sr-users February 2007

sr-users@lists.kamailio.org

158 participants
286 discussions

[Users] Corrupted 'To' after uac_restore_from
by Mahesh Paolini-Subramanya 08 Feb '07

08 Feb '07

I'm seeing this *very* weird error where uac_restore_from is ending up with a corrupted 'From' header. The weird thing is that it happens with only one end point (a Televantage SIP Trunk). With virtually *everything* else - and I am talking about tens of thousands of end points, spanning all sorts of sip devices, we are having *no* problems. The call is set up as follows -> the endpoint sends the INVITE, which is forwarded to a PSTN gateway The ACKs and the 200 OK all have their From/To headers appropriately rewritten The problem arises when the PSTN gateway initiates the BYE As you can see below, the 'To' header gets corrupted The *only* thing I can think of is that it has something to do with the structure of the Televantage tags (the '+' signs?). Any ideas? FYI, Openser 1.1.0 (latest CVS checkout), and modparam("uac","from_restore_mode","auto") modparam("uac","rr_store_param","aptelavsf") ---INVITE--- endpoint->proxy From: "Member service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8d200003+32cc4d0e. To: <sip:17038677006@69.25.47.136>;tag=277e3aa2408f2ad73a234f5bbb99133b.c13b. ---INVITE--- proxy->gateway From: "Membersevice Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136> Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> --- 183 Session Progress--- gateway->proxy From: "Membersevice Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>;tag=1386496. Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> ---183 Session Progress--- proxy->endpoint From: "Member service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>;tag=1386496 ---200 OK--- gateway->proxy From: "Membersevice Queue"<sip:6462174165@flareon.aptela.com>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>;tag=1386496 Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> ---200 OK--- proxy->endpoint From: "Member service"<sip:user100.crunchhq@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3. To: <sip:17038677006@69.25.47.136>;tag=1386496 Record-Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> ---BYE--- gateway->proxy To: "Membersevice Queue"<sip:6462174165@69.25.47.136>;tag=10.10.35.100+1+8c020003+dd5437f3. From: <sip:17038677006@66.52.236.25>;tag=1386496. Route: <sip:69.25.47.136;lr;ftag=10.10.35.100+1+8c020003+dd5437f3;nat=yes;aptelavsf=dGVzdDciIDR0YndrISJGfHtsaWBbPWQ7NiQ4LCJlISgv> ---BYE--- proxy->endpoint To: "Membersevice Queue"<sip:user100.cru>6'(!.l asr}XV.R\[>;tag=10.10.35.100+1+8c020003+dd5437f3. From: <sip:17038677006@66.52.236.25>;tag=1386496. -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

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[Users] qop="auth" with uac_auth
by Andreas Heise 08 Feb '07

08 Feb '07

Hello, since a lot of providers has change there authentication to qop="auth" the uac_auth(); function of the uac module can't use anymore. by google I found a lot of requests for qop with uac_auth and also the official feature request 1345887. Is a target date known for a solution, maybe it's possible with 1.2.0? Feature Request [ 1345887 ] Implement qop functionality in uac module p.s.: I know it's on the road map, but it is needed so often...... thanks. Andreas

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[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.

2 1

[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.

1 0

[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 >

4 6

[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

3 4

[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

4 6

[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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sr-users February 2007