sr-users August 2008

sr-users@lists.kamailio.org

144 participants
201 discussions

[OpenSER-Users] uaCSTA support in openser

by Hugo Koblmueller

high.all! i'm wondering if there is any support of uaCSTA in openser (planned)? i'm just working on the integration of asterisk (*) environment to OCS 2007 environment, having openSER in the middle (mainly for TCP/UDP translation and smoothing out the protocol deficienes on both sides). in this setup the * having the openSER in front is talking to the OCS (and vice versa) via the OCS mediation server, which is moreorless sending standard SIP messages, which enables normal softphone (integration to *) of the office communicator. this configuration is already working... now i'm planning to go for the CTI integration, where there is no OCS mediation server in between OCS and openSER, doing the translation of SIP/CSTA to SIP. i'm thinking about using openSER for this task, that's why i'm looking for a CSTA module or perl programm, which is capable of this functionality. afaik for the CTI communication there isn't the full complexity of CSTA needed, just a subset mainly for call setup and call clearing. anyone having experience on this topic? thx & cheers -hugo Great Ideas for Small Devices Hugo Koblmueller Senior Staff Engineer Software Development COMNEON electronic technology GmbH & Co. OHG Freistaedter Strasse 400 4040 Linz Austria hugo.koblmueller(a)comneon.com tel: fax: mobile: Skype ID: +43 (5) 1777 - 15730 <http://www.plaxo.com/click_to_call?lang=en&src=jj_signature&To=%2B43+%285%2 9+1777+%2D+15730&Email=hugo(a)koblmueller.com> +43 (5) 1777 - 15810 +43 (676) 82051280 <http://www.plaxo.com/click_to_call?lang=en&src=jj_signature&To=%2B43+%28676 %29+82051280&Email=hugo(a)koblmueller.com> drhookson Want to always have my latest info? <https://www.plaxo.com/add_me?u=21475050628&src=client_sig_212_1_banner_join &invite=1&lang=en> Want a signature like <http://www.plaxo.com/signature?src=client_sig_212_1_banner_sig&lang=en> this?

13 years, 2 months

[Serusers] Deadly embrace with rtpproxy - Is it necessary? Can it be turned off?

by Frank Durda IV

I'm running into a problem with rtpproxy on this point, quoting from the README: - - - - - - - - - - - - after the session has been created, the proxy listens on the port it has allocated for that session and waits for receiving at least one UDP packet from each of two parties participating in the call. Once such packet is received, the proxy fills one of two ip:port structures associated with each call with source ip:port of that packet. When both structures are filled in, the proxy starts relaying UDP packets between parties; - - - - - - - - - - - However, a number of clients frequently fail to emit any audio when originating a call until they hear something from the TDM gateway, such as ring-back or the called party answering. So although rtpproxy is receiving a stream of audio, such as a voice mail menu robot, the calling party can't hear any of it unless they happen to make some noise or randomly and blindly press a DTMF key. This seems to be made worse on links with silence suppression, so there is no background noise to trigger two-way audio. This is being encountered between Class 4 carriers, so we don't have the option to get someone to adjust their phone/PBX settings or have them breathe heavier. Is there a setting adjustment to get rtpproxy to just pass the RTP packets from directed calling and called sources even if one party hasn't happened to make noise yet? I personally don't understand why this requirement for seeing audio from both sides before starting the flow in either direction if audio starts coming in even exists. It seems to have no benefit but is bound to cause this deadly embrace problem in many situations that may be beyond the control of the owners of the equipment passing traffic along to the site where rtpproxy is in use. Suggestions? Fix? I have looked at the latest snapshot of rtpproxy and the README is unchanged since 1.1 so apparently this behavior is still the same. Thanks in advance!

15 years

[OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?

by Iñaki Baz Castillo

Hi, maybe this question is a bit off-topic so I'm sorry for that. My question is about SIP providers using OpenSer that associate PSTN numbers to their local clients (SIP accounts): Usually the client must register to OpenSer in order to receive calls. Then it will appear in location table with "Contact=sip:clientX@IP". Suppose clientX has two PSTN numbers associated in a ENUM entry: +34999000111 +34999000222 When anyone in PSTN world calls to +34999000222 the call will arrive to the OpenSer from a gateway in an INVITE like: INVITE sip:+34999000222@gateway SIP/2.0 To: <sip:+34999000222@gateway> The OpenSer will do the lookup in location table and finally send this INVITE to the clientX: INVITE sip:clientX@IP_clientX SIP/2.0 To: <sip:+34999000222@gateway> The info about the called PSTN number is just available in "To" header, so a way to get different behaviour for each associated PSTN number is matching "To" URI. Is common to do it? which other alternatives are there? Thanks for any comment. Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

15 years, 1 month

[OpenSER-Users] Openser1.3.2 + snmp

by Ricardo Alves Ferreira Silva

Hi all, Trying to install Openser1.3.2.... everything is OK and fine... but when I try to add support for snmp there are errors.. Dig dep into google, recheck all dependencies and didn't succeed... My env is FreeBSD 7.0-STABLE i386 The full net-snmp package is installed: atila# pkg_info | grep net-snmp net-snmp-5.3.2_3 An extendable SNMP implementation These are the error messages.... make modules/libsnmpstats modules.... Compiling alarm_checks.c gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -DMOD_NAME='"snmpstats"' -DNAME='"openser"' -DVERSION='"1.3.2-notls"' -DARCH='"i386"' -DOS='"freebsd"' -DCOMPILER='"gcc 4.2.1"' -D__CPU_i386 -D__OS_freebsd -D__SMP_no -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -DHAVE_TIMEGM -DHAVE_NETINET_IN_SYSTM -DHAVE_KQUEUE -DHAVE_SELECT -c alarm_checks.c -o alarm_checks.o In file included from alarm_checks.c:38: openserObjects.h:41:38: error: net-snmp/net-snmp-config.h: No such file or directory openserObjects.h:42:40: error: net-snmp/net-snmp-includes.h: No such file or directory openserObjects.h:43:52: error: net-snmp/agent/net-snmp-agent-includes.h: No such file or directory If I edit each file and hardcode the path ... /usr/local/include/net-snmp/... there are no errors... What am I missing in making sure the path to the include files are OK... Tried a couple options but didn't succeed... tehre are tons of include files into the src.... Regards Ricardo Ferreira Yes, we do run Openser www.vipway.com.br Enterprise VoIP Services

15 years, 5 months

[Serusers] ser-0.9.6: modules/postgres bugreport & patch

by Alexandre Snarskii

Hi! I'm running ser-0.9.6, on FreeBSD 6.1-stable, database backend is postgresql version 8.1.3. Today I got errors in logfiles, saying: messages.2.bz2:May 29 14:35:03 <XXX> /usr/local/sbin/ser[51448]: ERROR:avpops:dbrow2avp: dbrow contains NULL fields The similar problem reported in: http://lists.iptel.org/pipermail/serusers/2005-May/019681.html with much more detailed description of error, database contents and config samples. Patch is trivial, and looks more like a fix to copy'n'paste error: in mysql/val.c function str2val states: if (!_s) { memset(_v, 0, sizeof(db_val_t)); VAL_TYPE(_v) = _t; VAL_NULL(_v) = 1; return 0; } VAL_NULL(_v) = 0; and the last line mentions that 'well, that's value is not NULL'. In postgresql/db_val.c, line 182, function str2valp, the same statement is the: if (!_s) { DLOG("str2valp", "got a null value"); VAL_TYPE(_v) = _t; VAL_NULL(_v) = 1; return 0; } without explicit notification that this is not-NULL value. More than, nowhere else in this function VAL_NULL(_v) not set to 0. So, if a value _v.val contained anyting but 0 at the function start, resulting value will be threated as NULL despite the fact, that _s is NOT NULL. Patch is obvious, just add VAL_NULL(_v)=0; after cited block (line 188) and everyting will be ok. At least for me it's ok for some hours :)

15 years, 6 months

[OpenSER-Users] Problems with SerMyAdmin Installation

by Mark Koopmann

Hi everybody, does anybody has experience with installing the java tool SerMyAdmin? The Tool is running but every action ends with an runtime exception. Trying to register a new user ends with the following message: Grails Runtime Exception Error Details Message: Caused by: java.lang.NullPointerException Class: RegisterUserController At Line: [98] Code Snippet: (empty) Stack Trace org.codehaus.groovy.runtime.InvokerInvocationException: java.lang.NullPointerException at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:290) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationDispatcher.invoke(ApplicationDispatcher.java:630) at org.apache.catalina.core.ApplicationDispatcher.processRequest(ApplicationDispatcher.java:436) at org.apache.catalina.core.ApplicationDispatcher.doForward(ApplicationDispatcher.java:374) at org.apache.catalina.core.ApplicationDispatcher.forward(ApplicationDispatcher.java:302) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.jsecurity.web.servlet.WebInterceptorFilter.doFilterInternal(WebInterceptorFilter.java:106) at org.jsecurity.web.servlet.SecurityContextFilter.doFilterInternal(SecurityContextFilter.java:93) at org.jsecurity.web.servlet.OncePerRequestFilter.doFilter(OncePerRequestFilter.java:106) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:233) at org.apache.catalina.core.StandardContextValve.invoke(StandardContextValve.java:175) at org.apache.catalina.core.StandardHostValve.invoke(StandardHostValve.java:128) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:102) at org.apache.catalina.core.StandardEngineValve.invoke(StandardEngineValve.java:109) at org.apache.catalina.connector.CoyoteAdapter.service(CoyoteAdapter.java:286) at org.apache.coyote.http11.Http11Processor.process(Http11Processor.java:844) at org.apache.coyote.http11.Http11Protocol$Http11ConnectionHandler.process(Http11Protocol.java:583) at org.apache.tomcat.util.net.JIoEndpoint$Worker.run(JIoEndpoint.java:447) at java.lang.Thread.run(Thread.java:595)Caused by: java.lang.NullPointerException at RegisterUser.toString(RegisterUser.groovy) at RegisterUserController$_closure8.doCall(RegisterUserController.groovy:98) at RegisterUserController$_closure8.doCall(RegisterUserController.groovy) ... 29 more Unfortunately, i dont have much experience with installing such java stuff. It would be great if one of you can give me a hint or something. I dont have any idea how to solve this problem. Thanks for any comment! Best regards Mark Koopmann

15 years, 7 months

[Kamailio-Users] call forward problem

by muhammad akl

hey all after making call forward feature using avpops module and its exported functions avp_db_load and avp_pushto i have a problem : assuming all calls to extension 12001 will be forwarded to extension 12008, in opensers's database i have added the following : username=12001 domain =mydomain attribute = callfwd value = sip:12008@mydomain type = 0 untill that every thing was fine and all calls comes to 12001 forward to 12008 without any problem , the problem comes from when i am using 12008 to call 12001 it calls itself ( mean 12008) so anyone can tell me what i have missed ? i have configured my openser.cfg like this in loaded modules section added loadmodules "avpops.so" > in module parameter section added : modparam("avpops", "avp_url", "mysql://openser:openserrw@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") in Invite message handler added : if(avp_db_load("$ru/username","$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); route(1); exit; } any suggestions ?

15 years, 7 months

[Kamailio-Users] FW: Call Hangup from Caller-End not working with TLS

by Ali Jawad

Hi All I am using using openser 1.3..if I make a call between two softphones on the same lan or a a pstn call to my mobile phone..and the called/receiver party does hang-up the call. It works fine in UDP mode and the call get's hang-up. However in TLS mode this does not work. Anything I might have missed here? Since both udp and tls use the same routes, and voice is fine and no one way audio ..etc. I notice this http://pastebin.com/m38c979f6 on rtp proxy. However in the logs of openser I can't see any errors. However on the wire shark I can see icmp destination unreachable...port unreachable. I would have said it is a NAT issue. However it works for simple UDP. However I did notice the following in the logs Aug 28 13:41:01 [8565] DBG:tm:set_timer: relative timeout is 4000000 Aug 28 13:41:01 [8565] DBG:tm:insert_timer_unsafe: [7]: 0xb6137794 (45900000) Aug 28 13:41:01 [8565] DBG:tm:retransmission_handler: retransmission_handler : done Aug 28 13:41:02 [8564] DBG:core:parse_msg: SIP Request: Aug 28 13:41:02 [8564] DBG:core:parse_msg: method: <BYE> Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri: <sip:michofr@193.227.186.146:3218;transport=UDP;rinstance=D98C1DD404B200 8F980980E97E42F8EC;nat=yes> Aug 28 13:41:02 [8564] DBG:core:parse_msg: version: <SIP/2.0> Aug 28 13:41:02 [8564] DBG:core:parse_headers: flags=2 Aug 28 13:41:02 [8564] DBG:core:parse_via_param: found param type 232, <branch> = Shouldn't the transport=TLS ? Regards

15 years, 7 months

[Kamailio-Users] Kamailio dispatcher Module redirection

by Asim Riaz

Hi All I am trying to configure Openser as loadblancer for two more Kamailio. I wan loardbalancer not stay in the middle once routed to other kamilio servers. my first question is, dispathcer list always take IPs or I can configure proxy names as well. at the moment I am checking the to_uri and then redirecting the traffic to my one of Kamilio on my network, it seems to work fine with Linksys but not other phones e.g polycom xlite or even sjphone. if(to_uri=~"sip:.+@sip.mydomain.ie <sip%3A.%2B(a)sip.mydomain.ie>") { ds_select_domain("1", "4"); sl_send_reply("300","Redirect"); exit; } any suggestion. Thanks in advance Regards Asim Riaz

15 years, 7 months

[Kamailio-Users] uac_redirect question

by Zahid Mehmood

Hi, Need a little help with get_redirects(), using openser 1.2. I am using the following in failure_route to capture contacts from a "multiple choices" reply from cisco media gateway. Following code sets the ruri to the last contact, however, the $ds show more information about other contacts. Is there a way to reset this so that only ruri information is used? Cisco response and xlog information is also given below. if(!get_redirects("*:*")) { xlog("L_ERROR", "Failed to fetch contact '$ct' from 301/302 - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci \n"); acc_db_request("480", "acc"); t_reply("480", "Temporarily Unavailable"); exit; } # get last URI from destination-set and set it as R-URI xlog("L_INFO", "Redirect from UAC intercepted 1 - M= $rm RURI=$ru D=$ds B=$bR \nF=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:tmp)/g"); $avp(s:tmp) = $ds; avp_subst("$avp(s:tmp)", "/.*(sip:.+@[^:;>]+).*$/\1/"); avp_pushto("$ru", "$avp(s:tmp)"); setflag(29); xlog("L_INFO", "Redirect from UAC intercepted 2 - M= $rm RURI=$ru D=$ds B=$bR \ntmp=$avp(s:tmp) \nF=$fu T=$tu IP=$si ID=$ci \n"); append_branch(); route(17); # process exit; } Thanks in advance for your advice. -- Zahid The response from gateway: U 2008/08/26 08:19:04.440411 10.10.0.32:5060 -> 10.10.0.98:5060 SIP/2.0 300 Multiple Choices. Via: SIP/2.0/UDP 10.10.0.98;branch=z9hG4bK901b.335cb695.0,SIP/2.0/UDP 10.10.12.140;branch=z9hG4bK612a4f20B9293665. From: "10521" <sip:10521@devproxy.myip.org>;tag=B395F9F2-4618AC3F. To: <sip:40001@devproxy.myip.org;user=phone>;tag=195535C4-11AC. Date: Tue, 26 Aug 2008 12:19:04 GMT. Call-ID: f7932e23-d89cd014-e3ea20a9(a)10.10.12.140. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 2 INVITE. Allow-Events: telephone-event. Diversion: <sip:40001@10.10.0.32>;reason=unconditional;counter=1. Contact: <sip:10512@10.10.0.98>,<sip:10512@10.10.0.32>. Content-Length: 0. xlog entries: Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted 1 - M=INVITE RURI=sip:40001@10.10.0.32:5060;transport=udp D=Contact: sip:40001@10.10.0.32:5060;transport=udp, <sip:10512@10.10.0.32 >;q=0.01, <sip:10512@10.10.0.98>;q=0.01 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140 Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted 2 - M=INVITE RURI=sip:10512@10.10.0.98 D=Contact: sip:10512@10.10.0.98, <sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98 >;q=0.01 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01 tmp=sip:10512@10.10.0.98 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140 Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted - M=INVITE RURI=sip:10512@10.10.0.98 D=Contact: sip:10512@10.10.0.98, <sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98 >;q=0.01, sip:10512@10.10.0.98 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01, sip:10512@10.10.0.98 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140

15 years, 7 months

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sr-users August 2008