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[OpenSER-Users] Help - As I picked up codes negative www_authorize()
by Marcio 23 May '08

23 May '08

Hi... Anybody could me help, I'm doing some tests and would like to pick up negative codes of the return www_authorize.... I trying with ($retcode) ($?) I've not had success..... Does anyone know how...? Any hints.....?? thx...

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[OpenSER-Users] Problems with sl module in openser1.3.1
by arturo arturo 23 May '08

23 May '08

Hi guys, Could some help me with this problem, I'm trying to run openser 1.3.1, but it dies complaining about the sl module. I issued just 'make all' command before this root@zeus:/home/aochoa/openser/openser-1.3.1-notls # ./openser May 22 15:57:16 [4482] DBG:core:yyparse: loading module /usr/local/lib/openser/modules/mysql.so May 22 15:57:16 [4482] DBG:core:yyparse: loading module /usr/local/lib/openser/modules/sl.so May 22 15:57:16 [4482] ERROR:core:version_control: *module compile flags mismatch for* /usr/local/lib/openser/modules/sl.so core: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT module: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT Ing Arturo Ochoa. Electrosystems

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[OpenSER-Users] Open Source VoIP in Porto, Portugal
by Johansson Olle E 23 May '08

23 May '08

Friends, I will be spending a few days in Porto, Portugal in the beginning of June. Any Asterisk and/or OpenSER users there that wants to go out and have dinner and Open Source Voip talk? Respond off list, and we'll see if we can meet. Have a nice weekend! /Olle --- * Olle E. Johansson - oej(a)edvina.net * Asterisk Training http://edvina.net/training/

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[OpenSER-Users] [OT] NGREP showing \r and \n symbols
by Iñaki Baz Castillo 23 May '08

23 May '08

Hi, does somebody know how to use NGREP command to show \r and \n symbols instead of a dot "." replacing them? By default NGREP prints a dot "." relacing any non-printable caracter. This symbol can be changed with "-p" option but waht I want is to show \r and \n literally. Thanks for any suggestion. PD: Any other recomendation (program) to get SIP traces showing \r and \n? -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [OpenSER-Users] Authentication problem in openSER (Henning Westerholt)
by Ahmed Huraimel 23 May '08

23 May '08

hello Henning, i would like to thank you very much. you save alot of work because i decided to change my database from LDAP to MySQL. just to make it clear to anyone who may face this problem. the solution to keep you passwords hashed in the LDAP server is to save it as HA1 = MD5(username:realm:password). after you build the HA1 save as clear in the LDAP server (i am using openLADP). if you save it as MD5 you will retrieve it as {MD5} blablabla which may cause a problem. so hash it then save it as clear. thanks for every one helps me in this issue regards, Ahmed Huraimel ALALI On Fri, May 23, 2008 at 12:00 PM, <users-request(a)lists.openser.org> wrote: > 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. Re: [OT] NGREP showing \r and \n symbols (Andreas Heise) > 2. Re: [OT] NGREP showing \r and \n symbols (I?aki Baz Castillo) > 3. Re: Problem with audio when either or both UAs are not behind > NAT (I?aki Baz Castillo) > 4. Re: Authentication problem in openSER (Henning Westerholt) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 22 May 2008 23:55:56 +0200 > From: Andreas Heise <aheise(a)gmx.de> > Subject: Re: [OpenSER-Users] [OT] NGREP showing \r and \n symbols > To: Ovidiu Sas <osas(a)voipembedded.com> > Cc: users(a)lists.openser.org > Message-ID: <4835EBEC.7070800(a)gmx.de> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi I?aki, > > I had the same issue with a Nortel CS1k switch, the easiest way to check > this > problem is to use the Wireshark feature "Display raw text for SIP message" > which can be enabled under preferences / protocols / SIP > > regards, > Andreas > > > Ovidiu Sas schrieb: > > Hello Inaki, > > > > Perform a raw capture using tcpdump and open the file with wireshark. > > Check the lower window in wireshark where you have the message > > displayed in hex format. > > There you will be able to spot '\n' and '\r'. To find the hex codes, > > perform a 'man ascii' on your linux box. > > > > > > Regards, > > Ovidiu Sas > > > > On Tue, May 20, 2008 at 10:53 AM, I?aki Baz Castillo <ibc(a)in.ilimit.es> > wrote: > > > >> El Tuesday 20 May 2008 16:37:03 Bogdan-Andrei Iancu escribi?: > >> > >>> Hi I?aki, > >>> > >>> have you tried the "-W byline" ? - it will format you the lines. > >>> > >> Hi Bogdan. Yes, in fact I use this option "-W byline", but this option > just > >> splits the trace into lines (macthing "\n"). This is not enough for me > since > >> \r is not showed. > >> > >> I need it since I send and receive traffic from a carrier using a > painful > >> Nortel CS2K softswitch that separes SDP lines with \r instead of the > >> mandatory \r\n adn they are applying dirty and not working patches > during > >> production state so I need a way to monitorize it. > >> > >> Thanks a lot. > >> > >> > >> > >> -- > >> I?aki Baz Castillo > >> ibc(a)in.ilimit.es > >> > >> _______________________________________________ > >> Users mailing list > >> Users(a)lists.openser.org > >> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >> > >> > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > ------------------------------ > > Message: 2 > Date: Fri, 23 May 2008 09:31:31 +0200 > From: I?aki Baz Castillo <ibc(a)in.ilimit.es> > Subject: Re: [OpenSER-Users] [OT] NGREP showing \r and \n symbols > To: users(a)lists.openser.org > Message-ID: <200805230931.31959.ibc(a)in.ilimit.es> > Content-Type: text/plain; charset="iso-8859-1" > > El Thursday 22 May 2008 23:55:56 Andreas Heise escribi?: > > Hi I?aki, > > > > I had the same issue with a Nortel CS1k switch > > It's nice to know that I'm not theonly sending/receiving traffic from a > Nortel > CS[12]K. ;) > > -- > I?aki Baz Castillo > ibc(a)in.ilimit.es > > > > ------------------------------ > > Message: 3 > Date: Fri, 23 May 2008 09:32:21 +0200 > From: I?aki Baz Castillo <ibc(a)in.ilimit.es> > Subject: Re: [OpenSER-Users] Problem with audio when either or both > UAs are not behind NAT > To: users(a)lists.openser.org > Message-ID: <200805230932.21444.ibc(a)in.ilimit.es> > Content-Type: text/plain; charset="ISO-8859-1" > > El Thursday 22 May 2008 18:54:53 Krunal Patel escribi?: > > Hi, > > > > I have registered an UA 12345 , which is public. > > In location table, I found NULL value for received. > > I dial any UA , irrespective to their NAT status. > > Call has been established but can't get audio both side. > > It does not use mediaproxy for audio signaling. > > > > How to resolve this problem? > > Do a full sip flow capture and analyze trhe address and port of media in > SDP > of INVITE and 200 OK. > > > -- > I?aki Baz Castillo > ibc(a)in.ilimit.es > > > > ------------------------------ > > Message: 4 > Date: Fri, 23 May 2008 10:57:56 +0200 > From: Henning Westerholt <henning.westerholt(a)1und1.de> > Subject: Re: [OpenSER-Users] Authentication problem in openSER > To: users(a)lists.openser.org > Cc: Ahmed Huraimel <huraimel(a)gmail.com> > Message-ID: <200805231057.56974.henning.westerholt(a)1und1.de> > Content-Type: text/plain; charset="utf-8" > > On Thursday 22 May 2008, Ahmed Huraimel wrote: > > i am investigating the authentication on openSER. I search for a proper > > explanations but unfortunately i did not find how it is exactly done so i > > did some experiments. i assumed that the response is generated as the > > following: note that i set the username and password with the same string > > "alali" > > > > Response = MD5( username + MD5(password) + realm + nonce) > > [..] > > could anyone tell me how exactly the authentication is done in openSER? > is > > the response generated is like the one i assumed? what + means in the > > response? does is mean concatenation or exoring? > > Hi Ahmed, > > the authentification in OpenSER/ SIP is based on HTTP auth. You find a > detailed explanation for the construction of the response for example at: > http://en.wikipedia.org/wiki/Digest_access_authentication , some further > informations at: http://www.voip-info.org/wiki/view/SIP+Authentication > > For the exact logic inside OpenSER just take a look at the auth module > source > code, it should be not that hard to understand. ;-) > > Cheers, > > Henning > > > > ------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > End of Users Digest, Vol 36, Issue 48 > ************************************* >

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[OpenSER-Users] Authentication problem in openSER
by Ahmed Huraimel 23 May '08

23 May '08

hello all, i am investigating the authentication on openSER. I search for a proper explanations but unfortunately i did not find how it is exactly done so i did some experiments. i assumed that the response is generated as the following: note that i set the username and password with the same string "alali" Response = MD5( username + MD5(password) + realm + nonce) and in the openser.cfg i tell the server to take the username from the header and set the password from the username since the username and the password are the same (only for testing). TEST1: setting modparam("auth", "calculate_ha1", 1) and telling the server to expect a clear password $avp(s:username) = $fU; $avp(s:password) = $fU; IT WORKS TEST2:setting modparam("auth", "calculate_ha1", 0) and telling the server to expect a hashed password $avp(s:username) = $fU; $avp(s:password) = $(fU{s.md5}); IT DOES NOT WORK could anyone tell me how exactly the authentication is done in openSER? is the response generated is like the one i assumed? what + means in the response? does is mean concatenation or exoring? if(!pv_www_authorize("")) { www_challenge(""/*realm*/,"0"/*qop*/); exit; } if (!save("location")) sl_reply_error(); sl_send_reply("200", "ok"); exit;

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[OpenSER-Users] Problem with audio when either or both UAs are not behind NAT
by Krunal Patel 23 May '08

23 May '08

Hi, I have registered an UA 12345 , which is public. In location table, I found NULL value for received. I dial any UA , irrespective to their NAT status. Call has been established but can't get audio both side. It does not use mediaproxy for audio signaling. How to resolve this problem? Thanks in advance. -- Regards, Krunal Patel

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[OpenSER-Users] Problems with sl module in openser1.3.1
by aochoa aochoa 22 May '08

22 May '08

Hi guys, Could some help me with this problem, I'm trying to run openser 1.3.1, but it dies complaining about the sl module. I issued just 'make all' command before this root@zeus:/home/aochoa/openser/openser-1.3.1-notls # ./openser May 22 15:57:16 [4482] DBG:core:yyparse: loading module /usr/local/lib/openser/modules/mysql.so May 22 15:57:16 [4482] DBG:core:yyparse: loading module /usr/local/lib/openser/modules/sl.so May 22 15:57:16 [4482] ERROR:core:version_control: *module compile flags mismatch for* /usr/local/lib/openser/modules/sl.so core: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT module: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT Ing Arturo Ochoa. Electrosystems

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[OpenSER-Users] Problems with sl module in openser1.3.1
by aochoa aochoa 22 May '08

22 May '08

Hi guys, Could some help me with this problem, I'm trying to run openser 1.3.1, but it dies complaining about the sl module. I issued just 'make all' command before this root@zeus:/home/aochoa/openser/openser-1.3.1-notls # ./openser May 22 15:57:16 [4482] DBG:core:yyparse: loading module /usr/local/lib/openser/modules/mysql.so May 22 15:57:16 [4482] DBG:core:yyparse: loading module /usr/local/lib/openser/modules/sl.so May 22 15:57:16 [4482] ERROR:core:version_control: *module compile flags mismatch for* /usr/local/lib/openser/modules/sl.so core: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT module: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT Ing Arturo Ochoa. Electrosystems

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[Serusers] Mysql Problem
by vijay tiwari 22 May '08

22 May '08

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[OpenSER-Users] OPENSER Dinner, Vienna, Austria, May 24
by Daniel-Constantin Mierla 22 May '08

22 May '08

Hello, just to announce the next OPENSER Social Networking Event in Vienna, Austria, May 24, 2008, 19:30 at Plutzer Brau: http://www.plutzerbraeu.at/plutzerbraeu.html <http://www.voipuser.org/ship_to.php?url=http://www.plutzerbraeu.at/plutzerb…> Among the participants that confirmed the presence already: - Elena-Ramona Modroiu, co-founder OpenSER, COO at http://www.asipto.com <http://www.voipuser.org/ship_to.php?url=http://www.asipto.com> - Andreas Granig, developer OpenSER, CEO at http://www.sipwise.com <http://www.voipuser.org/ship_to.php?url=http://www.sipwise.com> - Daniel-Constantin Mierla, co-founder OpenSER, CEO at http://www.asipto.com <http://www.voipuser.org/ship_to.php?url=http://www.asipto.com> The primary goal of the event is to strengthen the relations within community and give opportunity to network for collaboration and business. Participation to the event is free, everybody pays for its expenses. Anyone interested in OpenSER, SIP or VoIP in general is welcome to join. You should contact us to reserve your seat and get notifications in case of changes of time or place. Previous similar event was in Barcelona, May 6 -- remarks and photos: http://www.asipto.com/photos/BARCELONA2008-ALBUM/ http://openser.blogspot.com/2008/05/openser-dinner-barcelona-remarks.html Next one is in Berlin, May 28 (more details soon). Cheers, Daniel -- http://www.asipto.com

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[OpenSER-Users] [Fwd: Re: facing problem]
by Amit Vijayvargiya 22 May '08

22 May '08

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Re: [Serusers] ser/sems question
by Stefan Sayer 22 May '08

22 May '08

o Steven C. Blair [05/22/08 13:52]: > Thanks Stefan. Is the README for sems-1.0.0 correct? I'm questioning step #4. If I am running this version of SEMS on a Linux box without a proxy server, our proxy resides elsewhere, do I need to perform step #4? no, that is optional, and you only need to do that if you want SER as frontend to SEMS. That's why it says 'Optional' in in http://ftp.iptel.org/pub/sems/doc/current/Readme.html Stefan > > Thanks,Steve > >> -----Original Message----- >> From: Stefan Sayer [mailto:stefan.sayer@iptego.com] >> Sent: Wednesday, May 21, 2008 4:45 PM >> To: Steven C. Blair >> Cc: serusers(a)iptel.org >> Subject: Re: [Serusers] ser/sems question >> >> Hello, >> >> o Steven C. Blair [05/20/08 20:13]: >>> I've been away from SEMS for a few years. II hear the new version of >>> SEMS makes an excellent conference server. Does anyone happen to know >> if >>> I can install SEMS on a standalone Linux machine and interface it, >> via >>> SIP, to a SER 0.9.7-pre3 proxy server? >> yes, you can use it now with any SIP proxy thanks to its own SIP stack. ... -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.com www.iptego.com iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

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[OpenSER-Users] Problem PUA_USRLOC, Presence acc. REGISTER/INVITE
by Schumann Sebastian 22 May '08

22 May '08

Dear all I have a problem, which I thought I can solve with PUA_USRLOC. I have a OpenSER SIP Proxy and want him to create presence information about the messages that pass him. Some examples: - REGISTER: Publish as online/available - REGISTER+Expires=0: Publish as offline - INVITE/200 OK: Publish as busy - BYE: Publish as available again. PUA_USRLOC seems to me can handle only simple on/offline according registration state, that is saved on the server. As I have only a proxy server and messages just pass by, this solution seems not the best. What would you recommend me to solve the problem? Change the module, add a module? As I don't have that much of experience in that area I was thinking about using perl module and trigger some perl functions that parse the messages that pass the proxy and make perl send the PUBLISHes to the presence server. Maybe some of you have other ideas or suggestions about that.... I would be happy if you share them with me. Best regards Sebastian

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[OpenSER-Users] Does somebody use $(rU{s.unescape.user}) to be RFC3261 compliant?
by Iñaki Baz Castillo 22 May '08

22 May '08

Hi, to be RFC3261 compliant a SIP proxy should accept hex encoded username in any URI, this is: sip:alice@domain == sip:%61l%69ce@domain For allowing this we must use, explicitely, the transformation $(rU{s.unescape.user}). Do OpenSer administrators allow this hex encoding? -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Small documentation mistake pua_usrloc
by Schumann Sebastian 22 May '08

22 May '08

Dear all I noticed a small mistake in the documentation: I had some errors in config file using the pua_usrloc modparam parameters with copy & paste. The problem were the semicolons ; behind the modparam definitions. As I had the error some time ago already I easily figured that out. Although, it might be good to remove them from the doc for other users in the future to not have the same problem as well. Best regards Sebastian

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[Serusers] Creating SIP trunk from OpenSER to Cisco CallManager
by G. Ross 22 May '08

22 May '08

Hi, I've been tasked with getting an OpenSER system to talk to a Cisco CallManager system.I've tried searching the docs, the mailing lists and google, but all my searches end up with references to the "trunk" in the source code repository. Can someone point me somewhere which could help me in this task ? Thanks, GTG

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[OpenSER-Users] facing problem
by Amit Vijayvargiya 22 May '08

22 May '08

Hi, I am facing a problem when user is online more then one machines, openser not able to send request to all machines. Thanks in advance Amit

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[Serusers] ser/sems question
by Steven C. Blair 21 May '08

21 May '08

I've been away from SEMS for a few years. II hear the new version of SEMS makes an excellent conference server. Does anyone happen to know if I can install SEMS on a standalone Linux machine and interface it, via SIP, to a SER 0.9.7-pre3 proxy server? Thanks,steve Senior Network Engineer, Information Systems and Computing Networking and Telecommunications , Suite 221A /6228 University of Pennsylvania Voice:215-573-8396 FAX:215-898-9348

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[OpenSER-Users] v1.3.2 , presence hash table error
by Jeremy George 21 May '08

21 May '08

Hello All, BLA was working nicely for me with v1.3.0 and Polycom 3.0.0 phones. I downloaded 1.3.2 and I'm now getting this error on the presence server. I tried adding the hash table modparams explicitly, but with got the same error message. modparam("presence", "subs_htable_size", 11) modparam("presence", "pres_htable_size", 11) Thoughts? And thanks. May 20 11:44:35 [12319] ERROR:presence:get_stored_info: record not found in hash_table May 20 11:44:35 [12319] ERROR:presence:handle_subscribe: getting stored info - Jeremy

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[OpenSER-Users] Unable to do transformation on expires param if display name present
by Robert Dyck 21 May '08

21 May '08

I am currently using 1.3.2 but this problem exists in older versions as well. My config script includes a transformation on the expires parameter in a Contact header. I have found that a Linksys SPA3000 will include the display name in the Contact header and that there is white space between the display name and the URI. The function parse_param_name in parse_param.c will stop parsing when it encounters the white space. As a workaround I commented the line in parse_param_name that checks for white space. This works for me but I do not know if there are side effects. This form of Contact is not common but it appears to be legal according to RFC 3261. 20.10 Contact A Contact header field value provides a URI whose meaning depends on the type of request or response it is in. A Contact header field value can contain a display name, a URI with URI parameters, and header parameters. This document defines the Contact parameters "q" and "expires". These parameters are only used when the Contact is present in a REGISTER request or response, or in a 3xx response. Additional parameters may be defined in other specifications. When the header field value contains a display name, the URI including all URI parameters is enclosed in "<" and ">". If no "<" and ">" are present, all parameters after the URI are header parameters, not URI parameters. The display name can be tokens, or a quoted string, if a larger character set is desired. Even if the "display-name" is empty, the "name-addr" form MUST be used if the "addr-spec" contains a comma, semicolon, or question mark. There may or may not be LWS between the display-name and the "<". Here is an excerpt from my script -- if ((is_present_hf("Expires") && $(hdr(Expires) {s.int}) == 0)\ ||($(ct{param.value,expires}) == '0')) { log(1, "looks like deregistration\n"); I added some debug messages to illustrate the problem -- May 19 22:16:32 [14447] DBG:core:parse_param_name: string passed to parse_param_name is RJD <sip:17476316719@192.168.1.4:5061>;expires=3600 May 19 22:16:32 [14447] DBG:core:parse_param_name: next char is R May 19 22:16:32 [14447] DBG:core:parse_param_name: next char is J May 19 22:16:32 [14447] DBG:core:parse_param_name: next char is D May 19 22:16:32 [14447] DBG:core:parse_param_name: next char is May 19 22:16:32 [14447] ERROR:core:parse_params: invalid character, ; expected May 19 22:16:32 [14447] DBG:core:run_transformations: return val is -1 May 19 22:16:32 [14447] CRITICAL:core:comp_scriptvar: cannot get left var value May 19 22:16:32 [14447] WARNING:core:do_action: error in expression

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Re: [OpenSER-Users] ERROR 1071 (42000) at line 2: Specified key was too long; max key length is 1000 bytes
by Henning Westerholt 21 May '08

21 May '08

On Wednesday 21 May 2008, Yazid Hadj Said wrote: > Hi Henning, > I configured mysql with latin1 (in my.cnf), but i don't know when can i > specify to openser to use latin1. In other word how can I verify if openser > use UTF-8? Hi Yazid, its not possible to use the UTF-8 charset at the moment with OpenSER and mysql. I think this is a limitation from mysql, look for example to this bug: http://bugs.mysql.com/bug.php?id=4541 The openserdbctl script failed to detect that your server uses UTF-8 as default charset, because there was an error present in the tool. This is fixed now, it should now work for you regardless if you use latin1 or utf8 as default charset. Cheers, Henning

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[OpenSER-Users] ERROR 1071 (42000) at line 2: Specified key was too long; max key length is 1000 bytes
by Yazid Hadj Said 21 May '08

21 May '08

Hi, When I try to create openser database (openserdbctl create) on Gentoo, I have the message: INFO: creating database openser ... ERROR 1071 (42000) at line 2: Specified key was too long; max key length is 1000 bytes ERROR: Creating core tables failed! I search when i can modify TABLE_TYPE but i not found. Regards, Yazid

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[OpenSER-Users] nat problems
by Pezhman Lali 21 May '08

21 May '08

Dear I am testing a lot of configs, for ser, openser and asterisk, but till now, I can not find a total solution for solving nat, do you think ser/openser+ (defined modules) can support all of nat services, around the world ? best,

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[OpenSER-Users] NAT problem in bridging mode
by Christian Koch 21 May '08

21 May '08

Hi, I have a problem with openser and rtpproxy. I'm trying to use them as a gateway between the public internet and a LAN. Clients in the internet may be natted, so I'm using nathelper. Calls are only made from LAN to outside or vice versa, but not from LAN to LAN or from outside to outside. The following should illustrate my configuation: ----- ------------------ UAC1 --- | NAT | ------------- | OpenSER/rtpproxy | ----------------UAC2 ----- ------------------ | | | | | dynamic public IP 2.3.4.5 192.168.103.121 192.168.103.189 (e.g. 1.2.3.4) UAC1 and UAC2 are both registered at OpenSER. Now I'm making a call from UAC1 to UAC2. SIP messages are passed just fine, but the RTP traffic from UAC2 to UAC1 is dropped at the NAT. I used tcpdump on the OpenSER/rtpproxy machine to figure out what happens to RTP and it shows the following (ports and IPs are just examples): stream1: 1.2.3.4:10000 -> 2.3.4.5:35000 ->RTP is forwarded by rtpproxy-> 192.168.103.121:35000 -> 192.168.103.189:11000 stream2: 1.2.3.4:20000 <- 2.3.4.5:35000 <-RTP is forwarded by rtpproxy<- 192.168.103.121:35000 <- 192.168.103.189:11000 Port 20000 in stream2 is the RTP-port used internally by UAC1 behind the NAT (this port is found in the INVITE from UAC1 to OpenSER). I understand, that rtpproxy sends the first packets to port 20000. But, after receiving some packets from port 10000, shouldn't it change the destination port to 20000 so they can pass the NAT? rtpproxy is started like this: "./rtpproxy -l 192.168.103.121/2.3.4.5 -f ". It produces the following output: [root@192 rtpproxy]# /usr/local/bin/rtpproxy -l 192.168.103.121/2.3.4.5 -f rtpproxy started, pid 22125 received command "UIE 9D740CB7-18A4-40B2-A96D-13FC3C5B27D3(a)192.168.103.189 192.168.103.189 49156 207860870326595;1" new session 9D740CB7-18A4-40B2-A96D-13FC3C5B27D3(a)192.168.103.189, tag 207860870326595;1 requested, type strong new session on a port 35000 created, tag 207860870326595;1 pre-filling caller's address with 192.168.103.189:49156 sending reply "35000 2.3.4.5 " received command "L 9D740CB7-18A4-40B2-A96D-13FC3C5B27D3(a)192.168.103.189 1.2.3.4 49154 207860870326595;1 5364140283;1" lookup on ports 35000/35000, session timer restarted pre-filling callee's address with 1.2.3.4:49154 sending reply "35000 192.168.103.121 In my openser.cfg I'm not really checking wheter a client is really natted, but I think it shouldn't be a problem to assume, that all clients are behind a NAT? I attached the openser.cfg to this mail (real public IP is changed to 2.3.4.5). Do you have any ideas how to fix this problem? Any help would be greatly appreciated! Thanks, Christian # $Id: nat-rtpproxy.cfg 51 2006-01-31 13:28:04Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ #/usr/local/bin/rtpproxy -l 192.168.103.121/2.3.4.5 -f debug=6 fork=yes log_stderror=no log_facility=LOG_LOCAL1 listen=192.168.103.121 # Internal IP listen=2.3.4.5 # External IP port=5060 children=4 dns=no rev_dns=no #set module path mpath="/usr/local/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "mi_fifo.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://xxx:yyy@192.168.103.122/openser") 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", "unix:/home/sys/bin/rtpproxy.sock") modparam("nathelper", "received_avp", "$avp(s:rcv)") modparam("registrar", "received_avp", "$avp(s:rcv)") modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 6) modparam("rr", "enable_full_lr", 1) route { setflag(1); # log this msg # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # 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"); return; }; if (method=="INVITE") { setflag(6); force_rport(); fix_nated_contact(); if (dst_ip == 192.168.103.121){ force_rtp_proxy("faie"); } else { force_rtp_proxy("faei"); } }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); return; }; if (method=="ACK") { route(1); return; } else if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; 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 # ---------------------------------------------------------------- setflag(6); fix_nated_register(); force_rport(); sl_send_reply("100", "Trying"); if (!save("location")) { log(1,"Error :-(.\n"); sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- setflag(6); lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- force_rport(); fix_nated_contact(); if (dst_ip == 192.168.103.121){ log(1,"INVITE 4 intern extern\n"); force_rtp_proxy("fie"); } else { log(1,"INVITE 4 extern intern\n"); force_rtp_proxy("fei"); } } onreply_route[1] { force_rtp_proxy("f"); fix_nated_contact(); }

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[OpenSER-Users] question with acl using openserctl and sermyadmin
by mark morreny 21 May '08

21 May '08

Hi, I am hoping someone can help me our with associating user with group in openser. If I use the command line: Question 1: debian:/home/ser# openserctl acl grant 1006 LD WARNING: Invalid privilege: acl 'LD' ignored debian:/home/ser# What is wrong with the syntax? Question 2: If I use sermyadmin to associate user to groups, I get the following error: org.codehaus.groovy.runtime.InvokerInvocationException: org.springframework.dao.DataIntegrityViolationException: could not insert: [UserGroup]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:290) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationDispatcher.invoke(ApplicationDispatcher.java:630) at org.apache.catalina.core.ApplicationDispatcher.processRequest(ApplicationDispatcher.java:436) at org.apache.catalina.core.ApplicationDispatcher.doForward(ApplicationDispatcher.java:374) at org.apache.catalina.core.ApplicationDispatcher.forward(ApplicationDispatcher.java:302) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.jsecurity.web.servlet.WebInterceptorFilter.doFilterInternal(WebInterceptorFilter.java:106) at org.jsecurity.web.servlet.SecurityContextFilter.doFilterInternal(SecurityContextFilter.java:93) at org.jsecurity.web.servlet.OncePerRequestFilter.doFilter(OncePerRequestFilter.java:106) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:233) at org.apache.catalina.core.StandardContextValve.invoke(StandardContextValve.java:175) at org.apache.catalina.core.StandardHostValve.invoke(StandardHostValve.java:128) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:102) at org.apache.catalina.core.StandardEngineValve.invoke(StandardEngineValve.java:109) at org.apache.catalina.connector.CoyoteAdapter.service(CoyoteAdapter.java:286) at org.apache.coyote.http11.Http11Processor.process(Http11Processor.java:844) at org.apache.coyote.http11.Http11Protocol$Http11ConnectionHandler.process(Http11Protocol.java:583) at org.apache.tomcat.util.net.JIoEndpoint$Worker.run(JIoEndpoint.java:447) at java.lang.Thread.run(Thread.java:595) Caused by: org.springframework.dao.DataIntegrityViolationException: could not insert: [UserGroup]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] at UserController$_closure6_closure11.doCall(UserController.groovy:111) at UserController$_closure6.doCall(UserController.groovy:101) at UserController$_closure6.doCall(UserController.groovy) ... 29 more Caused by: org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] at $Proxy3.saveOrUpdate(Unknown Source) ... 32 more Caused by: com.mysql.jdbc.exceptions.MySQLIntegrityConstraintViolationException: Duplicate entry '1005-192.168.1.104-' for key 2 at com.mysql.jdbc.SQLError.createSQLException(SQLError.java:1011) at com.mysql.jdbc.SQLError.createSQLException(SQLError.java:956) at com.mysql.jdbc.MysqlIO.checkErrorPacket(MysqlIO.java:3491) at com.mysql.jdbc.MysqlIO.checkErrorPacket(MysqlIO.java:3423) at com.mysql.jdbc.MysqlIO.sendCommand(MysqlIO.java:1936) at com.mysql.jdbc.MysqlIO.sqlQueryDirect(MysqlIO.java:2060) at com.mysql.jdbc.ConnectionImpl.execSQL(ConnectionImpl.java:2542) at com.mysql.jdbc.PreparedStatement.executeInternal(PreparedStatement.java:1734) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:209) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:1937) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:1922) at org.apache.tomcat.dbcp.dbcp.DelegatingPreparedStatement.executeUpdate(DelegatingPreparedStatement.java:102) ... 33 more What is wrong with my setup of sermyadmin? Thank you in advance for your help. Regards, Mark

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[OpenSER-Users] Help, for configure my client (Xlite).
by ZaidKa 21 May '08

21 May '08

1 0

[OpenSER-Users] UDP Buffer Size / OpenSER / Linux
by Carsten Bock 21 May '08

21 May '08

Hi, short question: Can anybody give me a hint regarding the Linux UDP-Buffer size? Are there any known good values for this or is the default value (128k) of Linux sufficient? I've read, regarding other SIP-Systems (namely IBM SIP Servlets and Communigate pro), the recommendation to increase the UDP-Buffer-size to 3 or even 4 MByte.... Are there any recommended / known best values for heavy load OpenSER Installations? Thanks in advance, Carsten Reference: http://publib.boulder.ibm.com/infocenter/wasinfo/v6r1/index.jsp?topic=/com.…

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[OpenSER-Users] Why can't I "append_branch()" in FAILURE_ROUTE if reply is 6XX ??
by Iñaki Baz Castillo 21 May '08

21 May '08

Hi, when I receive a 480 or 603 from a local user I run "on_failure_route" and create new branch to forward the INVITE to a media server in which I reply an early-media "The number you are calling is not available now...". It works perfectly if the error is 4XX but it doesn't work if it's a 6XX. I get this error: --------------------- ## onreply_route[ON_REPLY_TO_USER]: 603: Decline ## failure_route[ON_FAILURE_TO_USER] --- TO_MEDIA_SERVER: Destino user_not_available May 21 12:25:25 [8502] ERROR:tm:t_forward_nonack: discarding fwd for a cancelled/6xx transaction May 21 12:25:25 [8502] ERROR:tm:w_t_relay: t_forward_nonack failed ---------------------- My code is very simple: ---------------------- failure_route[ON_FAILURE_TO_USER] { if (t_check_status("480|603")) { $rU = "user_not_available"; route(ROUTE_TO_MEDIA_SERVER); exit; } } route[ROUTE_TO_MEDIA_SERVER] { rewritehostport("MEDIA_SERVER_IP:MEDIA_SERVER_PORT"); append_branch(); t_relay(); } --------------------- It seems that if the reply is a 6XX then OpenSer can't generate a new_branch in failure_route. May 21 12:25:25 [8502] ERROR:tm:t_forward_nonack: discarding fwd for a cancelled/6xx transaction Why? doesn't it make sense to forward to some destination even if the negative reply is a 4XX or 6XX? Thanks a lot for any help. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Call failed 403 Forbiden
by luzango mfupe 20 May '08

20 May '08

Hi mates, Everytime i do try to make my 2 xlite clients talk (which i correctly added them into the database), i encountered with the same problm, Openser perfoms authentication and return call failed 403 forbiden signal. My setup comprise of the first box with openser 1.3 and mediaproxy the second box with Mysql and two Xlite clients in two other boxes all are in an internal network, as far as am concerned my NetAdmin have already opened ports 5060 and 3306 for me. I need your right direction on this probm. below is my ngrep snapshot mzee:/# ngrep -d eth1 -W byline port 5060 interface: eth1 (168.172.200.0/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) # U 168.172.200.70:1824 -> 168.172.200.87:5060 INVITE sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87>SIP/2.0. Via: SIP/2.0/UDP 168.172.200.70:1824 ;branch=z9hG4bK-d87543-ef08fb62b30f1a54-1--d87543-. Max-Forwards: 70. Contact: <sip:dreamteam@168.172.200.70:1824>. To: "musk"<sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87>>. From: "dream"<sip:dreamteam@168.172.200.87 <sip%3Adreamteam(a)168.172.200.87> >;tag=af4bd714. Call-ID: OTAyNWJhOTdiZjBlNmUwNDYyZWI0YzdkM2JhZTUxMTU.. CSeq: 1 INVITE. Session-Expires: 95. Min-SE: 90. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: timer. User-Agent: X-Lite release 1011s stamp 41150. Content-Length: 426. . v=0. o=- 9 2 IN IP4 168.172.200.70. s=CounterPath X-Lite 3.0. c=IN IP4 168.172.200.70. t=0 0. m=audio 52166 RTP/AVP 107 119 100 106 0 105 98 8 101. a=alt:1 1 : uZB2dYm+ NKBRK8Ep 168.172.200.70 52166. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:119 BV32-FEC/16000. a=rtpmap:100 SPEEX/16000. a=rtpmap:106 SPEEX-FEC/16000. a=rtpmap:105 SPEEX-FEC/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. # U 168.172.200.87:5060 -> 168.172.200.70:1824 SIP/2.0 403 Forbidden. Via: SIP/2.0/UDP 168.172.200.70:1824 ;branch=z9hG4bK-d87543-ef08fb62b30f1a54-1--d87543-. To: "musk"<sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87> >;tag=9a17bd4180f96d7136f8b30b25c6947e.d7e9. From: "dream"<sip:dreamteam@168.172.200.87 <sip%3Adreamteam(a)168.172.200.87> >;tag=af4bd714. Call-ID: OTAyNWJhOTdiZjBlNmUwNDYyZWI0YzdkM2JhZTUxMTU.. CSeq: 1 INVITE. Server: OpenSER (1.3.0-notls (i386/linux)). Content-Length: 0. . # U 168.172.200.70:1824 -> 168.172.200.87:5060 ACK sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87> SIP/2.0. Via: SIP/2.0/UDP 168.172.200.70:1824 ;branch=z9hG4bK-d87543-ef08fb62b30f1a54-1--d87543-. To: "musk"<sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87> >;tag=9a17bd4180f96d7136f8b30b25c6947e.d7e9. From: "dream"<sip:dreamteam@168.172.200.87 <sip%3Adreamteam(a)168.172.200.87> >;tag=af4bd714. Call-ID: OTAyNWJhOTdiZjBlNmUwNDYyZWI0YzdkM2JhZTUxMTU.. CSeq: 1 ACK. Content-Length: 0. WBR, LU. -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA

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[OpenSER-Users] Wesip question
by mark morreny 20 May '08

20 May '08

Hi, Is it possible to implement a media proxy using wesip? If so, is there a reference info available? Other than wesip, is there any other tool that can be used to extend Openser possibility through a scripting language? Thanks, Mark

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[OpenSER-Users] db_extra: the avp value does not correspond to the real value
by Bruno Machado 20 May '08

20 May '08

Hi friends. OK. People... I'm almost getting crazy. I don't understand what is happening. I already looked for an answer, but I didn't find nothing yet. I am recording 8 extra values in ACC, but the SQL command is using a weird value for one of them. So, I removed 6 of them to facilitate the analyze. Look at the code: . . modparam("acc", "db_extra", "valueA=$avp(i:15); valueB=$avp(i:16)") . . . avp_write("i:15","$avp(i:15)"); avp_write("i:16","$avp(i:16)"); if(!t_relay()) { sl_reply_error(); exit; }; avp_print(); exit; OK. Looking at these lines, I would say: "the sql INSERT command will use the value 15 for avp:15 and 16 for avp:16". Now, look at the result of "avp_print": 2(11112) INFO:avpops:print_avp: p=0xb3b4a900, flags=0x0000 2(11112) INFO: id=<16> 2(11112) INFO: val_int=<16> 2(11112) INFO:avpops:print_avp: p=0xb3b4a8e8, flags=0x0000 2(11112) INFO: id=<15> 2(11112) INFO: val_int=<15> Nice. AVP 15 has the value 15 and the AVP 16 has 16. But something is happening.... It is the insert command: insert into acc (from_uri,to_uri,sip_method...............valueA,valueB,time) values .........,'16','16','2008-05-19 12:48:38'). 16 twice!! Why? This question is running in loop inside my head. It should be "....'15','16','2008-05-19 12:48:38'.....". The openser code above is executed just once by the INVITE. I can bet this error is the like when we sum "2 and 3" and the result is "6". You look at the equation, but you don't see it at all. And, when you ask your younger brother to see, he says: "you are stupid... 2+3=5". I never had this problem. I use several avps to log in the database and it is the first time that it happens. You can see that I'M NOT CHANGING the values of the avps. Am I doing a stupid thing? Or, maybe, am I being blind? By the way, I'm using version 1.1.1. Tks people. OK. Apparently, there's nothing wrong. Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/

2 1

[OpenSER-Users] DNS query
by Yazid Hadj Said 20 May '08

20 May '08

Hi, I want to perform a DNS query with the Record-Route header's content, and replace this content by the response (replacing "domain.com" by xxx.xxx.xxx.xxx). someone knows how? Regards, Yazid

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[OpenSER-Users] OPENSER Workshop, Berlin, May 26-27
by Daniel-Constantin Mierla 20 May '08

20 May '08

Hello, for those interested, there will be two OPENSER sessions hosted by Asterisk Tag Conference, Berlin, Germany, May 26-27, 2008. The event gathers an important group of people from VoIP area, including the inventors of Asterisk, Adhearsion, PGP, Yate, ..., for more details visit: http://asterisk-tag.org In the first day, the OPENSER presentation will focus on how to use the proxy to increase the capacity of Asterisk-based platforms, the second day will host 2 hours workshop targeting to reveal where OPENSER fits in IP communication services, Voice over IP and beyond, and to give the starting point to use it. FYI: the following days is the Linux Tag conference/exhibition, also in Berlin: http://www.linuxtag.org . A preview notice -- on the 28th of May will be an OPENSER dinner (more beer than food, as it happens in Germany) -- place and time will be announced very soon. Cheers, Daniel -- CEO ASIPTO http://www.asipto.com

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[OpenSER-Users] Authentication with LDAP databse
by Ahmed Huraimel 20 May '08

20 May '08

Hello, i am successfully integrated an openLDAP server with my openSER SIP proxy server. however i am facing a security problem. let me explain it briefly. **** Successful Registration with password save as clear in openLDAP DB **** * # Configration # - user name was stored in clear in openSER database - modparam("auth", "calculate_ha1", 1) which means the server will assume that the "password_spec" pseudo-variable contains plaintext passwords and it will calculate HA1 strings on the fly. # Senario # - after the UAC receives Authentication request he will build the response = MD5(username + MD5(passowrd) + realm + nonce) - then the server will build the challenge by searching the the user in the database and retrieving the password in clear then hash the password with MD5 build the challenge such that challenge=MD5(username + MD5(passowrd) + realm + nonce) . . - by comparing the the response the with the challenge the user will be authenticated. - *it works * **** Successful Registration with password save as MD5 in openLDAP DB **** * # Configration # - user name was stored in MD5 in openSER database - modparam("auth", "calculate_ha1", 0) which means the server assumes the pseudo-variable contains the HA1 strings directly and will not calculate them. # Senario # - after the UAC receives Authentication request he will build the response = MD5(username + MD5(password) + realm + nonce) - then the server will build the challenge by searching the the user in the database and retrieving the password in MD5 then challenge such that challenge=MD5(username + MD5(password) + realm + nonce) . - by comparing the the response the with the challenge the user will be authenticated. - *401 unauthorized !!!! ***** CONCLUSION **** *there for possible scenarios: 1- password clear + calculate_ha1= 0 ==> 401 unauthorized !!!! 2- password clear + calculate_ha1= 1 ==> Authorized 3- password MD5 + calculate_ha1= 0 ==> 401 unauthorized !!!! 4- password MD5 + calculate_ha1= 1 ==> 401 unauthorized !!!! * * ----------------------------------------------------------------------------------------- * asuumptions: *1- the password might be not hashed. if so then why modparam("auth", "calculate_ha1", ) used? does it mean that the password might be received hashed or not? *2- in scenario(2) the sip server hash the password by setting calculate_ha1= 1. if the password is already hashed in the database then scenario(3) should work unless there is a conflict with the hash. is this might be related to hash type or size?* or something else that i do not know!!! *question*: 1- why scenario(3) does not work? where might be the problem? 2- what to do if i want to change the hash algorithm used? for example i need to SSH1 instead of MD5 because nowadays MD5 is proved to be weak algorithm regards, Ahmed ALALI

2 3

[OpenSER-Users] How to install OpenSER on VMware?
by Habib, Yeasin 20 May '08

20 May '08

Hi, I am successful till the addition of voip overlay. After that I have also include the line source /usr/portage/local/layman/make.conf in /etc/make.conf. Now when I try to install openser using the command ACCEPT_KEYWORDS=!x86 emerge openser, every time I got the following reply back from the console emerge: there are no ebuilds to satisfy "openser" Does anybody know how I can solve this problem and install and configure OpenSER on top my OpenIMSCore Best regards Yeasin Habib You can install layman, add voip overlay and install OpenSER using standard gentoo way (emerge openser) 1) start OpenIMS image 2) Login 3) emerge --sync. On this step emerge will download information about all packages. Maybe about 200 mb 4) emerge layman 5) edit make.conf 6) add voip overlay (layman -a voip) 7) install openser (ACCEPT_KEYWORDS=~x86 emerge openser) On Tue, Apr 22, 2008 at 9:53 AM, Rob <otrebor79 at alice.it<https://lists.berlios.de/mailman/listinfo/openimscore-users>> wrote: > Hi, > I've downloaded this image of OpenIMSCore for VMware > http://www.digilo.com/openims/files/OpenIMS.2008-03-30.r545.rar > I've installed and runned it. > Now I want install on it a Presence Server so I've downloaded the > packages of the OpenSER 1.2.0 version 1.20. > Now, how can I install it onto che VMware image? > > Rob > > _______________________________________________ > OpenIMSCore-Users mailing list > OpenIMSCore-Users at lists.berlios.de<https://lists.berlios.de/mailman/listinfo/openimscore-users> > https://lists.berlios.de/mailman/listinfo/openimscore-users > -- Vladimir Romanov

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[OpenSER-Users] Newbie quesion on configuring Openser as SIP Registrar
by Pete Kay 20 May '08

20 May '08

Hi, I followed instruction from http://www.jeremy-mcnamara.com/2007/03/28/how-to-configure-openser-sip-regi… setup Openser. #modparam("nathelper", "rtpproxy_disable", 1) #modparam("registrar", "nat_flag", 6) #modparam("registrar", "use_domain", 0) The above three lines will cause the server to give error: May 20 23:29:51 [2784] CRITICAL:core:yyerror: parse error in config file, line 93, column 19-20: Can't set module parameter I saw from the document that those are valid arguments, and I don't understand why Openser complain about those parameters. Then, I comment out those three lines, and get another errors: May 20 23:48:25 [3218] CRITICAL:mediaproxy:mod_init: bad config - nat ping enabled, but no nat bflag set in the usrloc module May 20 23:48:25 [3218] ERROR:core:init_mod: failed to initialize module mediaproxy May 20 23:48:25 [3218] ERROR:core:main: error while initializing modules I am very new to openser. Any help or suggestion will be greatly appreciated. Here is my openser.cfg ####### Modules Section ######## #set module path mpath="/usr/local/lib/openser/modules/" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/uac.so" loadmodule "/usr/local/lib/openser/modules/speeddial.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- #modparam("usrloc|auth_db|domain|speeddial|acc", "db_url", "mysql://localhost:openserrw@localhost/openser") modparam("auth_db", "db_url","mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "use_domain", 0) modparam("domain", "db_mode", 1) #modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 60) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 0) modparam("registrar", "default_expires", 60) modparam("registrar", "min_expires", 30) #modparam("registrar", "nat_flag", 6) #modparam("registrar", "use_domain", 0) modparam("rr", "enable_full_lr", 1) #modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") #modparam("auth", "rpid_avp", "s:rpid")

1 0

[OpenSER-Users] problem in presence module
by Amit Vijayvargiya 20 May '08

20 May '08

Hi, i m add presence module in openser 1.3.0 it's working fine.... but some time it given error in stoped.... see the logs... May 20 14:58:09 [6432] INFO:pua_usrloc:ul_publish: should not send ul publish May 20 14:58:09 [6435] ERROR:presence:handle_subscribe: Missing or unsupported event header field value May 20 14:58:09 [6435] ERROR:presence:handle_subscribe: event= message-summary May 20 14:58:09 [6433] ERROR:mysql:db_mysql_submit_query: driver error: Duplicate entry 'sip:userA@domain.com-userB-domain.com-presence' for key 2 May 20 14:58:09 [6433] ERROR:mysql:db_mysql_insert: error while submitting query May 20 14:58:09 [6433] ERROR:presence:insert_db_subs_auth: in sql insert May 20 14:58:09 [6433] ERROR:presence:handle_subscribe: while inserting record in watchers table May 20 14:58:09 [6433] ERROR:mysql:db_mysql_submit_query: driver error: Malformed packet May 20 14:58:09 [6433] ERROR:mysql:db_mysql_query: error while submitting query May 20 14:58:09 [6433] ERROR:presence:get_subs_db: while querying database May 20 14:58:09 [6433] ERROR:presence:get_subs_dialog: getting dialogs from database May 20 14:58:09 [6443] CRITICAL:core:receive_fd: EOF on 10 May 20 14:58:09 [6429] INFO:core:handle_sigs: child process 6435 exited by a signal 11 May 20 14:58:09 [6429] INFO:core:handle_sigs: core was generated May 20 14:58:09 [6429] INFO:core:handle_sigs: terminating due to SIGCHLD May 20 14:58:09 [6442] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6432] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6437] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6433] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6434] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6436] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6439] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6440] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6441] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6443] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6438] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6429] NOTICE:presence:destroy: destroy module ... plz help me wat is the problem???? Thanks & Regards, Amit Vijayvargiya

1 0

[OpenSER-Users] problem in audio when using media proxy
by Krunal Patel 20 May '08

20 May '08

Hi, We are using openser with media proxy and radiator. We're having audio problems when we send calls to mediatrix gateway for termination. Internal calls work fine. We hear audio both side and we can see the session and bytes transfer properly in media proxy monitoring webpage. When we send calls to mediatrix gateway for termination, calls connect properly but we don't hear audio. We see inactive session in mediaproxy status page with no bytes transferring. One doubt I have is it is not entering reply route when it receives replies like 183, 180, etc. Has anyone encountered similar issue? How can we fix it? Thanks in advance! -- regards, Krunal Patel

3 4

[OpenSER-Users] openser.vim
by Daniel-Constantin Mierla 19 May '08

19 May '08

Hello, just found vim syntax specs for openser configuration file. vim fans might find it interesting: http://www.vim.org/scripts/script.php?script_id=2242 All thanks to Stanislaw Pitucha. Cheers, Daniel -- http://www.asipto.com

2 1

[Serusers] Release schedule for 2.0?
by Marcus Better 18 May '08

18 May '08

Hi, I'm working on packaging ser for Debian. We currently ship 0.9.7, but have also packaged a 2.0 snapshot which seems to be working well. We now need to decide which version to ship in the upcoming "lenny" release of Debian. Do you have any information that you can share about the release plans for SER 2.0? This would help us a lot in making this decision. Thanks, Marcus

3 3

Re: [OpenSER-Users] null socket list
by Daniel-Constantin Mierla 17 May '08

17 May '08

On 05/17/08 00:53, Michael Barinek wrote: > mysql works as well. Thanks again. perfect, thanks for testing. Daniel > > > > Michael Barinek wrote: >> I think that worked, Thank you! I'm setting up mysql and will have >> more info soon. Thanks again. >> >> >> Daniel-Constantin Mierla wrote: >>> Hello, >>> >>> found a similar case and hopefully the fix I just committed works. >>> You have to install branch 1.3 from SVN and then test. Let me know >>> if it is working for you. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 03/26/08 22:18, Michael Barinek wrote: >>>> I have openser-1.3.1-notls installed although am unable to start >>>> openser. Any suggestions/ideas regarding the below error? Any >>>> existing docs/links would be great. >>>> >>>> Thanks >>>> >>>> /// >>>> >>>> Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not >>>> rev. resolve 10.138.86.56 >>>> Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not >>>> rev. resolve 10.138.87.100 >>>> Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not >>>> rev. resolve 10.37.129.2 >>>> Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not >>>> rev. resolve 10.211.55.2 >>>> Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not >>>> rev. resolve 10.138.86.56 >>>> Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not >>>> rev. resolve 10.138.87.100 >>>> Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not >>>> rev. resolve 10.37.129.2 >>>> Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not >>>> rev. resolve 10.211.55.2 >>>> Listening on >>>> udp: 127.0.0.1 [127.0.0.1]:5060 >>>> udp: 10.138.86.56 [10.138.86.56]:5060 >>>> udp: 10.138.87.100 [10.138.87.100]:5060 >>>> udp: 10.37.129.2 [10.37.129.2]:5060 >>>> udp: 10.211.55.2 [10.211.55.2]:5060 >>>> tcp: 127.0.0.1 [127.0.0.1]:5060 >>>> tcp: 10.138.86.56 [10.138.86.56]:5060 >>>> tcp: 10.138.87.100 [10.138.87.100]:5060 >>>> tcp: 10.37.129.2 [10.37.129.2]:5060 >>>> tcp: 10.211.55.2 [10.211.55.2]:5060 >>>> Aliases: >>>> tcp: my.machine.name:5060 >>>> udp: my.machine.name:5060 >>>> >>>> Mar 20 08:59:14 [18124] WARNING:core:main: no fork mode and more >>>> than one listen address found(will use only the first one) >>>> Mar 20 08:59:14 [18124] INFO:core:init_tcp: using kqueue as the TCP >>>> io watch method (auto detected) >>>> Mar 20 08:59:14 [18124] NOTICE:core:main: version: openser >>>> 1.3.1-notls (i386/darwin) >>>> Mar 20 08:59:14 [18124] INFO:core:main: using 32 Mb shared memory >>>> Mar 20 08:59:14 [18124] INFO:core:main: using 1 Mb private memory >>>> per process >>>> Mar 20 08:59:14 [18124] INFO:sl:mod_init: Initializing StateLess >>>> engine >>>> Mar 20 08:59:14 [18124] INFO:tm:mod_init: TM - initializing... >>>> Mar 20 08:59:14 [18124] CRITICAL:tm:uac_init: null socket list >>>> Mar 20 08:59:14 [18124] ERROR:tm:mod_init: uac_init failed >>>> Mar 20 08:59:14 [18124] ERROR:core:init_mod: failed to initialize >>>> module tm >>>> Mar 20 08:59:14 [18124] ERROR:core:main: error while initializing >>>> modules >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)lists.openser.org >>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >> -- http://www.asipto.com

1 0

[OpenSER-Users] auth_radius:radius_authorize_sterman: rc_auth failed
by Simon J Xuereb 16 May '08

16 May '08

Hi, I am trying to configure openser + freeradius for authentication & accounting. I have my freeradius configured with openldap and it's working successfully. tested with radclient -f digest 127.0.0.1:1812 auth testing123 results are successfull. However I cannot get it to work with openser. I see nothing happening in my "/usr/sbin/radius -X" output coming from openser when i try to register. My enviroment was tested on Fedora 8 and Fedora 9 both with rpm's and same results. auth_radius:radius_authorize_sterman: rc_auth failed no output is seen in my /usr/sbin/radius -X Below please find my openser.cfg & radiusclient.conf & servers ######## radiusclient.conf # General settings # specify which authentication comes first respectively which # authentication is used. possible values are: "radius" and "local". # if you specify "radius,local" then the RADIUS server is asked # first then the local one. if only one keyword is specified only # this server is asked. auth_order radius,local # maximum login tries a user has login_tries 4 # timeout for all login tries # if this time is exceeded the user is kicked out login_timeout 60 # name of the nologin file which when it exists disables logins. # it may be extended by the ttyname which will result in # a terminal specific lock (e.g. /etc/nologin.ttyS2 will disable # logins on /dev/ttyS2) nologin /etc/nologin # name of the issue file. it's only display when no username is passed # on the radlogin command line issue /etc/radiusclient-ng/issue # RADIUS settings # RADIUS server to use for authentication requests. this config # item can appear more then one time. if multiple servers are # defined they are tried in a round robin fashion if one # server is not answering. # optionally you can specify a the port number on which is remote # RADIUS listens separated by a colon from the hostname. if # no port is specified /etc/services is consulted of the radius # service. if this fails also a compiled in default is used. authserver 127.0.0.1 authserver 10.0.0.10 authserver localhost # RADIUS server to use for accouting requests. All that I # said for authserver applies, too. # acctserver 127.0.0.1:1813 # file holding shared secrets used for the communication # between the RADIUS client and server servers /etc/radiusclient-ng/servers # dictionary of allowed attributes and values # just like in the normal RADIUS distributions #dictionary /usr/share/radiusclient-ng/dictionary dictionary /etc/radiusclient-ng/dictionary.openser # program to call for a RADIUS authenticated login login_radius /usr/sbin/login.radius # file which holds sequence number for communication with the # RADIUS server seqfile /var/run/radius.seq # file which specifies mapping between ttyname and NAS-Port attribute mapfile /etc/radiusclient-ng/port-id-map # default authentication realm to append to all usernames if no # realm was explicitly specified by the user # the radiusd directly form Livingston doesnt use any realms, so leave # it blank then default_realm # time to wait for a reply from the RADIUS server radius_timeout 10 # resend request this many times before trying the next server radius_retries 3 # local address from which radius packets have to be sent bindaddr * # LOCAL settings # program to execute for local login # it must support the -f flag for preauthenticated login login_local /bin/login ######################################################## #/etc/radiusclient-ng/servers #Server Name or Client/Server pair Key #---------------- --------------- #portmaster.elemental.net hardlyasecret #portmaster2.elemental.net donttellanyone localhost testing123 127.0.0.1 testing123 10.0.0.10 testing123 ####################################################### ##########openser.cfg # -- rr params -- debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 port=5060 # ------------------ module loading ---------------------------------- mpath="/usr/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "avpops.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "uri.so" loadmodule "auth.so" loadmodule "auth_radius.so" loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc","db_url","mysql://openser:openser@localhost/openser") modparam("usrloc", "db_mode", 2) # -- acc params -- modparam("auth_radius","radius_config","/etc/radiusclient-ng/radiusclient.conf") # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # authenticate registers if (method=="REGISTER") { if (!radius_www_authorize("")) { www_challenge("", "0"); exit; }; save("location"); exit; }; route(1); } # generic forward route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } # Thanks for your help SJX

1 0

Re: [OpenSER-Users] problem - 484 address incomplete for the correct TO NUMBER
by Kaeman Chris 16 May '08

16 May '08

Like i already took this as a benefit of doubt. They used G711u for a California calls and it was successful and used G711u NSE to the NewYork and the call failed with 484. so i asked them to use the G711u for NewYork and not G711u NSE still the call failed with G711u (excluding NSE). I enquired with the provider already that they support G711u but not G711u NSE. but still with G711u they failed and that too with 484. thats really wierd and i am stuck up in the middle. KC LetMeKnow <sunkara.raviprakash.feb14(a)gmail.com> wrote: Hi Chris, This Problem is happened to me, when i used EyeBeam---> OpenSER--> Asterisk as PSTN Gateway, Asterisk is seems to send 484, this is due to the imcompartable of the Codecs in Eyebean and Asterisk gateway........................................... Check out.... logs from these customer signaling ip-addr - 23.21.76.23 customer media ip-addr - 231.43.25.1 supplier media ip-addr - 43.25.32.85 supplier signaling ip-addr - 13.45.213.245 my signaling ip-addr - 123.123.1.234 supplier have producing this t SIP headers and found no issues. Also, when i use the same called number and make the calls through the same provider it works from my end with 200 OK. its really interesting and seems to be some issues with the headers. I am using openser and following are the logs. sorry - its too long, but i need someones help to figure out whats going on really. Also, following is the definition for the ips and telephone numbers. FROM number - 17592034561 TO number - 19846572302 customer signaling ip-addr - 23.21.76.23 customer media ip-addr - 231.43.25.1 supplier media ip-addr - 43.25.32.85 supplier signaling ip-addr - 13.45.213.245 my signaling ip-addr - 123.123.1.234 --------------------------------------------------------------------- logs as below; Y#HE& _ d E pl@ 7 ! @ V)ÇèNÃ Ä Ä |MàINVITE sip:+17592034561@123.123.1.234 SIP/2.0 Record-Route: <sip:23.21.76.23;lr;ftag=5d411c09;did=3b7.c57bab42> To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Contact: <sip:+19846572302@43.25.32.85:8394> Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3007n stamp 17816 Authorization: Digest username="19846572302",realm="23.21.76.23",nonce="48235aef53d38bbdccb25be97f6ac3599ce752eb",uri="sip:+17592034561@23.21.76.23",response="2993367e31be586485909086d44a328d",algorithm=MD5 Content-Length: 270 v=0 o=- 10556151 10556316 IN IP4 43.25.32.85 s=eyeBeam c=IN IP4 231.43.25.1 t=0 0 m=audio 54864 RTP/AVP 18 101 a=alt:1 1 : 7DBA1F5A 00000092 10.0.0.1 8360 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=direction:active a=nortpproxy:yes ÄY#H¯' ¨ ¨ 0H% E èÌ@ @ £ ÇèNÃ@ V) Ä Ä ¯ SIP/2.0 100 Trying To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 ÄY#H ; ® ® 0H% E èÍ@ @ ¢ÿÇèNÃ@ V) Ä Ä ¯ SIP/2.0 100 Giving a try To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 ÄY#H; M M 0H% E =èÎ@ @ G;ÇèNÃ@®N Ä Ä ) ÑINVITE sip:17592034561@13.45.213.245 SIP/2.0 Record-Route: <sip:123.123.1.234;lr=on;ftag=5d411c09;osp-o=t5198095097179410177_s23.21.76.23_a1210276292_c1> Record-Route: <sip:23.21.76.23;lr;ftag=5d411c09;did=3b7.c57bab42> To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Contact: <sip:+19846572302@43.25.32.85:8394> Max-Forwards: 68 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3007n stamp 17816 Authorization: Digest username="19846572302",realm="23.21.76.23",nonce="48235aef53d38bbdccb25be97f6ac3599ce752eb",uri="sip:+17592034561@23.21.76.23",response="2993367e31be586485909086d44a328d",algorithm=MD5 Content-Length: 270 v=0 o=- 10556151 10556316 IN IP4 43.25.32.85 s=eyeBeam c=IN IP4 231.43.25.1 t=0 0 m=audio 54864 RTP/AVP 18 101 a=alt:1 1 : 7DBA1F5A 00000092 10.0.0.1 8360 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=direction:active a=nortpproxy:yes ÄY#HÖº Ê Ê _ d E º** 2 Ws@®NÇèNÃ Ä Ä ¦ SIP/2.0 100 Trying Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 To: <sip:+17592034561@23.21.76.23>;tag=gK0cf56d6a Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Content-Length: 0 ÄY#HY Ö Ö _ d E Æ+ 2 V@®NÇèNÃ Ä Ä ² SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 To: <sip:+17592034561@23.21.76.23>;tag=gK0cf56d6a Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Content-Length: 0 ÄY#H v v 0H% E fè×@ @ K ÇèNÃ@®N Ä Ä R úACK sip:17592034561@13.45.213.245 SIP/2.0 Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Call-ID: 5d1bb15abd63917a To: <sip:+17592034561@23.21.76.23>;tag=gK0cf56d6a CSeq: 2 ACK User-Agent: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 --------------------------------- Download prohibited? No problem. CHAT from any browser, without download. _______________________________________________ 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 91-9999882776 --------------------------------- Check out the all-new face of Yahoo! India. Click here. --------------------------------- Check out the all-new face of Yahoo! India. Click here.

1 0

[OpenSER-Users] Implementing international destination allow/deny system with PDT module (patch included for 1.3.X)
by Iñaki Baz Castillo 16 May '08

16 May '08

Hi, I've implemented a system to control international destinations (allow/deny) using "PDT" module (based on prefix). In OpenSer 1.4 there will be a new module "userblacklist" which can handle this issue in a more ellegant way, but for OpenSer 1.3.X we can use "pdt" module (with a little patch to pdt.c) to implement it. I've submitted the patch and the how-to: https://sourceforge.net/tracker/index.php?func=detail&aid=1965211&group_id=… Hope it could be useful. Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

1 0

[OpenSER-Users] Res: sermyadmin almost works
by Flavio Goncalves 16 May '08

16 May '08

Please, check the updated instructions to install at: www.sermyadmin.org Flavio ----- Mensagem original ---- De: whudson05 <whudson05(a)gmail.com> Para: users(a)lists.openser.org Enviadas: Quarta-feira, 14 de Maio de 2008 11:29:26 Assunto: Re: [OpenSER-Users] sermyadmin almost works I'm also at the same stage. I've looked at the mySQL queries and they appear to be correct. Have you added the user admin to subscriber? Lucas Catranis wrote: > > Anyone Get this setup correctly? I've been at it for a while and after > getting it to insert new users I cannot login as an admin. I'm not sure > where the problem is but Any help would be appreciated. > -- View this message in context: http://www.nabble.com/sermyadmin-almost-works-tp17219053p17232461.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com. _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users

2 1

[OpenSER-Users] problem - 484 address incomplete for the correct TO NUMBER
by Kaeman Chris 15 May '08

15 May '08

Hello, I am having the following problem. please help me to figure out the issue. I have my client calling me and getting 484 Address incomplete. I checked all the SIP headers and found no issues. Also, when i use the same called number and make the calls through the same provider it works from my end with 200 OK. its really interesting and seems to be some issues with the headers. I am using openser and following are the logs. sorry - its too long, but i need someones help to figure out whats going on really. Also, following is the definition for the ips and telephone numbers. FROM number - 17592034561 TO number - 19846572302 customer signaling ip-addr - 23.21.76.23 customer media ip-addr - 231.43.25.1 supplier media ip-addr - 43.25.32.85 supplier signaling ip-addr - 13.45.213.245 my signaling ip-addr - 123.123.1.234 --------------------------------------------------------------------- logs as below; Y#HE& _d E pl@ 7!@V)ÇèNÃÄÄ|MàINVITE sip:+17592034561@123.123.1.234 SIP/2.0 Record-Route: <sip:23.21.76.23;lr;ftag=5d411c09;did=3b7.c57bab42> To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Contact: <sip:+19846572302@43.25.32.85:8394> Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3007n stamp 17816 Authorization: Digest username="19846572302",realm="23.21.76.23",nonce="48235aef53d38bbdccb25be97f6ac3599ce752eb",uri="sip:+17592034561@23.21.76.23",response="2993367e31be586485909086d44a328d",algorithm=MD5 Content-Length: 270 v=0 o=- 10556151 10556316 IN IP4 43.25.32.85 s=eyeBeam c=IN IP4 231.43.25.1 t=0 0 m=audio 54864 RTP/AVP 18 101 a=alt:1 1 : 7DBA1F5A 00000092 10.0.0.1 8360 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=direction:active a=nortpproxy:yes ÄY#H¯' ¨ ¨ 0H% EèÌ@ @£ÇèNÃ@V)ÄÄ¯SIP/2.0 100 Trying To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 ÄY#H; ® ® 0H% EèÍ@ @¢ÿÇèNÃ@V)ÄÄ¯ SIP/2.0 100 Giving a try To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 ÄY#H; M M 0H% E=èÎ@ @G;ÇèNÃ@®NÄÄ) ÑINVITE sip:17592034561@13.45.213.245 SIP/2.0 Record-Route: <sip:123.123.1.234;lr=on;ftag=5d411c09;osp-o=t5198095097179410177_s23.21.76.23_a1210276292_c1> Record-Route: <sip:23.21.76.23;lr;ftag=5d411c09;did=3b7.c57bab42> To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Contact: <sip:+19846572302@43.25.32.85:8394> Max-Forwards: 68 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3007n stamp 17816 Authorization: Digest username="19846572302",realm="23.21.76.23",nonce="48235aef53d38bbdccb25be97f6ac3599ce752eb",uri="sip:+17592034561@23.21.76.23",response="2993367e31be586485909086d44a328d",algorithm=MD5 Content-Length: 270 v=0 o=- 10556151 10556316 IN IP4 43.25.32.85 s=eyeBeam c=IN IP4 231.43.25.1 t=0 0 m=audio 54864 RTP/AVP 18 101 a=alt:1 1 : 7DBA1F5A 00000092 10.0.0.1 8360 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=direction:active a=nortpproxy:yes ÄY#HÖº Ê Ê _d E º** 2Ws@®NÇèNÃÄÄ¦ SIP/2.0 100 Trying Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 To: <sip:+17592034561@23.21.76.23>;tag=gK0cf56d6a Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Content-Length: 0 ÄY#HY Ö Ö _d E Æ+ 2V@®NÇèNÃÄÄ² SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 To: <sip:+17592034561@23.21.76.23>;tag=gK0cf56d6a Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Content-Length: 0 ÄY#H v v 0H% Efè×@ @K ÇèNÃ@®NÄÄRúACK sip:17592034561@13.45.213.245 SIP/2.0 Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Call-ID: 5d1bb15abd63917a To: <sip:+17592034561@23.21.76.23>;tag=gK0cf56d6a CSeq: 2 ACK User-Agent: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 --------------------------------- Download prohibited? No problem. CHAT from any browser, without download.

1 0

[OpenSER-Users] OpenSER 1.3.2 is released
by Bogdan-Andrei Iancu 15 May '08

15 May '08

Hi, I'm happy to announce that OpenSER 1.3.2 is out - after almost 6 months from the first 1.3 release, 1.3.2 is available and it includes valuable experiences and feedback from operating 1.3.x in live platform. This means more stability, more reliability. This is a minor release, so it includes only fixes. That means that configuration and DB structure was not changes and it is fully compatible with all the other releases from 1.3 branch. A comprehensive compilation of fixes (thanks to Henning Westerholt) is available via the Changelog - see http://www.openser.org/pub/openser/1.3.2/ChangeLog For more about how to download this version, see http://www.openser.org/mos/view/News/NewsItem/OpenSER-1.3.2-is-released/ If people want to contribute with packages generated on different OS, please let me know - I will be glad to upload them on the repository. Regards, Bogdan

1 0

[OpenSER-Users] [RFC] - siptrace change
by Bogdan-Andrei Iancu 15 May '08

15 May '08

Hi, I would like to get your opinion/comments on some siptrace issue. In 1.3, a new capability was added in siptrace - to trace the ACK resulted from sending stateless replies. The problem I discovered with this is that it cannot be controlled by selecting the messages or transactions to be be traced. Usually you use siptrace to trace only certain transactions / messages - tracing all traffic is usually not a realistic option. So, with the stateless ACK, the siptrace module cannot apply the selection from script and trace them all, with no filtering at all. So, if you want to trace a traffic for a single user, all ACK going through the platform will be traced. This reduces the usability of the module and you get a lot of garbage tracing. My suggestion will be to disable stateless ACK tracing until a way to control/filter it is found. I'm asking this considering the upcoming 1.3.2 release from tomorrow. Regards, Bogdan

3 8

Re: [OpenSER-Users] problem in audio when using media proxy
by Krunal Patel 15 May '08

15 May '08

Hi, Actually I am new to openser. Here i am sending trace from openser. So please check out SDP. *************************************************************************************************************************************** Message Body: INVITE sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com>SIP/2.0 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;rport;branch=z9hG4bK7F77064E8D8FD3FD41779B0F67EF5C05 From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>> Contact: <sip:5555555555@122.xxx.xxx.35:5061> Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28079 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 291 v=0 o=5555555555 3961416264 3961416447 IN IP4 122.xxx.xxx.35 s=X-Lite c=IN IP4 122.xxx.xxx.35 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv 0(14396) parse_headers: flags=100 0(14396) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(14396) parse_headers: flags=80 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) parse_headers: flags=8000000 0(14396) #############################record_route############################### 0(14396) #######################After record_route########################3 0(14396) ##################################Before loose_route########################## 0(14396) parse_headers: flags=200 0(14396) DEBUG: get_hdr_body : content_length=291 0(14396) found end of header 0(14396) find_first_route: No Route headers found 0(14396) loose_route: There is no Route HF 0(14396) DEBUG: t_newtran: T on entrance=0xffffffff 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) parse_headers: flags=78 0(14396) t_lookup_request: start searching: hash=48297, isACK=0 0(14396) DEBUG: RFC3261 transaction matching failed 0(14396) DEBUG: t_lookup_request: no transaction found 0(14396) DBG: trans=0xb60e0960, callback type 1, id 0 entered 0(14396) ##########################method=INVITE################################## 0(14396) ##############################client_nat_test(7)############################## 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) DEBUG:check_content_type: type <application/sdp> found valid 0(14396) parse_headers: flags=80 0(14396) parse_headers: flags=10000 0(14396) pre_auth(): Credentials with given realm not found 0(14396) build_auth_hf(): 'Proxy-Authenticate: Digest realm=" voip.domain.com", nonce="482c8213b23f69ac2b268e593620cb2367dc643b" ' 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) WARNING: script writer didn't release transaction 0(14396) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(14396) DEBUG: add_to_tail_of_timer[2]: 0xb60e09a8 (10704) 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 0(14396) SIP Request: 0(14396) method: <ACK> 0(14396) uri: <sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> > 0(14396) version: <SIP/2.0> 0(14396) parse_headers: flags=2 0(14396) Found param type 235, <rport> = <n/a>; state=6 0(14396) Found param type 232, <branch> = <z9hG4bK7F77064E8D8FD3FD41779B0F67EF5C05>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) preparing to run routing scripts... 0(14396) parse_headers: flags=8 0(14396) DEBUG: add_param: tag=b802e73ab92fec121a4f880263d541a3.3a34 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [83]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) DEBUG: sl_filter_ACK : local ACK found -> dropping it! 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 0(14396) SIP Request: 0(14396) method: <INVITE> 0(14396) uri: <sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> > 0(14396) version: <SIP/2.0> 0(14396) parse_headers: flags=2 0(14396) Found param type 235, <rport> = <n/a>; state=6 0(14396) Found param type 232, <branch> = <z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) preparing to run routing scripts... 0(14396) In request log route 0(14396) 0(14396) ############################## NEW REQUEST RECEIVED ############################## 0(14396) parse_headers: flags=10 0(14396) DEBUG: add_param: tag=2027736222 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={5555555555}, ruri={ sip:5555555555@voip.domain.com:5061} 0(14396) request from 122.xxx.xxx.35 [Thu May 15 20:28:55 2008]: INVITE sip:5555555555@voip.domain.com:5061 -> sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> : 0(14396) request from 122.xxx.xxx.35 [Thu May 15 20:28:55 2008]: Request: INVITE sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> 0(14396) From: sip:5555555555@voip.domain.com:5061;tag=2027736222 0(14396) parse_headers: flags=8 0(14396) DEBUG:parse_to:end of header reached, state=10 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [41]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> > ] 0(14396) To: sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> ;tag=<null> 0(14396) SIP message-id: 201 0(14396) parse_headers: flags=40 0(14396) Call-id: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 0(14396) parse_headers: flags=20 0(14396) get_hdr_field: cseq <CSeq>: <28080> <INVITE> 0(14396) CSeq: 28080 0(14396) Contact: <sip:5555555555@122.xxx.xxx.35:5061> 0(14396) Message Body: INVITE sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com>SIP/2.0 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;rport;branch=z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>> Contact: <sip:5555555555@122.xxx.xxx.35:5061> Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 INVITE Proxy-Authorization: Digest username="5555555555",realm="voip.domain.com ",nonce="482c8213b23f69ac2b268e593620cb2367dc643b",response="7a7d3cf46a5b7cf90cc623664ca52e4d",uri=" sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>" Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 291 v=0 o=5555555555 3961416264 3961416447 IN IP4 122.xxx.xxx.35 s=X-Lite c=IN IP4 122.xxx.xxx.35 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv 0(14396) parse_headers: flags=100 0(14396) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(14396) parse_headers: flags=80 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) parse_headers: flags=8000000 0(14396) #############################record_route############################### 0(14396) #######################After record_route########################3 0(14396) ##################################Before loose_route########################## 0(14396) parse_headers: flags=200 0(14396) DEBUG: get_hdr_body : content_length=291 0(14396) found end of header 0(14396) find_first_route: No Route headers found 0(14396) loose_route: There is no Route HF 0(14396) DEBUG: t_newtran: T on entrance=0xffffffff 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) parse_headers: flags=78 0(14396) t_lookup_request: start searching: hash=48303, isACK=0 0(14396) DEBUG: RFC3261 transaction matching failed 0(14396) DEBUG: t_lookup_request: no transaction found 0(14396) DBG: trans=0xb60df1c8, callback type 1, id 0 entered 0(14396) ##########################method=INVITE################################## 0(14396) ##############################client_nat_test(7)############################## 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) DEBUG:check_content_type: type <application/sdp> found valid 0(14396) parse_headers: flags=80 0(14396) check_nonce(): comparing [482c8213b23f69ac2b268e593620cb2367dc643b] and [482c8213b23f69ac2b268e593620cb2367dc643b] 0(14396) DEBUG:auth_radius:radius_authorize_sterman: Success 0(14396) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair 225 0(14396) ERROR:avp_radius:avp_load_user: rc_auth failed 0(14396) DEBUG:usrloc:db_load_urecord: aor not found in DB 0(14396) lookup(): '999999999(a)voip.domain.com' Not found in usrloc 0(14396) grep_sock_info - checking if host==us: 23==23 && [voip.domain.com] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) alias_db_lookup: no alias found for R-URI 0(14396) does_uri_exit(): User in request uri does not exist 0(14396) parse_headers: flags=78 0(14396) ------[sip:5555555555@voip.domain.com:5061] ---checkeding---- 0(14396) radius_is_user_in(): Success 0(14396) -------- media route 5 ---- 0(14396) ########################## isbflagset(6) isbflagset(7)#################################### 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) parse_headers: flags=8000000 0(14396) DEBUG:lcr:load_gws: add matched_gws[0]=[0,0] 0(14396) load_gws(): DEBUG: Added gw_uri_avp <sip:|0@213.XXX.XXX.28 :5060;transport=udp> 0(14396) -----------------lcr is loaded into the ------ 0(14396) load_gws(): DEBUG: Added ruri_user_avp <999999999> 0(14396) ------------- Gateway is loaded --------- 0(14396) DEBUG: mk_proxy: doing DNS lookup... 0(14396) parse_headers: flags=2000 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) clen_builder: content-length: 313 (313) 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) DBG:check_against_rule_list: using list dns 0(14396) DEBUG:tm:set_timer: relative timeout is 500000 0(14396) DEBUG: add_to_tail_of_timer[4]: 0xb60df314 (10701100000) 0(14396) DEBUG:tm:set_timer: relative timeout is 20 0(14396) DEBUG: add_to_tail_of_timer[0]: 0xb60df330 (10720) 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 0(14396) SIP Reply (status): 0(14396) version: <SIP/2.0> 0(14396) status: <100> 0(14396) reason: <Trying> 0(14396) parse_headers: flags=2 0(14396) get_hdr_field: cseq <CSeq>: <28080> <INVITE> 0(14396) DEBUG: add_param: tag=5430ebc2c5d6575 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [61]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) Found param type 232, <branch> = <z9hG4bKfacb.bd19b862.0>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) grep_sock_info - checking if host==us: 13==23 && [195.XXX.XXX.51] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) DEBUG:forward_reply: found module nathelper, passing reply to it 0(14396) DEBUG:forward_reply: found module tm, passing reply to it 0(14396) DEBUG: t_check: start=0xffffffff 0(14396) parse_headers: flags=22 0(14396) parse_headers: flags=8 0(14396) DEBUG: t_reply_matching: hash 48303 label 646681051 branch 0 0(14396) DEBUG:tm:REF_UNSAFE: after is 1 0(14396) DEBUG: t_reply_matching: reply matched (T=0xb60df1c8)! 0(14396) DEBUG: t_check: end=0xb60df1c8 0(14396) DEBUG:tm:reply_received: org. status uas=0, uac[0]=0 local=0 is_invite=1) 0(14396) DEBUG:tm:t_should_relay_response: T_code=0, new_code=100 0(14396) DEBUG:tm:relay_reply: branch=0, save=0, relay=-1 0(14396) DEBUG:tm:set_timer: relative timeout is 50 0(14396) DEBUG: add_to_tail_of_timer[1]: 0xb60df330 (10750) 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 0(14396) SIP Reply (status): 0(14396) version: <SIP/2.0> 0(14396) status: <183> 0(14396) reason: <Session Progress> 0(14396) parse_headers: flags=2 0(14396) get_hdr_field: cseq <CSeq>: <28080> <INVITE> 0(14396) DEBUG: add_param: tag=5430ebc2c5d6575 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [61]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) Found param type 232, <branch> = <z9hG4bKfacb.bd19b862.0>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) grep_sock_info - checking if host==us: 13==23 && [195.XXX.XXX.51] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) DEBUG:forward_reply: found module nathelper, passing reply to it 0(14396) DEBUG:forward_reply: found module tm, passing reply to it 0(14396) DEBUG: t_check: start=0xffffffff 0(14396) parse_headers: flags=22 0(14396) parse_headers: flags=8 0(14396) DEBUG: t_reply_matching: hash 48303 label 646681051 branch 0 0(14396) DEBUG:tm:REF_UNSAFE: after is 1 0(14396) DEBUG: t_reply_matching: reply matched (T=0xb60df1c8)! 0(14396) DEBUG: t_check: end=0xb60df1c8 0(14396) DEBUG:tm:reply_received: org. status uas=0, uac[0]=100 local=0 is_invite=1) 0(14396) DEBUG:tm:t_should_relay_response: T_code=0, new_code=183 0(14396) DEBUG:tm:relay_reply: branch=0, save=0, relay=0 0(14396) old size: 884, new size: 822 0(14396) build_res_from_sip_res: copied size: orig:318, new: 256, rest: 566 msg= SIP/2.0 183 Session Progress Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 INVITE From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;branch=z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED;rport=5061 Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Content-Length: 209 Content-Type: application/sdp Contact: sip:999999999@213.XXX.XXX.28:5060 User-Agent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3.2.8.45 00a0ba009b05 v=0 o=MxSIP 0 7983 IN IP4 213.XXX.XXX.28 s=SIP Call c=IN IP4 213.XXX.XXX.28 t=0 0 m=audio 19462 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 0(14396) WARNING:vqm_resize: resize(0) called 0(14396) DEBUG:tm:relay_reply: sent buf=0x819a510: SIP/2.0 1..., shmem=0xb60e8218: SIP/2.0 1 0(14396) DEBUG:tm:set_timer: relative timeout is 50 0(14396) DEBUG: add_to_tail_of_timer[1]: 0xb60df330 (10750) 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 1(14397) DEBUG: timer routine:4,tl=0xb60df314 next=(nil), timeout=10701100000 1(14397) DEBUG: timer routine:2,tl=0xb60e09a8 next=(nil), timeout=10704 1(14397) DEBUG: wait_handler : removing 0xb60e0960 from table 1(14397) DEBUG: delete transaction 0xb60e0960 1(14397) DEBUG: wait_handler : done 0(14396) SIP Reply (status): 0(14396) version: <SIP/2.0> 0(14396) status: <200> 0(14396) reason: <OK> 0(14396) parse_headers: flags=2 0(14396) get_hdr_field: cseq <CSeq>: <28080> <INVITE> 0(14396) DEBUG: add_param: tag=5430ebc2c5d6575 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [61]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) Found param type 232, <branch> = <z9hG4bKfacb.bd19b862.0>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) grep_sock_info - checking if host==us: 13==23 && [195.XXX.XXX.51] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) DEBUG:forward_reply: found module nathelper, passing reply to it 0(14396) DEBUG:forward_reply: found module tm, passing reply to it 0(14396) DEBUG: t_check: start=0xffffffff 0(14396) parse_headers: flags=22 0(14396) parse_headers: flags=8 0(14396) DEBUG: t_reply_matching: hash 48303 label 646681051 branch 0 0(14396) DEBUG:tm:REF_UNSAFE: after is 1 0(14396) DEBUG: t_reply_matching: reply matched (T=0xb60df1c8)! 0(14396) DEBUG: t_check: end=0xb60df1c8 0(14396) DEBUG:tm:reply_received: org. status uas=183, uac[0]=183 local=0 is_invite=1) 0(14396) DEBUG:tm:t_should_relay_response: T_code=183, new_code=200 0(14396) DEBUG:tm:relay_reply: branch=0, save=0, relay=0 0(14396) old size: 891, new size: 829 0(14396) build_res_from_sip_res: copied size: orig:304, new: 242, rest: 587 msg= SIP/2.0 200 OK Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 INVITE From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;branch=z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED;rport=5061 Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Content-Length: 209 Content-Type: application/sdp Supported: replaces Contact: sip:999999999@213.XXX.XXX.28:5060 User-Agent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3.2.8.45 00a0ba009b05 v=0 o=MxSIP 0 7983 IN IP4 213.XXX.XXX.28 s=SIP Call c=IN IP4 213.XXX.XXX.28 t=0 0 m=audio 19462 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 0(14396) DEBUG: add_to_tail_of_timer[2]: 0xb60df210 (10710) 0(14396) DEBUG:tm:relay_reply: sent buf=0x819a510: SIP/2.0 2..., shmem=0xb60e05e0: SIP/2.0 2 0(14396) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 0(14396) SIP Reply (status): 0(14396) version: <SIP/2.0> 0(14396) status: <200> 0(14396) reason: <OK> 0(14396) parse_headers: flags=2 0(14396) get_hdr_field: cseq <CSeq>: <28080> <INVITE> 0(14396) DEBUG: add_param: tag=5430ebc2c5d6575 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [61]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) Found param type 232, <branch> = <z9hG4bKfacb.bd19b862.0>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) grep_sock_info - checking if host==us: 13==23 && [195.XXX.XXX.51] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) DEBUG:forward_reply: found module nathelper, passing reply to it 0(14396) DEBUG:forward_reply: found module tm, passing reply to it 0(14396) DEBUG: t_check: start=0xffffffff 0(14396) parse_headers: flags=22 0(14396) parse_headers: flags=8 0(14396) DEBUG: t_reply_matching: hash 48303 label 646681051 branch 0 0(14396) DEBUG:tm:REF_UNSAFE: after is 1 0(14396) DEBUG: t_reply_matching: reply matched (T=0xb60df1c8)! 0(14396) DEBUG: t_check: end=0xb60df1c8 0(14396) DEBUG:tm:reply_received: org. status uas=200, uac[0]=200 local=0 is_invite=1) 0(14396) DEBUG:tm:t_should_relay_response: T_code=200, new_code=200 0(14396) DEBUG:tm:t_should_relay_response: 200 OK for INVITE after final sent 0(14396) DEBUG:tm:relay_reply: branch=0, save=0, relay=0 0(14396) old size: 891, new size: 829 0(14396) build_res_from_sip_res: copied size: orig:304, new: 242, rest: 587 msg= SIP/2.0 200 OK Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 INVITE From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;branch=z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED;rport=5061 Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Content-Length: 209 Content-Type: application/sdp Supported: replaces Contact: sip:999999999@213.XXX.XXX.28:5060 User-Agent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3.2.8.45 00a0ba009b05 v=0 o=MxSIP 0 7983 IN IP4 213.XXX.XXX.28 s=SIP Call c=IN IP4 213.XXX.XXX.28 t=0 0 m=audio 19462 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 0(14396) DEBUG:tm:relay_reply: sent buf=0x819a510: SIP/2.0 2..., shmem=0xb60e05e0: SIP/2.0 2 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 0(14396) SIP Request: 0(14396) method: <ACK> 0(14396) uri: <sip:999999999@213.XXX.XXX.28:5060> 0(14396) version: <SIP/2.0> 0(14396) parse_headers: flags=2 0(14396) Found param type 235, <rport> = <n/a>; state=6 0(14396) Found param type 232, <branch> = <z9hG4bK384293560A7C9508E28212EC0D72CDFA>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) preparing to run routing scripts... 0(14396) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(14396) In request log route 0(14396) 0(14396) ############################## NEW REQUEST RECEIVED ############################## 0(14396) parse_headers: flags=10 0(14396) DEBUG: add_param: tag=2027736222 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={5555555555}, ruri={ sip:5555555555@voip.domain.com:5061} 0(14396) request from 122.xxx.xxx.35 [Thu May 15 20:29:02 2008]: ACK sip:5555555555@voip.domain.com:5061 -> sip:999999999@213.XXX.XXX.28:5060: 0(14396) request from 122.xxx.xxx.35 [Thu May 15 20:29:02 2008]: Request: ACK sip:999999999@213.XXX.XXX.28:5060 0(14396) From: sip:5555555555@voip.domain.com:5061;tag=2027736222 0(14396) parse_headers: flags=8 0(14396) DEBUG: add_param: tag=5430ebc2c5d6575 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [61]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) To: sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> ;tag=5430ebc2c5d6575 0(14396) SIP message-id: 206 0(14396) parse_headers: flags=40 0(14396) Call-id: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 0(14396) parse_headers: flags=20 0(14396) get_hdr_field: cseq <CSeq>: <28080> <ACK> 0(14396) CSeq: 28080 0(14396) Contact: <sip:5555555555@122.xxx.xxx.35:5061> 0(14396) Message Body: ACK sip:999999999@213.XXX.XXX.28:5060 SIP/2.0 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;rport;branch=z9hG4bK384293560A7C9508E28212EC0D72CDFA From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Contact: <sip:5555555555@122.xxx.xxx.35:5061> Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 ACK Max-Forwards: 70 Content-Length: 0 0(14396) parse_headers: flags=100 0(14396) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(14396) #############################record_route############################### 0(14396) #######################After record_route########################3 0(14396) ##################################Before loose_route########################## 0(14396) parse_headers: flags=200 0(14396) is_preloaded: No 0(14396) grep_sock_info - checking if host==us: 14==23 && [213.XXX.XXX.28] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) grep_sock_info - checking if host==us: 14==23 && [213.XXX.XXX.28] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) DEBUG:check_self: host != me 0(14396) grep_sock_info - checking if host==us: 23==23 && [voip.domain.com] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) after_loose: Topmost route URI: 'sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on' is me 0(14396) parse_headers: flags=200 0(14396) DEBUG: get_hdr_body : content_length=0 0(14396) found end of header 0(14396) find_next_route: No next Route HF found 0(14396) after_loose: No next URI found 0(14396) #################IN Loose_route############################ 0(14396) DEBUG: has_totag: totag found 0(14396) ############################has_totag() method==INVITE method==ACK################################ 0(14396) parse_headers: flags=80 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) parse_headers: flags=8000000 0(14396) ###########################setbflag6################################# 0(14396) ###############################force_rport############################# 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) #############################use_media_proxy############################### 0(14396) parse_headers: flags=78 0(14396) DEBUG: t_newtran: T on entrance=0xffffffff 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) parse_headers: flags=78 0(14396) t_lookup_request: start searching: hash=48303, isACK=1 0(14396) parse_headers: flags=38 0(14396) DEBUG:tm:REF_UNSAFE: after is 1 0(14396) DEBUG: t_lookup_request: e2e proxy ACK found 0(14396) DEBUG:tm:t_newtran: building branch for end2end ACK 0(14396) DEBUG:tm:t_relay: forwarding ACK 0(14396) DEBUG: mk_proxy: doing DNS lookup... 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) DBG:check_against_rule_list: using list dns 0(14396) DEBUG:forward_request: sending: ACK sip:999999999@213.XXX.XXX.28:5060 SIP/2.0 Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Via: SIP/2.0/UDP 195.XXX.XXX.51;branch=z9hG4bKfacb.bd19b862.2 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;rport=5061;branch=z9hG4bK384293560A7C9508E28212EC0D72CDFA From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Contact: <sip:5555555555@122.xxx.xxx.35:5061> Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 ACK Max-Forwards: 69 Content-Length: 0 . 0(14396) DEBUG:forward_request: orig. len=519, new_len=593, proto=1 0(14396) -------------- cheked t reply --- 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list 0xb60e04c8 0(14396) receive_msg: cleaning up 0(14396) SIP Request: 0(14396) method: <ACK> 0(14396) uri: <sip:999999999@213.XXX.XXX.28:5060> 0(14396) version: <SIP/2.0> 0(14396) parse_headers: flags=2 0(14396) Found param type 235, <rport> = <n/a>; state=6 0(14396) Found param type 232, <branch> = <z9hG4bK384293560A7C9508E28212EC0D72CDFA>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) preparing to run routing scripts... 0(14396) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(14396) In request log route 0(14396) 0(14396) ############################## NEW REQUEST RECEIVED ############################## 0(14396) parse_headers: flags=10 0(14396) DEBUG: add_param: tag=2027736222 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={5555555555}, ruri={ sip:5555555555@voip.domain.com:5061} 0(14396) request from 122.xxx.xxx.35 [Thu May 15 20:29:02 2008]: ACK sip:5555555555@voip.domain.com:5061 -> sip:999999999@213.XXX.XXX.28:5060: 0(14396) request from 122.xxx.xxx.35 [Thu May 15 20:29:02 2008]: Request: ACK sip:999999999@213.XXX.XXX.28:5060 0(14396) From: sip:5555555555@voip.domain.com:5061;tag=2027736222 0(14396) parse_headers: flags=8 0(14396) DEBUG: add_param: tag=5430ebc2c5d6575 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [61]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) To: sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> ;tag=5430ebc2c5d6575 0(14396) SIP message-id: 207 0(14396) parse_headers: flags=40 0(14396) Call-id: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 0(14396) parse_headers: flags=20 0(14396) get_hdr_field: cseq <CSeq>: <28080> <ACK> 0(14396) CSeq: 28080 0(14396) Contact: <sip:5555555555@122.xxx.xxx.35:5061> 0(14396) Message Body: ACK sip:999999999@213.XXX.XXX.28:5060 SIP/2.0 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;rport;branch=z9hG4bK384293560A7C9508E28212EC0D72CDFA From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Contact: <sip:5555555555@122.xxx.xxx.35:5061> Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 ACK Max-Forwards: 70 Content-Length: 0 0(14396) parse_headers: flags=100 0(14396) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(14396) #############################record_route############################### 0(14396) #######################After record_route########################3 0(14396) ##################################Before loose_route########################## 0(14396) parse_headers: flags=200 0(14396) is_preloaded: No 0(14396) grep_sock_info - checking if host==us: 14==23 && [213.XXX.XXX.28] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) grep_sock_info - checking if host==us: 14==23 && [213.XXX.XXX.28] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) DEBUG:check_self: host != me 0(14396) grep_sock_info - checking if host==us: 23==23 && [voip.domain.com] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) after_loose: Topmost route URI: 'sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on' is me 0(14396) parse_headers: flags=200 0(14396) DEBUG: get_hdr_body : content_length=0 0(14396) found end of header 0(14396) find_next_route: No next Route HF found 0(14396) after_loose: No next URI found 0(14396) #################IN Loose_route############################ 0(14396) DEBUG: has_totag: totag found 0(14396) ############################has_totag() method==INVITE method==ACK################################ 0(14396) parse_headers: flags=80 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) parse_headers: flags=8000000 0(14396) ###########################setbflag6################################# 0(14396) ###############################force_rport############################# 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) #############################use_media_proxy############################### 0(14396) parse_headers: flags=78 0(14396) DEBUG: t_newtran: T on entrance=0xffffffff 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) parse_headers: flags=78 0(14396) t_lookup_request: start searching: hash=48303, isACK=1 0(14396) parse_headers: flags=38 0(14396) DEBUG:tm:REF_UNSAFE: after is 1 0(14396) DEBUG: t_lookup_request: e2e proxy ACK found 0(14396) DEBUG:tm:t_newtran: building branch for end2end ACK 0(14396) DEBUG:tm:t_relay: forwarding ACK 0(14396) DEBUG: mk_proxy: doing DNS lookup... 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) DBG:check_against_rule_list: using list dns 0(14396) DEBUG:forward_request: sending: ACK sip:999999999@213.XXX.XXX.28:5060 SIP/2.0 Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Via: SIP/2.0/UDP 195.XXX.XXX.51;branch=z9hG4bKfacb.bd19b862.2 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;rport=5061;branch=z9hG4bK384293560A7C9508E28212EC0D72CDFA From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Contact: <sip:5555555555@122.xxx.xxx.35:5061> Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 ACK Max-Forwards: 69 Content-Length: 0 . 0(14396) DEBUG:forward_request: orig. len=519, new_len=593, proto=1 0(14396) -------------- cheked t reply --- 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list 0xb60e04c8 0(14396) receive_msg: cleaning up 1(14397) DEBUG: timer routine:2,tl=0xb60df210 next=(nil), timeout=10710 1(14397) DEBUG: wait_handler : removing 0xb60df1c8 from table 1(14397) DEBUG: delete transaction 0xb60df1c8 1(14397) DEBUG: wait_handler : done *************************************************************************************************************************************** -- Thanks & regards, Krunal Patel On Thu, May 15, 2008 at 12:03 AM, LetMeKnow < sunkara.raviprakash.feb14(a)gmail.com> wrote: > hey krunal, > i hopes ur fine and also all members of Ecosmob are fine, > > i have read ur mail, > yours problem is mention properly in mailing list. > > In ur openser script file some problem in there and u not handling the nat > in ur script. > make sure u have use both modules nathelper and mediaproxy, > > Before doing on this checkout is openser system is located inside the nat > or not. > > while test use ngrep tool on openser system, see whether u have got the > properly signaling happening not. > > check out after 200 ok of invite msg > > > On Wed, May 14, 2008 at 5:40 PM, Krunal Patel <krunal.patel(a)ecosmob.com> > wrote: > >> Hi, >> >> We are using openser with media proxy and radiator. We're having audio >> problems when we send calls to mediatrix gateway for termination. >> Internal calls work fine. We hear audio both side and we can see the >> session and bytes transfer properly in media proxy monitoring webpage. >> When we send calls to mediatrix gateway for termination, calls connect >> properly but we don't hear audio. We see inactive session in mediaproxy >> status page with no bytes transferring. >> One doubt I have is it is not entering reply route when it receives >> replies like 183, 180, etc. >> Has anyone encountered similar issue? How can we fix it? >> Thanks in advance! >> >> -- >> regards, >> Krunal Patel >> >> >> _______________________________________________ >> 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 > 91-9999882776 >

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Re: [OpenSER-Users] [OpenSER-Devel] [RFC] - siptrace change
by Bogdan-Andrei Iancu 15 May '08

15 May '08

Hi Norman, yup, that is exactly my problem - I use it only for doing siptrace per user and on platform with high traffic, the stateless ACK tracing produces a lot of residual data in DB. Regards, Bogdan Norman Brandinger wrote: > This change would be good. We have reduced our use of siptrace for > this very reason. > > Norm > > Bogdan-Andrei Iancu wrote: >> Hi, >> >> I would like to get your opinion/comments on some siptrace issue. >> >> In 1.3, a new capability was added in siptrace - to trace the ACK >> resulted from sending stateless replies. >> >> The problem I discovered with this is that it cannot be controlled by >> selecting the messages or transactions to be be traced. Usually you >> use siptrace to trace only certain transactions / messages - tracing >> all traffic is usually not a realistic option. >> >> So, with the stateless ACK, the siptrace module cannot apply the >> selection from script and trace them all, with no filtering at all. >> So, if you want to trace a traffic for a single user, all ACK going >> through the platform will be traced. >> >> This reduces the usability of the module and you get a lot of garbage >> tracing. >> >> My suggestion will be to disable stateless ACK tracing until a way to >> control/filter it is found. I'm asking this considering the upcoming >> 1.3.2 release from tomorrow. >> >> Regards, >> Bogdan >> >> _______________________________________________ >> Devel mailing list >> Devel(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/devel >> >> >> > >

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[OpenSER-Users] parallel forking
by Allan Chao ( 趙廷睿 ) 15 May '08

15 May '08

Dear All: I have found there are some devices can't process two 180 ringing after use OPENSER to do parallel forking, so, i have a idea that is used OPENSER to delete To tag in the To header before relay it , and it works well with eyebeam. But i can't find any spec to support my idea because it is against "early dialog" prograss and my question is how does UE(eyebeam or some client) to create dialog if there is no any To tag in 180 ringing but 200 OK has in the latter ? i think the UA will generate local ring tone but no any dialog after receives 180 ringing with no To tag? is it right ? thanks, allan DISCLAIMER: Sample Disclaimer added in a VBScript.

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[OpenSER-Users] cdrtool error: "cannot figure out the destination id ..."
by Adrian Georgescu 15 May '08

15 May '08

You must provision in the destinations table with all prefixes for PSTN destinations (at least all country codes). The software will find the longest match from that table. Adrian >>>>>> Trying cdrtool prepaid rating, got stuck with something. When a request like "MaxSessionTime From=sip:me at example.com To=sip:0031650222 at example.com Duration=7200 Lock=0" is given, cdrtool always return "0", no matter what amount of balance is available in prepaid table. This is what I see in syslog: "CDRTool rating[5759]: Error: cannot figure out the destination id for sip:0031650222 at example.com" Cdrtool version is 6.3.2. Apparently rating engine needs the variable $CDR->DestinationID set (rating_lib.phtml, line #5926), but this is not happening. It isn't clear to me how this is initialised (In weak defense, I'm totally clueless about php), and it doesn't work even if DestinationID parameter is added in the above stated request.

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[OpenSER-Users] username mangling
by David Terry 14 May '08

14 May '08

Hi Openser, My question is regarding case-handling of SIP usernames with Openser 1.3.1 presence modules. I'm using a stand-alone presence server with pua/pua_usrloc modules. This is the test case: 1. user "sip:1234@10.10.90.51" sends SUBSCRIBE for user "sip:Jane@10.10.90.51" 2. user "sip:Jane@10.10.90.51" sends REGISTER After 2, watcher does not receive NOTIFY from Openser/presence. DB contents is as follows: active_watchers/presentity_uri contains "sip:Jane@10.10.90.51" pua/pres_uri contains "sip:jane@10.10.90.51" presentity/username contains "jane". location/username contains "jane" location/contact contains "sip:Jane@10.10.90.51:10001" I'm not sure why case is different in these tables, and whether this causes the missed NOTIFY. Can anyone suggest what might be wrong? Here's the Openser route block: route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (is_method("SUBSCRIBE|PUBLISH")) route(2); if (method == "REGISTER") { pua_set_publish(); save("location"); exit; } } route[2] { # presence handling if (! t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); }; exit; }

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[OpenSER-Users] SerMyAdmin - Cannot log in as admin
by Wyndham Hudson 14 May '08

14 May '08

Hi, I've set up serMyAdmin on the tomcat server and everything appears to be working, however I don't seem to be able to log in as the admin. I've added a user admin to subscriber with password etc. but it still does not allow me in. MySQL is recieveing the queries and processing them as I have checked in the general-query-log. I'm currently running openser 1.2.x with serMyAdmin 0.5. Any help would be appreciated. Thanks

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[OpenSER-Users] remove message body
by Andreas Granig 14 May '08

14 May '08

Hi, Is there any possibility to remove the whole message body of a SIP message? replace_body_all and subst_body seem to operate per line, leaving a bunch of "\n" behind. There's a two year old feature request for that on the tracker (https://sourceforge.net/tracker/?func=detail&atid=743023&aid=1546501&group_…). Any news on that? Any other possibilities? Andreas

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[OpenSER-Users] Segmentation Fault in Openser 1.3.1 - 64 bits.
by Sergio Gutierrez 14 May '08

14 May '08

Hello everybody. Today we got a crash in our production deployment of OpenSER, which is running version 1.3.1 on Solaris SPARC. OpenSER was compiled on 64 bits mode.file is: The backtrace of the core file is: Program terminated with signal 11, Segmentation fault. #0 0xffffffff79719774 in insert_dlg_timer_unsafe (tl=0xffffffff61ea8ad0) at dlg_timer.c:91 91 if ( ptr->timeout <= tl->timeout ) (gdb) bt #0 0xffffffff79719774 in insert_dlg_timer_unsafe (tl=0xffffffff61ea8ad0) at dlg_timer.c:91 #1 0xffffffff79719a8c in update_dlg_timer (tl=0xffffffff61ea8ad0, timeout=300) at dlg_timer.c:163 #2 0xffffffff7970fe9c in dlg_onroute (req=0x100341fb0, route_params=0xffffffff7fffe2b0, param=0x0) at dlg_handlers.c:612 #3 0xffffffff7d80c278 in run_rr_callbacks (req=0x100341fb0, rr_params=0xffffffff7d910468) at rr_cb.c:83 #4 0xffffffff7d807b0c in after_loose (_m=0x100341fb0, preloaded=0) at loose.c:865 #5 0xffffffff7d803744 in loose_route (_m=0x100341fb0, _s1=0x0, _s2=0x0) at loose.c:901 #6 0x000000010000f088 in do_action (a=0x1002e3e30, msg=0x100341fb0) at action.c:816 #7 0x0000000100009e48 in run_action_list (a=0x1002e3e30, msg=0x100341fb0) at action.c:132 #8 0x0000000100086a08 in eval_elem (e=0x1002e3ed0, msg=0x100341fb0, val=0x0) at route.c:1069 #9 0x000000010008db44 in eval_expr (e=0x1002e3ed0, msg=0x100341fb0, val=0x0) at route.c:1360 #10 0x000000010008dd00 in eval_expr (e=0x1002e3f18, msg=0x100341fb0, val=0x0) at route.c:1381 #11 0x000000010000e0b8 in do_action (a=0x1002e53c8, msg=0x100341fb0) at action.c:672 #12 0x0000000100009e48 in run_action_list (a=0x1002d8878, msg=0x100341fb0) at action.c:132 #13 0x000000010000a4b0 in run_actions (a=0x1002d8878, msg=0x100341fb0) at action.c:112 #14 0x000000010000a6fc in run_top_route (a=0x1002d8878, msg=0x100341fb0) at action.c:172 #15 0x00000001000781ec in receive_msg ( buf=0x1002aaf68 "INVITE sip:76907295@200.114.42.182<sip%3A76907295(a)200.114.42.182>;user=phone SIP/2.0\r\nVia: SIP/2.0/UDP 200.30.79.17:5066;branch=z9hG4bKa8fae90df\r\nRoute: <sip:200.13.225.243;lr;ftag=885a2b1ed6b8a7d8;did=7cf.1721>\r\nCall-ID: f6a12831f"..., len=747, rcv_info=0xffffffff7ffff860) at receive.c:156 #16 0x00000001000db620 in udp_rcv_loop () at udp_server.c:438 #17 0x00000001000321b8 in main_loop () at main.c:834 #18 0x00000001000373bc in main (argc=7, argv=0xffffffff7ffffbe8) at main.c:1373 I would thank any hint you could provide about it. Best regards. Sergio Gutiérrez.

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[OpenSER-Users] sermyadmin almost works
by Lucas Catranis 14 May '08

14 May '08

Anyone Get this setup correctly? I've been at it for a while and after getting it to insert new users I cannot login as an admin. I'm not sure where the problem is but Any help would be appreciated. -- View this message in context: http://www.nabble.com/sermyadmin-almost-works-tp17219053p17219053.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[OpenSER-Users] Where exactly are " sip-clients" and "rtp-clients" supposed to be?
by luzango mfupe 14 May '08

14 May '08

Hi mates, In my O'SER 1.3 config file from svn i have this modules parameters which shows the location of rtp and sip clients but in reality these files doesnt exist in the said location or anywhere in my box, what must be wrong with my config?? below are the two lines: modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-clients") -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA website:http://www.tuune.mobi

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[OpenSER-Users] cdrtool error: "cannot figure out the destination id ..."
by Sajith T S 14 May '08

14 May '08

Hi! Trying cdrtool prepaid rating, got stuck with something. When a request like "MaxSessionTime From=sip:me@example.com To=sip:0031650222@example.com Duration=7200 Lock=0" is given, cdrtool always return "0", no matter what amount of balance is available in prepaid table. This is what I see in syslog: "CDRTool rating[5759]: Error: cannot figure out the destination id for sip:0031650222@example.com" Cdrtool version is 6.3.2. Apparently rating engine needs the variable $CDR->DestinationID set (rating_lib.phtml, line #5926), but this is not happening. It isn't clear to me how this is initialised (In weak defense, I'm totally clueless about php), and it doesn't work even if DestinationID parameter is added in the above stated request. Since there seems to be other people using cdrtool with success, I guess might be doing something really stupid. I haven't trudged too far away from a standard cdrtool install. Can anyone tell me what possibly could have gone wrong? Or even better: teach a man how to fish (ie, how to trace out this), so that he might shut up when there's no fish... :) Thanks, Sajith. -- Don't you wish you had more energy... or less ambition? 9DB8FF06 : CB80 0BA6 7D13 B10A 6FBB D43E B4D2 28AD 9DB8 FF06

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[OpenSER-Users] Authentication challenge failure
by Bogdan-Andrei Iancu 14 May '08

14 May '08

Hi, I just found and fix a really strange bug in authentication module, when building the auth challenge header. I say strange, because I found it while using a UAC that implements a very strict view on the auth process. Also this UAC tries to reuse the nonces. This bug had as effect the UACs stopping to re-register with openser after an openser restart. Quite unpleasant effect to have all the UACs dropping out if you do a server restart :( . More technically, the bug consists in openser's failure to append the stale parameter in the challenge request if the nonce is not recognize as local - this can happen after a restart, when openser uses a new schema to generate nonces. Scenario: 1) start openser -> it will set SCHEMA1 for generating nonces 2) UAC registers with authentication and receives during challenge the nonce NONCE1 (based on SCHEMA1) 3) OpenSER restarts and sets a new SCHEMA2 for generating nonces 4) UAC tries to re-register using the previous nonce it received - NONCE1. 5) OpenSER rejects the auth as received NONCE1 does not follow current SCHEMA2. 6) OpenSER sends a new challenge to the UAC, but so far, the stale parameter was not added to indicate that the nonce is invalid 7) UAC simply drops any registration attempts as it thinks that the password it has is wrong -> it authentication was rejected and no stale indication was received. With the fix, openser now adds the stale parameter in the challenge and to indicate to UAC a nonce issue if the nonce is not recognized. The script auth functions were already reporting (as return code) NONCE_STALE indication in this case, but the challange was not properly computed. Have anybody experience such problem also? or I was the first coming across a such UAC :) . Regards, Bogdan

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[OpenSER-Users] compilation on solaris 10/Spar
by Liangyou Wang (liangwan) 14 May '08

14 May '08

Hi experts, Anyone has experience on compilation on solaris 10/Spar? I am start user of openser and need use openser to support sctp. now I have download the source codes of openser-1.3.1-tls. then I compiled it, there have many errors that need your supports. e.g. openser-1.3.1-tls> make all Makefile.defs:717: *** missing separator. Stop. then I commented Makefile.defs line717 and 718. # $(warning Old gcc detected ($(CC_SHORTVER)), use gcc >= 3.1 \ # for better results) Errors were still there. openser-1.3.1-tls> make all Makefile:402: warning: overriding commands for target `/opt/CiscoMGC$(DESTDIR)' Makefile:393: warning: ignoring old commands for target `/opt/CiscoMGC$(DESTDIR)' yacc -d -b cfg cfg.y "cfg.y", line 391: warning: redeclaration of precedence of SLASH. conflicts: 1 shift/reduce flex cfg.lex statistics.c:58: warning: #warning STATISTICS: Architecture with no support for atomic operations. Using Locks!! Makefile:402: warning: overriding commands for target `/opt/CiscoMGC$(DESTDIR)' Makefile:393: warning: ignoring old commands for target `/opt/CiscoMGC$(DESTDIR)' Compiling action.c gcc -g -O9 -funroll-loops -Wall -mv8 -DNAME='"openser"' -DVERSION='"1.3.1-notls"' -DARCH='"sparc64"' -DOS='"solaris"' -DCOMPILER='"gcc 2.91.66"' -D__CPU_sparc64 -D__OS_solaris -D__SMP_no -DCFG_DIR='"$(DESTDIR)/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -D_POSIX_PTHREAD_SEMANTICS -DHAVE_DEVPOLL -DHAVE_SELECT -c action.c -o action.o parser/msg_parser.h: In function `char_msg_val': In file included from action.h:27, from action.c:44: parser/msg_parser.h:300: parse error before `)' forward.h: In function `msg_send': In file included from action.c:49: forward.h:95: parse error before `)' forward.h:103: parse error before `)' forward.h:107: parse error before `)' mem/../lock_alloc.h: In function `lock_set_alloc': In file included from mem/../locking.h:66, from mem/../statistics.h:115, from mem/shm_mem.h:33, from ut.h:52, from action.c:54: mem/../lock_alloc.h:68: parse error before `)' ut.h: In function `int2bstr': In file included from action.c:54: ut.h:172: parse error before `)' ut.h: In function `un_escape': ut.h:365: parse error before `)' ut.h: In function `shm_str_dup': ut.h:498: parse error before `)' ut.h: In function `pkg_str_dup': ut.h:515: parse error before `)' ut.h: In function `str_strcmp': ut.h:532: parse error before `)' ut.h:535: parse error before `int' ut.h:539: `minlen' undeclared (first use in this function) ut.h:539: (Each undeclared identifier is reported only once ut.h:539: for each function it appears in.) ut.h:547: `alen' undeclared (first use in this function) ut.h:547: `blen' undeclared (first use in this function) ut.h: In function `str_strcasecmp': ut.h:563: parse error before `)' ut.h:566: parse error before `int' ut.h:570: `minlen' undeclared (first use in this function) ut.h:578: `alen' undeclared (first use in this function) ut.h:578: `blen' undeclared (first use in this function) action.c: In function `run_action_list': action.c:140: parse error before `)' action.c: In function `do_assign': action.c:193: parse error before `)' action.c:202: parse error before `)' action.c:222: parse error before `)' action.c:228: parse error before `)' action.c: In function `do_action': action.c:302: parse error before `)' action.c:309: parse error before `)' action.c:335: parse error before `)' action.c:498: parse error before `)' action.c:524: parse error before `)' action.c:682: parse error before `)' action.c:710: parse error before `)' action.c:723: parse error before `)' action.c:757: parse error before `)' action.c:763: parse error before `)' action.c:807: parse error before `)' action.c:819: parse error before `)' action.c:870: parse error before `)' action.c:896: parse error before `)' action.c:904: parse error before `)' action.c:930: parse error before `)' action.c:939: parse error before `)' action.c:945: parse error before `)' action.c:966: parse error before `)' make: *** [action.o] Error 1 Your any help is much appreciated! BR, Eric

2 1

[OpenSER-Users] [OT?] voip-to-voip rating using cdrtool
by Adrian Georgescu 14 May '08

14 May '08

You can rate sip calls to specific SIP URIs, which are not telephone numbers in CDRTool. The SIP URIs must be fully qualified, that is you can set a price for user(a)example.com but you cannot do a wildcard matching for *(a)example.com Just add the SIP URI in the destination table as any other PSTN prefix. Regards, Adrian >>>>>>>>> Apologies if this is off-topic here, but this seemed like a good friendly place to ask. Apologies for being so naïve also... Question is, how can I get cdrtool to rate voip-to-voip calls when destination is non-numeric? As in, say, sip:holmes at example.com calls sip:watson at example.com. How do I get cdrtool to match *(a)example.com or *(a)example2.com? (FWIW, I'm just trying to see how this works... no plan of rolling out this Evil Scheme into a real world deployment.) While cdrtool has decent documentation, I wish this was a little more easy on me. Has anyone read/reviewed this book yet? http://www.packtpub.com/building-telephony-systems-with-openser/book I'm specifically interested in the "Billing with Freeradius and CDRTool" part. Thanks, Sajith.

2 1

[OpenSER-Users] presence not working with X-lite
by Amit Vijayvargiya 14 May '08

14 May '08

Hi, we are done all the configuration of presence module in openser.cfg file set all the parameters..........i.e. modparam("presence|presence_xml", "db_url", "mysql://openser:openser@localhost/openser") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:domain:5060") modparam("presence", "fallback2db", 1) modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "to_tag_pref", 'pres') modparam("presence", "expires_offset", 10) modparam("xcap_client", "db_url", "mysql://openser:openser@localhost/openser") modparam("xcap_client", "xcap_table","xcap") modparam("presence_xml", "db_url", "mysql://openser:openser@localhost/openser") modparam("presence_xml", "force_active", 0) modparam("presence_xml", "xcap_table", "xcap") modparam("presence_xml", "pidf_manipulation", 1) modparam("presence_xml", "integrated_xcap_server", 1) and route the PUBLISH and SUBSCRIBE massages if( is_method("PUBLISH|SUBSCRIBE")) route(2); route[2] is::::: route[2] { sl_send_reply("100","trying"); if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { if($hdr(Sender)!= NULL) handle_publish("$hdr(Sender)"); else handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; } when we are run X-lite is asking you want to allow or deny the user-B........... but not showing online even we are allow the user-B.......... when we are allow the user x-lite not send any type of message...... we see in Notify openser sending a notify with watcher-list but with the status pending inside XML format. please tell me where we are wrong???????????? there is a problem with X-lite or our configuration????? Thanks in advance Amit Vijayvargiya

1 0

[Serusers] Presence Server
by Allan Chao ( 趙廷睿 ) 14 May '08

14 May '08

Dear All: I want to use SER's Presence module to do presence server , but someone say it's not stable , so , 1. Does anyone could tell me why it doesn't stable ? because it looks work after i do some testing with eyebeam. 2. how does openser's presence ? is it better than SER's presence ? thanks, allan DISCLAIMER: Sample Disclaimer added in a VBScript.

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[Serusers] TLS problem ser 2_0_0 - ERROR: tls_init.c:366: Unable to set the memory allocation functions
by .:: Francesco la Torre ::. 13 May '08

13 May '08

Have you found any solution for this issue ? I'm facing with a similar problem (not the same because the source of same error is line 520). Thanks in advance Francesco la Torre IIT/CNR-Pisa

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[OpenSER-Users] Presence, multiple status info handling
by Dan-Cristian Bogos 13 May '08

13 May '08

Guys, while dealing with presence/presence_xml modules, I came into one situation when one user crashes simultaneously and each time displaying different presence info (online, away, offline). Have noticed that the presence server is smart enough to inform all the other watchers about three different status information for the same user. My question is, how can the watcher know the most recent status information. Have looked on both order of info in the tuple as well as on ids but did not find any logic. Can u please guide me about possibilities I have? Thank you in advance, DanB PS: FYI, I am pasting such a tuple example. <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="t9062"> <status xmlns="urn:ietf:params:xml:ns:pidf"> <basic>closed</basic> </status> </tuple> <dm:person xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" id="p8838"><rpid:activities/></dm:person> <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="t5358"> <status><basic>open</basic></status><contact>sip:user@domain.com<sip%3Auser(a)domain.com> </contact><note>Online</note> </tuple> <dm:person xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" id="p4226"><rpid:activities><rpid:away/></rpid:activities></dm:person> <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="t7139"> <status><basic>open</basic></status><contact>sip:user@domain.com<sip%3Auser(a)domain.com> </contact><note>Away</note> </tuple> <dm:person xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" id="p4606"><rpid:activities/></dm:person> <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="t5632"> <status><basic>open</basic></status><contact>sip:user@domain.com<sip%3Auser(a)domain.com> </contact><note>Online</note> </tuple></presence>

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[OpenSER-Users] Routing Logic
by VoIP Forums www.Go4Calls.com 13 May '08

13 May '08

Hi Friends, I have the following route plan in my openser.cfg all blocks are working well instead of 800800. if (uri=~"sip:800800[1-9][0-9]+@.*") { rewritehostport("officePBX-IP:5060"); route(1); exit; } else if ($(rU{s.len})>=8) { rewritehostport("MyPSTNprovider-IP:5061"); route(1); exit; }else{ lookup("location"); route(1); exit; }; When i am trying to call 800800 the error message appearing is "513 Message too big" 800800 is routed to the asterisk for our internal office's extensions. Pls guide us how i can solve this issue. Regards, www.Go4Calls.Com VoIP Forums _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

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[OpenSER-Users] Authentcation using DISC is successful
by Dilip 13 May '08

13 May '08

Hello Sir, I have been able to make the DISC code compatible with openser-1.1.1-notls. I have eliminated some of the bugs which were there in the source code of DISC.But the thing is that the disc code is not able to work with x86_64 architecture. But it is successfully running with i386 arch. Could u please suggest me the neccessary task,i have to perform in order to make DISC compiled on x86_64 arch. I tried taking the successfully running code from i386 arch and copying it on x86_64 arch and then to compile it, but it is giving some architecture compilation problem which is given below. In file included from ../mem/shm_mem.h:53, from avp.c:33: ../mem/../lock_ops.h:182:2: error: #error "no locking method selected" ../mem/../lock_ops.h:265:2: error: #error "no lock set method selected" In file included from avp.c:33: ../mem/shm_mem.h:79: error: expected '=', ',', ';', 'asm' or '__attribute__' before '*' token ../mem/shm_mem.h: In function '_shm_malloc': ../mem/shm_mem.h:111: warning: implicit declaration of function 'lock_get' ../mem/shm_mem.h:111: error: 'mem_lock' undeclared (first use in this function) ../mem/shm_mem.h:111: error: (Each undeclared identifier is reported only once ../mem/shm_mem.h:111: error: for each function it appears in.) ../mem/shm_mem.h:113: warning: implicit declaration of function 'lock_release' avp.c: In function 'AAACreateAVP': avp.c:124: warning: passing argument 3 of '_shm_malloc' discards qualifiers from pointer target type avp.c:140: warning: passing argument 3 of '_shm_malloc' discards qualifiers from pointer target type avp.c: In function 'AAAFreeAVP': avp.c:321: error: 'mem_lock' undeclared (first use in this function) avp.c:321: warning: passing argument 4 of 'qm_free' discards qualifiers from pointer target type avp.c:323: warning: passing argument 4 of 'qm_free' discards qualifiers from pointer target type avp.c: In function 'AAACloneAVP': avp.c:438: warning: passing argument 3 of '_shm_malloc' discards qualifiers from pointer target type avp.c:448: warning: passing argument 3 of '_shm_malloc' discards qualifiers from pointer target type avp.c:451: error: 'mem_lock' undeclared (first use in this function) avp.c:451: warning: passing argument 4 of 'qm_free' discards qualifiers from pointer target type make[2]: *** [avp.lo] Error 1 Please Do help me ,to sort out the problem. regards, Dilip

1 0

[Serusers] Asterisk->SER rtp problems
by Thorsten 13 May '08

13 May '08

Hi, I have an asterisk server running with an private IP. This asterisk forwards all calls to a SER server with a public IP. The SER server then forwards its calls to a public SIP provider. The problem now is that SER tries to stay in the loop which it shouldn't because there is no media proxy running. I don't get any audio because of this issue. But if I register the asterisk box directly to the SIP provider it works. Does anybody know how to fix this. My ser.cfg debug=4 # debug level (cmd line: -dddddddddd) #debug=0 fork=no log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #listen=0.0.0.0 #listen=82.98.89.140 port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/opt/ser/lib/ser/modules/mysql.so" loadmodule "/usr/local/ser/lib/ser/modules/sl.so" loadmodule "/usr/local/ser/lib/ser/modules/tm.so" loadmodule "/usr/local/ser/lib/ser/modules/rr.so" loadmodule "/usr/local/ser/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/ser/lib/ser/modules/usrloc.so" loadmodule "/usr/local/ser/lib/ser/modules/registrar.so" loadmodule "/usr/local/ser/lib/ser/modules/textops.so" loadmodule "/usr/local/ser/lib/ser/modules/avpops.so" #loadmodule "/usr/local/ser/lib/ser/modules/group.so" loadmodule "/usr/local/ser/lib/ser/modules/xlog.so" loadmodule "/usr/local/ser/lib/ser/modules/auth.so" loadmodule "/usr/local/ser/lib/ser/modules/nathelper.so" loadmodule "/usr/local/ser/lib/ser/modules/uri.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/opt/ser/lib/ser/modules/auth.so" #loadmodule "/opt/ser/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) #modparam("registrar", "nat_flag", 6) #modparam("nathelper", "natping_interval", 30) # Ping interval 30 s #modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="ACK") { route(1); break; } if (method=="REGISTER") { #record_route(); save("location"); break; }; if (method=="INVITE") { #if (uri =~ "sip:[0-9]@*") { # if (nat_uac_test("19")) { # fix_nated_contact(); # fix_nated_sdp("3"); # } # route(3); # break; #} if (uri =~ "sip:[0-9]@*") { # record_route(); route(3); break; } }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } route[3] { if (uri =~ "sip:[0-9]@*") { log(1, "Forwarding to mg3.net-m.de \n"); #rewritehostport("192.168.13.102:5060"); rewritehostport("62.214.145.199:5060"); #forward(62.214.145.199, 5060); route(1); break; } } My extensions.conf [toser] exten => _X.,1,Dial(sip/${EXTEN}(a)82.98.89.139) Thanks for any help Ciao Thorsten

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[OpenSER-Users] how to set FROM to a different uri
by sye 13 May '08

13 May '08

HI, I use aliases table set sip forwarding. During the forwarding I want to do is set FROM uri to a different sip address. ( I want to hide the original FROM uri). How do I do it? I check the cook book and don't see any function that I can use to do that work. Any hint? Thanks, Stanley

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[OpenSER-Users] regarding presence module facing problem
by Amit Vijayvargiya 13 May '08

13 May '08

hi, i m add presence module in openser 1.3. but it not working properly. when x-lite send a subscribe to client-B, it receive notify but with different body. i see in the table presentity it shows right body. where is the problem????? my configuration like this for subscribe and publish...... route[2] { sl_send_reply("100","trying"); append_to_reply("Contact: <sip:sip.pyrogroup.com:5060>\r\n"); if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { if($hdr(Sender)!= NULL) handle_publish("$hdr(Sender)"); else handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; } Thanks & Regards, Amit

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[Serusers] SER 2.0 Question
by Santiago Gimenez Ocano 12 May '08

12 May '08

Hi, I'm using SER 2.0 rc1. Thing is I have found excelent documentation (like tm module) on iptel pages, but I couldn't find information for other modules like acc_radius. There's a web site or something where I can find this kind of help? Thanks, Santiago _________________________________________________________________ Do more with your photos with Windows Live Photo Gallery. http://www.windowslive.com/share.html?ocid=TXT_TAGLM_Wave2_photos_022008

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Re: [OpenSER-Users] 200 OK retransmissions on missing ACK can cause subsequent calls to fail
by Sean O'Donnell 09 May '08

09 May '08

1 0

Re: [OpenSER-Users] 200 OK retransmissions on missing ACK can cause subsequent calls to fail
by Sean O'Donnell 09 May '08

09 May '08

2 1

[Serusers] SER-Server-ID, Radius Dictionary Issue
by Santiago Gimenez Ocano 09 May '08

09 May '08

Hi all, I'm trying to use acc_radius module in SER 2.0.0. But when it's loading I get this error: 0(9009) ERROR: acc: can't get code for the SER-Server-ID attribute 0(9009) init_mod(): Error while initializing module acc_radius ERROR: error while initializing modules I think this is a dictionary error, when I add the attribute I get the following message: 0(9031) ERROR: sr_module.c:887: Error while fixing parameter, AVP, SELECT, and int conversions failed I assume I'm making some mistake, but I don't know where. I don't think that is in ser.cfg and maybe is a module or dictionary problem. Can anybody help me? Thanks, Santiago _________________________________________________________________ Get your fix of news, sports, entertainment and more on MSN Mobile http://www.msnmobilefix.com/Default.aspx

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Re: [Serusers] SER - Strange Calling Issues
by Arun Kumar 09 May '08

09 May '08

> > Hi ALL, > > I'm facing some strange issue with my VoIP Setup. I've configure 5 servers > : > > 1. 192.168.20.1 = SER / MediaProxy > 2. 192.168.20.2 = MySQL > 3. 192.168.20.3 = Presence > 4. 192.168.20.4 = Asterisk for PSTN with A2Billing > 5. 192.168.20.5 = Voicemail > > *First Problem:* > I've created few users on my SER and they are able to call each other > (Users are behind NAT) *but if User A calls User B and User A disconnect > call from his side the User B phone keeps on ringing until User B > disconnects the call. > > Second Problem(Only some time): > *when user dial PSTN number they hear ringing on there side but other > side has picked up the phone and they started to talk also but calling side > is still hearing the ring. > > *Third: > *Some time users complain that there calls are getting disconnect (PSTN > Calls) after 20-30 second, I feel its more related to my PSTN Provider but > how can I prove so I can tell my provider about this problem. > > Please provide your feedback. thanks in advance. > > I've attached my ser.cfg file b'coz if I'll paste message will be big so. > =================== > >

1 0

[Serusers] CANCEL doubts
by Nuno Ribeiro 08 May '08

08 May '08

Hi all, I have a doubt about the CANCEL messages and I would like to hear your opinion. The routes that should be inserted in the CANCEL message should be based int the ones that are part of the received INVITE or in the ones that are part of the transmitted INVITE? From my point of view the CANCELs should be created based in the routes of the transmitted INVITE, in order to the stateless proxies be able to route the CANCEL requests properly. The RFC says the following: 9.1 Client Behavior A CANCEL request SHOULD NOT be sent to cancel a request other than INVITE. Since requests other than INVITE are responded to immediately, sending a CANCEL for a non-INVITE request would always create a race condition. The following procedures are used to construct a CANCEL request. The Request-URI, Call-ID, To, the numeric part of CSeq, and From header fields in the CANCEL request MUST be identical to those in the request being cancelled, including tags. A CANCEL constructed by a client MUST have only a single Via header field value matching the top Via value in the request being cancelled. Using the same values for these header fields allows the CANCEL to be matched with the request it cancels (Section 9.2 indicates how such matching occurs). However, the method part of the CSeq header field MUST have a value of CANCEL. This allows it to be identified and processed as a transaction in its own right (See Section 17). If the request being cancelled contains a Route header field, the CANCEL request MUST include that Route header field's values. This is needed so that stateless proxies are able to route CANCEL requests properly. Thanks in advance, Nuno Ribeiro

2 1

[OpenSER-Users] recover from previous strict routing triggered incorrectly in ALG scenario
by Andrew Pogrebennyk 08 May '08

08 May '08

Hi, Im'm cross-posting this from OpenSER-Devel because I am not 100% sure this is a bug. I am trying to create openser configuration for proxy that connects private (VPN) network with public network. I started with a config similar to http://voipembedded.com/resources/openser_cr.cfg, added a call to force_send_socket() to send from public IP and rtpproxy/nathelper in bridge mode. As per config loose_route() is called from openser.cfg to route a sequential request within a dialog through record-routing. Now when a BYE from callee comes in, it contains two Route headers: Route: <a.b.c.d;r2=on;lr;ftag=...> where a.b.c.d is server's public IP, Route: <10.0.0.100;r2=on;lr;ftag=...> where 10.0.0.100 is a private IP and username(a)a.b.c.d in Request-URI. Take a look at loose_route(): /* * Do loose routing as defined in RFC3261 */ int loose_route(struct sip_msg* _m, char* _s1, char* _s2) { int ret; if (find_first_route(_m) != 0) { LM_DBG("There is no Route HF\n"); return -1; } if (parse_sip_msg_uri(_m)<0) { LM_ERR("failed to parse Request URI\n"); return -1; } ret = is_preloaded(_m); if (ret < 0) { return -1; } else if (ret == 1) { return after_loose(_m, 1); } else { #ifdef ENABLE_USER_CHECK if (is_myself(&_m->parsed_uri.user, &_m->parsed_uri.host, _m->parsed_uri.port_no)) { #else if (is_myself(&_m->parsed_uri.host, _m->parsed_uri.port_no)) { #endif return after_strict(_m); } else { return after_loose(_m, 0); } } } Obviously, is_myself() check passes and after_strict() is called. when after_strict() is called, it looks if the URI given in Route contains lr parameter which marks loose routers, if so this block of code is executed: { LM_DBG("Next hop: '%.*s' is loose router\n", uri.len, ZSW(uri.s)); if(get_maddr_uri(&uri, &puri)!=0) { LM_ERR("failed to check maddr\n"); return RR_ERROR; } if (set_dst_uri(_m, &uri) < 0) { LM_ERR("failed to set dst_uri\n"); return RR_ERROR; } /* Next hop is a loose router - Which means that is is not endpoint yet * In This case we have to recover from previous strict routing, that * means we have to find the last Route URI and put in in R-URI and * remove the last Route URI. */ if (rt != hdr->parsed) { /* There is a previous route uri which was 2nd uri of mine * and must be removed here */ rem_off = hdr->body.s; rem_len = rt->nameaddr.name.s - hdr->body.s; if (!del_lump(_m, rem_off - _m->buf, rem_len, 0)) { LM_ERR("failed to remove Route HF\n"); return RR_ERROR; } } res = find_rem_target(_m, &hdr, &rt, &prev); if (res < 0) { LM_ERR("searching for last Route URI failed\n"); return RR_ERROR; } else if (res > 0) { /* No remote target is an error */ return RR_ERROR; } uri = rt->nameaddr.uri; if(get_maddr_uri(&uri, 0)!=0) { LM_ERR("checking maddr failed\n"); return RR_ERROR; } if (rewrite_uri(_m, &uri) < 0) { LM_ERR("failed to rewrite R-URI\n"); return RR_ERROR; } /* The first character if uri will be either '<' when it is the * only URI in a Route header field or ',' if there is more than * one URI in the header field */ LM_DBG("The last route URI: '%.*s'\n", rt->nameaddr.uri.len, ZSW(rt->nameaddr.uri.s)); if (prev) { rem_off = prev->nameaddr.name.s + prev->len; rem_len = rt->nameaddr.name.s + rt->len - rem_off; } else { rem_off = hdr->name.s; rem_len = hdr->len; } if (!del_lump(_m, rem_off - _m->buf, rem_len, 0)) { LM_ERR("failed to remove Route HF\n"); return RR_ERROR; } } Therefore recover from previous strict routing is triggered: contents of request line is destroyed and 10.0.0.100 is put instead: Apr 15 16:16:24 vpn-proxy oser[11027]: DBG:core:rewrite_uri: rewriting Request-URI with 'sip:10.0.0.100;r2=on;lr;ftag=aa40c2a5-54fb-1810-970d-0015c5bf2da5' Apr 15 16:16:24 vpn-proxy oser[11032]: [1] Got in dialog request not catched by loose_route(): BYE sip:10.0.0.100;r2=on;lr;ftag=aa40c2a5-54fb-1810-970d-0015c5bf2da5 SIP/2.0^M Via:... And we respond with 404 to in-dialog requests not caught by loose_route(). Previous hop is SER and I don't know why OpenSER considered it a strict router. Looks like a peculiarity of ALG mode. Please advice what should I do to solve this problem. Is it a configuration issue or perhaps loose_route() should not call after_strict() in case of double record-routing? -- Thank you, Andrew Pogrebennyk

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[OpenSER-Users] segfault with general protection error
by Marcello Lupo 08 May '08

08 May '08

Hi to all, today i got a segfault with general protection error openser 1.2.2 notls under Linux 2.6.16.27-0.9-smp #1 SMP Tue Feb 13 09:35:18 UTC 2007 x86_64 x86_64 x86_64 GNU/Linux. It is due to: May 8 18:29:25 voip /data/openser/sbin/openser[19569]: val2str: Destination buffer too short May 8 18:29:25 voip /data/openser/sbin/openser[19569]: print_values: Error while converting value to string May 8 18:29:25 voip /data/openser/sbin/openser[19569]: ERROR:acc:acc_db_request: Error while inserting to database May 8 18:29:26 voip /data/openser/sbin/openser[19560]: child process 19573 exited by a signal 11 May 8 18:29:27 voip /data/openser/sbin/openser[19560]: core was generated I examined the core dump and it has been caused from the same thing that happened to me on October 2007 on another installation. I already posted the question to the list without getting any response. http://lists.openser.org/pipermail/users/2007-October/013796.html I post now the new gdb trace. I'm a very beginner using gdb but like very much to know why this is happening. Core was generated by `/data/openser/sbin/openser'. Program terminated with signal 11, Segmentation fault. #0 0x00002b83e4be4150 in escape_string_for_mysql () from /usr/lib64/ libmysqlclient.so.15 (gdb) bt #0 0x00002b83e4be4150 in escape_string_for_mysql () from /usr/lib64/ libmysqlclient.so.15 #1 0x00002b83e49465d1 in val2str (_c=0x2b83e4de05a0, _v=<value optimized out>, _s=0x2b83e4a48f8d "'33301','','469','','4','','','','102 INVITE','','12.13.14.15','')", _len=0x7fffc67d7804) at val.c:316 #2 0x00002b83e4942952 in db_mysql_print_values (_c=0x6b8768, _b=0x2b83e4a48e0e "'INVITE','5ec34ebc6ad9825','as54a9f138','647f8f41f2ed74cfb28b83633bb8efdb(a)voip.test.biz ','200','OK','2008-05-08 18:29:25','sip:035123456@voip.test.biz;tag=5ec34ebc6ad9825','sip:031454647@voip "..., _l=65170, _v=<value optimized out>, _n=28) at dbase.c:145 #3 0x00002b83e4943967 in db_mysql_insert (_h=0x6b9018, _k=<value optimized out>, _v=0x2b83e65aa5c0, _n=28) at dbase.c:584 #4 0x00002b83e64956d3 in acc_db_request (rq=0x2b83e6c9dc28) at acc.c: 378 #5 0x00002b83e6497244 in tmcb_func (t=<value optimized out>, type=<value optimized out>, ps=<value optimized out>) at acc_logic.c:376 #6 0x00002b83e52958f1 in run_trans_callbacks (type=128, trans=0x2b83e7673690, req=<value optimized out>, rpl=<value optimized out>, code=<value optimized out>) at t_hooks.c:205 #7 0x00002b83e52a3e76 in relay_reply (t=0x2b83e7673690, p_msg=<value optimized out>, branch=0, msg_status=200, cancel_bitmap=0x7fffc67d7ac8) at t_reply.c:1155 #8 0x00002b83e52a5b0d in reply_received (p_msg=0x6b9138) at t_reply.c: 1388 #9 0x000000000041929e in forward_reply (msg=0x6b9138) at forward.c:489 #10 0x000000000043cbd2 in receive_msg ( buf=0x60e700 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 80.122.12.3;branch=z9hG4bKdd28.b33b4137.0;received=80.122.12.3\r\nVia: SIP/2.0/UDP 20.21.22.23:5060;rport=5060;branch=z9hG4bK31ffac575\r \nRecord-Route: <sip:031454647"..., len=850, rcv_info=0x7fffc67d7b90) at receive.c:195 #11 0x000000000046d45f in udp_rcv_loop () at udp_server.c:451 #12 0x0000000000428dd0 in main_loop () at main.c:834 #13 0x000000000042a899 in main (argc=<value optimized out>, argv=<value optimized out>) at main.c:1393 Thanks in advance to anyone can help me to debug this thing. Bye, Marcello

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[OpenSER-Users] [OT?] voip-to-voip rating using cdrtool
by Sajith T S 08 May '08

08 May '08

Apologies if this is off-topic here, but this seemed like a good friendly place to ask. Apologies for being so naïve also... Question is, how can I get cdrtool to rate voip-to-voip calls when destination is non-numeric? As in, say, sip:holmes@example.com calls sip:watson@example.com. How do I get cdrtool to match *(a)example.com or *(a)example2.com? (FWIW, I'm just trying to see how this works... no plan of rolling out this Evil Scheme into a real world deployment.) While cdrtool has decent documentation, I wish this was a little more easy on me. Has anyone read/reviewed this book yet? http://www.packtpub.com/building-telephony-systems-with-openser/book I'm specifically interested in the "Billing with Freeradius and CDRTool" part. Thanks, Sajith. -- Don't you wish you had more energy... or less ambition? 9DB8FF06 : CB80 0BA6 7D13 B10A 6FBB D43E B4D2 28AD 9DB8 FF06

1 0

[OpenSER-Users] Privacy Question
by Helmut Kuper 08 May '08

08 May '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, today I found an incomming Privacy-Header Field from my PSTN-Gateway-Provider like this: Privacy: header;user;id Openser 1.4 says: May 8 11:22:25 [29986] ERROR:core:parse_privacy: invalid privacy value RFC 3323 says: Privacy-hdr = "Privacy" HCOLON priv-value *(";" priv-value) priv-value = "header" / "session" / "user" / "none" / "critical" ~ / token The values of priv-value today are restricted to the above options, ~ although further options can be defined as appropriate (see Section ~ 7). Each legitimate priv-value can appear zero or one times in a ~ Privacy header. So that means, the incomming Privacy-header is OK, doesn't it? a Bug? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.8 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkgiyI0ACgkQ4tZeNddg3dzbfACgjyLMZsVFW30bdTkiEJu+4pH2 XPoAnAhBEQ0IvecHApVqdOP7nSLh563x =4nWD -----END PGP SIGNATURE-----

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[OpenSER-Users] regarding presence module facing problem
by Amit Vijayvargiya 08 May '08

08 May '08

Hi All, i m add presence module in openser 1.3. but it not working properly. openser given this error --> DBG:tm:run_reqin_callbacks: trans=0xb60fa590, callback type 1, id 0 entered May 8 21:09:14 [20993] DBG:core:parse_headers: flags=ffffffffffffffff May 8 21:09:14 [20993] DBG:presence:search_event: start event= [message-summary] May 8 21:09:14 [20993] ERROR:presence:handle_subscribe: Missing or unsupported event header field value May 8 21:09:14 [20993] ERROR:presence:handle_subscribe: event= message-summary May 8 21:09:14 [20993] DBG:core:parse_headers: flags=ffffffffffffffff May 8 21:09:14 [20993] DBG:core:check_via_address: params 172.22.24.192, 172.22.24.192, 0 May 8 21:09:14 [20993] DBG:tm:cleanup_uac_timers: RETR/FR timers reset Thanks & Regards Amit

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[OpenSER-Users] OPenSER DISC Problem
by Dilip 08 May '08

08 May '08

Hello EveryOne, In the openser auth-diameter module,the first phase of digest AUTHETICATION of SIP is completly happening.But In the second phase when OPENSER sends the request to diameter client with authorization header,DISC server cant authorize the USER.Because the response of OPENSER and DISC SERVER are different. As per the code, In the DISC Server code, sipauth module is there which contains db.c file. In this file upto calc_response function call comes ,which calculate the response .This response and OPENSER response are different. Is that any configuration File Problem ? I have attached config file here. What was the reason for it? How can I solve this response problem? Thanks To Evrybody Regards, Dilip # # $Id: openser.cfg,v 1.6.2.1 2006/07/17 15:51:03 klaus_darilion Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) #/* Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes #*/ #check_via=no # (cmd. line: -v) #dns=no # (cmd. line: -r) #rev_dns=no # (cmd. line: -R) port=5060 children=4 listen=192.168.1.1 #fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib64/openser/modules/mysql.so" #loadmodule "/usr/local/lib64/openser/modules/acc.so" loadmodule "/usr/local/lib64/openser/modules/sl.so" loadmodule "/usr/local/lib64/openser/modules/tm.so" loadmodule "/usr/local/lib64/openser/modules/rr.so" loadmodule "/usr/local/lib64/openser/modules/maxfwd.so" loadmodule "/usr/local/lib64/openser/modules/usrloc.so" loadmodule "/usr/local/lib64/openser/modules/registrar.so" loadmodule "/usr/local/lib64/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib64/openser/modules/auth.so" loadmodule "/usr/local/lib64/openser/modules/auth_db.so" loadmodule "/usr/local/lib64/openser/modules/auth_diameter.so" ##Dilip loadmodule "/usr/local/lib64/openser/modules/domain.so" ##Dilip ############################################################ #modparam("dispatcher", "list_file", "/usr/local/etc/openser/dispatcher.list") #modparam("acc", "log_flag", 1) #modparam("acc", "log_missed_flag", 1) #modparam("acc", "service_type", 15) ############################################################# # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc|auth_db","db_url","mysql://openser:openserrw@localhost/openser") #Dilip #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", 1) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "passwd_h") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) ############################################################################################################ # Add By Dilip modparam("auth_diameter", "use_domain", 1) modparam("auth_diameter", "diameter_client_host", "linux173") modparam("auth_diameter", "diameter_client_port", 3000) ############################################################################################################ # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; #Add By Dilip if (method=="INVITE") { if (!diameter_www_authorize("linux173")) { www_challenge("", "0"); exit; }; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("192.168.1.173", "subscriber")) { # www_challenge("192.168.1.173", "0"); # exit; # }; # Add by Dilip if(!diameter_www_authorize("linux173")) #if(!diameter_www_authorize("")) { #/* user is not authorized */ www_challenge("linux173", "1"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; }

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[OpenSER-Users] openseradmin-03 modification
by Zen Kato 08 May '08

08 May '08

Hi, I installed openseradmin-0.3 on openser-1.2.3-notls. There are some errors,but works fine. Is it possible for the following modifications using Ruby-on-Rail even for a Ruby beginner? user login (only 2 screens) ============ http://localhost:3000/user/login Login Required Login : 0103 Password: 0103 [login>>] --------------------------------- [logout] Name: Domain: Username: 0103 Password: 0103 Email Address: 0103@localhost Caller ID: Voicemail: No Voicemail Password: Created At: Last Updated At: [Edit] ---------------------------------- [logout] Username: 0103 Password: 0103 Domain: First Name: Last Name: Email Address: 0103@localhost Caller ID: Voicemail[ ] Voicemail Password: [Edit] show ---------------------------------- admin login =========== http://localhost:3000/admin/login Login Required Login : openser Password: openserrw [login>>] no modification for admin screens. -- Zen Kato

1 0

[OpenSER-Users] Increase MAX_BRANCHES value?
by Zappasodi Daniele 07 May '08

07 May '08

Hello, I'm interested in this question too. I have tried to modify the tm module and I have found a solution that extends the maximum number of branches and it seems to work, but I'm still testing it. Can anyone, more expert than me on the tm module, give a look to the attached file with my patch? Bye Daniele Hi, thanks for your answer... I increased the SHM_MEM_SIZE to 64 and left KG_MEM_POOL_SIZE 4*1024*1024 . Anyway i found that is not possible to raise MAX_BRANCHES to values higher than 30 due to tm module internal limitation. Bye, Marcello ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

2 3

Re: [OpenSER-Users] srv load-balancing
by Stanisław Pitucha 07 May '08

07 May '08

----- "Bogdan-Andrei Iancu" <bogdan(a)voice-system.ro> wrote: > OpenSER it is not ignoring the weights - the received records are > ordered by weight. The tricky question is about records with the same > > weight - which to be used? openser is right now using the first > returned > by the DNS server. Ok - maybe I should expand that a bit: It's not disallowed behaviour, but rfc2782 specifies standard way for handling weights and sip protocol doesn't really need other handling afaik: --->8--- from http://www.faqs.org/rfcs/rfc2782.html --- The following algorithm SHOULD be used to order the SRV RRs of the same priority: To select a target to be contacted next, arrange all SRV RRs (that have not been ordered yet) in any order, except that all those with weight 0 are placed at the beginning of the list. Compute the sum of the weights of those RRs, and with each RR associate the running sum in the selected order. Then choose a uniform random number between 0 and the sum computed (inclusive), and select the RR whose running sum value is the first in the selected order which is greater than or equal to the random number selected. --->8--- So even if they have same weight: server1=50 and server2=50, openser should get random number X between 0 and 100 and select server1 for X<=50, or server2 for X>50. Currently openser is not using SRV records correctly imho - it's client responsibility to get all those RRs and select one of them randomly from all servers with selected priority (whether they have same weight or not). Please correct me if I'm wrong. Cheers, Stan

2 1

[OpenSER-Users] Prepaid Service with SER + B2BUA + Radius
by Ilker YILMAZ 07 May '08

07 May '08

Dear Group, I would like to have your comments on setting up successful prepaid service using SER + Sippy B2BUA and a Radius Server. Did anybody accomplished to drop a call when a user's balance is empty with B2BUA? I will really appreciate if you can share your experinces on this issue.. Thank You Ilker

1 0

[OpenSER-Users] 200 OK retransmissions on missing ACK can cause subsequent calls to fail
by Sean O'Donnell 07 May '08

07 May '08

Hi all, Im using openser as a call distributor/proxy between a soft-switch/SBC and voicemail platform. Im seeing a problem with openser in that it is sometimes cancels an in-progress call (fr_inv_timer firing) because it didnt match the 200/OK with the call. After some investigation, I noticed that this was happening after a missing ACK on a previous call caused the voicemail platform to retransmit 200/OK responses beyond the TM wt_timer expiration, which in turn left several openser child processes (those that received a 200 after wt_timer expiration) in a state such that they might not properly match transactions on subsequent calls. My setup: I have openser 1.2.0 operating on a linux box with two network interfaces, with one interface (call it the outside interface) taking incoming calls from the soft-switch, and the other (inside) connected to the VM platform. I have openser configured to use both interfaces (see config below) and the TM wt_timer set to 5 seconds (default). As this is a voicemail system, all of the call traffic is inbound from the soft-switch. Given the traffic flow, for the most part the openser child processes servicing the inside interface are handling responses (180,183,200) from the VM platform. Call scenario: When an INVITE arrives from the soft-switch, openser forwards it to the VM platform. The VM platform responds with a 180 and then a 200. I've noticed several instances where the soft-switch did not respond with an ACK. This caused the VM platform to retransmit the 200 several times over a 10 second period. These were absorbed correctly by openser for the duration of wt_timer. After the timer expired, however, each openser child process that received a retransmitted 200 logged something like this: 4(2715) DEBUG: t_reply_matching: hash 45870 label 727647196 branch 0 4(2715) DEBUG: t_reply_matching: no matching transaction exists 4(2715) DEBUG: t_reply_matching: failure to match a transaction 4(2715) DEBUG: t_check: end=(nil) When I look at the TM code, the static variable T in t_lookup.c is now NULL for this child process. On a subsequent inbound call, the INVITE is passed to the VM correctly, and the 180 transaction matches (causing the fr_inv_timer to be armed). If the 200 is read by child proc 2715, I see: 4(2715) DEBUG: t_check: start=(nil) 4(2715) DEBUG: t_check: T previously sought and not found The 200 is forwarded back to the soft-switch, which responds with an ACK. Both end-points think the call is up, but since openser never matched the 200 with the call, the fr_inv_timer fires and cancels the call. Basically, child proc 2715 wont match any transaction after this unless it happens to process a request. I think this problem is made worse by the fact that Im using two network interfaces, and that the openser children on the inside interface handle (for the most part) only responses. This problem was touched on here: http://lists.openser.org/pipermail/users/2007-November/014188.html but I didnt see any follow up. Also, Ive checked openser 1.2.3 and 1.3.1 for fixes, but I dont think this has been addressed. I have a work around, I think, by upping the wt_timer to something like 15 seconds, but I was wondering if there is any scenario in which leaving T=NULL is desirable. Thanks in advance Sean ==== CONFIG ===== # # VMPROXY-PROXY - route between softswitch and media servers # # ----------- global configuration parameters ------------------------ check_via=yes # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=8 # multi-homed (i.e. listen on two or more interfaces) mhomed=1 alias=sipproxy1 alias=sipproxy1.vmproxy.com port=5060 #inside address listen=192.168.1.100 #outside address listen=10.55.2.100 disable_dns_blacklist=yes # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/openser/lib/openser/modules/mysql.so" mpath="/usr/local/openser/lib/openser/modules" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "exec.so" loadmodule "xlog.so" loadmodule "textops.so" loadmodule "options.so" loadmodule "mi_fifo.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/openser/lib/openser/modules/auth.so" #loadmodule "/usr/local/openser/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- # local memory #modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # uncomment this for multiple record routes #modparam("rr", "enable_double_rr", 1) # -- tm params -- modparam("tm", "fr_timer", 5) modparam("tm", "fr_inv_timer", 60) # -- mi_fifo parms modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo_proxy") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); return; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); return; }; xlog("L_INFO","$Tf VMPROXY: <$rm> message From:<$fu> To:<$tu> srcip:<$si>\n"); # # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol # record_route(); # # loose-route processing # messages that already have routes on them are proxied here # if (loose_route()) { t_relay(); return; }; # # message not for us, route it through # if ( !(uri==myself) ) { if (!t_relay()) { sl_reply_error(); }; return; }; xlog("L_DBG","$Tf VMPROXY: <$rm> for me\n"); # handle options meant for me if( method == "OPTIONS" ) { options_reply(); return; }; if( method == "REGISTER" || method == "PUBLISH" || method == "SUBSCRIBE" || method == "MESSAGE" ) { sl_send_reply("405","Method Not Allowed"); return; }; # route INVITE based on source if( method == "INVITE" ) { exec_dset("/usr/local/vmproxy/bin/routeINVITE.sh"); xlog("L_DBG","$Tf VMPROXY: <$rm> after routeINVITE: new destination <$ru>\n"); t_on_failure("1"); }; # route NOTIFY to softswitch if( method == "NOTIFY" ) { exec_dset("/usr/local/vmproxy/bin/routeNOTIFY.sh"); xlog("L_DBG","$Tf VMPROXY: <$rm> after routeNOTIFY: new destination <$ru>\n"); t_on_failure("2"); }; # # NOTE: this logic is only executed if the request didn't have # a route on it (tsk,tsk). If it did, it would have been handed via # loose route function above. if( method == "ACK" || method == "BYE" || method == "CANCEL" || method == "PRACK" ) { exec_dset("/usr/local/vmproxy/bin/routeACK.sh"); xlog("L_DBG","$Tf VMPROXY: <$rm> after routeACK: new destination <$ru>\n"); }; # routing script error? if( uri=="sip:err@127.0.0.1" ) { sl_reply_error(); return; }; # forward to current uri if (!t_relay()) { sl_reply_error(); }; } # # Initial INVITE route failed, retry # failure_route[1] { # endpoint canceled call if( t_was_cancelled()) { xlog("L_DBG", "$Tf VMPROXY: call <id:$ci> cancelled\n"); return; } # don't retry on 40[0-7|5xx failures if( !t_check_status("40[0-7]|420|487|488|5[0-9][0-9]")) { xlog("L_ERR","$Tf VMPROXY: ERROR routing call <$rm> from <$fu> to <$ru>\n"); exec_dset("/usr/local/vmproxy/bin/routeINVITE2.sh"); xlog("L_ERR","$Tf VMPROXY: Rerouting call <$rm> to <$ru>\n"); append_branch(); t_relay(); } } # # Initial NOTIFY route failed, retry # failure_route[2] { # endpoint canceled call if( t_was_cancelled()) { xlog("L_DBG", "$Tf VMPROXY: call <id:$ci> cancelled\n"); return; } # don't retry on 40[0-7|5xx failures if( !t_check_status("40[0-7]|5[0-9][0-9]")) { xlog("L_ERR","$Tf VMPROXY: ERROR routing <$rm> from <$fu> to <$ru>\n"); exec_dset("/usr/local/vmproxy/bin/routeNOTIFY2.sh"); xlog("L_ERR","$Tf VMPROXY: Rerouting <$rm> to <$ru>\n"); append_branch(); t_relay(); } }

2 1

[Serusers] SER Nokia CANCEL Problem
by Abdul Qadir 07 May '08

07 May '08

Hi , I tried to call from one nokia sip (E61 and other models )phone to another nokia sip phone. The call works fine. The problem comes only when I call from Phone A to Phone B and then immediately cancel the call(from Phone A). The Phone A will hangup the call as it sent CANCEL but the SER will ignore this CANCEL and still send INVITE to Phone B resulting in a ghost call situation. I tried to capture a log of message and found that Phone A "CANCEL" message is received on SER even before any provisional response from Phone B. Therefor SER doesnot relay this CANCEL request to Phone B. I even checked RFC which clearly says that UAC should not send CANCEL untill it receives any provisional response. I talked to Nokia expert and they said the 100 Trying message from your server is considered as provisional response, therefor behaviour of client is absolutely correct. Is there any way I can stop 100 Trying message and still run statefull SER, so that I can verify what nokia said. Any ideas suggestions are welcome. Thanking you all in advance. Best Regards, Abdul Qadir --------------------------------- Don't be flakey. Get Yahoo! Mail for Mobile and always stay connected to friends.

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[OpenSER-Users] Pseudovariable for whole SIP-message
by Peter Lemenkov 07 May '08

07 May '08

Hello All! Is there a pseudovariable denoting the whole SIP-message? If not - are there plans for implementing such functionality? -- With best regards!

2 2

Re: [OpenSER-Users] Version table
by Henning Westerholt 07 May '08

07 May '08

On Wednesday 07 May 2008, Bård Aase wrote: >> [...] > Thanks, I will check this out now, I'm using OpenSER 1.3.1 and the > sip_buddies table looks like this: > [..] You'll find the subscriber DB docs (for trunk) at: http://www.openser.org/docs/db-tables/openser-db-devel.html#GEN-DB-SUBSCRIB… AFAIK there were some fiels removed, but except this it should be the same format as in the 1.3 branch. > The openser.cfg configuration for auth_db: > [..] > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "user_column", "name") > modparam("auth_db", "domain_column", "domain") > modparam("auth_db", "password_column", "secret") > modparam("auth_db", "password_column_2", "ha1b") > modparam("auth_db", "use_domain", 0) > modparam("auth_db", "db_url", > "mysql://UNDISCLOSED:UNDISCLOSED@UNDISCLOSED/asterisk") > > All the fields referred to here exists in the asterisk.sip_buddies > table, so my guess now is that this is just a matter of disabling the > version check, or bumping the version number. I think the same. If the semantics of the colums in question are also the same, then it should work just fine. Regards, Henning

1 0

[OpenSER-Users] Fwd: Version table
by Bård Aase 07 May '08

07 May '08

1 0

Re: [OpenSER-Users] [OpenSER-Devel] Memory dump for OpenSER 1.3.1 which slowly leaks memory
by Henning Westerholt 07 May '08

07 May '08

On Tuesday 06 May 2008, Julian Yap wrote: > Private memory is default. Should I increase this? Is there a > recommended rule of thumb? > > Shared memory is set for 128Mb. There were memory errors at 32Mb but > none since this was increased to 128Mb. > > Using the setting 'children = 8'. (moved to user list) Hi Julien, you only need to increase this if you use some modules that needs to load a big dataset on startup (e.g. carrierroute), and if you get allocation errors for the private memory. If you run with 128 MB shared memory, then the server should not use more then this memory, i.e. the depletion of memory you observed is not caused from OpenSER. The ammount of memory that is used from the server is: ((actual number of children) x private memory) + shared memory Perhaps you can adapt your "monitoring" to log the actual biggest memory consumer on the system? Cheers, Henning

1 0

[OpenSER-Users] Version table
by Bård Aase 07 May '08

07 May '08

I am trying to switch from SER to OpenSER, and I'm using mysql for data storage. Asterisk's sip_buddies table is supposed to contain authentication data. The configuration is set up, the openser database is appearantly set up and working, but when I try to authenticate against asterisk.sip_buddies i get the following errors: May 6 12:36:58 [25672] ERROR:auth_db:auth_fixup: invalid table version (use openser_mysql.sh reinstall) May 6 12:36:58 [25672] ERROR:core:fix_actions: fixing failed (code=-1) at cfg line 399 May 6 12:36:58 [25672] CRITICAL:core:fix_expr: fix_actions error May 6 12:36:58 [25672] ERROR:core:main: failed to fix configuration with err code -1 The version tables in asterisk and openser is as follows: SELECT * FROM asterisk.`version` 5 SELECT * FROM openser.`version` 6 What is the differences between database version 5 and 6? Since the asterisk database is shared between multiple asterisks and SER instances, its not trivial to upgrade the tables if that would have implications on asterisk and/or ser Thanks in advance. -- Bård Aase gTalk/Jabber: elzapp(a)elzapp.com MSN: msn(a)elzapp.com http://blog.elzapp.com

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[OpenSER-Users] Log errors when processing a contact list with RLS module
by Pablo Guijarro Enríquez 07 May '08

07 May '08

Hi, I am running openser v1.3.1 and I cannot get the RLS module to work properly when handling a contact list already retrieved from database. Some errors related with the pua module appear in the log when the server seems to be trying to generate the subscribe requests for each of the contacts in the list (see attached, lines 60 to 62). Apr 10 13:02:23 comm ./openser[20184]: ERROR:pua:subs_build_hdr: getting event from list Apr 10 13:02:23 comm ./openser[20184]: ERROR:pua:send_subscribe: while building extra headers Apr 10 13:02:23 comm ./openser[20184]: ERROR:rls:process_list_and_exec: infunction given as a parameter Those requests are never sent. What could this be due to? Thanks! Pablo

2 9

[OpenSER-Users] Problems with SIMPLE-XMPP presence
by Pablo Guijarro Enríquez 06 May '08

06 May '08

Hi everybody, I have some problems to get presence information exchanged between SIP users and xmpp ones. SIP clients (X-Lite) depend on an openser server v1.3.1, with all necessary modules working within it, and xmpp clients (Psi) rely on an xmpp server (ejabberd) which is in the same machine. The link between both sip and xmpp servers is established when openser starts, and the exchange of instant messages between sip and xmpp users works fine. So does presence too, as long as there are only sip users or only xmpp users involved, but it does not work between the two worlds in any direction. Moreover, I do not see any packet being exchanged between the sip and the xmpp servers when a user from one domain subscribe to one from the other, or when they change their status. I dont know what the problem can be. No errors appear in the log and I thought adding xmpp presence to openser would be straightforward once the IM was already working. Openser config file and part of the log file (the subscription to an xmpp user) are attached. Hope someone can give me some clue. Thanks in advance! Paul

2 11

[Serusers] Issue for subst()
by inge 06 May '08

06 May '08

Hi all, I find a solution to replace in my From the string "anonymous" by "92". So, I do the following : subst('/^(From|f): (.*) <sip:(.*)@(.*)>;/\1: \"Anonymous\" <sip:92@ \4>;/g'); The result is : From: "Anonymous" <sip:anonymous@From: "Anonymous" <sip:92@0172292965@domain.tld>;tag=21910a7967574581o0. I try a lot of pattern, but it doesn't run. Thanks for your support. Regards, Adrien

1 0

[Serusers] SER MySQL Memory Problem
by Arun Kumar 06 May '08

06 May '08

Hi All, I'm facing a strange issue: I've installed SER and MySQL on two machines, SER(200K users in subscriber table) is running with 6GB and MySQL with 5GB of memory but in just 2 days MySQL is using almost 5GB of memory and my SER response is getting very slow some time number of transaction waiting will be 1300+ :( any solution to this will be really helpful. thanks in advance. arun

1 0

[OpenSER-Users] null socket list
by Michael Barinek 05 May '08

05 May '08

I have openser-1.3.1-notls installed although am unable to start openser. Any suggestions/ideas regarding the below error? Any existing docs/links would be great. Thanks /// Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not rev. resolve 10.138.86.56 Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not rev. resolve 10.138.87.100 Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not rev. resolve 10.37.129.2 Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not rev. resolve 10.211.55.2 Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not rev. resolve 10.138.86.56 Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not rev. resolve 10.138.87.100 Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not rev. resolve 10.37.129.2 Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not rev. resolve 10.211.55.2 Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 10.138.86.56 [10.138.86.56]:5060 udp: 10.138.87.100 [10.138.87.100]:5060 udp: 10.37.129.2 [10.37.129.2]:5060 udp: 10.211.55.2 [10.211.55.2]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 10.138.86.56 [10.138.86.56]:5060 tcp: 10.138.87.100 [10.138.87.100]:5060 tcp: 10.37.129.2 [10.37.129.2]:5060 tcp: 10.211.55.2 [10.211.55.2]:5060 Aliases: tcp: my.machine.name:5060 udp: my.machine.name:5060 Mar 20 08:59:14 [18124] WARNING:core:main: no fork mode and more than one listen address found(will use only the first one) Mar 20 08:59:14 [18124] INFO:core:init_tcp: using kqueue as the TCP io watch method (auto detected) Mar 20 08:59:14 [18124] NOTICE:core:main: version: openser 1.3.1-notls (i386/darwin) Mar 20 08:59:14 [18124] INFO:core:main: using 32 Mb shared memory Mar 20 08:59:14 [18124] INFO:core:main: using 1 Mb private memory per process Mar 20 08:59:14 [18124] INFO:sl:mod_init: Initializing StateLess engine Mar 20 08:59:14 [18124] INFO:tm:mod_init: TM - initializing... Mar 20 08:59:14 [18124] CRITICAL:tm:uac_init: null socket list Mar 20 08:59:14 [18124] ERROR:tm:mod_init: uac_init failed Mar 20 08:59:14 [18124] ERROR:core:init_mod: failed to initialize module tm Mar 20 08:59:14 [18124] ERROR:core:main: error while initializing modules

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