sr-users November 2007

sr-users@lists.kamailio.org

138 participants
248 discussions

[OpenSER-Users] [Fwd: Re: [OpenSER-Devel] pua module - unsubscribe]

by Anca Vamanu

-------- Original Message -------- Subject: Re: [OpenSER-Devel] pua module - unsubscribe Date: Tue, 13 Nov 2007 11:57:03 +0200 From: Anca Vamanu <anca(a)voice-system.ro> To: Reinhold Buchinger <reinhold.buchinger(a)gmail.com> References: <473420DF.8020809(a)gmail.com> <47342410.4080307(a)voice-system.ro> <47348B90.8070700(a)gmail.com> <47381EEA.6090702(a)voice-system.ro> <4738CC58.6040105(a)gmail.com> Hello, The succession on checks was wrong. Please test with the latest revision. Thanks and regards, Anca Reinhold Buchinger wrote: > Hi, > > Unfortunately, there is also an unwanted side-effect. If a > subscription is renewed expires is set to -1 for the call-back > parameter (infinite subscribes). In case of an error this also > triggers a new subscription with expires = 0.... > > regards, > Reinhold > > Anca Vamanu schrieb: >> Hello, >> >> It is not about retransmission there, but about the cases in which a >> request is sent inside a dialog stored by pua but which is not >> recognized by the client (the case of a client signing off and no >> Subscribe with expires= 0 sent to pua to announce it that the dialog >> has ended). The logics are like this: if someone asked me to send a >> request (for publish it's the same) and I matched it with a stored >> dialog and it failed, try again with an initial request. This is to >> ensure a a certain reliability. >> Therefore, in that case that you mentioned, when a second attempt is >> made, it is no longer about an unsubscribe. The request sent will be >> an fetching Subscribe(one with expires= 0)- I have made the changes >> to make this case possible. However,even if an unsubscribe was >> desired, the effects are the same, the dialog is already no longer >> valid, the pua record is deleted, and a Notify is sent. >> >> regards, >> Anca Vamanu >> >> Reinhold Buchinger wrote: >>> Hi, >>> >>> Thanks for your answer. But there is still a problem with >>> subs_cback_func(...) if a retransmission is triggered. This message >>> cannot be send with expires=0. >>> >>> subs.expires= (hentity->desired_expires>0)? >>> hentity->desired_expires- (int)time(NULL)+ 10:-1; >>> >>> Regards, >>> Reinhold >>> >>> >>> Anca Vamanu schrieb: >>>> Hello, >>>> >>>> The expires= 0 case is covered in subs_build_hdr(...) : >>>> >>>> if( expires<= min_expires) >>>> subs_expires= int2str(min_expires, &len); else >>>> subs_expires= int2str(expires+ 10, &len); >>>> >>>> if the module parameter min_expires maintains its default value, >>>> which is 0. >>>> I will test to see if indeed it is not possible now to unsubscribe. >>>> >>>> regards, >>>> Anca Vamanu >>>> >>>> Reinhold Buchinger wrote: >>>>> Hi, >>>>> >>>>> Using the pua module, I cannot send a subscription with expires=0. >>>>> If you look at subs_build_hdr(...) in send_subscribe.c this case >>>>> is not covered. >>>>> >>>>> Regards, >>>>> Reinhold >>>>> >>>>> _______________________________________________ >>>>> Devel mailing list >>>>> Devel(a)lists.openser.org >>>>> http://lists.openser.org/cgi-bin/mailman/listinfo/devel >>>>> >>>>> >>>> >>>> >>> >>> >> >> > >

16 years, 10 months

Re: [Serusers] Voicemail setup on SER v0.9.6

by roayu

Which version of SEMS should I download? Is it included the SER as well? I would prefer a stable version =) Thanks. Roa Yu -----Original Message----- From: SIP [mailto:sip@arcdiv.com] Sent: Monday, November 12, 2007 10:53 AM To: roayu Subject: Re: [Serusers] Voicemail setup on SER v0.9.6 You CAN have those on the same box. Although it's not required. You could have one SER on one box and then the other SER/SEMS on another box of you preferred. roayu wrote: > Hi Eneref, > > Thanks for your reply. So, do u mean that I'm gonna have the follwing > applications in the SAME box? > - 2 SER servers ( 1 existing SER + 1 new SER for SEMS ) with different port > - 1 SEMS > > Thanks. > > Roa Yu > > > -----Original Message----- > From: Eneref [mailto:eneref@arcdiv.com] > Sent: Friday, November 09, 2007 4:14 PM > To: roayu > Subject: Re: [Serusers] Voicemail setup on SER v0.9.6 > > Yes. You need either SEMS or Asterisk or some such. Meshing SEMS with > SER, though it LOOKS complex, is really not that bad. > > The easiest thing to do is to run another SER server to handle voicemail > (on a different port from the first SER if it's on the same box) and > forward messages to it if voicemail is required (on time outs, or > whatever conditions you set). There are some good instructions on how > to do that in the SEMS docs. > > This also makes it quite easy to swap out with Asterisk or any other > sort of voicemail-capable server should the need arise in future. > > N. > > roayu wrote: > >> Hi there! >> >> >> >> Could anyone let me know how to setup Voicemail on SER v.0.9.6 ? Do I >> need to install SEMS as well? >> >> >> >> Thanks >> >> >> >> >> >> Regards, >> >> Roa Yu >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

16 years, 10 months

[OpenSER-Users] how to implement calling between asterisk and asterisk?

by gonghai78

i am new to openser.my english is poor.my question is: xxxx@123 ---> openser() <---- xxxx@456 | | asterisk 123<----------->asterisk 456 how to implement calling between xxxx@123 and xxxx@456 in openser.cfg? thank you for your help!

16 years, 10 months

[Serusers] ERROR: warning_builder: buffer size exceeded & WARNING: warning skipped -- too big

by Tomasz Zieleniewski

Hi, I am using rel_2_0_0 from cvs. I encounter following errors in my ser output log: ERROR: warning_builder: buffer size exceeded WARNING: warning skipped -- too big Please tell what do they mean. Best regerds Tomasz

16 years, 10 months

Re: [Serusers] R: R: R: Loadbalancing for interco

by Jan Andres

Hi, On Mon, Nov 12, 2007 at 06:48:20PM +0100, Stefano Capitanio wrote: > Yes, it seems really strange also to me. > > I've tried with both SER-0.9.6 and OpenSER-1.2.2 on a Gentoo-linux > And in both cases there is the same problem... > > Do you think that can be a problem of installation/compiling? > Have ever heard something like that? Just a wild guess -- is there some network component where an application layer gateway (ALG) for SIP is enabled? Maybe the SIP ALG kernel module is loaded on the server itself and is doing the rewriting? Regards, Jan -- Jan Andres <jan.andres(a)freenet.ag> VoIP Systems Engineer phone: +49 431 9020-557 freenet Cityline GmbH, Hamburger Chaussee 2-4, D-24114 Kiel, Germany Ein Unternehmen der freenet AG Amtsgericht Kiel, HRB 6202 Geschäftsführer: Eckhard Spoerr, Axel Krieger --------------------------------------------------------------------- "The question of whether a computer can think is no more interesting than the question of whether a submarine can swim." -- E. W. Dijkstra

16 years, 10 months

[OpenSER-Users] question about the INVITE method

by 程千

hi: I isntall the openser-1.2.2-notls under the centos5. I config the INVITE method like this: if (is_method("INVITE")) { # enable Record-Route record_route(); # Is this to one of our numbers if (lookup("location")) { append_hf("P-hint: userloc applied\r\n"); } else { sl_send_reply("404","Not found"); } } route(1); ... route[1] { if (isflagset(2)) { if (is_method("INVITE")) { t_on_reply("1"); t_on_failure("1"); use_media_proxy(); } else if (is_method("BYE|CANCEL")) { end_media_session(); } } # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); } exit; } But when the end ua is not online,the ser proxy seems to be in a loop lock. Thanks a lot!

16 years, 10 months

[OpenSER-Users] TCP better than UDP??

by lu

Hi, I came accros one Nokia-S60 Guideline regarding SIP setting in the NAT/Firewall traversal section which say:"It is recommended to use TCP as the transport instead of UDP since even doubled battery life can be achieved with a UI always connected to a SIP service". I need OPenSER users to comment on this since what i know is that UDP is more reliable transportation. WBR lu.

16 years, 10 months

[OpenSER-Users] ERROR: receive_fd: EOF on 15

by Dave Singer

ERROR: receive_fd: EOF on 15 child process 26410 exited by a signal 11 core was generated INFO: terminating due to SIGCHLD INFO: signal 15 received .... repeated ... INFO:mi_fifo:mi_destroy: seems that fifo child is already dead! I've added xlog lines to narrow down where it is crashing. It hasn't crashed since that so not sure exactly where it is crashing in the script. I think it is crashing some where around a avp_db_load: if ( avp_db_load("$ru/username","*/usr_preferences_to") ) It doesn't always crash on setting up a call... just when I'm not watching for it. Thanks Dave PS my first time posting to a user list so not sure just how things work for it. Pointers welcome. here is some pert info about my install: version: openser 1.2.1-notls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 1827 2007-03-12 15:22:53Z bogdan_iancu $ main.c compiled on 11:12:59 Sep 6 2007 with gcc 4.1.1

16 years, 10 months

[Serusers] Some calls do not go to voicemail

by Steven C. Blair

We are experiencing a problem for which we can use some help. When a subscriber receives a call from either the PSTN or another IP phone we set the inv_timeout to a value which represents about four rings. For most subscribers an unanswered call will timeout after the specified number of rings then, via a failure route, their call will be re-route to our Asterisk server for voicemail. Some subscribers, no set pattern of users, have their calls timeout and the caller experience silence for about 45 seconds. During this period the caller usually hangs up. I cannot seem to figure out why. We set modparam("tm", "fr_inv_timer_avp", "inv_timeout") and then avp_write("20", "inv_timeout"); before relaying an incoming call. Has anyone else experienced any similar problems? 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

16 years, 10 months

[Serusers] SER 0.9.6 "desc_time_order" issue

by Mitya

Hi Everybody, I would like to set up thefollowing with ser 0.9.6 1) an UAC should be able to register more times with the same q value note: lets say, there is a power outage on the client side and it gets a new IP from the ISP, registers again, the "old" registration has not expired yed, it is still alive, because it did not unreggistered itself 2) when SER calls this UAC, only the most recently registered one should be invited 3) to limit the maximum number of contacts is not possible (I need this feature) From the documentation: ------------------------ ----- CUT ------ 1.3.5. desc_time_order (integer) [...] If set to 1 then all contacts will be ordered in descending modification time order. In this case the most recently updated/created contact will be used. [...] 1.3.3. append_branches (integer) [...] If the parameter is set to 0, only Request-URI will be overwritten with the highest-q rated contact and the rest will be left unprocessed. [...] ----- CUT ------ I also found this thread: ------------------------- http://lists.iptel.org/pipermail/serusers/2003-March/000758.html From Jan: You will have to use modparam("registrar", "append_branches", 0) in your script and after that only the most recently updated/created contact will be used (even if there are many of them). This is exactly what I need !! my relevant config is the following: modparam("usrloc", "db_mode", 1) modparam("registrar", "append_branches", 0) modparam("registrar", "desc_time_order", 1) modparam("registrar", "max_contacts", 50) Sympthom: --------- When I call the UA, always the firstly registered one gets the INVITE, and "desc_time_order" DOES NOT work :( Can sy explain me what should I do now ? Many thanks. Mitya

16 years, 10 months

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sr-users November 2007