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[Users] $avp(s:rpid) and another question...
by Helge Waastad 09 Mar '07

09 Mar '07

Hi, two small questions this time :-) 1.) From 1.2.x modules dok, default rpid_avp for auth is: modparam("auth", "rpid_avp", "$avp(s:rpid)") However setting this gives a failure: parse_avp_name: error - use type (s: or i:) in front of avp name Mar 8 21:33:46 proxy-02 /usr/sbin/openser[16128]: ERROR:auth:init_rpid_avp: invalid rpid AVP specs "$avp(s:rpid)" Mar 8 21:33:46 proxy-02 /usr/sbin/openser[16128]: ERROR:auth:mod_init: failed to process rpid AVPs Mar 8 21:33:46 proxy-02 /usr/sbin/openser[16128]: init_mod(): Error while initializing module auth The "old" way still works: modparam("auth","rpid_avp","s:rpid") 2.) Earlier I had modparam("tm","fr_inv_timer_avp","s:inv_timeout") and: avp_db_load("$tu",s:inv_timeout) [inv_timeout attribute in database] What would be the equivalent with the new avp configuration? br hw

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[Users] sdp negotiation fails - no ACK - rtpproxy
by Benko 09 Mar '07

09 Mar '07

Hello! I'm struggling with a issue since several days, since i make no more efforts i'd like to ask you for help: I'm trying to attach a exchange 2007 server to openser, most of the features work(inbound call, refer) but i can't get a outbound call to work as exc doesn't seem to like the SDP in my "200 OK" answer, i get no ACK and no rtp-stream is sent from exchange. My setup looks like this: ________ _______ _____ ________ |asterisk|=======|openser|======|(NAT)|=======|exchange| |________| |_______| |_____| |________| 2x.x.x.107 2x.x.x.106 62.x.x.193 10.1.99.97 A rtpproxy is running on the openser too, exchange is talking tcp... In my openser-config i use the nathelper-module and rtpproxy to traverse nat for the clients. This setup is working for all phones i've tested so far, so imho it is useable. Exchange has a feature that allows dialouts to read voicemail to e.g. a mobile phone. It is using a standard INVITE dialog, in a local network without nat i've been succesfull to attach exchange to sipx and a phone registered at sipx and i guess it will also work with a local openser who doesn't use nat-traversal features. However, with the public openser where i have a NAT, exchange doesn't seem to accept my sdp-proposal, i get no ACK and the rtp stream is only sent by asterisk, not by exchange. Unfortunately exchange offers no valuable debugging information, so i have no idea what exactly it doesn't like about my answer - i did some trial and error with different nat-helper-functions, activated and deactivated them but so far it didn't make a difference(or the call didn't got through at all due to missing nat-traversal). I've uploaded wireshark-traces exported to text of the working local dialog with sipX(http://pastebin.ca/383516) and the non-working external dialog (http://pastebin.ca/383540) with openser, maybe someone can take a look at it and check if there's something wrong on my(=openser) side of the dialog - Thanks in advance! best regards Christian

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[Users] Strange Problem involving PSTN call in
by Sam Lee 09 Mar '07

09 Mar '07

Hey Guys, I have had this very strange problem .. I have 2 PSTN gateways to allow for inbound and outbound calls. INBOUND calls from PSTN will reach one of these gateways (depending on the provider) and then directly routed to OPENSER. Gateways are cisco AS5350. The very strange thing is , on certain SIP-UAs that are being used outside (most of them ADSL users) , calls that come inbound via Gateway A can always ring the SIP-UA , calls coming inbound via Gateway B will not get a response. AND when i did a packet capture on the openser server , both gateways forwarded SIP requests inwards to the server and both are being processed . User Location is found and being routed. Both calls are being routed to the same UA , same port . So much for strange , anyone has any idea ? Regards, Sam

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[Users] Openser 1.1.x avp question
by Mike O'Connor 09 Mar '07

09 Mar '07

Hi Guys Would this work ? (I have a working system and do not want to break it to test) avp_db_load("$ruri/username", "*"); if( is_avp_set("$avp(s:callfwdimmediate)")) { .... }; if( is_avp_set("$avp(s:callfwdbusy)")) { .... }; if( is_avp_set("$avp(s:callfwdnoanswer)")) { .... }; Thanks Mike

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Re: [Users] WeSIP b2bua problem
by tele 09 Mar '07

09 Mar '07

Hi Gines! It's almost clear to me how to do, this what i have done and it works. now i need last clarification how forward the 200 OK from BYE request, i cannot get it working. for forwarding the 200 ok from the BYE in the doBye request i save downstreamLeg.setAttribute("byereq", req) and then in the doResponse i get the bye request and create the response from it, something like this but i'm wrong. if (resp.getStatus() == 200 && resp.getMethod().equalsIgnoreCase("BYE")) { SipServletRequest upstreamRequest = (SipServletRequest)upstreamLeg.getAttribute("byereq"); SipServletResponse upstreamResponse = upstreamRequest.createResponse(resp.getStatus(),resp.getReasonPhrase()); //Copy the content from the downstream response to the upstream response if (resp.getContentType() != null) { upstreamResponse.setContent(resp.getRawContent(), resp.getContentType()); } upstreamResponse.send(); } this is what i have done for working ACK. protected void doAck(SipServletRequest req) throws ServletException, IOException { //Retrieve the upstream request to respond it SipSession downstreamLeg = (SipSession)req.getSession().getApplicationSession().getAttribute("downstreamLeg"); SipServletResponse downstreamResponse = (SipServletResponse) downstreamLeg.getAttribute("200ok"); SipServletRequest ackRequest = downstreamResponse.createAck(); //Copy the content from the downstream response to the upstream response if (req.getContentType() != null) { ackRequest.setContent(req.getRawContent(), req.getContentType()); } ackRequest.send(); } protected void doBye(SipServletRequest req) throws ServletException, IOException { SipSession downstreamLeg = (SipSession)req.getSession().getApplicationSession().getAttribute("downstreamLeg"); SipServletRequest byeRequest = downstreamLeg.createRequest("BYE"); // Copy the content from the downstream response to the upstream response if (req.getContentType() != null) { byeRequest.setContent(req.getRawContent(), req.getContentType()); } byeRequest.send(); } protected void doResponse(SipServletResponse resp) throws ServletException, IOException { if (resp.getStatus() == 200 && resp.getMethod().equalsIgnoreCase("INVITE")) { SipSession downstreamLeg = (SipSession) resp.getSession().getApplicationSession().getAttribute("downstreamLeg"); downstreamLeg.setAttribute("200ok",resp); } //Retrieve the upstream request to respond it SipSession upstreamLeg = (SipSession)resp.getSession().getApplicationSession().getAttribute("upstreamLeg"); } On Mon, 2007-03-05 at 00:29 +0100, Ginés Gómez wrote: > The problem is that you didn't implement the doACK method. Implement > it following a similar technique as the one used in doResponse. Save > the 200OK response in the downstream session so you can retrieve it > when the ACK arrives from the upstream then generate ACK using > response.createACK() and copy the content using getContentType/ > setContentType geContent/setContent if required > > > Hope it helps > > Gines > > > Attached there is a .zip with all the log, trace and configuration. > > > > thanks for the support, > > > > :tele > > <wesip_test.zip> > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users >

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[Serusers] INFO: Daily generated documentation
by Greger V. Teigre 09 Mar '07

09 Mar '07

Hi all, The daily generation of module documentation is now the following: * SER 2.0 aka Ottendorf module documentation can be found at http://www.iptel.org/views/moduledocs or in table from: http://www.iptel.org/views/tableofmodules * Module documentation for the current development (trunk) can be found at http://siprouter.onsip.org/doc/trunk/modules/ * Doxygen documentation for trunk can be found at http://siprouter.onsip.org/doc/doxygen/ All are generated nightly around 4am (CET). Thus, if you want to see documentation for modules added in the current development version, use http://siprouter.onsip.org/doc/trunk/modules/ All links on iptel.org have been updated, so these links can easily be found. g-)

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[Users] accounting of final replies
by Henning Westerholt 08 Mar '07

08 Mar '07

Hello all, i need to do additional accounting for final replies with the acc module, especially distinguish local from forwarded replies. We patched 0.9.5 for this functionality. We add a flag to the message, that is set by tm, and read from the acc module and then added to the log message. Is this possible in 1.2 with the extra accounting functions and pseudo variables? Or is the approach described above still prefered? I know that there exist the t_local_replied() function in tm, and it is perhaps with that possible to log a message in the config file with add_log_request(). But it would be nice if this information could be included in the normal accounting log message. Thanks, Henning

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[Users] Routing question
by Lucas Diaz 08 Mar '07

08 Mar '07

I have openser listening in 2 different interfaces, and it works great. My question is about users with different ip ranges: do they have to be able to see each other (same ip range or through a gateway), or openser must/can allow/route that communication? Thanx a lot. Lucas

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[Users] Working on 3xx FWD
by Sam Lee 08 Mar '07

08 Mar '07

Hey guys, I have a current setup like this . PSTN user calls in to SIP boxes via PSTN Gateways Gateway will send the call to the OPENSER servers and in turn rings the SIP UA IF SIP UA tries to forward it out to the PSTN again , the problem i am getting now is NO ring back on the CALLER . (called party (PSTN) rings and has no problem starting conversation . Just that no ring back). I am currently making use of the UAC_Redirect in the openser server to route based on the 3xx move temporarily message. Is there a way to send a provisional message to the calling party PSTN gateway (originating GW) that it is trying / ringing so that the ORIGINATING GW will generate ring back to the caller ? Please let me know if you can't understand . THanks for your help guys! Regards, Sam

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[Users] how to add user to database
by sukerry 08 Mar '07

08 Mar '07

Hi, I have installed openser 1.2.0 in my federo, however I do not know how to use a tool to add users to database. Any hint will be appreciated. Thanks kerry

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