[00:06] hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) left irc: Read error: 104 (Connection reset by peer) [00:06] hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) joined #openser. [00:10] habakuk (n=chatzill@c-24-6-224-243.hsd1.ca.comcast.net) joined #openser. [00:28] still wont register [00:28] pre_auth(): Credentials with given realm not found [00:28] never had probs before [00:28] lol [00:41] kl [00:41] did you give the proper realm ? [00:54] yep [00:54] error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy [00:54] now [00:54] still 401 [00:54] hmm [00:54] thats using hte generated one from site you gave [00:54] your just dont reg [00:55] are you running the media proxy python script shit ? [00:55] yes.. but should auth anyhow [00:55] the agprocrap one [00:56] Route 2 Req Auth For < [00:57] tuch in route 2 ? and its showing the bad sip crap.. its actualy showint [00:57] xlog("L_INFO", " Route 2 Req Auth For $(ct), URI = $(ru)\n"); [00:57] i see CT as my nated device.. uri as the domain.. nothin else.. $(ru) shows sip:myproxyname [00:58] not sure man - other than Caller*ID that config works in all respects for me [00:58] hmm its sendin f register but not coming back [00:59] firewall/nat puckin things up? [00:59] im nated the proxy isnts [00:59] i guess that why [01:00] OUTE 2 client nat test 7 [01:00] ok [01:01] if (!search("^Contact:\ +\*") && client_nat_test("7")) { [01:01] that is true in my case [01:02] wats diff in proxy auth and web auth ? [01:05] one gets used for registrations [01:05] the other for invites [01:06] hmm any [01:06] it seems.. that $(au) is emtpy [01:07] for soem reason [01:07] sweet, I finally got session timers to work [01:08] could media proxy mess things up on reg ? [01:08] not sure on config for that [01:08] i doubt it [01:09] could not connect to media proxy is a major problem [01:09] well.. its just not coming back to me.. the register atempt on my local asterisk sip debug shows they never come back [01:10] Bobthehunter: you never get a 401 you mean? [01:10] exact [01:10] if (!search("^Contact:\ +\*") && client_nat_test("7")) { [01:10] this = TRE [01:11] ok can you run ngrep on openser box? [01:11] somebody is blocking [01:11] where is the 401 going ? [01:11] im running ngrep [01:11] all ok [01:11] just not coming back [01:11] so openser is not sending it? [01:11] exactly [01:12] if i dial my normal asterisk trunk .. that works.. but dialing hte ser trunk never gets reg'ed [01:12] ngrep time [01:12] so i know TCP setting sare ok [01:12] time ? [01:12] I've never used mediaproxy. I've always used rtpproxy [01:12] yes metoo.. but it seems ill need this crap [01:13] as this point i think mediproxy is interferecing somehow [01:13] if openser cannot connect to it, maybe [01:13] bt it shoudnt [01:13] habakuk: how do you deal with proper accounting then ? [01:13] ok run openser debug 9 and redirect to a file and attemp to register [01:13] that shold help you [01:14] says invalid realm [01:14] but it should be ok its SAME config as my other that works [01:14] JerJer: sst [01:14] not every ua supports sst [01:14] so? [01:14] nor do i want to run development code [01:14] my gw supports sst [01:14] thats all I need [01:15] any real voip provider cannot require a specific UA [01:15] AH ok im tired of this bull shit [01:15] timeto get a real hardware box [01:15] or get scammed out of 40k by agprojects [01:15] JerJer: reread the docs. only one side needs to support it [01:15] its still unproven code [01:15] i.e. if gw supports and ua doesn't you are golden [01:16] JerJer: then don't use it [01:16] i'm gonna hold off - let someone else deal with the headaches [01:16] it works for me [01:16] i don't plan on using it [01:17] WTF [01:17] Authorization is OK [01:17] so my pap is ok but asterisk is not [01:17] i might attempt to extend rtpproxy to report back accounting details [01:18] JerJer: cool [01:18] if that doesn't work, I'll write a new mediaproxy app and properly document it [01:19] problem I have with that kind of accounting, is that you have to route the media [01:19] which I don't like [01:19] my situation requires dealing with the media [01:19] ok found it.. seems i need oreg to ip not domain.. that REALLY weird [01:20] openser isn't responding to your domain name then [01:20] i have all the alias [01:20] i had to add my domain name into /etc/hosts [01:20] you have to deal with media if you want accurate accounting [01:21] JT: agreed [01:21] however sst does seem interesting - if you have your own GW [01:22] well most gws support it [01:22] i doubt L3 does [01:22] the only problem I have with this is how the dialog module is implemented [01:22] since their shit doesn't even support DNS [01:23] of course l3 supports it [01:23] not according to the switch monkey i talked to [01:23] unless you are 50K / month they won't talk to you [01:23] Registration successful [01:23] we send traffic to l3 all day (and night) long [01:24] GX as well [01:24] then some to Qwerst [01:24] well take it from me that they support it. used to work there [01:24] you must have gotten in there before they changed their rules [01:24] we do well more than 50k a month [01:25] there you go, then its not an issue then [01:25] we've got private gig-e fiber transit to them [01:26] JerJer: so what do you use for a platform to handle that kind of volume? [01:26] mainly lame ass broadsoft shit [01:26] then a growing amount of SER+Asterisk and SER+my own b2bua [01:26] ah ok [01:26] broadsoft is $$ isn't it? [01:27] stupid amount of $$$ [01:28] I didn't buy it, so its not my liability [01:28] ok well If you want to sell it I have a buyer for you [01:29] one of the companies I am involved with is owns buildings in Detoilet, Chicago, Portland and LA [01:29] -is [01:29] we can't sell it until I can make something else work just as good (if not better) [01:30] SER+Asterisk is close but not quite [01:30] right [01:32] OpenSER+ProxyDispatcher+MediaProxy for simple call term/orig seems to be promising [01:33] any way the issue I have with the dialog module, is if the bye doesn't have the correct record-route header ( because of a broken ua) it messes up the dialog code. Still investigating how this affects the session timer [01:33] cool. haven't looked at the dispatcher a whole lot, but looks good [01:34] it is pretty trivial [01:34] wahts a proper config file for mediaproxy ?\ [01:34] it listens to the unix socket and repeats the info out to a UDP socket that mediaproxy's listen on [01:34] using SRV lookups [01:35] Bobthehunter: AG Projects wont' tell me, but I can show you mine [01:35] i see error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy [01:35] still [01:35] yrah [01:35] please do [01:36] probably the listen allow things i medssed [01:37] http://www.pastebin.ca/372377 [01:37] ip addy is external ip of that box ? [01:37] yes [01:38] error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy [01:38] still [01:38] weird [01:38] is your mediaproxy modparam to the unix socket getting set correctly ? [01:39] do you see the unix socket in that dir ? [01:39] 0 Feb 25 19:33 mediaproxy.sock [01:40] 6 Feb 25 19:33 mediaproxy.pid [01:41] is openser running as root / [01:41] or does openser have read/write perms to that socket ? [01:41] yep [01:42] root 12064 0.0 0.4 41504 4300 pts/3 S+ 19:33 0:00 openser -d -D -f ./generated.cfg [01:42] does syslog show any thing from mediaproxy ? [01:42] run mediaproxy with --no-fork [01:43] yeah errors [01:43] lol [01:43] pythn crap [01:43] there ya go [01:44] No module named pyrad.client [01:44] seems its picky lol [01:45] i think a C based 'mediaproxy' application would kick some serious ass [01:46] named MySQLdb [01:46] oh my god.. its gonna lpay me all night [01:46] then if we can keep the UDP in kernel space we could scale it even farther [01:49] i said acocunting=none and still wants it [01:49] lame program [01:49] what ver you runnin? i have to uncomment shit to get accounting to load [01:50] yum search python|grep -5 mysql [01:50] i run debian [01:50] MySQLdb [01:50] ;0 [01:51] MySQL-python [01:51] finaly [01:51] got it [01:51] ok enough for a sunday [01:52] just need invite auth to fix.. seems reg works [01:58] are you talking about asterisk invite issues? [01:58] if so you need to set fromuser and username [01:58] yes i know [01:59] i user user/username/fromuser [01:59] overited callerid but who care at this point [01:59] 503 "PSTN Termination Currently Unavailable" [01:59] ok [01:59] so now.. i need my own LCR strat.. since oser's sucks a bit [02:03] LCR strat? whats that? [02:03] strategy [02:10] Bobthehunter (n=Bobthehu@145-27.mc.cite.net) left irc: "Leaving" [02:15] JerJer still there? I was thinking of what you were saying with the rtpproxy enhancements [02:20] anyway my idea is, take the concept of mediaproxy, rtpproxy with natping, and simply extend that to something like sip-ping, if no response in time X then send bye to far side [02:53] Nick change: CrazyTux -> CrazyTuxer [03:11] CrazyTuxer (n=CrazyTux@64.95.219.140) left irc: Read error: 113 (No route to host) [03:29] habakuk (n=chatzill@c-24-6-224-243.hsd1.ca.comcast.net) left irc: Read error: 110 (Connection timed out) [04:25] Nick change: hohum -> compiletime [04:25] Nick change: compiletime -> hohum [05:03] variable_office (n=variable@cerberus.iswan.net) left irc: "Leaving" [05:22] Larchen (n=Larchen@195.49.168.10) joined #openser. [05:22] Hello [05:55] hey all, I think I found a bug. anyone home? [05:56] I have avpops and LCR with different db_urls and avp_db_query is failing [06:00] oh I get it.. avpops uses a avp_url instead of a db_url [06:01] those should be the same acording to the doc :-) [06:02] it needs both of them? [06:02] http://www.openser.org/docs/modules/1.2.x/avpops.html#AEN93 [06:03] hmm. .it didn't work with just a db_url param [06:08] hmm.. looks like the modules db_urls are overriding each other? [06:08] I have my avps and lcrs in different databses [06:08] on diff servers [06:08] and whichever one is listed last, will work.. [06:09] this is in 1.1 [06:10] osas: any idea there? ever heard of that problem? [06:12] no scratch that.. looks like LCR is taking the DB url of the last listed db_url in the file!.. weird.. [06:13] osas: did you write those bits? I know you worked on LCR [06:21] hello? [06:43] habakuk (n=chatzill@c-24-6-224-243.hsd1.ca.comcast.net) joined #openser. [07:18] CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) joined #openser. [08:35] is there a way to statefully send a cancel( or Bye) ? if ($var($DLG_STATUS < 4) sl_sendreply("CANCEL","cancel"); doesn't seem to work [08:40] CANCEL isn't a reply [08:40] the UAC module might be able to do what you want [08:41] hmm.. maybe not... [08:41] brettnem: hmm.. don't see any options [08:42] basically I want to cancel a call if the ack to the 200 ok isn't kosher [08:42] I have several broken UA's that I want to block [08:43] ah [08:43] problem is.. this is the earliest I can tell if they are broken or not [08:43] well there is this [08:43] http://www.iptel.org/faq/how_can_i_manually_send_a_bye_message_to_a_pstn_gateway_0 [08:43] but that's kinda hokey [08:44] I thought there was a way to manualy construct and send a message [08:44] well this technically should be a cancel [08:44] same idea [08:44] read the RFC to be sure you have it formatted right. or copy a real oine [08:45] hmm.. thats kind of lame though, I'd rather not have to do that way [08:46] yeah, it's lame [08:46] This might give you some leads.. [08:46] http://www.arcknowledge.com/gmane.comp.voip.openser.devel/2005-11/msg00314.html [08:46] have you looked at the dialog module? Basically its designed to keep the dialog in a RR cookie [08:47] so if a broken ua doesn't contain that cookie, then dialogue is never closed.. [08:47] ah, so you want to deny users who don't keep the cookie? [08:48] yes [08:48] hmm [08:49] hmm.. but it sounds like you are testing the cookie capability of your gateway, not your client [08:50] well it depends who originated the call [08:50] right [08:50] in this case I want to test the UA that initiated the call [08:50] CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) left irc: Remote closed the connection [08:50] well eventually both.. [08:50] just starting with this case anyway [08:51] ok so looks like this function needs to be implemented still [08:51] yeah, I don't really know enough about that module to be of help. seems like the cookie wouldn't show up until you had subsequent requests within the dialog [08:52] yes if UA's are working properly. Case in point SJPhone doesn't work correctly [08:53] raise some hell [08:54] heh right [09:01] ok well i have sort of a solution [09:04] I just drop the ACK, and eventually the UA will time out and send a bye [09:25] qdk (n=qdk@90.184.3.249) left irc: Read error: 110 (Connection timed out) [09:41] is it possible to check in LooseRoute if an INVITE allready has an Transaction (reINVITE) .. like is method==INVITE && t_new_trans ..? But what if there is no transaktion? [09:42] than the function will create a new Transaction and the Invite went through, right? [09:44] mmarkus: I think you can do something like that with the dialog module [09:44] you can then check $DLG_status in LOOSE_ROUTE() section [09:53] qdk (n=qdk@213.150.62.32) joined #openser. [10:01] has anyone ever seen this error? "comp_scriptvar: cannot get left var value " : if ( $DLG_status != 4 ) { #do something }; [10:01] shokoi (n=rmaglasa@58.71.34.138) joined #openser. [10:01] confusing because it looks right from what I've read of Pseudo Variables [10:02] hi, is there a way to cancel an INVITE request upon receipt of an OK response? [10:03] shokoi: I had that same question earlier [10:03] apparently not [10:03] my solution is to drop the ack to that 200 ok [10:03] habakuk: thanks, i just joined the irc [10:04] how to do that? [10:04] can i just call drop()? [10:04] <_Bilbo_> doesnt sl_send_reply work? [10:05] well test for when you receive the ACK and just exit(); [10:05] _Bilbo: this works only in the ROUTE block [10:05] I'm using the dialog module, so I'm testing if an ack is out of dialog, I just reject it [10:05] habakuk: i've tried that, exit() and drop() and still the OK reply is progpagated to the calling UA [10:06] looks interesting, which functions are use using? [10:06] did you write new ones? [10:07] right.. well my "solution" if you can call it that, basically never allows the call to be set up as the ack to the 200 is never received [10:07] no only existing ones [10:08] think my requirement is a bit different than yours [10:08] whats your requirement? [10:08] <_Bilbo_> shokoi: in what section do you need that? [10:08] _Bilbo: in ONREPLY [10:09] <_Bilbo_> ack [10:09] habakuk: i wrote a new module to do something, lets call it func(), this function returns success or failure [10:09] ok [10:09] i'l call this function everytime i receive an OK during INVITE [10:10] ok [10:10] so if the func() returns failure, i need to cancel or terminate the current session establishment [10:10] ah.. got you [10:10] if that is the correct term [10:11] thanks [10:11] habakuk: have you experience this ? [10:11] I don't think its possible [10:11] but.. [10:11] really? [10:11] have you seen this patch? [10:11] http://sourceforge.net/tracker/index.php?func=detail&aid=1642890&group_id=139143&atid=743022 [10:11] not yet [10:12] i'll try it later, thanks [10:12] looks like it might be kind of what you are looking for.. [10:12] but not exactly [10:13] thanks for the info [10:15] sure [10:32] shokoi (n=rmaglasa@58.71.34.138) left irc: "using sirc version 2.211+KSIRC/1.3.12" [10:33] DanB (n=danbogos@87.139.12.167) joined #openser. [10:33] Morning Guys! [10:41] habakuk: thanx i will have a look [10:44] morning [12:15] L|NUX (n=linux@unaffiliated/lnux/x-10290) joined #openser. [13:18] qdk (n=qdk@213.150.62.32) got netsplit. [13:18] Larchen (n=Larchen@195.49.168.10) got netsplit. [13:18] JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) got netsplit. [13:18] _Vile (n=vile@bc182112.bendcable.com) got netsplit. [13:18] mmarkus (n=markus@markus.netzquadrat.de) got netsplit. [13:18] Eimann (i=eimann@chaconne.etherkiller.de) got netsplit. [13:18] L-info (n=Adam@g0962184.demon.co.uk) got netsplit. 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[14:59] _Bilbo_ (n=bastian@mail.collax.com) returned to #openser. [14:59] mmarkus (n=markus@markus.netzquadrat.de) returned to #openser. [14:59] brettnem (n=brettnem@72.29.102.158) returned to #openser. [15:27] L|NUX (n=linux@unaffiliated/lnux/x-10290) joined #openser. [15:35] Curi (n=Curi@200.24.227.122) joined #openser. [15:37] anyone knows how can i rewrite the user information from the SIP packet? [15:38] <_Bilbo_> "user information"? [15:39] like the auth user and digest [15:40] for example: i want to receive an INVITE and replace the user who's sending the INVITE [15:40] <_Bilbo_> have a look at the "textops" module and it's functions [15:40] ok, thnx [15:40] <_Bilbo_> and/or you might want to have a look at the upcoming perl module :) [15:41] i was afraid i would need that [15:41] <_Bilbo_> depends on what you want to do [15:41] btw, there's no way to do this in asterisk ? [15:42] _Bilbo_: are you promoting your module? ;-) [15:43] Curi: you might wanna check this: http://openser.org/docs/modules/1.2.x/uac.html#AEN160 [15:44] <_Bilbo_> osas: sure... :) [15:44] <_Bilbo_> i just fixed a bug that /should/ have been detected _much_ earlier [15:44] osas: great [15:44] <_Bilbo_> any testing thus is greatly appreciated [15:44] I was playing with the svn this morning [15:44] the link on the website is wrong ... [15:44] Morning. [15:44] I ended up dl the whole repo [15:44] morning [15:46] <_Bilbo_> osas: well, that's the default sourceforge message... [15:46] <_Bilbo_> they do say you "probably need to append "/trunk" to the path" or something like that [15:46] I sent an e-mail to the list with the right command [15:47] <_Bilbo_> oh, that was you :) [15:47] yup [15:49] <_Bilbo_> downloading should be available through http, also - without SSL slowdown [16:24] is it safe to use EXEC module? [16:25] i mean in production [16:28] <_Bilbo_> imho it's regarded as "stable", yes [16:28] <_Bilbo_> but safety vastly depends on your environment [16:28] humm [16:39] tcseke (n=chatzill@22-36.adsl.etel.hu) joined #openser. [16:53] CrazyTux (n=CrazyTux@64.95.219.140) joined #openser. [17:16] Pazzo (n=ugelt@dialin-225136.rol.raiffeisen.net) left irc: "Verlassend" [18:02] apardo (n=apardo@87.217.144.244) joined #openser. [18:05] qdk (n=qdk@213.150.62.32) left irc: Read error: 113 (No route to host) [18:19] tcseke (n=chatzill@22-36.adsl.etel.hu) left irc: "Chatzilla 0.9.75 [Firefox 1.0.7/20050915]" [18:39] CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) joined #openser. [18:45] qdk (n=qdk@90.184.3.249) joined #openser. [18:46] Curi (n=Curi@200.24.227.122) left #openser. [18:46] qdk (n=qdk@90.184.3.249) left irc: Client Quit [18:46] qdk (n=qdk@90.184.3.249) joined #openser. [18:56] quetwo (n=quetwo@pplant-336.user.msu.edu) joined #openser. [18:56] hey everybody [18:57] hi [18:58] hows it hangin? [18:58] ok [19:01] slightly to the left [19:01] hah [19:01] hey, are there any configuration guides other than the one available from onsip? [19:01] the angle of the dangle is proprotional to the beat of the meat [19:01] http://www.sipwise.com/wizard [19:01] run thru that [19:02] i tried, but i'm at a loss of where to configure it from there [19:02] quetwo: start with the basic one provided with the distribution [19:02] play with it [19:02] i'm feeling that what i'm trying to acomplish is not very standard [19:03] enhance it with what you want [19:03] and then you will have a good handle on it [19:03] but of course ... who needs something standard ;-) [19:03] i have been, and i can get SIP clients to register to it, etc. but just need a push in the right direction to get it past that [19:04] serves me right for trying to use it with an Avaya PBX ;P [19:06] i pretty much just need it to proxy the requests from the public network to my private network, and replace the record-path on the outgoing path with my firewall's public address [19:06] i don't need it to keep track of registrations, etc. as the Avaya already does that [19:08] aha [19:08] you need an sbc [19:08] nod [19:08] session border controller [19:08] you want to proxy calls and media from two non routable networks [19:09] right? [19:10] close -- i don't need to proxy the voice streams, as my media-processor (the device that generates the G.711mu signaling) has both public and private addresses [19:10] I see [19:10] i just need something to modify the SIP data packets to point to the correct address [19:10] than you need only the signalling [19:10] nod [19:10] you ca do that by instructing openser to bind to both IPs [19:11] use the mhomed=1 option to route from the right interface [19:12] ok [19:14] then i would use the method=="INVITE" and client_nat_test("3") to change the record route path? [19:15] you can do that on the public interface ... [19:15] Action: osas goes to lunch [19:40] habakuk (n=chatzill@c-24-6-224-243.hsd1.ca.comcast.net) left irc: Read error: 110 (Connection timed out) [19:42] L|NUX (n=linux@unaffiliated/lnux/x-10290) left irc: "Leaving" [19:56] CrazyTux (n=CrazyTux@64.95.219.140) left irc: No route to host [20:13] Action: osas is back [20:13] fun [20:13] hehe [20:19] CrazyTux (n=CrazyTux@70.142.26.244) joined #openser. [20:20] osas: would i ever want to issue the record_route() function in my senerio? [20:44] yup [20:44] gotta go now ... [20:44] I will be back l8r [20:46] Defraz_ (n=t0tal@fw.centrisys.com) joined #openser. [20:59] sloth_ (n=chatzill@209.10.153.194) joined #openser. [21:25] DanB (n=danbogos@87.139.12.167) left irc: Read error: 110 (Connection timed out) [22:19] osas (n=osas@CABLE-72-53-75-252.cia.com) left irc: Read error: 110 (Connection timed out) [22:48] osas (n=nosas@CABLE-72-53-75-252.cia.com) joined #openser. [22:49] welcom back [23:12] quetwo (n=quetwo@pplant-336.user.msu.edu) left irc: [23:58] sipper (n=chatzill@65-113-98-146.dia.static.qwest.net) joined #openser. [00:00] --- Tue Feb 27 2007