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36 participants
37545 discussions

[Users] NAT ping best methods
by Jayesh Nambiar 27 Apr '07

27 Apr '07

Hi, I am using Openser 1.2 and all the UAs that register to it are only PAP2 or SPAs. But all these UAs are on a private network. I tried using the NAT ping method from Openser to keep the NAT port open, but still sometimes the NAT port gets blocked by some NAT boxes. I tried keeping the interval as 25 seconds. So then I tried sending Keep Alive Notify messages from the UAs itself towards my Openser, but there I see a strange problem. The problem is like the device does not send the notify message from the same port which it registered from in the first place. So the port which is stored in the locaion table gets blocked and again the user is not able to receive calls. The user is able to receive calls only after it registers again but then again it is inconsistent. Has anyone worked with PAP2 and Openser and has got some best method to maintain the NAT ports open so that the user receive calls consistently. I also wanted to know what is more efficient, the UAs sending keep-alives or Openser sending NAT-ping? Thanks in advance, w/regards, Jayesh Add online presence to your web pages, find out how here - http://messenger.yahoo.com/addpresence.php

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[Users] openser: group module
by Helmut Kuper 27 Apr '07

27 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I tried to call is_user_in funktion with a variable as group name. I want to create the group name on the fly, when a INVITE arrives. if(is_user_in("To", $var(a))) { ... } I got this error message: 0(27274) xl_parse_svname: parsing [$var(a)] 0(27274) xl_lookup_spec_name: found [fu] [18] 0(27274) find_cmd_export_t: found <xlog>(2) in module xlog [/home/hk/openser//lib/openser/modules/] 0(27274) parse error (479,27-34): syntax error 0(27274) parse error (479,34-35): bad arguments Is there a way to do dynamic group names für is_user_id? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGMNVj4tZeNddg3dwRApHgAJ4qP30SN9NgH5yzVzx3qDuawQskywCbBfE+ Uum6I3PsxVVrzFa/KvchnmA= =6hHJ -----END PGP SIGNATURE-----

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[Serusers] Is there a bug in SERWeb ?
by inge 27 Apr '07

27 Apr '07

Hi all, I use SERWeb especially for call history. The core of this function is method.get_acc_entries.php. For some calls, I see that they are posted two times. During my investigation I see that the problem come from SQL request in method.get_acc_entries. In fact, it seems that when both calling and called party hang out in the same time, there are two BYE generated in acc table. Maybe the bug have been already reported, but I don't see nothing about this. Anyone one have an idea to correct it ? Thanks for your su

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[Serusers] Where can I Download a soft phone?
by karl lu 27 Apr '07

27 Apr '07

Hi, I'm new here. I'm wondering if anyone knows where I can download a softphone, which will work with SER. Thank you very much, Karl

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[Serusers] Ottendorf TM timers - global setting?
by Mahatma Sarasvati 27 Apr '07

27 Apr '07

Hello all, I see that one can set timeout variables per user in Ottendorf. Is it possible to avoid this and set them on a global or domain basis? My users aren't so particular and I'd like to avoid the effort of having to set the timers for every user. By the way, the default ser.cfg in Ottendorf is fantastic. Thanks for producing a well thought out config that demonstrates a good range of functionality. I've been using SER (+Asterisk) since 0.8.11 in my corporate environment (130 employees) so I'm not new, but I appreciate not having to take the time to sift though the code to figure out how the basics work. I'm eagerly awaiting more documentation on Ottendorf!! Thanks, Mahatma __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Users] WeSIP don't work with JDK1.6.0?
by GP.Coder 27 Apr '07

27 Apr '07

Hello,All I'm having problems running OpenSER(openser-1.2.0-notls_src.tar.gz) with WeSIP(wesip-02-04-07-beta-0.1.4.tar.gz) AS. SeasMessageEvent [SeasRouter] - New message received, from 192.168.0.119:3639 to 192.168.0.119:5060 over UDP SeasTransactionEvent [SeasRouter] - hash_index = 57932,label=1021493454 flags=0 ExpressMessageChannel [ExpressMChannel[0]] - peerAddress = 192.168.0.119/3639 ExpressMessageChannel [ExpressMChannel[0]] - About to process INVITE sip:2001@192.168.0.200 SIP/2.0 SipConnector [ExpressMChannel[0]] - createProcessor: Reusing existing processor SipProcessor [ExpressMChannel[0]] - An incoming message is being assigned ExpressMessageChannel [ExpressMChannel[0]] - Done processing INVITE sip:2001@192.168.0.200 SIP/2.0 SipProcessor [SipProcessor[4]] - <<<<<<<<< Request Received <<<<<<<<< INVITE sip:2001@192.168.0.200 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.119:15524 Max-Forwards: 69 From: "GP.Coder" <sip:5006@192.168.0.229>;tag=e495a30ed682466dad6c6fe809870efe;epid=1403e45675 To: <sip:2001@192.168.0.200> Call-ID: dec886aa3e954c6492b92fc8adef9de8(a)192.168.0.119 CSeq: 1 INVITE Contact: <sip:192.168.0.119:15524> User-Agent: RTC/1.2 Content-Type: application/sdp Content-Length: 479 SeasHashTable [SipProcessor[4]] - Inserting hashIdx:57932 label:1021493454 into hashTable SipProcessor [SipProcessor[4]] - Transaction [null] is new Transaction?? --> true EngineSipMapper [SipProcessor[4]] - Request from 192.168.0.229 mapped to: ConvergedEngine[Standalone].StandardHttpHost[localhost] ConvergedEngineValve [SipProcessor[4]] - Invoking Host:localhost SipProcessor [SipProcessor[4]] - error processing event... java.lang.NullPointerException at java.util.Hashtable.get(Hashtable.java:334) at com.voztele.sipservlet.sessions.StandardSipManager.findUncomplete(StandardSipManager.java:570) at com.voztele.sipservlet.valves.ConvergedHostValve.invokeSipRequest(ConvergedHostValve.java:218) at com.voztele.sipservlet.valves.ConvergedHostValve.invokeSIP(ConvergedHostValve.java:126) at com.voztele.sipservlet.valves.ConvergedHostValve.invoke(ConvergedHostValve.java:104) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.valves.ErrorDispatcherValve.invoke(ErrorDispatcherValve.java:124) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:594) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:118) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:594) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at com.voztele.sipservlet.valves.ConvergedEngineValve.invoke(ConvergedEngineValve.java:114) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at com.voztele.sipservlet.processor.SipProcessor.invokeContainer(SipProcessor.java:198) at com.voztele.sipservlet.processor.SipProcessor.processRequest(SipProcessor.java:142) at com.voztele.sipservlet.processor.SipProcessor.process(SipProcessor.java:272) at com.voztele.sipservlet.processor.SipProcessor.run(SipProcessor.java:253) at java.lang.Thread.run(Thread.java:619) SipConnector [SipProcessor[4]] - recycle: Recycling processor SipProcessor[4] GP.Coder 2007-04-26

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[Serusers] rtp proxy problem when re-inviting with changed local port in sdp
by Thomas Nilsen 27 Apr '07

27 Apr '07

Is it possible to get rtpproxy/nathelper to start a totally new rtpproxy session on re-invite and not use the previous pair of ports on the rtp proxy? In this case: A calls B behind nat. Sdp are rewritten to go though rtpproxy on two ports. X & Y. Some time goes by with conversation. Then the call chages codec, line or something (not important). In this case B ua changes rtp port locally also (incremented by 2). This is no problem. The re-invite is sent with new port. But problem now is that rtpproxy sees this and updates the sdp as always and it uses the previous X & Y port that was used on the rtpproxy before re-invite. The problem now is this. Packets from B to A goes correctly since A's local rtp port was same both before and after re-invite. But packets from A to B is arrivind on B's local port for the first sdp session, not the second. It seems that rtpproxy wont change this.. I assume this is a security measure for rtpproxy. But it should have some function to force new session ports on rtpproxy by parameters to force_rtpproxy() or something. As far as I can tell this is only a problem if the sip-ua is changing local port.. I thought maby calling unforce_rtp_proxy() before forcing rtp_proxy again would help, but don't work. Probably because of some internal working of nathelper. Anyone got an idea how to solve this? I would really appreciate any input on this.. - Thomas

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RE: [Users] [OT] WeSIP don't work with JDK1.6.0?
by "C. Bergström" 26 Apr '07

26 Apr '07

I apologize for the noise/rant of this post, but I feel something needs to be said. "By now support will be made trough the USERS openser mailing list" Source: http://www.wesip.com/mediawiki/index.php/SupportMailingList 1) I don't object to you making a commercial product, but please do not direct your users to this list. 2) I'm no license expert and won't confirm this by downloading your software, but seems you are in violation of the GPL by linking into openser and not releasing your software under a compatible license. 3) "WeSIP is patent pending technology" Source: http://www.wesip.com/mediawiki/index.php/Main_Page Where exactly is this technology pending patent? Do you have a reference number by some chance? 4) I'm going to be a bit of a jerk and say you probably don't have the capital to successfully litigate any international licensing violations of your software. So why was this license chosen exactly? To point of this is to not waste more of everyone's time, but to encourage the company behind this seemingly fine product to GPL their source so others can contribute. Thanks C.

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[Users] connect xmpp component to jabber xcp?
by Matt Winter 26 Apr '07

26 Apr '07

Hello, I'm having a problem connecting my openser xmpp module(in component mode) to a jabber xcp server. Has anyone done this? Here's my openser.cfg file. Thanks. # more /etc/openser/openser.cfg # # $Id: openser-xmpp.cfg 1827 2007-03-12 15:22:53Z bogdan_iancu $ # # simple quick-start config script for XMPP GW # # make sure in your main SIP server that you send # only the adequate SIP MESSAGES to XMPP GW # # # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=4 listen=udp:10.1.1.234:5060 alias=xmpp.jabbersrv.corp.com # ------------------ module loading ---------------------------------- mpath="/lib/openser/modules/" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # XMPP loadmodule "xmpp.so" modparam("xmpp", "domain_separator", "*") modparam("xmpp", "gateway_domain", "xmpp.jabbersrv.corp.com") modparam("xmpp", "xmpp_domain", "xmpp.jabbersrv.corp.com") modparam("xmpp", "xmpp_host", "jabbersrv.corp.com") modparam("xmpp", "backend", "component") modparam("xmpp", "xmpp_password", "BCD11234DC") # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo_xmpp") # -- usrloc params -- # -- 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; ### absorb retransmissions ### if (!t_newtran()) { sl_reply_error(); return; } if (method == "MESSAGE") { log("*** xmpp-handled MESSAGE message.\n"); if (xmpp_send_message()) { t_reply("200", "Accepted"); } else { t_reply("404", "Not found"); } return; } log("*** xmpp: unhandled message type\n"); t_reply("503", "Service unavailable"); return; }

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[Users] sending NOTIFY messages
by Mark Price 26 Apr '07

26 Apr '07

Hi, Are there any generally used methods for sending SIP NOTIFY messages to a phone that is registered through openser? I.e. for sending events to sip phones? For example, I imagine a script that would register to the mydomain.org openser server as an administrative user and send a NOTIFY packet to user(a)mydomain.org. I'm hoping not to reinvent the wheel here. Mark Price

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