sr-users March 2008

sr-users@lists.kamailio.org

169 participants
237 discussions

[Serusers] Need way to specify SOURCE IP address on packets of SIP messages being emitted by SER

by Frank Durda IV

I have a problem that I thought SER could solve but now that I need it, the method isn't obvious. First, as an example, let's say I have three independent organizations running Asterisk boxes, which send me the SIP and RTP audio for calls. Let's call them Company A Asterisk box at 31.1.2.3 Company B Asterisk box at 42.4.5.6 Company C Asterisk box at 53.7.8.9 So they are one side of the SER box all communicating SIP to a single interface. On the other side is a TDM PSTN switch capable of accepting SIP calls. The PSTN switch has exactly one IP address, which we will call 101.2.3.4. Inside the PSTN switch, for calls originating in TDM headed to SIP, a "Route List" directs calls for a given trunk group (everything is visualized in TDM terms even if the terms don't really apply) to a specific IP address, where the SIP INVITE should go for this call. In the case of a SIP to TDM call, when an INVITE comes in, the PSTN switch looks through that list of Routes that point to IP addresses and when it finds an IP address match, it knows which trunk group to associate with this call for billing. This also acts as a simple friend/foe mechanism, because if there is no route pointing at your SIP source, it rejects your INVITES. So, if Company A, B and C were communicating directly with the PSTN switch (no SER), you would simply need three trunk groups where each IP address pointed directly to the Asterisk boxes at those companies, like this: TGN SIP IP 1111 -> 31.1.2.3 2222 -> 42.4.5.6 3333 -> 53.7.8.9 and all would be well. However, when you insert a SER box between those the Asterisk boxes belonging to those companies and the PSTN switch, problems arise. I originally decided to address the issue by establishing two groups of three IP addresses, one set that the Asterisk servers would individually send their INVITEs to, and three that the PSTN switch could send its INVITES to. SER already supports the idea of listening on numerous IP addresses, so that was easy (although the documentation for "listen" and "alias" is weak and a code review was needed to determine what alias actually did, something other than what the "alias" option of an ifconfig command does...). Anyway, we have a setup like this: interface interface interface em1 on SER em0 on SER sig0 on PSTN 10.1.0.0/16 10.2.0.0/16 101.2.3.0/28 Company A 31.1.2.3 ---- 10.1.2.3 SER 10.2.0.1 ---- 101.2.3.4 Company B 42.4.5.6 ---- 10.1.5.6 SER 10.2.0.2 ---- 101.2.3.4 Company C 53.7.8.9 ---- 10.1.8.9 SER 10.2.0.3 ---- 101.2.3.4 Inside the ser.cfg, it is simple enough to rewrite the "host address" (where you want the packet to go) to reflect the new destination once that is determined, and for TDM to SIP calls work okay from the perspective of the PSTN switch, but then it really never cared whether the SER box was there or not. As long as the PSTN switch knows who to bill, it is vaguely happy. To accomplish that, the routes inside the PSTN switch were altered to point to the SIP IP addresses that the PSTN switch would have contact with, as in: TGN SIP IP 1111 -> 10.2.0.1 2222 -> 10.2.0.2 3333 -> 10.2.0.3 (Changing the IPs in the PSTN switch turns out to be a non-trivial operation.) The more serious problem arises with a call going SIP to TDM, because when the SER box sends a packet on to the PSTN box at 101.2.3.4, it needs to also re-write the SOURCE IP of the packet to reflect the appropriate IP address on the em0 interface, not just the first IP address that is discovered on the interface. What it does right now is send everything out under the first IP address listed on that interface, usually 10.2.0.1, so calls fail to set up properly as the replies go back to company A even if coming from company B or C. Is there a function in SER that lets you set the right address to use on outgoing packets? As all of the IP addresses that would be used are in fact ones set on that interface, it should be just a question of setting the right one, but I haven't found a function for ser.cfg or loadable module that clearly states that it will do this. So in summary, when Company B sends an INVITE to 10.1.5.6 from their 42.4.5.6 system, I need to have SER send an invite to 101.2.3.4 from 10.2.0.2 and never any other address that happens to be on that same interface. And when the reply to the INVITE comes to 10.2.0.2 from 101.2.3.4, I need SER to forward it on to 42.4.5.6 from 10.1.5.6. Note that while I am showing an example with three companies, there will be dozens, even hundreds of unique IPs involved here, hence the enormous netblocks set aside for each interface on the SER box. I'm certainly open to alternative methods to get SER to alter the headers of the packets it forwards exactly as needed, but keep in mind that I have basically zero control over how the PSTN switch does things, and it has exactly one IP address (three actually, but one does all the SIP work) that you send to and it originates from, yet is keenly interested in where packets it receives came from (for billing purposes) so you can't expect any help from the PSTN switch side. So is there a function callable from ser.cfg to force set the SOURCE IP address on SIP messages on a message by message basis? We are currently using ser-0.9.6. Thanks for your assistance, Frank Durda

16 years, 1 month

[Serusers] Is there any way to modify ser.cfg without rebooting ser?

by lin9k＠yahoo.com

Hi, Anybody can illuminate how to modify ser.cfg and make the new modified version effective for ser without rebooting ser? Thanks in advance, Ling ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

16 years, 1 month

[Serusers] Problems with links on iptel.org (Getting Started, module docs etc)

by Greger Viken Teigre

Due to a domain/registrar problem, the onsip.org domain used in referencing a server with SER Getting Started, module docs, etc has caused broken links on iptel.org the last week or so. The plan is to move everything over to iptel.org servers/domain, but I haven't had time yet. I have now updated the referenced links on iptel.org site to use a temporary domain name, so everything should be ok now. Please raise your voice if there are still onsip.org reference lying around on iptel.org. g-)

16 years, 1 month

[OpenSER-Users] Issue with Openser Serial Forking

by Aadilkhan Maniyar

Hi All, I am trying to integrate Openser 1.2.3 and Asterisk 1.4.18. The registrations are handled by Openser and Asterisk acts as a B2BUA. In one of the cases the combo should support Serial Forking. I register an user, ua1 from two different contacts say ua1@contact1 first and ua1@contact2 later. Hence the latest registration of ua1 is ua1@contact2. Now when I call ua1, ua1@contact2 gets two calls, one directly from Openser and second from Asterisk (since I am forwarding the INVITEs from openser to Asterisk). I would like to implement serial forking in Openser. Can someone help me out here.. Regards, Aadil

16 years, 1 month

[Serusers] acc.so issue

by Santiago Gimenez Ocano

Hi all, I'm trying to use the acc.so module but for some reason I can't make it load in my ser 0.9.7 (x86_64/linux). This is the stderror output: 0(3184) ERROR: load_module: could not open module </usr/local/lib/ser/modules/acc.so>: /usr/local/lib/ser/modules/acc.so: undefined symbol: parse_orig_uri and this is the list of the modules I'm loading in ser.cfg file: loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri_radius.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/acc.so" Does any one have experienced this? Does any one can helpme? Regards, Santiago _________________________________________________________________ Blog your life in 3D with Windows Live Writer. http://www.windowslive.com/overview.html?ocid=TXT_TAGLM_Wave2_wl_writer_022…

16 years, 1 month

Re: [OpenSER-Users] LCR Routing.... Choosing Best Route

by Douglas Garstang

Thanks. I'll do all that as soon as I find documentation that gives me some clues. :) ----- Original Message ---- From: Alex Hermann <alex(a)speakup.nl> To: users(a)lists.openser.org Sent: Friday, March 28, 2008 12:36:51 AM Subject: Re: [OpenSER-Users] LCR Routing.... Choosing Best Route On Friday 28 March 2008, Douglas Garstang wrote: > Carrier Prefix Cost > --------------------------- > Carrier1 1650 0.3 > Carrier2 165 0.2 > What do people do in this case? Don't use the LCR module to compare, but do it by other means by selecting the longest matching prefix per carrier and then sort the list on price. Store the list of associated uri's in the gw_uri_avp and use next_gw() to traverse them. Alex. _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

16 years, 1 month

[OpenSER-Users] calling external phone number

by Sam Wun

Hi, I have media proxy, openser, and monit installed in FreeBSD. I have also had the x-lite windows client software working with the openser server. Now, if I want to make phone call from home to external phone number thru the Openser server, what else I have to do to accomplish that? Thanks Sam ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

16 years, 1 month

[OpenSER-Users] [openser install] error while creating the openser DB - presence tables

by Pascal Maugeri

Hi While creating the openser DB (/usr/local/sbin/openserdbctl create) I get the following error: INFO: creating presence tables into openser ... ERROR 1170 (42000) at line 2: BLOB/TEXT column 'rlsubs_did' used in key specification without a key length ERROR: Failed to create rls-presence tables! FYI I've downloaded the source code from 1.3 branch (rev. 3957) and disabled the following modules in my installation: exclude_modules?= jabber cpl-c postgres osp unixodbc avp_radius auth_radius group_radius uri_radius xmpp presence_mwi pua pua_bla pua_mi pua_usrloc pua_xmpp rls mi_xmlrpc perl snmpstats perlvdb ldap carrierroute h350 xcap_client db_berkeley seas (From the original Makefile I only enabled mysql, presence and presence_xml modules) Should I enable any other module ? It does work when I choose to not install the presence related tables. Thanks in advance for any help or suggestion. Regards, Pascal

16 years, 1 month

[OpenSER-Users] javax.sip.TransactionUnavailableException: no route! error

by shiebar＠post.tau.ac.il

Hi, Does anyone have a idea what can cause this error message? I have installed OpenSER with WeSIP and small HelloWorld Servlet. When tring to send INVITE message I get the following WeSIP trace: SipProcessor [ExpressMChannel[6]] - An incoming message is being assigned SipProcessor [SipProcessor[4]] - <<<<<<<<< Request Received <<<<<<<<< INVITE sip:shie_sip2@192.168.2.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.102:5061;rport;branch=z9hG4bK23208 To: <sip:shie_sip2@192.168.2.100> From: "Shie" <sip:shie_sip@192.168.2.100>;tag=9844 Call-ID: 1206715243-3208-SHIE(a)10.108.3.122 CSeq: 751 INVITE Max-Forwards: 19 User-Agent: NCH Swift Sound Express Talk 3.06 Contact: <sip:shie_sip@192.168.2.102:5061> Allow: INVITE,CANCEL,ACK,BYE,INFO,OPTIONS,NOTIFY,REFER Supported: replaces Content-Type: application/sdp Content-Length: 352 SeasHashTable [SipProcessor[4]] - Inserting hashIdx:9320 label:781080043 into hashTable SipProcessor [SipProcessor[4]] - Transaction [z9hG4bK23208] is new Transaction?? --> true EngineSipMapper [SipProcessor[4]] - Request from 192.168.2.100 mapped to: ConvergedEngine[Standalone].StandardHttpHost[localhost] ConvergedEngineValve [SipProcessor[4]] - Invoking Host:localhost ConvergedHostValve [SipProcessor[4]] - SipSession not found for requestINVITE, trying match against Rules ConvergedHost [SipProcessor[4]] - Mapped request to:/HelloWorldServlet ConvergedContextValve [SipProcessor[4]] - new session created :1206715243-3208-SHIE@10.108.3.122 StandardProxy [SipProcessor[4]] - sip:shie_sip2@192.168.2.100 Added to proxying targets. StandardProxy [SipProcessor[4]] - doProxy: proxying to sip:shie_sip2@192.168.2.100 StandardProxy [SipProcessor[4]] - ========== Request Sent ============== INVITE sip:shie_sip2@192.168.2.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.100:5060,SIP/2.0/UDP 192.168.2.102:5061;rport;branch=z9hG4bK23208 To: <sip:shie_sip2@192.168.2.100> From: "Shie" <sip:shie_sip@192.168.2.100>;tag=9844 Call-ID: 1206715243-3208-SHIE(a)10.108.3.122 CSeq: 751 INVITE Max-Forwards: 18 User-Agent: NCH Swift Sound Express Talk 3.06 Contact: <sip:shie_sip@192.168.2.102:5061> Allow: INVITE,CANCEL,ACK,BYE,INFO,OPTIONS,NOTIFY,REFER Supported: replaces Content-Type: application/sdp Record-Route: <sip:192.168.2.100:5060;lr=on;transport=udp> Content-Length: 352 StandardProxy [SipProcessor[4]] - javax.sip.TransactionUnavailableException: no route! SipConnector [SipProcessor[4]] - recycle: Recycling processor SipProcessor[4] ExpressMessageChannel [ExpressMChannel[6]] - Done processing INVITE sip:shie_sip2@192.168.2.100 SIP/2.0 Thanks in advance, Shie

16 years, 1 month

[OpenSER-Users] How to check if $rd lies in an IP range

by Josh Mahonin

Hi folks, In my setup, I've got two disjoint subnets (call then A and B) that cannot communicate directly to each other, but devices on each can both communicate to my OpenSER server and Asterisk box (both on their own subnet, C). There is no NAT involved, so I only want to use rtpproxy when it's the case that device from subnet A attempts to call a device on subnet B, or vice-versa. I would ideally not like to use rtp proxy for communication between A-C and A-B (this will enable RTP media between both subnets, but that solution will not scale very well...) I'm attempting do something like this: if (src_ip == a.b.c.d/24 && dst_ip == w.x.y.z/24) use rtp proxy But unfortunately, on an INVITE, after a lookup, dst_ip is set to the OpenSER server. The pseudovariable $rd is set to the value I'd like to check against, but it complains loudly when I attempt to substitute dst_ip for $rd. Is there any way to use avp_check() or the like to verify that the value in $rd lies in a given subnet? I don't want to match just one IP, but a whole range. I found a similar question on the SER mailing list asked several years ago, with no response. Thanks, Josh

16 years, 1 month

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sr-users March 2008