[00:06] Nick change: DoberMann -> DoberMann[ZZZzzz [00:07] Nick change: DoberMann[ZZZzzz -> DoberMann[PullA] [00:07] Nick change: DoberMann[PullA] -> DoberMann[ZZZzzz [00:55] CunningPike (n=CunningP@204.239.10.234) left irc: Remote closed the connection [01:15] osas (n=nnnnnnno@CABLE-72-53-75-252.cia.com) left irc: Read error: 110 (Connection timed out) [01:55] NormB (n=NormB@smoothwall.goes.com) left irc: "Leaving" [03:34] DoberMann[PullA] (n=james@AToulouse-156-1-100-194.w90-30.abo.wanadoo.fr) joined #openser. [03:47] DoberMann[ZZZzzz (n=james@AToulouse-156-1-144-253.w90-30.abo.wanadoo.fr) left irc: Read error: 110 (Connection timed out) [04:03] brettnem_ (n=brettnem@72.29.102.158) joined #openser. [04:04] DoberMann[PullA] (n=james@AToulouse-156-1-100-194.w90-30.abo.wanadoo.fr) got netsplit. [04:04] techie (n=gus@voip.routedsystems.com) got netsplit. [04:04] brettnem (n=brettnem@72.29.102.158) got netsplit. [04:04] _VoiceMeUp_Com (n=_VoiceMe@145-27.mc.cite.net) got netsplit. [04:04] Nick change: brettnem_ -> brettnem [04:04] Possible future nick collision: brettnem [04:13] DoberMann[PullA] (n=james@AToulouse-156-1-100-194.w90-30.abo.wanadoo.fr) returned to #openser. [04:13] techie (n=gus@voip.routedsystems.com) returned to #openser. [04:13] _VoiceMeUp_Com (n=_VoiceMe@145-27.mc.cite.net) returned to #openser. [04:18] DoberMann_ (n=james@AToulouse-156-1-100-194.w90-30.abo.wanadoo.fr) joined #openser. [04:30] maxedout (n=maxstout@user-38lm3ok.cable.mindspring.com) joined #openser. [04:32] DoberMann[PullA] (n=james@AToulouse-156-1-100-194.w90-30.abo.wanadoo.fr) left irc: Read error: 113 (No route to host) [04:34] i'm experiencing the problem mentioned here http://mail.gnome.org/archives/ekiga-devel-list/2007-January/msg00063.html with nathelper.cfg, i'm trying force_rtp_proxy("c"), but it doesn't appear to be working, what's the correct syntax for this? [04:53] i'm only trying this for the 2 uses of force_rtp_proxy() in the nathelper.cfg, do i need to use it elsewhere? [04:55] Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) joined #openser. [05:18] CrazyTux (n=CrazyTux@64.95.219.140) left irc: Connection timed out [05:20] maxedout: what is a ekiga? [06:06] erider, its an open-source software audio/video voip client [06:06] erider, but according to the thread, i should be able to use force_rtp_proxy("c"), to compensate for the bug [06:07] erider, however... i can't get this to work... i'm thinking maybe i should try mediaproxy if i can't get rtpproxy to behave [06:08] CrazyTux (n=CrazyTux@216.142.88.85) joined #openser. [07:26] ber__ (i=brad@neu.cow.org) joined #openser. [07:27] ber_ (i=brad@neu.cow.org) got netsplit. [07:27] plr_ (i=juhali@shell.evtek.fi) got netsplit. [07:27] plr_ (i=juhali@shell.evtek.fi) returned to #openser. [07:38] ber_ (i=brad@neu.cow.org) got lost in the net-split. [08:06] Nick change: DoberMann_ -> DoberMann [08:28] CrazyTux (n=CrazyTux@216.142.88.85) left irc: Read error: 113 (No route to host) [08:28] CrazyTux (n=CrazyTux@64.95.219.140) joined #openser. [08:30] CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) joined #openser. [08:38] maxedout: is seems that the usage is "c=...." [08:46] can someone help with a suggestion for voip client to test openser on 64 bit linux [08:55] time for me to try to sleep again [08:57] CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) left irc: Connection timed out [09:01] Nick change: DoberMann -> DoberMann[PullA] [09:08] DanB (n=danbogos@87.139.12.167) joined #openser. [09:16] CrazyTux (n=CrazyTux@64.95.219.140) left irc: Read error: 110 (Connection timed out) [09:17] CrazyTux (n=CrazyTux@216.142.88.85) joined #openser. [10:06] apardo (n=apardo@87.217.145.63) joined #openser. [10:23] Chris-NB (n=chris@ng1.kurtkrenn.com) joined #openser. [10:23] hi [10:23] anyone using the presence module? [10:31] Hi Chris-NB: shoot [10:31] I've problems with these modules [10:32] I'm testing with Gaim 2.0.0beat6 and X-Lite [10:32] I have used them in test enviroment, and no problems [10:32] and a Grandestream BT100 [10:32] in Gaim I can see the other two phones's presence [10:32] but not in X-Lite [10:33] and I think it's related with the active_watchers table [10:33] when gaim registers, two entries are generated (database is pgsql) [10:33] do u see anything in the presence table? [10:34] these two entires (two buddies) remain there until gaim is closed and timeout is passed [10:34] jep. [10:34] as soon as u will have the info in the presence table, your active_watchers will be notified [10:34] did u set use of xcap to no? [10:34] codestr0m (n=asura@88.232.131.179) joined #openser. [10:35] but when I register X-Lite I can see in debug that an entry in active_watchers is generated and almost simuntanously deleted! ??? have no clue why [10:35] nop. do I have to? or do you mean force_active? [10:35] modparam("presence", "force_active", 1) [10:35] have done that [10:36] yeah, I call them in my own language ;-) [10:36] : ) np. as long as I know what you mean ; ) [10:37] so, how does it work? a client registeres at the server and his status is entered into presentity [10:37] it should be something else in your params [10:38] if he want's to get the status of others it is entered into active_watchers? [10:38] yup, this should be the behaviour [10:38] ok [10:38] but I can see this in the logs: [10:38] did u set xlite as presence agent? [10:39] and add your buddies in your contact list? [10:40] I told X-Lite it should use presence agent and entered buddies into contact list and checked 'show status' [10:40] here are the logs: http://rafb.net/p/BN0KWF23.html [10:41] as you can see, the insertation and deletion is at the same time [10:41] I've no clue why? [10:42] there shold be something in your settings then [10:43] can u post your settings for presence module? [10:43] and what version of openser are you using? [10:44] I am using the latest SVN one, I remember there were some fixes after the release for presence module [10:45] http://rafb.net/p/o1dN0x92.txt [10:46] OpenSER is from SVN built 4 days ago [10:46] apardo (n=apardo@87.217.145.63) left irc: Read error: 104 (Connection reset by peer) [10:47] qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) left irc: Read error: 113 (No route to host) [10:47] qdk (n=qdk@213.150.62.32) joined #openser. [10:50] *hmmm I've just updated the src. there are new versions of presence. probably I should recompile it [10:50] what about the config? does it look ok? [10:51] patrickv0x (n=patrick@64.235.249.36) left irc: Read error: 110 (Connection timed out) [10:53] ok, I see no issue from the convig, it is only that I am not enforcing new transaction, if you want to have some work arround I can give u a simple config which I am using... [10:53] it could be better than no ideea... [10:55] would be great [10:56] http://pastebin.ca/446087 [10:56] I did not really test calls through this config, only presence [10:57] and messages [10:58] thanks [10:58] messages work fine with my config [11:15] *hmmmm [11:15] I've upgraded to the new version, but without any changes : / [11:16] did u try my config? [11:16] in the active_watchers table there are only the entries from gaim [11:16] I've commented out the t_release(); [11:16] but I'll replace the config [11:16] if you have nothing to loose, try my config for tests [11:17] nothing to loose, it's only testing [11:18] I have bigger issue with the new version [11:19] openser behaves strange on TCP connections [11:19] anybody here with enough experience in TCP? [11:20] is this 'OK' ? [11:20] 2(24971) PRESENCE:get_xcap_tree:The query in table xcap for [username]=133 , domain=192.168.68.130 returned no result [11:21] Last message repeated 2 time(s). [11:33] DanB, with your config it works for X-Lite [11:33] lasonic (n=lasonic@watto.labri.fr) joined #openser. [11:33] Gaim hangs. but don't know yet if this is a problem of my pc [11:33] ok, then u have something to start with... [11:34] lasonic (n=lasonic@watto.labri.fr) left #openser ("Kopete 0.12.3 : http://kopete.kde.org"). [11:34] jep. thanks for your help! [11:35] danke : D [11:37] Chris-NB: nichts zu danken ;-) [11:37] : ) [12:24] it is fact now, after upgrading from 1.1 to 1.2 openser doesn't listen anymore on TCP socket, any hint? [12:41] SoftIce (n=phil@vc-196-207-45-253.3g.vodacom.co.za) joined #openser. [12:42] hello, please can somebody advise me here, say for instance im routing traffic from south africa to the us [12:42] the latency to the us is around 250ms at best cases 240ms [12:42] and from south africa to the uk its around 140 [12:42] so would it be advised to route via uk then to the us? [12:43] also would i be able to forward all my sip traffic to an ser in the UK and then divert it to the US ? [12:44] SoftIce: jitter + packet loss will be more of a concern than latency in a lot of cases [12:45] I see [12:46] codestr0m: so you saying rather setup rsvp + other qos stradegies? [12:46] SoftIce: I would consider one way delay, not ping latency... [12:46] SoftIce:.. well believe it or not, but depending on how you've got your setup. I can make a call from eastern europe sound more crystal clear than sometimes in the US [12:46] if its all SIP traffic, a simple method would be to put your proxy in the middle between the two endpoints [12:47] the RTP should go direct anyway, no? [12:47] L-info: so rather have a server say in the UK [12:47] thats about mid way to us and SA [12:48] and then have all my stuff registering there? [12:48] sure.. 200ms or less is fine for SIP.. its the RTP that you need to worry about [12:48] ye, thing is its clear [12:48] but im getting issues with my carrier [12:48] but they supposed to be massive [12:48] define 'issues'? [12:48] SoftIce: run mtr against the different pops.. [12:48] now im wondering if im getting congested, etc due to the latency on my end? [12:48] press J to see the jitter and that will give you an idea. pick the clearest route [12:49] granted ping isn't sip traffic, but it can give you some ideas [12:49] (or rtp) [12:49] I see [12:49] and out of intrest what is your guys views on asterisk? [12:50] depends on what you want yoo use it for [12:50] I don't hang out in this channel, but you may try to ask that in #asterisk [12:50] well just pretty much for routing calls, billing, etc [12:51] codestr0m: naa, I wanted some outside opinion [12:51] :) [12:51] codestr0m: what could be the reason for this [12:51] getting congested on my carrier [12:51] SoftIce: well. I wouldn't have suggested it if I thought you wouldn't get some.. ;) [12:52] yet nobody else is complaining, is it that they could have jitterbuffer set to X [12:52] to put it politely.. a lot of carriers outright suck. and sometimes it's the peering that sucks.. there could be tons of possible reasons [12:52] not jitterbuffer but jitter, and due to a latency over 250 they start to drop calls? [12:52] codestr0m: do you know ipsmarx.com by any chance? [12:52] do you have latency of over 250ms on rtp? [12:52] SoftIce: when I holiday. my latency is over 250ms all the time with little problem connecting to a proxy in the west coast of US [12:53] codestr0m: an rtp proxy? [12:53] or sip proxy? [12:53] L-info: yes I would have to, latency to the US from south africa is at minimum 240 ms [12:53] L-info: yeah. I force the rtp through my pop because I know my internation route is cleaner than the carrier [12:54] i have fibre links but light can only travel so fast :) [12:54] SoftIce: what's an IP [12:54] I'll mtr it from a few places in the world to give you an idea [12:54] sip.voipstream.co.za [12:54] its an atm link on my side [12:55] 27 hopes.. 544ms latency on my provider 1.. which is totally incorrect, but i don't route much to south africa.. [12:56] from eastern europe. same thing. about 500ms... [12:57] par6.alter.net 0.0% 49 182.4 174.6 161.1 207.3 10.3 [12:57] 17. s6-1-0.cr2.cpt1.alter.net 0.0% 49 352.5 347.9 321.5 541.3 36.9 [12:57] 18. srp5-0-0.cr1.cpt1.al [12:58] 500 [12:58] the one hop on alter.net is adding nearly 200ms alone [12:58] what is the difference between mtr and just a regual ping reply from a ip? [12:58] ok do it to another box of mine [12:58] 196.15.190.74 [13:00] 440 ish.. and lots and lots of tunnels. 9 hops jitter isn't great, but not bad considering it's so far [13:01] 350ish from 2nd pop.. [13:01] so that is a better route than the first ip I gave you sip.voipstream? [13:02] with a server in ams-ix is probably going to be your best choice from the 30 second view I can see [13:02] strange thing is [13:02] that first ip I gave you is UUNet [13:02] ad the second 1 I gave you is a link to my office [13:02] yeah. 2nd router is waaay cleaner, but still this isn't real traffic. this is ping. which can be qos down by any carrier in the middle [13:02] i cannot understand why uunet routing is so shocking [13:02] I see [13:03] hey DanB, I've a working config now. that has everything I had in my old and working presence : D [13:03] http://www.pastebin.ca/446198 [13:04] codestr0m: that is my mtr to my carrier [13:04] http://p.caboo.se/54617 [13:05] can you mtr to this ip [13:05] 216.89.79.2 [13:06] I've worked closely with that company.. and considered putting a pop there [13:06] http://p.caboo.se/54618 [13:07] so you're egress is cleaner on the one ip and the ingress is cleaner on the other, but that's only a couple IPs.. and I probably won't be sending you traffic [13:08] codestr0m: do you deny people if their latency is to high? [13:08] like if X then dont accept a call? [13:08] SoftIce: nope, but I run pretty much a private network [13:10] JGJones (n=jgjones@host217-41-30-182.in-addr.btopenworld.com) joined #openser. [13:35] NormB (n=NormB@smoothwall.goes.com) joined #openser. [13:35] Chris-NB: that's a good news, I found also my problem with not listening on TCP [13:37] whar was the problem? [13:38] I was running the process for tests in foreground [13:39] but what is strange it is that openser was showing as bound to TCP socket, netstat not... [13:39] so, false info given by openser [13:40] hmmm [13:40] interesting [13:41] it took me half day to figer this one out, but now I know at least the bits which needs to be set in a TCP handshake ;-) [13:42] three-way-handshake : D [13:43] yeah, again my own language ... [13:43] I will redefine once full IT :-) [13:45] FuL|OUT (n=fn@a83-132-175-96.cpe.netcabo.pt) left irc: "changing servers" [13:46] nice : D [13:47] send me a copy of it : D [14:39] JGJones (n=jgjones@host217-41-30-182.in-addr.btopenworld.com) left irc: Read error: 110 (Connection timed out) [14:41] JGJones (n=jgjones@host213-123-201-218.in-addr.btopenworld.com) joined #openser. [14:59] tcseke (n=chatzill@22-36.adsl.etel.hu) joined #openser. [15:00] osas (n=nnnnnnnn@CABLE-72-53-75-252.cia.com) joined #openser. [15:03] SoftIce (n=phil@vc-196-207-45-253.3g.vodacom.co.za) left irc: Read error: 104 (Connection reset by peer) [15:19] miconda (n=daniel@81.180.83.75) joined #openser. [15:48] is there any way to force the IP address of the Via field inserted by openser ? [15:58] QbY (n=Kelvin@66.236.241.67.ptr.us.xo.net) joined #openser. [15:59] Anyone have a suggestion for this error? WARNING: database engine not found - tried 'MYSQL' [16:01] JGJones (n=jgjones@host213-123-201-218.in-addr.btopenworld.com) left irc: "Ex-Chat" [16:01] QbY, do you have the mysql module loaded ? [16:01] very first one. [16:02] i'm having to move to another proxy fast.. my hd is failing.. [16:02] so i've taken the config from my old machine and dumped it onto the new one [16:02] hrmm [16:02] appears mysql.so is not in /usr/local/lib/openser/modules [16:03] you found the issue [16:03] that i did.. just not thinking this morning.. [16:03] easy , just go to openser_src_dir/modules/mysql/ ; make and cp mysql.so to there [16:04] anyone knows where can i find a example of lcr for openser 1.2 ? [16:14] sack_: what kind of example do u need? [16:16] well nothing particular , just to figure out how is working ... i need to provide a lcr solution , but nothing specific yet [16:20] umm i found one from Ser ... hope it could help me [16:24] lcr is the same across openser releases [16:24] Chris-NB (n=chris@ng1.kurtkrenn.com) left irc: Read error: 113 (No route to host) [16:24] there was a recent fix for lcr in 1.2 and head [16:25] osas, i saw i'm using svn . Thanks [16:25] for testing should be ok [16:25] but not for production [16:25] well i'm want to test it for now [16:26] but i guessed i could find any simple example in twiki of lcr [16:27] s/twiki/wiki/ [16:32] good morning all. [16:38] bkw_ (i=brian@ppp-70-128-123-137.dsl.tulsok.swbell.net) joined #openser. [16:44] Morning CrazyTux! [17:01] CrazyTux (n=CrazyTux@216.142.88.85) left irc: No route to host [17:01] CrazyTux (n=CrazyTux@216.142.88.85) joined #openser. [17:26] NormB (n=NormB@smoothwall.goes.com) left irc: "Leaving" [17:30] is there a way to extend the timeout of rtpproxy? [17:40] which scales better mediaproxy or rtpproxy? [17:45] which works better? [17:45] i can't even find a good mediaproxy.cfg [17:46] its in the media proxy distro [17:46] I just edited the file [17:46] and changed two lines [17:47] which were to uncoment them [17:48] CrazyTux2 (n=CrazyTux@64.95.219.140) joined #openser. [17:49] CrazyTux (n=CrazyTux@216.142.88.85) left irc: Nick collision from services. [17:49] CrazyTux2 (n=CrazyTux@64.95.219.140) left irc: Client Quit [17:49] CrazyTux (n=CrazyTux@64.95.219.140) joined #openser. [17:49] qdk (n=qdk@213.150.62.32) left irc: Read error: 113 (No route to host) [17:50] qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) joined #openser. [17:53] bkw_, i just downloaded media proxy, and can't find an openser config for it [17:54] maxdoubt, thats because you're suppose to create one ;o) [17:54] maxdoubt, google, there are various examples. [17:54] maxdoubt, also, the 'ser getting started' guide goes over mediaproxy. [17:55] CrazyTux, i barely understand openser's syntax still... i was very close to having a working setup with openser nathelper and rtpproxy, but the media stream wouldn't connect.... this is all a lot to take in a short period of time [17:56] quite frustrating [17:57] maxdoubt, are you at an autozone right now? :P [17:57] CrazyTux, yessir [17:57] maxdoubt, haha, cars --- voip hows that one work out [17:57] CrazyTux, they wanna do video conferencing w/ stores in Mexico [17:58] CrazyTux, not that our network could handle it... but its been a fun learning curve for me [17:58] CrazyTux, an SIP crash course lets say [17:58] Yes. [17:58] stimpie (n=michiel@ip565faf27.direct-adsl.nl) joined #openser. [17:58] maxdoubt, well, what exactly is the problem you are having currently, perhaps, someone can help get your config working. [17:59] CrazyTux, i just don't understand why the media stream wouldn't connect through rtpproxy... the udp looks right [17:59] maxdoubt, behind nat? [17:59] CrazyTux, one client is behind nat [17:59] CrazyTux, both clients can call each other [18:00] maxdoubt, one way audio, or both? [18:00] CrazyTux, both [18:00] so both no audio [18:00] maxdoubt, this is with RTPproxy? [18:00] CrazyTux, oh, the call gets disconnected before media streams get setup [18:01] CrazyTux, yes, rtpproxy [18:01] CrazyTux, sometimes, one client will stay connected... but no media connection ever occurs [18:01] CrazyTux, as far as i can decipher in the maze of debugging dump [18:03] CrazyTux, any ideas? can i send you dump for possible interpretation? [18:03] maxdoubt, pastebin [18:03] CrazyTux, ok... what dump would be most useful? openser, rtpproxy, ekiga? [18:03] tcseke (n=chatzill@22-36.adsl.etel.hu) left irc: "ChatZilla 0.9.78.1 [Firefox 2.0.0.3/2007030919]" [18:03] maxdoubt, ngrep [18:03] maxdoubt, sip signaling (openser) [18:08] DanB (n=danbogos@87.139.12.167) left #openser. [18:08] Nix (n=Nix@81.213.125.220) joined #openser. [18:13] CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) joined #openser. [18:18] CrazyTux, http://pastebin.ca/446590 [18:19] CrazyTux, sorry... did you want me to grep something? [18:23] CrazyTux, does it give you a sense of what's going wrong? [18:35] maxdoubt, back. [18:35] maxdoubt, can you give me like a tetheral or ngrep sipcap dump [18:36] CrazyTux, cool [18:37] CrazyTux, off which box? [18:37] maxdoubt, openser [18:38] CrazyTux, gimme 10 mins or so... [18:38] maxdoubt, alright I'll be around [19:02] CrazyTux, heh... our proxy blocks my post for some reason, but here's my sip cap http://pastebin.ca/446693 [19:06] DanB (n=danbogos@87.139.12.167) joined #openser. [19:07] NormB (n=NormB@smoothwall.goes.com) joined #openser. [19:15] maxdoubt (n=mackstou@169.198.254.6) left irc: Remote closed the connection [19:16] maxdoubt (n=mackstou@169.198.254.6) joined #openser. [19:27] CrazyTux, ping [19:27] Defraz_ (n=t0tal@fw.fuzecore.com) joined #openser. [19:31] Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) left irc: "Trillian (http://www.ceruleanstudios.com" [19:34] CrazyTux (n=CrazyTux@64.95.219.140) left irc: Read error: 110 (Connection timed out) [19:49] Defraz_ (n=t0tal@fw.fuzecore.com) left irc: Read error: 54 (Connection reset by peer) [19:52] Defraz (n=t0tal@fw.fuzecore.com) joined #openser. [20:14] could i get some help interpreting the dump i'm getting from openser? [20:28] miconda (n=daniel@81.180.83.75) left #openser. [20:29] supjigatr (n=syslod@152.53.16.10) joined #openser. [20:29] Hello. [20:31] CrazyTux (n=CrazyTux@64.95.219.140) joined #openser. [20:31] How goes it. [20:31] I'm having a problem with forwarded calls. I have a sip trace that indicates the UA sending a 400 bad request. [20:32] Eh. I've been better. This forwarding problem has been a bear to figure out. [20:32] http://pastebin.ca/446840 [20:32] A post of the sip trace. [20:45] CrazyTux, did you ever look at that dump I sent? [20:47] if you want to debug SIP, you need to check the payload of a packet [20:47] ngrep is a better tool [20:48] SIP is a complex protocol ... just looking at a method name won't reveal anything ... [20:48] osas, it sure is a complex protocol [20:48] jebus! [20:51] fulgas (n=fn@85.138.14.173) joined #openser. [20:51] osas, i'm trying to figure out how to use ngrep [20:51] ngrep -W byline -qt -d any port 5060 [20:55] osas. I also have the payload. [20:55] I have a full wireshark capture of 5060. [20:57] take a look at the headers and see what's wrong ... [20:57] I'm trying and I dont' get it . [20:57] see if you have any hints inside the 400 Bad Request [20:58] No hints. [20:58] post your trace [20:58] CrazyTux (n=CrazyTux@64.95.219.140) left irc: Connection timed out [20:59] CrazyTux (n=CrazyTux@64.95.219.140) joined #openser. [21:26] http://pastebin.ca/446955 [21:26] I don't understand the second invite and the ACK from openser which precedes the bad request from the linksys. [21:30] Nix (n=Nix@81.213.125.220) left irc: Remote closed the connection [21:54] DanB (n=danbogos@87.139.12.167) left irc: "Gotta Go!" [22:06] supjigatr (n=syslod@152.53.16.10) left irc: [22:58] hohum (n=dcorbe@mercury.sunrocket.com) left irc: Read error: 110 (Connection timed out) [23:00] osas, L-info happen to be around/ [23:08] here's my ngrep dump of the payload... trying to figure out what's wrong http://pastebin.ca/447116 [23:13] maxdoubt: what client are you using to test your setup? [23:13] erider, ekiga [23:14] erider, weren't you looking for a linux client last night? [23:14] do you know if it works on 64 bit linux system [23:14] yes [23:14] erider, i don't know, but you could try [23:14] erider, its a pretty good little tool, if only i could get it to work with openser [23:15] darn [23:15] I have just setup openser and I need to test it [23:18] maxedout: I guess I need to stop openser [23:19] maxdoubt: do you know the stop command [23:20] erider, killall openser [23:20] erider, there's probably a more appropriate method [23:21] maxdoubt: I get a error with I launch ekiga while its runnign [23:21] maxdoubt: I get a error when I launch ekiga while its runnign [23:21] erider, yeah... figuring out this openser stuff is a pain [23:22] erider, are you using the nathelper/rtpproxy configuration? [23:22] no [23:22] erider, which configuration are you using? [23:23] basic setup I don't know if it has a name [23:23] erider, ah... well then both you and i are on the very hard path of figuring out how the hell openser works [23:24] erider, best advice i can give.. is to start looking at the example config files, try to figure out some of the syntax, ngrep your sip packets... and hope somehow it makes sense [23:24] maxdoubt: I have not got that far [23:24] :/ [23:25] maxdoubt: have you uses the ekiga.net service? [23:25] maybe some friendly more experienced users will help us out ;-) osas CrazyTux ? [23:26] erider, yep... it works fine [23:26] maxdoubt, if you stick around tell later tonight I'll take a look [23:26] CrazyTux, alright, i'll be on... thanks for helping out us noobs [23:27] maxdoubt: have you used skpye [23:27] erider, yep [23:28] and where does ekiga allow you to call [23:28] erider, ekiga is an h323/sip client... if you buy an account through a gateway that can connect to the pstn, you should be able to call anywhere... ekiga recommends diamondcard as a gateway [23:29] erider, visit the ekiga website, and checkout diamondcard's service... it should give you a sense of what you can do [23:30] ok thanks max [23:30] erider, i believe most of the rates are comparable if not cheaper than skype [23:30] cool! [23:30] but what about quality [23:30] diamondcard: no way.. they are just a reseller for companies like teliax [23:31] ah... well see... there's more available [23:31] and the reason it's recommened is because of the devs helping on the project [23:31] i figured this much [23:31] ah I see [23:31] codestr0m, do you have any experience getting ekiga to work with openser + nathelper? [23:31] yeah. I know the owner to some degree and have worked with a number of the companies [23:32] I would like to make free calls [23:32] erider, the only way to do that is call pc <-> pc [23:32] ah [23:32] thats not fair :( [23:32] erider, or setup your own network [23:32] maxdoubt: not with ekiga no sorry.. I may have time in the near future to test that client, but it had some recent security issues and went straight to the bottom of my list [23:33] codestr0m, can you recommend any other linux A/V sip clients? [23:33] give twinkle a try [23:33] I knew that question was coming ;) [23:33] I have twinkle [23:33] what do we need openser to do for us? [23:34] ? [23:34] well a better question what does it allow us to do [23:34] erider, openser can do what you program it to do [23:34] if you have the client can you just use a gateway [23:35] erider, openser is for people trying to setup their own network [23:35] erider, or who need proxies, and gateways and such [23:35] thats what I figure and that is what I'm trying to do [23:36] but will I be about the call to pstn from my network [23:36] s/about/able [23:36] s/about/able to* [23:37] erider, if you have a phone line, you should be able to connect that to hardware that can talk to the network, and then you should be able to call out through the pstn [23:38] erider, i'm not sure how larger networks do it, but i'm guessing they have multiple-line connection to the pstn, either through a telco.. and then they can connect their voip network to it [23:39] I'm a long way from knowing how to do that [23:39] erider, sigh... yeah, me too [23:39] erider, twinkle doesn't do video does it? [23:39] I can even dial into my silly sip server [23:40] can't* [23:40] I don't think so [23:40] but I don't know I have not got it working yet [23:40] erider, what do you expect to happen when you dial in? [23:40] just to see that its connected [23:41] erider, oh, you're just trying to register? [23:41] its local and its not connected to the outside world yet [23:41] yup [23:41] erider, you need to learn how to set up openser as a registrar [23:41] erider, look at openser config files [23:42] I have some step by step directions for setup but I stop using them because they wanted to use x-lite which doesn't work on my system [23:43] erider, unfortunately, i don't think its just something you can follow instructions and have it work.. you need to sincerely understand it [23:43] I'm missing some [23:43] yeah [23:43] erider, how sip works, how openser redirects routes and handles sip traffic [23:43] Action: erider needs to find some info on line to get up to speed [23:44] erider, you and me both [23:45] where is a good starting point [23:45] erider, i can get ekiga to register with the openser configuration i'm using... but none of the media streams can connect [23:45] maxdoubt: where can I start from the beginning [23:46] erider, probably sip rfc [23:46] wow I hate reading those rfc pages [23:46] they read like a very bad book [23:47] erider, yeah... but to understand openser, you have to understand the protocol you're routing [23:47] erider, openser's syntax doesn't look that bad... there's docs for all the functions and such [23:48] maxdoubt: ok [23:50] maxdoubt: I wonder if I should start with sms first instead of voice and video [23:50] erider, couldn't hurt to get the very basics down [23:51] can you point me in the right direction [23:53] erider, i'm as clueless as you are :-) [23:54] maxdoubt: well can some here help [23:54] someone* [23:55] stimpie (n=michiel@ip565faf27.direct-adsl.nl) left irc: "Leaving" [00:00] --- Thu Apr 19 2007