sr-users August 2008

sr-users@lists.kamailio.org

144 participants
201 discussions

[Kamailio-Users] How to test and trace the RLS messages?

by mahesh.peddi＠info-spectrum.com

Hi All, I am working on openser 1.3.2 version with RLS as "integrated_xcap_server". Its running without erros. But my query is how to test, And please send me call flow between IMS components and RLS server. Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

15 years, 8 months

Re: [Serusers] Content-Type: application/isup read encoded message

by Jiri Kuthan

I think the best learning experience is OOB http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/etc/ser-oob.cfg?re… it is not really simple, on the contrary -- it is quite complete, but for that it gives you all kinds of examples that can be probably useful to you. -jiri michael(a)teldelnort.com wrote: > Hello jiri, > > Ok, > > Do yo have an idea of a simple example that will help me familiarize the > usage in ser.cfg? > > -michael > > -----Original Message----- > *From:* Jiri Kuthan [mailto:jiri@iptel.org] > *Sent:* Tuesday, August 19, 2008 03:44 PM > *To:* michael(a)teldelnort.com > *Subject:* Re: [Serusers] Content-Type: application/isup read > encoded message > > I don't think so for this particular one -- just try to use it as > other select operators. -jirimichael(a)teldelnort.com wrote:> Hello > Jiri,> > Thanks for the response.> > Is there any documentation on > how to use select operator with @context?> > > I am trying to filter > a search on the context for "II Digits:"> > -michael> > > > -----Original Message-----> *From:* Jiri Kuthan > [mailto:jiri@iptel.org]> *Sent:* Tuesday, August 19, 2008 02:09 AM> > *To:* michael(a)teldelnort.com> *Cc:* serusers(a)iptel.org> *Subject:* > Re: [Serusers] Content-Type: application/isup read> encoded message> > > I think the select operator, @content, should get you access to > the> body. There is however noparsing of SDP to my knowledge > (unless> someone corrects me, I remember it has been> > frequentlymentioned.)-jirimichael(a)teldelnort.com wrote:> Hello,> >> > How can I evaluate the Content-Type: multipart/mixed;boundary= with> > SER?> > Content-Type: application/isup; base=ansi92; version=ansi> > >> Is there a module that allows SER read the contents of the > encoded> message?> I am try to get the > > "Originating Line > Information> ----> II Digits:"> > from the message.> > -michael> > > >> > ------------------------------------------------------------------------>> > > _______________________________________________> Serusers > mailing> list> Serusers(a)lists.iptel.org>> > http://lists.iptel.org/mailman/listinfo/serusers

15 years, 9 months

[Kamailio-Users] Ratelimit outgoing SIP INVITE messages after dispatcher

by sergejf

Hello, We are connected to several VoIP carriers that impose a calls per second (CPS) rate limit on their SIP trunks. For example, on TRUNK1 we can only make call attempts at 10 CPS. Our calls are originally made by a dialer that does not support rate limiting, hence it always blasts calls as fast as it can. Upon receiving these SIP INVITE messages we determine what carrier to use and perform a mapping to a dispatcher group. Call attempts are then load-balanced between all SIP trunks from the chosen carrier. My question is whether it is possible to use the ratelimit module in conjunction with the dispatcher (or carrierroute) to ensure we don't overload the carrier SIP trunks? If this is possible, then I wonder how to configure ratelimit to support a 10 CPS call attempts rate. Thanks in advance for any advice on the topic! Serge -- View this message in context: http://www.nabble.com/Ratelimit-outgoing-SIP-INVITE-messages-after-dispatch… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

15 years, 9 months

[Serusers] Content-Type: application/isup read encoded message

by michael＠teldelnort.com

Hello, How can I evaluate the Content-Type: multipart/mixed;boundary= with SER? Content-Type: application/isup; base=ansi92; version=ansi Is there a module that allows SER read the contents of the encoded message? I am try to get the "Originating Line Information ---- II Digits:" from the message. -michael

15 years, 9 months

[Kamailio-Users] About Asterisk2Billing

by Ali Jawad

Hi All Was anyone able to get the accounting part of asterisk2billing to work with Openser ? This looks like a pretty mature solution to me. Thanks With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

15 years, 9 months

[Kamailio-Users] Accounting for OPenser

by Ali Jawad

Hi I am looking for a accounting solution, that I can use with Openser without involving Asterisk. Has anyone tested something similar ? I need the solution to work for my prepaid customers, for example he has got 10$ once the credit is used. Deny any more call.s With Regards Ali Jawad System Administrator

15 years, 9 months

[Kamailio-Users] SIPp dispatcher load testing failure with too many hops - why?

by sergejf

Hello, I'm attempting to load test a simple dispatcher script in OpenSER 1.3.x using SIPp (built-in UAC and UAS scenarios). My dispatcher.list only has one address 8.XX.XX.12 (a SIPp instance, UAS). SIPp on 8.XX.XX.10 sends the INVITE messages to 63.XXX.XXX.110. Everything goes well until the UAC sends ACK and BYE in quick sequence. OpenSER tries to t_relay() it to itself, which results in a "too many hops" error: 2008-08-18 11:16:56:708 1219072616.708649: Aborting call on unexpected message for Call-Id '1-3548(a)8.XX.XX.10': while expecting '200' (index 8), received 'SIP/2.0 483 To Many Hops Via: SIP/2.0/UDP 8.XX.XX.10:5061;branch=z9hG4bK-3548-1-7 From: sipp <sip:sipp@8.XX.XX.10:5061>;tag=3548SIPpTag001 To: sut <sip:service@63.XXX.XXX.110:5060>;tag=2148SIPpTag011 Call-ID: 1-3548(a)8.XX.XX.10 CSeq: 2 BYE Server: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 I have tried modifying the SIPp UAC script to re-use the Contact: header from the SIP 200 OK message in the R-URI of the SIP ACK and SIP BYE, to no avail. I have loaded the TM module but I suppose that since the SIP 200 OK is the final message of the transaction, SIP ACK and SIP BYE won't automatically be relayed to 8.XX.XX.12. Is this an OpenSER issue or a SIPp scripting issue? Any advice would be appreciated. Please find below some relevant lines from my openser.cfg: route{ xlog("TRACE:ROUTE: src($si:$sp) dst($Ri:$Rp) msg($mb)\n"); # initial checkings if ( !mf_process_maxfwd_header("10") ) { xlog("SCRIPT:ERROR: $rm (from $si:$sp) too many hops\n"); sl_send_reply("483","To Many Hops"); exit; }; if (method==CANCEL) { if (t_check_trans()) t_relay(); exit; } # routing if (has_totag()) { xlog("SCRIPT0:INFO: $rm RURI=[$ru] - routing to dst-uri [$du] cnt [$avp(i:273)] dst set [$avp(i:271)]\n"); loose_route(); if (method=="INVITE") record_route(); route(1); exit; } The SIP ACK and SIP BYE follow the route defined in the above if {} statement. The SIP INVITE follows the dispatcher route shown below: record_route(); # perform load balancing # set algorithm (4 = round robin) $avp(alg) = 4; # set the group $avp(grp) = 2; if (method=="INVITE") xlog("CALL_START: RURI=[$ru] group=[$avp(grp)] callid=$ci\n"); # do balancing if(!ds_select_dst("$avp(grp)", "$avp(alg)")) { xlog("CALL_DROP:INTERNAL: no destinations for [$ru] group=[$avp(grp)], callid=$ci\n"); sl_send_reply("404", "no dst"); exit; } if(avp_check("$avp(i:273)", "eq/i:1")) { if(!avp_pushto("$ruri","$avp(i:271)")) { xlog("CALL_DROP:INTERNAL: cannot push to ruri [$avp(i:271)], group=[$avp(grp)], callid=$ci\n"); sl_send_reply("500", "cannot get dst"); exit; } } else { # redundancy t_on_failure("1"); } xlog("SCRIPT:INFO: $rm RURI=[$ru] - routing to dst-uri [$du] cnt [$avp(i:273)] dst set [$avp(i:271)]\n"); # do forward route(1); exit; -- View this message in context: http://www.nabble.com/SIPp-dispatcher-load-testing-failure-with-too-many-ho… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

15 years, 9 months

[Kamailio-Users] poll: project name

by Daniel-Constantin Mierla

Hello everybody, some people expressed inconveniences regarding the new name, so we would like to get your opinion about, therefore this is a poll that will get us to a decision. Here is the question: Do you like the name "Kamailio"? 1) Yes, I love it. Please keep it. 2) It is OK. 3) I do not mind the name at all 4) I do not like it 5) I hate the name and we should change it There is no need of explanations, as it is up to personal taste and decision. Just vote by option. Cheers, Daniel PS. for those interested... As short history about renaming story. On the 28th of March the board was informed about issues with SER part of project name. You can easily find who has the trademarks. On the 7th of April the board was informed about an agreement for the deadline as of September 30, 2008 to use old name. From there on, all old releases, documents, etc. can be referred by the old name, but nothing new. Other few discussions about what should/could be kept/changed followed and on 12th of May started the discussion about the new name with two main direction, something similar, to remind about old project or just something completely new, to be used as brand name. Couple of proposals and discussions regarding possible names followed up to 16th of July when was the deadline and the start of vote. Majority was reached on 17th but it was required to have opinions from all members of the board, last came from Bogdan, on the 28th of July. All discussions about migration were on the board mailing list. No change were done to openser.org configuration, just the new domain with kamailio in it was pointed to same server. Every step done was reported to board, following a schedule discussed there. Another discussion was about making the renaming and 1.4.0 release in same day. Being practically impossible to do all the work in one day, it was decided to do first the announcement about renaming. That happened on 28th when all agreed with the new name. As you can see, there was no 5 months of renaming duration. There was no change in openser.org DNS configuration undertaken with/without the knowledge of all board members. The new name domains were reserved when majority was achieved, in any case; dns entries for new name were pointed to old server shortly after booking the domains. -- Daniel-Constantin Mierla http://www.asipto.com

15 years, 9 months

[Serusers] SIP and IP multicast

by Steven C. Blair

Has anyone had any experience using SIP via SER to setup an IP multicast session with a group of phones?

15 years, 9 months

[Kamailio-Users] Looking for a VOIP Engineer

by Jonathan K. Creasy

Voxitas Inc. is a national VOIP provider based in St. Louis, MO. and we are currently looking for qualified applicants to fill Tier Three support positions and a VOIP Engineer position. The reason I am posting to this list is because we are looking to fill the VOIP Engineer position with someone who has OpenSER experience and is comfortable configuring and maintaining these systems. For details please contact our VP of Operations Zack Sargent. Zack Sargent VP Operations and Engineering Voxitas, Inc. 210 N. Tucker, St. Louis, MO. 63101 314-266-4000 x108 zsargent(a)voxitas.com Thank you! -Jonathan

15 years, 9 months

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sr-users August 2008