- sr-users - kamailio.org

[OpenSER-Users] OpenSER 1.3 - release name
by Bogdan-Andrei Iancu 30 Nov '07

30 Nov '07

Hi, I would like to make a survey to see if people will consider a nice policy to give to the further openser releases not only version numbers, but also names. If so, I would like to be the first to suggest a name for the upcoming openser 1.3 release -"blackcat" (mainly because of the 13 version :D ). Any other suggestions or comments on the topic are welcomed. Thanks and regards, Bogdan

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[OpenSER-Users] OpenSER 1.3 - release date
by Bogdan-Andrei Iancu 30 Nov '07

30 Nov '07

Hi, There are still some bugs that need to be fixed before the release of 1.3. I think 2 weeks should from now on should be enough, so I suggest to try to set the release date to 13th of December. Please let me know your thoughts. Regards and a nice weekend, Bogdan

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[OpenSER-Users] does CPL change SDP parameter on reply message?
by Daniel 30 Nov '07

30 Nov '07

Hi all, if I use <PROXY> in a CPL script, then, how can I change SDP parameter (I want to use media proxy ) in the reply message (200 OK) ? regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] CPL and on reply route
by Daniel 30 Nov '07

30 Nov '07

Hi all, I use CPL script with "Proxy" element, like this: <cpl> <incoming> <lookup source="registration"> <success> <proxy ordering="sequential" /> </success> </lookup> </incoming> </cpl> and I use onreply_route(), where I make the changes on 180/200 reply messages in case of NAT (change callee's SDP parameter and change Callee's Contact hf). When an INVITE is forwarded, through the <PROXY> element, to the callee, then reply messagges (180 Ringing and 200OK) hasn't processed by Onreply_route. So I can't make my NAT control. Why? Regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] openser 1.3 sometimes fails to start
by Robert Dyck 30 Nov '07

30 Nov '07

I am testing openser 1.3.0-pre1-notls svnversion 2:3201. I believe it is failing while starting the child processes for the modules. If it starts successfully it seems to perform normally. I believe the following excerpt from the console might hold the clue. Thanks Rob Nov 29 14:40:30 [10492] DBG:xlog:child_init: init_child [0] pid [10492] Nov 29 14:40:30 [10492] DBG:core:init_mod_child: type=PROC_MAIN, rank=0, module=avpops Nov 29 14:40:30 [10498] DBG:core:io_watch_add: io_watch_add(0x8160900, 6, 4, 0xb5f824a4), fd_no=0 Nov 29 14:40:30 [10498] DBG:core:io_watch_add: io_watch_add(0x8160900, 7, 4, 0xb5f82530), fd_no=1 Nov 29 14:40:30 [10498] DBG:core:io_watch_add: io_watch_add(0x8160900, 8, 4, 0xb5f825bc), fd_no=2 Nov 29 14:40:30 [10498] DBG:core:io_watch_add: io_watch_add(0x8160900, 9, 4, 0xb5f82648), fd_no=3 Nov 29 14:40:30 [10498] DBG:core:io_watch_add: io_watch_add(0x8160900, 12, 4, 0xb5f826d4), fd_no=4 Nov 29 14:40:30 [10498] DBG:core:io_watch_add: io_watch_add(0x8160900, 11, 4, 0xb5f82760), fd_no=5 Nov 29 14:40:30 [10498] ERROR:core:io_watch_add: epoll_ctl failed: Invalid argument [22] Nov 29 14:40:30 [10498] CRITICAL:core:tcp_main_loop: failed to add process 6 (TCP main) unix socket to the fd list Nov 29 14:40:30 [10498] CRITICAL:core:tcp_main_loop: exiting...Nov 29 14:40:30 [10497] INFO:nathelper:rtpp_test: rtp proxy <unix:/var/run/rt pproxy.sock> found, support for it enabled Nov 29 14:40:30 [10497] DBG:core:init_mod_child: type=CHILD, rank=3, module=xlog Nov 29 14:40:30 [10497] DBG:xlog:child_init: init_child [3] pid [10497] Nov 29 14:40:30 [10497] DBG:core:init_mod_child: type=CHILD, rank=3, module=avpops Nov 29 14:40:30 [10497] DBG:core:io_watch_add: io_watch_add(0x8160a60, 11, 1, (nil)), fd_no=0 Nov 29 14:40:30 [10492] INFO:core:handle_sigs: child process 10498 exited normally, status=255 Nov 29 14:40:30 [10492] INFO:core:handle_sigs: terminating due to SIGCHLD Nov 29 14:40:30 [10493] INFO:core:sig_usr: signal 15 received Memory status (pkg): qm_status (0x8181760): heap size= 1048576 used= 44448, used+overhead=71504, free=977072 max used (+overhead)= 71504 dumping all alloc'ed. fragments: 0. N address=0x8186908 frag=0x8186900 size=32 used=1 1. N address=0x8186938 frag=0x8186930 size=16 used=1 2. N address=0x8186958 frag=0x8186950 size=32 used=1 3. N address=0x8186988 frag=0x8186980 size=32 used=1 4. N address=0x81869b8 frag=0x81869b0 size=16 used=1 5. N address=0x81869d8 frag=0x81869d0 size=16 used=1 6. N address=0x81869f8 frag=0x81869f0 size=32 used=1 7. N address=0x8186a28 frag=0x8186a20 size=32 used=1 8. N address=0x8186a58 frag=0x8186a50 size=64 used=1 9. N address=0x8186aa8 frag=0x8186aa0 size=64 used=1 10. N address=0x8186af8 frag=0x8186af0 size=64 used=1

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Re: [OpenSER-Users] auth_db module - get_ha1(): Error while querying database
by Daniel-Constantin Mierla 30 Nov '07

30 Nov '07

Hello, from the logs seems that the content of subscriber table is corrupted. Please attach /usr/local/etc/openser/dbtext/subscriber file and send it to me. Cheers, Daniel On 11/30/07 11:30, Mihail Gluhowchenko wrote: > Hello, > In file log on level 8 when try 9 openser down. > <list of dir(/usr/local/etc/openser/dbtext)> > acc > active_watchers > address > aliases > dbaliases > domaingrp > gw > gw_grp > lcr > location > missed_calls > pdt > presentity > pua > re_grp > silo > speed_dial > subscriber > trusted > uri > usr_preferences > version > watchers > xcap_xml > <end_list_of_dir(/usr/local/etc/openser/dbtext)> > In README file for dbtext error - > break; replace exit; > modparam("auth_db", "calculate_ha1", 1) > replace > modparam("auth_db", "calculate_ha1", yes) > > Regards, > Mihail Gluhowchenko. > > 2007/11/29, Daniel-Constantin Mierla <daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro>>: > > Hello, > > can you list the content of directory: /usr/local/etc/openser/dbtext ? > > Also, if you can set the debug=9 in cofig, run again, get the > output and > send it to me, would help a lot. > > Thanks, > Daniel > > On 11/28/07 13:46, Mihail Gluhowchenko wrote: > > Hello All; > > I`m trying to tune openser-1.2.2 with dbtext data base > > But have some problems. > > I make db use - textdb.sh > > Then add users use - sc.dbtext > > and add to openser.cfg: > > #set module path > > mpath="/usr/local/lib/openser/modules/" > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > >

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[Serusers] fixup_hostport2proxy
by Tuan Nguyen 30 Nov '07

30 Nov '07

Hi everyone, i have a problem when i try with two version of SER: SER 0.9.6 and SER 2.1to connect with SEMS (the 1. SER proxy ). With SER 0.9.6 i don't have problem when start it, but with SER 2.1 it's so: 0(13446) ERROR: mk_proxy: could not resolve hostname: "localhost" 0(13446) ERROR: fixup_hostport2proxy: bad host name in URI <localhost> I guess, problem locate in this code: log("Forwarding to 5070"); rewritehostport("localhost:5070"); t_relay_to_udp("localhost","5070"); Anyone know that where can i fix this problem with new version of SER? Thanks for responce. Tuan.

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[OpenSER-Users] FYI: [Fwd: [Sip] SIPit 21 summary]
by Klaus Darilion 30 Nov '07

30 Nov '07

-------- Original-Nachricht -------- Betreff: [Sip] SIPit 21 summary Datum: Mon, 19 Nov 2007 14:55:53 -0600 Von: Robert Sparks <rjsparks(a)nostrum.com> An: sip List <sip(a)ietf.org> SIPit 21 was hosted by the BII Group and the Beijing University of Posts and Telecommunications, November 5-9 in Beijing, China. The event was a little smaller than the most recent SIPits: There were 94 attendees from 38 companies visiting from 15 countries. There were 44 teams and around 70 distinct implementations. The majority of the testing I saw was focusing on advanced scenarios rather than basic registration and call setup. As with SIPit 20, the most common area of interoperability problems centered around offer/answer, particularly when attempting to negotiate alternate versions of streams or to explicitly state parameters to be used with a given stream. Forty of the attending teams completed the survey - from those answers: The roles represented (some implementations act in more than one role): 21 endpoints 10 proxy/registrars 3 standalone registrars 5 event servers 4 gateways 6 automaton UAs (voicemail, conference, appserver, etc) 8 b2bua/sbcs 4 UAs with signaling, but without media 1 test/monitoring tool Implementations using each transport for SIP messages: UDP 100% TCP 82% TLS 49% (server auth only) TLS 6% (mutual auth) SCTP 3% DTLS 0% 36% of the implementations supported SIP over IPv6 (up from 25% at SIPit20) 18% supported SIP over IPSec For DNS we had support for: Full RFC3263 : 61% SRV only : 21% A records only : 9% no DNS support : 9% Support for various items: 61% rport 15% sigcomp 22% enum 21% sending multiplexing STUN and SIP 28% receiving multiplexed STUN and SIP 22% RFC4320 fixes 12% identity 70% session-timer There was one implementation of parts of the session-policy framework. There was one implementation of the sip-consent framework. There were two implementations of parts of the sip-config framework (I do not know yet if they worked together). There were three implementations of outbound and four of GRUU. There were two implementations of MSRP present. The endpoints implemented these methods: 100% INVITE, CANCEL, ACK, BYE 87% REGISTER 87% OPTIONS 87% NOTIFY <- still a lot of unsolicted NOTIFYs 87% REFER 77% SUBSCRIBE 77% INFO 70% UPDATE 67% PRACK 47% MESSAGE 33% PUBLISH Support for various extensions in the endpoints: 43% RFC3323 privacy 41% RFC3327 path 13% RFC3840 pref 13% RFC3841 caller-prefs 59% RFC3891 replaces 20% RFC4488 norefersub 0% RFC4538 target-dialog <- There were several folks starting to talk about supporting this Support for various things in the endpoints: 10% ICE (but there was no interoperability - see below) 0% ICE-TCP 13% STUNbis 17% TURN (again, there was no interoperability) 75% symmetric RTP 25% SRTP 0% RTP over DTLS This is how the endpoints answered how they supported multipart/mime: 7% I break if someone sends me multipart/mime 30% I pretend multipart mime doesn't exist if someone sends it to me 20% I ignore multipart/mime, but will proxy it or hand it to my application if it shows up 17% I try to do something useful with multipart/mime I receive, but I never send it 3% I ignore multipart/mime I receive, but I try to send it to do something useful 20% I try to do something useful with multipart/mime I send and receive There were 4 endpoints that would send media over TCP - none of them supported RFC4145 comedia. One of those supported media over TLS. Implementation is all over the map for fork-loop-fix. However, only 3 of the proxy/b2buas present were still vulnerable to the simple form of the attack described in the draft. There were no implementations present with support for location- conveyance or the geopriv common-policy framework. There was one implementation present with support for the RFC4967 dial-string parameter, and one with support for the ecrit service-urn. Of the SIP-Events implementations, the following packages were supported: 62% refer 48% message-summary 38% presence 29% dialog 19% reg 19% conf There was one implementation of each of the following packages: ua-profile certificate/credentials vx-rtcpxp (sipping-rtcp-summary) There was only one implementation of event-lists present. Only 20% of the SIP-Events implementations supported winfo. We had a multiparty sesssion for a full morning focusing on NAT traversal. Basic use of STUN with SIP is hightly interoperable. No two implementations of TURN could even start trying to talk to each other (each having implemented to different points in the recent torrent of changes). I don't think we'll get much implementation feedback until the spec stops making big changes so frequently. No two implementations of ICE worked together. The closest was between two implementations that have worked in the past, but failed during this session when the connectivity checks arrived before the SDP. _______________________________________________ Sip mailing list https://www1.ietf.org/mailman/listinfo/sip This list is for NEW development of the core SIP Protocol Use sip-implementors(a)cs.columbia.edu for questions on current sip Use sipping(a)ietf.org for new developments on the application of sip

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[Serusers] Segmentation Fault (SER Not Starting)
by Alexandr Volkov 29 Nov '07

29 Nov '07

Hi All, I am having trouble when trying to start SER. I am using Ser 0.9.6 on centos4. When I am trying to start it gives: Segmentation Fault; but only if in 'location' table there is a row with callId starting by number: <field name="callid">49c72d7ed685500e7d844d1cf1357dbd</field> if I change this field in database (simple add any character), ser will start. <field name="callid">a49c72d7ed685500e7d844d1cf1357dbd</field> seems like a bug

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[Serusers] ERROR: forward_reply: no 2nd via found in reply - via parsing error
by Tomasz Zieleniewski 29 Nov '07

29 Nov '07

Hi, I have a chcek via parameter set in the ser.cfg file When ser receives reply with such via header it returns ERROR: forward_reply: no 2nd via found in reply This Via seems to be ok? Via: SIP/2.0/UDP 192.168.0.165:7160;branch=z9hG4bK1589.3bc310b3.1,SIP/2.0/UDP 192.168.0.165:7060;branch=z9hG4bK1589.e1306596.0,SIP/2.0/UDP 192.168.129.74 ;rport=5060;branch=z9hG4bK1589.606a3e61.0,SIP/2.0/UDP 192.168.129.74:5160 ;branch=z9hG4bK1589.a5bd9fb4.0,SIP/2.0/UDP 192.168.129.74;branch= z9hG4bK1589.506a3e61.0,SIP/2.0/UDP 192.168.0.102:7060 ;rport=7060;branch=z9hG4bK-d87543-a228576fa32c0b22-1--d87543- What is wrong? Cheers tomasz

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[Serusers] Unexpected disconnect - lost SIP packets?
by Jeremy A 29 Nov '07

29 Nov '07

Hello, I'm need some help with SIP protocol / SER configuration to resolve a problem. I have a system with a SER instance and a SIP client application running on the same box. SER is 10.50.32.10:5060 APP is 10.50.32.10:5070 Remote SIP server 10.1.39.10 Remote phone is 10.1.39.202 The user agent is registered with the SER instance as extension 1009. It has aliases 1001-1008 and is actually called as 231x which maps onto 100x which then connects to extension 1009. The remote phone makes a call to the user agent application via the remote SIP server and the SER instance. It dials 2311 which gets automagically converted to extension 1009 The call is established but the user application times out and detects a disconnect despite the calling phone not hanging up. In my wireshark trace it seems that SER is ignoring some form of keep-alive from the remote SIP server and this is causing the timeout. The lines of interest include 39, 42, 45, 50, 53, 56, 59 Lines 28 and others similar are on the loopback interface and are packets between port 5060 and 5070 Can someone please advise if the SIP protocol has problems and/or if SER configuration has problems? 22 0.218826 10.50.32.10 -> 127.0.0.1 SIP Request: REGISTER sip:localhost.localdomain 23 0.219570 127.0.0.1 -> 10.50.32.10 SIP Status: 200 OK (1 bindings) 24 19.472555 10.1.39.10 -> 10.50.32.10 SIP/SDP Request: INVITE sip:1001@butler.fesa.sto;transport=udp, with session description 25 19.473394 10.50.32.10 -> 10.1.39.10 SIP Status: 100 trying -- your call is important to us 26 19.473596 10.50.32.10 -> 10.50.32.10 SIP/SDP Request: INVITE sip:1009@10.50.32.10:5070;LINEID=1d5a67e38809, with session description 27 19.484244 10.50.32.10 -> 10.50.32.10 SIP Status: 100 Trying 28 19.725476 10.50.32.10 -> 10.50.32.10 SIP Status: 180 Ringing 29 19.725828 10.50.32.10 -> 10.1.39.10 SIP Status: 180 Ringing 32 21.770798 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 33 21.782437 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 38 21.789396 10.50.32.10 -> 10.1.39.202 RTCP Source port: 8001 Destination port: 16391 39 21.800069 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 40 22.278096 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 41 22.278713 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 42 22.298101 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 43 23.285536 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 44 23.288988 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 45 23.307839 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 48 25.298214 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 49 25.299105 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 50 25.317892 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 51 29.306496 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 52 29.306758 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 53 29.328420 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 54 33.314736 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 55 33.318736 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 56 33.338302 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 57 37.323175 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 58 37.324791 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 59 37.347936 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 --> client detects disconnect here Thanks jeremy

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[Serusers] Presence in SIP
by Carlos Angeles 29 Nov '07

29 Nov '07

Hello I have installed Ser-2.0 with the PA module.... I am testing the server with x-lite... When a user is NOTIFYING some event change the server is answering with a Status: 420 Bad Extension.... Is thst SER doesn´t support this kind of presence events? Thanks Carlos _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE

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[OpenSER-Users] Handling re-INVITES --hellp required
by srinivas Antarvedi 29 Nov '07

29 Nov '07

Hello all, i have users one is on global ip and another behind NAT am using asterisk as media server leg 1: caller : Global ip UAC. callee: asterisk leg2: caller :asterisk callee: NATed UAC. sdp of NATed client is handled at openser reply route at first stage when asterisk re-invites the NATed UAC to bridge the two call-Leg's the sdp from NATed UAC is not changed ,, even if i call t_on_reply in the loose route section of the script.. it is still showing privat ip so finally after 2 or 3 sec's there was an end to the dialog can anybody have any idea to handle re-invite's 200 ok SDP mangling? please help me out.. Thanks in advance regards srinivas -- Srinivas Antarvedi

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[OpenSER-Users] auth_db module - get_ha1(): Error while querying database
by Mihail Gluhowchenko 29 Nov '07

29 Nov '07

Hello All; I`m trying to tune openser-1.2.2 with dbtext data base But have some problems. I make db use - textdb.sh Then add users use - sc.dbtext and add to openser.cfg: #set module path mpath="/usr/local/lib/openser/modules/" # use dbtext database loadmodule "dbtext.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # Uncomment this if you want digest authentication loadmodule "auth.so" loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- #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 -- # use dbtext database for persistent storage modparam("usrloc", "db_mode", 2) modparam("usrloc|auth_db", "db_url", "dbtext:///usr/local/etc/openser/dbtext") # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "password") modparam("auth_db", "domain_column", "domain") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- 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"); log(1, "Too Many Hops\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); log(1, "Message too big\n"); 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); }; # 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") { log(1, "this is a request REGISTER users\n"); if (!www_authorize("my.domain", "subscriber")) { log(1, "this is a www_challenge\n"); www_challenge("my.domain", "0"); exit; }; save("location"); exit; }; but when trying REGISTER user use(sjphone) in log i see: Nov 28 17:30:03 devel s-eltex[12797]: this is a request for my.domain users Nov 28 17:30:03 devel s-eltex[12797]: this is a request REGISTER users Nov 28 17:30:03 devel s-eltex[12797]: this is a www_challenge Nov 28 17:30:03 devel s-eltex[12794]: this is a request for my.domain users Nov 28 17:30:03 devel s-eltex[12794]: this is a request REGISTER users Nov 28 17:30:03 devel s-eltex[12794]: get_ha1(): Error while querying database In sjphone LOG i see: Service unavailable server error; What do I adjust not correctly?

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[OpenSER-Users] t_relay can not handle PRACK retransmission
by Klaus Darilion 29 Nov '07

29 Nov '07

Hi! I have encountered this problem at an openser 1.1.1 proxy: Client Proxy Client ---PRACK------> ------PRACK-----> retansmission -----PRACK-------> here at this point (when openser receives the PRACK retransmission) t_relay fails: ERROR: t_newtran: transaction already in process 0xb66e7880 Is this a known problem? Were there some fixes to 1.1 or 1.2 branch? regards klaus

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[OpenSER-Users] Branch limitations
by Daniel 29 Nov '07

29 Nov '07

Hi all, I've read this from iptel website. It is true for SER 0.11.0. What about Openser 1.2 ? " * Script processing of multiple branches on forking Warning ser's request processing language allows to make request decisions based on current URI. When a request if forked to multiple destinations, only the first branch's URI is used as input for script processing. This might lead to unexpected results. Whenever a URI resolves to multiple different next-hop URIs, only the first is processed which may result in handling not appropriate for the other branch. For example, a URI might resolve to an IP phone SIP address and PSTN gateway SIP address. If the IP phone address is the first, then script execution ignores the second branch. If a script includes checking gateway address in request URI, the checks never match. That might result in ignoring of gateway admission control rules or applying them unnecessarily to non-gateway destinations. List of known problems is publicly available at the ser webpage at [70]http://www.iptel.org/ser/ . See the "ISSUES" link. -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] OpenSER: Support of advanced RegExpr?
by Helmut Kuper 28 Nov '07

28 Nov '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, is openser supporting advanced regular expressions as described here: http://www.regular-expressions.info/refadv.html regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHTVg64tZeNddg3dwRAl+JAKCst6W4QdNjJ6ZT6fWRaE9otPcX+wCdEKbw iQy1J1Md3CKE7pUgx4lld3k= =sRw7 -----END PGP SIGNATURE-----

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[OpenSER-Users] OpenXCAP: 'presence-rules' PUT sent: '409 Conflict' received
by ddonate 28 Nov '07

28 Nov '07

Hi, I am using OpenSER and openXCAP to handle 'resource list' and 'presence rules' of some buddies. The 'resource-list' works fine: I can GET my file (404 the 1st time, and I PUT one). But I have a problem with the 'presence-rules' PUT, when I receive a 'winfo' NOTIFY (when the user 'john' subscribes to my presence): I send an HTTP request but I receive a '409 Conflict': - My request: PUT /pres-rules/users/diego(a)openser/presence_rules.xml HTTP/1.1 User-Agent: IMS_DialcomClient Accept: text/* Host: 192.168.1.208:8000 Content-Length: 284 ... <?xml version='1.0' encoding='UTF-8'?><ruleset xmlns='urn:ietf:params:xml:ns:presence-rules'><rule id='sip:john@openser_rule'><conditions><identity><one id='sip:john@openser'/></identity></conditions><actions><sub-handling>allow</sub-handling></actions></rule></ruleset> - The traces in the openXCAP are: -------------------- Failed to validate document against XML schema: <string>:1:ERROR:SCHEMASV:SCHEMAV_CVC_ELT_1: Element '{urn:ietf:params:xml:ns:pres-rules}ruleset': No matching global declaration available for the validation root. -------------------- I know it is related to the schema, but I have not managed to fix it... Thanks in advance Diego -- View this message in context: http://www.nabble.com/OpenXCAP%3A-%27presence-rules%27-PUT-sent%3A-%27409-C… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[OpenSER-Users] routing table number
by Patrick Miccio 28 Nov '07

28 Nov '07

Hi everyone, I came across another problem when using a newly compiled OpenSER binary (from 1.1.1-notls source with no modifications). When starting OpenSER with -c -f configfile I would get the following errors: ----- 0(18837) parse error (354,1-2): invalid routing table number ----- Looking around with google I found out that this is because in the original config.h the following is set: ------ define RT_NO 40 /* routing tables number */ ------ In my openser.cfg there is already 44 routing tables ;) So my question is: can you somehow set RT_NO in openser.cfg, or is that only possible in config.h + recompiling? digital greetings, Patrick. This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

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[Serusers] ser presence question
by Yufei Tao 28 Nov '07

28 Nov '07

Hi I want to be able to use PUBLISH/SUBSCRIBE/NOTIFY for presence information of users. So users can update their presence information by PUBLISH and watchers be informed by NOTIFY when there are changes in the presence status. I am using pa.so, mysql. Not using XCAP. My problems are 1. The OK from SER to PUBLISH has no SIP_ETag. Also SER prints out debug messages saying: publish_presentity: no tuple for 01(a)sertest.com I wonder if there's something wrong with my PUBLISH and PIDF so SER is not processing it? 2. The watcher subscribes to the presentity 01(a)sertest.com. But the NOTIFY message the watcher gets always has basic status being closed, despite that the presentity has PUBLISHed its status as open. I guess this may be due to problem 1 - SER never understood the PUBLISH? Currently my UA sends PUBLISH message: PUBLISH sip:01@sertest.com:5060 SIP/2.0 via: SIP/2.0/UDP 192.168.168.149:40070;branch=z9hG4bK314278337 max-forwards: 70 to: <sip:01@sertest.com:5060> from: <sip:01@sertest.com:5060>;tag=314279337 call-id: cd674de3-0-1196253409 cseq: 46046 PUBLISH contact: <sip:01@192.168.168.149:40070> expires: 3600 user-agent: Red v1.00 0.6.3.42 - von event: presence content-type: application/pidf+xml content-length: 274 <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:01@sertest.com"> <tuple id="status"> <status> <basic>open</basic> </status> <contact priority="0.500000">sip:01@sertest.com</contact> </tuple> </presence> and SER replies with the following msg where there is no SIP-ETag field: SIP/2.0 200 OK via: SIP/2.0/UDP 192.168.168.149:40070;branch=z9hG4bK314278338 to: <sip:01@sertest.com:5060>;tag=482760500770a188d9ef565befb876b4-7979 from: <sip:01@sertest.com:5060>;tag=314279338 call-id: cd674de3-0-1196253409 cseq: 46047 PUBLISH Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 192.168.168.241:5060 "Noisy feedback tells: pid=14095 req_src_ip=192.168.168.149 req_src_port=40070 in_uri=sip:01@sertest.com:5060 out_uri=sip:01@sertest.com:5060 via_cnt==1" The parts in my ser.cfg that are relevant to SUBSCRIBE and PUBLISH: modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 10) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) # for SUBSCRIBE/NOTIFY modparam("pa", "default_expires", 3600) modparam("pa", "use_db", 1) modparam("pa", "db_url", "mysql://ser:heslo@localhost/ser") modparam("pa", "pa_domain", "sertest.com")# -- same as or different from registrar ... ... ... ... if (method=="SUBSCRIBE") { if (!t_newtran()) { sl_reply_error(); break; }; handle_subscription("registrar"); break; } if (method=="PUBLISH") { if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); }; handle_publish("registrar"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; };

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[OpenSER-Users] OpenSer: Group Module Parameter "re_exp_column"
by Helmut Kuper 28 Nov '07

28 Nov '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, the example in OpenSER 1.3 and devel module doc for module group is wrong. It gives this line as example: modparam("group", "reg_exp_column", "re") but it should be: modparam("group", "reg_exp_column", "reg_exp") minor thing, but anoying, when u just copy and paste modparams into your config without reading the chapter titles .... Maybe there should be a block with all modparams and their default values in each module's doc... regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHTDhm4tZeNddg3dwRAh/5AKC0r9jbisrrR+YZC0BKtk6hzGz8oQCdEtcv 2oHfwzkoe43hGB4YO4dv6mM= =ysim -----END PGP SIGNATURE-----

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[OpenSER-Users] multiple contact
by Daniel 28 Nov '07

28 Nov '07

Hi all, I have a user A registering with 2 differrent contact, one of each behind NAT. If I set "modparam("registrar", "append_branches", 1)", and if q-values are setted as 1, the proxy, aftel lookup (location) will call both of contacts in parallel mode. But how can I manage the Natted contact and set mediaproxy ? regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] editing the via header
by Othmar Pasteka 28 Nov '07

28 Nov '07

hello, I would like to know when the via header of the openser proxy is inserted (the new one). and/or if there is a practical way to manipulate it by me in a different module, like that some additional information is added after the branch tag. I browsed through the documentation and also googled but couldn't find valuable information how the via headers are treated in this respect. I found the reference in parser/msg_parser.h:struct sip_msg, that they reference via1/via2 but I assume via1 is not the current one which will be added by the local openser proxy. any additional information is greatly appreciated. thanks. regards, Othmar

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[OpenSER-Users] [CPL] Why aren't loaded bflags when <lookup source="registration"> ?
by Iñaki Baz Castillo 28 Nov '07

28 Nov '07

Hi, I set: modparam("cpl-c","proxy_route", 10) and in route[10] I set RtpProxy if flag(5) (NAT in the caller) or bflag(6) (NAT in the called) is present. But unfortunatelly the bflag(6) is never set when usingCPL. For example the following and very simple CPL: <cpl> <incoming> <lookup source="registration"> <success> <proxy /> </success> </lookup> </incoming> </cpl> I understand that the operation of CPL is "encapsulated" and so, but since we can spec some parameters as: modparam("cpl-c","lookup_domain","location") modparam("cpl-c","lookup_append_branches", 1) why then can't bflags be loaded? If it is not possible then I can't use <lookup source="registration"> and need to do, neccesarialy, a spiral with: <location url="sip:SAME_USER;cpl=no"> <proxy /> </location> and don't apply CPL in the spiral incoming message if URI has "cpl=no". Any explanation of this limitation could be really great. Thanks a lot. -- Iñaki Baz Castillo

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Re: [Serusers] nathelper doesn't ping registered users
by Jiri Kuthan 27 Nov '07

27 Nov '07

I'm kind of finding this conversation a bit confusing -- what is cl in this context actually good for? I mean the "boundary proxy" (what is it actually good for?) can have shadow usrloc (it is presumably in the path anyhow?) if natping shall be sent from there. sip-based natping can be run from the "inside proxy" as well, very much as any other SIP request. That causes heavier traffic but also increases the success chance (for example behind my firewall and with my SIP phone, non-SIP pings does not help because client doesn't send enough outbound traffic) Just wild guesses -- I'm not really understanding the specifics of this proxy chain. -jiri At 15:13 27/11/2007, Tomasz Zieleniewski wrote: >unfortunatelly it is not available >any chance to have it for ser 2.0?? > >regards >tomasz > >On Nov 27, 2007 2:51 PM, Greger V. Teigre <greger(a)teigre.com> wrote: >> >> usrloc-cl is a cacheless version of usrloc module, i.e. nothing is stored >> in memory. I believe Martin Hoffmann took over maintenance of it, but I'm >> not sure about the status on 2.0.0. If your edge proxy uses usrloc-cl >> instead of usrloc, it will pull locations to ping from the DB instead of >> from memory. >> >> >> g-) >> >> Tomasz Zieleniewski wrote: >> what is usrloc-cl?? >> What do you mean by that and how can I do that? >> >> -tomasz >> >> On Nov 27, 2007 8:58 AM, Greger V. Teigre <greger(a)teigre.com> wrote: >> >> >> You use usrloc-cl? >> If not, unless you replicate, the non-REGISTER handling server will not >> have the usrloc in memory. >> g-) >> >> >> >> Tomasz Zieleniewski wrote: >> Hi Jiri, >> >> On Nov 25, 2007 10:08 PM, Jiri Kuthan <jiri(a)iptel.org> wrote: >> >> >> At 10:56 25/11/2007, Tomasz Zieleniewski wrote: >> >> >> Hi, >> >> I don't have those lines because I have a registrar on seperate SIP Proxy. >> This proxy here is the boundary one, but they both use the same database >> and have access to location table common. >> >> Well, I'm not sure I'm getting the context of this configuration so I don't >> feel certain with providing some hints (in particular two proxy serveer >> in a series both using usrloc seems unusual to me). A hand-waving estimate >> is you need to configure the "boundary proxy" to change all natted clients >> to form "as seen by the network" and the "actual proxy" to send pings to >> all contacts. >> >> >> Boundary proxy uses usrloc only for the purpose of pinging the natted >> users. >> Nathelper module requires usrloc. Because all reuqest travels 'boundary >> proxy' >> before entering network the 'boundary proxy' must ping users to keep >> my nat bindings >> right. they are both on seperate machines. >> >> -tomek >> >> >> -jiri >> >> >> >> >> >> -tomasz >> >> On Nov 25, 2007 10:40 AM, Jiri Kuthan <jiri(a)iptel.org> wrote: >> >> >> Hi Tomasz, >> >> well it could be that reusing an existing config and commiting than your >> specific >> changes to it would save you quite some laborous exercises rather than >> writing >> it from scratch.... >> http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/etc/ser-oob.cfg?re… >> >> The lines in question are (out of which I think you are missing the >> registrar params) >> >> >> FLAG_NAT : 3, # the UAC is behind a NAT >> >> ... >> # identify natted contacts using a flag >> modparam("registrar", "load_nat_flag", "FLAG_NAT") >> modparam("registrar", "save_nat_flag", "FLAG_NAT") >> ... >> modparam("rr", "cookie_filter", "(account|uac_nat)") >> ... >> modparam("nathelper", "ping_nated_only", 1 ) >> # if this option is not set, simple 4-bytes ping is sent >> modparam("nathelper", "natping_method", "OPTIONS" ) >> #temporary statefull natping test (only in future versions) >> #modparam("nathelper", "natping_stateful", 1) >> ... >> >> setflag(FLAG_NAT); >> >> >> -jiri >> >> >> At 22:20 24/11/2007, Tomasz Zieleniewski wrote: >> >> >> I'm sorry for the CC >> here is my config: >> >> debug=3 # debug level (cmd line: -dddddddddd) >> memdbg=5 >> memlog=5 >> log_facility=LOG_LOCAL0 >> fork=yes >> log_stderror=no >> check_via=yes # (cmd. line: -v) >> dns=no # (cmd. line: -r) >> rev_dns=no # (cmd. line: -R) >> port=5060 >> tls_port_no=5061 >> children=4 >> user=ser >> group=ser >> mhomed=yes >> disable_tcp=no >> tcp_accept_aliases=yes >> enable_tls=no >> sip_warning=0 >> reply_to_via=no >> alias="voip.touk.pl" >> >> dns_retr_time=1 >> dns_retr_no=1 >> dns_servers_no=1 >> dns_use_search_list=no >> dns_try_ipv6=no >> # dns cache & failover >> use_dns_cache=on >> use_dns_failover=on >> # dns_cache_flags=0 >> dns_cache_negative_ttl=300 >> dns_cache_min_ttl=60 >> dns_cache_max_ttl=86400 # 1 day >> dns_cache_mem=2048 # 2 MB >> dns_cache_gc_interval=60 >> >> tcp_connection_lifetime=3600 >> tcp_max_connections=2048 >> tcp_connect_timeout=1 >> >> loadmodule ... >> .... >> >> # specify the path to you database here >> modparam("acc_db|auth_db|avp_db|domain|gflags|usrloc|uri_db|speeddial|msilo|lcr", >> "db_url", "mysql://ser:ser@localhost/ser") >> >> # -- usrloc params -- >> modparam("usrloc", "db_mode", 1) >> >> # -- registrar params -- >> modparam("registrar", "default_expires", 300) >> modparam("registrar", "min_expires", 60) >> modparam("registrar", "max_expires", 300) >> modparam("registrar", "append_branches", 1) >> modparam("registrar", "received_param", "rcv") >> modparam("registrar", "max_contacts", 2) >> >> # -- auth params -- >> modparam("auth_db", "calculate_ha1", yes) >> modparam("auth_db", "password_column", "password") >> >> # -- rr params -- >> modparam("rr", "enable_full_lr", 1) >> modparam("rr", "cookie_filter", "uac_nat") >> modparam("rr", "cookie_secret", "secret") >> modparam("rr", "user_part_avp", "route_user") >> modparam("rr", "next_route_avp", "route_next") >> >> # -- gflags params -- >> modparam("gflags", "load_global_attrs", 1) >> >> # -- domain params -- >> modparam("domain", "load_domain_attrs", 1) >> >> # -- ctl params -- >> modparam("ctl", "binrpc", "unixs:/tmp/ser-bsp_ctl") >> modparam("ctl", "fifo", "fifo:/tmp/ser-bsp_fifo") >> # modparam("ctl", "binrpc", "tcp:localhost:2046") >> modparam("ctl", "mode", 0777) >> >> # -- tm params -- >> modparam("tm", "restart_fr_on_each_reply", 1) >> >> # -- auth_radius params -- >> modparam("auth_radius", "radius_config", >> "/usr/local/etc/radiusclient-ng/radiusclient.conf") >> modparam("auth_radius", "service_type", 15) >> >> # -- options params -- >> modparam("options", "accept", "*/*") >> modparam("options", "accept_encoding", "") >> modparam("options", "accept_language", "en") >> >> # -- ratelimit params -- >> modparam("ratelimit","timer_interval", 10) >> modparam("ratelimit","invite_limit", 100) >> modparam("ratelimit","register_limit", 100) >> modparam("ratelimit","subscribe_limit", 100) >> >> # -- xmlrpc params -- >> modparam("xmlrpc", "route","RPC") >> >> # -- nathelper params -- >> modparam("nathelper", "natping_interval", 20) >> modparam("nathelper", "ping_nated_only", 0) >> modparam("nathelper", "rtpproxy_disable", 1) >> modparam("nathelper", "tcpping_crlf", 1) >> >> # -- tls params -- >> modparam("tls", "config", "/home/ser/ser/ca/tls-bsp.cfg") >> modparam("tls", "tls_force_run", 0) >> modparam("tls", "tls_log",3) >> modparam("tls", "handshake_timeout", 10) >> modparam("tls", "send_timeout", 10) >> modparam("tls", "tls_disable_compression", 0) >> #modparam("tls", "private_key", "/etc/certs/key.pem") >> #modparam("tls", "certificate", "/etc/certs/cert.pem") >> #modparam("tls", "ca_list", "/etc/certs/ca_list.pem") >> #modparam("tls", "require_certificate", 0) >> #modparam("tls", "verify_certificate", 1) >> #modparam("tls", "tls_method", "TLSv1") >> >> route{ >> ... >> } >> >> On Nov 24, 2007 9:35 PM, Atle Samuelsen <clona(a)cyberhouse.no> wrote: >> >> >> Hi, >> >> Always CC the list. >> >> then I dont know.. post your full config so people can see if they see >> something >> -A >> >> * Tomasz Zieleniewski <tzieleniewski(a)gmail.com> [071124 20:38]: >> >> >> >> Hi, >> >> I had that parameter set to 0, but still no udp packets... >> >> On Nov 24, 2007 6:43 PM, Atle Samuelsen <clona(a)cyberhouse.no> wrote: >> >> >> * Tomasz Zieleniewski <tzieleniewski(a)gmail.com> [071124 16:50]: >> >> >> >> HI, >> >> I have the following nathelper settings: >> modparam("nathelper", "natping_interval", 20) >> modparam("nathelper", "ping_nated_only", 1) >> modparam("nathelper", "rtpproxy_disable", 1) >> modparam("nathelper", "tcpping_crlf", 1) >> >> But i cant see any udp packets being sent to registered users. >> Please point me what did I miss. >> >> Hi, >> >> without seeing any more of your config, I'm guessing that you do not set >> the natflag, so if you change ping_nated_only to 0, I think you will see >> some udp packets. >> >> - Atle >> (shot in the whild world of sip) >> >> >> >> >> Best >> Tomasz >> _______________________________________________ >> 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 >> >> >> -- >> Jiri Kuthan http://iptel.org/~jiri/ >> >> >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> -- >> Jiri Kuthan http://iptel.org/~jiri/ >> >> >> >> _______________________________________________ >> 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 >> >> >> >> -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] nathelper doesn't ping registered users
by Jiri Kuthan 27 Nov '07

27 Nov '07

At 10:56 25/11/2007, Tomasz Zieleniewski wrote: >Hi, > >I don't have those lines because I have a registrar on seperate SIP Proxy. >This proxy here is the boundary one, but they both use the same database >and have access to location table common. Well, I'm not sure I'm getting the context of this configuration so I don't feel certain with providing some hints (in particular two proxy serveer in a series both using usrloc seems unusual to me). A hand-waving estimate is you need to configure the "boundary proxy" to change all natted clients to form "as seen by the network" and the "actual proxy" to send pings to all contacts. -jiri >-tomasz > >On Nov 25, 2007 10:40 AM, Jiri Kuthan <jiri(a)iptel.org> wrote: >> Hi Tomasz, >> >> well it could be that reusing an existing config and commiting than your specific >> changes to it would save you quite some laborous exercises rather than writing >> it from scratch.... >> http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/etc/ser-oob.cfg?re… >> >> The lines in question are (out of which I think you are missing the registrar params) >> >> >> FLAG_NAT : 3, # the UAC is behind a NAT >> >> ... >> # identify natted contacts using a flag >> modparam("registrar", "load_nat_flag", "FLAG_NAT") >> modparam("registrar", "save_nat_flag", "FLAG_NAT") >> ... >> modparam("rr", "cookie_filter", "(account|uac_nat)") >> ... >> modparam("nathelper", "ping_nated_only", 1 ) >> # if this option is not set, simple 4-bytes ping is sent >> modparam("nathelper", "natping_method", "OPTIONS" ) >> #temporary statefull natping test (only in future versions) >> #modparam("nathelper", "natping_stateful", 1) >> ... >> >> setflag(FLAG_NAT); >> >> >> -jiri >> >> >> At 22:20 24/11/2007, Tomasz Zieleniewski wrote: >> >I'm sorry for the CC >> >here is my config: >> > >> >debug=3 # debug level (cmd line: -dddddddddd) >> >memdbg=5 >> >memlog=5 >> >log_facility=LOG_LOCAL0 >> >fork=yes >> >log_stderror=no >> >check_via=yes # (cmd. line: -v) >> >dns=no # (cmd. line: -r) >> >rev_dns=no # (cmd. line: -R) >> >port=5060 >> >tls_port_no=5061 >> >children=4 >> >user=ser >> >group=ser >> >mhomed=yes >> >disable_tcp=no >> >tcp_accept_aliases=yes >> >enable_tls=no >> >sip_warning=0 >> >reply_to_via=no >> >alias="voip.touk.pl" >> > >> >dns_retr_time=1 >> >dns_retr_no=1 >> >dns_servers_no=1 >> >dns_use_search_list=no >> >dns_try_ipv6=no >> ># dns cache & failover >> >use_dns_cache=on >> >use_dns_failover=on >> ># dns_cache_flags=0 >> >dns_cache_negative_ttl=300 >> >dns_cache_min_ttl=60 >> >dns_cache_max_ttl=86400 # 1 day >> >dns_cache_mem=2048 # 2 MB >> >dns_cache_gc_interval=60 >> > >> >tcp_connection_lifetime=3600 >> >tcp_max_connections=2048 >> >tcp_connect_timeout=1 >> > >> >loadmodule ... >> >.... >> > >> ># specify the path to you database here >> >modparam("acc_db|auth_db|avp_db|domain|gflags|usrloc|uri_db|speeddial|msilo|lcr", >> >"db_url", "mysql://ser:ser@localhost/ser") >> > >> ># -- usrloc params -- >> >modparam("usrloc", "db_mode", 1) >> > >> ># -- registrar params -- >> >modparam("registrar", "default_expires", 300) >> >modparam("registrar", "min_expires", 60) >> >modparam("registrar", "max_expires", 300) >> >modparam("registrar", "append_branches", 1) >> >modparam("registrar", "received_param", "rcv") >> >modparam("registrar", "max_contacts", 2) >> > >> ># -- auth params -- >> >modparam("auth_db", "calculate_ha1", yes) >> >modparam("auth_db", "password_column", "password") >> > >> ># -- rr params -- >> >modparam("rr", "enable_full_lr", 1) >> >modparam("rr", "cookie_filter", "uac_nat") >> >modparam("rr", "cookie_secret", "secret") >> >modparam("rr", "user_part_avp", "route_user") >> >modparam("rr", "next_route_avp", "route_next") >> > >> ># -- gflags params -- >> >modparam("gflags", "load_global_attrs", 1) >> > >> ># -- domain params -- >> >modparam("domain", "load_domain_attrs", 1) >> > >> ># -- ctl params -- >> >modparam("ctl", "binrpc", "unixs:/tmp/ser-bsp_ctl") >> >modparam("ctl", "fifo", "fifo:/tmp/ser-bsp_fifo") >> ># modparam("ctl", "binrpc", "tcp:localhost:2046") >> >modparam("ctl", "mode", 0777) >> > >> ># -- tm params -- >> >modparam("tm", "restart_fr_on_each_reply", 1) >> > >> ># -- auth_radius params -- >> >modparam("auth_radius", "radius_config", >> >"/usr/local/etc/radiusclient-ng/radiusclient.conf") >> >modparam("auth_radius", "service_type", 15) >> > >> ># -- options params -- >> >modparam("options", "accept", "*/*") >> >modparam("options", "accept_encoding", "") >> >modparam("options", "accept_language", "en") >> > >> ># -- ratelimit params -- >> >modparam("ratelimit","timer_interval", 10) >> >modparam("ratelimit","invite_limit", 100) >> >modparam("ratelimit","register_limit", 100) >> >modparam("ratelimit","subscribe_limit", 100) >> > >> ># -- xmlrpc params -- >> >modparam("xmlrpc", "route","RPC") >> > >> ># -- nathelper params -- >> >modparam("nathelper", "natping_interval", 20) >> >modparam("nathelper", "ping_nated_only", 0) >> >modparam("nathelper", "rtpproxy_disable", 1) >> >modparam("nathelper", "tcpping_crlf", 1) >> > >> ># -- tls params -- >> >modparam("tls", "config", "/home/ser/ser/ca/tls-bsp.cfg") >> >modparam("tls", "tls_force_run", 0) >> >modparam("tls", "tls_log",3) >> >modparam("tls", "handshake_timeout", 10) >> >modparam("tls", "send_timeout", 10) >> >modparam("tls", "tls_disable_compression", 0) >> >#modparam("tls", "private_key", "/etc/certs/key.pem") >> >#modparam("tls", "certificate", "/etc/certs/cert.pem") >> >#modparam("tls", "ca_list", "/etc/certs/ca_list.pem") >> >#modparam("tls", "require_certificate", 0) >> >#modparam("tls", "verify_certificate", 1) >> >#modparam("tls", "tls_method", "TLSv1") >> > >> >route{ >> >... >> >} >> > >> >On Nov 24, 2007 9:35 PM, Atle Samuelsen <clona(a)cyberhouse.no> wrote: >> >> Hi, >> >> >> >> Always CC the list. >> >> >> >> then I dont know.. post your full config so people can see if they see >> >> something >> >> -A >> >> >> >> * Tomasz Zieleniewski <tzieleniewski(a)gmail.com> [071124 20:38]: >> >> >> >> > Hi, >> >> > >> >> > I had that parameter set to 0, but still no udp packets... >> >> > >> >> > On Nov 24, 2007 6:43 PM, Atle Samuelsen <clona(a)cyberhouse.no> wrote: >> >> > > * Tomasz Zieleniewski <tzieleniewski(a)gmail.com> [071124 16:50]: >> >> > > >> >> > > > HI, >> >> > > > >> >> > > > I have the following nathelper settings: >> >> > > > modparam("nathelper", "natping_interval", 20) >> >> > > > modparam("nathelper", "ping_nated_only", 1) >> >> > > > modparam("nathelper", "rtpproxy_disable", 1) >> >> > > > modparam("nathelper", "tcpping_crlf", 1) >> >> > > > >> >> > > > But i cant see any udp packets being sent to registered users. >> >> > > > Please point me what did I miss. >> >> > > >> >> > > Hi, >> >> > > >> >> > > without seeing any more of your config, I'm guessing that you do not set >> >> > > the natflag, so if you change ping_nated_only to 0, I think you will see >> >> > > some udp packets. >> >> > > >> >> > > - Atle >> >> > > (shot in the whild world of sip) >> >> > > >> >> > > >> >> > > > >> >> > > > Best >> >> > > > Tomasz >> >> > > > _______________________________________________ >> >> > > > 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 >> >> >> >> -- >> Jiri Kuthan http://iptel.org/~jiri/ >> >> >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[OpenSER-Users] OpenSER Core CookBook link on openser.org
by Ovidiu Sas 27 Nov '07

27 Nov '07

It would be really handy to have a link to the CoreBook page from the main doc page: http://www.openser.org/docs/ This page starts with a shortcut for the wiki (the first link), but after that it would be nice to have links to the CoreBook/version page just like we have for modules, PV and transformations. Regards, Ovidiu Sas

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[OpenSER-Users] OpensSer Release Name
by Walter Rodrigues Filho 27 Nov '07

27 Nov '07

Dear Bogdan - Since OpenSer has deep roots in Rumania, and considering the suggestions like: * blackcat (because of the 13 thing...) * Santa "Curz"(instead of Cruz it looks like a CURSE...) * deadly SINS (like luxuria, gula...) They are all drive you into either BAD or superstitious concepts.... heheh IN THIS LINE, I would suggest then people names related somehow to Rumanian history (eventough they may have not been Rumanians but after all OpenSer is OPEN to other nationalities to improve and contribute on it...) * DRACULA * IVAN * ATTILA * CORVINUS All of them have had something to do with Rumania in certain time like all members of this opensource community. IN CONCLUSION: All of them is good but "blackcat" is much to much similar to american missiles (tomcat and things..). We really should have here a LATIN BASED NAME or Rumanian history related name (eventhough the names could be of temporary CONQUERORS of Rumania...hehe). END OF LINE> Walter Rodrigues Filho Brazil

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[OpenSER-Users] wesip startup error
by Daniel 27 Nov '07

27 Nov '07

Hi all, I'm using wesip with openser 1.2.2. I've configured both of the configuration files (openser.cfg with SEAS module and server.xml) but when I run the wesip startup script I get this Exception: Starting WESIP Application Server VozTelecom... [root@SERVER conf]# Exception during startup processing java.lang.reflect.InvocationTargetException at java.lang.reflect.Method.invoke (libgcj.so.7) at org.apache.catalina.startup.Bootstrap.main (Bootstrap.java:156) Caused by: java.lang.NoSuchMethodError: method java.lang.Class.getSimpleNamewith signature ()Ljava.lang.String; was not found. at org.apache.catalina.connector.http.HttpConnector.start ( HttpConnector.java:1194) at org.apache.catalina.core.StandardService.start (StandardService.java :452) at org.apache.catalina.core.StandardServer.start (StandardServer.java :2142) at org.apache.catalina.startup.Catalina.start (Catalina.java:474) at org.apache.catalina.startup.Catalina.execute (Catalina.java:358) at org.apache.catalina.startup.Catalina.process (Catalina.java:129) at java.lang.reflect.Method.invoke (libgcj.so.7) ...1 more Can somebody help me? Regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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Re: [OpenSER-Users] OpenSER 1.3 - release name
by lu 27 Nov '07

27 Nov '07

Hi Guys, I like the whole idea of naming and my self i propose this great OpenSource Tool to be Code named "UHURU". It is a Swahili word which means Freedom, it can be freedom of anything, be it colonialization, of expression, way of doing things etc.etc. ....am not John the Baptist though..teh! teh!..teh!.. WBR, LU.

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[Serusers] 483 Too many Hops Error with SER Presence Agent
by Carlos Angeles 27 Nov '07

27 Nov '07

Hi... I have installed SER-0.10.99 with presence capabilities following the presence hadnbook, the server is running OK, but I have problems when I tried to register a client (x-lite). I am getting 483: Too many hops... What can be the problem????? Thank you for your help, Carlos Config file: debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) #memlog=5 # memory debug log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=2 alias="test-domain.com" #user=ser #group=ser #open_fd_limit=1024 # sets the open file descriptors limit mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) #tcp_poll_method="sigio_rt" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/base/ser/directory/lib/ser/modules//sl.so" loadmodule "/base/ser/directory/lib/ser/modules/avp.so" loadmodule "/base/ser/directory/lib/ser/modules//avpops.so" loadmodule "/base/ser/directory/lib/ser/modules//tm.so" loadmodule "/base/ser/directory/lib/ser/modules//rr.so" loadmodule "/base/ser/directory/lib/ser/modules//maxfwd.so" loadmodule "/base/ser/directory/lib/ser/modules//usrloc.so" loadmodule "/base/ser/directory/lib/ser/modules//registrar.so" loadmodule "/base/ser/directory/lib/ser/modules//textops.so" loadmodule "/base/ser/directory/lib/ser/modules//mysql.so" loadmodule "/base/ser/directory/lib/ser/modules//dialog.so" loadmodule "/base/ser/directory/lib/ser/modules//rls.so" loadmodule "/base/ser/directory/lib/ser/modules/pa.so" loadmodule "/base/ser/directory/lib/ser/modules//presence_b2b.so" loadmodule "/base/ser/directory/lib/ser/modules//uri.so" loadmodule "/base/ser/directory/lib/ser/modules//uri_db.so" loadmodule "/base/ser/directory/lib/ser/modules//domain.so" loadmodule "/base/ser/directory/lib/ser/modules//fifo.so" loadmodule "/base/ser/directory/lib/ser/modules//xmlrpc.so" loadmodule "/base/ser/directory/lib/ser/modules//xlog.so" #loadmodule "/base/ser/directory/lib/ser/modules//unixsock.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/base/ser/directory/lib/ser/modules//auth.so" loadmodule "/base/ser/directory/lib/ser/modules//auth_db.so" loadmodule "/base/ser/directory/lib/ser/modules//msilo.so" # ----------------- setting module-specific parameters --------------- # modparam("msilo","registrar","sip:registrar@test-domain.com") modparam("msilo","use_contact",0) modparam("msilo","expire_time",7200) # -- usrloc params -- # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("rls", "min_expiration", 200) modparam("rls", "max_expiration", 300) modparam("rls", "default_expiration", 300) modparam("rls", "auth", "none") modparam("rls", "xcap_root", "http://localhost/xcap") modparam("rls", "reduce_xcap_needs", 1) modparam("rls", "db_mode", 0) modparam("rls", "db_url", "mysql://ser:heslo@localhost:3306/ser") modparam("pa", "use_db", 0) # allow storing authorization requests for offline users into database modparam("pa", "use_offline_winfo", 1) # how often try to remove old stored authorization requests modparam("pa", "offline_winfo_timer", 600) # how long stored authorization requests live modparam("pa", "offline_winfo_expiration", 600) modparam("pa", "db_url", "mysql://ser:heslo@localhost:3306/ser") # mode of PA authorization: none, implicit or xcap modparam("pa", "auth", "xcap") modparam("pa", "auth_xcap_root", "http://localhost/xcap") # do not authorize watcherinfo subscriptions modparam("pa", "winfo_auth", "none") # use only published information if set to 0 modparam("pa", "use_callbacks", 1) # don't accept internal subscriptions from RLS, ... modparam("pa", "accept_internal_subscriptions", 0) # maximum value of Expires for subscriptions modparam("pa", "max_subscription_expiration", 600) # maximum value of Expires for publications modparam("pa", "max_publish_expiration", 120) # how often test if something changes and send NOTIFY modparam("pa", "timer_interval", 10) # route for generated SUBSCRIBE requests for presence modparam("presence_b2b", "presence_route", "") # waiting time from error to new attepmt about SUBSCRIBE modparam("presence_b2b", "on_error_retry_time", 60) # how long wait for NOTIFY with Subscription-Status=terminated after unsubscribe modparam("presence_b2b", "wait_for_term_notify", 33) # how long before expiration send renewal SUBSCRIBE request modparam("presence_b2b", "resubscribe_delta", 30) # minimal time to send renewal SUBSCRIBE request from receiving previous response modparam("presence_b2b", "min_resubscribe_time", 60) # default expiration timeout modparam("presence_b2b", "default_expiration", 3600) # process internal subscriptions to presence events modparam("presence_b2b", "handle_presence_subscriptions", 1) modparam("usrloc", "db_mode", 0) modparam("domain", "db_mode", 1) modparam("domain|uri_db|acc|auth_db|usrloc|msilo", "db_url", "mysql://ser:heslo@localhost:3306/ser") modparam("fifo", "fifo_file", "/tmp/ser_fifo") # ------------------------- request routing logic ------------------- # main routing logic route{ # XML RPC if (method == "POST" || method == "GET") { create_via(); dispatch_rpc(); break; } # 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>= max_len ) { 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 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 (!lookup_domain("To")) { xlog("L_ERR", "Unknown domain to: %tu from: %fu\n"); route(1); break; } if (method=="SUBSCRIBE") { if (!t_newtran()) { sl_reply_error(); break; }; if (@to.tag=="") { # only for new subscriptions (with empty to tag) if (lookup_user("To")) { # existing user -> it is subscription to PA if (handle_subscription("registrar")) { if ((@msg.event=~"presence\.winfo")) { # new watcher info subscription # sends one watcher info NOTIFY message with all saved authorization requests xlog("L_ERR", "dumping stored winfo to %fu\n"); dump_stored_winfo("registrar", "presence"); } else { # new presence subscription if ((@msg.event=~"presence") && (%subscription_status=="pending")) { # if offline user and new pending subscription if (!target_online("registrar")) { #%subscription_status="waiting"; # store it as waiting subscription xlog("L_ERR", "storing 'pending' winfo to: %tu, from: %fu\n"); store_winfo("registrar"); } } } } break; } if ((@msg.supported=~"eventlist")) { # such user doesn't exist and Supported header field # -> probably RLS subscription if (lookup_domain("From")) { if (lookup_user("From")) { if (is_simple_rls_target("$uid-list")) { # log(1, "it is simple subscription!\n"); # takes From UID and makes XCAP query for user's # list named "default" if (!query_resource_list("default")) { t_reply("404", "No such user list"); break; } } } } if (!have_flat_list()) { # query_resource_list failed or was not called # do standard RLS query acording to To/AOR if (!query_rls_services()) { log(1, "XCAP query failed\n"); t_reply("404", "No such list URI"); break; } } # uncomment this if you want to authenticate first SUBSCRIBE request to resource list # if (!proxy_authenticate("test-domain.com", "credentials")) { # proxy_challenge( "test-domain.com", "0"); # break; # }; handle_rls_subscription("1"); } else { # not resource list subscription -> invalid user xlog("L_ERR", "subscription to invalid user %tu\n"); t_reply("404", "User not found"); } break; } else { # renewal subscriptions - try to handle it as RLS and if failed, handle it as PA subscription # FIXME: better will be test like existing_rls_subscription() # and existing_subscription("registrar") if (!handle_rls_subscription("0")) { handle_subscription("registrar"); } break; } }; # get user (common for all other messages than SUBSCRIBE) if (!lookup_user("To")) { # log(1, "Unknown user - message should be forwarded?"); # # break; append_hf("P-hint: unknown user\r\n"); route(1); break; } if (method=="PUBLISH") { if (!t_newtran()) { # log(1, "newtran error\n"); sl_reply_error(); break; }; handle_publish("registrar"); # deliver messages to online user # TODO: only if user goes from offline to online? if (target_online("registrar")) { # log(1, "Dumping stored messages\n"); # dump stored messages - route it through myself (otherwise routed via DNS!) if (m_dump("sip:127.0.0.1")) { xlog("L_ERR", "MSILO: offline messages for %fu dumped\n"); } } break; }; if (method=="NOTIFY") { if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); break; }; # handle notification sent in internal subscriptions (presence_b2b) if (!handle_notify()) { t_reply("481", "Unable to handle notification"); } break; }; if (method=="MESSAGE") { if (authorize_message("http://localhost/xcap")) { # use usrloc for delivery if (lookup("location")) { log(1, "Delivering MESSAGE using usrloc\n"); t_on_failure("1"); if (!t_relay()) { sl_reply_error(); } break; } else { # store messages for offline user xlog("L_ERR", "MSILO: storing MESSAGE for %tu\n"); if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); break; }; # store only text messages NOT isComposing... ! if (search("^(Content-Type|c):.*application/im-iscomposing\+xml.*")) { log(1, "it is only isComposing message - ignored\n"); t_reply("202", "Ignored"); break; } if (m_store("0", "sip:127.0.0.1")) { # log(1, "MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; } else { log(1, "MSILO: error storing offline message\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; break; } break; } else { # log(1, "unauthorized message\n"); sl_reply("403", "Forbidden"); } break; } if (method=="REGISTER") { # uncomment this if you want to authenticate REGISTER request # if (!www_authenticate("", "credentials")) { # www_challenge("", "0"); # break; # }; save("location"); # dump stored messages - route it through myself (otherwise routed via DNS!) if (m_dump("sip:127.0.0.1")) { xlog("L_ERR", "MSILO: offline messages for %fu dumped\n"); } 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(); }; } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1, "MSILO: MESSAGE forward failed - storing it\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("0", "")) { t_reply("202", "Accepted"); } else { log(1, "MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } _________________________________________________________________ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE

2 1

[Serusers] SER and rtpproxy stability
by Steven Wang 27 Nov '07

27 Nov '07

Hello We use SER and rtpproxy for a couple of years over public network. It worked perfectly before. The users from different LNA can talk to each other over SIP call (xlite or Linksys adapters). We could fax over SIP from other countries. However, fax stopped working shortly (about 6-month). But SIP call worked fine till the beginning of this year. Suddenly, sip-phone only hears ring. There is no sound on both sides. Do anyone have the similar problem or encounter stability issues in this setup (SER + Rtpproxy)? Best Steven

2 1

Re: [Serusers] [Sems] How to setup conference?
by roayu 27 Nov '07

27 Nov '07

Hi, >From the documentation (mini-Howto on how to configure SER and SEMS to work together), the SER that it's mentioned, does it means the SIP-stack SER or the Proxy SER? I'm having 2 SERs in one machine and I'm a bit confuse... Hope someone can give me a hint =) Thanks. Roa Yu -----Original Message----- From: sems-bounces(a)lists.iptel.org [mailto:sems-bounces@lists.iptel.org] On Behalf Of roayu Sent: Monday, November 26, 2007 10:40 PM To: 'sems(a)lists.iptel.org' Subject: [Sems] How to setup conference? Hi, I'm new to SEMS and i managed to setup 2 SERs and 1 SEMS to work together. However, i'm not sure how to configure the Conference. Can anyone guide me on the conference setup? Thanks. Roa Yu _______________________________________________ Sems mailing list Sems(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/sems

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Re: [Serusers] Can SER support MySQL 5.0.22
by roayu 27 Nov '07

27 Nov '07

Hi, I managed to create the mysql.so after re-intall the MySQL, especially the mysql-dev package. Thanks for your help! Roa Yu -----Original Message----- From: SIP [mailto:sip@arcdiv.com] Sent: Monday, November 26, 2007 8:36 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Can SER support MySQL 5.0.22 First off, do you have the mysql-dev package installed? As for whether or not it will run with mysql 5.0.22, it can, but it's tricky. The makefile for the mysql module doesn't look everywhere for the libmysqlclient libs when it's compiling. It expects the libs to be in a particular place, and if they're not there, it ignores the libpath, compiles ANYway, and they just fails to work once you start up SER. It gives NO errors on compile, which makes it utterly difficult to track down. The latest RPMs for RHEL/CentOS from the MySQL community site place the libmysqlclient.so.X libs into /usr/lib and NOT /usr/lib/mysql as SER expects. SER will simply NOT find it in that location. What we've ended up having to do is symlinking the /usr/lib/libmysqlclient* into /usr/lib/mysql, recompile the mysql module, and install it in the correct place. Then, SER runs happily and readily without weird errors. Also note that there is a patch that must be applied to use MySQL 5.0.X for SER 0.9.6 see http://www.iptel.org/faq/why_does_my_mysql_5_0_x_connection_time_out_when_us ing_ser_0_9_x N. roayu wrote: > > Hi there! > > > > Would like to ask, can SER 0.9.6 support MySQL 5.0.22 ? I got the > following errors when I tried start SER and to connect to MySQL: > > > > [root@localhost ~]# /usr/local/sbin/ser > > 0(3291) ERROR: load_module: could not open module > </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.14: cannot > open shared object file: No such file or directory > > 0(3291) parse error (31,13-14): failed to load module > > ERROR: bad config file (1 errors) > > > > > > Thanks. > > > > Roa Yu > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

1 0

[OpenSER-Users] OpenSER Sending INVITE twice
by Douglas Garstang 27 Nov '07

27 Nov '07

I have an OpenSER 1.1 install here, and for some reason, OpenSER is sending the INVITE message twice to the upstream host. The time difference between them is about 1/5 of a second. Nothing is received between the first and second INVITE's. Why is OpenSER doing this? I have xlog() statements everywhere in openser.cfg, and OpenSER is only logging ONE outgoing INVITE message, eventhough it's sending two. It is logging the multiple TRYING messages that come back however. OpenSER is not calling failure_route because nothing is logged in there. What could be going wrong? Why is it doing this? Doug. ____________________________________________________________________________________ Get easy, one-click access to your favorites. Make Yahoo! your homepage. http://www.yahoo.com/r/hs

2 1

[Serusers] Invoking force_rtp_proxy() after fix_nated_sdp() error
by Tomasz Zieleniewski 26 Nov '07

26 Nov '07

Hi, Invoking force_rtp_proxy() after fix_nated_sdp() causes error that in the connection filed one ip address apears after second (previous one is not removed) v=0. o=- 3 2 IN IP4 192.168.1.3. s=CounterPath X-Lite 3.0. c=IN IP4 82.210.160.59192.168.129.74. t=0 0. m=audio 35036 RTP/AVP 107 119 100 106 0 105 98 8 101. 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. a=direction:active. a=oldmediaip:192.168.1.3. a=nortpproxy:yes. in the example above fix_nated_sdp put the 82.210.160.59 address and force_rtp_proxy instead of replacing the 82.210.160.59 appended it at the end of the line Cheers TOmasz

2 2

[OpenSER-Users] Opensedbctl Error
by TCB 26 Nov '07

26 Nov '07

Hello all, Just checkout the lates svn version of OSer. trying to run openserdbctl create with DB=PGSQL i get the following errors. ERROR: relation "subscriber" does not exist ERROR: relation "dbaliases" does not exist ERROR: Grant privileges to database failed! Any pointers welcome. OSer details: Repository UUID: 689a6050-402a-0410-94f2-e92a70836424 Revision: 3203 -- TC

2 4

[OpenSER-Users] Openser-tls - Registration Error: 503
by Marcio Ribeiro 26 Nov '07

26 Nov '07

Hi, i'm using openser-tls 1.0.1, and i intend to estabilish a security call between 2 users agents. When using eyeBeam 1.5.6 in the automatic mode i can register and make calls, no problem. But when i choose TLS in the softphone i always get Registration Error: 503 - Service Unavailable. the eyeBeam log shows: [07-11-24]13:01:01.386 | Info (debug) RESIP:TRANSPORT | "Adding message to tx buffer to: [ V4 xxx.xx.xx.xxx:5061 TLS target domain=xxx.xx.xx.xxxreceived on: Transport: [ V4 0.0.0.0:35085 TLS target domain=unspecified connectionId=0 ] connectionId=0 ]" | [07-11-24]13:01:01.487 | Info (debug) RESIP:TRANSPORT | "Processing write for [ V4 xxx.xx.xx.xxx:5061 TLS target domain=xxx.xx.xx.xxx received on: Transport: [ V4 0.0.0.0:35085 TLS target domain=unspecified connectionId=0 ] connectionId=0 ]" | [07-11-24]13:01:01.487 | Info (debug) RESIP:TRANSPORT | "Creating fd=1660 V4/TCP" | [07-11-24]13:01:01.487 | Info (debug) RESIP:TRANSPORT | "Opening new connection to [ V4 xxx.xx.xx.xxx:5061 TLS target domain=xxx.xx.xx.xxxreceived on: Transport: [ V4 0.0.0.0:35085 TLS target domain=unspecified connectionId=0 ] connectionId=0 ]" | [07-11-24]13:01:01.488 | Info (debug) RESIP:TRANSPORT | "ConnectionBase::ConnectionBase, who: [ V4 xxx.xx.xx.xxx:5061 TLS target domain=xxx.xx.xx.xxx received on: Transport: [ V4 0.0.0.0:35085 TLS target domain=unspecified connectionId=0 ] connectionId=0 ] 024AA840" | [07-11-24]13:01:01.488 | Info (more) RESIP:TRANSPORT | "Creating TLS connection for domain [ V4 xxx.xx.xx.xxx:5061 TLS target domain= xxx.xx.xx.xxx received on: Transport: [ V4 0.0.0.0:35085 TLS target domain=unspecified connectionId=0 ] connectionId=0 ] on 1660" | [07-11-24]13:01:01.488 | Info (debug) RESIP:TRANSPORT | "Trying to form TLS connection - acting as client" | [07-11-24]13:01:02.405 | Info (more) RESIP:TRANSPORT | "Exception writing to socket 1660 code: 10061; closing connection" | [07-11-24]13:01:02.406 | Info (debug) RESIP:TRANSPORT | "ConnectionBase::~ConnectionBase 024AA840" | [07-11-24]13:01:02.406 | Info (more) RESIP:TRANSACTION | "Sending ConnectionTerminated 4 to TUs" | [07-11-24]13:01:02.406 | Info (more) RESIP:TRANSACTION | "Try sending request to a different dns result" | [07-11-24]13:01:02.406 | Info (more) RESIP:TRANSACTION | "Ran out of dns entries for xxx.xx.xx.xxx. Send 503" | [07-11-24]13:01:02.406 | Info (debug) RESIP | "Helper::makeResponse(SipReq: REGISTER xxx.xx.xx.xxx tid=a1596d01462f2d03 cseq=REGISTER contact= marcio@yyy.yy.yy.yy:35085 / 1 from(tu) code=503 reason=" | [07-11-24]13:01:02.408 | Warning (min) RESIP:DNS | "local hostname does not contain a domain part" | [07-11-24]13:01:02.408 | Info (debug) RESIP:TRANSACTION | "Send to TU: TU: DialogUsageManager size=1 SIP/2.0 503 Service Unavailable Via: SIP/2.0/TLS yyy.yy.yy.yy :35085;branch=z9hG4bK-d87543-a1596d01462f2d03-1--d87543-;rport To: ""Marcio""<sip:marcio@xxx.xx.xx.xxx>;tag=b76ee406 From: ""Marcio""<sip:marcio@xxx.xx.xx.xxx>;tag=bf4bff41 Call-ID: 507d385424726b5fOGFmYzRkOTMwZjFmOTNjNzkyYmE1MjI4MDY0YzAwNzg. CSeq: 1 REGISTER Warning: 499 DUMNONIA """" Content-Length: 0 searching on google doesn't give-me much stuf about this. I really need to implement secure communication... the softphone doesn't matter. I have no idea of what to do, and i really need help... thanks, Marcio Ribeiro

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[OpenSER-Users] error: use_media_proxy(): out of memory
by Tomasz Zieleniewski 26 Nov '07

26 Nov '07

Hi, I am using media proxy for nat handling when I invoke use_media_proxy() for some calls I get the following error: error: use_media_proxy(): out of memory please point me what is wrong?? (I know that some problem with memory:) but why?) Cheers Tomasz

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Re: [Serusers] error: use_media_proxy(): out of memory
by Jiri Kuthan 26 Nov '07

26 Nov '07

memory leak? nathelper may provide you with similar functionality and there is a good chance you will not see this problem. -jiri At 13:20 26/11/2007, Tomasz Zieleniewski wrote: >Hi, > >I am using media proxy for nat handling when I invoke >use_media_proxy() for some calls from my >pstn gateway I get the following error: >error: use_media_proxy(): out of memory >please point me what is wrong?? >(I know that some problem with memory:) but why?) > >cheers >tomasz >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] Can SER support MySQL 5.0.22
by roayu 26 Nov '07

26 Nov '07

Hi there! Would like to ask, can SER 0.9.6 support MySQL 5.0.22 ? I got the following errors when I tried start SER and to connect to MySQL: [root@localhost ~]# /usr/local/sbin/ser 0(3291) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.14: cannot open shared object file: No such file or directory 0(3291) parse error (31,13-14): failed to load module ERROR: bad config file (1 errors) Thanks. Roa Yu

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[Serusers] setavpflag() - how to access avp afterwards
by Tomasz Zieleniewski 26 Nov '07

26 Nov '07

Hi, How can I access avps from the Route header set with the usage of setavpflag() method after receiving of the message? Are they somehow accessible through global avps?? Cheers tomasz

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[Serusers] error: use_media_proxy(): out of memory
by Tomasz Zieleniewski 26 Nov '07

26 Nov '07

Hi, I am using media proxy for nat handling when I invoke use_media_proxy() for some calls from my pstn gateway I get the following error: error: use_media_proxy(): out of memory please point me what is wrong?? (I know that some problem with memory:) but why?) cheers tomasz

1 0

[OpenSER-Users] WeSIP and Click2dial application
by Daniel 26 Nov '07

26 Nov '07

Hi all, Can somebody help me to install an run WeSip with Click2dial application ? Is there some documentation file about it ? Regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] Modify $au to lowercase
by Peter P GMX 26 Nov '07

26 Nov '07

Hello, is there any chance to modify the authorization user name? I have the following scenario. We do allow only lowercase usernames when users are registering. So the md5 password is created and stored based on lowercase letters. When a user receives his account data he might enter his username in camelcase in his user agent. So when he tries to register, it fails, as the md5 hash based on the camelcased username is different. I tried to use the perl module, put the username to lowercase and put it back to OpenSER via avp_pushto. But Openser only allows $ru,$du,$br to be modified. Do I have a chance to modify $au? Best regards Peter

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Re: [Serusers] Can SER support 3-way Conference
by roayu 26 Nov '07

26 Nov '07

Hi, I managed to start SEMS as well as the 2 SERs. Now, I'm not sure how to start a conference using SEMS. I would like to have the follwing conference: UA1 call to UA2, the UA2 call to UA3. Then UA2 assign the conference call. UA1 --> UA2 (Agent who starts the conference) --> UA3 Conference : UA1 + UA2 + UA3 Can anyone guide me on how to do so? Thanks. Roa Yu -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of roayu Sent: Monday, November 26, 2007 9:44 AM To: Stefan Sayer Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Can SER support 3-way Conference Hi Stefan, Ops, I thought I installed the g++ ... Anyway, thanks for ur help! Really appreciate that. Btw, previously I do have one SER running in the machine. Now, I'm having 2 SERs and 1 SEMS. So, do I need to configure anything on my 1st SER? Or I only need to configure on the SIP-stack SER? If no, where else that I need to configure? Thanks. Roa Yu -----Original Message----- From: Stefan Sayer [mailto:stefan.sayer@iptego.de] Sent: Saturday, November 24, 2007 7:02 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Can SER support 3-way Conference Hello, why not have a look at what is print on the screen? make[3]: g++: Command not found You either havent g++ installed (you will need at least g++ 3.4 - but I would recommend 4.1, see README; on debian: apt-get install g++-4.1), or you don't have a symlink to it: ln -s /usr/bin/g++-x.y /usr/bin/g++. hth Stefan roayu wrote: > Hi Stefan, > > The following are the output I got after re-run the 'make install' on SEMS. > > [root@localhost sems-0.10.0-rc2]# make install > > make[1]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps' > > make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' > make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' > make[3]: Nothing to be done for `deps'. > make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' > make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' > g++ -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE > -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -c > AnnounceB2B.cpp -o AnnounceB2B.o > make[3]: g++: Command not found > make[3]: *** [AnnounceB2B.o] Error 127 > make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' > make[2]: [all] Error 2 (ignored) > mkdir -p /usr/local/lib/sems/audio/ann_b2b > mkdir -p /usr/local/lib/sems/plug-in/ > touch /usr/local/lib/sems/plug-in//ann_b2b.so ; \ > install -m 755 ann_b2b.so /usr/local/lib/sems/plug-in/ > install: cannot stat `ann_b2b.so': No such file or directory > make[2]: *** [install] Error 1 > make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' .... -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.de www.iptego.de iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[OpenSER-Users] Mediaproxy relay and mixed network situation
by Antoine Fressancourt 26 Nov '07

26 Nov '07

Hello list, I have a question about media relay in a quite particular situation. I have two users, say A and B, connected on a softphone and a "hard" phone. These two devices are on two different networks, and media proxy must be used to communicate between the two. When A calls B with his softphone, if B is registered on both the softphone and the "hard" one, the two phones ring. but when B answers with the softphone, the request is flagged to use the media proxy. Do you know how I can do in order to avoid this behaviour ? Thank you in advance for your answer, Antoine

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Re: [Serusers] Can SER support 3-way Conference
by roayu 26 Nov '07

26 Nov '07

Hi Stefan, Ops, I thought I installed the g++ ... Anyway, thanks for ur help! Really appreciate that. Btw, previously I do have one SER running in the machine. Now, I'm having 2 SERs and 1 SEMS. So, do I need to configure anything on my 1st SER? Or I only need to configure on the SIP-stack SER? If no, where else that I need to configure? Thanks. Roa Yu -----Original Message----- From: Stefan Sayer [mailto:stefan.sayer@iptego.de] Sent: Saturday, November 24, 2007 7:02 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Can SER support 3-way Conference Hello, why not have a look at what is print on the screen? make[3]: g++: Command not found You either havent g++ installed (you will need at least g++ 3.4 - but I would recommend 4.1, see README; on debian: apt-get install g++-4.1), or you don't have a symlink to it: ln -s /usr/bin/g++-x.y /usr/bin/g++. hth Stefan roayu wrote: > Hi Stefan, > > The following are the output I got after re-run the 'make install' on SEMS. > > [root@localhost sems-0.10.0-rc2]# make install > > make[1]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps' > > make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' > make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' > make[3]: Nothing to be done for `deps'. > make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' > make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' > g++ -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE > -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -c > AnnounceB2B.cpp -o AnnounceB2B.o > make[3]: g++: Command not found > make[3]: *** [AnnounceB2B.o] Error 127 > make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' > make[2]: [all] Error 2 (ignored) > mkdir -p /usr/local/lib/sems/audio/ann_b2b > mkdir -p /usr/local/lib/sems/plug-in/ > touch /usr/local/lib/sems/plug-in//ann_b2b.so ; \ > install -m 755 ann_b2b.so /usr/local/lib/sems/plug-in/ > install: cannot stat `ann_b2b.so': No such file or directory > make[2]: *** [install] Error 1 > make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' .... -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.de www.iptego.de iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

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[Serusers] Please join me @ Sarfu
by Sarfaraz Chougule 25 Nov '07

25 Nov '07

1 0

[Serusers] Please join me @ Sarfu
by Sarfaraz Chougule 25 Nov '07

25 Nov '07

1 0

Re: [Serusers] nathelper doesn't ping registered users
by Jiri Kuthan 25 Nov '07

25 Nov '07

Hi Tomasz, well it could be that reusing an existing config and commiting than your specific changes to it would save you quite some laborous exercises rather than writing it from scratch.... http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/etc/ser-oob.cfg?re… The lines in question are (out of which I think you are missing the registrar params) FLAG_NAT : 3, # the UAC is behind a NAT ... # identify natted contacts using a flag modparam("registrar", "load_nat_flag", "FLAG_NAT") modparam("registrar", "save_nat_flag", "FLAG_NAT") ... modparam("rr", "cookie_filter", "(account|uac_nat)") ... modparam("nathelper", "ping_nated_only", 1 ) # if this option is not set, simple 4-bytes ping is sent modparam("nathelper", "natping_method", "OPTIONS" ) #temporary statefull natping test (only in future versions) #modparam("nathelper", "natping_stateful", 1) ... setflag(FLAG_NAT); -jiri At 22:20 24/11/2007, Tomasz Zieleniewski wrote: >I'm sorry for the CC >here is my config: > >debug=3 # debug level (cmd line: -dddddddddd) >memdbg=5 >memlog=5 >log_facility=LOG_LOCAL0 >fork=yes >log_stderror=no >check_via=yes # (cmd. line: -v) >dns=no # (cmd. line: -r) >rev_dns=no # (cmd. line: -R) >port=5060 >tls_port_no=5061 >children=4 >user=ser >group=ser >mhomed=yes >disable_tcp=no >tcp_accept_aliases=yes >enable_tls=no >sip_warning=0 >reply_to_via=no >alias="voip.touk.pl" > >dns_retr_time=1 >dns_retr_no=1 >dns_servers_no=1 >dns_use_search_list=no >dns_try_ipv6=no ># dns cache & failover >use_dns_cache=on >use_dns_failover=on ># dns_cache_flags=0 >dns_cache_negative_ttl=300 >dns_cache_min_ttl=60 >dns_cache_max_ttl=86400 # 1 day >dns_cache_mem=2048 # 2 MB >dns_cache_gc_interval=60 > >tcp_connection_lifetime=3600 >tcp_max_connections=2048 >tcp_connect_timeout=1 > >loadmodule ... >.... > ># specify the path to you database here >modparam("acc_db|auth_db|avp_db|domain|gflags|usrloc|uri_db|speeddial|msilo|lcr", >"db_url", "mysql://ser:ser@localhost/ser") > ># -- usrloc params -- >modparam("usrloc", "db_mode", 1) > ># -- registrar params -- >modparam("registrar", "default_expires", 300) >modparam("registrar", "min_expires", 60) >modparam("registrar", "max_expires", 300) >modparam("registrar", "append_branches", 1) >modparam("registrar", "received_param", "rcv") >modparam("registrar", "max_contacts", 2) > ># -- auth params -- >modparam("auth_db", "calculate_ha1", yes) >modparam("auth_db", "password_column", "password") > ># -- rr params -- >modparam("rr", "enable_full_lr", 1) >modparam("rr", "cookie_filter", "uac_nat") >modparam("rr", "cookie_secret", "secret") >modparam("rr", "user_part_avp", "route_user") >modparam("rr", "next_route_avp", "route_next") > ># -- gflags params -- >modparam("gflags", "load_global_attrs", 1) > ># -- domain params -- >modparam("domain", "load_domain_attrs", 1) > ># -- ctl params -- >modparam("ctl", "binrpc", "unixs:/tmp/ser-bsp_ctl") >modparam("ctl", "fifo", "fifo:/tmp/ser-bsp_fifo") ># modparam("ctl", "binrpc", "tcp:localhost:2046") >modparam("ctl", "mode", 0777) > ># -- tm params -- >modparam("tm", "restart_fr_on_each_reply", 1) > ># -- auth_radius params -- >modparam("auth_radius", "radius_config", >"/usr/local/etc/radiusclient-ng/radiusclient.conf") >modparam("auth_radius", "service_type", 15) > ># -- options params -- >modparam("options", "accept", "*/*") >modparam("options", "accept_encoding", "") >modparam("options", "accept_language", "en") > ># -- ratelimit params -- >modparam("ratelimit","timer_interval", 10) >modparam("ratelimit","invite_limit", 100) >modparam("ratelimit","register_limit", 100) >modparam("ratelimit","subscribe_limit", 100) > ># -- xmlrpc params -- >modparam("xmlrpc", "route","RPC") > ># -- nathelper params -- >modparam("nathelper", "natping_interval", 20) >modparam("nathelper", "ping_nated_only", 0) >modparam("nathelper", "rtpproxy_disable", 1) >modparam("nathelper", "tcpping_crlf", 1) > ># -- tls params -- >modparam("tls", "config", "/home/ser/ser/ca/tls-bsp.cfg") >modparam("tls", "tls_force_run", 0) >modparam("tls", "tls_log",3) >modparam("tls", "handshake_timeout", 10) >modparam("tls", "send_timeout", 10) >modparam("tls", "tls_disable_compression", 0) >#modparam("tls", "private_key", "/etc/certs/key.pem") >#modparam("tls", "certificate", "/etc/certs/cert.pem") >#modparam("tls", "ca_list", "/etc/certs/ca_list.pem") >#modparam("tls", "require_certificate", 0) >#modparam("tls", "verify_certificate", 1) >#modparam("tls", "tls_method", "TLSv1") > >route{ >... >} > >On Nov 24, 2007 9:35 PM, Atle Samuelsen <clona(a)cyberhouse.no> wrote: >> Hi, >> >> Always CC the list. >> >> then I dont know.. post your full config so people can see if they see >> something >> -A >> >> * Tomasz Zieleniewski <tzieleniewski(a)gmail.com> [071124 20:38]: >> >> > Hi, >> > >> > I had that parameter set to 0, but still no udp packets... >> > >> > On Nov 24, 2007 6:43 PM, Atle Samuelsen <clona(a)cyberhouse.no> wrote: >> > > * Tomasz Zieleniewski <tzieleniewski(a)gmail.com> [071124 16:50]: >> > > >> > > > HI, >> > > > >> > > > I have the following nathelper settings: >> > > > modparam("nathelper", "natping_interval", 20) >> > > > modparam("nathelper", "ping_nated_only", 1) >> > > > modparam("nathelper", "rtpproxy_disable", 1) >> > > > modparam("nathelper", "tcpping_crlf", 1) >> > > > >> > > > But i cant see any udp packets being sent to registered users. >> > > > Please point me what did I miss. >> > > >> > > Hi, >> > > >> > > without seeing any more of your config, I'm guessing that you do not set >> > > the natflag, so if you change ping_nated_only to 0, I think you will see >> > > some udp packets. >> > > >> > > - Atle >> > > (shot in the whild world of sip) >> > > >> > > >> > > > >> > > > Best >> > > > Tomasz >> > > > _______________________________________________ >> > > > 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 -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] nathelper doesn't ping registered users
by Atle Samuelsen 24 Nov '07

24 Nov '07

Hi, Always CC the list. then I dont know.. post your full config so people can see if they see something -A * Tomasz Zieleniewski <tzieleniewski(a)gmail.com> [071124 20:38]: > Hi, > > I had that parameter set to 0, but still no udp packets... > > On Nov 24, 2007 6:43 PM, Atle Samuelsen <clona(a)cyberhouse.no> wrote: > > * Tomasz Zieleniewski <tzieleniewski(a)gmail.com> [071124 16:50]: > > > > > HI, > > > > > > I have the following nathelper settings: > > > modparam("nathelper", "natping_interval", 20) > > > modparam("nathelper", "ping_nated_only", 1) > > > modparam("nathelper", "rtpproxy_disable", 1) > > > modparam("nathelper", "tcpping_crlf", 1) > > > > > > But i cant see any udp packets being sent to registered users. > > > Please point me what did I miss. > > > > Hi, > > > > without seeing any more of your config, I'm guessing that you do not set > > the natflag, so if you change ping_nated_only to 0, I think you will see > > some udp packets. > > > > - Atle > > (shot in the whild world of sip) > > > > > > > > > > Best > > > Tomasz > > > _______________________________________________ > > > Serusers mailing list > > > Serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > >

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[Serusers] nathelper doesn't ping registered users
by Tomasz Zieleniewski 24 Nov '07

24 Nov '07

HI, I have the following nathelper settings: modparam("nathelper", "natping_interval", 20) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "tcpping_crlf", 1) But i cant see any udp packets being sent to registered users. Please point me what did I miss. Best Tomasz

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Re: [Serusers] Can SER support 3-way Conference
by roayu 24 Nov '07

24 Nov '07

Hi Stefan, The following are the output I got after re-run the 'make install' on SEMS. [root@localhost sems-0.10.0-rc2]# make install make[1]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' make[3]: Nothing to be done for `deps'. make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' g++ -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -c AnnounceB2B.cpp -o AnnounceB2B.o make[3]: g++: Command not found make[3]: *** [AnnounceB2B.o] Error 127 make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' make[2]: [all] Error 2 (ignored) mkdir -p /usr/local/lib/sems/audio/ann_b2b mkdir -p /usr/local/lib/sems/plug-in/ touch /usr/local/lib/sems/plug-in//ann_b2b.so ; \ install -m 755 ann_b2b.so /usr/local/lib/sems/plug-in/ install: cannot stat `ann_b2b.so': No such file or directory make[2]: *** [install] Error 1 make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/announcement' make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/announcement' make[3]: Nothing to be done for `deps'. make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/announcement' make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/announcement' g++ -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -c Announcement.cpp -o Announcement.o make[3]: g++: Command not found make[3]: *** [Announcement.o] Error 127 make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/announcement' make[2]: [all] Error 2 (ignored) mkdir -p /usr/local/lib/sems/audio/announcement mkdir -p /usr/local/lib/sems/plug-in/ touch /usr/local/lib/sems/plug-in//announcement.so ; \ install -m 755 announcement.so /usr/local/lib/sems/plug-in/ install: cannot stat `announcement.so': No such file or directory make[2]: *** [install] Error 1 make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/announcement' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/announce_transfer' make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/announce_transfer' make[3]: Nothing to be done for `deps'. make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/announce_transfer' make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/announce_transfer' g++ -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -c AnnounceTransfer.cpp -o AnnounceTransfer.o make[3]: g++: Command not found make[3]: *** [AnnounceTransfer.o] Error 127 make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/announce_transfer' make[2]: [all] Error 2 (ignored) mkdir -p /usr/local/lib/sems/audio/announce_transfer mkdir -p /usr/local/lib/sems/plug-in/ touch /usr/local/lib/sems/plug-in//announce_transfer.so ; \ install -m 755 announce_transfer.so /usr/local/lib/sems/plug-in/ install: cannot stat `announce_transfer.so': No such file or directory make[2]: *** [install] Error 1 make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/announce_transfer' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/conf_auth' find . -iname "*\.pyc" -o -iname "*\.py~" | xargs rm -f rm -f sems-app-conf_auth*.tar.gz python2.4 ../ivr/py_comp -q . install -d /"/usr/local/lib/sems/ivr" install -m "644" -o "root" -g "root" *.pyc /"/usr/local/lib/sems/ivr" install -d /"/usr/local/lib/sems/ivr"/. install -m "644" -o "root" -g "root" ./*.pyc /"/usr/local/lib/sems/ivr"/. make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/conf_auth' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/conference' make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/conference' make[3]: Nothing to be done for `deps'. make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/conference' make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/conference' g++ -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -c AmRingTone.cpp -o AmRingTone.o make[3]: g++: Command not found make[3]: *** [AmRingTone.o] Error 127 make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/conference' make[2]: [all] Error 2 (ignored) mkdir -p /usr/local/lib/sems/audio/conference mkdir -p /usr/local/lib/sems/plug-in/ touch /usr/local/lib/sems/plug-in//conference.so ; \ install -m 755 conference.so /usr/local/lib/sems/plug-in/ install: cannot stat `conference.so': No such file or directory make[2]: *** [install] Error 1 make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/conference' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/early_announce' make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/early_announce' make[3]: Nothing to be done for `deps'. make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/early_announce' make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/early_announce' g++ -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -c EarlyAnnounce.cpp -o EarlyAnnounce.o make[3]: g++: Command not found make[3]: *** [EarlyAnnounce.o] Error 127 make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/early_announce' make[2]: [all] Error 2 (ignored) mkdir -p /usr/local/lib/sems/plug-in/ touch /usr/local/lib/sems/plug-in//early_announce.so ; \ install -m 755 early_announce.so /usr/local/lib/sems/plug-in/ install: cannot stat `early_announce.so': No such file or directory make[2]: *** [install] Error 1 make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/early_announce' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/ivr' python2.4 py_comp -l -q py &&\ cp py/*.pyc ../../core/lib make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/ivr' g++ -MM IvrDialogBase.cpp -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -I/usr/include/python2.4 -fno-strict-aliasing > IvrDialogBase.d /bin/sh: g++: command not found make[3]: *** [IvrDialogBase.d] Error 127 make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/ivr' make[2]: [all] Error 2 (ignored) echo "Installing Python object files..." Installing Python object files... for f in log.pyc ; do \ if [ -n "py/$f" ]; then \ touch /usr/local/lib/sems/plug-in//$f; \ install -m 755 py/$f /usr/local/lib/sems/plug-in//$f; \ fi ; \ done mkdir -p /usr/local/lib/sems/plug-in/ touch /usr/local/lib/sems/plug-in//ivr.so ; \ install -m 755 ivr.so /usr/local/lib/sems/plug-in/ install: cannot stat `ivr.so': No such file or directory make[2]: *** [install] Error 1 make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/ivr' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/mailbox' find . -iname "*\.pyc" -o -iname "*\.py~" | xargs rm -f rm -f sems-app-mailbox*.tar.gz python2.4 ../ivr/py_comp -q . install -d /"/usr/local/lib/sems/ivr" install -m "644" -o "root" -g "root" *.pyc /"/usr/local/lib/sems/ivr" install -d /"/usr/local/lib/sems/ivr"/imap_mailbox install -m "644" -o "root" -g "root" imap_mailbox/*.pyc /"/usr/local/lib/sems/ivr"/imap_mailbox make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/mailbox' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/pin_collect' find . -iname "*\.pyc" -o -iname "*\.py~" | xargs rm -f rm -f sems-app-pincollect*.tar.gz python2.4 ../ivr/py_comp -q . install -d /"/usr/local/lib/sems/ivr" install -m "644" -o "root" -g "root" *.pyc /"/usr/local/lib/sems/ivr" install -d /"/usr/local/lib/sems/ivr"/. install -m "644" -o "root" -g "root" ./*.pyc /"/usr/local/lib/sems/ivr"/. make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/pin_collect' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/py_sems' make[2]: *** No rule to make target `python_files', needed by `all'. Stop. make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/py_sems' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/registrar_client' make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/registrar_client' make[3]: Nothing to be done for `deps'. make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/registrar_client' make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/registrar_client' g++ -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -c ContactInfo.cpp -o ContactInfo.o make[3]: g++: Command not found make[3]: *** [ContactInfo.o] Error 127 make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/registrar_client' make[2]: [all] Error 2 (ignored) mkdir -p /usr/local/lib/sems/plug-in/ touch /usr/local/lib/sems/plug-in//registrar_client.so ; \ install -m 755 registrar_client.so /usr/local/lib/sems/plug-in/ install: cannot stat `registrar_client.so': No such file or directory make[2]: *** [install] Error 1 make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/registrar_client' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/voicemail' make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/voicemail' make[3]: Nothing to be done for `deps'. make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/voicemail' make[3]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/voicemail' g++ -I ../../core -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' -c AnswerMachine.cpp -o AnswerMachine.o make[3]: g++: Command not found make[3]: *** [AnswerMachine.o] Error 127 make[3]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/voicemail' make[2]: [all] Error 2 (ignored) mkdir -p /usr/local/etc/sems/ touch /usr/local/etc/sems/default.template.sample install -m 644 default.template.sample /usr/local/etc/sems/ if [ ! -f /usr/local/etc/sems/default.template ]; then \ cp -f /usr/local/etc/sems/default.template.sample \ /usr/local/etc/sems/default.template; \ fi mkdir -p /usr/local/lib/sems/audio/voicemail mkdir -p /usr/local/lib/sems/plug-in/ touch /usr/local/lib/sems/plug-in//voicemail.so ; \ install -m 755 voicemail.so /usr/local/lib/sems/plug-in/ install: cannot stat `voicemail.so': No such file or directory make[2]: *** [install] Error 1 make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/voicemail' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' mkdir -p /usr/local/etc/sems/etc installing etc/ann_b2b.conf make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/ann_b2b' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/announcement' mkdir -p /usr/local/etc/sems/etc installing etc/announcement.conf make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/announcement' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/announce_transfer' mkdir -p /usr/local/etc/sems/etc installing etc/announce_transfer.conf make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/announce_transfer' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/conf_auth' mkdir -p /usr/local/etc/sems/etc installing etc/conf_auth.conf make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/conf_auth' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/conference' mkdir -p /usr/local/etc/sems/etc installing etc/conference.conf make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/conference' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/early_announce' mkdir -p /usr/local/etc/sems/etc installing etc/early_announce.conf make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/early_announce' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/ivr' mkdir -p /usr/local/etc/sems/etc installing etc/ivr.conf make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/ivr' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/mailbox' mkdir -p /usr/local/etc/sems/etc installing etc/mailbox.conf installing etc/mailbox_query.conf make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/mailbox' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/pin_collect' mkdir -p /usr/local/etc/sems/etc installing etc/pin_collect.conf make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/pin_collect' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/py_sems' mkdir -p /usr/local/etc/sems/etc installing etc/py_sems.conf make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/py_sems' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/registrar_client' mkdir -p /usr/local/etc/sems/etc make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/registrar_client' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/apps/voicemail' mkdir -p /usr/local/etc/sems/etc installing etc/voicemail.conf make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps/voicemail' make[1]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/apps' make[1]: Entering directory `/usr/local/src/sems-0.10.0-rc2/core' make[2]: Entering directory `/usr/local/src/sems-0.10.0-rc2/core' g++ -MM AmAudio.cpp -Wall -Wno-reorder -fPIC -g -O2 -D_DEBUG -D_THREAD_SAFE -D_REENTRANT -DVERSION='"0.10.0-rc2"' -DARCH='"i386"' -DOS='"linux"' > AmAudio.d /bin/sh: g++: command not found make[2]: *** [AmAudio.d] Error 127 make[2]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/core' make[1]: [all] Error 2 (ignored) sed -e "s#/usr/.*lib/sems/audio/#/usr/local/lib/sems/audio/#g" \ -e "s#/usr/.*lib/sems/plug-in/#/usr/local/lib/sems/plug-in/#g" \ -e "s#/usr/.*etc/sems#/usr/local/etc/sems/#g" \ < sems.conf.sample > /usr/local/etc/sems/sems.conf.default chmod 644 /usr/local/etc/sems/sems.conf.default if [ ! -f /usr/local/etc/sems/sems.conf ]; then \ cp -f /usr/local/etc/sems/sems.conf.default \ /usr/local/etc/sems/sems.conf; \ fi touch /usr/local/sbin/sems install -m 755 sems /usr/local/sbin/ install: cannot stat `sems': No such file or directory make[1]: *** [install-bin] Error 1 make[1]: Leaving directory `/usr/local/src/sems-0.10.0-rc2/core' Thanks. Roa Yu :) -----Original Message----- From: Stefan Sayer [mailto:stefan.sayer@iptego.de] Sent: Friday, November 23, 2007 6:32 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Can SER support 3-way Conference Hello, can you re-run 'make install' in sems' source directory, and post the output? Thanks Stefan roayu wrote: > Hi Ram! > > I've installed SER (sip stack) and SEMS. I managed to start both SER but I > failed to start SEMS. > > I got the following error message: > "/usr/local/sbin/sems : Permission denied" > > Then I found that the permission for /usr/local/sbin/sems is not 755 and the > file size of it is 0 (zero). > > Why am I having this error message? Can anyone there tell me why? > > Thanks. > > Roa Yu :) > > > -----Original Message----- > From: ram [mailto:talk2ram@gmail.com] > Sent: Wednesday, November 21, 2007 9:06 PM > To: roayu > Subject: Re: [Serusers] Can SER support 3-way Conference > > On Nov 20, 2007 8:39 AM, roayu <roayu(a)ctisys.net> wrote: >> >> >> Hi there, >> >> >> >> Can SER alone support 3-way Conference w/o integrate with other >> applications? >> >> >> > > Hi > > SER is proxy not a Media gateway > > Look at SEM or ASTERISK for that > > ram > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.de www.iptego.de iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

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[OpenSER-Users] A suggestion for 1.3 nathelper
by Robert Dyck 24 Nov '07

24 Nov '07

When I started using openser I used a patched version of 1.0 nathelper. This patch created two formats for fix_nated_contact. There was the original format -- fix_nated_contact() and the new format fix_nated_contact(1.2.3.4). The original format appears to be useful for service providers. The new format is useful when you are using openser as an ALG for a home or small office phone system. The contact gets mangled before being sent to the registrar. Because redoing the patch for each new version of openser that comes along is silly, I had previously asked the list for an alternative. Someone suggested using encode_contact from the mangler modules. This works but I think it is sub-optimal when you have multiple phones with the same user name. Each one creates its own contact with the registrar. The service providers server does the forking and there is a lot of excess messaging with the INVITEs and CANCELs when the call is answered by one of the phones. With fix_nated_contact(1.2.3.4) there is only one location registered and the local openser does the forking and the CANCELs are confined to the LAN. Comments? Rob

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[OpenSER-Users] In 1.3 how to use set_rtp_proxy_set
by Robert Dyck 23 Nov '07

23 Nov '07

If I have only one rtp proxy should I use set_rtp_proxy_set("0")? I think openser should be smart enough to figure it out if there is only one rtp proxy. It looks like there is a cut and paste error in the 1.3 nathelper wiki. 1.6.3. set_rtp_proxy_set() Sets the Id of the rtpproxy set to be used for the next [un]force_rtp_proxy() command. This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, BRANCH_ROUTE. Example 1-17. fix_nated_contact usage ... set_rtp_proxy_set("2"); force_rtp_proxy(); Rob ...

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Re: [Serusers] [Serdev] [Tracker] Created: (SER-332) tcp branching error
by Tomasz Zieleniewski 23 Nov '07

23 Nov '07

once again because to big attachments:) On Nov 23, 2007 11:48 AM, Tomasz Zieleniewski <tzieleniewski(a)gmail.com> wrote: > sorry I forgot to place the attachments > > > On Nov 23, 2007 11:47 AM, Tomasz Zieleniewski <tzieleniewski(a)gmail.com> wrote: > > Hi, > > > > Some more debug information: > > > > User contacts: > > ser@sen:~/ser/ser-2.0/ssp$ ./serctl/ser_ctl usrloc show touk(a)voip.rd.touk.pl > > sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP 3330 -1.0 > > sip:touk@192.168.0.164:1348;rinstance=e51fd20db73fc732;transport=TCP 3405 -1.0 > > > > Contents of the location table: > > mysql> select * from location where uid like 'touk@%'; > > +----------------------+--------------------------+----------------------------------------------------------------------+----------+---------------------+----+----------------------------------------------+------+-------+----------------------------------+----------+ > > | uid | aor | contact > > | received | expires > > | q | callid | cseq > > | flags | user_agent | instance | > > +----------------------+--------------------------+----------------------------------------------------------------------+----------+---------------------+----+----------------------------------------------+------+-------+----------------------------------+----------+ > > | touk(a)voip.rd.touk.pl | sip:touk@voip.rd.touk.pl | > > sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP | > > NULL | 2007-11-23 11:17:03 | -1 | > > MTQ1ZWZmZWE2MWIxZjViYmVkNzZlNmU0YTEwNjRmMTA. | 4 | 0 | X-Lite > > release 1011s stamp 41150 | NULL | > > | touk(a)voip.rd.touk.pl | sip:touk@voip.rd.touk.pl | > > sip:touk@192.168.0.164:1348;rinstance=e51fd20db73fc732;transport=TCP | > > NULL | 2007-11-23 11:18:18 | -1 | > > NDk0ZDA3ZmU5NmFlZmYyMjA5Mjc2ZTZiYTFmMTMxODY. | 4 | 0 | X-Lite > > release 1011s stamp 41150 | NULL | > > +----------------------+--------------------------+----------------------------------------------------------------------+----------+---------------------+----+----------------------------------------------+------+-------+----------------------------------+----------+ > > 2 rows in set (0.00 sec) > > > > > > Here is what does the internal proxy (192.168.0.165:7160) which is > > registrar and perform location service, > > the destination set contains both contacts and the first contact used > > is the one which is unavailable. > > this proxy sends first invite with this contact to boundary sip proxy > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: xlset_attr(bsp_route, > > <$global_addr_bsp;lr=on>) > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: attr_destination($bsp_route) > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: ($fr_inv_timer) > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: t_on_reply(ONREPLY_ROUTE) > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: t_on_failure(FAILURE_ROUTE) > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: t_on_branch(BRANCH_ROUTE) > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: destination set: > > Contact: sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP, > > sip:touk@192.168.0.164:1348;rinstance=e51fd20db73fc732;transpo > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: next route: <null> > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: message's next hop: > > sip:voip.rd.touk.pl:7060;lr=on > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: (!t_relay()) > > Nov 23 11:20:27 sen ser[28434]: branch_route[BRANCH_ROUTE]: <INVITE> > > ruri:<sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP> > > t:<sip:touk@voip.rd.touk.pl> f:<sip:hellboy@voip.rd.touk.pl> c > > Nov 23 11:20:27 sen ser[28434]: branch_route[BRANCH_ROUTE]: > > append_hf(P-Warn: forked branch) > > Nov 23 11:20:27 sen ser[28434]: branch_route[BRANCH_ROUTE]: <INVITE> > > ruri:<sip:touk@192.168.0.164:1348;rinstance=e51fd20db73fc732;transport=TCP> > > t:<sip:touk@voip.rd.touk.pl> f:<sip:hellboy@voip.rd.touk.pl> c > > Nov 23 11:20:27 sen ser[28434]: branch_route[BRANCH_ROUTE]: > > append_hf(P-Warn: forked branch) > > Nov 23 11:20:27 sen ser[28434]: ERROR: tcp_blocking_connect: poll > > error: flags 18 > > Nov 23 11:20:27 sen ser[28434]: ERROR: tcp_blocking_connect: SO_ERROR > > (111) Connection refused > > Nov 23 11:20:27 sen ser[28434]: ERROR: tcpconn_connect: > > tcp_blocking_connect failed > > Nov 23 11:20:27 sen ser[28434]: ERROR: tcp_send: connect failed > > Nov 23 11:20:27 sen ser[28434]: msg_send: ERROR: tcp_send failed > > Nov 23 11:20:27 sen ser[28434]: ERROR: t_send_branch: sending request > > on branch 1 failed > > Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: drop > > > > here is what happens in boundary sip proxy: > > is receives the invite tries to do the t_rqlay but there is the tcp error. > > t_relay() fails and it sends the sl_reply_error to internal sip proxy: > > Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: destination set: <null> > > Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: next route: <null> > > Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: message's next hop: > > sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP > > Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: (!t_relay()) > > Nov 23 11:31:38 sen ser[31861]: ERROR: tcp_blocking_connect: poll > > error: flags 18 > > Nov 23 11:31:38 sen ser[31861]: ERROR: tcp_blocking_connect: SO_ERROR > > (111) Connection refused > > Nov 23 11:31:38 sen ser[31861]: ERROR: tcpconn_connect: > > tcp_blocking_connect failed > > Nov 23 11:31:38 sen ser[31861]: ERROR: tcp_send: connect failed > > Nov 23 11:31:38 sen ser[31861]: msg_send: ERROR: tcp_send failed > > Nov 23 11:31:38 sen ser[31861]: ERROR: t_send_branch: sending request > > on branch 0 failed > > Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: (!t_relay()) -> TRUE > > Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: sl_reply_error() > > Nov 23 11:31:38 sen ser[31861]: ERROR: sl_reply_error used: I'm > > terribly sorry, server error occurred (1/SL) > > > > in the attachment there are ngrep and wireshark log from the same situation > > > > if you need more info please let me know > > waiting for your feedback > > > > Cheers > > tomasz > > > > > > > > On Nov 22, 2007 8:40 PM, Andrei Pelinescu-Onciul <andrei(a)iptel.org> wrote: > > > On Nov 16, 2007 at 14:43, Tomasz Zieleniewski (JIRA) <tracker(a)iptel.org> wrote: > > > > tcp branching error > > > > ------------------- > > > > > > > > Key: SER-332 > > > > URL: http://tracker.iptel.org/browse/SER-332 > > > > Project: SER > > > > Issue Type: Bug > > > > Components: core > > > > Affects Versions: 2.0 > > > > Environment: Linux sen 2.6.18-5-amd64 #1 SMP Tue Oct 2 20:37:02 UTC 2007 x86_64 GNU/Linux > > > > Reporter: Tomasz Zieleniewski > > > > Priority: Critical > > > > > > > > > > > > Hi, > > > > > > > > I have the following situation. > > > > I have two ser rel_2_0_0 instances one acting as a boundary proxy the > > > > other on as internal proxy (registrar, location). > > > > > > > > I have the following situation I have a user which has registered two > > > > contacs one woth tcp transport the other one with udp. > > > > > > > > All messages coming to and outgoing from my network travels through > > > > boundary sip proxy. > > > > > > > > When internal ser perform lookup_location it get to contacts and the > > > > destination set contains two records: > > > > Nov 15 11:03:13 sen ser[3673]: route[FORWARD]: destination set: > > > > Contact: sip:xlite@192.168.0.164:1221;rinstance=d35f9cda6bd79b0a;transport=TCP, > > > > sip:xlite@192.168.0.164:7060 > > > > > > > > Then internal ser forward message to boundary SIP proxy by changing > > > > the destination of the message and with the usage of t_relay(). > > > > I see from log that there are two branches involved: > > > > Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: <INVITE> > > > > ruri:<sip:xlite@192.168.0.164:1221;rinstance=d35f9cda6bd79b0a;transport=TCP> > > > > t:<sip:00@voip.rd.touk.pl> f:<sip:lkl@voip.rd.touk.pl> > > > > Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: > > > > append_hf(P-Warn: forked branch) > > > > Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: <INVITE> > > > > ruri:<sip:xlite@192.168.0.164:7060;rinstance=cfca8523e3747cb0> > > > > t:<sip:00@voip.rd.touk.pl> f:<sip:lkl@voip.rd.touk.pl> > > > > Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: > > > > append_hf(P-Warn: forked branch) > > > > > > > > both INVITES reaches boudary SIP proxy. > > > > > > > > Because first contact was unreachable there was an tcp error: > > > > ser[3516]: ERROR: tcp_blocking_connect: poll error: flags 18 > > > > Nov 15 11:03:13 sen ser[3516]: ERROR: tcp_blocking_connect: SO_ERROR > > > > (111) Connection refused > > > > Nov 15 11:03:13 sen ser[3516]: ERROR: tcpconn_connect: > > > > tcp_blocking_connect failed > > > > Nov 15 11:03:13 sen ser[3516]: ERROR: tcp_send: connect failed > > > > Nov 15 11:03:13 sen ser[3516]: msg_send: ERROR: tcp_send failed > > > > Nov 15 11:03:13 sen ser[3516]: ERROR: t_send_branch: sending request > > > > on branch 0 failed > > > > > > > > after which ser invokes sl_reply_error() which results in 477 next hop > > > > error reponse message sent to internal ser instance. > > > > The problem arises when boudary sip proxy processes second INVITE > > > > which correspond to second destination. > > > > when boundary SIP proxy receives it doesn't send it to user agent but > > > > instead reponses to internal ser with > > > > 500 I'm terribly sorry, server error occurred (1/SL) > > > > the only thing I see in the boundary SIP proxy log is: > > > > ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) > > > > > > > > Did I miss something here or this is an error? > > > > Waiting for Your feedback > > > > > > > > > > Could you make some traffic dump between the 2 proxies and send it to me > > > (andrei(a)iptel.org)? > > > It's sounds quite strange (it should work without any problems) and a > > > look at the packets might help. > > > > > > > > > Thanks, > > > Andrei > > > _______________________________________________ > > > Serdev mailing list > > > Serdev(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serdev > > > > > >

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Re: [Serusers] [Serdev] [Tracker] Created: (SER-332) tcp branching error
by Tomasz Zieleniewski 23 Nov '07

23 Nov '07

Hi, Some more debug information: User contacts: ser@sen:~/ser/ser-2.0/ssp$ ./serctl/ser_ctl usrloc show touk(a)voip.rd.touk.pl sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP 3330 -1.0 sip:touk@192.168.0.164:1348;rinstance=e51fd20db73fc732;transport=TCP 3405 -1.0 Contents of the location table: mysql> select * from location where uid like 'touk@%'; +----------------------+--------------------------+----------------------------------------------------------------------+----------+---------------------+----+----------------------------------------------+------+-------+----------------------------------+----------+ | uid | aor | contact | received | expires | q | callid | cseq | flags | user_agent | instance | +----------------------+--------------------------+----------------------------------------------------------------------+----------+---------------------+----+----------------------------------------------+------+-------+----------------------------------+----------+ | touk(a)voip.rd.touk.pl | sip:touk@voip.rd.touk.pl | sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP | NULL | 2007-11-23 11:17:03 | -1 | MTQ1ZWZmZWE2MWIxZjViYmVkNzZlNmU0YTEwNjRmMTA. | 4 | 0 | X-Lite release 1011s stamp 41150 | NULL | | touk(a)voip.rd.touk.pl | sip:touk@voip.rd.touk.pl | sip:touk@192.168.0.164:1348;rinstance=e51fd20db73fc732;transport=TCP | NULL | 2007-11-23 11:18:18 | -1 | NDk0ZDA3ZmU5NmFlZmYyMjA5Mjc2ZTZiYTFmMTMxODY. | 4 | 0 | X-Lite release 1011s stamp 41150 | NULL | +----------------------+--------------------------+----------------------------------------------------------------------+----------+---------------------+----+----------------------------------------------+------+-------+----------------------------------+----------+ 2 rows in set (0.00 sec) Here is what does the internal proxy (192.168.0.165:7160) which is registrar and perform location service, the destination set contains both contacts and the first contact used is the one which is unavailable. this proxy sends first invite with this contact to boundary sip proxy Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: xlset_attr(bsp_route, <$global_addr_bsp;lr=on>) Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: attr_destination($bsp_route) Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: ($fr_inv_timer) Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: t_on_reply(ONREPLY_ROUTE) Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: t_on_failure(FAILURE_ROUTE) Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: t_on_branch(BRANCH_ROUTE) Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: destination set: Contact: sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP, sip:touk@192.168.0.164:1348;rinstance=e51fd20db73fc732;transpo Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: next route: <null> Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: message's next hop: sip:voip.rd.touk.pl:7060;lr=on Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: (!t_relay()) Nov 23 11:20:27 sen ser[28434]: branch_route[BRANCH_ROUTE]: <INVITE> ruri:<sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP> t:<sip:touk@voip.rd.touk.pl> f:<sip:hellboy@voip.rd.touk.pl> c Nov 23 11:20:27 sen ser[28434]: branch_route[BRANCH_ROUTE]: append_hf(P-Warn: forked branch) Nov 23 11:20:27 sen ser[28434]: branch_route[BRANCH_ROUTE]: <INVITE> ruri:<sip:touk@192.168.0.164:1348;rinstance=e51fd20db73fc732;transport=TCP> t:<sip:touk@voip.rd.touk.pl> f:<sip:hellboy@voip.rd.touk.pl> c Nov 23 11:20:27 sen ser[28434]: branch_route[BRANCH_ROUTE]: append_hf(P-Warn: forked branch) Nov 23 11:20:27 sen ser[28434]: ERROR: tcp_blocking_connect: poll error: flags 18 Nov 23 11:20:27 sen ser[28434]: ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused Nov 23 11:20:27 sen ser[28434]: ERROR: tcpconn_connect: tcp_blocking_connect failed Nov 23 11:20:27 sen ser[28434]: ERROR: tcp_send: connect failed Nov 23 11:20:27 sen ser[28434]: msg_send: ERROR: tcp_send failed Nov 23 11:20:27 sen ser[28434]: ERROR: t_send_branch: sending request on branch 1 failed Nov 23 11:20:27 sen ser[28434]: route[FORWARD]: drop here is what happens in boundary sip proxy: is receives the invite tries to do the t_rqlay but there is the tcp error. t_relay() fails and it sends the sl_reply_error to internal sip proxy: Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: destination set: <null> Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: next route: <null> Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: message's next hop: sip:touk@192.168.0.164:1338;rinstance=191d670c84591ea3;transport=TCP Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: (!t_relay()) Nov 23 11:31:38 sen ser[31861]: ERROR: tcp_blocking_connect: poll error: flags 18 Nov 23 11:31:38 sen ser[31861]: ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused Nov 23 11:31:38 sen ser[31861]: ERROR: tcpconn_connect: tcp_blocking_connect failed Nov 23 11:31:38 sen ser[31861]: ERROR: tcp_send: connect failed Nov 23 11:31:38 sen ser[31861]: msg_send: ERROR: tcp_send failed Nov 23 11:31:38 sen ser[31861]: ERROR: t_send_branch: sending request on branch 0 failed Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: (!t_relay()) -> TRUE Nov 23 11:31:38 sen ser[31861]: route[FORWARD]: sl_reply_error() Nov 23 11:31:38 sen ser[31861]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) in the attachment there are ngrep and wireshark log from the same situation if you need more info please let me know waiting for your feedback Cheers tomasz On Nov 22, 2007 8:40 PM, Andrei Pelinescu-Onciul <andrei(a)iptel.org> wrote: > On Nov 16, 2007 at 14:43, Tomasz Zieleniewski (JIRA) <tracker(a)iptel.org> wrote: > > tcp branching error > > ------------------- > > > > Key: SER-332 > > URL: http://tracker.iptel.org/browse/SER-332 > > Project: SER > > Issue Type: Bug > > Components: core > > Affects Versions: 2.0 > > Environment: Linux sen 2.6.18-5-amd64 #1 SMP Tue Oct 2 20:37:02 UTC 2007 x86_64 GNU/Linux > > Reporter: Tomasz Zieleniewski > > Priority: Critical > > > > > > Hi, > > > > I have the following situation. > > I have two ser rel_2_0_0 instances one acting as a boundary proxy the > > other on as internal proxy (registrar, location). > > > > I have the following situation I have a user which has registered two > > contacs one woth tcp transport the other one with udp. > > > > All messages coming to and outgoing from my network travels through > > boundary sip proxy. > > > > When internal ser perform lookup_location it get to contacts and the > > destination set contains two records: > > Nov 15 11:03:13 sen ser[3673]: route[FORWARD]: destination set: > > Contact: sip:xlite@192.168.0.164:1221;rinstance=d35f9cda6bd79b0a;transport=TCP, > > sip:xlite@192.168.0.164:7060 > > > > Then internal ser forward message to boundary SIP proxy by changing > > the destination of the message and with the usage of t_relay(). > > I see from log that there are two branches involved: > > Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: <INVITE> > > ruri:<sip:xlite@192.168.0.164:1221;rinstance=d35f9cda6bd79b0a;transport=TCP> > > t:<sip:00@voip.rd.touk.pl> f:<sip:lkl@voip.rd.touk.pl> > > Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: > > append_hf(P-Warn: forked branch) > > Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: <INVITE> > > ruri:<sip:xlite@192.168.0.164:7060;rinstance=cfca8523e3747cb0> > > t:<sip:00@voip.rd.touk.pl> f:<sip:lkl@voip.rd.touk.pl> > > Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: > > append_hf(P-Warn: forked branch) > > > > both INVITES reaches boudary SIP proxy. > > > > Because first contact was unreachable there was an tcp error: > > ser[3516]: ERROR: tcp_blocking_connect: poll error: flags 18 > > Nov 15 11:03:13 sen ser[3516]: ERROR: tcp_blocking_connect: SO_ERROR > > (111) Connection refused > > Nov 15 11:03:13 sen ser[3516]: ERROR: tcpconn_connect: > > tcp_blocking_connect failed > > Nov 15 11:03:13 sen ser[3516]: ERROR: tcp_send: connect failed > > Nov 15 11:03:13 sen ser[3516]: msg_send: ERROR: tcp_send failed > > Nov 15 11:03:13 sen ser[3516]: ERROR: t_send_branch: sending request > > on branch 0 failed > > > > after which ser invokes sl_reply_error() which results in 477 next hop > > error reponse message sent to internal ser instance. > > The problem arises when boudary sip proxy processes second INVITE > > which correspond to second destination. > > when boundary SIP proxy receives it doesn't send it to user agent but > > instead reponses to internal ser with > > 500 I'm terribly sorry, server error occurred (1/SL) > > the only thing I see in the boundary SIP proxy log is: > > ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) > > > > Did I miss something here or this is an error? > > Waiting for Your feedback > > > > Could you make some traffic dump between the 2 proxies and send it to me > (andrei(a)iptel.org)? > It's sounds quite strange (it should work without any problems) and a > look at the packets might help. > > > Thanks, > Andrei > _______________________________________________ > Serdev mailing list > Serdev(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serdev >

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Re: [Serusers] Can SER support 3-way Conference
by roayu 23 Nov '07

23 Nov '07

Hi Ram! I've installed SER (sip stack) and SEMS. I managed to start both SER but I failed to start SEMS. I got the following error message: "/usr/local/sbin/sems : Permission denied" Then I found that the permission for /usr/local/sbin/sems is not 755 and the file size of it is 0 (zero). Why am I having this error message? Can anyone there tell me why? Thanks. Roa Yu :) -----Original Message----- From: ram [mailto:talk2ram@gmail.com] Sent: Wednesday, November 21, 2007 9:06 PM To: roayu Subject: Re: [Serusers] Can SER support 3-way Conference On Nov 20, 2007 8:39 AM, roayu <roayu(a)ctisys.net> wrote: > > > > Hi there, > > > > Can SER alone support 3-way Conference w/o integrate with other > applications? > > > Hi SER is proxy not a Media gateway Look at SEM or ASTERISK for that ram

2 1

[OpenSER-Users] CPL serial forking behaviour
by Daniel 23 Nov '07

23 Nov '07

Hi all, I'd like to understand CPL the behaviour or serial forking made by CPL. This is my CPL script (for USER A): <incoming> <lookup source="registration"> <success> <proxy ordering="sequential" /> </success> </lookup> </incoming> </cpl> The first question is : In this case, if I have 2 contacts for USER A, what is its behaviour ? I have made a couple of tests about it. USER A has 2 contacts : USERA@IP_1 and USERA@IP_2 and the CPL script as above. USER B has 1 contact : USERB@IP_3 The location table is: ID username contact q 318 USERB USERB@IP_3 1 319 USERA USERA@IP_1 1 320 USERA USERA@IP_2 1 In this case the CPL script calls both of Contacts of USER A, at the same time. Then, I've changed the two contact's q parameters: ID username contact q 318 USERB USERB@IP_3 1 319 USERA USERA@IP_1 1 320 USERA USERA@IP_2 0.1 In this case CPL script calls first USERA@IP_2 (low q paramenter) and then USERA@IP_1 (high q-value) ID username contact q 318 USERB USERB@IP_3 1 319 USERA USERA@IP_1 0.1 320 USERA USERA@IP_2 1 In this case CPL script calls first USERA@IP_1 (low q paramenter) and then USERA@IP_2 (high q-value) ID username contact q 318 USERB USERB@IP_3 1 319 USERA USERA@IP_1 0.1 320 USERA USERA@IP_2 0.5 In this case CPL script calls first USERA@IP_1 (low q paramenter) and then USERA@IP_2 (high q-value) ID username contact q 318 USERB USERB@IP_3 1 319 USERA USERA@IP_1 0.1 320 USERA USERA@IP_2 0.1 In this case the CPL script calls both of Contacts of USER A, at the same time. It seem work in the reverse mode: it call first the lowest q value and then the highest. So, does PROXY element (with ordering="sequential") care about q-parameter ? Can somebody help me? Regards, Daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

1 1

[OpenSER-Users] Help with unixODBC + sqlite driver
by kionez 22 Nov '07

22 Nov '07

Hi, i found a message on openser-user mailing list: http://openser.org/pipermail/users/2007-April/010628.html I'm trying to get working my openser with unixodbc+sqlite3, but i don't know why it fails with same errors.. i think is unixodbc openser's module, because with "isql -v mysqlitedb" i can access to my sqlite db... someone can help me? :) thanx Fulvio

2 4

[OpenSER-Users] RINSTANCE
by Thiago Maluf 22 Nov '07

22 Nov '07

Somebody know what is RINSTANCE in SIP Contact? ---------------------------------------------------------------- THIAGO MALUF RESENDE e-mail: malufrj(a)gmail.com

2 1

[OpenSER-Users] Edit SIP Header, Contact field
by Thiago Maluf 22 Nov '07

22 Nov '07

I Want edit the Contact field. How I do it? Thanks in advanced, Thiago -- ---------------------------------------------------------------- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: malufrj(a)gmail.com

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[OpenSER-Users] [CPL] ERROR:core:set_dst_uri: not enough pkg memory
by Iñaki Baz Castillo 22 Nov '07

22 Nov '07

I have a simple CPL for a user "200" that just should lookup for the contact in the "location" table: <cpl> <incoming> <lookup source="registration"> <success> <proxy /> </success> </lookup> </incoming> </cpl> In the script I do: cpl_run_script("incoming","FORCE_STATEFUL"); and I get an error: ERROR:core:set_dst_uri: not enough pkg memory ERROR:cpl_c:cpl_proxy_to_loc_set: do_action failed ERROR:cpl_c:cpl_run_script: runtime error and the script continues but with the URI changed. It's strange: it's obvious that CPL does a DB query to "location" table asking for 200@domain (I've monitorized it in mysql logs) but after that it shows the errors above and the OpenSer script continues (with the URI set to the contact URI). Why can it occur? maybe it's a bug? Thanks. -- Iñaki Baz Castillo

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Re: [OpenSER-Users] openser SBC and upper registration.
by Ovidiu Sas 21 Nov '07

21 Nov '07

This issue was already discussed. Check the following link: http://lists.openser.org/pipermail/users/2007-May/010857.html Regards, Ovidiu Sas On Nov 16, 2007 6:53 PM, rohail khan <rohail_12(a)yahoo.com> wrote: > Well save() does not work on reply route, its for request route , what I did > is from reply route i used t_check_status("200") and route it to another > route block let say route[6] and I used save_noreply() to save the contact > and it saving now but the problem is that like SBC I was modifying > registration request from UA and changing the CONTACT before relaying to my > other server but when I am saving the contact info its saving with openser > contact info no information for UA so call can still can not route to UA. > any idea > > Thanks > > > Ovidiu Sas <osas(a)voipembedded.com> wrote: > You can call save() in the reply_route. You can check there is the > response is a 200ok or not. > In this particular case you will lose some informations that are used > to handle NATed UA. > Test and see if it will fit your needs. > > > Regards, > Ovidiu sas > > On Nov 15, 2007 5:53 AM, rohail khan wrote: > > Hi > > I am trying to configure openser for uppser registration, so far I am > > relaying the registration after modifying contact in the registration of > > openser IP address but I want to save the contact only if the registration > > is sucessfull from upper registration server. If I use "save_noreply" its > > save the location no matter if the UA is registered with upper > registration > > server > > > > any idea > > thanks in advance > > > > > > > > ________________________________ > > Never miss a thing. Make Yahoo your homepage. > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > ________________________________ > Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try it > now.

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Re: [Serusers] Implicit Registration Set
by Jiri Kuthan 21 Nov '07

21 Nov '07

p.s. which may be good enough. we know from interop events that having multiple contacts with only one actual registration has actually caused some interop issues. anyhow -- it depends on what your use case is, some of these things -- should they pay off -- are scriptable. At 23:01 21/11/2007, Jiri Kuthan wrote: >It will return one contact. -jiri > >At 07:44 21/11/2007, Magnus Sörman wrote: >>OK, but using alias, will that give me back both contacts in the 200 OK for a Register (unfortunately I don't have the possibility to try this right now)? >> >> /Magnus >> >>-----Original Message----- >>From: Jiri Kuthan [mailto:jiri@iptel.org] >>Sent: den 15 november 2007 15:00 >>To: Magnus Sörman; serusers(a)lists.iptel.org >>Subject: Re: [Serusers] Implicit Registration Set >> >>well it depends -- doing what you are suggesting literally is quite tricky and I'm not sure this IMS feature is worth the effort. IF you would like to register and be then available under all of your aliases (both "that address" >>and "his tel-uri"), you can easily use aliases. >> >>-jiri >> >>At 14:52 15/11/2007, Magnus Sörman wrote: >>>Content-class: urn:content-classes:message >>>Content-Type: multipart/alternative; >>> boundary="----_=_NextPart_001_01C8278E.D8F42EB4" >>> >>>Hi, >>> >>>How do you/could you configure SER to use implicit registration set? >>>If e.g. a user with a sip URI registers, then both that address and his tel-uri should be registered. >>> >>>Thanks, >>> /Magnus Sorman >>>_______________________________________________ >>>Serusers mailing list >>>Serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >>-- >>Jiri Kuthan http://iptel.org/~jiri/ > > > >-- >Jiri Kuthan http://iptel.org/~jiri/ -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] Implicit Registration Set
by Jiri Kuthan 21 Nov '07

21 Nov '07

It will return one contact. -jiri At 07:44 21/11/2007, Magnus Sörman wrote: >OK, but using alias, will that give me back both contacts in the 200 OK for a Register (unfortunately I don't have the possibility to try this right now)? > > /Magnus > >-----Original Message----- >From: Jiri Kuthan [mailto:jiri@iptel.org] >Sent: den 15 november 2007 15:00 >To: Magnus Sörman; serusers(a)lists.iptel.org >Subject: Re: [Serusers] Implicit Registration Set > >well it depends -- doing what you are suggesting literally is quite tricky and I'm not sure this IMS feature is worth the effort. IF you would like to register and be then available under all of your aliases (both "that address" >and "his tel-uri"), you can easily use aliases. > >-jiri > >At 14:52 15/11/2007, Magnus Sörman wrote: >>Content-class: urn:content-classes:message >>Content-Type: multipart/alternative; >> boundary="----_=_NextPart_001_01C8278E.D8F42EB4" >> >>Hi, >> >>How do you/could you configure SER to use implicit registration set? >>If e.g. a user with a sip URI registers, then both that address and his tel-uri should be registered. >> >>Thanks, >> /Magnus Sorman >>_______________________________________________ >>Serusers mailing list >>Serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers > > > >-- >Jiri Kuthan http://iptel.org/~jiri/ -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] Linux version -information request-
by Jiri Kuthan 21 Nov '07

21 Nov '07

Almost any unless you have a real historical one -- debian, RH, gentoo, etc, ideally with 2.6 kernel. -jiri At 21:04 21/11/2007, Mary Allen (mrallen) wrote: >Content-class: urn:content-classes:message >Content-Type: multipart/alternative; > boundary="----_=_NextPart_001_01C82C79.91616C64" > >Hi, > >I need to install the SERi proxy. Would you please let me know what version of Linux should I put in my system before the installation of SERi? > >Thanks, >Mary >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] Linux version -information request-
by Mary Allen (mrallen) 21 Nov '07

21 Nov '07

Hi, I need to install the SERi proxy. Would you please let me know what version of Linux should I put in my system before the installation of SERi? Thanks, Mary

1 0

[OpenSER-Users] [OT] Looking for a good SIP PBX
by Iñaki Baz Castillo 21 Nov '07

21 Nov '07

Hi, first of all I'm sorry for this OffTopic. I'm looking for a good SIP PBX since Asterisk SIP implementation doesn't let me using it in a multidomain SIP enviroment properly with OpenSer ("From" domain not respected when generates a SIP channel, no good behaviour in parallel forking, etc). I can use it just as media server for now. So I'm looking for other SIP PBX server. One I've in mind is, of course, WeSIP, but I'd like to know if any of you has good experiences integrating other PBX's as FreeSwitch or SipX/SipFoundry. If possible, it would be really great a SIP PBX with virtual domains support, so the PBX logic for independent SIP domains would exist in the same server/PBX (less and easy maintenance). The aim of this PBX is to allow SIP incoming calls and during the execution of the dialplan maybe call to that domain users (registered in a multidomain main OpenSer). Please, any opinion or experience about it with some of those PBX's? I'm going now to install and try of all them, but some help would be really appreciated. Thanks a lot. Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [Serusers] Implicit Registration Set
by Jiri Kuthan 21 Nov '07

21 Nov '07

well it depends -- doing what you are suggesting literally is quite tricky and I'm not sure this IMS feature is worth the effort. IF you would like to register and be then available under all of your aliases (both "that address" and "his tel-uri"), you can easily use aliases. -jiri At 14:52 15/11/2007, Magnus Sörman wrote: >Content-class: urn:content-classes:message >Content-Type: multipart/alternative; > boundary="----_=_NextPart_001_01C8278E.D8F42EB4" > >Hi, > >How do you/could you configure SER to use implicit registration set? >If e.g. a user with a sip URI registers, then both that address and his tel-uri should be registered. > >Thanks, > /Magnus Sorman >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

2 1

[Serusers] November 79% OFF
by Canadian Doctor Damon Herrera 21 Nov '07

21 Nov '07

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Best Price for VIAGRA & CIALIS ONLINE, All Credit Cards Accepted, FREE DELIVERY http://www.foodenter.com [...] Content analysis details: (18.1 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.0 UNPARSEABLE_RELAY Informational: message has unparseable relay lines 2.7 DRUG_ED_ONLINE BODY: Fast Viagra Delivery 0.5 DRUG_ED_CAPS BODY: Mentions an E.D. drug 3.5 BAYES_99 BODY: Bayesian spam probability is 99 to 100% [score: 0.9999] 1.6 URIBL_SBL Contains an URL listed in the SBL blocklist [URIs: foodenter.com] 4.1 URIBL_JP_SURBL Contains an URL listed in the JP SURBL blocklist [URIs: foodenter.com] 2.1 URIBL_WS_SURBL Contains an URL listed in the WS SURBL blocklist [URIs: foodenter.com] 3.0 URIBL_OB_SURBL Contains an URL listed in the OB SURBL blocklist [URIs: foodenter.com] 0.5 DRUGS_ERECTILE Refers to an erectile drug

1 0

[Serusers] Another alternative for handling RTP packets
by Carlos Loarca 20 Nov '07

20 Nov '07

Hi everyone: I have been working with NATHELPER or MEDIAPROXY to handle audio and video calls. However I have got into the point that there should be another way to work Firewalls and NAT transversal rather than mediaproxies and rtp-proxies. I have heard about STUN servers with ICE support but never seen it working and I wonder if this works. And I dont know any server like this, and I know that the softphones or IPphones need to support it. If I dont want to force the audio or video to come to my server, what kind of solution is the most recommendable? Best regards, /Carlos ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

2 1

Re: [OpenSER-Users] CPL serial call attempt
by Iñaki Baz Castillo 20 Nov '07

20 Nov '07

El Martes, 20 de Noviembre de 2007, Daniel Grotti escribió: > Hi Inaki, > Thank you for your help. > I've a question, does Serial Forking (with LCR) work well with NAT ? Sure, there is not difference. Just force RtpProxy (or MediaProxy) in each branch (when neccesary). -- Iñaki Baz Castillo

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[OpenSER-Users] Openser 1.3 trunk: unix_sock
by Helmut Kuper 20 Nov '07

20 Nov '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, after upgrading from openser 1.2 to 1.3 trunk the core commands unix_sock, sock_mode, sock_user, sock_group are missing. Does anybody know what happend to this commands ? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHQuo/4tZeNddg3dwRAkx9AKCPRZKGfyOxbox73Asf+aaIgAWLnACfc/Uu IepPGsTGfNwJt2p1frTRSwM= =VFPA -----END PGP SIGNATURE-----

2 3

[OpenSER-Users] Natping & register
by Stanisław Pitucha 20 Nov '07

20 Nov '07

Hey Is it possible to do with openser/nathelper what some other sip/nat gateways do? - for initial register, keep location, but pass registration packets upstream -> save("location", 0x02) - when there's a nat flag set, rewrite response to client with very low "Expire:" header - accept frequent register packets from client, but register on real server, when original expire time is reached It's basically natping done the other way - should be more stable, because when one natping doesn't reach the client, it's lost till next register / invite from that phone. With registers, phone tries to sent packets itself, so when one is lost, next one will reactivate nat anyway. Is there any module that does it now? Or does it need some avp & database hacking?

1 0

[OpenSER-Users] cpl-c module problems
by Aki Rehn 20 Nov '07

20 Nov '07

Hi, Im having some cpl-c module problems.. Both are related to time-switch. What it seems to me like is that bymonth parameter is parsed somehow wrong which then causes overflow. I added bit more debug and attr_name and attr_len seem to be completely off-the-track. (The actual line number of the overflow in cpl_switches.h is 763 because of the one line added.) Input: <time dtstart="20071101T090000" dtend="20171101T090000" byhour="7,8,9,10,11,12,13,14,15,16,17,18,19,20"> Output: openser[5160]: DEBUG:cpl_run_script: processing CPL node openser[5160]: DEBUG:cpl_run_script: processing time-switch node openser[5160]: DEBUG:cpl-c:run_time_switch: checking recv. time stamp <1195111240> openser[5160]: DEBUG:cpl-c:set_TZ: switching TZ as "TZ=Europe/Helsinki" openser[5160]: ---> fill = Thu Nov 15 09:20:40 2007 openser[5160]: DEBUG:cpl_c:testing, attr_name: [0], attr_len: [16] openser[5160]: DEBUG:cpl_c:run_time_node: attribute [0] found :[20071101T090000] openser[5160]: DEBUG:cpl_c:testing, attr_name: [1], attr_len: [16] openser[5160]: DEBUG:cpl_c:run_time_node: attribute [1] found :[20171101T090000] openser[5160]: DEBUG:cpl_c:testing, attr_name: [19], attr_len: [12592] openser[5160]: ERROR:cpl_c: overflow detected ip=0xb5a3c8ec ptr=0xb5a3fa6e in func. cpl_switches.h, line 764 openser[5160]: DEBUG:cpl-c:set_TZ: switching TZ as "TZ=" openser[5160]: ERROR:cpl_c:cpl_run_script: script error Second problem is the byhour attribute, which doesnt seem to be supported at all. Input: <time dtstart="20071101T090000" dtend="20171101T090000" byhour="7,8,9,10,11,12,13,14,15,16,17,18,19,20"> Output: openser[5162]: DEBUG:cpl_run_script: processing CPL node openser[5162]: DEBUG:cpl_run_script: processing time-switch node openser[5162]: DEBUG:cpl-c:run_time_switch: checking recv. time stamp <1195111623> openser[5162]: DEBUG:cpl-c:set_TZ: switching TZ as "TZ=Europe/Helsinki" openser[5162]: ---> fill = Thu Nov 15 09:27:03 2007 openser[5162]: DEBUG:cpl_c:testing, attr_name: [0], attr_len: [16] openser[5162]: DEBUG:cpl_c:run_time_node: attribute [0] found :[20071101T090000] openser[5162]: DEBUG:cpl_c:testing, attr_name: [1], attr_len: [16] openser[5162]: DEBUG:cpl_c:run_time_node: attribute [1] found :[20171101T090000] openser[5162]: DEBUG:cpl_c:testing, attr_name: [9], attr_len: [39] openser[5162]: DEBUG:cpl_c:run_time_node: attribute [9] found :[7,8,9,10,11,12,13,14,15,16,17,18,19,20] openser[5162]: ERROR:cpl_c:run_time_switch: unsupported attribute [9] found in TIME node openser[5162]: DEBUG:cpl-c:set_TZ: switching TZ as "TZ=" openser[5162]: ERROR:cpl_c:cpl_run_script: script error Br, Aki Rehn

2 2

[OpenSER-Users] Error using openser
by Marcio Ribeiro 20 Nov '07

20 Nov '07

Hi, I'm trying to set up openser with TLS support with mysql database, on Debian. In fact, i could not even start openser with succes. When using "openserctl start": Starting OpenSER: ERROR: PID file /var/run/openser.pid does not exist -- OpenSER start failed when using "openser": Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 164.41.14.213 [164.41.14.213]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 164.41.14.213 [164.41.14.213]:5060 Aliases: tcp: unbvoip3.cic.unb.br:5060 tcp: localhost:5060 tcp: localhost.localdomain:5060 udp: unbvoip3.cic.unb.br:5060 udp: localhost:5060 udp: localhost.localdomain:5060 *: voip.unb.br:* but it's not running, since my "ps aux | grep openser" cannot find it. Sugestions??? Thanks, Marcio

4 6

[Serusers] Can SER support 3-way Conference
by roayu 20 Nov '07

20 Nov '07

Hi there, Can SER alone support 3-way Conference w/o integrate with other applications? Thanks. Roa Yu :-)

1 0

Re: [Serusers] Voicemail setup on SER v0.9.6
by roayu 20 Nov '07

20 Nov '07

Thanks Stefan! I will try on it. Roa Yu =) -----Original Message----- From: Stefan Sayer [mailto:stefan.sayer@iptego.de] Sent: Thursday, November 15, 2007 7:41 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Voicemail setup on SER v0.9.6 roayu wrote: > Thanks Stefan! > > I need to confirm that after the installation, I will be having 2 SERs + 1 > SEMS, rite? ( 1 SER for normal sip, 1 SER as SIP stack, 1 SEMS) yes. this is the easiest way to do it. Regards Stefan > > Thanks. > > Roa Yu > -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.de www.iptego.de iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

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[Serusers] from uri???
by Roderick 19 Nov '07

19 Nov '07

Hi list, I'm looking how to forward a call to a PBX if it comes from a certain number. Just to illustrate, this would call forward to a pbx if you're dialing 123: if (uri=~"^sip:123@") { #Forward to PBX if calling 123 } What I want is something like if (from=~"^sip:123@") { #Forward to PBX if calling FROM 123 } Is there something like this??? Thanks for your help, Roderick

1 0

[OpenSER-Users] check_from() and calls from a PBX
by Iñaki Baz Castillo 19 Nov '07

19 Nov '07

Hi, I've a PBX (Asterisk) registered in OpenSer as user "pbx@domain". When Asterisk receives a call from PSTN it calls to OpenSer users with From: From: <sip:PSTN_number@domain> But of course, this can't validate "check_from()" since From username (PSTN_number) and auth username (pbx) are differents. As I solution I don't do "check_from()" if From username is "pbx" but of course, it can create security problems. Other solution is to force "From" username in PBX outgoing calls, but with this the callerid number will be "PBX" (that is not very cool for end users xD). Another solution is using "trusted" ips, but this is not valid for me since Asterisk is behind a NAT and dynamic IP, and because there can be multiples PBX for each domain of OpenSer and I don't want they to have privileges. Is there any other approach for this? Thanks for any suggestion. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

3 2

[OpenSER-Users] lookup("location") and CPL locations set
by Daniel 19 Nov '07

19 Nov '07

Hi all, I have a question: in this scenario route(1) { ..... ...... lookup ("location"); cpl_run_script("incoming", "force_stateful"); .... ... } is the location (R-URI address), re-writed by the "lookup" function, part of the CPL locations set ? Or CPL doesn't care about this and it see the locations set empty? Regards daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

2 1

[OpenSER-Users] Openser performance with rtpproxy
by Stanisław Pitucha 19 Nov '07

19 Nov '07

Hey I've got a simple openser configuration working with rtpproxy. It's mostly copied from http://www.voip-info.org/wiki/view/OpenSER+And+RTPProxy but I don't handle registration and do only nat helping (for current testing I don't even save location - Contact / URI headers are enough). I've tried setting up sipp (no authorization, just invite, send 7 secs rtp, bye) and standard UAS scenario on the other end. I start to get retransmissions / timeouts when I use 2 hosts and ~70 simultaneous connections (total). Typical delay is ~250ms (not network delay - it's on gigabit LAN). I've used the same sipp scenarios for asterisk before and got up to 700 sessions running correctly, so I believe the problem is with openser config here. How can I check what causes the problems exactly? Is there any typical performance problem? I've tried googling already - couldn't find anything obvious. I'm running svn version. Thanks for ideas.

1 0

[OpenSER-Users] [CPL] ERROR:cpl-c:run_priority_switch: error parsing attr [0][9]
by Iñaki Baz Castillo 19 Nov '07

19 Nov '07

Hi, I'm just starting with CPL and I don't know if this mail should go to devel list or users list. I load a very simple CPL script sucessfully: $ openserctl fifo LOAD_CPL sip:ibc@domain /etc/openser/cpl/ibc(a)domain.xml 200 OK The XML is this one: <?xml version="1.0" encoding="UTF-8"?> <cpl> <incoming> <time-switch> <time dtstart="9" dtend="5" wkst="MO|TU|WE|TH|FR"> <lookup source="registration"> <success> <proxy /> </success> </lookup> </time> <otherwise> <location url="sip:angel@domain"> <proxy /> </location> </otherwise> </time-switch> </incoming> </cpl> But when OpenSer runs: cpl_run_script("incoming","force_stateful"); I get in the logs: ERROR:cpl-c:run_priority_switch: error parsing attr [0][9] ERROR:cpl_c:cpl_run_script: script error I can't understand. I suppose that that XML check is done when loading the file with FIFO command. Why then I get a "error parsing" during OpenSer script? is it normal? Thanks for any explanation. Best regards. -- Iñaki Baz Castillo

3 2

[OpenSER-Users] CPL serial call attempt
by Daniel 19 Nov '07

19 Nov '07

Hi all, I have this problem. I have 2 contacts for one user subscriber, so in LOCATION tables, after REGISTER session I have: ID user contact q 12 A A@IP_X 1 13 A A@IP_Y 1 14 B B@IP_Z 1 so, B calls A and A has an incoming CPL script. In this script I'd like to call the 2 A's contacts in sequential order: The script is after looup("location"), so the R-URI is just A@IP_X (because it has lowest ID) SRIPT user A: <cpl> <incoming> <location url="sip:A@IP_Y" priority="0.3"> <proxy timeout="10" ordering="sequential" /> </location> </incoming> </cpl> Whit this script I hoped to add A@IP_Y to location set (that already included A@IP_X) with priority 0.3, and then call A@IP_X as first (it has priority=q=1) and then, if I no aswer after 10sec, call A@IP_Y (that it has lowest priority). CPL don't add A@IP_Y to the location set, but it rewrites the location (it removes the old R-URI, A@IP_X) with A@IP_Y and it call only this contact. Is there a sense in what I've write? if yes, where is the problem? regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

4 4

[OpenSER-Users] record_route() for responses
by Klaus Darilion 19 Nov '07

19 Nov '07

Hi! I just was thinking about the follwing scenario_ client a--->proxy--->client b the proxy does record routing a calls b b send back 200 OK, but does NOT mirror the Record-route headers. in-dialog requests sent from a to b will bypass the proxy, thus bypass proxy security features (e.g. removing the credentials). Thus IMO we should have some settings which allow the proxy to add RR header also for relayed response if during request processing record-route was called and the proxy misses itself as the top RR header. what do you think? regards klaus

7 15

[OpenSER-Users] Record-Route: "did" - parameter
by Franz Edler 19 Nov '07

19 Nov '07

Hi, can anyone explain what the "did" parameter in Record-Route header does? Record-Route: <sip:192.168.0.5;lr=on;ftag=e45c4b13;did=c92.09e2afc1> I suppose it is inserted in case of a "routing spiral", because it is inserted when the INVITE request is routed through the proxy a second time. Cheers Franz

2 1

[OpenSER-Users] OpenSER + Remote Database scalability issue
by John Barry 19 Nov '07

19 Nov '07

Hello, I have notice that when I configure http-digest authentication for INVITE messages in OpenSER 1.2.2, there is a big difference in the call throughput when you use a database in a remote machine. For example, when using a Mysql database in a remote machine with a RTT= 0.187 ms, the total call throughput decreases by almost 60% in comparison when using the database in the same machine where OpenSER is installed. Why is this happening? How can improve this behavior? I would like to have the database in a separate machine. Thanks, JB74 _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

2 1

[OpenSER-Users] openxcap - GET a document
by Reinhold Buchinger 18 Nov '07

18 Nov '07

Hi, I'm trying to fetch a document from the openxcap server with a normal browser. My openxcap configuration uses the database backend and I created a db table equal to the xcap table used in OpenSER. For testing I inserted some data but I'm not sure what the fields "source" and "doc-uri" should contain. Because whenever I try to fetch a document with my webbrowser I receive a 404 Not Found. OpenXCAP doesn't show any logs... Thanks a lot for your help! Best regards, Reinhold

2 4

[OpenSER-Users] openser SBC and upper registration.
by rohail khan 16 Nov '07

16 Nov '07

Hi I am trying to configure openser for uppser registration, so far I am relaying the registration after modifying contact in the registration of openser IP address but I want to save the contact only if the registration is sucessfull from upper registration server. If I use "save_noreply" its save the location no matter if the UA is registered with upper registration server any idea thanks in advance --------------------------------- Never miss a thing. Make Yahoo your homepage.

2 1

[OpenSER-Users] save("location") not saving in openser 1.3
by Dave Singer 16 Nov '07

16 Nov '07

Any one interested in helping with OpenSER 1.3? Compiled openser 1.3 from svn, installed new test db from 1.3 svn db scripts (tables should have right structure). when a phone tries to register it hits this script section. if (!www_authorize("", "subscriber")) { xlog("L_INFO", "user didn't send credentials: f-uri <$fu> r-uri <$ru>\n"); www_challenge("", "0"); exit; }; xlog("L_INFO", "user registered: f-uri <$fu>\n"); save("location"); exit; nothing is actually saving in location table. I sometimes get extra messages in the log file: core:db_print_set: Error in snprintf usrloc:db_update_ucontact: updating database failed usrloc:wb_timer: updating contact in db failed I am able to use openserctl, from the openser server with all the same db setting as the openser.cfg file has, to make changes so It "can't" be permissions. It is able to verify with the www_authorize that we should save it. just not able to do the actual save. I'm using mysql 5.x (which is also serving up the production openser 1.2-svnserver with a diff db) and openser -V renders: version: openser 1.3.0-pre1-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 3015 2007-11-05 13:08:50Z henningw $ main.c compiled on 14:57:42 Nov 5 2007 with gcc 4.1.1 Thanks Dave

2 2

[OpenSER-Users] cancel going to 127.0.0.1
by Richard Bennett 16 Nov '07

16 Nov '07

Hi, I seem to remember reading about this problem before, but couldn't find any pointers by searching... On the following setup with Openser 1.1: UA -> Openser/Mediaproxy -> PSTN-Gateway most calls pass just fine and cancel fine, but some don't stop properly. If the UA cancels the call the cancel goes: gateway ip= 1.2.3.204 proxy ip= 1.2.3.201 ua ip = 81.1.2.223 U 81.1.2.223:5071 -> 1.2.3.201:5060 CANCEL sip:0475711111@sipc.example.com SIP/2.0. Via: SIP/2.0/UDP 192.168.1.11:5071;branch=z9hG4bK-2f6ae216. From: 002 <sip:101211111@sipc.example.com>;tag=a27b4fe18ca60f4ao0. To: <sip:0475711111@sipc.example.com>. U 1.2.3.201:5060 -> 127.0.0.1:5060 CANCEL sip:0475711111@sipc.example.com SIP/2.0. Record-Route: <sip:1.2.3.201;lr=on;ftag=a27b4fe18ca60f4ao0>. Via: SIP/2.0/UDP 1.2.3.201;branch=z9hG4bK7f9f.b68d8185.0. Via: SIP/2.0/UDP 192.168.1.11:5071;received=81.1.2.223;branch=z9hG4bK-2f6ae216. From: 002 <sip:101211111@sipc.example.com>;tag=a27b4fe18ca60f4ao0. To: <sip:0475711111@sipc.example.com>. I 127.0.0.1 -> 1.2.3.201 3:3 ....E..o..@.@...Y............[h.CANCEL sip:0475711111@sipc.example.com SIP/2.0. Record-Route: <sip:1.2.3.201;lr=on;ftag=a27b4fe18ca60f4ao0>. Via: SIP/2.0/UDP 1.2.3.201;branch=z9hG4bK7f9f.b68d8185.0. Via: SIP/2.0/UDP 192.168.1.11:5071;received=81.1.2.223;branch=z9hG4bK-2f6ae216. From: 002 <sip:101211111@sipc.example.com>;tag=a27b4fe18ca60f4ao0. To: <sip:0475711111@sipc.example.com>. And then it keeps looping between 127.0.0.1 and 1.2.3.201. For some reason openser sends the CANCEL to the loopback interface instead of on to the gateway. Is this a recognizable problem or should I post a full trace and config file? Thanks for looking, Richard.

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[OpenSER-Users] Newbie issue: Cannot exec HelloWorldServlet on WeSIP
by Alvaro Mari Diego 16 Nov '07

16 Nov '07

Hi everyone, I'm new to WeSIP, so I've got stuck pretty fast, as expected :) I got WeSIP running on OpenSER 1.2.0. I'm trying to run a HelloWorld sip servlet to test WeSIP, which responses a 200 OK and a MESSAGE request saying "Hello, World", to any incoming request to a sip:greetings@amd-openims.test URI. I mounted my sar file, and installed it in wesip webapps with the web manager application, and got WeSIP properly connected to OpenSER' seas. Openser is configured to route any MESSAGE request to WeSIP AS. But when I try to send a message to sip:greetings@amd-openims.test from a client logged as alice(a)amd-openims.test, WeSIP console displays this: ******************************************************************** SeasMessageEvent [SeasRouter] - New message received, from 138.4.7.163:6060 to 138.4.7.163:5080 over UDP SeasTransactionEvent [SeasRouter] - hash_index = 54974,label=620758046 flags=0 ExpressMessageChannel [ExpressMChannel[0]] - peerAddress = 138.4.7.163/6060 ExpressMessageChannel [ExpressMChannel[0]] - About to process MESSAGE sip:greetings@amd-openims.test SIP/2.0 SipConnector [ExpressMChannel[0]] - createProcessor: Reusing existing processor SipProcessor [ExpressMChannel[0]] - An incoming message is being assigned SipProcessor [SipProcessor[4]] - <<<<<<<<< Request Received <<<<<<<<< MESSAGE sip:greetings@amd-openims.test SIP/2.0 Route: <sip:138.4.7.163:5080;lr=on>,<sip:iscmark@scscf.amd-openims.test:6060;lr=on;s=1;h=0;d=0> Via: SIP/2.0/UDP 138.4.7.163:6060;branch=z9hG4bKeb6d.ff7360e1.0,SIP/2.0/UDP 138.4.7.163:4060;branch=z9hG4bKeb6d.000ef08.0,SIP/2.0/UDP 138.4.7.149:63926;branch=z9hG4bK-d87543-b370c6252d6ad226-1--d87543-;rport=63926 Max-Forwards: 14 To: "Servicio HelloWorld" <sip:greetings@amd-openims.test> From: "alice" <sip:alice@amd-openims.test>;tag=737cbe08 Call-ID: Y2FiYmY0MmM2NjFhYzU5NWUzNTE2MmM5MWY2ZjdhMGI. CSeq: 2 MESSAGE Allow: INVITE,CANCEL,ACK,BYE,INFO,OPTIONS,MESSAGE,SUBSCRIBE,NOTIFY,REFER Content-Type: application/unknown? User-Agent: X-Lite release 1011s stamp 41150 P-Charging-Vector: icid-value="P-CSCFabcd4739818c00000007";icid-generated-at=138.4.7.163;orig-ioi="amd-openims.test" Content-Length: 262 SeasHashTable [SipProcessor[4]] - Inserting hashIdx:54974 label:620758046 into hashTable SipProcessor [SipProcessor[4]] - Transaction [z9hG4bKeb6d.ff7360e1.0] is new Transaction?? --> true EngineSipMapper [SipProcessor[4]] - Request from 138.4.7.163 mapped to: ConvergedEngine[WeSIP_engine].StandardHttpHost[localhost] ConvergedEngineValve [SipProcessor[4]] - Invoking Host:localhost ConvergedHostValve [SipProcessor[4]] - SipSession not found for requestMESSAGE, trying match against Rules ConvergedHostValve [SipProcessor[4]] - No servlet-mapping matched this Request. SipResponse [SipProcessor[4]] - >>>>>>>>> Sending Response >>>>>>>>> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 138.4.7.163:6060;branch=z9hG4bKeb6d.ff7360e1.0,SIP/2.0/UDP 138.4.7.163:4060;branch=z9hG4bKeb6d.000ef08.0,SIP/2.0/UDP 138.4.7.149:63926;branch=z9hG4bK-d87543-b370c6252d6ad226-1--d87543-;rport=63926 Max-Forwards: 14 To: "Servicio HelloWorld" <sip:greetings@amd-openims.test> From: "alice" <sip:alice@amd-openims.test>;tag=737cbe08 Call-ID: Y2FiYmY0MmM2NjFhYzU5NWUzNTE2MmM5MWY2ZjdhMGI. CSeq: 2 MESSAGE Content-Length: 0 LingerList [SipProcessor[4]] - scheduled transaction z9hG4bKeb6d.ff7360e1.0 for deletion ExpressMessageChannel [SipProcessor[4]] - sending with spiral=false SipConnector [SipProcessor[4]] - recycle: Recycling processor SipProcessor[4] ActionRouter$ActionSender [ActionSender] - Writing 509 bytes into Action Socket SeasHashTable [ExpressMChannel[0]] - Removing hashIdx:54974 label:620758046 from hashTable SeasHashTable [ExpressMChannel[0]] - Transaction with id:z9hG4bKeb6d.ff7360e1.0, hash_idx:54974, label:620758046 removed from the Seas Hash Table ExpressMessageChannel [ExpressMChannel[0]] - Done processing MESSAGE sip:greetings@amd-openims.test SIP/2.0 ******************************************************************** And, as I feared, got a 481 response in my client. I don't know what I'm doing wrong. I wonder it could be a misconfiguration in the WeSIP server.xml, or maybe something wrong in my servlet. I attach you in a zip my HelloWorld Servlet, openser.cfg file, and wesip server.xml file. Thanks in advance, Cheers, -- Alvaro

2 1

[OpenSER-Users] ERROR: init_io_wait: could not alloc epoll array
by Ariadne Ramos 16 Nov '07

16 Nov '07

I have Openser 1.1.1 installed and working fine for long time. But now when I restart the service openser with no changes made in configuration this doesn't start. I have to restart many times to get the service up again. The error that appear in logs is the following: Nov 15 11:32:06 prxc /usr/local/sbin/openser[22658]: ERROR: init_io_wait: could not alloc epoll array Nov 15 11:32:06 prxc /usr/local/sbin/openser[22658]: ERROR: tcp_receive_loop: exiting... Do you know why this is happening? I appreciate your help Thanks in advanced Ariadne

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[OpenSER-Users] q-value
by Daniel 16 Nov '07

16 Nov '07

Hi, Is it possible to change q parameter from Client side? I'm using xlite and Sjphone, but I don't be able to change this value so, Proxy, by default, set as 1 every q-value. any clue? Regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] [OT] Do ITSP SIP_PSTN gateways allow REFER?
by Iñaki Baz Castillo 16 Nov '07

16 Nov '07

Hi, I will connect my OpenSer with a Nortel (SIP to PSTN gateway) for incoming/outgoings call to/from PSTN. I have a doubt: if an OpenSer user (200(a)mydomain.org) receives an incoming call from gateway, could he REFER the call to 201(a)mydomain.org? This is: do ITSP gateways allow REFER to SIP users subcribers to my OpenSer? I understand they could not accept REFER's to PSTN number because accounting issues (gateway UAC receving a REFER and calling to PSTN?). Thanks for any explanation. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[Serusers] rtpproxy and rtcp
by Miles Scruggs 15 Nov '07

15 Nov '07

Just wondering if there is any way to collect rtcp information from rtpproxy? Thanks Miles

1 0

[OpenSER-Users] attention!!! -- mailing list domains
by Daniel-Constantin Mierla 15 Nov '07

15 Nov '07

Hello, as I see that some send emails to xyz(a)lists.openser.org and cc to xyz(a)openser.org. I want to make sure that everybody understands that only the new domain "lists.openser.org" MUST be used now. First is confusing and, also, we do not need duplicates. If you get problems with lists.openser.org, please report. The old domain "openser.org" for mailing lists will stay in place about one more month from now on, to be sure that most of discussion threads started before migration get closed. The reason at migration time was not to create chaos by not being able to reply, or not having enough time to learn about the new domain. The old mailing lists domains are just aliased to the new ones, the messages end in the same archive, same server. Thanks, Daniel

1 0

[OpenSER-Users] Permissions module addition
by Daryl G. Jurbala 15 Nov '07

15 Nov '07

I'm looking for an addition to the permissions module. I'm using the address table to match valid originators. Each originator needs to be placed into a specific route tree (in carrierroute). Right now, I can do something like: $var(orig) = "unauth"; if (allow_source_address("<fist route tree>")) { $var(orig) = "<first route tree>" } if (allow_source_address("<second route tree>")) { $var(orig) = "<second route tree>" } <etc., etc. for each route tree I have> if $var(orig)=="unauth" { sl_send_reply("403", "Unauthorized IP"); exit; } else { ....process the call..... What I'd like to see is a permissions function that would allow me to do something like: $var(orig) = allow_source_address_group() if $var(orig)=="-1" { sl_send_reply("403", "Unauthorized IP"); exit; } else { ....process the call..... Where the proposed allow_source_address_group() function would search all IPs in the address table and return either a -1 if no matching address is found, or return the value of the group column if one is found. Anyone out there willing to make these changes? If so, please contact me. I'm willing to put a bounty on this. Thanks, Daryl G. Jurbala Director of Network Operations Global Convergence Solutions Tel: +1 732 853 0513

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[Serusers] ACK Problem : Not routed to NAT'd port
by Ricardo Martinez 15 Nov '07

15 Nov '07

Hello list. I'm facing a problem with my SER configuration and a call to a NAT'd endpoint. When I call a NAT'd endpoint registered in my SER proxy, the call is established, but the last ACK message is not properly routed by SER. You can see this in the next messsage flow. 100.100.100.50 : Callee 100.100.100.99 : SER 100.100.100.231 : Called U 100.100.100.50:5060 -> 100.100.100.99:5060 INVITE sip:008991000404@sip.mydomain.com SIP/2.0. Via: SIP/2.0/UDP 100.100.100.50;rport;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000010f. Contact: <sip:551100000121@100.100.100.50;user=phone>. U 100.100.100.99:5060 -> 100.100.100.50:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. U 100.100.100.99:5060 -> 100.100.100.231:62447 INVITE sip:551100000684@100.100.100.231;user=phone SIP/2.0. Record-Route: <sip:100.100.100.99;ftag=979715625143;lr=on>. Via: SIP/2.0/UDP 100.100.100.99;branch=z9hG4bKe4c7.150a3f81.0. Via: SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. Contact: <sip:551100000121@100.100.100.50;user=phone>. U 100.100.100.231:62447 -> 100.100.100.99:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 100.100.100.99;branch=z9hG4bKe4c7.150a3f81.0,SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. (The call is answered) U 100.100.100.231:62447 -> 100.100.100.99:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.100.99;branch=z9hG4bKe4c7.150a3f81.0,SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. Contact: <sip:551100000684@100.100.100.231;user=phone>. U 100.100.100.99:5060 -> 100.100.100.50:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. Contact: <sip:551100000684@100.100.100.231;user=phone>. U 100.100.100.50:5060 -> 100.100.100.99:5060 ACK sip:551100000684@100.100.100.231;user=phone SIP/2.0. Via: SIP/2.0/UDP 100.100.100.50;rport;branch=z9hG4bK404c94810000005a43b2b7f00000182a00000112. Route: <sip:100.100.100.99;ftag=979715625143;lr=on>. U 100.100.100.99:5060 -> 100.100.100.231:5060 ACK sip:551100000684@100.100.100.231;user=phone SIP/2.0. Record-Route: <sip:100.100.100.99;ftag=979715625143;lr=on>. Via: SIP/2.0/UDP 100.100.100.99;branch=0. Via: SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7f00000182a0000 0112. U 100.100.100.231:62447 -> 100.100.100.99:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.100.99;branch=z9hG4bKe4c7.150a3f81.0,SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. Contact: <sip:551100000684@100.100.100.231;user=phone>. U 100.100.100.99:5060 -> 100.100.100.50:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7ee000053dc0000 010f. Contact: <sip:551100000684@100.100.100.231;user=phone>. U 100.100.100.50:5060 -> 100.100.100.99:5060 ACK sip:551100000684@100.100.100.231;user=phone SIP/2.0. Via: SIP/2.0/UDP 100.100.100.50;rport;branch=z9hG4bK404c94810000005a43b2b7f000001d6e00000114. Route: <sip:100.100.100.99;ftag=979715625143;lr=on>. . U 100.100.100.99:5060 -> 100.100.100.231:5060 ACK sip:551100000684@100.100.100.231;user=phone SIP/2.0. Record-Route: <sip:100.100.100.99;ftag=979715625143;lr=on>. Via: SIP/2.0/UDP 100.100.100.99;branch=0. Via: SIP/2.0/UDP 100.100.100.50;rport=5060;branch=z9hG4bK404c94810000005a43b2b7f000001d6e0000 0114. (.........) (Timeout and the call is released) U 100.100.100.231:62447 -> 100.100.100.99:5060 BYE sip:551100000121@100.100.100.50;user=phone SIP/2.0. Via: SIP/2.0/UDP 192.168.1.20;rport;branch=z9hG4bKc0a801140000002643b2a9be000036e10000001f. Route: <sip:100.100.100.99;ftag=979715625143;lr=on>. . U 100.100.100.99:5060 -> 100.100.100.50:5060 BYE sip:551100000121@100.100.100.50;user=phone SIP/2.0. Record-Route: <sip:100.100.100.99;ftag=93794163011;lr=on>. Via: SIP/2.0/UDP 100.100.100.99;branch=z9hG4bK15c7.924c3535.0. Via: SIP/2.0/UDP 192.168.1.20;received=100.100.100.231;rport=62447;branch=z9hG4bKc0a801140000 002643b2a9be000036e10000001f. U 100.100.100.50:5060 -> 100.100.100.99:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 100.100.100.99;branch=z9hG4bK15c7.924c3535.0,SIP/2.0/UDP 192.168.1.20;rport=62447;received=100.100.100.231;branch=z9hG4bKc0a801140000 002643b2a9be000036e10000001f. U 100.100.100.99:5060 -> 100.100.100.231:62447 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.20;rport=62447;received=100.100.100.231;branch=z9hG4bKc0a801140000 002643b2a9be000036e10000001f. I don't understand why the the ACK message is routed to port 5060 and not to the port 62447 like the INVITE or the final OK. Is this maybe a configuration problem?. My ser.cfg is based on the onsip.org configuration documents plus some modifications by me. Can someone help me here Thanks in advance Ricardo Martinez.-

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[OpenSER-Users] Illegal characteres in SIP URI?
by Iñaki Baz Castillo 15 Nov '07

15 Nov '07

Hi, could somebody tell me which characteres are allowd in a SIP URI? which RFC defines it? I ask it because I do a DB query with $ru so a SQL injection it's possible if RURI contains single or double ' So at the begining of the script I'd like to reject a message if the RURI contains illegal symbols. Could be useful a core function for this? Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 3

Re: [OpenSER-Users] Aliases table and db aliases
by Iñaki Baz Castillo 15 Nov '07

15 Nov '07

El Thursday 15 November 2007 16:08:27 escribió: > *Hi,* Better replies to the mailist ;) > Iñaki Baz Castillo wrote: > > * alias with DB_ALIASES module: > > > > table dbaliases: > > alias_username: 1234 > > alias_domain: IP_PROXY > > username: bob > > domain: IP_PROXY > > > > db_aliases_lookup() will rewrite URI when it's 1234@IP_PROXY and it'll > > become bob@IP_PROXY > > > > Later lookup("location") will look for the contact of that AOR. > > *Ok. > db_aliases talble is filled in by myself through "openserctl db_alias > add" command.* > > > * alias with ALIASES table: > > > > table aliases: > > username: 1234 > > domain: IP_PROXY > > contact: sip:bob@IP_PROXY > > > > lookup("aliases") will rewrite URI when it's 1234@IP_PROXY and it'll > > become bob@IP_PROXY > > > > Later lookup("location") will look for the contact of that AOR. > > *so, is ALIASES table also created by myself through "openserctl" > command*s ? No idea. In fact I fill these tables via SQL. > > ok now? > > *ok, thank you very much. > daniel > * Regards. -- ilimit... *Iñaki Baz Castillo* ibc(a)in.ilimit.es ÀREA SISTEMES 0034 937 333 375 VOLTA 1, PIS 5 08224 TERRASSA.BCN Aquest enviament és confidencial i està destinat únicament a la persona a qui s'ha enviat. Pot contenir informació privada sotmesa al secret professional, la distribució de la qual està prohibida per la legislació vigent.

1 0

Re: [OpenSER-Users] Aliases table and db aliases
by Daniel 15 Nov '07

15 Nov '07

Thanks. But I don't understand the use of aliases table. I have defined some subscriber users, with username (like BOB) and password (BOBPWD). So, if I wanted to access the system, I would send a REGISTER message with correct username and password (authorization required). Then my proxy address will be "sip:BOB@IP_PROXY" and my Contact will be "sip:BOB@MY _IP_ADRRESS". These information are saved into USER LOCATION table (save("location") function). Now, through DB_aliases module and dbaliases table, I can define some aliases for BOB (like "sip:1234@IP_PROXY", etc...), so if an INVITE for bob has "sip:1234@IP_PROXY" as R-URI, with db_aliases_lookup() function I can rewrite R-URI with Bob's URI ("bob@IP_PROXY") and then call the lookup("location") function. But I don't understand how I can use the ALIASES table. Are there any example explaining its use ? Sorry. Regards, daniel Iñaki Baz Castillo wrote: El Thursday 15 November 2007 14:31:52 Daniel escribi�: Hi all, what is the different between Aliases table and DB aliases table? What is their use ? "aliases" table is same as "location" table but it should be used to look for alias with the function lookup("aliases"). Of course, aliases are simpler than the complex location or aliases tables that manage more info about users, so in order to implement alias system in a simpler way the module ALIAS_DB exists and its table dbaliases. Note that dbaliases table is much simpler (it's just valid for aliases), Look at the module doc: http://www.openser.org/docs/modules/1.3.x/alias_db.html -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

2 1

[Serusers] Implicit Registration Set
by Magnus Sörman 15 Nov '07

15 Nov '07

Hi, How do you/could you configure SER to use implicit registration set? If e.g. a user with a sip URI registers, then both that address and his tel-uri should be registered. Thanks, /Magnus Sorman

1 0

[OpenSER-Users] Aliases table and db aliases
by Daniel 15 Nov '07

15 Nov '07

Hi all, what is the different between Aliases table and DB aliases table? What is their use ? regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

2 1

[OpenSER-Users] Presence module help
by Stanisław Pitucha 15 Nov '07

15 Nov '07

Hello Can someone tell me couple of details about presence code? I.e.: 1. There's PUBL_TYPE and WINFO_TYPE define. They are used as flags, but then they are just marking winfo / other packages. What's the effect of PUBL_TYPE on event package exactly? 2. What is the meaning of 'get_rules_doc', 'apply_auth_nbody', 'get_auth_status' funcs? Unfortunately I wasn't able figure it out from source completely. Thanks Stanisław Pitucha

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[OpenSER-Users] OpenSER with multidomain configuration
by satish patel 15 Nov '07

15 Nov '07

Dear all I am going to install openser with multidomain support do anyone have sample .cfg file or configuration guide i have find many document but it is very hard to implement so just suggest me how to test and configure it ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org --------------------------------- Never miss a thing. Make Yahoo your homepage.

2 1

[Serusers] SER Dispatcher Module
by Arun Kumar 15 Nov '07

15 Nov '07

Hi Here is my setup Users ----> SER ----> Asterisk ----> PSTN works fine. when ever user dial able to hear balance message and calls goes fine. Just now I replaced my setup to use Dispatcher module Users -----> SER (Dispatcher module) ----------> Asterisk Boxes -----> PSTN when user makes calls not able to hear balance message all the times but able to make PSTN calls ? In my config I just changed replace IP of my asterisk box with dispatcher config: Old config: strip(1); rewritehostpost("x.x.x.x:5060"); route(4); NAT Detet. New conifg: strip(1); ds_select_dst("1", "0"); forward(uri:host, uri:port); route(4); dispatcher file: 1 sip:x.x.x.x:5060 1 sip.y.y.y.y:5060 Need help on above. Thanks, Arun

2 2

[Serusers] DTMF Problem
by Arun Kumar 15 Nov '07

15 Nov '07

Hi Here is my setup: USER ------> PSTN -----> Asterisk A ----> IAX2 Trunk ----> Asterisk B ---------> SER --------> Asterisk C I'm not able to receive DTMF passed by USER on Asterisk C. All my asterisk boxs are configured with same DTMF type (auto) but no luck. Please help on this issue. Thanks, Arun

1 0

[OpenSER-Users] MyISAM or InnoDB in MySQL tables?
by Iñaki Baz Castillo 15 Nov '07

15 Nov '07

Hi, MySQL tables of OpenSer created by the script are "MyISAM". Is there any reason for that instead of using InnoDB? As I know: MyISAM: - Is faster reading and writting (except writting various entries to same table since it blocks the entire table). - No transactions or rollbacks. - Small disc space requeriments. InnoDB: - Transactions, rollbacks, restrictions in foreing keys. - Faster writting various entries in same table (entry lock instead of table lock). - More disc space and RAM requeriments. Yes, it can be obvious that for OpenSer the engine MyISAM can be a good choice but I would like to know if there is more data for this choice. Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [Serusers] Voicemail setup on SER v0.9.6
by roayu 15 Nov '07

15 Nov '07

Thanks Stefan! I need to confirm that after the installation, I will be having 2 SERs + 1 SEMS, rite? ( 1 SER for normal sip, 1 SER as SIP stack, 1 SEMS) Thanks. Roa Yu -----Original Message----- From: Stefan Sayer [mailto:stefan.sayer@iptego.de] Sent: Tuesday, November 13, 2007 6:27 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Voicemail setup on SER v0.9.6 Hello, roayu wrote: > Which version of SEMS should I download? Is it included the SER as well? > I would prefer a stable version =) I would recommend the stable SEMS 0.10.0 release. Bundled with its SER its available at ftp://ftp.iptel.org/pub/sems/sems-0.10.0-bundle.tar.gz You will find all information at http://iptel.org/sems. Regards Stefan Sayer > > 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 >> > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.de www.iptego.de iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

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[OpenSER-Users] pua_usrloc
by Venter, Andries <andries＠sun.ac.za> 15 Nov '07

15 Nov '07

listen=udp:196.211.9.222 port=5065 loadmodule "presence.so" loadmodule "avpops.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "mi_fifo.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "acc.so" loadmodule "domain.so" loadmodule "pua.so" loadmodule "pua_mi.so" loadmodule "pua_usrloc.so" modparam("usrloc|auth_db|uri_db|acc", "db_url", mysql://openser:password@localhost/openser") modparam("presence|pua", "db_url", "mysql://openser:password@localhost/openser") modparam("pua_usrloc", "default_domain", "196.211.9.222") modparam("presence", "force_active", 1) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip:196.211.9.222:5065") if( is_method("PUBLISH|SUBSCRIBE")) { route(4); } if (method=="REGISTER") { xlog("L_INFO", "Kyk of ek moet PUBLISH........................\n"); if(!search("^User-Agent: X-Lite")) { xlog("L_INFO", "JA - ek moet PUBLISH........................\n"); pua_set_publish(); } save("location"); exit; }; # presence handling route route[4] { xlog("L_INFO","Route 4........................................\n"); sl_send_reply("100","trying"); # absorb retransmissions if (! t_newtran()) { xlog("L_INFO", "Not new transaction.................................\n"); sl_reply_error(); exit; }; # append_to_reply("Contact: <sip:196.211.9.222:5065>\r\n"); if(is_method("PUBLISH")) { xlog("L_INFO", "Handling PUBLISH.................................\n"); handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { xlog("L_INFO", "Handling SUBSCRIBE.................................\n"); handle_subscribe(); t_release(); }; exit; }

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Re: [OpenSER-Users] pua_usrloc
by Venter, Andries <andries＠sun.ac.za> 15 Nov '07

15 Nov '07

My config is as follows listen=udp:196.211.9.222 port=5065 loadmodule "presence.so" loadmodule "avpops.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "mi_fifo.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "acc.so" loadmodule "domain.so" loadmodule "pua.so" loadmodule "pua_mi.so" loadmodule "pua_usrloc.so" modparam("usrloc|auth_db|uri_db|acc", "db_url", mysql://openser:password@localhost/openser") modparam("presence|pua", "db_url", "mysql://openser:password@localhost/openser") modparam("pua_usrloc", "default_domain", "196.211.9.222") modparam("presence", "force_active", 1) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip:196.211.9.222:5065") if( is_method("PUBLISH|SUBSCRIBE")) { route(4); } if (method=="REGISTER") { xlog("L_INFO", "Kyk of ek moet PUBLISH........................\n"); if(!search("^User-Agent: X-Lite")) { xlog("L_INFO", "JA - ek moet PUBLISH........................\n"); pua_set_publish(); } save("location"); exit; }; # presence handling route route[4] { xlog("L_INFO","Route 4........................................\n"); sl_send_reply("100","trying"); # absorb retransmissions if (! t_newtran()) { xlog("L_INFO", "Not new transaction.................................\n"); sl_reply_error(); exit; }; # append_to_reply("Contact: <sip:196.211.9.222:5065>\r\n"); if(is_method("PUBLISH")) { xlog("L_INFO", "Handling PUBLISH.................................\n"); handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { xlog("L_INFO", "Handling SUBSCRIBE.................................\n"); handle_subscribe(); t_release(); }; exit; }

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[OpenSER-Users] pua_usrloc
by Venter, Andries <andries＠sun.ac.za> 15 Nov '07

15 Nov '07

I would like to get presence published for all phones. I call pua_set_publish() before doing save ("location"), but no PUBLISH messages are generated when the REGISTER request comes through. I am using OpenSER 1.2.2. Any ideas?

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[Serusers] tcp branching errors
by Tomasz Zieleniewski 15 Nov '07

15 Nov '07

Hi, I have the following situation. I have two ser rel_2_0_0 instances one acting as a boundary proxy the other on as internal proxy (registrar, location). I have the following situation I have a user which has registered two contacs one woth tcp transport the other one with udp. All messages coming to and outgoing from my network travels through boundary sip proxy. When internal ser perform lookup_location it get to contacts and the destination set contains two records: Nov 15 11:03:13 sen ser[3673]: route[FORWARD]: destination set: Contact: sip:xlite@192.168.0.164:1221;rinstance=d35f9cda6bd79b0a;transport=TCP, sip:xlite@192.168.0.164:7060 Then internal ser forward message to boundary SIP proxy by changing the destination of the message and with the usage of t_relay(). I see from log that there are two branches involved: Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: <INVITE> ruri:<sip:xlite@192.168.0.164:1221;rinstance=d35f9cda6bd79b0a;transport=TCP> t:<sip:00@voip.rd.touk.pl> f:<sip:lkl@voip.rd.touk.pl> Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: append_hf(P-Warn: forked branch) Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: <INVITE> ruri:<sip:xlite@192.168.0.164:7060;rinstance=cfca8523e3747cb0> t:<sip:00@voip.rd.touk.pl> f:<sip:lkl@voip.rd.touk.pl> Nov 15 11:03:13 sen ser[3673]: branch_route[BRANCH_ROUTE]: append_hf(P-Warn: forked branch) both INVITES reaches boudary SIP proxy. Because first contact was unreachable there was an tcp error: ser[3516]: ERROR: tcp_blocking_connect: poll error: flags 18 Nov 15 11:03:13 sen ser[3516]: ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused Nov 15 11:03:13 sen ser[3516]: ERROR: tcpconn_connect: tcp_blocking_connect failed Nov 15 11:03:13 sen ser[3516]: ERROR: tcp_send: connect failed Nov 15 11:03:13 sen ser[3516]: msg_send: ERROR: tcp_send failed Nov 15 11:03:13 sen ser[3516]: ERROR: t_send_branch: sending request on branch 0 failed after which ser invokes sl_reply_error() which results in 477 next hop error reponse message sent to internal ser instance. The problem arises when boudary sip proxy processes second INVITE which correspond to second destination. when boundary SIP proxy receives it doesn't send it to user agent but instead reponses to internal ser with 500 I'm terribly sorry, server error occurred (1/SL) the only thing I see in the boundary SIP proxy log is: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) Did I miss something here or this is an error? Waiting for Your feedback Best regards Tomasz

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[OpenSER-Users] OpenSER CVS
by Bogdan-Andrei Iancu 15 Nov '07

15 Nov '07

Hi, If there are good arguments against, Wednesday 14th, 12:00 AM GMT, the CVS repository from SF will be disabled. Regards, Bogdan

1 2

[OpenSER-Users] RPID or PAI just for calls to PSTN?
by Iñaki Baz Castillo 15 Nov '07

15 Nov '07

Hi, I'll add RPID or PAI header in calls to PSTN gateway, but the fact is that I add those headers after user Authentication, even if it's a local call to other domain user and so. Could be problematic in any way to insert those headers when they are not needed? Of course, if 200@domain INVITE's 201@domain and during auth OpenSer add "RPID=+0034123123123..." I want 201 sees 200 as callerid. Maybe some UAS use received RPID or PAI as callerid? maybe I should add those headers just if the call goes to PSTN gateway? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Preserving domain name on route()
by Daniel Corbe 15 Nov '07

15 Nov '07

So, externally I'm using DNS SRV records to get traffic to my proxy server. Once it hits my proxy server though, I want to preserve the domain name portion of the URI, but still route the traffic around from server to server. Currently I'm using rewritehostport to forward traffic onto another box. This of course changes the field I'm interested in preserving. Is there another way to route traffic around? The only other option I see is to use 1 set of SRV records externally and another set of SRV records internally. Thanks. -Daniel

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[OpenSER-Users] [OT] Nortel CS2000 and "RPID" / "P-Asserted-Identity"
by Iñaki Baz Castillo 15 Nov '07

15 Nov '07

Hi, I need to send SIP traffic from my OpenSer to a Nortel CS2000 (SIP_2_PSTN gateway). I have a bad new: the provider tells me that they set the callerid with the "From" I send. I've asked them to support "RPID" or "P-Asserted-Identity" headers (still waiting for a reply). In case I can only use "From" I can modify it with UAC module, but I also know this is very anti-RFC3261 behaviour. Does anyone know if Nortel CS2000 can use "RPID" or "P-Asserted-Identity" to set the PSTN calling number? What do you think about changing From header? for example: 200(a)domain.com -> 00123223223234@IP Thanks for any response. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Permissions Modile - BYE handling
by Douglas Garstang 15 Nov '07

15 Nov '07

So, I've been playing around with the permissions module. It's working ok, except... I am sending calls from Asterisk to OpenSER, and then onto an ITSP. Calls only flow in this outbound direction. There are NO inbound calls. I put the IP addresses of the Asterisk servers into the address table and used allow_source_address() to confirm the calls where allowed. However, I found that I also had to put the IP's of the ITSP's in the address table or otherwise when the remote end hangs up and sends a BYE message the call to allow_source_address() returns false. This isn't that big a deal, but it seems messy. It also means that the ITSP, being in the address list, could send calls to us, and we don't want that. How do I work around this? Do I do: if ( allow_source_address("1") || method=="BYE" ) { .... } What about CANCEL? I haven't tried that. The same thing goes for putting IP's in the trusted table and using if_trusted(). Douglas. ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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[OpenSER-Users] Exported Statistics
by Daniel Corbe 14 Nov '07

14 Nov '07

Hello, Several modules have "exported statistics" which may be extremely useful to me in monitoring my network. The module documentation isn't clear on how to get access to these statistics. Can anyone tell me how I would probe my OpenSER box for this information? Thanks. -Daniel

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Re: [OpenSER-Users] Problem with RINSTANCE in X-LITE 3.0
by Thiago Maluf 14 Nov '07

14 Nov '07

Hi Mr. Castillo, I tried it, but don't work. My problem is: - Each Register created by X-LITE 3.0 is a new contact because each message have parameter RINSTANCE with new value. - So OpenSER will keep in your location table all the contacts received. - I want break it, so for each Register Message Openser will update this contact in your location table. My location table (Replace My Ip Number to xxxxxxxxxxxx): id | username | domain | contact | received | path | expires | q | callid | cseq | last_modified | flags | cflags | user_agent | socket | methods ----+----------+--------+-------------------------------------------------------------+----------+------+---------------------+------+----------------------------------------------+------+---------------------+-------+--------+----------------------------------+--------------------------+--------- 36 | teste1 | | sip:teste1@xxxxxxxxx:27420;rinstance=457f14a9ad568c74 | | | 2007-11-14 16:54:05 | 0.00 | YjY5YmFkZDRkZjBmOTliOWFiNmY2ODViMmRmMTg3ZjQ. | 2 | 2007-11-14 15:54:05 | 0 | 0 | X-Lite release 1011s stamp 41150 | udp:xxxxxxxxxxxx:5060 | 5951 37 | teste1 | | sip:teste1@xxxxxxxxxxxx:9150;rinstance=d491e76ae7d17c63 | | | 2007-11-14 17:01:09 | 0.00 | OTdiNDM4Y2M4NWEzMjIwMmU5MWE1Y2E5M2E3N2VkNjY. | 2 | 2007-11-14 16:01:09 | 0 | 0 | X-Lite release 1011s stamp 41150 | udp:xxxxxxxxxxxx:5060 | 5951 38 | teste1 | | sip:teste1@xxxxxxxxxxxxx:32448;rinstance=a88d552b74e3ebbc | | | 2007-11-14 17:07:13 | 0.00 | ZTE4YTE4OTVlOGJjYTExM2IxM2RiZjFlOGMyNzI3Nzk. | 2 | 2007-11-14 16:07:13 | 0 | 0 | X-Lite release 1011s stamp 41150 | udp:xxxxxxxxxxxx:5060 | 5951 Then this problem is just with X-LITE 3.0 Already talked with CounterPath' Support and this error is knowledge them and in the next version will be removed. But still in demo this new version, and will need long time to be released. Wait your help. Thiago! -------------------------------------------------------------------------------------------- El Wednesday 14 November 2007 14:55:00 Thiago Maluf escribi?: > Hi list, > The X-Lite 3.0 create one new rinstance to each REGISTER Message. > Therefore the next register will be one new contact in location table. > Because this new register have new contact. > So it's my problem. I should in soon config my OpenSER to just allow one > register by user. And with this problem, I won't config it. > Thanks in Advanced, Ok, then you can use: modparam("registrar", "method_filtering", 1) so just contacts that "Allow" the actual method (INVITE, MESSAGE or whatever) will be called. For this you should, in each REGISTER, do a DB query (or FIFO command) to test if there was a previous entry for this location, and in that case modify the "methods" field of new entry: http://openser.svn.sourceforge.net/viewvc/openser/trunk/parser/msg_parser.h… -- I?aki Baz Castillo ibc(a)in.ilimit.es -- ---------------------------------------------------------------- THIAGO MALUF RESENDE e-mail: malufrj(a)gmail.com

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[OpenSER-Users] Problem with RINSTANCE in X-LITE 3.0
by Thiago Maluf 14 Nov '07

14 Nov '07

Hi list, The X-Lite 3.0 create one new rinstance to each REGISTER Message. Therefore the next register will be one new contact in location table. Because this new register have new contact. So it's my problem. I should in soon config my OpenSER to just allow one register by user. And with this problem, I won't config it. Thanks in Advanced, Thiago ---------------------------------------------------------------- THIAGO MALUF RESENDE Voip Developer e-mail: malufrj(a)gmail.com

2 1

[OpenSER-Users] Clients behind the same NAT
by Daniel 14 Nov '07

14 Nov '07

HI all, I've read the solutions proposed at http://www.openser.org/dokuwiki/doku.php/examples:caller-callee-behind-same… 2 client behind the same NAT. The clients may be behind the same NAT, but on different sub-net (for example, one on 192.168.10.x and another on 192.168.2.x). In this case, checking source and destination IP, to discriminate against behind or not the same NAT, is not enough - they can't reach each othere anyway. What do you think? Regards, daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

3 4

[OpenSER-Users] $ruri PV missing from pseudovariables wiki page
by Ovidiu Sas 14 Nov '07

14 Nov '07

Hi all, It seems that somehow the description for $ruri PV is missing from the pseudovariables wiki page. Regards, Ovidiu Sas

6 6

[OpenSER-Users] presence module - wrong Subscription-State when doing un-subscribe ?
by Pascal Maugeri 14 Nov '07

14 Nov '07

Hi I'm doing stability testing against presence module and after "several" requests an un-subscribe request (SUBSCRIBE with Expires=0) triggers a NOTIFY with a payload and the following Subscription-State header value: Subscription-State: active;expires=0 In my humble opinion this is incorrect as the RFC 3265 section says in section 3.3.6: Note that the NOTIFY messages triggered by SUBSCRIBE messages with "Expires" headers of 0 will contain a "Subscription-State" value of "terminated", and a "reason" parameter of "timeout". Ref.: http://tools.ietf.org/html/rfc3265#section-3.3.6 Best regards, Pascal

2 4

[OpenSER-Users] Using Carrierroute - Weighted fail over instead of percentage groups...
by Kyle Stone 14 Nov '07

14 Nov '07

I'm wondering if it's possible to have weighted failover groups instead of % balanced groups.. In Example: domain register { prefix 011 { max_targets = 2 target int.domain.com { hash_index = 1 status = 1 comment = "Try me first.." } target int2.domain.com { hash_index = 2 status = 1 comment = "Try me second.." } } } And if the one with the lowest hash_index responds with a 5XX/6XX/Temporary etc error, send the call to the next highest hash_index, before looking for a shorter prefix? Essobi

2 1

[OpenSER-Users] CPL + DBTEXT
by Chris Heiser 13 Nov '07

13 Nov '07

So I'm playing with CPL, and I notice that the dbtext table isn't getting updated. Anyone seen this behavior before? I'm running 1.2.2-tls --Chris

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[OpenSER-Users] Reg. conditional processing of invite
by padmaja.rv＠vodcalabs.com 13 Nov '07

13 Nov '07

HiThanks for the reply, I however, do not know how to implement what you mentioned in your earlier mail. can you detail the steps? One of the openser users has suggested the following: ******************************************* if (is_method('INVITE')) { # check if from URI domain equals 911operator.com if ($fd == '911operator.com') { # relay stateless using request URI forward(); exit; } else { # relay statefull using request URI if (!t_relay()) { sl_reply_error(); }; exit; } } However, the function forward(); does not parse and gives error bad arguments. Can you suggest how to get this code working? I need it working asap. Thanks again, Padmaja Hi Padmaja, There is no other way around but just to implement a call back service on your proxy. Save in DB the last caller hitting the 911 service and when receiving the 8911, just load it and fwd to it. regards, bogdan Padmaja wrote: > Hi all, > > I have an issue, where the sip user dials 911 number and hangs up. The 911 > operator then calls back to the user using the call back command provided > earlier say 8911 (there is a b2bua that interprets the call back number and > routes the call to the specific user). This number is to be routed via the > openser proxy to the b2bua. However, when the 911 calls back, proxy treats > this number as a username and since there are no registrations to this 8911 > number (though this entry is present at the proxy, no device has registered > with this number), it sends back "404 Not found". How can this be > circumvented? > > Thanks, > Padmaja

1 0

[OpenSER-Users] Drop reply
by Alex Hermann 13 Nov '07

13 Nov '07

Hello all, is it possible to completely drop a reply from failure_route? I know it is possible to change the reply with t_reply, but I need the possibility to send nothing back to the UAC. -- Greetings, Alex Hermann

3 5

[OpenSER-Users] carrierroute Error
by Mark Baker 13 Nov '07

13 Nov '07

I am attempting to use carrierroute for the first time, and the problem that I am having is that when I attempt to place a call, I see the following in stderror: /usr/local/sbin/openser[32664]: New request - M=INVITE RURI= sip:1678XXXXXXX@domain.com F=sip:1404XXXXXXX@domain.com T= sip:1678XXXXXXX@domain.com IP=204.XXX.XXX.XXX ID=133DAB62-97D8-4A2C-8D68-BAA4EC4B310C(a)192.168.5.147 /usr/local/sbin/openser[32664]: Callee is not local - M=INVITE RURI= sip:1678XXXXXXX@domain.com F=sip:1404XXXXXXX@domain.com T= sip:1678XXXXXXX@domain.com IP=204.XXX.XXX.XXX ID=133DAB62-97D8-4A2C-8D68-BAA4EC4B310C(a)192.168.5.147 /usr/local/sbin/openser[32664]: Not normalized callee '1678XXXXXXX' to E.164format - M=INVITE RURI= sip:1678XXXXXXX@domain.com F=sip:1404XXXXXXX@domain.com T= sip:1678XXXXXXX@domain.com IP=204.XXX.XXX.XXX ID=133DAB62-97D8-4A2C-8D68-BAA4EC4B310C(a)192.168.5.147 /usr/local/sbin/openser[32664]: carrierroute:carrier_rewrite_msg: Domain too big. (We only have 5 domains, you wanted 0.) /usr/local/sbin/openser[32664]: Request leaving server, D-URI='<null>' - M=INVITE RURI=sip:1678XXXXXXX@domain.com F=sip:1404XXXXXXX@domain.com T= sip:1678XXXXXXX@domain.com IP=204.XXX.XXX.XXX ID=133DAB62-97D8-4A2C-8D68-BAA4EC4B310C(a)192.168.5.147 /usr/local/sbin/openser[32667]: New request - M=INVITE RURI= sip:1678XXXXXXX@domain.com F=sip:1404XXXXXXX@domain.com T= sip:1678XXXXXXX@domain.com IP=72.XXX.XXX.XXX ID=133DAB62-97D8-4A2C-8D68-BAA4EC4B310C(a)192.168.5.147 /usr/local/sbin/openser[32660]: NAT-Reply - S=100 D=Trying F= sip:1404XXXXXXX@domain.com T=sip:1678XXXXXXX@domain.com IP=72.XXX.XXX.XXX ID=133DAB62-97D8-4A2C-8D68-BAA4EC4B310C(a)192.168.5.147 /usr/local/sbin/openser[32667]: Proxy authentication failed - M=INVITE RURI= sip:1678XXXXXXX@domain.com F=sip:1404XXXXXXX@domain.com T= sip:1678XXXXXXX@domain.com IP=72.XXX.XXX.XXX ID=133DAB62-97D8-4A2C-8D68-BAA4EC4B310C(a)192.168.5.147 /usr/local/sbin/openser[32663]: NAT-Reply - S=407 D=Proxy Authentication Required F=sip:1404XXXXXXX@domain.com T=sip:1678XXXXXXX@domain.com IP= 72.XXX.XXX.XXX ID=133DAB62-97D8-4A2C-8D68-BAA4EC4B310C(a)192.168.5.147 /usr/local/sbin/openser[32663]: Failure route for PSTN entered - M=INVITE RURI=sip:1678XXXXXXX@domain.com F=sip:1404XXXXXXX@domain.com T= sip:1678XXXXXXX@domain.com IP=204.XXX.XXX.XXX ID=133DAB62-97D8-4A2C-8D68-BAA4EC4B310C(a)192.168.5.147 Thanks for any direction!!

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Re: [Serusers] Damaged message body
by Jiri Kuthan 13 Nov '07

13 Nov '07

At 10:55 02/11/2007, Stefan.Brozinski(a)materna.de wrote: >Hello everybody > >there is a bug in SER that removes the last byte from a message body if >this happens to be a binary zero. > >This may be the case when the Content-Type is something binary (i.e. >*not* using "Content-Type: text/something"). > >The bad code is in udp_server.c. It is present in SER 0.9.x a well as in >2.0. I have reported it in the bug tracker as SER-324 and attached a >comment which points out the bad section of the code. > >How shall I proceed to fix this bug? Ideally submit a patch to JIRA...http://tracker.iptel.org/browse/SER -jiri >Best regards >Stefan >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] R: R: R: Loadbalancing for interco
by Jiri Kuthan 13 Nov '07

13 Nov '07

At 18:48 12/11/2007, 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? Not really :-( Calling forward can't change contacts -- perhaps you are mistakenly starting SER with other config file than you think? -jiri >-Stefano > >> -----Messaggio originale----- >> Da: Jiri Kuthan [mailto:jiri@iptel.org] >> Inviato: lunedì 12 novembre 2007 18.28 >> A: Stefano Capitanio; serusers(a)lists.iptel.org >> Oggetto: Re: [Serusers] R: R: Loadbalancing for interco >> >> I think there is some confusion here ... SER doesn't change contacts by >> the action referred to >> bellow. >> >> -jiri >> >> At 17:54 12/11/2007, Stefano Capitanio wrote: >> >It is the simplest ser.cfg you can imagine: >> > >> > Route { >> > forward(z.z.z.z,5060); >> > } >> > >> > >> >The message enter in SER (y.y.y.y) as: >> > >> > U x.x.x.x:5060 -> y.y.y.y:5060 >> > REGISTER sip:213.203.128.126 SIP/2.0. >> > Via: SIP/2.0/UDP >> x.x.x.x:5060;rport;branch=z9hG4bKFCDF5014775145F7BBFB33B621BC8BC0. >> > From: openser <sip:123456102@y.y.y.y>;tag=1540538748. >> > To: openser <sip:123456102@y.y.y.y>. >> > Contact: "openser" <sip:123456102@x.x.x.x:5060>. >> > Call-ID: 23FDB6F533AC4BCF840FD35F34B385FD(a)y.y.y.y. >> > CSeq: 60558 REGISTER. >> > Expires: 120. >> > Max-Forwards: 70. >> > User-Agent: X-Lite release 1103m. >> > Content-Length: 0. >> > >> >And goes out as: >> > >> > U y.y.y.y:5060 -> z.z.z.z:5060 >> > REGISTER sip:y.y.y.y SIP/2.0. >> > Via: SIP/2.0/UDP y.y.y.y:5060;branch=0. >> > Via: SIP/2.0/UDP >> 195.110.129.41:5060;rport=5060;branch=z9hG4bKFCDF5014775145F7BBFB33B621BC8 >> BC0. >> > From: openser <sip:123456102@y.y.y.y>;tag=1540538748. >> > To: openser <sip:123456102@y.y.y.y>. >> > Contact: "openser" <sip:123456102@y.y.y.y:5060>. >> > Call-ID: 23FDB6F533AC4BCF840FD35F34B385FD(a)y.y.y.y. >> > CSeq: 60558 REGISTER. >> > Expires: 120. >> > Max-Forwards: 70. >> > User-Agent: X-Lite release 1103m. >> > Content-Length: 0. >> > >> > >> > >> >> -----Messaggio originale----- >> >> Da: Jiri Kuthan [mailto:jiri@iptel.org] >> >> Inviato: lunedì 12 novembre 2007 17.41 >> >> A: Stefano Capitanio; serusers(a)lists.iptel.org >> >> Oggetto: Re: [Serusers] R: Loadbalancing for interco >> >> >> >> I haven't seen your config file, but normally it does not change >> Contacts. >> >> It changes contacts if it is configured to deal with NATs. >> >> >> >> -jiri >> >> >> >> At 17:20 12/11/2007, Stefano Capitanio wrote: >> >> >Ok, thanks for your answer, I understand your position. >> >> > >> >> >Anyway I cannot understand why SER modify the Contact header without >> any >> >> instruction about that in the config file...is there any reason >> concerning >> >> RFC compliance? >> >> > >> >> >Best regards, >> >> >Stefano >> >> > >> >> >> -----Messaggio originale----- >> >> >> Da: Jiri Kuthan [mailto:jiri@iptel.org] >> >> >> Inviato: lunedì 12 novembre 2007 17.09 >> >> >> A: Stefano Capitanio; serusers(a)lists.iptel.org >> >> >> Oggetto: Re: [Serusers] Loadbalancing for interco >> >> >> >> >> >> Well, load-balancing is not easy. To deal with issues like you are >> >> >> describing, >> >> >> your best choice is a load-balancer which is capable of working in >> >> >> transparent >> >> >> mode. We have such in our intelligence, some work, some less so, let >> me >> >> >> know >> >> >> if you need some intelligence on this. >> >> >> >> >> >> -jiri >> >> >> >> >> >> At 12:23 12/11/2007, Stefano Capitanio wrote: >> >> >> >Content-class: urn:content-classes:message >> >> >> >Content-Type: multipart/alternative; >> >> >> > boundary="----_=_NextPart_001_01C8251E.664EA61A" >> >> >> > >> >> >> >Hi, >> >> >> > >> >> >> >we are trying to do the same with our servers but we have some >> problem >> >> >> with registrations: >> >> >> >it seem that when the REGISTER message pass through SER, the host- >> part >> >> of >> >> >> Contact field is modified with the local address of SER. >> >> >> >Is it a misconfiguration problem? >> >> >> >Do you have experience in balancing also the registrations? >> >> >> > >> >> >> >Thanks, >> >> >> >Stefano >> >> >> > >> >> >> > >> >> >> >On Thu, 2007-09-13 at 11:02 +0200, inge wrote: >> >> >> > >> >> >> >> Hi all, >> >> >> > >> >> >> >> >> >> >> > >> >> >> >> Is there a way to have something like a loadbalancing on SER for >> >> >> > >> >> >> >> outgoing calls ? >> >> >> > >> >> >> >> >> >> >> > >> >> >> >> I want to distribute the calls between two gateways. Ideally, >> with a >> >> >> > >> >> >> >> coefficient (ie. 60% for the first and 40% for the second). >> >> >> > >> >> >> >> >> >> >> > >> >> >> >> Thanks for your support. >> >> >> > >> >> >> >> >> >> >> > >> >> >> >> Adrien .L >> >> >> >_______________________________________________ >> >> >> >Serusers mailing list >> >> >> >Serusers(a)lists.iptel.org >> >> >> >http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> >> Jiri Kuthan http://iptel.org/~jiri/ >> >> >_______________________________________________ >> >> >Serusers mailing list >> >> >Serusers(a)lists.iptel.org >> >> >http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> >> >> >> >> -- >> >> Jiri Kuthan http://iptel.org/~jiri/ >> >_______________________________________________ >> >Serusers mailing list >> >Serusers(a)lists.iptel.org >> >http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> -- >> Jiri Kuthan http://iptel.org/~jiri/ >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[OpenSER-Users] Checking Reply to INVITE
by Douglas Garstang 13 Nov '07

13 Nov '07

I need to be able to check the SIP response code to INVITE messages only. I don't care about BYE, CANCEL or anything else. I tried putting a hint header in the INVITE when I sent it out, but the replies came back from the upstream provider with the header removed. I tried setting a flag when an INVITE was received and using that. Didn't work. How can I do this? Is there a way to see what method a response is for? If there is, I could ignore everything that wasn't a response to an INVITE. Thanks, Doug. ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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[OpenSER-Users] missing cflags column in User Locations (location) Table definition in wiki
by Piotr Sobolewski 13 Nov '07

13 Nov '07

HI There is one column missing in User Locations (location) Table definition in wiki link below http://www.openser.org/dokuwiki/doku.php/database:1.2.x#user_locations missing column name is cflags. -- Piotr Sobolewski sobolewski(a)gmail.com

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[OpenSER-Users] Number of sessions
by Bruno Machado 13 Nov '07

13 Nov '07

Hi I`d like to know if is possible to use 'openserctl' to find the number of sessions OpenSER is managing. I think that it isn't possible. Is there another way? Thanks. Bruno Machado Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/

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[OpenSER-Users] xml body parser
by Daniel 13 Nov '07

13 Nov '07

Hi all, Is openser able to parse and interpret a xml script contained, for example, in the body of INVITE method ? If it was possible, what kind of module I'd have to use ? Regards, Daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] [Fwd: Re: [OpenSER-Devel] pua module - unsubscribe]
by Anca Vamanu 13 Nov '07

13 Nov '07

-------- 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 >>>>> >>>>> >>>> >>>> >>> >>> >> >> > >

1 0

Re: [Serusers] Voicemail setup on SER v0.9.6
by roayu 13 Nov '07

13 Nov '07

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 >

2 1

[OpenSER-Users] how to implement calling between asterisk and asterisk?
by gonghai78 13 Nov '07

13 Nov '07

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!

1 0

[Serusers] ERROR: warning_builder: buffer size exceeded & WARNING: warning skipped -- too big
by Tomasz Zieleniewski 13 Nov '07

13 Nov '07

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

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Re: [Serusers] R: R: R: Loadbalancing for interco
by Jan Andres 13 Nov '07

13 Nov '07

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

1 0

[OpenSER-Users] question about the INVITE method
by 程千 13 Nov '07

13 Nov '07

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!

2 1

[OpenSER-Users] TCP better than UDP??
by lu 13 Nov '07

13 Nov '07

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.

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[OpenSER-Users] ERROR: receive_fd: EOF on 15
by Dave Singer 13 Nov '07

13 Nov '07

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

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[Serusers] Some calls do not go to voicemail
by Steven C. Blair 12 Nov '07

12 Nov '07

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

2 1

[Serusers] SER 0.9.6 "desc_time_order" issue
by Mitya 12 Nov '07

12 Nov '07

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

1 1

[OpenSER-Users] Mediaproxy --help required
by srinivas Antarvedi 12 Nov '07

12 Nov '07

Hello all , i have doubts regarding the mediaproxy do i have to run my openser server on public ip if i need to use the mediaproxy server on the same box as that of openser? -- Srinivas Antarvedi

2 1

Re: [Serusers] R: R: Loadbalancing for interco
by Jiri Kuthan 12 Nov '07

12 Nov '07

I think there is some confusion here ... SER doesn't change contacts by the action referred to bellow. -jiri At 17:54 12/11/2007, Stefano Capitanio wrote: >It is the simplest ser.cfg you can imagine: > > Route { > forward(z.z.z.z,5060); > } > > >The message enter in SER (y.y.y.y) as: > > U x.x.x.x:5060 -> y.y.y.y:5060 > REGISTER sip:213.203.128.126 SIP/2.0. > Via: SIP/2.0/UDP x.x.x.x:5060;rport;branch=z9hG4bKFCDF5014775145F7BBFB33B621BC8BC0. > From: openser <sip:123456102@y.y.y.y>;tag=1540538748. > To: openser <sip:123456102@y.y.y.y>. > Contact: "openser" <sip:123456102@x.x.x.x:5060>. > Call-ID: 23FDB6F533AC4BCF840FD35F34B385FD(a)y.y.y.y. > CSeq: 60558 REGISTER. > Expires: 120. > Max-Forwards: 70. > User-Agent: X-Lite release 1103m. > Content-Length: 0. > >And goes out as: > > U y.y.y.y:5060 -> z.z.z.z:5060 > REGISTER sip:y.y.y.y SIP/2.0. > Via: SIP/2.0/UDP y.y.y.y:5060;branch=0. > Via: SIP/2.0/UDP 195.110.129.41:5060;rport=5060;branch=z9hG4bKFCDF5014775145F7BBFB33B621BC8BC0. > From: openser <sip:123456102@y.y.y.y>;tag=1540538748. > To: openser <sip:123456102@y.y.y.y>. > Contact: "openser" <sip:123456102@y.y.y.y:5060>. > Call-ID: 23FDB6F533AC4BCF840FD35F34B385FD(a)y.y.y.y. > CSeq: 60558 REGISTER. > Expires: 120. > Max-Forwards: 70. > User-Agent: X-Lite release 1103m. > Content-Length: 0. > > > >> -----Messaggio originale----- >> Da: Jiri Kuthan [mailto:jiri@iptel.org] >> Inviato: lunedì 12 novembre 2007 17.41 >> A: Stefano Capitanio; serusers(a)lists.iptel.org >> Oggetto: Re: [Serusers] R: Loadbalancing for interco >> >> I haven't seen your config file, but normally it does not change Contacts. >> It changes contacts if it is configured to deal with NATs. >> >> -jiri >> >> At 17:20 12/11/2007, Stefano Capitanio wrote: >> >Ok, thanks for your answer, I understand your position. >> > >> >Anyway I cannot understand why SER modify the Contact header without any >> instruction about that in the config file...is there any reason concerning >> RFC compliance? >> > >> >Best regards, >> >Stefano >> > >> >> -----Messaggio originale----- >> >> Da: Jiri Kuthan [mailto:jiri@iptel.org] >> >> Inviato: lunedì 12 novembre 2007 17.09 >> >> A: Stefano Capitanio; serusers(a)lists.iptel.org >> >> Oggetto: Re: [Serusers] Loadbalancing for interco >> >> >> >> Well, load-balancing is not easy. To deal with issues like you are >> >> describing, >> >> your best choice is a load-balancer which is capable of working in >> >> transparent >> >> mode. We have such in our intelligence, some work, some less so, let me >> >> know >> >> if you need some intelligence on this. >> >> >> >> -jiri >> >> >> >> At 12:23 12/11/2007, Stefano Capitanio wrote: >> >> >Content-class: urn:content-classes:message >> >> >Content-Type: multipart/alternative; >> >> > boundary="----_=_NextPart_001_01C8251E.664EA61A" >> >> > >> >> >Hi, >> >> > >> >> >we are trying to do the same with our servers but we have some problem >> >> with registrations: >> >> >it seem that when the REGISTER message pass through SER, the host-part >> of >> >> Contact field is modified with the local address of SER. >> >> >Is it a misconfiguration problem? >> >> >Do you have experience in balancing also the registrations? >> >> > >> >> >Thanks, >> >> >Stefano >> >> > >> >> > >> >> >On Thu, 2007-09-13 at 11:02 +0200, inge wrote: >> >> > >> >> >> Hi all, >> >> > >> >> >> >> >> > >> >> >> Is there a way to have something like a loadbalancing on SER for >> >> > >> >> >> outgoing calls ? >> >> > >> >> >> >> >> > >> >> >> I want to distribute the calls between two gateways. Ideally, with a >> >> > >> >> >> coefficient (ie. 60% for the first and 40% for the second). >> >> > >> >> >> >> >> > >> >> >> Thanks for your support. >> >> > >> >> >> >> >> > >> >> >> Adrien .L >> >> >_______________________________________________ >> >> >Serusers mailing list >> >> >Serusers(a)lists.iptel.org >> >> >http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> >> >> >> >> -- >> >> Jiri Kuthan http://iptel.org/~jiri/ >> >_______________________________________________ >> >Serusers mailing list >> >Serusers(a)lists.iptel.org >> >http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> -- >> Jiri Kuthan http://iptel.org/~jiri/ >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

2 1

[OpenSER-Users] testing null AVPs
by Ovidiu Sas 12 Nov '07

12 Nov '07

Hi all, How should we test for non existing AVPs? method 1: is_avp_set("$avp(i:123)") method 2: $(avp(i:123)) != null Both of the above methods are working. Is there a difference between them? If no, then maybe is_avp_set should be deprecated. Regards, Ovidiu Sas

2 1

Re: [Serusers] R: Loadbalancing for interco
by Jiri Kuthan 12 Nov '07

12 Nov '07

I haven't seen your config file, but normally it does not change Contacts. It changes contacts if it is configured to deal with NATs. -jiri At 17:20 12/11/2007, Stefano Capitanio wrote: >Ok, thanks for your answer, I understand your position. > >Anyway I cannot understand why SER modify the Contact header without any instruction about that in the config file...is there any reason concerning RFC compliance? > >Best regards, >Stefano > >> -----Messaggio originale----- >> Da: Jiri Kuthan [mailto:jiri@iptel.org] >> Inviato: lunedì 12 novembre 2007 17.09 >> A: Stefano Capitanio; serusers(a)lists.iptel.org >> Oggetto: Re: [Serusers] Loadbalancing for interco >> >> Well, load-balancing is not easy. To deal with issues like you are >> describing, >> your best choice is a load-balancer which is capable of working in >> transparent >> mode. We have such in our intelligence, some work, some less so, let me >> know >> if you need some intelligence on this. >> >> -jiri >> >> At 12:23 12/11/2007, Stefano Capitanio wrote: >> >Content-class: urn:content-classes:message >> >Content-Type: multipart/alternative; >> > boundary="----_=_NextPart_001_01C8251E.664EA61A" >> > >> >Hi, >> > >> >we are trying to do the same with our servers but we have some problem >> with registrations: >> >it seem that when the REGISTER message pass through SER, the host-part of >> Contact field is modified with the local address of SER. >> >Is it a misconfiguration problem? >> >Do you have experience in balancing also the registrations? >> > >> >Thanks, >> >Stefano >> > >> > >> >On Thu, 2007-09-13 at 11:02 +0200, inge wrote: >> > >> >> Hi all, >> > >> >> >> > >> >> Is there a way to have something like a loadbalancing on SER for >> > >> >> outgoing calls ? >> > >> >> >> > >> >> I want to distribute the calls between two gateways. Ideally, with a >> > >> >> coefficient (ie. 60% for the first and 40% for the second). >> > >> >> >> > >> >> Thanks for your support. >> > >> >> >> > >> >> Adrien .L >> >_______________________________________________ >> >Serusers mailing list >> >Serusers(a)lists.iptel.org >> >http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> -- >> Jiri Kuthan http://iptel.org/~jiri/ >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] Loadbalancing for interco
by Jiri Kuthan 12 Nov '07

12 Nov '07

Well, load-balancing is not easy. To deal with issues like you are describing, your best choice is a load-balancer which is capable of working in transparent mode. We have such in our intelligence, some work, some less so, let me know if you need some intelligence on this. -jiri At 12:23 12/11/2007, Stefano Capitanio wrote: >Content-class: urn:content-classes:message >Content-Type: multipart/alternative; > boundary="----_=_NextPart_001_01C8251E.664EA61A" > >Hi, > >we are trying to do the same with our servers but we have some problem with registrations: >it seem that when the REGISTER message pass through SER, the host-part of Contact field is modified with the local address of SER. >Is it a misconfiguration problem? >Do you have experience in balancing also the registrations? > >Thanks, >Stefano > > >On Thu, 2007-09-13 at 11:02 +0200, inge wrote: > >> Hi all, > >> > >> Is there a way to have something like a loadbalancing on SER for > >> outgoing calls ? > >> > >> I want to distribute the calls between two gateways. Ideally, with a > >> coefficient (ie. 60% for the first and 40% for the second). > >> > >> Thanks for your support. > >> > >> Adrien .L >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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[OpenSER-Users] Digest Authentication and nonce count
by Klaus Darilion 12 Nov '07

12 Nov '07

Hi! the ..._challenge() functions support adding the qop=auth to the challenge. This will cause the SIP client to use the nonce count in the digest response. Is this nonce count actually used in openser? I guess this would require stateful handling of nonce and nonce-count. regards klaus

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[Serusers] Loadbalancing for interco
by Stefano Capitanio 12 Nov '07

12 Nov '07

Hi, we are trying to do the same with our servers but we have some problem with registrations: it seem that when the REGISTER message pass through SER, the host-part of Contact field is modified with the local address of SER. Is it a misconfiguration problem? Do you have experience in balancing also the registrations? Thanks, Stefano On Thu, 2007-09-13 at 11:02 +0200, inge wrote: > Hi all, > > Is there a way to have something like a loadbalancing on SER for > outgoing calls ? > > I want to distribute the calls between two gateways. Ideally, with a > coefficient (ie. 60% for the first and 40% for the second). > > Thanks for your support. > > Adrien .L

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[OpenSER-Users] SIP Response code in a Failure Route Block
by Douglas Garstang 12 Nov '07

12 Nov '07

Can someone tell me why the pseudo variable $rr (SIP reply reason) is not present in a failure_route? If a call fails, this pseudo var contains the SIP response code, '404' for a Not Found for example. I would think that it should DEFINITELY be set in a failure route. OTHER pseudo variables are set in a failure route. Why not the SIP response code? This is probably the most important pseudo variable to have available in a failure route! Douglas. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[OpenSER-Users] Forking based on XML document in SDP
by Pascal Maugeri 12 Nov '07

12 Nov '07

Hi I need to fork an INVITE request to a list of SIP URIs. The list of URIs is defined in an XML document, in the payload of the incoming request. I've read in a previous email (contributed by John Peters) that one may fork the request using avpops: > modparam("avpops","avp_aliases","myGroup=i:665") > avp_write("sip:u1@host", "$myGroup"); > avp_write(" sip:u2@host", "$myGroup"); > avp_pushto("$ruri", "$myGroup/g"); For the time being the only solution I see is to process the XML document in a external application using EXEC module that will return the list of SIP URIs for forking. Before sending the forked request(s) the XML document should be removed from the payload (I have no idea yet how to do this). Do you think there is smarter way to manage it ? Maybe I should not try to do this using OpenSER but rather forward the request to an external SIP service that will do the job ? Thanks in advance for any advice, Pascal

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[OpenSER-Users] Loose route problem with ;nat=yes
by Daniel 12 Nov '07

12 Nov '07

Hi all, there is a little problem in loose_route () function. Loose_route() function is not able to recognize the address of the proxy server (in the ROUTE header field) if ROUTE header field has the ";nat=yes" tag. So, addresses like " Route: 192.168.2.1;nat=yes;lr=on;ftag=123455 " are read by loose_route as foreign address. Meanwhile "Route: 192.168.2.1;lr=on;ftag=123455" are recognized correctly. where 192.168.2.1 = proxy server address Regerds, Daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

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[OpenSER-Users] Reg. conditional processing of invite
by Padmaja 12 Nov '07

12 Nov '07

Hi all, I have an issue, where the sip user dials 911 number and hangs up. The 911 operator then calls back to the user using the call back command provided earlier say 8911 (there is a b2bua that interprets the call back number and routes the call to the specific user). This number is to be routed via the openser proxy to the b2bua. However, when the 911 calls back, proxy treats this number as a username and since there are no registrations to this 8911 number (though this entry is present at the proxy, no device has registered with this number), it sends back "404 Not found". How can this be circumvented? Thanks, Padmaja ----- Original Message ----- From: <users-request(a)lists.openser.org> To: <users(a)lists.openser.org> Sent: Saturday, November 03, 2007 3:46 AM Subject: Users Digest, Vol 30, Issue 6 > 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. Core dumped when deploying presence module (Sergio Gutierrez) > 2. Re: MediaProxy 1.9.0 - Radius (Brian Heath) > 3. Re: Forking Madness (Chris Heiser) > 4. Re: MediaProxy 1.9.0 - Radius (Mike O'Connor) > 5. Re: MediaProxy 1.9.0 - Radius (Ovidiu Sas) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 2 Nov 2007 15:59:56 -0500 > From: "Sergio Gutierrez" <saguti(a)gmail.com> > Subject: [OpenSER-Users] Core dumped when deploying presence module > To: users(a)lists.openser.org, users(a)openser.org > Message-ID: > <8a4af3e20711021359p3327c719t4ae680de4ac3e155(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hello to all members. > > Currently we are trying to deploy presence module in the following > environment: > > 1. Openser 1.2.1 > 2. Solaris 10 on Sparc System > 3. libxml 2.6.30 > > Openser starts up fine, but when happens any event as a Status change of a > user agent, Openser crashes. > Below I send an extract from the logfile, and the backtrace of the core. > Thanks in advance for your attention. > > Sergio Guti?rrez > EPM Telecomunicaciones > Medellin - Colombia > > **************************************************** >>From Openser logfile. > > 0(5493) Nuevo Requerimiento - M=SUBSCRIBE > RURI=sip:5192901@200.13.225.250F= > sip:5192902@200.13.225.250 T=sip:5192901@200.13.225.250 > IP=200.116.28.88ID=YmE2MGVhMGQzMTI3ZmU0YTkzMDJjNjIxODNmMGIwOWE. > 0(5493) comp_scriptvar: str 20 : 5192901 > 0(5493) DEBUG:maxfwd:is_maxfwd_present: value = 70 > 0(5493) parse_headers: flags=ffffffffffffffff > 0(5493) get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> > 0(5493) DEBUG: get_hdr_body : content_length=0 > 0(5493) found end of header > 0(5493) PRESENCE: handle_subscribe: 'expires' found > 0(5493) PRESENCE: handle_subscribe: lexpire= 3600 > 0(5493) PRESENCE: handle_subscribe: 'To' header ALREADY PARSED: < > sip:5192901@200.13.225.250> > 0(5493) PRESENCE:handle_subscribe: generating to_tag > 0(5493) > [p_user]= 5192901 [p_domain]= 200.13.225.250 > [w_user]= 5192902 [w_domain]= 200.13.225.250 > 0(5493) [event]= presence > [staus]= pending > [expires]= 3600 > 0(5493) [to_tag]= 10.5493.1194034490.40 > [from_tag]= 1f783845 > 0(5493) PRESENCE: update_subscription ... > 0(5493) > [p_user]= 5192901 [p_domain]= 200.13.225.250 > [w_user]= 5192902 [w_domain]= 200.13.225.250 > 0(5493) [event]= presence > [staus]= pending > [expires]= 3600 > 0(5493) [to_tag]= 10.5493.1194034490.40 > [from_tag]= 1f783845 > 0(5493) expires: 3600 > 0(5493) PRESENCE:update_subscription:Inserting into database: > n_query_cols:14 > 0(5493) [0] = to_user 5192901 > 0(5493) [1] = to_domain 200.13.225.250 > 0(5493) [2] = from_user 5192902 > 0(5493) [3] = from_domain 200.13.225.250 > 0(5493) [4] = event presence > 0(5493) [5] = event_id > 0(5493) [6] = callid YmE2MGVhMGQzMTI3ZmU0YTkzMDJjNjIxODNmMGIwOWE. > 0(5493) [7] = to_tag 10.5493.1194034490.40 > 0(5493) [8] = from_tag 1f783845 > 0(5493) [9] = contact sip:5192902@200.116.28.88:47166 > 0(5493) [10] = status pending > 0(5493) [11] = cseq 1 > 0(5493) [12] = expires 1194038090 > 0(5493) [13] = version 0 > 0(5493) parse_headers: flags=ffffffffffffffff > 0(5493) check_via_address(200.116.28.88, 192.168.0.13, 0) > 0(5493) DBG:sl:run_sl_callbacks: callback id 0 entered > 0(5493) trace_sl_onreply_out: trace off... > 0(5493) PRESENCE: get_subs_dialog:n= 1 > 0(5493) PRESENCE:notify:dialog informations: > 0(5493) > [p_user]= 5192901 [p_domain]= 200.13.225.250 > [w_user]= 5192901 [w_domain]= 200.13.225.250 > 0(5493) [event]= presence.winfo > [staus]= active > [expires]= 3518 > 0(5493) [to_tag]= 10.5493.1194034408.34 > [from_tag]= 39320458 > 0(5493) presence:uandd_to_uri: uri=sip:5192901@200.13.225.250 > 0(5493) presence:uandd_to_uri: uri=sip:5192902@200.13.225.250 > 0(5493) presence:uandd_to_uri: uri=sip:5192901@200.13.225.250 > 0(5493) PRESENCE: notify: build notify to user= 5192901 domain= > 200.13.225.250 for event= presence.winfo > 0(5493) > [p_user]= 5192901 [p_domain]= 200.13.225.250 > [w_user]= 5192901 [w_domain]= 200.13.225.250 > 0(5493) [event]= presence.winfo > [staus]= active > [expires]= 3518 > 0(5493) [to_tag]= 10.5493.1194034408.34 > [from_tag]= 39320458 > 0(5493) PRESENCE:build_str_hdr: expires = 3518 > 0(5493) PRESENCE:build_str_hdr: subs_expires : 3518 > 0(5493) PRESENCE: build_str_hdr: headers: > Event: presence.winfo > Contact: <sip:200.13.225.250:5060> > Subscription-State: active;expires=3518 > Content-Type: application/watcherinfo+xml > > 0(5493) PRESENCE:notify: headers:Event: presence.winfo > Contact: <sip:200.13.225.250:5060> > Subscription-State: active;expires=3518 > Content-Type: application/watcherinfo+xml > > 0(5493) CONTACT = sip:5192901@200.13.254.180:17116 > 0(5493) presence:uandd_to_uri: uri=sip:5192901@200.13.225.250 > 0(5493) parse_rr_body(): No body for record-route > 0(5493) PRESENCE: notify:Send notify for presence on callback 1(5494) > DEBUG: timer routine:4,tl=fc5c9a80 next=0, timeout=353600000 > 1(5494) DEBUG: retransmission_handler : request resending (t=fc5c9910, > NOTIFY si ... ) > 1(5494) DEBUG:tm:set_timer: relative timeout is 1000000 > 1(5494) DEBUG: add_to_tail_of_timer[5]: fc5c9a80 (354600000) > 1(5494) DEBUG: retransmission_handler : done > 1(5494) DEBUG: timer routine:5,tl=fc5c9a80 next=0, timeout=354600000 > 1(5494) DEBUG: retransmission_handler : request resending (t=fc5c9910, > NOTIFY si ... ) > 1(5494) DEBUG:tm:set_timer: relative timeout is 2000000 > 1(5494) DEBUG: add_to_tail_of_timer[6]: fc5c9a80 (356600000) > 1(5494) DEBUG: retransmission_handler : done > 1(5494) DEBUG: timer routine:6,tl=fc5c9a80 next=0, timeout=356600000 > 1(5494) DEBUG: retransmission_handler : request resending (t=fc5c9910, > NOTIFY si ... ) > 1(5494) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(5494) DEBUG: add_to_tail_of_timer[7]: fc5c9a80 (360600000) > 1(5494) DEBUG: retransmission_handler : done > > > ******************************************************************************** > Backtrace of the core: > > #0 0xfe7f77bc in shm_dup_subs (subs=0xffbff1b0, to_tag=Cannot access > memory > at address 0x96 > ) at notify.c:1819 > 1819 cb_param->wi_subs->to_user.s = (char*)cb_param + size; > (gdb) bt > #0 0xfe7f77bc in shm_dup_subs (subs=0xffbff1b0, to_tag=Cannot access > memory > at address 0x96 > ) at notify.c:1819 > #1 0xfe7f8238 in notify (subs=0x1779c8, watcher_subs=0x15, n_body=0x0, > force_null_body=1024) at notify.c:1562 > #2 0xfe7f8f18 in query_db_notify (p_user=0x234, p_domain=0xdff08, > event=0xfe809a28 "presence.winfo", watcher_subs=0xffbff1b0, etag=0x0) > at notify.c:957 > #3 0xfe8028a8 in update_subscription (msg=0x175fd8, subs=0xffbff1b0, > rtag=0xffbff228, to_tag_gen=1) at subscribe.c:476 > #4 0xfe805080 in handle_subscribe (msg=0x175fd8, str1=0xffbff0e0 "", > str2=0x800 <Address 0x800 out of bounds>) at subscribe.c:1252 > #5 0x00018378 in do_action (a=0x145270, msg=0x175fd8) at action.c:883 > #6 0x00019bb8 in run_action_list (a=0x145270, msg=0x175fd8) at > action.c:131 > #7 0x000192bc in do_action () > #8 0x00019bb8 in run_action_list (a=0x1452c8, msg=0x175fd8) at > action.c:131 > #9 0x00019df8 in run_top_route (a=0x143e10, msg=0x175fd8) at action.c:111 > #10 0x00043b60 in receive_msg (buf=0x12dc00 "", len=1024, > rcv_info=0xe0400) > at receive.c:156 > #11 0x0006f294 in udp_rcv_loop () at udp_server.c:465 > #12 0x000338bc in main_loop () at main.c:834 > #13 0x00035c44 in main (argc=9, argv=0xe7800) at main.c:1399 > (gdb) quit >

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Re: [Serusers] Voicemail setup on SER v0.9.6
by roayu 12 Nov '07

12 Nov '07

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 >

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[OpenSER-Users] MediaProxy 1.9.0 - Radius
by Mike O'Connor 11 Nov '07

11 Nov '07

Hi Guys I really think there is something wrong with MediaProxy, I can not get the radius or the mysql accounting to report any details, as best as I can see there is never any attempt to even try. There is nothing in the logs about there being anything wrong at all. I had added print statement thought out the code and I can see that the code for both mysql update and the code for sending radius updates never gets call. I have not been able to find the connection between the radius or mysql thread and the rtphandler but the print statement a in the actual calls to the mysql or radius and the print messages are never seen in the logs. Other prints added by me do get logged. Has any one actual tried the current code and got do the stop updates ? Mike

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[OpenSER-Users] Openser bridging
by Max Lock 11 Nov '07

11 Nov '07

Hi Folks, I'm struggling with bridging openser between a public and a private network. The private 10.10.12.0/24 network connects openser and an asterisk server that plays a file and hangs up. I've got 2 sip clients connected to the public and private sides of the openser server. When I make calls via openser to the asterisk box, I only get audio on the private networks sip client, and neither client gets hung up correctly at the end of the call. I've put the SIP debug from the external client here: http://pastebin.com/m388a707d My openser and rtpproxy configs are here: http://pastebin.com/m3ed309d3 Can anyone give me any pointers? I've been staring at this for days :( -Cheers Max

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[OpenSER-Users] Regarding the LCR module and AVPs...
by Kyle Stone 09 Nov '07

09 Nov '07

I'm using 1.2.X and the LCR module... following syntax fails at start up... $avp(i:20) = 1; if(!load_gw( $avp(i:20) ) ) { with an invalid arg to load_gw... anyone have a suggestion?

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Re: [OpenSER-Users] lack of notify, subscribe problem
by Anca Vamanu 09 Nov '07

09 Nov '07

Stanisław Pitucha wrote: > ----- "Anca Vamanu" <anca(a)voice-system.ro> wrote: > >> I am not sure if snom 190 supports server presence. I've searched a >> bit >> but did not find sufficient info. >> Does the phone send other Subscribe messages apart from the ones with >> >> event: dialog? Does it send any Publish message? >> > > It has "Publish Presence" option turned on. It has also line monitoring with lights. I've found this site with example notify packets: > http://www.abptech.com/support/faqs/faq_snomphone_LED_explained.html > I have both support broken registrar on and filtering off, whatever that means for snom phone. That means it supports something. > Still - it subscribes for dialog properly (as far as I can tell from rfc4235), so I don't understand the bad event response... Or maybe openser just doesn't support 'dialog' events? > Yes, as I have mentioned before, OpenSer does not support 'dialog' event. It has support only for dialog with sla parameter (Event: dialog;sla) , used for BLA implementation (draft draft-anil-sipping-bla-03.txt). Please send a ngrep capture with a Subscribe message with Event: presence , its reply and the Notify that follows. Anca

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[Serusers] proxy_authorize and radius_proxy_authorize doesn't work for ACK
by Tomasz Zieleniewski 09 Nov '07

09 Nov '07

Hi, When I invoke proxy_authorize and radius_proxy_authorize for an ACK message the is no uid avp present. All other messages which are authorized work fine. Why is that so?? Is it the right behaviour?? Bests regards Tomasz

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Re: [Serusers] proxy_authorize and radius_proxy_authorize doesn't work for ACK
by Jiri Kuthan 09 Nov '07

09 Nov '07

At 17:49 08/11/2007, Tomasz Zieleniewski wrote: >Hi Piotr:) > >RFC only implies that proxy should not challenge the ACK message and >moreover it says that client sending ACK should duplicate >Authorization and Proxy-Authorization headers. This it what my client >does. But according to the description of the proxy_authorize function >it only verifies that credentials are valid. This function doesn't >cause the challenge response to be sent to the client. This is >performed with the usage of proxy_challenge function. >"The function verifies credentials according to RFC2617. If the >credentials are verified successfully then the function will succeed >and mark the credentials as authorized (marked credentials can be >later used by some other functions). If the function was unable to >verify the credentials for some reason then it will fail and the >script should call proxy_challenge which will challenge the user >again." >In my opinion, that is why proxy_authorize function should generate >the uid avp for ACK request if verification of credential gives >positive result. Well -- it should be stored somewhere in transaction state and someway I guess you should be able to get access to it (not sure if with or without some hack) ... but do you relaly need that? ACK has a transport function and as such, I don't see the use value for doing more processing with it than abosrbing/forwarding... -jiri -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] proxy_authorize and radius_proxy_authorize doesn't work for ACK
by Jiri Kuthan 09 Nov '07

09 Nov '07

At 15:53 08/11/2007, Tomasz Zieleniewski wrote: >Hi, > >When I invoke proxy_authorize and radius_proxy_authorize for an ACK >message the is no uid avp present. >All other messages which are authorized work fine. >Why is that so?? >Is it the right behaviour?? yes. ACk has only a transport function and does not really impact SIP processing. -jiri >Bests regards >Tomasz >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Serusers] textops module - are the inserted hfs visible after insertion in the same route block?
by Tomasz Zieleniewski 09 Nov '07

09 Nov '07

Hi, I have the following problem after I insert a partical hf value woth the usage of the textops module (insert_attr_hf) the is_present_hf function returns negative result. is this correct behaviour?? Regards Tomasz

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Re: [OpenSER-Users] lack of notify, subscribe problem
by Stanisław Pitucha 09 Nov '07

09 Nov '07

----- "Anca Vamanu" <anca(a)voice-system.ro> wrote: > If you do not use an xcap server, you should set the force_active > presence module parameter in 1.2.x and presence_xml module parameter > in > 1.3 to positive value. > More info here: > http://www.openser.org/dokuwiki/doku.php/presence:module-exports I'm using force-active=1 - still, no NOTIFY information is sent. Relevant parts from config: modparam("presence|pua", "db_url", "mysql://openserrw:openserrw@192.168.0.134/openser") modparam("presence", "server_address", "sip:192.168.0.134:9876") modparam("presence", "fallback2db", 1) modparam("presence_xml", "db_url", "mysql://openserrw:openserrw@192.168.0.134/openser") modparam("presence_xml", "force_active", 1) modparam("presence_xml", "xcap_table", "xcap_xml") modparam("pua_usrloc", "default_domain", "XXX") in route: handle_publish, handle_subscribe on is_method("PUBLISH") and is_method("SUBSCRIBE") (looks mostly like http://www.openxcap.org/wiki/Installation, but I'm not using openxcap) Anything else I should have added there? > > 2. Phone issue. Trying to start subscribe on snom 190 - messages > There was an wrong match with the dialog;sla event that version 1.3 > has support for. It is fixed now. Thanks. With last svn update it stops after subscribe, I don't even get 202 OK like before. Messages from Snom 190: SUBSCRIBE sip:AAA@XXX;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.112:2078;branch=z9hG4bK-cilxh214xgmc;rport From: <sip:BBB@XXX>;tag=bhscby89jz To: <sip:AAA@XXX;user=phone> Call-ID: 3c26700c5573-eye598917qwu@snom190 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Contact: <sip:BBB@192.168.0.112:2078;line=g4ev9jb3> Event: dialog Accept: application/dialog-info+xml Expires: 3600 Content-Length: 0 It gets "489 bad event" response. >From openser output: ERROR:presence:handle_subscribe: Missing or unsupported event header field value ERROR:presence:handle_subscribe: event= dialog Thanks.

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Re: [OpenSER-Users] Presence module - NOTIFY not sent after un-SUBSCRIBE
by Pascal Maugeri 09 Nov '07

09 Nov '07

OK I got it now working after the update you recommended it. When I un-subscribe I get now the final notify with: Subscription-State: terminated;reason=timeout Thank you very much to everybody for your help and prompt responses. Regards, Pascal On Nov 9, 2007 11:19 AM, Anca Vamanu <anca(a)voice-system.ro> wrote: > > You have to take the latest svn repository for the 1.2.x branch: > > # svn co > https://openser.svn.sourceforge.net/svnroot/openser/branches/1.2 openser > > More info: http://www.openser.org/mos/view/Download/. > > Anca > > > Pascal Maugeri wrote: > > Do you mean I have to update again with 1.2.2 or should I update to "trunk" ? > > > > Thanks > > Pascal > > > > On Nov 9, 2007 11:00 AM, Anca Vamanu <anca(a)voice-system.ro> wrote: > > > >> Hello Pascal , > >> > >> Indeed the 1.2.x version did not sent Notify when unSubscribe was > >> received. It has been fixed in trunk but a back port was forgotten. > >> It was fixed with the latest commit. > >> > >> Thanks and regards, > >> Anca Vamanu > >> > >> > >> Pascal Maugeri wrote: > >> > >>> Hi > >>> > >>> I've update to 1.2.2 > >>> (http://openser.svn.sourceforge.net/svnroot/openser/tags/1.2.2/) and I > >>> get the same problem. > >>> > >>> This is the log of openser I get when I try an unsubscribe: > >>> > >>> 6(18125) PRESENCE:update_subscription: querying database > >>> 6(18125) PRESENCE:update_subscription: expires =0 -> deleting from database > >>> 6(18125) parse_headers: flags=ffffffffffffffff > >>> 6(18125) check_via_address(10.1.1.7, 10.1.1.7, 0) > >>> 6(18125) PRESENCE: get_subs_dialog:The query for subscribtion for > >>> [user]= pascal,[domain]= mydomain.net for [event]= presence.winfo > >>> returned no result > >>> 6(18125) PRESENCE:query_db_notify: Could not get subs_dialog from database > >>> 6(18125) PRESENCE:update_subscription:Could not send notify for presence.winfo > >>> 6(18125) DEBUG: cleanup_uac_timers: RETR/FR timers reset > >>> 6(18125) DEBUG: add_to_tail_of_timer[2]: 0xb60af2b8 (414) > >>> 6(18125) DEBUG:tm:UNREF_UNSAFE: after is 0 > >>> 6(18125) DEBUG:destroy_avp_list: destroying list (nil) > >>> 6(18125) receive_msg: cleaning up > >>> 13(18132) DEBUG: timer routine:2,tl=0xb60af2b8 next=(nil), timeout=414 > >>> 13(18132) DEBUG: wait_handler : removing 0xb60af270 from table > >>> 13(18132) DEBUG: delete transaction 0xb60af270 > >>> 13(18132) DEBUG: wait_handler : done > >>> > >>> It looks there is something wrong with my configuration > >>> "query_db_notify: Could not get subs_dialog from database", doesn't it > >>> ? > >>> > >>> FYI, subscription and notifications work OK. > >>> > >>> Regards > >>> Pascal > >>> > >>> > >>> On Nov 8, 2007 9:40 AM, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > >>> > >>> > >>>> Hi Pascal, > >>>> > >>>> be sure you are using the latest version for 1.2 - try using the SVN > >>>> snapshots for 1.2.2 > >>>> > >>>> Regards, > >>>> Bogdan > >>>> > >>>> > >>>> Pascal Maugeri wrote: > >>>> > >>>> > >>>>> Hi > >>>>> > >>>>> I'm testing the presence module (OpenSER 1.2) and noticed that there > >>>>> is no final NOTIFY request sent when I send a SUBSCRIBE to > >>>>> un-subscribe from a subscription. > >>>>> > >>>>> Is there anything to be configured ? > >>>>> > >>>>> Regards, > >>>>> Pascal > >>>>> > >>>>> > >>>>> _______________________________________________ > >>>>> 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 > >>> > >>> > >>> > >> > > > > > >

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Re: [OpenSER-Users] Storing SIP Response Codes with avpops?
by Douglas Garstang 09 Nov '07

09 Nov '07

Daniel, I found that I was able to do this, and openser accepts it, and inserts a database record... avp_db_query("insert into sip_responses (response_code, response_reason) values ('$rs','abc')"); That is in a failure block. However, in the failure block, $rs is 0. I tried putting the avp_db_query into the reply block, and openser tells me that avp_db_query isn't allowed in the reply block, SO I then tried putting this: avp_write("$rs","$avp(result)"); into the reply block, and openser tells me that avp_write can't be used in a reply block. Argh! I just want to get a sip response code for the call and put it into the database. I'm using version 1.1. Douglas. ----- Original Message ---- From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> To: Douglas Garstang <dougmig33(a)yahoo.com> Cc: users(a)lists.openser.org Sent: Thursday, November 8, 2007 1:30:50 PM Subject: Re: [OpenSER-Users] Storing SIP Response Codes with avpops? What version are you using? I just tested with trunk version and all ok? For the other issue, just set: modparam("avpops", "avp_table", "usr_preferences") Daniel On 11/08/07 23:20, Douglas Garstang wrote: > Thanks, but when I try that, I get the following error from openser: > > Nov 7 23:17:32 localhost openser[7035]: ERROR:xl_parse_spec: bad pvar > name "rr{s.escape.common})') " > Nov 7 23:17:32 localhost openser[7035]: > ERROR:avpops:fixup_db_query_avp: wrong format[insert into > sip_responses (response_code, response_reason) values ('$rs', > '$(rr{s.escape.common})') ] > Nov 7 23:17:32 localhost openser[7035]: ERROR: fix_actions: fixing > failed (code=-1) at cfg line 139 > > OpenSER also complains that i need to specify an avp table, but since > I am specifying the table in the query, I don't understand why that's > required. > > Douglas. > > > ----- Original Message ---- > From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> > To: Douglas Garstang <dougmig33(a)yahoo.com> > Cc: users(a)lists.openser.org > Sent: Thursday, November 8, 2007 11:57:35 AM > Subject: Re: [OpenSER-Users] Storing SIP Response Codes with avpops? > > Hello, > > On 11/08/07 21:40, Douglas Garstang wrote: > > All, > > > > I'd like to store SIP response codes in mysql using avpops. Is this > > possible? > > > > I tried to do this a few years ago and couldn't find any > > documentation. Unfortunately, I still can't find any. > > > > Can someone post an example? All I can find online is reference > > documentation for avpops. > > > > I'd really like to see a full example with the modparams defining the > > table structure, and how I would use avp_write and avp_db_store to > > write the $rr response code. > the easiest way is to use avp_db_query(), you can give as parameter any > sql query. In your case, you can use in onreply_route: > > avp_db_query("insert into sip_responses (response_code, response_reason) > values ('$rs', '$(rr{s.escape.common})') "); > > You need to create the table sip_responses, with the two string columns. > > Daniel > > > > > Thanks. > > Douglas. > > > > > > __________________________________________________ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam protection around > > http://mail.yahoo.com > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[OpenSER-Users] [Fwd: Re: Presence module - NOTIFY not sent after un-SUBSCRIBE]
by Anca Vamanu 09 Nov '07

09 Nov '07

-------- Original Message -------- Subject: Re: [OpenSER-Users] Presence module - NOTIFY not sent after un-SUBSCRIBE Date: Fri, 09 Nov 2007 12:19:57 +0200 From: Anca Vamanu <anca(a)voice-system.ro> To: Pascal Maugeri <pascal.maugeri1(a)gmail.com> References: <990aed650711050832u17bcebeey5fea7e4b593ebf2b(a)mail.gmail.com> <990aed650711071336i16268d29v7826e6c40ec580dd(a)mail.gmail.com> <4732CB99.2020104(a)voice-system.ro> <990aed650711090100l16fc5cd2h32c94ad0504ac12a(a)mail.gmail.com> <47342FBF.7060208(a)voice-system.ro> <990aed650711090210q4112985bt16cbbd28255cdd5f(a)mail.gmail.com> You have to take the latest svn repository for the 1.2.x branch: # svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.2 openser More info: http://www.openser.org/mos/view/Download/. Anca Pascal Maugeri wrote: > Do you mean I have to update again with 1.2.2 or should I update to "trunk" ? > > Thanks > Pascal > > On Nov 9, 2007 11:00 AM, Anca Vamanu <anca(a)voice-system.ro> wrote: > >> Hello Pascal , >> >> Indeed the 1.2.x version did not sent Notify when unSubscribe was >> received. It has been fixed in trunk but a back port was forgotten. >> It was fixed with the latest commit. >> >> Thanks and regards, >> Anca Vamanu >> >> >> Pascal Maugeri wrote: >> >>> Hi >>> >>> I've update to 1.2.2 >>> (http://openser.svn.sourceforge.net/svnroot/openser/tags/1.2.2/) and I >>> get the same problem. >>> >>> This is the log of openser I get when I try an unsubscribe: >>> >>> 6(18125) PRESENCE:update_subscription: querying database >>> 6(18125) PRESENCE:update_subscription: expires =0 -> deleting from database >>> 6(18125) parse_headers: flags=ffffffffffffffff >>> 6(18125) check_via_address(10.1.1.7, 10.1.1.7, 0) >>> 6(18125) PRESENCE: get_subs_dialog:The query for subscribtion for >>> [user]= pascal,[domain]= mydomain.net for [event]= presence.winfo >>> returned no result >>> 6(18125) PRESENCE:query_db_notify: Could not get subs_dialog from database >>> 6(18125) PRESENCE:update_subscription:Could not send notify for presence.winfo >>> 6(18125) DEBUG: cleanup_uac_timers: RETR/FR timers reset >>> 6(18125) DEBUG: add_to_tail_of_timer[2]: 0xb60af2b8 (414) >>> 6(18125) DEBUG:tm:UNREF_UNSAFE: after is 0 >>> 6(18125) DEBUG:destroy_avp_list: destroying list (nil) >>> 6(18125) receive_msg: cleaning up >>> 13(18132) DEBUG: timer routine:2,tl=0xb60af2b8 next=(nil), timeout=414 >>> 13(18132) DEBUG: wait_handler : removing 0xb60af270 from table >>> 13(18132) DEBUG: delete transaction 0xb60af270 >>> 13(18132) DEBUG: wait_handler : done >>> >>> It looks there is something wrong with my configuration >>> "query_db_notify: Could not get subs_dialog from database", doesn't it >>> ? >>> >>> FYI, subscription and notifications work OK. >>> >>> Regards >>> Pascal >>> >>> >>> On Nov 8, 2007 9:40 AM, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: >>> >>> >>>> Hi Pascal, >>>> >>>> be sure you are using the latest version for 1.2 - try using the SVN >>>> snapshots for 1.2.2 >>>> >>>> Regards, >>>> Bogdan >>>> >>>> >>>> Pascal Maugeri wrote: >>>> >>>> >>>>> Hi >>>>> >>>>> I'm testing the presence module (OpenSER 1.2) and noticed that there >>>>> is no final NOTIFY request sent when I send a SUBSCRIBE to >>>>> un-subscribe from a subscription. >>>>> >>>>> Is there anything to be configured ? >>>>> >>>>> Regards, >>>>> Pascal >>>>> >>>>> >>>>> _______________________________________________ >>>>> 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 >>> >>> >>> >> > >

1 0

[OpenSER-Users] TLS Registrations start dropping?
by Chris Heiser 09 Nov '07

09 Nov '07

Information on TLS and OpenSER with this specific issues seems to be scarce. I have 2 different setups. One with about 130 registrations, one with about 250. Both with SSLv23 turned on, and TLS is enabled in the phones. What's odd is after some unknown amount of time, phones start dropping like files and fail to re-register. (Our registration timer is set at 60 seconds to aid in failover). What's interesting is it appears that the SSL handshake completely fails when this starts to happen. The requests from the phones are smaller in size (733 bytes about 970 for a good one). Suspicion would normally lie with the phones. They're obviously not doing something right! Here's the kicker. If I fail over to another OpenSER proxy (we move the IP over, use the same certs, etc...) everyone comes back up, and everything is happy for hours, maybe a day, maybe a week, and then it all starts going downhill again. I've increased the number of OpenSER children, I've played with the tcp_persistent_flag. None of these things have stopped the madness. Anyone have any ideas or thoughts? --Chris

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[OpenSER-Users] Presence module - NOTIFY not sent after un-SUBSCRIBE
by Pascal Maugeri 09 Nov '07

09 Nov '07

Hi I'm testing the presence module (OpenSER 1.2) and noticed that there is no final NOTIFY request sent when I send a SUBSCRIBE to un-subscribe from a subscription. Is there anything to be configured ? Regards, Pascal

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[OpenSER-Users] registrations not saved in the mysql database
by Ovidiu Sas 09 Nov '07

09 Nov '07

Hi all, I was trying to keep persistent registrations in the mysql db, but it seems that something is broken in the latest svn: version: openser 1.3.0-pre1-notls (i386/linux) flags: STATS: Off, EXTRA_DEBUG, 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: 2:3034M @(#) $Id: main.c 3015 2007-11-05 13:08:50Z henningw $ main.c compiled on 11:41:03 Nov 6 2007 with gcc 4.1.2 The config that I used for testing is pretty simple: # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=1 disable_tcp=yes # ------------------ module loading ---------------------------------- mpath="/usr/local/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") loadmodule "usrloc.so" modparam("usrloc", "db_url", "mysql://openser:openserrw@127.0.0.1/openser") modparam("usrloc", "db_mode", 1) loadmodule "registrar.so" #------------------------- request routing logic ------------------- route{ if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { save("location"); exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } Here are the relevant debug logs: Nov 6 12:19:04 localhost /usr/local/sbin/openser[8963]: core:db_double2str: Error in snprintf Nov 6 12:19:04 localhost /usr/local/sbin/openser[8963]: mysql:val2str: error while converting string to double Nov 6 12:19:04 localhost /usr/local/sbin/openser[8963]: core:db_print_values: Error while converting value to string Nov 6 12:19:04 localhost /usr/local/sbin/openser[8963]: usrloc:db_insert_ucontact: inserting contact in db failed Nov 6 12:19:04 localhost /usr/local/sbin/openser[8963]: usrloc:insert_ucontact: failed to insert in database I can see the in memory registration via `openserctl ul show' command, but the mysql database is not updated (verified via `openserctl db show location') Regards, Ovidiu Sas

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Re: [OpenSER-Users] CPL "Location" and "Proxy" behavior
by Bogdan-Andrei Iancu 09 Nov '07

09 Nov '07

Hi Daniel, Could please check the latest SVN version of 1.3 - I made a fix there. If everything ok, I will backport it to 1.2 also. Regards, Bogdan Daniel Grotti wrote: > *Hi Bogdan,* >> Hi Daniel, >> >> Hmmm....I see - you say (for outgoing scenario) that the location set >> is initialized with the A destination instead of B destination, right? > *yes, it's right.* >> If so, please open a bug report on the tracker. > *well, on the bug Repert, if I don't set any location (as in the last > script I have sended), I can see this DEBUG line: > " DEBUG:cpl_proxy_to_loc_set: rewriting Request-URI with < > A@A_ip_address>" > > where : "A@A_ip_address" is the caller IP address, not the destination > address (that is B). > * > > *regards. > daniel > * >> regards, >> Bogdan >> > > > >> Daniel Grotti wrote: >>> Hi bogdan, >>> If it's my cpl outgoing script configuration, from USER A >>> ****************************************************** >>> <cpl> >>> <outgoing> >>> <address-switch field="destination" subfield="user"> >>> <address is="B"> >>> <proxy /> >>> </address> >>> </address-switch> >>> </outgoing> >>> <cpl> >>> ******************************************************* >>> When I make a call from A to B, then the script has called, but in >>> debug I can see that CPL rewrites R-URI with source address >>> (A@A_ip_address) instead of destination address (B@B_ip_address). So >>> the message goes back to user A. >>> >>> regards, >>> Daniel >>> >>> Bogdan-Andrei Iancu wrote: >>>> Hi Daniel, >>>> >>>> there are no differences - the implementation follows the RFC. For >>>> incoming calls, the location set is empty, but it can be populated >>>> by the CPL script during execution. So, when you do "proxy", you >>>> may have multiple entries in the location set. >>>> >>>> The comment in the cpl_sig.c file refers to some implementation >>>> details - how to process the available location set, depeding if it >>>> is the first time doing proxy (for the call) or a second time. >>>> >>>> regards, >>>> bogdan >>>> >>>> daniel grotti wrote: >>>>> Hi all, >>>>> this is what CPL_SIG.c says about forwarding a given message to >>>>> the given location set: >>>>> >>>>> " forwards the msg to the given location set; if flags has set the >>>>> CPL_PROXY_DONE, all locations will be added as branches, >>>>> otherwise, the first one will set as RURI (this is ha case when >>>>> this is the first proxy of the message) >>>>> The given list of location will be freed, returning 0 instead. >>>>> Returns: 0 - OK " >>>>> >>>>> >>>>> My question is : which and what is "the first one" location ? Can >>>>> you explain it better? >>>>> Because RFC 3880 say that : "For the incoming top-level call >>>>> processing action, the location set is initialized to the empty set. >>>>> For the outgoing action, it is initialized to the destination >>>>> address of the call." >>>>> >>>>> So RFC is very clear. I think there are considerable differences >>>>> between RFC and the behavior of CPL interpreter. >>>>> >>>>> Thanks. >>>>> Daniel >>>>> >> >> >

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[OpenSER-Users] CPL processing in OpenSER
by Chris Heiser 09 Nov '07

09 Nov '07

So voip-info.org's wiki page on CPL-C notest that the XML processing is resource consuming and thus common features can be implemented in other OpenSER modules. I was under the impression that the upload of a CPL script into openser caused a binary compile of the script that gets used at runtime to eliminate the terrible overhead of libxml, etc... Or am I completely off base? --Chris

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[OpenSER-Users] lack of notify, subscribe problem
by Stanisław Pitucha 09 Nov '07

09 Nov '07

Hey I've got 2 issues with pub/sub: 1. Openser does correct subscription on grandstream 2020 phone. But then anything happens: register, invite, whatever from monitored phone, openser doesn't generate any notifications for subscribers. Any pointers on it? Is there some variable, that I should set to enable those? 2. Phone issue. Trying to start subscribe on snom 190 - messages sent are: ----- SUBSCRIBE ... ... Event: dialog Accept: application/dialog-info+xml ----- 202 OK ----- NOTIFY ... Event: dialog;sla Subscription-State: active;expires=3600\r\n ----- 489 Bad Event ----- Anything I can do about it? I've tried it with svn version with the same effect. Couldn't really find anything in docs that I could change in config. pub/sub is handled with simple handle_publish/handle_subscribe - nothing fancy in config - just location storing on register and message passing. Thanks for all info.

2 1

[OpenSER-Users] UAC_REPLACE_FROM function
by Allan Chao ( 趙廷睿 ) 09 Nov '07

09 Nov '07

Dear All: I want to use uac_replace_from function to change from header username, it will be successed in the first time , but when i use in the second time , the from header username become <username1 username2> . but i want it become <username2> , how can i solve it , thanks. allan

2 1

[OpenSER-Users] cdrtool radus openser.cfg configuration
by ayeddjalil 09 Nov '07

09 Nov '07

Hi every one It's my first time there, I am trying to install cdrtool, I installed openser with radius and mysql accounting, I follow the install file with cdrtool, but it seam it's not very clear I added these line in my openser.cfg as they said in the install file 268 loadmodule "acc.so" 269 270 modparam("acc", "failed_transaction_flag", 1) 271 modparam("acc", "report_cancels", 0) 272 modparam("acc", "report_ack", 0) 273 modparam("acc", "early_media", 0) 274 275 modparam("acc", "log_level", 1) 276 modparam("acc", "log_flag", 1) 277 modparam("acc", "log_missed_flag", 1) 278 279 modparam("acc", "radius_config", "/etc/openser/radius/client.conf") 280 modparam("acc", "radius_flag", 1) 281 modparam("acc", "radius_missed_flag", 1) 282 modparam("acc", "radius_extra", "Sip-RPid=$avp(s:rpid); \ 283 Source-IP=$si; \ 284 Source-Port=$sp; \ 285 Canonical-URI=$avp(can_uri); \ 286 Billing-Party=$avp(billing_party); \ 287 Divert-Reason=$avp(s:divert_reason); \ 288 X-RTP-Stat=$avp(s:rtp_statistics); \ 289 From-Header=$hdr(from); \ 290 User-Agent=$hdr(user-agent); \ 291 Contact=$hdr(contact); \ 292 Event=$hdr(event); \ 293 SIP-Proxy-IP=$avp(s:sip_proxy_ip)") alone opense do not start intill I added this loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "avp_radius.so" modparam("acc", "service_type", 15) modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("acc|auth_radius|group_radius|avp_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") the next problem is when adding setflag(ACCOUNTING_FLAG); avp_write("SER_IP","$avp(s:sip-proxy)"); avp_write("$ru", "$avp(can_uri)"); errors comes, here what I get from the command /sbin/openser -c 0(22857) parse error (123,9-24): syntax error 0(22857) parse error (123,9-24): missing '(' or ')'? 0(22857) parse error (123,9-24): bad command: missing ';'? 0(22857) parse error (123,24-25): bad command!) 0(22857) parse error (124,24-25): unknown command, missing loadmodule? 0(22857) parse error (125,25-26): unknown command, missing loadmodule? ERROR: bad config file (6 errors) 0(22857) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was not allocated -> nothing to destroy is thier any one has a sample of openser.cfg file for cdrtool Please help -- View this message in context: http://www.nabble.com/cdrtool-radus-openser.cfg-configuration-tf4776366.htm… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

Re: [Serusers] Spam Re: Voicemail setup on SER v0.9.6
by roayu 09 Nov '07

09 Nov '07

Hi Mashudi, Thanks for your reply. From the documentation, it seems that I need to install both SER and SEMS under the same directory. 1) I've already installed the SER under /usr/src, can I also install SEMS under /usr/src instead of /usr/local/src ? 2) By the way, I would like to know that does this voicemail able to trigger the voicemail on the IP hard phone? For instance, if there's a new voicemail, can the IP hard phone got the signal of the new voicemail? Thanks. Regards, Roa Yu -----Original Message----- From: mashudi [mailto:mashudiflexi@telkom.co.id] Sent: Saturday, November 10, 2007 2:04 AM To: roayu Subject: Spam Re: [Serusers] Voicemail setup on SER v0.9.6 dear roayu, you can read on http://www.iptel.org/sems/sems_application_development_tutorial just follow the document, you will able run simple voice mail. roayu wrote: > ********************** > Your mail has been scanned by > Telkom Anti Virus System > *********************** > > 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 >

1 0

[Serusers] Voicemail setup on SER v0.9.6
by roayu 09 Nov '07

09 Nov '07

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

1 0

[OpenSER-Users] Forking Madness
by Chris Heiser 09 Nov '07

09 Nov '07

So I've been all over the place looking at various examples, threads, issues, and solutions to all sorts of forking scenarios. What I haven't been able to find is a solution to my odd corner case. Let me explain the forking scenario: Call comes in for user A: - Use avp_db_load to find that user A wants to parallel fork to 123 and 456 - Call to 123 fails because there's no registration - Call to 456 times out == At this point we serial fork to the next step which might hbe - Just call my cell What's unclear to me is how to actually achieve this logic. I mean, getting the initial parallel fork is quite simple using avp_pushto. What's unclear is how to setup the failure route to know that you can ignore a failure in any of the forks until the last fork as failed, and then it's time to move on. Something like, "are there other branches pending?" would be a nice check to determine if the failure route should just drop or if it should append a branch (or two) for the next serial fork. Comments? Ideas? Thoughts? Am I totally insane? Should I be using a specific module for this? --Chris Heiser

4 9

[OpenSER-Users] openser + cdrtool and multiple cdrs
by Morten Isaksen 08 Nov '07

08 Nov '07

Hi! I few times a day I have 2 cdrs for one call but with reverse direction. I have looked at the radius server detail records and as far as I can see this happens: A Acct-Status-Type = Start with Acct-Session-Id = X1 and Sip-To-Tag = X2 and Sip-From-Tag = X3 is logged. I few seconds later Acct-Status-Type = Start with Acct-Session-Id = X1 and Sip-To-Tag = X3 and Sip-From-Tag = X2 is logged. (To and From tag is reversed) When the call is over a Update and a Stop is logged. The Acct-Status-Type is not a unique key in the raddact table and there for I get a CDR for each direction. My theory is that this happens when the server recieves a re-INVITE. Can I prevent this in openser.cfg? Is it safe to add Acct-Session-Id as a unique key to the raddact table? Or is the Session id not globaly unique? -- Morten Isaksen http://www.misak.dk/blog/

1 0

Re: [OpenSER-Users] Storing SIP Response Codes with avpops?
by Douglas Garstang 08 Nov '07

08 Nov '07

Thanks, but when I try that, I get the following error from openser: Nov 7 23:17:32 localhost openser[7035]: ERROR:xl_parse_spec: bad pvar name "rr{s.escape.common})') " Nov 7 23:17:32 localhost openser[7035]: ERROR:avpops:fixup_db_query_avp: wrong format[insert into sip_responses (response_code, response_reason) values ('$rs', '$(rr{s.escape.common})') ] Nov 7 23:17:32 localhost openser[7035]: ERROR: fix_actions: fixing failed (code=-1) at cfg line 139 OpenSER also complains that i need to specify an avp table, but since I am specifying the table in the query, I don't understand why that's required. Douglas. ----- Original Message ---- From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> To: Douglas Garstang <dougmig33(a)yahoo.com> Cc: users(a)lists.openser.org Sent: Thursday, November 8, 2007 11:57:35 AM Subject: Re: [OpenSER-Users] Storing SIP Response Codes with avpops? Hello, On 11/08/07 21:40, Douglas Garstang wrote: > All, > > I'd like to store SIP response codes in mysql using avpops. Is this > possible? > > I tried to do this a few years ago and couldn't find any > documentation. Unfortunately, I still can't find any. > > Can someone post an example? All I can find online is reference > documentation for avpops. > > I'd really like to see a full example with the modparams defining the > table structure, and how I would use avp_write and avp_db_store to > write the $rr response code. the easiest way is to use avp_db_query(), you can give as parameter any sql query. In your case, you can use in onreply_route: avp_db_query("insert into sip_responses (response_code, response_reason) values ('$rs', '$(rr{s.escape.common})') "); You need to create the table sip_responses, with the two string columns. Daniel > > Thanks. > Douglas. > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[OpenSER-Users] Storing SIP Response Codes with avpops?
by Douglas Garstang 08 Nov '07

08 Nov '07

All, I'd like to store SIP response codes in mysql using avpops. Is this possible? I tried to do this a few years ago and couldn't find any documentation. Unfortunately, I still can't find any. Can someone post an example? All I can find online is reference documentation for avpops. I'd really like to see a full example with the modparams defining the table structure, and how I would use avp_write and avp_db_store to write the $rr response code. Thanks. Douglas. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: [OpenSER-Users] OpenSER SIP registrar - third party registration
by Klaus Darilion 08 Nov '07

08 Nov '07

Pascal Maugeri wrote: > Thanks for your prompt answer. > > On 10/19/07, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: >> It depends on what kind of information should be stored in the other >> server. If you "your server" needs to see the same contact information >> as openser received you could use t_replicate to replicate the REGISTER >> to "your server". In this case "your server" must not challenge the >> REGISTER. > > My server does not need to store any information but the information > that a user has been registered. > > I believe the solution I need is to use this t_replicate but I need to > replicate the register once the registration is achieved successfully. > So how can I do this replicate to my server once the complete dialog > REGISTER-401-REGISTER-OK has been achieved between the UAC and OpenSER > ? Just use t_replicate after save(), e.g: http://lists.iptel.org/pipermail/serdev/2006-June/007518.html regards klaus ps: please Cc: the mailing-list

3 2

[OpenSER-Users] [AS WeSIP & OpenIMSCore]
by Alscion MailingList 08 Nov '07

08 Nov '07

Hi, Does anybody successed to trigg the WeSIP AS with Fokus OpenIMSCore? Do you have some informations about that? Thx.

1 0

[OpenSER-Users] X-lite (for Linux) and BYE Scenario
by Daniel Grotti 08 Nov '07

08 Nov '07

Hi All, I've noticed that using X-lite (for Linux) there are some problems with BYE method. When X-lite generates BYE, it puts in the Route Header field the address known through the record-route added by proxy. As "SER - Gettin Started" shown, an address like : IP_proxy:5060;nat=yes: " /.... .... if(method=="INVITE" && client_nat_test("7")) { record_route_preset ("proxy_ip_address :5060;nat=yes"); .... } /" 1. When we have a BYE message sended by X-lite (user A), it is like this: "/Request-Line: BYE sip: B@proxy_ip_address : 5060 SIP/2.0 Via: SIP/2.0/UDP A_address:5061; rport;branch=xyz From: <sip: A@proxy_ip_address:5061>;tag=123 To: <sip: B@proxy_ip_address:5060>;tag=456 Contact:<sip:A@A_address:5061> * Route: <sip: proxy_ip_address:5060; nat=yes; ftag=9098789; lr=on>* .... ..... .... "/ 2. Now the message BYE is parsed by the Proxy, and it uses the loose_route(): /if(loose_route())/ { forward the message }; but Proxy doesn't recognizes its address and then rewrites the R-URI with the address in Route header field and sends the BYE message.....to itself! then Proxy parses the new BYE (sended by itself) with the URI :/* <sip: proxy_ip_address:5060; nat=yes; ftag=9098789; lr=on>,*/ the the answer to user A is : 404 Destination not found! Why doesn't OpenSER, in the loose_route() function, recognize its address in Route header field? This is what should it do. And then, delete the route header field and then forward the message using R-URI. I think there is something wrong.... Regards, Daniel -- --------------------------------------------------- Daniel Grotti DEIS - University of Bologna Via Venezia, 52 - 47023 Cesena (FC) - ITALY Contacts e-mail : daniel.grotti(a)unibo.it Skype name : Daniel Grotti ---------------------------------------------------

1 0

[Serusers] What are the possible HFNAME values in hf selects in ser rel_2_0_0?
by Tomasz Zieleniewski 08 Nov '07

08 Nov '07

Hi, What are the possible HFNAME values in hf selects in ser rel_2_0_0? When I try to add my own header and then use the a select for it the select is empty. I do it for P-Asserted-Identity header I did the following selects and no results: xlog("L_INFO", " route[AUTHENTICATION]: P-ASS: %(a)hf_value.p-asserted-identity \n"); xlog("L_INFO", " route[AUTHENTICATION]: P-ASS: %(a)hf_value.P_Asserted_Identity \n"); xlog("L_INFO", " route[AUTHENTICATION]: P-ASS: %(a)hf_value.p_asserted_identity \n"); xlog("L_INFO", " route[AUTHENTICATION]: P-ASS: %(a)hf_value2.p-asserted-identity \n"); xlog("L_INFO", " route[AUTHENTICATION]: P-ASS: %(a)hf_value2.P_Asserted_Identity \n"); xlog("L_INFO", " route[AUTHENTICATION]: P-ASS: %(a)hf_value2.p_asserted_identity \n"); Please point me what do I miss. Regards Tomasz

2 1

[Serusers] (no subject)
by Shad Mortazavi 08 Nov '07

08 Nov '07

Shad Mortazavi UK Managing Director Nexus Management (EMEA) Ltd -------------------- T: +44 (0)844 484 2574 M: +44 (0)781 721 0758 Sent from Windows Mobile.

1 0

Re: [OpenSER-Users] CPL "Location" and "Proxy" behavior
by Bogdan-Andrei Iancu 08 Nov '07

08 Nov '07

Hi Daniel, Hmmm....I see - you say (for outgoing scenario) that the location set is initialized with the A destination instead of B destination, right? If so, please open a bug report on the tracker. regards, Bogdan Daniel Grotti wrote: > Hi bogdan, > If it's my cpl outgoing script configuration, from USER A > ****************************************************** > <cpl> > <outgoing> > <address-switch field="destination" subfield="user"> > <address is="B"> > <proxy /> > </address> > </address-switch> > </outgoing> > <cpl> > ******************************************************* > When I make a call from A to B, then the script has called, but in > debug I can see that CPL rewrites R-URI with source address > (A@A_ip_address) instead of destination address (B@B_ip_address). So > the message goes back to user A. > > regards, > Daniel > > Bogdan-Andrei Iancu wrote: >> Hi Daniel, >> >> there are no differences - the implementation follows the RFC. For >> incoming calls, the location set is empty, but it can be populated by >> the CPL script during execution. So, when you do "proxy", you may >> have multiple entries in the location set. >> >> The comment in the cpl_sig.c file refers to some implementation >> details - how to process the available location set, depeding if it >> is the first time doing proxy (for the call) or a second time. >> >> regards, >> bogdan >> >> daniel grotti wrote: >>> Hi all, >>> this is what CPL_SIG.c says about forwarding a given message to the >>> given location set: >>> >>> " forwards the msg to the given location set; if flags has set the >>> CPL_PROXY_DONE, all locations will be added as branches, >>> otherwise, the first one will set as RURI (this is ha case when >>> this is the first proxy of the message) >>> The given list of location will be freed, returning 0 instead. >>> Returns: 0 - OK " >>> >>> >>> My question is : which and what is "the first one" location ? Can >>> you explain it better? >>> Because RFC 3880 say that : "For the incoming top-level call >>> processing action, the location set is initialized to the empty set. >>> For the outgoing action, it is initialized to the destination >>> address of the call." >>> >>> So RFC is very clear. I think there are considerable differences >>> between RFC and the behavior of CPL interpreter. >>> >>> Thanks. >>> Daniel >>>

1 0

[OpenSER-Users] CPL "Location" and "Proxy" behavior
by daniel grotti 08 Nov '07

08 Nov '07

Hi all, this is what CPL_SIG.c says about forwarding a given message to the given location set: " forwards the msg to the given location set; if flags has set the CPL_PROXY_DONE, all locations will be added as branches, otherwise, the first one will set as RURI (this is ha case when this is the first proxy of the message) The given list of location will be freed, returning 0 instead. Returns: 0 - OK " My question is : which and what is "the first one" location ? Can you explain it better? Because RFC 3880 say that : "For the incoming top-level call processing action, the location set is initialized to the empty set. For the outgoing action, it is initialized to the destination address of the call." So RFC is very clear. I think there are considerable differences between RFC and the behavior of CPL interpreter. Thanks. Daniel ----------------------------- Daniel Grotti DEIS - Universita' di Bologna ----------------------------- Via Venezia, 52 47023 Cesena (FC) - ITALY ----------------------------- email:daniel.grotti@unibo.it -----------------------------

2 1

[Serusers] Openser bridging
by Max Lock 08 Nov '07

08 Nov '07

Hi Folks, I'm still struggling with bridging openser between a public and a private network. The private 10.10.12.0/24 network connects openser and an asterisk server that plays a file and hangs up. I've got 2 sip clients connected to the public and private sides of the openser server. When I make calls via openser to the asterisk box, I only get audio on the private networks sip client, and neither client gets hung up correctly at the end of the call. I've put the SIP debug from the external client here: http://pastebin.com/m388a707d My openser and rtpproxy configs are here: http://pastebin.com/m3ed309d3 Can anyone give me any pointers? I've been staring at this for days :( -Cheers Max

2 1

[Serusers] Integrate LCS & SER Server
by elilkumaran s 07 Nov '07

07 Nov '07

Sir/Madam I'm a System administrator of Institute. I use MS LCS 2005 SP1 Enterprise edition & Office Communicator in my work place. I have linux SER Server & CISCO ATA 186 Telephone adapter. I have to run to Messenger client X-Lite for SER Server and Office Communicator for LCS. Because I place TP in Guard Office they can call Staff through SER server / X-Lite. But can't call Office Communicator! CISCO ATA 186 can't registering in LCS, I thing it not possible directly. ( Because LCS use Active Directory Account through Kerbas & NTLM authentication). So I want integrate LCS & SER Server to facilitate TP call to Office Communicator & Office Communicator call TP. I use a pool in LCS as normal TCP (not TLS). SER Server also use normal (with out DB -Any one can registrar with any name) Could you kindly tell the process for integrate LCS & SER Server? Thank You Your truly Elil

2 1

[OpenSER-Users] Core dumped when deploying presence module
by Sergio Gutierrez 07 Nov '07

07 Nov '07

Hello to all members. Currently we are trying to deploy presence module in the following environment: 1. Openser 1.2.1 2. Solaris 10 on Sparc System 3. libxml 2.6.30 Openser starts up fine, but when happens any event as a Status change of a user agent, Openser crashes. Below I send an extract from the logfile, and the backtrace of the core. Thanks in advance for your attention. Sergio Gutiérrez EPM Telecomunicaciones Medellin - Colombia **************************************************** >From Openser logfile. 0(5493) Nuevo Requerimiento - M=SUBSCRIBE RURI=sip:5192901@200.13.225.250F= sip:5192902@200.13.225.250 T=sip:5192901@200.13.225.250 IP=200.116.28.88ID=YmE2MGVhMGQzMTI3ZmU0YTkzMDJjNjIxODNmMGIwOWE. 0(5493) comp_scriptvar: str 20 : 5192901 0(5493) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(5493) parse_headers: flags=ffffffffffffffff 0(5493) get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> 0(5493) DEBUG: get_hdr_body : content_length=0 0(5493) found end of header 0(5493) PRESENCE: handle_subscribe: 'expires' found 0(5493) PRESENCE: handle_subscribe: lexpire= 3600 0(5493) PRESENCE: handle_subscribe: 'To' header ALREADY PARSED: < sip:5192901@200.13.225.250> 0(5493) PRESENCE:handle_subscribe: generating to_tag 0(5493) [p_user]= 5192901 [p_domain]= 200.13.225.250 [w_user]= 5192902 [w_domain]= 200.13.225.250 0(5493) [event]= presence [staus]= pending [expires]= 3600 0(5493) [to_tag]= 10.5493.1194034490.40 [from_tag]= 1f783845 0(5493) PRESENCE: update_subscription ... 0(5493) [p_user]= 5192901 [p_domain]= 200.13.225.250 [w_user]= 5192902 [w_domain]= 200.13.225.250 0(5493) [event]= presence [staus]= pending [expires]= 3600 0(5493) [to_tag]= 10.5493.1194034490.40 [from_tag]= 1f783845 0(5493) expires: 3600 0(5493) PRESENCE:update_subscription:Inserting into database: n_query_cols:14 0(5493) [0] = to_user 5192901 0(5493) [1] = to_domain 200.13.225.250 0(5493) [2] = from_user 5192902 0(5493) [3] = from_domain 200.13.225.250 0(5493) [4] = event presence 0(5493) [5] = event_id 0(5493) [6] = callid YmE2MGVhMGQzMTI3ZmU0YTkzMDJjNjIxODNmMGIwOWE. 0(5493) [7] = to_tag 10.5493.1194034490.40 0(5493) [8] = from_tag 1f783845 0(5493) [9] = contact sip:5192902@200.116.28.88:47166 0(5493) [10] = status pending 0(5493) [11] = cseq 1 0(5493) [12] = expires 1194038090 0(5493) [13] = version 0 0(5493) parse_headers: flags=ffffffffffffffff 0(5493) check_via_address(200.116.28.88, 192.168.0.13, 0) 0(5493) DBG:sl:run_sl_callbacks: callback id 0 entered 0(5493) trace_sl_onreply_out: trace off... 0(5493) PRESENCE: get_subs_dialog:n= 1 0(5493) PRESENCE:notify:dialog informations: 0(5493) [p_user]= 5192901 [p_domain]= 200.13.225.250 [w_user]= 5192901 [w_domain]= 200.13.225.250 0(5493) [event]= presence.winfo [staus]= active [expires]= 3518 0(5493) [to_tag]= 10.5493.1194034408.34 [from_tag]= 39320458 0(5493) presence:uandd_to_uri: uri=sip:5192901@200.13.225.250 0(5493) presence:uandd_to_uri: uri=sip:5192902@200.13.225.250 0(5493) presence:uandd_to_uri: uri=sip:5192901@200.13.225.250 0(5493) PRESENCE: notify: build notify to user= 5192901 domain= 200.13.225.250 for event= presence.winfo 0(5493) [p_user]= 5192901 [p_domain]= 200.13.225.250 [w_user]= 5192901 [w_domain]= 200.13.225.250 0(5493) [event]= presence.winfo [staus]= active [expires]= 3518 0(5493) [to_tag]= 10.5493.1194034408.34 [from_tag]= 39320458 0(5493) PRESENCE:build_str_hdr: expires = 3518 0(5493) PRESENCE:build_str_hdr: subs_expires : 3518 0(5493) PRESENCE: build_str_hdr: headers: Event: presence.winfo Contact: <sip:200.13.225.250:5060> Subscription-State: active;expires=3518 Content-Type: application/watcherinfo+xml 0(5493) PRESENCE:notify: headers:Event: presence.winfo Contact: <sip:200.13.225.250:5060> Subscription-State: active;expires=3518 Content-Type: application/watcherinfo+xml 0(5493) CONTACT = sip:5192901@200.13.254.180:17116 0(5493) presence:uandd_to_uri: uri=sip:5192901@200.13.225.250 0(5493) parse_rr_body(): No body for record-route 0(5493) PRESENCE: notify:Send notify for presence on callback 1(5494) DEBUG: timer routine:4,tl=fc5c9a80 next=0, timeout=353600000 1(5494) DEBUG: retransmission_handler : request resending (t=fc5c9910, NOTIFY si ... ) 1(5494) DEBUG:tm:set_timer: relative timeout is 1000000 1(5494) DEBUG: add_to_tail_of_timer[5]: fc5c9a80 (354600000) 1(5494) DEBUG: retransmission_handler : done 1(5494) DEBUG: timer routine:5,tl=fc5c9a80 next=0, timeout=354600000 1(5494) DEBUG: retransmission_handler : request resending (t=fc5c9910, NOTIFY si ... ) 1(5494) DEBUG:tm:set_timer: relative timeout is 2000000 1(5494) DEBUG: add_to_tail_of_timer[6]: fc5c9a80 (356600000) 1(5494) DEBUG: retransmission_handler : done 1(5494) DEBUG: timer routine:6,tl=fc5c9a80 next=0, timeout=356600000 1(5494) DEBUG: retransmission_handler : request resending (t=fc5c9910, NOTIFY si ... ) 1(5494) DEBUG:tm:set_timer: relative timeout is 4000000 1(5494) DEBUG: add_to_tail_of_timer[7]: fc5c9a80 (360600000) 1(5494) DEBUG: retransmission_handler : done ******************************************************************************** Backtrace of the core: #0 0xfe7f77bc in shm_dup_subs (subs=0xffbff1b0, to_tag=Cannot access memory at address 0x96 ) at notify.c:1819 1819 cb_param->wi_subs->to_user.s = (char*)cb_param + size; (gdb) bt #0 0xfe7f77bc in shm_dup_subs (subs=0xffbff1b0, to_tag=Cannot access memory at address 0x96 ) at notify.c:1819 #1 0xfe7f8238 in notify (subs=0x1779c8, watcher_subs=0x15, n_body=0x0, force_null_body=1024) at notify.c:1562 #2 0xfe7f8f18 in query_db_notify (p_user=0x234, p_domain=0xdff08, event=0xfe809a28 "presence.winfo", watcher_subs=0xffbff1b0, etag=0x0) at notify.c:957 #3 0xfe8028a8 in update_subscription (msg=0x175fd8, subs=0xffbff1b0, rtag=0xffbff228, to_tag_gen=1) at subscribe.c:476 #4 0xfe805080 in handle_subscribe (msg=0x175fd8, str1=0xffbff0e0 "", str2=0x800 <Address 0x800 out of bounds>) at subscribe.c:1252 #5 0x00018378 in do_action (a=0x145270, msg=0x175fd8) at action.c:883 #6 0x00019bb8 in run_action_list (a=0x145270, msg=0x175fd8) at action.c:131 #7 0x000192bc in do_action () #8 0x00019bb8 in run_action_list (a=0x1452c8, msg=0x175fd8) at action.c:131 #9 0x00019df8 in run_top_route (a=0x143e10, msg=0x175fd8) at action.c:111 #10 0x00043b60 in receive_msg (buf=0x12dc00 "", len=1024, rcv_info=0xe0400) at receive.c:156 #11 0x0006f294 in udp_rcv_loop () at udp_server.c:465 #12 0x000338bc in main_loop () at main.c:834 #13 0x00035c44 in main (argc=9, argv=0xe7800) at main.c:1399 (gdb) quit

3 2

Re: [Serusers] Integrate LCS & SER Server
by Jiri Kuthan 07 Nov '07

07 Nov '07

At 18:17 05/11/2007, elilkumaran s wrote: >Sir/Madam > >I'm a System administrator of Institute. > >I use MS LCS 2005 SP1 Enterprise edition & Office Communicator in my work place. >I have linux SER Server & CISCO ATA 186 Telephone adapter. > >I have to run to Messenger client X-Lite for SER Server and Office Communicator for LCS. > >Because I place TP in Guard Office they can call Staff through SER server / X-Lite. >But can't call Office Communicator! Consider this "howto": - log in as administrator in the Windows machine on which LCS is running - start debian installation CD - when asked "replace existing partitions", answer yes - when finished, install SER debian packages, along with mysql - continue testing your SIP end-devices ;-) >CISCO ATA 186 can't registering in LCS, I thing it not possible directly. ( Because LCS use Active Directory Account through Kerbas & NTLM authentication). >So I want integrate LCS & SER Server to facilitate TP call to Office Communicator & Office Communicator call TP. >I use a pool in LCS as normal TCP (not TLS). SER Server also use normal (with out DB -Any one can registrar with any name) > >Could you kindly tell the process for integrate LCS & SER Server? Seriously -- why would you like to have two servers when you can have one, whichever you prefer better? -jiri >Thank You >Your truly >Elil > > > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

3 2

[OpenSER-Users] mi_fifo leaves little openser_receiver_* fifos on the floor
by Chris Heiser 07 Nov '07

07 Nov '07

I have a cronjob that periodically runs openserctl to gather stats out of openser. I've noticed that in my fifo directory, there are anywhere from 1.5 ~ 3k openser_receiver files laying around. Is there something screwy with mi_fifo in v1.2.2-tls that's preventing it from cleaning up after itself? --Chris

2 2

Re: [OpenSER-Users] OpenSER as a conditional stateless proxy
by lu 07 Nov '07

07 Nov '07

Hi, Iam too interested in this discussion, can some one explicitly elaborate how iam gonna satisfy all the conditions in a situation whereby OpenSER is relaying to Asterisk for local PSTN calls and to another a voip provider for International PSTN calls, ofcourse this applies for any numeric uri. I have a Sipwise generated config with offnet-termination which i tried to edit route [5] 'invite-find-calee' using t_relay() and rewritehostport conditions, but my concern is the role of LCR module which is supposed to be taking care of the GWs especially in route[7] "Request route invite-to-external", will it gonna disrupt/conflict my set conditions?? ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[5] { if(alias_db_lookup("dbaliases")) { xlog("L_INFO", "Callee was aliased - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); #CHECK WHERE THE REQUEST IS FROM???-edit by me if($fd=="sip.domain") { #FORWARD THEM TO ASTERISK -edit by me rewritehostport("asterisk IP"); #AND THEN TAKE THEM TO REQUEST ROUTE 'base-outbound'-edit by me route(3); } else if(uri=~"Sip:00[0-9]+@"){ #RELAY THEM TO INTERNETIONAL PSTN-edit by me t_relay("IP_of_international_PSTN_ provider"); route (3); } if(!is_domain_local("$rd")) { setflag(20); route(7); } if(does_uri_exist()) { xlog("L_INFO", "Callee is local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); } else { xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP= $si ID=$ci\n"); route(7); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[7] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); exit; } if(!isflagset(23)) { # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded if(uri =~ "^sip:[0-9]+@") { # only route numeric users to PSTN if(!load_gws()) { xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T= $tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } if(!next_gw()) { xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T= $tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } setflag(21); t_on_failure("1"); route(3); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP= $si ID=$ci\n"); sl_send_reply("404", "User Not Found"); exit; } WBR, LU. On Tue, 2007-11-06 at 06:35 +0000, 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: Openser as a conditional stateless proxy (Christian Schlatter) > 2. Presence RLS module: initialization error (Schumann Sebastian) > 3. CPL and XML schema (Daniel) > 4. Re: MediaProxy 1.9.0 - Radius (Miles Scruggs) > 5. Re: Is it possible to insert avp to reply message? (Tung Tran) > 6. Re: Is it possible to insert avp to reply message? > (I?aki Baz Castillo) > 7. Cellphones and voicemail (Chris Heiser) > 8. dispatcher with GeoIP (Victor Sergeev) > 9. Re: Cellphones and voicemail (Christian Schlatter) > 10. Re: retransmissions after final response (Vasile Zaharia) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 05 Nov 2007 10:16:18 -0500 > From: Christian Schlatter <cs(a)unc.edu> > Subject: Re: [OpenSER-Users] Openser as a conditional stateless proxy > To: Padmaja <padmaja.rv(a)vodcalabs.com> > Cc: users(a)lists.openser.org > Message-ID: <472F33C2.90900(a)unc.edu> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Padmaja wrote: > > Hi, > > > > Thanks for the reply, > > Could you provide me the example script for stateless forwarding of > > invite when a call is received from say a 911 operator to the server? > > I suppose this should be implemented by checking the From field of the > > invite and if it is from a certain 911, then it should be forwarded > > statelessly (or is this logic incorrect?). I am new to the Openser > > stuff, so I do not wish to change the script myself. > > > Your logic is correct, you can either check the From: header or the > request URI or any other part of an incoming SIP message to make the > routing decision. > > A simple example looks like: > > if (is_method('INVITE')) > { > # check if from URI domain equals 911operator.com > if ($fd == '911operator.com') > { > # relay stateless using request URI > forward(); > exit; > } > else > { > # relay statefull using request URI > if (!t_relay()) > { > sl_reply_error(); > }; > exit; > } > } > > Please always CC users(a)lists.openser.org so that others can take > advantage of the discussion. > > /Christian > > > > > Thanks again, > > Padmaja > > > > ----- Original Message ----- From: "Christian Schlatter" <cs(a)unc.edu> > > To: "Padmaja" <padmaja.rv(a)vodcalabs.com> > > Cc: <users(a)lists.openser.org> > > Sent: Monday, November 05, 2007 4:53 AM > > Subject: Re: [OpenSER-Users] Openser as a conditional stateless proxy > > > > > >> Padmaja wrote: > >>> Hi all, > >>> > >>> Can any one tell me if the same running instance of Openser can be > >>> configured as a stateful proxy for some user accounts and for some > >>> numbers like emergency services, it acts like a stateless proxy, just > >>> forwarding the request to the destination? > >> > >> Yes, it only depends on what forwarding function is used in the > >> routing script: > >> > >> - t_forward(..) --> stateless > >> > >> - t_relay(..) --> stateful > >> > >> /Christian

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[OpenSER-Users] SIP Topology Question
by Brandon Armstead 07 Nov '07

07 Nov '07

Hey Guys, just looking for some input on this, thanks! Basic Topology ------------------------------- Core Route Servers - Server A, B Registration Server, Collects USRLOC which all reports to the SAME DATABASE. Server C (User A initial registration server) User A -> it will tell me location Server D User A -> it will tell me location Any complications, I should be aware of? So Call Flow will look like so: INBOUND CALL: (Server A|Server B) -> (Server C|Server D) -> (END USER) Now should I be weary of any "lockings" such as a client end such as cable/dsl modem, the ata itself, etc? Thanks. -- Brandon Armstead

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