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[Serusers] Is there something weird about this INVITE ?
by SIP 27 Jul '07

27 Jul '07

One of our clients, using a Nokia E90, is having problems connecting to anything. His INVITE looks like this: U 123.236.98.24:55031 -> 11.22.33.44:5060 INVITE sip:1747XXXXXXX@our.proxy.com;user=phone SIP/2.0. Route: <sip:our.proxy.com;lr;transport=UDP>. Via: SIP/2.0/UDP 192.168.1.229:5060;branch=z9hG4bK3e89pocn1m7mkuiefdjljsj;rport. From: <sip:1101XXXXXXX@our.proxy.com>;tag=ls57mprkkdhc7kku07de. To: <sip:1747XXXXXXX@our.proxy.com;user=phone>. Contact: <sip:1101XXXXXXX@123.236.98.24:55045;transport=UDP>. Supported: 100rel,sec-agree. CSeq: 980 INVITE. Call-ID: pyBXpNNeoIcFBM7qVqVKrBneC2kxeM. Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK. Expires: 120. Privacy: none. User-Agent: Nokia RA-6 V 07.24.0.3. P-Preferred-Identity: sip:1101XXXXXXX@our.proxy.com. Max-Forwards: 70. Proxy-Authorization: Digest qop=auth,realm="our.proxy.com",nonce="46a713c0c887788778a731de0355c04db1896d23",algorithm=MD5,username="1101XXXXXXX",cnonce="0dad64affaabb2fa9c93611ec864e363",nc=00000001,uri="sip:1747XXXXXXX@our.proxy.com;user=phone",response="197f8a8b4b3d39281a147c2d9d2b6a28". Content-Type: application/sdp. Accept: application/sdp. Content-Length: 447. . v=0. o=Nokia-SIPUA 63353630343979125 63353630343979125 IN IP4 123.236.98.24. s=-. c=IN IP4 123.236.98.24. t=0 0. m=audio 49152 RTP/AVP 96 0 8 97 18 98 13. a=sendrecv. a=ptime:20. a=maxptime:200. a=fmtp:96 mode-change-neighbor=1. a=fmtp:18 annexb=no. a=fmtp:98 0-15. a=rtpmap:96 AMR/8000/1. a=rtpmap:0 PCMU/8000/1. a=rtpmap:8 PCMA/8000/1. a=rtpmap:97 iLBC/8000/1. a=rtpmap:18 G729/8000/1. a=rtpmap:98 telephone-event/8000/1. a=rtpmap:13 The server received this packet, but never responds to it. It simply never sends a reply. It's as though it doesn't recognise this as a proper INVITE. The ONLY thing I noticed about this that's completely non-standard, is that every other packet seems to have this period after it (indicating, perhaps, the end of a line... I don't know enough about the display of ngrep to tell). This INVITE does not. It goes almost immediately into the next packet, and it's almost as though this one never ends. Has anyone seen this before? Is there anything weird about this INVITE that would cause SER to ignore it? N.

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[OpenSER-Users] Re: [Serusers] CDRz?
by Parminder.Nayyar＠relianceada.com 27 Jul '07

27 Jul '07

Hello Friends ... Jai, Thanks for showing a pointer towards teh acc module... but i really cudn't grasp even a bit out of it... all it says is about logs being dumped, with specific data like \: The fixed minimal accouting information is: Request Method name >From header TAG parameter To header TAG parameter Call-Id 3-digit Status code from final reply Reason phrase from final reply Time stamp when transaction was completed There is no session/dialog accounting (yet)-- OpenSER maintains no sessions. If one needs to correlate INVITEs with BYEs for generating proper CDRs for example for purpose of billing, then it is better done in the entity which collects accounting information. If a UA fails in middle of conversation, a proxy will never learn it. In general, a better practice is to account from an end-device (such as PSTN gateway), which best knows about call status (including media status and PSTN status in case of the gateway). Now, my requirement is to generate a CDR for billing purpose... In syslog there are not any logs present, and even if i get the logs there.. using changes in config file, how do i use this raw information to generated CDRz for billing purposes??? I have searched for quite a few tools online which work on the output of SER to generate CDR or for CDR collection... but I need a free software, they are all paid services... Can someone help me where should i go further..? I'm lost ... :( Thanks a lot! Parminder Nayyar "Jai Rangi" <jprangi(a)gmail.com> 07/23/2007 10:52 PM To "Parminder.Nayyar(a)relianceada.com" <Parminder.Nayyar(a)relianceada.com> cc Subject Re: [Serusers] CDRz? acc module can be used to all CDRs. But I found that useless for billing purpose. Jai www.bingotelecom.com On 7/22/07, Parminder.Nayyar(a)relianceada.com < Parminder.Nayyar(a)relianceada.com> wrote: Hi Can anyone tell me which module to explore , if i want to check about CDRz and billing?? Please help..... Parminder Nayyar _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] can SIP client connect sip proxy via SOCKS5 proxy?
by Liu Wenlong 27 Jul '07

27 Jul '07

Is there any protocol support it? thx -- aRES iN aNYWhere

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[OpenSER-Users] t_on_branch -- using drop() question?
by Zahid Mehmood 26 Jul '07

26 Jul '07

I was looking at the openser config generated using the online sipwise wizard. They implement conditional call forwarding using: if($avp(s:cfc) != NULL) { avp_pushto("$ru", "$avp(s:cfc)"); setflag(29); append_branch(); t_on_branch("1"); xlog("L_INFO", "CFC detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(13); } branch_route[1] { if(is_domain_local("$rd")) { xlog("L_INFO", "Dropping local branch - M=$rm RURI= $ru F=$fu T=$tu IP=$si ID=$ci\n"); drop(); } } Suppose an invite gets to this point and ruri is changed to another local user. does this drop() in branch_route prevents that invite to be relayed? does it still go through route(13) ? I'm still a newbie trying to better understand the working of openser functions. I'll greatly appreciate if someone can briefly describe when it makes sense to use "append_branch()" Thanks in advance for your help. -- Zahid

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[OpenSER-Users] openSER crashes while accessing emtpy pseudo-variable $fn in if-clause ?
by Rainer Strobel 26 Jul '07

26 Jul '07

Hello, while playing around with the pseudo variables whithin openser 1.2.1 I got a strange behavior. I tried to check if the from-header-name of an invite message is set to (A|a)nonymous with an simple if-clause. if($fn =~"(A|a)nonymous") { ...do something ; } It works fine while the from-name variable is not empty. If the from-header-name in the INVITE message is not given openser crashes exactly on that comparison. After that I started playing around using other comparisons, for example 'if($fn=~"foo")' or ' if($fn == "foo")' , with the same set-up. Openser crashed each time processing that if-clause. I tested this with openser-1.2.1-notls on i386 openSUSE 10.2 with MySQL 5.0.26 and gcc 4.1.2 Can anybody confirm this problem ? Debuggin informations are available and can be send additional if necessary. Regards. Rainer Strobel --------------------------------------------------------------------- HFO Telecom AG Network Management Bahnhofstraße 18 95028 Hof Tel.: + 49 (0) 92 81 / 14 48 132 Fax: + 49 (0) 92 81 / 14 48 52 132 E-Mail: strobel(a)hfo-telecom.de Internet: www.hfo-telecom.de

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[OpenSER-Users] Problem with handling 3xx responses and contact with maddr
by Watkins, Bradley 26 Jul '07

26 Jul '07

I'm using OpenSER to connect to a Nortel CS1000M, and I'm having some problems. The way the Nortel works is that it has what they call a Network Routing Service (redirect server, really), that issues either a 302 (if a single SIP gateway exists on the CS1000) or a 300 (if multiple). Here's a sample 300 from my system (IPs and domains changed to protect the guilty): U 10.1.1.3:5060 -> 10.0.1.1:5060 SIP/2.0 300 Multiple Choices. Via: SIP/2.0/UDP 10.0.1.1;branch=z9hG4bKa66e.99fc1a25.1;received=10.0.1.1. Via: SIP/2.0/UDP 10.1.3.114;branch=z9hG4bK7d42804867A25509. From: "0004f2145263" <sip:0004f2145263@dtw-openser01.sub.domain.com>;tag=5F7CDD45-DDCE1F92. To: <sip:12345@dtw-openser01.sub.domain.com;user=phone>;tag=34887. Call-ID: 58dd5646-e86d140f-c2ead634(a)10.1.3.114. CSeq: 1 INVITE. Contact: <sip:12345;phone-context=udp@domain.com:5060;maddr=10.1.1.1;transport=ud p;x-nt-redirect=redirect-server;x-nt-redirect=redirect-server>. Contact: <sip:12345;phone-context=udp@domain.com:5060;maddr=10.1.1.2;transport=tc p;x-nt-redirect=redirect-server;x-nt-redirect=redirect-server>. Content-Length: 0. The problem I have is that the Contacts have an maddr, and I need relay to it. However, I can't seem to be able to use transformations to get the maddr for relaying. Here are some log messages with comments inline: Jul 16 19:50:49 dtw-openser01 /usr/local/sbin/openser[4306]: Branches are: <sip:12345;phone-context=udp@domain.com:5060;maddr=10.1.1.1;transport=ud p;x-nt-redirect=redirect-server>;q=0.01 This is just an xlog of $bR, but I'm not sure where the q= value is coming from (get_redirects?). Jul 16 19:50:49 dtw-openser01 /usr/local/sbin/openser[4306]: ERROR: parse_uri: bad uri, state 0 parsed: <<sip> (4) / <<sip:12345;phone-context=udp@domain.com:5060;maddr=10.1.1.1;transport=u dp;x-nt-redirect=redirect-server>;q=0.01> (120) Jul 16 19:50:49 dtw-openser01 /usr/local/sbin/openser[4306]: tr_eval_uri: invalid uri [<sip:12345;phone-context=udp@domain.com:5060;maddr=10.1.1.1;transport=u dp;x-nt-redirect=redirect-server>;q=0.01] These are what I get as a result of trying to xlog $(bR{uri.maddr}) Anyway, I would be greatly appreciative of any help someone might offer on the matter. Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

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[OpenSER-Users] [multidomain] Using different RtpProxy for each domain
by Iñaki Baz Castillo 26 Jul '07

26 Jul '07

Hi, I've configured a OpenSer in multidomain with a RtpProxy. Now I'd like to use a RtpProxy for each domain, so it could be based in $rd. But at first point I don't know how to configure OpenSer to use varios RtpProxies in different machines. I've tried with: modparam("nathelper", "rtpproxy_sock", "udp:localhost:22222,udp:192.168.1.58:22222") and: modparam("nathelper", "rtpproxy_sock", "udp:localhost:22222udp:192.168.1.58:22222") but it fails when starting giving an error of parsing. If I use: modparam("nathelper", "rtpproxy_sock", "udp:localhost:22222,udp") modparam("nathelper", "rtpproxy_sock", "udp:192.168.1.58:22222") just the last line is used. Any idea of how to use various RtpProxies? My idea is to have some of them availables and use them based in $rd using: force_rtp_proxy("","192.168.1.58") and so. Any help please? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] What does "script writer didn't release transaction" mean?
by isetien001＠ikasle.ehu.es 26 Jul '07

26 Jul '07

Hi! I get the following warning when I start openser. Any idea of what it is? I'm using Openser with the xmpp module. Thanks, Izortze. # openser start Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 158.227.64.88 [158.227.64.88]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 158.227.64.88 [158.227.64.88]:5060 Aliases: tcp: b001948.bi.ehu.es:5060 tcp: localhost:5060 udp: b001948.bi.ehu.es:5060 udp: localhost:5060 0(21183) init_tcp: using epoll_lt as the io watch method (auto detected) 0(0) INFO: statistics manager successfully initialized 0(0) StateLess module - initializing 0(0) TM - initializing... b001948:/usr/local/etc/openser# 0(0) Maxfwd module- initializing 0(0) INFO:ul_init_locks: locks array size 512 0(0) TextOPS - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 107520 0(0) INFO: udp_init: SO_RCVBUF is finally 215040 0(0) INFO: udp_init: SO_RCVBUF is initially 107520 0(0) INFO: udp_init: SO_RCVBUF is finally 215040 1(21186) INFO:mi_fifo:mi_child_init(1): extra fifo listener processes created 5(21190) WARNING: script writer didn't release transaction 6(21191) WARNING: script writer didn't release transaction 7(21192) WARNING: script writer didn't release transaction 6(21191) WARNING: script writer didn't release transaction 8(21193) WARNING: script writer didn't release transaction 7(21192) WARNING: script writer didn't release transaction 8(21193) WARNING: script writer didn't release transaction 5(21190) WARNING: script writer didn't release transaction 7(21192) WARNING: script writer didn't release transaction 5(21190) WARNING: script writer didn't release transaction 7(21192) WARNING: script writer didn't release transaction

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[Serusers] ser.cfg syntax error
by Emiliano Ogando 26 Jul '07

26 Jul '07

This is the beginning of my ser.cfg file I've got syntax error on line 12, when I run the ser process. The ser_fifo file should be created when I compile ser or when I run this script?? debug=3 fork=no log_stderror=yes listen=192.9.208.66 port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" #line 12 Regards Emiliano Ogando Dirección General de Gestión Informática Ministerio del Interior 25 de Mayo #101 TE: (54-11) 4346-1663 Fax: (54-11) 4346-1633 Web: <http://www.mininterior.gov.ar/> http://www.mininterior.gov.ar Buenos Aires - República Argentina Nota: todos los mails enviados desde esta cuenta eogando(a)mininterior.gov.ar<mailto:eogando@mininterior.gov.ar> serán válidos únicamente si se encuentran firmados digitalmente, certificados por la autoridad Oficina Nacional de Tecnologías de la Información, de la Jefatura de Gabinete de Ministros de la Presidencia de la Nación Argentina.

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[OpenSER-Users] Regarding parse error
by Ashish Barot 26 Jul '07

26 Jul '07

Hello all, right now i had just try to install openser-1.2.1-notls it get installed with out any error while doing make and make install but after that if i try to start service then it is giving me error openser: parse error (23,1-5): syntax error Jul 27 04:46:39 barot1 openser: parse error (23,1-5): i get above error when i try to run "openser start" command. i am using Fedora Core-6 as my OS. can anybody guide me on this? Thanks. Best Regads, Ashish Barot.

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[OpenSER-Users] Concept Question Regaring OpenSer and MediaProxy
by Christopher Aloi 25 Jul '07

25 Jul '07

Hello All - First - I am new to SER; but versed in telephony, so apologies for any incorrect verbiage. I've been reading a bit on OpenSer and MedaProxy. I plan to build an proxy to connect two separate segments of our internal network. I will need SER to pass signaling and media from networkA to networkB. There will be no NAT. My question is -> All the reading i've done regarding proxying media with OpeSer relates to NAT traversal solutions; do the core concepts still apply for traversing multiple networks w/no NAT? Thank, Chris

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[OpenSER-Users] Language encoding
by Greg Fausak 25 Jul '07

25 Jul '07

I hope someone can help me with a language encoding problem. I don't have much experience with non-ascii From headers. What is the correct way to encode in the SIP headers? The reason I ask is because we're having problems with parse_from() barfing on 8859-1 characters. Do we need to just use utf-8? Are other charsets supported? Thanks, -- Greg Fausak greg(a)thursday.com

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[Serusers] FW: SER Fail Over
by Ravin Suri 25 Jul '07

25 Jul '07

Can you elaborate what is "heartbeat"? Thanks for your help. - Ravin _____ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Jai Rangi Sent: Tuesday, July 24, 2007 9:30 PM To: Arun Kumar Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SER Fail Over Arun, You can use heartbeat. That way you will be able to use the same IP for both ser servers. Thats what we have in our callingcard application. -Jai www.bingotelecom.com <http://www.bingotelecom.com> On 7/24/07, Arun Kumar <arunvoip(a)gmail.com> wrote: Hi I've two sites running my ser with two different database (master-slave). How do I configure fail over between my ser ? thanks arun _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[OpenSER-Users] Wrong/Double text appears in Contact header during sequential forking
by liran tal 25 Jul '07

25 Jul '07

Hey everyone, I'm using sequential forking and on one of the scenarios there appears to be a problem. When OpenSER attempts to find the first most relevant destination for the call the SIP headers are ok. If the first destination that OpenSER attempts to contact is offline/unreachable it continues to the next one in turn in which it produces a bad Contact header which looks like this: Contact: <sip:101@192.168.0.1:5060sip:101@192.168.0.1:5060> As you can see it's writing the sip information twice for some reason. Has anyone seen this happen before? Also, where should I be looking at to find the problem? Thanks, Lir.

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[OpenSER-Users] OpeSER Aministrator Training Course at VoN Itally, Rome - Details
by Bogdan-Andrei Iancu 25 Jul '07

25 Jul '07

Hi everybody, As promised, we finalized all the arrangements and here are the last details about the "OpenSER Admin Training Course" at VoN Italy, Rome. Prices, dates, agenda, registration forms and more others can be found on the OpenSER project site, at: http://www.openser.org/mos/view/OpenSER-Admin-Course---Rome-2007/ The registration process is on, so feel free and join :). Thanks and regards, Bogdan

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[Serusers] SER-routing-problem
by Vlad Costea 25 Jul '07

25 Jul '07

Hello. I have just installed SER on RedHat 9 Linux with the purpose to route voice packages between a PSTN Gateway AS5350 and a Grandstream HandyTone 286.Letme tell you how it should work: I have assigned a phone number, that when it is picked-up, the IVR from the AS5350 router respods and after the press of a key it sends the packages to the HandyTone and from there to a normal phone; So far so good because it partialy works, that means that the phone rings butt there is no voice. I have managed to solve this problem using a hardware Sip Server between the two devices ( in this way both HT286 and AS5350 act as clients), but this is not possible any more because it was not my server; so i have tried with a software solution and I've installed SER. As I said before , I haven't managed to configure it to work as I wish and this is why I'am asking for your help; it would realy help me if you can provide a ser.cfg example that would do just the routing part (no ack no authentification , no mysql,...., just routing). Please excuse my bad english. Thank you very much !

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[OpenSER-Users] Info about OpenSer Von Italy course
by Iñaki Baz Castillo 25 Jul '07

25 Jul '07

Hi, I'm interested in going to that course but I get no info about it. I sent some mails with no response. I'd just need to know if I could buy the plane tickets and reserve hotel now or I should wait for the registration process. Could somebody give me some info about it? Thanks for all and regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] linux kernel + iptables + sip nat module + RTP streams
by Marc LEURENT 25 Jul '07

25 Jul '07

Good Morning, I need to set a VOIP architecture with natted users. And I'm trying to prevent bottleneck (with a rtpproxy architecture) and I have a full access to the nat server (debian box / iptables MASQUERADE NAT). So, I've compiled nf_nat_sip / nf_conntrack_sip modules that changed the received RTP stream IP with the public one of my nat box. So the RTP stream is arriving to the natbox! Do you have any idea of a solution to forward RTP streams to the end user Best Regards, Marc

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[OpenSER-Users] Problem matching GTALK with Openser
by isetien001＠ikasle.ehu.es 25 Jul '07

25 Jul '07

Hi! I have load the XMPP module to my openser configuration to integrate GTALK in my SIP system designed with Openser. The XMPP module works in the 'component' mode. I have added the server Jabberd 2 to my system and I have checked that it works correctly with different Jabber clients. When I start Openser and try to connect GTALK to it I get the following error: 10(8016) ERROR:parse_first_line: error parsing the method 10(8016) ERROR:parse_first_line: bad message 10(8016) ERROR: parse_msg: message=<CONNECT www.google.com:443 HTTP/1.0 User-Agent: Google Talk Host: www.google.com Content-Length: 0 Proxy-Connection: Keep-Alive > 10(8016) ERROR: receive_msg: parse_msg failed 10(8016) ERROR: receive_msg: no via found in request 10(8016) ERROR: receive_msg: no via found in request Any idea of what it's wrong? Thanks in advance! Izortze. P.D= This is my openser.cfg file: $Id: openser.cfg 1676 2007-02-21 13:16:34Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes #log_stderror=no # (cmd line: -E) children=4 # Uncomment these lines to enter debugging mode #fork=no log_stderror=yes # check_via=no rev_dns=no dns=no port=5060 #fifo='/tmp/openser_fifo' # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- #set module path mpath="/usr/local/lib/openser/modules/" # Uncomment this if you want to use SQL database 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 "mi_fifo.so" loadmodule "xmpp.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "auth.so" #loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- 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) #-- xmpp module parameters -- modparam("xmpp","backend","component") modparam("xmpp","domain_separator","*") modparam("xmpp","gateway_domain","b001948.bi.ehu.es") modparam("xmpp","xmpp_domain","b001948.bi.ehu.es") modparam("xmpp","xmpp_password","secret") modparam("xmpp","xmpp_host","b001948.bi.ehu.es") # ------------------------- 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; }; # filter XMPP destinations if (uri =~ "sip:.+\*b001948.bi.ehu.es@b001948.bi.ehu.es") { # absorb retransmissions if (!t_newtran()) { sl_reply_error(); exit; } # handle IM if (is_method("MESSAGE")) { log("message to XMPP: from <$fu> to <$ru>\n"); if (xmpp_send_message()) { sl_send_reply("202", "Accepted"); } else { sl_send_reply("404", "Not found"); } exit; } # unsupported type of SIP message log("message to XMPP: method <$rm> not supported yet\n"); sl_send_reply("503", "Service unavailable"); exit; } else { # 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; #}; 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; }

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[OpenSER-Users] Multidomain support
by Iñaki Baz Castillo 25 Jul '07

25 Jul '07

Hi, I want to mantain independent domains in OpenSer. In my case I've a OpenSer with a single DNS A record and various CNAME (I still don't want to play with SRV and so). so: DNS A = openser.domain.org CNAME = sip1.domain.org CNAME = sip2.domain.org And I want users of sip1.domain.org and sip2.domain.org, as independent groups. I just want to avoid SIP interdomain messages, so 200(a)sip1.domain.org CAN'T invite 300(a)sip2.domain.org even if he does authentication. I've loaded "domain" module and use "is_uri_host_local()" and "is_from_local()" functions, but for now I only used one domain. My question is very general: for implement (or avoid) interdomain comunication, do I need to use the "domainpolicy" [1] module? I've read its doc and know it's based in 3 drafts [2][3][4], but all of them seems to be based in the complex NAPTR record and so. Is it the way? I think I could just compare the FROM domain with the TO domain in order to avoid interdomain communication, but of course I'd like in the future the possiblity of allowing some domains to contact some other domains. Is then "domainpolicy" the solution I should learn? Any other doc about it? Thanks for any help. [1] http://www.openser.org/docs/modules/1.2.x/domainpolicy.html [2] http://tools.ietf.org/id/draft-lendl-domain-policy-ddds-02.txt [3] http://tools.ietf.org/id/draft-lendl-speermint-federations-02.txt [4] http://tools.ietf.org/html/draft-lendl-speermint-technical-policy-00 -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Is not possible to register in OpenSer dokuwiki?
by Iñaki Baz Castillo 25 Jul '07

25 Jul '07

I just see the option to login but not register, is not possible? Anyway it allows anonymous edit. PD: This is because I see an error in: http://www.openser.org/dokuwiki/doku.php/pseudovariables:1.2.x#domain_of_de… There was: Domain of destination URI $dd - reference to domain of destination uri but that is not correct at least in 1.2.1 version. I've changed that to $fd. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] "openserctl online" command showing user@domain
by Iñaki Baz Castillo 25 Jul '07

25 Jul '07

Hi, I've configured a multidomain OpenSer but "openserctl online" just shows the usernames and not the domain. Is it possible to get a list of registered user with syntax "user@domain"? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] TM request function - callback data parameter
by Tiberius DULUMAN 25 Jul '07

25 Jul '07

Hello, I'm confused with the callback data parameter of request function of TM module: does TM takes the ownership of the callback data? I've seen that, if there is an error, t_request function frees the callback data. Also, I've seen empty_tmcb_list function that free that callback data, too. I've checked other modules, to see how do they use this parameter, and I've got even more confused. msilo.c : it initialize the pointer from an int; there is a problem in request function from uac.c file, after the 'err:' label, when 'if (cp) shm_free(cp)' is executed. pua.c and send_publish.c: the callback functions free the callback data parameter; if the TM took the ownership, then that memory is freed twice. So, what is the right way to handle the memory pointed by callback data? Thank you _________________________________________________________________ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/

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[OpenSER-Users] forward one user to a spezific ip
by NicklasZ 25 Jul '07

25 Jul '07

Hi what do i have to do to forward only 1 user to a spezific IP address on INVITE? if(method=="INVITE") { if (uri=~"^sip:\200" || src_ip == xxx.xxx.xxx.xxx) { ??????? it should be forwarded on fixed ip yyy.yyy.yyy.yyy:5060 } }; i think its very easy but i'm a greenhorn ;) Thanks Andy

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[OpenSER-Users] regarding openser.cfg file
by Ashish Barot 25 Jul '07

25 Jul '07

Respected All, right now i had installed the openser-1.0.1 in my /usr/local/src/openser diectory, so i want to know which one is my main configuraion file, [i.e., openser.cfg] as i am finding that file at two places, and they are [1] /usr/local/etc/openser/openser.cfg [2] /usr/local/src/openser/openser-1.0.1/etc/openser.cfg Now i want to know, when we start openser, which file will be called ? As i had check both of those files, and there is no difference between them. so i want to understand that, which file is called first? Thanks. Best Regards, Ashish Barot.

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