sr-users July 2008

sr-users@lists.kamailio.org

147 participants
207 discussions

[OpenSER-Users] OpenSER as NAT traversal proxy HELP!
by Joris Dobbelsteen 30 Jul '08

30 Jul '08

Dear, I'm really trying to use OpenSER as a NAT traversal SIP proxy, since my home phone keeps breaking voice channels (the box was not intended behind NAT and I'm, of course, using a configuration that no so well supported). What is the idea: SIP transactions should travel this way: ZyXEL UA <-> SIP Proxy <-> NAT Firewall (iptables) <-> {Internet} RTP should travel this way: ZyXEL UA <-> NAT Firewall & RTPProxy <-> {Internet} My current test is using X-Lite with voipbuster, but that doesn't really work. It seems that registers are functioning, at least X-Lite reports itself being registered. Voice calls always end up in timeouts, so something is really going wrong here, it might be authentication problems? An added problem is that I have just sufficient knowledge of SIP to see what it is doing, without really knowing what to expect exactly. Furthermore I have virtually no knowledge of OpenSER. I've quite a hard time even grasping the configuration I typed in. This is not really helpful What I do know: * SIP Proxy traffic is flowing. * SIP INVITES don't work at all. * SIP to RTP is communication, but I don't know if RTP is actually flowing. I stole most of the configuration from the "04 NAT Traversal" slides of the "Italy 2007 Admin course", to which there is link on the documentation site. I adapted it to make it work with the debian supplied OpenSER 1.1. How do I get this all working? What am I getting wrong? I really really appeciate any help I can get to get it working! - Joris Config is this: # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) fork=yes # Set to no to enter debugging mode log_stderror=no # (cmd line: -E) Set to yes to enter debugging mode check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) advertised_address="82.168.191.xx" advertised_port=5060 port=5060 children=4 fifo="/tmp/openser_fifo" # # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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 "textops.so" loadmodule "nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "auth.so" loadmodule "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) # -- nathelper params --- modparam("nathelper", "rtpproxy_sock", "udp:192.168.10.6:22222") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) #modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@82.168.191.xx") # ------------------------- 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; }; # NAT detection route(2); # 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==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("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # 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("sip.familiedobbelsteen.nl", "subscriber")) { www_challenge("sip.familiedobbelsteen.nl", "0"); exit; }; if (isflagset(5)) { # set branch flag -- when someone will call this user # INVITE will have branch flag 6 set after loopup("location") setflag(6); # if you want OPTIONS natpings uncomment next # setflag(7); }; 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 (subst_uri('/(sip:.*);nat=yes/\1/i')) { setflag(6); }; if (isflagset(5) || isflagset(6)) { route(3); }; if (!t_relay()) { sl_reply_error(); }; exit; } route[2] { force_rport(); if(nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } route[3] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")) { force_rtp_proxy("", "82.168.191.xx"); t_on_failure("2"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[2] { if (isflagset(6)||isflagset(5)) { unforce_rtp_proxy(); }; } onreply_route[1] { if ((isflagset(5) || isflagset(6)) && status =~ "(183)|(2[0-9][0-9])") { force_rtp_proxy(); }; search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (isflagset(6)) { fix_nated_contact(); }; exit; }

7 17

[OpenSER-Users] reply route failure
by Fero Vlkolinsky 30 Jul '08

30 Jul '08

Hello, I am getting the folowing error on communication with one of our application servers: tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 404 This is in situation where I have two triggerings to the AS (orig and term) and term subscriber is not registered to OpenSER. OpenSER thus returns 404 not found which is then sent back to OpenSER to be forwarded to orig leg of the AS. I had the very same behavior with other AS in our lab and found out that it was because of tag added to To header in AS for the 404 response. So i removed the tag and the call worked fine. However with this AS it doesn't seem to work. Is there a way to hack pick_branch so that I see what are the existing branches and why it does not match any? Or is there any easier way to find out the reason? Thanks a lot, Fero --- reklama ----------------------------------------------------- Hľadáš DOM – BYT – POZEMOK – KANCELÁRIU ? http://www.reality.sk

2 1

[Kamailio-Users] Add a table to openser DB
by Diego Costantini 30 Jul '08

30 Jul '08

Hi, I need to use something like this (openser on a pc with 2 network cards): http://www.voip-info.org/wiki/index.php?page=SER+example+outboundproxy since I couldn't find any documentation to properly create a "location_external" table, I followed this procedure: http://www.tutorialspoint.com/mysql/mysql-clone-tables.htm Now I have the table I need, but when I start openser I get this error: Jul 30 15:49:32 [8374] ERROR:usrloc:register_udomain: Invalid table version (use openser_mysql.sh reinstall) Jul 30 15:49:32 [8374] ERROR:registrar:domain_fixup: failed to register domain Jul 30 15:49:32 [8374] ERROR:core:fix_actions: fixing failed (code=-1) at cfg line 451 Jul 30 15:49:32 [8374] CRITICAL:core:fix_expr: fix_actions error Jul 30 15:49:32 [8374] ERROR:core:main: failed to fix configuration with err code -1 Jul 30 15:49:32 [8374] NOTICE:presence:destroy: destroy module ... How can I make Openser happy to accept my new table? Diego

2 1

[Kamailio-Users] Getting SNMPStats to work
by Alex Massover 30 Jul '08

30 Jul '08

Hello! I'm trying to setup SNMPStats on Debian Sid. I have: dev5:~# cat /etc/openser/openser.cfg | grep snmp loadmodule "snmpstats.so" modparam("snmpstats", "snmpgetPath", "/usr/bin/") modparam("snmpstats", "sipEntityType", "proxyServer") modparam("snmpstats", "MsgQueueMinorThreshold", 1234) modparam("snmpstats", "MsgQueueMajorThreshold", 5678) modparam("snmpstats", "dlg_minor_threshold", 500) modparam("snmpstats", "dlg_minor_threshold", 600) dev5:~# cat /etc/snmp/snmpd.conf | grep agent master agentx agentXSocket tcp:localhost:705 dev5:~# cat /etc/snmp/snmpstats.conf agentXSocket tcp:localhost:705 dev5:~# snmpd -f -D -Dagentx -x tcp:localhost:705 2>&1 | less No log handling enabled - turning on stderr logging registered debug token agentx, 1 netsnmp_assert !"registration != duplicate" failed agent_registry.c:535 netsnmp_subtree_load() netsnmp_assert !"registration != duplicate" failed agent_registry.c:535 netsnmp_subtree_load() netsnmp_assert !"registration != duplicate" failed agent_registry.c:535 netsnmp_subtree_load() agentx_register_app_config_handler: registering .conf token for "agentxsocket" agentx_register_app_config_handler: registering .conf token for "agentxperms" agentx_register_app_config_handler: registering .conf token for "agentxRetries" agentx_register_app_config_handler: registering .conf token for "agentxTimeout" /etc/snmp/snmpd.conf: line 115: Warning: Unknown token: system.sysDescr.0. Turning on AgentX master support. agentx/config: port spec: tcp:localhost:705 agentx/master: initializing... agentx/master: initializing... DONE NET-SNMP version 5.4.1 agentx/master: unexpected callback op 4 agentx/master: handle pdu (req=0x365b5eac,trans=0x0,sess=0x0) agentx/master: open 0x86b37e8 agentx/master: opened 0x86c2028 = 6 with flags = a0 agentx/master: send response, stat 0 (req=0x365b5eac,trans=0x0,sess=0x0) But it doesn't work: dev5:~# snmpget -v2c -c public localhost openserSIPEntityType.0 OPENSER-SIP-COMMON-MIB::openserSIPEntityType.0 = No Such Object available on this agent at this OID Am I missing something? -- Best Regards, Alex Massover

1 0

[Kamailio-Users] OpenSER + Asterisk (Voicemail) Issue....
by Gerard A. Matthew 30 Jul '08

30 Jul '08

I'm having this little issue when implementing the voicemail feature. My openser.cfg looks like this in the failure route: if(!t_was_cancelled()) { revert_uri(); rewritehostport("voicemail.mydomain.com:5061"); append_branch(); #PREVENT SOME CRAZY VOICEMAIL LOOP xlog("L_INFO", "INFO: CALL TO VOICEMAIL"); setflag(10); route(1); } On my asterisk end after the time out, i'm viewing the following: SELECT * FROM sipusers WHERE name = 'XXXXXX' i.e XXXXXX = the PSTN number i'm using to call into the IP phone that's connected to OpenSER There seems to be a simple mixup with the number that is sent to asterisk. Obviously there is no user with the PSTN number, however there is one with the called number. Any idea as to what wold be causing this? Have I provided enough information? Regards, Gerard.

2 1

[OpenSER-Users] Polycom Phones produce no wringing when dialing pots lines on the pstn
by Alex G 30 Jul '08

30 Jul '08

I'm sure i'm not the first to experience this. My polycom phones when calling out a pstn gateway recieve far end ringing when dialing cellphones in the US, but when i try to call a pots line i get no ringing, only silence until the end point answers the call. Does anyone have a solution for this? From what i can see in the sip captures, the gateway sends a 180 ringing on the pots line calls, but when i call a cellphone i recieve a 183 Session Progress where the phones do provide the ringing. A polycom to polycom call shows the same 180 ringing, but in this case they do provide ringing. Just to be clear, the ringing i am talking about is the rinigng you here in the earpiece after a dial. Any insight or help would be very appreciated. I have sip captures that i can post, but opted not to incase someone has seen this and knows the answer first. Thanks so much!

3 2

Re: [Serusers] use_dns_cache and call forwarding
by caio 29 Jul '08

29 Jul '08

Hi Miklos, This happens only when I try to do a call to a subcriber having forwarding enabled. Normal calls success ok. I'm not sure about the append_branch() usage and how to work with 'on busy' and 'no answer' forwarding features. Thanks your comments.. Claudio On Thu, Jul 24, 2008 at 4:58 AM, Miklos Tirpak <miklos(a)iptel.org> wrote: > Hi, > > On 07/23/2008 05:22 PM, caio wrote: >> >> Hi guys, >> I have a problem with call forwarding (on busy or not answer).., the >> call fall into the failure_route(1) block, and AVP checking and pushto >> does its job. >> >> Here a snipped of code of failure_route(1) (flag 27 means call fwd if >> no answer): >> >> if (isflagset(27) && t_check_status("408")) { # if fwd no >> answer is set and reply msg is 408 >> >> if (avp_pushto("$ruri", "s:fwdnoanswer")) { >> avp_delete("s:fwdnoanswer"); >> resetflag(27); >> >> avp_print(); >> >> log(1, "LOG: --> appending new branch..."); >> if (!append_branch()){ >> t_reply("500", "Too many branches?!"); >> drop; >> } >> >> log(1, "LOG: --> Calling route_6..."); >> route(6); >> break; >> >> Route(6) finally calls route(1), where t_relay is called. >> But have a dns failure..and the call never is forwarded. >> >> == ser.log == >> >> DEBUG: mk_proxy: doing DNS lookup... >> get_record: lookup(_sip._udp.wsa.lab, 33) failed_ >> sip_resolvehost: no SRV record found for wsa.lab, trying 'normal' >> lookup..._ >> check_via_address(190.244.33.5, 190.244.33.5, 0)_ >> ERROR: udp_send: sendto(sock,0x2ab32599c720,910,0,0x2ab325999e68,16): >> Operation not permitted(1)_ >> msg_send: ERROR: udp_send failed_ >> ERROR: t_forward_nonack: sending request failed_ >> DEBUG: add_to_tail_of_timer[4]: 0x2ab325999e80_ >> DEBUG: add_to_tail_of_timer[0]: 0x2ab325999ea0_ >> ERROR: w_t_relay (failure mode): forwarding failed_ >> >> A question regarding this trouble.....Can I disable use_dns_cache ? >> SER version is 0.9.7. > > there is no DNS cache in version 0.9.x as I recall, but anyway, your problem > does not seem to be related to DNS, but to the UDP message sending: SER > fails to send out the UDP packet, "Operation not permitted". Is there no > firewall or routing problem at the proxy side? > > Miklos > >> What do you recommend? >> If you need my ser.cfg I can paste it... >> >> Thanks.. >> > -- caio -- caio

2 8

[Kamailio-Users] rtpproxy scalability.
by Alex Balashov 29 Jul '08

29 Jul '08

Does anyone have any reliable data on rtpproxy dimensioning, i.e. just about how many calls per box (and what kind of specs) it can really handle in actually-existing practice, not in theory? The one time I used it for NAT traversal was on a pretty beefy quad-CPU box, and I started seeing clicks and distortions in the calls beyond a load of 90 or so. However, I was never able to decisively ascertain that the cause is not the network. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

1 0

[Kamailio-Users] new list for announcements
by Henning Westerholt 29 Jul '08

29 Jul '08

The Kamailio project has now a dedicated list for announcements. There will be only important messages posted to this list, like release announcements, security advisories, informations about important changes in the project organisation and community events. The posting frequency will be problably not more then a few per month. All postings will be moderated. The address of the list is: kamailio-announce(a)lists.kamailio.org, to subscribe visit: http://lists.kamailio.org/cgi-bin/mailman/listinfo/kamailio-announce Best regards, Henning

1 0

Re: [Kamailio-Users] Prepaid system using openSER
by Darren Sessions 29 Jul '08

29 Jul '08

I’ve used SER and now OpenSER (uhhm, Kamailio) for the sip proxy and cdr generation (along with media proxy and cdrtool) and Asterisk as the ivr piece for years. Using a proxy like OpenSER/Kamailio to front your IVR servers is your best bet. The combination has worked extremely well for me. - Darren -- www.darrensessions.com On 7/28/08 6:52 PM, "Rana Dhekial" <dhekial(a)msn.com> wrote: > Hi, > > We have a requirement to build a prepaid system using OpenSER. I > took a look at CDRTool which looks quite promising. But I am not > sure how to build the Call control module and provide IVR type > information to the end users. > > If any one has done similar type of job, please respond. > > thanks, > > Rana > > Keep your kids safer online with Windows Live Family Safety. Help > protect your kids. <http://www.windowslive.com/family_safety/overview.html?ocid=TXT_TAGLM_WL_fa… > >

1 0

← Newer
1
2
3
4
5
6
7
...
21
Older →

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users July 2008