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37464 discussions

[Users] which is better: avp_db_load with scheme or avp_db_query?
by Alan Crosswell 13 Nov '06

13 Nov '06

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

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[Serusers] Asterisk + SIP integration for SIPLE. Newb needs advice.
by Ryan Churches 13 Nov '06

13 Nov '06

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?"

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[Users] CANCEL/ACK processing
by Mark Kent 13 Nov '06

13 Nov '06

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...

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[Users] Some problem in Hung up ...... help me.........
by raviprakash sunkara 13 Nov '06

13 Nov '06

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

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[Users] exec module : why not in onreply_route ?
by Jean-François SMIGIELSKI 13 Nov '06

13 Nov '06

Hello, all. I would know why we should avoid using the exec module exported functions outside out the request_route and on_failure_route blocks? (and so ... why it is forbidden). Answers carry useful informations (e.g. SDP) that we could want to parse with an external command. Maybe is there a more appropriate way to achieve this, and I mail you for this reason. Thanks once again, JF Smigielski ________________________________________________________________________ iBELGIQUE, exprimez-vous ! http://web.ibelgique.com/

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[Serusers] off-topic: Sofphones with Redirect Suport
by Felipe Louback 13 Nov '06

13 Nov '06

Hi all, I need a softphone that supports redirect. Let me explain: In all free softphones I download, I have an "outbound proxy" field. So, if my server returns and 302 Moved Temporarily, the softphone will get the Contact header field and try to call the user again using the outbound proxy, and the outbound will reply with an 302 and I will get an loop. I need a softphone who will use a location server to query the location of a user and after that the softphone will make the call directly, without using a proxy. Could anyone point me a softphone that will do this? Best Regards, Felipe -- Master Student - Electrical Engineering Department Computer Engineering and Telecommunications Research Group Universidade Federal de Minas Gerais - Brazil "Come to me, all you who are weary and burdened, and I will give you rest." Matthews 11:28

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[Serusers] SER pre-release ser_ctl
by SER LIST 13 Nov '06

13 Nov '06

Hello, When I try to run the ser_ctl command to do something like ser_ctl version or any other command, I get the error ser_ctl: (111, 'Connection refused'): error SER is running while I type the command. Any hints will be appreciated. _________________________________________________________________ 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

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Re: [Serusers] SER pre-release - PRESENCE ISSUES
by SER LIST 12 Nov '06

12 Nov '06

Thank you. settig the xcap_root for the xcap module fixed the two errors relating to xcap_root. BUT I am still getting the two errors relating to the lookup_domain command, namely: ------------------------------------- ser: parse error (192,24-25): unknown command, missing loadmodule? ser: parse error (261,26-27): unknown command, missing loadmodule? ser: ERROR: bad config file (2 errors) ser startup failed ---------------------------------- I am using the presence example file. The excerpt from my ser.cfg file is (I have marked the LINES where the error occurs): ---- start --- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/avp.so" loadmodule "/usr/local/lib/ser/modules/avpops.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/domain.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/dialog.so" loadmodule "/usr/local/lib/ser/modules/rls.so" loadmodule "/usr/local/lib/ser/modules/pa.so" loadmodule "/usr/local/lib/ser/modules/xcap.so" loadmodule "/usr/local/lib/ser/modules/presence_b2b.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/fifo.so" loadmodule "/usr/local/lib/ser/modules/xmlrpc.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" { LINES TRUNCATED} if (uri==myself) { if (!lookup_domain("To")) { <-------------------------- LINE 192 xlog("L_ERR", "Unknown domain to: %tu from: %fu\n"); route(1); break; } if (method=="SUBSCRIBE") { if (!t_newtran()) { sl_reply_error(); break; }; (lines truncated) if (lookup_domain("From")) { <---------------------- LINE 261 if (lookup_user("From")) { if (is_simple_rls_target("$uid-list")) { # log(1, "it is simple subscription!\n"); # takes From UID and makes XCAP query for user's # list named "default" if (!query_resource_list("default")) { t_reply("404", "No such user list"); break; } } } } ----------- END --------------------- Any suggestions?? THanks in advance >From: samuel <samu60(a)gmail.com> >To: "SER LIST" <sergrp(a)hotmail.com> >CC: serusers(a)iptel.org, serdev(a)iptel.org >Subject: Re: [Serusers] SER pre-release - PRESENCE ISSUES >Date: Fri, 10 Nov 2006 10:14:17 +0100 > >XCAP parameters where removed from rls and pa and a new module, called >xcap contains the xcap root so you have to delete pa and rls xcap >parameter, load the new module xcap.so and set the xcap_root parameter >of xcap module to the directory having the doc structure. > >About the lookup_domain, there can be a typo. Can you post the config >file syntax?? > >Hope it helps! > >Samuel > >2006/11/10, SER LIST <sergrp(a)hotmail.com>: >>I installed the SER pre-release along w/ all modules. The problem is that >>presence now does not run (it was working w/ the presence-release-10.99). >>The errors I get are: >>---------------------------------------------------------------------------------------------------------- >>set_mod_param_regex: parameter <xcap_root> not found in module <rls> >>ser: parse error (82,20-21): Can't set module parameter >>set_mod_param_regex: parameter <auth_xcap_root> not found in module <pa> >>parse error (97,20-21): Can't set module parameter >>ser: parse error (186,24-25): unknown command, missing loadmodule? >>parse error (255,26-27): unknown command, missing loadmodule? >>ERROR: bad config file (4 errors) >>------------------------------------------------------------------------------------------------------------- >>I checked the pa module code and also the rls code and the parameters >>auth_xcap_root and xcap_root are NOT defined !!!! >> >>The error on line 185 and 255 refer to the "Lookup_domain" function. I >>verified that the domain module has been compiled and included in the >>ser.cfrg file _________________________________________________________________ Ready for the world's first international mobile film festival celebrating the creative potential of today's youth? Check out Mobile Jam Fest for your a chance to WIN $10,000! www.mobilejamfest.com

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[Users] Forwarding requests
by Harry Harcrow 12 Nov '06

12 Nov '06

I have my SER server running on a Linux machine and I am attempting to forward some requests to a Cisco SIP server. I have been running X-lite clients on windows and the same Linux box that the SER server is running on. I have no problem completing calls between the X-lite clients OR with either X-lite client calling the Cisco SIP server directly. However, when I try to make a call from either X-lite client through the SER server I eventually get timeout methods. I am trapping those calls and wrote the following route block: route [5] { record_route(); forward(uri:host, uri:port); } I have tried replacing the forward with send() and t_relay() method calls, but nothing seems to work. The header from the X-lite client directly to the Cisco SIP server, which works is: INVITE sip:2158200@65.221.7.xxx SIP/2.0 Via: SIP/2.0/UDP 166.34.149.xxx:5061;rport;branch=z9hG4bK5963AD09C90F6F3A16F55A271C98F573 From: Harry Harcrow <sip:1000@166.34.149.xxx:5061>;tag=1279481120 To: <sip:2158200@65.221.7.xxx> Contact: <sip:1000@166.34.149.xxx:5061> Call-ID: 28E5B723-9B02-BF38-1A81-1FECC0A85D5E(a)166.34.149.xxx CSeq: 6254 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 313 The header from SER to the Cisco server, which does not work is: INVITE sip:2158200@65.221.7.xxx SIP/2.0 Record-Route: <sip:166.34.149.xxx;ftag=2061592395;lr=on> Via: SIP/2.0/UDP 166.34.149.xxx;branch=0 Via: SIP/2.0/UDP 166.34.149.xxx:5061;rport=5061;branch=z9hG4bK4EA9A90B8CE7CE52BFB69DA431560E5 7 From: Harry Harcrow <sip:1000@166.34.149.xxx:5061>;tag=2061592395 To: <sip:2158200@65.221.7.xxx> Contact: <sip:1000@166.34.149.xxx:5061> Call-ID: 38CA166F-59A6-E44B-F34E-9E1D6307698B(a)166.34.149.xxx CSeq: 3009 INVITE Max-Forwards: 16 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 313 The only thing that I can see different is the record_route record and an additional Via record. Does anyone have any idea what I am doing wrong? Thanks Harry Harcrow

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[Serusers] Suggestion
by Eddie Johnson Jr 12 Nov '06

12 Nov '06

I am new to setting up SER and this is the test setup. 1. DHCP running from router 2. SJ Phone on laptop 192.168.X.X 3. Grandstream Budgetone 101 192.168.X.X 4. SIP Express Router 192.168.X.X 5. Asterisk Server 192.168.X.X attached to tdm400p card is an analog phone Will this work internally? There is no connection to a telco. The SJ phone is showing the following: nat/firewall: Port Restricted Cone NAT SIP calls use outbound proxy: 192.168.X.X (SER) Will I have to register the SJ, Grandstream and Asterisk PBX? Ed

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