sr-users September 2004

sr-users@lists.kamailio.org

190 participants
382 discussions

RE: [Serusers] NatHelper & Mediaproxy question.

by Ricardo Martinez

someone who can help me or guide me? Ricardo.- > -----Mensaje original----- > De: Ricardo Martinez [SMTP:rmartinez@redvoiss.net] > Enviado el: Lunes, 27 de Septiembre de 2004 06:56 p.m. > Para: 'serusers(a)lists.iptel.org' > Asunto: [Serusers] NatHelper & Mediaproxy question. > > Hello list. > I have reached a point where i need some advices and a little help. > I'm using mediaproxy to solve the NAT'd client "problems" this module > works > fine, and solves all the scenarios that i'm interested but for i noticed i > alway have to enable the mediaproxy, even for the calls between two > clients > with public.ip, this is a problem for me, i would expected that the media > in > this case goes directly between the endpoints. > Mi first question is : > Is this behaivor normal? Is that the way the mediaproxy works? > > Second Question. > If i use nathelper, would i have the same problem?. Among the nathelper > users do you have any similar experience. Does anyone knows if Nathelper > solve this issue?. > > I really, really hope that someone can help me with these questions. > Thanks in advance. > > Best Regards > Ricardo > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

20 years, 1 month

[Serusers] how can i forward sip message only once

by Chen Youcheng

hello, i am using Windows Message 5.0 as sip client. The sip message with the same call-id is sent three times. But i want to forward the message only once. Idee? Yongcheng Chen __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

20 years, 1 month

[Serusers] how can i forward sip message only once

by Chen Youcheng

20 years, 1 month

[Serusers] FW: Redirect to PSTN Gateway doesn't create an accounting

by Andrew K. Caruana

Dear all, Has this problem been solved? We are encountering the same problem. Any help shall be greatly appreciated. Thanks Andrew & Charles [Serusers] Redirect to PSTN Gateway doesn't create an accounting Kamen Sharlandjiev stone at netbg.com Mon May 31 11:37:28 CEST 2004 * Previous message: [Serusers] Redirect to PSTN Gateway doesn't create an accounting * Next message: [Serusers] Commercial Support / Development of SER * Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 have you set modparms like: db_flag, db_log, db_level... are you recording route? (I think yes if you recive BYE messages...) <-this is the hard part... INVITE and ACK messages are sample... just define this modparams ... and call him in your ser.cfg example: modparam("acc", "db_url", "mysql://$user:$pass@$hostname/$db_name") modparam("acc", "db_flag", 3) modparam("acc", "log_flag", 3) modparam("acc", "log_level", 3) ... if (!(method=="REGISTER")) if (loose_route()) { t_relay(); break; }; setflag(3); <<- HERE is the log flag set... and it will return to your mysql the INVITE and ACK ... if you allready recieve BYE ... your problem is solved. >On Monday 31 May 2004 12:08, Glynn Condez wrote: > Hi all, > > I'm thinking why my ser couldn't log accounting if I redirect a call to an > FXO voip gateway? > > I observed that I didn't saw any INVITE, ACK message on the logs except BYE > message. > > Here's my setup: > > sip ip phone ---> ser --- fx0 voip gateway > > heres my config: > > if ( (uri=~"^sip:123[0-9]*@.*") ) { > route(5); > break; > }; > > route[5] { > if ( (uri=~"^sip:123[0-9]*@.*") ) { > rewritehostport("fx0.voipgateway.com:5060"); > #rewritehostport("fx0.voipgateway.com:5090"); > forward(uri:host, uri:port); > break; > }; > } - -- Regards, Kamen Sharlandjiev System Administrator NetBG Communication Tel: +359 2 962 43 52 +359 2 962 53 93 - -- Public GPG key at: http://pgp.mit.edu pub 1024D/C6347D3D 2003-03-19 Kamen Sharlandjiev (Comment) <stone at netbg.com> Key fingerprint = D6AA 55BD F0CB FC6B 8C22 A13B 91D1 55C3 C634 7D3D -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFAuvzYkdFVw8Y0fT0RAuWIAKDAYQH/3rJSSauLVdcZyDgwftfuQwCfVsIT yxLpn7GacrE4Uf2hoVCaK/s= =kLuS -----END PGP SIGNATURE----- * Previous message: [Serusers] Redirect to PSTN Gateway doesn't create an accounting * Next message: [Serusers] Commercial Support / Development of SER * Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] More information about the Serusers mailing list

20 years, 1 month

[Serusers] Problem compiling radiusclient-ng on Solaris

by Thomas Jungbauer

hi all, it happens that i have to use an sun solaris 8 server for installing ser, radiusclient and freeradius. unfortunately i have got a problem while making radiusclient: make[2]: Entering directory `/var/home/tju/SER-Server/radiusclient-0.4.3/src' /bin/bash ../libtool --mode=link gcc -g -O2 -o radlogin radlogin.o radius.o local.o ../lib/libradiusclient.la -lcrypt -lsocket -lnsl gcc -g -O2 -o .libs/radlogin radlogin.o radius.o local.o ../lib/.libs/libradiusclient.so -lcrypt -lsocket -lnsl -R/usr/local/lib Undefined first referenced symbol in file strsep ../lib/.libs/libradiusclient.so ld: fatal: Symbol referencing errors. No output written to .libs/radlogin collect2: ld returned 1 exit status make[2]: *** [radlogin] Error 1 make[2]: Leaving directory `/var/home/tju/SER-Server/radiusclient-0.4.3/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/var/home/tju/SER-Server/radiusclient-0.4.3' make: *** [all] Error 2 normally the function strsep() should be declared at string.h....but solaris is different. there is no declaration of this function or i have not found it. does anyone had the same problem while compiling on a solaris? regards thomas PS: meanwhile i tried a quckfix and added the following code into %radiusclient%/lib/dict.c (which uses strsep): #ifndef HAVE_STRSEP static char * strsep(char **stringp, char *delim) { char *start = *stringp; char *cp; char ch; if (start == NULL) return NULL; for (cp = start; ch = *cp; cp++) { if (strchr(delim, ch)) { *cp++ = 0; *stringp = cp; return start; } } *stringp = NULL; return start; } #endif

20 years, 1 month

[Serusers] mssql

by Pavel Siderov

Is there a way to use mssql instead of mysql?

20 years, 1 month

[Serusers] handling REFER/transfers in multihomed mode

by Jeremy M. Dolan

Hi all. Wondering the proper way to handle REFER requests, as I haven't found much helpful documentation, and nothing in the way of examples. Our setup is basically: Cisco 7960s <---> SER <---> Vega Gateway <---> PSTN | Private 1918 net | Internet | (The Cisco phones are on a 1918 private network, SER is running on the IP gateway in multihomed mode, and the PSTN gateway is on the Internet) SER works fine for normal calls, with force_rtp_proxy("FAII") or force_rtp_proxy("FAEI") properly translating INVITE requests. However, if the phone receives an inbound call from the PSTN, and tries to transfer it, to either a local phone, or a remote number, this fails. The Refer-To: header is set to number(a)1918.ip.addr by the phone, e.g., 5551212(a)10.0.0.1, and then SER relays this to the Vega gateway unchanged. The Vega's INVITE attempt to that URI obviously gets nowhere. I don't see any function in SER or its modules that address this. All of the NAT and multihomed handling stuff seems only to translate the RURI, Contact header, and SDP payloads, never the Refer-To. I've gotten transfer to work now for certain cases by basically switching out the IP with a seemingly overbearing subst(): subst('/^Refer-To:(.*)@1918.ip.addr/Refer-To:\1@SERs.outside.ip/'); but this seems rather inelegant, to put it mildly. Isn't there another way to handle this in SER? Perhaps my relatively poor understanding of SIP at the protocol level is causing the problem here. As I understand it, in the above scenario, SER cannot "complete" the transfer request itself, so it *must* forward the REFER further on in the call path, that being the Vega here. It is not possible for SER to take the one half of the call path, from the Vega to SER, and close it (with a BYE, or however), and then connect the other half (which is still a live connection between SER and the user) to the new user specified in Refer-To:. Am I correct in thinking that this is how a SIP proxy (even one with RTP proxying and NAT mangling capabilities) works? For this, is a B2BUA needed? Though I hate to present any more info, especially without knowing first if my assumptions are flawed, there is one more highly related issue I may as well include here. We are relaying outbound calls through Masergy's VoIP service. Their servers, however, do not support REFER at all. What are our scenarios for being able to transfer (just among phones inside the office) outbound calls that we've made through Masergy? If I was correct in the preceding paragraph, SER itself could not help here. Is there a recommended way to handle this? Thank you in advance for ANY info in response to the above. Do any of the SER developers, or third-party gurus, offer any sort of paid support? I was surprised not to find anything under "services" on the iptel.org site. We might be interested, now or in the near future, in having someone who actually knows what they're doing take a little bit more in-depth look at our setup, and hopefully head off all of the problems we've undoubtedly set ourselves up for. :) Thanks again, Jeremy -- Jeremy M. Dolan <mailto:jmd@pobox.com> <http://jmd.us/> PGP: 1024D/3C68A1BA 9470 210C A476 FFBB 6D11 0223 0D1C ABFC 3C68 A1BA

20 years, 1 month

[Serusers] medisproxy error

by 牛坤

Dear serusers, I encountered the error mediaproxy/getSDPMessage(): SDP message has zero length. use_media_proxy();failed to get the SDP message What might cause the problem? Maybe it's a naive problem.:) Look forward to your reply. Sincerely, Kun

20 years, 1 month

[Serusers] Via Appending and t_relay vs. forward

by Michael Shuler

I am using a L4 switch to load balance across a few SER servers. The L4 switch has a virtual IP address on it that needs to appear as the Record-Route and I am assuming the Via headers. The record route is easily take care of by the record_route_preset() function and that seems to cause everything to work fine with the appropriate alias settings added so the myself variable works correctly but I don't really see how to control what SER uses as the Via IP. I am using the following to proxy the INVITE: if(!t_relay()) { xlog("L_INFO", "Failed sending requesting %rm URI (%ru)"); sl_reply_error(); }; I am assuming there is either some global setting that can be changed or some sortof rewrite function but I just can't seem to find it :( Any help would be appreciated!!! Thanks in advance! ---------------------------------------- Michael Shuler, C.E.O. BitWise Communications, Inc. (CLEC) And BitWise Systems, Inc. (ISP) 682 High Point Lane East Peoria, IL 61611 Office: (217) 585-0357 Cell: (309) 657-6365 Fax: (309) 213-3500 E-Mail: mike(a)bwsys.net Customer Service: (877) 976-0711

20 years, 1 month

[Serusers] radius accounting

by mohammad mirzaee

HI ALL; I have couple of sip UA registered in my ser box which is enabled with "acc module". I just want to do simple acconting when they call each other and to keep track the duration of call can anybody give me a sample ser.cfg which do accounting for just registered users Warmest Regards mohammad

20 years, 1 month

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sr-users September 2004