loop detection xlog("L_ERR", "OPENSER_NAME: $mi $rm $fu -> $ru status 482 Loop Detected\n"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE") { record_route(); if (client_nat_test("3") && isflagset(6)) { add_rr_param(";nat=yes"); }; }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (isflagset(6)) { if (has_totag() && (method=="INVITE" || method=="ACK" || method=="BYE")) { setflag(1); # not really clear which of the setflags triggered the "BYE" and "CANCEL" to be accounted setflag(2); if (client_nat_test("3")||search("^Route:.*;nat=yes")){ xlog("L_NOTICE","Loose Route: Using MediaProxy"); use_media_proxy(); }; }; }; route(6); return; }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (isflagset(6)) { if (method=="BYE" || method=="CANCEL") { setflag(1); setflag(2); end_media_session(); }; }; # ----------------------------------------------------------------- # Unauthorized relay # ----------------------------------------------------------------- if (!is_uri_host_local()) { xlog("L_NOTICE","Prevent unauthorized Access is_uri_host_local=false, URI: $u, From: $fu -> Request: $ru\n"); if (is_from_local() || allow_trusted()) { # We prevent unautherised relays xlog("is_from_local or allow_trusted is true\n"); append_hf("P-hint: outbound\r\n"); route(10); # need auth } else { xlog("L_NOTICE","Uauthorized Relay: Reply a Forbidden\n"); sl_send_reply("403", "Forbidden"); }; return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri=~"sip:daemon@" ) { sl_send_reply("410", "daemon has gone for a beer"); return; }; # Calls matching means it is a local call without prefix # So prefix needs to be added expect for special numbers if (uri=~"sip:[1-9][0-9]*@" ) { # Check if uri contains special numbers ( 110,112 and some other 11[0-9]+ numbers ) # that should not be changed if (!uri=~"sip:11.*") { # add prefix to number # load user prefix and store it in $avp(s:800) # the prefix is stored in the subscriber table in an supplement attribute prefix if(!avp_db_load("$fu","$avp(s:800)/$prefix-scheme")) { xlog("Failed to load User Prefix"); } else {; # prefix R-URI xlog("Adding Prefix: $avp(s:800) to $ru"); subst_uri('/^sip:([0-9]+)@(.*)$/sip:$avp(s:800)\1@\2;orig_uri=\1-\2/i'); xlog("R-URI: $ru"); } } } if (method=="INVITE") { route(3); return; } else if (method=="ACK") { route(1); return; } else if (method=="BYE" || method=="CANCEL") { setflag(1); setflag(2); route(1); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); #alias_db_lookup("dbaliases"); if (uri!=myself) { xlog("L_NOTICE", "Lookup Aliases failed"); route(10); return; }; if (!lookup("location")) { xlog("L_NOTICE", "Lookup Location failed"); route(8); return; }; route(1); # Do not proxy authenticate - needed by system to send messages return; } route[1] { # ----------------------------------------------------------------- # Default Route relay Handler with NAT ENABLED # ----------------------------------------------------------------- if (isflagset(6)) { if (client_nat_test("3")) { setflag(4); force_rport(); fix_contact(); }; if (isflagset(4) || isflagset(5)) { xlog("L_NOTICE","Route(1) using Mediaproxy"); use_media_proxy(); }; t_on_reply("1"); }; route(6); # Relay the call return; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- # This cool register script came from http://openser.org/docs/scripting.html sl_send_reply("100", "Trying"); if (!route(7)) { # verify the user return(0); }; if (!search("^Contact:[ ]*\*") && client_nat_test("7") && isflagset(6)) { setflag(3); setflag(5); force_rport(); fix_contact(); }; if (!is_user_in("From", "noauth")) { #no authentication required if (!www_authorize("", "subscriber")) { if (isflagset(13)) { xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu Authorization for $fu ($si) requested\n"); }; www_challenge("", "0"); return(0); } else { if (isflagset(13)){ xlog("L_NOTICE","OPENSER_NAME: $mi route[$rm][2] \ $fu Authorization OK\n"); }; }; # Prohibit attempts to grab someone else's To address using valid credentials if (!check_to()) { if (isflagset(13)){ xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu -> $ru REPLY 401 - Unauthorized check_to() failed\n"); }; sl_send_reply("401", "Unauthorized - account and username must match"); return(0); }; consume_credentials(); }; if (!save("location")) { if (isflagset(13)) { xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu save(location) Failed\n"); }; sl_reply_error(); return(0); }else { if (isflagset(13)) { xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu save(location)OK\n"); }; }; return; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- setflag(1); #turn on accounting setflag(2); #also missed calls lookup("aliases"); #in the event that an alias resolves to another domain. #alias_db_lookup("dbaliases"); if (uri!=myself) { xlog("L_NOTICE","No Local Call Check Auth I: r-uri: <$ru> from-uri:<$fu>\n"); route(10); #check authentication return; }; if (!lookup("location")) { xlog("L_NOTICE","Lookup Location Failed: r-uri: <$ru> from-uri:<$fu>\n"); if (src_ip==OPENSER_PBX_IP || !isflagset(7)) { # loopback prevention route(8); return; } else { # if it's not here, go hassle asterisk. xlog("L_NOTICE","Trunk Routing: r-uri: <$ru> from-uri:<$fu>\n"); route(4); return; }; }; t_on_failure("1"); #call waiting,busy,DND,voicemail route(1); # local normal routing return; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- avp_write("$avp(i:120)", "$avp(s:inv_timeout)"); xlog("L_NOTICE","Loading Gateways form Database"); if (!load_gws()){ xlog("L_ERR","****************************** LoadGateways Failed **********************"); sl_send_reply("500", "Internal server error, unable to load phone gateways"); return; }; if (!next_gw()) { xlog("L_NOTICE","******************** Didn'T found Any Gateways *************************"); sl_send_reply("503", "Service not available - No gateways"); return; }; # check if we route through BT # if so, change From Header domain to IP ( @callway-voip.de -> @212.65.6.4 ) # BT Gateway is in GatewayGroup 1 if (to_gw("2")) { xlog("L_NOTICE","Relaying to External Gateway, so rewriting From-HEADER:$fu\n"); #subst('/182 Queued/180 Ringing/ig'); #replace_all("182 Queued","180 Ringing"); subst('/From:([^@]*)@callway-voip.de/From:\1@212.65.6.4/ig'); }; #if (isflagset(13)){ xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][4] \ time [$Tf] Next Gateway $ruri_user_avp method <$rm> r-uri <$ru> 2nd via <$hdr(via[1])>\n"); #}; route(1); return; } route[5] { # ----------------------------------------------------------------- # TLS Handling # ----------------------------------------------------------------- # if (!is_uri_host_local()) { # 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; # }; # }; if (!is_uri_host_local()) { remove_hf("Alert-Info"); if (method=="INVITE" && proto==tls) { append_hf("Alert-info: 1\r\n"); #cisco 7960 append_hf("Alert-info: Bellcore-dr4\r\n"); #cisco ATA # append_hf("Alert-info: http://foo.bar/x.wav\r\n"); #snom }; }; return; } route[6] { # ----------------------------------------------------------------- # Route relay Handler # ----------------------------------------------------------------- #subst('/182 Queued/180 Ringing/ig'); #replace_all("182 Queued","180 Ringing"); if (!t_relay()) { if (isflagset(6)) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; }; xlog("L_NOTICE","Route6: T_relay failed"); sl_reply_error(); }; return; } route[7] { # ----------------------------------------------------------------- # User verify # ----------------------------------------------------------------- # Check for known domain(s) if (!is_from_local()) { if (isflagset(13)){ xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu Unknown domain\n"); sl_send_reply("403", "Unknown domain"); }; return(0); }; # Check to see that the user has not been disabled if (is_user_in("From", "disabled")) { if (isflagset(13)){ xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $fu Your account has been disabled\n"); }; sl_send_reply("403", "Your account has been disabled"); return(0); }; if (is_user_in("To", "disabled")) { if (isflagset(13)) { xlog("L_NOTICE", "OPENSER_NAME: $mi route[$rm][2] \ $tu That account has been disabled\n"); }; sl_send_reply("403", "That account has been disabled"); return(0); }; return(1); } route[8] { # ----------------------------------------------------------------- # Handling of unavailable user # ----------------------------------------------------------------- # non-Voip -- just send "off-line" if (!(method=="INVITE" || method=="ACK" || method=="CANCEL")) { sl_send_reply("404", "Not Found"); return; }; t_newtran(); t_reply("404", "Not Found"); # we account missed incoming calls; previous statteful processing # guarantees that retransmissions are not accounted acc_db_request("404 missed call", "missed_calls"); return; } route[10] { # ----------------------------------------------------------------- # Default Message Handler with Proxy Authentication # ----------------------------------------------------------------- if(method=="ACK") { #these you never proxy authenticate route(1); return; }; if(method=="BYE" || method=="CANCEL") { #these you never proxy authenticate setflag(1); route(1); return; }; xlog("L_INFO", "Proxy auth $fd $dP destination:$du $dd $ds"); if (!route(7)) { #verify the user return(0); }; if (!is_user_in("From", "noauth")) { #no authentication required if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; # if (is_user_in("Credentials", "local")) { # Uncomment to use the group options route(1); # }else{ # sl_send_reply("403", "Denied, you are not allowed this route"); # }; return; } onreply_route[1] { # ----------------------------------------------------------------- # Return Route # ----------------------------------------------------------------- #replace_all("182 Queued","180 Ringing"); if ((isflagset(4) || isflagset(5)) && isflagset(6) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { xlog("L_NOTICE","OnReplyRoute(1) Using Mediaproxy"); use_media_proxy(); }; }; if (client_nat_test("1") && isflagset(6)) { fix_contact(); }; return; } failure_route[1] { # ----------------------------------------------------------------- # No answer, cancel, busy handling # ----------------------------------------------------------------- return; } route[13] { # ----------------------------------------------------------------- # Debuging info # ----------------------------------------------------------------- if (status=="100") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 100 Trying\n"); } else if (status=="180") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 180 Ringing\n"); } else if (status=="181") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 181 Call Is Being Forwarded\n"); } else if (status=="182") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 182 Queued\n"); } else if (status=="183") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 183 Session Progress\n"); } else if (status=="200") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 200 OK\n"); } else if (status=="202") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 202 Accepted\n"); } else if (status=="300") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 300 Multiple Choices $ct\n"); } else if (status=="301") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 301 Moved Permanently $ct\n"); } else if (status=="302") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 302 Moved Temporarily $ct\n"); } else if (status=="305") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 305 Use Proxy\n"); } else if (status=="380") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 380 Alternative Service\n"); } else if (status=="400") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 400 Bad Request\n"); } else if (status=="401") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 401 Unauthorized\n"); } else if (status=="402") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 402 Payment Required\n"); } else if (status=="403") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 403 Forbidden\n"); } else if (status=="404") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 404 Not Found\n"); } else if (status=="405") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 405 Method Not Allowed\n"); } else if (status=="406") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 406 Not Acceptable\n"); } else if (status=="407") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 407 Proxy Authentication Required\n"); } else if (status=="408") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 408 Request Timeout\n"); } else if (status=="409") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 409 Conflict\n"); } else if (status=="410") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 410 Gone\n"); } else if (status=="411") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 411 Length Required\n"); } else if (status=="412") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 412 Conditional Request Failed\n"); } else if (status=="413") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 413 Request Entity Too Large\n"); } else if (status=="414") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 414 Request-URI Too Long\n"); } else if (status=="415") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 415 Unsupported Media Type\n"); } else if (status=="416") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 416 Unsupported URI Scheme\n"); } else if (status=="420") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 420 Bad Extension\n"); } else if (status=="421") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 421 Extension Required\n"); } else if (status=="422") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 422 Session Interval Too Small\n"); } else if (status=="423") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 423 Interval Too Brief\n"); } else if (status=="429") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 429 Provide Referrer Identity\n"); } else if (status=="480") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 480 Temporarily Unavailable\n"); } else if (status=="481") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 481 Call Leg/Transaction Does Not Exist\n"); } else if (status=="482") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 482 Loop Detected\n"); } else if (status=="483") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 483 Too Many Hops\n"); } else if (status=="484") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 484 Address Incomplete\n"); } else if (status=="485") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 485 Ambiguous\n"); } else if (status=="486") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 486 Busy Here\n"); } else if (status=="487") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 487 Busy Here; Request Terminated\n"); } else if (status=="488") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 488 Not Acceptable Here\n"); } else if (status=="489") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 489 Bad Event\n"); } else if (status=="491") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 491 Request Pending\n"); } else if (status=="493") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 493 Undecipherable\n"); } else if (status=="494") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 494 Security Agreement Required\n"); } else if (status=="500") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 500 Server Internal Error\n"); } else if (status=="501") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 501 Not Implemented\n"); } else if (status=="502") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 502 Bad Gateway\n"); } else if (status=="503") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 503 Service Unavailable\n"); } else if (status=="504") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 504 Server Timeout\n"); } else if (status=="505") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 505 Version Not Supported\n"); } else if (status=="513") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 513 Message Too Large\n"); } else if (status=="580") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 580 Precondition Failure\n"); } else if (status=="600") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 600 Busy Everywhere\n"); } else if (status=="603") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 603 Decline\n"); } else if (status=="604") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 604 Does Not Exist Anywhere\n"); } else if (status=="606") { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status 606 Not Acceptable\n"); } else { xlog("L_NOTICE", "OPENSER_NAME: $mi $rm $fu -> $ru status not listed\n"); }; xlog("L_NOTICE", "OPENSER_NAME: $mi $mb\n"); return; }

2 1

[Serusers] jabber2foreign IM
by Allan Chao ( 策略性新技術開發處申請 ) 05 Feb '07

05 Feb '07

Dear All: I am a ser user, i want to use sip client that can commuicate with foreign IM client(msn), my flow is sip client -> ser(jabber module) (0.10 version) -> jabber server(has msn gateway) -> MSN network. but i did not see any function that can register to MSN server(just like jabber client browse transport in the jabber server and then register to MSN network) and there is no function or program using msn id and password. i doubt jabber module does not implement this part of program.Maybe there is another mathod can solve it. Does anyone can help me ? thanks. allan DISCLAIMER: Sample Disclaimer added in a VBScript.

2 1

[Users] Exec - environment variables - not found?
by Kjell Arild Frigstad 04 Feb '07

04 Feb '07

Hello! I have just upgraded from 1.0.0 til 1.1.1, and I having problem with exec_msg (the exec_msg working, but environment variabels is "not found".) It was OK in 1.0.0 version. The coding: exec_msg("echo SIP_SRCIP $SIP_SRCIP >>/var/log/messages"); At startup (with debuggging) : 0(0) fixing exec_msg, line 160 0(0) xl_parse_format: parsing [echo SIP_SRCIP $SIP_SRCIP >>/var/log/messages] 0(0) xl_lookup_spec_name: not found [SIP_SRCIP] 0(0) xl_parse_format: format parsed OK: [2] items Error message when running the command (with debugging) : 0(28888) xl_get_spec_value: error - null sp->itf 0(28888) exec:w_exec_msg: executing [echo SIP_SRCIP >>/var/log/messages] Error message when running in "normal" mode Feb 4 21:51:24 test /usr/local/sbin/openser[28947]: xl_get_spec_value: error - null sp->itf SIP_SRCIP As you see the text SIP_SRCIP is beeing displayd, but not the variabel. Any ide what can bee wrong? Best regards K.A.Frigstad

2 1

[Users] auth_radius extra attributes
by Cseke Tamas 03 Feb '07

03 Feb '07

Hello, can anybody tell me how can i send extra attributes to radius in authpacket, in acctpacket i can with radius_extra parameter, is there a way to use it in auth too? Thanks any help, Tamas

1 0

[Serusers] Problem with mysql.so on new server
by sip 03 Feb '07

03 Feb '07

So... I've had ser 0.9.6 running on RH EL3, and FC3 and 4. I'm building out a server on RH EL4 for SER, but I've run into a snag. I'm sure it's something simple, but it has me stumped at the moment. I've recompiled mysql.so on the new server... compiles fine (as do the rest of the modules). However, when I try to start up SER with mysql.so loaded, it gives me this: ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: undefined symbol: log Undefined symbol: log Never seen this before. Anyone know what it means or, more importantly, how to fix it (and no... I'm not going to scrap it all and go with Ottendorf yet, so that's not a solution ;) )? N.

4 12

[Serusers] Which locking mechanism is useful for usrloc in production environment?
by Ramu Yadav 03 Feb '07

03 Feb '07

Hello, At present we are using "sys v semaphores"(USE_SYSV_SEM) for locking. But we see a some delay while acquiring the lock to modify the usrloc's location table. We are in a thought to use "posix semaphores" (USE_POSIX_SEM), will this improve any performance for usrloc location table. Thoughts and Ideas will be appreciated. -- Ramu Yadav

4 5

[Users] NEW FEATURE: better timers
by Bogdan-Andrei Iancu 02 Feb '07

02 Feb '07

Hi everybody, following to some discussions / requests from users, openser devel branch has now core support for timers will resolution less than a second. It uses time functions that can give a microsecond accuracy, but for performance reasons, the utimer (microtimer) uses a ticking period of 100 milliseconds. Of course, this can be changed at compile time. First "beneficiary" of this timer, is TM, to gain more accuracy for retransmissions: * retransmission timers use the the core utimer (with microsecond resolution), having 100 milliseconds intervals. * this will increase the retransmissions precision to an error of maximum 100 milliseconds. * T1 (first retransmission) is done at 500 milliseconds (instead of 1 sec), as the RFC3261 suggests. * T1 and T2 (see RFC3261) can be configured via module parameters - the inter-steps are automatically computed on startup based on the RFC3261 specifications. Along with increasing the timer accuracy, the TM timer lists were improved for faster operations: * faster timers lists - use shortcut pointers (based on timeout values) for faster walking through the list -> faster insert and expire * for elements to be deleted from the timer lists, do not overwrite the timeout value as this will trigger more processing when inserting new elements into the list The performance tests are not completed - I need more machines to stress the proxy :D -, but I will come back with results during the testing phase. regards, bogdan

1 0

[Users] removed avp_write(); avp_aliases moved in core
by Daniel-Constantin Mierla 02 Feb '07

02 Feb '07

Hello, two major changes were applied to avpops module in development branch (1.2.0): - avp_write() has been removed - same functionality is provided by assignment operator in routing blocks, together with latest commit of PV transformations - avp_aliases parameter has been removed from avpops module and introduced as core parameter. you can use it directly in core, in global parameters section in form of: avp_aliases="uuid=I:660;email=s:email_addr;fwd=i:753" The value is the same as it was for avpops parameter. Now the AVP aliases can be used directly in routing blocks. Cheers, Daniel

1 0

[Users] new feature: pseudo-variable transformations
by Daniel-Constantin Mierla 02 Feb '07

02 Feb '07

Hello, currently in CVS trunk of development version is a new feature 'transformations'. A transformations is basically a function that can be applied to a pseudo-variable (PV). In this way, the value of PV is not altered but the result is something different. They are intended to facilitate access to different attributes of PV (like strlen of value, parts of value, substrings) or complete different value of PV (encoded in hexa, md5 value, escape/unescape PV value for DB operations...). The format is '{id[,parameters]}' and they can be used inside parenthesis of pseudo-variables (note that even short name pseudo-variables must be enclosed in '(' and ')' to be able to apply transformations) For example: - $(ru{s.len}) - returns the lenth of R-URI - $(avp(i:3){s.md5}) - returns the md5 string over value of $avp(i:3) The following classes of transformations are implemented 1) string - the name of transformation starts with 's.'. Available transformations in this class: - {s.len} - returns strlen of PV value - {s.md5} - returs md5 over PV value - {s.substr,offset,length} - returns substring starting at offset having size of 'length'. If offset is negative, then it is counted from the end of PV value, -1 being the last char. In case of positive value, 0 is first char. Length must be positive, in case of 0, substring to the end of PV value is returned. offset and length can be PV as well. Example: "abcd"{s.substr,1,0} = "bcd" - {s.select,index,separator} - returns a field from PV value. The field is selected based on separator and index. The separator must be a character used to identify the fields. Index must be a integer value or a PV. If index is negative, the count of fields starts from end of PV value, -1 being last field. If index is positive, 0 is the first field. Example: "12,34,56"{s.select,1,,} = "34" ; "12,34,56"{s.select,-2,,} = "12" - {s.encode.hexa} - returns encoding in hexa of PV value - {s.decode.hexa} - returns decoding from hexa of PV value - {s.escape.common} - returns escaped string of PV value. Characters escaped are ''', '"', '\' and 0. Useful when doing DB queries (care should be taken for non latin character set) - {s.unescape.common} - returns unescaped string of PV value. Reverse of above transformation. 2) uri - the name of transformation starts with 'uri.'. The PV value is considered to be a SIP URI. This transformation returns parts of SIP URI (see struct sip_uri). Available transformations in this class: - {uri.user} - returns the user part - {uri.host} or {uri.domain} - returns the domain part - {uri.passwd} - returns the password - {uri.port} - returns the port - {uri.params} - returns the URI parameters in a string - {uri.param,name} - returns the value of parameter with name 'name' - {uri.headers} - returns URI headers - {uri.transport} - returns the value of transport parameter - {uri.ttl} - returns the value of ttl parameter - {uri.uparam} - returns the value of user parameter - {uri.maddr} - returns the value of maddr parameter - {uri.method} - returns the value of method parameter - {uri.lr} - returns the value of lr parameter - {uri.r2} - returns the value of r2 parameter 3) parameters list - the name of the transformation starts with 'param.'. The PV value is considered to be a string like "name1=value1;name2=value2;...". The transformations returns the value for a specific parameter, or the name of a parameter at a specific index. Available transformations in this class: - {param.value,name} - returns the value of parameter 'name' Example: "a=1;b=2;c=3"{param.value,c} = "3" - {param.name,index} - returns the name of parameter at position 'index'. Example: "a=1;b=2;c=3"{param.name,1} = "b" Within a PV, many transformation can be applied, being executed from left to right. Example: $var(x) = "a=1;b=22;c=333"; $(var(x){param.value,$(var(x){param.name,1})}{s.len}) = 2 - the length of the value of parameter at position 1 (remember 0 is first position, 1 is second position) The transformations can be used anywhere, being considered parts of PV -- in xlog, avpops or other modules' functions and parameters, in right side assignment expressions or in comparisons. Documentation page for transformations was opened at: http://openser.org/dokuwiki/doku.php/transformations:devel Feedback is welcome! Cheers, Daniel

1 0

[Serusers] meeting (was: Re: [Serdev] SER's core design features(process model/parser/lumps/script) - was: So who/what is SER for, anyway?
by Jiri Kuthan 02 Feb '07

02 Feb '07

We could use the IETF meeting in Prague in March. Actually, I have anticipated that and there is a hotel room reserved already :-) Could folks willing to show up and debate roadmap or whatever else appears important technically speak up now? -jiri At 12:08 25/01/2007, Olle E Johansson wrote: >As I am fighting the same issues and try to make chan_sip easier for >the administrator, would it be >a good idea with a joint Open Source SIP developer meeting somewhere >in Europe this spring? > >We have a lot of common issues, need to settle with the same >terminology and propably fight >the same fights with RFCs to implement... > >A lot of of users use both Asterisk and SER/OPENSER, so they need >interoperability. > >For my work on a new chan_sip, please check http://www.codename- pineapple.org > >/O >_______________________________________________ >Serdev mailing list >Serdev(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serdev -- Jiri Kuthan http://iptel.org/~jiri/

7 9

[Serusers] www_challenge() error in auth module
by raj gopal 02 Feb '07

02 Feb '07

Hi, I can not use www_challenge(realm,qop)function in ser.cfg file even though auth module is loaded. The readme file has documentation for it. When I use the following in config file and start SER, the following error is thrown if (!radius_www_authorize("")) { www_challenge("", "1"); break; }; parse error (197,49-50): unknown command, missing loadmodule? Can any one help me to find out whats wrong with this module/function. -Venkat

1 0

[Serusers] automated response
by Patrick Timmons 02 Feb '07

02 Feb '07

IMPORTANT NOTE: if you are receiving this auto responder after accessing Acme Packet's web portal or sending an email to a mailing list, rest assured that your message has been received and will be acted upon by the appropriate personnel. I am taking Friday, February 2 through Tuesday, February 6 off and will be mostly unavailable during this time. I will not have access to a computer, but will be reachable via my cell phone in the event of emergencies. For support issues, please contact support(a)acmepacket.com. For non-support issues, please contact Erin Medeiros, Vice President of Professional Services, at emedeiros(a)acmepacket.com. Thank you, Patrick Timmons Vice President, Systems Engineering Acme Packet, Inc.

1 0

[Serusers] ser 0.10.99 - call forwarding issue
by tzieleniewski 01 Feb '07

01 Feb '07

hi! I am trying to set up the call forwarding feature and I have some question bout the idea how to realize this. My Idea was to do it with the usage of avp's. First thing I do is the check if there is a call_forward parameter set in the database in user_attrs table. I do this by invoking load_attrs("$fu", "$f.uid"); and then I do the check if ($tu.call_forward) On the "what's new site" in the avps and selects article I found another method which performs similar task: lookup_user("$tu","@ruri") What is the difference between those to methods?? Are they part of the SER core? previously in ser 0.9.x I have it done by, first loading the avp by avp_db_load() and then in case of call forwarding by making avp_pushto("$ruri","something") or rewriteuri() and then t_relay(). What is now a preferred way to realized similar functions. I mean can I utilize those methods like load_attrs or lookup_user() and then substitute the ruri without avp_pushto or rewriteuri?? best tomasz

2 5

Re: [Users] Acc'ing + Flatstore, Log Format?
by Bogdan-Andrei Iancu 01 Feb '07

01 Feb '07

Brandon, the log_fmt parameter affects only the syslog accounting (logging via syslog to log files). Flatstore is implemented as Write-Only DB engine, so it be used by the DB part of the accounting module. You need to play with the "db_*" related parameters from the ACC module. regards, bogdan Brandon Armstead wrote: > Bogdan-Andrei Iancu wrote: > >> Hi Brandon, >> >> what version of OpenSER are you using? There were important changes >> in the latest version, so it is important to know it. >> >> regards, >> bogdan >> >> Brandon Armstead wrote: >> >>> Hello everyone, >>> >>> Just a quick question, I have the following in my configuration: >>> modparam("acc", "log_fmt", "miocrdfs01"), however, in the actual >>> logging it is giving me back 14 different log "information", yet the >>> format "miocrdfs01" is only 10, does flatstore use a different >>> formatting mechanism, or does it force it to be that 14, and only >>> that 14, or anyone know whats going on? Thanks >>> >>> Regards, >>> Brandon >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >> >> > Hello Bodgan, > > I'm currently using OpenSER version 1.1.1-notls. Thanks again! > > Regards, > Brandon >

1 0

[Serusers] src_ip=="%$foo"
by Atle Samuelsen 01 Feb '07

01 Feb '07

Hi guys and girls, is it possible to do a if(src_ip=="%$foo") where $foo contains more then one IP address? - Atle

1 0

[Serusers] Presence client error
by Jens-Martin Groenne 01 Feb '07

01 Feb '07

I'm testing the presence module on 0.10.99-dev in CVS. When using the newest X-lite client I am getting the following error messages in /var/log/messages /usr/local/sbin/ser[30891]: INFO: rls_mod.c:155: reading RLS data from database /usr/local/sbin/ser[30879]: INFO: subscribe.c:286: Unsupported event package /usr/local/sbin/ser[30879]: ERROR: subscribe.c:590: Error while checking message /usr/local/sbin/ser[30879]: INFO: subscribe.c:603: handle_subscription about to send_reply and return -2 /usr/local/sbin/ser[30885]: Warning: run_failure_handlers: no UAC support (0, 256) /usr/local/sbin/ser[30885]: WARN: rl_subscription.c:398: destroying subscription from callback due to 420 response on NOTIFY /usr/local/sbin/ser[30885]: TRACE: rl_subscription.c:400: RLS: subscription destroyed!!! In X-lite I get this error message: [07-01-30]15:44:45.526 | Debug | RESIP:TRANSACTION | "Match!" | [07-01-30]15:44:45.526 | Debug | RESIP:TRANSACTION | "Send to TU: TU: DialogUsageManager size=0 NOTIFY sip:tuser@192.168.1.35:34492 SIP/2.0 Via: SIP/2.0/UDP 85.112.135.70;branch=z9hG4bK7ed6.d015cec1.0 Max-Forwards: 70 Require: eventlist Contact: <sip:85.112.135.70:5060> To: ""Jens""<sip:tuser@test-domain.com>;tag=d204727a From: <sip:tuser@test-domain.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-1a1f Call-ID: OTYwMDE1NmQ5MDQwYjFmMTBlZjQ0NTNhZTExNWM4MGM. CSeq: 10 NOTIFY Content-Type: multipart/related;type=""application/rlmi+xml"";start=""qwW1099041416pPdxX1162343142"";boundary=""RewXdpxR741013657xA447338541""; User-Agent: Sip EXpress router(0.10.99-dev66 (i386/linux)) Subscription-State: active;expires=300 Event: presence Content-Length: 338 --RewXdpxR741013657xA447338541 Content-Transfer-Encoding: binary Content-ID: qwW1099041416pPdxX1162343142 Content-Type: application/rlmi+xml;charset=""UTF-8"" <?xml version=""1.0""?> <list xmlns=""urn:ietf:params:xml:ns:rlmi"" uri=""sip:tuser@test-domain.com"" version=""0"" fullState=""true""> </list> --RewXdpxR741013657xA447338541-- " | [07-01-30]15:44:45.526 | Debug | RESIP | "SipMessage::getContents: multipart/related" | [07-01-30]15:44:45.526 | Debug | RESIP:DUM | "Decrypted message:02815028 " | [07-01-30]15:44:45.526 | Info | RESIP:DUM | "Got: SipReq: NOTIFY tuser@192.168.1.35:34492 tid=7ed6.d015cec1.0 cseq=NOTIFY contact=85.112.135.70:5060 / 10 from(wire)" | [07-01-30]15:44:45.526 | Info | RESIP:DUM | "Received an unsupported option tag(s): SipReq: NOTIFY tuser@192.168.1.35:34492 tid=7ed6.d015cec1.0 cseq=NOTIFY contact=85.112.135.70:5060 / 10 from(wire)" | [07-01-30]15:44:45.526 | Debug | RESIP | "Helper::makeResponse(SipReq: NOTIFY tuser@192.168.1.35:34492 tid=7ed6.d015cec1.0 cseq=NOTIFY contact=85.112.135.70:5060 / 10 from(wire) code=420 reason=" | [07-01-30]15:44:45.526 | Debug | RESIP | "SEND: SipResp: 420 tid=7ed6.d015cec1.0 cseq=NOTIFY / 10 from(tu)" | [07-01-30]15:44:45.527 | Debug | RESIP:DUM | "Failed required options validation NOTIFY sip:tuser@192.168.1.35:34492 SIP/2.0 Via: SIP/2.0/UDP 85.112.135.70;branch=z9hG4bK7ed6.d015cec1.0 Max-Forwards: 70 Require: eventlist Contact: <sip:85.112.135.70:5060> To: ""Jens""<sip:tuser@test-domain.com>;tag=d204727a From: <sip:tuser@test-domain.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-1a1f Call-ID: OTYwMDE1NmQ5MDQwYjFmMTBlZjQ0NTNhZTExNWM4MGM. CSeq: 10 NOTIFY Content-Type: multipart/related;type=""application/rlmi+xml"";boundary=""RewXdpxR741013657xA447338541"";start=""qwW1099041416pPdxX1162343142"" User-Agent: Sip EXpress router(0.10.99-dev66 (i386/linux)) Subscription-State: active;expires=300 Event: presence Content-Length: 334 --RewXdpxR741013657xA447338541 Content-Type: application/rlmi+xml;charset=""UTF-8"" Content-Transfer-Encoding: binary Content-Id: qwW1099041416pPdxX1162343142 <?xml version=""1.0""?> <list xmlns=""urn:ietf:params:xml:ns:rlmi"" uri=""sip:tuser@test-domain.com"" version=""0"" fullState=""true""> </list> - Does any know know how to solve this problem? Is there any other clients that works well with the ser presence module? Kind regards, Jens-Martin Groenne

2 5

[Fwd: Re: [Users] Acc'ing + Flatstore, Log Format?]
by Brandon Armstead 01 Feb '07

01 Feb '07

1 0

[Users] Acc'ing + Flatstore, Log Format?
by Brandon Armstead 01 Feb '07

01 Feb '07

Hello everyone, Just a quick question, I have the following in my configuration: modparam("acc", "log_fmt", "miocrdfs01"), however, in the actual logging it is giving me back 14 different log "information", yet the format "miocrdfs01" is only 10, does flatstore use a different formatting mechanism, or does it force it to be that 14, and only that 14, or anyone know whats going on? Thanks Regards, Brandon

2 1

[Serusers] Reconnect to Mysql
by Shaun Hofer 01 Feb '07

01 Feb '07

Hi When ever MySQL Server (5.0.24a) on a remote machine is restarted, SER can't reconnect to the MySQL server. I see this problem has occurred before via mailing list: ERROR message : MySQL server has gone away (http://lists.iptel.org/pipermail/serusers/2006-April/027961.html) I am running debian package of SER 0.9.6-2 (ser-mysql-module 0.9.6-2) and MySQL Client 5.0.32 . I have tested that the MySQL client reconnects after server has been restarted. Is there something I'm missing that I'm still getting this problem ? Thanks --Shaun

1 1

[Serusers] Problem with sl_send_reply: I won't send a reply for ACK.
by Ricardo Martinez 31 Jan '07

31 Jan '07

Hello list. I'm facing a problem with a UAC and i was hoping that someone can give me a hand here. I have a IP Phone calling to a PSTN number through SER and then a GW. 10.0.0.243 : IP Phone 10.0.0.246 : SER SIP Proxy 10.0.0.239 : GW SIP PSTN When the call is established and the "200 - OK" message arrives from the GW to the Proxy, the proxy re-route the message back to the Client, and finally the client respond with an ACK. Here is when te problem begins, i'm not sure if the ACK is the problem or maybe is a bug with my SER box. I'm using the Getting Started ser.cfg from iptel.org. You can see the debug here: U 10.0.0.239:5060 -> 10.0.0.246:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.0.0.246;branch=z9hG4bK34e7.0648f244.0. Via: SIP/2.0/UDP 10.0.0.243:5060;rport=5060;branch=z9hG4bK407006395. From: <sip:5501234567@sipvoiss.desa.mydomain.net>;tag=139103625. To: <sip:0101005622408196@sipvoiss.desa.mydomain.net>;tag=d745f073a4. Call-ID: 90212623(a)10.0.0.243. CSeq: 21 INVITE. Supported: timer, replaces, early-session. User-Agent: A SIP Gateway. Contact: sip:005622408196@10.0.0.239. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO. Content-Type: application/sdp. Content-Length: 247. Record-Route: <sip:10.0.0.246;ftag=139103625;lr=on>. . v=0. o=005622408196 1170173661 1170173661 IN IP4 10.0.0.239. s=A Gateway SDP. c=IN IP4 10.0.0.239. t=1170173661 0. m=audio 23614 RTP/AVP 18 101. a=rtpmap:18 G729/8000/1. a=rtpmap:101 telephone-event/8000/1. a=fmtp:101 0-15. a=ptime:20. U 10.0.0.246:5060 -> 10.0.0.243:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.0.0.243:5060;rport=5060;branch=z9hG4bK407006395. From: <sip:5501234567@sipvoiss.desa.mydomain.net>;tag=139103625. To: <sip:0101005622408196@sipvoiss.desa.mydomain.net>;tag=d745f073a4. Call-ID: 90212623(a)10.0.0.243. CSeq: 21 INVITE. Supported: timer, replaces, early-session. User-Agent: A SIP Gateway. Contact: sip:005622408196@10.0.0.239. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO. Content-Type: application/sdp. Content-Length: 247. Record-Route: <sip:10.0.0.246;ftag=139103625;lr=on>. . v=0. o=005622408196 1170173661 1170173661 IN IP4 10.0.0.239. s=A Gateway SDP. c=IN IP4 10.0.0.239. t=1170173661 0. m=audio 23614 RTP/AVP 18 101. a=rtpmap:18 G729/8000/1. a=rtpmap:101 telephone-event/8000/1. a=fmtp:101 0-15. a=ptime:20. U 10.0.0.243:5060 -> 10.0.0.246:5060 ACK sip:0101005622408196@sipvoiss.desa.mydomain.net SIP/2.0. Via: SIP/2.0/UDP 10.0.0.243:5060;rport;branch=z9hG4bK1227697472. Route: <sip:10.0.0.246;ftag=139103625;lr=on>. From: <sip:5501234567@sipvoiss.desa.mydomain.net>;tag=139103625. To: <sip:0101005622408196@sipvoiss.desa.mydomain.net>;tag=d745f073a4. Call-ID: 90212623(a)10.0.0.243. CSeq: 21 ACK. Contact: <sip:5501234567@10.0.0.243:5060>. Max-Forwards: 70. User-Agent: S SIP User Agent / 1.10. Content-Length: 0. . Is this ACK ok?. The ACK hits the "Sanity Checks" (Max Forwards) and then breaks sending to console : "Warning: sl_send_reply: I won't send a reply for ACK!!" I was reading the RFC3261 because i'm not sure about the R-URI from this endpoint. Let me explain : RFC3261 : Section 12.1.2 The route set MUST be set to the list of URIs in the Record-Route header field from the response, taken in reverse order and preserving all URI parameters. If no Record-Route header field is present in the response, the route set MUST be set to the empty set. This route set, even if empty, overrides any pre-existing route set for future requests in this dialog. The remote target MUST be set to the URI from the Contact header field of the response. So, if this is correct the R-URI from the ACk must be set to sip:005622408196@10.0.0.239, and not the sip:0101005622408196@sipvoiss.desa.mydomain.net. Is this ok?. I think this is causing the ACK problem in my SER box. Can someone help me here? Thanks in advance. Best Regards, Ricardo Martinez.-

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[Serusers] a problem about radius and digest - openser patch
by TZieleniewski 31 Jan '07

31 Jan '07

_______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Users] ERROR: module version mismatch for /usr/local/lib/openser/modules/mysql.so; core: openser 1.1.1-tls (i386/freebsd); module: openser 1.1.1-notls (i386/freebsd)
by Hansi 31 Jan '07

31 Jan '07

Hello All, I have installed openser-1.1.1 with tls on my box. However, when I try to run openser. It returns me this error: ERROR: module version mismatch for /usr/local/lib/openser/modules/mysql.so; core: openser 1.1.1-tls (i386/freebsd); module: openser 1.1.1-notls (i386/freebsd). Would someone be kind enough to help me how to compile a mysql.so which supports tls? Thank you. "I'm a great believer in luck and I find the harder I work, the more I have of it." --------------------------------- No need to miss a message. Get email on-the-go with Yahoo! Mail for Mobile. Get started.

1 0

[Serusers] Multiple Gateway Help
by Giovanni Chirico 30 Jan '07

30 Jan '07

Hi, How I can send calls to multiple Gateway? I would like to use the "dispatcher module". I made my configuration with dispatcher module, the calls are redirect to the gateway defined in the dispatcher.list file, but when I hung up the call the client always ring. (No BYE message). In attachment my ser.cfg file. Thanks in advance, John

1 0

[Serusers] Getting 513 - Message too big for REGISTER message
by Sudhakar Patil 30 Jan '07

30 Jan '07

Hi, I am running into the attached errors. Appreciate your help. To add further clarification, looks like if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); exit; }; max_len is getting converted to 0 (zero) and hence I keep getting 513 error message. I replaced max_len with numeric values but the effect is the same in that when comp_num in route.c is called the right value is always 0 (zero). Regards, Sudhakar. ____________________________________________________________________________________ Expecting? Get great news right away with email Auto-Check. Try the Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/newmail_tools.html

2 1

[Users] approaching testing phase for 1.2.0
by Daniel-Constantin Mierla 30 Jan '07

30 Jan '07

Hello everybody, we are approaching the testing phase for release 1.2.0. This week is planned to be the latest before cvs freezing. Then we will take care only of bug fixing, documentation update and small tunings. By this week we should be able to fix pending issues like script variables, dns failover and black lists, timers -- most of the code being already available on public cvs. If you have something else in mind, please send a mail to devel mailing list. Cheers, Daniel

2 3

[Serusers] does ser support presence?
by Su Kerry 30 Jan '07

30 Jan '07

2 1

[Serusers] Re: a problem about radius and digest
by TZieleniewski 29 Jan '07

29 Jan '07

1 0

[Serusers] nathelper / rtpproxy
by Nhadie 29 Jan '07

29 Jan '07

Hi All, Can i use nathelper/rtpproxy on extension to extension and not use when an extension calls PSTN? Please see config if it makes sense. Thank You if (nat_uac_test("3")) { if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewritingn"); fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; if ((uri=~"^(sip:)?00[0-9]*@([a-z]+.)?mydomain.com") || (uri=~"^(sip:)?00[0-9]*@2.2.2.2")) { if (is_user_in("From", "onstun")) { if (isflagset(8)) {xlog("L_INFO", "THIS GROUP SHOULD NOT USE RTPPROXYn");}; # Strip first to digits (00) strip(2); prefix("37333#"); rewritehostport("10.10.10.10:5060"); consume_credentials(); t_relay_to_udp("10.10.10.10","5060"); break; } else { if (isflagset(8)) {xlog("L_INFO", "THIS WILL USE RTPPROXYn");}; if (isflagset(6)) route(1); <----- WILL CHECK SET FLAG 6 so that it will USE NATHELPER/RTPPROXY WILL THIS WORK? strip(2); prefix("37333#"); rewritehostport("10.10.10.10:5060"); consume_credentials(); t_relay_to_udp("10.10.10.10","5060"); break; } route[1] { xlog("L_INFO", "User-Agent behind NATn"); force_rtp_proxy(); if (method=="INVITE") { t_on_reply("1"); }; append_hf("P-Behind-NAT: Yesrn"); break; } ________________________________________________ Message sent using UebiMiau 2.7

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[Serusers] Ser 0.9.6 + RADIUS
by Daniel Corbe 29 Jan '07

29 Jan '07

I cannot get my SER to talk to my RADIUS server, its just blindly 401ing things without ever making contact with RADIUS. SER config looks like this if (method == "REGISTER") { if (!radius_www_authorize("")) { www_challenge("", "1"); break; }; save("location"); }; radiusclient-ng servers file: 192.168.1.103 heslo radiusclient.conf: auth_order radius,local login_tries 4 login_timeout 60 authserver 192.168.1.103:1812 acctserver 192.168.1.103:1813 dictionary /usr/local/etc/radiusclient-ng/dictionary FreeRADIUS clients.conf: client 192.168.1.109 { secret = heslo shortname = proxy1 nastype = other Not sure what to do!

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Res: [Users] 407 and algorithm=MD5
by Bruno Machado 29 Jan '07

29 Jan '07

Hi Bogdan The section 3.2.1 (The WWW-Authenticate Response Header) from RFC 2617 says: "algorithm A string indicating a pair of algorithms used to produce the digest and a checksum. If this is not present it is assumed to be "MD5". " I send a INVITE to Openser but it send back a 407 (Proxy Authentication). The parameter response was OK. But, without the parameter 'algorithm', the proxy didn't allow the authentication. It neither consult at Radius. Looking at the RFC above, the SIP proxy must to use MD5 by default, but it don't accept the message without the parameter. Can it be a bug of OpenSER? Or my interpretation is wrong? (I bet that I'm wrong :) ) Best regards. Bruno Machado ----- Mensagem original ---- De: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> Para: Bruno Machado <bsmufu(a)yahoo.com.br> Cc: users(a)openser.org Enviadas: Segunda-feira, 29 de Janeiro de 2007 14:21:31 Assunto: Re: [Users] 407 and algorithm=MD5 Hi Bruno, you cannot configure to have the algorithm param added. If I'm not wrong, the digest RFC says this is an optional param and the default value is MD5. regards, bogdan Bruno Machado wrote: > Hi. > > When I use the function proxy_challenge(), the 407 don't have the > parameter 'algorithm=MD5'. Look below: > > ... > Proxy-Authenticate: Digest realm="proxy.com.br", > nonce="45be00f941f137ff037436c77f80a9531a02155f". > ... > > How can I put this parameter in digest informations? > > ... > Proxy-Authenticate: Digest realm="proxy.com.br", > nonce="45be00f941f137ff037436c77f80a9531a02155f", > algorithm="MD5". > ... > > > Thanks in advance. > > Bruno > > __________________________________________________ > Fale com seus amigos de graça com o novo Yahoo! Messenger > http://br.messenger.yahoo.com/ > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > __________________________________________________ Fale com seus amigos de graça com o novo Yahoo! Messenger http://br.messenger.yahoo.com/

1 0

[Users] new features: script variables and pseudo-variables in script
by Daniel-Constantin Mierla 29 Jan '07

29 Jan '07

Hello everybody, the development version allows now ability to use pseudo-variables directly in configuration file. In addition to old pseudo-variables you know, a new type was introduced: script variables - the format is $var(name) where name can be any string of digits and characters. Basically this are variables that can have integer or string values and they can be set directly in the configuration file via 'assign operation. The big advantage is that they are directly referenced at runtime, avoiding searching each access time. They are specific per processes and survive over multiple sip message processing (be sure you set the value before testing it -- e.g., $var(counter) = $var(counter) + 1 at the beginning of main route is a counter of messages processed). The script variables are initialized at startup to '0'. You can shift between value types: e.g., '$var(a)=1; $var(a)="openser";' is valid. The switch statement can be used now with any valid pseudo-variable ('retcode' must be now replaced with '$retcode' to get old behavior). switch($rU) will compare the r-uri username with 'case' values. The case 'values' can be string or integer. switch($rU) { case "101": xlog("dialed number is 101\n"); break; case "102": xlog("dialed number is 102\n"); break; default: xlog("dialed number is $rU\n"); } Assignments of script variables, AVPs, R-URI parts (username, domain, and URI) and dst URI can be done as well in script. $avp(i:11) = 1; - sets the value of $avp(i:11) to integer '1' $avp(i:11) = "1"; - sets the value of $avp(i:11) to string '1' $ru = "sip:"+$var(user)+"@"+$fd; - sets the R-URI to string composed from value of variable $var(user) and From URI domain '+' can be used to concatenate strings or do arithmetic addition. Other arithmetic operators: - '-' - minus - '/' - divide - '*' - multiply - '%' - modulo - '|' - bitwise OR - '&' - bitwise AND - '^' - bitwise XOR - '~' - bitwise NOT To make sure that the priority of the operators is right, do use parenthesis, it is the best to get expected result: $var(a) = 2 + (7&(~4)); If you want to evaluate arithmetic operations and test the value in conditions, use the test operator ' [ ... ] ' . This is required to convert the result to internal convention for drop, true and false. if ( [ $var(a)&4 ] ) { xlog("bit 3 is set in var a"); } Old '*' used to match any port or proto was changed to 'any' to avoid the conflict with multiply operator. Some operations overlap with several avpops functions. Until sure that all cases are covered in script variant, both will go further, avpops version being scheduled for deprecation. It was a big lump of changes, help in testing and feedback is very much appreciated. Cheers, Daniel

2 1

[Users] 407 and algorithm=MD5
by Bruno Machado 29 Jan '07

29 Jan '07

Hi. When I use the function proxy_challenge(), the 407 don't have the parameter 'algorithm=MD5'. Look below: ... Proxy-Authenticate: Digest realm="proxy.com.br", nonce="45be00f941f137ff037436c77f80a9531a02155f". ... How can I put this parameter in digest informations? ... Proxy-Authenticate: Digest realm="proxy.com.br", nonce="45be00f941f137ff037436c77f80a9531a02155f", algorithm="MD5". ... Thanks in advance. Bruno __________________________________________________ Fale com seus amigos de graça com o novo Yahoo! Messenger http://br.messenger.yahoo.com/

2 1

[Users] want to try out openSER
by Ahsen 29 Jan '07

29 Jan '07

Hello ! I have an problem with the SER ..therefore I want to install the OpenSER, but before I install it I want to know whether it can solve my problem or not. I got an shell script which starts a java application , there are no problem with it. But when I start the script with SERs exec module (exec_msg() ) the java application does not support german charset and replaces german spellings with "?". The problem do not occur when I start the shell script manually. I have testet a lot of thinks but the problem still exists. So can someone tell whether he/she has such a problem with OpenSER or not or have an idea how I can solve the problem with SER ? If not I would try out OpenSER ! thanks Ahsen

2 1

[Users] Help. in OpenSER in Post-paid and pre-paid
by raviprakash sunkara 27 Jan '07

27 Jan '07

Hello Users, I'm Using The Following software of My Sip productions, 1) OpenSER is a Sip Server,proxy, in front-end 2) Asterisk is PBX server, in Back-end 3) Radius for AAA to OpenSER, 4) Asterisk-java for IVR for Billing Customer care if I didn't any mistakes in these mail , forgive me . My Question is 1) If Caller calls to Callee , Before the call established to callee. The Caller has to know his Balance Account. if the Balance is there, then only the call has to establish, 2) I want the OpenSER total in RealTime application Server like Java application . Call Authorization ( Question no 1 ). If Asterisk is Front-end , ---> problem is Solved, How can a b2bua implement with OpenSER, please when -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

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[Serusers] ERROR: load_module: could not open module </usr/lib/ser/modulestextops.so>
by subhransu behera 27 Jan '07

27 Jan '07

Hi All I am using a FC-6 machine and installed ser-0.9.6-9.fc6. But while starting the serctl I got the following the error. So do you have any clue about it? Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed ********************************************************* 0(4265) ERROR: load_module: could not open module </usr/lib/ser/modulessl.so>: /usr/lib/ser/modulessl.so: cannot open shared object file: No such file or directory 0(4265) parse error (30,13-14): failed to load module 0(4265) ERROR: load_module: could not open module </usr/lib/ser/modulestm.so>: /usr/lib/ser/modulestm.so: cannot open shared object file: No such file or directory 0(4265) parse error (31,13-14): failed to load module 0(4265) ERROR: load_module: could not open module </usr/lib/ser/modulesrr.so>: /usr/lib/ser/modulesrr.so: cannot open shared object file: No such file or directory 0(4265) parse error (32,13-14): failed to load module 0(4265) ERROR: load_module: could not open module </usr/lib/ser/modulesmaxfwd.so>: /usr/lib/ser/modulesmaxfwd.so: cannot open shared object file: No such file or directory 0(4265) parse error (33,13-14): failed to load module 0(4265) ERROR: load_module: could not open module </usr/lib/ser/modulesusrloc.so>: /usr/lib/ser/modulesusrloc.so: cannot open shared object file: No such file or directory 0(4265) parse error (34,13-14): failed to load module 0(4265) ERROR: load_module: could not open module </usr/lib/ser/modulesregistrar.so>: /usr/lib/ser/modulesregistrar.so: cannot open shared object file: No such file or directory 0(4265) parse error (35,13-14): failed to load module 0(4265) ERROR: load_module: could not open module </usr/lib/ser/modulestextops.so>: /usr/lib/ser/modulestextops.so: cannot open shared object file: No such file or directory 0(4265) parse error (36,13-14): failed to load module 0(4265) set_mod_param_regex: No module matching usrloc found | 0(4265) parse error (47,21-22): Can't set module parameter 0(4265) set_mod_param_regex: No module matching rr found | 0(4265) parse error (65,19-20): Can't set module parameter 0(4265) parse error (75,34-35): unknown command, missing loadmodule? 0(4265) parse error (76,22-23): unknown command, missing loadmodule? 0(4265) parse error (80,23-24): unknown command, missing loadmodule? 0(4265) parse error (89,32-33): unknown command, missing loadmodule? 0(4265) parse error (93,18-19): unknown command, missing loadmodule? 0(4265) parse error (95,19-20): unknown command, missing loadmodule? 0(4265) parse error (102,19-20): unknown command, missing loadmodule? 0(4265) parse error (120,11-12): unknown command, missing loadmodule? 0(4265) parse error (124,12-13): unknown command, missing loadmodule? 0(4265) parse error (126,20-21): unknown command, missing loadmodule? 0(4265) parse error (132,17-18): unknown command, missing loadmodule? 0(4265) parse error (133,24-25): unknown command, missing loadmodule? 0(4265) parse error (137,18-19): unknown command, missing loadmodule? 0(4265) parse error (145,15-16): unknown command, missing loadmodule? 0(4265) parse error (146,18-19): unknown command, missing loadmodule? ERROR: bad config file (24 errors) [root@host223 ~]# /usr/sbin/serctl start Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed [root@host223 ~]# rpm -q ser ser-0.9.6-9.fc6 [root@host223 ~]# [root@host223 ~]# /usr/sbin/serctl start Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed [root@host223 ~]# vi /var/log/messages [root@host223 ~]# /usr/sbin/ser -c 0(4316) ERROR: load_module: could not open module </usr/lib/ser/modulessl.so>: /usr/lib/ser/modulessl.so: cannot open shared object file: No such file or directory 0(4316) parse error (30,13-14): failed to load module 0(4316) ERROR: load_module: could not open module </usr/lib/ser/modulestm.so>: /usr/lib/ser/modulestm.so: cannot open shared object file: No such file or directory 0(4316) parse error (31,13-14): failed to load module 0(4316) ERROR: load_module: could not open module </usr/lib/ser/modulesrr.so>: /usr/lib/ser/modulesrr.so: cannot open shared object file: No such file or directory 0(4316) parse error (32,13-14): failed to load module 0(4316) ERROR: load_module: could not open module </usr/lib/ser/modulesmaxfwd.so>: /usr/lib/ser/modulesmaxfwd.so: cannot open shared object file: No such file or directory 0(4316) parse error (33,13-14): failed to load module 0(4316) ERROR: load_module: could not open module </usr/lib/ser/modulesusrloc.so>: /usr/lib/ser/modulesusrloc.so: cannot open shared object file: No such file or directory 0(4316) parse error (34,13-14): failed to load module 0(4316) ERROR: load_module: could not open module </usr/lib/ser/modulesregistrar.so>: /usr/lib/ser/modulesregistrar.so: cannot open shared object file: No such file or directory 0(4316) parse error (35,13-14): failed to load module 0(4316) ERROR: load_module: could not open module </usr/lib/ser/modulestextops.so>: /usr/lib/ser/modulestextops.so: cannot open shared object file: No such file or directory 0(4316) parse error (36,13-14): failed to load module 0(4316) set_mod_param_regex: No module matching usrloc found | 0(4316) parse error (47,21-22): Can't set module parameter 0(4316) set_mod_param_regex: No module matching rr found | 0(4316) parse error (65,19-20): Can't set module parameter 0(4316) parse error (75,34-35): unknown command, missing loadmodule? 0(4316) parse error (76,22-23): unknown command, missing loadmodule? 0(4316) parse error (80,23-24): unknown command, missing loadmodule? 0(4316) parse error (89,32-33): unknown command, missing loadmodule? 0(4316) parse error (93,18-19): unknown command, missing loadmodule? 0(4316) parse error (95,19-20): unknown command, missing loadmodule? 0(4316) parse error (102,19-20): unknown command, missing loadmodule? 0(4316) parse error (120,11-12): unknown command, missing loadmodule? 0(4316) parse error (124,12-13): unknown command, missing loadmodule? 0(4316) parse error (126,20-21): unknown command, missing loadmodule? 0(4316) parse error (132,17-18): unknown command, missing loadmodule? 0(4316) parse error (133,24-25): unknown command, missing loadmodule? 0(4316) parse error (137,18-19): unknown command, missing loadmodule? 0(4316) parse error (145,15-16): unknown command, missing loadmodule? 0(4316) parse error (146,18-19): unknown command, missing loadmodule? ********************************************************* Kindly assist me how can I start the ser. Regards, Subhransu __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

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[Users] tm:t_write_unix: write_to_unixsock failed
by Peter Lemenkov 26 Jan '07

26 Jan '07

Hello, All! I'm trying to use openser as SIP-router and SEMS as voicemail/conference/etc. But then I tried it the following error occured then openSER tried to contact SEMS: 0(31580) ERROR: tsend_datagram_ev: failed to send: (12) Cannot allocate memory 0(31580) ERROR:tm:write_to_unixsock: writev failed: Cannot allocate memory 0(31580) ERROR:tm:t_write_unix: write_to_unixsock failed I pointed right SEMS socked in OpenSER config, but looks like no one single byte wasn't written in in. omega1 ~ # openser -V version: openser 1.2.0-dev18-notls (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. @(#) $Id: main.c,v 1.27 2007/01/11 10:58:47 bogdan_iancu Exp $ main.c compiled on 15:27:13 Jan 15 2007 with gcc 3.4.6 omega1 ~ # I'm using latest SEMS from SVN (rev. 192), but stable release don't work also. Here is my openser.cfg: ======================================== if (!t_newtran()){ sl_send_reply("500","could not create transaction"); return; }; if (method=="INVITE"){ # avp_db_load("$ru", "$avp(email)"); if(!t_write_unix("/tmp/sems_sock","conference")) { t_reply("500","error contacting sems"); }; } return; ======================================== What I'm dong wrong? -- With best regards!

2 3

[Users] MSRP support
by Danilo Vilela Dantas 26 Jan '07

26 Jan '07

Hi all, two doubts about OpenSER version 1.2: 1. This version will support MSRP sessions ? It will implements a MSRP Relay? 2. It will support chunking messages ? Thank you very much and Best regards, Danilo Vilela Researcher Inovax Engenharia de Sistemas Rio de Janeiro, Brazil

2 1

[Serusers] ser and radius - WARNING! No "known good" password found for the user.
by tzieleniewski 25 Jan '07

25 Jan '07

Hi! I am trying to use freeradius as an authentication server for ser. I use the latest cvs version. I am o newbee with free radius so a kindly ask for your help becouse someone who has manage to done radius support probably wouldn/t have any probelms to point me what I do wrong. I configured ser so it sends the requests messages to radius but I encountered a strange error. Both are running on the same machine so it is enough to use the 127.0.0.1 client configuration from client.conf. I defined realm that would service the request comming with the "voip.touk.pl" domain part in the proxy.conf: realm voip.touk.pl { type = radius authhost = LOCAL accthost = LOCAL nostrip } I also included the digest modules in the authentication and authorization sections in the radiusd.conf. At the end I added the user data in the users file: hellboy Auth-Type := Digest, User-Password := "hellboy", Huntgroup-Name == "voip.touk.pl" Fall-Through = No, User-Name = `%{User-Name}`, Reply-Message = "Accepted: %{User-Name}" And know when SER sends the authentication request radius rejects it and there is this "strange" info" in radius debug rlm_pap: WARNING! No "known good" password found for the user. Authentication may fail because of this. Another thing wich is strange is: modcall[authorize]: module "suffix" returns noop for request 0 When my suffix is defined in the radiusd.conf as: realm suffix { format = suffix delimiter = "@" } here is the whole free radius degub outout: rad_recv: Access-Request packet from host 127.0.0.1 port 32794, id=225, length=299 User-Name = "hellboy(a)voip.touk.pl" Digest-Attributes = 0x0a0968656c6c626f79 Digest-Attributes = 0x010e766f69702e746f756b2e706c Digest-Attributes = 0x022a34356238633234313732386462316132343230333639303633313264366333626338363135306563 Digest-Attributes = 0x04177369703a7465737440766f69702e746f756b2e706c Digest-Attributes = 0x0308494e56495445 Digest-Attributes = 0x050661757468 Digest-Attributes = 0x090a3030303030303031 Digest-Attributes = 0x08223133304432333745384238443837353945304441383438444443323133353542 Digest-Response = "bd2e0fd81858b8114d326394b7b92001" Service-Type = 0x0000000f00000000 SER-Service-Type = 0x0000000300000000 SER-Uri-User = "hellboy" NAS-Port = 0x000013c400000000 NAS-IP-Address = 0x7f00000100000000 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 0 modcall[authorize]: module "preprocess" returns ok for request 0 rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 0 rlm_realm: Looking up realm "voip.touk.pl" for User-Name = "hellboy(a)voip.touk.pl" rlm_realm: Found realm "voip.touk.pl" rlm_realm: Proxying request from user hellboy to realm voip.touk.pl rlm_realm: Adding Realm = "voip.touk.pl" rlm_realm: Authentication realm is LOCAL. modcall[authorize]: module "suffix" returns noop for request 0 modcall[authorize]: module "files" returns noop for request 0 modcall[authorize]: module "expiration" returns noop for request 0 modcall[authorize]: module "logintime" returns noop for request 0 rlm_pap: WARNING! No "known good" password found for the user. Authentication may fail because of this. modcall[authorize]: module "pap" returns noop for request 0 modcall: group authorize returns ok for request 0 rad_check_password: Found Auth-Type DIGEST auth: type "digest" Processing the authenticate section of radiusd.conf modcall: entering group authenticate for request 0 rlm_digest: Cleartext-Password or Digest-HA1 is required for authentication. modcall[authenticate]: module "digest" returns invalid for request 0 modcall: group authenticate returns invalid for request 0 auth: Failed to validate the user. Login incorrect: [hellboy(a)voip.touk.pl/<via Auth-Type = DIGEST>] (from client ip4.localhost port 0) Found Post-Auth-Type Processing the post-auth section of radiusd.conf modcall: entering group REJECT for request 0 radius_xlat: 'hellboy(a)voip.touk.pl' attr_filter: Matched entry DEFAULT at line 11 modcall[post-auth]: module "attr_filter.access_reject" returns updated for request 0 modcall: group REJECT returns updated for request 0 Best tomasz

2 1

[Serusers] Ser - Asterisk small problem
by karim basraoui 25 Jan '07

25 Jan '07

Hi, I am using an architecture with Ser and Asterisk. When Ser forwards the calls to asterisk, everything is all right. the caller has a "ringing" message and the callee rings. when the callee picks up, the caller has a "connecting" message but the callee hungs up suddenly. my asterisk shows the message: -- Got SIP Response 503 "server error" back from Ser_ip -- how can i tell to asterisk to setup the call ? thank you ! -- BASRAOUI Karim

2 1

[Serusers] SER forwarding to Asterisk
by karim basraoui 25 Jan '07

25 Jan '07

Hi, The topic may have been already discussed, but still there are some problems. I have an architecture like this : UA1 ->SER -> ASTERISK -> UA2 when UA1 sends an INVITE to UA2, SER forwards the call to ASTERISK, i can hear UA2 ringing. however, when UA2 answers, i can see "connected" on UA1 but UA2 hungs up by itself, as if ASTERISK cancels the call. is there any solution to oblige ASTERISK to forward the "200 OK" to SER? thanks in advance -- BASRAOUI Karim

1 1

[Users] authentication protocol
by Artur Hayne 25 Jan '07

25 Jan '07

Hi, Its possible the openser change the authentication protocol of the digest for otherone? Thanks Artur. __________________________________________________ Fale com seus amigos de graça com o novo Yahoo! Messenger http://br.messenger.yahoo.com/

2 1

[Serusers] Re: [OpenIMSCore-Users] Getting 513 - Message too big for REGISTER message
by Dragos Vingarzan 25 Jan '07

25 Jan '07

Strange, this looks like a SER bug... Just comment these line and if you want to pursue it further, please report it on the serdev(a)iptel.org mailing lists. It seems to be only relevant to Solaris though. -Dragos Sudhakar Patil wrote: > Hello, > I am using FOKUS OpenIMSCore modules and configuration > that are built around SER. > > I am running the software on Solaris 8 OS with Sparc > v9 architecture and 64 bit compilation. > > I am using xten client. It sends REGISTER message but > gets 513: Message too long in the response. The actual > message size is 546 bytes. I have placed some DBG > commands in SER receive.c and action.c. Everywhere the > message size seems to be correct and is 546 bytes. > > The .cfg file of sip2ims module has the following code > snippet: > > if (msg:len >= max_len ) { > sl_send_reply("513", "Message too big"); > exit; > }; > > I tracked the code eventually to route.c, comp_num > function where the comparison is made between msg:len > and max_len. However, when I print left and right > numbers, the right number is always 0 and hence the > expression is always TRUE and I get 513 always. > (Please see 1(22318) In comp_num. left 546, right 0 > debug output below) > > Below is the output of sip2ims module. > > Any help in fixing this problem is greatly > appreciated. > > > 1(22318) qm_malloc(100224ce8, 1392) called from > receive.c: receive_msg(92) > 1(22318) qm_malloc(100224ce8, 1392) returns address > 100265d38 frag. 100265d08 (size=1392) on 1 -th hit > 1(22318) SIP Request: > 1(22318) method: <REGISTER> > 1(22318) uri: <sip:open-ims.test> > 1(22318) version: <SIP/2.0> > 1(22318) parse_headers: flags=2 > 1(22318) qm_malloc(100224ce8, 64) called from > parser/msg_parser.c: parse_headers(297) > 1(22318) qm_malloc(100224ce8, 64) returns address > 1002657d0 frag. 1002657a0 (size=64) on 1 -th hit > 1(22318) qm_malloc(100224ce8, 232) called from > parser/msg_parser.c: get_hdr_field(111) > 1(22318) qm_malloc(100224ce8, 232) returns address > 100265870 frag. 100265840 (size=232) on 1 -th hit > 1(22318) qm_malloc(100224ce8, 64) called from > parser/parse_via.c: parse_via(2142) > 1(22318) qm_malloc(100224ce8, 64) returns address > 1002659b8 frag. 100265988 (size=64) on 1 -th hit > 1(22318) Found param type 232, <branch> = > <z9hG4bK-d87543-27653f391711d351-1--d87543->; state=6 > 1(22318) qm_malloc(100224ce8, 64) called from > parser/parse_via.c: parse_via(2142) > 1(22318) qm_malloc(100224ce8, 64) returns address > 100265a58 frag. 100265a28 (size=64) on 1 -th hit > 1(22318) Found param type 235, <rport> = <n/a>; > state=17 > 1(22318) end of header reached, state=5 > 1(22318) parse_headers: Via found, flags=2 > 1(22318) parse_headers: this is the first via > 1(22318) After parse_msg... > 1(22318) preparing to run routing scripts... > 1(22318) Message length 546 > 1(22318) completed exec_pre_cb > 1(22318) Before run_actions: message length 546 > 1(22318) Message length in run_actions546 > 1(22318) In ROUTE_T 1 546 > 1(22318) Message length in run_actions546 > 1(22318) Message length in run_actions546 > 1(22318) parse_headers: flags=100 > 1(22318) qm_malloc(100224ce8, 64) called from > parser/msg_parser.c: parse_headers(297) > 1(22318) qm_malloc(100224ce8, 64) returns address > 100265af8 frag. 100265ac8 (size=64) on 1 -th hit > 1(22318) DEBUG:maxfwd:is_maxfwd_present: value = 70 > 1(22318) DBG:maxfwd:process_maxfwd_header: value 70 > decreased to 16 > 1(22318) priting e structure 0 > -------------------------------------------- > 1(22318) In comp_num. left 546, right 0 > -------------------------------------------- > 1(22318) Message length in run_actions546 > 1(22318) qm_malloc(100224ce8, 21) called from ut.c: > as_asciiz(135) > 1(22318) qm_malloc(100224ce8, 24) returns address > 100265b98 frag. 100265b68 (size=24) on 1 -th hit > 1(22318) parse_headers: flags=8 > 1(22318) qm_malloc(100224ce8, 64) called from > parser/msg_parser.c: parse_headers(297) > 1(22318) qm_malloc(100224ce8, 64) returns address > 100265c10 frag. 100265be0 (size=200) on 1 -th hit > 1(22318) qm_malloc(100224ce8, 64) called from > parser/msg_parser.c: parse_headers(297) > 1(22318) qm_malloc(100224ce8, 64) returns address > 100266308 frag. 1002662d8 (size=64) on 1 -th hit > 1(22318) qm_malloc(100224ce8, 88) called from > parser/msg_parser.c: get_hdr_field(151) > 1(22318) qm_malloc(100224ce8, 88) returns address > 1002663a8 frag. 100266378 (size=88) on 1 -th hit > 1(22318) end of header reached, state=9 > 1(22318) DEBUG: get_hdr_field: <To> [34]; > uri=[sip:alice@open-ims.test] > 1(22318) DEBUG: to body > ["alice"<sip:alice@open-ims.test> > ] > 1(22318) parse_headers: flags=ffffffffffffffff > 1(22318) qm_malloc(100224ce8, 64) called from > parser/msg_parser.c: parse_headers(297) > 1(22318) qm_malloc(100224ce8, 64) returns address > 100266460 frag. 100266430 (size=64) on 1 -th hit > 1(22318) qm_malloc(100224ce8, 64) called from > parser/msg_parser.c: parse_headers(297) > 1(22318) qm_malloc(100224ce8, 64) returns address > 100266500 frag. 1002664d0 (size=64) on 1 -th hit > 1(22318) qm_malloc(100224ce8, 64) called from > parser/msg_parser.c: parse_headers(297) > 1(22318) qm_malloc(100224ce8, 64) returns address > 1002665a0 frag. 100266570 (size=64) on 1 -th hit > 1(22318) qm_malloc(100224ce8, 40) called from > parser/msg_parser.c: get_hdr_field(130) > 1(22318) qm_malloc(100224ce8, 40) returns address > 100266640 frag. 100266610 (size=40) on 1 -th hit > 1(22318) get_hdr_field: cseq <CSeq>: <1> <REGISTER> > 1(22318) qm_malloc(100224ce8, 64) called from > parser/msg_parser.c: parse_headers(297) > 1(22318) qm_malloc(100224ce8, 64) returns address > 1002666c8 frag. 100266698 (size=64) on 1 -th hit > 1(22318) qm_malloc(100224ce8, 64) called from > parser/msg_parser.c: parse_headers(297) > 1(22318) qm_malloc(100224ce8, 64) returns address > 100266768 frag. 100266738 (size=64) on 1 -th hit > 1(22318) qm_malloc(100224ce8, 64) called from > parser/msg_parser.c: parse_headers(297) > 1(22318) qm_malloc(100224ce8, 64) returns address > 100266808 frag. 1002667d8 (size=64) on 1 -th hit > 1(22318) qm_malloc(100224ce8, 64) called from > parser/msg_parser.c: parse_headers(297) > 1(22318) qm_malloc(100224ce8, 64) returns address > 1002668a8 frag. 100266878 (size=64) on 1 -th hit > 1(22318) DEBUG: get_hdr_body : content_length=0 > 1(22318) qm_malloc(100224ce8, 64) called from > parser/msg_parser.c: parse_headers(297) > 1(22318) qm_malloc(100224ce8, 64) returns address > 100266948 frag. 100266918 (size=64) on 1 -th hit > 1(22318) found end of header > 1(22318) qm_free(100224ce8, 100266948), called from > parser/msg_parser.c: parse_headers(313) > 1(22318) qm_free: freeing frag. 100266918 alloc'ed > from parser/msg_parser.c: parse_headers(297) > 1(22318) check_via_address(192.168.120.1, > 192.168.120.1, 0) > 1(22318) qm_malloc(100224ce8, 13) called from > msg_translator.c: rport_builder(353) > 1(22318) qm_malloc(100224ce8, 16) returns address > 100266948 frag. 100266918 (size=64) on 1 -th hit > 1(22318) qm_malloc(100224ce8, 609) called from > msg_translator.c: build_res_buf_from_sip_req(1824) > 1(22318) qm_malloc(100224ce8, 616) returns address > 1002669e8 frag. 1002669b8 (size=616) on 1 -th hit > 1(22318) qm_free(100224ce8, 100266948), called from > msg_translator.c: build_res_buf_from_sip_req(1984) > 1(22318) qm_free: freeing frag. 100266918 alloc'ed > from msg_translator.c: rport_builder(353) > 1(22318) qm_free(100224ce8, 1002669e8), called from > sl_funcs.c: sl_send_reply(176) > 1(22318) qm_free: freeing frag. 1002669b8 alloc'ed > from msg_translator.c: > build_res_buf_from_sip_req(1824) > 1(22318) qm_free(100224ce8, 100265b98), called from > sl.c: w_sl_send_reply(178) > 1(22318) qm_free: freeing frag. 100265b68 alloc'ed > from ut.c: as_asciiz(135) > 1(22318) After run_actions > 1(22318) DEBUG:destroy_avp_list: destroying list 0 > 1(22318) DEBUG:destroy_avp_list: destroying list 0 > 1(22318) DEBUG:destroy_avp_list: destroying list 0 > 1(22318) DEBUG:destroy_avp_list: destroying list 0 > 1(22318) DEBUG:destroy_avp_list: destroying list 0 > 1(22318) DEBUG:destroy_avp_list: destroying list 0 > 1(22318) receive_msg: cleaning up > 1(22318) qm_free(100224ce8, 1002659b8), called from > parser/parse_via.c: free_via_param_list(2361) > 1(22318) qm_free: freeing frag. 100265988 alloc'ed > from parser/parse_via.c: parse_via(2142) > 1(22318) qm_free(100224ce8, 100265a58), called from > parser/parse_via.c: free_via_param_list(2361) > 1(22318) qm_free: freeing frag. 100265a28 alloc'ed > from parser/parse_via.c: parse_via(2142) > 1(22318) qm_free(100224ce8, 100265870), called from > parser/parse_via.c: free_via_list(2373) > 1(22318) qm_free: freeing frag. 100265840 alloc'ed > from parser/msg_parser.c: get_hdr_field(111) > 1(22318) qm_free(100224ce8, 1002657d0), called from > parser/hf.c: free_hdr_field_lst(209) > 1(22318) qm_free: freeing frag. 1002657a0 alloc'ed > from parser/msg_parser.c: parse_headers(297) > 1(22318) qm_free(100224ce8, 100265af8), called from > parser/hf.c: free_hdr_field_lst(209) > 1(22318) qm_free: freeing frag. 100265ac8 alloc'ed > from parser/msg_parser.c: parse_headers(297) > 1(22318) qm_free(100224ce8, 100265c10), called from > parser/hf.c: free_hdr_field_lst(209) > 1(22318) qm_free: freeing frag. 100265be0 alloc'ed > from parser/msg_parser.c: parse_headers(297) > 1(22318) qm_free(100224ce8, 1002663a8), called from > parser/parse_to.c: free_to(784) > 1(22318) qm_free: freeing frag. 100266378 alloc'ed > from parser/msg_parser.c: get_hdr_field(151) > 1(22318) qm_free(100224ce8, 100266308), called from > parser/hf.c: free_hdr_field_lst(209) > 1(22318) qm_free: freeing frag. 1002662d8 alloc'ed > from parser/msg_parser.c: parse_headers(297) > 1(22318) qm_free(100224ce8, 100266460), called from > parser/hf.c: free_hdr_field_lst(209) > 1(22318) qm_free: freeing frag. 100266430 alloc'ed > from parser/msg_parser.c: parse_headers(297) > 1(22318) qm_free(100224ce8, 100266500), called from > parser/hf.c: free_hdr_field_lst(209) > 1(22318) qm_free: freeing frag. 1002664d0 alloc'ed > from parser/msg_parser.c: parse_headers(297) > 1(22318) qm_free(100224ce8, 100266640), called from > parser/parse_cseq.c: free_cseq(102) > 1(22318) qm_free: freeing frag. 100266610 alloc'ed > from parser/msg_parser.c: get_hdr_field(130) > 1(22318) qm_free(100224ce8, 1002665a0), called from > parser/hf.c: free_hdr_field_lst(209) > 1(22318) qm_free: freeing frag. 100266570 alloc'ed > from parser/msg_parser.c: parse_headers(297) > 1(22318) qm_free(100224ce8, 1002666c8), called from > parser/hf.c: free_hdr_field_lst(209) > 1(22318) qm_free: freeing frag. 100266698 alloc'ed > from parser/msg_parser.c: parse_headers(297) > 1(22318) qm_free(100224ce8, 100266768), called from > parser/hf.c: free_hdr_field_lst(209) > 1(22318) qm_free: freeing frag. 100266738 alloc'ed > from parser/msg_parser.c: parse_headers(297) > 1(22318) qm_free(100224ce8, 100266808), called from > parser/hf.c: free_hdr_field_lst(209) > 1(22318) qm_free: freeing frag. 1002667d8 alloc'ed > from parser/msg_parser.c: parse_headers(297) > 1(22318) qm_free(100224ce8, 1002668a8), called from > parser/hf.c: free_hdr_field_lst(209) > 1(22318) qm_free: freeing frag. 100266878 alloc'ed > from parser/msg_parser.c: parse_headers(297) > 1(22318) qm_free(100224ce8, 100265d38), called from > receive.c: receive_msg(239) > 1(22318) qm_free: freeing frag. 100265d08 alloc'ed > from receive.c: receive_msg(92) > > Regards, > Sudhakar. > > > > > > > > ____________________________________________________________________________________ > The fish are biting. > Get more visitors on your site using Yahoo! Search Marketing. > http://searchmarketing.yahoo.com/arp/sponsoredsearch_v2.php > _______________________________________________ > OpenIMSCore-Users mailing list > OpenIMSCore-Users(a)lists.berlios.de > https://lists.berlios.de/mailman/listinfo/openimscore-users > > -- ----------------------------------------- Dipl. Eng. Dragos Vingarzan FOKUS/NGNI Kaiserin-Augusta-Allee 31 10589 Berlin,Germany Phone +49 (0)30 - 3463 - 7385 Mobile +49 (0)163 - 159 - 5221 eMail vingarzan(a)fokus.fraunhofer.de Web www.fokus.fraunhofer.de We could change the world if God would give us the source code... -----------------------------------------------------------------

1 0

[Serusers] Signalig through 2 proxies
by Gustavo Passos Tourinho 25 Jan '07

25 Jan '07

Hello there. I'm have some strange problem and was wondering if anyone of you can help me. I have 2 scenarios to call from a IP terminal to a PSTN. Scenario1: Terminal >> Proxy1 (statefull)>> Proxy2 (stateless)>> GW (PSTN) Scenario2: Terminal >> Proxy1 (statefull)>> GW (PSTN) The problem is when I use scenario 1. When the terminal sends BYE request, the GW never response with OK. In the other hand, using scenario 2, BYE request is respond with OK (by the GW). The signaling is perfectly normal. The signaling is the same, except by the via header add by second proxy. Thanks in advanced, Gustavo

2 2

[Users] avp_db_load
by Brandon Armstead 25 Jan '07

25 Jan '07

Hello everyone, I'm trying to load an avp from the database, now from my understanding I simply use the avp_db_load() function. Now I previously setup my avp_table to be usr_preferences, now I'm doing the following: route[5] { xlog("L_INFO", "time [$Tf] method <$rm> r-uri <$ru> 2nd via <$hdr(via[1])>\n"); avp_db_load("$rU/username", "$avp(s:e911)"); xlog("L_INFO", "CURRENT AVP VALUE: $avp(s:e911)"); if(is_avp_set("$avp(s:e911)")) { xlog("L_INFO", "CURRENT \$ru: $ru"); avp_pushto("$ru", "$avp(s:e911)"); xlog("L_INFO", "AFTER \$ru: $ru"); } route(1); } For route 5, and I have a AVP row for 'e911' in the database, with the username, domain, and the attribute set, the attribute ofcourse being e911, type 0, and a value set. However, I keep getting NULL back for the AVP I'm trying to pull, does anyone care to explain what I'm doing wrong? Thanks. -- Brandon Armstead

6 9

[Serusers] password length & ha1 calculation
by chiahuey 25 Jan '07

25 Jan '07

Hi there, If using plain text password and need to calculate ha1 on the fly, is there any requirement as in there is a minimum length for the password? I am using ser-0.9.6, and facing a strange issue here, if password length < 8, will have inconsistency in successful registration or invite. Always get (401 unauthorized or 407 proxy authentication required). if password length >= 8, will be consistently successful. As we are planning to port the users from another platform, can't change the password for all users. Anyone have any clue in the password length issue? Thanks in advance. Regards, Chia

2 1

[Serusers] non-symmetric rtp with rtpproxy??
by Cesc 24 Jan '07

24 Jan '07

Hi, I know this probably is a useless question, as rtpproxy says it is a symmetric proxy ... but i have to ask, for peace of mind :) In my setup, i have on one side a symmetric rtp phone (A) ... connected to a ser proxy + rtpproxy ... and on the other side, a non-symmetric phone (B). This is all on a lan, so no firewalls and stuff ... but I MUST use the rtpproxy and I MUST use this non-symmetric phone (more like a gateway, actually). A ------ SER+rtpproxy ------ B (nonsym) So, the funny thing happens that i end up reciving audio from B to A ... and then I hear A's own audio after being bounced by B (funny, huh?). The bouncing is due to rtpproxy changing the rtp port where he sends to B ... he starts with the one announced on SDP ... then, after the first RTP received from B, he switches to this new port ... mmmm I guess this is a lost cause ... but maybe someone has a brilliant idea?? :D (like a hack in rtpproxy which suddenly turns it into non-symmetrical-rtp-supported-endpoint proxy? :D ) Regards, Cesc

3 4

[Users] List of Functions
by Brandon Armstead 24 Jan '07

24 Jan '07

Hello everyone, I'm wondering if you guys have a list of functions available, core, module, etc, but one compiled list of functions, such as I'm having a hard time finding what module rewritehostport() belongs to, or is it a core function? Like is there an easier way to overview the functions available, whether it be core, or supplied via a module? Thanks, any input appreciated! -- Brandon Armstead

2 2

RE: [Users] openser, spa9000, etc.
by juliano Duque 24 Jan '07

24 Jan '07

Scott, By default You cannot reach the extensions registered to SPA9000 from the extensions registered at Openser. You will only reach the SPA9000 line registered at Openser. You can reach the other extensions registered at SPA 9000 if the SPA9000 line registered at Openser forward the call to the other extensions or if you set the extensions on SPA9000 to receive calls without authentication but for that to work you need to create aliases for that extensions on Openser pointing it to SPA9000 Regards, Juliano Duque

2 1

[Users] Openser and Grandstream
by Shane Burrell 24 Jan '07

24 Jan '07

We have a few Grandstream ATAs that lockup every few days. It requires a hard reboot. Anyone else experienced this? TIA Shane

1 0

[Users] VoIP tranzit proxy
by Ion Minzu 24 Jan '07

24 Jan '07

Hello , Hello, I have the following situation: UA's ----> openser ----> proxy ------> international 172.19.5.0/24 172.19.5.2 172.19.5.3 traffic A.B.C.D E.F.G.H The problem is about proxy, I need a proxy which would be function as transit between openser (private IP) and a router with a public IP (it should pass the NAT too). I have tested siproxd(http://siproxd.sourceforge.net/) for the case when it sits between the openser (public IP) and the users with private IP - and it works. In the first case siproxd doesn't want to work ( neither as a transparent sip proxy), openser sends INVITE to proxy, it replies 408 Request Time Out. The question is: can RTPproxy or mediaproxy help me in this case? Or maybe there is another solution to solve my problem? -- Best regards, Ion Minzu, mailto:ion.minzu@cts.md

4 7

[Users] About Openser version 1.2 (doubts, questions, contribution)
by Danilo Vilela Dantas 24 Jan '07

24 Jan '07

Hi, we are working on a VoIP project and we will use OpenSER as a SIP proxy. This project needs presence and instant message funcionality. We checked out and compiled the CVS HEAD version of OpenSER, but we have a few questions: 1. The modules that implements this two funcionalities (Presence and IM) are ok to use? 2. If question 1 is no, what problems could we have ? Is there any list of bug reports? 3. Is there any date for version 1.2 to be released ? 4. Can we contribute to OpenSER Project ? Thank you very much and Best regards Danilo Vilela Researcher Inovax Engenharia de Sistemas Rio de Janeiro, Brazil

2 1

[Users] Problem for using presence agent using MSN messenger
by neil thomson 24 Jan '07

24 Jan '07

Hi All, I have installed the openser server on linux machine. I followed the procedure given in the link http://www.openser.org/index.php?option=com_content&task=view&id=26&Itemid=… . The server is running fine, but when I try to connect a client (MSN messenger v 5.1) the subscribe message is failing. I have installed the mysql database also given in the website http://openser.org/pub/openser/latest/packages/rhel-3.0/ . The detailed steps I followed are given below. Thanks in advance, Neil

2 1

[Users] avp_db_query, NULL AVP?
by Brandon Armstead 24 Jan '07

24 Jan '07

Hello everyone, How is this possible? I have the table usr_preferences, this query runs fine: if(avp_db_query("SELECT value FROM usr_preferences WHERE username = '$fU' AND attribute = 'e911'", "$avp(i:911)")), in other words if I take the query that should be constructed here, and is constructed here, and I run it in mysql cli, I get back a result, however this function "does not" create the AVP for me, no matter what I do, I run this, and I run xlog("L_INFO", "the avp \$avp(i:911) equals.... $avp(i:911)"); however it says its NULL every single time, what am I doing wrong, I've read and reread the AVP documentation over a million times, however can not get this simply working, all help is appreciated, thank you. -- Brandon Armstead

2 2

[Users] SMS module modem diagnosis
by Cristian Caprar 24 Jan '07

24 Jan '07

How can I diagnose a problem in the SMS module when opening a modem? I see the following line in the log: DEBUG: modem_process: opening modem After about 5 minutes, I see the next one: ERROR:modem_process: cannot open modem Siemens_S45! Resource temporarily unavailable All happens in libsms_modem.c, line 423, where open() is called. The call blocks for about 5 minutes and afterwards I see the above error in the log. I tried to connect to the modem via minicom (on /dev/ttyS0) and everything is just fine. Thanks, Cristian

2 1

[Users] BUG: io_watch_del: invalid fd
by Dan-Cristian Bogos 24 Jan '07

24 Jan '07

Hi Guys, Since a while, I am getting problems with OpenSER while stopping. The error is bellow: Jan 22 09:48:27 localhost /usr/local/sbin/openser[10018]: BUG: io_watch_del: invalid fd 42, not in [0, 0) Jan 22 09:48:27 localhost /usr/local/sbin/openser[10018]: BUG: io_watch_del: invalid fd 45, not in [0, 0) Jan 22 09:48:27 localhost /usr/local/sbin/openser[10018]: BUG: io_watch_del: invalid fd 43, not in [0, 0) Jan 22 09:48:27 localhost /usr/local/sbin/openser[10018]: BUG: io_watch_del: invalid fd 49, not in [0, 0) Jan 22 09:48:27 localhost /usr/local/sbin/openser[10018]: BUG: io_watch_del: invalid fd 48, not in [0, 0) Jan 22 09:48:27 localhost /usr/local/sbin/openser[10018]: BUG: io_watch_del: invalid fd 44, not in [0, 0) Jan 22 09:48:27 localhost /usr/local/sbin/openser[10018]: BUG: io_watch_del: invalid fd 50, not in [0, 0) Jan 22 09:48:27 localhost /usr/local/sbin/openser[10018]: BUG: io_watch_del: invalid fd 47, not in [0, 0) Jan 22 09:48:27 localhost /usr/local/sbin/openser[10018]: BUG: io_watch_del: invalid fd 51, not in [0, 0) If I will start and stop in the first 10 secs, the error will be not in, but stopping it later will always give out the error. I have seen a similar problem, here http://www.openser.org/pipermail/users/2006-September/006982.html, but no solution provided. Did anybody find anything which could help? Is this error serious enough to make me worry more? At a first glance, if I will stop openser, there will come a loot of errors in postgresql database, something related to not closing the sessions properly. My version: version: openser 1.1.0-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. @(#) $Id: main.c,v 1.20 2006/07/04 17:25:54 bogdan_iancu Exp $ main.c compiled on 20:52:20 Oct 10 2006 with gcc 3.3.5 Please let me know if I can help with further info or tests. Cheers, Dan

2 1

[Users] Newbie needs some (transparent) proxy help
by ryadave＠gmail.com 24 Jan '07

24 Jan '07

Hi I've just had a project dumped on me to build a sip proxy to sit in front of our Nortel MCS setup (and ultimately other VoIP gateways). Previous admin was supposed to have done this but he recently been given the boot and hadn't even touched this project from what I can see. And I'm a VoIP newbie which doesn't help much - keen to learn but defo a newbie Setup is fairly straight forward from what I can tell but I seem to be doing something completely wrong so seeking help. New sip proxy (OpenSER) is to sit in front of the MCS (and in time other boxes as I mentioned) and it should proxy traffic to/from the sip servers based on the end users realms/domains (i.e. 1234(a)test1.sip goes to server X and 1234(a)tes2.sip goes to server Y) - I'll need to add an authentication layer here but one step at a time. There's no NAT'ing or anything else going on - everything (clients, sip server & sip proxy) sits on public IP space. All I've been able to get to work so far is registration but can't make or receive calls. I've spend 2 days now trying to figure this out (newbie eh ?) and am under increading pressure from management to get this sorted so am getting desperate. I've googled this to death and tried all the various configs that sounded like they would do the job - but none have. I'm sure someone has already done this and would appreciate if you could share the config with me ? thanks dave

2 1

[Users] CALL FORWARD / PSTN GW
by Eduardo A. Pérez L. 24 Jan '07

24 Jan '07

Can you send how to remove the RPID or how to change it? Im having same problem

2 1

[Serusers] Info about my iptel.org blog
by Greger V. Teigre 24 Jan '07

24 Jan '07

Hi, As some of you might have seen (there's a link on the http://iptel.org/ main page), I keep a blog covering iptel.org related things. I comment on threads of particular on serusers and serdev, as well as report on status and my thoughts around iptel.org projects, up to now with a focus on SER and the upcoming new release. http://sipstuff.blogspot.com/ The intention is to improve communication about what is happening on iptel.org, as well as covering things of interest that is maybe not suited for the mailing lists or as a page on iptel.org. However, the posts and opinions are entirely mine and are by no means official attitudes or opinions of iptel.org. ;-) I encourage comments (comments are allowed in the blog), suggestions, discussions on the mailing lists, or private notes related to what I cover in the blog! Cheers, Greger aka g-)

1 0

[Users] TLS between Openser and UAs
by Jayesh Nambiar 23 Jan '07

23 Jan '07

Hi, I am trying to implement Openser with TLS. But my requirement is as follows: I wanted that all the UAs that register to my Openser Server should come through TLS and when I forward that call to my PSTN gateway it should go through UDP. While forwarding I understand that it can be done using t_relay_to_udp() but how can the first part be achieved. I needed that all the registrations and subsequent messages between Openser and UAs be in TLS. Is this possible, and if yes what all certificates will be required on my server side and the UA side? The documents only mention about TLS transport between servers/proxies. Please someone help me as I am new to TLS and my understanding related to TLS is not that good. Thanks, w/regards, Jayesh __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

2 1

[Users] Routing inbound calls
by Bill Neely 23 Jan '07

23 Jan '07

We are trying to route inbound calls with openser. In other words, I would like calls from DID 123-345-1234 to go to extension 123 and calls from DID 234-456-6789 to go to extension 234. I assume that lookup("aliases") is the way to do this. But I need an example of what would be in the aliases database. Any example would help. TIA. Bill -- 702-874-3833 1-866-553-3833

2 1

[Users] Please Advice!!
by Jayesh Nambiar 23 Jan '07

23 Jan '07

Hi, We are planning to use Openser in a large scale deployment. We are planning to stick to Linksys ATAs and IP Phones. Now all these phones have good NAT handling capablities. I just wanted to understand if it is better to handle these parameters on the UA side instead of Openser side. Will it cause some overhead to reduce at the Openser side. By using these UAs is it also possible to eliminate the use of Mediaproxy/RTPproxy at the Server side. I have the Openser working fine with Mediaproxy but I just wanted some expert suggestion from the people who are using these kind of UAs. Thanks, w/regards, Jayesh. __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

4 6

[Serusers] SERWEB Installation
by vincenzo mennella 23 Jan '07

23 Jan '07

Hi, I tried to install serweb 0.9.6 (as followed by the docs INSTALL and I've extract serweb directorys under /srv/www/htdocs default dir for apache 2) and ser 0.9.6 on SUSE 10.2 - Linux 2.6.18.2-34-default i686, mysql 5.0.26-12, php5 5.2.0-10, but now I can see the login page, but the problem is that when i try to put username and password in the localhost admin interface and then click "login" button, firefox 2.0 give me a another windows where I have to choose how to open index.php. I think that something is wrong. Can anybody help me? The contents of /var/log/apache2/error_log is: [Fri Jan 19 16:19:36 2007] [notice] child pid 6316 exit signal Segmentation fault (11) [Fri Jan 19 16:19:44 2007] [notice] child pid 6154 exit signal Segmentation fault (11) [Fri Jan 19 16:19:44 2007] [notice] child pid 6317 exit signal Segmentation fault (11) The contents of /var/log/serweb is : Jan 19 16:19:44 serweb [debug] User login: values from login form: username: admin@localhost, password: heslo Jan 19 16:19:44 serweb [debug] User login: checking password of user with username: admin, domain: localhost Jan 19 16:19:44 serweb [debug] User login: authentication succeeded, uuid: 4531d4cf51584ec6b48a59bcb5a34cef Jan 19 16:19:44 serweb [debug] User login: redirecting to page: users.php Jan 19 16:19:44 serweb [debug] User login: values from login form: username: admin@localhost, password: heslo Jan 19 16:19:44 serweb [debug] User login: checking password of user with username: admin, domain: localhost Jan 19 16:19:44 serweb [debug] User login: authentication succeeded, uuid: 4531d4cf51584ec6b48a59bcb5a34cef Jan 19 16:19:44 serweb [debug] User login: redirecting to page: users.php This is my PEAR LIST (installed with command pear install <package> Installed packages, channel pear.php.net: ========================================= Package Version State Archive_Tar 1.3.1 stable Auth_SASL 1.0.2 stable Console_Getopt 1.2 stable DB 1.7.7 stable HTML_Template_IT 1.2.1 stable HTTP 1.4.0 stable Log 1.9.9 stable MDB2 2.3.0 stable Mail 1.1.14 stable Net_SMTP 1.2.8 stable Net_Socket 1.0.6 stable Net_UserAgent_Detect 2.2.0 stable PEAR 1.4.11 stable XML_Parser 1.2.8 stable XML_RPC 1.5.1 stable classkit 0.4 beta These are some questions that I need to ask: 1. Have I to download another PEAR pachage ? 2. The problem is FIREFOX, APACHE 2 OR SERWEB ? 3. Have I to check another config file? Thank you very much Regards, vincenzo ___________________________________ Vinci i biglietti per FIFA World Cup in Germania! yahoo.it/concorso_messenger

3 5

Re: [Serusers] SERWEB Installation
by vincenzo mennella 23 Jan '07

23 Jan '07

I Arun I am not be able to start PHP 4 with your indication, because i have dependencies problem . thanks vincenzo --- Arun Kumar <arunvoip(a)gmail.com> ha scritto: > Hi, > > no need to remove php 5.x rpms just download the php > 4.x and apache compile > in their default location and then install serweb > using these php and > apache. > > thanks > > > On 1/21/07, vincenzo mennella <v_mennella(a)yahoo.it> > wrote: > > > > hi ARUN > > I will try , but I before have to unistall php5.2 > rpm > > package with YAST ( SUSE) and then install php4. > Is it > > correct ? > > > > Can I use this version php 4.4.4. at this link ?: > > http://www.php.net/downloads.php > > > > vincenzo > > > > --- Arun Kumar <arunvoip(a)gmail.com> ha scritto: > > > > > Hi, > > > > > > This version of PHP 5.x has some memory leak > problem > > > so I suggest you > > > install PHP 4.x and install you serweb same > trick > > > worked for me. > > > > > > thanks > > > > > > On 1/20/07, vincenzo mennella > <v_mennella(a)yahoo.it> > > > wrote: > > > > > > > > Hi ARUN > > > > I Tried this command: > > > > #PHP -v > > > > PHP 5.2.0 with SUHOSIN-PATCH 0.9.6.1(CLI) > > > > zend engine v.2.2.0 > > > > Regards > > > > Thank you > > > > vincenzo > > > > > > > > --- Arun Kumar <arunvoip(a)gmail.com> ha > scritto: > > > > > > > > > what is the version of php you are using ? > > > > > > > > > > > > > > > On 1/19/07, vincenzo mennella > > > <v_mennella(a)yahoo.it> > > > > > wrote: > > > > > > > > > > > > Hi, > > > > > > I tried to install serweb 0.9.6 (as > followed > > > by > > > > > the > > > > > > docs INSTALL and I've extract serweb > > > directorys > > > > > under > > > > > > /srv/www/htdocs default dir for apache 2) > and > > > ser > > > > > > 0.9.6 on SUSE 10.2 - Linux > 2.6.18.2-34-default > > > > > i686, > > > > > > mysql 5.0.26-12, php5 5.2.0-10, but now I > can > > > see > > > > > the > > > > > > login page, but the problem is that when i > try > > > to > > > > > put > > > > > > username and password in the "localhost > admin > > > > > > interface" and then click "login" button, > > > firefox > > > > > 2.0 > > > > > > give me a another "windows" where I have > to > > > choose > > > > > > "how to open" index.php. I think that > > > something > > > > > is > > > > > > wrong. Can anybody help me? > > > > > > > > > > > > The contents of > /var/log/apache2/error_log > > > is: > > > > > > [Fri Jan 19 16:19:36 2007] [notice] child > pid > > > 6316 > > > > > > exit signal Segmentation fault (11) > > > > > > [Fri Jan 19 16:19:44 2007] [notice] child > pid > > > 6154 > > > > > > exit signal Segmentation fault (11) > > > > > > [Fri Jan 19 16:19:44 2007] [notice] child > pid > > > 6317 > > > > > > exit signal Segmentation fault (11) > > > > > > > > > > > > The contents of /var/log/serweb is : > > > > > > Jan 19 16:19:44 serweb [debug] User login: > > > values > > > > > from > > > > > > login form: username: admin@localhost, > > > password: > > > > > heslo > > > > > > Jan 19 16:19:44 serweb [debug] User login: > > > > > checking > > > > > > password of user with username: admin, > domain: > > > > > > localhost > > > > > > Jan 19 16:19:44 serweb [debug] User login: > > > > > > authentication succeeded, uuid: > > > > > > 4531d4cf51584ec6b48a59bcb5a34cef > > > > > > Jan 19 16:19:44 serweb [debug] User login: > > > > > redirecting > > > > > > to page: users.php > > > > > > Jan 19 16:19:44 serweb [debug] User login: > > > values > > > > > from > > > > > > login form: username: admin@localhost, > > > password: > > > > > heslo > > > > > > Jan 19 16:19:44 serweb [debug] User login: > > > > > checking > > > > > > password of user with username: admin, > domain: > > > > > > localhost > > > > > > Jan 19 16:19:44 serweb [debug] User login: > > > > > > authentication succeeded, uuid: > > > > > > 4531d4cf51584ec6b48a59bcb5a34cef > > > > > > Jan 19 16:19:44 serweb [debug] User login: > > > > > redirecting > > > > > > to page: users.php > > > > > > > > > > > > This is my PEAR LIST (installed with > command > > > pear > > > > > > install <package> > > > > > > > > > > > > Installed packages, channel pear.php.net: > > > > > > ========================================= > > > > > > Package Version > > > > > State > > > > > > Archive_Tar 1.3.1 > > > > > stable > > > > > > Auth_SASL 1.0.2 > > > > > stable > > > > > > Console_Getopt 1.2 > > > > > stable > > > > > > DB 1.7.7 > > > > > stable > > > > > > HTML_Template_IT 1.2.1 > > > > > stable > > > > > > HTTP 1.4.0 > > > > > stable > > > > > > Log 1.9.9 > > > > > stable > > > > > > MDB2 2.3.0 > > > > > stable > > > > > > Mail 1.1.14 > > > > > stable > > > > > > Net_SMTP 1.2.8 > > > > > stable > > > > > > Net_Socket 1.0.6 > > > > > stable > > > > > > Net_UserAgent_Detect 2.2.0 > > > > > stable > > > > > > PEAR 1.4.11 > > > > > stable > > > > > > XML_Parser 1.2.8 > > > > > stable > > > > > > XML_RPC 1.5.1 > > > > > stable > > > > > > classkit 0.4 > > > > > beta > > > > > > > > > > > > These are some questions that I need to > ask: > > > > > > 1. Have I to download another PEAR > > > pachage ? > > > > > > 2. The problem is FIREFOX, APACHE 2 > OR > > > SERWEB > > > > > ? > > > > > > 3. Have I to check another config > file? > === message truncated === ___________________________________ Vinci i biglietti per FIFA World Cup in Germania! yahoo.it/concorso_messenger

1 0

[Users] Support for T.38
by Max Gregorian 23 Jan '07

23 Jan '07

Hi, Does anyone know if OpenSER supports T.38 for Fax over IP? Or has anyone managed to get it to work? I did some quick tests and what I got was the destination Fax machine rang, but the fax message did not arrive. I don't think the destination fax picked up. Setup was: Fax (PC modem) -> IP Gateway -> OpenSER -> PSTN -> Fax machine Is it possible the Fax tones are not being properly generated by OpenSER, and so not being recognised by the Fax machine? (For reference, the Gateway used is advertised as supporting T.38)

3 3

[Users] About the realm
by Howard Tang 22 Jan '07

22 Jan '07

Hi, I have (silly) question about the realm for authenication which I am not able to fix it. Currently, i have the following config. if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("1.1.1.1", "subscriber")) { www_challenge("1.1.1.1", "0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); save("location"); exit; }; if (method=="INVITE") { if (!proxy_authorize("1.1.1.1","subscriber")) { proxy_challenge("1.1.1.1","0"); exit; } if (!check_from()&& !does_uri_exist()) { sl_send_reply("403", "Use From=ID"); exit; } consume_credentials(); }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; My problem is user can only register or make phone call using 1.1.1.1 as a domain. I have setup a domain name on this ip address. for example sip.proxy.com. when i set up sip.proxy.com as the domain/proxy address in the UA, i was not able to register and received a 503 service unavailable. What do i need to do to allow user register using ip address or the host/domain name at the same time? currently i have it working for ip address only. In the subscriber tables-domain column, i have ip address for the users, do i need to add another entry with the domain name in the domain column instead of he ip address? if that is the case, i will have 2 username which the same number in the table. Please help. -- Howard Tang

2 1

[Users] Load Balancing
by raviprakash sunkara 22 Jan '07

22 Jan '07

Hello Users, How can I perform the load Balancing in My SIP server of Both OpenSER and Asterisk , Currently I have One OpenSER server and Asterisk Server, For OpenSER is to need use these modules, and is any 1) LCR and Dispatcher modules, 2) OSP Modules ( also need ) Please can anyone help me ...... -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

4 3

[Users] OpenSER v1.1.1 released
by Daniel-Constantin Mierla 22 Jan '07

22 Jan '07

Hello, version 1.1.1 of OpenSER has been released. It is an update of version 1.1.0 and includes the fixes for the issues discovered in version 1.1.0, therefore all people using 1.1.0 should upgrade to this new version. For people using sms, permissions, postgres, unixodbc, osp, it is highly recommended to move to this release, in the case no CVS updates were done after v1.1.0 (if you installed and maintain OpenSER from CVS, then you have to use branch rel_1_1_0, as it was for v1.1.0). The compatibility of the configuration file and database has been preserved (you do not need to do database reinstallation). Some details about v1.1.1 can be found at: http://www.openser.org/index.php?option=com_content&task=view&id=47&Itemid=… Detailed changelog: http://www.openser.org/pub/openser/1.1.1/ChangeLog Sources can be downloaded from: http://openser.org/pub/openser/1.1.1/ The documentation for modules is posted at: http://openser.org/docs/modules/1.1.x/ If someone wants to contribute with packages for different distros, please send the link from where can be downloaded to team(a)openser.org. Cheers, Daniel

2 1

[Serusers] ? presence ser 0.10.99? - db/db.c:83: Database driver 'mysql' not found
by tzieleniewski 22 Jan '07

22 Jan '07

HI!! I am working with the presence functionality. Today's morning I compiled the latest CVS source and after I run ser I received the following error: Jan 22 11:19:27 rd /home/ser/ser/ser-0.10.99/presence/ser[3861]: ERROR: db/db.c:83: Database driver 'mysql' not found Jan 22 11:19:27 rd /home/ser/ser/ser-0.10.99/presence/ser[3861]: uri_db:mod_init: Unable to bind to the database module Jan 22 11:19:27 rd /home/ser/ser/ser-0.10.99/presence/ser[3861]: init_mod(): Error while initializing module uri_db my ser.cfg looks like this: .. loadmodule "/home/ser/src/sip_router/modules/uri/uri.so" loadmodule "/home/ser/src/sip_router/modules/uri_db/uri_db.so" loadmodule "/home/ser/src/sip_router/modules/rls/rls.so" loadmodule "/home/ser/src/sip_router/modules/pa/pa.so" loadmodule "/home/ser/src/sip_router/modules/presence_b2b/presence_b2b.so" ... modparam("domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") I have the previous version of ser running as proxy+registrar and there is no such error. Waiting for Your help:) Bests TOmasz

2 1

[Users] Comapring keywords in openser...
by Premnath, KN 22 Jan '07

22 Jan '07

Hi, How to compare core keywords in openser? I wanted to compare to_uri and uri is same.. Eg: if(uri == to_uri)... But there is a syntax error thrown, stating "only myself or string" can be compared... Then could I use if(uri=="$to_uri") Thank You. With warm regards, Premnath, KN The new company Nokia Siemens Networks is expected to start operations in the first quarter of 2007, subject to customary regulatory approvals, the completion of standard closing conditions, and the agreement of a number of detailed implementation steps. Disclaimer: This e-mail may contain trade secrets or privileged, undisclosed or otherwise confidential information. If you have received this e-mail in error, you are hereby notified that any review, copying or distribution of it is strictly prohibited. Please inform us immediately and destroy the original transmittal. Thank you for your cooperation.

2 1

[Users] Asterisk Integration
by Víg Pál 22 Jan '07

22 Jan '07

Hi List members, I am planning to bulid a system for large scale deployment. Looking at Openser features and Asterisk capabilities I decided to build a system on these platforms, but faced the problem of integration. What is the best method of integration ? I read some columns in the docs, and can not decide... I came from the network world and can understand protocols and IP communication quite well. I think I should do the following : CLIENT <---> Internet <---> Openser machines (load balanced) <---> Asterisk machines (behind NAT, also load balanced, doing mainly codec conversion and switching logic) My guesses are : - use dispatcher on the frontend(s) - do not use dispatcher, but rewrite the packets by some programmatical way - any other solution ? I'd like to spare some energy on this not to start developing some not working solution ;) Thanks in advance, Pal Vig

2 1

[Users] avp_check
by Bruno Machado 22 Jan '07

22 Jan '07

Hi. Is it possible to use a avp variable in functions' second parameter when I'm using regular expression? For instance: avp_check("$ruri","re/sip:$avp(s:prefix).*$") I didn't see any code doing that. Thanks in advance. Bruno __________________________________________________ Fale com seus amigos de graça com o novo Yahoo! Messenger http://br.messenger.yahoo.com/

2 1

[Users] Grandstream weirdness.
by Shane Burrell 21 Jan '07

21 Jan '07

Not sure if this is related to signaling or openser but I have a few grandstreams that lock up a similar times that have there signaling done via openser. Doesn't seem to happen with Asterisk. Basically the grandstream seems to hang. Has anyone seen this? Something I should to do keep the grandstream awake? TIA Shane

3 2

[Serusers] MySQL Accounting
by Rosa De Santis 20 Jan '07

20 Jan '07

Hello all. I have accounting with MySQL working fine. I have records on my acc table. Please, I'd like to know if there is a tool to process the acc table entries to get completed calls on another MySQL table, taking into account missing BYEs or CANCELs. I found that CDRtool can do this and more, but its not free and it seems to be complicated. Thanks in advance for any help Rosa. _________________________________________________________________ Descubre Live.com - tu mundo en línea reunido: noticias, deportes, el tiempo, y mucho más. http://www.live.com/getstarted

2 1

[Users] Best practice for DNS failover using OpenSER?
by Kerker Staffan 19 Jan '07

19 Jan '07

Hi I'm trying to get something to work that I thought should work out of the box, but when implemented didn't work the way I thought it would... Please comment. I'm using OpenSER as a core SIP proxy, among other things to connect to a POTS network via two independent Cisco PGW2200 SS7 GW. These gateways have fully DNS resolvable SIP domains (rfc 3263, NAPTR, SRV and A records). NAPTR (udp only) => SRV (udp only) => 2x A records (one for each redundant GW). Calls directed to POTS nicely load shares between the two A records. So far so good. Now, if I disable one of the Gateways, I hang every second call. OpenSER does not try the second A record address if the first doesn't answer. How can I solve this? Shouldn't OpenSER fail over to the second A record listed in the NAPTR => SRV resolving? Or will OpenSER continue to resend all SIP INVITES until timers fire? Would it help if the proxy recieved an ICMP port/destination unreachable from the network? Is there anyway to get around this? In the other direction, from POTS to sip, the PGW2200 nicely switches over to the second of my two OpenSER servers if I shut one of them down. These servers have the same DNS entries (but for another SIP domain, NAPTR => SRV => 2x A record). I would love some best practice implementation clues regarding OpenSER and multiple GW fail over, if anyone of you have such knowledge or experience. Best regards, /Staffan --- Staffan Kerker, Saab Communication Ljungadalsgatan 2, 35180 Växjö, Sweden p. +46 470 42185 c. +46 705 391365 m. staffan.kerker(a)saabgroup.com w. http://www.saabgroup.com

8 16

[Users] patch release openser 1.1.1
by Daniel-Constantin Mierla 19 Jan '07

19 Jan '07

Hello, we plan to release a patch update to 1.1.0, numbered 1.1.1 at the beginning of the next week. It will be a packaging of the current version of the branch rel_1_1_0. People using OpenSER from cvs are already using this if they did 'cvs update' recently, but those using packages released for 1.1.0 from htpp://openser.org should update to it. This is a query to gather any know issues you are aware of, to include them there. There is not going to be any backport, major release 1.2.0 not being too far. Cheers, Daniel

2 1

[Users] avp type field
by Mark Price 19 Jan '07

19 Jan '07

Hi, I found documentation for what the type column means (i.e. possible values and what they imply), but can't seem to find it. Maybe that could be documnted in http://www.openser.org/docs/OpenSER_Database_Structure.html or http://www.openser.org/docs/modules/1.1.x/avpops.html Also, is there a reason why the datatype of the type field is int(11)? 11 seems like an odd number. Mark

2 1

[Serusers] Config for presence on 0.10
by Jens-Martin Groenne 19 Jan '07

19 Jan '07

Hi, Is there an updated config example with presence for 0.10? I find that the example in the presence handbook is outdated. Kind regards, Jens-Martin Groenne

2 1

[Users] ADDON: new nathelper parameter
by Bogdan-Andrei Iancu 18 Jan '07

18 Jan '07

Hi everybody, Thanks to a contribution from Jeremie, nathelper module offers the possibility to configure the "a=nortpproxy:yes\r\n" string. This is a string marker added by nathelper when rtpproxy is enabled. The new "nortpproxy_str" parameter: http://www.openser.org/docs/modules/1.2.x/nathelper.html#AEN217 allows you to both change or disable this string marker. For more see: http://openser.org/pipermail/users/2006-September/006519.html regards, Bogdan

4 7

[Users] allow trusted proxies with dynamic IPs
by Cristian Caprar 18 Jan '07

18 Jan '07

Hi, Is it possible to allow_trusted some PSTN gateways which have a dynamic IP address? >From what it looks like, the trusted table only allows the usage of a source IP address. A hostname that is DNS resolvable or an IP/mask pair would solve this. What can be done? Thanks, Cristian

2 4

[Users] $hdr($avp(s:name))
by JF 18 Jan '07

18 Jan '07

Hi all, Is there support for dynamic avp/hdr names, like $hdr($avp(s:name)) I'm getting <null> when I try this... and I've checked that $avp(s:name) has a valid header name (Contact). Thanks, JF

2 3

[Users] User Registration Security on Openser
by Howard Tang 18 Jan '07

18 Jan '07

Hi, I realized some one is able to make call and registered to my sip proxy while he/she is not in the subscriber table. I couldn't find his/her username in the subscriber table, but i was able to see him/her in the location table. I am able to see he/she made 10 calls from my sip proxy. Anyone have an idea on what i have done wrong? I have included the authentication part of code here. if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("x.x.x.x", "subscriber")) { www_challenge("x.x.x.x", "0"); exit; }; consume_credentials(); save("location"); exit; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("x.x.x.x","0"); exit; } consume_credentials(); }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; -- Howard Tang

5 5

[Serusers] Resource List Notification Problem
by raj gopal 18 Jan '07

18 Jan '07

Hi, I have configured apache and SER as described in presence hand book. I am facing some problem in group resource list subscription and notification. SUBSCRIBE sip:venkatfriends@10.187.83.218 SIP/2.0 Supported: eventlist Subject: Subscription Request Via: SIP/2.0/UDP 10.187.83.222:5061;rport;branch=z9hG4bK08652 Max-Forwards: 70 To: <sip:venkatfriends@10.187.83.218> From: <sip:venkat@10.187.83.218>;tag=z9hG4bK42075489 Call-ID: 382598915012(a)10.187.83.222 CSeq: 1 SUBSCRIBE Contact: <sip:venkat@10.187.83.222:5061> Expires: 7200 User-Agent: XXXXXXXXXXXXXXX Event: presence Content-Length: 0 Accept: application/pidf+xml,multipart/related,application/rlmi+xml SIP/2.0 200 OK Via: SIP/2.0/UDP 10.187.83.222:5061;rport=5061;branch=z9hG4bK08652 To: <sip:venkatfriends@10.187.83.218 >;tag=c20c251a135762f9e7a5a4d8e6d1fe64-93f1 From: <sip:venkat@10.187.83.218>;tag=z9hG4bK42075489 Call-ID: 382598915012(a)10.187.83.222 CSeq: 1 SUBSCRIBE Require: eventlist Contact: <sip:10.187.83.218:5060> Expires: 6200 Server: Sip EXpress router (0.10.99-dev64 (i386/linux)) Content-Length: 0 NOTIFY sip:venkat@10.187.83.222:5061 SIP/2.0 Via: SIP/2.0/UDP 10.187.83.218;branch=z9hG4bK7fa8.3ee87012.0 To: <sip:venkat@10.187.83.218>;tag=z9hG4bK42075489 From: <sip:venkatfriends@10.187.83.218 >;tag=c20c251a135762f9e7a5a4d8e6d1fe64-93f1 CSeq: 10 NOTIFY Call-ID: 382598915012(a)10.187.83.222 Content-Length: 340 User-Agent: Sip EXpress router(0.10.99-dev64 (i386/linux)) Subscription-State: active;expires=6200 Contact: <sip:10.187.83.218:5060> Max-Forwards: 70 Event: presence Require: eventlist Content-Type: multipart/related;type="application/rlmi+xml";start="qwW388551513pPdxX1615182410";boundary="RewXdpxR742478745xA757064664"; --RewXdpxR742478745xA757064664 Content-Transfer-Encoding: binary Content-ID: qwW388551513pPdxX1615182410 Content-Type: application/rlmi+xml;charset="UTF-8" <?xml version="1.0"?> <list xmlns="urn:ietf:params:xml:ns:rlmi" uri=" sip:venkatfriends@10.187.83.218" version="0" fullState="true"> </list> --RewXdpxR742478745xA757064664-- I have created all entries in resource-list.xml and service entry in index file. But, the server sends NOTIFY with no resource list entries and doesn't give any error messages. I am using full-no-failover.cfg configuration file. Whats wrong here ?????? Can anyone help me.... -Venkat

2 1

[Users] SipSAK with OpenSER
by Bill Neely 18 Jan '07

18 Jan '07

Received the following from Klaus a few weeks ago: Does anybody have an example of how to use sipsak with openser. Does openser call sipsak, or is it the other way around? Xantek, Inc. wrote: I am sure this must have been covered somewhere, but I cannot find it. I am trying to get openser to register with username and password and forward a call to that server. This is for outbound sip termination to the PSTN. openser is a proxy thus basically openser can'T do that. Now the good news. For registration at the termination provider you can use sipsak. For sending authentication credentials to the termination provider openser has the "uac" module. But be aware that this module does not always works as it does not increase the cseq when sending the INVITE with credentials (search the archive for details). E.g. the uac module wont work if the termination providers uses Asterisk (as Asterisk checks the cseq) regards klaus -- 702-874-3833 1-866-553-3833

3 2

[Users] CID insertion
by nick 18 Jan '07

18 Jan '07

I have my user authentication in a MYSQL database and use openser administrator to manage users. With every user I can add the Caller ID which ends up in the database (I've confirmed the existence in the DB). I'd like to add a Remote Party ID header into my calls directed to my PSTN termination provider (this is the header they want). I have had a look at the Auth module, and if I understood correctly, needed to add the lines modparam("auth", "rpid_avp", "i:13") to obtain the rpid field from the database and append_rpid_hf(); to actually add the RPID field to my outgoing INVITES, etc. however, I still can't see the header being added to the calls in outgoing SIP traces, and I have the feeling that I'm missing something fundamental.. I've tried also adding this line to load the credentials from the database: modparam("auth_db", "load_credentials", "rpid") Do I need to set the load credentials to be i:13=rpid to load the rpid field into the avp??? It's not 100% clear from the documentation where and how these avps are used (automatically if they're filled, or if they need to be somehow referenced in the function that uses them) and exactly which function is necessary (do I need the auth_db function or not?)..

2 2

[Users] Module Permissions
by Bruno Machado 18 Jan '07

18 Jan '07

Hi I'd like to use the 'permissions' module but there is a something I need to change: the table openser.trusted has a column that matches the 'From URI' with it (from_pattern). Is there a way to change it to RURI? I mean, I don't want to analyze the header From: I want the function 'allow_trusted()' matching the RURI instead. Is it possible? Thanks in advance. Bruno Machado __________________________________________________ Fale com seus amigos de graça com o novo Yahoo! Messenger http://br.messenger.yahoo.com/

2 1

Re: [Users] mi_xmlrpc - versions of dependancies for - libxml, libxmlrpc-c3, libxmlrpc-c3-dev
by Bogdan-Andrei Iancu 17 Jan '07

17 Jan '07

Hi Steven, yes, in the official distribution there is an older version of xmlrpc-c3. Not sure why this, but we decided to build the module against the version you currently find in all distro to avoid forcing the users to compile/install by hand additional software. regards, bogdan Steven Halerman wrote: >Thank you for the quick response. I didn't see >binaries for Fedora, so i downloaded the latest - >xmlrpc-c-1.06.09 (earlier versions would not cleanly >compile with GCC 3.4.4) from >http://sourceforge.net/projects/xmlrpc-c but it does >not have the xmlrpc_abyss.h file. Upon further >digging, i noticed in server_abyss.h, a note that >mentions > >Abyss contents handlers >xmlrpc_server_abyss_rpc2_handler() and >xmlrpc_server_abyss_default_handler() were available >in older Xmlrpc-c, but starting with Release 1.01, >they are not. Instead, call >xmlrpc_server_abyss_set_handlers() to install them. > >Steven > >--- Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> >wrote: > > > >>Hi Steven, >> >>I have libxmlrpc-c3-dev version 0.9.10-4 on a >>unstable debian and >>compiles with no problem. The "xmlrpc_abyss.h" file >>belong to the >>libxmlrpc-c3-dev package: >>/usr/include/xmlrpc_abyss.h could you check >>if you have this hdr file installed on your machine? >> >>regards, >>bogdan >> >>Steven Halerman wrote: >> >> >> >>>I have been trying to make mi_xmlrpc (from cvs >>> >>> >>head). >> >> >>>I am running Fedora Core 3 - gcc version 3.4.4 >>> >>>I get the following error: >>> >>>xr_server.c:37:26: xmlrpc_abyss.h: No such file or >>>directory >>>mi_xmlrpc.c:44:26: xmlrpc_abyss.h: No such file or >>>directory >>>In file included from >>>/usr/local/include/xmlrpc-c/base.h:9, >>> from >>> >>> >>/usr/local/include/xmlrpc.h:1, >> >> >>> from mi_xmlrpc.h:35, >>> from mi_xmlrpc.c:38: >>>/usr/local/include/xmlrpc-c/util.h:20:56: >>>xmlrpc-c/c_util.h: No such file or dir >>>ectory >>>In file included from >>>/usr/local/include/xmlrpc-c/base.h:9, >>> from >>> >>> >>/usr/local/include/xmlrpc.h:1, >> >> >>> from mi_xmlrpc.h:35, >>> from mi_xmlrpc.c:38: >>>/usr/local/include/xmlrpc-c/util.h:150: error: >>> >>> >>syntax >> >> >>>error before "GNU_PRINTF_A >>>TTR" >>>/usr/local/include/xmlrpc-c/util.h:158: error: >>> >>> >>syntax >> >> >>>error before "GNU_PRINTF_A >>>TTR" >>>mi_xmlrpc.c:44:26: xmlrpc_abyss.h: No such file or >>>directory >>>mi_xmlrpc.c: In function `mod_child_init': >>>mi_xmlrpc.c:156: warning: implicit declaration of >>>function `xmlrpc_server_abyss_ >>>init_registry' >>>mi_xmlrpc.c:166: error: >>>`xmlrpc_server_abyss_rpc2_handler' undeclared >>> >>> >>(first >> >> >>>use >>>in this function) >>>mi_xmlrpc.c:166: error: (Each undeclared identifier >>> >>> >>is >> >> >>>reported only once >>>mi_xmlrpc.c:166: error: for each function it >>> >>> >>appears >> >> >>>in.) >>>mi_xmlrpc.c:172: error: >>>`xmlrpc_server_abyss_default_handler' undeclared >>>(first >>>use in this function) >>>make[1]: *** [mi_xmlrpc.o] Error 1 >>> >>>what versions of >>> libxml >>> libxmlrpc-c3 >>> libxmlrpc-c3-dev >>>did you use to successfully compile? >>> >>>Thanks in advance >>>Steven >>> >>>

1 0

[Users] interrupts after 30 secconds
by Ion Minzu 17 Jan '07

17 Jan '07

Hello, I have a problem: When someone begins a call from PSTN to my voip network, the session interrupts after 30 secconds. The UA don't receive the ACK message(6)[the ACK doesn't arrive].The scheme below indicate this situation. P.S. This situation happens just with one type of equipment: atcom AG468; with other equipment (atcom AT320ED, audiocodes, atcom AG168) all works fine. Very strange. Thunks. |cisco-gateway| | openser | |UA's| X.X.X.X - public ip adress eth0-X.X.X.Y - public IP 172.17.6.0/24 | eth1-172.17.6.2 | | Invite | Invite | 1 |------------------------------->|----------------------------->| | Trying | Trying | 2 |<-------------------------------|<-----------------------------| | Ringing | Ringing | 3 |<-------------------------------|<-----------------------------| | PRACK | | 4 |------------------------------->| | | 200 ok | 200 ok | 5 |<-------------------------------|<-----------------------------| | ACK | | 6 |------------------------------->| | |................................|..............................|30 s | Bye | Bye | 7 |<-------------------------------|<-----------------------------| | 200 OK | | |------------------------------->| | | | | -- Best regards, Ion Minzu, Specialist Tehnologii Informationale, Administrator de sistem al Centrului de certificare, Administrator VoIP, I.S."Centrul de Telecomunicatii Speciale", tel:250-517 (office), 069501208 (mob), 382869185 (ICQ) mailto:ion.minzu@cts.md

5 5

[Users] Fwd: [kphone-users] New KPhone in the CVS
by Cesc 17 Jan '07

17 Jan '07

FYI ---------- Forwarded message ---------- From: Klaus Fleischmann <kgfleischmann(a)t-online.de> Date: Jan 16, 2007 7:01 PM Subject: [kphone-users] New KPhone in the CVS To: kphone-devel <kphone-devel(a)lists.sourceforge.net>, kphone-users <kphone-users(a)lists.sourceforge.net > Hi, last night I uploaded new code for kphoneSI, the most important features are: - SIMPLE presence: presence server support (PUBLISH, subscription to watcherlist) - SIMPLE presence end to end: After answered subsription, the buddy will be shown in the contact window (in red) even if he not subscribe us too. - new Call Widget design - help file and dtmf panels with own window - the last error message will be shown in the call widget instead of alarm windows - some more ALSA improvements - document and man page update If you are not used to CVS-downloads, see http://sourceforge.net/docman/?group_id=47742 or http://sourceforge.net/docman/display_doc.php?docid=34779&group_id=47742 for details. Yours Klaus ------------------------------------------------------------------------- Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV _______________________________________________ kphone-users mailing list kphone-users(a)lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/kphone-users

2 1

[Users] REMOVED: old fifo implementation
by Bogdan-Andrei Iancu 17 Jan '07

17 Jan '07

Hi everybody, the old fifo implementation (from core) was removed and replaced by the MI FIFO module. See http://www.openser.org/docs/modules/1.2.x/mi_fifo.html why? See: http://www.openser.org/pipermail/users/2006-December/008154.html To maintain the previous FIFO behaviour, you just have to: 1) remove "fifo=/tmp/openser_fifo" from your config 2) load the mi_fifo module 3) configure modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") that's all :) for an working example, see the default openser.cfg which was already updated: http://openser.cvs.sourceforge.net/openser/sip-server/etc/openser.cfg?revis… regards, bogdan

2 11

[Users] Rtpproxy and re-invites
by Joshua Laroff 17 Jan '07

17 Jan '07

Hello, I am new to openser and rtpproxy. I have tried to configure openser so that reinvites are also handled by rtpproxy. It seems however that they are not. I have included my config below: debug=9 fork=yes log_stderror=no check_via=no dns=no rev_dns=no port=5060 children=4 fifo="/tmp/openser_fifo" # ------------------ Module Loading Section ------------------------- mpath="/usr/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "domain.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "avpops.so" loadmodule "permissions.so" loadmodule "options.so" #------------------- Module Parameters ----------------------------- modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") #--- nathelper params --- modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:8899") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_from", "sip:pinger@openser.org") #--- usrloc params --- modparam("usrloc", "db_mode", 2) #--- registrar params --- modparam("registrar", "nat_flag", 6) #--- rr params --- modparam("rr", "enable_full_lr", 1) #--- tm params --- modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "i:45") #--- permissions params --- modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") #------------------------ Main Routing Block ------------------------------ route { # ----------------------------------------- # Sanity check Section # ----------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ------------------------------------------- # Record Route Section # ------------------------------------------- if (method!="REGISTER") { record_route(); }; # ------------------------------------------- # Call Tear Down Section # ------------------------------------------- if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); }; # ------------------------------------------- # - Loose Route Section # ------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); exit; }; if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); }; if (nat_uac_test("19")) { setflag(6); }; }; route(4); route(1); exit; }; # --------------------------------------------- # - Call Type Processing Section # --------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; exit; }; if (method=="ACK") { route(1); exit; } if (method=="CANCEL") { route(1); exit; } else if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; } else if (uri==myself) { if (method==OPTIONS) { options_reply(); } } lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); } route[1] { # -------------------------------------------- # - Default Message Handler # -------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # -------------------------------------------- # - REGISTER Message Handler # -------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ------------------------------------------------ # - INVITE Message Handler # ------------------------------------------------ if (!allow_trusted()) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); }; if (nat_uac_test("19")) { setflag(6); }; if (uri=~"^sip:1[0-9]{10}@") { strip(1); }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(5); exit; }; if (uri=~"^sip:011[0-9]*@") { # International PSTN route(4); route(5); exit; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { # Domestic PSTN route(4); route(5); exit; }; sl_send_reply("404", "User Not Found"); exit; }; route(4); route(5); } route[4] { # --------------------------------------- # - NAT Traversal Section # --------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; } route[5] { # ---------------------------------------- # - PSTN GATEWAY Handler # ---------------------------------------- rewritehost("204.x.x.x"); # avp_write("i:45", "inv_timeout"); route(1); } onreply_route[1] { # --------------------------------------- # - REPLY ROUTE # --------------------------------------- if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } } if (nat_uac_test("1")) { fix_nated_contact(); }; }

2 1

[Users] mi_xmlrpc - versions of dependancies for - libxml, libxmlrpc-c3, libxmlrpc-c3-dev
by Steven Halerman 17 Jan '07

17 Jan '07

I have been trying to make mi_xmlrpc (from cvs head). I am running Fedora Core 3 - gcc version 3.4.4 I get the following error: xr_server.c:37:26: xmlrpc_abyss.h: No such file or directory mi_xmlrpc.c:44:26: xmlrpc_abyss.h: No such file or directory In file included from /usr/local/include/xmlrpc-c/base.h:9, from /usr/local/include/xmlrpc.h:1, from mi_xmlrpc.h:35, from mi_xmlrpc.c:38: /usr/local/include/xmlrpc-c/util.h:20:56: xmlrpc-c/c_util.h: No such file or dir ectory In file included from /usr/local/include/xmlrpc-c/base.h:9, from /usr/local/include/xmlrpc.h:1, from mi_xmlrpc.h:35, from mi_xmlrpc.c:38: /usr/local/include/xmlrpc-c/util.h:150: error: syntax error before "GNU_PRINTF_A TTR" /usr/local/include/xmlrpc-c/util.h:158: error: syntax error before "GNU_PRINTF_A TTR" mi_xmlrpc.c:44:26: xmlrpc_abyss.h: No such file or directory mi_xmlrpc.c: In function `mod_child_init': mi_xmlrpc.c:156: warning: implicit declaration of function `xmlrpc_server_abyss_ init_registry' mi_xmlrpc.c:166: error: `xmlrpc_server_abyss_rpc2_handler' undeclared (first use in this function) mi_xmlrpc.c:166: error: (Each undeclared identifier is reported only once mi_xmlrpc.c:166: error: for each function it appears in.) mi_xmlrpc.c:172: error: `xmlrpc_server_abyss_default_handler' undeclared (first use in this function) make[1]: *** [mi_xmlrpc.o] Error 1 what versions of libxml libxmlrpc-c3 libxmlrpc-c3-dev did you use to successfully compile? Thanks in advance Steven ____________________________________________________________________________________ Cheap talk? Check out Yahoo! Messenger's low PC-to-Phone call rates. http://voice.yahoo.com

2 1

[Serusers] inv_timeout from DB
by inge 17 Jan '07

17 Jan '07

Hi all, I learn about it in the archive, but I don't really get the good way. If I put in a usr_preferences table like, a value like i:40, how can I learn it and write into the inv_timeout ? I'm not very far of the good result, but this is the point where I block. Thanks for your support.

1 1

[Serusers] Ser 0.10.x CVS and MySQL module
by Daniel Corbe 17 Jan '07

17 Jan '07

For some reason the mysql module for ser compiled cleanly but SER refuses to load it. IE # ser -c 0(28131) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: Cannot load specified object 0(28131) parse error (55,13-14): failed to load module ERROR: bad config file (1 errors) It does exist, is installed and was compile from the same sources as the SER core. I'm not sure what I'm doing wrong. Thanks. -Daniel

3 6

[Users] RTP without proxying, Linksys 3102
by Bill Neely 16 Jan '07

16 Jan '07

Has anyone come up with a method for getting a NATED Linksys 3102 to talk to another NATED UA? I see that it has something to do with the 3102 being a symmetric NAT device. I have tried setting fix_nated_register and force_rport, but this doesn't do it (still no RTP going back and forth). I know we could fix this by setting up a proxy server, but we have bandwidth reasons for not wanting to do this. Can provide openser.cfg if anyone would like to look at it. -- 702-874-3833 1-866-553-3833

2 2

[Users] SER Accounting
by Arun Kumar 16 Jan '07

16 Jan '07

Hi, My ser and asterisk are working fine and able to send calls each other. When my ser sends call to my asterisk, my asterisk ask for pin to place call but as soon as it ask for pin ser starts accouting (ack). Is there any why that my accounting starts only when it connects to actual number that I want to dial. thanks arun

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[Users] SER wrong accounting
by Arun Kumar 16 Jan '07

16 Jan '07

Hi, I've configured my ser to send calls to my asterisk server and doing well. When call reaches to my asterisk server, asterisk will ask for pin to place a call. Here is my question: When I place it sends invite to my asterisk and accounting will store this invite time and when asterisk ask for pin (ack) my ser stores this as ack time and when I disconnect the call (bye) it will store the bye time. But my call is not connected to my actual number. Is there any why that accounting only start when actual call get connect. thanks arun

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[Serusers] MetaSwitch
by Bill Hissong 16 Jan '07

16 Jan '07

Anyone have SER talking to MetaSwitch? I am looking for pointers on the MetaSwitch as well as SER. I have them talking, just not nicely. Bill

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[Users] openser+freeradius+mysql
by yuan zhao 16 Jan '07

16 Jan '07

Hi all, I use the document to build l the system with openser+freeradius+mysql( http://www.openser.org/docs/openser-radius-1.0.x.html), +---------+ +---------+ | PHONE 1 | <--SIP-- + +-------------------+ | STORAGE | +---------+ | SIP SERVER | +----+----+ #===> | (OpenSER) | | ........... | |-------------------| +-------+-------+ | | RADIUS CLIENT|<==AAA==> | RADIUS SERVER | +---------+ | | (radiusclient-ng) | | (FreeRADIUS) | | PHONE N |<--SIP--+ +-------------------+ +---------------+ +---------+ I have two linux PCs.the one install Openser and radius client ,and another install freeradius and DATABASE of mysql .But the document said the Openser also need mysql ,then I have a question, what is the function about the mysql of openser? Maybe it can place the users inforamtion of register,but for freeradius, what about the function ? Then I have another question,I install the system step by step with the document ,and the ua can register correctly,but a few minutes it will break down, the ua said it's time out ,can you give me some suggestion ?can I find other openesr+ freeradius documents except *openser-radius-1.0.x.html* . thaks.

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[Users] Accessing data from reply route in failure route
by T.R. Missner 16 Jan '07

16 Jan '07

Hello, I have a need to pass info contained in a X-header within a 300 message to the failure-route or branch-route for inclusion in a new INVITE which results from the redirect. Is there a mechanism available that would allow me to do this? AVPs are not available in the reply route. Thanks, T.R.

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[Users] NEW FEATURE: branch and script flags
by Bogdan-Andrei Iancu 16 Jan '07

16 Jan '07

Hi everybody, Following some discussions with Juha regarding some extension for the branch flags, we decided it is better to have a re-design of flags in openser to allow a better flexibility and extensibility without any consistency penalties. What we had so far: - message flags (or transaction flags) which are transaction persistent - pseudo-branch flags - a set of flags from the message flags were handled by TM as branch flags (saved per branch and not only per transaction). What we have now: - message flags (or transaction flags) will works as they do now, but the branch mask will be removed (rolling back as in 0.9.x versions). These flags are transaction persistent. - branch flags (NEW) are saved also in transaction, but per branch; also they will be saved in usrloc (per contact). A new set of functions were added for manipulating these flags from script. So, there flags will be registration persistent and branch persistent. - script flags (NEW) are no-message-related flags - they are only script persistent and you can strictly use them for scripting. Once you exit a top level route, they will be lost. These flags are useful and they offer an option to de-congest the message flags - many flags have no need to be saved as they just reflect some scripting status. What changes brings this: - NAT flag will become a branch flag all the time. - you can have custom flags to be saved into usrloc (via branch flags). - more flags in script (via script flags). Corresponding Functions: Message/transaction flags: setflag(flag_idx) resetflag(flag_idx) isflagset(flag_idx) Branch flags: setbflag/setbranchflag(branch_idx,flag_idx) resetbflag/resetbranchflag(branch_idx,flag_idx) isbflagset/isbranchflagset(branch_idx,flag_idx) or, the shorter format, working on the default (branch 0) flags: setbflag(flag_idx) resetbflag(flag_idx) isbflagset(flag_idx) Script flags: setsflag/setscriptflag(flag_idx) resetsflag/resetscriptflag(flag_idx) issflagset/isscriptflagset(flag_idx) Flags and Pseudo Variables Message/transaction flags $mf (decimal) , $mF (hexa) Branch flags $bf (decimal) , $bF (hexa) Script flags $sf (decimal) , $sF (hexa) Flags and routes: Message/transaction flags These flags will show up in all routes where messages related to the initial request are processed. So, they will be visible and changeable in onbranch, failure and onreply routes; the flags will be visible in all branch routes; if you change a flag in a branch route, the next branch routes will inherit the change. Branch flags There flags will show up in all routes where messages related to initial branch request are processed. So, in branch route you will see different sets of flags (as they are different branches); in onreply route yo will see the branch flags corresponding to the branch the reply belongs to; in failure route, the branch flags corresponding to the branch the winning reply belongs to will be visible. In request route, you can have multiple branches (as a result of a lookup(), enum query, append_branch(), etc) - the default branch is 0 (corresponding to the RURI); In reply routes there will be only one branch , the 0 one. In branch route the default branch is the current process branch (having index 0); In failure route, initialy there is only one branch (index 0), corresponding the the failed branch. Script flags There flags are available only in script and are reset after each top level route execution (routes internally triggered by OpenSER). They will be persistent per main route, onreply_route, branch_route, failure_route. Note they will be inherit in routes called from other routes. Example: nat flag handling .......... # 3 - the nat flag modparam("usrloc","nat_bflag",3) .......... route { .......... if (nat detected) setbflag(3); # set branch flag 3 for the branch 0 .......... if (is_method("REGISTER")) { # the branch flags (including 3) will be saved into location save("location"); exit; } else { # lookup will load the branch flag from location if (!lookup("location")) { sl_send_reply("404","Not Found"); exit; } t_on_branch("1") t_relay(); } } onbranch_route[1] { xlog("-------branch=$T_branch_idx, branch flags=$bF\n"); if (isbflagset(3)) { #current branch is marked as natted ......... } } if no parallel forking is done, you can get rid of the branch route and add instead of t_on_branch(): ........ if (isbflagset(3)) { #current branch is marked as natted ......... } ......... I will upload this description on the wiki page to be more accessible - in the mean while, any feedback (reports, better ideas, bugd) are welcome! Regards, Bogdan

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[Serusers] [Fwd: problem with german characters with exec_msg]
by Shazad 16 Jan '07

16 Jan '07

I want to add something... the IM sends an text which is an txtfile, generated with an java program, started by the shell script... exec_msg->shell script-> java->txtfile (the characters are replaced already here not in IM)->Instant Message But starting the java program direct or with the shell script the characters are not replaced..only with SER the problem occurs Hope someone give me some hint. Thanks

1 0

[Serusers] problem with german characters with exec_msg
by Shazad 16 Jan '07

16 Jan '07

Hello ! I got an problem with the exec_msg function. When I use it to start an shell script which sends an Instant Message with german characters the Instant Message arrives to the UA with special german characters replaced with "?" symbol ... When I start the shell script manually on the shell , the error does not occur. I dont know , whether there exists special parameters for exec module or the the exec_msg function to set. Can someone tell me, hoe I can solve the problem with , that the SER doesnt replace special german characters with an "?" . Thanks ___________________________________________________________ Der frühe Vogel fängt den Wurm. Hier gelangen Sie zum neuen Yahoo! Mail: http://mail.yahoo.de

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[Users] No memory left errors and Private Memory Size
by Max Gregorian 16 Jan '07

16 Jan '07

Firstly, Happy New Year to all, I am seeing quite a few of these memory errors in the openser log file: *build_contact(): No memory left*. These seem to be occurring when the UA tries to register with Openser. In the last instance I saw, this lasted almost half an hour before clearing up. So I guess what I am trying to find out really is: 1. Is this related to the Private memory size issue I remember reading about a while ago? I am using Openser 1.0.1 and so far have not messed with the default memory size settings from the sources I downloaded. The link on the Openser website about increasing Private memory seems to imply that this could be what is producing these errors. (I refer to this link - http://openser.org/dokuwiki/doku.php/troubleshooting:private-memory-size). 2. Also, if I really have to increase the private memory, I would like to know how to go about it. Since I already have openser installed and the server is already in use, do I have to wipe the server and partitions and re-install the modified Openser or is there an easier way? Can I just recompile the sources and re-install openser over the previous one and get it to overwrite the existing directory (without experiencing any problems with left over files)? 3. How much Private memory is enough to stop these errors? For example, for a system with 1GB of installed RAM should it be using say 768MB? 4. Finally, can anyone tell me the difference between *private memory* and *share memory* in this context? The only difference I am aware of is to do with, for example, shared graphics memory (e.g. for onboard graphics cards on some systems which use up some amount of installed physical RAM memory instead of their own onboard graphics memory). Is this the same thing? Sorry if some of these questions sound a bit basic but I'm coming from a Windows background. Thanks in advance.

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[Users] something wrong with the OpenSER's install
by chenlu 16 Jan '07

16 Jan '07

hi,I meet trouble about the installing of OpenSER.when I download the code with CVS,I use 'make all',a moment later,there is something happened: openserunix.o(.text+0x8f): In function `main': /usr/local/src/openser-1.0.1/sip-server/utils/openserunix/openserunix.c:73: warning: the use of `tmpnam' is dangerous, better use `mkstemp' make[1]: Leaving directory `/usr/local/src/openser-1.0.1/sip-server/utils/openserunix' How can I do ? Need your help,my friends.

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[Users] Openser with Asterisk
by Naveen 16 Jan '07

16 Jan '07

Dear Users, Can some one please post me the sample configuration guide to configure Openser with asterisk? Thanks and regards Naveen

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[Users] can not connect mysql
by chenlu 16 Jan '07

16 Jan '07

hi,my friends,I want to ask you a question,when I exec thish command '/usr/local/sbin/openser_mysql.sh create',then it has something wrong that ' ERROR 2002(HY000):Cannot connect to local MySQL server through socket ‘/var/lib/mysql/mysql.sock ',I do not know how to do with it,need your help.

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[Serusers] make 'ing lib (xcap, presence, cds) - compling errors
by "tzieleniewski" 16 Jan '07

16 Jan '07

Hi! I was trying to compile the latest cvs sources of ser. As always before compiling ser I tried to compile and make presence stuff contained in lib directory. I need the latest cvs in order to use modparam("pa", "auth_xcap_root", "http://localhost/xcap-root/voip.touk.pl/"). Unfortunatelly there were some errors which I didn't have with my previous version of ottendorf (perhapse i missed something - please help mewith this one): ser@rd:~/src/sip_router/lib$ pwd /home/ser/src/sip_router/lib ser@rd:~/src/sip_router/lib$ make Making in cds make[1]: Wejście do katalogu `/home/ser/src/sip_router/lib/cds' cc -shared -Wall -I/home/ser/src/sip_router/lib -I/home/ser/src/sip_router/lib/cds/.. -o libcds.so cds.o dbid.o dstring.o hash_table.o memory.o msg_queue.o ptr_vector.o ref_cntr.o rr_serialize.o serialize.o ser_profile.o simple_profile.o sip_utils.o sstr.o vector.o -L/home/ser/src/sip_router/lib/cds -L/home/ser/src/sip_router/lib/qsa /usr/bin/ld: cds.o: relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC cds.o: could not read symbols: Bad value collect2: ld returned 1 exit status make[1]: *** [libcds.so] Błąd 1 make[1]: Opuszczenie katalogu `/home/ser/src/sip_router/lib/cds' make: *** [cds] Błąd 2 and ser@rd:~/src/sip_router/lib$ make -f Makefile.ser install prefix="/home/ser/ser/ser-0.10.99" Making install in cds make[1]: Wejście do katalogu `/home/ser/src/sip_router/lib/cds' cc -shared -DNAME='"ser"' -DVERSION='"0.10.99-dev64"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_x86_64 -D__OS_linux -DSER_VER=10099 -DCFG_DIR='"/home/ser/ser/ser-0.10.99/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DHAVE_SELECT -DSER -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron -Wall -Wall -I/home/ser/src/sip_router/lib -I/home/ser/src/sip_router/lib/.. -ldl -lresolv -L/home/ser/src/sip_router/lib/cds -L/home/ser/src/sip_router/lib/qsa -o lib_ser_cds.so cds.o dbid.o dstring.o hash_table.o memory.o msg_queue.o ptr_vector.o ref_cntr.o rr_serialize.o serialize.o ser_profile.o simple_profile.o sip_utils.o sstr.o vector.o /usr/bin/ld: cds.o: relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC cds.o: could not read symbols: Bad value collect2: ld returned 1 exit status make[1]: *** [lib_ser_cds.so] Błąd 1 make[1]: Opuszczenie katalogu `/home/ser/src/sip_router/lib/cds' make: [cds] Błąd 2 (zignorowany) Making install in xcap make[1]: Wejście do katalogu `/home/ser/src/sip_router/lib/xcap' cc -shared -DNAME='"ser"' -DVERSION='"0.10.99-dev64"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_x86_64 -D__OS_linux -DSER_VER=10099 -DCFG_DIR='"/home/ser/ser/ser-0.10.99/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DHAVE_SELECT -DSER -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron -Wall -Wall -I/home/ser/src/sip_router/lib -I/home/ser/src/sip_router/lib/.. -I/usr/include/libxml2 -ldl -lresolv -L/home/ser/src/sip_router/lib/cds -L/home/ser/src/sip_router/lib/qsa -lxml2 -l_ser_cds -o lib_ser_xcap.so common_policy.o msg_rules.o parse_common_rules.o parse_msg_rules.o parse_pres_rules.o pres_rules.o resource_list.o resource_lists_parser.o rls_services_parser.o xcap_client.o xml_utils.o /usr/bin/ld: cannot find -l_ser_cds collect2: ld returned 1 exit status make[1]: *** [lib_ser_xcap.so] Błąd 1 make[1]: Opuszczenie katalogu `/home/ser/src/sip_router/lib/xcap' make: [xcap] Błąd 2 (zignorowany) Making install in presence make[1]: Wejście do katalogu `/home/ser/src/sip_router/lib/presence' cc -shared -DNAME='"ser"' -DVERSION='"0.10.99-dev64"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_x86_64 -D__OS_linux -DSER_VER=10099 -DCFG_DIR='"/home/ser/ser/ser-0.10.99/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DHAVE_SELECT -DSER -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron -Wall -Wall -I/home/ser/src/sip_router/lib -I/home/ser/src/sip_router/lib/.. -I/usr/include/libxml2 -ldl -lresolv -L/home/ser/src/sip_router/lib/cds -L/home/ser/src/sip_router/lib/qsa -lxml2 -l_ser_cds -o lib_ser_presence.so domain_maintainer.o lpidf.o notifier_domain.o pidf.o pres_doc.o qsa.o qsa_params.o utils.o xml_utils.o xpidf.o /usr/bin/ld: cannot find -l_ser_cds collect2: ld returned 1 exit status make[1]: *** [lib_ser_presence.so] Błąd 1 make[1]: Opuszczenie katalogu `/home/ser/src/sip_router/lib/presence' make: [presence] Błąd 2 (zignorowany) Bests TOmasz

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[Serusers] About getting started chapter.9
by Sho Aihara 16 Jan '07

16 Jan '07

hello, I tried to test SER with GettingStarted.pdf document, But now I have trouble on "Chapter 9. PSTN Gateway Connectivity". I made ser.cfg like this chapter. Then I try to restart SER, SIP clients didn't authorize. Here is trace log. 2(3926) SIP Request: 2(3926) method: <REGISTER> 2(3926) uri: <sip:191.0.0.2> 2(3926) version: <SIP/2.0> 2(3926) parse_headers: flags=1 2(3926) Found param type 232, <branch> = <z9hG4bK-d87543-3e309a0333127c58-1--d87543->; state=6 2(3926) Found param type 235, <rport> = <n/a>; state=17 2(3926) end of header reached, state=5 2(3926) parse_headers: Via found, flags=1 2(3926) parse_headers: this is the first via 2(3926) After parse_msg... 2(3926) preparing to run routing scripts... 2(3926) parse_headers: flags=128 2(3926) DEBUG:maxfwd:is_maxfwd_present: value = 70 2(3926) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 2(3926) parse_headers: flags=256 2(3926) end of header reached, state=9 2(3926) DEBUG: get_hdr_field: <To> [40]; uri=[sip:8580002@191.0.0.2] 2(3926) DEBUG: to body ["8580002"<sip:8580002@191.0.0.2> ] 2(3926) get_hdr_field: cseq <CSeq>: <1> <REGISTER> 2(3926) DEBUG: get_hdr_body : content_length=0 2(3926) found end of header 2(3926) find_first_route: No Route headers found 2(3926) loose_route: There is no Route HF 2(3926) is_local(): Realm '191.0.0.2' is not local 2(3926) DEBUG: add_param: tag=f7391c78 2(3926) end of header reached, state=29 2(3926) is_local(): Realm '191.0.0.2' is not local 2(3926) parse_headers: flags=-1 2(3926) check_via_address(1910.0.2, 192.168.1.20, 0) 2(3926) DEBUG:destroy_avp_list: destroying list (nil) 2(3926) receive_msg: cleaning up Ip address 192.168.1.20 is SIP client (CounterPath X-Lite) and 191.0.0.2 is SER's. And I added "191.0.0.2" ip address for "trusted" table (MySQL database). Someone can help me? Thanks --------- Aihara sahria.hao(a)gmail.com

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[Serusers] parameter <auth_xcap_root> not found in module <pa> - ser 0.10.99
by "tzieleniewski" 16 Jan '07

16 Jan '07

Hi! I am trying to enable ser working as the presence server.I use the lastest CVS version I doing everything according to the cookbook and get the following errors when I check the ser.cfg file: ser@rd:~/ser/ser-0.10.99/presence$ ./ser -c -f no-db.ser.cfg 0(23175) set_mod_param_regex: parameter <xcap_root> not found in module <rls> 0(23175) parse error (58,20-21): Can't set module parameter 0(23175) set_mod_param_regex: parameter <auth_xcap_root> not found in module <pa> 0(23175) parse error (69,20-21): Can't set module parameter ERROR: bad config file (2 errors) the corresponding lines in the ser.cfg look like this: loadmodule "/home/ser/src/ser-0.10.99/modules/rls/rls.so" loadmodule "/home/ser/src/ser-0.10.99/modules/pa/pa.so" ... modparam("rls", "xcap_root", "http://localhost/xcap-root/voip.touk.pl") modparam("pa", "auth_xcap_root", "http://localhost/xcap-root/voip.touk.pl") and here are the xcap files: ser@rd:~/xcap/xcap-root$ pwd /home/ser/xcap/xcap-root ser@rd:~/xcap/xcap-root$ find . -print . ./voip.touk.pl ./voip.touk.pl/pres-rules ./voip.touk.pl/pres-rules/users ./voip.touk.pl/pres-rules/users/tzl ./voip.touk.pl/pres-rules/users/tzl/presence-rules.xml ./voip.touk.pl/pres-rules/users/template ./voip.touk.pl/pres-rules/users/template/presence-rules.xml ./voip.touk.pl/pres-rules/users/common ./voip.touk.pl/pres-rules/users/common/presence-rules.xml ./voip.touk.pl/im-rules ./voip.touk.pl/im-rules/users ./voip.touk.pl/im-rules/users/tzl ./voip.touk.pl/im-rules/users/tzl/im-rules.xml ./voip.touk.pl/im-rules/users/template ./voip.touk.pl/im-rules/users/template/im-rules.xml ./voip.touk.pl/im-rules/users/common ./voip.touk.pl/im-rules/users/common/im-rules.xml ./voip.touk.pl/rls-services ./voip.touk.pl/rls-services/global ./voip.touk.pl/rls-services/global/index ./voip.touk.pl/resource-lists ./voip.touk.pl/resource-lists/users ./voip.touk.pl/resource-lists/users/tzl ./voip.touk.pl/resource-lists/users/common ./voip.touk.pl/resource-lists/users/common/resource-list.xml ./voip.touk.pl/resource-lists/users/template ./voip.touk.pl/resource-lists/users/template/resource-list.xml I ask for your help. Bests TOmasz

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[Users] FYI: CPLEd new release
by Bogdan-Andrei Iancu 16 Jan '07

16 Jan '07

Hi everybody. there were a lot of people asking about CPLEd, if it still available, still under development, etc.....so here is a new release: CPLEd 0.4.0 "reborn" The project was revived on SF: https://sourceforge.net/projects/cpled/ documentation, CVS access, release downloads, tracker are available on the project site. Also a new mailing list is available: cpled-users(a)lists.sourceforge.net Regards, Bogdan

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[Serusers] ser autostart
by ali daher 16 Jan '07

16 Jan '07

Hi all is there a way to configure ser to run when the computer boots? regards _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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[Serusers] Truncate username ?
by inge 16 Jan '07

16 Jan '07

Hi all, Is it possible to truncate username in avp_db_load(), or with an other way ? In fact, I want to do a usr_preferences table like, but I don't want to add all my number. Just the first 6 digits. Something like 012345xxxx. So I don't have to add in the table each number between 012345000 to 0123459999. I think we can't do that directly in avp_db_load, but maybe it's possible to do something like that : truncate with an avp_check using regular expression and put the result into a variable, then use this variable like the source parameter in avp_db_load(). Someone have an idea ? Thanks for your support.

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Re: [Serusers] SERWEB Installation
by vincenzo mennella 16 Jan '07

16 Jan '07

which permission do you mean? what have I to check? could you explain with any command or file ? thank you vincenzo --- Arun Kumar <arunvoip(a)gmail.com> ha scritto: > check for permission > > On 1/16/07, vincenzo mennella <v_mennella(a)yahoo.it> > wrote: > > > > Can someone please help me to explain this error? > > > > When I try to open firefox > ...http://lolcalhost/serweb > > there is the following error in > > /var/log/apache2/erro_log: > > > > [Mon Jan 15 13:48:00 2007] [error] [client > 127.0.0.1] > > client denied by server configuration: > > /var/www/html/error_log > > thanks > > vincenzo > > > > > > > > > > > > > > ___________________________________ > > Vinci i biglietti per FIFA World Cup in Germania! > > yahoo.it/concorso_messenger > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > ___________________________________ Vinci i biglietti per FIFA World Cup in Germania! yahoo.it/concorso_messenger

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[Serusers] SERWEB - PECL package for Suse 10.2
by vincenzo mennella 16 Jan '07

16 Jan '07

hi, Some question about installation SERWEB. 1. Where is possible to find PECL package for PHP5 version 5.2.0-10? 2. Please, i am looking for instruction for installation with SUSE 10.2. Thanks vincenzo ___________________________________ Vinci i biglietti per FIFA World Cup in Germania! yahoo.it/concorso_messenger

2 2

[Serusers] SERWEB Installation
by vincenzo mennella 16 Jan '07

16 Jan '07

Can someone please help me to explain this error? When I try to open firefox ...http://lolcalhost/serweb there is the following error in /var/log/apache2/erro_log: [Mon Jan 15 13:48:00 2007] [error] [client 127.0.0.1] client denied by server configuration: /var/www/html/error_log thanks vincenzo ___________________________________ Vinci i biglietti per FIFA World Cup in Germania! yahoo.it/concorso_messenger

2 2

[Serusers] Ottendorf compilation errors
by raj gopal 16 Jan '07

16 Jan '07

Hi, I am newbie to linux and SER. While installing Ottendorf version of SER , I am facing following compilation errors /usr/bin/ld: cannot find -l_ser_cds collect2: ld returned 1 exit status make[1]: *** [dialog.so] Error 1 /usr/bin/ld: cannot find -l_ser_xcap collect2: ld returned 1 exit status make[1]: *** [pa.so] Error 1 /usr/bin/ld: cannot find -l_ser_xcap collect2: ld returned 1 exit status make[1]: *** [presence_b2b.so] Error 1 /usr/bin/ld: cannot find -l_ser_xcap collect2: ld returned 1 exit status make[1]: *** [rls.so] Error 1 What would be the problem??? -Venkat

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[Serusers] SER 0.10.99 (Carpo) - Cancel from CP-7912G not recognized
by Theodor Meier 16 Jan '07

16 Jan '07

Hello, the VoIP Provider Carpo (www.carpo.com) uses Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)). When using the CP-7912G with this server, the CANCEL reuqests aren't processed. When testing myself, has the openser (it was easier to set up with deb-packages) the same behaviour like the SER? Following a log from the Cisco CP-7912G with the 4 CANCEL that SER doesn't recognize. Is it due to configuration or aren't the SIP-Messages from 7912g not right? This log was recorded without the CP-7912G ConnectMode Parameter set to 0×00400000 (set bit 22 to use the received ip address for NAT) Before the 2 INVITE: ---- to SER INVITE sip:01608438483@sip.carpo.de;user=phone SIP/2.0 ---- from SER SIP/2.0 407 Proxy Authentifikation erforderlich ---- To SER ACK sip:01608438483@sip.carpo.de;user=phone SIP/2.0 ---- [0:0] Tx Msg to 80.95.252.5:5060 INVITE sip:01608438483@sip.carpo.de;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.178.41:5062;branch=z9hG4bKa96c7242c55a1bc From: Use of a Display Name <sip:wa.mayer@sip.carpo.de>;tag=3931549548 To: <sip:01608438483@sip.carpo.de;user=phone> Call-ID: 3762823499(a)192.168.178.41 CSeq: 2 INVITE Contact: Use of a Display Name <sip:wa.mayer@192.168.178.41:5062;transport=udp> User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Proxy-Authorization: Digest username="wa.mayer",realm="carpo",nonce="45961c6006af8c7d88236d8de4aa44b64d504738",uri="sip:01608438483@sip. carpo.de",response="77323afc4bea0fb5e2cceb50bff9e404" Expires: 300 Content-Length: 285 Content-Type: application/sdp v=0 o=wa.mayer 2928 2928 IN IP4 192.168.178.41 s=Cisco 7912 SIP Call c=IN IP4 192.168.178.41 t=0 0 m=audio 30004 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 ---- [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.178.41:5062;rport=5062;received=84.150.194.222;branch=z9hG4bKa96c7242c55a1bc From: Use of a Display Name <sip:wa.mayer@sip.carpo.de>;tag=3931549548 To: <sip:01608438483@sip.carpo.de;user=phone> Call-ID: 3762823499(a)192.168.178.41 CSeq: 2 INVITE P-Nat: Yes Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)) Content-Length: 0 Warning: 392 80.95.252.14:5070 "Noisy feedback tells: pid=22183 req_src_ip=80.95.252.5 req_src_port=5060 in_uri=sip:01608438483@sip.carpo.de;user=phone out_uri=sip:491608438483@217.188.44.79 via_cnt==2" Response=100; Trying 0:30;2,0,0,0, ---- [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 183 Sitzungsfortschritt Record-Route: <sip:80.95.252.14:5070;avp=B50DCABydHBwcm94eQMAeWVz;lr=on> Record-Route: <sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.41:5062;rport=5062;received=84.150.194.222;branch=z9hG4bKa96c7242c55a1bc From: Use of a Display Name <sip:wa.mayer@sip.carpo.de>;tag=3931549548 To: <sip:01608438483@sip.carpo.de;user=phone>;tag=e0aef7928df800c0a39d563783d2a332 Call-ID: 3762823499(a)192.168.178.41 CSeq: 2 INVITE Contact: <sip:aa+wa.mayer@80.95.252.29:5090> Content-Type: application/sdp Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)) Content-Length: 382 Warning: 392 80.95.252.29:5090 "Noisy feedback tells: pid=5490 req_src_ip=80.95.252.14 req_src_port=5070 in_uri=sip:491608438483@217.188.44.79 out_uri=sip:491608438483@217.188.44.79 via_cnt==0" v=0 o=- 4067017 0 IN IP4 62.53.226.18 s=Cisco SDP 0 c=IN IP4 80.95.247.69 t=0 0 m=audio 2544 RTP/AVP 0 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194,200-202 a=X-sqn:0 a=X-cap: 1 audio RTP/AVP 100 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 192-194,200-202 a=X-cap: 2 image udptl t38 a=nortpproxy:yes AVT: payload=101 Response=183; Session Progress 18x Response has SDP Frames per packet=2 SetG723Mode: 2 0 [0]Enable encoder 0 RTP TX[0]:SSRC_ID = 9e8b6574 RTP Tx Init: 0, 0 [0]Enable encoder 0 [0]Start/Resume Media-> 80.95.247.69:2544 (pt=0) ---- [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 180 Klingelt Record-Route: <sip:80.95.252.14:5070;avp=B50DCABydHBwcm94eQMAeWVz;lr=on> Record-Route: <sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.41:5062;rport=5062;received=84.150.194.222;branch=z9hG4bKa96c7242c55a1bc From: Use of a Display Name <sip:wa.mayer@sip.carpo.de>;tag=3931549548 To: <sip:01608438483@sip.carpo.de;user=phone>;tag=e0aef7928df800c0a39d563783d2a332 Call-ID: 3762823499(a)192.168.178.41 CSeq: 2 INVITE Content-Type: application/sdp Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)) Content-Length: 0 Warning: 392 80.95.252.29:5090 "Noisy feedback tells: pid=5490 req_src_ip=80.95.252.14 req_src_port=5070 in_uri=sip:491608438483@217.188.44.79 out_uri=sip:491608438483@217.188.44.79 via_cnt==0" Response=180; Ringing ---- [0]DPKT 1st: 91375780 91375620, pt 0 SCC Cmd[0:0]CANCEL [0:0]Cancel INVITE [0]CANCEL Retry 0 [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip::80.95.252.14:5070;avp=B50DCA SIP/2.0 Route: <sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.41:5062;branch=z9hG4bKa96c7242c55a1bc From: Use of a Display Name <sip:wa.mayer@sip.carpo.de>;tag=3931549548 To: <sip:01608438483@sip.carpo.de;user=phone> Call-ID: 3762823499(a)192.168.178.41 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="wa.mayer",realm="carpo",nonce="45961c6006af8c7d88236d8de4aa44b64d504738",uri="sip:01608438483@sip. carpo.de",response="31b050a88b23b31a915df6f6ef8cff28" Content-Length: 0 [0]CANCEL Retry 1 ----- [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip::80.95.252.14:5070;avp=B50DCA SIP/2.0 Route: <sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.41:5062;branch=z9hG4bKa96c7242c55a1bc From: Use of a Display Name <sip:wa.mayer@sip.carpo.de>;tag=3931549548 To: <sip:01608438483@sip.carpo.de;user=phone> Call-ID: 3762823499(a)192.168.178.41 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="wa.mayer",realm="carpo",nonce="45961c6006af8c7d88236d8de4aa44b64d504738",uri="sip:01608438483@sip. carpo.de",response="31b050a88b23b31a915df6f6ef8cff28" Content-Length: 0 [0]CANCEL Retry 2 [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip::80.95.252.14:5070;avp=B50DCA SIP/2.0 Route: <sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.41:5062;branch=z9hG4bKa96c7242c55a1bc From: Use of a Display Name <sip:wa.mayer@sip.carpo.de>;tag=3931549548 To: <sip:01608438483@sip.carpo.de;user=phone> Call-ID: 3762823499(a)192.168.178.41 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="wa.mayer",realm="carpo",nonce="45961c6006af8c7d88236d8de4aa44b64d504738",uri="sip:01608438483@sip. carpo.de",response="31b050a88b23b31a915df6f6ef8cff28" Content-Length: 0 [0]CANCEL Retry 3 ----- [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip::80.95.252.14:5070;avp=B50DCA SIP/2.0 Route: <sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.41:5062;branch=z9hG4bKa96c7242c55a1bc From: Use of a Display Name <sip:wa.mayer@sip.carpo.de>;tag=3931549548 To: <sip:01608438483@sip.carpo.de;user=phone> Call-ID: 3762823499(a)192.168.178.41 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="wa.mayer",realm="carpo",nonce="45961c6006af8c7d88236d8de4aa44b64d504738",uri="sip:01608438483@sip. carpo.de",response="31b050a88b23b31a915df6f6ef8cff28" Content-Length: 0 [0]CANCEL Retry 4 ---- [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip::80.95.252.14:5070;avp=B50DCA SIP/2.0 Route: <sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.41:5062;branch=z9hG4bKa96c7242c55a1bc From: Use of a Display Name <sip:wa.mayer@sip.carpo.de>;tag=3931549548 To: <sip:01608438483@sip.carpo.de;user=phone> Call-ID: 3762823499(a)192.168.178.41 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="wa.mayer",realm="carpo",nonce="45961c6006af8c7d88236d8de4aa44b64d504738",uri="sip:01608438483@sip. carpo.de",response="31b050a88b23b31a915df6f6ef8cff28" Content-Length: 0 BYE/CAN Failed (Proxy No Resp) SCC Cmd[0:0]RELEASE >>> SIP Soft Reset [0:0] <<< ___________________________________________________________ Telefonate ohne weitere Kosten vom PC zum PC: http://messenger.yahoo.de

5 5

[Serusers] SER 0.10.99 (Carpo) - Cancel from CP-7912G not recognized - second
by Theodor Meier 15 Jan '07

15 Jan '07

I found another CANCEL problem where the Request-URI has not two double dots (sip:: ..) The call goes from Carpo to Carpo, wheras in my previous mail the call went to a gateway. What I found in the log is, that the branch Parameter is Via: SIP/2.0/UDP 192.168.178.42:5066;rport=5066;received=84.150.187.204;branch=z9hG4bK68fec26c25c87141 till PRACK, there and in the OK from the server this changes. Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK4abf1bf17596f52 the CANCEL request from the phone has the initial branch value. Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141 finally the CANCEL is rejected SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141;rport=5066;received=84.150.187.204 What is the reason for the rejection of the CANCEL? Is this a server or a phone problem? --------------------------- calling 23200302 SCC Cmd[0:0]CLIP SCC Cmd[0:0]CALL proxy = sip.carpo.de:5060 [0]INVITE Retry 0 [0:0] Tx Msg to 80.95.252.5:5060 INVITE sip:23200302@sip.carpo.de;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK56e50100b962975c From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone> Call-ID: 2749835686(a)84.150.187.204 CSeq: 1 INVITE Contact: <sip:paula.mayer@84.150.187.204:5066;transport=udp> User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Expires: 300 Content-Length: 289 Content-Type: application/sdp v=0 o=paula.mayer 17339 17339 IN IP4 192.168.178.42 s=Cisco 7912 SIP Call c=IN IP4 84.150.187.204 t=0 0 m=audio 30006 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 [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 407 Proxy Authentifikation erforderlich Via: SIP/2.0/UDP 192.168.178.42:5066;rport=5066;received=84.150.187.204;branch=z9hG4bK56e50100b962975c From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=68e836eb5e2cffe6b7aefb548a3e0a9b.43de Call-ID: 2749835686(a)84.150.187.204 CSeq: 1 INVITE P-Nat: Yes Proxy-Authenticate: Digest realm="carpo", nonce="45abcd5aa631dd0be187ade4e8203010f5fb9de8" Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)) Content-Length: 0 Warning: 392 80.95.252.14:5070 "Noisy feedback tells: pid=22174 req_src_ip=80.95.252.5 req_src_port=5060 in_uri=sip:23200302@sip.carpo.de;user=phone out_uri=sip:23200302@sip.carpo.de;user=phone via_cnt==2" Response=407; Proxy Authentication Required [0]TxMsg to 80.95.252.5:5060 ACK sip:23200302@sip.carpo.de;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK56e50100b962975c From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=68e836eb5e2cffe6b7aefb548a3e0a9b.43de Call-ID: 2749835686(a)84.150.187.204 CSeq: 1 ACK User-Agent: Cisco-CP7912/8.0.0-060111A Content-Length: 0 [0]INVITE Retry 0 [0:0] Tx Msg to 80.95.252.5:5060 INVITE sip:23200302@sip.carpo.de;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone> Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 INVITE Contact: <sip:paula.mayer@84.150.187.204:5066;transport=udp> User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Proxy-Authorization: Digest username="paula.mayer",realm="carpo",nonce="45abcd5aa631dd0be187ade4e8203010f5fb9de8",uri="sip:23200302@sip.carpo.de",response="271e020ff6474a1304077083c9beca99" Expires: 300 Content-Length: 289 Content-Type: application/sdp v=0 o=paula.mayer 17358 17358 IN IP4 192.168.178.42 s=Cisco 7912 SIP Call c=IN IP4 84.150.187.204 t=0 0 m=audio 30006 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 [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.178.42:5066;rport=5066;received=84.150.187.204;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone> Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 INVITE P-Nat: Yes Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)) Content-Length: 0 Response=100; Trying [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 180 Klingelt Via: SIP/2.0/UDP 192.168.178.42:5066;rport=5066;received=84.150.187.204;branch=z9hG4bK68fec26c25c87141 Record-Route: <sip:80.95.252.5;lr=on> Record-Route: <sip:80.95.252.11:5070;avp=B50DCABydHBwcm94eQMAeWVz;lr=on> Record-Route: <sip:80.95.252.5;lr=on> From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=3015611499 Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 INVITE Require: 100rel RSeq: 1 Contact: <sip:wa.mayer@84.150.187.204:5062;transport=udp> Server: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Content-Length: 0 Response=180; Ringing [0]PRACK Retry 0 [0:0] Tx Msg to 80.95.252.5:5060 PRACK sip:wa.mayer@84.150.187.204:5062 SIP/2.0 Route: <sip:80.95.252.5;lr=on>,<sip:80.95.252.11:5070;avp=B50DCABydHBwcm94eQMAeWVz;lr=on>,<sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK4abf1bf17596f52 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=3015611499 Call-ID: 2749835686(a)84.150.187.204 CSeq: 3 PRACK RAck: 1 2 INVITE User-Agent: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces, 100rel Proxy-Authorization: Digest username="paula.mayer",realm="carpo",nonce="45abcd5aa631dd0be187ade4e8203010f5fb9de8",uri="sip:23200302@sip.carpo.de",response="34b5d81e9d984c39b09f586b09b12f04" Content-Length: 0 [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.178.42:5066;rport=5066;received=84.150.187.204;branch=z9hG4bK4abf1bf17596f52 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=3015611499 Call-ID: 2749835686(a)84.150.187.204 CSeq: 3 PRACK Server: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Supported: replaces Content-Length: 0 SCC Cmd[0:0]CANCEL [0:0]Cancel INVITE [0]CANCEL Retry 0 [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip:wa.mayer@84.150.187.204:5062 SIP/2.0 Route: <sip:80.95.252.5;lr=on>,<sip:80.95.252.11:5070;avp=B50DCABydHBwcm94eQMAeWVz;lr=on>,<sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone> Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="paula.mayer",realm="carpo",nonce="45abcd5aa631dd0be187ade4e8203010f5fb9de8",uri="sip:23200302@sip.carpo.de",response="5a04c4c23e034a4192e964fa468c9461" Content-Length: 0 [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141;rport=5066;received=84.150.187.204 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=a8c29d96e482b9ddbb33ac37669b77b1.da3a Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)) Content-Length: 0 Warning: 392 80.95.252.5:5060 "Noisy feedback tells: pid=16875 req_src_ip=84.150.187.204 req_src_port=5066 in_uri=sip:wa.mayer@84.150.187.204:5062 out_uri=sip:wa.mayer@84.150.187.204:5062 via_cnt==1" [0]CANCEL Retry 1 [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip:wa.mayer@84.150.187.204:5062 SIP/2.0 Route: <sip:80.95.252.5;lr=on>,<sip:80.95.252.11:5070;avp=B50DCABydHBwcm94eQMAeWVz;lr=on>,<sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone> Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="paula.mayer",realm="carpo",nonce="45abcd5aa631dd0be187ade4e8203010f5fb9de8",uri="sip:23200302@sip.carpo.de",response="5a04c4c23e034a4192e964fa468c9461" Content-Length: 0 [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141;rport=5066;received=84.150.187.204 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=a8c29d96e482b9ddbb33ac37669b77b1.da3a Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)) Content-Length: 0 Warning: 392 80.95.252.5:5060 "Noisy feedback tells: pid=16872 req_src_ip=84.150.187.204 req_src_port=5066 in_uri=sip:wa.mayer@84.150.187.204:5062 out_uri=sip:wa.mayer@84.150.187.204:5062 via_cnt==1" [0]CANCEL Retry 2 [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip:wa.mayer@84.150.187.204:5062 SIP/2.0 Route: <sip:80.95.252.5;lr=on>,<sip:80.95.252.11:5070;avp=B50DCABydHBwcm94eQMAeWVz;lr=on>,<sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone> Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="paula.mayer",realm="carpo",nonce="45abcd5aa631dd0be187ade4e8203010f5fb9de8",uri="sip:23200302@sip.carpo.de",response="5a04c4c23e034a4192e964fa468c9461" Content-Length: 0 [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141;rport=5066;received=84.150.187.204 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=a8c29d96e482b9ddbb33ac37669b77b1.da3a Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)) Content-Length: 0 Warning: 392 80.95.252.5:5060 "Noisy feedback tells: pid=16874 req_src_ip=84.150.187.204 req_src_port=5066 in_uri=sip:wa.mayer@84.150.187.204:5062 out_uri=sip:wa.mayer@84.150.187.204:5062 via_cnt==1" [0]CANCEL Retry 3 [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip:wa.mayer@84.150.187.204:5062 SIP/2.0 Route: <sip:80.95.252.5;lr=on>,<sip:80.95.252.11:5070;avp=B50DCABydHBwcm94eQMAeWVz;lr=on>,<sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone> Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="paula.mayer",realm="carpo",nonce="45abcd5aa631dd0be187ade4e8203010f5fb9de8",uri="sip:23200302@sip.carpo.de",response="5a04c4c23e034a4192e964fa468c9461" Content-Length: 0 [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141;rport=5066;received=84.150.187.204 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=a8c29d96e482b9ddbb33ac37669b77b1.da3a Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux3:00;2,0,0,0, )) Content-Length: 0 Warning: 392 80.95.252.5:5060 "Noisy feedback tells: pid=16871 req_src_ip=84.150.187.204 req_src_port=5066 in_uri=sip:wa.mayer@84.150.187.204:5062 out_uri=sip:wa.mayer@84.150.187.204:5062 via_cnt==1" [0]CANCEL Retry 4 [0:0] Tx Msg to 80.95.252.5:5060 CANCEL sip:wa.mayer@84.150.187.204:5062 SIP/2.0 Route: <sip:80.95.252.5;lr=on>,<sip:80.95.252.11:5070;avp=B50DCABydHBwcm94eQMAeWVz;lr=on>,<sip:80.95.252.5;lr=on> Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone> Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL User-Agent: Cisco-CP7912/8.0.0-060111A Supported: replaces, 100rel Proxy-Authorization: Digest username="paula.mayer",realm="carpo",nonce="45abcd5aa631dd0be187ade4e8203010f5fb9de8",uri="sip:23200302@sip.carpo.de",response="5a04c4c23e034a4192e964fa468c9461" Content-Length: 0 [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141;rport=5066;received=84.150.187.204 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=a8c29d96e482b9ddbb33ac37669b77b1.da3a Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 CANCEL Server: Sip EXpress router (0.10.99-iptelorg/tekelec-SOPv2 (i386/linux)) Content-Length: 0 Warning: 392 80.95.252.5:5060 "Noisy feedback tells: pid=16868 req_src_ip=84.150.187.204 req_src_port=5066 in_uri=sip:wa.mayer@84.150.187.204:5062 out_uri=sip:wa.mayer@84.150.187.204:5062 via_cnt==1" BYE/CAN Failed (Proxy No Resp) SCC Cmd[0:0]RELEASE >>> SIP Soft Reset [0:0] <<< 3:30;2,0,0,0, [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 480 Temporarily Not Available Via: SIP/2.0/UDP 192.168.178.42:5066;rport=5066;received=84.150.187.204;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=3015611499 Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 INVITE Server: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Content-Length: 0 [0]TxMsg to 80.95.252.5:5060 ACK sip:23200302@sip.carpo.de;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.178.42:5066;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=3015611499 Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 ACK User-Agent: Cisco-CP7912/8.0.0-060111A Content-Length: 0 [0]Rx Msg from 80.95.252.5:5060 SIP/2.0 480 Temporarily Not Available Via: SIP/2.0/UDP 192.168.178.42:5066;rport=5066;received=84.150.187.204;branch=z9hG4bK68fec26c25c87141 From: sip:paula.mayer@sip.carpo.de;tag=168806121 To: <sip:23200302@sip.carpo.de;user=phone>;tag=3015611499 Call-ID: 2749835686(a)84.150.187.204 CSeq: 2 INVITE Server: Cisco-CP7912/8.0.0-060111A Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Content-Length: 0 ___________________________________________________________ Der frühe Vogel fängt den Wurm. Hier gelangen Sie zum neuen Yahoo! Mail: http://mail.yahoo.de

2 1

[Serusers] Re: Users presence status
by Vaclav Kubart 15 Jan '07

15 Jan '07

According to PIDF RFC there MUST be <basic> with value "open" or "closed". I guess, that this PUBLISH request should be rejected with non-2xx response. Is it? BTW, PA is not able to handle "partial PIDF" - you have to use PIDF. Presence status can be stored into database if you set "use_db" module parameter to 1. You can publish document like this: <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" entity="pres:venkat@10.187.83.218"> <tuple id="000s3"> <status> <basic>open</basic> </status> <contact>venkat(a)10.187.83.218</contact> </tuple> <dm:person id="p1"> <rpid:activities> <busy/> </rpid:activities> </dm:person> </presence> I'm not sure if it is strictly according to RPID specification, but might be. Much simpler it could be like this: <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" entity="pres:venkat@10.187.83.218"> <tuple id="000s3"> <status> <basic>open</basic> </status> <contact>venkat(a)10.187.83.218</contact> <note>busy</note> </tuple> </presence> Vaclav On Mon, Jan 15, 2007 at 01:34:04PM +0530, raj gopal wrote: > Thanks for reply, > > I am sending PUBLISH method with following simplified PIDF content. > > <pidf-part:presence xmlns:pidf-part="urn:ietf:params:xml:ns:pidf-partial" > xmlns="urn:ietf:params:xml:ns:pidf" > xmlns:et="urn:ietf:params:xml:ns:pidf:rpid-tuple" > xmlns:ep="urn:ietf:params:xml:ns:pidf:status:rpid-status" > xmlns:dc="urn:ietf:params:xml:ns:pidf:status:devcaps" entity=" > venkat(a)10.187.83.218" version="1" state="full"> > <tuple id="000s3"> > <status> > <basic>busy</basic> > </status> > <contact>venkat(a)10.187.83.218</contact> > </tuple> > > But, server sends following NOTIFY msg > > <?xml version="1.0" encoding="UTF-8"?> > <presence xmlns="urn:ietf:params:xml:ns:pidf" entity=" > pres:venkat@10.187.83.218"> > <tuple id="0xb626fba4x52c779c5x45ab1b6d"> > <status><basic>open</basic></status> > <contact priority="0.00">sip:10.187.83.222:5060</contact> > </tuple> > </presence> > > The server sends either closed/open. But it dont send busy status. > > Am i doing something wrong? is status is stored in database?? > > -venkat > > > On 1/15/07, Vaclav Kubart <vaclav.kubart(a)iptel.org> wrote: > > > >Hello, > >SER understands only basic status what means open/closed, but it allows > >clients to publish other info in extension elements (look into PIDF > >specification - RFC 3863). Such elements are not parsed by SER, but they > >are sent in NOTIFY to watchers. > > > >If you don't want to use PIDF extension elements, you can send > >additional status info in <note> elements - many clients do it in this > >way. > > > > Vaclav > > > >On Mon, Jan 15, 2007 at 12:11:31PM +0530, raj gopal wrote: > >> Hi Vaclav, > >> > >> I am using SER as sip and presence server. According to my understanding > >SER > >> supports "open" and "close" presence status. But, my application > >required > >> presence status like busy, tolunch etc.. When client sends PUBLISH with > >his > >> presence status , how does SER process it? while server sending NOTIFY, > >does > >> it consider the status published by client?? > >> > >> Please suggest me to achieve above requirement. > >> > >> -Venkat > >

1 0

[Serusers] Users presence status
by raj gopal 15 Jan '07

15 Jan '07

Hi, Does presence module supports user presence status other than closed/open. -venkat

2 1

Re: [Users] Multi Call Legs Accounting
by Daniel-Constantin Mierla 14 Jan '07

14 Jan '07

Hello, are you proxying the media stream all the time now? Or just when the one of the party is behind the nat? I have in mynd the sceanrio when the initial callee is behind nat, it sends back redirect, and the new destination is public... Cheers, Daniel On 01/12/07 14:36, Howard Tang wrote: > Hi Daniel, > > Thank you very much, the problem fixed. > > Regards, > Howard > > On 1/11/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> Hello, >> >> as i see in your failure route you close mediaproxy nat handling -- try >> to keep it on. ALso, the contacts in redirect message might be natted >> even when initial destination wasn't -- you have to deal with this case >> as well (start nat handling). >> >> Cheers, >> Daniel >> >> >> On 01/09/07 14:23, Howard Tang wrote: >> > Hi Daniel, >> > >> > by using the following code in the failure_route block, i am able to >> > have a better accounting record for billing. However, i am having one >> > way audio problem which i believe it is related to NAT. I have tested >> > forward a call on a UA which is on public ip, which works prefectly. >> > However, when i tested it with UA which is behind NAT, one way audio >> > happen when the UA send 302 reply to forward a call.. >> > >> > failure_route[1] >> > { >> > end_media_session(); >> > # Handle 302 Redirects >> > if (t_check_status("302")) { >> > get_redirects("*"); >> > t_relay(); >> > }; >> > } >> > >> > Anyone has an idea on how i can fix this? >> > >> > Regards, >> > Howard >> > >> > >> > On 1/9/07, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> >> Hello, >> >> >> >> when the call A -> B is forwarded to C, then you need to have in your >> >> script: >> >> >> >> avp_write("A", "$avp(i:110)"); >> >> avp_write("B", "$avp(i:111)"); >> >> avp_write("B", "$avp(i:110)"); >> >> avp_write("C", "$avp(i:111)"); >> >> >> >> A, B, C are the SIP addresses of the parties involved in this call. >> >> >> >> Cheers, >> >> Daniel >> >> >> >> On 01/08/07 23:14, Howard Tang wrote: >> >> > Hi, >> >> > >> >> > Did anyone successfully setup call accounting for call forward? >> I have >> >> > been searching for a working example for few days on google, I >> still >> >> > couldn't find any working example. >> >> > >> >> > The config I have at the moment is >> >> > >> >> > modparam("acc", "multi_leg_enabled", 1) >> >> > modparam("acc", "src_leg_avp_id", 110) >> >> > modparam("acc", "dst_leg_avp_id", 111) >> >> > >> >> > In the main route, >> >> > if (!method=="REGISTER" && !method=="NOTIFY") { >> >> > avp_write("$from/username", "$avp(i:110)"); >> >> > avp_write("$ruri/username", "$avp(i:111)"); >> >> > }; >> >> > >> >> > Let say A call B and B forward call to C, the record in the acc >> table >> >> > is all >> >> > A to C without B in any one of the record for billing purpose. The >> >> > Call forward was set on the UA. >> >> > >> >> > Can anyone give me a hand on this please? >> >> > >> >> >> > >> >

2 1

[Serusers] proxy_challenge and E60
by Stefan Tichy 14 Jan '07

14 Jan '07

Hi, a nokia E60 is registered to SER 0.9.6. The response to a INVITE by the E60 is not a proper 407 response but just a small UDP paket (46 bytes). I cannot be shure that it is really the response and not some data send from time to time, but there is no other response. Ethereal does not identify this paket as SIP if (!proxy_authorize("domain","subscriber")) { log(3,"proxy_challenge"); proxy_challenge("domain","0"); break; } The phone is registered and can be called. Just INVITE does not work. Compression and security flags in phones config are set to off. The firmware of the phone is somewhat old, but probably I am not the only one using E60 with old firmware. The INVITE itself looks not bad and does work for asterisk 1.4 server and other systems. Even if the INVITE would contain garbage SER should send some SIP response with 5XX code. It is strange. Any help appreciated -- Stefan Tichy <seruser(a)pi4tel.de>

4 4

[Users] prevent double registration
by Howard Tang 14 Jan '07

14 Jan '07

Hi, What do i need to do if i want to prevent same user using two different UAs to register to openser in different location? e.g. A has 2 ATA, one at 222.222.222.222 and one in 111.111.111.111, he has put same login detials in the ATA, and try to register openser on 123.123.123.123 Both ATA will successfully register to the proxy, i want to prevent this, what can be done? Regards, Howard

3 4

[Users] Reduce Retry Time
by Arun Kumar 14 Jan '07

14 Jan '07

Hi, I've configured the ser with my lcr and its working fine. If first gateway fails after a wait of 30sec it loads another gw and passes the call to my asterisk server. Now I want to reduce this 30sec retry time to 5sec so in case my first gw is not reachable then after 5sec it should load the next gateway. thanks arun

4 5

[Serusers] Status of SER 2.0 (Ottendorf) documentation
by Greger V. Teigre 14 Jan '07

14 Jan '07

Just a quick summary of documentation status for SER 2.0 (Ottendorf): ------------------- * http://www.iptel.org/ser/doc/010whatsnew contains a set of book pages introducing new features * ser-basic.cfg and ser.cfg in etc/ directory of source package contains a non-database and mysql database example that should be a good basis for testing * http://www.iptel.org/new_major_ser_pre_release_ottendorf_is_out_for_testing provides and overview of new stuff and a step-wise guide to get started * Module documentation is available online at http://www.iptel.org/views/moduledocs (I personally prefer the table overview for quick browsing: http://www.iptel.org/views/tableofmodules). Most of the important modules are available online, but see below for module documentation status What's left? ---------------- * Overall review of all readme files in source tree, as well as review of completeness/correctness of module documentation * Completion of module documentation * Update SER - Getting Started document and configuration files * Complete What's New 0.10.x as a migration guide; evaluate what to extract in a separate manual for SER functionality (updated admin guide) * Make sure all new features have been documented properly either in module docs or migration guide If you want to contribute in some way, please volunteer! (ex. to review module documentation for your favorite module...) g-) ---------------------------------------------------------------------------- Most of the modules are documented online at iptel.org. The documentation is checked out every night (4am CET) from the cvs and generated, so any changes made the previous day will be available online the next day. The modules below are not yet available online and the following have no readme or docbook documentation in CVS (anybody with documentation for these, please make a sound!): acc_db acc_radius ctl dialog fifo xmlrpc These modules have README files available in CVS, but have not been converted to docbook format yet (in the works): db_ops diversion eval mediaproxy postgres timer tls

1 0

[Users] get_hdr_field: bad to header...
by Lars Sundqvist 14 Jan '07

14 Jan '07

Hello, We have a few hundred users registered to a openser we are running and everything works fine, however i get som error messages in the logs I would like to get rid of... //sbin/openser[28867]: ERROR: get_hdr_field: bad to header /sbin/openser[28867]: ERROR: bad header field /sbin/openser[28867]: ERROR: get_hdr_field: bad to header /sbin/openser[28867]: ERROR: bad header field /sbin/openser[28867]: ERROR:parse_from_header: bad from header /sbin/openser[28867]: xl_get_from: ERROR cannot parse FROM header /sbin/openser[28867]: ERROR: get_hdr_field: bad to header /sbin/openser[28867]: ERROR: bad header field /sbin/openser[28867]: xl_get_to: ERROR cannot parse TO header /sbin/openser[28867]: [REGISTER] from [<null>] to [<null>] /sbin/openser[28867]: ERROR: get_hdr_field: bad to header /sbin/openser[28867]: ERROR: bad header field /sbin/openser[28867]: ERROR:maxfwd:is_maxfwd_present : parsing MAX_FORWARD header failed! /sbin/openser[28867]: ERROR:parse_from_header: bad from header /sbin/openser[28867]: insert_RR(): From parsing failed /sbin/openser[28867]: record_route(): Error while inserting Record-Route line /sbin/openser[28867]: ERROR: get_hdr_field: bad to header /sbin/openser[28867]: ERROR: bad header field /sbin/openser[28867]: find_first_route: Error while parsing headers /sbin/openser[28867]: analyzing REGISTER request /sbin/openser[28867]: ERROR: get_hdr_field: bad to header /sbin/openser[28867]: ERROR: bad header field /sbin/openser[28867]: find_credentials(): Error while parsing headers /sbin/openser[28867]: pre_auth(): Error while looking for credentials /sbin/openser[28867]: ERROR: get_hdr_field: bad to header /sbin/openser[28867]: ERROR: bad header field /sbin/openser[28867]: ERROR: get_hdr_field: bad to header /sbin/openser[28867]: ERROR: bad header field /sbin/openser[28867]: ERROR: build_res_buf_from_sip_req: alas, parse_headers failed /sbin/openser[28867]: pre_auth(): Error while sending 400 reply / Anyone has a clue what I can do to debug this to find out where the request are coming from or what can be done to stop the invalid(?) requests from being processed and clutter my logs? Thanks Lars Sundqvist

6 9

[Serusers] How to create sip account Ser 0.10
by jean sigue 12 Jan '07

12 Jan '07

All, I have a fresh install ser 0.10 running with mysql. The utility serctl cannot be use to create sip account because the credentials table replaces the subscriber table. Can someone tell me how to create an account ? [root@itiah MySQL-python-1.2.1_p2]# ps -ef | grep ser root 87 7 0 Jan08 ? 00:00:00 [kseriod] root 1696 1 0 Jan08 ? 00:00:00 /usr/bin/hidd --server mysql 9972 9936 0 Jan08 pts/0 00:00:01 /usr/libexec/mysqld --defaults-file=/etc/my.cnf --basedir=/usr --datadir=/var/lib/mysql --user=myl --pid-file=/var/run/mysqld/mysqld.pid --skip-external-locking --socket=/var/lib/mysql/mysql.sock root 21031 1 0 Jan09 ? 00:00:00 ser root 21038 21031 0 Jan09 ? 00:00:00 ser root 21040 21031 0 Jan09 ? 00:00:00 ser root 21042 21031 0 Jan09 ? 00:00:00 ser root 21044 21031 0 Jan09 ? 00:00:00 ser root 21046 21031 0 Jan09 ? 00:00:00 ser root 21048 21031 0 Jan09 ? 00:00:00 ser root 21050 21031 0 Jan09 ? 00:00:00 ser root 21052 21031 0 Jan09 ? 00:00:00 ser root 21053 21031 0 Jan09 ? 00:00:00 ser root 21056 21031 0 Jan09 ? 00:00:00 ser root 21058 21031 0 Jan09 ? 00:00:00 ser root 21060 21031 0 Jan09 ? 00:00:00 ser root 21062 21031 0 Jan09 ? 00:00:00 ser root 21064 21031 0 Jan09 ? 00:00:00 ser root 21066 21031 0 Jan09 ? 00:00:00 ser root 21068 21031 0 Jan09 ? 00:00:00 ser root 21070 21031 0 Jan09 ? 00:00:10 ser root 21072 21031 0 Jan09 ? 00:00:00 ser root 21075 21031 0 Jan09 ? 00:00:00 ser root 21076 21031 0 Jan09 ? 00:00:00 ser root 21077 21031 0 Jan09 ? 00:00:00 ser root 21078 21031 0 Jan09 ? 00:00:00 ser root 21079 21031 0 Jan09 ? 00:00:00 ser root 21080 21031 0 Jan09 ? 00:00:00 ser root 21081 21031 0 Jan09 ? 00:00:00 ser root 21082 21031 0 Jan09 ? 00:00:00 ser root 21083 21031 0 Jan09 ? 00:00:00 ser root 21084 21031 0 Jan09 ? 00:00:00 ser root 17014 31475 0 19:24 pts/2 00:00:00 grep ser

3 3

[Serusers] SER + Asterisk Voice Mail, delay before redirect
by inge 12 Jan '07

12 Jan '07

Hi all, I work on an existing configuration of SER. I'm not able to find the parameter wich define the time in seconds before SER redirect call to Asterisk Voice Mail. In fact, I want to change this timing to put it in SER DB with a usr_preferences table like. Someone have an idea about the kind of line which could be define it ? Thanks for your help.

3 6

[Users] adding aliases does not work
by Cristian Caprar 12 Jan '07

12 Jan '07

Hi, Adding aliases with openserctl does not seem to be working anymore in the latest CVS snapshot. I run the command: /usr/local/sbin/openserctl alias add 00162123456 sip:user@domain.local And the output is: database engine 'MYSQL' loaded Control engine 'FIFO' loaded is_user: user counter=0 check_db_alias: alias counter=0 entering fifo_cmd ul_add aliases 00162123456(a)domain.local sip:user@domain.local 0 1.00 0 0 4294967295 ERROR: Error opening openser's FIFO /tmp/openser_fifo ERROR: Make sure you have line fifo=/tmp/openser_fifo in your config The same command used to work with a snapshot I got around 20 dec. OpenSER starts correctly. Thanks, Cristian

2 2

[Users] Meaning of CS_DIRTY
by Bill Neely 12 Jan '07

12 Jan '07

I get the follwing response from openserctl show ul ~~~Contact(0x284a6880)~~~ domain : 'location' aor : '1020111' Contact : 'sip:1020111@192.168.0.198:5060' Expires : 3592 q : Call-ID : '4ef617d2-8e70ad17(a)192.168.0.198' CSeq : 4249 User-Agent: 'Linksys/SPA3102-5.1.5(GWa)' received : 'sip:67.188.231.188:62292' Path : '' State : CS_DIRTY Flags : 0 Sock : 66.224.20.35:5060 (0x8123fb0) Methods : 5183 next : 0x0 prev : 0x0 ~~~/Contact~~~~ Can anybody tell me the meaning of the CS_DIRTY state reply? -- 702-874-3833 1-866-553-3833

2 1

[Serusers] ser logging into the file? - ser 0.10.99
by TZieleniewski 12 Jan '07

12 Jan '07

Hi ! Is there in ser 0.10.99 a way to send the output to the file instead of syslog?? Previously for ser 0.9.7 I had it done throught the serctl tool. I had the logstderror=yes in my ser.cfg and the output was redirected to log file pointed by SER_LOGFILE inside serctl. Tomasz

4 4

[Serusers] SER with LCR Config
by Arun Kumar 12 Jan '07

12 Jan '07

Hi All, I've posted the working config of LCR with SER and NAT on voipbhiwani.blogspot.com b'coz list is not allowing this much big config file. thanks Arun

1 0

Re: [Serusers] ser-0.10.99-dev64 (ser_ctl)]
by TZieleniewski 11 Jan '07

11 Jan '07

jean sigue napisał(a): > I create the database but the credentials table replaces the old > subscriber table 0.9.6 > > >> From: Hendrik Scholz <hendrik.scholz(a)freenet-ag.de> >> To: jean sigue <jeansigue(a)hotmail.com> >> CC: serusers(a)lists.iptel.org >> Subject: Re: [Serusers] ser-0.10.99-dev64 (ser_ctl) >> Date: Thu, 11 Jan 2007 14:46:36 +0100 >> >> Hi! >> >> jean sigue wrote: >> > MySql password: >> > ERROR 1146 (42S02) at line 1: Table 'ser.subscriber' doesn't exist >> >> Did you create the MySQL tables? >> ser_mysql.sh create (or the *.sql directly) should create them. >> Either way make sure you populate the domain table as well. >> >> Just my $.02, >> Hendrik >> >> -- >> freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114 Kiel, Germany >> Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559 >> Internet: http://www.freenet.de, eMail: hendrik.scholz(a)freenet-ag.de > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > create the database with -n option (name of the new database) the same db user but new database for other ser instance there are some other usefull options too:)

1 0

[Serusers] ser-0.10.99-dev64 (ser_ctl)
by jean sigue 11 Jan '07

11 Jan '07

Hi, I am trying to create a sip account. When I enter the command [root@itiah runkit-0.9]# serctl add 7777777(a)vaboomz.com mypass ge(a)vaboomz.com MySql password: ERROR 1146 (42S02) at line 1: Table 'ser.subscriber' doesn't exist error: 500 Command 'ul_show_contact' not found Can anyome tell me where to find ser_ctl or more importantly how to fix it .

2 2

RE: [Users] using multiple DNS servers
by Will Quan 11 Jan '07

11 Jan '07

Yes, I think its DNS related because the callflows look perfect when both my DNS servers are up and running. This scenario only occurs when the primary dns server is offline. I thought maybe my debug trace would tell more of the story. Sorry I dont have a ngrep trace. << primary dns is offline; assuming res_search() considers the secondary in resolv.conf >>Jan 10 20:55:20 www openser[21124]: get_record: lookup(5934.rdtw1.qa2.l1.voiplabs.net, 35) failedJan 10 20:55:20 www openser[21124]: DEBUG:sip_resolvehost2: no valid NAPTR record found for 5934.rdtw1.qa2.l1.voiplabs.net, trying direct SRV lookup...Jan 10 20:55:20 www openser[21346]: get_record: lookup(5934.rdtw1.qa2.l1.voiplabs.net, 35) failedJan 10 20:55:20 www openser[21346]: DEBUG:sip_resolvehost2: no valid NAPTR record found for 5934.rdtw1.qa2.l1.voiplabs.net, trying direct SRV lookup...Jan 10 20:55:20 www openser[21140]: is_from_user_enum(): order 100, pref 100, flen 1, flags 'u', slen 7, services 'SIP+E2U', rlen 43, regexp '!^(.*)$!sip:\1@5934.rdtw1.qa2.l1.voiplabs.net!'Jan 10 20:55:20 www openser[21140]: reg_replace(): pattern: '^(.*)$', replacement: 'sip:\1@5934.rdtw1.qa2.l1.voiplabs.net', string: '+14556660002'Jan 10 20:55:20 www openser[21140]: is_from_user_enum(): resulted in replacement: 'sip:+14556660002@5934.rdtw1.qa2.l1.voiplabs.net'Jan 10 20:55:20 www openser[21140]: DEBUG:sip_resolvehost2: no port, no proto -> do NAPTR lookup!Jan 10 20:55:21 www openser[21233]: get_record: lookup(5934.rdtw1.qa2.l1.voiplabs.net, 35) failedJan 10 20:55:21 www openser[21233]: DEBUG:sip_resolvehost2: no valid NAPTR record found for 5934.rdtw1.qa2.l1.voiplabs.net, trying direct SRV lookup...<<processing 200 OK (invite) for the first time >>Jan 10 20:55:22 www openser[21124]: SIP Reply (status):Jan 10 20:55:22 www openser[21124]: version: <SIP/2.0>Jan 10 20:55:22 www openser[21124]: status: <200>Jan 10 20:55:22 www openser[21124]: reason: <OK>Jan 10 20:55:22 www openser[21124]: parse_headers: flags=2Jan 10 20:55:22 www openser[21124]: Found param type 232, <branch> = <z9hG4bK7d34.d8cbc746.0>; state=16Jan 10 20:55:22 www openser[21124]: end of header reached, state=5Jan 10 20:55:22 www openser[21124]: parse_headers: Via found, flags=2Jan 10 20:55:22 www openser[21124]: parse_headers: this is the first viaJan 10 20:55:22 www openser[21124]: After parse_msg...Jan 10 20:55:22 www openser[21124]: forward_reply: found module tm, passing reply to itJan 10 20:55:22 www openser[21124]: DEBUG: t_check: msg id=19 global id=18 T start=0xffffffffJan 10 20:55:22 www openser[21124]: parse_headers: flags=22Jan 10 20:55:22 www openser[21124]: Found param type 232, <branch> = <z9hG4bK175-64.101.130.210>; state=16Jan 10 20:55:22 www openser[21124]: end of header reached, state=5Jan 10 20:55:22 www openser[21124]: parse_headers: Via found, flags=22Jan 10 20:55:22 www openser[21124]: parse_headers: this is the second viaJan 10 20:55:22 www openser[21124]: DEBUG: add_param: tag=35Jan 10 20:55:22 www openser[21124]: DEBUG:parse_to:end of header reached, state=29Jan 10 20:55:22 www openser[21124]: DBUG:parse_to: display={"18001033799"}, ruri={sip:+18001033799@rdtw1.qa2.l1.voiplabs.net}Jan 10 20:55:22 www openser[21124]: DEBUG: get_hdr_field: <To> [64]; uri=[sip:+18001033799@rdtw1.qa2.l1.voiplabs.net] Jan 10 20:55:22 www openser[21124]: DEBUG: to body ["18001033799" <sip:+18001033799@rdtw1.qa2.l1.voiplabs.net>]Jan 10 20:55:22 www openser[21124]: get_hdr_field: cseq <CSeq>: <1> <INVITE>Jan 10 20:55:22 www openser[21124]: parse_headers: flags=8Jan 10 20:55:22 www openser[21124]: DEBUG: t_reply_matching: hash 17367 label 1685896333 branch 0Jan 10 20:55:22 www openser[21124]: DEBUG:tm:REF_UNSAFE: after is 1Jan 10 20:55:22 www openser[21124]: DEBUG: t_reply_matching: reply matched (T=0x283fe3a8)!Jan 10 20:55:22 www openser[21124]: parse_headers: flags=8Jan 10 20:55:22 www openser[21124]: DBG: trans=0x283fe3a8, callback type 2, id 0 enteredJan 10 20:55:22 www openser[21124]: parse_headers: flags=8Jan 10 20:55:22 www openser[21124]: DEBUG: t_check: msg id=19 global id=19 T end=0x283fe3a8Jan 10 20:55:22 www openser[21124]: DEBUG:tm:reply_received: org. status uas=180, uac[0]=180 local=0 is_invite=1)Jan 10 20:55:22 www openser[21124]: DEBUG:tm:t_should_relay_response: T_code=180, new_code=200Jan 10 20:55:22 www openser[21124]: DEBUG:tm:relay_reply: branch=0, save=0, relay=0Jan 10 20:55:22 www openser[21124]: old size: 844, new size: 783Jan 10 20:55:22 www openser[21124]: build_res_from_sip_res: copied size: orig:289, new: 228, rest: 555 msg= SIP/2.0 200 OK^M Record-Route: <sip:10.88.128.76;lr;ftag=64.101.130.210>^M Record-Route: <sip:10.88.128.77;lr;ftag=64.101.130.210>^M Record-Route: <sip:10.88.128.74;lr;ftag=64.101.130.210>^M Record-Route: <sip:64.101.130.210:5062>^M Via: SIP/2.0/UDP 64.101.130.210:5062;branch=z9hG4bK175-64.101.130.210^M From: "14556660002" <sip:+14556660002@5934.rdtw1.qa2.l1.voiplabs.net>;tag=64.101.130.210^M To: "18001033799" <sip:+18001033799@rdtw1.qa2.l1.voiplabs.net>;tag=35^M Call-ID: 1-1140(a)64.101.130.210^M CSeq: 1 INVITE^M Max-Forwards: 70^M Content-Length: 145^M Contact: "18001033799" <sip:+118001033799@10.88.129.252:6001>^M Content-Type: application/sdp^M ^M v=0^M o=user1 53655765 2353687637 IN IP4 10.88.129.252^M s=SIP Call^M c=IN IP4 10.88.129.252^M t=0 0^M m=audio 10800 RTP/AVP 0^M a=rtpmap:0 PCMU/8000^M Jan 10 20:55:22 www openser[21124]: DEBUG: update_totag_set: new totag Jan 10 20:55:22 www openser[21124]: DEBUG: add_to_tail_of_timer[2]: 0x283fe3f0Jan 10 20:55:22 www openser[21124]: DEBUG:tm:relay_reply: sent buf=0x81351c8: SIP/2.0 2..., shmem=0x283fe090: SIP/2.0 2Jan 10 20:55:22 www openser[21124]: DBG: trans=0x283fe3a8, callback type 128, id 0 enteredJan 10 20:55:22 www openser[21124]: ACC: transaction answered: method=INVITE, i-uri=sip:+18001033799@rdtw1.qa2.l1.voiplabs.net, o-uri=sip:+18001033799@10.88.128.77:5060, call_id=1-1140(a)64.101.130.210, from="14556660002" <sip:+14556660002@5934.rdtw1.qa2.l1.voiplabs.net>;tag=64.101.130.210, code=200, uid=n/a, fromtag=64.101.130.210, fromuri=sip:+14556660002@5934.rdtw1.qa2.l1.voiplabs.net, fromuri=sip:+14556660002@5934.rdtw1.qa2.l1.voiplabs.net, fromuser=+14556660002, fromdomain=5934.rdtw1.qa2.l1.voiplabs.net, to="18001033799" <sip:+18001033799@rdtw1.qa2.l1.voiplabs.net>;tag=35, totag=35, touri=sip:+18001033799@rdtw1.qa2.l1.voiplabs.net, touser=+18001033799, userpart=+18001033799, domain=rdtw1.qa2.l1.voiplabs.netJan 10 20:55:22 www openser[21124]: DEBUG: cleanup_uac_timers: RETR/FR timers resetJan 10 20:55:22 www openser[21124]: DEBUG:tm:UNREF_UNSAFE: after is 0Jan 10 20:55:22 www openser[21124]: DEBUG:destroy_avp_list: destroying list 0x0Jan 10 20:55:22 www openser[21124]: receive_msg: cleaning up<< processing ACK from orig >>Jan 10 20:55:22 www openser[21124]: SIP Request:Jan 10 20:55:22 www openser[21124]: method: <ACK>Jan 10 20:55:22 www openser[21124]: uri: <sip:+18001033799@10.88.129.253>Jan 10 20:55:22 www openser[21124]: version: <SIP/2.0>Jan 10 20:55:22 www openser[21124]: parse_headers: flags=2Jan 10 20:55:22 www openser[21124]: end of header reached, state=5Jan 10 20:55:22 www openser[21124]: parse_headers: Via found, flags=2Jan 10 20:55:22 www openser[21124]: parse_headers: this is the first viaJan 10 20:55:22 www openser[21124]: After parse_msg...Jan 10 20:55:22 www openser[21124]: preparing to run routing scripts...Jan 10 20:55:22 www openser[21124]: DEBUG : sl_filter_ACK: to late to be a local ACK!Jan 10 20:55:22 www openser[21124]: parse_headers: flags=100Jan 10 20:55:22 www openser[21124]: DEBUG: add_param: tag=35Jan 10 20:55:22 www openser[21124]: DEBUG:parse_to:end of header reached, state=29Jan 10 20:55:22 www openser[21124]: DBUG:parse_to: display={"18001033799"}, ruri={sip:+18001033799@rdtw1.qa2.l1.voiplabs.net}Jan 10 20:55:22 www openser[21124]: DEBUG: get_hdr_field: <To> [64]; uri=[sip:+18001033799@rdtw1.qa2.l1.voiplabs.net] Jan 10 20:55:22 www openser[21124]: DEBUG: to body ["18001033799" <sip:+18001033799@rdtw1.qa2.l1.voiplabs.net>]Jan 10 20:55:22 www openser[21124]: get_hdr_field: cseq <Cseq>: <2> <ACK>Jan 10 20:55:22 www openser[21124]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Jan 10 20:55:22 www openser[21124]: parse_headers: flags=200Jan 10 20:55:22 www openser[21124]: is_preloaded: NoJan 10 20:55:22 www openser[21124]: grep_sock_info - checking if host==us: 13==12 && [10.88.129.253] == [10.88.128.74]Jan 10 20:55:22 www openser[21124]: grep_sock_info - checking if port 5060 matches port 5060Jan 10 20:55:22 www openser[21124]: check_self: host != meJan 10 20:55:22 www openser[21124]: grep_sock_info - checking if host==us: 12==12 && [10.88.128.74] == [10.88.128.74]Jan 10 20:55:22 www openser[21124]: grep_sock_info - checking if port 5060 matches port 5060Jan 10 20:55:22 www openser[21124]: after_loose: Topmost route URI: 'sip:10.88.128.74;lr;ftag=64.101.130.210' is meJan 10 20:55:22 www openser[21124]: after_loose: URI to be processed: 'sip:10.88.128.77;lr;ftag=64.101.130.210'Jan 10 20:55:22 www openser[21124]: after_loose: Next URI is a loose routerJan 10 20:55:22 www openser[21124]: DEBUG: t_newtran: msg id=20 , global msg id=19 , T on entrance=0xffffffffJan 10 20:55:22 www openser[21124]: parse_headers: flags=ffffffffffffffffJan 10 20:55:22 www openser[21124]: DEBUG: get_hdr_body : content_length=0Jan 10 20:55:22 www openser[21124]: found end of headerJan 10 20:55:22 www openser[21124]: parse_headers: flags=78Jan 10 20:55:22 www openser[21124]: t_lookup_request: start searching: hash=17364, isACK=1Jan 10 20:55:22 www openser[21124]: DEBUG: proceeding to pre-RFC3261 transaction matchingJan 10 20:55:22 www openser[21124]: DEBUG: t_lookup_request: no transaction foundJan 10 20:55:22 www openser[21124]: DEBUG:tm:t_relay: forwarding ACKJan 10 20:55:22 www openser[21124]: DEBUG: mk_proxy: doing DNS lookup...Jan 10 20:55:22 www openser[21124]: check_via_address(64.101.130.210, 64.101.130.210, 0)Jan 10 20:55:22 www openser[21124]: Sending: ACK sip:+18001033799@10.88.129.253 SIP/2.0^M Via: SIP/2.0/UDP 10.88.128.74;branch=0^M Via: SIP/2.0/UDP 64.101.130.210:5062^M From: "14556660002" <sip:+14556660002@5934.rdtw1.qa2.l1.voiplabs.net>;tag=64.101.130.210^M To: "18001033799" <sip:+18001033799@rdtw1.qa2.l1.voiplabs.net>;tag=35^M Call-ID: 1-1140(a)64.101.130.210^M Cseq: 2 ACK^M Contact: "14556660002" <sip:14556660002@64.101.130.210>^M Max-Forwards: 69^M Record-Route: <sip:64.101.130.210:5062>^M Route: <sip:10.88.128.77;lr;ftag=64.101.130.210>,<sip:10.88.128.76;lr;ftag=64.101.130.210>,"18001033799" <sip:+118001033799@10.88.129.252:6001>^M Content-Length: 0^M ^M .Jan 10 20:55:22 www openser[21124]: orig. len=608, new_len=605, proto=1Jan 10 20:55:22 www openser[21124]: DEBUG:destroy_avp_list: destroying list 0x0Jan 10 20:55:22 www openser[21124]: receive_msg: cleaning up<< strange behavior starts here ... >>Jan 10 20:55:22 www openser[21346]: DEBUG:do_srv_lookup: SRV(_sip._udp.5934.rdtw1.qa2.l1.voiplabs.net) = proxy.5934.rdtw1.qa2.l1.voiplabs.net:5060Jan 10 20:55:22 www openser[21346]: rdtw1.qa2.l1.voiplabs.net-outbound: time_t=1168484116 ^LINE=180 ^FILE=/var/local/ser/ROUTLET/config.pre ^call_id=1-1140(a)64.101.130.210 ^cseq=1 ^contact="14556660002" <sip:+14556660002@64.101.130.210> ^from=sip:+14556660002@5934.rdtw1.qa2.l1.voiplabs.net ^fromtag=64.101.130.210 ^to=sip:+18001033799@rdtw1.qa2.l1.voiplabs.net ^totag=<null> ^method=INVITE ^ruri=sip:+18001033799@rdtw1.qa2.l1.voiplabs.net ^messageid=8^remark=sJan 10 20:55:22 www openser[21346]: rdtw1.qa2.l1.voiplabs.net-RECORDROUTE: time_t=1168484116 ^LINE=180 ^FILE=/var/local/ser/ROUTLET/config.pre ^call_id=1-1140(a)64.101.130.210 ^cseq=1 ^contact="14556660002" <sip:+14556660002@64.101.130.210> ^from=sip:+14556660002@5934.rdtw1.qa2.l1.voiplabs.net ^fromtag=64.101.130.210 ^to=sip:+18001033799@rdtw1.qa2.l1.voiplabs.net ^totag=<null> ^method=INVITE ^ruri=sip:+18001033799@10.88.128.77:5060 ^messageid=8^remark=rdtw1.qa2.l1.voiplabs.netJan 10 20:55:22 www openser[21346]: DEBUG: t_newtran: msg id=8 , global msg id=7 , T on entrance=0xffffffffJan 10 20:55:22 www openser[21346]: parse_headers: flags=ffffffffffffffffJan 10 20:55:22 www openser[21346]: parse_headers: flags=78Jan 10 20:55:22 www openser[21346]: t_lookup_request: start searching: hash=17367, isACK=0Jan 10 20:55:22 www openser[21346]: DEBUG: RFC3261 transaction matched, tid=175-64.101.130.210Jan 10 20:55:22 www openser[21346]: DEBUG:tm:REF_UNSAFE: after is 1Jan 10 20:55:22 www openser[21346]: DEBUG: t_lookup_request: transaction found (T=0x283fe3a8)<< about to reTX the 200 OK (invite) *even though we just finished processing ACK* >>Jan 10 20:55:22 www openser[21346]: DEBUG:tm:t_retransmit_reply: buf=0x28229b40: SIP/2.0 2..., shmem=0x283fe090: SIP/2.0 2 ! ! ! ! ! ! ! !!!!!!!!Jan 10 20:55:22 www openser[21346]: DEBUG:tm:UNREF_UNSAFE: after is 0Jan 10 20:55:22 www openser[21346]: DEBUG:destroy_avp_list: destroying list 0x0Jan 10 20:55:22 www openser[21346]: receive_msg: cleaning up<< orig sends another ACK; to be expected since we just reTX the 200 OK >>Jan 10 20:55:22 www openser[21124]: SIP Request:Jan 10 20:55:22 www openser[21124]: method: <ACK>Jan 10 20:55:22 www openser[21124]: uri: <sip:+18001033799@10.88.129.253>Jan 10 20:55:22 www openser[21124]: version: <SIP/2.0>Jan 10 20:55:22 www openser[21124]: parse_headers: flags=2Jan 10 20:55:22 www openser[21124]: end of header reached, state=5Jan 10 20:55:22 www openser[21124]: parse_headers: Via found, flags=2Jan 10 20:55:22 www openser[21124]: parse_headers: this is the first viaJan 10 20:55:22 www openser[21124]: After parse_msg...Jan 10 20:55:22 www openser[21124]: preparing to run routing scripts...Jan 10 20:55:22 www openser[21124]: DEBUG : sl_filter_ACK: to late to be a local ACK!Jan 10 20:55:22 www openser[21124]: parse_headers: flags=100Jan 10 20:55:22 www openser[21124]: DEBUG: add_param: tag=35Jan 10 20:55:22 www openser[21124]: DEBUG:parse_to:end of header reached, state=29Jan 10 20:55:22 www openser[21124]: DBUG:parse_to: display={"18001033799"}, ruri={sip:+18001033799@rdtw1.qa2.l1.voiplabs.net}Jan 10 20:55:22 www openser[21124]: DEBUG: get_hdr_field: <To> [64]; uri=[sip:+18001033799@rdtw1.qa2.l1.voiplabs.net] Jan 10 20:55:22 www openser[21124]: DEBUG: to body ["18001033799" <sip:+18001033799@rdtw1.qa2.l1.voiplabs.net>]Jan 10 20:55:22 www openser[21124]: get_hdr_field: cseq <Cseq>: <2> <ACK>Jan 10 20:55:22 www openser[21124]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Jan 10 20:55:22 www openser[21124]: parse_headers: flags=200Jan 10 20:55:22 www openser[21124]: is_preloaded: NoJan 10 20:55:22 www openser[21124]: grep_sock_info - checking if host==us: 13==12 && [10.88.129.253] == [10.88.128.74]Jan 10 20:55:22 www openser[21124]: grep_sock_info - checking if port 5060 matches port 5060Jan 10 20:55:22 www openser[21124]: check_self: host != meJan 10 20:55:22 www openser[21124]: grep_sock_info - checking if host==us: 12==12 && [10.88.128.74] == [10.88.128.74]Jan 10 20:55:22 www openser[21124]: grep_sock_info - checking if port 5060 matches port 5060Jan 10 20:55:22 www openser[21124]: after_loose: Topmost route URI: 'sip:10.88.128.74;lr;ftag=64.101.130.210' is meJan 10 20:55:22 www openser[21124]: after_loose: URI to be processed: 'sip:10.88.128.77;lr;ftag=64.101.130.210'Jan 10 20:55:22 www openser[21124]: after_loose: Next URI is a loose routerJan 10 20:55:22 www openser[21124]: DEBUG: t_newtran: msg id=21 , global msg id=20 , T on entrance=0xffffffffJan 10 20:55:22 www openser[21124]: parse_headers: flags=ffffffffffffffffJan 10 20:55:22 www openser[21124]: DEBUG: get_hdr_body : content_length=0Jan 10 20:55:22 www openser[21124]: found end of headerJan 10 20:55:22 www openser[21124]: parse_headers: flags=78Jan 10 20:55:22 www openser[21124]: t_lookup_request: start searching: hash=17364, isACK=1Jan 10 20:55:22 www openser[21124]: DEBUG: proceeding to pre-RFC3261 transaction matchingJan 10 20:55:22 www openser[21124]: DEBUG: t_lookup_request: no transaction foundJan 10 20:55:22 www openser[21124]: DEBUG:tm:t_relay: forwarding ACKJan 10 20:55:22 www openser[21124]: DEBUG: mk_proxy: doing DNS lookup...Jan 10 20:55:22 www openser[21124]: check_via_address(64.101.130.210, 64.101.130.210, 0)Jan 10 20:55:22 www openser[21124]: Sending: ACK sip:+18001033799@10.88.129.253 SIP/2.0^M Via: SIP/2.0/UDP 10.88.128.74;branch=0^M Via: SIP/2.0/UDP 64.101.130.210:5062^M From: "14556660002" <sip:+14556660002@5934.rdtw1.qa2.l1.voiplabs.net>;tag=64.101.130.210^M To: "18001033799" <sip:+18001033799@rdtw1.qa2.l1.voiplabs.net>;tag=35^M Call-ID: 1-1140(a)64.101.130.210^M Cseq: 2 ACK^M Contact: "14556660002" <sip:14556660002@64.101.130.210>^M Max-Forwards: 69^M Record-Route: <sip:64.101.130.210:5062>^M Route: <sip:10.88.128.77;lr;ftag=64.101.130.210>,<sip:10.88.128.76;lr;ftag=64.101.130.210>,"18001033799" <sip:+118001033799@10.88.129.252:6001>^M Content-Length: 0^M ^M .Jan 10 20:55:22 www openser[21124]: orig. len=608, new_len=605, proto=1Jan 10 20:55:22 www openser[21124]: DEBUG:destroy_avp_list: destroying list 0x0Jan 10 20:55:22 www openser[21124]: receive_msg: cleaning upJan 10 20:55:23 www openser[21376]: DEBUG: timer routine:1,tl=0x283fe4d4 next=0x0Jan 10 20:55:23 www openser[21376]: DEBUG: timer routine:4,tl=0x283fe4c4 next=0x0<< repeat reTX 200 OK (invite) / ACK several times >>Any Ideas?Will> Date: Thu, 11 Jan 2007 17:23:10 +0100> From: klaus.mailinglists(a)pernau.at> To: sip(a)voiplabs.net> CC: users(a)openser.org> Subject: Re: [Users] using multiple DNS servers> > Will Quan wrote:> > I have a primary and secondary dns server setup in resolv.conf, with> > a timeout option of 1 second.The idea is when the primary fails, my> > secondary server picks up the workload.When openser is running, and> > primary dns is down, I am seeing several re-transmissions of 200 OK> > (invite) to my originator *after* openser has finished processing> > ACK. I was thinking the final responses would stop reTX after the ACK> > was t_relayed. Have you seen such behavior? And better yet- Is there> > a solution for this?Thanks, Will> > Are you sure this problem is related to DNS failover?> > regards> klaus> > > > > > > ------------------------------------------------------------------------> > > > > > _______________________________________________ Users mailing list > > Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users> > > -- > Klaus Darilion> nic.at>

1 0

[Users] NEW OPTION: "auto_aliases" core parameter
by Bogdan-Andrei Iancu 11 Jan '07

11 Jan '07

Hi everybody, I added a nee core parameter named "auto_aliases" - it controls if aliases should be automatically discovered and added during fixing listening sockets. Formally, this was done with no control over it (to turn it off), which was a security risk (in my opinion) as the server was using auto-discover aliases that were not configured. To maintain backward compatibility, the default value is "on"; Ex: auto_aliases=no or auto_aliases=0 regards, bogdan

2 4

[Users] msg_exec program in background
by Dimo 11 Jan '07

11 Jan '07

Hi, I am having trouble with msg_exec. I want it to start the execution of a command which may wait for some time but i do not need openser to wait for the command to end in order to proceed with script execution. I tried this: msg_exec("command &"); and this makes openser process hang forever with a defunct child. Is there any way someone may suggest that i can make openser run a command and not wait for it to end? nohup does not do the trick also. Best, Dimo

2 2

[Users] presence module database tables
by Cristian Caprar 11 Jan '07

11 Jan '07

Hi, I am having trouble creating the xcap_xml table of the new presence module in mysql (latest 4.1.22 on RHEL 4). The table definition from the docs is: CREATE TABLE `xcap_xml` ( `user` varchar(256) NOT NULL, `domain` varchar(256) NOT NULL, `xcap` text NOT NULL, `doc_type` varchar(64) NOT NULL, PRIMARY KEY (`user`,`domain`,`doc_type`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1; I get the following error: ERROR 1170 (42000): BLOB/TEXT column 'user' used in key specification without a key length In my opinion, the problem is the definition of the user and domain columns, which are varchar(256), instead of varchar(255) and are automatically promoted to text columns. If changing the size to 255, everything is fine. Does the presence db code do some length checks that can be affected by this change? Is the table definition correct or am I missing something? Thanks, Cristian

2 2

[Serusers] New version of SER with presence support
by raj gopal 11 Jan '07

11 Jan '07

Hi, When the new version of stable SER with complete presence support is planned for release???? Venkat

3 5

[Users] How can i add my own events on KphoneSI
by Kc Ayuba 11 Jan '07

11 Jan '07

Hi, I'm using the latest openser version with presence supports. My User Agent=kphoneSI. My question is: How can i add my own events on the KphoneSI? Kind regards, Kc

3 4

[Users] Multi Call Legs Accounting
by Howard Tang 11 Jan '07

11 Jan '07

Hi, Did anyone successfully setup call accounting for call forward? I have been searching for a working example for few days on google, I still couldn't find any working example. The config I have at the moment is modparam("acc", "multi_leg_enabled", 1) modparam("acc", "src_leg_avp_id", 110) modparam("acc", "dst_leg_avp_id", 111) In the main route, if (!method=="REGISTER" && !method=="NOTIFY") { avp_write("$from/username", "$avp(i:110)"); avp_write("$ruri/username", "$avp(i:111)"); }; Let say A call B and B forward call to C, the record in the acc table is all A to C without B in any one of the record for billing purpose. The Call forward was set on the UA. Can anyone give me a hand on this please? -- Howard Tang

2 4

[Users] Updating call-info...
by Premnath, KN 11 Jan '07

11 Jan '07

Hello, How to update call-info in openser (for an INVITE sip message)? I want the call-info to contain the actual IP-address of called party. (that we normally do for $ruri). Thank You. With warm regards, Premnath, KN The new company Nokia Siemens Networks is expected to start operations in the first quarter of 2007, subject to customary regulatory approvals, the completion of standard closing conditions, and the agreement of a number of detailed implementation steps. Disclaimer: This e-mail may contain trade secrets or privileged, undisclosed or otherwise confidential information. If you have received this e-mail in error, you are hereby notified that any review, copying or distribution of it is strictly prohibited. Please inform us immediately and destroy the original transmittal. Thank you for your cooperation.

2 5

[Serusers] No Sound when use LCR
by Arun Kumar 11 Jan '07

11 Jan '07

Hi, I've configured my SER to send calls to my Asterisk server when user connects to ser and dial a number ser forwards this to my Asterisk server and my asterisk server ask for my pin number saying "Please enter your pin number". In this config I'm using rewritehostport(ip:port) to forward the calls and its working fine. But, Now I've implemented the LCR when I see ngrep calls are going to my asterisk server but I'm not able to hear any sound from my asterisk server saying "enter your pin number". Please help in this issue. Here is my ser.cfg file : debug=3 fork=no log_stderror=yes listen=192.168.2.210 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:arun@localhost/ser" 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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/lcr.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" modparam("auth_db|permissions|uri_db|usrloc|domain", "db_url", " mysql://ser:arun@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") # modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") # modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 120) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("lcr","db_url", "mysql://ser:arun@localhost/ser") modparam("lcr","gw_table","gw") modparam("lcr","gw_name_column","gw_name") modparam("lcr","ip_addr_column","ip_addr") modparam("lcr","port_column","port") modparam("lcr","grp_id_column","grp_id") modparam("lcr","lcr_table","lcr") modparam("lcr","prefix_column","prefix") modparam("lcr","from_uri_column","from_uri") modparam("lcr","priority_column","priority") modparam("lcr","contact_avp","1401") modparam("lcr","fr_inv_timer_avp","fr_inv_timer_avp") modparam("lcr","fr_inv_timer",90) modparam("lcr","fr_inv_timer_next",60) # ------ Account and Radius modules-------------- modparam("acc", "db_url", "mysql://ser:arun@localhost/ser") modparam("acc","log_level",1) modparam("acc","log_flag",1) modparam("acc", "db_flag", 1) modparam("acc","radius_flag",1) modparam("acc","service_type",15) #modparam("acc","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; if (!load_gws()) { sl_send_reply("500", "Server Internal Error - Cannot load gateways"); break; } else { xlog("L_INFO", "\n[SER]: - [$Tf] Method: <$rm> R-uri: <$ru> From: <$fu>To <$tu> : Successfully Loaded Gateway List\n"); }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.168.2.210:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { setflag(1); end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden Cant cross 1st bridge"); break; }; if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { sl_send_reply("404", "I am trusted host, "); route(4); route(1); } else { sl_send_reply("403", "Forbidden, I am here"); }; break; }; if (method=="ACK") { setflag(1); route(1); break; } if (method=="CANCEL") { setflag(1); route(1); break; } else if (method=="INVITE") { setflag(1); route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { setflag(1); end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (!is_uri_host_local()) { route(4); route(1); break; }; if (!next_gw()) { t_reply("503", "Service not available - No more gateways"); break; } else { t_on_failure("2"); t_relay(); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } #route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- # rewritehostport("192.168.2.202:5060"); # INSERT YOUR PSTN GATEWAY IP ADDRESS # avp_write("i:45", "inv_timeout"); # route(1); #} onreply_route[1] { xlog("L_INFO", "In onreply route"); if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } failure_route[2] { if (!next_gw()) { t_reply("503", "Service not available - No more gateways"); break; } else { t_on_failure("2"); t_relay(); break; } } thanks arun

2 2

[Serusers] RTPProxy and SER, Multiple Interface Server
by Gerardo Amaya 11 Jan '07

11 Jan '07

Hello All. After investigating quite a while about this issue I'm kind of desperate now, I will appreciate your help on this. I have a SER Server with multiple interfaces, I'm able to handle all the SIP packets with the multi homed enabled. Now the part that I'm having real trouble with is the RTP audio streams. After struggling with MediaProxy I finally started using RTPProxy for this, since it has bridge mode available. Now the problem is that I have more than one network to connect audio from, so the internal and external configuration is not enough for me. So my questions are: 1.-Can I have RTPProxy Bridge mode working for all the interfaces in my server? 2.-Can I have more than one RTPProxy socket available for each of the multiple interfaces I'm connecting in order to have the audio working, so each route will use the proxy of their own network? 3.-If #2 is true, is this where the force_socket and force_rtp_proxy functions play a role in my configuration? Please help!! Thanks you so much in advance Gerardo Amaya

2 3

[Serusers] 0.10.99 - ERROR: authorize.c:359: Cannot obtain digest realm from parameter '$fd.digest_realm'
by "tzieleniewski" 11 Jan '07

11 Jan '07

HI All! I have encountered the following error and don't know waht to do with it:) ERROR: authorize.c:359: Cannot obtain digest realm from parameter '$fd.digest_realm' the situation takes places when the following sip massage arives at ser: # U 2007/01/09 11:43:02.274949 127.0.0.1:7000 -> 127.0.0.1:5060 REGISTER sip:scscf.ims.touk.pl SIP/2.0. Via: SIP/2.0/UDP 127.0.0.1:7000;rport;branch=z9hG4bK2575034C94BE793615C5BEE814464E5B. From: tzl <sip:tzl@scscf.ims.touk.pl>;tag=1829574745. To: tzl <sip:tzl@scscf.ims.touk.pl>. Contact: "tzl" <sip:tzl@127.0.0.1:7000>. Call-ID: 3E67EAEFDC8BFC73844E4D0F849DF676(a)scscf.ims.touk.pl. CSeq: 38402 REGISTER. Expires: 1800. Max-Forwards: 70. User-Agent: X-Lite release 1105d. Content-Length: 0. . My Register logic looks like this: route[REGISTER] { if (method=="REGISTER") { if (!$t.did) { sl_send_reply("403", "Register forwarding forbidden"); drop; } if (!www_authenticate("$fd.digest_realm", "credentials")) { if ($? == -2) { sl_send_reply("500", "Internal Server Error"); } else if ($? == -3) { sl_send_reply("400", "Bad Request"); } else { if ($digest_challenge) { append_to_reply("%$digest_challenge"); } sl_send_reply("401", "Unauthorized"); } drop; }; } and here are my database contensts: mysql> select * from uri; +------------------------------+--------------------------+----------+-------+--------+ | uid | did | username | flags | scheme | +------------------------------+--------------------------+----------+-------+--------+ | tzl(a)sipproxy.hellboy.touk.pl | sipproxy.hellboy.touk.pl | tzl | 61 | sip | +------------------------------+--------------------------+----------+-------+--------+ mysql> select * from domain; +--------------------------+--------------------------+-------+ | did | domain | flags | +--------------------------+--------------------------+-------+ | sipproxy.hellboy.touk.pl | sipproxy.hellboy.touk.pl | 37 | +--------------------------+--------------------------+-------+ mysql> select * from domain_attrs; +--------------------------+--------------+------+--------------------------+-------+ | did | name | type | value | flags | +--------------------------+--------------+------+--------------------------+-------+ | sipproxy.hellboy.touk.pl | sw_admin | 2 | 0 | 32 | | sipproxy.hellboy.touk.pl | digest_realm | 2 | sipproxy.hellboy.touk.pl | 32 | +--------------------------+--------------+------+--------------------------+-------+ mysql> select * from credentials; +---------------+--------------------------+--------------------------+----------+-------+----------------------------------+----------------------------------+------------------------------+ | auth_username | did | realm | password | flags | ha1 | ha1b | uid | +---------------+--------------------------+--------------------------+----------+-------+----------------------------------+----------------------------------+------------------------------+ | tzl | sipproxy.hellboy.touk.pl | sipproxy.hellboy.touk.pl | tzl | 33 | df4a49c51b082f68b792d7e2177c6ed8 | 8ba18e21048f2f587ad3386b54b99e5e | tzl(a)sipproxy.hellboy.touk.pl | | admin | sipproxy.hellboy.touk.pl | sipproxy.hellboy.touk.pl | heslo | 32 | | | 0 | +---------------+--------------------------+--------------------------+----------+-------+----------------------------------+----------------------------------+------------------------------+ Thanku for any help Bests Tomasz

6 8

[Serusers] xlite and ser
by zahra bahar 11 Jan '07

11 Jan '07

my configure in xlite:in system setting in network I put these: Auto detect IP:yes Primary stun server:blank in sip proxy I selected default and: enabled:yes display name:user username:1234 Aotorisation user:1234 password:1234 Domain/Realm:10.10.10.138 sip proxy:10.10.10.138:5060 use outband proxy:default send internal ip:Always register:default Is it any wrong?this xlite phone is on a pc that ser server is.another one is on another PC and only in username and display and password differs with this one. I am in the back of a NAT but my two clients and server are there. what is so important in setting xlite? thanks. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com --------------------------------- Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates.

2 1

[Serusers] documentation generation - ser 0.10.99
by TZieleniewski 10 Jan '07

10 Jan '07

Hi, Is there some method to generate modules documentation from the cvs source?? Im running make install-modules-doc and nothing happens?? Bests Tomasz

3 3

[Serusers] Routing Based on Codec
by Nathan E. Pralle 10 Jan '07

10 Jan '07

Has anyone ever tried routing calls based on codec in SER? If so, do you have pointers for getting started? Thanks, Nathan -- -------------------- www.nathanpralle.com --------------------

1 0

[Users] Logging levels
by Cristian Caprar 10 Jan '07

10 Jan '07

Can anyone explain the link between the debug core parameter and the logging levels used in the log functions (log and xlog)? The levels start from -3 (ALERT) and go up to 5 (DEBUG), but the parameter value is from 0 to 9. Thanks, Cristian

2 3

[Users] Multiple networks source address
by T.R. Missner 10 Jan '07

10 Jan '07

Hello, I ran into a seemingly strange issue using openser to proxy an invite from the public internet to a private ipsec tunnel on my network. The problem appears to be caused by the fact that the packet is sent out to on the correct interface ( eth1 in my case ) but still is using the source address of et0 in the ip header. This is causing my ipsec tunnel to ignore the packet. Is there anyway to control the source addr in the IP header? Using a simple python script to send a upd packet across the same path I have verified this is not a tunnel issue. Example et0: 4.X.X.229 packet arrives on this interface eth1: 10.X.X.13 packet is sent out this interface RURI is re-written and t_relay is used to send the packet to destination: 10.X.X.115 Here is a partially redacted packet as captured by tethereal: Frame 1 (982 bytes on wire, 982 bytes captured) Arrival Time: Jan 2, 2007 16:49:35.536412000 [Time delta from previous packet: 0.000000000 seconds] [Time since reference or first frame: 0.000000000 seconds] Frame Number: 1 Packet Length: 982 bytes Capture Length: 982 bytes [Frame is marked: False] [Protocols in frame: eth:ip:udp:sip:sdp] Ethernet II, Src: DellComp_f6:27:19 (00:06:5b:f6:27:19), Dst: DellComp_ec:8d:a3 (00:06:5b:ec:8d:a3) Destination: DellComp_ec:8d:a3 (00:06:5b:ec:8d:a3) Address: DellComp_ec:8d:a3 (00:06:5b:ec:8d:a3) .... ...0 .... .... .... .... = IG bit: Individual address (unicast) .... ..0. .... .... .... .... = LG bit: Globally unique address (factory default) Source: DellComp_f6:27:19 (00:06:5b:f6:27:19) Address: DellComp_f6:27:19 (00:06:5b:f6:27:19) .... ...0 .... .... .... .... = IG bit: Individual address (unicast) .... ..0. .... .... .... .... = LG bit: Globally unique address (factory default) Type: IP (0x0800) Internet Protocol, Src: 4.X.X.229 (4.X.X.229), Dst: 10.X.X.108 (10.X.X.108) ------> should source from 10.X.X.13 not 4.X.X.229 Version: 4 Header length: 20 bytes Differentiated Services Field: 0x10 (DSCP 0x04: Unknown DSCP; ECN: 0x00) 0001 00.. = Differentiated Services Codepoint: Unknown (0x04) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 968 Identification: 0x0076 (118) Flags: 0x04 (Don't Fragment) 0... = Reserved bit: Not set .1.. = Don't fragment: Set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 64 Protocol: UDP (0x11) Header checksum: 0x1dfe [correct] [Good: True] [Bad : False] Source: 4.X.X.229 (4.X.X.229) Destination: 10.X.X.108 (10.X.X.108) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 6060 (6060) Source port: 5060 (5060) Destination port: 6060 (6060) Length: 948 Checksum: 0x1c67 [incorrect, should be 0x222a] Session Initiation Protocol Request-Line: INVITE sip:+1720560XXXX@10.X.X.108:6060 SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Record-Route: <sip:4.X.X.229;lr;ftag=as0473b81f> Via: SIP/2.0/UDP 4.X.X.229;branch=z9hG4bK7a42.e0b9bac5.0 Transport: UDP Sent-by Address: 4.X.X.229 ..... Any clues on how to fix this problem? Thanks, T.R.

4 4

Re: [Serusers] Please help with LCR
by Steve Blair 10 Jan '07

10 Jan '07

Great! The only other advice I'd give you is to think about the gateway list, from matching and prefix. I ran into a few odd cases after deploying LCR where I had to define additional rules to handle URIs with slightly different syntax. -Steve Arun Kumar wrote: > Hi Steve, > > thanks very much your code is working fine now ser is able to use my > both gw and able to send calls on both gw if first fail then it will > send to other. > > thanks very much for helping me. > > thanks > arun > > On 1/10/07, *Steve Blair* <blairs(a)isc.upenn.edu > <mailto:blairs@isc.upenn.edu>> wrote: > > > Arun: > > This depends a little on your configuration. Generally speaking the > !load_gws and !next_gw would be in route block 0. You will most likely > want to do any per-call checks or rewrites before determining which > gateway should receive the call. The failure_route can be inserted > after > your route blocks. > > -Steve > > Arun Kumar wrote: > > Hi, > > > > thanks, One more thing can you please guide me where do I've to > place > > this code in my ser.cfg. > > > > thanks > > > > arun > > > > On 1/10/07, *Steve Blair * <blairs(a)isc.upenn.edu > <mailto:blairs@isc.upenn.edu> > > <mailto: blairs(a)isc.upenn.edu <mailto:blairs@isc.upenn.edu>>> wrote: > > > > > > Arun: > > > > I would try something like the following: > > > > if (!load_gws()) { > > sl_send_reply("500", "Server Internal Error - > Cannot load > > gateways"); > > break; > > } else { > > xlog("L_INFO", "\n[SER]: - [$Tf] Method: <$rm> > R-uri: <$ru> > > From: <$fu> > > To <$tu> : Successfully Loaded Gateway List\n"); > > }; > > > > ...... > > > > if (!next_gw()) { > > t_reply("503", "Service not available - No more > gateways"); > > break; > > } else { > > t_on_failure("2"); <--- This will be where you > load the > > next > > gateway > > t_relay(); > > break; > > }; > > > > failure_route[2] { > > > > if (!next_gw()) { > > xlog("L_CRIT","No more gateways for <$tu>\n"); > > t_reply("503", "Service not available - No more > gateways"); > > break; > > } else { > > t_on_failure("2"); > > t_relay(); > > break; > > } > > } > > > > -Steve > > > > Arun Kumar wrote: > > > Hi > > > > > > thanks for your prompt reply but I've already checked and I'm > > not able > > > to make my lcr working by seeing these two pages I got > this conf but > > > > > > This config is sending the call to my first gateway but > when My > > first > > > gateway is down its not sending the calls to other > gateways. Plese > > > help me to make this working. Here is my config file. > > > > > > ------------------------ > > > debug=3 > > > fork=no > > > log_stderror=yes > > > > > > listen= 192.168.2.210 <http://192.168.2.210> > <http://192.168.2.210> > > <http://192.168.2.210/> # INSERT YOUR IP > > > ADDRESS HERE > > > port=5060 > > > children=4 > > > > > > dns=no > > > rev_dns=no > > > fifo="/tmp/ser_fifo" > > > fifo_db_url="mysql://ser:pass@localhost/ser" > > > > > > 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/auth.so" > > > loadmodule "/usr/local/lib/ser/modules/auth_db.so" > > > loadmodule "/usr/local/lib/ser/modules/uri.so" > > > loadmodule "/usr/local/lib/ser/modules/uri_db.so" > > > loadmodule "/usr/local/lib/ser/modules/domain.so" > > > loadmodule "/usr/local/lib/ser/modules/nathelper.so" > > > loadmodule "/usr/local/lib/ser/modules/avpops.so" > > > loadmodule "/usr/local/lib/ser/modules/textops.so" > > > loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" > > > loadmodule "/usr/local/lib/ser/modules/permissions.so" > > > loadmodule "/usr/local/lib/ser/modules/acc.so" > > > loadmodule "/usr/local/lib/ser/modules/auth_radius.so" > > > loadmodule "/usr/local/lib/ser/modules/lcr.so" > > > > > > > > > modparam("auth_db|permissions|uri_db|usrloc|domain", > "db_url", " > > > mysql://ser:pass@localhost/ser") > > > modparam("auth_db", "calculate_ha1", 1) > > > modparam("auth_db", "password_column", "password") > > > modparam("nathelper", "rtpproxy_disable", 1) > > > modparam("nathelper", "natping_interval", 0) > > > > > > modparam("mediaproxy","natping_interval", 30) > > > modparam("mediaproxy","mediaproxy_socket", > > "/var/run/mediaproxy.sock") > > > # > > > > > > modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") > > > > # > > > > > > modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") > > > > > > > > modparam("usrloc", "db_mode", 2) > > > > > > modparam("registrar", "nat_flag", 6) > > > > > > modparam("rr", "enable_full_lr", 1) > > > > > > modparam("tm", "fr_inv_timer", 120) > > > modparam("tm", "fr_inv_timer_avp", "inv_timeout") > > > > > > modparam("permissions", "db_mode", 1) > > > modparam("permissions", "trusted_table", "trusted") > > > > > > modparam("lcr","db_url", "mysql://ser:pass@localhost/ser") > > > modparam("lcr","gw_table","gw") > > > modparam("lcr","gw_name_column","gw_name") > > > modparam("lcr","ip_addr_column","ip_addr") > > > modparam("lcr","port_column","port") > > > modparam("lcr","grp_id_column","grp_id") > > > modparam("lcr","lcr_table","lcr") > > > modparam("lcr","prefix_column","prefix") > > > modparam("lcr","from_uri_column","from_uri") > > > modparam("lcr","priority_column","priority") > > > modparam("lcr","contact_avp","1401") > > > modparam("lcr","fr_inv_timer_avp","fr_inv_timer_avp") > > > modparam("lcr","fr_inv_timer",90) > > > modparam("lcr","fr_inv_timer_next",60) > > > # ------ Account and Radius modules-------------- > > > modparam("acc", "db_url", " mysql://ser:pass@localhost/ser") > > > > > > modparam("acc","log_level",1) > > > modparam("acc","log_flag",1) > > > modparam("acc", "db_flag", 1) > > > > > > modparam("acc","radius_flag",1) > > > modparam("acc","service_type",15) > > > > > > #modparam("acc","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") > > > > > > > > > > route { > > > > > > # > > > ----------------------------------------------------------------- > > > # Sanity Check Section > > > # > > > ----------------------------------------------------------------- > > > if (!mf_process_maxfwd_header("10")) { > > > sl_send_reply("483", "Too Many Hops"); > > > break; > > > }; > > > > > > if (msg:len > max_len) { > > > sl_send_reply("513", "Message Overflow"); > > > break; > > > }; > > > > > > # > > > ----------------------------------------------------------------- > > > # Record Route Section > > > # > > > ----------------------------------------------------------------- > > > if (method=="INVITE" && client_nat_test("3")) { # > setflag(1); > > > # INSERT YOUR IP ADDRESS HERE > > > record_route_preset(" 192.168.2.210:5060 > <http://192.168.2.210:5060> > > <http://192.168.2.210:5060> > > > <http://192.168.2.210:5060/>;nat=yes"); > > > } else if (method!="REGISTER") { > > > record_route(); > > > }; > > > # > > > ----------------------------------------------------------------- > > > # Call Tear Down Section > > > # > > > ----------------------------------------------------------------- > > > if (method=="BYE" || method=="CANCEL") { setflag(1); > > > end_media_session(); > > > }; > > > > > > # > > > ----------------------------------------------------------------- > > > # Loose Route Section > > > # > > > ----------------------------------------------------------------- > > > if (loose_route()) { > > > > > > if ((method=="INVITE" || method=="REFER") && > !has_totag()) { > > > sl_send_reply("403", "Forbidden Cant cross 1st > bridge"); > > > break; > > > }; > > > > > > if (method=="INVITE") { > > > > > > if (!allow_trusted()) { > > > > > > if (!proxy_authorize("","subscriber")) { > > > proxy_challenge("","0"); > > > break; > > > } else if (!check_from()) { > > > sl_send_reply("403", "Use From=ID"); > > > break; > > > }; > > > > > > consume_credentials(); > > > }; > > > > > > if > (client_nat_test("3")||search("^Route:.*;nat=yes")){ > > > setflag(6); > > > use_media_proxy(); > > > }; > > > }; > > > > > > route(1); > > > break; > > > }; > > > # > > > ----------------------------------------------------------------- > > > # Call Type Processing Section > > > # > > > ----------------------------------------------------------------- > > > if (!is_uri_host_local()) { > > > if (is_from_local() || allow_trusted()) { > > > sl_send_reply("404", "I am trusted host, "); > > > route(4); > > > route(1); > > > } else { > > > sl_send_reply("403", "Forbidden, I am here"); > > > }; > > > break; > > > }; > > > > > > if (method=="ACK") { setflag(1); > > > route(1); > > > break; > > > } if (method=="CANCEL") { setflag(1); > > > route(1); > > > break; > > > } else if (method=="INVITE") { setflag(1); > > > # sl_send_reply("410", "I am sending your call, "); > > > route(3); > > > break; > > > } else if (method=="REGISTER") { > > > route(2); > > > break; > > > }; > > > > > > lookup("aliases"); > > > if (!is_uri_host_local()) { > > > route(4); > > > route(1); > > > break; > > > }; > > > > > > if (!lookup("location")) { > > > sl_send_reply("404", "User Not Found"); > > > break; > > > }; > > > > > > route(1); > > > } > > > > > > route[1] { > > > > > > # > > > ----------------------------------------------------------------- > > > # Default Message Handler > > > # > > > ----------------------------------------------------------------- > > > > > > t_on_reply("1"); > > > > > > if (!t_relay()) { > > > > > > if (method=="INVITE" || method=="ACK") { setflag(1); > > > end_media_session(); > > > }; > > > > > > sl_reply_error(); > > > }; > > > } > > > > > > route[2] { > > > > > > # > > > ----------------------------------------------------------------- > > > # REGISTER Message Handler > > > # > > > ----------------------------------------------------------------- > > > > > > sl_send_reply("100", "Trying"); > > > > > > if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { > > > setflag(6); > > > fix_nated_register(); > > > force_rport(); > > > }; > > > > > > if (!www_authorize("","subscriber")) { > > > www_challenge("","0"); > > > break; > > > }; > > > if (!check_to()) { > > > sl_send_reply("401", "Unauthorized"); > > > break; > > > }; > > > > > > consume_credentials(); > > > > > > if (!save("location")) { > > > sl_reply_error(); > > > }; > > > } > > > > > > route[3] { > > > > > > # > > > ----------------------------------------------------------------- > > > # INVITE Message Handler > > > # > > > ----------------------------------------------------------------- > > > > > > if (client_nat_test("3")) { > > > setflag(7); > > > force_rport(); > > > fix_nated_contact(); > > > }; > > > > > > if (!allow_trusted()) { > > > > > > if (!proxy_authorize("","subscriber")) { > > > proxy_challenge("","0"); > > > break; > > > } else if (!check_from()) { > > > sl_send_reply("403", "Use From=ID"); > > > break; > > > }; > > > > > > consume_credentials(); > > > }; > > > > > > route(4); > > > > > > if(!load_gws()){ > > > sl_send_reply("500", "Internal server error, unable to load > > gateways"); > > > break; > > > } > > > if(!next_gw()){ > > > sl_send_reply("503", "Service not available, no > gateways found"); > > > break; > > > }; > > > > > > if (!is_uri_host_local()) { > > > route(4); > > > route(1); > > > break; > > > }; > > > > > > > > > # route(1); > > > } > > > > > > route[4] { > > > > > > # > > > ----------------------------------------------------------------- > > > # NAT Traversal Section > > > # > > > ----------------------------------------------------------------- > > > > > > if (isflagset(6) || isflagset(7)) { > > > if (!isflagset(8)) { > > > setflag(8); > > > use_media_proxy(); > > > }; > > > }; > > > } > > > > > > onreply_route[1] { > > > > > > if ((isflagset(6) || isflagset(7)) && > > > (status=~"(180)|(183)|2[0-9][0-9]")) { > > > > > > if (!search("^Content-Length:[ ]*0")) { > > > use_media_proxy(); > > > }; > > > }; > > > > > > if (client_nat_test("1")) { > > > fix_nated_contact(); > > > }; > > > } > > > > > > > > > thanks very much > > > > > > arun > > > > > > On 1/10/07, * Greger V. Teigre* <greger(a)teigre.com > <mailto:greger@teigre.com> > > <mailto: greger(a)teigre.com <mailto:greger@teigre.com>> > > > <mailto: greger(a)teigre.com <mailto:greger@teigre.com> > <mailto:greger@teigre.com <mailto:greger@teigre.com>>>> wrote: > > > > > > A quick search on iptel.org <http://iptel.org> > <http://iptel.org> > > <http://iptel.org> would have given > > > you these: > > > http://www.iptel.org/lcr_module_drop_in_for_0_9_x > > > http://www.iptel.org/lcr_with_mysql_accounting > <http://www.iptel.org/lcr_with_mysql_accounting> > > <http://www.iptel.org/lcr_with_mysql_accounting> > > > > > > > > > Arun Kumar wrote: > > >> Hi > > >> > > >> please help with LCR can some one show his lcr config > file > > >> please. I'm not able to make it working. > > >> > > >> please help > > >> > > >> thanks > > >> arun > > >> > > > ------------------------------------------------------------------------ > > >> > > >> _______________________________________________ > > >> Serusers mailing list > > >> Serusers(a)lists.iptel.org > <mailto:Serusers@lists.iptel.org> <mailto:Serusers@lists.iptel.org > <mailto:Serusers@lists.iptel.org>> > > <mailto:Serusers@lists.iptel.org > <mailto:Serusers@lists.iptel.org> <mailto:Serusers@lists.iptel.org > <mailto:Serusers@lists.iptel.org>>> > > >> http://lists.iptel.org/mailman/listinfo/serusers > > >> > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Serusers mailing list > > > Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > <mailto: Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org>> > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > >

1 0

Re: [Serusers] Please help with LCR
by Steve Blair 10 Jan '07

10 Jan '07

Arun: This depends a little on your configuration. Generally speaking the !load_gws and !next_gw would be in route block 0. You will most likely want to do any per-call checks or rewrites before determining which gateway should receive the call. The failure_route can be inserted after your route blocks. -Steve Arun Kumar wrote: > Hi, > > thanks, One more thing can you please guide me where do I've to place > this code in my ser.cfg. > > thanks > > arun > > On 1/10/07, *Steve Blair * <blairs(a)isc.upenn.edu > <mailto:blairs@isc.upenn.edu>> wrote: > > > Arun: > > I would try something like the following: > > if (!load_gws()) { > sl_send_reply("500", "Server Internal Error - Cannot load > gateways"); > break; > } else { > xlog("L_INFO", "\n[SER]: - [$Tf] Method: <$rm> R-uri: <$ru> > From: <$fu> > To <$tu> : Successfully Loaded Gateway List\n"); > }; > > ...... > > if (!next_gw()) { > t_reply("503", "Service not available - No more gateways"); > break; > } else { > t_on_failure("2"); <--- This will be where you load the > next > gateway > t_relay(); > break; > }; > > failure_route[2] { > > if (!next_gw()) { > xlog("L_CRIT","No more gateways for <$tu>\n"); > t_reply("503", "Service not available - No more gateways"); > break; > } else { > t_on_failure("2"); > t_relay(); > break; > } > } > > -Steve > > Arun Kumar wrote: > > Hi > > > > thanks for your prompt reply but I've already checked and I'm > not able > > to make my lcr working by seeing these two pages I got this conf but > > > > This config is sending the call to my first gateway but when My > first > > gateway is down its not sending the calls to other gateways. Plese > > help me to make this working. Here is my config file. > > > > ------------------------ > > debug=3 > > fork=no > > log_stderror=yes > > > > listen= 192.168.2.210 <http://192.168.2.210> > <http://192.168.2.210/> # INSERT YOUR IP > > ADDRESS HERE > > port=5060 > > children=4 > > > > dns=no > > rev_dns=no > > fifo="/tmp/ser_fifo" > > fifo_db_url="mysql://ser:pass@localhost/ser" > > > > 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/auth.so" > > loadmodule "/usr/local/lib/ser/modules/auth_db.so" > > loadmodule "/usr/local/lib/ser/modules/uri.so" > > loadmodule "/usr/local/lib/ser/modules/uri_db.so" > > loadmodule "/usr/local/lib/ser/modules/domain.so" > > loadmodule "/usr/local/lib/ser/modules/nathelper.so" > > loadmodule "/usr/local/lib/ser/modules/avpops.so" > > loadmodule "/usr/local/lib/ser/modules/textops.so" > > loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" > > loadmodule "/usr/local/lib/ser/modules/permissions.so" > > loadmodule "/usr/local/lib/ser/modules/acc.so" > > loadmodule "/usr/local/lib/ser/modules/auth_radius.so" > > loadmodule "/usr/local/lib/ser/modules/lcr.so" > > > > > > modparam("auth_db|permissions|uri_db|usrloc|domain", "db_url", " > > mysql://ser:pass@localhost/ser") > > modparam("auth_db", "calculate_ha1", 1) > > modparam("auth_db", "password_column", "password") > > modparam("nathelper", "rtpproxy_disable", 1) > > modparam("nathelper", "natping_interval", 0) > > > > modparam("mediaproxy","natping_interval", 30) > > modparam("mediaproxy","mediaproxy_socket", > "/var/run/mediaproxy.sock") > > # > > > modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") > > # > > > modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") > > > > > modparam("usrloc", "db_mode", 2) > > > > modparam("registrar", "nat_flag", 6) > > > > modparam("rr", "enable_full_lr", 1) > > > > modparam("tm", "fr_inv_timer", 120) > > modparam("tm", "fr_inv_timer_avp", "inv_timeout") > > > > modparam("permissions", "db_mode", 1) > > modparam("permissions", "trusted_table", "trusted") > > > > modparam("lcr","db_url", "mysql://ser:pass@localhost/ser") > > modparam("lcr","gw_table","gw") > > modparam("lcr","gw_name_column","gw_name") > > modparam("lcr","ip_addr_column","ip_addr") > > modparam("lcr","port_column","port") > > modparam("lcr","grp_id_column","grp_id") > > modparam("lcr","lcr_table","lcr") > > modparam("lcr","prefix_column","prefix") > > modparam("lcr","from_uri_column","from_uri") > > modparam("lcr","priority_column","priority") > > modparam("lcr","contact_avp","1401") > > modparam("lcr","fr_inv_timer_avp","fr_inv_timer_avp") > > modparam("lcr","fr_inv_timer",90) > > modparam("lcr","fr_inv_timer_next",60) > > # ------ Account and Radius modules-------------- > > modparam("acc", "db_url", " mysql://ser:pass@localhost/ser") > > > > modparam("acc","log_level",1) > > modparam("acc","log_flag",1) > > modparam("acc", "db_flag", 1) > > > > modparam("acc","radius_flag",1) > > modparam("acc","service_type",15) > > > #modparam("acc","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") > > > > > > route { > > > > # > ----------------------------------------------------------------- > > # Sanity Check Section > > # > ----------------------------------------------------------------- > > if (!mf_process_maxfwd_header("10")) { > > sl_send_reply("483", "Too Many Hops"); > > break; > > }; > > > > if (msg:len > max_len) { > > sl_send_reply("513", "Message Overflow"); > > break; > > }; > > > > # > ----------------------------------------------------------------- > > # Record Route Section > > # > ----------------------------------------------------------------- > > if (method=="INVITE" && client_nat_test("3")) { # setflag(1); > > # INSERT YOUR IP ADDRESS HERE > > record_route_preset("192.168.2.210:5060 > <http://192.168.2.210:5060> > > <http://192.168.2.210:5060/>;nat=yes"); > > } else if (method!="REGISTER") { > > record_route(); > > }; > > # > ----------------------------------------------------------------- > > # Call Tear Down Section > > # > ----------------------------------------------------------------- > > if (method=="BYE" || method=="CANCEL") { setflag(1); > > end_media_session(); > > }; > > > > # > ----------------------------------------------------------------- > > # Loose Route Section > > # > ----------------------------------------------------------------- > > if (loose_route()) { > > > > if ((method=="INVITE" || method=="REFER") && !has_totag()) { > > sl_send_reply("403", "Forbidden Cant cross 1st bridge"); > > break; > > }; > > > > if (method=="INVITE") { > > > > if (!allow_trusted()) { > > > > if (!proxy_authorize("","subscriber")) { > > proxy_challenge("","0"); > > break; > > } else if (!check_from()) { > > sl_send_reply("403", "Use From=ID"); > > break; > > }; > > > > consume_credentials(); > > }; > > > > if (client_nat_test("3")||search("^Route:.*;nat=yes")){ > > setflag(6); > > use_media_proxy(); > > }; > > }; > > > > route(1); > > break; > > }; > > # > ----------------------------------------------------------------- > > # Call Type Processing Section > > # > ----------------------------------------------------------------- > > if (!is_uri_host_local()) { > > if (is_from_local() || allow_trusted()) { > > sl_send_reply("404", "I am trusted host, "); > > route(4); > > route(1); > > } else { > > sl_send_reply("403", "Forbidden, I am here"); > > }; > > break; > > }; > > > > if (method=="ACK") { setflag(1); > > route(1); > > break; > > } if (method=="CANCEL") { setflag(1); > > route(1); > > break; > > } else if (method=="INVITE") { setflag(1); > > # sl_send_reply("410", "I am sending your call, "); > > route(3); > > break; > > } else if (method=="REGISTER") { > > route(2); > > break; > > }; > > > > lookup("aliases"); > > if (!is_uri_host_local()) { > > route(4); > > route(1); > > break; > > }; > > > > if (!lookup("location")) { > > sl_send_reply("404", "User Not Found"); > > break; > > }; > > > > route(1); > > } > > > > route[1] { > > > > # > ----------------------------------------------------------------- > > # Default Message Handler > > # > ----------------------------------------------------------------- > > > > t_on_reply("1"); > > > > if (!t_relay()) { > > > > if (method=="INVITE" || method=="ACK") { setflag(1); > > end_media_session(); > > }; > > > > sl_reply_error(); > > }; > > } > > > > route[2] { > > > > # > ----------------------------------------------------------------- > > # REGISTER Message Handler > > # > ----------------------------------------------------------------- > > > > sl_send_reply("100", "Trying"); > > > > if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { > > setflag(6); > > fix_nated_register(); > > force_rport(); > > }; > > > > if (!www_authorize("","subscriber")) { > > www_challenge("","0"); > > break; > > }; > > if (!check_to()) { > > sl_send_reply("401", "Unauthorized"); > > break; > > }; > > > > consume_credentials(); > > > > if (!save("location")) { > > sl_reply_error(); > > }; > > } > > > > route[3] { > > > > # > ----------------------------------------------------------------- > > # INVITE Message Handler > > # > ----------------------------------------------------------------- > > > > if (client_nat_test("3")) { > > setflag(7); > > force_rport(); > > fix_nated_contact(); > > }; > > > > if (!allow_trusted()) { > > > > if (!proxy_authorize("","subscriber")) { > > proxy_challenge("","0"); > > break; > > } else if (!check_from()) { > > sl_send_reply("403", "Use From=ID"); > > break; > > }; > > > > consume_credentials(); > > }; > > > > route(4); > > > > if(!load_gws()){ > > sl_send_reply("500", "Internal server error, unable to load > gateways"); > > break; > > } > > if(!next_gw()){ > > sl_send_reply("503", "Service not available, no gateways found"); > > break; > > }; > > > > if (!is_uri_host_local()) { > > route(4); > > route(1); > > break; > > }; > > > > > > # route(1); > > } > > > > route[4] { > > > > # > ----------------------------------------------------------------- > > # NAT Traversal Section > > # > ----------------------------------------------------------------- > > > > if (isflagset(6) || isflagset(7)) { > > if (!isflagset(8)) { > > setflag(8); > > use_media_proxy(); > > }; > > }; > > } > > > > onreply_route[1] { > > > > if ((isflagset(6) || isflagset(7)) && > > (status=~"(180)|(183)|2[0-9][0-9]")) { > > > > if (!search("^Content-Length:[ ]*0")) { > > use_media_proxy(); > > }; > > }; > > > > if (client_nat_test("1")) { > > fix_nated_contact(); > > }; > > } > > > > > > thanks very much > > > > arun > > > > On 1/10/07, * Greger V. Teigre* <greger(a)teigre.com > <mailto:greger@teigre.com> > > <mailto: greger(a)teigre.com <mailto:greger@teigre.com>>> wrote: > > > > A quick search on iptel.org <http://iptel.org> > <http://iptel.org> would have given > > you these: > > http://www.iptel.org/lcr_module_drop_in_for_0_9_x > > http://www.iptel.org/lcr_with_mysql_accounting > <http://www.iptel.org/lcr_with_mysql_accounting> > > > > > > Arun Kumar wrote: > >> Hi > >> > >> please help with LCR can some one show his lcr config file > >> please. I'm not able to make it working. > >> > >> please help > >> > >> thanks > >> arun > >> > ------------------------------------------------------------------------ > >> > >> _______________________________________________ > >> Serusers mailing list > >> Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > <mailto:Serusers@lists.iptel.org <mailto:Serusers@lists.iptel.org>> > >> http://lists.iptel.org/mailman/listinfo/serusers > >> > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > > http://lists.iptel.org/mailman/listinfo/serusers > > > >

1 0

[Serusers] Please help with LCR
by Arun Kumar 10 Jan '07

10 Jan '07

Hi please help with LCR can some one show his lcr config file please. I'm not able to make it working. please help thanks arun

3 3

[Serusers] OT? Duplicated digits.
by Juan 10 Jan '07

10 Jan '07

My system is DIDprovider ---SER---Cisco AS5350---Calling card platform. Since December the 1st we noticed a large increase in the amount of wrong numbers received at the CC platform. As this problem arrises once the call is established and there is no RTP relay I don't think is a SER question but I resort to your major knowledge to see if someone can enlight me about where to look to solve the problem. BTW. No changes were made in the equipment at neither end. Thanks in advance Juan

2 1

[Serusers] BYE and SER.cfg
by Benito Caracuel 10 Jan '07

10 Jan '07

Hello all, In my ser.cfg I do: route{.... if (method=="BYE"){log(1,"enter in method bye")do ABCD.....} ....} When SER receives a BYE, SER doesn´t do "ABCD...." and if I put SER in debug mode, I can`t read the log "enter in method bye" but I can read the line"method: <BYE>" why doesn`t work "if method="BYE"? Thanks. _________________________________________________________________ Llama a tus amigos de PC a PC: ¡Es GRATIS! http://get.live.com/messenger/overview

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