sr-users August 2005

sr-users@lists.kamailio.org

220 participants
558 discussions

by Daniel Corbe

Hello, I'm going to attempt to build a box that listens on a SPAN switch port which has all my proxy traffic mirrored to it. I was curious if anyone has ever attempted something like this before, and what software/tools were used to listen for signaling (SIP) traffic and subsequently log the traffic out to disk. -Daniel

18 years, 8 months

Re: [Serusers] NAT example

by Kamran Ahmad

hello Why uri!=myself what could be the reason using "SER 0.9.3 source package for Getting Started" from Onsip.org when i try to use port 5060 then it only work for public ip UA and dont work for nated users. But when i change port to 5069 it try to listen for nated UA but the problem is still why uri!=myself i am using and for configuration ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-rtpproxy.3.05.cfg after lot of changes in ports exec_msg("cat >> /tmp/call.log"); if (uri!=myself) { xlog("L_ERR","uri!=myself"); route(1); break; }; i found this /var/log/messages -------------------------------------------------- Aug 30 21:58:09 SIP ser[10560]: uri!=myself and /tmp/call.log -------------------------------------------------- REGISTER sip:SERPUBIP SIP/2.0 Content-Length: 0 Contact: <sip:8000@192.168.0.21:5060>;events="message-summary" Call-ID: B692E8F0-DDC4-42A7-8A87-79B8133FA79B(a)192.168.0.21 Max-Forwards: 16 From: <sip:8000@SERPUBIP>;tag=587922581424 CSeq: 5 REGISTER To: <sip:8000@SERPUBIP> Via: SIP/2.0/UDP 203.99.58.250:52904;rport;branch=z9h8001009User-Agent: SJLabs-SJphone/1.40.258 --- Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > Don't worry, > These messages comes from the startup process, when > SER tries to get the PTR DNS records of the IP > addresses present in the config file. These pointers > map IP addresses to domain names, instead of the > "standard" domain->IP mapping provided by the A|AAA > DNS records. > > Probably your SER instance is running, have you > tried ps?? > > Enjoy! > > Samuel. > > > Unclassified. > >>> Kamran Ahmad <p_kami(a)yahoo.com> 08/30/05 10:35AM > >>> > i tried this > > ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-rtpproxy.3.05.cfg > > > but it is not working > > /var/log/messages > Aug 30 11:12:42 SIP ser: WARNING: fix_socket_list: > could not rev. resolve ip > Aug 30 11:12:42 SIP ser: WARNING: fix_socket_list: > could not rev. resolve ip > > > --- "Greger V. Teigre" <greger(a)teigre.com> wrote: > > > Try to Getting Started document from > > http://onsip.org/ You will find both a > > working config, as well as detailed explanation of > > the theory and actual > > commands. > > g-) > > > > Kamran Ahmad wrote: > > > Hello > > > > > > i am following this link for ser-0.9.3 but Nated > > > Clients > > > are not working. Can any one please tell > > me > > > what is the problem in my configuration. > > > > > > > > > http://www.voip-info.org/tiki-index.php?page=SER+nat+support > > > > > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam > protection around > http://mail.yahoo.com > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

18 years, 8 months

Re: [Users] #DEFINE equivalent?

by Kanakatti Mahesh Subramanya

because i didn't think of it? :-) cheers, and thanx Bogdan-Andrei Iancu wrote: > Kanakatti Mahesh Subramanya wrote: > >> I totally agree. >> >> Personally, I've been ripping out large chunks of my config file, and >> replacing it with varioud avpops based stuff - it may be *slightly* >> more convoluted, but it is completely database configurable, and >> hence easier to maintain >> >> re: rr, its actually neither. Its a static, which i'm using to >> identify NAT'ed clients to account for re-INVITEs > > > why don't you use the append_rr_param() function to added the NAT > identifier to the RR hdr? do a normal record_route() and if nat is > detected, add the param > > regards, > bogdan > >> >> cheers >> >> Bogdan-Andrei Iancu wrote: >> >>> Hi Kanakatti, >>> >>> I'm for m4'ing (for the moment) for two reasons: >>> 1) since we can do it with m4, I prefer to use the cycles for >>> something still missing; >>> 2) using dynamic parameters (like via AVPs) reduce the >>> performance - time to look for the AVP and most critical the >>> impossibility of calling fixup functions at startup (which will >>> force to fix params at runtime, for each processing). >>> >>> regarding the record_route_preset() - do you use dynamic parameters >>> or dynamic rr URI? >>> >>> regards, >>> bogdan >> >>

18 years, 8 months

[Serusers] Reviewers needed for Open Solaris SIP API design review

by Rao Shoaib

Dear Colleagues, We are conducting a community design review for the open solaris SIP API at http://www.opensolaris.org/jive/thread.jspa?threadID=1936&tstart=0 We are very interested in hearing your comments on the design. Please post your comments at the above location. Thanks. Rao.

18 years, 8 months

[Serusers] supporting multiple calls from one user

by Iqbal

Hi Does ser support dialing to multiple destinations using one username/password, especially for setups where hardware may have multiple voice channels, but only space for one username. Anyone have any test setups Iqbal

18 years, 8 months

[Serusers] SER outgoing through TCP?

by Pat Wang

Hi, Does anyone know how to configure SER proxy signalling through TCP? For example, UAC1 -------> SER -------> UAC2, I would like to have SER proxy the INVITE (and the following) message to UAC2 using TCP. Is that possible and how to setup on ser.cfg? Or if it possible to do TCP signalling between two SERs: SER ------TCP---------SER? Thanks a million, Pat

18 years, 8 months

[Serusers] Routing - Looking up database table for subscriber accounts not registered at the moment

by Frank Fischer

Hi all i just started working with SIP and i'm currently trying to implement the routing logic as needed in my setup. Doing this i came across a requirement i couldn't find a solution for. Maybe someone of you may help me? I have the following situation: I have users with (only) e164Numbers as usernames in my SIP domain and i have a PSTN Gateway behind which i may reach also e164Numbers. Now, i have a subscriber base in my domain where not all of the subscribers with a e164Number may be registered at a certain moment (but still have a valid account and phonenumber). When i get a call at such a moment, i will try to look up the callee username in the location database. If i find one, fine. If not, there are two possibilities left - first, it could be a user only reachable in the pstn, so route to the pstn gateway or second it might be a user in my domain with a valid account but not registered at the moment, so reject or redirect to a voicemail or what ever. If I would work with alphanumerical numbers only within my SIP cloud, i knew how this could be done - all numerical numbers automatically would be PSTN numbers. But how can this be done in my setup? Thanks a lot for your help and kind Regards Frank

18 years, 8 months

[Serusers] ACK

by Sebastian Kühner

Hi! I noticed a "phenomenon" that I don't understand: I make accounting with ser. If all works fine, ser puts into the database: sip_method: ACK i_uri: sip:0182011542614xxxxxx@xxx.xxx.xxx.xxx:5060 (xxx.xxx.xxx.xxx = IP of PSTN-Gateway) ... and after the call, the BYE message reaches my server. But sometimes, the ACK is logged in the database like this (in the most cases form users "far away"): sip_method: ACK i_uri: sip:0182011542614xxxxxx@my-ip:5060 In this case, the BYE message doesn't reach my server. Does anybody know what could be the reason and why ser logs the ACK from himself? Many thanks! Sebastian

18 years, 8 months

Re: [Serusers] Radius Server sending reply to another machine!!!

by Ashutosh Kumar

Many thanks Jan, Adding bindaddr directive to radiusclient.conf worked!!!!. Now, the problem remains as to why is the radiusclient/ser picking up the dictionary.ascend, when I have included the dictionary.ser at the end of my dictionary file. Also, I have auth_radius:service_type modparamed' to value of 15. Please guide. Regards, Ashutosh Kumar -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Tuesday, August 30, 2005 2:40 PM To: Ashutosh Kumar Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Radius Server sending reply to another machine!!! Check the configuration file of the radius client library, perhaps you have set the bindaddr parameter to the wrong IP address ? Jan. On 30-08-2005 03:57, Ashutosh Kumar wrote: > Hi, > I am facing a strange bewildering problem out here. Earlier I was > running ser -0.9 on machine a.b.c.116. Now we shifted it to a.b.c.113 for > testing for radius testing purposes. > But in the radius packet dump, it is showing as the radius packet still > originating from a.b.c.116 machine, despite the fact, my new revised ser.cfg > doesn't contain any listen address or alias for .116 box. > Also, the /etc/hosts doesn't mention .116 address. Also, it is not the case > that .116 and .113 are IPs homes on same machine. My ser.cfg code for radius > authorization is as follows. > > ... > ..... > > if (!radius_www_authorize("")) { > # if (!www_authorize("","subscriber")) { > www_challenge("","0"); > break; > }; > ... > ..... > > In the DB also, there is no table . The Domains table has no records. But > still SER is usng .116 address to send radius packets from. Here is the > radius packet dump. > > [root@sip radiusclient]# tcpdump -i eth1 -s 1500 -w - udp | raddump > -vvvvvvvvvvvvvvvvvvvvvsn > tcpdump: listening on eth1 > Looks like a(n) libpcap/tcpdump packet trace > 17:45:13.051205 radius a.b.c.113:50912 > p.q.r.s:1812 Access-Request 180 > User-Name = test(a)a.b.c.112 > Ascend-PW-Warntime = 168195173 > Ascend-PW-Warntime = 17774137 > Ascend-PW-Warntime = 36320307 > Ascend-PW-Warntime = 68383593 > Ascend-PW-Warntime = 51008069 > Ascend-Menu-Item = 75dd216d3efa9bfb9c2d7e013d54f7bb > Service-Type = 15 (unknown) > Ascend-PW-Lifetime = 1952805748 > NAS-IP-Address = a.b.c.116 > NAS-Port = 5060 > > (Is there any table /file in ser other than ser.cfg or mysql://ser that > contains the domain information that might have been lying there unchanged > since machine migration?) > > Also, why is SER using incorrect SER dictionary NAS client (Ascend)( > shouldn't it have been Cisco or sometinhg like that). I have included the > dictionary.ser included in ser-0.9 source in the radiusclient conf file. > > Help me, its driving me crazy. > > > Regards, > Ashutosh Kumar > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

18 years, 8 months

[Users] CANCEL problem

by Klaus Darilion

Hi! I just having problems cancelling forked calls. openser forwards the incoming CANCEL to both phones. As you see, openser added 2 via headers to the last CANCEL. The via header of branch 1 and branch 2. Maybe something got wrong when introducing branch routes? I'm using todays CVS. klaus 83.136.33.3:1200 -> 83.136.32.83:5060 CANCEL sip:klaus@enum.at SIP/2.0 Via: SIP/2.0/UDP 10.10.0.76:8418 Max-Forwards: 70 From: "klaus.darilion(a)nic.at43.at" <sip:klaus.darilion@nic.at43.at>;tag=b3669e387d784f1dbd080f83fdd62306;epid=9012264a99 To: <sip:klaus@enum.at> Call-ID: e654c424b1124ce29b93e7dd6a8ad44c CSeq: 2 CANCEL User-Agent: RTC/1.3 Content-Length: 0 # U 83.136.32.83:5060 -> 83.136.33.19:5060 CANCEL sip:klaus@83.136.33.19:5060 SIP/2.0 Record-Route: <sip:83.136.32.83;ftag=b3669e387d784f1dbd080f83fdd62306;nat=yes;lr=on> Via: SIP/2.0/UDP 83.136.32.83;branch=z9hG4bK131f.f3d7378.0 Via: SIP/2.0/UDP 10.10.0.76:8418;rport=1200;received=83.136.33.3 Max-Forwards: 16 From:"klaus.darilion(a)nic.at43.at" <sip:klaus.darilion@nic.at43.at>;tag=b3669e387d784f1dbd080f83fdd62306;epid=9012264a99 To: <sip:klaus@enum.at> Call-ID: e654c424b1124ce29b93e7dd6a8ad44c CSeq: 2 CANCEL User-Agent: RTC/1.3 Content-Length: 0 P-Behind-NAT: Yes # U 83.136.32.83:5060 -> 83.136.33.3:8718 CANCEL sip:klaus@10.10.0.50:8718 SIP/2.0 Record-Route: <sip:83.136.32.83;ftag=b3669e387d784f1dbd080f83fdd62306;nat=yes;lr=on> Via: SIP/2.0/UDP 83.136.32.83;branch=z9hG4bK131f.f3d7378.0 Via: SIP/2.0/UDP 83.136.32.83;branch=z9hG4bK131f.f3d7378.1 Via: SIP/2.0/UDP 10.10.0.76:8418;rport=1200;received=83.136.33.3;rport=1200;received=83.136.33.3 Max-Forwards: 16 From: "klaus.darilion(a)nic.at43.at" <sip:klaus.darilion@nic.at43.at>;tag=b3669e387d784f1dbd080f83fdd62306;epid=9012264a99 To: <sip:klaus@enum.at> Call-ID: e654c424b1124ce29b93e7dd6a8ad44c CSeq:2 CANCEL User-Agent: RTC/1.3 Content-Length: 0 P-Behind-NAT: Yes

18 years, 8 months

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sr-users August 2005