sr-users February 2006

sr-users@lists.kamailio.org

189 participants
374 discussions

[Serusers] More info: lookup(): '' Not found in usrloc
by Kim Culhan 25 Feb '06

25 Feb '06

The ser.cfg in use is: features-callfwd.cfg 15 2005-09-15 17:37:10Z The portion of the config file with this logic appears to be: lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found - lookup_location"); break; }; route(1); } regards -kim -- w8hdkim(a)gmail.com

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[Serusers] ser and firewall on same server
by Corey 25 Feb '06

25 Feb '06

Dual homed server, operating as a vpn server, has one public and one private ip. Remote client pc's ( scattered around the internet ) connect to server's private ip (via openvpn) tunnel, server NATs traffic originating from private/vpn interface through server's public ip off to the internet at large; all this works fine: pc's route all traffic through vpn. Now, if this server is also running an instance of ser, via which the client pc's register - therefore, the NAT'ing takes place on the server _itself_, rather than on some intermediary server/router _between_ the pc's and ser server - is it still necessary to run mediaproxy or nathelper? I'm still cutting my teeth on voip in general, and ser in particular - however our environment seems quite different to what I have seen and read in all the docs and material regarding sip/ser and the NAT issues involved -- all of these seem to assume that the NAT'ing takes place on external/3rd-party servers/routers rather than on the ser server itself. As such, I'm having all kinds of fun trying to figure out what the heck I'm doing - whether I'm on the right track, and am merely suffering from some small misconfig somewhere that is preventing all this to work, or whether this scenerio is so outlandish that I'm unlikely to ever succeed period. So what I'm hoping to get, is some indication from you experts on whether this environment I've tried to conceptualize above is possible or not; and any tips or suggestions you may have. I'll be happy to provide further clarification on anything that I haven't described clearly. Many thanks! Cheers, Corey

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[Serusers] local ACK found -> dropping it!
by Corey 25 Feb '06

25 Feb '06

Having some troubles with a new voip gateway service we're using; hoping someone could provide any advice. To sum up: Call is initiated from pots line; gets routed to voip gateway provider; gateway forwards invite to our ser server; one ring then busy signal is heard on originating/requesting pots line; the destined softphone UA, registered via our ser server, never gets call. As far as I can fathom, ser recieves an INVITE forwarded/routed from the gateway, then ser requests auth/cred, then gateway responds with an ACK and at that point everything just stops. Assistance is much appreciated, as I'm unable to determine what may be causing the issue. 208.xxx.xxx.xx = 3rd-party/commercial voip gateway server 206.xx.xx.x = our ser proxy server 1234567890 = originating pots line 8889997777 = softphone UA number ( aliased to sip uri 'blah(a)206.xx.xx.x' ) ngrep: # U 208.xxx.xxx.xx:5060 -> 206.xx.xx.x:5060 INVITE sip:8889997777@206.xx.xx.x SIP/2.0..Via: SIP/2.0/UDP 208.xxx.xxx.xx:5060;branch=z9hG4bK40f0830c;rport..From: "John Doe" <sip:1234567890@208.xxx.xxx.xx>;tag=as5f35b551..To: <s ip:8889997777@206.xx.xx.x>..Contact: <sip:1234567890@208.xxx.xxx.xx>..Call-ID: 02c132b659e84c2321debbf226051e3d@208.xxx.xxx.xx..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Max-Forwards: 70 ..Date: Fri, 24 Feb 2006 22:28:05 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Content-Type: application/sdp..Content-Length: 397....v=0..o=root 32669 32669 IN IP4 208.xxx.xxx.xx..s=session..c=IN IP4 208.xxx.xxx.xx..t=0 0..m=audio 10784 RTP/AVP 0 18 8 3 111 4 97 101..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:8 PCMA/80 00..a=rtpmap:3 GSM/8000..a=rtpmap:111 G726-32/8000..a=rtpmap:4 G723/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -.. # U 206.xx.xx.x:5060 -> 208.xxx.xxx.xx:5060 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 208.xxx.xxx.xx:5060;branch=z9hG4bK40f0830c;rport=5060..From: "John Doe" <sip:1234567890@208.xxx.xxx.xx>;tag=as5f35b551..To : <sip:8889997777@206.xx.xx.x>;tag=2d38da4c38ea547b795405b32554c3bb.f3cb..Call-ID: 02c132b659e84c2321debbf226051e3d@208.xxx.xxx.xx..CSeq: 102 INVITE..Proxy-Authenticate: Digest realm="208.1 39.204.245", nonce="43fe339b33d7c58a0d89e4711838584d"..Server: Sip EXpress router (0.9.4 (i386/linux))..Content-Length: 0..Warning: 392 206.xx.xx.x:5060 "Noisy feedback tells: pid=2 974 req_src_ip=208.xxx.xxx.xx req_src_port=5060 in_uri=sip:8889997777@206.xx.xx.x out_uri=sip:8889997777@206.xx.xx.x via_cnt==1".... # U 208.xxx.xxx.xx:5060 -> 206.xx.xx.x:5060 ACK sip:8889997777@206.xx.xx.x SIP/2.0..Via: SIP/2.0/UDP 208.xxx.xxx.xx:5060;branch=z9hG4bK40f0830c;rport..From: "John Doe" <sip:1234567890@208.xxx.xxx.xx>;tag=as5f35b551..To: <sip: 8889997777(a)206.xx.xx.x>;tag=2d38da4c38ea54795405554c3bb...Contact: <sip:1234567890@208.xxx.xxx.xx>..Call-ID: 02c132b659e84c2321debbf226051e3d@208.xxx.xxx.xx..CSeq: 102 ACK..User-A gent: Asterisk PBX..Max-Forwards: 70..Content-Length: 0.... ... end of transmission; no further exchange after gateway ACK; originating pots line hears a single ring, then fast busy signal; softphone UA never recieves INVITE or anything ser debug: IP Request: 0(2974) method: <INVITE> 0(2974) uri: <sip:8889997777@206.xx.xx.x> 0(2974) version: <SIP/2.0> 0(2974) parse_headers: flags=1 0(2974) Found param type 232, <branch> = <z9hG4bK40f0830c>; state=6 0(2974) Found param type 235, <rport> = <n/a>; state=17 0(2974) end of header reached, state=5 0(2974) parse_headers: Via found, flags=1 0(2974) parse_headers: this is the first via 0(2974) After parse_msg... 0(2974) preparing to run routing scripts... 0(2974) parse_headers: flags=128 0(2974) end of header reached, state=9 0(2974) DEBUG: get_hdr_field: <To> [31]; uri=[sip:8889997777@206.xx.xx.x] 0(2974) DEBUG: to body [<sip:8889997777@206.xx.xx.x> ] 0(2974) get_hdr_field: cseq <CSeq>: <102> <INVITE> 0(2974) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2974) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2974) DEBUG: add_param: tag=as5f35b551 0(2974) end of header reached, state=29 0(2974) parse_headers: flags=256 0(2974) DEBUG: get_hdr_body : content_length=397 0(2974) found end of header 0(2974) find_first_route: No Route headers found 0(2974) loose_route: There is no Route HF 0(2974) is_local(): Realm '206.xx.xx.x' is local 0(2974) parse_headers: flags=16384 0(2974) pre_auth(): Credentials with given realm not found 0(2974) build_auth_hf(): 'Proxy-Authenticate: Digest realm="208.xxx.xxx.xx", nonce="43fe35e639b33d7c58a0d89e47df7fe11838584d" ' 0(2974) parse_headers: flags=-1 0(2974) check_via_address(208.xxx.xxx.xx, 208.xxx.xxx.xx, 0) 0(2974) DEBUG:destroy_avp_list: destroying list (nil) 0(2974) receive_msg: cleaning up 0(2974) SIP Request: 0(2974) method: <ACK> 0(2974) uri: <sip:8889997777@206.xx.xx.x> 0(2974) version: <SIP/2.0> 0(2974) parse_headers: flags=1 0(2974) Found param type 232, <branch> = <z9hG4bK40f0830c>; state=6 0(2974) Found param type 235, <rport> = <n/a>; state=17 0(2974) end of header reached, state=5 0(2974) parse_headers: Via found, flags=1 0(2974) parse_headers: this is the first via 0(2974) After parse_msg... 0(2974) preparing to run routing scripts... 0(2974) parse_headers: flags=4 0(2974) DEBUG: add_param: tag=2d38da4c38ea547b795405b32554c3bb.f3cb 0(2974) end of header reached, state=29 0(2974) DEBUG: get_hdr_field: <To> [73]; uri=[sip:8889997777@206.xx.xx.x] 0(2974) DEBUG: to body [<sip:8889997777@206.xx.xx.x>] 0(2974) DEBUG: sl_filter_ACK : local ACK found -> dropping it! 0(2974) DEBUG:destroy_avp_list: destroying list (nil) 0(2974) receive_msg: cleaning up

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[Users] Custom CNAM module
by Brandon Price 24 Feb '06

24 Feb '06

I am interested in a custom openser module. It needs to perform cnam (calling party name) resolution through a third party webserver. The module should perform the following actions on an incoming call where SIP Display info: == " "(nothing). 1.Query our local cnam_custom db (mysql) to determine if the customer being called has a custom cnam for the number calling them 2.Query our local cnam_cache db (mysql) to determine if we have already queried the third party database for the number calling in 3.Query the third party database (https) to determine cnam if neither of the first two queries returned anything 4.Insert the result into the SIP Display info: portion of the INVITE so that the correct name will show up on my users callerid. If 1,2,or 3 return a non null value skip to to 4. I would really love to do this with an external script that grabs the values, inserts them into the database, and AVP's but have been this far been unable to build a working solution. I tried... avp_db_load("$from","cnam/$cnam_scheme"); xlog("cnam value [$avp(s:cnam)]\n"); uac_replace_from("$cnam",""); or something like that to no avail. Please reply with your hourly rate if you will need to write the module, if you can do it with $avp's?, and include total cost for the job. I would prefer to do it with avp's. Thank you.

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R: [Serusers]
by Piccari Gianluca 24 Feb '06

24 Feb '06

Hi all, does anyone know if SER supports Number Portability. Thank you in advance

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[Serusers] serweb question
by frankie chiang 24 Feb '06

24 Feb '06

Dear all, 1. How can I change username prefix 8 to other under user registration ? 2. How can I set default timezone for all users ? Best Regard, __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers]
by Piccari Gianluca 24 Feb '06

24 Feb '06

Hi all, does anyone know if SER supports Number Portability. Thank you in advance -----Messaggio originale----- Da: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]Per conto di gokhan uslu Inviato: venerdì 24 febbraio 2006 16.54 A: serusers(a)lists.iptel.org Oggetto: [Serusers] radius authentication with ip address of a user Hi, I have installed ser and freeradius into same machine. everything is working fine. i have created users at the freeradius. but while creating a user, in the user information i had to use the ip of the SER not the ip of the user gateway(a sip gateway) I mean i want to register the user according to its username , password and ALSO IP. my user`s ip: 83.100.100.23 my SER ip: 212.175.10.36 when i edit user file for freeradius and when i write this: myuser(a)83.100.100.23 for user name it does not work but when i write myuser(a)212.175.10.36 then it can authenticate. but this is unusefull for me at the moment. I must be authenticated according to user`s ip, not according to SER`s ip please help meee thanks a lot _____ Relax. Yahoo! Mail virus <http://us.rd.yahoo.com/mail_us/taglines/virusall/*http://communications.yah…> scanning helps detect nasty viruses!

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[Serusers] radius authentication with ip address of a user
by gokhan uslu 24 Feb '06

24 Feb '06

Hi, I have installed ser and freeradius into same machine. everything is working fine. i have created users at the freeradius. but while creating a user, in the user information i had to use the ip of the SER not the ip of the user gateway(a sip gateway) I mean i want to register the user according to its username , password and ALSO IP. my user`s ip: 83.100.100.23 my SER ip: 212.175.10.36 when i edit user file for freeradius and when i write this: myuser(a)83.100.100.23 for user name it does not work but when i write myuser(a)212.175.10.36 then it can authenticate. but this is unusefull for me at the moment. I must be authenticated according to user`s ip, not according to SER`s ip please help meee thanks a lot --------------------------------- Relax. Yahoo! Mail virus scanning helps detect nasty viruses!

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[Users] BYE retransmits too quick!
by Greg Fausak 24 Feb '06

24 Feb '06

I'm doing an interop test with a well known ITPS, they are kicking back these BYE retransmits, like: SIP MESSAGE 18 216.138.115.30:5061(www.testnode-3.com) -> 209.247.16.2:5060(deneps551.Denver1) UDP Frame 18 17/Feb/06 17:09:36.8142 TimeFromPreviousSipFrame=0.0010 TimeFromStart=8.0692 BYE sip:209.247.16.2:5060 SIP/2.0 Record-Route: <sip:216.138.115.30:5061;ftag=245-192.168.10.1;lr> Record-Route: <sip:216.138.115.27;ftag=245-192.168.10.1;lr> Via: SIP/2.0/UDP 216.138.115.30:5061;branch=z9hG4bK036d.a99621d6.0 Via: SIP/2.0/UDP 216.138.115.27;branch=z9hG4bK036d.1fe4ca41.0 Via: SIP/2.0/UDP 198.41.9.65:5060;branch=z9hG4bK245-192.168.10.1.256 Via: SIP/2.0/UDP 192.168.10.23:5060;branch=z9hG4bK1371-192.168.10.23.1374 From: <sip:+18663072489@216.138.115.30:5061>;tag=245-192.168.10.1 To: "Unknown" <sip:+17205626376@209.247.16.2>;tag=VPSF50603522629634 Call-ID: DEN05020060217230928060500(a)209.244.48.214 CSeq: 101 BYE Max-Forwards: 67 Record-Route: <sip:198.41.9.65:5060> Contact: "8663072489" <sip:+18663072489@198.41.9.65:1074> SIP MESSAGE 19 216.138.115.30:5061(www.testnode-3.com) -> 209.247.16.2:5060(deneps551.Denver1) UDP Frame 19 17/Feb/06 17:09:36.8424 TimeFromPreviousSipFrame=0.0282 TimeFromStart=8.0974 BYE sip:209.247.16.2:5060 SIP/2.0 Record-Route: <sip:216.138.115.30:5061;ftag=245-192.168.10.1;lr> Record-Route: <sip:216.138.115.27;ftag=245-192.168.10.1;lr> Via: SIP/2.0/UDP 216.138.115.30:5061;branch=z9hG4bK036d.a99621d6.0 Via: SIP/2.0/UDP 216.138.115.27;branch=z9hG4bK036d.1fe4ca41.0 Via: SIP/2.0/UDP 198.41.9.65:5060;branch=z9hG4bK245-192.168.10.1.256 Via: SIP/2.0/UDP 192.168.10.23:5060;branch=z9hG4bK1371-192.168.10.23.1374 From: <sip:+18663072489@216.138.115.30:5061>;tag=245-192.168.10.1 To: "Unknown" <sip:+17205626376@209.247.16.2>;tag=VPSF50603522629634 Call-ID: DEN05020060217230928060500(a)209.244.48.214 CSeq: 101 BYE Max-Forwards: 67 Record-Route: <sip:198.41.9.65:5060> Contact: "8663072489" <sip:+18663072489@198.41.9.65:1074> Why is the BYE being sent so quickly? This is an intermediate UAS proxy, and every now and then it spits out an extra BYE like this. Shouldn't this wait 500ms before doing a retry? The rest of the trace is clean. I can forward the whole thing if there is any interest. -g -- Greg Fausak greg(a)thursday.com

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Re: [Users] Openser TLS: problem on port 5061
by Klaus Darilion 24 Feb '06

24 Feb '06

Hi Martin! please always cc the list Martin Petraschek wrote: > I want openser to listen on port 5060 for UDP and TCP, and on port 5061 > for TCP/TLS. How could I achieve this? My current config is as follows: > > listen=udp:10.1.2.9:5060 > listen=tcp:10.1.2.9:5060 > disable_tls = 0 > listen = tls:10.1.2.9:5061 this is fine > tls_port_no=5061 this is not needed > > Is this correct? > > > What is the request URI and destination before sending (try > > xlog("L_ERR","duri=$du, ruri=$ru") before t_relay()) > > With the openser.cfg as shown above the error message I described in my > mail previous mail went away. However, when I try to make a call, I get > the following: > > Feb 24 15:01:48 server2 /usr/local/sbin/openser[4246]: duri=, > ruri=sip:user2@192.168.0.47:3985;transport=tls;line=ojn9itpa > Feb 24 15:01:49 server2 /usr/local/sbin/openser[4223]: child process > 4246 exited by a signal 11 Sorry, i have no idea why it crashes here. Try to increase the debug level (debug=4) and watch the logfile. regards klaus > Feb 24 15:01:49 server2 /usr/local/sbin/openser[4223]: core was generated > Feb 24 15:01:49 server2 /usr/local/sbin/openser[4223]: INFO: terminating > due to SIGCHLD > > If you need more information, please tell me. > > Thanks, > > Martin > > > >

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sr-users February 2006