sr-users November 2006

sr-users@lists.kamailio.org

189 participants
306 discussions

by Benito Caracuel

Hello All, Basic Concept: I have to change the Path MSRP into SDP field in an INVITE and relay the new INVITE. SDP EXAMPLE: v=0 o=686873200 3372497915 3372497916 IN IP4 80.27.211.205 s=SipSession c=IN IP4 80.27.211.205 t=0 0 m=message 49152 TCP/MSRP * a=direction:active a=accept-types:text/plain a=path:msrp://80.27.211.205:49152/63331682315468750;tcp When I receive the SDP, I have to change "msrp://80.27.211.205:49152", so I execute an script and I obtain the new SDP into file.txt. Now, I need to replace the old msrp path by the new sdp path, which is into file.txt In my ser.cfg: if (method=="INVITE") { exec_msg("printenv SRCIP > /tmp/invite.txt; cat >> /tmp/invite.txt; /bin/bash /usr/local/ser_snapshot/sbin/controlador"); #I have the new SDP in /tmp/test.txt exec_msg("set `cat /tmp/test.txt`; %var = $1"); #I have the new SDP in $var replace_all("msrp:\/\/[0-9\.:]{2,20}","%$var"); # I don't know why $var = null!!!! I need that $var = cat /tmp/test.txt but i don't know how to do this!!!!! t_relay(); break; }; Please help me, Thank you very much. Ben. _________________________________________________________________ Busca desde cualquier página Web con una protección excepcional. Consigue la Barra de herramientas de Windows Live hoy mismo y GRATUITAMENTE. http://www.toolbar.live.com

17 years, 10 months

[Serusers] Presence - PA AND RLS

by SER LIST

Hi, I am not sure whether PA module access the XCAP server or not? I have users presence rules configured on XCAP server, but all my clients are getting a NOTIFY w/ pending status. What function call in PA accesses the XCAP server? It seems to me that the RLS accesses the XCAP. Thanks in advance _________________________________________________________________ Buy, Load, Play. The new Sympatico / MSN Music Store works seamlessly with Windows Media Player. Just Click PLAY. http://musicstore.sympatico.msn.ca/content/viewer.aspx?cid=SMS_Sept192006

17 years, 10 months

[Serusers] Presence / Eyebeam Question

by SER LIST

I have the following issue: I have two clients (eyebeam softphones). If I modparam(pa, auth, none) then I can see the status of each other. But when modparam(pa, auth, xcap) I cannot see each other's status. I do get a POPUP window on each softphone asking me whether I want to allow the other party to see my status. Even after I say "Allow", I still can't see the status. I am able to write the presence-rules to XCAP server from the eyebeam client. Eyebeam writes the file presrules.xml and rls looks for presence-rules.xml. So I have a softlink between the two files. Do I need to do something to make this work? Thanks _________________________________________________________________ Not only does Windows Live OneCare provide all-in-one PC care to keep your computer protected and well-maintained, but it also makes creating backup files a breeze. Try it today! http://ideas.live.com/programpage.aspx?versionid=b2456790-90e6-4d28-9219-5d…

17 years, 10 months

[Users] exec module : why not in onreply_route ?

by Jean-François SMIGIELSKI

Hello, On 11/13/06 Daniel-Constantin Mierla wrote: [...] > basically should work in onreply route as well, I haven't tried it. You > will not have access to transaction's avps, otherwise you can play and > see if you encounter any problem. Just edit the modules/exec/exec_mod.c, > add ONREPLY_ROUTE to exported functions' flags, recompile and reinstall. > If everything is ok, then we can include it in the trunk. [...] That is what I did last week and till now, I did not *yet* meet any problem with that. The exec module is a really convenient way to test various functionnalities without having to write a custom module. Very useful even in a reply block. Regards, JF Smigielski. ________________________________________________________________________ iBELGIQUE, exprimez-vous ! http://web.ibelgique.com/

17 years, 10 months

[Users] update documentation

by Klaus Darilion

Hi Daniel,Bogdan! Please update the 1.2.x documentation (http://openser.org/docs/modules/1.2.x/): - add domainpolicy module - add new ENUM module features thanks Klaus -- Klaus Darilion nic.at

17 years, 10 months

[Users] Errors whie doing openser integration with radius server

by Sanjeev Manoli

Hi, I am getting following errors as seen in /var/log/message while running openser with radius integration, ***************************************************************************************** Nov 6 20:33:45 lx-dev monit[13565]: 'openser' start: /etc/init.d/openser Nov 6 20:33:45 lx-dev monit[13565]: 'openser' failed to start Nov 6 20:33:45 lx-dev openser: init_tcp: using epoll_lt as the io watch method (auto detected) Nov 6 20:33:45 lx-dev openser: INFO: statistics manager successfully initialized Nov 6 20:33:45 lx-dev openser: StateLess module - initializing Nov 6 20:33:45 lx-dev openser: TM - initializing... Nov 6 20:33:46 lx-dev openser: Maxfwd module- initializing Nov 6 20:33:46 lx-dev openser: AVPops - initializing Nov 6 20:33:46 lx-dev openser: TextOPS - initializing Nov 6 20:33:46 lx-dev openser: ACC - initializing Nov 6 20:33:46 lx-dev openser: AUTH module - initializing Nov 6 20:33:46 lx-dev openser: xl_parse_item: error - bad parameters Nov 6 20:33:46 lx-dev openser: ERROR:avpops:fixup_check_avp: unable to get pseudo-variable in P1 Nov 6 20:33:46 lx-dev openser: ERROR: fix_actions: fixing failed (code=-2) at cfg line 146 Nov 6 20:33:46 lx-dev openser: ERROR: fix_expr : fix_actions error ***************************************************************************************** I am using openser (Version: openser-1.1.0-tls) and radius server ( freeradius-1.1.3) along with radiusclient-ng (radiusclient-ng-0.5.2). I exactly followed the following radius integration documentation from openser web site, http://openser.org/docs/openser-radius-1.0.x.html For your reference I am also attaching the openser.cfg file that I am using. If I remove the radius integration related part from openser.cfg then my openser server starts fine, I have tested it with kphone SIP UA and it works fine. One other question is I get parse error for following 2 statements in openser.cfg (I commented them to make forward progress). Please advice the right syntax to use following modparam statements. modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("avpops", "avp_aliases", "day=i:101;time=i:102") This is kind of urgent for me and I am clueless at this point so really appreciate all your help. Thanks, - San

17 years, 10 months

[Users] which is better: avp_db_load with scheme or avp_db_query?

by Alan Crosswell

Hi, I want to pull the First+Last names of the subscriber for the $ru so I can stick an RPID in a 180 or 183 response (Polycom phones display the called party name when you do this). Is it better practice to define two schemes and use avp_db_load to get the first and last name columns out of subscriber table or to use avp_db_query and do it in one shot? Thanks. /a

17 years, 10 months

[Serusers] Asterisk + SIP integration for SIPLE. Newb needs advice.

by Ryan Churches

Ive administrate several Asterisk servers, and sometimes one company will have multiple servers with their own respective users. Unfortunately users of one server are unable to see the presence state of a user on another server. I understand that SER can (a) be my presence agent or (b) can pass SUBSCRIBErs to a respective asterisk box, but doesnt that result in (a) clients subscribing to SER cannot see the presence states of local channels on asterisk (i.e. a parking spot), or (b) clients not being able to see the presence of clients SUBSCRIBing to a different asterisk server (back to square 1)? Can someone help me wrap my mind around how this could/should work. Perhaps it can't be made to work at all. All i'm looking for is a general direction to research in. The question as best i can phrase it susinctly is "How can I use SER to allow clients of disparate asterisk servers on disparate lans to have a centralized presence roster?"

17 years, 10 months

[Users] CANCEL/ACK processing

by Mark Kent

Observant readers will recognize from the following that this question is related to earlier posts from me... that is: there is something fundamental that I just don't get. If someone is eager to answer all my questions, then please search ahead for the question marks (?). Also, what is the bigger picture that I am missing? Please use as many words as necessary to explain. Thanks, -mark The players: CUSTOMER-IP: the UserAgent OPENSER-IP: my openser box OPENSER.FQDN: the fully qualified domain name of my openser box PSTNGW-IP: the PSTN gateway My openser box gets this CANCEL: U 2006/11/06 18:40:10.689785 CUSTOMER-IP:5060 -> OPENSER-IP:5060 CANCEL sip:011445551212@OPENSER.FQDN:5060 SIP/2.0 From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP> Call-ID: CALLID@CUSTOMER-IP CSeq: 2 CANCEL Via: SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH Max-Forwards: 70 Supported: 100rel,replaces Content-Length: 0 It ends up in my standard stateful relay handler where I t_relay() it (should I?) and this gets generated: # U 2006/11/06 18:40:10.691301 OPENSER-IP:5060 -> PSTNGW-IP:5060 CANCEL sip:445551212@PSTNGW-IP;user=phone SIP/2.0 Record-Route: <sip:011445551212@OPENSER-IP;lr=on;ftag=TAG> From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP> Call-ID: CALLID@CUSTOMER-IP CSeq: 2 CANCEL Via: SIP/2.0/UDP OPENSER-IP;branch=BRANCH2.0 Via: SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH Max-Forwards: 69 Supported: 100rel,replaces Content-Length: 0 and this also happens: # U 2006/11/06 18:40:10.691342 OPENSER-IP:5060 -> CUSTOMER-IP:5060 SIP/2.0 200 canceling From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP>;tag=TAG2 Call-ID: CALLID@CUSTOMER-IP CSeq: 2 CANCEL Via: SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH Content-Length: 0 How was that last packet generated, where was it done? It's not in my openser.cfg, should I have sent back just such a message? If so, would this automagically sent message still have been sent? Next I get these two packets: # U 2006/11/06 18:40:10.761582 PSTNGW-IP:5060 -> OPENSER-IP:5060 SIP/2.0 200 OK Call-ID: CALLID@CUSTOMER-IP From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP> Content-Length: 0 CSeq: 2 CANCEL Via: SIP/2.0/UDP OPENSER-IP;branch=BRANCH2.0,SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH Contact: sip:445551212@PSTNGW-IP:5060;user=phone Record-Route: <sip:011445551212@OPENSER-IP;lr=on;ftag=TAG> # U 2006/11/06 18:40:10.761616 PSTNGW-IP:5060 -> OPENSER-IP:5060 SIP/2.0 487 Request Terminated Call-ID: CALLID@CUSTOMER-IP From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP>;tag=20505 Content-Length: 0 CSeq: 2 INVITE Via: SIP/2.0/UDP OPENSER-IP;branch=BRANCH2.0,SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH and my openser box replies with : # U 2006/11/06 18:40:10.762238 OPENSER-IP:5060 -> PSTNGW-IP:5060 ACK sip:445551212@PSTNGW-IP;user=phone SIP/2.0 Via: SIP/2.0/UDP OPENSER-IP;branch=BRANCH2.0 From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG Call-ID: CALLID@CUSTOMER-IP To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP>;tag=20505 CSeq: 2 ACK Content-Length: 0 and sends on this to the UAC: # U 2006/11/06 18:40:10.762377 OPENSER-IP:5060 -> CUSTOMER-IP:5060 SIP/2.0 487 Request Terminated Call-ID: CALLID@CUSTOMER-IP From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP>;tag=20505 Content-Length: 0 CSeq: 2 INVITE Via: SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH Now, how was that generated? Is this part of the "tm" module? And then when I get this ACK: # U 2006/11/06 18:40:10.772875 CUSTOMER-IP:5060 -> OPENSER-IP:5060 ACK sip:011445551212@OPENSER.FQDN:5060 SIP/2.0 From: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP>;tag=TAG To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP>;tag=20505 Call-ID: CALLID@CUSTOMER-IP CSeq: 2 ACK Via: SIP/2.0/UDP CUSTOMER-IP:5060;branch=BRANCH Max-Forwards: 70 Contact: <sip:CUST-PHONE-NUMBER@CUSTOMER-IP:5060;transport=UDP> Content-Length: 0 What should I do with it? I do nothing... if I should do nothing is there any special way I should do it? Or is this just handled by the tm module? What I don't get is the CANCEL is directed AT my openser box: CANCEL sip:011445551212@OPENSER.FQDN:5060 SIP/2.0 To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP> just like the ACK: ACK sip:011445551212@OPENSER.FQDN:5060 SIP/2.0 To: <sip:011445551212@OPENSER.FQDN:5060;transport=UDP>;tag=20505 and I t_relay() the CANCEL but sit on the ACK...

17 years, 10 months

[Users] Some problem in Hung up ...... help me.........

by raviprakash sunkara

Hello Users, When I keep my Sip Server into firewall_router is not hung up, But with Direct Modem to My Sip Server then it hung ups finely, Sip server listen on 192.168.2.90 with public ip xx.xx.xx.xx.xx. See the Loation tables +----------+--------+----------------------------+------------------------+------+---------------------+-------+-----------------------------------------------+-------+---------------------+-------+----------------------+-----------------------+---------+ | username | domain | contact | received | path | expires | q | callid | cseq | last_modified | flags | user_agent | socket | methods | +----------+--------+----------------------------+------------------------+------+---------------------+-------+-----------------------------------------------+-------+---------------------+-------+----------------------+-----------------------+---------+ | 102 | | sip:102@59.144.72.234:5060 | sip:59.144.72.234:5060 | NULL | 2006-11-07 03:17:42 | -1.00 | 414D9B3694024D7D94F21AF17E63460E(a)xx.xx.xx.xx | 61160 | 2006-11-07 02:47:42 | 1 | X-Lite release 1103m | udp:192.168.2.90:5060 | NULL | | 101 | | sip:101@61.17.248.68:5060 | sip:61.17.248.68:5060 | NULL | 2006-11-07 03:17:04 | -1.00 | 532BD0D57F89495CB42AFA7B7D760BE2(a)xx.xx.xx.xx | 9830 | 2006-11-07 02:47:04 | 1 | X-Lite release 1103m | udp:192.168.2.90:5060 | NULL | +----------+--------+----------------------------+------------------------+------+---------------------+-------+-----------------------------------------------+-------+---------------------+-------+----------------------+-----------------------+---------+ my openser nat configure is like route[2] { if(nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_resgiter(); force_rport(); setflag(6); } else { if (method == "INVITE") { fix_nated_sdp("2") }; fix_nated_contact() force_rport() setflag(6); }; } In X-lite log..... After the INVITE Method got response from server ( xx.xx.xx.xx.xx) .... ACK is sending like That ..... and also Bye method After the INvite is posseed Ack Send >> 192.168.2.90:5060 ... But CANCEL method send properly............ Help me -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

17 years, 10 months

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sr-users November 2006