- sr-users - kamailio.org

[OpenSER-Users] Working in TCP but NOT in UDP?
by David Villasmil Govea 09 Jan '08

09 Jan '08

Hello all, I've got a working conf on which RTPs work perfectly with TCP, but fails to work with UDP... can anyont tell me why?? Here's my conf: # # sample config file to be used with nathelper/rtpproxy # # start RTPProxy with: rtpproxy -l your_public_ip -s udp:localhost:8899 # # ----------- global configuration parameters ------------------------ debug=3 # 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) port=5060 children=10 #fifo="/tmp/openser_fifo" # --- module loading mpath="/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 "dispatcher.so" loadmodule "nathelper.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "xlog.so" #loadmodule "/usr/lib/openser/modules/acc.so" #modparam("acc", "early_media", 0) #modparam("acc", "report_cancels", 1) #modparam("acc", "db_flag", 2) #modparam("acc", "db_url", "mysql://user:pwd@localhost/openser") #modparam("acc", "report_ack", 0) #modparam("acc", "db_missed_flag", 3) #modparam("acc", "failed_transaction_flag", 4) #modparam("acc", "db_extra", "from_uri=$fu ; to_uri=$tu ; from_user=$fU; source_ip=$si") # --- setting module parameters modparam("usrloc|auth_db","db_url","mysql://user:pwd@localhost/openser") # --- setting module parameters #modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") # -- usrloc params -- modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 3) # -- registrar params -- #modparam("registrar", "received_avp", "$avp(s:rcv)") # -- auth params -- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # -- rr params -- modparam("rr", "enable_full_lr", 1) # -- nathelper modparam("registrar|nathelper", "received_avp", "$avp(i:42)") modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@whatever.com") #modparam("nathelper", "received_avp", "$avp(s:rcv)") # -- xlog modparam("xlog", "buf_size", 8192) modparam("registrar", "max_contacts", 5) # --- main routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; # NAT detection route(2); if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("OPENSER_IP", "subscriber")) { www_challenge("OPENSER_IP", "0"); exit; }; if (isflagset(5)) { setbflag(6); # if you want OPTIONS natpings uncomment next # setflag(7); }; save("location"); exit; }; # # - TERMINATIONS # if (method=="INVITE") { if (!proxy_authorize("OPENSER_IP","subscriber")) { if($si=="GW_IP"){ xlog("L_NOTICE", "\n************** ALLOWING INCOMING CALL AS IT COMES FROM $si ************\n"); }else{ proxy_challenge("OPENSER_IP","0"); exit; } } if (uri=~"^sip:00"){ xlog("L_NOTICE", "\n************** GOT OUTSIDE CALL ************\n"); strip(2); prefix("M3ll4m0d4v1d"); rewritehostport("GW_IP:5060"); sl_send_reply("100","Trying"); forward(); exit(); } } if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ setbflag(6); }; if (isflagset(5)||isbflagset(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|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[1] { if (isbflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (isbflagset(6)) { fix_nated_contact(); } exit; } MANY thanks to all. David

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[OpenSER-Users] Carrirroute
by TCB 09 Jan '08

09 Jan '08

Would appreciate some help with CarrierRoute module I get the following error ERROR:carrierroute:carrier_rewrite_msg: error during rewrite_uri_recursor This is my Route route[22] { # route calls based on hash over callid # choose route domain 0 of the default carrier if(!cr_rewrite_uri("0", "call_id")){ sl_send_reply("403", "Not allowed"); } else { # In case of failure, re-route the request t_on_failure("3"); # Relay the request to the gateway t_relay(); } } -- TC

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[OpenSER-Users] INVITE, exec_dset(), and BYE
by Juan Carlos Castro y Castro 09 Jan '08

09 Jan '08

I have a scenario in which an OpenSER machine distributes load among several Asterisk machines for outgoing PSTN. I use an external program which I call through exec_dset() to select which server a call goes to. I use record_route(). All calls are completed OK, but when I dial from some SIP user-agents and hang up from the caller side, OpenSER gets crazy and doesn't know where to route the BYE to. Here's the reason: Good scenario: - I make a call to <5551234@mydomain> - OpenSER calls exec_dset, which turns it into <providercode#5551234@chosenserver> - Call is answered - I hang up - My UA sends a BYE like this: "BYE sip:providercode#5551234@chosenserver SIP/2.0" - I get an OK back, all is well. (Sometimes the # in the uri above is sent as %23, but it works either way) Bad scenario: - I make a call to <5551234@mydomain> - OpenSER calls exec_dset, which turns it into <providercode#5551234@chosenserver> - Call is answered - I hang up - My UA sends a BYE like this: "BYE sip:5551234@mydomain SIP/2.0" - My openser.cfg doesn't know how to handle that and I get a "Loop Detected" back. The destination never gets a hangup signal. Is the second UA from the "bad" scenario disrespecting the RFC? Do I have any recourse to route a BYE that comes like that to the right Asterisk server? Thanks, Juan

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[OpenSER-Users] OpenXCAP: no percent-encoded URI allowed? can't fetch element that beneath multiple-folded elements?
by KevinKinnan 09 Jan '08

09 Jan '08

The OpenXCAP 0.9.7(version 0.9.7) allows the HTTP URI that is not percent-encoded while the percent-encoded URI is rejected by the server. There is no problem of installation. Two types of http URIs are shown below: Accept: http://xcap-root/resource-lists/users/sip:alice@example.com/index/~~/resour…"friends"] Reject: http://xcap-root/resource-lists/users/sip:alice@example.com/index/~~/resour… Another problem is about fetching element that within multiple-folded elements, for example: <list name="friend"> <entry uri="sip:alice@example.com"> <display-name>Alice</display-name> </entry> <entry uri="sip:bob@example.com"> <display-name>Bob</display-name> </entry> </list> <list name="closefriends"> <entry uri="sip:tom@example.com"> <display-name>Tom</display-name> </entry> <entry uri="sip:mike@example.com"> <display-name>Mike</display-name> </entry> </list> If fetching the element of "mike", the responsed to the following URI is "404 Not found". http://xcap-root/resource-lists/users/sip:alice@example.com/index/~~/resour… However, an alternative way of uing element sequence number works. http://xcap-root/resource-lists/users/sip:alice@example.com/index/~~/resour… Hope anyone can give me some clues, thanks a billion Regards, Kevin _________________________________________________________________ MSN 中文网，最新时尚生活资讯，白领聚集门户。 http://cn.msn.com

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[OpenSER-Users] Working in TCP but NOT in UDP?
by David Villasmil Govea 09 Jan '08

09 Jan '08

> > > Hello all, > > > > I've got a working conf on which RTPs work perfectly with TCP, but > > fails to work with UDP... can anyont tell me why?? > > Could you detail more what the issue is? Do you mean that the SIP works > fine > over TCP and fails over UDP? RTP has nothing to do here. > > > -- > I?aki Baz Castillo > Hello, That's exactly what I mean, signalling being TCP or UPD got nothing to do here.... but it DOES work on TCP and NOT over UDP! Weird... David

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Re: [OpenSER-Users] how to restrict invite when regiest unsuccessful
by liu york 09 Jan '08

09 Jan '08

yes, when i used same authenticating on both regiest and invite.the problem was resolved.many many thx. ======= 2008-01-08 15:16:35 您在来信中写道：======= > > >Alex Balashov schrieb: >> On Tue, 8 Jan 2008, liu york wrote: >> >>> I want to restrict invite when regiest unsuccessful even the invite >>> request have right authentication. actually,i just want to only one >>> person can call to pstn,others cant call even have sipid and >>> password.now i has set by modparam("registrar", "max_contacts", 10).but >>> cant restrict the invite by the sipphone no need regiestion. >> >> Are you authenticating INVITEs differently than you are authenticating >> registrations? In other words, are the authentication realm and required >> credentials different for your proxy_challenge() than they are for >> your www_challenge() for handling REGISTER requests? >> >> If they are, it seems that all you should have to do is do a lookup() >> and see if it fails. Presumably, you are doing a save() into your >> registrar/usrloc only on valid REGISTER challenges? >> >> >> i.e. >> >> if(is_method("INVITE")) { >> if(!lookup("location")) { > > if(!registered("location")) { >is sufficient > >btw: this does not give you any security > >regards >klaus > > >> sl_send_reply("403", "Forbidden -- REGISTER invalid."); >> exit; >> } >> } >> >> >> -- >> Alex Balashov >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : +1-678-954-0670 >> Direct : +1-678-954-0671 >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　york 　　　　　　　　york(a)goldentek.biz 　　　　　　　　　　2008-01-09

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Re: [OpenSER-Users] CDRTool help?
by Andy Smith 08 Jan '08

08 Jan '08

Sorry, not all of the text I originally sent has appeared on the list archives, so here is the full email another time! :P Hi, can anyone provide any advice on getting CDRTool to work? I have installed CDRTool 6.0 on a Debian Linux system 2.6 Kernel with Apache/2.2.6 (Unix) PHP/5.2.5 I am attempting to get it to work with OpenSER only (I have commented out the Asterisk and sip_trace sections in global.inc) and I am recieving the error: Error initializing CDRTool datasource >From in my web browser. It seems that in callsearch.phtml "$cdr_source" is not being set, but I dont understand why not and I am not sure how to get more debugging info out of the whole thing (Im not used to PHP), I dont see any errors in the syslog :( With regards global.inc, I believe all database usernames and passwords are setup and have successfully tested connecting to all databases with this data - and I believe all db_class sections have the corresponding sections as required (the username, password, tablename etc). thanks for any help! Andy.

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Re: [OpenSER-Users] CDRTool help?
by Andy Smith 08 Jan '08

08 Jan '08

mmm, again! :S well here is the bottom half of my email.... It seems that in callsearch.phtml "$cdr_source" is not being set, but I dont understand why not and I am not sure how to get more debugging info out of the whole thing (Im not used to PHP), I dont see any errors in the syslog :( With regards global.inc, I believe all database usernames and passwords are setup and have successfully tested connecting to all databases with this data - and I believe all db_class sections have the corresponding sections as required (the username, password, tablename etc). thanks for any help! Andy.

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[OpenSER-Users] CDRTool help?
by Andy Smith 08 Jan '08

08 Jan '08

Hi, can anyone provide any advice on getting CDRTool to work? I have installed CDRTool 6.0 on a Debian Linux system 2.6 Kernel with Apache/2.2.6 (Unix) PHP/5.2.5 I am attempting to get it to work with OpenSER only (I have commented out the Asterisk and sip_trace sections in global.inc) and I am recieving the error: Error initializing CDRTool datasource >From in my web browser. It seems that in callsearch.phtml "$cdr_source" is not being set, but I dont understand why not and I am not sure how to get more debugging info out of the whole thing (Im not used to PHP), I dont see any errors in the syslog :( With regards global.inc, I believe all database usernames and passwords are setup and have successfully tested connecting to all databases with this data - and I believe all db_class sections have the corresponding sections as required (the username, password, tablename etc). thanks for any help! Andy.

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[OpenSER-Users] Problem adding permanent user locations using openserctl
by Ashley Unitt 08 Jan '08

08 Jan '08

I've recently installed a vanilla OpenSER v1.3.0 (a new install) and I'm trying to set up permanent user locations using openserctl as I have some devices that won't register on their own. I am currently running without a database. When trying to add a user using the command line "openserctl ul add 100 sip:192.168.1.40" I get the error: ERROR: 400; check if you use aliases in OpenSER. Can someone please give me some pointers on what to try? Thanks Ashley I've modified the openserctlrc as follows: # $Id: openserctlrc,v 1.2 2006/07/05 19:37:20 miconda Exp $ # # openser control tool resource file # # here you can set variables used in the openserctl ## your SIP domain SIP_DOMAIN=openser.development.newvoicemedia.com ## database type: MYSQL or PGSQL, by defaulte none is loaded # DBENGINE=MYSQL ## database host # DBHOST=localhost ## database name # DBNAME=openser ## database read/write user # DBRWUSER=openser ## database read only user # DBROUSER=openserro ## password for database read only user # DBROPW=openserro ## database super user # DBROOTUSER="root" ## type of aliases used: DB - database aliases; UL - usrloc aliases ## - default: none ALIASES_TYPE=""UL ## control engine: FIFO or UNIXSOCK ## - default FIFO # CTLENGINE="FIFO" ## path to FIFO file # OSER_FIFO="FIFO" ## check ACL names; default on (1); off (0) # VERIFY_ACL=1 ## ACL names - if VERIFY_ACL is set, only the ACL names from below list ## are accepted # ACL_GROUPS="local ld int voicemail free-pstn" ## verbose - debug purposes - default '0' VERBOSE=1 Free Phone: 0800 280 2888 Tel: +44 (0)1256 892020 Fax: +44 (0)1256 892200 Registered Office: NewVoiceMedia Ltd, Laverstoke Grange, Laverstoke, Whitchurch, Hants, RG28 7PF NewVoiceMedia Registered in England No: 3602868

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[OpenSER-Users] Openser + Mediaproxy behind NAT
by Halomoan Chow 08 Jan '08

08 Jan '08

Hi Is it possible to put Openser + Mediaproxy behind NAT while serving internal phones and external phones ? Internal phone <-> Openser+MediaProxy <-> NAT <-> external phone (maybe behind nat also) Openser is act as a registrar in this case. If you have successful story with this setting, please share. Thank you so much Regards, Halomoan

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[OpenSER-Users] [OT] Error in Rome VON material about OpenSer and Asterisk Realtime
by Iñaki Baz Castillo 08 Jan '08

08 Jan '08

Hi, just as curiosity I comment that there is an error (an Asterisk error in fact) in the material of the Rome VON course of 2007: In file "config/08-asterisk.conf" it appears: CREATE VIEW sipusers AS SELECT username as name, [...] domain as defaultip, domain as ipaddr, [...] FROM openser.subscriber; There is a bug in case of SIP SRV domains because Asterisk doesn't perform a SRV DNS query for the location of registerd users (just for outbound calls if "srvlookup=yes") so if OpenSer has a SRV SIP domain "my_srv_domain.com" but DNS type A of that domain points to a different IP then Asterisk will fail (it will resolve to the A DNS entry). A simple solution is to modify the view as follows: CREATE VIEW sipusers AS SELECT username as name, [...] __OpenSer_IP__ as defaultip, __OpenSer_IP__ as ipaddr, [...] FROM openser.subscriber; Of course this is a bug in Asterisk since it should, according to RFC3261, perform a SRV query for the location of registered users in case they are not an IP. Just it ;) Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Question about append-hf and remove_hf
by Helmut Kuper 08 Jan '08

08 Jan '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I add some private headerfields (append-hf) during main route resp. sub-routes like "route(3)" and in certain cases I want to remove (remove_hf) some of them in main route befor sending request out. Unfortunately that doesn't work. Even when I add a header field and test for it directly afterwards or try to remove it, openser doesn't find anything. Test-Code is like this: append_hf("P-SI: sad\r\n"); if(is_present_hf("P-SI")) xlog("L_INFO", "######## Found\n"); if(remove_hf("P-SI")) xlog("L_INFO", "######## Removed\n"); textops doc has a hint at append_hf: " ... Note: Headers which are added in main route cannot be removed in further routes (e.g. failure routes). So, the idea is not to add there any headers that you might want to remove later. To add headers temporarely use the branch route because the changes you do there are per-branch. " Not sure if this is the reason for my problem. Is this a bug? Is there a way to add header fields temporarely? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHg1XQ4tZeNddg3dwRArKlAJ9vTWjg93/MeR8w+7R3XtP90p2eDwCgq/90 OfBsmaeapmBrGez4+Rt8NXQ= =3cte -----END PGP SIGNATURE-----

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Re: [OpenSER-Users] how to restrict invite when regiest unsuccessful
by liu york 08 Jan '08

08 Jan '08

OK,i see,let me try. many thx. ======= 2008-01-08 15:16:35 您在来信中写道：======= > > >Alex Balashov schrieb: >> On Tue, 8 Jan 2008, liu york wrote: >> >>> I want to restrict invite when regiest unsuccessful even the invite >>> request have right authentication. actually,i just want to only one >>> person can call to pstn,others cant call even have sipid and >>> password.now i has set by modparam("registrar", "max_contacts", 10).but >>> cant restrict the invite by the sipphone no need regiestion. >> >> Are you authenticating INVITEs differently than you are authenticating >> registrations? In other words, are the authentication realm and required >> credentials different for your proxy_challenge() than they are for >> your www_challenge() for handling REGISTER requests? >> >> If they are, it seems that all you should have to do is do a lookup() >> and see if it fails. Presumably, you are doing a save() into your >> registrar/usrloc only on valid REGISTER challenges? >> >> >> i.e. >> >> if(is_method("INVITE")) { >> if(!lookup("location")) { > > if(!registered("location")) { >is sufficient > >btw: this does not give you any security > >regards >klaus > > >> sl_send_reply("403", "Forbidden -- REGISTER invalid."); >> exit; >> } >> } >> >> >> -- >> Alex Balashov >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : +1-678-954-0670 >> Direct : +1-678-954-0671 >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　york 　　　　　　　　york(a)goldentek.biz 　　　　　　　　　　2008-01-08

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[OpenSER-Users] how to restrict invite when regiest unsuccessful
by liu york 08 Jan '08

08 Jan '08

hi I want to restrict invite when regiest unsuccessful even the invite request have right authentication. actually,i just want to only one person can call to pstn,others cant call even have sipid and password.now i has set by modparam("registrar", "max_contacts", 10).but cant restrict the invite by the sipphone no need regiestion. so plz help,many thx.

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[OpenSER-Users] presence(XCAP) permission rules?
by KevinKinnan 07 Jan '08

07 Jan '08

According to the instruction(README) of 'presence_xml', if using a integrated xcap server like OpenXCAP, the value of 'force_active' should set to 0. Then, the xcap server is queried and the subscription states is according to user defined permission rules. If no rules are defined for a certain watcher, the subscriptions remains in pending state and the Notify sent will have no body. Is the 'user defined permission rules' mentioned above specified in RFC 5025(Presence Authorization Rules) or 'XCAP usage for setting presence authorization'? They seems different. I have tried both of them and stored xml in xcap table for a user, say alice(she allows bob to wather her pre-info). But it was not working, subscription state was still 'pending'. Thank you! Regards, Kevin _________________________________________________________________ MSN圣诞礼物火热登场，免费发放中，快来领取吧！ http://im.live.cn/emoticons/?ID=18

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[OpenSER-Users] XCAP permission rules?
by KevinKinnan 07 Jan '08

07 Jan '08

My OpenSER works with openxcap, force_active is set to 0. The Subscription query the "xcap" table for permission rules of specified watcher. I create a simple xml rule which is shown below, and store it in xcap table. But the subscription-state is still "pending". I found another presence authorization rules that uses different statments like <provide-services> or <transformations>. which one should I use? Thank you! <?xml version="1.0" encoding="UTF-8"?> <permission-statements> <statement id="as8f"> <applies-to> <domain>open-ims.test</domain> </applies-to> <permissions> <accept/> </permissions> </statement> </permission-statements>Regards, Lei _________________________________________________________________ 天凉了，添衣了，心动了，“七件”了 http://get.live.cn

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[OpenSER-Users] new CDRTool release 6.1.0
by Adrian Georgescu 07 Jan '08

07 Jan '08

The software can be downloaded from: http://download.dns-hosting.info/CDRTool/ Changelog since 5.1.1: cdrtool (6.1.0) unstable; urgency=low * Fixed update of billing_rates entries from csv files * Overwrite defaults from sip_settings class per NGNPro connection * Added contrib directory * Added freeradius patch for freeradius cvs version from Norm Brandinger * Log progress of importing rating files * Show rating engine daemon uptime in syslog * Many fixes in provisioning client * Lookup customers and rates in MySQL, there is no memory limit anymore * Importing rating does not lock the rating engine anymore * Fixed billing of calls over the minimum duration cdrtool (6.0) unstable; urgency=low * Fixed links from Group-by output to discrete CDRs * After renormalization, re-cache the monthly usage only for the accounts that had CDRs changed during renormalization * Fixed application type for rating engine (default application is audio) * Added info to ENUM generator * Fixed node IP parsing in sip trace * Fixed default login account creation * Don't log ENUM tld in rate info if is set to none * Added RatingProfiles management in sip settings page * Added prepaid_cards management in Rating tables page * Fixed name of domain table when SIP Thor data-source is enabled * Added link in the trace window to jump to a different SIP Proxy * Import/export delimiter for rating tables can be adjusted, default is , * Must apply database structure changes from setup/mysql/ alter_tables.mysql * NGNPro client completed * Save settings before existing provisioning pages cdrtool (5.3.3) unstable; urgency=low * Fixed ENUM tld match and logging * Show visual ENUM usage * Provisioning engine update cdrtool (5.3.2) unstable; urgency=low * Fixed memory leak cdrtool (5.3.1) unstable; urgency=low * Removed reference to soapFilter which was unused * Skip ENUM tld 'none' in discounts * Update presence functions in provisioning client * Fixed initalization of gatewaygroups * Add function for showing customer text box * Allow specification of custom port names and classes per reseller * Fixed phplib typo that caused crash on php5 cdrtool (5.3.0) unstable; urgency=low * Added ENUM tld based discounts in the rating engine * Added ENUM tld management in the Rating page * Renamed field radius.Framed-Protocol to radius.ENUMtld to accomodate discounts based on ENUM top level domain. This functionality requires changes to the database structures and reconfigure several components: See doc/Upgrade.txt and setup/mysql/alter_tables.mysql for more information * Removed unused SOAP directory * Added NGNPro client provisioning library * Added sip settings page * Added new rights for provisioning tasks * Change main menu item names cdrtool (5.2.6) unstable; urgency=low * Log in syslog the duration of reloading of the rating tables and of the sip accounts into memcacche cdrtool (5.2.5) unstable; urgency=low * Replaced multiple SOAP definitions from global.inc with a pointer to an entry from the $soapEngines defined in soap_engines.inc * Fixed read of HTTP vars in sip trace, public traces are now visible when global vars are turned off cdrtool (5.2.4) unstable; urgency=low * Skip billing_rates.trafficRate absent from csv import * Update rating documentation * Added link to FreeRADIUS-CDRTool by Dan-Cristian Bogos * Fixed sip trace purging * Fix for virtualhost allowing .htaccess customization for cdrtool cdrtool (5.2.3) unstable; urgency=low * Allow usernames without domain part in quota check * Added publish of SIP domain statistics using SIP SIMPLE cdrtool (5.2.2) unstable; urgency=low * Added SIP trace for SIP Thor datasource cdrtool (5.2.1) unstable; urgency=low * Synced UPDATE queries for current and older month (in SQL stored procedures) (you must apply changes from setup/mysql/alter_tables.mysql) cdrtool (5.2.0) unstable; urgency=low * Eliminate the race condition that appears when normalization process is performed between the time radius database is updated when a BYE arrives and the MediaProxy updates the same radius record. cdrtool (5.1.9) unstable; urgency=low * Removed print of sql query cdrtool (5.1.8) unstable; urgency=low * Fixed saving the agreement with license page cdrtool (5.1.7) unstable; urgency=low * Added SOAP/XML management for LCR * Save when the user agrees with the usage license to avoid display the license agreement at each login cdrtool (5.1.6) unstable; urgency=low * Fixed calculus of historical rates when rate is found in alternative profile cdrtool (5.1.5) unstable; urgency=low * Hide price when grouping results for login accounts without showPrice right cdrtool (5.1.4) unstable; urgency=low * Fixed a typo in cdrlib.phtml cdrtool (5.1.3) unstable; urgency=low * Fixed pagination of exported records cdrtool (5.1.2) unstable; urgency=low * Show status of multiple mysql clusters Must migrate global.inc setting $CDRTool['replicated_databases'] to $CDRTool['mysql_clusters'] See setup/global.inc.in for example Regards, Adrian

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[Serusers] SEMS- unix socket problem
by dim 06 Jan '08

06 Jan '08

hi everybody, You can see attached my conf files for ser, ser-sems and sems. I run two instances of ser : 1. the first on port 5060 2. the second which is the sip stack for sems, on 5070 My conf files show that when for example i call 101 or 102 , the call is directed to sems, but the debug of my 2nd ser shows this : 2(8658) write_to_unixsock: Error in connect: Connection refused 2(8658) ERROR:tm:t_write_unix: write_to_unixsock failed 2(8658) write_to_unixsock: Error in connect: Connection refused 2(8658) ERROR:tm:t_write_unix: write_to_unixsock failed Probably these is some kind of problem with my unix socket configuration in my conf files... can anyone help me ??

1 0

[OpenSER-Users] Run on multiple ports
by Arun Kumar 05 Jan '08

05 Jan '08

Hi, how do I run ser/openser on multiple ports. ? I've configured my ser with two listen stat. listen = ip:5060 listen = ip:5070 UA:5070 -------> Cisco (Blocked 5060) --------> SER -------------> UB:5060 UA is able to call UB, but UB is not able to call UA, I know 5060 is blocked but it should goto 5070 ? please guide on this ? Thanks, Arun

4 5

Re: [OpenSER-Users] newbie help - handeling 302s and 404s
by Jesus Rodriguez 05 Jan '08

05 Jan '08

Hi Lazer, >i need a simple config that achieves the following (w/ out a db) > >if a user is not registered then all calls should go to a default gw >192.168.10.1 > >the default gw will only respond w/ a 302 or a 404 > >if a 302 is received then openser should forward the call on to the >new gw listed in the reply > >if a 404 is received the call should be forwarded to a different gw 172.16.20.1 > >if the send gw is done then the call should die. Try with somehting like this (not tested): if (!lookup("location")) { route(5); exit; } route[5] { rewritehostport("192.168.10.1:5060"); t_on_failure("1"); route(1); exit; } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1] { if (t_check_status("404")) { rewritehostport("172.16.20.1:5060"); append_branch(); t_relay(); } else { get_redirects("*"); t_relay(); } } Saludos JesusR. >here is what i was thinking of but as a newbie i may be off my rocker >i seem to get an error > > ERROR: t_check_status: cannot check status for a reply which has no >T-state established > >but again my whole approach may be wrong. > > > if (!lookup("location")) { > >rewritehostport ("192.168.10.1:5060"); >if (t_check_status("(404)|(480)")) { > log("404 or 480 negative reply\n"); >rewritehostport ("172.16.20.1:5060"); >} >else{ >t_on_failure("1"); > t_relay(); >} > > > failure_route[1] { > > get_redirects("*"); > t_relay(); >} > } Saludos JesusR. ------------------------------------ Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 -------------------------------------

3 4

[OpenSER-Users] Problem with "uac_replace_from()" and Linksys PAP: "vsf" parameter in Record-Route
by Iñaki Baz Castillo 05 Jan '08

05 Jan '08

Hi, I use "uac_replace_from" with: modparam("uac","from_restore_mode","auto") modparam("uac","rr_store_param","vsf") So when a call arrives and must be sent to an Asterisk voicemail server I change the From: Linksys PAP -> OpenSer INVITE sip:*500@openser.ilimit.es SIP/2.0 Via: SIP/2.0/UDP 192.168.1.109:5061;branch=z9hG4bK-91c6d79a From: PAP809 <sip:809@openser.ilimit.es>;tag=bd9af3255e5ce6ao1 To: <sip:*500@openser.ilimit.es> Call-ID: 148834d2-a640054a(a)192.168.1.109 OpenSer -> Asterisk INVITE sip:*500@80.94.0.111 SIP/2.0 Record-Route: <sip:80.94.0.110;lr=on;ftag=bd9af3255e5ce6ao1;vsf=AAAAAAAAAB8AAAAAAAAAAAAAAAAAAAAAAEBvcGVuc2VyLmlsaW1pdC5lcw--> Via: SIP/2.0/UDP 80.94.0.110;branch=z9hG4bKb114.2e2e6ca5.0 Via: SIP/2.0/UDP 192.168.1.109:5061;rport=5061;received=212.121.235.18;branch=z9hG4bK-91c6d79a From: PAP809 <sip:809_openser.ilimit.es@openser.ilimit.es>;tag=bd9af3255e5ce6ao1 To: <sip:*500@openser.ilimit.es> Call-ID: 148834d2-a640054a(a)192.168.1.109 Asterisk -> OpenSer SIP/2.0 200 OK Via: SIP/2.0/UDP 80.94.0.110;branch=z9hG4bKb114.2e2e6ca5.0;received=80.94.0.110 Via: SIP/2.0/UDP 192.168.1.109:5061;rport=5061;received=212.121.235.18;branch=z9hG4bK-91c6d79a Record-Route: <sip:80.94.0.110;lr=on;ftag=bd9af3255e5ce6ao1;vsf=AAAAAAAAAB8AAAAAAAAAAAAAAAAAAAAAAEBvcGVuc2VyLmlsaW1pdC5lcw--> From: PAP809 <sip:809_openser.ilimit.es@openser.ilimit.es>;tag=bd9af3255e5ce6ao1 To: <sip:*500@openser.ilimit.es>;tag=as32a464eb Call-ID: 148834d2-a640054a(a)192.168.1.109 OpenSer -> Linksys PAP SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.109:5061;rport=5061;received=212.121.235.18;branch=z9hG4bK-91c6d79a Record-Route: <sip:80.94.0.110;lr=on;ftag=bd9af3255e5ce6ao1;vsf=AAAAAAAAAB8AAAAAAAAAAAAAAAAAAAAAAEBvcGVuc2VyLmlsaW1pdC5lcw--> From: PAP809 <sip:809@openser.ilimit.es>;tag=bd9af3255e5ce6ao1 To: <sip:*500@openser.ilimit.es>;tag=as32a464eb Call-ID: 148834d2-a640054a(a)192.168.1.109 The problem is that this "200 OK" from OpenSer to Linksys PAP is not recognized by Linksys PAP who doesn't reply with an ACK (just ignores the "200 OK"), so the "200 OK" is resent again and again by Asterisk. This issue just occurs with the Linksys PAP: Product Name: PAP2-NA Software Version: 2.0.12(LS) Hardware Version: 0.03.4 There is no NAT problem and if I set: modparam("uac","from_restore_mode","none") then the problem dissapeares, so the problem is the existence of "vsf" parameter in Record-Route. If this parameter doesn't exist the problem doesn't occur. Again I repeat that this issue doesn't occur with others phones. Any idea of why this occurs with Linksys PAP? Thanks for any suggestion. PD: Changing the parameter name does solve nothing. PPD: What about if I don't restore the From in the replies? According to RFC3261 the only important is the matching of From_tag, To_tag and Call-id, so, maybe I just should set: modparam("uac","from_restore_mode","none") and forget this issue? -- Iñaki Baz Castillo ibc(a)in.ilimit.es

3 3

[OpenSER-Users] Digest auth on openwrt without mysql ?
by Michael Bradley Jr 04 Jan '08

04 Jan '08

Hi, is it possible to run openser with digest auth on an openwrt router *without* mysql? If yes can someone please provide me a hint/link? Did try loading "/usr/lib/openser/module/auth.so" without great success, since my box lacks a mysql server version! Thanks in advance Michaek

4 3

[OpenSER-Users] # characters in URIs getting reformatted.
by Alex Balashov 04 Jan '08

04 Jan '08

I am using OpenSER to front some Cisco AS5300s. The way the trunk groups on those are set up, a prefix of #755 is required to dial out, so I transform the URI in OpenSER: $rU = "#755" + $rU; The problem is, certain messages never get acknowledged because OpenSER reformats the URI to use escaped sequences. I am not sure if this is RFC compliant. For instance, a 200 OK comes from the media gateway, and the ACK that OpenSER sends back has this for the URI scheme: ACK sip:%237557069999999@ip.address:5060 This is in response to an initial INVITE to: sip:#7557069999999@ip.address:5060 ... what should I make of this? Thanks! -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

4 5

Re: [OpenSER-Users] Digest auth on openwrt without mysql ?
by Henning Westerholt 04 Jan '08

04 Jan '08

On Friday 04 January 2008, Michael Bradley Jr wrote: > Hi Henning, > > I'm running openser on an openwrt (whiterussion 0.9) box and I'm quite new > to openser. > I did install openser and would like to quickly add a domain and some users > that could digest auth on my box. > I know RTFM but can you tell me the direction to go for this quick setup? Hi, look at http://www.openser.org/docs/modules/1.3.x/ for the modules docs. Basically you need to load a database driver of your choice and the db_auth module. Take a look at the docs for this module for the available functions for authentification, in the example config there are also some hints. I don't have experience with OpenSER on the openWRT, unfortunally, so i don't can give you more detailed hints about this special setup. You need to specify your database type in the /etc/openser/openserctlrc file, and then you could use the openserdbctl tool to create a new database. After that you could add a new user with the openserctl tool, and authentification should work. It it don't, increase the debug level to get some hints about the problems. Run this tools without any command to get some help, there is also some documentation about the installation on the wiki. Hope this helps - cheers, Henning

1 0

[OpenSER-Users] Fetching multiple rows from DB.
by Alex Balashov 04 Jan '08

04 Jan '08

So, is there a legitimate way to retrieve multiple rows using avp_db_query() and iterate through them with a loop structure? Does AVP support the notion of an "array" construction? And if not, how do I get OpenSER to apply a set of rules codified in custom database structures? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

3 2

[OpenSER-Users] UAC stops responding to opensers 401 unauthorized challenge after few hours
by Raju Abhyankar 03 Jan '08

03 Jan '08

I need some help and guidance here as I am fairly new to openser but have made some progress.... Can some one throw some light on this issue. I am using a standard openser script from sipwise.com. I have several ATA or UAC registered with openser-1.1.1-tls. THe registration process works just fine. I can make calls etc. no issues. I am using Mediaproxy for nat traversal and no issues there too. Since I am using the internet (not private LAN) to connect these devices, I have set my register expiration time to 1200 seconds. Now after a few hours of operation, I see that some of the ATA or UAC (HT486) deregister. I have used "ngrep" and I can see: 1. UAC sends REGISTER request 2. openser responds by saying "100" "Trying" 2. Openser generates a 401 unauthorized 3. UAC once again generates REGISTER request and this goes on. The only way to correct this issue is to reboot the ATA by powering it off and on. Any ideas why this may happen? When I run the same devices using Asterisk I dont see the devices deregister but I would like to use Asterisk only a PSTN gateway. I have included a trace of this below Thanks in advance Raju ==================================================== From: <sip:88@lmnopq.com>;tag=ed8cb29199123c47. To: <sip:88@lmnopq.com>. Contact: <sip:88@YY.YYY.YYY.YY:50826>. Supported: replaces, timer. Call-ID: a679dcef0eebdbec(a)192.168.1.5. CSeq: 100 REGISTER. Expires: 60. User-Agent: TigerNetCom 1.02.12.0.9.11. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE. Content-Length: 0. . U 2008/01/03 13:42:29.518524 XXX.XX.XXX.XXX:5060 -> YY.YYY.YYY.YY:50825 SIP/2.0 100 Trying. Via: SIP/2.0/UDP YY.YYY.YYY.YY:50826;branch=z9hG4bK0826bc66f89e21de;rport=50825. From: <sip:88@lmnopq.com>;tag=ed8cb29199123c47. To: <sip:88@lmnopq.com>. Call-ID: a679dcef0eebdbec(a)192.168.1.5. CSeq: 100 REGISTER. Server: OpenSer (1.1.1-tls (i386/linux)). Content-Length: 0. . U 2008/01/03 13:42:29.518566 XXX.XX.XXX.XXX:5060 -> YY.YYY.YYY.YY:50825 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP YY.YYY.YYY.YY:50826;branch=z9hG4bK0826bc66f89e21de;rport=50825. From: <sip:88@lmnopq.com>;tag=ed8cb29199123c47. To:<sip:88@lmnopq.com>;tag=5cc123387d097550f225581b658b043b.4f69. Call-ID: a679dcef0eebdbec(a)192.168.1.5. CSeq: 100 REGISTER. WWW-Authenticate: Digest realm="lmnopq.com", nonce="477d57f148e6dc9dd21502b6c60d4bfe9d061657". Server: OpenSer (1.1.1-tls (i386/linux)). Content-Length: 0. After this the register with same cseq shows up and the device does not register....

1 0

[Serusers] SER fails to match ACK to previous INVITE
by Srinivas Kotamarti 03 Jan '08

03 Jan '08

Hi, I have recently downloaded SER & SEMS and configured them for early_announce application as described in the early_announce module documentation. I am using SIPp to generate an incoming call to test the setup. Upon receiving ACK for some reason SER fails to match it to the previous INVITE and tries to forward it to the destination statelessly and picks it up again and forwards it. It continues to do this until the max forward limit is reached. I cannot figure out what is wrong. Any help is really appreciated. Here is the Route section from ser-sems.cfg file that I am using # ------------------------- 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"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; if (uri==myself) { # filter unsupported requests if (!(method=="REGISTER" || method=="ACK" || method=="INVITE" || method=="BYE" || method=="CANCEL" )) { sl_send_reply("501", "method not understood here"); break; } if (method=="REGISTER") { # make UAs which want to register unhappy sl_send_reply("501","registration couldn't be accepted"); break; }; if (!t_newtran()){ sl_send_reply("500","could not create transaction"); break; }; if (method == "INVITE") { avp_write("Final-Reply-Code=405;Final-Reply-Reason=Prohiboted", "$sess_params"); if(!t_write_unix("/tmp/sems_sock","early_announce/early_headers")){ log("could not contact early_announce\n"); t_reply("404","not found"); }; break; } else if (method=="BYE" || method=="CANCEL") { if(!t_write_unix("/tmp/sems_sock","bye")) { t_reply("500","error contacting sems"); }; break; } else if (method=="ACK"){ # absorb ACKs t_relay(); break; }; break; } if (!t_relay()) { sl_reply_error(); }; } Debug logs from SER 0(27347) SIP Request: 0(27347) method: <ACK> 0(27347) uri: <sip:service@10.107.2.246> 0(27347) version: <SIP/2.0> 0(27347) parse_headers: flags=1 0(27347) Found param type 232, <branch> = <0>; state=16 0(27347) end of header reached, state=5 0(27347) parse_headers: Via found, flags=1 0(27347) parse_headers: this is the first via 0(27347) After parse_msg... 0(27347) preparing to run routing scripts... 0(27347) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(27347) parse_headers: flags=128 0(27347) Found param type 232, <branch> = <0>; state=16 0(27347) end of header reached, state=5 0(27347) parse_headers: Via found, flags=128 0(27347) parse_headers: this is the second via 0(27347) Found param type 232, <branch> = <z9hG4bK-27351-1-3>; state=16 0(27347) end of header reached, state=5 0(27347) parse_headers: Via found, flags=128 0(27347) DEBUG: add_param: tag=79EBE47B-477CE4230002578E-B7F6D6C0 0(27347) end of header reached, state=29 0(27347) DEBUG: get_hdr_field: <To> [76]; uri=[sip:service@10.107.2.246:5060]<mailto:[sip:service@10.107.2.246:5060]> 0(27347) DEBUG: to body [sut <sip:service@10.107.2.246:5060>] 0(27347) get_hdr_field: cseq <CSeq>: <1> <ACK> 0(27347) DEBUG:maxfwd:is_maxfwd_present: value = 15 0(27347) DEBUG: add_param: tag=27351SIPpTag001 0(27347) end of header reached, state=29 0(27347) parse_headers: flags=256 0(27347) DEBUG: get_hdr_body : content_length=0 0(27347) found end of header 0(27347) find_first_route: No Route headers found 0(27347) loose_route: There is no Route HF 0(27347) grep_sock_info - checking if host==us: 12==12 && [10.107.2.246] == [10.107.2.246] 0(27347) grep_sock_info - checking if port 5060 matches port 5060 0(27347) DEBUG: t_newtran: msg id=4 , global msg id=3 , T on entrance=(nil) 0(27347) parse_headers: flags=-1 0(27347) parse_headers: flags=60 0(27347) t_lookup_request: start searching: hash=30348, isACK=1 0(27347) DEBUG: proceeding to pre-RFC3261 transaction matching 0(27347) DEBUG: t_lookup_request: no transaction found 0(27347) DEBUG: t_newtran: msg id=4 , global msg id=4 , T on entrance=(nil) 0(27347) parse_headers: flags=-1 0(27347) parse_headers: flags=60 0(27347) t_lookup_request: start searching: hash=30348, isACK=1 0(27347) DEBUG: proceeding to pre-RFC3261 transaction matching 0(27347) DEBUG: t_lookup_request: no transaction found 0(27347) SER: forwarding ACK statelessly 0(27347) DEBUG: mk_proxy: doing DNS lookup... 0(27347) check_via_address(10.107.2.246, 10.107.2.246, 0) 0(27347) Sending: ACK sip:service@10.107.2.246 SIP/2.0 Record-Route: <sip:10.107.2.246;ftag=27351SIPpTag001;lr=on> Record-Route: <sip:10.107.2.246;ftag=27351SIPpTag001;lr=on> Record-Route: <sip:10.107.2.246;ftag=27351SIPpTag001;lr=on> Via: SIP/2.0/UDP 10.107.2.246;branch=0 Via: SIP/2.0/UDP 10.107.2.246;branch=0 Via: SIP/2.0/UDP 10.107.2.246;branch=0 Via: SIP/2.0/UDP 10.107.2.246:5061;branch=z9hG4bK-27351-1-3 From: sipp <sip:sipp@10.107.2.246:5061>;tag=27351SIPpTag001 To: sut <sip:service@10.107.2.246:5060>;tag=79EBE47B-477CE4230002578E-B7F6D6C0 Call-ID: 1-27351(a)10.107.2.246<mailto:1-27351@10.107.2.246> CSeq: 1 ACK Contact: sip:sipp@10.107.2.246:5061 Max-Forwards: 14 Subject: Performance Test Content-Length: 0 --Srinivas Kotamarti

1 0

[Serusers] SER fails to match ACK to previous INVITE
by Srinivas Kotamarti 03 Jan '08

03 Jan '08

Hi, I have recently downloaded SER & SEMS and configured them for early_announce application as described in the early_announce module documentation. I am using SIPp to generate an incoming call to test the setup. Upon receiving ACK for some reason SER fails to match it to the previous INVITE and tries to forward it to the destination statelessly and picks it up again and forwards it. It continues to do this until the max forward limit is reached. I cannot figure out what is wrong. Any help is really appreciated. Here is the Route section from ser-sems.cfg file that I am using # ------------------------- 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"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; if (uri==myself) { # filter unsupported requests if (!(method=="REGISTER" || method=="ACK" || method=="INVITE" || method=="BYE" || method=="CANCEL" )) { sl_send_reply("501", "method not understood here"); break; } if (method=="REGISTER") { # make UAs which want to register unhappy sl_send_reply("501","registration couldn't be accepted"); break; }; if (!t_newtran()){ sl_send_reply("500","could not create transaction"); break; }; if (method == "INVITE") { avp_write("Final-Reply-Code=405;Final-Reply-Reason=Prohiboted", "$sess_params"); if(!t_write_unix("/tmp/sems_sock","early_announce/early_headers")){ log("could not contact early_announce\n"); t_reply("404","not found"); }; break; } else if (method=="BYE" || method=="CANCEL") { if(!t_write_unix("/tmp/sems_sock","bye")) { t_reply("500","error contacting sems"); }; break; } else if (method=="ACK"){ # absorb ACKs t_relay(); break; }; break; } if (!t_relay()) { sl_reply_error(); }; } Debug logs from SER 0(27347) SIP Request: 0(27347) method: <ACK> 0(27347) uri: <sip:service@10.107.2.246> 0(27347) version: <SIP/2.0> 0(27347) parse_headers: flags=1 0(27347) Found param type 232, <branch> = <0>; state=16 0(27347) end of header reached, state=5 0(27347) parse_headers: Via found, flags=1 0(27347) parse_headers: this is the first via 0(27347) After parse_msg... 0(27347) preparing to run routing scripts... 0(27347) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(27347) parse_headers: flags=128 0(27347) Found param type 232, <branch> = <0>; state=16 0(27347) end of header reached, state=5 0(27347) parse_headers: Via found, flags=128 0(27347) parse_headers: this is the second via 0(27347) Found param type 232, <branch> = <z9hG4bK-27351-1-3>; state=16 0(27347) end of header reached, state=5 0(27347) parse_headers: Via found, flags=128 0(27347) DEBUG: add_param: tag=79EBE47B-477CE4230002578E-B7F6D6C0 0(27347) end of header reached, state=29 0(27347) DEBUG: get_hdr_field: <To> [76]; uri=[sip:service@10.107.2.246:5060]<mailto:[sip:service@10.107.2.246:5060]> 0(27347) DEBUG: to body [sut <sip:service@10.107.2.246:5060>] 0(27347) get_hdr_field: cseq <CSeq>: <1> <ACK> 0(27347) DEBUG:maxfwd:is_maxfwd_present: value = 15 0(27347) DEBUG: add_param: tag=27351SIPpTag001 0(27347) end of header reached, state=29 0(27347) parse_headers: flags=256 0(27347) DEBUG: get_hdr_body : content_length=0 0(27347) found end of header 0(27347) find_first_route: No Route headers found 0(27347) loose_route: There is no Route HF 0(27347) grep_sock_info - checking if host==us: 12==12 && [10.107.2.246] == [10.107.2.246] 0(27347) grep_sock_info - checking if port 5060 matches port 5060 0(27347) DEBUG: t_newtran: msg id=4 , global msg id=3 , T on entrance=(nil) 0(27347) parse_headers: flags=-1 0(27347) parse_headers: flags=60 0(27347) t_lookup_request: start searching: hash=30348, isACK=1 0(27347) DEBUG: proceeding to pre-RFC3261 transaction matching 0(27347) DEBUG: t_lookup_request: no transaction found 0(27347) DEBUG: t_newtran: msg id=4 , global msg id=4 , T on entrance=(nil) 0(27347) parse_headers: flags=-1 0(27347) parse_headers: flags=60 0(27347) t_lookup_request: start searching: hash=30348, isACK=1 0(27347) DEBUG: proceeding to pre-RFC3261 transaction matching 0(27347) DEBUG: t_lookup_request: no transaction found 0(27347) SER: forwarding ACK statelessly 0(27347) DEBUG: mk_proxy: doing DNS lookup... 0(27347) check_via_address(10.107.2.246, 10.107.2.246, 0) 0(27347) Sending: ACK sip:service@10.107.2.246 SIP/2.0 Record-Route: <sip:10.107.2.246;ftag=27351SIPpTag001;lr=on> Record-Route: <sip:10.107.2.246;ftag=27351SIPpTag001;lr=on> Record-Route: <sip:10.107.2.246;ftag=27351SIPpTag001;lr=on> Via: SIP/2.0/UDP 10.107.2.246;branch=0 Via: SIP/2.0/UDP 10.107.2.246;branch=0 Via: SIP/2.0/UDP 10.107.2.246;branch=0 Via: SIP/2.0/UDP 10.107.2.246:5061;branch=z9hG4bK-27351-1-3 From: sipp <sip:sipp@10.107.2.246:5061>;tag=27351SIPpTag001 To: sut <sip:service@10.107.2.246:5060>;tag=79EBE47B-477CE4230002578E-B7F6D6C0 Call-ID: 1-27351(a)10.107.2.246<mailto:1-27351@10.107.2.246> CSeq: 1 ACK Contact: sip:sipp@10.107.2.246:5061 Max-Forwards: 14 Subject: Performance Test Content-Length: 0 --Srinivas Kotamarti

1 0

[OpenSER-Users] Daily Snapshots for 1.3.x release
by Anatoly Pidruchny 03 Jan '08

03 Jan '08

Hello, Is it planned to add daily snapshats for the 1.3.x release? They are missing here: http://www.openser.org/downloads/snapshots/ Thanks, /Anatoly.

1 0

[Serusers] Unix Socket Problem
by frq ser 03 Jan '08

03 Jan '08

Hi, Thanks for your reply Stefan. Here is the log from SEMS 0.10.0, which I just installed. This log is in two parts. The first one (with a few lines missing from the top) is displayed when sems is started. The second part is displayed when I send the INVITE message. Thanks again for your help. Frq. SEMS-0.10.0 log (Part 1) . . . /usr/local/lib/sems/plug-in': (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/stat s.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'stats' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/anno uncement.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'announcement' load ed. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/wav. so ... (24406) DEBUG: addCodec (AmPlugIn.cpp:540): codec id 1 inserted (24406) DEBUG: addCodec (AmPlugIn.cpp:540): codec id 2 inserted (24406) DEBUG: addPayload (AmPlugIn.cpp:572): payload 'PCMU' inserted with id 0 and order 100 (24406) DEBUG: addPayload (AmPlugIn.cpp:572): payload 'PCMA' inserted with id 8 and order 108 (24406) DEBUG: addFileFormat (AmPlugIn.cpp:610): file format Wav inserted (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/ilbc .so ... (24406) DEBUG: addCodec (AmPlugIn.cpp:540): codec id 4 inserted (24406) DEBUG: addPayload (AmPlugIn.cpp:572): payload 'iLBC' inserted with id 97 and order 197 (24406) DEBUG: addFileFormat (AmPlugIn.cpp:610): file format iLBC inserted (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/webc onference.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'webconference' loa ded. (24406) DEBUG: loadDiPlugIn (AmPlugIn.cpp:479): component 'webconference' loaded . (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/ann_ b2b.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'ann_b2b' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/sess ion_timer.so ... (24406) DEBUG: loadSehPlugIn (AmPlugIn.cpp:457): session component 'session_time r' loaded. (24406) DEBUG: loadDiPlugIn (AmPlugIn.cpp:479): component 'user_timer' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/uac_ auth.so ... (24406) DEBUG: loadSehPlugIn (AmPlugIn.cpp:457): session component 'uac_auth' lo aded. (24406) DEBUG: loadDiPlugIn (AmPlugIn.cpp:479): component 'uac_auth' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/echo .so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'echo' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/earl y_announce.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'early_announce' lo aded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/reg_ agent.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'reg_agent' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/regi strar_client.so ... (24406) DEBUG: loadDiPlugIn (AmPlugIn.cpp:479): component 'registrar_client' loa ded. (24406) DEBUG: loadSIPehPlugIn (AmPlugIn.cpp:502): sip event handler component ' registrar_client' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/voic email.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'voicemail' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/conf erence.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'conference' loaded . (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/ivr. so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'ivr' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/gsm. so ... (24406) DEBUG: addCodec (AmPlugIn.cpp:540): codec id 3 inserted (24406) DEBUG: addPayload (AmPlugIn.cpp:572): payload 'GSM' inserted with id 3 a nd order 103 (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/anno unce_transfer.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'announce_transfer' loaded. (24406) DEBUG: load (AmPlugIn.cpp:172): AmPlugIn: modules loaded. (24406) DEBUG: load (AmPlugIn.cpp:174): AmPlugIn: Initializing plugins... (24406) DEBUG: start (AmThread.cpp:135): Thread 0 is just created. (24406) DEBUG: start (AmThread.cpp:135): Thread 0 is just created. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'first_parti cipant' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'join_sound' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'drop_sound' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'enter_pin' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'wrong_pin' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'entering_co nference' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '0' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '1' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '2' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '3' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '4' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '5' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '6' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '7' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '8' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '9' to promp t collection. (24406) DEBUG: onLoad (WebConference.cpp:104): Using adaptive playout buffer as playout technique. (24406) DEBUG: onLoad (WebConference.cpp:127): Webconference will strip 5 leadin g characters from direct room access usernames (24406) DEBUG: start (AmThread.cpp:135): Thread 0 is just created. (24406) DEBUG: init (AmCtrlInterface.cpp:231): AmUnixCtrlInterface::init @ /tmp/ sems_rsp_sock (24406) DEBUG: onLoad (AnnounceB2B.cpp:66): AnnounceFile = default_en.wav (24406) DEBUG: onLoad (Conference.cpp:258): Using adaptive playout buffer as pla yout technique. (24406) DEBUG: onLoad (Echo.cpp:52): session_timer_f == 0x0000000009500490 (24406) DEBUG: add_env_path (AmUtils.cpp:891): setting PYTHONPATH to: '/usr/loca l/lib/sems/plug-in' (24406) DEBUG: Ivr-Python: Python-Ivr logging started (24406) DEBUG: onLoad (Ivr.cpp:469): ** IVR compile time configuration: (24406) DEBUG: onLoad (Ivr.cpp:470): ** built with PYTHON support. (24406) DEBUG: onLoad (Ivr.cpp:475): ** Text-To-Speech disabled (24406) DEBUG: onLoad (Ivr.cpp:478): ** IVR run time configuration: (24406) DEBUG: onLoad (Ivr.cpp:479): ** script path: '/usr/local/lib /sems/ivr/' (24406) DEBUG: onLoad (Ivr.cpp:495): directory '/usr/local/lib/sems/ivr/' opened (24406) DEBUG: _start (AmThread.cpp:68): Thread 3083512736 is starting. (24406) DEBUG: run (SIPRegistrarClient.cpp:190): SIPRegistrarClient starting... (24406) DEBUG: _start (AmThread.cpp:68): Thread 3084868512 is starting. (24406) DEBUG: _start (AmThread.cpp:68): Thread 3083512736 is starting. (24406) DEBUG: run (SIPRegistrarClient.cpp:190): SIPRegistrarClient starting... (24406) INFO: onLoad (Ivr.cpp:522): Application script registered: conf_auth. (24406) INFO: onLoad (Ivr.cpp:522): Application script registered: mailbox. (24406) INFO: onLoad (Ivr.cpp:522): Application script registered: mailbox_query . (24406) INFO: onLoad (Ivr.cpp:522): Application script registered: pin_collect. (24406) ERROR: onLoad (RegistrationAgent.cpp:69): Account for registration not c orrectly configured. (24406) ERROR: onLoad (RegistrationAgent.cpp:70): RegistrationAgent will not reg ister any accounts. (24406) DEBUG: init (StatsUDPServer.cpp:144): udp_port = 5040 (24406) DEBUG: init (StatsUDPServer.cpp:186): socket bound at port 5040 (24406) DEBUG: start (AmThread.cpp:135): Thread 0 is just created. (24406) DEBUG: start (AmThread.cpp:135): Thread 0 is just created. (24406) DEBUG: loadEmailTemplates (AnswerMachine.cpp:298): loading /usr/local/et c/sems//default.template ... (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'first_parti cipant' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'join_sound' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'drop_sound' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'enter_pin' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'wrong_pin' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt 'entering_co nference' to prompt collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '0' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '1' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '2' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '3' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '4' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '5' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '6' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '7' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '8' to promp t collection. (24406) DEBUG: setPrompt (AmPromptCollection.cpp:81): adding prompt '9' to promp t collection. (24406) DEBUG: onLoad (WebConference.cpp:104): Using adaptive playout buffer as playout technique. (24406) DEBUG: onLoad (WebConference.cpp:127): Webconference will strip 5 leadin g characters from direct room access usernames (24406) DEBUG: load (AmPlugIn.cpp:221): AmPlugIn: Initialized plugins. (24406) DEBUG: main (sems.cpp:377): Starting session container (24406) DEBUG: _start (AmThread.cpp:68): Thread 3078294432 is starting. (24406) DEBUG: _start (AmThread.cpp:68): Thread 3078294432 is starting. (24406) DEBUG: _start (AmThread.cpp:68): Thread 3076889504 is starting. (24406) DEBUG: start (AmThread.cpp:135): Thread 3076889504 is just created. (24406) DEBUG: main (sems.cpp:380): Starting media processor (24406) DEBUG: init (AmMediaProcessor.cpp:61): Starting 1 MediaProcessorThreads. (24406) DEBUG: start (AmThread.cpp:135): Thread 0 is just created. (24406) DEBUG: main (sems.cpp:383): Starting mailer (24406) DEBUG: start (AmThread.cpp:135): Thread 0 is just created. (24406) DEBUG: main (sems.cpp:386): Starting RTP receiver (24406) DEBUG: start (AmThread.cpp:135): Thread 0 is just created. (24406) DEBUG: init (AmCtrlInterface.cpp:231): AmUnixCtrlInterface::init @ /tmp/ sems_sock (24406) DEBUG: _start (AmThread.cpp:68): Thread 3075836832 is starting. (24406) DEBUG: _start (AmThread.cpp:68): Thread 3074784160 is starting. (24406) DEBUG: _start (AmThread.cpp:68): Thread 3073731488 is starting. Ser's unix socket: /tmp/ser_sock our unix socket: /tmp/sems_sock reply unix socket: /tmp/sems_rsp_sock plug-in path: /usr/local/lib/sems/plug-in daemon mode: 0 local IP: 192.168.3.208 default application: <not set> (24406) DEBUG: main (sems.cpp:371): Loading plug-ins (24406) DEBUG: AmPlugIn (AmPlugIn.cpp:102): adding built-in codecs... (24406) DEBUG: addCodec (AmPlugIn.cpp:540): codec id 0 inserted (24406) DEBUG: addCodec (AmPlugIn.cpp:540): codec id 6 inserted (24406) DEBUG: addPayload (AmPlugIn.cpp:572): payload 'telephone-event' inserted with id 96 and order 196 (24406) DEBUG: load (AmPlugIn.cpp:136): AmPlugIn: loading modules in directory ' /usr/local/lib/sems/plug-in': (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/stat s.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'stats' loaded. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/anno uncement.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'announcement' load ed. (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/wav. so ... (24406) DEBUG: addCodec (AmPlugIn.cpp:540): codec id 1 inserted (24406) DEBUG: addCodec (AmPlugIn.cpp:540): codec id 2 inserted (24406) DEBUG: addPayload (AmPlugIn.cpp:572): payload 'PCMU' inserted with id 0 and order 100 (24406) DEBUG: addPayload (AmPlugIn.cpp:572): payload 'PCMA' inserted with id 8 and order 108 (24406) DEBUG: addFileFormat (AmPlugIn.cpp:610): file format Wav inserted (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/ilbc .so ... (24406) DEBUG: addCodec (AmPlugIn.cpp:540): codec id 4 inserted (24406) DEBUG: addPayload (AmPlugIn.cpp:572): payload 'iLBC' inserted with id 97 and order 197 (24406) DEBUG: addFileFormat (AmPlugIn.cpp:610): file format iLBC inserted (24406) DEBUG: load (AmPlugIn.cpp:146): loading /usr/local/lib/sems/plug-in/webc onference.so ... (24406) DEBUG: loadAppPlugIn (AmPlugIn.cpp:435): application 'webconference' loa --------------------------------------------------------------------------------------- And the second part of the log is this: (24406) DEBUG: run (AmServer.cpp:93): revents = 1 (24406) DEBUG: cacheMsg (AmCtrlInterface.cpp:186): recv-ed: <0.3 sip_request.announcement INVITE 200 192.168.3.208 192.168.3.208 5070 sip:200@192.168.3.208:5070 sip:rsf@192.168.3.77 sip:rsf@192.168.3.77 sip:200@192.168.3.208:5070 9787963-3319555348-76751(a)192.168.3.77 100 47668:470495812 <sip:192.168.3.208;lr=on> sip:192.168.3.208;lr=on P-MsgFlags: 0 . > (24406) DEBUG: execute (AmInterfaceHandler.cpp:127): req.method = <INVITE> (24406) DEBUG: execute (AmInterfaceHandler.cpp:128): req.user = <200> (24406) DEBUG: execute (AmInterfaceHandler.cpp:129): req.domain = <192.168.3.208> (24406) DEBUG: execute (AmInterfaceHandler.cpp:130): req.dstip = <192.168.3.208> (24406) DEBUG: execute (AmInterfaceHandler.cpp:131): req.port = <5070> (24406) DEBUG: execute (AmInterfaceHandler.cpp:132): req.r_uri = <sip:200@192.168.3.208:5070> (24406) DEBUG: execute (AmInterfaceHandler.cpp:133): req.from_uri = <sip:rsf@192.168.3.77> (24406) DEBUG: execute (AmInterfaceHandler.cpp:134): req.from = <sip:rsf@192.168.3.77> (24406) DEBUG: execute (AmInterfaceHandler.cpp:135): req.to = <sip:200@192.168.3.208:5070> (24406) DEBUG: execute (AmInterfaceHandler.cpp:136): req.callid = <9787963-3319555348-76751(a)192.168.3.77> (24406) DEBUG: execute (AmInterfaceHandler.cpp:137): req.from_tag = <> (24406) DEBUG: execute (AmInterfaceHandler.cpp:138): req.to_tag = <> (24406) DEBUG: execute (AmInterfaceHandler.cpp:141): cseq_str = <100> (24406) DEBUG: execute (AmInterfaceHandler.cpp:147): cseq = <100>(100) (24406) DEBUG: execute (AmInterfaceHandler.cpp:149): req.key = <47668:470495812> (24406) DEBUG: execute (AmInterfaceHandler.cpp:150): req.route = <<sip:192.168.3.208;lr=on>> (24406) DEBUG: execute (AmInterfaceHandler.cpp:151): req.next_hop = <sip:192.168.3.208;lr=on> (24406) DEBUG: execute (AmInterfaceHandler.cpp:156): hdrs = <P-MsgFlags: 0 > (24406) DEBUG: execute (AmInterfaceHandler.cpp:161): body = <> (24406) ERROR: run (AmServer.cpp:110): AmRequestHandler::execute: mandatory parameter is empty (from|to|callid|from_tag) (24406) DEBUG: run (AmServer.cpp:93): revents = 0 On 1/2/08, Stefan Sayer <stefan.sayer(a)iptego.de> wrote: > Hello, > > > frq ser wrote: > > Hi > > > > I am running the simple configuration in which SER 0.9.7-pre1 serves > > as Proxy (at port 5060) and SER-SEMS is the media server (at port > > 5070). At port 5070, the SER version is 0.9.6-pre1 and SEMS version is > > 0.10.0-rc2. I have placed log commands in the config files just to > > record the process flow. > > > > Using SIPSAK, I am sending an INVITE (for SER-SEMS at :5070) through > > the proxy. My sipsak message is this: > > C:>sipsak -f invite.txt -s sip:200@192.168.3.208:5060 -vvv > > > > The request enters the proxy ser.cfg file and is redirected to the > > SER-SEMS (see the log messages below). The user name (200@...) is used > > by if statement to invoke the announcement application. However, I > > don't see any response from the unix socket (no announcement played) > > nor do I see any error message, and the connection finally times out. > > > > Can anyone please tell me how I can check whether the unix sockets are > > doing their job or not? Why does the process not return any error and > > why does it just time out? > what does the sems log look like if you start sems with -D 3 -E ? does > it run and receive the message? > > thanks > Stefan > > > > > > > Thanks in advance for your help. > > Frq > > > > PS: Here are the (somewhat condensed) versions of SER and SEMS outputs > > (log messages mostly), SIPSAK responses, and the relevant config > > files. I apologise for sending a huge email. > p.s. I would recommend sems 0.10.0 - > ftp://ftp.iptel.org/pub/sems/sems-0.10.0.tar.gz > > -- > Stefan Sayer > VoIP Services > > stefan.sayer(a)iptego.de > www.iptego.de > > iptego GmbH > Am Borsigturm 40 > 13507 Berlin > Germany > > Amtsgericht Charlottenburg, HRB 101010 > Geschaeftsfuehrer: Alexander Hoffmann >

3 4

[OpenSER-Users] Meeting at CCC in Berlin, OSMB in Nuremberg
by Henning Westerholt 03 Jan '08

03 Jan '08

Hello all, i'll do two presentations about OpenSER at the Chaos Communication Congress [1], December 27th-30th, 2007 in Berlin and the Open Source Meets Business congress [2], January 22th–24th, 2008 in Nuremberg. If some people there are interested in a meeting for discussions around OpenSER and other related (or unrelated..) topics, please drop me a private email. Regards, Henning [1] http://events.ccc.de/congress/2007/Main_Page [2] http://www.heise.de/events/2008/open_source_meets_business/en/

3 4

[Serusers] Using SER with Microsoft OCS 2007
by Nolan Garrett 02 Jan '08

02 Jan '08

Hello! I am interested in using SER to convert TCP SIP traffic from OCS to UDP SIP traffic that I can use with an external PSTN gateway provider. Has anyone successfully managed this? Basically, the configuration would look like OCS -> Mediation Server -> SER -> External Provider Would the Mediation Server still need an external (non-NAT'd) IP? It seems like it would, unless I use mediaproxy or RTPproxy, which would add an additional layer of complexity. I guess I am looking for some validation that what I am trying to do will work. My next question, which I can't seem to find an answer to, is how I take the TCP SIP traffic from OCS, convert it to UDP SIP, and also authenticate with that external provider. Any ideas on how to use t_relay_to_udp with authentication? Thank you! Nolan

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Re: [OpenSER-Users] avp_pushto question
by Jeremy George 02 Jan '08

02 Jan '08

Perfect. Thank you. - Jeremy >-----Original Message----- >From: Iñaki Baz Castillo [mailto:ibc@aliax.net] >Sent: Wednesday, January 2, 2008 01:27 PM >To: users(a)lists.openser.org >Subject: Re: [OpenSER-Users] avp_pushto question > >El Miércoles, 2 de Enero de 2008, Jeremy George escribió: >> while($var(idx) < 5) { >> if ($(avp(i:1)[$var(idx)]) != NULL) { >> xlog("L_NOTICE", "avp(i:1)[$var(idx)]=$(avp(i:1)[$var(idx)]) >> avp(i:2)[$var(idx)]=$(avp(i:2)[$var(idx)])\n"); avp_pushto("$ru/username", >> "$(avp(i:2)[$var(idx)])"); >> rewritehostport("10.10.10.10:5060"); >> xlog("L_NOTICE", "$ru\n"); >> } else { >> xlog("L_NOTICE", "avp(i:1)[$var(idx)] == NULL\n"); >> } >> $var(idx) = $var(idx) + 1; >> } >> } else { >> xlog("L_NOTICE", "avp_db_query() returned no records\n"); >> }; > >Forget using avp_pushto, you can do: > > $rU = $(avp(i:2)[$var(idx)]); > >instead of: > > avp_pushto("$ru/username", "$(avp(i:2)[$var(idx)])"); > >-- >Iñaki Baz Castillo > >_______________________________________________ >Users mailing list >Users(a)lists.openser.org >http://lists.openser.org/cgi-bin/mailman/listinfo/users >

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[OpenSER-Users] avp_pushto question
by Jeremy George 02 Jan '08

02 Jan '08

Hello all, I have a small problem with avp_db_query() and avp_pushto() that I could use some help with. The manual select statement returns the expected values: mysql> select username,attribute,value from usr_preferences where username='55555'; +----------+-----------+------------+ | username | attribute | value | +----------+-----------+------------+ | 55555 | 1 | 2035552222 | | 55555 | 2 | 2035551111 | +----------+-----------+------------+ 2 rows in set (0.00 sec) The problem I have is that, when I run the configuration segment below only the first instance of avp_pushto in the loop produces the correct value. avp_delete("*"); $var(idx) = 0; if (avp_db_query("SELECT attribute, value FROM usr_preferences WHERE username='$rU' ORDER BY attribute")) { while($var(idx) < 5) { if ($(avp(i:1)[$var(idx)]) != NULL) { xlog("L_NOTICE", "avp(i:1)[$var(idx)]=$(avp(i:1)[$var(idx)]) avp(i:2)[$var(idx)]=$(avp(i:2)[$var(idx)])\n"); avp_pushto("$ru/username", "$(avp(i:2)[$var(idx)])"); rewritehostport("10.10.10.10:5060"); xlog("L_NOTICE", "$ru\n"); } else { xlog("L_NOTICE", "avp(i:1)[$var(idx)] == NULL\n"); } $var(idx) = $var(idx) + 1; } } else { xlog("L_NOTICE", "avp_db_query() returned no records\n"); }; So, the avp_db_query() works exactly as expected. The problem is that the second avp_pushto() doesn't change the value in the ruri username. avp(i:1)[0]=1 avp(i:2)[0]=2035552222 sip:2035552222@10.10.10.10:5060;user=phone avp(i:1)[1]=2 avp(i:2)[1]=2035551111 <-- This is the mismatch sip:2035552222@10.10.10.10:5060;user=phone <-- This is the mismatch avp(i:1)[2] == NULL avp(i:1)[3] == NULL avp(i:1)[4] == NULL As always, thanks for any help. - Jeremy

2 1

[Serusers] Unix Socket Problem
by frq ser 02 Jan '08

02 Jan '08

Hi I am running the simple configuration in which SER 0.9.7-pre1 serves as Proxy (at port 5060) and SER-SEMS is the media server (at port 5070). At port 5070, the SER version is 0.9.6-pre1 and SEMS version is 0.10.0-rc2. I have placed log commands in the config files just to record the process flow. Using SIPSAK, I am sending an INVITE (for SER-SEMS at :5070) through the proxy. My sipsak message is this: C:>sipsak -f invite.txt -s sip:200@192.168.3.208:5060 -vvv The request enters the proxy ser.cfg file and is redirected to the SER-SEMS (see the log messages below). The user name (200@...) is used by if statement to invoke the announcement application. However, I don't see any response from the unix socket (no announcement played) nor do I see any error message, and the connection finally times out. Can anyone please tell me how I can check whether the unix sockets are doing their job or not? Why does the process not return any error and why does it just time out? Thanks in advance for your help. Frq PS: Here are the (somewhat condensed) versions of SER and SEMS outputs (log messages mostly), SIPSAK responses, and the relevant config files. I apologise for sending a huge email. ------------------------------------------------------------------------------------------- SER output [root@xxx ~]# ser Listening on udp: 192.168.3.208 [192.168.3.208]:5060 tcp: 192.168.3.208 [192.168.3.208]:5060 aliases... WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 107520 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 2(0) INFO: fifo process starting: 5058 2(0) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(0) WARNING: no fifo_db_url given - fifo DB commands disabled! 0(5056) Inside the main route.... 0(5056) just passed the sanity checks... 0(5056) inside method-register... 0(5056) inside uri-myself... 0(5056) Inside route1... 0(5056) Inside the main route.... 0(5056) just passed the sanity checks... 0(5056) inside method-register... 0(5056) inside uri-myself... 0(5056) Inside route1... ------------------------------------------------------------------------------------------- SER-SEMS output [root@xxx ~]# /opt/ser-sems/sbin/ser -f /opt/ser-sems/etc/ser/ser.cfg Listening on udp: 192.168.3.208 [192.168.3.208]:5070 tcp: 192.168.3.208 [192.168.3.208]:5070 aliases .... WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 107520 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 2(0) INFO: fifo process starting: 5061 2(0) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(0) WARNING: no fifo_db_url given - fifo DB commands disabled! 0(5059) Inside route block.... 0(5059) Inside if-invite... 0(5059) Inside if-uri-200... 0(5059) Inside route block.... ------------------------------------------------------------------------------------------- INVITE message used by SIPSAK (invite.txt ) is this: INVITE sip:200@192.168.3.208:5070 SIPi/2.0 From: sip:rsf@192.168.3.77 To: sip:200@192.168.3.208:5070 Contact: sip:rsf@192.168.3.77 CSeq: 100 INVITE Call-ID: 9787963-3319555348-76751(a)192.168.3.77 Max-Forwards: 400 Content-Length: 0 ------------------------------------------------------------------------------------------- SIPSAK responses: fqdnhostname: 192.168.3.77 our Via-Line: Via: SIP/2.0/UDP 192.168.3.77:2646;branch=z9hG4bK.7271c2b5;rport;a lias New message with Via-Line: INVITE sip:200@192.168.3.208:5070 SIPi/2.0 Via: SIP/2.0/UDP 192.168.3.77:2646;branch=z9hG4bK.7271c2b5;rport;alias From: sip:rsf@192.168.3.77 To: sip:200@192.168.3.208:5070 Contact: sip:rsf@192.168.3.77 CSeq: 100 INVITE Call-ID: 9787963-3319555348-76751(a)192.168.3.77 Max-Forwards: 400 Content-Length: 0 request: INVITE sip:200@192.168.3.208:5070 SIPi/2.0 Via: SIP/2.0/UDP 192.168.3.77:2646;branch=z9hG4bK.7271c2b5;rport;alias From: sip:rsf@192.168.3.77 To: sip:200@192.168.3.208:5070 Contact: sip:rsf@192.168.3.77 CSeq: 100 INVITE Call-ID: 9787963-3319555348-76751(a)192.168.3.77 Max-Forwards: 400 Content-Length: 0 send to: UDP:192.168.3.208:5060 message received received from: UDP:192.168.3.208:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.3.77:2646;branch=z9hG4bK.7271c2b5;rport=2647;alias From: sip:rsf@192.168.3.77 To: sip:200@192.168.3.208:5070 CSeq: 100 INVITE Call-ID: 9787963-3319555348-76751(a)192.168.3.77 Server: Sip EXpress router (0.9.7-pre1 (i386/linux)) Content-Length: 0 Warning: 392 192.168.3.208:5060 "Noisy feedback tells: pid=5056 req_src_ip=192. 168.3.77 req_src_port=2647 in_uri=sip:200@192.168.3.208:5070 out_uri=sip:200@192 .168.3.208:5070 via_cnt==1" ** reply received after 0.000 ms ** SIP/2.0 100 trying -- your call is important to us provisional received; still waiting for a final response message received received from: UDP:192.168.3.208:5060 SIP/2.0 101 Trying - just wait a minute! Via: SIP/2.0/UDP 192.168.3.77:2646;branch=z9hG4bK.7271c2b5;rport=2647;alias From: sip:rsf@192.168.3.77 To: sip:200@192.168.3.208:5070 CSeq: 100 INVITE Call-ID: 9787963-3319555348-76751(a)192.168.3.77 Server: Sip EXpress router (0.9.6-sems (i386/linux)) Content-Length: 0 Warning: 392 192.168.3.208:5070 "Noisy feedback tells: pid=5059 req_src_ip=192. 168.3.208 req_src_port=5060 in_uri=sip:200@192.168.3.208:5070 out_uri=sip:200@19 2.168.3.208:5070 via_cnt==2" ** reply received after 16.000 ms ** SIP/2.0 101 Trying - just wait a minute! provisional received; still waiting for a final response message received received from: UDP:192.168.3.208:5060 Content-Length set to 0 New message with changed Content-Length: ACK sip:200@192.168.3.208:5070 SIPi/2.0 Via: SIP/2.0/UDP 192.168.3.77:2646;branch=z9hG4bK.7271c2b5;rport;alias From: sip:rsf@192.168.3.77 To: sip:200@192.168.3.208:5070 Contact: sip:rsf@192.168.3.77 CSeq: 100 ACK Call-ID: 9787963-3319555348-76751(a)192.168.3.77 Max-Forwards: 400 Content-Length: 0 reply with copyed To: ACK sip:200@192.168.3.208:5070 SIPi/2.0 Via: SIP/2.0/UDP 192.168.3.77:2646;branch=z9hG4bK.7271c2b5;rport;alias From: sip:rsf@192.168.3.77 To: sip:200@192.168.3.208:5070;tag=cc619c805c36ff52db55353ff7c0aad3-be50 Contact: sip:rsf@192.168.3.77 CSeq: 100 ACK Call-ID: 9787963-3319555348-76751(a)192.168.3.77 Max-Forwards: 400 Content-Length: 0 request: ACK sip:200@192.168.3.208:5070 SIPi/2.0 Via: SIP/2.0/UDP 192.168.3.77:2646;branch=z9hG4bK.7271c2b5;rport;alias From: sip:rsf@192.168.3.77 To: sip:200@192.168.3.208:5070;tag=cc619c805c36ff52db55353ff7c0aad3-be50 Contact: sip:rsf@192.168.3.77 CSeq: 100 ACK Call-ID: 9787963-3319555348-76751(a)192.168.3.77 Max-Forwards: 400 Content-Length: 0 send to: UDP:192.168.3.208:5060 SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 192.168.3.77:2646;branch=z9hG4bK.7271c2b5;rport=2647;alias From: sip:rsf@192.168.3.77 To: sip:200@192.168.3.208:5070;tag=cc619c805c36ff52db55353ff7c0aad3-be50 CSeq: 100 INVITE Call-ID: 9787963-3319555348-76751(a)192.168.3.77 Server: Sip EXpress router (0.9.6-sems (i386/linux)) Content-Length: 0 Warning: 392 192.168.3.208:5070 "Noisy feedback tells: pid=5060 req_src_ip=192. 168.3.208 req_src_port=5060 in_uri=sip:200@192.168.3.208:5070 out_uri=sip:200@19 2.168.3.208:5070 via_cnt==0" ** reply received 29235.000 ms after first send and -15.000 ms after last send ** SIP/2.0 408 Request Timeout final received ------------------------------------------------------------------------------------------- Ser.cfg (for proxy server at :5060) debug=3 fork=no log_stderror=yes listen=192.168.3.208 #2.13 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" 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" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) route { log(1, "Inside the main route....\n"); # ------------------------------------------------------------------------ # 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; }; log(3, "just passed the sanity checks...\n"); # ------------------------------------------------------------------------ # Record Route Section # ------------------------------------------------------------------------ if (method!="REGISTER") { record_route(); log(3, "inside method-register...\n"); }; # ------------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------------ if (loose_route()) { log(3, "inside loose route...\n"); route(1); break; }; # ------------------------------------------------------------------------ # Call Type Processing Section # ------------------------------------------------------------------------ if (uri!=myself) { log(3, "inside uri-myself...\n"); route(1); break; }; . . . route(1); } route[1] { # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ log(3, "Inside route1...\n"); if (!t_relay()) { sl_reply_error(); }; } . . ----------------------------------------------------------------------------------------- SER (Stack for SEMS at :5070) config file is this: # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) #/* 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=192.168.3.208 port=5070 fifo="/tmp/ser_fifo" unix_sock="/tmp/ser_sock" # ------------------ module loading ---------------------------------- loadmodule "/opt/ser-sems/lib/ser/modules/sl.so" loadmodule "/opt/ser-sems/lib/ser/modules/tm.so" loadmodule "/opt/ser-sems/lib/ser/modules/rr.so" loadmodule "/opt/ser-sems/lib/ser/modules/maxfwd.so" loadmodule "/opt/ser-sems/lib/ser/modules/usrloc.so" loadmodule "/opt/ser-sems/lib/ser/modules/registrar.so" loadmodule "/opt/ser-sems/lib/ser/modules/textops.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) modparam("tm", "pass_provisional_replies", 1) # ------------------------- request routing logic ------------------- route{ log(1, "Inside route block....\n"); # 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"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # ----------------------------------------------------------------------- # Redirection-related processing # ---------------------------------------------------------------------- if (method == "ACK" || method == "INVITE" || method == "BYE" || method == "CANCEL"){ # switch to stateful mode if (!t_newtran()){ sl_send_reply("500", "Could not create transaction"); break; }; # prevent timeout on the other side t_reply("101", "Trying - just wait a minute!"); if (method == "INVITE"){ log(2, "Inside if-invite...\n"); # Redirect the call to the 'conference' plugin # in case the URI begins with 100 if (uri =~"sip:100*@"){ # assumes that SMES configuration parameter # 'socket_name=' has been set to '/tmp/sems_sock' log(3, "Inside if-uri-100...\n"); if (!t_write_unix("/tmp/sems_sock","conference")){ t_reply("500","Error contacting SEMS 1"); }; break; }; # Redirect the call to the 'announcement' plugin in case # the URI begins with 200 if (uri =~"sip:200*@"){ log(3, "Inside if-uri-200...\n"); if (!t_write_unix("/tmp/sems_sock","announcement")){ t_reply("500","Error contacting SEMS 2"); log(3, "just past 200-unixsocket...\n"); }; break; }; # No service number, redirect to voicemail # Load email address into AVP so that voicemail gets the # callee's email address (see below for avpops modparams) #avp_db_load("$uri", "$email/$email_scheme"); #if(!t_write_unix("/tmp/sems_sock","voicemail")){ # t_reply("500","Error contacting SEMS"); #}; break; } . . } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } ----------------------------------------------------------------------------------------- Finally, the relevant parts of sems.conf file are these: # $Id: sems.conf.sample 265 2007-03-07 21:31:54Z sayer $ # # sems.conf.sample # # Sip Express Media Server (sems) # # sample configuration file # . . socket_name=/tmp/sems_sock reply_socket_name=/tmp/sems_rsp_sock ser_socket_name=/tmp/ser_sock listen=192.168.3.208 sip_port=5070 . . .

2 1

[OpenSER-Users] Some issues with Session-Timer module
by Franz Edler 31 Dec '07

31 Dec '07

Hi, I have just made some tests with sst-module (OpenSER v1.3) and I found the following two issues: I case the min-se value of sst-module is higher than the Session-Expires header field of a received INVITE OpenSER rejects the INVITE request with 422 response (that's as expected). But: 1. the Min-SE header field in 422 response (packet 18 in attached trace) does not end with CR/LF so that it is concatenated with the next header field (Server header) syslog prints in this case: ERROR:sst:sst_dialog_terminate_CB: DID 0xb5582d80 failed (canceled). Terminating session. 2. OpenSER in addition continues to forward the already rejected INVITE transaction with "100 Giving a try"(packet 19) and forwards the INVITE to the UAS (packet 20). The next issue I discovered is with the UAC (Xlite): The repeated INVITE (caused by the 422 Response) does not increment the CSeq header field so that it is regarded as same request leading to a "482 Merged request" rejection at the UAS (Eyebeam). I have attached the tcpdump-trace and the according SIP-scenario trace. Any comments appreciated. Regards Franz

1 1

[OpenSER-Users] (no subject)
by VoIP Forums www.Go4Calls.com 30 Dec '07

30 Dec '07

Hi Friends, How we can authorize Clients with their static IP Address in MySQL subscriber table? We don't want to provide them username or PWS and they also don't need to register. Regards, www.Go4Calls.Com VoIP Forums _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

2 1

[OpenSER-Users] dialog module: dialog data remain in DB after "dlg_end_dlg"
by Franz Edler 29 Dec '07

29 Dec '07

Hi, I just found that after forced end of a dialog (MI command "dlg_end_dlg") the dialog data remain in the database. The dialog list in memory (MI command "dlg_list") is cleared after forced end of a dialog as expected. Shouldn't the dialog data in the DB also be deleted? Regards Franz

1 0

Re: [OpenSER-Users] ERROR:avpops:check_avp: value types don't match
by Jayesh Nambiar 29 Dec '07

29 Dec '07

Thanks a lot. Its working now. It had to be like: if(avp_check("$avp(i:103)", "eq/i:0/g")) { do certain things } else { do certain things } w/regards, Jayesh ----- Original Message ---- From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> To: Jayesh Nambiar <voip_freak(a)yahoo.co.in> Cc: openser <users(a)openser.org> Sent: Friday, 28 December, 2007 2:54:17 PM Subject: Re: [OpenSER-Users] ERROR:avpops:check_avp: value types don't match Hi Jayesh, try to use the avp_print() debug function to see what avps you have loaded after the avp_db_query() and check the type. Note you need to use debug=6 . http://www.openser.org/docs/modules/1.3.x/avpops.html#AEN611 regards, bogdan Jayesh Nambiar wrote: > Hi All, > I am using the function avp_db_query to query a table and fetch few > values. The query is as follows: > "select subscriptionid, ip, media_status from domestic where > prefix=$rU", "$avp(i:101), $avp(i:102), $avp(i:103)". > > Now the media_status column is a column with datatype as integer which > will contain either 0 or 1. I need to check the value in this column > so I use the following: > > if(avp_check("$avp(i:103)", "eq/0/i")) { > do certain things; > } else { > do certain things; > }; > > Even when the value at media_status column is 0 it does not get into > that condition, it always executes the else condition and I see the > following in the logs: > ERROR:avpops:check_avp: value types don't match > > Can someone please help me to overcome this error. When I keep the > datatype of media_status column as varchar it works fine, but I dont > want to do that as that column is going to conatin only integer values. > > Thanks in advance. > > w/regards, > Jayesh > > > ------------------------------------------------------------------------ > Forgot the famous last words? Access your message archive online. > Click here. > <http://in.rd.yahoo.com/tagline_webmessenger_4/*http://in.messenger.yahoo.co…> > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > Get the freedom to save as many mails as you wish. To know how, go to http://help.yahoo.com/l/in/yahoo/mail/yahoomail/tools/tools-08.html

1 0

[Serusers] establishment call failure
by dimitris(yahoo) 29 Dec '07

29 Dec '07

hi, i am experiencing on little problem establishishing calls with my SER. First of all , i am running ser under linux opensuse, (ser version: 0.9.6 ) , i also installed rtpproxy and made the needed changes to my ser.cfg. When i "connetct" 2 different users with my SER locally , using softphone, everything looks fine. Gennerally the registration procedure using MySQL works fine(locally or not). When i am connected locally, a remote user can call me with no problems... The problem is the following, and i would appreciate if you could help me... When i try to call a remote user, the call cannot be established from my side. At the other side, the establishment of the call is ok. The message is the following: No ACK recieved... PS: I am only working now, till the level of the establishment of the calls... When i solve this problem , i will continue to the voice and video transmition, where i will need your help for sure... PS2: In order rtpproxy to run on the same computer with SER, i type rtpproxy with no parameters ?? can you help me with that too ??? Thank you in advance I send you also my ser.cfg

2 1

[OpenSER-Users] A Happy New Year
by Bogdan-Andrei Iancu 28 Dec '07

28 Dec '07

Hi everybody, I hope you found under the tree something that made you smile and happy, something that made you think about 2007 as a fulfilled year. I and the Voice System team wish you A Happy New Year, a good party and rest over the years and see you in 2008 ;) Best regards, Bogdan

1 0

[OpenSER-Users] xlog: color PV moved in xlog module ?
by Iñaki Baz Castillo 28 Dec '07

28 Dec '07

Hi, a change in 1.3 is: xlog: color PV moved in xlog module Could somebody explain what does it mean? Thanks a lot. -- Iñaki Baz Castillo

2 2

[OpenSER-Users] ERROR:avpops:check_avp: value types don't match
by Jayesh Nambiar 28 Dec '07

28 Dec '07

Hi All, I am using the function avp_db_query to query a table and fetch few values. The query is as follows: "select subscriptionid, ip, media_status from domestic where prefix=$rU", "$avp(i:101), $avp(i:102), $avp(i:103)". Now the media_status column is a column with datatype as integer which will contain either 0 or 1. I need to check the value in this column so I use the following: if(avp_check("$avp(i:103)", "eq/0/i")) { do certain things; } else { do certain things; }; Even when the value at media_status column is 0 it does not get into that condition, it always executes the else condition and I see the following in the logs: ERROR:avpops:check_avp: value types don't match Can someone please help me to overcome this error. When I keep the datatype of media_status column as varchar it works fine, but I dont want to do that as that column is going to conatin only integer values. Thanks in advance. w/regards, Jayesh Now you can chat without downloading messenger. Go to http://in.messenger.yahoo.com/webmessengerpromo.php

2 1

[OpenSER-Users] LCR module: missing the link between lcr and gw
by CSB 28 Dec '07

28 Dec '07

I have read and re-read the overview of the lcr module but still think I'm failing to understand it. What I want to achieve is routing to different gateways based on number dialled e.g. 001, 0044, 0086 should result in different gateways. Looking at the tables, I would guess that the link is lcr -> gw_grp -> gw as shown in the following example: lcr prefix from_uri grp_id priority 001 1 1 001 2 2 0044 3 1 0086 4 1 0086 5 2 gw_grp grp_id grp_name 1 US 1 2 IS 2 3 UK 1 4 China 1 5 China 2 gw gw_name grp_id ip_addr port uri_scheme transport strip prefix A 1 192.1 5060 1 011 B 2 192.2 5060 1 011 etc I don't think I have understood correctly because the documentation implies that gw_grp is optional. Is there a tutorial for this module that could calrify my understanding? I'm also trying to understand what the various avps are used for. Any advice appreciated. Regards Cameron

2 3

[OpenSER-Users] LCR module gw database IP address column
by CSB 28 Dec '07

28 Dec '07

According to http://www.openser.org/docs/OpenSER_Database_Structure.html the ip_addr column of the gw table is int. Am I missing something obvious? How does the IP address (e.g. 127.0.0.1) get stored in such a column? Regards Cameron

2 2

[OpenSER-Users] SEMS 0.10.0 incompatible with openser 1.3.0?
by CHEN XUEQIN 28 Dec '07

28 Dec '07

Hi all: I just tested SEMS media server 0.10.0 with openser 1.3.0. The test case is : 1. openser get SIP message, indentify message by SIP URI, and deliver to SEMS by UNIX socket [openser quote] modparam("mi_datagram", "socket_name", "/tmp/ser_sock") .. if (uri=~"sip:100.*@") { # assumes that Sems configuration parameter 'socket_name=' # has been set to /tmp/sems_sock if(!t_write_unix("/tmp/sems_sock","conference")) { t_reply("500","error contacting sems"); }; exit; }; [/openser quote] 2. SEMS get SIP message, give a response through mi_datagram, the response is like [response snip] :t_reply:/tmp/1D8FFAE5-47746119000C9DF2-B6E03B90 200 OK 41621:1987570916 7D41B2DF-47746119000C89B3-B7CA76C0 Contact: <sip:100@192.168.35.40> Content-Type: application/sdp . v=0^M o=username 0 0 IN IP4 192.168.35.40^M s=session^M c=IN IP4 192.168.35.40^M t=0 0^M m=audio 10002 RTP/AVP 0 8 3 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:3 GSM/8000^M a=rtpmap:103 telephone-event/8000^M a=fmtp:101 0-15^M . [/response snip] But the problem is that openser says " command not available". The detail log is: Dec 28 10:36:09 [17682] ERROR:mi_datagram:identify_command: the request's first line is invalid :no newline after the second : Dec 28 10:36:09 [17682] ERROR:mi_datagram:mi_datagram_server: command not available Dec 28 10:36:09 [17682] DBG:mi_datagram:mi_datagram_server: received :t_reply:/tmp/70FAB4E3-47746119000CA82E-B6E03B90 500 could not send response 41621:1987570916 3E69BD05-47746119000CA86F-B6E03B90 When I replaced openser 1.3.0 with ser-0.9.6 bundled with SEMS 0.10.0, everything was OK. So, is openser 1.3.0 incompatible with SEMS 0.10.0? Or I made a mistake in openser.cfg? Any clue is welcome. Thanks in advance. Regards, Chen Xueqin

2 2

[OpenSER-Users] Problem installing openser1.3.0 with carrieroute module
by Ricardo Martinez 27 Dec '07

27 Dec '07

Hello list. I'm getting this error trying to install the last version of openser. route_config.c:39:21: error: confuse.h: No such file or directory Can someone help me here? Thanks in advance Regards, Ricardo Martinez.-

3 3

[OpenSER-Users] ser + NAT not audio both side
by satish patel 27 Dec '07

27 Dec '07

Dear all I have installed openser with mediaproxy and it is working fine without nat but behind the nat Xlite register successfully and call established but thing is that no audio on both side after 1 min call automatically disconnected Debug error [root@proxy ~]# /usr/local/mediaproxy/mediaproxy.py --no-fork warning: accounting is enabled but the accounting module is missing. accounting is not available! Listening for commands on local socket `/var/run/mediaproxy.sock' Listening for remote commands is disabled Using IP address `203.124.16.109 # This is where OpenSER installed' for the RTP/RTCP proxy request YTEzN2ZiNTIwZDhkNDI0MDE1N2NjYmZkY2FlODc1MWY. 202.131.100.102:8128:audio 59.96.156.122 example.com local 192.168.104.164 remote X-Lite=20release=201011s=20stamp=2041150 info=from:123@example.com,to:111@example.com,fromtag:a9746f3e,totag: session YTEzN2ZiNTIwZDhkNDI0MDE1N2NjYmZkY2FlODc1MWY.: started. listening on 203.124.16.109:#,This,is,where,OpenSER,installed,60000 execution time: 1.72 ms lookup YTEzN2ZiNTIwZDhkNDI0MDE1N2NjYmZkY2FlODc1MWY. 192.168.104.164:13930:audio 192.168.104.164 example.com local example.com unknown X-Lite=20release=201002tx=20stamp=2029712 info=from:123@example.com,to:111@example.com,fromtag:a9746f3e,totag:e376233a execution time: 0.28 ms lookup YTEzN2ZiNTIwZDhkNDI0MDE1N2NjYmZkY2FlODc1MWY. 192.168.104.164:13930:audio 192.168.104.164 example.com local example.com unknown X-Lite=20release=201002tx=20stamp=2029712 info=from:123@example.com,to:111@example.com,fromtag:a9746f3e,totag:e376233a execution time: 0.37 ms lookup YTEzN2ZiNTIwZDhkNDI0MDE1N2NjYmZkY2FlODc1MWY. 192.168.104.164:13930:audio 192.168.104.164 example.com local example.com unknown X-Lite=20release=201002tx=20stamp=2029712 info=from:123@example.com,to:111@example.com,fromtag:a9746f3e,totag:e376233a execution time: 0.37 ms session YTEzN2ZiNTIwZDhkNDI0MDE1N2NjYmZkY2FlODc1MWY.: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) session YTEzN2ZiNTIwZDhkNDI0MDE1N2NjYmZkY2FlODc1MWY.: ended (did timeout). Is there any nat issuse of RTP port problem ??? ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org --------------------------------- Never miss a thing. Make Yahoo your homepage.

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[OpenSER-Users] avp_db_query information
by raviprakash sunkara 27 Dec '07

27 Dec '07

Hello Users Merry X-mas and Adv . Happy New Year So long back ago , I posted the same imformation to mailing listing. Is in Black Cat of OpenSER 1.3.0, avp_db_query can execute the function and stores procedures of MySQL database? -- Thanks &Regards Ravi Prakash Sunkara VoIP Development Tech Lead

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Re: [OpenSER-Users] [OT] Any voicemail server supporting RFC 3842 (MWI) ?
by Iñaki Baz Castillo 27 Dec '07

27 Dec '07

2007/12/27, Juha Heinanen <jh(a)tutpro.com>: > Iñaki Baz Castillo writes: > > > Thanks, I'll try it. > > i made a mistake, you need to use mailbox application rather than > voicemail application. Yes, I already realized of that ;) The problem is that mailbox/mailbox_query apps seem to use IMAP as storage while I prefer to use MySQL. Anyway there is not too much info about those two SEMS applications, I think that for now I'll try with Asterisk + ODBC storage. Thanks for all. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[OpenSER-Users] [OT] Any voicemail server supporting RFC 3842 (MWI) ?
by Iñaki Baz Castillo 27 Dec '07

27 Dec '07

Hi, Presence_MWI module support PUBLISH or SUBSCRIBE for "Event: message-summary" according to RFC 3842), so a voicemail server could publish voicemail status. Asterisk (as a voicemail server) just allows unsollicited message waiting indication (not RFC 3842). Is there any GPL voicemail system supporting RFC 3842? Regards. -- Iñaki Baz Castillo

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[OpenSER-Users] avp_db_query() problem
by Jeremy George 26 Dec '07

26 Dec '07

Hello All, I expect I have a syntax fluff with an avp_db_query() example, but I can't find it. A pointer to the solution would be greatly appreciated. I'm running mysql 5: Spongy% mysql -V mysql Ver 14.12 Distrib 5.0.27, for pc-linux-gnu (i686) using readline 5.0 and openser 1.3.0: Spongy% openser -V version: openser 1.3.0-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. svnrevision: unknown @(#) $Id: main.c 3304 2007-12-11 12:55:11Z bogdan_iancu $ main.c compiled on 12:09:35 Dec 14 2007 with gcc 3.4.6 A direct query of the mysql usr_preferences table works as expected: mysql> select attribute, value from usr_preferences where username='55555' order by attribute; +-----------+-------+ | attribute | value | +-----------+-------+ | 1 | 55510 | | 2 | 55520 | | 3 | 55530 | | 4 | 55540 | | 5 | 55500 | +-----------+-------+ 5 rows in set (0.00 sec) This configuration taken almost directly from avp_db_query.html mpath="/usr/local/lib/openser/modules" ... loadmodule "mysql.so" loadmodule "avpops.so" loadmodule "xlog.so" ... avp_delete("*"); if (avp_db_query("SELECT attribute, value FROM usr_preferences WHERE username='55555' ORDER BY attribute")) { xlog("L_NOTICE", "avp(i:1[1])=$avp(i:1[1]) avp(i:2[1])=$avp(i:2[1])\n"); xlog("L_NOTICE", "avp(i:1[2])=$avp(i:1[2]) avp(i:2[2])=$avp(i:2[2])\n"); xlog("L_NOTICE", "avp(i:1[3])=$avp(i:1[3]) avp(i:2[3])=$avp(i:2[3])\n"); xlog("L_NOTICE", "avp(i:1[4])=$avp(i:1[4]) avp(i:2[4])=$avp(i:2[4])\n"); xlog("L_NOTICE", "avp(i:1[5])=$avp(i:1[5]) avp(i:2[5])=$avp(i:2[5])\n"); } else { xlog("L_NOTICE", "avp_db_query() returned no records\n"); }; produces these errors: Dec 26 14:16:28 [25226] WARNING:core:main: no fork mode Dec 26 14:16:28 [25226] INFO:core:init_tcp: using epoll_lt as the TCP io watch method (auto detected) Dec 26 14:16:28 [25226] NOTICE:core:main: version: openser 1.3.0-tls (i386/linux) Dec 26 14:16:28 [25226] INFO:core:main: using 32 Mb shared memory Dec 26 14:16:28 [25226] INFO:core:main: using 1 Mb private memory per process Dec 26 14:16:28 [25226] INFO:sl:mod_init: Initializing StateLess engine Dec 26 14:16:28 [25226] INFO:tm:mod_init: TM - initializing... Dec 26 14:16:28 [25226] INFO:acc:mod_init: initializing... Dec 26 14:16:28 [25226] INFO:maxfwd:mod_init: initializing... Dec 26 14:16:28 [25226] INFO:usrloc:ul_init_locks: locks array size 512 Dec 26 14:16:28 [25226] INFO:registrar:mod_init: initializing... Dec 26 14:16:28 [25226] INFO:auth:mod_init: initializing... Dec 26 14:16:28 [25226] INFO:auth_db:mod_init: initializing... Dec 26 14:16:28 [25226] INFO:textops:mod_init: initializing... Dec 26 14:16:28 [25226] INFO:exec:mod_init: exec - initializing Dec 26 14:16:28 [25226] INFO:options:mod_init: initializing... Dec 26 14:16:28 [25226] INFO:avpops:avpops_init: initializing... Dec 26 14:16:28 [25226] INFO:xlog:mod_init: initializing... Dec 26 14:16:28 [25226] ERROR:core:parse_avp_name: invalid ID <1[1]> not a number Dec 26 14:16:28 [25226] ERROR:core:pv_parse_avp_name: bad avp name [1[1]] Dec 26 14:16:28 [25226] ERROR:core:pv_parse_spec: pvar "avp" has an invalid name param [1[1]] Dec 26 14:16:28 [25226] ERROR:core:pv_parse_spec: wrong char [)/41] in [$avp(i:1[1]) avp(i:2[1])=$avp(i:2[1]) ] at [11 (5)] Dec 26 14:16:28 [25226] ERROR:xlog:xdbg_fixup: ERROR: wrong format[avp(i:1[1])=$avp(i:1[1]) avp(i:2[1])=$avp(i:2[1]) ] Dec 26 14:16:28 [25226] ERROR:core:fix_actions: fixing failed (code=-1) at cfg line 288 Dec 26 14:16:28 [25226] ERROR:core:main: failed to fix configuration with err code -1 Thanks much for any help. - Jeremy

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[OpenSER-Users] avp_db_query problem
by Jeremy George 26 Dec '07

26 Dec '07

Hello All, I expect I have a syntax fluff with an avp_db_query() example, but I can't find it. A pointer to the solution would be greatly appreciated. I'm running mysql 5: Spongy% mysql -V mysql Ver 14.12 Distrib 5.0.27, for pc-linux-gnu (i686) using readline 5.0 and openser 1.3.0: Spongy% openser -V version: openser 1.3.0-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. svnrevision: unknown @(#) $Id: main.c 3304 2007-12-11 12:55:11Z bogdan_iancu $ main.c compiled on 12:09:35 Dec 14 2007 with gcc 3.4.6 A direct query of the mysql usr_preferences table works as expected: mysql> select attribute, value from usr_preferences where username='55555' order by attribute; +-----------+-------+ | attribute | value | +-----------+-------+ | 1 | 55510 | | 2 | 55520 | | 3 | 55530 | | 4 | 55540 | | 5 | 55500 | +-----------+-------+ 5 rows in set (0.00 sec) This configuration taken almost directly from avp_db_query.html mpath="/usr/local/lib/openser/modules" ... loadmodule "mysql.so" loadmodule "avpops.so" loadmodule "xlog.so" ... avp_delete("*"); if (avp_db_query("SELECT attribute, value FROM usr_preferences WHERE username='55555' ORDER BY attribute")) { xlog("L_NOTICE", "avp(i:1[1])=$avp(i:1[1]) avp(i:2[1])=$avp(i:2[1])\n"); xlog("L_NOTICE", "avp(i:1[2])=$avp(i:1[2]) avp(i:2[2])=$avp(i:2[2])\n"); xlog("L_NOTICE", "avp(i:1[3])=$avp(i:1[3]) avp(i:2[3])=$avp(i:2[3])\n"); xlog("L_NOTICE", "avp(i:1[4])=$avp(i:1[4]) avp(i:2[4])=$avp(i:2[4])\n"); xlog("L_NOTICE", "avp(i:1[5])=$avp(i:1[5]) avp(i:2[5])=$avp(i:2[5])\n"); } else { xlog("L_NOTICE", "avp_db_query() returned no records\n"); }; produces these errors: Dec 26 14:16:28 [25226] WARNING:core:main: no fork mode Dec 26 14:16:28 [25226] INFO:core:init_tcp: using epoll_lt as the TCP io watch method (auto detected) Dec 26 14:16:28 [25226] NOTICE:core:main: version: openser 1.3.0-tls (i386/linux) Dec 26 14:16:28 [25226] INFO:core:main: using 32 Mb shared memory Dec 26 14:16:28 [25226] INFO:core:main: using 1 Mb private memory per process Dec 26 14:16:28 [25226] INFO:sl:mod_init: Initializing StateLess engine Dec 26 14:16:28 [25226] INFO:tm:mod_init: TM - initializing... Dec 26 14:16:28 [25226] INFO:acc:mod_init: initializing... Dec 26 14:16:28 [25226] INFO:maxfwd:mod_init: initializing... Dec 26 14:16:28 [25226] INFO:usrloc:ul_init_locks: locks array size 512 Dec 26 14:16:28 [25226] INFO:registrar:mod_init: initializing... Dec 26 14:16:28 [25226] INFO:auth:mod_init: initializing... Dec 26 14:16:28 [25226] INFO:auth_db:mod_init: initializing... Dec 26 14:16:28 [25226] INFO:textops:mod_init: initializing... Dec 26 14:16:28 [25226] INFO:exec:mod_init: exec - initializing Dec 26 14:16:28 [25226] INFO:options:mod_init: initializing... Dec 26 14:16:28 [25226] INFO:avpops:avpops_init: initializing... Dec 26 14:16:28 [25226] INFO:xlog:mod_init: initializing... Dec 26 14:16:28 [25226] ERROR:core:parse_avp_name: invalid ID <1[1]> not a number Dec 26 14:16:28 [25226] ERROR:core:pv_parse_avp_name: bad avp name [1[1]] Dec 26 14:16:28 [25226] ERROR:core:pv_parse_spec: pvar "avp" has an invalid name param [1[1]] Dec 26 14:16:28 [25226] ERROR:core:pv_parse_spec: wrong char [)/41] in [$avp(i:1[1]) avp(i:2[1])=$avp(i:2[1]) ] at [11 (5)] Dec 26 14:16:28 [25226] ERROR:xlog:xdbg_fixup: ERROR: wrong format[avp(i:1[1])=$avp(i:1[1]) avp(i:2[1])=$avp(i:2[1]) ] Dec 26 14:16:28 [25226] ERROR:core:fix_actions: fixing failed (code=-1) at cfg line 288 Dec 26 14:16:28 [25226] ERROR:core:main: failed to fix configuration with err code -1 Thanks much for any help. - Jeremy

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[OpenSER-Users] SER problem with NAT
by satish patel 26 Dec '07

26 Dec '07

Dear all I have installed SER with multidomain support and it is working fine i have host 2 domain abc.com and example.com now i dont have any NAT between server and client But now i have few more client. they are behind NAT so i have find nathelper help and configure with mediaproxy now what happend when i test this configuration in my demo setup [192.168.1.2_SIP_Client]------------[SER + Mediaproxy + nathaper]------[192.168.1.3_SIP_Client] MY server IP address is 192.168.1.1 everyone in same subnet no NAT between them ser configuration with NAT + Mediaproxy and mediaproxy running on same machine.. When i trying to register Xlite on server i got error 483 Too many Hops why is there any domain related problem or something else when i remove name server in /etc/resolve.conf i got Timeout error My --------- ser.cfg # $Id: nat-mediaproxy.cfg 51 2006-01-31 13:28:04Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ debug=7 fork=yes log_stderror=yes listen=192.168.104.20 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=yes rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://root:tulip@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/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" modparam("auth_db|domain|uri_db|usrloc", "db_url", "mysql://root:tulip@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("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") modparam("usrloc|registrar|auth_db|avpops|group", "use_domain", 1) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) 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")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.168.104.20:5060;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"); break; }; if (method=="INVITE") { 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 (uri!=myself) { route(4); route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (is_from_local()) { if (method=="REGISTER") { route(2); break; }; }; lookup("aliases"); if (uri!=myself) { 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") { end_media_session(); }; sl_reply_error(); }; } # REGISTER Message Handling # ------------------------- route[2] { 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 (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; 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(); }; } ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org --------------------------------- Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

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[Serusers] ser nathelper too many hope isse
by satish patel 26 Dec '07

26 Dec '07

Dear all I have install ser with nathelper but when i try to register xlite i got this error messge 3(31047) WARNING:vqm_resize: resize(0) called 3(31047) DEBUG: add_to_tail_of_timer[2]: 0x40586698 3(31047) DEBUG: reply relayed. buf=0x81220c0: SIP/2.0 4..., shmem=0x405a01d0: SIP/2.0 4 3(31047) DEBUG: cleanup_uac_timers: RETR/FR timers reset 3(31047) DEBUG:destroy_avp_list: destroying list (nil) 3(31047) receive_msg: cleaning up 3(31047) SIP Reply (status): 3(31047) version: <SIP/2.0> 3(31047) status: <483> 3(31047) reason: <Too Many Hops> 3(31047) parse_headers: flags=1 3(31047) Found param type 232, <branch> = <z9hG4bK79cd.4f2da674.0>; state=16 3(31047) end of header reached, state=5 3(31047) parse_headers: Via found, flags=1 3(31047) parse_headers: this is the first via 3(31047) After parse_msg... 3(31047) forward_reply: found module tm, passing reply to it 3(31047) DEBUG: t_check: msg id=258 global id=257 T start=0xffffffff 3(31047) parse_headers: flags=17 3(31047) Found param type 232, <branch> = <z9hG4bK79cd.3f2da674.0>; state=16 3(31047) end of header reached, state=5 3(31047) parse_headers: Via found, flags=17 3(31047) parse_headers: this is the second via 3(31047) Found param type 232, <branch> = <z9hG4bK79cd.2f2da674.0>; state=16 3(31047) end of header reached, state=5 3(31047) parse_headers: Via found, flags=17 3(31047) Found param type 232, <branch> = <z9hG4bK79cd.1f2da674.0>; state=16 3(31047) end of header reached, state=5 3(31047) parse_headers: Via found, flags=17 3(31047) Found param type 232, <branch> = <z9hG4bK79cd.0f2da674.0>; state=16 3(31047) end of header reached, state=5 3(31047) parse_headers: Via found, flags=17 3(31047) Found param type 232, <branch> = <z9hG4bK79cd.fe2da674.0>; state=16 3(31047) end of header reached, state=5 3(31047) parse_headers: Via found, flags=17 3(31047) Found param type 235, <rport> = <5061>; state=6 3(31047) Found param type 232, <branch> = <z9hG4bK00C839C4E76F483741C969446436A708>; state=16 3(31047) end of header reached, state=5 3(31047) parse_headers: Via found, flags=17 3(31047) DEBUG: add_param: tag=2ec2004a41f4343ae885c4e522e0f260.52d6 3(31047) end of header reached, state=29 3(31047) DEBUG: get_hdr_field: <To> [65]; uri=[sip:444@abc.com] 3(31047) DEBUG: to body [ser <sip:444@abc.com>] 3(31047) get_hdr_field: cseq <CSeq>: <41216> <REGISTER> 3(31047) DEBUG: t_reply_matching: hash 56471 label 1198183156 branch 0 3(31047) DEBUG: t_reply_matching: reply matched (T=0x40584cb8)! 3(31047) DEBUG: t_check: msg id=258 global id=258 T end=0x40584cb8 3(31047) DEBUG: reply_received: org. status uas=0, uac[0]=0 local=0 is_invite=0) 3(31047) parse_headers: flags=64 3(31047) DEBUG: get_hdr_body : content_length=0 3(31047) found end of header 3(31047) ->>>>>>>>> T_code=0, new_code=483 3(31047) DEBUG: relay_reply: branch=0, save=0, relay=0 3(31047) old size: 920, new size: 857 3(31047) build_res_from_sip_res: copied size: orig:90, new: 27, rest: 830 msg= SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP 192.168.104.20;branch=z9hG4bK79cd.3f2da674.0 Via: SIP/2.0/UDP 192.168.104.20;branch=z9hG4bK79cd.2f2da674.0 Via: SIP/2.0/UDP 192.168.104.20;branch=z9hG4bK79cd.1f2da674.0 Via: SIP/2.0/UDP 192.168.104.20;branch=z9hG4bK79cd.0f2da674.0 Via: SIP/2.0/UDP 192.168.104.20;branch=z9hG4bK79cd.fe2da674.0 Via: SIP/2.0/UDP 192.168.104.20:5061;rport=5061;branch=z9hG4bK00C839C4E76F483741C969446436A708 From: ser <sip:444@abc.com>;tag=889909258 To: ser <sip:444@abc.com>;tag=2ec2004a41f4343ae885c4e522e0f260.52d6 Call-ID: 7F06D3E766CAF64EAA23E6D089B86795(a)abc.com CSeq: 41216 REGISTER Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 192.168.104.20:5060 "Noisy feedback tells: pid=31047 req_src_ip=192.168.104.20 req_src_port=5060 in_uri=sip:abc.com out_uri=sip:abc.com via_cnt==18" $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Meet people who discuss and share your passions. Join them now.

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[Serusers] ser + nat + mediaproxy registration issue
by satish patel 26 Dec '07

26 Dec '07

I have installed ser-0.9.x and install mediaproxy on same machine where ser running Now when i going to register Xlite i got error message 483-Too Many Hop http://onsip.org example configuration my configuration is # $Id: nat-mediaproxy.cfg 51 2006-01-31 13:28:04Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ debug=7 fork=yes log_stderror=yes listen=192.168.104.20 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://root:tulip@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/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" modparam("auth_db|domain|uri_db|usrloc", "db_url", "mysql://root:tulip@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) 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")) { # # INSERT YOUR IP ADDRESS HERE # record_route_preset("192.168.104.20:5060;nat=yes"); # } else if (method!="REGISTER") { # record_route(); # }; record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # ----------------------------------------------------------------- # 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"); break; }; if (method=="INVITE") { 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 (uri!=myself) { route(4); route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { 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") { 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 (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; 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(); }; } $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Chat on a cool, new interface. No download required. Click here.

1 0

[OpenSER-Users] [CPL] Is possible to group <not-present> and <otherwise>?
by Iñaki Baz Castillo 24 Dec '07

24 Dec '07

Hi. The following script doesn't proxy to "sip:voicemail@domain" if no "Subject" header exits: <cpl> <incoming> <string-switch field="subject"> <string contains="hello"> <reject status="reject" reason="Subjet = hello" /> </string> <otherwise> <location url="sip:voicemail@domain" /> <proxy /> </location> </otherwise> </string-switch> </incoming> </cpl> For that I need to add <not-present> and clone <otherwise> content: <cpl> <incoming> <string-switch field="subject"> <string contains="hello"> <reject status="reject" reason="Subjet = hello" /> </string> <not-present> <location url="sip:voicemail@domain" /> <proxy /> </location> </not-present> <otherwise> <location url="sip:voicemail@domain" /> <proxy /> </location> </otherwise> </string-switch> </incoming> </cpl> Is there any way to group <not-present> and <otherwise>? -- Iñaki Baz Castillo <ibc(a)aliax.net>

1 1

[OpenSER-Users] Christmas Wishes
by Dan-Cristian Bogos 24 Dec '07

24 Dec '07

With this occasion I would like to wish you all "Merry Christmas and Prosperous New Year" and express my hopes in a continuous grow of OpenSER project during the year to come . Best regards, DanB

2 1

[OpenSER-Users] [CPL] subfield="tel" without "user=phone" parameter?
by Iñaki Baz Castillo 24 Dec '07

24 Dec '07

Hi, in order to match a callee URI as a telephone number (discarding punctuation and separator characters) I should user "tel" as "subfield" value, but this is only valid if "user=phone" parameters exists in the matching URI. RFC 3880 says: The "tel" subfield is defined to be the "user" part of the URI, with visual separators stripped, if the "user=phone" parameter is given to the URI, or if the server is otherwise configured to recognize the user part as a telephone number. Is there any way OpenSer CPL module to recognize "From" user part as telephone number even if no "user=phone" parameter exists? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Re: [OpenSER-Users] [CPL] recurse="yes" doesn't work ? [SOLVED]
by Iñaki Baz Castillo 24 Dec '07

24 Dec '07

2007/12/24, Iñaki Baz Castillo <ibc(a)aliax.net>: > But in fact, the 302 arrives to the callee as if not "recurse" > parameter exists. Maybe I haven't understood the meaning of this > "recurse" parameter? or could it be a bug? Yes, a bug of mine XD For this feature working I must set a CPL parameter ("0 = dissable" by default): modparam("cpl-c","proxy_recurse",2) Now it works :) -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[OpenSER-Users] [CPL] recurse="yes" doesn't work ?
by Iñaki Baz Castillo 24 Dec '07

24 Dec '07

Hi, I try to set "recurse" parameter in order to automatically proxy the call to "Contact" URI present in a 302 response from called UAS: <lookup source="registration"> <success> <proxy timeout="8" recurse="yes"> </proxy> </success> </lookup> Now I set a redirection in the called UAS so it replies a 302. Reading RFC 3880 I assume that the proxy should create a branch to the 302 "Contact" URI instead of passing the 302 to the callee: 6.1. Proxy [...] The second optional parameter is "recurse", which can take two values, "yes" or "no". This specifies whether the server should automatically attempt to place further call attempts to telephony addresses in redirection responses that were returned from the initial server. [...] But in fact, the 302 arrives to the callee as if not "recurse" parameter exists. Maybe I haven't understood the meaning of this "recurse" parameter? or could it be a bug? Thanks for any explanation. Best regards. -- Iñaki Baz Castillo <ibc(a)aliax.net>

1 0

[OpenSER-Users] Stopping branches from onreply_route?
by Helmut Kuper 24 Dec '07

24 Dec '07

Hi, is there a way to stop all branches of an initial INVITE from onreply_route, if a specific branch replies with code 486 ? regrads helmut

2 2

[OpenSER-Users] SIP alternate port, cross connect with 5060 port
by Raju Abhyankar 24 Dec '07

24 Dec '07

Hello, I have setup openser (1.1.1-tls) to talk to our UA's on port 5060 and some UA's on port 4800 (as provider blocks 5060). I am using Mediaproxy (1.9.0). Currently the following works: 1. UA using 5060 makes call to UA using 4800: Phone bell rings and audio two way no issues. 2. UA using 4800 makes call to UA using 5060. Phone bell rings but no audio in either direction All UA's use RTP port 5004. What can I add to the following loose route section so that I can make "2" work above. My loose route section in openser is: ##################################################### listen= udp:XXX.XXX.XXX.XXX:5060 listen= udp:XXX.XXX.XXX.XXX:4800 # Loose route section if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) {(20) sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { (21)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; }; ##################################################### Please do provide if there is a solution to this issue. Your inputs appreciated. regards, Roger

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[OpenSER-Users] T38 re invite not being sent to VoIP provider
by Mike O'Connor 23 Dec '07

23 Dec '07

Hi Guys I'm trying to get T38 working between our VoIP Provider and my epygi PBX. I have the epygi registered with openSER when a phone call is received from our provider openSER forwards the call to the epygi. When this unit detects the fax tone it does a re invite. (the re invite has the SDP indicating it want a T38) This re invite never gets to our VoIP provider. OpenSER is blocking the request. The invite would seem to be hitting our loose routing if (loose_route()) { xlog("L_NOTICE", "Running loose_route logic"); if ((is_method("INVITE|REFER") && ! has_totag())) { route(ROUTE_AUTHTEST); if (retcode == FALSE) return; }; if (method=="INVITE") { route(ROUTE_AUTHTEST); if (retcode == FALSE) return; }; append_hf("P-hint: rr-enforced\r\n"); route(ROUTE_DEFAULTHANDLER); return; }; route[ROUTE_DEFAULTHANDLER] { t_on_reply("REPLYROUTE"); if (!t_relay()) { sl_reply_error(); }; } Any ideas as to what I might have wrong ? Keep in mind that we are using OpenSER as a part of VoIP service to a small number of customers. Thanks Mike

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[OpenSER-Users] Saving Active Calls
by VoIP Forums www.Go4Calls.com 23 Dec '07

23 Dec '07

Hi Friends, How we can save Active Calls in openser mysql database? Please advise us i need to display it on our billing system. Regards, www.Go4Calls.Com VoIP Forums _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

2 1

[OpenSER-Users] Merry Christmas and a Happy New Year
by luzango.mfupe＠tuune.mobi 23 Dec '07

23 Dec '07

Mates, May you feel the Angels enfold you in their wings. May you always find serenity in the simple things. May the Light of Heaven shine upon your path, and bring you to the completion of your work in Peace and Joy and Grace. Wishing You All the Best this Christmas & a Happy new Year Welcome Year 2008 Tuune Mobile Bldg 6 Room 106,Staatsatillerie Rd FSATIE Tshwane University of Technology Pretoria West Republic of South Africa Website:WWW.Tuune.Mobi Tel:012-382-5953

1 0

[OpenSER-Users] pv transformation does not work when Contact contains spaces
by Robert Dyck 23 Dec '07

23 Dec '07

When I included a pv transformation in my cfg script to extract the expires value from a Contact I found that it would not work if the Contact string contained spaces. I does not seem to affect registration. $(ct{param.value,expires}) The following patch in 1.3.0 works for me -- --- parser/parse_param.c-orig 2007-12-22 23:23:05.000000000 -0800 +++ parser/parse_param.c 2007-12-22 23:23:38.000000000 -0800 @@ -281,7 +281,7 @@ while(_s->len) { switch(_s->s[0]) { - case ' ': + //case ' ': case '\t': case '\r': case '\n': An example of a message that fails Contact expires transformation -- Session Initiation Protocol Request-Line: REGISTER sip:proxy01.sipphone.com SIP/2.0 Method: REGISTER Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.1.4:5061;branch=z9hG4bK-1849b3b3 From: Gateway <sip:17476316719@proxy01.sipphone.com>;tag=a0bda46794e0291o1 To: Gateway <sip:1234567@proxy01.sipphone.com> Call-ID: 6d0c06a5-8d11d464(a)192.168.1.4 CSeq: 51635 REGISTER Max-Forwards: 70 Authorization: Digest username="1234567",realm="proxy01.sipphone.com",nonce="476e0e7866cafbf20434684456644a3d2022c85f",uri="sip:proxy01.sipphone.com",algorithm=MD5,response="57d8e4aea67f9bb45ced0394fe54d659" Contact: Gateway <sip:1234567@192.168.1.4:5061>;expires=3600 User-Agent: Linksys/SPA3000-3.1.10(GWd) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura

1 0

[Serusers] (more details) Solution for the NAT IP changed?
by Gonzalo J. Sambucaro 23 Dec '07

23 Dec '07

Folcks, The problem =========== supose that there is an established rtp connection between two endpoints and the media proxy is in the middle doing the relay of the rtp streams, let´s say (MP = mediaproxy) EP A <--------->[NAT with IP1]<-----------> MP <--------------------> EP B EP A sends rtp to MP_IP:MP_PORT passing through the NAT box. EP B sends rtp to MP_IP:MP_PORT without passing through a NAT box. The MP know that the caller = NAT_IP1:NAT_PORT1, and the called = EP_B_IP:EP_B_PORT Now, supose that the NAT box change their PUBLIC IP from IP1 to IP2, so this escenary EP A <------->[NAT with IP1]<---------> MP <-------------> EP B will change to this EP A <------->[NAT with IP2]<---------> MP <--------------> EP B so the MP should detect that change of IPs and continue relaying the rtp streams but now to IP2:PORT2 instead of IP1:PORT1. Well, that was the situation y have experienced. The solution ============ To fix this, I was thinking this solution: 1) When the first rtp packet of a source arrives, save the SSRC field in the MP. - Save the SSRC of the caller. - Save the SSRC of the called. 2) If arrives a rtp packet with unknown source IP but with the same SSRC field of some of the two streams, updates the binding (with the new IP detected) between the caller and the MP or between the called and the MP according to the field SSRC previously saved. Note: SSRC (RFC 3550 RTP), (from the rfc: "The SSRC identifier carried in the RTP header and in various fields of RTCP packets is a random 32-bit number that is required to be globally unique within an RTP session ") What do you think about the solution? Regards, -- Gonzalo J. Sambucaro Ingeniería de Software Tel: +54-341-4230504 MSLC gonzalo(a)mslc.com.ar www.mslc.com.ar Ocampo y Esmeralda - Vivero de Empresas de Base Tecnológica Ciudad Universitaria Rosario UNR, CCT CONICET Rosario - Santa Fé - Argentina

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[Serusers] Forking request - destination in branches?
by Tomasz Zieleniewski 22 Dec '07

22 Dec '07

Hi, I have following issue. All my outgoing requests are forwarded through boundary proxy by setting the destination before t_relay() my problem is when there is another branch created. This second branch is routed in the standard SIP way directly to destination in request URI. How can I force also branches to go through my boundary proxy. Can I somehow check in branch_route that this branch is not the first one (not first message) but the second and and change the destination and make t_relay()? Thank You for any feedback Regards Tomasz

2 4

[Serusers] SER + asterisk integration config
by satish patel 22 Dec '07

22 Dec '07

Dear all anybody have done this job integration SER with Asterisk ????? $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- 5, 50, 500, 5000 - Store N number of mails in your inbox. Click here.

2 1

[OpenSER-Users] UAC gives
by Raju Abhyankar 22 Dec '07

22 Dec '07

Hello, I am using openser-1.1 and have the standard script from onsip. I am using Mediaproxy for NAT traversal. Most of my clients register for days with no issues. A few clients after say a day of operation suddenly start giving getting de-registered. I can see the ATA or UAC trying to register to openser but openser gives a reply back to the UAC saying "401 Unauthorized". The UAC's which give this issue are behind standard home grade wireless routers (netgear/linksys type). Any idea why this may happen. Thanks in advance Roger

1 0

[OpenSER-Users] Any Sip Phone support examples/web_im/click_to_dial.php? Mine failed, UA says "481 Call Leg/Transaction Does Not Exist"
by CHEN XUEQIN 21 Dec '07

21 Dec '07

Hi all: I know that the example uses REFER sip method and follow such process: 1. prepare two SIP softphone(eyebeam 1.5.16.1 ) , register to openser 2. user enter Caller and Callee's SIP Address 3. Web Server receive http request, then construct FIFO command, using t_uac_dlg MI command, build Refer SIP message with From and To header using user input 4. t_uac_dlg is invoked, send sip REFER message to Caller Sip softphone, softphone return 481 Call Leg/Transaction Does Not Exist, so the call can't be setup. I have seen the notice"/Unfortunately, this example does not work. The reason is use of REFER for third-party call-control has not been standardized due to resistance of proponents of B2BUA use (which is somewhat bloated and not always operational)". But I still wonder if any softphone support this example? My openser version: 1.3.0 Thanks. Regards, Chen Xueqin /

2 3

[OpenSER-Users] Merry Christmas and a Happy New Year 2008!
by Daniel-Constantin Mierla 21 Dec '07

21 Dec '07

I wish Merry Christmas and a Happy New Year 2008 to everybody helping with the project, from code to feedback and reports. It was a full year of activities, with two major releases, lot of enhancements, fixes, and participations to events around the world. I think OpenSER fulfills most of the requirements its users need and it will keep being focused on robustness, performance, flexibility and new enhancements. Looking forward to a 2008 of successes. Cheers, Daniel

1 0

[OpenSER-Users] OpenSER SST module for session timer
by Pat wang 21 Dec '07

21 Dec '07

Happy holidays! Not sure what happened to my previous email the format is kind of messy. Sorry about that. I am playing with SST module for session timer and have some questions that I would like someone can help me out. 1. First question is the 422 message is not sent immediately after receiving INVITE with SE header smaller than default 1800 sec. My OpenSER script is very strait forward as the example in the module document: if (method=="INVITE") { if (sstCheckMin("1")) { blah blah.... exit; } } Debug on OpenSER shows the result returns -1 when an INVITE contains SE and Min-SE timer at 90. It is until the INVITE hits lookup("location") it replies a 422. 2. Right after OpenSER sends a 422 it proxies the INVITE to the called party at that same time. So the call flow looks like: Caller OpenSER Callee INVITE (SE 90)=======> <======= 422 ==== ======INVITE===> === INVITE (SE 1800)==> 3. The 422 Session Timer Too Small message put MIN-SE and Server headers in one line which looks like: MIN-SE: 1800Server: OpenSER........... Has anyone tried SST module before and how did it work for you? Many thanks for you help. Best regards, Patrick _________________________________________________________________ The best games are on Xbox 360. Click here for a special offer on an Xbox 360 Console. http://www.xbox.com/en-US/hardware/wheretobuy/

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[OpenSER-Users] (no subject)
by Pat wang 21 Dec '07

21 Dec '07

Happy holidays! I am playing with SST module for session timer and have some questions that I would like someone can help me out. 1. First question is the 422 message is not sent immediately after receiving INVITE with SE header smaller than default 1800 sec. My OpenSER script is very strait forward as the example in the module document: if (method=="INVITE") { if (sstCheckMin("1")) { blah blah.... exit; }}Debug on OpenSER shows the result returns -1 when an INVITE contains SE and Min-SE timer at 90. It is until the INVITE hits lookup("location") it replies a 422. 2. Right after OpenSER sends a 422 it proxies the INVITE to the called party at that same time. So the call flow looks like: Caller OpenSER Callee INVITE (SE 90)=======> <======= 422 ==== ======INVITE===> === INVITE (SE 1800)==> 3. The 422 Session Timer Too Small message put MIN-SE and Server headers in one line which looks like: MIN-SE: 1800Server: OpenSER........... Has anyone tried SST module before and how did it work for you? Many thanks for you help. Best regards,Patrick _________________________________________________________________ Share life as it happens with the new Windows Live. http://www.windowslive.com/share.html?ocid=TXT_TAGHM_Wave2_sharelife_122007

1 0

[OpenSER-Users] Newbie
by ravimondi＠wlink.com.np 21 Dec '07

21 Dec '07

Hello all, I would like to know where could the materials for openser configurations be found for a newbie like me? Thank you

2 1

[Serusers] fifo question
by Strassburger, Patrick 21 Dec '07

21 Dec '07

Hi all, I am struggling arround since hours about fifo. I get all the time a syntax error. Tired several things like chmod, cr /tmp/ser_fifo as a file or directory, created a /temp/ser_fifo. Changed ser.cfg to fifo="/temp/ser_fifo" fifo_mode=0666 with / without "" does not help all the time a syntax error. Can someone explain - what fifo is (file or folder) - would gues it is a file (first in first out)? - does fifo needs to be created before? I changed /temp/ or /tmp/ to 666 but this does not help either. starting ser shows me [root@ser /]# ser -f /etc/ser/ser.cfg -c 0(8574) parse error (23,1-5): syntax error 0(8574) parse error (23,1-5): ERROR: bad config file (2 errors) wherin line 23 is the one with fifo. Kind regards Patrick # ------------- Initial global variables debug=3 fork=yes log_stderror=no listen=<ip adress> # put your real IP here port=5060 alias=sip.mycompany.com children=4 server_signature=yes dns=no rev_dns=no check_via=no syn_branch=yes memlog=1000 sip_warning=yes reply_to_via=no uid="nobody" gid="nobody" mhomed=0 fifo="/temp/ser_fifo" fifo_mode=0666 fifo_db_url="mysql://ser:heslo@localhost/ser" ************************************************************************************* The message is intended for the named addressee only and may not be disclosed to or used by anyone else, nor may it be copied in any way. The contents of this message and its attachments are confidential and may also be subject to legal privilege. If you are not the named addressee and/or have received this message in error, please advise us by e-mailing security(a)colt.net and delete the message and any attachments without retaining any copies. Internet communications are not secure and COLT does not accept responsibility for this message, its contents nor responsibility for any viruses. No contracts can be created or varied on behalf of COLT Telecommunications, its subsidiaries or affiliates ("COLT") and any other party by email Communications unless expressly agreed in writing with such other party. Please note that incoming emails will be automatically scanned to eliminate potential viruses and unsolicited promotional emails. For more information refer to www.colt.net or contact us on +44(0)20 7390 3900.

2 1

Re: [OpenSER-Users] Ignore CPL script
by Jesus Rodriguez 20 Dec '07

20 Dec '07

Hello Martin, Please, keep always the list in copy as more people can be interested in the replies. > I am beginner in this. I have this as project to school. In which > part of > config file I have to write > cpl_run_script("incoming","force_stateful") ? You can have something like: if (is_method("INVITE") { # run incoming script if ( !cpl_run_script("incoming","force_stateful") ) { # script execution failed sl_send_reply("500","CPL script execution failed"); exit; }; }; Regards. Saludos JesusR. >> Hi Martin, >> >> >> El 20/12/2007, a las 16:28, Martin Kardi escribió: >> >> [...] >> >>> and this is easy cpl script: >>> >>> <?xml version="1.0" encoding="UTF-8"?> >>> >>> <cpl> >>> <incoming> >>> <reject status="400" reason="Busy" /> >>> </incoming> >>> </cpl> >>> >>> Upload of this cpl script to DB is correct, record in table "cpl" is >>> created >>> for "user" but when I am calling to this "user", openser ignores >>> this script >>> and phone is ringing. >>> I don't know what is problem, could somebody help me? >> >> >> You need to call the cpl_run_script() function to execute the >> uploaded >> cpl script. Something like: >> >> cpl_run_script("incoming","force_stateful") >> >> Regards. >> >> Saludos >> JesusR. >> >> ------------------------------------ >> Jesus Rodriguez >> VozTelecom Sistemas, S.L. >> jesusr(a)voztele.com >> http://www.voztele.com >> Tel. 902360305 >> ------------------------------------- >> >> >> >> >> > > > Saludos JesusR. ------------------------------------ Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 -------------------------------------

1 0

[Serusers] Solution for the NAT IP changed?
by Gonzalo J. Sambucaro 20 Dec '07

20 Dec '07

Hello folks, I was Working in the NAT IP changed solution, my idea is to change media proxy server (MPS) in a such way when in a established connection and receive a RTP package from unknown IP source (because a NAT server changed their IP), using de field SSRC (RFC 3550 RTP) updates the binding (with the new IP detected) between the caller and the MPS or between the called and the MPS according to the field SSRC previously saved. What do you think about my solution? Regards, Gonzalo Sambucaro. -- Gonzalo J. Sambucaro Ingeniería de Software Tel: +54-341-4230504 MSLC gonzalo(a)mslc.com.ar www.mslc.com.ar Ocampo y Esmeralda - Vivero de Empresas de Base Tecnológica Ciudad Universitaria Rosario UNR, CCT CONICET Rosario - Santa Fé - Argentina

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[OpenSER-Users] OpenSER 1.2.3 patch release is out!
by Bogdan-Andrei Iancu 20 Dec '07

20 Dec '07

Hi everybody, The third patch release of 1.2.x series is out as version 1.2.3. It preserves full compatibility with 1.2 family regarding the configuration file. OpenSER 1.2.3 brings all kind of fixes (mainly from the 1.3 version). No new features were added. The 1.2.3 release includes: - small fixes (compile warnings, porting across UNIX platform errors - bug fixing - mi_xmlrpc update to work with latest 1.0 libxmlrpc-c3 releases - functionality fixes (Route header handling in local sequential requests) - small improvements - more self - sanity checks, more verbose error codes in script (like t_relay() in TM). Tomorrow I will start uploading the tarballs on the web sites. Best regards, Bogdan

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[OpenSER-Users] Ignore CPL script
by Martin Kardi 20 Dec '07

20 Dec '07

Hi, I have problem with CPL. This is my config file for openser: debug=3 # 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) 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 = "/build/buildd/openser-1.1.0/debian/openser/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/build/buildd/openser-1.1.0/debian/openser/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/build/buildd/openser-1.1.0/debian/openser/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database # mysql.so must be loaded ! 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" #cpl modul -- dopisali sme tu my !!!!!! loadmodule "/usr/lib/openser/modules/cpl-c.so" # Uncomment this if you want digest authentication #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) #---CPL-C Parameter modparam("cpl-c","db_url","mysql://openser:ser@localhost/openser") modparam("cpl-c","cpl_table","cpl") modparam("cpl-c","cpl_dtd_file","/etc/openser/cpl-06.dtd") modparam("cpl-c","log_dir","/var/log/openser/cpl") #modparam("cpl-c","proxy_recurse",2) #modparam("cpl-c","proxy_route",0) #modparam("cpl-c","nat_flag",6) modparam("cpl-c","lookup_domain","location") #cpl_run_script("incoming","force_statefull") # ------------------------- 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; }; # 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("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; cpl_process_register(); 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; and this is easy cpl script: <?xml version="1.0" encoding="UTF-8"?> <cpl> <incoming> <reject status="400" reason="Busy" /> </incoming> </cpl> Upload of this cpl script to DB is correct, record in table "cpl" is created for "user" but when I am calling to this "user", openser ignores this script and phone is ringing. I don't know what is problem, could somebody help me? thanks Martin -- View this message in context: http://www.nabble.com/Ignore-CPL-script-tp14438450p14438450.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[OpenSER-Users] Need help troubleshooting flags in onreply route - v.1.2.2
by Dan-Cristian Bogos 20 Dec '07

20 Dec '07

Guys, I would need your help in troubleshooting some pretty serious issue for me. Scenario: relaying an INVITE from openser to a softswitch and recognizing the flags (global and branch ones) in onreply. Everything seems to work fine with relaying the requests, but when the answers are matched in onreply, the flags are gone. Bellow is the log configuration which gives the output used by me to troubleshoot in logs: in route: xlog("First branch: $br, all branches: $bR, flags set: $mf , branch flags: $bf"); xlog("Relaying to: $avp(s:route), message: $mb"); xlog("OUT to Softswitch"); in onreply route: xlog ("ENTERING ON REPLY ROUTE"); xlog("First branch: $br, all branches: $bR, flags set: $mf , branch flags: $bf"); xlog("In onreply, message: $mb"); Please see the output bellow: Dec 20 10:07:51 ellie /usr/local/sbin/openser[15295]: First branch: <null>, all branches: <null>, flags set: 130 , branch flags: 1024 Dec 20 10:07:51 ellie /usr/local/sbin/openser[15295]: Relaying to: pstn1, message: INVITE sip:INT_NUMBER_MASKED@sip.itsyscom.com SIP/2.0^M Via: SIP/2.0/UDP 10.10.11.10:7091;rport;branch=z9hG4bK2127028508^M From: <sip:dan@sip.itsyscom.com>;tag=2128765320^M To: <sip:INT_NUMBER_MASKED@sip.itsyscom.com>^M Call-ID: 886593274(a)10.10.11.10^M CSeq: 21 INVITE^M Contact: <sip:dan@10.10.11.10:7091>^M Proxy-Authorization: Digest username="dan", realm="sip.itsyscom.com", nonce="476a40226a7cfaa9a11de5b6e7d6742d45ec6bd4", uri="sip:INT_NUMBER_MASKED@sip.itsyscom.com", response="08872e4a06f011ea7d6f947f1874a882", algorithm=MD5^M Max-Forwards: 4^M User-Agent: Linphone-1.5.1/eXosip^M Subject: Phone call^M Expires: 120^M Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE^M Content-Type: application/sdp^M Content-Length: 310^M ^M v=0^M o=dan 123456 654321 IN IP4 10.10.11.10^M s=A conversation^M c=IN IP4 10.10.11.10^M t=0 0^M m=audio 7078 RTP/AVP 3 0 8 111 110 101^M a=rtpmap:3 GSM/8000/1^M a=rtpmap:0 PCMU/8000/1^M a=rtp Dec 20 10:07:51 ellie /usr/local/sbin/openser[15295]: OUT to Softswitch Dec 20 10:07:51 ellie /usr/local/sbin/openser[15295]: ENTERING ON REPLY ROUTE Dec 20 10:07:51 ellie /usr/local/sbin/openser[15295]: First branch: <null>, all branches: <null>, flags set: 0 , branch flags: 0 Dec 20 10:07:51 ellie /usr/local/sbin/openser[15295]: In onreply, message: SIP/2.0 100 Trying^M Via: SIP/2.0/UDP IP_MASKED;branch=z9hG4bK9221.886b3817.0;received=IP_MASKED;rport=5060^M Via: SIP/2.0/UDP 10.10.11.10:7091;received=ORIGINATOR_IP;rport=8026;branch=z9hG4bK2127028508^M Record-Route: <sip:IP_MASKED;lr=on;ftag=2128765320;nat=yes>^M From: <sip:dan@sip.itsyscom.com>;tag=2128765320^M To: <sip:INT_NUMBER_MASKED@sip.itsyscom.com>^M Call-ID: 886593274(a)10.10.11.10^M CSeq: 21 INVITE^M Server: GlobalE SoftSWITCH/1.3.0.^M Content-Length: 0^M ^M Dec 20 10:07:53 ellie /usr/local/sbin/openser[15296]: ENTERING ON REPLY ROUTE Dec 20 10:07:53 ellie /usr/local/sbin/openser[15296]: First branch: <null>, all branches: <null>, flags set: 0 , branch flags: 0 Dec 20 10:07:53 ellie /usr/local/sbin/openser[15296]: In onreply, message: SIP/2.0 180 Ringing^M Via: SIP/2.0/UDP IP_MASKED;branch=z9hG4bK9221.886b3817.0;received=IP_MASKED;rport=5060^M Via: SIP/2.0/UDP 10.10.11.10:7091;received=ORIGINATOR_IP;rport=8026;branch=z9hG4bK2127028508^M Record-Route: <sip:IP_MASKED;lr=on;ftag=2128765320;nat=yes>^M From: <sip:dan@sip.itsyscom.com>;tag=2128765320^M To: <sip:INT_NUMBER_MASKED@sip.itsyscom.com>;tag=975999519^M Call-ID: 886593274(a)10.10.11.10^M CSeq: 21 INVITE^M Server: GlobalE SoftSWITCH/1.3.0.^M Contact: <sip:INT_NUMBER_MASKED@IP_MASKED:5070>^M Allow: ACK, INVITE, BYE, CANCEL, OPTIONS^M Content-Type: application/sdp^M Content-Length: 187^M ^M v=0^M o=yate 1198145273 1198145273 IN IP4 208.50.127.243^M s=SIP Call^M c=IN IP4 208.50.127.243^M t=0 0^M m=audio 23166 RTP/AVP 8 101^M a=rtpmap:8 PCMA/8000^M a=rtpmap:101 telephone-event/8000^M Can any of you help me troubleshooting this? I get the feeling that there is no transaction matched by openser for the replies. Version I am using (compiled from sources, with some params modified in configure.h): openser -V version: openser 1.2.2-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. svnrevision: 2:3368M @(#) $Id: main.c 3173 2007-11-20 08:26:35Z bogdan_iancu $ Thxs in advance for any help. DanB

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[Serusers] SER To Address
by Arun Kumar 20 Dec '07

20 Dec '07

Hi, When I send calls to PSTN the From filed is my user@SER IP is correct but in To Filed I want Number@ProviderIP but SER send Number@SERIP ? Thanks, Arun

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Re: [OpenSER-Users] Openser and Nokia E61
by David Villasmil Govea 19 Dec '07

19 Dec '07

Ravi or Juha, if you've connected E61 succesfully, could you post the parameters here? I just can't... Thanks a lot David

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Re: [OpenSER-Users] What is the difference?
by David Villasmil Govea 19 Dec '07

19 Dec '07

> > > El Wednesday 19 December 2007 13:20:07 I?aki Baz Castillo escribi?: > > El Wednesday 19 December 2007 13:15:35 David Villasmil Govea escribi?: > > > What's the difference between > > > > > > setbflag and setflag > > > ?? > > > > http://www.openser.org/dokuwiki/doku.php/utils:flags > > Note that if you use setbflag into a "route" you set the bflag for ALL the > branches (there could be more than one branch because pararell forking and > so). > > If you want to set bflag just for a branch you should use "setbflag" > into "branch_route". > > And NEVER do "isbflagset" into a "route" since the result can be > unexpectable > (if a branch has that bflag active and other one not active, "isbflagset" > in "route" could return 1 or 0 depending on which branch examinates > before). > > I ask, because i had this cfg file that was working perfectly with nat/rtpproxy. I replace it with a new cfg that was using bflag instead of flag. client-rtpproxy-client rtps stopped working. When i replace those bflags with regular flags it started working again!... Also, as i read there, "per branch" means each location... talking about location, is there a reason a client would have 98 almost exactly alike locations? but for the id, they're the same... weird. And i started seeing lots of "max branch exceeded" on the syslog. We do not limit branches because a client could have his softphone and ata logged in... thanks a lot

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[OpenSER-Users] Problems with presence
by Jon Farmer 19 Dec '07

19 Dec '07

Hi I am trying to get presence working and getting the following error. Can anyone suggest what it wrong? I am using openser 1.2.2 notls and xlite and eyebeam for testing. Dec 19 17:11:47 esosser openser[22325]: PRESENCE: generate_ETag: etag= a.1198083751.22325.1 / 20 Dec 19 17:11:47 esosser openser[22325]: PRESENCE: handle_publish: No E-Tag and no body present Dec 19 17:11:47 esosser openser[22325]: PRESENCE: handle_publish: ERROR occured Regards Jon __________________________________________________________ Sent from Yahoo! Mail - a smarter inbox http://uk.mail.yahoo.com

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[OpenSER-Users] SoftPhone with CPL script upload support
by Martin Kardi 19 Dec '07

19 Dec '07

Hi, I try find some softphone with uploading CPL scripts support? Please advise me somebody. Thanks Martin -- View this message in context: http://www.nabble.com/SoftPhone-with-CPL-script-upload-support-tp14419452p1… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[OpenSER-Users] ERROR:avp:add_avp: no more shm mem
by Marc LEURENT 19 Dec '07

19 Dec '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good Morning Dan and Good Morning everybody, I've got Freeradius 1.1.7 + (python module enable + rlm_digest patches) OpenSER 1.2.2 + (Radius ACC enable) CDRTool 5.3.2 and I don't understand these outputs... OpenSER is crashing after receiving Authorization from Freeradius... Any idea?? Freeradius sent AVP to OpenSER: SIP-AVP = "callMaxDur:2957" SIP-AVP = "credit:10.0000" SIP-AVP = "authentic#1" And openser is trying to parse it but the there is a problem... 2(30175) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair 225 2(30175) DEBUG:auth_radius:extract_avp: string is <ur:2957> 2(30175) DEBUG:auth_radius:extract_avp: AVP name is <ur> 2(30175) DEBUG:auth_radius:extract_avp: AVP val is <2957> 2(30175) ERROR:avp:add_avp: no more shm mem 2(30175) ERROR:auth_radius:generate_avps: Unable to create a new AVP 10(30190) ERROR: receive_fd: EOF on 6 Thanks ### Small Freeradius output ### A2 = INVITE:sip:0677832974@sip.wifirst.net H(A1) = b82eaab7b4eaab2db812d3a2695f0e22 H(A2) = 83134a8c6b9ea1a20de012abf16e87f5 KD = b82eaab7b4eaab2db812d3a2695f0e22:47679476280e04ea50ded04b24f7d3f75068eea7:83134a8c6b9ea1a20de012abf16e87f5 EXPECTED 40ed14d07aa16035e67c21886a802379 RECEIVED 40ed14d07aa16035e67c21886a802379 modcall[authenticate]: module "digest" returns ok for request 71 modcall: leaving group authenticate (returns ok) for request 71 Login OK: [mleurent(a)sip.wifirst.net/<no User-Password attribute>] (from client private port 5060) Processing the post-auth section of radiusd.conf modcall: entering group post-auth for request 71 rlm_sql (sql): Processing sql_postauth radius_xlat: 'mleurent(a)sip.wifirst.net' rlm_sql (sql): sql_set_user escaped user --> 'mleurent(a)sip.wifirst.net' radius_xlat: 'INSERT into radpostauth (id, user, pass, reply, date) values ('', 'mleurent(a)sip.wifirst.net', 'Chap-Password', 'Access-Accept', NOW())' rlm_sql (sql) in sql_postauth: query is INSERT into radpostauth (id, user, pass, reply, date) values ('', 'mleurent(a)sip.wifirst.net', 'Chap-Password', 'Access-Accept', NOW()) rlm_sql (sql): Reserving sql socket id: 2 rlm_sql (sql): Released sql socket id: 2 modcall[post-auth]: module "sql" returns ok for request 71 modcall: leaving group post-auth (returns ok) for request 71 Sending Access-Accept of id 180 to 172.24.0.8 port 47861 SIP-AVP = "callMaxDur:2957" SIP-AVP = "credit:10.0000" SIP-AVP = "authentic#1" Finished request 71 Going to the next request - --- Walking the entire request list --- Waking up in 6 seconds... ### Full OpenSER output ### 2(30175) SIP Request: 2(30175) method: <INVITE> 2(30175) uri: <sip:0677832974@sip.wifirst.net> 2(30175) version: <SIP/2.0> 2(30175) parse_headers: flags=2 2(30175) get_hdr_field: cseq <CSeq>: <2> <INVITE> 2(30175) Found param type 232, <branch> = <z9hG4bK3c2eaf3d-baab-dc11-90b2-ed571fe4ba74>; state=6 2(30175) Found param type 235, <rport> = <n/a>; state=17 2(30175) end of header reached, state=5 2(30175) parse_headers: Via found, flags=2 2(30175) parse_headers: this is the first via 2(30175) After parse_msg... 2(30175) preparing to run routing scripts... 2(30175) parse_headers: flags=100 2(30175) DEBUG:parse_to:end of header reached, state=10 2(30175) DBUG:parse_to: display={}, ruri={sip:0677832974@sip.wifirst.net} 2(30175) DEBUG: get_hdr_field: <To> [34]; uri=[sip:0677832974@sip.wifirst.net] 2(30175) DEBUG: to body [<sip:0677832974@sip.wifirst.net> ] 2(30175) DEBUG: get_hdr_body : content_length=235 2(30175) DEBUG:maxfwd:is_maxfwd_present: value = 70 2(30175) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 50 2(30175) !!!! STARTING ROUTE 2: TEST NAT 2(30175) !!!!!!!!! YAHH NAT DETECTED - FIXING CONTACT!!!!!!!!!!!!! 2(30175) parse_headers: flags=80 2(30175) DEBUG: add_param: tag=2eb08b3d-baab-dc11-90b2-ed571fe4ba74 2(30175) DEBUG:parse_to:end of header reached, state=29 2(30175) DBUG:parse_to: display={"Marc LEURENT"}, ruri={sip:mleurent@sip.wifirst.net} 2(30175) parse_headers: flags=200 2(30175) found end of header 2(30175) find_first_route: No Route headers found 2(30175) loose_route: There is no Route HF 2(30175) grep_sock_info - checking if host==us: 15==12 && [sip.wifirst.net] == [213.91.4.219] 2(30175) grep_sock_info - checking if port 5060 matches port 5060 2(30175) JUSTE BEFORE SETTING FLAGS FOR RADIUS 2(30175) DEBUG: has_totag: no totag 2(30175) I AM SETTING THE FLAGS FOR RADIUS 2(30175) SETTING FLAGS 1 FOR RADIUS 2(30175) SETTING FLAGS 2 FOR RADIUS 2(30175) grep_sock_info - checking if host==us: 15==12 && [sip.wifirst.net] == [213.91.4.219] 2(30175) grep_sock_info - checking if port 5060 matches port 5060 2(30175) grep_sock_info - checking if host==us: 15==12 && [sip.wifirst.net] == [213.91.4.219] 2(30175) grep_sock_info - checking if port 5060 matches port 5060 2(30175) check_nonce(): comparing [47679476280e04ea50ded04b24f7d3f75068eea7] and [47679476280e04ea50ded04b24f7d3f75068eea7] 2(30175) DEBUG:auth_radius:radius_authorize_sterman: Success 2(30175) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair 225 2(30175) DEBUG:auth_radius:extract_avp: string is <ur:2957> 2(30175) DEBUG:auth_radius:extract_avp: AVP name is <ur> 2(30175) DEBUG:auth_radius:extract_avp: AVP val is <2957> 2(30175) ERROR:avp:add_avp: no more shm mem 2(30175) ERROR:auth_radius:generate_avps: Unable to create a new AVP 10(30190) ERROR: receive_fd: EOF on 6 10(30190) DBG: handle_ser_child: dead child 2, pid 30175 (shutting down?) 10(30190) DBG: io_watch_del (0x5b5c20, 6, -1, 0x0) fd_no=13 called 0(30173) child process 30175 exited by a signal 11 0(30173) core was generated 0(30173) INFO: terminating due to SIGCHLD 4(30177) INFO: signal 15 received 4(30177) Memory status (pkg): 4(30177) fm_status (0x5f6d80): 4(30177) heap size= 1048576 4(30177) used= 77952, used+overhead=122432, free=926144 4(30177) max used (+overhead)= 129560 4(30177) dumping free list: 4(30177) hash = 1 fragments no.: 4, unused: 0 bucket size: 8 - 8 (first 8) 4(30177) hash = 2 fragments no.: 6, unused: 0 bucket size: 16 - 16 (first 16) 4(30177) hash = 3 fragments no.: 4, unused: 0 bucket size: 24 - 24 (first 24) 4(30177) hash = 4 fragments no.: 1, unused: 0 bucket size: 32 - 32 (first 32) 4(30177) hash = 5 fragments no.: 1, unused: 0 bucket size: 40 - 40 (first 40) 4(30177) hash = 6 fragments no.: 4, unused: 0 bucket size: 48 - 48 (first 48) 4(30177) hash = 7 fragments no.: 25, unused: 0 bucket size: 56 - 56 (first 56) 4(30177) hash = 8 fragments no.: 6, unused: 0 bucket size: 64 - 64 (first 64) 4(30177) hash = 9 fragments no.: 14, unused: 0 bucket size: 72 - 72 (first 72) 4(30177) hash = 10 fragments no.: 2, unused: 0 bucket size: 80 - 80 (first 80) 4(30177) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 4(30177) hash = 15 fragments no.: 1, unused: 0 bucket size: 120 - 120 (first 120) 4(30177) hash = 28 fragments no.: 1, unused: 0 bucket size: 224 - 224 (first 224) 4(30177) hash = 29 fragments no.: 1, unused: 0 bucket size: 232 - 232 (first 232) 4(30177) hash = 52 fragments no.: 2, unused: 0 bucket size: 416 - 416 (first 416) 4(30177) hash = 62 fragments no.: 1, unused: 0 bucket size: 496 - 496 (first 496) 4(30177) hash = 64 fragments no.: 1, unused: 0 bucket size: 512 - 512 (first 512) 4(30177) hash = 65 fragments no.: 1, unused: 0 bucket size: 520 - 520 (first 520) 4(30177) hash = 146 fragments no.: 1, unused: 0 bucket size: 1168 - 1168 (first 1168) 4(30177) hash = 176 fragments no.: 1, unused: 0 bucket size: 1408 - 1408 (first 1408) 4(30177) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 917296) 4(30177) TOTAL: 79 free fragments = 926336 free bytes 4(30177) ----------------------------- 5(30178) INFO: signal 15 received 5(30178) Memory status (pkg): 5(30178) fm_status (0x5f6d80): 5(30178) heap size= 1048576 5(30178) used= 77776, used+overhead=121040, free=927536 5(30178) max used (+overhead)= 121504 5(30178) dumping free list: 5(30178) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 5(30178) hash = 41 fragments no.: 1, unused: 0 bucket size: 328 - 328 (first 328) 5(30178) hash = 55 fragments no.: 1, unused: 0 bucket size: 440 - 440 (first 440) 5(30178) hash = 56 fragments no.: 1, unused: 0 bucket size: 448 - 448 (first 448) 5(30178) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 926496) 5(30178) TOTAL: 5 free fragments = 927728 free bytes 5(30178) ----------------------------- 1(30174) INFO: signal 15 received 1(30174) Memory status (pkg): 1(30174) fm_status (0x5f6d80): 1(30174) heap size= 1048576 3(30176) INFO: signal 15 received 3(30176) Memory status (pkg): 3(30176) fm_status (0x5f6d80): 3(30176) heap size= 1048576 3(30176) used= 77936, used+overhead=122400, free=926176 3(30176) max used (+overhead)= 129576 3(30176) dumping free list: 3(30176) hash = 1 fragments no.: 6, unused: 0 bucket size: 8 - 8 (first 8) 3(30176) hash = 2 fragments no.: 4, unused: 0 bucket size: 16 - 16 (first 16) 3(30176) hash = 3 fragments no.: 1, unused: 0 bucket size: 24 - 24 (first 24) 3(30176) hash = 4 fragments no.: 2, unused: 0 bucket size: 32 - 32 (first 32) 3(30176) hash = 6 fragments no.: 3, unused: 0 bucket size: 48 - 48 (first 48) 3(30176) hash = 7 fragments no.: 24, unused: 0 bucket size: 56 - 56 (first 56) 3(30176) hash = 8 fragments no.: 3, unused: 0 bucket size: 64 - 64 (first 64) 3(30176) hash = 9 fragments no.: 16, unused: 0 bucket size: 72 - 72 (first 72) 3(30176) hash = 10 fragments no.: 4, unused: 0 bucket size: 80 - 80 (first 80) 1(30174) used= 77992, used+overhead=122472, free=926104 1(30174) max used (+overhead)= 129696 1(30174) dumping free list: 1(30174) hash = 1 fragments no.: 6, unused: 0 bucket size: 8 - 8 (first 8) 1(30174) hash = 2 fragments no.: 5, unused: 0 bucket size: 16 - 16 (first 16) 1(30174) hash = 3 fragments no.: 3, unused: 0 bucket size: 24 - 24 (first 24) 1(30174) hash = 6 fragments no.: 6, unused: 0 bucket size: 48 - 48 (first 48) 1(30174) hash = 7 fragments no.: 29, unused: 0 bucket size: 56 - 56 (first 56) 1(30174) hash = 8 fragments no.: 3, unused: 0 bucket size: 64 - 64 (first 64) 1(30174) hash = 9 fragments no.: 16, unused: 0 bucket size: 72 - 72 (first 72) 1(30174) hash = 11 fragments no.: 2, unused: 0 bucket size: 88 - 88 (first 88) 1(30174) hash = 19 fragments no.: 1, unused: 0 bucket size: 152 - 152 (first 152) 1(30174) hash = 28 fragments no.: 1, unused: 0 bucket size: 224 - 224 (first 224) 1(30174) hash = 29 fragments no.: 1, unused: 0 bucket size: 232 - 232 (first 232) 1(30174) hash = 52 fragments no.: 2, unused: 0 bucket size: 416 - 416 (first 416) 1(30174) hash = 96 fragments no.: 1, unused: 0 bucket size: 768 - 768 (first 768) 1(30174) hash = 150 fragments no.: 1, unused: 0 bucket size: 1200 - 1200 (first 1200) 1(30174) hash = 176 fragments no.: 1, unused: 0 bucket size: 1408 - 1408 (first 1408) 1(30174) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 917848) 1(30174) TOTAL: 79 free fragments = 926296 free bytes 1(30174) ----------------------------- 3(30176) hash = 13 fragments no.: 1, unused: 0 bucket size: 104 - 104 (first 104) 3(30176) hash = 14 fragments no.: 1, unused: 0 bucket size: 112 - 112 (first 112) 3(30176) hash = 15 fragments no.: 1, unused: 0 bucket size: 120 - 120 (first 120) 3(30176) hash = 28 fragments no.: 2, unused: 0 bucket size: 224 - 224 (first 224) 3(30176) hash = 29 fragments no.: 1, unused: 0 bucket size: 232 - 232 (first 232) 3(30176) hash = 52 fragments no.: 2, unused: 0 bucket size: 416 - 416 (first 416) 3(30176) hash = 58 fragments no.: 1, unused: 0 bucket size: 464 - 464 (first 464) 3(30176) hash = 59 fragments no.: 1, unused: 0 bucket size: 472 - 472 (first 472) 3(30176) hash = 102 fragments no.: 1, unused: 0 bucket size: 816 - 816 (first 816) 3(30176) hash = 122 fragments no.: 1, unused: 0 bucket size: 976 - 976 (first 976) 3(30176) hash = 128 fragments no.: 1, unused: 0 bucket size: 1024 - 1024 (first 1024) 3(30176) hash = 176 fragments no.: 1, unused: 0 bucket size: 1408 - 1408 (first 1408) 3(30176) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 916008) 3(30176) TOTAL: 78 free fragments = 926368 free bytes 3(30176) ----------------------------- 10(30190) INFO: signal 15 received 10(30190) Memory status (pkg): 10(30190) fm_status (0x5f6d80): 10(30190) heap size= 1048576 10(30190) used= 127600, used+overhead=170816, free=877760 10(30190) max used (+overhead)= 170816 10(30190) dumping free list: 10(30190) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 877952) 10(30190) TOTAL: 1 free fragments = 877952 free bytes 10(30190) ----------------------------- 9(30188) INFO: signal 15 received 9(30188) Memory status (pkg): 9(30188) fm_status (0x5f6d80): 9(30188) heap size= 1048576 9(30188) used= 135792, used+overhead=179024, free=869552 9(30188) max used (+overhead)= 179024 9(30188) dumping free list: 9(30188) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 869744) 9(30188) TOTAL: 1 free fragments = 869744 free bytes 9(30188) ----------------------------- 6(30181) INFO: signal 15 received 6(30181) Memory status (pkg): 6(30181) fm_status (0x5f6d80): 6(30181) heap size= 1048576 6(30181) used= 135792, used+overhead=179024, free=869552 6(30181) max used (+overhead)= 179024 6(30181) dumping free list: 6(30181) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 869744) 6(30181) TOTAL: 1 free fragments = 869744 free bytes 6(30181) 8(30185) INFO: signal 15 received 8(30185) Memory status (pkg): 8(30185) fm_status (0x5f6d80): 8(30185) heap size= 1048576 8(30185) used= 135792, used+overhead=179024, free=869552 8(30185) max used (+overhead)= 179024 8(30185) dumping free list: 8(30185) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 869744) 8(30185) TOTAL: 1 free fragments = 869744 free bytes 8(30185) ----------------------------- - ----------------------------- 7(30182) INFO: signal 15 received 7(30182) Memory status (pkg): 7(30182) fm_status (0x5f6d80): 7(30182) heap size= 1048576 7(30182) used= 135792, used+overhead=179024, free=869552 7(30182) max used (+overhead)= 179024 7(30182) dumping free list: 7(30182) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 869744) 7(30182) TOTAL: 1 free fragments = 869744 free bytes 7(30182) ----------------------------- 1(30174) INFO: signal 15 received 1(30174) Memory status (pkg): 1(30174) fm_status (0x5f6d80): 1(30174) heap size= 1048576 1(30174) used= 85664, used+overhead=128704, free=919872 1(30174) max used (+overhead)= 128704 1(30174) dumping free list: 1(30174) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 1(30174) hash = 42 fragments no.: 1, unused: 0 bucket size: 336 - 336 (first 336) 1(30174) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 919640) 1(30174) TOTAL: 3 free fragments = 920064 free bytes 1(30174) ----------------------------- 0(30173) pool_remove: Connection still kept in the pool 0(30173) pool_remove: Connection still kept in the pool 0(30173) XLOG: destroy module ... 0(30173) pool_remove: Removing connection from the pool 0(30173) pool_remove: Connection still kept in the pool 0(30173) INFO:mi_fifo:mi_destroy: seems that fifo child is already dead! 0(30173) pool_remove: Removing connection from the pool 0(30173) DEBUG: tm_shutdown : start 0(30173) DEBUG: unlink_timer_lists : emptying DELETE list 0(30173) DEBUG: tm_shutdown : emptying hash table 0(30173) DEBUG: tm_shutdown : releasing timers 0(30173) DEBUG: tm_shutdown : removing semaphores 0(30173) DEBUG: tm_shutdown : destroying tmcb lists 0(30173) DEBUG: tm_shutdown : done 0(30173) Memory status (pkg): 0(30173) fm_status (0x5f6d80): 0(30173) heap size= 1048576 0(30173) used= 56464, used+overhead=99648, free=948928 0(30173) max used (+overhead)= 112768 0(30173) dumping free list: 0(30173) hash = 1 fragments no.: 7, unused: 0 bucket size: 8 - 8 (first 8) 0(30173) hash = 2 fragments no.: 3, unused: 0 bucket size: 16 - 16 (first 16) 0(30173) hash = 3 fragments no.: 7, unused: 0 bucket size: 24 - 24 (first 24) 0(30173) hash = 4 fragments no.: 33, unused: 0 bucket size: 32 - 32 (first 32) 0(30173) hash = 6 fragments no.: 9, unused: 0 bucket size: 48 - 48 (first 48) 0(30173) hash = 7 fragments no.: 5, unused: 0 bucket size: 56 - 56 (first 56) 0(30173) hash = 8 fragments no.: 1, unused: 0 bucket size: 64 - 64 (first 64) 0(30173) hash = 9 fragments no.: 2, unused: 0 bucket size: 72 - 72 (first 72) 0(30173) hash = 12 fragments no.: 9, unused: 0 bucket size: 96 - 96 (first 96) 0(30173) hash = 13 fragments no.: 1, unused: 0 bucket size: 104 - 104 (first 104) 0(30173) hash = 14 fragments no.: 3, unused: 0 bucket size: 112 - 112 (first 112) 0(30173) hash = 21 fragments no.: 2, unused: 0 bucket size: 168 - 168 (first 168) 0(30173) hash = 156 fragments no.: 2, unused: 0 bucket size: 1248 - 1248 (first 1248) 0(30173) hash = 842 fragments no.: 1, unused: 0 bucket size: 6736 - 6736 (first 6736) 0(30173) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 936000) 0(30173) TOTAL: 86 free fragments = 949120 free bytes 0(30173) ----------------------------- 0(30173) Memory status (shm): 0(30173) fm_status (0x2ab19f9d0000): 0(30173) heap size= 33554432 0(30173) used= 144, used+overhead=36720, free=33517712 0(30173) max used (+overhead)= 2746832 0(30173) dumping free list: 0(30173) hash = 1 fragments no.: 56, unused: 0 bucket size: 8 - 8 (first 8) 0(30173) hash = 2 fragments no.: 12, unused: 0 bucket size: 16 - 16 (first 16) 0(30173) hash = 3 fragments no.: 9, unused: 0 bucket size: 24 - 24 (first 24) 0(30173) hash = 4 fragments no.: 10, unused: 0 bucket size: 32 - 32 (first 32) 0(30173) hash = 5 fragments no.: 3, unused: 0 bucket size: 40 - 40 (first 40) 0(30173) hash = 6 fragments no.: 4, unused: 0 bucket size: 48 - 48 (first 48) 0(30173) hash = 7 fragments no.: 47, unused: 0 bucket size: 56 - 56 (first 56) 0(30173) hash = 8 fragments no.: 3, unused: 0 bucket size: 64 - 64 (first 64) 0(30173) hash = 9 fragments no.: 6, unused: 0 bucket size: 72 - 72 (first 72) 0(30173) hash = 10 fragments no.: 2, unused: 0 bucket size: 80 - 80 (first 80) 0(30173) hash = 12 fragments no.: 1, unused: 0 bucket size: 96 - 96 (first 96) 0(30173) hash = 15 fragments no.: 1, unused: 0 bucket size: 120 - 120 (first 120) 0(30173) hash = 17 fragments no.: 1, unused: 0 bucket size: 136 - 136 (first 136) 0(30173) hash = 21 fragments no.: 5, unused: 0 bucket size: 168 - 168 (first 168) 0(30173) hash = 22 fragments no.: 1, unused: 0 bucket size: 176 - 176 (first 176) 0(30173) hash = 35 fragments no.: 1, unused: 0 bucket size: 280 - 280 (first 280) 0(30173) hash = 76 fragments no.: 1, unused: 0 bucket size: 608 - 608 (first 608) 0(30173) hash = 104 fragments no.: 1, unused: 0 bucket size: 832 - 832 (first 832) 0(30173) hash = 110 fragments no.: 1, unused: 0 bucket size: 880 - 880 (first 880) 0(30173) hash = 112 fragments no.: 1, unused: 0 bucket size: 896 - 896 (first 896) 0(30173) hash = 128 fragments no.: 2, unused: 0 bucket size: 1024 - 1024 (first 1024) 0(30173) hash = 150 fragments no.: 1, unused: 0 bucket size: 1200 - 1200 (first 1200) 0(30173) hash = 193 fragments no.: 1, unused: 0 bucket size: 1544 - 1544 (first 1544) 0(30173) hash = 224 fragments no.: 1, unused: 0 bucket size: 1792 - 1792 (first 1792) 0(30173) hash = 258 fragments no.: 2, unused: 0 bucket size: 2064 - 2064 (first 2064) 0(30173) hash = 727 fragments no.: 1, unused: 0 bucket size: 5816 - 5816 (first 5816) 0(30173) hash = 831 fragments no.: 1, unused: 0 bucket size: 6648 - 6648 (first 6648) 0(30173) hash = 1024 fragments no.: 2, unused: 0 bucket size: 8192 - 8192 (first 8192) 0(30173) hash = 2049 fragments no.: 2, unused: 0 bucket size: 16384 - 32768 (first 20480) 0(30173) hash = 2056 fragments no.: 1, unused: 0 bucket size: 2097152 - 4194304 (first 2621440) 0(30173) hash = 2059 fragments no.: 1, unused: 0 bucket size: 16777216 - 33554432 (first 30805984) 0(30173) TOTAL: 181 free fragments = 33517712 free bytes 0(30173) ----------------------------- 0(30173) shm_mem_destroy 0(30173) destroying the shared memory lock 0(30173) terminating due to SIGCHLD -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHZ5doN4+o+2LtdFwRAknyAKCXJHhyHZlAooGoyONmFphFcZE3BgCfbJOO aN4wxnGPjLgxqVvx5YTgGac= =pkmq -----END PGP SIGNATURE-----

3 5

[Serusers] Newbie Problems: MySQL support in SER
by frq ser 19 Dec '07

19 Dec '07

Hi, Further to my previous email, there is another problem for which I need help. While trying to add MySQL support, I could not find the file src/sip_router/Makefile (as asked in the Getting started document (http://siprouter.onsip.org/doc/gettingstarted/ch11s05.html). The Makefile for the SER in the base directory (/lhome/.../ser/Makefile) does not refer to MySQL in the exclude_modules line (Note the ? after exclude_modules). Here is the excerpt from the Makefile: # if not set on the cmd. line or the env, exclude this modules: exclude_modules?= cpl ext extcmd \ postgres snmp \ im \ jabber mysql \ cpl-c \ auth_radius group_radius uri_radius avp_radius \ pa Does that mean that MySQL is already installed? How can I check if MySQL is already installed on my machine? I also found the following code in the Makefile but I am not sure what it means. .PHONY: dbinstall dbinstall: -@echo "Initializing ser database" scripts/ser_mysql.sh create -@echo "Done" . . . I'd be very grateful if anyone could help. Frq

1 1

[OpenSER-Users] Res: FIFO / SEMS
by Bruno Machado 19 Dec '07

19 Dec '07

Great job, Iancu. It is what I really needed. I tried to use the 'dialog' module and the session was torn down without any problems. Perfect. I have just one more question. I would like to log the generated BYE in ACC. Is it possible when OpenSER receive this BYE request from FIFO? Thanks a lot again bruno ----- Mensagem original ---- De: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> Para: Bruno Machado <bsmufu(a)yahoo.com.br> Cc: users(a)lists.openser.org Enviadas: Segunda-feira, 17 de Dezembro de 2007 11:19:29 Assunto: Re: [OpenSER-Users] FIFO / SEMS Hi Bruno, In OpenSER 1.3 "BlackCat", the "dialog" module provides a way to terminate an ongoing dialog. See the generic docs: http://www.openser.org/docs/modules/1.3.x/dialog.html and about sending BYEs: http://www.openser.org/docs/modules/1.3.x/dialog.html#AEN461 Regards, bogdan Bruno Machado wrote: > Hi all > > I am reading about FIFO and I am wandering in my thoughts, searching > for a solution > that tear down a session. I don't want to use Asterisk anymore for > this purpose. > Can I send a BYE message through FIFO? Should I use "t_uac_dlg"? I was > reading > about "t_uac_cancel" but it generates a CANCEL and, of course, it > won't resolve the problem. > > I didn't install SEMS yet. So, I would like to know if someone uses > this software > (OpenSER + SEMS) within this purpose. This link: > > http://svn.berlios.de/wsvn/sems/trunk/doc/Readme.sw_prepaid_sip?op=file&rev… > <http://svn.berlios.de/wsvn/sems/trunk/doc/Readme.sw_prepaid_sip?op=file&rev…> > > has this: > > "When the call duration exceeds the credit, the call is terminated by > SEMS." > > > So, I can realize it sends a BYE to OpenSER through FIFO and it will be > forwarded to the voip devices. Am I right? I will try to use the > 't_uac_dlg' now. > Any comments are welcome. > > Cheers > Bruno Machado > > > > ------------------------------------------------------------------------ > Abra sua conta no Yahoo! Mail > <http://br.rd.yahoo.com/mail/taglines/mail/*http://br.mail.yahoo.com/>, > o único sem limite de espaço para armazenamento! > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/

1 0

[OpenSER-Users] Openser-1.3 packages are ready in Fedora testing repository
by Jan ONDREJ (SAL) 19 Dec '07

19 Dec '07

Hello, i am pleased to announce, that OpenSER has been approved as a part of Fedora distribution. openser-1.3.0 is ready in fedora-7-testing and fedora-8-testing repository. It will be moved to stable after aprox. a week. If you have openser < 1.3.0 installed on your fedora or EPEL, please turn off your automatic updates or update it manually before the final release. Follow these steps to update: read upgrading guide at: http://openser.org/dokuwiki/doku.php/install:1.2.2-to-1.3.0 run this command: yum --enablerepo=updates-testing update 'openser*' check your configuration files and just then restart openser Jan ONDREJ (SAL) PS: I am not a member of this mailinglist. If you wish to reply, please send email directly to me.

1 0

[OpenSER-Users] What is the difference?
by David Villasmil Govea 19 Dec '07

19 Dec '07

What's the difference between setbflag and setflag ?? Thanks David

2 2

[OpenSER-Users] Openser and Nokia E61
by David Villasmil Govea 19 Dec '07

19 Dec '07

Hello all, Anyone know how to connect a nokia e61 to openser? I've configured it to connect to asterisk and no problem whatsoever.... but seems impossible to connect to openser 1.2 Cheers David

3 2

[OpenSER-Users] need Help in TLS Problem in openser 1.3 Black CAT
by raviprakash sunkara 19 Dec '07

19 Dec '07

Hello Users Merry XMAS Resently I installed and Configure the openser 1.3 with TLS Support in FC6 with openSSL 0.9.8b Am Trying implementing the TLS of my OpenSER proxy First I trying TLS for Registering to My OpenSER Proxy Server. The Certificatation am using the default while in installing the openser 1.3 Am using eyebeam 1.5 with TLS. on the Screen am getting 503 service unavilable I don't know what happening in Server side. Please Help in this . -- Thanks &Regards Ravi Prakash Sunkara VoIP Development Tech Lead.

2 1

[OpenSER-Users] error_route not working
by Jayesh Nambiar 19 Dec '07

19 Dec '07

Hi All, I am using openser 1.3. Yesterday I added the error_route in my script to handle the parsing errors. But I see that the script logic is not going into error_route and I still see the message log filled up with messages like below: INFO:core:parse_first_line: bad message Dec 19 08:02:17 sw1 /usr/local/sbin/openser[26964]: ERROR:core:parse_msg: message=<INVITE sip:61: 394086416(a)207.251.84.106 SIP/2.0 If the error route is present in the script should'nt the script go to the error route in the case like above? My error_route is as follows: error_route { xlog("--- error route class=$(err.class) level=$(err.level) info=$(err.info) rcode=$(err.rcode) rreason=$(err.rreason) ---\n"); xlog("--- error from [$si:$sp]\n+++++\n$mb\n++++\n"); sl_send_reply("$err.rcode", "$err.rreason"); exit; } Can somebody give me some pointers on why error_route is not working in this case or am I missing something. Thanks in advance w/regards, Jayesh Save all your chat conversations. Find them online at http://in.messenger.yahoo.com/webmessengerpromo.php

1 0

[Serusers] nat problem
by Pezhman Lali 19 Dec '07

19 Dec '07

Dear I have a lot of problems with nat and registering, for example, grandstream does not work with a lot of internet providers, also linksys and ... sometimes the sip phone was registered, but the call is one way or no way. please help me, if my config has problem. Best Mani listen=1.1.1.1 debug=0 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL5 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="postgres://postgres:mypass@localhost:5432/test" alias="xxx.com" alias="xxx.com" alias="xxx.com" alias="xxx.com" alias="xxx.com" sip_warning=yes # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/postgres.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/acc.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/nathelper.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/xlog.so" loadmodule "/usr/local/lib/ser/modules/uri.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("acc", "log_level", 1) #modparam("acc", "log_flag", 1) modparam("acc", "db_url", "postgres://postgres:mypass@localhost:5432/test") modparam("acc", "db_missed_flag", 2) modparam("acc", "db_flag", 1) #modparam("acc", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("usrloc", "db_mode", 1) modparam("usrloc", "user_column", "user_id") modparam("usrloc", "db_url", "postgres://postgres:mypass@localhost:5432/test") # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) modparam("auth_db", "use_domain", 0) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "user_column", "user_id") modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "postgres://postgres:mypass@localhost:5432/test") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("rr", "enable_double_rr", 0) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 0) modparam("usrloc", "use_domain", 0) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("nathelper", "rtpproxy_sock","unix:/var/run/rtpproxy.sock") # ------------------------- 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"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; if (method=="INVITE" || method=="BYE" || method=="ACK") { setflag(1); }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". 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 }; }; # 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); break; }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); break; }; if (method == "INVITE") { # if (!lookup("location")) { # xlog("L_ERR","%Tf: Rejecting %ru From %fu To %tu \n"); # sl_send_reply("404", "Not Registered"); # break; # }; if (!uri =~ "sip:43[0-9]{6}@*") { log(1, "Forwarding to Asterisk\n"); forward(2.2.2.2,5060); break; } } # 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") { if (!www_authorize("xxx.com", "subscriber")) { www_challenge("xxx.com", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); #acc_db_request("404 missed call", "missed_calls"); break; }; setflag(2); }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; } ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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[Serusers] multi-domain support solution
by satish patel 19 Dec '07

19 Dec '07

Dear all I have solve my multidomain configuration with ser-0.9.6 and it is working fine for me and i am sending you my configuration for incomming users whos want do some stuff wid ser i have find many document on multi-support and i have consolidate all document and make my ser.cfg script or config cat /usr/local/etc/ser/ser.cfg # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=7 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.104.20 port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/domain.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/call.so" #loadmodule "/usr/local/lib/ser/modules/vm.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/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", 0) # -- 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) modparam("domain", "db_url", "mysql://root:mypasswd@localhost/ser") modparam("domain", "db_mode", 1) # Use chaching modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") modparam("usrloc|registrar|auth_db|avpops|group", "use_domain", 1) #modparam("usrloc", "use_domain", 1) #modparam("registrar", "use_domain", 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"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # 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) #-----------Multi-domain Logic work blow in stanza--------# if (is_from_local()) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); break; save("aliases"); break; }; #----Alias check--- # lookup("aliases"); # if (!uri==myself) { # route(1); # break; # }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } #-----END---# Now to test configuration start ser #serctl start #serctl domain add abc.com #serctl add 222(a)abc.com tulip 222(a)abc.com #serctl add 333(a)abc.com tulip 333(a)abc.com this to user 222 and 333 for domain abc.com **Notes:- you should configure SRV recode on your DNS otherwise it will not work for you NOW configure your Xlite and try to call each other Then try to add new domain and do same thing which u have done before abc.com Thanks and Best of Luck $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Meet people who discuss and share your passions. Join them now.

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[Serusers] Fwd: Newbie Problems: ser.cfg
by frq ser 18 Dec '07

18 Dec '07

---------- Forwarded message ---------- From: frq ser <frq.ser(a)googlemail.com> Date: Dec 18, 2007 11:32 PM Subject: Newbie Problems: ser.cfg To: serusers(a)iptel.org Hi, I am trying to understand the process flow in the following ser.cfg file, which controls a SER proxy server at root@ip.address:5060. I have added "log()" command at almost every step to see how things proceed. The problem is that when I generate an OPTIONS message through SIPSAK (sipsak -T -vv -s sip:root@ip.address:5060), the process branches from route(0) to route(3), sends a message (200 Message received, probably from Sipsak), then comes back to route(0), completes this route, and then loops back to start of route(0) (as evident from the last log message "Inside main route..."). This is despite the "break" command in the main route block. Here's the output of the SER: [root@drone08 ~]# ser Listening on udp: 192.168.3.208 [192.168.3.208]:5060 tcp: 192.168.3.208 [192.168.3.208]:5060 Aliases: tcp: 192.168.3.208:5060 udp: 192.168.3.208:5060 WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 107520 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 2(0) INFO: fifo process starting: 5116 2(0) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(0) WARNING: no fifo_db_url given - fifo DB commands disabled! 0(5114) Inside the main route.... 0(5114) Inside the main route.... 0(5114) just passed the sanity checks... 0(5114) inside method-register... 0(5114) inside method-options... 0(5114) Inside route3... 0(5114) End of route3... 0(5114) just past redir-route3.. 0(5114) Inside the main route.... The corresponding SIPSAK output is: C:\..Sipsak>sipsak -v -T -s sip:root@192.168.3.208:5060 warning: IP extract from warning activated to be more informational 0: 192.168.3.208 (0.000 ms) SIP/2.0 483 Too Many Hops 1: 192.168.3.208 (0.000 ms) SIP/2.0 201 Options message received, probably from Sipsak without Contact header On the other hand, when I send an INVITE message, this problem doesn't happen. The process completes the main route and doesn't loop back (ie I don't get the "Inside the main route..." message). [root@drone08 ~]# ser Listening on udp: 192.168.3.208 [192.168.3.208]:5060 tcp: 192.168.3.208 [192.168.3.208]:5060 Aliases: tcp: 192.168.3.208:5060 udp: 192.168.3.208:5060 WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 107520 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 2(0) INFO: fifo process starting: 4860 2(0) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(0) WARNING: no fifo_db_url given - fifo DB commands disabled! 0(4858) Inside the main route.... 0(4858) just passed the sanity checks... 0(4858) inside method-register... 0(4858) outside methods-options... 0(4858) inside lookup-location... All I want to know is that when I send an OPTIONS msg, why the process doesn't terminate at the break command in the (if (method == "OPTIONS")) in the main route. Although it is not a critical problem, I just want to improve my understanding. I'll be glad if any of you could help. Fr. PS: The config file (ser.cfg) is given below: # $Id: hello-world.cfg 46 2006-01-25 04:21:30Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ debug=3 fork=no log_stderror=yes listen= 192.168.3.208 #2.13 port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" 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" # chg_rsf dec07 #loadmodule "/usr/local/lib/ser/modules/textops.so" #loadmodule "/usr/local/lib/ser/modules/avpops.so" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) #modparam("tm", "pass_provisional_replies", 1) route { log(1, "Inside the main route....\n"); # ------------------------------------------------------------------------ # 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; }; log(3, "just passed the sanity checks...\n"); # ------------------------------------------------------------------------ # Record Route Section # ------------------------------------------------------------------------ if (method!="REGISTER") { record_route(); log(3, "inside method-register...\n"); }; # ------------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------------ if (loose_route()) { log(3, "inside loose route...\n"); route(1); break; }; # ------------------------------------------------------------------------ # Call Type Processing Section # ------------------------------------------------------------------------ if (uri!=myself) { log(3, "inside uri-myself...\n"); route(1); break; }; if (method=="ACK") { log(3, "inside method-ack...\n"); route(1); break; } else if (method=="REGISTER") { log(3, "inside method-register...\n"); route(2); break; }; if (method=="OPTIONS"){ log(3, "inside method-options...\n"); route(3); log(3, "just past redir-route3..\n"); break; log(3, "just past break...\n"); }; log(3, "outside methods-options...\n"); lookup("aliases"); if (uri!=myself) { log(3, "inside uri-myself 2....\n"); route(1); break; }; if (!lookup("location")) { log(3, "inside lookup-location...\n"); sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ log(3, "Inside route1...\n"); if (!t_relay()) { sl_reply_error(); }; } route[2] { # ------------------------------------------------------------------------ # REGISTER Message Handler # ------------------------------------------------------------------------ log(3, "Inside route2...\n"); if (!save("location")) { #sl_reply_error(); sl_send_reply("201", "Register message received at Proxy Server"); }; } route[3]{ # ----------------------------------------------------------------------- # OPTIONS Message Handler # ----------------------------------------------------------------------- log(3, "Inside route3...\n"); if (!t_relay()){ sl_reply_error(); }; sl_send_reply("201", "Options message received, probably from Sipsak"); #forward(192.168.3.208, 5070); log(3, "End of route3...\n"); #t_release(); }

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[OpenSER-Users] Exposing the NIC IP's in openser.cfg
by Robert Dyck 18 Dec '07

18 Dec '07

I have been using openser as an ALG for the purpose of NAT traversal. Unfortunately my public IP is dynamically assigned. Although openser is aware of the IP's available on the host, I have searched unsuccessfully the documentation for a way to expose this information to the configuration script. Have I missed something? I am using the following ugly work around. DHCP client writes to /var/openser/my-ip From the rtpproxy init script - OPTIONS="-l 192.168.1.1/`cat /var/openser/my_ip` -s udp:localhost:12221" from openser.cfg - route{ # get our public ip, this happens only once if !$var(ip_initialized) == "yes" { xlog("L_NOTICE", "****************************************************\n"); xlog("L_NOTICE", "*** Initializing the IP for masquerading SIP Contact\n"); exec_avp("cat /var/openser/my_ip", "$avp(s:pub_ip)"); pv_printf("$var(my_ip)", "$avp(s:pub_ip)"); xlog("L_NOTICE", "*** New public IP is <$var(my_ip)>\n"); $var(ip_initialized) = "yes"; }; mangle contacts

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