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[OpenSER-Users] Mediaproxy listening but handling zero bytes of audio
by Andy Smith 05 Feb '08

05 Feb '08

Hi, Im trying to test mediaproxy for the first time, I am forcing all traffic through mediaproxy in the openser.cfg (there's nothing wrong with that in principal is there?). I have one SIP handset hanging directly off OpenSER and another connected to Asterisk which has a channel configured to OpenSER. I can get the call answered but there is audio only in one direction and even more strangely I see that mediaproxy has not handled any data at all in the audio stream. This is what I see from mediaproxy: Feb 5 21:26:05 server proxydispatcher[13968]: request 75c5e34d0e567d5f1fa22f583ecc7af0(a)10.1.1.1 10.1.1.1:16042:audio 10.1.1.1 10.1.1.1 remote server.domain.net local Asterisk=20PBX info=from:0042*001@10.1.1.1,to:0044208888888@server.domain.net,fromtag:as1a72639d,totag: Feb 5 21:26:05 server mediaproxy[13965]: request 75c5e34d0e567d5f1fa22f583ecc7af0(a)10.1.1.1 10.1.1.1:16042:audio 10.1.1.1 10.1.1.1 remote server.domain.net local Asterisk=20PBX info=totag:,to:0044208888888@server.domain.net,from:0042*001@10.1.1.1,fromtag:as1a72639d Feb 5 21:26:05 server mediaproxy[13965]: session 75c5e34d0e567d5f1fa22f583ecc7af0(a)10.1.1.1: started. listening on 10.1.2.2:60006 Feb 5 21:26:05 server mediaproxy[13965]: execution time: 1.49 ms Feb 5 21:26:05 server proxydispatcher[13968]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '10.1.2.2 60006' Feb 5 21:26:05 server proxydispatcher[13968]: execution time: 5.03 ms Feb 5 21:26:53 server proxydispatcher[13968]: delete 75c5e34d0e567d5f1fa22f583ecc7af0(a)10.1.1.1 info= Feb 5 21:26:53 server mediaproxy[13965]: delete 75c5e34d0e567d5f1fa22f583ecc7af0(a)10.1.1.1 info= Feb 5 21:26:53 server mediaproxy[13965]: session 75c5e34d0e567d5f1fa22f583ecc7af0(a)10.1.1.1: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) Feb 5 21:26:53 server mediaproxy[13965]: session 75c5e34d0e567d5f1fa22f583ecc7af0(a)10.1.1.1: ended. Feb 5 21:26:53 server mediaproxy[13965]: execution time: 0.60 ms Feb 5 21:26:53 server proxydispatcher[13968]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '' Feb 5 21:26:53 server proxydispatcher[13968]: execution time: 1.51 ms This "0/0/0 packets, 0/0/0 bytes (caller/called/relayed)" being the part that I dont understand, well it proves its not handling the audio stream, but other than put a use_media_proxy command in the openser route there isnt anything else I can do to get the data stream to route via media proxy is there? My openser route is very simple: route{ if (!mf_process_maxfwd_header("15")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); exit; }; if (method=="OPTIONS") { exit; }; if (method=="REGISTER") { record_route(); if (!save("location")) { sl_reply_error(); }; exit; }; if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); if(is_method("BYE")) { # log it all the time acc_rad_request("200 ok"); acc_log_request("200 ok"); end_media_session(); } route(1); }; if(is_method("INVITE") && !has_totag()) { # set the acc flags record_route(); use_media_proxy(); setflag(1); # setflag(2); $avp(s:can_uri) = $ru; }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (!lookup("location")) { # log to acc as missed call acc_rad_request("404 Not Found"); acc_log_request("404 Not Found"); sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } # SECTION 5 - SECONDARY ROUTE BLOCKS route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } Anyone any ideas what might be going on or how I can investigate the issue further? thanks for any help! Andy.

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[OpenSER-Users] Remote Mediaproxy Server
by Abdul 05 Feb '08

05 Feb '08

Hi All, I read readme file of Mediaproxy that there is possibility to install mediaproxy on different server than Openser. I have doubt how i can connect to that server my Mediaproxy server is in NAT mode and the IP is : 192.168.1.77 and my openser server is NO-NAT Mode and IP is : 212.XXX.XXX.XXX Is there any possibility to communicate to Mediaserver? I tried the following in openser.cfg but i got error "CANNOT CONNECT TO MEDIAPROXY SERVER" I forced media proxy to use 30001 port to accept remote command. loadmodule "mediaproxy.so" modparam("mediaproxy", "mediaproxy_socket", "192.168.1.77:30001") modparam("mediaproxy", "sip_asymmetrics", "/usr/local/src/sip-server/modules/mediaproxy/config/sip-asymmetric-clients") modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/src/sip-server/modules/mediaproxy/config/rtp-asymmetric-clients") modparam("mediaproxy", "natping_interval", 60) -------- Regard, --------------------------------- Looking for last minute shopping deals? Find them fast with Yahoo! Search.

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Re: [OpenSER-Users] Remote Mediaproxy Server
by Andy Smith 05 Feb '08

05 Feb '08

Hi Abdul, looking at the documentation this isnt a valid option: modparam("mediaproxy", "mediaproxy_socket", "192.168.1.77:30001")Check http://openser.org/docs/modules/1.3.x/mediaproxy.htmlThe mediaproxy docs included with the mediaproxy software gives details of how to configure remote mediaproxy servers, from my understanding you need to have a local proxydispatcher.sock socket (defined in openser.cfg) then you need to follow the INSTALL and/or README files included with mediaproxy to correctly configure your remote mediaproxy servers (and how they are located by the proxydispatcher) cheers Andy.

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[Serusers] commercial help requested
by Kenneth 05 Feb '08

05 Feb '08

We need someone who can quickly assist us with creating the P-asserted header on SER. RFC 3325. Configuration. If there is an expert out there, please contact us with your qualifications and rates. Kenneth

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[Serusers] Multiple Subscription (PA)
by Carlos Angeles 05 Feb '08

05 Feb '08

Hi... I am using the SER Presence agent for polling information from the server by sending SUBSCRIBE with expiration = 0 (just fetching the information). However, the server only responds to SUBSCRIBES with a time interval of 5 secondes between them. Is there any timer or parameter I can modify in order to reduce this time. Thanks Carlos _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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[Serusers] ser rel_2_0_0_rc6 from cvs doesn't start and breaks down
by Tomasz Zieleniewski 05 Feb '08

05 Feb '08

Hi, I have just downloaded the latest cvs sources of rel_2_0_0. I compiled and installed it as usually: make proper make prefix=/homedir install after this each time I start ser it breaks down here my log from console: Starting SER SIP Proxy: ser $Listening on udp: 127.0.0.1 [127.0.0.1]:7060 udp: 192.168.129.38 [192.168.129.38]:7060 tcp: 127.0.0.1 [127.0.0.1]:7060 tcp: 192.168.129.38 [192.168.129.38]:7060 Aliases: tcp: venom:7060 tcp: venom.touk.pl:7060 tcp: localhost:7060 udp: venom:7060 udp: venom.touk.pl:7060 udp: localhost:7060 *: sip.rd.touk.pl:* WARNING: no fork mode and more than one listen address found (will use only the the first one) 0(11117) init_tcp: using epoll_lt as the io watch method (auto detected) 0(11117) Maxfwd module- initializing 0(11117) WARNING: xl_mod_init: more IP 192.168.129.38 not used 0(11117) WARNING: xl_mod_init: more IP 192.168.129.38 not used 0(11117) WARNING: xl_mod_init: more IP 192.168.129.38 not used 0(11117) WARNING: xl_mod_init: more IP 192.168.129.38 not used 0(11117) WARNING: xl_mod_init: more IP 192.168.129.38 not used 0(11117) WARNING: xl_mod_init: more IP 192.168.129.38 not used 0(11117) WARNING: xl_mod_init: more IP 192.168.129.38 not used 0(11117) WARNING: xl_mod_init: more IP 192.168.129.38 not used 0(11117) WARNING: xl_mod_init: more IP 192.168.129.38 not used 0(11117) AVPops - initializing 0(11117) using 'none' subscription authorization for presence! 0(11117) TRACE: qsa_interface.c:55: PA: PA_CONTENT_TYPE: 0x2ba1844692e0 0(11117) INFO: xcap_mod.c:55: xcap module initialization 0(11117) TRACE: virtual_subscription.c:55: RLS: RLS_PRESENCE_INFO: 0x2ba1844692e0 0(11117) TRACE: virtual_subscription.c:63: RLS: RLS_RAW: 0x2ba184479598 0(11117) using 'none' rls-subscription authorization! 0(11117) TRACE: qsa_events.c:331: presence_b2b: b2b_CONTENT_TYPE: 0x2ba1844692e0 0(11117) WARNING: tls: mod_init: tls support is disabled (set enable_tls=1 in the config to enable it) exec - initializing ERROR: error -1 while trying to fix configuration Best regards tomasz

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[OpenSER-Users] Modifing the SIP header fields in OpenSER
by Skender Benattia 05 Feb '08

05 Feb '08

Hello everyone; I would like to rewrite the "To" field of all the SIP messages OpenSER receives in order to make some tests. Such a function exists for the originating part using rewriteuri or rewritehost, but I could not find anything for the destination part. Is it possible ? If yes, does somebody have a clue ? It would be very helpful. Thank you very much in advances. Best Regards, Skander

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Re: [OpenSER-Users] CDRTool rating on test system (without PSTN)
by Andy Smith 05 Feb '08

05 Feb '08

:P This was simple to fix, turns out my admin account in CDRTool didnt have full admin privileges as far as CDRTool was concerned. Not sure if Id accidentically messed it up or if the default is like that... Well anyway, in case anyone else is as dumb as me in future :D now the answer is on the list, cheers Andy. >However via the web interface in the CDR Section for the cost it is >displayed as "x.xxx" and no data is displayed in the detail section "Rating >Information" either... Any ideas what the issue is?

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[OpenSER-Users] Accurate CDR with RTPProxy
by VoIP Forums www.Go4Calls.com 05 Feb '08

05 Feb '08

Hi all, Is it possible to get accurate CDR of openser using RTPProxy? If ATA did not send BYE message to openser can RTPProxy disconnect the call using some rtptimeout then ACCT module log the correct duration in MySQL database? 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/

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[Serusers] ser 2.1.0-dev16-tcp avpops module compilation error
by Tomasz Zieleniewski 05 Feb '08

05 Feb '08

Hi, I was trying to compile sources from trunk and I encoutered following error: make[1]: Entering directory `/home/sip/src/ser-trunk/modules/avpops' gcc -fPIC -DPIC -m64 -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron -Wall -DNAME='"ser"' -DVERSION='"2.1.0-dev16-tcp"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.2.3"' -D__CPU_x86_64 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/home/sip/ser/trunk/test/etc/ser/"' -DMODS_DIR='"/home/sip/ser/trunk/test/lib/ser/modules/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -c avpops.c -o avpops.o In file included from avpops_impl.h:41, from avpops_parse.h:38, from avpops.c:50: avpops_db.h:66: error: expected ')' before '*' token make[1]: *** [avpops.o] Error 1 make[1]: Leaving directory `/home/sip/src/ser-trunk/modules/avpops' make: *** [modules] Error 1 Best regards tomasz

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[OpenSER-Users] Missing information in Migration Guide from 1.2.2 to 1.3.0.
by Sergio Gutierrez 05 Feb '08

05 Feb '08

Hello everybody. Trying to migrate from 1.2.2 version to 1.3.0 version, I discovered there is a lost item in migration guide; The structure of gw table changes, because in 1.3.0 version, it has an additional column, so this table must be rebuilt. Best regards. Sergio Gutiérrez

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[OpenSER-Users] Viewing OpenSER debug output
by Brian Heath 04 Feb '08

04 Feb '08

Hi all, Is there a way that you can 're-connect' to an openser debug output process, so you don't have to restart openser every time you want to view the debug data? I've currently got it set accordingly: ################# # openser.cfg # ################ debug=9 fork=yes log_stderror=yes log_facility=LOG_LOCAL1 Alternately, if there's a way I can pipe it all into a text file, and then use tail -f that would be optimal. But I don't mind it as it all flies down my screen - it's at least all in my buffer. Thanks in advance for your help, Brian

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Re: [OpenSER-Users] MSILO back notify not working
by Iñaki Baz Castillo 04 Feb '08

04 Feb '08

On Monday 04 February 2008 18:27:28 Dan-Cristian Bogos wrote: > Hi Iñaki, > > in order to send message confirmation, you need to instruct openser to do > so, it will not do automatically this for you. > > In the documentation you have some sample config how to do it ( > http://www.openser.org/docs/modules/1.3.x/msilo.html#AEN390): > > # MSILO - storing as offline message > if (m_store("$ru")) > { > log("MSILO: offline message stored\n"); > if (!t_reply("202", "Accepted")) > { > sl_reply_error(); > }; > }else{ > log("MSILO: offline message NOT stored\n"); > if (!t_reply("503", "Service Unavailable")) > { > sl_reply_error(); > }; > }; > break; > }; Hi Dan, I've read the fully MSILO module documentation, and my openser script implements the code above. But it does not send any notification MESSAGE to the sender. In fact I don't see in the code you paste where the notification MESSAGE is generated by OpenSer. As I said, message storage works ok for me, as well the message retrieving when destination user registers, but there is not notification MESSAGE (and I've set all the required parameters). Any suggestion? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] XMPP GW Error "invalid content-type 0x100ff"
by Ali Jawad 04 Feb '08

04 Feb '08

Hi All I have the following setup X-Lite Client -> Openser With XMPP GW Module <-> ejabberD on the same server <- exodus XMPP client I have enabled presence on the Openser server this allowed me to chat using the Openser server. The next step was to use the XMPP GW module to enable XMPP features using an ejabberd server. I changed the configuration as shown below and logged in using test2 on the exodus XMPP client. I added test2 to the x-lite client and chose send instant message, the x-lite client immediately returned "Error: Your contact is not online." The Openser logs return the following: Feb 4 17:05:52 [11929] DBG:xmpp:cmd_send_message: cmd_send_message Feb 4 17:05:52 [11929] DBG:core:parse_headers: flags=ffffffffffffffff Feb 4 17:05:52 [11929] ERROR:xmpp:cmd_send_message: invalid content-type 0x100ff My Config is below: http://pastebin.com/m7707753a

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Re: [OpenSER-Users] Mediaproxy error no attribute 'autocommit' RESOLVED
by Andy Smith 04 Feb '08

04 Feb '08

Ok, simple fix... Updated to MySQL_python-1.2.2 and all ok now! :D ----- Original Message ----- From: Andy Smith To: users(a)lists.openser.org Sent: Monday, February 04, 2008 4:01 PM Subject: Mediaproxy error no attribute 'autocommit' Hi there, Im trying to run a Mediaproxy and Im recieving this error, Feb 4 14:53:14 server proxydispatcher[22800]: 'Connection' object has no attribute 'autocommit' I´ve had a google for this and its seems like it may be a bug python-sqlite or sqlite bug , but also I tried searching for it on the python issue traker and didnt find anything. Is anyone else aware of this issue and any solution? Im running: Python 2.3.4 MySQL-python.i386 1.0.0-1.RHEL4.1 python-sqlite.i386 1.1.7-1.2.1 sqlite.i386 3.3.6-2 mysql server 5.1.22-rc thanks for any help! Andy.

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[OpenSER-Users] Problem with OpenSER when MySQL database fails
by Arturo Díaz Almagro 04 Feb '08

04 Feb '08

Hello all, Hope anybody could help me. We have a SIP telephony system based on OpenSER with a redundant MySQL database ring. When the "master" database dies, a monit process detect it and modify properly the database entry in the DNS zone file. Two tests can be perform to check the availability of the system when MySQL crashes: - kill mysql process in the database host machine. In this case OpenSER asks the DNS inmediately for the database IP address and the service keeps going as expected. No problem here. - shutdown the interface of the machine where mysql is hosted. In this case OpenSER does not ask the DNS for the new address. It keeps trying over and over the connection with the master and until the interface is not turn on the system keeps hung. So, if the mysql's host interface is shutdown the system hangs but if the mysql process dies the system works properly. In the first case both machines (OpenSER and MySQL) has IP visibility, not in the second, so the problem should be related to that. Has anybody experienced this behaviour? Does anybody know how to solve it? Thanks a lot. -- Arturo Díaz Contact me on FWD: 870436 Skype: arturo.diaz.almagro

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[OpenSER-Users] MSILO back notify not working
by Iñaki Baz Castillo 04 Feb '08

04 Feb '08

Hi, I've configured MSILO module. It works well except the fact that I don't get any notification back to the sender about the storage of the message sent by him. This is: when user_A sends a MESSAGE to user_B and this is stored by MSILO module I don't see any MESSAGE from OpenSer to user_A notifing it. I have set: ------------------------------ # -- msilo -- modparam("msilo", "registrar", "sip:user-offline@mydomain.org") modparam("msilo", "reminder", "sip:reminder@mydomain.org") modparam("msilo", "expire_time", 86400) modparam("msilo", "check_time", 30) modparam("msilo", "send_time", 60) modparam("msilo", "clean_period", 5) modparam("msilo", "use_contact", 1) modparam("msilo", "add_date", 1) modparam("msilo", "max_messages", 20) ------------------------------ As I read in the doc: ------------------------------ 1.3.3. registrar (string) The SIP address used to inform users that destination of their message is not online and the message will be delivered next time when that user goes online. If the parameter is not set, the module will not send any notification. ------------------------------ But I've set that parameter, but when a message is stored I don't see any MESSAGE originated by OpenSer to notify the sender. Shoul I see a MESSAGE from OpenSer? I don't understant too what "reminder" parameter means: ------------------------------ 1.3.4. reminder (string) The SIP address used to send reminder messages. If this value is not set, the reminder feature is disabled. ------------------------------ Any explanation please? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] ERROR:mysql:db_mysql_convert_row: no private memory
by Iñaki Baz Castillo 04 Feb '08

04 Feb '08

Hi, why could this error occur? It has ocurred me now several times but never seen it before ¿? Feb 4 15:31:31 [27228] ERROR:mysql:db_mysql_convert_row: no private memory left Feb 4 15:31:31 [27228] ERROR:mysql:db_mysql_convert_rows: error while converting row #1 Feb 4 15:31:31 [27228] ERROR:mysql:db_mysql_convert_result: error while converting rows Feb 4 15:31:31 [27228] ERROR:mysql:db_mysql_store_result: error while converting result Feb 4 15:31:31 [27228] ERROR:core:db_do_raw_query: error while storing resultFeb 4 15:31:31 [27228] ERROR:usrloc:get_all_db_ucontacts: raw_query failed I will investigate it but any comment will be welcome. Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Mediaproxy error no attribute 'autocommit'
by Andy Smith 04 Feb '08

04 Feb '08

Hi there, Im trying to run a Mediaproxy and Im recieving this error, Feb 4 14:53:14 server proxydispatcher[22800]: 'Connection' object has no attribute 'autocommit' I´ve had a google for this and its seems like it may be a bug python-sqlite or sqlite bug , but also I tried searching for it on the python issue traker and didnt find anything. Is anyone else aware of this issue and any solution? Im running: Python 2.3.4 MySQL-python.i386 1.0.0-1.RHEL4.1 python-sqlite.i386 1.1.7-1.2.1 sqlite.i386 3.3.6-2 mysql server 5.1.22-rc thanks for any help! Andy.

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[OpenSER-Users] Call transfer
by Jan ONDREJ (SAL) 04 Feb '08

04 Feb '08

Hello, can somebody give me a working example of configuration file, supporting "Call transfer" or proper "REFER" passing? Call transfer does not work with default openser.cfg and also with my extended configuration or sipwise generated configuration. My phones supports call transfer and it works well with asterisk, but no success with openser. Can somebody give me an example config or hints, what to do to make a working call transfer? Thank you. SAL

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Re: [OpenSER-Users] CDRTool rating on test system (without PSTN) zero cost problem
by A.smith 04 Feb '08

04 Feb '08

Ok, Rating engine seems to be working ok when I have a canonicalURI starting "00" and in the database I see this for a rated call: Span: 1 Duration: 75 s StartTime: 2008-02-02 12:33:15 (UK) ProfileId: 442 for weekend RateId: 442 for 0-24h Rate: 0.0200 / 60 s Price: 0.0250 However via the web interface in the CDR Section for the cost it is displayed as "x.xxx" and no data is displayed in the detail section "Rating Information" either... Any ideas what the issue is? thanks Andy. ________________________________________________ Message sent using UK Grid Webmail 2.7.9

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Re: [OpenSER-Users] CDRTool rating on test system (without PSTN) zero cost problem
by A.smith 04 Feb '08

04 Feb '08

Hey Dan! Yep ur right! I had to have a leading "00" on the canonicalURI for to rate the call! Thanks alot for the info! cheers Andy! :) --------- Original Message -------- From: Dan-Cristian Bogos <dan.bogos(a)gmail.com> To: Andy Smith <a.smith(a)ukgrid.net> Cc: users(a)lists.openser.org Subject: Re: [OpenSER-Users] CDRTool rating on test system (without PSTN) zero cost problem Date: 04/02/08 11:28 > Hi Andy, > > I suspect that your problem is related to > var $intAccessCode = '00'; inside cdrlib.phtml. > Based on that all the numbers which should not be altered and be > considered billable should contain "00" as international prefix. > > Have a look on that, perhaps it helps. > > Cheers, > DanB > > On Feb 3, 2008 7:17 PM, Andy Smith <a.smith(a)ukgrid.net> wrote: > > > > > > Hi all, > > > > im trying to "trick" CDRTool into rating calls when my OpenSER isnt actually > > routing anything to a PSTN, Im trying just putting the handset username as > > an internation number without the leading zera (ie 44208888888) but CDRTool > > is rating ever call as zero cost. > > > > Maybe its something to do with this, from the CDRTool rating document: > > > > "The rating logic checks if the TLD exists in the billing_enum_tlds table. > > If it does, the rating engine tries to match the regexp field against the > > username part of the destination from the ENUM response, which has been > > saved in the Canonical URI. The match must return a fully qualified E164 > > number otherwise the call is considered to have ended to a no E164 > > destination and the call will be free of charge." > > > > What constitutes a fully qualified E164 number? 44208888888(a)mydomain.com > > should be ok shouldnt it? I also tried > > +44208888888(a)mydomain.com also without any success... > > > > > > Its quite furstrating cos I cant see the logic as its applied by the rating > > system which would explain how it reaches the zero cost > > result. It is certainly looking at the calls as they are made as its logging > > info to the syslog saying so, eg: > > > > Feb 1 16:11:39 server CDRTool[22869]: ShowPrice Id=94 Timestamp=1201882245 > > Duration=6 inputTraffic=0 outputTraffic=0 From=003(a)domain.net > > To=44208888888(a)domain.net Gateway=x.x.x.x Application=audio ENUMtld= > > Feb 1 16:11:39 server CDRTool[10751]: Normalization done in 0 s, memory > > usage: 4.96 MB > > Feb 1 16:11:39 server CDRTool[10751]: Unlock ser_radius:radacct > > > > > > Any ideas people?? Mediaproxy isnt actually required for billing is it? I > > understood it is optional for the documentation...?? > > > > Thanks for any help! Andy. > > > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > ________________________________________________ Message sent using UK Grid Webmail 2.7.9

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[OpenSER-Users] CDRTool rating on test system (without PSTN) zero cost problem
by Andy Smith 04 Feb '08

04 Feb '08

Hi all, im trying to "trick" CDRTool into rating calls when my OpenSER isnt actually routing anything to a PSTN, Im trying just putting the handset username as an internation number without the leading zera (ie 44208888888) but CDRTool is rating ever call as zero cost. Maybe its something to do with this, from the CDRTool rating document: "The rating logic checks if the TLD exists in the billing_enum_tlds table. If it does, the rating engine tries to match the regexp field against the username part of the destination from the ENUM response, which has been saved in the Canonical URI. The match must return a fully qualified E164 number otherwise the call is considered to have ended to a no E164 destination and the call will be free of charge." What constitutes a fully qualified E164 number? 44208888888(a)mydomain.com should be ok shouldnt it? I also tried +44208888888(a)mydomain.com also without any success... Its quite furstrating cos I cant see the logic as its applied by the rating system which would explain how it reaches the zero cost result. It is certainly looking at the calls as they are made as its logging info to the syslog saying so, eg: Feb 1 16:11:39 server CDRTool[22869]: ShowPrice Id=94 Timestamp=1201882245 Duration=6 inputTraffic=0 outputTraffic=0 From=003(a)domain.net To=44208888888(a)domain.net Gateway=x.x.x.x Application=audio ENUMtld= Feb 1 16:11:39 server CDRTool[10751]: Normalization done in 0 s, memory usage: 4.96 MB Feb 1 16:11:39 server CDRTool[10751]: Unlock ser_radius:radacct Any ideas people?? Mediaproxy isnt actually required for billing is it? I understood it is optional for the documentation...?? Thanks for any help! Andy.

2 1

[OpenSER-Users] Recommended distribution for OpenXCAP
by Sergio Gutierrez 04 Feb '08

04 Feb '08

Hello everybody. What is the linux distribution you recommend to run OpenXCAP Server? What distribution allows an easy installation, and offers better performance? Thanks in advance for your answer. Best regards. Sergio Gutiérrez

2 1

[OpenSER-Users] MSRP Relay release 0.9.1
by Adrian Georgescu 04 Feb '08

04 Feb '08

Hello, AG Projects launches MSRPRelay, an Open Source implementation of IETF standard RFC 4976. MSRP relay is an extension to the MSRP protocol (RFC 4975). Its main role is to help NAT traversal of Interactive Messaging and file transfer sessions for SIP/MSRP endpoints located behind NAT. Background ----------- A series of related instant messages between two or more parties can be viewed as part of a "message session", that is, a conversational exchange of messages with a definite beginning and end. This is in contrast to individual messages each sent independently. Messaging schemes that track only individual messages can be described as "page-mode" messaging, whereas messaging that is part of a "session" with a definite start and end is called "session-mode" messaging. Page-mode messaging is enabled in SIP via the SIP MESSAGE method, as defined in RFC 3428. Session-mode messaging has a number of benefits over page-mode messaging, however, such as explicit rendezvous, tighter integration with other media-types, direct client-to-client operation, and brokered privacy and security. Message Session Relay Protocol (MSRP) is a protocol for transmitting a series of related instant messages in the context of a session. Message sessions are treated like any other media stream when set up via a rendezvous or session creation protocol such as the Session Initiation Protocol (SIP). MSRP Sessions are defined in RFC 4975. Out of the box, the software is designed to work in combinations with the subscriber database used by OpenSER (see http://openser.org). For more information visit http://msrprelay.org Regards, Adrian

2 1

[OpenSER-Users] Openser problem in logs
by Ali Jawad 04 Feb '08

04 Feb '08

Hi All When I start openser using /usr/local/sbin/openser it loads then I login using one user and I keep gettingthe logs below all the time is this normal Feb 1 14:03:54 [29077] DBG:core:parse_headers: flags=78 Feb 1 14:03:54 [29077] DBG:tm:t_lookup_request: start searching: hash=50458, isACK=0 Feb 1 14:03:54 [29077] DBG:tm:matching_3261: RFC3261 transaction matched, tid=-d87543-5c484340d1308708-1--d87543- Feb 1 14:03:54 [29077] DBG:tm:t_lookup_request: REF_UNSAFE: after is 1 Feb 1 14:03:54 [29077] DBG:tm:t_lookup_request: transaction found (T=0x4069bb80) Feb 1 14:03:54 [29077] DBG:tm:t_retransmit_reply: nothing to retransmit Feb 1 14:03:54 [29077] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Feb 1 14:03:54 [29077] DBG:core:destroy_avp_list: destroying list (nil) Feb 1 14:03:54 [29077] DBG:core:receive_msg: cleaning up Feb 1 14:03:54 [29078] DBG:tm:utimer_routine: timer routine:7,tl=0x4069bccc next=(nil), timeout=12700000 Feb 1 14:03:54 [29078] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x4069bb80, SUBSCRIBE ... ) Feb 1 14:03:54 [29078] DBG:tm:set_timer: relative timeout is 4000000 Feb 1 14:03:54 [29078] DBG:tm:insert_timer_unsafe: [7]: 0x4069bccc (16700000) Feb 1 14:03:54 [29078] DBG:tm:retransmission_handler: retransmission_handler : done Feb 1 14:03:58 [29075] DBG:core:parse_msg: SIP Request: Feb 1 14:03:58 [29075] DBG:core:parse_msg: method: <SUBSCRIBE> Feb 1 14:03:58 [29075] DBG:core:parse_msg: uri: <sip:ali@jabber.splendor.net> Feb 1 14:03:58 [29075] DBG:core:parse_msg: version: <SIP/2.0> Feb 1 14:03:58 [29075] DBG:core:parse_headers: flags=2 Feb 1 14:03:58 [29075] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d87543-5c484340d1308708-1--d87543->; state=6 Feb 1 14:03:58 [29075] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Feb 1 14:03:58 [29075] DBG:core:parse_via: end of header reached, state=5 Feb 1 14:03:58 [29075] DBG:core:parse_headers: via found, flags=2 Feb 1 14:03:58 [29075] DBG:core:parse_headers: this is the first via Feb 1 14:03:58 [29075] DBG:core:receive_msg: After parse_msg... Feb 1 14:03:58 [29075] DBG:core:receive_msg: preparing to run routing scripts... Feb 1 14:03:58 [29075] DBG:core:parse_headers: flags=100 Feb 1 14:03:58 [29075] DBG:maxfwd:is_maxfwd_present: value = 70 Feb 1 14:03:58 [29075] DBG:core:parse_headers: flags=200 Feb 1 14:03:58 [29075] DBG:core:parse_to: end of header reached, state=10 Feb 1 14:03:58 [29075] DBG:core:parse_to: display={"ssafass"}, ruri={sip:ali@jabber.splendor.net} Feb 1 14:03:58 [29075] DBG:core:get_hdr_field: <To> [40]; uri=[sip:ali@jabber.splendor.net] Feb 1 14:03:58 [29075] DBG:core:get_hdr_field: to body ["ssafass"<sip:ali@jabber.splendor.net> ] Feb 1 14:03:58 [29075] DBG:core:get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> Feb 1 14:03:58 [29075] DBG:core:get_hdr_field: content_length=0 Feb 1 14:03:58 [29075] DBG:core:get_hdr_field: found end of header Feb 1 14:03:58 [29075] DBG:rr:find_first_route: No Route headers found Feb 1 14:03:58 [29075] DBG:rr:loose_route: There is no Route HF Feb 1 14:03:58 [29075] DBG:core:grep_sock_info: checking if host==us: 19==15 && [jabber.splendor.net] == [193.227.186.153] Feb 1 14:03:58 [29075] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Feb 1 14:03:58 [29075] DBG:uri_db:does_uri_exist: User in request uri does exist Feb 1 14:03:58 [29075] DBG:core:db_free_rows: freeing 1 rows Feb 1 14:03:58 [29075] DBG:core:db_free_rows: row[0]=0x81841e0 Feb 1 14:03:58 [29075] DBG:core:db_free_rows: 0x81841e0=pkg_free() RES_ROWS Feb 1 14:03:58 [29075] DBG:core:rewrite_uri: rewriting Request-URI with 'sip:ali@193.227.186.146:6242;rinstance=63a8660ec1b35169' Feb 1 14:03:58 [29075] DBG:tm:t_newtran: transaction on entrance=0xffffffff Feb 1 14:03:58 [29075] DBG:core:parse_headers: flags=ffffffffffffffff Feb 1 14:03:58 [29075] DBG:core:parse_headers: flags=78 Feb 1 14:03:58 [29075] DBG:tm:t_lookup_request: start searching: hash=50458, isACK=0 Feb 1 14:03:58 [29075] DBG:tm:matching_3261: RFC3261 transaction matched, tid=-d87543-5c484340d1308708-1--d87543- Feb 1 14:03:58 [29075] DBG:tm:t_lookup_request: REF_UNSAFE: after is 1 Feb 1 14:03:58 [29075] DBG:tm:t_lookup_request: transaction found (T=0x4069bb80) Feb 1 14:03:58 [29075] DBG:tm:t_retransmit_reply: nothing to retransmit Feb 1 14:03:58 [29075] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Feb 1 14:03:58 [29075] DBG:core:destroy_avp_list: destroying list (nil) Feb 1 14:03:58 [29075] DBG:core:receive_msg: cleaning up Feb 1 14:03:58 [29078] DBG:tm:utimer_routine: timer routine:7,tl=0x4069bccc next=(nil), timeout=16700000 Feb 1 14:03:58 [29078] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x4069bb80, SUBSCRIBE ... ) Feb 1 14:03:58 [29078] DBG:tm:set_timer: relative timeout is 4000000 Feb 1 14:03:58 [29078] DBG:tm:insert_timer_unsafe: [7]: 0x4069bccc (20700000) Feb 1 14:03:58 [29078] DBG:tm:retransmission_handler: retransmission_handler : done Feb 1 14:04:02 [29077] DBG:core:parse_msg: SIP Request: Feb 1 14:04:02 [29077] DBG:core:parse_msg: method: <SUBSCRIBE> Feb 1 14:04:02 [29077] DBG:core:parse_msg: uri: <sip:ali@jabber.splendor.net> Feb 1 14:04:02 [29077] DBG:core:parse_msg: version: <SIP/2.0> Feb 1 14:04:02 [29077] DBG:core:parse_headers: flags=2 Feb 1 14:04:02 [29077] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d87543-5c484340d1308708-1--d87543->; state=6 Feb 1 14:04:02 [29077] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Feb 1 14:04:02 [29077] DBG:core:parse_via: end of header reached, state=5 Feb 1 14:04:02 [29077] DBG:core:parse_headers: via found, flags=2 Feb 1 14:04:02 [29077] DBG:core:parse_headers: this is the first via Feb 1 14:04:02 [29077] DBG:core:receive_msg: After parse_msg... Feb 1 14:04:02 [29077] DBG:core:receive_msg: preparing to run routing scripts... Feb 1 14:04:02 [29077] DBG:core:parse_headers: flags=100 Feb 1 14:04:02 [29077] DBG:maxfwd:is_maxfwd_present: value = 70 Feb 1 14:04:02 [29077] DBG:core:parse_headers: flags=200 Feb 1 14:04:02 [29077] DBG:core:parse_to: end of header reached, state=10 Feb 1 14:04:02 [29077] DBG:core:parse_to: display={"ssafass"}, ruri={sip:ali@jabber.splendor.net} Feb 1 14:04:02 [29077] DBG:core:get_hdr_field: <To> [40]; uri=[sip:ali@jabber.splendor.net] Feb 1 14:04:02 [29077] DBG:core:get_hdr_field: to body ["ssafass"<sip:ali@jabber.splendor.net> ] Feb 1 14:04:02 [29077] DBG:core:get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> Feb 1 14:04:02 [29077] DBG:core:get_hdr_field: content_length=0 Feb 1 14:04:02 [29077] DBG:core:get_hdr_field: found end of header Feb 1 14:04:02 [29077] DBG:rr:find_first_route: No Route headers found Feb 1 14:04:02 [29077] DBG:rr:loose_route: There is no Route HF Feb 1 14:04:02 [29077] DBG:core:grep_sock_info: checking if host==us: 19==15 && [jabber.splendor.net] == [193.227.186.153] Feb 1 14:04:02 [29077] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Feb 1 14:04:02 [29077] DBG:uri_db:does_uri_exist: User in request uri does exist Feb 1 14:04:02 [29077] DBG:core:db_free_rows: freeing 1 rows Feb 1 14:04:02 [29077] DBG:core:db_free_rows: row[0]=0x817fca0 Feb 1 14:04:02 [29077] DBG:core:db_free_rows: 0x817fca0=pkg_free() RES_ROWS Feb 1 14:04:02 [29077] DBG:core:rewrite_uri: rewriting Request-URI with 'sip:ali@193.227.186.146:6242;rinstance=63a8660ec1b35169' Feb 1 14:04:02 [29077] DBG:tm:t_newtran: transaction on entrance=0xffffffff Feb 1 14:04:02 [29077] DBG:core:parse_headers: flags=ffffffffffffffff Feb 1 14:04:02 [29077] DBG:core:parse_headers: flags=78 Feb 1 14:04:02 [29077] DBG:tm:t_lookup_request: start searching: hash=50458, isACK=0 Feb 1 14:04:02 [29077] DBG:tm:matching_3261: RFC3261 transaction matched, tid=-d87543-5c484340d1308708-1--d87543- Feb 1 14:04:02 [29077] DBG:tm:t_lookup_request: REF_UNSAFE: after is 1 Feb 1 14:04:02 [29077] DBG:tm:t_lookup_request: transaction found (T=0x4069bb80) Feb 1 14:04:02 [29077] DBG:tm:t_retransmit_reply: nothing to retransmit Feb 1 14:04:02 [29077] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Feb 1 14:04:02 [29077] DBG:core:destroy_avp_list: destroying list (nil) Feb 1 14:04:02 [29077] DBG:core:receive_msg: cleaning up Feb 1 14:04:02 [29078] DBG:tm:utimer_routine: timer routine:7,tl=0x4069bccc next=(nil), timeout=20700000 Feb 1 14:04:02 [29078] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x4069bb80, SUBSCRIBE ... ) Feb 1 14:04:02 [29078] DBG:tm:set_timer: relative timeout is 4000000 Feb 1 14:04:02 [29078] DBG:tm:insert_timer_unsafe: [7]: 0x4069bccc (24700000) Feb 1 14:04:02 [29078] DBG:tm:retransmission_handler: retransmission_handler : done Feb 1 14:04:06 [29075] DBG:core:parse_msg: SIP Request: Feb 1 14:04:06 [29075] DBG:core:parse_msg: method: <SUBSCRIBE> Feb 1 14:04:06 [29075] DBG:core:parse_msg: uri: <sip:ali@jabber.splendor.net> Feb 1 14:04:06 [29075] DBG:core:parse_msg: version: <SIP/2.0> Feb 1 14:04:06 [29075] DBG:core:parse_headers: flags=2 Feb 1 14:04:06 [29075] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d87543-5c484340d1308708-1--d87543->; state=6 Feb 1 14:04:06 [29075] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Feb 1 14:04:06 [29075] DBG:core:parse_via: end of header reached, state=5 Feb 1 14:04:06 [29075] DBG:core:parse_headers: via found, flags=2 Feb 1 14:04:06 [29075] DBG:core:parse_headers: this is the first via Feb 1 14:04:06 [29075] DBG:core:receive_msg: After parse_msg... Feb 1 14:04:06 [29075] DBG:core:receive_msg: preparing to run routing scripts... Feb 1 14:04:06 [29075] DBG:core:parse_headers: flags=100 Feb 1 14:04:06 [29075] DBG:maxfwd:is_maxfwd_present: value = 70 Feb 1 14:04:06 [29075] DBG:core:parse_headers: flags=200 Feb 1 14:04:06 [29075] DBG:core:parse_to: end of header reached, state=10 Feb 1 14:04:06 [29075] DBG:core:parse_to: display={"ssafass"}, ruri={sip:ali@jabber.splendor.net} Feb 1 14:04:06 [29075] DBG:core:get_hdr_field: <To> [40]; uri=[sip:ali@jabber.splendor.net] Feb 1 14:04:06 [29075] DBG:core:get_hdr_field: to body ["ssafass"<sip:ali@jabber.splendor.net> ] Feb 1 14:04:06 [29075] DBG:core:get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> Feb 1 14:04:06 [29075] DBG:core:get_hdr_field: content_length=0 Feb 1 14:04:06 [29075] DBG:core:get_hdr_field: found end of header Feb 1 14:04:06 [29075] DBG:rr:find_first_route: No Route headers found Feb 1 14:04:06 [29075] DBG:rr:loose_route: There is no Route HF Feb 1 14:04:06 [29075] DBG:core:grep_sock_info: checking if host==us: 19==15 && [jabber.splendor.net] == [193.227.186.153] Feb 1 14:04:06 [29075] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Feb 1 14:04:06 [29075] DBG:uri_db:does_uri_exist: User in request uri does exist Feb 1 14:04:06 [29075] DBG:core:db_free_rows: freeing 1 rows Feb 1 14:04:06 [29075] DBG:core:db_free_rows: row[0]=0x817fcc8 Feb 1 14:04:06 [29075] DBG:core:db_free_rows: 0x817fcc8=pkg_free() RES_ROWS Feb 1 14:04:06 [29075] DBG:core:rewrite_uri: rewriting Request-URI with 'sip:ali@193.227.186.146:6242;rinstance=63a8660ec1b35169' Feb 1 14:04:06 [29075] DBG:tm:t_newtran: transaction on entrance=0xffffffff Feb 1 14:04:06 [29075] DBG:core:parse_headers: flags=ffffffffffffffff Feb 1 14:04:06 [29075] DBG:core:parse_headers: flags=78 Feb 1 14:04:06 [29075] DBG:tm:t_lookup_request: start searching: hash=50458, isACK=0 Feb 1 14:04:06 [29075] DBG:tm:matching_3261: RFC3261 transaction matched, tid=-d87543-5c484340d1308708-1--d87543- Feb 1 14:04:06 [29075] DBG:tm:t_lookup_request: REF_UNSAFE: after is 1 Feb 1 14:04:06 [29075] DBG:tm:t_lookup_request: transaction found (T=0x4069bb80) Feb 1 14:04:06 [29075] DBG:tm:t_retransmit_reply: nothing to retransmit Feb 1 14:04:06 [29075] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Feb 1 14:04:06 [29075] DBG:core:destroy_avp_list: destroying list (nil) Feb 1 14:04:06 [29075] DBG:core:receive_msg: cleaning up Feb 1 14:04:06 [29078] DBG:tm:utimer_routine: timer routine:7,tl=0x4069bccc next=(nil), timeout=24700000 Feb 1 14:04:06 [29078] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x4069bb80, SUBSCRIBE ... ) Feb 1 14:04:06 [29078] DBG:tm:set_timer: relative timeout is 4000000 Feb 1 14:04:06 [29078] DBG:tm:insert_timer_unsafe: [7]: 0x4069bccc (28700000) Feb 1 14:04:06 [29078] DBG:tm:retransmission_handler: retransmission_handler : done Feb 1 14:04:10 [29077] DBG:core:parse_msg: SIP Request: Feb 1 14:04:10 [29077] DBG:core:parse_msg: method: <SUBSCRIBE> Feb 1 14:04:10 [29077] DBG:core:parse_msg: uri: <sip:ali@jabber.splendor.net> Feb 1 14:04:10 [29077] DBG:core:parse_msg: version: <SIP/2.0> Feb 1 14:04:10 [29077] DBG:core:parse_headers: flags=2 Feb 1 14:04:10 [29077] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d87543-5c484340d1308708-1--d87543->; state=6 Feb 1 14:04:10 [29077] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Feb 1 14:04:10 [29077] DBG:core:parse_via: end of header reached, state=5 Feb 1 14:04:10 [29077] DBG:core:parse_headers: via found, flags=2 Feb 1 14:04:10 [29077] DBG:core:parse_headers: this is the first via Feb 1 14:04:10 [29077] DBG:core:receive_msg: After parse_msg... Feb 1 14:04:10 [29077] DBG:core:receive_msg: preparing to run routing scripts... Feb 1 14:04:10 [29077] DBG:core:parse_headers: flags=100 Feb 1 14:04:10 [29077] DBG:maxfwd:is_maxfwd_present: value = 70 Feb 1 14:04:10 [29077] DBG:core:parse_headers: flags=200 Feb 1 14:04:10 [29077] DBG:core:parse_to: end of header reached, state=10 Feb 1 14:04:10 [29077] DBG:core:parse_to: display={"ssafass"}, ruri={sip:ali@jabber.splendor.net} Feb 1 14:04:10 [29077] DBG:core:get_hdr_field: <To> [40]; uri=[sip:ali@jabber.splendor.net] Feb 1 14:04:10 [29077] DBG:core:get_hdr_field: to body ["ssafass"<sip:ali@jabber.splendor.net> ] Feb 1 14:04:10 [29077] DBG:core:get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> Feb 1 14:04:10 [29077] DBG:core:get_hdr_field: content_length=0 Feb 1 14:04:10 [29077] DBG:core:get_hdr_field: found end of header Feb 1 14:04:10 [29077] DBG:rr:find_first_route: No Route headers found Feb 1 14:04:10 [29077] DBG:rr:loose_route: There is no Route HF Feb 1 14:04:10 [29077] DBG:core:grep_sock_info: checking if host==us: 19==15 && [jabber.splendor.net] == [193.227.186.153] Feb 1 14:04:10 [29077] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Feb 1 14:04:10 [29077] DBG:uri_db:does_uri_exist: User in request uri does exist Feb 1 14:04:10 [29077] DBG:core:db_free_rows: freeing 1 rows Feb 1 14:04:10 [29077] DBG:core:db_free_rows: row[0]=0x8179d78 Feb 1 14:04:10 [29077] DBG:core:db_free_rows: 0x8179d78=pkg_free() RES_ROWS Feb 1 14:04:10 [29077] DBG:core:rewrite_uri: rewriting Request-URI with 'sip:ali@193.227.186.146:6242;rinstance=63a8660ec1b35169' Feb 1 14:04:10 [29077] DBG:tm:t_newtran: transaction on entrance=0xffffffff Feb 1 14:04:10 [29077] DBG:core:parse_headers: flags=ffffffffffffffff Feb 1 14:04:10 [29077] DBG:core:parse_headers: flags=78 Feb 1 14:04:10 [29077] DBG:tm:t_lookup_request: start searching: hash=50458, isACK=0 Feb 1 14:04:10 [29077] DBG:tm:matching_3261: RFC3261 transaction matched, tid=-d87543-5c484340d1308708-1--d87543- Feb 1 14:04:10 [29077] DBG:tm:t_lookup_request: REF_UNSAFE: after is 1 Feb 1 14:04:10 [29077] DBG:tm:t_lookup_request: transaction found (T=0x4069bb80) Feb 1 14:04:10 [29077] DBG:tm:t_retransmit_reply: nothing to retransmit Feb 1 14:04:10 [29077] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Feb 1 14:04:10 [29077] DBG:core:destroy_avp_list: destroying list (nil) Feb 1 14:04:10 [29077] DBG:core:receive_msg: cleaning up Feb 1 14:04:10 [29078] DBG:tm:utimer_routine: timer routine:7,tl=0x4069bccc next=(nil), timeout=28700000 Feb 1 14:04:10 [29078] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x4069bb80, SUBSCRIBE ... ) Feb 1 14:04:10 [29078] DBG:tm:set_timer: relative timeout is 4000000 Feb 1 14:04:10 [29078] DBG:tm:insert_timer_unsafe: [7]: 0x4069bccc (32700000) Feb 1 14:04:10 [29078] DBG:tm:retransmission_handler: retransmission_handler : done Feb 1 14:04:11 [29075] DBG:core:udp_rcv_loop: probing packet received from 193.227.186.146 11518 Feb 1 14:04:12 [29078] DBG:tm:timer_routine: timer routine:0,tl=0x4069bce8 next=0x4069bdfc, timeout=31 Feb 1 14:04:12 [29078] DBG:tm:final_response_handler: stop retr. and send CANCEL (0x4069bb80) Feb 1 14:04:12 [29078] DBG:tm:t_should_relay_response: T_code=0, new_code=408 Feb 1 14:04:12 [29078] DBG:tm:t_pick_branch: picked branch 0, code 408 Feb 1 14:04:12 [29078] DBG:tm:is_3263_failure: dns-failover test: branch=0, last_recv=408, flags=0 Feb 1 14:04:12 [29078] DBG:tm:t_should_relay_response: trying DNS-based failover Feb 1 14:04:12 [29078] DBG:tm:relay_reply: branch=0, save=0, relay=0 Feb 1 14:04:12 [29078] DBG:core:parse_headers: flags=ffffffffffffffff Feb 1 14:04:12 [29078] DBG:core:check_via_address: params 193.227.186.146, 192.168.0.176, 0 Feb 1 14:04:12 [29078] DBG:core:_shm_resize: resize(0) called Feb 1 14:04:12 [29078] DBG:tm:insert_timer_unsafe: [2]: 0x4069bbc8 (36) Feb 1 14:04:12 [29078] DBG:tm:relay_reply: sent buf=0x8179e70: SIP/2.0 4..., shmem=0x40699c70: SIP/2.0 4 Feb 1 14:04:12 [29078] DBG:tm:final_response_handler: done Feb 1 14:04:12 [29078] DBG:tm:timer_routine: timer routine:0,tl=0x4069bdfc next=(nil), timeout=31

2 1

[OpenSER-Users] anonymous LDAP bind issue
by antalsia＠free.fr 04 Feb '08

04 Feb '08

Hi, I'm trying to implement LDAP authentication with anonymous LDAP bind. I set the ldap configuration file without ldap_bind_dn, ldap_bind_password attributes. This step works fine. Unfortunately, I can't figure out how to set the openser.cfg file. I need to pass the bind DN and the user password to the ldap_search function ; that's ok for the bind DN but I don't know how to procede for the password. Can someone post an example please ? Regards,

2 4

Re: [OpenSER-Users] Mediaproxy Vs RTP Proxy
by Andy Smith 03 Feb '08

03 Feb '08

Hi Ali, there is a good description on the mediaproxy module doc page, near the bottom (nathelper). Should clear things up for you! :P http://openser.org/docs/modules/1.3.x/mediaproxy.html cheers Andy.

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[Serusers] Help needed with SER imprementations
by voip termination 02 Feb '08

02 Feb '08

Hello, I´m new to SER Express and I needed to use SER, on the following imprementations, but do not know which to use. I want to use this imprementations for free in network internet calling, calling card via asterisk (a2billing), and PSTN calls for all the network users. 1. I need to setup SER, to routes all my incoming and outgoing sip calls, (in network users to users) free sip calls 2. Routes all PSTN calls to Asterisk for billing via a2billing and then Asterisk will do the PSTN call terminations of the calls. 3. All the sip user to be authenticated by Asterisk / a2billing for the billing via their user name and password. 4. SER to use asterisk for the users voicemail and other pbx features. Can anyone please help me out? Looking to hearing from you. Regards, Henry -- ================================ "Never Let Go Of A Dream Untill You Can Wakeup And Make It Happens" ================================

1 0

[OpenSER-Users] match src_ip
by Andreas Heise 02 Feb '08

02 Feb '08

Hi all, I search for a simple way to match src_ip from 192.168.1.6 - 192.168.1.11. Any ideas, the following aren't working :-( ... if(src_ip==192.168.1.[6-11]) if(src_ip==192.168.1.[6|7|8|9|10|11]) thanks, Andreas

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[OpenSER-Users] Openser, inbound SIP "trunks" and REFER
by Robby Dermody 01 Feb '08

01 Feb '08

All, I have a situation where I have a OpenSER server frontending a set of Asterisk boxes (call them box A1 and A2). Calls come in and are routed to one of the boxes (after doing some custom HTTP lookups using a custom perl module, rewriting the RURI, etc). This works fine, however at some point, the call will most likely need to be transferred from one internal Asterisk box to the other Asterisk box (e.g. A1 to A2). At this point, what we have set to happen is that A1 would send a REFER up through the proxy, the proxy caches an auth token record from info in the outgoing REFER and writes the token ID into the Refer-To header (along with its own address as the recipient) as the REFER goes out to the SIP trunk, and then when the subsequent invite is made from the sip trunk (to something like route_34983948-34343@<sipProxyIP> with that 34983948-34343 being the token ID) the SIP proxy will look up that auth token, rewrite the RURI with the cached info and direct the call to the appropriate internal asterisk box...A2 in this case. So that all is done and would normally work. However our SIP provider says that it only provides "inbound DID service" and thus do not have their dialplan set up to handle REFERs with custom URIs (even if it is back down to the same openser proxy, as it always is in our case). So looking at this, I have two questions: 1. Could resolving this problem just be as simple as finding a new SIP DID provider? I am a bit worried that providers may consider use of REFERs in any way as "SIP termination" (e.g. outgoing service) and either not office REFER support or charge accordingly for outbound access, even though this is clearly still inbound. 2. Those Asterisk boxes in our environment are behind a NAT, so our openser proxy does employ nathelper/rtpproxy. Given this, another alternative I see is for the openser proxy to intercept the REFER for the DID->A2 message, craft an INVITE for proxy->A2, negotiate the conversation, and reconstruct the internal RTP leg between it and A2 instead of A1. This way, the SIP trunk is none the wiser and does not even know this transfer has happened, it just gets some dead air for a second or two and everything is done behind the scenes. I know that a SIP proxy shouldn't inject new SIP messages into the stream, and the REFER way seems to be the most RFC-compliant and technically sound method. However, there is the real-world threat in my mind that it may not be very well supported among inbound DID providers, so I am interested in the community's opinion on the options. Thanks, Robby

1 0

[OpenSER-Users] Route calls to PSTN Gateway
by Ali Jawad 01 Feb '08

01 Feb '08

Hi All I have successfully setup OpenSer with a mysql backend. I am able to make calls using sip phones between registered users. What I need to do know is routing calls that start with 00 to my PSTN gateway. But first I need to setup OpenSer to route calls to my PSTN ..! I am using OpenSer1.3 and I have checked sipwise.com but honestly I prefer changing my own config file. But from sipwise.com I concluded that I need the config below But are the nathelper and mediaproxy compiled by default or do I have to recompile with special options ? Other than that where do I set the IP of my PSTN gateway ? Any suggestions or help would be welcome thanks. loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 0) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "mediaproxy.so" modparam("mediaproxy", "mediaproxy_socket","/var/run/proxydispatcher.sock") modparam("mediaproxy", "sip_asymmetrics","/etc/openser/sip-asymmetric-clients") modparam("mediaproxy", "rtp_asymmetrics","/etc/openser/rtp-asymmetric-clients") modparam("mediaproxy", "natping_interval", 60) loadmodule "uri.so"

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[OpenSER-Users] Wrong To: Field In SIP Packet when Sending to PSTN GW
by Ali Jawad 01 Feb '08

01 Feb '08

Hi All I am using OpenSER as a proxy to make outbound calls my config is very simple if the number dialled is not an openser account route it to the PSTN gw. if(does_uri_exist()){ # local uri does exist, is probably a user. # lookup location if(lookup("location")){ route(1); return; } } else { # probably a call to pstn.... route(2); return; } and route[2] { # pstn handling, simply route out to pstn. sethostport("xx.xx.xx.xx:5060"); route(1); } The problem is that once the SIP packet arrives at the PSTN GW it does NOT have the correct TO: set. Therefore the call does not get routed . In the example below TO: is sip:calledNumber@myOpenserDomain instead of sip:calledNumber@PSTN.GATEWAY.IP.IP Caller: ali [!at] jabber.splendor.net (replace the [!at] with a @) Callee: 009613041708 OpenSerDomain: jabber.splendor.net U +0.289348 PSTN.GW.IP.IP:5060 -> 193.237.226.252:5060 SIP/2.0 404 Not Found . Via: SIP/2.0/UDP 193.237.226.252;rport;branch=z9hG4bK6828.10c0315.0. Via: SIP/2.0/UDP 192.168.0.176:65068;received=193.227.186.146;branch=z9hG4bK-d87543-be62c 55d821be10d-1--d87543-;rport=65068. Record-Route: <sip:193.237.226.252;lr=on>. From: "ssafass" <sip:ali@jabber.splendor.net>;tag=f36d6608. To: "009613041705" <sip:009613041705@jabber.splendor.net>;tag=GR52RWG346-34. Call-ID: 0942e159a72eab40ZmViZWY4YTVlOTRlOGJmZTM5ZDdkZGJiZjFmMTlmMjk.. CSeq: 1 INVITE. Contact: "0000" <sip:PSTN.GW.IP.IP:5060>. User-Agent: eyeBeam release 1003s stamp 31159. Content-Length: 0. I did a siptrace on the interface of the SIP proxy http://pastebin.com/d56426d63 <http://www.voipuser.org/ship_to.php?url=http://pastebin.com/d56426d63> This is my config: http://pastebin.com/m128ca16e <http://www.voipuser.org/ship_to.php?url=http://pastebin.com/m128ca16e>

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[OpenSER-Users] Not able to connect Application Server to the SEAS module of OpenSER.
by Rajat Dudeja 01 Feb '08

01 Feb '08

Hi Folks, My openser is running on SCTP. I'm using SEAS module. I am using a simple client socket (TCP Socket) program as Application Server(AS) to connect to SEAS module. As is written to connect to port 5080 of server. But it does not connect as is evident from the line (openser' console log): Feb 1 17:30:47 [12031475] INFO:seas:dispatcher_main_loop: polling [2 ServSock] [1 pipe] [0 App Servers] [0 Uncomplete AS] I'm sending a REGISTER message from a SCTP Client to OpenSER on port 5070. I'm expecting the same REGISTER to be passed to the AS. I'm using the following openser.cfg file : ####### Global Parameters ######### #debug=3 #log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=1 dns=no /* uncomment the following lines to enable debugging */ debug=9 #fork=no log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "//etc/openser/tls/user/user-cert.pem" #tls_private_key = "//etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "//etc/openser/tls/user/user-calist.pem" #port=5060 /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ listen=SCTP:157.227.110.27:5070 ####### Modules Section ######## #set module path #mpath="//lib/openser/modules/" mpath="/home/ops/openser/lib/openser/modules/" 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 "uri_db.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" #Rajat-Jiten added 01.02.08 loadmodule "seas.so" # ----------------- setting module-specific parameters --------------- #Rajat-Jiten added 01.02.08 # ----- seas params ----- modparam("seas", "listen_sockets","157.227.110.27:5080") # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/abc/openser_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # ----- usrloc params ----- modparam("usrloc", "db_mode", 0) # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ #modparam("auth_db", "calculate_ha1", yes) #modparam("auth_db", "password_column", "password") #modparam("auth_db", "db_url", # "mysql://openser:openserrw@192.168.1.3/openser_1_3") #modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@192.168.1.3/openser_1_3") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@192.168.1.3/openser_1_3") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #modparam("presence|presence_xml", "db_url", # "mysql://openser:openserrw@192.168.1.3/openser_1_3") #modparam("presence_xml", "force_active", 1) #modparam("presence", "server_address", "sip:192.168.1.2:5060") ####### Routing Logic ######## route{ if(!as_relay_t("app_server")){ t_reply("500","Application Server error"); } } =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

1 0

[OpenSER-Users] openser+opendiameter integration problem----->KINDLY PAY ATTENTION
by bhargav.b＠renovau.com 01 Feb '08

01 Feb '08

hello everybody, i have some problem regarding the integratoion b/w open diameter and openser for clients authorization. i have recently installed openser and opendiameter as needed.both are working fine independently.but for the integration i need somebodies kind help.i have run all the sample examples of both. even my softphone[xlite]is doing fine with openser.can anybody please tell me how can i configure the openser.cfg to support diameter? please also sujjest me regarding avp's creation for the diameter+openser support. thank you very much bhargav.b bhargav.b(a)renovau.com

4 4

[OpenSER-Users] openser+opendiameter integration problem----->KINDLY PAY ATTENTION
by bhargav.b＠renovau.com 01 Feb '08

01 Feb '08

hello everybody, i have some problem regarding the integratoion b/w open diameter and openser. i have recently installed openser and opendiameter as needed.both are working fine independently.but for the integration i need somebodies kind help.i have run all the sample examples of both. even my softphone[xlite]is doing fine with openser.can anybody please tell me how can i configure the openser.cfg to support diameter? please also sujjest me regarding avp's creation for the diameter+openser support. thank you very much bhargav.b bhargav.b(a)renovau.com

1 0

Re: [OpenSER-Users] Reg. Proxy domains
by Padmaja 01 Feb '08

01 Feb '08

Hi.. That is the whole issue,,, The proxy 1 and proxy2 do not know each other. How can I configure openser so that it recognizes proxy1 as another proxy and sends/receives invites? I do not want any of the proxies acting as out bound proxy. Both should be primary proxies for their respective domains and yet they should be able to recognize the other proxy and forward the requests to it. In other words, what I am asking how do the proxies talk to each other? Thanks, Padmaja ----- Original Message ----- From: raviprakash sunkara To: Padmaja Sent: Thursday, January 31, 2008 1:55 PM Subject: Re: [OpenSER-Users] Reg. Proxy domains Hello Padmaja Akka, IS urs Proxy1 sending the INVITE to Proxy2.. Can u trace by using ngrep or trace and Send to u .... On Jan 31, 2008 12:39 PM, Padmaja <padmaja.rv(a)vodcalabs.com> wrote: Hi all, Please see the situation below: UA1 ----------------------> Proxy 1(not openser)---------------------- Proxy 2(openser)<----------------------UA2 UA1 is registered to proxy 1 and UA 2 to openser. I wish to make a call from UA 1 to UA 2. Proxy 1 and proxy 2 are in the LAN but do not know of each other. Is there a way we can configure the openser to send/receive the invites to proxy 1? I think the calls will be handled on domain basis, so openser will treat all the UAs registered to it as local and allow them to call each other. But how to make it allow inter domain calls? Please let me know. Thanks, Padmaja _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users -- Thanks &Regards Ravi Prakash Sunkara VoIP Development Tech Lead

2 1

[OpenSER-Users] XMPP Gateway
by Ali Jawad 31 Jan '08

31 Jan '08

Dear All I am trying to achieve a setup where users can both chat using XMPP and do voice calls through VOIP. I have already setup eJabberD and I like it's features and I am almost done setting up OpenSer. I have read about the XMPP Gateway module ? If I do use this plugin will this synchronize the users created between the two systems if I do create them on the OpenSer server ? Has anyone used the module ? If I do use it will the contact lists for the users be stored on the SIP server or the XMPP server ? What about presence will the XMPP be responsible for the presence changes or the OpenSer server ? Thanks

1 0

[OpenSER-Users] Mediaproxy Vs RTP Proxy
by Ali Jawad 31 Jan '08

31 Jan '08

Dear All I know that there are two possible setups one usng mediaproxy and another one using rtpproxy. How do these two proxies differ from each other ? And if I chose NOT to use any of these proxies would I be able to make calls through a PSTN gateway located on the Internet ? Thanks.

1 0

[OpenSER-Users] Help Sending calls to PSTN
by Ali Jawad 31 Jan '08

31 Jan '08

Hi there my config file is http://pastebin.com/d33669ee8 I have set modparam("nathelper", "ping_nated_only", 1) modparam("registrar", "nat_flag", 6) This gives me the following error: Jan 31 16:47:26 [16380] ERROR:core:set_mod_param_regex: parameter <nat_flag> not found in module <registrar> Jan 31 16:47:26 [16380] CRITICAL:core:yyerror: parse error in config file, line 173, column 19-20: Can't set module parameter If I set modparam("nathelper", "ping_nated_only", 0) and ### modparam("registrar", "nat_flag", 6) This allows me to run openser but when I make a call to PSTN the following warning is sent out Jan 31 16:55:45 [16484] ERROR:nathelper:send_rtpp_command: can't connect to RTP proxy Jan 31 16:55:45 [16484] ERROR:nathelper:send_rtpp_command: proxy </var/run/rtpproxy.sock> does not respond, disable it Jan 31 16:55:45 [16484] WARNING:nathelper:rtpp_test: can't get version of the RTP proxy Jan 31 16:55:45 [16484] WARNING:nathelper:rtpp_test: support for RTP proxy </var/run/rtpproxy.sock> has been disabled temporarily Calls between softphones work but a call to the PSTN ends up with the following Jan 31 16:57:26 [16509] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 International PSTN Jan 31 16:57:26 [16509] DBG:core:parse_headers: flags=80 Jan 31 16:57:26 [16509] DBG:core:parse_headers: flags=ffffffffffffffff Jan 31 16:57:26 [16509] DBG:nathelper:check_content_type: type <application/sdp> found valid Jan 31 16:57:26 [16509] DBG:core:parse_headers: flags=40 ===>Jan 31 16:57:26 [16509] ERROR:nathelper:force_rtp_proxy2_f: no available proxies Routed to PSTN gateway Jan 31 16:57:26 [16509] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Jan 31 16:57:26 [16509] DBG:core:destroy_avp_list: destroying list (nil) Jan 31 16:57:26 [16509] DBG:core:receive_msg: cleaning up Jan 31 16:57:27 [16514] DBG:tm:utimer_routine: timer routine:4,tl=0x406ddf7c next=(nil), timeout=25100000 Jan 31 16:57:27 [16514] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x406dde30, INVITE si ... ) Jan 31 16:57:27 [16514] DBG:tm:set_timer: relative timeout is 1000000 Jan 31 16:57:27 [16514] DBG:tm:insert_timer_unsafe: [5]: 0x406ddf7c (26100000) Jan 31 16:57:27 [16514] DBG:tm:retransmission_handler: retransmission_handler : done Jan 31 16:57:28 [16514] DBG:tm:utimer_routine: timer routine:5,tl=0x406ddf7c next=(nil), timeout=26100000 Jan 31 16:57:28 [16514] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x406dde30, INVITE si ... ) If I use tcpdump I don't see any traffic leaving for my PSTN GW. With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

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[OpenSER-Users] dp and dP pseudo-vars
by Victor Gamov 31 Jan '08

31 Jan '08

Hi gentlemen! I use openser-1.2.2 and I want to access to dp & dP variables but sometimes they are empty. I try to access it before and after t_relay() and result is the same -- variable is empty. When this variables are set? Is it possible that they are not set at all? Thanks! -- CU, Victor Gamov

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[OpenSER-Users] Reg. Proxy domains
by Padmaja 31 Jan '08

31 Jan '08

Hi all, Please see the situation below: UA1 ----------------------> Proxy 1(not openser)---------------------- Proxy 2(openser)<----------------------UA2 UA1 is registered to proxy 1 and UA 2 to openser. I wish to make a call from UA 1 to UA 2. Proxy 1 and proxy 2 are in the LAN but do not know of each other. Is there a way we can configure the openser to send/receive the invites to proxy 1? I think the calls will be handled on domain basis, so openser will treat all the UAs registered to it as local and allow them to call each other. But how to make it allow inter domain calls? Please let me know. Thanks, Padmaja

1 0

[Serusers] SER/FIFO not accessible
by Steve B. 31 Jan '08

31 Jan '08

I am new to SER. I have installed SER 0.9.6-2 on a fresh Kubuntu server using apt. I have been able to get SER to run using the sample ser.cfg files (only changing module paths from /usr/local/lib... to /usr/lib/...). I am trying to setup the mysql support. In a nutshell, when I run serctl I get an error: # serctl add sbelt(a)ser.mytestsite.com sbelt sbelt(a)mytestsite.com MySql password: (here I enter 42serro11) error: SER/FIFO not accessible: 2 # Am I using the correct password? I am not sure which Mysql Password is being requested here. I have also tried the mysql root password with the same result. I have seen other posts similar that seem to indicate it is permissions on the /tmp directory. I tried adding to ser.cfg fifo_mode=0777. I have verified the current user can write to the /tmp directory. Frankly, I do not know if this /tmp/ser_fifo is all a red herring. Thanks for any help on how I can figure out what I am doing wrong. Steve B. Here is my ser.cfg # $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 fork=no log_stderror=yes listen=192.168.32.103 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_mode=0777 fifo_db_url="mysql://serro:47serro11@localhost/ser" loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/uri_db.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://serro:47serro11@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) 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!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (method=="ACK") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); 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 (!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(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); }

1 0

[Serusers] SER and user=phone parameter
by simon 30 Jan '08

30 Jan '08

Hi all, i have some issue with my SER server and somes IP phones. I'm using a SER 2.0 Somes phones doesn't what login in the server because its send their username <sip:login@domain.tld:5060;user=phone> The SER server try to search the domain named : "domain.tld:5060;user=phone" and doesn't find it. How can I configure my SER server to make the difference between domain and parameter ? Thanks, Simon

3 2

[OpenSER-Users] Disconnect A SIP User, how ?
by Ali Jawad 30 Jan '08

30 Jan '08

Dear All I need a way to detect if a user is logged in and disconnect him if possible. A better solution would be to detect if he is in a call and disconnect him. My questions would be 1. How to detect if a user is connected ? 2. How to detect if a user is currently in a call ? 3. How to disconnect a user 4. How to stop the call for a user (If possible) 5. If dialog is needed is my install below sufficient ? I was told in the IRC channel with regard to Q 3 that I can use the dialog module to achieve that through sending bye using dlg_end_dlg mi command. I checked the documentation at http://www.openser.org/docs/modules/1.3.x/dialog.html#AEN578 but I do not have modparam("dialog" // and so on lines in my openser.cfg file I have installed 1.3 with MySQL, Jabber and Presence support using the following First I edited the MakeFile and deleted the modules I want from the exclude_modules entry, it showed up as follows exclude_modules?= cpl-c postgres osp unixodbc \ avp_radius auth_radius group_radius uri_radius xmpp \ pua pua_bla pua_mi \ pua_usrloc pua_xmpp rls mi_xmlrpc perl perlvdb \ ldap carrierroute h350 xcap_client db_berkeley seas NICER=1 make all make prefix=/usr/local install Does that mean that the dialog module is installed? I could not find a loadmodule "dialog" line!! With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

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Re: [OpenSER-Users] CFG problem with 3rd party tool SOLVED(ish)
by Andy Smith 30 Jan '08

30 Jan '08

Hi, ok, I found and solved one issue. The acc radius_extra entries seem to have a maximum lenght that openser can parse so when I break the configuration into two "lines" like this: modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$ruri; \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat)") modparam("acc", "radius_extra", "User-Agent=$hdr(user-agent); \ Event=$hdr(event); \ Contact=$hdr(contact); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") #Billing-Party=$avp(billing_party); \ It all works, I hope OpenSER isnt ignoring the first entry in the cfg file!!!! :S. But its only working if I remove the entry "Billing-Party=$avp(billing_party)", if I change this to "Billing-Party=$avp(s:billing_party)", then all starts fine. Just hope thats how its meant to be! :P Andy.

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Re: [OpenSER-Users] CFG problem with 3rd party tool
by Andy Smith 30 Jan '08

30 Jan '08

If I was guessing (without fully understanding the syntax!) Id say the problem was Billing-Party=$avp(billing_party); should be Billing-Party=$avp(s:billing_party); Although in theory both of the above config entries seem to be valid, though different, openser.cfg entries. According to this: http://www.openser.org/docs/modules/1.2.x/avpops.html :S Also, I tried with the second format (with s:) and that then gives me another error when I start openser: 0(0) INFO:xl_parse_name: using hdr type (27) instead of <user-agent> 0(0) ERROR:acc:parse_acc_extra: parse failed in <User-Name=$Au; Calling-Station-Id=$from; Called-Station-Id=$to; Sip-Translated-Request-URI=$ruri; Sip-RPid=$avp(s:rpid); Source-IP=$si; Source-Port=$sp; Canonical-URI=$avp(s:can_uri); Billing-Party=$avp(s:billing_party); Divert-Reason=$avp(s:divert_reason); X-RTP-Stat=$hdr(X-RTP-Stat); User-Agent=$hdr(user-agent); \ Contact=$hdr(contact); Event=$hdr(event); SIP-Proxy-IP=$avp(s:sip_proxy_ip); ENUM-TLD=$avp(s:enum_tld)> around position 319 0(0) acc:parse_acc_extra: error 0(0) ERROR:acc:mod_init: failed to parse rad_extra param 0(0) init_mod(): Error while initializing module acc ERROR: error while initializing modules

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Re: [OpenSER-Users] Does CDRTool require Radius server?
by Andy Smith 30 Jan '08

30 Jan '08

Hi Iñaki, ah yeah, I hadnt noticed this before, and diameter. However have you tested this to see if it writes data in exactly the same format to a DB as Freeradius would? For sure CDRTool has been written to work with freeradius and will therefore need the data in a freeradius format/schema... Andy. ----- Original Message ----- @Andy: > but Radius will do all other updates to DB accounting data. But the point here is that accounting is done by OpenSer ACC module, which can use radius or mysql to fill "radacct" table.

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[Serusers] Question on Accounting
by Nhadie 30 Jan '08

30 Jan '08

Hi, I'm recently added voip on our services using ser. i bill my users using the time of INVITE and BYE. but got a complain from one of my users, why even if the call was not answered they still get billed, what can i use on ser to check whether a call has been answered or not, then i will bill only base on the answered call. Thanks in advanced. Regards, Ron

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[OpenSER-Users] Does CDRTool require Radius server?
by Iñaki Baz Castillo 30 Jan '08

30 Jan '08

Hi, I'm starting reading about CDRTool and as AFAIK: - MediaProxy can update call media info directly into database (without using radius). - OpenSer can log accounting in database (with "exrta-accounting"). So my question is: does CDRTool require a radius server? why exactly? Thanks for any tip ;) -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[Serusers] ser.cfg (forward to ser-sems)
by dim 30 Jan '08

30 Jan '08

hi.. I have attached my "running" ser.cfg...in which i want to include something like : if (uri =~ "sip:1.*@") { log("Forwarding to 5070"); rewritehostport("localhost:5070"); t_relay_to_udp("localhost","5070"); break; } ... in order to uris start from 1 to be forwarded to my other ser (stack for sems).. Where exactly in my ser.cfg, should i put this ??? thank you very much... dimitris

2 1

[Serusers] Check on-connected status on SER
by roayu 30 Jan '08

30 Jan '08

Hi, I would like to know whether SER can know the following status? For instance, (1) UA1(1000) is on conversation with UA2(1005) ? (2) UA1(1000) is on conversation with UA2(1005), then UA1 transfer to UA(1006)? Is there any logs that we can keep track? Thanks Roayu

2 1

[Serusers] SER Config
by Rami Abualhasan 30 Jan '08

30 Jan '08

Hi, I'm trying to configure ser to be able to act as a sip registerer and proxy sip simple message between the sip UAC and UAS. I'd like also to enable mediaproxy and natping. It will be really appreciated if some one can give the correct version of ser or openser that can fulfil my requirement with a config file which will load the modules I need. Regards

2 1

[OpenSER-Users] CFG problem with 3rd party tool
by Andy Smith 29 Jan '08

29 Jan '08

Hi list, can someone more familiar with openser CFG syntax tell me why follong the instructions for CDRTool, here: http://download.dns-hosting.info/CDRTool/doc/INSTALL.txt , breaks OpenSER?? With the following acc module settings OpenSER won't start up: modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$ruri; \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Billing-Party=$avp(billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat); \ User-Agent=$hdr(user-agent); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") I get this when starting OpenSER: 0(0) xl_parse_vname: error - bad avp name [billing_party] 0(0) ERROR:acc:parse_acc_extra: parse failed in <User-Name=$Au; Calling-Station-Id=$from; Called-Station-Id=$to; Sip-Translated-Request-URI=$ruri; Sip-RPid=$avp(s:rpid); Source-IP=$si; Source-Port=$sp; Canonical-URI=$avp(s:can_uri); Billing-Party=$avp(billing_party); Divert-Reason=$avp(s:divert_reason); X-RTP-Stat=$hdr(X-RTP-Stat); User-Agent=$hdr(user-agent); \Contact=$hdr(contact); Event=$hdr(event); SIP-Proxy-IP=$avp(s:sip_proxy_ip); ENUM-TLD=$avp(s:enum_tld)> around position 198 0(0) acc:parse_acc_extra: error Thanks for any help! Andy.

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Re: [OpenSER-Users] Does CDRTool require Radius server?
by Andy Smith 29 Jan '08

29 Jan '08

Hi Iñaki, my understanding of the documentation is that mediaproxy can update the (Radius) DB in theevent of a missing SIP BYE, but Radius will do all other updates to DB accounting data. Doesthat sound right to you? Do you have any documentation to the contrary etc?cheers Andy.Hi, I'm starting reading about CDRTool and as AFAIK: - MediaProxy can update call media info directly into database (without using radius). - OpenSer can log accounting in database (with "exrta-accounting"). So my question is: does CDRTool require a radius server? why exactly? Thanks for any tip ;)

1 0

[Serusers] SER/OpenSER Consultant
by Brian Peters 29 Jan '08

29 Jan '08

Hi, We are looking for someone for offer consultancy services for a SER/OpenSER deployment. Please advise if you offer those services or know of anyone that does. Thank you, Brian Peters mailto: brian(a)proximosystems.com

1 0

[OpenSER-Users] ABout Diameter
by dilip.m＠renovau.com 29 Jan '08

29 Jan '08

hello everyone, Anyone has used opendiameter with openser. i can run diameter client and diameter server but about the users i dont know.. How can we add users in the diameter? SO please anyone have any idea then tell me. Regards, Dilip

2 2

[OpenSER-Users] OpenSER XMPP Gateway Feature
by Ali Jawad 29 Jan '08

29 Jan '08

Dear OpenSER Users I would like some info about the XMPP gateway feature, I do have a running XMPP server ejabbard, and I would like to use a client that uses both XMPP and SIP. So I am setting up the OpenSER server and I noticed the XMPP gateway feature. What does it do ? I what scenarios can it be applied ? Thanks With Regards Ali Jawad

1 1

[OpenSER-Users] how to implement handoff?
by Aaash shah 29 Jan '08

29 Jan '08

hi i want to implement soft handoff using SIP, so please give me some information regarding it. And also i dont know how to use openSER and how to register it! i am trying to do register by kphone but registration failed. so please give me some information regarding this problems. thanks shital -- "Never Ignore Anyone Who Cares 4 u, Becoz one day u will Realize that u have lots a diamond while u were buzy in collecting stones...."

2 1

[OpenSER-Users] Strip Digits
by Jonathan Creasy 29 Jan '08

29 Jan '08

I am sending an INVITE to Box A for "F+18007622265@boxa" Part of GW Table: gw_name grp_id ip_addr port uri_scheme transport strip prefix toll-free failover 5 XXXXXXXX /NULL/ /NULL/ /NULL/ 1 Part of LCR Table: id prefix from_uri grp_id priority 170 F+1800 5 1 This sends the call to box B XXXXXXXX per the GW table: Jan 29 04:03:16 boxb-sip /usr/local/sbin/openser[28118]: Req 41070 INVITE From <sip:asterisk@XXX.XXX.XXX.XXX> To <sip:F+18007622265@boxa> RURI <sip:+18007622265@boxb> It strips the F off in the RURI field but not in the TO field so boxb then fails the call as it doesn't know how to route a call to F+1800NXXXXXX. What can I do to make it re-format the to: field? The original problem I am trying to solve is that toll-free calls are routed to a system. If that system cannot handle the call I want that system to then place the call back to the OpenSER box (boxa) but instead of using the grp_id of 2 that +1800 calls would go to (and route to the system in question), I want to indicate that boxa should use group 5 and send the calls to boxb. There is probably a very easy way of doing this which doesn't involve a mangled dial string but I'm missing it so if anyone would like to assist me with this that would be great.

1 0

[OpenSER-Users] [OT] Any UAC sending INVITE with empty SDP?
by Iñaki Baz Castillo 28 Jan '08

28 Jan '08

Hi, in order to try managing NAT when the SDP negoziation is done via "200 OK" - "ACK" (instead of "INVITE - "200 OK") I need a UAC that can send INVITE with empty SDP in order to force the called to generate the SDP offer in the reply. Is there any UAC allowing it? Thanks a lot. -- Iñaki Baz Castillo

4 6

[OpenSER-Users] Increased heap size.
by Sergio Gutierrez 28 Jan '08

28 Jan '08

Hello everybody. Currently, I am watching a non normal behaviour in OpenSER 1.2.1, no tls. My operating System is Solaris 10, running on Sparc The process which appears as timer, is consuming almost the 50% of the whole physical memory of the machine. After a deep inspection of memory distribution, appears that the most of memory consumption is at process heap, as presented with pmap(1M) command: 1978: /toip/sbin/openser -n 9 -m 256 -f /toip/etc/openser/prueba1.cfg Address Kbytes RSS Anon Locked Mode Mapped File 00010000 760 752 - - r-x-- openser 000DC000 24 24 24 - rwx-- openser 000E2000 56 32 24 - rwx-- [ anon ] 000F0000 192 128 - - rwx-- openser 00120000 3906432 3906432 3627200 - rwx-- [ heap ] EE800000 262144 3624 - - rw-s- [ anon ] ... Thanks in advance for any hint about it. Best regards. Sergio Gutiérrez. EPM Telecomunicaciones S.A. E.S.P. Medellin, Colombia.

2 5

[OpenSER-Users] dual nic at single machine openser
by flatworm 28 Jan '08

28 Jan '08

I got some routing problem configuring two NICs at debian. Openser is listening all possible network interface. I've configured one for static global IP address and one for internal private IP address. The reason why I've configured like this is for maintenance and development purpose. The static global IP line is for the service and the private IP line is for monitoring and administrations. But when I try to connect to openser of static global IP address, it was failed to register. 1) My machine was using private IP address of same subnet of openser connected to our private network. 2) My machine is trying to connect openser of static global IP address. In case of configuring dual NICs on Openser machine, is there any routing problem ? So I restricted the openser only listening static global IP address then it allows connection and succeeds in registration.

2 1

[OpenSER-Users] Implementing the Ringing Groups
by raviprakash sunkara 28 Jan '08

28 Jan '08

Hello Users Like Asterisk PBX I implemented the Ringing Group like Ring all, Round Robin and Memory Hunt , RingAll and Round Robin Ringing is working fine with the following UserAgent LinkSys 941 and X-lite, GrandStream, And SJphones, with these is working, When comes to Memory Hunt Ring feature only with Sjphone are working fine, but with User Agent not working the problem is Previous User-Agent number is Not ringing. Can any one tell me solution Why these user-Agents are not working ........ Here is my Memory Hunt logic First A rings Second time A and B rings ( in Failure route Ist time ) Third Time A, B and C ( in Failure route 2 nd time ) Fourth time A, B,C and D rings ( in Failure route 3 time ) -- Thanks &Regards Ravi Prakash Sunkara VoIP Development Tech Lead

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[OpenSER-Users] Are there are any special settings to run openser over SCTP?
by rajat.dudeja＠tcs.com 28 Jan '08

28 Jan '08

2 1

Re: [OpenSER-Users] Is possible to store various values in an AVP?
by Iñaki Baz Castillo 28 Jan '08

28 Jan '08

El Sábado, 26 de Enero de 2008, escribió: > Hello, > > Why not just make allowed_from_numbers a regex, and feed it directly? > I.e. > > $avp(allowed_from_numbers) = "(100|200|300|400)"; > > if($avp(allowed_from_numbers) =~ $fU){ > # allowed > } else { > # not allowed > } > > > Hope this helps! Yes, sure, it's a good idea ;) Thanks. -- Iñaki Baz Castillo

2 1

[OpenSER-Users] Fw: Are there are any special settings to run openser over SCTP?
by rajat.dudeja＠tcs.com 28 Jan '08

28 Jan '08

1 0

[OpenSER-Users] rtpproxy seems to be idle
by Kovács Zoltán 26 Jan '08

26 Jan '08

Hi, my plan is to give the possibility for multiple VoIP users within the same LAN to call each other using OpenSER as the SIP server. I found that if there is a NATed LAN, two different VoIP users inside the NATed LAN cannot talk to each other properly, because IP parts of the RTP packets are of the form of the local network and it is not acceptable by the VoIP software which is prepared for the IP of the router instead. (I hope I understand clearly what is going on here.) As a workaround, I found a suggestion in the internet: using rtpproxy with OpenSER. As I can understand, all RTP packets are sent to the rtpproxy process via OpenSER (nathelper module). rtpproxy will behave as a bridge between the two NATed VoIP clients lying in the same LAN. Both clients will initiate their RTP/UDP traffic on their own side and answers will come from the rtpproxy process just for the same UDP pipe. Using this method the clients do not need to allow incoming UDP traffic, only outgoing traffic should be enabled (which is usually allowed by default in most LANs). Here comes my question. I successfully set up OpenSER (version 1.1.x, but I will migrate to 1.2.x soon). I think I also successfully set up rtpproxy, but it seems that there is no real traffic between OpenSER and rtpproxy. Here I attach my command line for running rtpproxy ( 200.200.200.200 is my public IP of the OpenSER server host; I run rtpproxy on the same server). -------------------------8X snip here------------------------------- # rtpproxy -f -l 200.200.200.200 -s unix:/var/run/rtpproxy.sock -r /home/rtpproxy -S /home/rtpproxy.work rtpproxy started, pid 24568 received command "V" sending reply "20040107 " received command "VF 20050322" sending reply "1 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "VF 20050322" sending reply "1 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "U 3331228473621-29341876517920(a)160.114.144.65 160.114.144.65 10004 196215176;1" new session 3331228473621-29341876517920(a)160.114.144.65 , tag 196215176;1 requested, type strong new session on a port 35000 created, tag 196215176;1 pre-filling caller's address with 160.114.144.65:10004 sending reply "35000 212.92.3.218 " session timeout RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped session on ports 35000/0 is cleaned up -------------------------8X snip here------------------------------- My debug=7 output for OpenSER, filtered for the string "rtp" comes here (I had to chown openser.openser /var/run/rtpproxy* for OpenSER to make it possible to communicate with rtpproxy at all): -------------------------8X snip here------------------------------- # tail -f daemon.log | grep rtp Jan 15 00:27:39 firewall openser: set_mod_param_regex: found <rtpproxy_sock> in module nathelper [/usr/lib/openser/modules/] Jan 15 00:27:39 firewall openser: set_mod_param_regex: found <rtpproxy_disable> in module nathelper [/usr/lib/openser/modules/] Jan 15 00:27:39 firewall openser: set_mod_param_regex: found <rtpproxy_disable_tout> in module nathelper [/usr/lib/openser/modules/] Jan 15 00:27:39 firewall openser: set_mod_param_regex: found <rtpproxy_tout> in module nathelper [/usr/lib/openser/modules/] Jan 15 00:27:39 firewall openser: set_mod_param_regex: found <rtpproxy_retr> in module nathelper [/usr/lib/openser/modules/] Jan 15 00:27:39 firewall openser: find_cmd_export_t: found <force_rtp_proxy>(0) in module nathelper [/usr/lib/openser/modules/] Jan 15 00:27:39 firewall openser: f<force_rtp_proxy> Jan 15 00:27:39 firewall /usr/sbin/openser[25093]: fixing force_rtp_proxy, line 443 Jan 15 00:27:39 firewall /usr/sbin/openser[25095]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25096]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25098]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25100]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25106]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25105]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25117]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25103]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25108]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25104]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25112]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25094]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25113]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25093]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25111]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25122]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25118]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25123]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25097]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25107]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25125]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25120]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25115]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25110]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25119]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25116]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25124]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25114]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25121]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25099]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25102]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25109]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25101]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:28:21 firewall /usr/sbin/openser[25112]: force_rtp_proxy2: proxy reply: 35000 212.92.3.218 Jan 15 00:28:27 firewall /usr/sbin/openser[25114]: build_res_from_sip_res: copied size: orig:77, new: 16, rest: 780 msg= SIP/2.0 200 Ok^M Via: SIP/2.0/UDP 160.114.144.65:1024;branch=z9hG4bK27994379851422672^M From: 06212527755 <sip:06212527755@sip.ephone.hu:5060>;tag=196215176^M To: 06212527722 < sip:06212527722@sip.ephone.hu:5060>;tag=7816428^M Contact: < sip:06212527722@160.114.144.65:5060>^M Record-Route: <sip:200.200.200.200;lr;ftag=196215176>^M Call-ID: 3331228473621-29341876517920(a)160.114.144.65^M CSeq: 1 INVITE^M Content-Type: application/sdp^M Server: X-Lite release 1105d^M Content-Length: 320^M ^M v=0^M o=06212527722 2057424946 2057430520 IN IP4 160.114.144.65^M s=X-Lite^M c=IN IP4 160.114.144.65^M t=0 0^M m=audio 8000 RTP/AVP 0 8 3 98 97 101^M a=rtpmap:0 pcmu/8000^M a=rtpmap:8 pcma/8000^M a=rtpmap:3 gsm/8000^M a=rtpmap:98 iLBC/8000^M a=rtpmap:97 speex/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-15^M a=sendrecv^M -------------------------8X snip here------------------------------- My nathelper related part of openser.cfg comes here: -------------------------8X snip here------------------------------- loadmodule " nathelper.so" modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "received_avp", 42) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 0) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") -------------------------8X snip here------------------------------- Of course, directories /home/rtpproxy* are empty. Could somebody please help me why there is no media stream between OpenSER and rtpproxy? Your help is greatly appreciated in advance. I am using Debian Etch. Best regards, Zoltan

2 4

[OpenSER-Users] Is possible to store various values in an AVP?
by Iñaki Baz Castillo 25 Jan '08

25 Jan '08

Hi, I'd like to store in an AVP various posible values and load it with "load_credentials" parameter of "auth" module. The DB column would be: "allowed_from_numbers" AndI'd like to store there the valid values a subscribe can use as From username. For example: allowed_from_numbers = "200;201;200" So in OpenSer I do: if ($avp(allowed_from_numbers) == $fU) route(FROM_VALID); But unfortunatelly I think this is not possible, is it? which would be the required syntax to store various values? Thanks. -- Iñaki Baz Castillo

1 0

[OpenSER-Users] Revealing Number Dialled
by Jon Farmer 25 Jan '08

25 Jan '08

Hi I have some users successfully using openser and they are able to make and receive calls with any problems. I now want to provide inbound service from other telephone numbers to these registed users and want to be able to put the number dialled into the SIP header. However I am unsure how to represent this in the SIP message. Can anyone help me? TIA Jon __________________________________________________________ Sent from Yahoo! Mail - a smarter inbox http://uk.mail.yahoo.com

2 1

Re: [OpenSER-Users] Schema for OpenSER LDAP module
by Schumann Sebastian 25 Jan '08

25 Jan '08

Thanks Christian! I will use this module then in addition, didn't actually realize it is available for OpenSER :-( So far, I just checked the LDAP module, and on http://www.openser.org/dokuwiki/doku.php/tutorials:openser-auth-ldap was no hint on the H.350.4 module when showing the sample LDAP tree. I will try to implement the H.350.4 into my LDAP server and authenticate against it with OpenSER and LDAP+H.350 module. Thanks so far! Best regards Sebastian -----Original Message----- From: Christian Schlatter [mailto:cs@unc.edu] Sent: Friday, 25. January 2008 4:57 PM To: Schumann Sebastian Cc: users(a)lists.openser.org Subject: Re: [OpenSER-Users] Schema for OpenSER LDAP module Sebastian, There is an ITU-T Recommendation called H.350 that standardizes LDAP schemas for storing SIP (and other session signaling protocols) account data. The documentation of the OpenSER 1.3 H350 module (http://www.openser.org/docs/modules/1.3.x/h350.html) further describes this standard. The "H.350 Cookbook" includes LDAP schema files ready for use with different LDAP server implementations: http://metric.it.uab.edu/vnet/cookbook/v2.1/node143.html /Christian Schumann Sebastian wrote: > Dear all > > I am wondering whether there is a schema file for OpenSER LDAP module, > to use at the LDAP server for data storage? > > On the tutorial from OpenSER Wiki, e.g. the attributes SIPUserName and > SIPPassword are used, but no standard schema actually defines them acc. > my research. > > Any help with this is as always appreciated! > > Thanks and a nice weekend to all of you! > > Regards > Sebastian > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users

1 0

[OpenSER-Users] Schema for OpenSER LDAP module
by Schumann Sebastian 25 Jan '08

25 Jan '08

Dear all I am wondering whether there is a schema file for OpenSER LDAP module, to use at the LDAP server for data storage? On the tutorial from OpenSER Wiki, e.g. the attributes SIPUserName and SIPPassword are used, but no standard schema actually defines them acc. my research. Any help with this is as always appreciated! Thanks and a nice weekend to all of you! Regards Sebastian

2 1

[OpenSER-Users] OpenSUSE packages required
by Dowling, Dan (Com US) 25 Jan '08

25 Jan '08

Hi, Can anyone tell me how to best determine which OpenSUSE packages are required for openser when used with asterisk? Best regards, Dan

1 0

[OpenSER-Users] presence: difference between watchers and active_watchers
by Klaus Darilion 25 Jan '08

25 Jan '08

Hi! Can someone explain me please the difference between between watchers and active_watchers? If I understand correctly, active_watchers are the SIP clients which have SUBSCRIBEd and thus will receive NOTIFYs. But what are "watchers"? regards klaus

2 1

[OpenSER-Users] Intial INVITES without SDP
by Patrick Miccio 25 Jan '08

25 Jan '08

Hi @ all, I am fighting a pretty annoying problem at the moment. As described in rfc3261 / 13.2.1 Creating the Initial INVITE o The initial offer MUST be in either an INVITE or, if not there, in the first reliable non-failure message from the UAS back to the UAC. In this specification, that is the final 2xx response. My problem is the following: I receive a initial INVITE without SDP, forward it to a PSTN Gateway. >From the Gateway I receive the 200 OK with SDP offer. In the onreply_route I make a call to the mediaproxy use_media_proxy(); Unfortunately use_media_proxy() seems to make a lookup command when used in a reply route. I would need to make a request command which would create the media session and then do a lookup with the ACK from the UAC. Is there a way around this? cheers, Patrick. This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

3 6

[OpenSER-Users] test
by NP南頻電信黃明杰網路電話部 jeff 25 Jan '08

25 Jan '08

南頻電信股份有限公司 NP Telecom Co., Ltd. 網路電話部 VOIP Dept 黃明杰 Jeff Huang E-Mail：jeff(a)np.net.tw 公司電話(Tel)：+886-2-8226-5008 Ext：867 公司網路代表號：070-288-1122 公司傳真(FAX)：+886-2-8226-5558 公司地址：23562台北縣中和市建八路十六號七樓之一 7F-1, NO. 16, Jian-ba Rd., Chung-ho City 23562, Taipei County, Taiwan, R.O.C. 公司網址：www.np.net.tw 統一編號：89459253 匯款銀行：彰化銀行雙和分行匯款帳號：5678-01-00999-1-00

1 0

[OpenSER-Users] WeSIP: Not SER connector available - Check the Configuration, new user issue help!
by rahul kadian 25 Jan '08

25 Jan '08

Hi, I am trying to configure WESIP and trying to run demo application "Click2dial" Everything seems to ve connected properly according to logs But when I call the manager Web-GUI I get as a status of OpenSER: "Not SER connector available - Check the Configuration" What might be the reason for this. Also while trying to run Click2dial , i get the error code "Call failed with code 2" or sometimes with code 3. Have checked the confi and server.xml according to the docs and seem to be running fine. If someone can let me know any solution to this , it would be of great help. -- Regards, Rahul.

1 0

[OpenSER-Users] SVN checkout of 1.2.3?
by Tobias Lindgren 24 Jan '08

24 Jan '08

Hi, I must be getting really tired, but checking out 1.2 from svn only brings me up to 1.2.2, not 1.2.3? Same with latest snapshot from 1.2. # $Id: ChangeLog 2609 2007-08-16 14:02:25Z miconda $ Br, /Tobias

2 2

Re: [OpenSER-Users] Carrierroute help..
by Henning Westerholt 24 Jan '08

24 Jan '08

On Tuesday 22 January 2008, Ricardo Martinez wrote: > Thanks Henning.... > Please see my answers below : > > [..] > According to this,in the "failure route" is checked if receives from the > gateway the messages "408" or "5XX", but since the gateway is offline there > is no way to receive any response from it. So, what is the behaviour of > the module here?. It keeps sending the calls to the de-2.carrier1 and > de-3.carrier1 gateways? or keeps sending the calls between the 3 gateways? Hi, this has nothing to do with the behaviour of carrierroute.. OpenSER will try to resend the invite, and then generate a local 408 after the timer of the tm module run out. And then you can enter a failure route. > > 2.- Now, suppose that "de-1.carrier1" gateway is only full, so i can > > probably have a reply from the gateway (maybe a "480" message"), so the > > carrierroute module now tries with a failover route (defa.carrier1?) or > > tries with any of the other two gateways still not full? > > I suppose you mean that 'de-1' is overloaded. No, the next domain is not > automatically entered. This deficiency is known, and will be addressed in > the future. But this code is not ready yet for a release. > > Yes sorry.. i meant "de-1" is overloaded, so in this case, i can receive a > "408" Message from the gateway indicating maybe "overload". So i handle > the answer with the failure route[2], so all the calls answered with the > "408" are re-routed to the "domain 1" route isn't?. Wouldn't be more > accurate to keep sending calls to the other gateways (de-2 and de-3)? > insted of make a failover to a default route? since the other two gateways > are not overloaded...... Well, 408 is not the right error for overloading, more appropriate would be 503. Carrierroute has no knowledge of the meaning and your policy regarding a 408/ 503, let alone that the two other gws are not overloaded.. A hardcoded logic in this module would not fit to everybody. At the moment you must go with the failure_routes, or use the lcr module instead. In the future there will be probably some better solution, as i wrote in my first mail. Cheers, Henning

1 0

[OpenSER-Users] Openser 1.3 How to rewrite transport protocol from udp to tcp after redirect
by Peter P GMX 24 Jan '08

24 Jan '08

Hello, I have the following scenario: SIP UA <=UDP=> OpenSER <=TCP> Microsoft Unified Messaging 2007 When I invite an account on the UM 2007 server the following happens 1 SIP UA ==Invite UDP=> OpenSER 2 SIP UA <== 407 Proxy Authentication Required UDP= OpenSER 3 SIP UA ==ACK UDP=> OpenSER 4 SIP UA ==Invite UDP=> OpenSER 5 OpenSER ==Invite TCP=> UM2007 6 OpenSER <=Trying TCP== UM2007 7 OpenSER <= 302 Moved Temporarily TCP== UM2007 8 OpenSER == ACK TCP=> UM2007 9 OpenSER == INVITE UDP=> UM2007 .....no answer from UM2007 (I left out some steps between Openser and UA) In Step 5 OpenSER invites "sip:11111@10.10.2.13;transport=UDP>" with TCP protocol (checked with ngrep) and UM2007 answers. In Step 7 UM2007 answers with CONTACT: <sip:11111@10.10.2.13:5065;transport=UDP> but I need to answer on TCP to UM2007. (I can change the message with add_uri_param("transport=tcp") in step 5 and I get a <sip:11111@10.10.2.13:5065;transport=TCP> back from UA2007, but in fact the message is corrupted and OpenSER replies to the UA with Error 500 then, see below) The point is: - I can invite UM2007 with TCP (although I was not able to produce correct headers as there is still "transport=udp;>" mentioned in the INVITE part) - but in the failure route when I force OpenSER to reinvite with get_redirects("*") and t_relay() I am not able to swith to TCP I tried several ways as described in discussions which I found in http://osdir.com/ml/voip.openser.user/2005-12/msg00256.html - add_uri_param("transport=tcp"); ==> added transport=tcp additionally to transport=udp ==> transport=udp;transport=tcp ==> 500 Internal Server error - $avp(s:foo)="sip:10.10.2.13;transport=tcp"; avp_pushto("$du","$avp(s:foo)"); ==>had actually no effect - replace_all("transport=udp", "transport=tcp"); ==> produced invalid headers as it simply replaced the headers wrongly as described in http://openser.org/pipermail/devel/2007-May/007702.html Questions: 1. How can I change the initial INVITe message (Step5) in order to have the right headers. (transport=TCP instead of transport=UDP)? 2. How can I answer to a redirect and change to a different transport protocol (Step 9)? Your help is very much appreciated. Best regards Peter

4 10

[OpenSER-Users] Lookinfg for an SCTP compliant SIP phone
by Rajat Dudeja 24 Jan '08

24 Jan '08

Hi folks, I'm new to openser. I'm using the sctp feature of it. I'm looking to test the openser's sctp functionality. For this I have used the sctp compliant kphone. However, it does not work. The phone does not even register. Please suggest if there is any other tool to it's functionality. Thanks and regards, Rajat =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

2 1

[Serusers] presence: difference between watchers and active_watchers
by Klaus Darilion 24 Jan '08

24 Jan '08

Hi! Can someone explain me please the difference between between watchers and active_watchers? If I understand correctly, active_watchers are the SIP clients which have SUBSCRIBEd and thus will receive NOTIFYs. But what are "watchers"? regards klaus

3 3

[OpenSER-Users] User Agent (Client)
by J Bhanot 24 Jan '08

24 Jan '08

Hi, I am running openser Server... I want to know that do we have any inbuit user agent (client) in openser which we can use to send messages to this running server.. Thanks, jb ____________________________________________ =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

2 1

[Serusers] ser , rtpproxy and pstn gateway
by mashudi 24 Jan '08

24 Jan '08

I have two separate network, for public and vpn , i have install ser and rtpproxy on different machine, all user from public network will register to proxy server on ip 10.44.9.18 and the rtp will be handle by another machine using ip 10.44.9.17 , all call comming from client on public ip will route to pstn gateway using cantata media gateway on ip 10.177.107.90 (vpn ip). When client call to pstn gateway, the signalling reach the callee, but after callee pickup the phone, the client using x-lite still in calling status, and rtpproxy not receive rtp from pstn gateway. the rtpproxy have two nic using two ip for public and vpn, and ser as proxy using two nic public ip and vpn ip. any one can help? here is the log for rtpproxy and ser. the rtpproxy log : Starting rtpproxy: rtpproxy started, pid 25745 received command "2573_0 V" sending reply "2573_0 20040107 " received command "2574_0 V" sending reply "2574_0 20040107 " received command "2572_0 V" sending reply "2572_0 20040107 " received command "2572_6 U NjQzNmQzNzFjZTBmMzM5MDUxYjQwODFlOTFmMTdhMzI. 10.11.11.152 30126 9d08dc55" new session NjQzNmQzNzFjZTBmMzM5MDUxYjQwODFlOTFmMTdhMzI., tag 9d08dc55 requested, type strong new session on a port 35010 created, tag 9d08dc55 pre-filling caller's address with 10.11.11.152:30126 sending reply "2572_6 35010 10.44.9.17 " the ser log: 0(2572) SIP Request: 0(2572) method: <NOTIFY> 0(2572) uri: <sip:02170743627@10.11.11.152:65100> 0(2572) version: <SIP/2.0> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-9533d56650536c16-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <n/a>; state=17 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) preparing to run routing scripts... 0(2572) parse_headers: flags=128 0(2572) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2572) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2572) parse_headers: flags=8 0(2572) DEBUG: add_param: tag=9e3f287c 0(2572) end of header reached, state=29 0(2572) DEBUG: get_hdr_field: <To> [56]; uri=[sip:02170743627@10.44.9.18] 0(2572) DEBUG: to body ["02170743627"<sip:02170743627@10.44.9.18>] 0(2572) DEBUG: add_param: tag=59420d70 0(2572) end of header reached, state=29 0(2572) parse_headers: flags=256 0(2572) is_preloaded: No 0(2572) grep_sock_info - checking if host==us: 12==10 && [10.11.11.152] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) grep_sock_info - checking if host==us: 12==14 && [10.11.11.152] == [10.177.107.124] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) grep_sock_info - checking if host==us: 12==10 && [10.11.11.152] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) grep_sock_info - checking if host==us: 12==14 && [10.11.11.152] == [10.177.107.124] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) check_self: host != me 0(2572) grep_sock_info - checking if host==us: 10==10 && [10.44.9.18] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 5060 0(2572) after_loose: Topmost route URI: 'sip:10.44.9.18;lr;ftag=9e3f287c' is me 0(2572) parse_headers: flags=256 0(2572) get_hdr_field: cseq <CSeq>: <13> <NOTIFY> 0(2572) DEBUG: get_hdr_body : content_length=461 0(2572) found end of header 0(2572) find_next_route: No next Route HF found 0(2572) after_loose: No next URI found 0(2572) DEBUG: t_newtran: msg id=55 , global msg id=54 , T on entrance=0xffffffffffffffff 0(2572) parse_headers: flags=-1 0(2572) parse_headers: flags=60 0(2572) t_lookup_request: start searching: hash=24011, isACK=0 0(2572) DEBUG: RFC3261 transaction matching failed 0(2572) DEBUG: t_lookup_request: no transaction found 0(2572) DEBUG: mk_proxy: doing DNS lookup... 0(2572) DEBUG: get_out_socket: socket determined: 0x6be9c8 0(2572) check_via_address(10.11.11.152, 10.11.11.152, 0) 0(2572) DEBUG: add_to_tail_of_timer[4]: 0x2ab57bba4208 0(2572) DEBUG: add_to_tail_of_timer[0]: 0x2ab57bba4228 0(2572) SER: new transaction fwd'ed 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 0(2572) SIP Reply (status): 0(2572) version: <SIP/2.0> 0(2572) status: <200> 0(2572) reason: <OK> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bKbcd5.79134365.0>; state=16 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) forward_reply: found module tm, passing reply to it 0(2572) DEBUG: t_check: msg id=56 global id=55 T start=0xffffffffffffffff 0(2572) parse_headers: flags=17 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-9533d56650536c16-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <65100>; state=16 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=17 0(2572) parse_headers: this is the second via 0(2572) DEBUG: add_param: tag=9e3f287c 0(2572) end of header reached, state=29 0(2572) DEBUG: get_hdr_field: <To> [56]; uri=[sip:02170743627@10.44.9.18] 0(2572) DEBUG: to body ["02170743627"<sip:02170743627@10.44.9.18>] 0(2572) get_hdr_field: cseq <CSeq>: <13> <NOTIFY> 0(2572) DEBUG: t_reply_matching: hash 24011 label 1446261143 branch 0 0(2572) DEBUG: t_reply_matching: reply matched (T=0x2ab57bba4030)! 0(2572) DEBUG: t_check: msg id=56 global id=56 T end=0x2ab57bba4030 0(2572) DEBUG: reply_received: org. status uas=0, uac[0]=0 local=0 is_invite=0) 0(2572) parse_headers: flags=64 0(2572) parse_headers: flags=64 0(2572) ->>>>>>>>> T_code=0, new_code=200 0(2572) DEBUG: relay_reply: branch=0, save=0, relay=0 0(2572) old size: 531, new size: 472 0(2572) build_res_from_sip_res: copied size: orig:267, new: 208, rest: 264 msg= SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.11.152:65100;branch=z9hG4bK-d87543-9533d56650536c16-1--d87543-;rport=65100 Record-Route: <sip:10.44.9.18;lr;ftag=59420d70> Contact: <sip:02170743627@10.11.11.152:65100> To: "02170743627"<sip:02170743627@10.44.9.18>;tag=9e3f287c From: "02170743627"<sip:02170743627@10.44.9.18>;tag=59420d70 Call-ID: MWNiZWE5ODMyN2U0MTJmMWMyNjQzMTNiNGJjZmRjYjA. CSeq: 13 NOTIFY User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 0(2572) WARNING:vqm_resize: resize(0) called 0(2572) DEBUG: add_to_tail_of_timer[2]: 0x2ab57bba40b0 0(2572) DEBUG: reply relayed. buf=0x6c9930: SIP/2.0 2..., shmem=0x2ab57bba6b10: SIP/2.0 2 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 1(2573) DEBUG: timer routine:4,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: timer routine:2,tl=0x2ab57bba40b0 next=(nil) 1(2573) DEBUG: wait_handler : removing 0x2ab57bba4030 from table 1(2573) DEBUG: delete transaction 0x2ab57bba4030 1(2573) DEBUG: wait_handler : done 0(2572) SIP Request: 0(2572) method: <INVITE> 0(2572) uri: <sip:02170890010@10.44.9.18> 0(2572) version: <SIP/2.0> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-2623ee4d6908a453-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <n/a>; state=17 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) preparing to run routing scripts... 0(2572) parse_headers: flags=128 0(2572) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2572) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2572) parse_headers: flags=8 0(2572) end of header reached, state=9 0(2572) DEBUG: get_hdr_field: <To> [43]; uri=[sip:02170890010@10.44.9.18] 0(2572) DEBUG: to body ["02170890010"<sip:02170890010@10.44.9.18> ] 0(2572) DEBUG: add_param: tag=9d08dc55 0(2572) end of header reached, state=29 0(2572) parse_headers: flags=256 0(2572) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(2572) DEBUG: get_hdr_body : content_length=420 0(2572) found end of header 0(2572) find_first_route: No Route headers found 0(2572) loose_route: There is no Route HF 0(2572) grep_sock_info - checking if host==us: 10==10 && [10.44.9.18] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 5060 0(2572) parse_headers: flags=16384 0(2572) pre_auth(): Credentials with given realm not found 0(2572) build_auth_hf(): 'Proxy-Authenticate: Digest realm="10.44.9.18", nonce="4798817c850ce0e3c233a35804d9be1afe7fe2a3" ' 0(2572) parse_headers: flags=-1 0(2572) check_via_address(10.11.11.152, 10.11.11.152, 0) 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 0(2572) SIP Request: 0(2572) method: <ACK> 0(2572) uri: <sip:02170890010@10.44.9.18> 0(2572) version: <SIP/2.0> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-2623ee4d6908a453-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <n/a>; state=17 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) preparing to run routing scripts... 0(2572) parse_headers: flags=4 0(2572) DEBUG: add_param: tag=a3f1a545ddce5205bb83afef24613e34.78aa 0(2572) end of header reached, state=29 0(2572) DEBUG: get_hdr_field: <To> [85]; uri=[sip:02170890010@10.44.9.18] 0(2572) DEBUG: to body ["02170890010"<sip:02170890010@10.44.9.18>] 0(2572) DEBUG: sl_filter_ACK : local ACK found -> dropping it! 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 0(2572) SIP Request: 0(2572) method: <INVITE> 0(2572) uri: <sip:02170890010@10.44.9.18> 0(2572) version: <SIP/2.0> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-363a1c17b37ac509-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <n/a>; state=17 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) preparing to run routing scripts... 0(2572) parse_headers: flags=128 0(2572) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2572) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2572) parse_headers: flags=8 0(2572) end of header reached, state=9 0(2572) DEBUG: get_hdr_field: <To> [43]; uri=[sip:02170890010@10.44.9.18] 0(2572) DEBUG: to body ["02170890010"<sip:02170890010@10.44.9.18> ] 0(2572) DEBUG: add_param: tag=9d08dc55 0(2572) end of header reached, state=29 0(2572) parse_headers: flags=256 0(2572) get_hdr_field: cseq <CSeq>: <2> <INVITE> 0(2572) DEBUG: get_hdr_body : content_length=420 0(2572) found end of header 0(2572) find_first_route: No Route headers found 0(2572) loose_route: There is no Route HF 0(2572) grep_sock_info - checking if host==us: 10==10 && [10.44.9.18] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 5060 0(2572) check_nonce(): comparing [4798817c850ce0e3c233a35804d9be1afe7fe2a3] and [4798817c850ce0e3c233a35804d9be1afe7fe2a3] 0(2572) HA1 string calculated: 0f7e2f302e87dfe46c5ca6566ad32b47 0(2572) check_response(): Our result = '82c47bef0adf71299b8fb157a15e0131' 0(2572) check_response(): Authorization is OK 0(2572) save_rpid(): rpid value is '' 0(2572) check_username(): Digest username and URI username match 0(2572) check_via_address(10.11.11.152, 10.11.11.152, 0) 0(2572) parse_headers: flags=64 0(2572) lookup(): '02170890010' Not found in usrloc 0(2572) grep_sock_info - checking if host==us: 10==10 && [10.44.9.18] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 5060 0(2572) parse_headers: flags=64 0(2572) parse_headers: flags=-1 0(2572) DEBUG:check_content_type: type <application/sdp> found valid 0(2572) parse_headers: flags=32 0(2572) ERROR: extract_mediaip: no `c=' in SDP 0(2572) DEBUG: t_newtran: msg id=59 , global msg id=56 , T on entrance=0xffffffffffffffff 0(2572) parse_headers: flags=-1 0(2572) parse_headers: flags=60 0(2572) t_lookup_request: start searching: hash=48512, isACK=0 0(2572) DEBUG: RFC3261 transaction matching failed 0(2572) DEBUG: t_lookup_request: no transaction found 0(2572) SER: new INVITE 0(2572) parse_headers: flags=-1 0(2572) check_via_address(10.11.11.152, 10.11.11.152, 0) 0(2572) WARNING:vqm_resize: resize(0) called 0(2572) DEBUG: reply sent out. buf=0x6c9930: SIP/2.0 1..., shmem=0x2ab57bba3ad8: SIP/2.0 1 0(2572) DEBUG: _reply_light: finished 0(2572) DEBUG: mk_proxy: doing DNS lookup... 0(2572) DEBUG: get_out_socket: socket determined: 0x6bebb8 0(2572) WARNING: get_send_socket: not listening on the requested socket, no fork mode? 0(2572) parse_headers: flags=2048 0(2572) parse_headers: flags=-1 0(2572) clen_builder: content-length: 436 (436) 0(2572) check_via_address(10.11.11.152, 10.11.11.152, 0) 0(2572) DEBUG: add_to_tail_of_timer[4]: 0x2ab57bba4208 0(2572) DEBUG: add_to_tail_of_timer[0]: 0x2ab57bba4228 0(2572) SER: new transaction fwd'ed 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 1(2573) DEBUG: timer routine:4,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[5]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:5,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[6]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:6,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[7]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:7,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[7]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: get_out_socket: socket determined: 0x6be9c8 0(2572) udp_rcv_loop: probing packet received from 10.11.11.152 19710 1(2573) DEBUG: timer routine:7,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[7]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:7,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[7]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:7,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[7]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:7,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[7]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:0,tl=0x2ab57bba4228 next=(nil) 1(2573) DEBUG: final_response_handler:stop retr. and send CANCEL (0x2ab57bba4030) 1(2573) ->>>>>>>>> T_code=100, new_code=408 1(2573) DEBUG: relay_reply: branch=0, save=0, relay=0 1(2573) parse_headers: flags=-1 1(2573) check_via_address(10.11.11.152, 10.11.11.152, 0) 1(2573) DEBUG: add_to_tail_of_timer[4]: 0x2ab57bba4150 1(2573) DEBUG: add_to_tail_of_timer[0]: 0x2ab57bba4170 1(2573) DEBUG: reply relayed. buf=0x6c84d0: SIP/2.0 4..., shmem=0x2ab57bba3ad8: SIP/2.0 4 1(2573) DEBUG: final_response_handler : done 1(2573) DEBUG: timer routine:7,tl=0x2ab57bba4208 next=(nil) 0(2572) SIP Request: 0(2572) method: <ACK> 0(2572) uri: <sip:02170890010@10.44.9.18> 0(2572) version: <SIP/2.0> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-363a1c17b37ac509-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <n/a>; state=17 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) preparing to run routing scripts... 0(2572) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(2572) parse_headers: flags=128 0(2572) DEBUG: add_param: tag=d548cf07be509c4dc64b42f06e6fb880-a911 0(2572) end of header reached, state=29 0(2572) DEBUG: get_hdr_field: <To> [85]; uri=[sip:02170890010@10.44.9.18] 0(2572) DEBUG: to body ["02170890010"<sip:02170890010@10.44.9.18>] 0(2572) get_hdr_field: cseq <CSeq>: <2> <ACK> 0(2572) DEBUG: get_hdr_body : content_length=0 0(2572) found end of header 0(2572) DEBUG: is_maxfwd_present: max_forwards header not found! 0(2572) DEBUG: add_param: tag=9d08dc55 0(2572) end of header reached, state=29 0(2572) parse_headers: flags=256 0(2572) find_first_route: No Route headers found 0(2572) loose_route: There is no Route HF 0(2572) grep_sock_info - checking if host==us: 10==10 && [10.44.9.18] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 5060 0(2572) DEBUG: t_newtran: msg id=60 , global msg id=59 , T on entrance=0xffffffffffffffff 0(2572) parse_headers: flags=-1 0(2572) parse_headers: flags=60 0(2572) t_lookup_request: start searching: hash=48512, isACK=1 0(2572) DEBUG: RFC3261 transaction matched, tid=-d87543-363a1c17b37ac509-1--d87543- 0(2572) DEBUG: t_lookup_request: transaction found (T=0x2ab57bba4030) 0(2572) DEBUG: add_to_tail_of_timer[2]: 0x2ab57bba40b0 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 1(2573) DEBUG: timer routine:4,tl=0x2ab57bba4150 next=(nil) 1(2573) DEBUG: timer routine:2,tl=0x2ab57bba40b0 next=(nil) 1(2573) DEBUG: wait_handler : removing 0x2ab57bba4030 from table 1(2573) DEBUG: delete transaction 0x2ab57bba4030 1(2573) DEBUG: wait_handler : done 0(2572) SIP Request: 0(2572) method: <NOTIFY> 0(2572) uri: <sip:02170743627@10.11.11.152:65100> 0(2572) version: <SIP/2.0> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-3217dc739e312401-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <n/a>; state=17 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) preparing to run routing scripts... 0(2572) parse_headers: flags=128 0(2572) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2572) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2572) parse_headers: flags=8 0(2572) DEBUG: add_param: tag=9e3f287c 0(2572) end of header reached, state=29 0(2572) DEBUG: get_hdr_field: <To> [56]; uri=[sip:02170743627@10.44.9.18] 0(2572) DEBUG: to body ["02170743627"<sip:02170743627@10.44.9.18>] 0(2572) DEBUG: add_param: tag=59420d70 0(2572) end of header reached, state=29 0(2572) parse_headers: flags=256 0(2572) is_preloaded: No 0(2572) grep_sock_info - checking if host==us: 12==10 && [10.11.11.152] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) grep_sock_info - checking if host==us: 12==14 && [10.11.11.152] == [10.177.107.124] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) grep_sock_info - checking if host==us: 12==10 && [10.11.11.152] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) grep_sock_info - checking if host==us: 12==14 && [10.11.11.152] == [10.177.107.124] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) check_self: host != me 0(2572) grep_sock_info - checking if host==us: 10==10 && [10.44.9.18] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 5060 0(2572) after_loose: Topmost route URI: 'sip:10.44.9.18;lr;ftag=9e3f287c' is me 0(2572) parse_headers: flags=256 0(2572) get_hdr_field: cseq <CSeq>: <14> <NOTIFY> 0(2572) DEBUG: get_hdr_body : content_length=425 0(2572) found end of header 0(2572) find_next_route: No next Route HF found 0(2572) after_loose: No next URI found 0(2572) DEBUG: t_newtran: msg id=61 , global msg id=60 , T on entrance=0xffffffffffffffff 0(2572) parse_headers: flags=-1 0(2572) parse_headers: flags=60 0(2572) t_lookup_request: start searching: hash=24006, isACK=0 0(2572) DEBUG: RFC3261 transaction matching failed 0(2572) DEBUG: t_lookup_request: no transaction found 0(2572) DEBUG: mk_proxy: doing DNS lookup... 0(2572) DEBUG: get_out_socket: socket determined: 0x6be9c8 0(2572) check_via_address(10.11.11.152, 10.11.11.152, 0) 0(2572) DEBUG: add_to_tail_of_timer[4]: 0x2ab57bba4208 0(2572) DEBUG: add_to_tail_of_timer[0]: 0x2ab57bba4228 0(2572) SER: new transaction fwd'ed 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 1(2573) DEBUG: timer routine:4,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, NOTIFY si ... ) 1(2573) DEBUG: add_to_tail_of_timer[5]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 0(2572) SIP Reply (status): 0(2572) version: <SIP/2.0> 0(2572) status: <200> 0(2572) reason: <OK> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK6cd5.c487f5f5.0>; state=16 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) forward_reply: found module tm, passing reply to it 0(2572) DEBUG: t_check: msg id=62 global id=61 T start=0xffffffffffffffff 0(2572) parse_headers: flags=17 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-3217dc739e312401-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <65100>; state=16 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=17 0(2572) parse_headers: this is the second via 0(2572) DEBUG: add_param: tag=9e3f287c 0(2572) end of header reached, state=29 0(2572) DEBUG: get_hdr_field: <To> [56]; uri=[sip:02170743627@10.44.9.18] 0(2572) DEBUG: to body ["02170743627"<sip:02170743627@10.44.9.18>] 0(2572) get_hdr_field: cseq <CSeq>: <14> <NOTIFY> 0(2572) DEBUG: t_reply_matching: hash 24006 label 1600092236 branch 0 0(2572) DEBUG: t_reply_matching: reply matched (T=0x2ab57bba4030)! 0(2572) DEBUG: t_check: msg id=62 global id=62 T end=0x2ab57bba4030 0(2572) DEBUG: reply_received: org. status uas=0, uac[0]=0 local=0 is_invite=0) 0(2572) parse_headers: flags=64 0(2572) parse_headers: flags=64 0(2572) ->>>>>>>>> T_code=0, new_code=200 0(2572) DEBUG: relay_reply: branch=0, save=0, relay=0 0(2572) old size: 531, new size: 472 0(2572) build_res_from_sip_res: copied size: orig:267, new: 208, rest: 264 msg= SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.11.152:65100;branch=z9hG4bK-d87543-3217dc739e312401-1--d87543-;rport=65100 Record-Route: <sip:10.44.9.18;lr;ftag=59420d70> Contact: <sip:02170743627@10.11.11.152:65100> To: "02170743627"<sip:02170743627@10.44.9.18>;tag=9e3f287c From: "02170743627"<sip:02170743627@10.44.9.18>;tag=59420d70 Call-ID: MWNiZWE5ODMyN2U0MTJmMWMyNjQzMTNiNGJjZmRjYjA. CSeq: 14 NOTIFY User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 0(2572) WARNING:vqm_resize: resize(0) called 0(2572) DEBUG: add_to_tail_of_timer[2]: 0x2ab57bba40b0 0(2572) DEBUG: reply relayed. buf=0x6c9930: SIP/2.0 2..., shmem=0x2ab57bba6b10: SIP/2.0 2 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up # $Id: nat-rtpproxy.cfg 51 2006-01-31 13:28:04Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ debug=9 fork=no log_stderror=yes mhomed=yes listen=10.44.9.18 # INSERT YOUR IP ADDRESS HERE listen=10.177.107.124 port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/uri_db.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/avpops.so" loadmodule "/usr/lib/ser/modules/permissions.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "udp:10.44.9.17:12345") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") 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!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); 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 (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; 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" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); 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 (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (uri=~"^sip:0217[0-9]*@") { route(4); route(5); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } route[5] { rewritehost("10.177.107.90"); avp_write("i:45", "inv_timeout"); route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; }

1 0

[Serusers] SIP Express Bundle - let's get started
by Mike Trest - Personal 23 Jan '08

23 Jan '08

Hi, The holidays and the new year work load seems to have taken all my time. Anyway, next week, I should be clear of regulatory filings and paperwork. That should give me time to resume building stand alone SER to test prototype of the SER Express Bundle. Any first-draft materials from anyone yet? ..mike..

4 5

[OpenSER-Users] UAC module (openser 1.2.1)
by Antonio Reale 23 Jan '08

23 Jan '08

Hi all, I'm trying to use UAC module and I'd like to know if it is possible to configure openser to register on another proxy on start up and wait for incoming calls, instead of use UAC module to terminate calls on an authenticated trunk. Thanks. Regards, Antonio

4 4

[Serusers] forking error - branch response sent to uac
by Tomasz Zieleniewski 23 Jan '08

23 Jan '08

hi, I encountered following error. I have two user agents register under the same uid. When there is an invite request both ua's ring. When one of them UA1 answers the call, by sending 200 response some strange error occurs. UA1's 200 response is immediately sent to UAC who initiated a call. after this ser sends cancel message to other UA2. The other UA2 responses with 200 to the cancel message and sends response to the initial invite: 487 Request Terminated. After this ser should response with ACK to the second UA2 but it isn't instead it forwards 487 response to UAC who sent the initial invite. Please tell why ser behaves this way? What could be the reason for such behavior? Regards Tomasz

1 0

[OpenSER-Users] Performance of Openser as Presence Server
by Noble Antony T 23 Jan '08

23 Jan '08

Hi, Do any one has an idea about the Performance benchmark of Openser as Presence Server..? Like how many Subscribers, how many subscribe/publish/notify messages, it can handle.. [ I want to know how much Subscribers/Subscriptions & how many Subscribe/Notify/Publish messages, Openser can handle. I am using Openser as only Presence Server ( not proxy/registrar )] If anybody have any figures about this performance, please let me know. Thanks -- Noble Antony Thattil.

1 0

[OpenSER-Users] Reg. Proxy authentication
by Padmaja 23 Jan '08

23 Jan '08

Hi all, Please see the call flow below: GW1 Proxy GW2 |-----invite----------->| | |<------407-----------| | |-----invite w/cred--->| | | |-----Invite------->| |<------------200 Ok---------------------| |----------------Ack--------------------->| | | | |<----ReInvite -----| | When the proxy receives the Reinvite below from GW 2, should it again challenge it for proxy authentication or simply forward the call to the GW1? The openser proxy in our lab simply forwards the Invite without asking for authentication this time. However, I need to test the situation, where the proxy asks for authentication for the Reinvite. Is this possible? Please let me know. Thanks, Padmaja

2 2

[OpenSER-Users] Forwarding to SRV records not working
by James Holden 23 Jan '08

23 Jan '08

Hi, I'm having problems with OpenSER forwarding calls to a domain where there are only SRV records. OpenSER seems not to perform any balancing or failover when a target server is unavailable or gives an error. I wish to forward calls to a number of onward gateways that at times may not respond at all, or may respond with something like "503 Service Unavailable". The actual destination address is derived from an enum lookup, but it may be for example 123456(a)example.com There are SRV records as follows: _sip._udp.example.com. 600 IN SRV 10 50 5060 gw1.example.com. _sip._udp.example.com. 600 IN SRV 10 50 5060 gw2.example.com. The behaviour I'd expect/like would be that OpenSER will distribute the calls between the two targets specified in the SRV records, and will fail over to the other target if one fails to respond or gives a fatal error such as a 503. What actually happens, is that the calls all go to the first one regardless. Are my expectations reasonable? Does OpenSER's SRV record implementation work like this or am I misled? Kind regards, James

3 3

Re: [OpenSER-Users] Users Digest, Vol 32, Issue 70
by Padmaja 23 Jan '08

23 Jan '08

----- Original Message ----- From: <users-request(a)lists.openser.org> To: <users(a)lists.openser.org> Sent: Wednesday, January 23, 2008 3:28 AM Subject: Users Digest, Vol 32, Issue 70 > Send Users mailing list submissions to > users(a)lists.openser.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.openser.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)lists.openser.org > > You can reach the person managing the list at > users-owner(a)lists.openser.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Forwarding to SRV records not working (James Holden) > 2. Re: Serial fork terminated after 5 call (Bogdan-Andrei Iancu) > 3. How to handle INVITES with different To: header (Peter P GMX) > 4. Re: How to handle INVITES with different To: header > (Bogdan-Andrei Iancu) > 5. Re: Forwarding to SRV records not working (Bogdan-Andrei Iancu) > 6. OpenSER: no NOTIFY when updating PUBLISH if integrating with > OpenXCAP? (KevinKinnan) > 7. OpenSER: no NOTIFY when updating PUBLISH if integrating with > OpenXCAP? (KevinKinnan) > 8. Re: Problems trying to build an Active/Active OSer 1.3 > cluster (Stefan Sayer) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 22 Jan 2008 16:41:42 +0000 > From: James Holden <james.holden(a)uk.aql.com> > Subject: [OpenSER-Users] Forwarding to SRV records not working > To: users(a)lists.openser.org > Message-ID: <20080122164142.GD4315@james-desktop> > Content-Type: text/plain; charset=us-ascii > > Hi, > > I'm having problems with OpenSER forwarding calls to a domain where > there are only SRV records. > > OpenSER seems not to perform any balancing or failover when a target > server is unavailable or gives an error. > > I wish to forward calls to a number of onward gateways that at times may > not respond at all, or may respond with something like "503 Service > Unavailable". > > The actual destination address is derived from an enum lookup, but it > may be for example 123456(a)example.com > > There are SRV records as follows: > > _sip._udp.example.com. 600 IN SRV 10 50 5060 gw1.example.com. > _sip._udp.example.com. 600 IN SRV 10 50 5060 gw2.example.com. > > The behaviour I'd expect/like would be that OpenSER will distribute the > calls between the two targets specified in the SRV records, and will > fail over to the other target if one fails to respond or gives a fatal > error such as a 503. > > What actually happens, is that the calls all go to the first one > regardless. > > Are my expectations reasonable? > > Does OpenSER's SRV record implementation work like this or am I misled? > > Kind regards, > > James > > > > ------------------------------ > > Message: 2 > Date: Tue, 22 Jan 2008 18:49:21 +0200 > From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > Subject: Re: [OpenSER-Users] Serial fork terminated after 5 call > To: Sergey Babichev <zaikini(a)yandex.ru> > Cc: users(a)lists.openser.org > Message-ID: <47961E91.7080003(a)voice-system.ro> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi Sergey, > > the most important thing when searching is to know where to search ;)... > See: > http://www.openser.org/docs/modules/1.2.x/nathelper.html#AEN370 > > Regards, > Bogdan > > Sergey Babichev wrote: >> Okey:)) And last question about documentation, I searched documentation >> about function nat_uac_test from nathelper module for version 1.2.x and >> I'm find it in documentation for version 0.9. Openser 1.2.x not support >> function nat_uac_test or documentation 1.2.x not full? >> >> >> >> >>> Hi Sergey, >>> >> >> >>> I'm afraid there is no docs about the errors - the idea is to have >>> >> >> >>> sef-explaining error messages and to avoid documenting them ;) >>> >> >> >>> Regards, >>> >> >> >>> bogdan >>> >> >> >>> Sergey Babichev wrote: >>> >> >> >>>> Thanks, where i can read about this error (discarding fwd for a >>>> cancelled transaction) and others? >>>> >> >> >> >> >> >> >> >> >>>> >>>> >> >> >>>>> Hi Sergey, >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> the "discarding fwd for a cancelled transaction" error is generated >>>>> when >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> you try to send a request for a transaction that was already cancelled >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> from the client side. So probably the PSTN GW cancels the call while >>>>> you >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> still want to fork a new branch. >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> regards, >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> Bogdan >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> Sergey Babichev wrote: >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>>> Problem is appear when i calling from PSTN. It's work fine when i >>>>>> calling from sip account to huntgroup. I think this is problem whith >>>>>> timer in my PSTN gateway. >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Hi all! >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> I want enable hunting for my extensions. It's work fine, but if my >>>>>>> hunt group consists of more then 5 extensions, call terminated and I >>>>>>> see in syslog error: >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: >>>>>>> ERROR:tm:t_forward_nonack: discarding fwd for a cancelled >>>>>>> transaction >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: >>>>>>> ERROR:tm:w_t_relay: t_forward_nonack failed >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: >>>>>>> DEBUG:tm:relay_reply: branch=5, save=0, relay=5 >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> What is problem may be? >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> -- >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Best regards, >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Sergey Babichev >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> _______________________________________________ >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Users mailing list >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Users(a)lists.openser.org >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> -- >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> Best regards, >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> Sergey Babichev >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> _______________________________________________ >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>> Users mailing list >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>> Users(a)lists.openser.org >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >> >> >> >> >>>> -- >>>> >> >> >> >> >>>> Best regards, >>>> >> >> >> >> >>>> Sergey Babichev >>>> >> >> >> >> >>>> >>>> >> >> >> >> -- >> >> Best regards, >> >> Sergey Babichev >> >> > > > > > ------------------------------ > > Message: 3 > Date: Tue, 22 Jan 2008 19:20:44 +0100 > From: Peter P GMX <Prometheus001(a)gmx.net> > Subject: [OpenSER-Users] How to handle INVITES with different To: > header > To: users(a)lists.openser.org > Message-ID: <479633FC.2090506(a)gmx.net> > Content-Type: text/plain; charset=ISO-8859-15; format=flowed > > I have the following scenario: > > I receive a request with PSTN_number(a)my.ip.add.res from my VoIP > In alias_db I have defined a rule that points to yyy(a)myip.com which is > a registered account > > After alias_db_lookup OpenSER invites as follows: > Invite yyy(a)myip.com > To: xxxxxxx(a)my.ip.add.res > > The UA (actually a PBX with VoIP support) answers with 404 not found. > > When I invite as follows: > Invite yyy(a)myip.com > To: yyy(a)myip.com > > everything works fine. > > My question: How do you handle these cases? > Rewriting the To: part seems dificult and will mess up a lot - hein? > > Kind regards > Peter > > > > > ------------------------------ > > Message: 4 > Date: Tue, 22 Jan 2008 21:41:32 +0200 > From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > Subject: Re: [OpenSER-Users] How to handle INVITES with different To: > header > To: Prometheus001(a)gmx.net > Cc: users(a)lists.openser.org > Message-ID: <479646EC.5090301(a)voice-system.ro> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi Peter, > > According to RFC3261, in SIP, routing is exclusively done based on RURI > (and eventually Route hdrs). TO header has 0 implication in routing. > So, your PBX is deprecated and out of standards and I suggest either > upgrade, either replacement. > > Regards, > Bogdan > > Peter P GMX wrote: >> I have the following scenario: >> >> I receive a request with PSTN_number(a)my.ip.add.res from my VoIP >> In alias_db I have defined a rule that points to yyy(a)myip.com which is >> a registered account >> >> After alias_db_lookup OpenSER invites as follows: >> Invite yyy(a)myip.com >> To: xxxxxxx(a)my.ip.add.res >> >> The UA (actually a PBX with VoIP support) answers with 404 not found. >> >> When I invite as follows: >> Invite yyy(a)myip.com >> To: yyy(a)myip.com >> >> everything works fine. >> >> My question: How do you handle these cases? >> Rewriting the To: part seems dificult and will mess up a lot - hein? >> >> Kind regards >> Peter >> > > > > ------------------------------ > > Message: 5 > Date: Tue, 22 Jan 2008 23:00:28 +0200 > From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > Subject: Re: [OpenSER-Users] Forwarding to SRV records not working > To: James Holden <james.holden(a)uk.aql.com> > Cc: users(a)lists.openser.org > Message-ID: <4796596C.7000004(a)voice-system.ro> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi James, > > Openser does not do balancing over records with same priority - it will > just use the first (as returned by DNS server). > > But failover based on NAPTR and SRV records is in place. > > Regards, > Bogdan > > > James Holden wrote: >> Hi, >> >> I'm having problems with OpenSER forwarding calls to a domain where >> there are only SRV records. >> >> OpenSER seems not to perform any balancing or failover when a target >> server is unavailable or gives an error. >> >> I wish to forward calls to a number of onward gateways that at times may >> not respond at all, or may respond with something like "503 Service >> Unavailable". >> >> The actual destination address is derived from an enum lookup, but it >> may be for example 123456(a)example.com >> >> There are SRV records as follows: >> >> _sip._udp.example.com. 600 IN SRV 10 50 5060 gw1.example.com. >> _sip._udp.example.com. 600 IN SRV 10 50 5060 gw2.example.com. >> >> The behaviour I'd expect/like would be that OpenSER will distribute the >> calls between the two targets specified in the SRV records, and will >> fail over to the other target if one fails to respond or gives a fatal >> error such as a 503. >> >> What actually happens, is that the calls all go to the first one >> regardless. >> >> Are my expectations reasonable? >> >> Does OpenSER's SRV record implementation work like this or am I misled? >> >> Kind regards, >> >> James >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> >> > > > > > ------------------------------ > > Message: 6 > Date: Wed, 23 Jan 2008 05:49:18 +0800 > From: KevinKinnan <kinnan2224(a)hotmail.com> > Subject: [OpenSER-Users] OpenSER: no NOTIFY when updating PUBLISH if > integrating with OpenXCAP? > To: <users(a)lists.openser.org> > Message-ID: <BAY126-W641A1FEB850E880B28F56C43E0(a)phx.gbl> > Content-Type: text/plain; charset="gb2312" > > > Thank you so much, Franz! > > Indeed, the problem is coming from xml. It is well-formed but seems the > server can't understand it. > The whole xml document I used is the similar with yours except the root > element with namespace which > is showing below: > > <cr: ruleset xmlns="urn:ietf:params:xml:ns:common-policy" > xmlns:pr="urn:ietf:params:xml:ns:pres-rules" > xmlns:cr="urn:ietf:params:xml:ns:common-policy"> > > > I got no error sending the xml doc with this as beginning. Only no Notify > message after updating Publish, > and sometimes the first Subscribe can't get through that is no Notify > sending back from server. It's really > weired! Anyway, I finally fix it with your advise, Thanks Indeed. > > Kevin, > > _________________________________________________________________ > ????? MSN ?????????? > http://mobile.msn.com.cn/ >

1 0

Re: [OpenSER-Users] Serial fork terminated after 5 call SOLVED
by Sergey Babichev 23 Jan '08

23 Jan '08

Thanks ))) > Hi Sergey, > the most important thing when searching is to know where to search ;)... > See: > http://www.openser.org/docs/modules/1.2.x/nathelper.html#AEN370 > Regards, > Bogdan > Sergey Babichev wrote: > > Okey:)) And last question about documentation, I searched documentation about function nat_uac_test from nathelper module for version 1.2.x and I'm find it in documentation for version 0.9. Openser 1.2.x not support function nat_uac_test or documentation 1.2.x not full? > > > > > > > > > >> Hi Sergey, > >> > > > > > >> I'm afraid there is no docs about the errors - the idea is to have > >> > > > > > >> sef-explaining error messages and to avoid documenting them ;) > >> > > > > > >> Regards, > >> > > > > > >> bogdan > >> > > > > > >> Sergey Babichev wrote: > >> > > > > > >>> Thanks, where i can read about this error (discarding fwd for a cancelled transaction) and others? > >>> > > > > > > > > > > > > > > > > > >>> > >>> > > > > > >>>> Hi Sergey, > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> the "discarding fwd for a cancelled transaction" error is generated when > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> you try to send a request for a transaction that was already cancelled > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> from the client side. So probably the PSTN GW cancels the call while you > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> still want to fork a new branch. > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> regards, > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Bogdan > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Sergey Babichev wrote: > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>>> Problem is appear when i calling from PSTN. It's work fine when i calling from sip account to huntgroup. I think this is problem whith timer in my PSTN gateway. > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Hi all! > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> I want enable hunting for my extensions. It's work fine, but if my hunt group consists of more then 5 extensions, call terminated and I see in syslog error: > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:w_t_relay: t_forward_nonack failed > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: DEBUG:tm:relay_reply: branch=5, save=0, relay=5 > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> What is problem may be? > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> -- > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Best regards, > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Sergey Babichev > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> _______________________________________________ > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Users mailing list > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Users(a)lists.openser.org > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> -- > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> Best regards, > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> Sergey Babichev > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> _______________________________________________ > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>> Users mailing list > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>> Users(a)lists.openser.org > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > > > > > > > > > >>> -- > >>> > > > > > > > > > >>> Best regards, > >>> > > > > > > > > > >>> Sergey Babichev > >>> > > > > > > > > > >>> > >>> > > > > > > > > -- > > > > Best regards, > > > > Sergey Babichev > > > > -- Best regards, Sergey Babichev

1 0

[OpenSER-Users] Problems trying to build an Active/Active OSer 1.3 cluster
by Raúl Alexis Betancor Santana 22 Jan '08

22 Jan '08

Hi all, I'm trying to build an Active/Active OpenSer 1.3 cluster with 2 nodes. All it's working ok for the REGISTER messages throught the use of t_replicate and add_path, but I've problems with the INVITE messages. Both servers share the same location table on a PostgreSQL Multimaster cluster (but that's irrelevant, the PostgreSQL cluster I mean...), using db_mode = 2 An that's the logic I used: - On REGISTER ->> - Auth Request - NAT ? - add_path - t_replicate(oser2) - On INVITE ->> - Auth Request - lookup - Load prefs from DB - relay The problem I'm having is that the lookup function changes the RURI of the message on node1 BEFORE taking into account the "Path:" element present inside the AoR, so when the INVITE gets forwarded to node2 (where the NATed UAC it's registered), node2 deny the relaying of the message because $rD of the RURI it's not local (remember that the RURI whas changed on node1). Is there any way to know on node1 that the INVITE request MUST obey the "Path:" so I could to something like: $old_ru = $ru lookup("location") if(path_exists()) { $ru = $old_ru } ... t_relay() ?? Any other aproach ? Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual S.L.

3 3

[OpenSER-Users] OpenSER: no NOTIFY when updating PUBLISH if integrating with OpenXCAP?
by KevinKinnan 22 Jan '08

22 Jan '08

Franz, Have you tested 'validity' rule and 'sphere' rule under Openser + Openxcap? Regards, Kevin _________________________________________________________________ 天凉了，添衣了，心动了，“七件”了 http://get.live.cn

1 0

[OpenSER-Users] OpenSER: no NOTIFY when updating PUBLISH if integrating with OpenXCAP?
by KevinKinnan 22 Jan '08

22 Jan '08

Thank you so much, Franz! Indeed, the problem is coming from xml. It is well-formed but seems the server can't understand it. The whole xml document I used is the similar with yours except the root element with namespace which is showing below: <cr: ruleset xmlns="urn:ietf:params:xml:ns:common-policy" xmlns:pr="urn:ietf:params:xml:ns:pres-rules" xmlns:cr="urn:ietf:params:xml:ns:common-policy"> I got no error sending the xml doc with this as beginning. Only no Notify message after updating Publish, and sometimes the first Subscribe can't get through that is no Notify sending back from server. It's really weired! Anyway, I finally fix it with your advise, Thanks Indeed. Kevin, _________________________________________________________________ 手机也能上 MSN 聊天了，快来试试吧！ http://mobile.msn.com.cn/

1 0

[OpenSER-Users] How to handle INVITES with different To: header
by Peter P GMX 22 Jan '08

22 Jan '08

I have the following scenario: I receive a request with PSTN_number(a)my.ip.add.res from my VoIP In alias_db I have defined a rule that points to yyy(a)myip.com which is a registered account After alias_db_lookup OpenSER invites as follows: Invite yyy(a)myip.com To: xxxxxxx(a)my.ip.add.res The UA (actually a PBX with VoIP support) answers with 404 not found. When I invite as follows: Invite yyy(a)myip.com To: yyy(a)myip.com everything works fine. My question: How do you handle these cases? Rewriting the To: part seems dificult and will mess up a lot - hein? Kind regards Peter

2 1

Re: [OpenSER-Users] Serial fork terminated after 5 call
by Sergey Babichev 22 Jan '08

22 Jan '08

Okey:)) And last question about documentation, I searched documentation about function nat_uac_test from nathelper module for version 1.2.x and I'm find it in documentation for version 0.9. Openser 1.2.x not support function nat_uac_test or documentation 1.2.x not full? > Hi Sergey, > I'm afraid there is no docs about the errors - the idea is to have > sef-explaining error messages and to avoid documenting them ;) > Regards, > bogdan > Sergey Babichev wrote: > > Thanks, where i can read about this error (discarding fwd for a cancelled transaction) and others? > > > > > > > > > >> Hi Sergey, > >> > > > > > >> the "discarding fwd for a cancelled transaction" error is generated when > >> > > > > > >> you try to send a request for a transaction that was already cancelled > >> > > > > > >> from the client side. So probably the PSTN GW cancels the call while you > >> > > > > > >> still want to fork a new branch. > >> > > > > > >> regards, > >> > > > > > >> Bogdan > >> > > > > > >> Sergey Babichev wrote: > >> > > > > > >>> Problem is appear when i calling from PSTN. It's work fine when i calling from sip account to huntgroup. I think this is problem whith timer in my PSTN gateway. > >>> > > > > > > > > > > > > > > > > > >>> > >>> > > > > > >>>> Hi all! > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> I want enable hunting for my extensions. It's work fine, but if my hunt group consists of more then 5 extensions, call terminated and I see in syslog error: > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:w_t_relay: t_forward_nonack failed > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: DEBUG:tm:relay_reply: branch=5, save=0, relay=5 > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> What is problem may be? > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> -- > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Best regards, > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Sergey Babichev > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> _______________________________________________ > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Users mailing list > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Users(a)lists.openser.org > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >>>> > > > > > >>>> > >>>> > > > > > > > > > > > > > > > > > >>> -- > >>> > > > > > > > > > >>> Best regards, > >>> > > > > > > > > > >>> Sergey Babichev > >>> > > > > > > > > > >>> _______________________________________________ > >>> > > > > > >>> Users mailing list > >>> > > > > > >>> Users(a)lists.openser.org > >>> > > > > > >>> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >>> > > > > > > > > > >>> > >>> > > > > > > > > -- > > > > Best regards, > > > > Sergey Babichev > > > > -- Best regards, Sergey Babichev

2 1

[OpenSER-Users] Strating up with openser
by J Bhanot 22 Jan '08

22 Jan '08

Hi, I am just starting up with this stack.. Could you kindly let me know what strategy i should follow so as to use this stack for building applications rapidly. Thanking in anticipation. Regards, Jagmohan Bhanot Tata Consultancy Services Mailto: j.bhanot(a)tcs.com Website: http://www.tcs.com ____________________________________________ Experience certainty. IT Services Business Solutions Outsourcing ____________________________________________ =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

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[OpenSER-Users] cdr records in openser - how to get them
by tloginbr-openserbr＠yahoo.com.br 22 Jan '08

22 Jan '08

Hi all, I'm pretty new to OpenSER, and I'm trying to get it working here. I have one OpenSER machine and a few asterisk machines behind this one. All my call go the OpenSER that does load balancing and fail over between my asterisk servers. Up to now I was getting the CDR from all my asterisk, but that is not going very well since I have to search in all asterisk machines to get the cdr for one extension. So I want to get the cdr directly from the openser thats where all calls have to pass. I found some people talking about CDRTool, but I think it does a little too much for me. I'm just looking for some tool that saves the information I need inside mysql so I can build my cdr report with some simple sql queries. Does anyone in here get the cdr out of the openser or is that not even possible? All help is very welcome here. Thanks in advance, Thiago Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/

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Re: [OpenSER-Users] Carrierroute help..
by Henning Westerholt 22 Jan '08

22 Jan '08

On Tuesday 22 January 2008, Ricardo Martinez wrote: > Hello. > I would like to make a question about the carrierroute module. > How does it work when one of the "level 0" gw's are down?. For example , > this is my carrierroute table : > > +----+---------+-------------+--------+------+-------+---------------+ > | id | carrier | scan_prefix | domain | prob | strip | rewrite_host | > +----+---------+-------------+--------+------+-------+---------------+ > | 1 | 1 | 49 | 0 | 0.3 | 0 | de-1.carrier1 | > | 2 | 1 | 49 | 0 | 0.3 | 0 | de-2.carrier1 | > | 3 | 1 | 49 | 0 | 0.3 | 0 | de-3.carrier1 | > | 4 | 1 | | 1 | 1 | 0 | defa.carrier1 | > +----+---------+-------------+--------+------+-------+---------------+ > > > > 1- Let's suppose "de-1.carrier1" gateway is down. So.. any INVITE to > that gateway will not have any answer (for the failure_route failover), how > the module handle this?, It tries with the others+.? Hi Ricardo, no, at the moment there is no automatically failover done from the carrierroute module. So if you want to route your calls to another gw, e.g. 'de-2', you must manual re-route it to another gw/ domain in the failure_route block. E.g. route to domain 0 in route[..], route to domain 1 in failure_route. > 2.- Now, suppose that "de-1.carrier1" gateway is only full, so i can > probably have a reply from the gateway (maybe a "480" message"), so the > carrierroute module now tries with a failover route (defa.carrier1?) or > tries with any of the other two gateways still not full? I suppose you mean that 'de-1' is overloaded. No, the next domain is not automatically entered. This deficiency is known, and will be addressed in the future. But this code is not ready yet for a release. You could use the lcr module, there exist a 'next_gw' function, if you don't want to manually specify the failure_routes. Cheers, Henning

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