sr-users May 2005

sr-users@lists.kamailio.org

230 participants
522 discussions

[Serusers] Re: SIP load balancing cont..

by a c

this worked with t_check_status(). One thing I could not figure out is how to get a log printout when I receive a 200 OK. How do I know if I received a 200 OK? I can't put this logic in the main route block. thank you ac --- a c <lra101(a)yahoo.com> wrote: > > I need to route to a failure route when there is a > certain kind of failure. For example, If I don't > receive a 'Trying' back on the initial attempt{no > response to Invite}, then I would route using the > failure route. For all other failures {i.e. 404, no > answer, 486, etc..}, I don't try any other routes. > How > can this be configured? > > thank you > ac > > > > Discover Yahoo! > Have fun online with music videos, cool games, IM > and more. Check it out! > http://discover.yahoo.com/online.html > __________________________________ Do you Yahoo!? Make Yahoo! your home page http://www.yahoo.com/r/hs

19 years, 6 months

[Serusers] session Timers

by Jorge Crichigno

Hi all, I have a a doubt related with RFC 4028 - Session Timers in the Session Initiation Protocol (SIP). I am developing a new ser module that supports this RFC. While doing tests, I found the following situation: phone1(addpac) SER phone 2 (grandstream) Invite -> <-422 (Lower Min-SE) Ack -> Invite -> Invite -> <-481 (no such call) <-481 (no such call) What I saw was that, in the second Invite, the addpac sends the Invite with the the "to tag" that was attached by SER before (422 response). I belive that this is the reason of the 481 response from grandstream. I think it is a bug of Addpac. Is that correct? Thanks in advance!

19 years, 6 months

[Serusers] Bug in ucontact.c?

by Francesco Bottà

Hi all, I've a question about removing alias from memory and DB. I've a SER instance in WRITE_BACK DB Mode (db_mode=2) and timer_interval param set to 10 seconds. When I should remove an alias from memory&DB (with 'serctl rm alias 'username'), in WRITE_BACK mode, the expire time of this cancelled alias go to something like 'expires=-1116854859' and increase in time but never disappear from memory neither from my DB...indeed, if I set WRITE-THROUGH mode (db_mode=1) all works ok, and the cancelled alias disappears immediately from mem and DB.... I've this SER ver installed: ser -V version: ser 0.9.2 (i386/linux) flags: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $ main.c compiled on 11:28:47 May 6 2005 with gcc 3.3 Any hints? What I'm missing? Many thanx Verbal

19 years, 6 months

Re: [Serusers] error in compile ser-0.9.0

by Kamran Ahmad

i am using bison-2.0.tar.gz and flex-2.5.4a.tar.gz but unable to solve this issue #ps -A 8961 ? 00:00:00 ser <defunct> 8962 ? 00:00:00 ser <defunct> 8965 ? 00:00:00 ser <defunct> 8966 ? 00:00:00 ser <defunct> 8968 pts/1 00:00:00 ps # serctl moni Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config --- Grigory Fishilevich <g.fishilevich(a)gmail.com> wrote: > hi, > > these messages have nothing to say. i think you have > a permissions > problem, you should take a look at /var/log/messages > > ? have you ser running or not? > __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

19 years, 6 months

[Serusers] SUBSCRIBE -> BAD REQEST (Instant Communication Application + PA Module)

by Jaroslaw Gawron

Hello Everybody I am trying to create Instant Communication Application based on MS RTC 1.2 library. I am also using SER 0.8.14 as a SIP server. Voice communication works perfect, messages works perfect but I have a big problem with sending user's status. When I tray to send SUBSCRIBE message I get BAD REQEST response. T 10.1.1.62:3749 -> 10.1.3.10:5060 [AP] SUBSCRIBE sip:pszymanek@sippbx.bb.wasko.pl SIP/2.0 Via: SIP/2.0/TCP 10.1.1.62:8718 Max-Forwards: 70 From: "yarek" <sip:yarek@sippbx.bb.wasko.pl>;tag=d290de95c2124ceca42a748963dd04e9;epid=885 a5b4691 To: <sip:pszymanek@sippbx.bb.wasko.pl> Call-ID: fb0ac597ee2b480f9315e88a9e6bcb06(a)10.1.1.62 CSeq: 2 SUBSCRIBE..Contact: <sip:yarek@sippbx.bb.wasko.pl:8718;maddr=10.1.1.62;transport=tcp> User-Agent: RTC/1.2..Expires: 0 Event: presence..Content-Length: 0 # T 10.1.3.10:5060 -> 10.1.1.62:3749 [AP] SIP/2.0 400 Bad Request Via: SIP/2.0/TCP 10.1.1.62:8718 From: "yarek" <sip:yarek@sippbx.bb.wasko.pl>;tag=d290de95c2124 ceca42a748963dd04e9;epid=885a5b4691 To: <sip:pszymanek@sippbx.bb.wasko.pl>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-3fbe Call-ID: fb0ac597ee2b480f9315e88a9e6bcb06(a)10.1.1.62 CSeq: 2 SUBSCRIBE..Error while parsing headersServer: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 10.1.3.10:5060 "Noisy feedback tells: pid=13753 req_src_ip= 10.1.1.62 req_src_port=3749 in_uri=sip:pszymanek@sippbx.bb.wasko.pl out_uri=sip:pszymanek@sippbx.bb.wasko.pl via_cnt==1" I configured the presence module, added the lines to ser.cfg if (method=="SUBSCRIBE") { log(1, "Subscribe\n"); if (t_newtran()) { log(1, "Registrar\n"); handle_subscription("registrar"); }; break; }; But it still doesn't work. Has anyone faced a similar problem ? Why it doesn't work ? Is the PA module compatible with MS RTC. Why this masseges are in bad format. Any ideas are very welcome. Best regards Jarosław Gawron email: J.Gawron(a)wasko.pl

19 years, 6 months

[Serusers] Append a header field with dynamic data

by Michael Stier

Hi, I want to append a new header field on a 183 - session progress response with dynamic data: P-hint: dynamic data The string "dynamic data" e. g. could be generated by an external script. Is there any way to reach this inside ser.cfg? My imagination would be something like this: if (status=~"183") { exec_msg('TEMPSTRING=`cat /tmp/string.txt`'); append_hf("P-hint: $TEMPSTRING"); }; Many greetings Michael

19 years, 6 months

Re: [Serusers] sip load balancing

by a c

hello Iqbal, just need the SIP messages load balance across servers. No RTP. thank you ac --- Iqbal <iqbal(a)gigo.co.uk> wrote: > mediaproxy can do it using dns. Am assuming you want > the rtp stream load > balanced, if its the sip messages, search throuygh > the archives there is > along discussion bout howto use LVS and load > balancing....in short > doesnt seem to work, cause SIP uses udp > > Iqbal > > a c wrote: > > >hello, > > > > got a question on SER. Could SER do load balance > >across SIP servers? > > > >for example: > > > >SIP carrier --> SER --> Media/App Servers > {multiple} > > > >Are there any Configuration examples I can look at? > > > >thank you > >ac > > > > > > > >__________________________________ > >Yahoo! Mail Mobile > >Take Yahoo! Mail with you! Check email on your > mobile phone. > >http://mobile.yahoo.com/learn/mail > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > > >. > > > > > > > __________________________________ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250

19 years, 6 months

Re: [Serusers] Advice needed

by Samuel Osorio Calvo

just an idea: does any PSTN provider supports TLS? If some does you have your auth problem solved with the TLS support from SER. Samuel Unclassified. >>> "Greger V. Teigre" <greger(a)teigre.com> 05/22/05 08:57AM >>> See inline. Michael Ulitskiy wrote: > On Saturday 21 May 2005 02:31 am, you wrote: >> I would say SER is what you need, except that you struggle with the >> authentication. You have the following scenarios: >> 1. PSTN termination with IP-based access control (easiest) >> 2. PSTN termination with authentication of all INVITEs (yes, that's >> the UAC module. You should contact the maintainer, Ramona-Elena >> Modroiu about the status. I thought it was reported to work, but >> haven't tried myself) >> 3. PSTN termination with registration and authentication of REGISTER >> (but not INVITEs). Use sipsak to generate a REGISTER for your box. >> >> #2 requires that all INVITEs are sent twice and is not a very good >> option. I would seek out PSTN providers who will give you #1. >> g-) > > UAC module doesn't work and I think won't work unless ser is made > call-statefull, 'cause it needs to adjust cseq within dialog. I > posted my findings to this list > several days ago (UAC module (backport to 0.9.0). Nobody replied so I > guess > nobody knows the way to make it work. I saw your post on serusers, yes, but not on serdev. Because you cannot make a module work, doesn't mean it doesn't work for all, so as I said, if you have found a bug, post it to serdev (preferably) or directly to the maintainer. That's the way open source software work... > As for ip auth I guess it's just not good enough. UDP invites don't > require any handshake it's not hard at all to spoof ip address. I > believe sending 2 invites worth the security it actually adds. Yes, but you can also do TCP. > Also I don't understand what you mean by #3. Taking ip address from > authenticated REGISTER and then doing IP auth on that? No, using sipsak to actually do a REGISTER on behalf of your ser. No IP auth, basically it makes your ser a registered client of the GW. Of course, if INVITEs still must be authenticated, you are back to the UAC module problem. g-) > Thanks, > > Michael > >> Michael Ulitskiy wrote: >>> Hello, >>> >>> I'd like ask for advice on what is in your opinion the best solution >>> in the following scenario. >>> I have a bunch of sip servers (asterisk boxes as my users need pbx >>> functionality) that can make sip call to each other and my PSTN >>> gateway. Now I want to purchase PSTN terminitaion in several >>> different markets (and probably more in the future). All those >>> terminations will require authentication. >>> I want all my boxes when they see non-local call to send it to a >>> central routing server that would determine where this call should >>> be sent and authenticate to the appropriate provider so that I don't >>> have to configure all credentials on all asterisk boxes. Also I want >>> it not to deal with the media at all. All media streams should go >>> directly from asterisk box to the PSTN termination provider. >>> So basically it should be central SIP router that is able to >>> authenticate calls if neccessary. >>> I thought I could do it with SER and its UAC module, but it appears >>> UAC module doesn't work and probably won't work (see my previous >>> post in this list about UAC backport to 0.9.0). >>> Also I don't want to use asterisk in this place as asterisk always >>> wants to stay in media path and I'd really like to avoid of getting >>> into hassle with re-invites. >>> So the question is what are my options and what you would advice >>> as a solution. Are there any software out there that can do it >>> (preferably open-source, of course) or what else you could suggest >>> to do to get desired results. >>> Thanks a lot, >> >> > > -- > See you later, > Michael > > > ------------------------------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

19 years, 6 months

Re: [Serusers] exec module question

by Samuel Osorio Calvo

Hi, I am not sure if SER config file get the error code from the exit() call of your script but I just think so. Assuming it gets the return code, youn should be aware that only three possibilities can be checked from the config file: ret>0 --> config command was successfull ret<0 --> ERROR but: ret=0 --> drop message!!! So, if you return 0 from your script, the message will be dropped, just af it a break had occur... To allow this structure: if (exec_msg("/usr/local/bin/check_register.pl")) { break; } else { if (!radius_www_authorize("")) { You should return a positive value if the scrpit ended OK or a negative value if an error occurred. BTW, using exec|* commands can be a worse solution.......just take a look at older posts about this subject. Hope it helps, Samuel. Unclassified. >>> Daniel Corbe <daniel.junkmail(a)gmail.com> 05/20/05 05:31PM >>> Hello, I've noticed that 90% of my SIP traffic is REGISTER requests. I am trying to implement some sort of caching mechanism (which will inherently be faster than doing a RADIUS lookup on every REGISTER request) using exec and a perl script which does a lookup from a memory-cached file. I'm using exec_msg() to call the perl script; however it is not behaving as I would expect. Here's the code snippet from my SER config. if (exec_msg("/usr/local/bin/check_register.pl")) { break; } else { if (!radius_www_authorize("")) { I am just in preliminary testing stage so I have the perl script merely exiting with 0 status. exec documentation is not clear as to whether it uses the shell return codes to determine the exit status of exec_msg If I am exit(0)ing from the perl script I would expect the if( to fail, not break and do the radius_www_authorize. Is that not correct? Thanks for the help! Regards, Daniel Corbe _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

19 years, 6 months

[Serusers] Why does the memory occupied by SER process not decrese?

by lemon

I am using ser 0.8.14. I wrote a program to test the stability of SER. The scenario like this caller----SER----callee. Caller program will send INVITE continuously.Callee program answer each call proxyed by SER. And then Caller program send BYE to finish the call. I run 'top' tool in linux to monitor the usage of memory and cup by SER. I found that with the call establishing speed increasing, the memory occupied by SER process increasing too.After I shut down the test program, the memory ocupied by SER process didn't decrease. Can anyone tell me why? And Does is mean memory leak? Best regards! Karman Wu ---------------------------------------------- 21CNÓÊÏäÈ«ÃæÉýG http://mail.21cn.com/huodong/0504/mail/ ²ÊÆÁÊÖ»ú°×ËÍÀ²!žÏ¿ìÀŽÄÃ! http://qipai.g.21cn.com ÖÇ»ÛŽóÌôÕœŸÍÔÚµÍŒÛ¶á±Š http://super.21cn.com/ ÌåÑéÁíÀàÔŒ»á£¬žÐÊÜ±ðÑùÀËÂþ http://y.21cn.com/club/ Ð¡ÁéÍš¶ÌÐÅÖÐÐÄ£¬¶ÌÐÅ÷ÈÁŠÎÞŒ« http://pas.21cn.com/ È«ÄÜÁÄ2005°æÉÁÁÁµÇ³¡£¡ http://callme.21cn.com

19 years, 6 months

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sr-users May 2005