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37519 discussions

[Users] incorrect documentation
by Mark Price 22 Nov '06

22 Nov '06

Hi, I see that the siptrace documentation for openser 1.1.0 says that the function siptrace() will log a packet. However, the correct function name is sip_trace(). Is there a way to go about submitting improvements/fixes to the documentation? Mark Price

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[Users] Loop?
by Kelvin Williams 22 Nov '06

22 Nov '06

Yesterday, something happened-the exact I do not know just yet. However, when looking at the logging that I have my OpenSER performing I seen a ton of INVITES, CANCELs, OpenSER became unusable for a time until I was able to block all SIP traffic at the firewall and let it calm down enough to restart it. During this process calls could not be made (in or out). Is there a way to prevent this type of behavior? It looked as though there were only a handful of calls that caused a continuous loop. What logic is recommended to test for a call that has been handled too many times by the proxy, and to immediately drop it? Thanks in advance.

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[Serusers] Open IMS Core released last Thursday
by Dragos Vingarzan 22 Nov '06

22 Nov '06

Hi *, as you did not had enough of me last week, here I go again: FOKUS recently launched the Open IMS Core. Keeping it short: - project at http://www.openimscore.org (SVN, mail-lists, etc on berlios) - general information at http://www.fokus.fraunhofer.de/ims/OSIMS.php?lang=en or mailto:info@open-ims.org - you get P/I/S-CSCF, HSS, SIP2IMS Gateway, CDiameterPeer (as module for SER, also standalone usable), JavaDiameterPeer (100% Java and decent performance). All in development, so if you look for stability, security or compatibility, you won't find it there yet. -Dragos -- ----------------------------------------- Dipl. Eng. Dragos Vingarzan FOKUS/NGNI Kaiserin-Augusta-Allee 31 10589 Berlin,Germany Phone +49 (0)30 - 3463 - 7385 Mobile +49 (0)163 - 159 - 5221 eMail vingarzan(a)fokus.fraunhofer.de Web www.fokus.fraunhofer.de We could change the world if God would give us the source code... -----------------------------------------------------------------

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[Users] merging string and pseudo-variable in avp_write
by Benko 22 Nov '06

22 Nov '06

Hello! I'd like to implement an avpops serial forking solution as described in http://www.voice-sistem.ro/docs/avpops/ar01s08.html#ex_serial_forking However, i would like to define the adresses as "$rU(a)mydomain.org". What is the correct syntax to merge the string and the variable in avp_write? I would like to do something like this: avp_write("sip:$rU@mydomain1.org", "$serial_fork"); avp_write("sip:$rU@mydomain2.org", "$serial_fork"); where $rU should be parsed. $serial_fork would have a value of e.g. "12632421(a)mydomain1.org" then. Or does this not work at all? How else could i define it? Can someone point me to a fundamental doc where the openser.cfg-syntax is described? My well-tried trial-and-error method doesn't seem to work here... thx Christian

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[Fwd: Re: [Users] Why is fix_nated_sdp appending 127.0.0.1 at the end of my c= lines?]
by nick 22 Nov '06

22 Nov '06

-------- Original Message -------- Subject: Re: [Users] Why is fix_nated_sdp appending 127.0.0.1 at the end of my c= lines? Date: Tue, 21 Nov 2006 16:46:42 +0100 From: nick <nick(a)mobilia.it> To: Ovidiu Sas <sip.nslu(a)gmail.com> References: <4562E2D7.7010403(a)mobilia.it> <4562E82E.4070503(a)pernau.at> <45631216.3000800(a)mobilia.it> <6f497e130611210650i409e6cfblfb39b0d8a0b9f2e2(a)mail.gmail.com> <45631873.8010400(a)mobilia.it> <6f497e130611210724p4b3ead9bha95cb848c29d276c(a)mail.gmail.com> Ovidiu Sas wrote: > It seems that you are trying to do a double processing of the SDP > (that's why you have 2 IPs inside SDP). Check you script if you use > force_rtp_proxy and fix_nated_sdp at the same time. > > > Regards, > Ovidiu Sas > > On 11/21/06, nick <nick(a)mobilia.it> wrote: >> Ovidiu Sas wrote: >> > check the '-d' argument. If you don't specify '-d', ngrep will only >> > listen to one interface. '-d any' will force ngrep to listen to all >> > interfaces, including the loopback. >> > >> > >> > Hope this helps, >> > Ovidiu Sas >> >> BTW, I changed my fix_nated_sdp() command to fix_nated_sdp("10") only, >> which is adding the origin IP instead of the mediaproxy one.. I don't >> mind this, but it still seems to be adding the 127.0.0.1 >> >> > Actually, I'm using mediaproxy and fix_nated_sdp at the same time, I am using fix_nated_sdp because for some reason, mediaproxy isn't changing the SDP information in the SIP requests (or at least, it wasn't beforehand). I'll have a look at where it might be overlapping. whoops forgot to cc

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[Users] $ct and $hdr(Content-Type) failing
by Mark Price 22 Nov '06

22 Nov '06

Hi, I am trying to log MESSAGE packets differentiated the content-type of the packet. I can see in the packet dump that they have types such as "text/html" and so on. I'm using the following statement in the code: if(method=="MESSAGE") { xlog("L_ALERT","$ct $rb\n"); } The message body specified by $rb is printing out fine, but $ct is printing out "<null>". I get the exact same result if I say: xlog("L_ALERT","$hdr(Content-Type) $rb\n"); Is this a bug in openser? Thanks, Mark Price

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[Users] new poll: location for next OpenSER event
by Daniel-Constantin Mierla 22 Nov '06

22 Nov '06

*Hello Everybody, due to very good feedback and results we got for the first OpenSER summit we want to collect your opinion about the location of the next one. It will help to make the decision since will reflect community distribution. The options are basically the continents, if you have other suggestions please use team(a)openser.org. The poll is at bottom side of right column at http://www.openser.org Cheers, Daniel *

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[Serusers] re: question - digest authentication and firewall
by Yair Hakak 21 Nov '06

21 Nov '06

hi all- i have the following setup: 1. ser running on 5060 2. nathelper/rtp 3. mysql database w/digest authentication now, i have a client on a private network (NAT) behind a firewall. port 5060 tcp is on the firewall is open as well as 35000 to 37000 UDP (rtpproxy is set to run between these ports). when i plug the client into a regular, non-firewalled home router, it registers properly (register->401->register again->OK) when i put the same client behind the firewall, it 401's twice. i am 100% sure the password is correct. is there something else that needs to be open? I dont really understand the authentication mechanism - does the password come via a different port or something like that?

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[Users] Why is fix_nated_sdp appending 127.0.0.1 at the end of my c= lines?
by nick 21 Nov '06

21 Nov '06

Everything seems fine, the SIP messages go tot he right place, the NAT tests are finding everything as they should.. and in the o= row, they're being replaced (more as a test, I don't think it's completely neccessary) without an issue. However, in the c= line, it adds the IP I've specified (for some reason, if I leave it blank it only puts 127.0.0.1; or a mashup of the public address and localhost (as mediaproxy runs on the same machine for the moment). here are the two sections of code I've added.. these are the only places I call fix_nated_sdp() if (method=="INVITE") { if (nat_uac_test("8")) { fix_nated_sdp("10","89.xxx.xxx.16"); }; t_on_failure("1"); } else if (method == "BYE" || method == "CANCEL") { end_media_session(); }; onreply_route[1] { if (status=~"(183)|(2[0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); }; if (nat_uac_test("8")) { fix_nated_sdp("10","89.xxx.xxx.16"); }; use_media_proxy(); }; } I'm using openser 1.1.0 notls, on a dual opteron server running CentOS 4.3 here is a trace of the code in action.. (names changed.. because I enjoy search and replace) U 89.xxx.xxx.8:30626 -> 89.xxx.xxx.16:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.xxx.xxx.16;branch=z9hG4bK674e.0d2a4f64.0. Via: SIP/2.0/UDP 192.168.20.16:38000;received=87.xxx.xxx.99;branch=z9hG4bK-d87543-b3783334b231975b-1--d87543-;rport=38000. Record-Route: <sip:89.xxx.xxx.16;lr;ftag=5c63711f>. Contact: <sip:nick@89.xxx.xxx.8:30626;rinstance=f0677dc7f7d4d2f0>. To: "Joe Blow"<sip:joe@blow.com>;tag=ee11f527. From: "Tom"<sip:tom@blow.com>;tag=5c63711f. Call-ID: NTFjYmM5Y2JhNWE5MzU2NzQ1NTMxYTg2NGQ3ZDYzNmM.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 237. . v=0. o=- 1 2 IN IP4 192.168.1.67. s=CounterPath X-Lite 3.0. c=IN IP4 192.168.1.67. t=0 0. m=audio 31556 RTP/AVP 0 8 3 101. a=alt:1 1 : 0W8e8Rrk y9fAinpH 192.168.1.67 31556. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. # U 89.xxx.xxx.16:5060 -> 87.xxx.xxx.99:38000 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.20.16:38000;received=87.xxx.xxx.99;branch=z9hG4bK-d87543-b3783334b231975b-1--d87543-;rport=38000. Record-Route: <sip:89.xxx.xxx.16;lr;ftag=5c63711f>. Contact: <sip:nick@89.xxx.xxx.8:30626;rinstance=f0677dc7f7d4d2f0>. To: "Joe Blow"<sip:joe@blow.com>;tag=ee11f527. From: "Tom"<sip:tom@blow.com>;tag=5c63711f. Call-ID: NTFjYmM5Y2JhNWE5MzU2NzQ1NTMxYTg2NGQ3ZDYzNmM.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 302. . v=0. o=- 1 2 IN IP4 89.xxx.xxx.16. s=CounterPath X-Lite 3.0. c=IN IP4 89.xxx.xxx.16127.0.0.1. t=0 0. m=audio 60070 RTP/AVP 0 8 3 101. a=alt:1 1 : 0W8e8Rrk y9fAinpH 192.168.1.67 31556. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=oldmediaip:192.168.1.67. a=oldmediaip:192.168.1.67. # U 87.xxx.xxx.99:38000 -> 89.xxx.xxx.16:5060 ACK sip:nick@89.xxx.xxx.8:30626;rinstance=f0677dc7f7d4d2f0 SIP/2.0. Via: SIP/2.0/UDP 192.168.20.16:38000;branch=z9hG4bK-d87543-7e77316e3b41bd7d-1--d87543-;rport. Max-Forwards: 70. Route: <sip:89.xxx.xxx.16;lr;ftag=5c63711f>. Contact: <sip:thomas@87.xxx.xxx.99:38000>. To: "Joe Blow"<sip:joe@blow.com>;tag=ee11f527. From: "Tom"<sip:tom@blow.com>;tag=5c63711f. Call-ID: NTFjYmM5Y2JhNWE5MzU2NzQ1NTMxYTg2NGQ3ZDYzNmM.. CSeq: 1 ACK. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 0. . # U 89.xxx.xxx.16:5060 -> 89.xxx.xxx.8:30626 ACK sip:nick@89.xxx.xxx.8:30626;rinstance=f0677dc7f7d4d2f0 SIP/2.0. Record-Route: <sip:89.xxx.xxx.16;lr=on;ftag=5c63711f>. Via: SIP/2.0/UDP 89.xxx.xxx.16;branch=z9hG4bK674e.0d2a4f64.2. Via: SIP/2.0/UDP 192.168.20.16:38000;received=87.xxx.xxx.99;branch=z9hG4bK-d87543-7e77316e3b41bd7d-1--d87543-;rport=38000. Max-Forwards: 69. Contact: <sip:thomas@87.xxx.xxx.99:38000>. To: "Joe Blow"<sip:joe@blow.com>;tag=ee11f527. From: "Tom"<sip:tom@blow.com>;tag=5c63711f. Call-ID: NTFjYmM5Y2JhNWE5MzU2NzQ1NTMxYTg2NGQ3ZDYzNmM.. CSeq: 1 ACK. User-Agent: X-Lite release 1006e stamp 34025. Content-Length: 0. P-hint: rr-enforced. . As you can see, the code is appending 127.0.0.1 for no good reason (at least that I can tell) Any ideas (bug fixes, or tricks...)? Nick

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[Users] Using LCR
by Bruno Machado 21 Nov '06

21 Nov '06

Hi all I'm working with LCR and I'm trying the follow situation: I have 5 devices that are using 3 carries. Look the LCR table below: prefix | from_uri | grp_id | priority 5511 % 1 0 5512 % 2 0 5521 % 3 0 55 % 1 0 It works fine. But I want my voip uses only the route 2 wherever I call. I tried to use 'sip:1234%' in from_uri column but didnt work how I want. If I insert a new line: 55 sip:1234% 2 0 and call to '5511XXXXXX', the select route will be '1' and not '2'. It will work if I change to: 5511 sip:1234% 2 0 This way works fine. But I dont want to duplicate all the lines in the table when I want to priority a route to a specific user. The 'preferred' route is that have a longest prefix in the table. The module LCR tells: "Matching gateways are then ordered for forwarding purpose (1) according to longest user part match, (2) according to gateway's priority (3) randomly" There are a way to force Openser to select the line that has the longest 'from_uri'? Thanks a lot. Bruno Machado _______________________________________________________ Novidade no Yahoo! Mail: receba alertas de novas mensagens no seu celular. Registre seu aparelho agora! http://br.mobile.yahoo.com/mailalertas/

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