sr-users February 2006

sr-users@lists.kamailio.org

189 participants
373 discussions

[Users] Question on realm_prefix with multiple prefixes

by Barry Flanagan

Hi, I seem to recall that realm_prefix for the various modules could take multiple prefixes like: modparam("auth", "realm_prefix", "sip.|sip2.|sip3.") ...but this does not appear to work. Does this capability exist, and if so, what is the syntax? All the docs examples only show a single prefix. Thanks. -Barry Flanagan

18 years, 9 months

[Users] newbie-nat

by Mark Anthony C. Delfin

Hello openser users I just installed openser 1.0 , rtpproxy, openser=mysql on centos. calls can be made between UAs with no NAT between them but if one UA is behind nat calls are unsuccessful. does anyone have a sample config for nat calls using openser + rtpproxy. Thanks in advance mark

18 years, 9 months

[Users] New Feature - Global Statistics Management support

by Bogdan-Andrei Iancu

Hi everybody, A new feature is present on the development version of openser - Global Statistics Management support: the core includes a Statistics Manager which roll is to collect statistics variables from all over openser and to offer management support (to create and operate them); it also provides a single, centralized point of access to allow external apps or internal modules to get any statistics. How it works? the core and the module exports different statistics variables (via the module interface). The Statistics Manager (SM) offers support for auto-initialization, for updating ans resetting variables. The statistics variables are kept into shared mem to be accessible to all processes. Synchronization support is offered by SM. Statistic variable capabilities ------------------------------- be defined with: NO_RESET (it cannot be reset -ex. used mem, register subscribers, etc) NO_SYNC (it will not be synchronized during ops; by their nature, some variables are already synchronized) be updated (added with new val) be reset (set to zero if allowed). Accessing the statistics ------------------------- The SM export two FIFO functions for offering access to the statistics: "get_statistics all" - gets all statistics "get_statistics stat_name" - gets the value of the stat_name variable "get_statistics module_name:" - get all statistics exported by module "module name"; to get statistics from core, use "core" as module_name "reset_statistics stat_name" - sets to 0 the value of the stat_name variable The SM internally offers an API to be used by other modules interested in accessing the statistics. Ex: a future snmp module Current statistics ------------------ For the moment only a few core statistics are defined (we can easily add them in time): received requests : "rcv_reqs" received replies : "rcv_rpls" dropped requests (by scripts or post-script callbacks) : "drp_reqs" dropped replies (by scripts or post-script callbacks) : "drp_rpls" error requests (basic parsing failed) : "err_reqs" error replies (bassic parsing failed) : "err_rpls" How to use it -------------- first to need to enable its compiling - enable the -DSTATISTICS flag in Makefile.def (by default is disabled). Recompile and run openser. Use fifo to get the statistics. Ex: openserctl fifo get_statistics all openserctl fifo get_statistics core: openserctl fifo get_statistics rcv_reqs Next steps ----------- optimization of the SM (better locking for sync, usage of atomic ops if possible) replace old statistics (see stats.c) - the code is not compiling since it depends on some snmp stuff... replace local module statistics (sl,tm,usrloc) with sttistics via SM add more statistics...ideas are welcomed new module to implement dynamic statistics variables to be used from script. regards, bogdan

18 years, 9 months

[Serusers] grandstream budgetone not able to make/take calls

by ryan nalupa

hello all, i'm ryan and still a SER noob. copied some cfg files from onsip.org and from voip-info.org. basically, i'm using two softphones, at first they're all ok. can make a call, can accept...when i try it on my budgetone hardphone, my desktop phone registers but cannot make a call, can't take calls either. as of now i still don't know what to do with it yet, can anyone help/guide me on how to configure this little problem? i'm using sarge and ser-0.9.4.... is this an issue with NAT? i still don't know how to config it yet, pls, help me anyone? :) ryan __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

18 years, 9 months

[Users] Avoid loop on call forwarding

by Jens Carl

Hay list, I've got following scenario: I have call forwarding with the help of OpenSER. There are no problems with forwarding within the SIP network. Also the forwarding to a PSTN destination is possible if the caller is from the SIP network. And also the forwarding to a SIP destination if the call comes from a PSTN destination. The only problem is when the caller is from the PSTN network and the callee tries to forward this call to another PSTN destination. one server one server Asterisk* OpenSER | | call: 12 | call: SIP43 | --------->|------------->| look for forwarding and find 56 | | makes new branch with new found R-URI call: 56 | call: 56 | relay the call to the PSTN gateway <---------|<-------------| | | * ASTERISK works as gateway (incoming and outgoing calls to PSTN) That should be the the chain of the call but the OpenSER/Asterisk detects a loop and the call is dropped. But this will be the most used option of your call forwarding functionality. Is there a possibility to avoid these loop? And how realise this? Best regards Jens

18 years, 9 months

[Users] Natping in "dispatcher scenario"

by Helge Waastad

Hi, I just wonder: I have a demo scenario in which I have a dispatcher (and using path header) ahead of the registrar(s) and other fun stuff. I am currently implementing the NAT ping function in the registrar(s), but the nathelper function just sends the ping to the contact stored in location. This, of course, does not work in fw scenarios since the relation between the UA and the SIP server is the dispatchers IP:port. Does anyone else find it interesting to have the ping also relayed via the OBP? If not, I just have to move the registrar more towards the frontend. Br, Helge Waastad Senior Engineer Smartnet

18 years, 9 months

[Serusers] SER: Measuring ASR & PDD

by Jean-Michel Hiver

Hi List, At the moment, I log 'INVITE', 'ACK' and 'BYE' with this: # ----------------------------------------------------------------- # Sets the acc accounting flag 4 billing # ----------------------------------------------------------------- if( (method=="INVITE") || (method=="ACK") || (method=="BYE")) { setflag(1); }; To measure (approximately) the ASR, I think it is sort of reasonable to use the formula (number of INVITES) / (number of BYEs + number of INVITES) The reasoning behind this being that INVITES which have been BYEd have necessarily been answered. Now, how would you measure Post Dialing Delay? To be doing that I would need to record the 'Ringing' signal but I don't know how to do this. Any ideas? Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

18 years, 9 months

[Users] DISPATCHER not sending outgoing INVITES 100% of the time

by Anders Brownworth

Hi, I haven't been able to get an outgoing INVITE for every incoming INVITE my dispatcher setup receives. What I'm expecting: When the first INVITE comes from 10.1.50.30 (Asterisk) to 10.1.50.31, (OpenSER 1.0.0 with dispatcher) I want to see an outgoing INVITE from 10.1.50.31 to one of the listed addresses. (10.1.50.[34-37]) What I'm seeing: A tcpdump shows that most of the time (90%) all I see is the incoming INVITE. Seemingly random INVITES produce an outgoing INVITE. Presumably the only substantive difference being the CallID. Interestingly, all the outgoing INVITEs are addressed to the same IP. (10.1.50.36) Am I not expecting the right thing? Can anyone shed some light on what I'm doing wrong here? Thanks. openser.cfg: ----------------------------------------------------------------------------- #debug=4 fork=yes log_stderror=no children=4 check_via=no dns=off rev_dns=off port=5060 mpath="/usr/local/lib/openser/modules" loadmodule "maxfwd.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "dispatcher.so" modparam( "dispatcher", "list_file", "/usr/local/etc/openser/dispatcher.list" ) route { if ( ! mf_process_maxfwd_header( "10" ) ) { sl_send_reply( "483", "To Many Hops" ); drop( ); }; ds_select_domain( "1", "0" ); forward( uri:host, uri:port ); } ----------------------------------------------------------------------- dispatcher.list: ----------------------------------------------------------------------- # gateways 1 sip:10.1.50.34:5060 1 sip:10.1.50.35:5060 1 sip:10.1.50.36:5060 1 sip:10.1.50.37:5060 ----------------------------------------------------------------------- tcpdump shows: ----------------------------------------------------------------------- 07:43:37.216888 IP 10.1.50.30.5060 > 10.1.50.31.5060: UDP, length: 685 . at . <http://openser.org/cgi-bin/mailman/listinfo/users>@... .2. .2.........INVITE sip:+18666775910 at 10.1.50.31 <http://openser.org/cgi-bin/mailman/listinfo/users> SIP/2.0 Via: SIP/2.0/UDP 10.1.50.30:5060;branch=z9hG4bK434da07b From: "+19195551212" <sip:+19195551212 at 10.1.50.30 <http://openser.org/cgi-bin/mailman/listinfo/users>>;tag=as3a379ead To: <sip:+18666775910 at 10.1.50.31 <http://openser.org/cgi-bin/mailman/listinfo/users>> Contact: <sip:+19195551212 at 10.1.50.30 <http://openser.org/cgi-bin/mailman/listinfo/users>> Call-ID: 7dd7978262a876832fc69a4364f89f22 at 10.1.50.30 <http://openser.org/cgi-bin/mailman/listinfo/users> CSeq: 102 INVITE User-Agent: BandwidthVoice Date: Fri, 03 Feb 2006 12:41:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 202 v=0 o=root 11014 11014 IN IP4 10.1.50.30 s=session c=IN IP4 10.1.50.30 t=0 0 m=audio 8360 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - ----------------------------------------------------------------------- with no outgoing INVITE. Things seem to be dead. Then I try the same call a few seconds later and I get the outgoing INVITE, but always to the same IP (10.1.50.36) ----------------------------------------------------------------------- 07:43:48.695958 IP 10.1.50.30.5060 > 10.1.50.31.5060: UDP, length: 686 . at . <http://openser.org/cgi-bin/mailman/listinfo/users>@... .2. .2.........INVITE sip:+18666775910 at 10.1.50.31 <http://openser.org/cgi-bin/mailman/listinfo/users> SIP/2.0 Via: SIP/2.0/UDP 10.1.50.30:5060;branch=z9hG4bK15490a4e From: "+19195551212" <sip:+19195551212 at 10.1.50.30 <http://openser.org/cgi-bin/mailman/listinfo/users>>;tag=as302ac772 To: <sip:+18666775910 at 10.1.50.31 <http://openser.org/cgi-bin/mailman/listinfo/users>> Contact: <sip:+19195551212 at 10.1.50.30 <http://openser.org/cgi-bin/mailman/listinfo/users>> Call-ID: 531db3695f15035620baae890d555216 at 10.1.50.30 <http://openser.org/cgi-bin/mailman/listinfo/users> CSeq: 102 INVITE User-Agent: BandwidthVoice Date: Fri, 03 Feb 2006 12:41:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 203 v=0 o=root 11015 11015 IN IP4 10.1.50.30 s=session c=IN IP4 10.1.50.30 t=0 0 m=audio 28484 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - 07:43:48.696191 IP 10.1.50.31.5060 > 10.1.50.36.5060: UDP, length: 747 E....*@. at ..g <http://openser.org/cgi-bin/mailman/listinfo/users> .2. .2$......S.INVITE sip:+18666775910 at 10.1.50.36 <http://openser.org/cgi-bin/mailman/listinfo/users>:5060 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.1.50.31;branch=0 Via: SIP/2.0/UDP 10.1.50.30:5060;branch=z9hG4bK15490a4e From: "+19195551212" <sip:+19195551212 at 10.1.50.30 <http://openser.org/cgi-bin/mailman/listinfo/users>>;tag=as302ac772 To: <sip:+18666775910 at 10.1.50.31 <http://openser.org/cgi-bin/mailman/listinfo/users>> Contact: <sip:+19195551212 at 10.1.50.30 <http://openser.org/cgi-bin/mailman/listinfo/users>> Call-ID: 531db3695f15035620baae890d555216 at 10.1.50.30 <http://openser.org/cgi-bin/mailman/listinfo/users> CSeq: 102 INVITE User-Agent: BandwidthVoice Date: Fri, 03 Feb 2006 12:41:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 203 v=0 o=root 11015 11015 IN IP4 10.1.50.30 s=session c=IN IP4 10.1.50.30 t=0 0 m=audio 28484 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - -

18 years, 9 months

[Users] CVS troubles...yet again

by Helge Waastad

Hi, just wonder if something is to be done regarding the cvs? I am experiencing problems updating the source. To day, I got can't create temporary directory /tmp/cvs-serv28859 No space left on device ....and I'm dying to get a bug-fix this morning :-) br hw -- Helge Waastad Senior Konsulent Systemavdelingen Smartnet

18 years, 9 months

[Users] postgres and long AVPs values BUG?

by JF

Hi, I increased the length of the value column in the usr_preferences table from 128 to 1024, but I got a core dump after these logs: 0(24138) val2str(): converting record_route, 12 0(24138) PG[224] str2valp got string <sip:10.112.64.59;ftag=605ff764c617d3cd28dbbdd72be8f9a2;lr=on>,<sip:10.112.64.12;ftag=605ff764c617d3cd28dbbdd72be8f9a2;lr=on>,<sip:10.112.64.12;ftag=605ff764c617d3cd28dbbdd72be8f9a2;lr=on>,<sip:10.112.64.59;ftag=605ff764c617d3cd28dbbdd72be8f9a2;lr=on> Segmentation fault (core dumped) This was caused by an avp_db_load() call from the script. I'm using OpenSER 1.0.0. I found the following comment on modules/postgres/db_res.c that might have something to do with it: /* I'm not sure about this */ /* PQfsize() gives us the native length in the database, so */ /* an int would be 4, not the strlen of the value */ /* however, strlen of the value would be easy to strlen() */ Also, in modules/postgres/db_val.c, there is a FIXME in val2str() function that seems to be related. Can anyone help solve this problem? It seems like a bug. Thanks, JF

18 years, 9 months

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sr-users February 2006