| 00:00.10 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 00:00.10 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:00.20 | ManxPower | [TK]D-Fender: I'm at DruidCON in Atlanta, GA. voiceroute.net For a GUI it is clever. Very clever at points |
| 00:01.12 | jaytee | ZOMG, ManxPower actually finds a GUI he doesn't hate? Alert the media! |
| 00:01.31 | ManxPower | These guys sure are linux and voip fanboys. |
| 00:02.07 | angryuser | druid had problems when went out, not sure about now |
| 00:02.10 | jaytee | ManxPower, I gotta know what it is you find clever about it |
| 00:03.24 | ManxPower | jaytee: they store the configs in the database like many of these GUIs, but it also lets you edit the config settings. It converts the current config in the DB into a text file, lets you edit it, then shove the new settings into the database. |
| 00:03.52 | ManxPower | First time I've ever said "cool" in reference to a GUI product. |
| 00:05.18 | jaytee | the biggest problem I have with most * gui addons is the way they limit your control over the dialplan |
| 00:06.39 | Carlos_PHX | We've tried repeatedly to go to a GUI, always find a serious limitation. Switchvox hosted is closest, but $$$$ |
| 00:06.50 | ManxPower | jaytee: I didn't mean to imply I'd use it for MYSELF. |
| 00:07.02 | jaytee | I understand |
| 00:07.07 | ManxPower | Carlos_PHX: take a look at voiceroute.net. |
| 00:08.01 | jaytee | I'm still looking at customizing the asteriskgui for just adding user accounts and sip phones in a simple restricted template manner. |
| 00:08.34 | drmessano | ohhh ohhh |
| 00:08.42 | jaytee | trouble? |
| 00:08.45 | drmessano | I want to transcode all my calls in MP3 |
| 00:08.53 | drmessano | allow=mp3 FTW |
| 00:09.19 | Carlos_PHX | The RIAA would sue you for violating copyright on calls. |
| 00:09.44 | *** join/#asterisk seaq (n=seaq@190.25.65.76) |
| 00:10.02 | drmessano | Im gonna bet you $10000000 that Skype uses mp3 for it's proprietary p2p VoIP |
| 00:10.06 | drmessano | All too fitting |
| 00:10.15 | bkw_ | no they use GIPS |
| 00:10.21 | bkw_ | and MP3 for voip codec is FTL |
| 00:10.27 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 00:10.39 | drmessano | Apparently making a joke is too |
| 00:10.55 | bkw_ | asterisk did have a codec_mp3 at one point |
| 00:11.26 | ManxPower | still does in asterisk-addons last I hard. |
| 00:11.35 | bkw_ | no thats format_mp3 |
| 00:12.51 | ManxPower | The more I use Asterisk the less useful mp3 seems. |
| 00:13.12 | seaq | hi all!, i'm analizing CDR reports from asterisk, but i've got this situation. A call comes in to a QUEUE, the call is answered by an agent and then is transfered to an internal extension. I cannot find the CDR report from the transfer. the CDR registers this? maybe i'm missing an option? thanks in advance. |
| 00:13.32 | ManxPower | It uses quite a bit of CPU, isn't easy to transcode to/from, is optimized for music, not spoken word. |
| 00:14.43 | ManxPower | seaq: Wait until the system is otherwise idle, try to reproduce the problem. All CDRs will be close to gather in the logs. also, if you have batching enabled it can take a few mins for the CDRs to post. |
| 00:16.17 | seaq | hmm ok. i'll try to check that way... |
| 00:18.08 | bkw_ | ManxPower: then for voip if you loose a packet |
| 00:18.12 | bkw_ | all hell breaks loose on mp3 |
| 00:18.37 | hardwire | ogg saves lives |
| 00:18.49 | angryuser | going to sleep, bye @all |
| 00:18.51 | ManxPower | bkw_: I'm sure we could come up with a dozen reasons if we tried. |
| 00:19.05 | Carlos_PHX | Anybody heard of a limit of how many phones can work behind a WRT54G? 10, 20, 50...? |
| 00:19.34 | ManxPower | Carlos_PHX: I doubt it would be easy to know how many. |
| 00:19.51 | ManxPower | If it works with 3 phones it should work with 20 phones. |
| 00:20.11 | Carlos_PHX | At some point it will run out of memory for NAT. |
| 00:20.24 | ManxPower | If it doesn't work with three phones, then you can try several things, mostly change source port settings on the phones |
| 00:20.34 | Carlos_PHX | Seeing an odd behavior with one and 15-ish phones, never put that many on one of those. |
| 00:20.38 | Qwell | if it doesn't work with three phones, throw it in the garbage. |
| 00:21.45 | ManxPower | Carlos_PHX: you would have to ask the maker of the router how many UDP NAT streams they can handle, what their UDP NAT timeout can be changed to. assume each call will have one port for SIP signaling, two ports for RTP |
| 00:22.21 | drmessano | Im gonna have to say 2173 |
| 00:22.22 | Carlos_PHX | Heh, well yeah, but have you tried to talk to Linksys consumer division? Just figured I'd toss it out here in case someone knows "hey, it borks at X phones." |
| 00:22.26 | ManxPower | Carlos_PHX: I'll bet nobody has tried to make that many phones work behind a WRT54G. People with that many phones usually seem to use a real router. |
| 00:22.47 | Carlos_PHX | 2173, or 15... |
| 00:22.59 | ManxPower | Carlos_PHX: go to a linksys forum, I'll bet they would know |
| 00:23.04 | Carlos_PHX | True |
| 00:23.24 | ManxPower | It's just UDP, the higher level protocol has nothing to do with NAT |
| 00:23.31 | bkw_ | ManxPower: yah i'm sure we could come up with more than that :P |
| 00:24.11 | ManxPower | bkw_: the codec has a dozen or so patents attached to it, all expiring at different times. |
| 00:24.24 | bkw_ | well it doesn't matter really they aren't enforcing it |
| 00:24.30 | bkw_ | they are all fucked anyway |
| 00:24.38 | bkw_ | 2014 G723.1 is open |
| 00:24.49 | bkw_ | and I think 2016 G729A is open |
| 00:25.22 | Carlos_PHX | Speex is open... :-p |
| 00:26.11 | hardwire | Speex Freely. |
| 00:26.15 | hardwire | GET IT? |
| 00:26.19 | hardwire | It's not a code. |
| 00:26.25 | hardwire | btw I love speex |
| 00:26.54 | Carlos_PHX | <crickets> |
| 00:27.12 | hardwire | that would be the CNG activating. |
| 00:27.18 | *** join/#asterisk dmz (n=dmz@12.25.86.34) |
| 00:27.25 | drmessano | heh |
| 00:27.32 | drmessano | Asterisk doesn't like DMZ's |
| 00:27.57 | hardwire | does it like FTZs? |
| 00:27.58 | Carlos_PHX | Why not? |
| 00:28.22 | Carlos_PHX | Well, if properly configured, why not. |
| 00:28.44 | hardwire | dmz don't like drmessano |
| 00:28.53 | hardwire | or JerJer |
| 00:30.13 | ManxPower | DMZs can sense hostility. |
| 00:30.44 | Carlos_PHX | And fear. |
| 00:31.20 | jaytee | and somewhere around the DMZ is "Charlie" and he don't *^#& surf! |
| 00:31.25 | ManxPower | If you do not stay calm and confident they can attack with no notice. |
| 00:31.32 | drmessano | Someone has asterisk working thru a DMZ? |
| 00:31.58 | Carlos_PHX | Er, yeah, me, I'm sure others. |
| 00:32.06 | drmessano | Hmm |
| 00:32.09 | ManxPower | drmessano: A DMZ is just a default internal address to port forward by default. |
| 00:32.11 | Carlos_PHX | Define DMZ |
| 00:32.12 | hardwire | drmessano: kinda depends on your setup |
| 00:32.40 | Carlos_PHX | Traditional definition is a third physical port and subnet off a router. |
| 00:32.46 | ManxPower | no reason to put Asterisk in the DMZ, just put it behind standard NAT |
| 00:32.49 | Carlos_PHX | With either NAT or just different IPs |
| 00:32.51 | drmessano | I find that the "DMZ" setting on most routers mangles packets way too much to be useful |
| 00:33.03 | Carlos_PHX | Ah, you're using a dumb router. |
| 00:33.07 | drmessano | No |
| 00:33.12 | ManxPower | drmessano: Stop using cheap ass routers. |
| 00:33.18 | Carlos_PHX | Indeed |
| 00:33.20 | drmessano | Who the fuck said I was? |
| 00:33.24 | Carlos_PHX | At least on the server side. |
| 00:33.33 | ManxPower | drmessano: now who is not getting the joke? |
| 00:33.42 | ManxPower | BTW, what routers do you use? |
| 00:33.52 | drmessano | This week? |
| 00:33.57 | jaytee | I'm just waiting for the guy who tries to hook up 8 Sony PS3's in an Asterisk cluster to show up in here. "Massively Parallel VOIP" |
| 00:34.10 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 00:34.12 | ctooley | DMZ in a real network is definitely not just a static NAT mapping for connections that aren't previously expected. |
| 00:34.20 | drmessano | I think I have a sonicwall hooked up right now.. not sure |
| 00:34.28 | Carlos_PHX | The DMZ setting which just statically forwards to a specific IP is not a real DMZ and should really be called something else. |
| 00:34.29 | drmessano | Been farting around too much to know |
| 00:34.34 | ManxPower | that should handle "DMZ" just fine. |
| 00:34.40 | Carlos_PHX | And yeah, that definition of a DMZ is screwy. |
| 00:36.00 | drmessano | I really could care less.. but when the average user tells me they have * in a DMZ and have audio issues, it's fail |
| 00:36.53 | Carlos_PHX | Whoa, you have average users who can spell DMZ?!? |
| 00:37.19 | drmessano | I know.. not everyone is as smart as you, but yes |
| 00:37.38 | drmessano | All 3 letters too |
| 00:37.58 | Carlos_PHX | In caps? |
| 00:38.02 | hardwire | drmessano: sonicwall can eat me |
| 00:38.13 | hardwire | great product, still, eat me |
| 00:39.00 | *** join/#asterisk mib_3cujvn (i=be34811c@gateway/web/ajax/mibbit.com/x-add97bccd8ad511b) |
| 00:39.00 | Carlos_PHX | I actually have an e-mail from a customer with a known working config for Sonicwall and SIP if you want it, there was a packet type in there I didn't expect. |
| 00:39.07 | Carlos_PHX | Besides the obvious SIP/RTP. |
| 00:39.13 | hardwire | yeh |
| 00:39.16 | hardwire | you have to map the RTP |
| 00:39.18 | hardwire | evil |
| 00:39.21 | drmessano | Not really.. my shit works |
| 00:39.36 | hardwire | I choose you.. NAT |
| 00:40.47 | Carlos_PHX | T.38 testing. I feel like it's 1981 and I'm trying to bring up an ISDN circuit. |
| 00:42.32 | *** join/#asterisk sucituanbo (n=john@c-24-21-121-148.hsd1.wa.comcast.net) |
| 00:44.23 | drmessano | hmm |
| 00:44.44 | drmessano | Comcast gives a free domain and free hosted sharepoint to all business customers |
| 00:45.00 | drmessano | I wonder if they would cancel my account if I put up "Pr0np0int" |
| 00:45.31 | hardwire | heh |
| 00:45.37 | Carlos_PHX | You should try it. |
| 00:45.42 | Carlos_PHX | For the good of humanity. |
| 00:45.54 | hardwire | humanatees |
| 00:46.20 | hardwire | ok so.. I found this server, you guys may know about it.. but I call it the 2G1C server. |
| 00:46.24 | drmessano | It would probably take too long |
| 00:46.33 | hardwire | http://www.siliconmechanics.com/c1159/1u-twin-servers.php |
| 00:47.33 | drmessano | Comcasts hosted shit is incredibly slow |
| 00:48.05 | drmessano | They let you get a free subdomain from their comcastbiz.net domain for your email.. HAW |
| 00:48.39 | drmessano | drmessano@pr0np0int.comcastbiz.net <--- not valid in most 25 character limit submission forms |
| 00:51.27 | lanning | Seagate (as a company) has the same issue. |
| 00:51.49 | lanning | I was "Robert.Lanning@seagate.com" = 27 characters |
| 00:52.07 | hardwire | 25 chars is insane |
| 00:52.19 | russellb | russell@russellbryant.net ... exactly 25, yay |
| 00:52.22 | hardwire | back in the day when creativity was a 3 letter domain name. |
| 00:52.31 | hardwire | and they cost less than a few million |
| 00:52.44 | drmessano | When I worked for clear channel, I was 13 characters behind from the start |
| 00:52.47 | hardwire | spencersr@brutetechnologies.com |
| 00:52.48 | hardwire | I lose |
| 00:53.08 | drmessano | Well, 17 for the tld + @ |
| 00:53.26 | hardwire | thankfully I have brutetech.com as well :) |
| 00:54.49 | lanning | 2983477824.283792873429@compuserve.com :) |
| 00:55.12 | drmessano | Was that 4 (7)s or 5? Wait, hang on.. |
| 00:55.39 | hardwire | nice |
| 00:56.41 | drmessano | I really despise people who need a full, drawn out e-mail ePenis |
| 00:57.06 | drmessano | johnrobertwilkersonjr@johnrobertwilkersonjrindustriesllc.com |
| 00:57.33 | hardwire | I use my middle initial |
| 00:57.34 | drmessano | "GFY" |
| 00:57.36 | hardwire | it drives people nuts |
| 00:57.37 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 00:57.42 | hardwire | it's R |
| 00:57.48 | hardwire | I'm often party to pirate jokes. |
| 00:58.06 | drmessano | I use someone elses middle initial |
| 00:58.17 | hardwire | me too. |
| 00:58.26 | hardwire | somebody else uses R I'm sure. |
| 00:58.32 | drmessano | Just decided one day I didnt want to use mine.. so I was all like "fuck it" |
| 00:58.36 | drmessano | Went with another |
| 00:59.20 | lanning | My name is "Jim"... That's spelled "T-O-M" :) |
| 00:59.21 | drmessano | Someone asks me "So, is your middle name John? James?" "No, my middle initial is fake, sorry" |
| 01:00.14 | Carlos_PHX | @me.com gets it as short as it can be. |
| 01:00.31 | drmessano | When I got married, my wife was all like "WHAT? That is NOT your middle initial!" I had to apologize for weeks |
| 01:01.42 | drmessano | It was a good thing though.. Like 2 months later I told her I was white. She was less angry for the deceit. |
| 01:02.39 | drmessano | [20:30] <ServX> [C-Net] Report: Skype service in China recording, censoring messages <-- Nice |
| 01:03.18 | drmessano | I'll assume that's not "Skype" but someone legally using the name there |
| 01:03.34 | v4mp | so how do i do the setup for a queue with music on hold ? |
| 01:04.30 | Carlos_PHX | v4mp: http://www.orderlyq.com/asteriskqueues.html |
| 01:04.55 | Carlos_PHX | And http://www.voip-info.org/wiki-Asterisk+call+queues |
| 01:06.57 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 01:07.29 | drmessano | Asterisk announces partnership with Skype, which was actually a huge technological divide being bridged.. so not to be outdone, Fonality announces a partneship with Gizmo5 and are now including a module in their GUI for direct Gizmo5 config.............. which really sounds like little of nothing to me... |
| 01:08.43 | drmessano | That would be like HappyClownPBX partnering with Emacs |
| 01:08.54 | *** join/#asterisk Xentac (n=xentac@archlinux/developer/Xentac) |
| 01:08.58 | drmessano | "Emacs.. the official file editor of HappyClownPBX" |
| 01:09.06 | hardwire | really? |
| 01:09.13 | drmessano | ~happyclownpbx |
| 01:09.13 | jbot | [HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
| 01:09.35 | drmessano | ~Diahatsumashiniriki Keyotason |
| 01:09.38 | drmessano | Hmmm |
| 01:09.45 | drmessano | ~Diahatsumashiniriki Keyotason 200LP-A11 |
| 01:10.05 | drmessano | Ah well, theres some variation of that in there too |
| 01:10.37 | drmessano | Diahatsumashiniriki Keyotason 200LP-A11 SIP phone = very ^_^ |
| 01:11.01 | jaytee | "in rich sensual full 5.1 Surround Sound" |
| 01:11.03 | *** join/#asterisk edwin_quijada (n=macaruch@190.166.207.236) |
| 01:11.10 | edwin_quijada | Hi! |
| 01:11.21 | drmessano | Yes, the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone uses Dolby 5.1 |
| 01:11.36 | edwin_quijada | I have problem with T1 card openvox with asterisk 1.4.21 |
| 01:11.41 | lanning | DTS in beta |
| 01:11.44 | drmessano | It's also 1080p and HDMI compliant |
| 01:12.06 | jaytee | HD Skype is just around the corner |
| 01:12.20 | edwin_quijada | there is any issue with Dell SC440 server and Asterisk? |
| 01:12.24 | Xentac | I'm using trixbox (which uses freepbx) and having a problem with my ring groups, even though I set them to ringall they only actually ring the first 4 extensions |
| 01:12.30 | Xentac | anyone ever experience this? |
| 01:12.42 | drmessano | trixbox? burn.. |
| 01:12.56 | drmessano | Xentac: CPU and RAM specs? |
| 01:13.00 | Xentac | I know, it's probably pretty bad form to ask about trixbox et al in the asterisk channel... |
| 01:13.19 | Xentac | 2.7GHz cpu, 1 gig ram |
| 01:14.01 | drmessano | Might be that nasty ringall 100% CPU problem |
| 01:14.09 | Xentac | I heard something about that |
| 01:14.23 | drmessano | How many total extensions? |
| 01:14.26 | drmessano | in the ringall |
| 01:14.30 | Xentac | top tells me that the cpu is barely being used, if that's any consolation |
| 01:14.37 | Xentac | about 7 |
| 01:15.03 | hardwire | drmessano: google just called you a liar.. and me a naive. |
| 01:15.07 | drmessano | Dunno.. run into a lot of problems with ringall of ~20 extensions+ ringing about 1/4 |
| 01:15.20 | Xentac | nods. |
| 01:15.35 | drmessano | hardwire: For what? |
| 01:15.42 | hardwire | that phone :) |
| 01:15.48 | drmessano | ROFL |
| 01:15.51 | Xentac | I don't suppose there's a known fix for that problem, eh? ;) |
| 01:15.59 | drmessano | "Not available for EXPORT" |
| 01:16.02 | drmessano | RTFM |
| 01:16.15 | Carlos_PHX | Does HappyClownPBX do MP3 codec? |
| 01:16.16 | drmessano | Silly user |
| 01:16.26 | drmessano | MP3, and MP5 |
| 01:16.30 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 01:16.45 | jaytee | HappyClownPBX, hahaha |
| 01:16.49 | Carlos_PHX | User call today almost drove me to the MP5 |
| 01:17.45 | *** join/#asterisk Corydon76-dig (i=white@pdpc/supporter/bronze/Corydon76-home) |
| 01:17.45 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 01:17.46 | drmessano | Im thinking about a total rewrite of HappyClownPBX.. Last build wouldn't fit on a BlueRay DVD |
| 01:17.54 | *** join/#asterisk sCOTTo (n=sCOTTo@203-206-176-217.perm.iinet.net.au) |
| 01:17.55 | drmessano | it MAY be too big |
| 01:18.57 | hardwire | Carlos_PHX: you almost offed yourself? |
| 01:18.57 | sCOTTo | hey guys! |
| 01:19.15 | hardwire | an mp5 is quite a way to go. |
| 01:19.17 | lanning | you are not supposed to ship around blueray discs, you are supposed to ship 1TB drives |
| 01:19.23 | sCOTTo | im a dumb shit - just so you know - I am trying to wrapp my head around asterisk to understand what it does |
| 01:19.28 | sCOTTo | hehehe |
| 01:19.32 | hardwire | so.. guys |
| 01:19.34 | sCOTTo | BRB - Coffee needed |
| 01:19.39 | hardwire | alaska sucks for teh bandwidth |
| 01:19.43 | drmessano | sCOTTo: Thank you.. You saved me from calling you a dumbass |
| 01:19.48 | hardwire | I could seriously make a living shipping 1tb drives to colos in the main US |
| 01:19.51 | drmessano | sCOTTo: That is ROI |
| 01:20.06 | hardwire | "here's your SSH login.. have fun.. just run "shipdrive.sh" when you're finished" |
| 01:20.08 | jaytee | can you see Russia? |
| 01:20.19 | drmessano | When putin rears his head |
| 01:20.34 | hardwire | jaytee: I can see Russia. |
| 01:20.41 | hardwire | well |
| 01:20.46 | hardwire | I "could" see Russia once |
| 01:20.53 | hardwire | but the neighbor made his igloo too tall |
| 01:20.57 | *** join/#asterisk pcrane (n=pcrane@120.89.80.110) |
| 01:20.57 | hardwire | my view is wrecked |
| 01:21.15 | v4mp | how do u reload queues again ? |
| 01:21.16 | sCOTTo | RIO ? |
| 01:21.21 | sCOTTo | ROI ? |
| 01:21.40 | ManxPower | sCOTTo: You need to REALLY understand contexts. Just when you think you understand them is when you realize how much you don't understand about them. Understanding, and I mean REALLY understanding that an extension is a number, an extension is not a phone. A phone is a phone. extensions point to phones. |
| 01:21.52 | jaytee | so there's actually a market for VOIP at the frozen ass end of the world? |
| 01:22.01 | drmessano | As Putin rears his head and comes into the air space of the United States of America, where do they go? Itâs Alaska. Itâs just right over the border. It is from Alaska that we send those out to make sure that an eye is being kept on this very powerful nation, Russia, because they are right there, they are right next to our state. |
| 01:22.22 | drmessano | Indeed, Mrs. Palin |
| 01:23.21 | drmessano | Please, vote for John McCain. At least make Obama feel like he had to work for it. |
| 01:23.42 | hardwire | jaytee: it's a most interesting place |
| 01:23.58 | Carlos_PHX | v4mp: The same way you reloaded them the first time. |
| 01:24.07 | Carlos_PHX | No seriously folks, I'll be here all week. |
| 01:24.08 | ManxPower | Aren't there army bases up there to protect us from the russians? |
| 01:24.16 | drmessano | hardwire: Can you see alaska from your porch? |
| 01:24.26 | eric2 | palin's house is probably like a base |
| 01:24.28 | Carlos_PHX | reload app_queue.so |
| 01:24.32 | hardwire | drmessano: the answer is yes. |
| 01:24.47 | hardwire | eric2: the answer is no |
| 01:24.51 | drmessano | I heard that 6000 years ago, you could see Dinosaurs from a porch in alaska |
| 01:24.58 | v4mp | Carlos_PHX, i cnt remember and pressing up on cli im not finding it only commands i haven't used today so lost somewhere unless i didn't reload it earlier :/ |
| 01:25.24 | Carlos_PHX | That was a joke, since you said "reload them again..." |
| 01:25.32 | Carlos_PHX | Sorry, bad humor, will return to drinking now. |
| 01:25.34 | eric2 | american politics are like a broadway show |
| 01:26.53 | v4mp | i found it now and found a problem time to fix :) |
| 01:27.04 | drmessano | Sarah Palin shares a very narrow maritime border with reality |
| 01:27.38 | Carlos_PHX | drmessano: Like a DMZ? |
| 01:28.14 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 01:28.23 | drmessano | No, like a Norton Firewall 2009 |
| 01:28.50 | v4mp | ok its still ignoring the queue :/ |
| 01:29.01 | grandpapadot | Hi all. I'm working on some jabber integration with openfire enterprise. I can't seem to get the asterisk jabber client (1.4.x) to stop filling the cli with updates even with debug set to off in jabber.conf, any ideas? |
| 01:29.06 | drmessano | Norton Internet Latency 2008 |
| 01:29.28 | drmessano | Openfire Enterprise? |
| 01:29.44 | drmessano | Enterprise is extinct |
| 01:30.23 | grandpapadot | Openfire in any case. |
| 01:31.22 | *** join/#asterisk drbrown (n=chatzill@rrcs-24-123-224-200.central.biz.rr.com) |
| 01:31.40 | drbrown | anyone have problems adjusting the volume for the ringer on a spectralink 8002? |
| 01:32.04 | drmessano | I dunno.. I'm using Asterisk Enterprise Edition 2008, so it's probably different |
| 01:32.26 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 01:33.30 | grandpapadot | Ahh.. A full reload fixed it.. |
| 01:34.02 | hardwire | a reload reload :) |
| 01:34.19 | v4mp | any idea what the reason would be that [queuename] is set in queues.conf and the Queue(queuename) is in extensions.conf what would be the reason for it still being UNKNOWN both conf |
| 01:34.24 | v4mp | s have been reloaded |
| 01:34.49 | Carlos_PHX | drbrown: Ah, if only my Specralink was actually functional. |
| 01:35.11 | drmessano | Is that like a spectralink? |
| 01:36.06 | drbrown | Carlos_PHX: I will not be purchasing another myself |
| 01:36.20 | hardwire | great googamooga |
| 01:36.31 | drbrown | Carlos_PHX: What's wrong with yours? |
| 01:36.36 | Carlos_PHX | They are very similar, but cheaper since they have one less letter in the logo. |
| 01:37.01 | drmessano | I guess the T was left out for lack of telephony functions |
| 01:37.06 | drmessano | Need to remember that |
| 01:37.41 | Qwell | Cisco is only 5 letters |
| 01:37.49 | Qwell | just throwing that out there |
| 01:37.52 | Carlos_PHX | drbrown: It's really just down to getting all the details worked out I guess, the config is the most complex for any phone. I had a short time to make it work, and it's not something that can be done quickly. Set it aside for now. |
| 01:38.08 | Carlos_PHX | Qwell: Yes, but you're buying the bridge. |
| 01:38.13 | Qwell | true |
| 01:38.18 | Carlos_PHX | "I've got a bridge to sell ya!" |
| 01:38.24 | Qwell | ~ciscolicense |
| 01:38.25 | jbot | extra, extra, read all about it, ciscolicense is unless you gave Cisco your first born, you probably aren't legally authorized to use their phones. see http://www.ntbox.com/cisco-openletter.html |
| 01:38.34 | *** join/#asterisk ThipThip (n=justin@cpe-69-205-232-160.stny.res.rr.com) |
| 01:38.47 | Carlos_PHX | Does anyone use that Cisco junk? |
| 01:38.48 | ThipThip | Hello. Who wants to help a complete and utter newb? |
| 01:38.49 | drmessano | Cisco is a 6 letter word for overpriced |
| 01:38.54 | drbrown | Carlos_PHX: The config through me off for a while as well, you have to provision it utilizing the tftp option within your dhcp server |
| 01:39.01 | Strom_C | but cisco is only five letters |
| 01:39.01 | drmessano | Shit, I thought I got 6 letters |
| 01:39.09 | drmessano | They lied to me |
| 01:39.17 | Carlos_PHX | Cisco would never lie. |
| 01:39.29 | ThipThip | I have some very basic, fundamental questions about VOIP. The first is - how do outgoing calls work? In other words, what do I need to set up a server which is capable of placing calls to telephones? |
| 01:39.51 | Strom_C | ThipThip: read these documents |
| 01:39.53 | Strom_C | ~101 |
| 01:39.53 | jbot | 101 is, like, Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 01:39.55 | Strom_C | ~book |
| 01:39.55 | jbot | hmm... book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 01:40.04 | drmessano | ~happyclownpbx |
| 01:40.05 | jbot | [HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
| 01:40.06 | Carlos_PHX | ThipThip: You need either a way to connect the server to the phone network (cards) or a SIP service provider. |
| 01:40.12 | jaytee | hahahahah |
| 01:40.19 | drmessano | or an IAX service provider, biggot |
| 01:40.25 | Carlos_PHX | ROFL |
| 01:40.25 | *** join/#asterisk axisys (n=axisys@117.18.229.130) |
| 01:40.45 | drmessano | IAX are people too |
| 01:40.52 | Carlos_PHX | I have a love/hate with IAX |
| 01:41.04 | Carlos_PHX | So I treat it passive-agressively and co-dependently. |
| 01:41.15 | drmessano | I love IAX, I hate how most ITSPs suck at implementing it |
| 01:41.38 | Carlos_PHX | There's that, but we've had some massive fails even between two Asterisk servers. |
| 01:41.50 | drmessano | Later 1.4 versions? |
| 01:42.00 | Carlos_PHX | No, 1.2.x early and mid. |
| 01:42.04 | drmessano | Ohhh |
| 01:42.25 | Carlos_PHX | I'm not saying it's rational. That's why it's passive-aggressive and co-dependent. |
| 01:42.31 | Carlos_PHX | Plus I have a lot of 1.2.x out there. |
| 01:42.38 | drmessano | Yeah.. 1.2 didn't really include IAX2.. It was more like a wrapper for TDMoE |
| 01:42.52 | drmessano | O.o |
| 01:43.06 | drmessano | Later 1.4 versions IAX2 is really solid |
| 01:43.31 | Carlos_PHX | Nothing like having your T1 seeing 18mbps of random IAX packet resends to send you up to the clock tower. |
| 01:43.46 | Carlos_PHX | We're doing well with 1.4 <> 1.4 IAX |
| 01:43.59 | russellb | yay |
| 01:44.26 | russellb | a lot of work has gone into chan_iax2 in 1.4 ... especially the latest versions |
| 01:45.08 | Carlos_PHX | Early 1.4 versions would spike the CPU with >25 calls, that was fun. But yeah, very solid now. |
| 01:45.29 | russellb | needs to benchmark the latest chan_iax2 in trunk it at some point .... |
| 01:45.33 | russellb | lots of performance improvements |
| 01:45.37 | Carlos_PHX | I was just commenting to someone today that our 1.4 main gateway, running in VMware, has been rock solid for months. |
| 01:45.44 | russellb | awesome. |
| 01:45.53 | drmessano | I've been almost obnoxious about using 1.4 IAX2 <> ITSP for a few home systems I had built just because it made a good low risk lab.. and it's been MUCH better than the last time I tried it.. on 1.4.12 I guess, and then 1.2.something before that |
| 01:46.21 | russellb | drmessano: very nice to hear |
| 01:46.58 | Carlos_PHX | Did you guys know that when top hits 45 or so, call quality goes all to hell? |
| 01:47.03 | drmessano | As long as the providers IAX2 isn't broken this week, it seems to work.. Its VERY much working with IPKALL... I blow 150 or so calls through one box I set up for a few friends and have had no complaints |
| 01:47.23 | drmessano | 150 / day |
| 01:49.07 | drmessano | I bet it's made lmadsen's DUNDI clustering kick more ass |
| 01:50.32 | Carlos_PHX | I gave up on IAX with non-Asterisk providers. Too much effort. I have enough fun with just SIP and now T.38. |
| 01:53.37 | *** join/#asterisk Deeewayne (i=dwayne@76.29.245.9) |
| 01:53.37 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:54.18 | sCOTTo | ManxPower, ok so an extension is an external phone number? so really what I want are internal phones on that system that connect over the internet to be able to call each other etc etc - does that sound right ? |
| 01:54.32 | *** join/#asterisk Steve_J-obs (n=Chris123@209.58.251.50) |
| 01:55.15 | Carlos_PHX | sCOTTo: An extension does not equal an external number. |
| 01:55.29 | sCOTTo | Carlos_PHX, yes I understand that. |
| 01:55.49 | sCOTTo | so I can have INTERNAL numbers that are located all over the world right ? |
| 01:56.08 | v4mp | does agents.conf need to be reloaded ? |
| 01:56.11 | Carlos_PHX | Of course. |
| 01:56.24 | Carlos_PHX | But the numbers aren't located anywhere. |
| 01:56.25 | Carlos_PHX | Phones are. |
| 01:56.41 | Carlos_PHX | Extensions point to phones or devices, they can be anywhere. |
| 01:56.53 | sCOTTo | Carlos - can I use a current VOIP number and settings to my current provider inside asterisk so that when I DO make an external phone call it automatically goes through that ?? |
| 01:56.57 | Carlos_PHX | It's CRITICAL to truly understand extensions vs. contexts vs. devices. |
| 01:57.13 | Carlos_PHX | Yes |
| 01:57.18 | sCOTTo | Carlos_PHX, is there a doc on that ? |
| 01:57.21 | Carlos_PHX | Asterisk is an endpoint just like a phone. |
| 01:57.26 | Carlos_PHX | Same settings as your phone. |
| 01:57.33 | Carlos_PHX | Look in the sample sip.conf |
| 01:57.45 | sCOTTo | ok |
| 01:57.50 | drmessano | Asterisk is a big orange wirenut |
| 01:57.51 | Carlos_PHX | Or the sip.conf instructions on voip-info. |
| 01:58.02 | sCOTTo | ok thanks guys |
| 01:58.04 | Carlos_PHX | It can be that too. Asterisk is what you make of it. |
| 01:58.07 | sCOTTo | bbs |
| 01:58.17 | drmessano | True |
| 01:58.48 | drmessano | I use Asterisk as my browser, for checking my e-mail, and it does my taxes |
| 01:58.52 | Carlos_PHX | exten => s,1,WingNut(orange|big) |
| 01:59.27 | Carlos_PHX | Oh, you wanted the wirenut app, didn't you? |
| 01:59.27 | drmessano | 1.2 caused me to get audited.. 1.4 has been much better for me |
| 01:59.28 | Steve_J-obs | hi everybody...Is there anyone here nice who would want to chat a bit about what's going on when a "PACKET2PACKET bridging" message occurs? |
| 01:59.48 | russellb | nothing important ... the message is more debug than anything |
| 02:00.02 | Carlos_PHX | 1.4 demagnetized all my credit cards, prank called DHS, set the freezer temp to defrost, and chased kids with my snowblower. |
| 02:00.03 | hardwire | Steve_J-obs: it means you're doing things correctly :) |
| 02:00.27 | hardwire | Steve_J-obs: are you familiar with packet2packet? |
| 02:00.38 | drmessano | I had that sort of issue with 1.6 right after I got out of prison |
| 02:00.38 | Steve_J-obs | well... correct...except that it is only happpening to me 20% of the time...80% of the time the call drops :( |
| 02:00.41 | Carlos_PHX | Steve_J-obs: It's basically a shortcut for non-transcoded packets. |
| 02:00.53 | *** join/#asterisk pcrane (n=pcrane@120.89.80.154) |
| 02:01.37 | Steve_J-obs | previous to that, invariably, I get "droppiong extra frame of g729 since we already have a VAD frame at the end" |
| 02:01.52 | drmessano | Prison teaches you a lot of things.. I learned to transcode with pen and paper in an 8x8 cell. |
| 02:02.11 | Steve_J-obs | which causes bridging to occur only 20% of the time |
| 02:02.18 | Carlos_PHX | drmessano: Have you ever been in a Turkish prison? |
| 02:02.50 | hardwire | Does eating Turkish food qualify? |
| 02:02.53 | jaytee | "do you like movies about gladiators?" |
| 02:03.30 | drmessano | Carlos_PHX: Are you coming on to me? |
| 02:03.42 | Carlos_PHX | Do you want me to be? |
| 02:03.58 | Steve_J-obs | besides carlos... that means the codec is either not transcoding well |
| 02:04.08 | drmessano | It can be our secret.. |
| 02:04.10 | v4mp | hmm any idea why the call wasn't sent to the agent ? the login from agent etc ran through fine and the extension to listen on was done |
| 02:04.21 | *** join/#asterisk pcrane (n=pcrane@120.89.80.154) |
| 02:04.23 | Carlos_PHX | Nobody reads IRC, nobody will ever know. |
| 02:04.51 | drmessano | Yeah, IRC is so 1963 or so |
| 02:05.05 | drmessano | I was IRC'ing in Saigon |
| 02:06.30 | hardwire | Saigon of course only existed until 1970. |
| 02:06.34 | Steve_J-obs | maybe I should ask how I can fix "dropping extra frame of g729 since we already blah blah" |
| 02:06.40 | hardwire | At which point drmessano floated to Thailand |
| 02:06.51 | drmessano | I spent most of the war running wireless ethernet cable 2 clicks south of pyuntang |
| 02:07.08 | drmessano | I was in Saigon long before 1970 |
| 02:07.14 | drmessano | Well |
| 02:07.17 | drmessano | "officially" |
| 02:07.25 | Carlos_PHX | You were always trying to get some pyuntang. |
| 02:09.56 | *** part/#asterisk v4mp (n=Gary@82.118.111.250) |
| 02:10.04 | *** join/#asterisk v4mp (n=Gary@82.118.111.250) |
| 02:10.24 | Carlos_PHX | Steve_J-obs: Is this between two phones, phone to PSTN, or ...? |
| 02:11.16 | hardwire | drmessano: it seems I have an incomplete history on you |
| 02:11.31 | hardwire | at which point were you inducted into the A-Team ? |
| 02:11.48 | drmessano | HA |
| 02:11.53 | Steve_J-obs | this is between to asterisks |
| 02:12.20 | Carlos_PHX | Steve_J-obs: IAX or SIP? Asterisk version? |
| 02:12.27 | Steve_J-obs | sip to sip |
| 02:12.39 | Steve_J-obs | my version is 1.4.21, the other I dont know |
| 02:12.55 | drmessano | The A-Team were a bunch of no-talent hacks.. They couldn't make a TV about the stuff I did.. it would scare people |
| 02:13.18 | russellb | Steve_J-obs: 1.4.22 is coming out tomorrow, you can try it now if you want |
| 02:13.23 | Carlos_PHX | Steve_J-obs: Do other CODECs work? |
| 02:13.24 | russellb | svn co http://svn.digium.com/svn/asterisk/tags/1.4.22 |
| 02:13.51 | Steve_J-obs | everything started happening since I added allow=g729 to sip.conf |
| 02:14.12 | Carlos_PHX | And I assume you have sufficient licenses for the number of calls? |
| 02:14.27 | Steve_J-obs | o yes...unlimited license |
| 02:14.27 | Carlos_PHX | Is 729 the only codec, or multiple? |
| 02:14.32 | drmessano | ROFL |
| 02:14.36 | Steve_J-obs | g729 is the only codec |
| 02:14.38 | russellb | "unlimited license" |
| 02:14.40 | russellb | i call BS! :-p |
| 02:14.42 | Carlos_PHX | Er |
| 02:14.42 | drmessano | "unlimited license" |
| 02:14.46 | drmessano | ha, beat me to it |
| 02:14.48 | *** join/#asterisk RypPn (i=TuMbL@80.177.214.249) |
| 02:15.03 | Carlos_PHX | < puts BS flag back in pocket, you guys got it. |
| 02:15.22 | Steve_J-obs | look, that part I don't know, I just know that the person that installed it for me handled that |
| 02:15.22 | drmessano | Steve_J-obs: Where did you buy that unlimited license from? |
| 02:15.29 | drmessano | HAHAHHAHAHA |
| 02:15.45 | drmessano | "I didn't download that cracked Photoshop.. my FRIEND did" |
| 02:15.55 | Carlos_PHX | The check is in the mail |
| 02:16.10 | Steve_J-obs | I know that I have sufficient licenses, and the person in charged of that installation said that to me |
| 02:16.17 | hardwire | Steve_J-obs: I think your support quota just got all filled up somehow |
| 02:16.29 | *** join/#asterisk sucituanbo (n=john@24.21.121.148) |
| 02:16.53 | russellb | Steve_J-obs: have you checked your CPU load? g729 eats up CPU |
| 02:16.54 | *** join/#asterisk voxter (n=voxter@mail.metrobridge.com) |
| 02:17.07 | Steve_J-obs | please guys.. I really dont know anything about this...it is not a licensing issue either since other calls are using it |
| 02:17.07 | phix | Steve_J-obs: gimme an iPhone |
| 02:17.19 | drmessano | Steve_J-obs: You can get support for that G729 codec you purchased.. Give Digium a call |
| 02:17.29 | Carlos_PHX | Yeah, that's my next suggestion. |
| 02:17.41 | hardwire | Steve_J-obs: its true.. it *is* supported by their paid professionals |
| 02:17.44 | Carlos_PHX | Though you saying other calls are using it doesn't mean it can't be a license issue. |
| 02:17.47 | Carlos_PHX | That's my point. |
| 02:17.55 | Carlos_PHX | Other calls may have all the licenses. |
| 02:17.57 | drmessano | Yes |
| 02:18.06 | voxter | Hey polycom dudes. In recent polycom firmware, when dialing numbers, the screen's cursor behaves like it is in character entry mode, leaving the cursor on the just-dialed digit for a couple seconds before advancing on the screen. Anyone know how to disable that behavior? |
| 02:18.09 | drmessano | Digium would love to hear from you if you're having a problem |
| 02:18.23 | russellb | we would? ^_^ |
| 02:18.28 | voxter | hardwire: hola! |
| 02:18.32 | drmessano | russellb: Oh yes |
| 02:18.32 | hardwire | its packet2packet tho.. so not like licenses are needed :) |
| 02:18.37 | Steve_J-obs | can anybody please explain why is this a licensing issue? |
| 02:18.38 | hardwire | voxter: don't you dare! |
| 02:18.46 | Carlos_PHX | He said p2p works, but not other calls. |
| 02:18.46 | phix | russellb == digium? |
| 02:18.53 | Carlos_PHX | Presumably transcoded? |
| 02:18.57 | russellb | I am not Digium. Digium is a compnay :) |
| 02:19.06 | phix | russellb: you work for them? |
| 02:19.07 | Carlos_PHX | That's why I was asking about the call destination. |
| 02:19.09 | russellb | nods |
| 02:19.14 | hardwire | russellb: you have the cutest accent. |
| 02:19.19 | drmessano | HA |
| 02:19.21 | russellb | hardwire: O.O |
| 02:19.22 | phix | russellb: awesome, what's your direct line :P |
| 02:19.28 | russellb | phix: 911 |
| 02:19.33 | hardwire | phix: 500 |
| 02:19.36 | phix | heh |
| 02:19.39 | Carlos_PHX | russellb: < Has been in a Turkish prison. |
| 02:19.39 | carrar | russellb, if you were Digium, what would you be? Quad T1 card? |
| 02:19.52 | russellb | I would be Asterisk. |
| 02:19.53 | Carlos_PHX | Did you just call him square? |
| 02:19.54 | carrar | heh |
| 02:19.57 | voxter | so has anyone seen this new behavior in polycom firmware? its driving me nuts |
| 02:20.01 | drmessano | Unlimited license... |
| 02:20.04 | hardwire | russellb invited the quad span card.. and every chuck norris joke. |
| 02:20.07 | phix | russellb: what's the country code on that :) |
| 02:20.09 | hardwire | so be careful ok? |
| 02:20.16 | russellb | phix: 800 km |
| 02:20.20 | Carlos_PHX | voxter: Yes, I've seen it, and continue to see it. |
| 02:20.21 | drmessano | russellb has an unlimited G729 license |
| 02:20.34 | carrar | Thats not hard to get |
| 02:20.37 | carrar | heh |
| 02:20.43 | drmessano | Apparently |
| 02:20.43 | voxter | Carlos_PHX: you dont know how to shut it off either, hm? I had to go onsite to a customer today as they were claiming it to be a bug. GRR. |
| 02:21.03 | Carlos_PHX | Yes, I know how to fix it. I'm replacing all the Polycoms with Linksys 942. |
| 02:21.03 | hardwire | I'd bother polycom relentlessly |
| 02:21.12 | voxter | Carlos_PHX: hah. |
| 02:21.17 | drmessano | I have a keygen for Asterisk 1.4 |
| 02:21.29 | russellb | drmessano: hax! |
| 02:21.30 | voxter | Carlos_PHX: Ive started pushing Aastra instead of polycom as of late. Simply because of the wonderful XML flexibility |
| 02:21.38 | Carlos_PHX | It's something to do with the dial plan in the phone and its pattern matching, but I'm sorry I can't give you detailed fixes. |
| 02:21.42 | Carlos_PHX | I've stumbled through it. |
| 02:21.49 | hardwire | voxter: the aastra phones have LCD's that make me want to cry |
| 02:21.56 | carrar | -- Executing [russellb@asterisk] Set("message", "Hi Russel!!") in new stack |
| 02:21.57 | hardwire | and not in a good way |
| 02:21.57 | voxter | hardwire: that too. and the buttons. |
| 02:22.06 | hardwire | heh |
| 02:22.14 | voxter | Ive brought it up with aastra many many times |
| 02:22.14 | Carlos_PHX | drmessano: Can it get around Asterisk Genuine Advantage? |
| 02:22.24 | drmessano | I was working on a keygen for 1.6 but apparently those &%$%#$ at Digium put some crap copy protect in there.. so I have a nice loader for it I wrote |
| 02:22.51 | russellb | carrar: hi2u2 |
| 02:23.20 | drmessano | Asterisk_1_6_Loader_DrMeSsAnO-o00oo0o0o.tar.rar.zip.gz.zoo |
| 02:23.22 | hardwire | drmessano: or you could just not use g729 |
| 02:23.29 | hardwire | seems like a good option to me |
| 02:23.46 | drmessano | It's not for G729 |
| 02:23.49 | drmessano | It's for Asterisk |
| 02:23.59 | hardwire | beats drmessano |
| 02:24.10 | hardwire | don't you dare tell Mr T |
| 02:24.17 | drmessano | has to explain EVERYTHING to hardwire |
| 02:24.22 | hardwire | yes |
| 02:24.25 | hardwire | you and my girlfriend |
| 02:24.51 | russellb | hardwire: you don't have a girlfriend ... you're on IRC too much |
| 02:25.21 | hardwire | russellb: she's so life like tho |
| 02:25.30 | *** join/#asterisk moy (n=moy@189.169.68.109) |
| 02:25.32 | hardwire | earlier today she stood there and proclaimed "this fabric still smells" |
| 02:25.36 | hardwire | yet she held no fabric |
| 02:25.42 | hardwire | and she never pointed at anything |
| 02:25.49 | hardwire | I could have swore I must have missed several words |
| 02:25.53 | hardwire | she too.. had to explain "everything" |
| 02:26.03 | drmessano | I have a patch for Asterisk that allows unlimited IAX2 as well |
| 02:26.07 | hardwire | apparently her new pants have the "new" smell. |
| 02:26.11 | hardwire | drmessano: lies |
| 02:26.18 | drmessano | Nope |
| 02:26.19 | russellb | drmessano: that's not possibly unless it wasn't IAX2 anymore. |
| 02:26.34 | russellb | the protocol has a built-in hard limit on number of concurrent calls on a given IAX2 endpoint. |
| 02:26.36 | hardwire | russellb: can you patch my dsl into a full T1? |
| 02:26.38 | russellb | fail. |
| 02:26.41 | hardwire | err |
| 02:26.43 | hardwire | drmessano: ^ |
| 02:26.46 | russellb | hardwire: for $ 800 km, sure |
| 02:26.56 | hardwire | kamillion? |
| 02:27.08 | russellb | 800 kilometer dollars |
| 02:27.13 | hardwire | hah |
| 02:27.20 | Qwell | dollars is a redundant unit of measurement |
| 02:27.28 | hardwire | prefers dollops |
| 02:27.30 | Qwell | 800km is currency |
| 02:27.37 | russellb | Qwell: sry :( |
| 02:27.40 | drmessano | Fine, i'm not gonna let anyone use my RTP patch for > 65,534 ports |
| 02:27.47 | russellb | drmessano: hax!! |
| 02:28.12 | jaytee | hehehe |
| 02:28.13 | hardwire | drmessano: you even got the 0 port? |
| 02:28.18 | hardwire | ffs |
| 02:28.28 | Qwell | my kernel uses 2.4 billion ports |
| 02:28.34 | drmessano | I've been running over 80,000 RTP ports in testing |
| 02:28.38 | Qwell | of course, I had to modify IP |
| 02:28.47 | russellb | you mean UDP |
| 02:28.56 | russellb | failllll |
| 02:28.57 | hardwire | my kernel uses n*16^2 ports.. where n represents my IP's |
| 02:29.04 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 02:29.12 | Qwell | hmm, is port a UDP layer thing? |
| 02:29.21 | Qwell | raw sockets don't have ports I guess.. |
| 02:29.21 | *** join/#asterisk i3inary (n=shaman@cpe-76-88-94-149.san.res.rr.com) |
| 02:29.39 | russellb | Qwell: sir yes sir |
| 02:29.40 | Qwell | makes sense. |
| 02:29.47 | Qwell | protocol layer |
| 02:29.54 | drmessano | my rtp.conf rtpstart=0 rtpend=81234 |
| 02:30.11 | hardwire | haha |
| 02:30.21 | drmessano | (with patch applied) |
| 02:30.41 | hardwire | seriously folks.. you can get more thn a few million ports IF YOU USE DNS SRV |
| 02:30.45 | hardwire | which drmessano has a patch for |
| 02:31.03 | Carlos_PHX | Script kiddies. I'm running thousands of concurrent calls on the same single port. |
| 02:31.06 | drmessano | Thats not entirely true |
| 02:31.23 | Carlos_PHX | I just have the users say who they want the packets to go to really fast before each word. |
| 02:31.28 | drmessano | DNS SRV doesn't use ports.. so you can run millions of calls |
| 02:31.40 | drmessano | It's portless |
| 02:32.00 | drmessano | Just use a * instead of a port number.. that's a wildcard |
| 02:32.12 | Carlos_PHX | No, that's a PBX. |
| 02:32.23 | Carlos_PHX | Or a big orange wirenut. |
| 02:33.34 | drmessano | _iax2._udp SRV 0 0 * pbx.domain.com. <--- sexy |
| 02:35.44 | voxter | i need to dig some more into utilizing SRV |
| 02:35.54 | hardwire | it's teh eazy |
| 02:35.59 | drmessano | SRV is actually neat |
| 02:36.05 | hardwire | and incredibly handy |
| 02:36.07 | drmessano | Started messing with it for XMPP |
| 02:36.09 | hardwire | and ties in well with clustering |
| 02:36.10 | hardwire | ftw |
| 02:36.16 | drmessano | Then SIP |
| 02:36.30 | hardwire | isn't it just _iax? |
| 02:36.36 | voxter | ya, ive got a simple 5060 entry for sip/iax and xmpp |
| 02:36.42 | voxter | but i havent dug into its power yet. |
| 02:36.51 | hardwire | it's about as powerful as your client :) |
| 02:36.55 | hardwire | so be careful |
| 02:36.57 | hardwire | bbl.. going home |
| 02:37.35 | voxter | im going to do the same |
| 02:37.44 | voxter | just moved into my new office today, woo! |
| 02:39.20 | drmessano | Actually, I have no idea if _iax2 or _iax is appropriate.. I've actually USED it for XMPP and SIP, seems like I found noted somewhere that * looked for _iax2, but I may just be imagining that |
| 02:40.48 | Carlos_PHX | We're just rolling out SRV with SER for fault tolerance, powerful stuff. |
| 02:40.50 | russellb | iax2 probably |
| 02:41.15 | russellb | actually, i'm wrong |
| 02:41.16 | Qwell | ruuusssselllllll |
| 02:41.19 | drmessano | I think I looked once, a long time ago, and found iax2.. may have dug through code |
| 02:41.20 | Qwell | Party Sunday |
| 02:41.22 | russellb | chan_iax2.c does an SRV lookup using "iax" |
| 02:41.23 | drmessano | Ok |
| 02:41.25 | Qwell | you should totally go |
| 02:41.26 | russellb | Qwell: yes! |
| 02:41.27 | drmessano | Sweet |
| 02:41.28 | russellb | i plan to. |
| 02:41.31 | Qwell | woot! |
| 02:41.45 | drmessano | I got invited |
| 02:41.46 | Qwell | drmessano: you're totally not invited |
| 02:41.46 | russellb | hopefully my brain will have fully recovered by then |
| 02:41.48 | Qwell | pwnt |
| 02:41.50 | russellb | heh |
| 02:41.52 | voxter | hahah |
| 02:41.55 | Qwell | psychic |
| 02:42.01 | voxter | Qwell lays the smack down. |
| 02:42.27 | Qwell | drmessano: not until you test :P |
| 02:42.29 | drmessano | I was completely invited by some guy from Digium that spoke at a trixbox con or something |
| 02:42.31 | Qwell | voxter: you too! |
| 02:42.37 | drmessano | Dont remember his name |
| 02:42.51 | russellb | drmessano: we call him "nub" around the office |
| 02:43.01 | Qwell | :( |
| 02:43.13 | russellb | <3 jk! |
| 02:43.18 | drmessano | Aww.. don't go all green in the face |
| 02:44.35 | drmessano | A lot of people don't know this about me.. but i've never actually seen a CLI |
| 02:45.02 | drmessano | Trixbox is as close to * as I will ever get.. <----- GUI ------ 10 foot pole -----> |
| 02:45.25 | russellb | sorry ... we're having an Asterisk CLI party, so clearly you can't come |
| 02:45.34 | drmessano | Someone told me to reload SIP the other day, and I was like "ctrl-alt-del" |
| 02:47.03 | drmessano | If it doesn't have a "Start" or "Kerry" button, I am like all totally like "Whoa, it's dark in here" |
| 02:47.14 | voxter | someone told me to load trixbox the other day and i was all 'load app_trixbox.so' : "cannot open app_trixbox.so: no such file or directory" |
| 02:47.20 | voxter | It was devastating. |
| 02:47.39 | russellb | every time someone installs trixbox, god kills a kitten |
| 02:47.44 | Qwell | trixbox totally needs a "Kerry" button |
| 02:47.56 | Qwell | drmessano: also, did I tell you about the awesomeness that was itexpo? |
| 02:48.02 | voxter | they should make the logo a kerry bobblehead |
| 02:48.18 | drmessano | No, I never got details.. |
| 02:48.19 | Qwell | Andrew would not acknowledge my existence. Kerry came up and introduced himself, Andrew came up with him...didn't say a word. |
| 02:48.26 | drmessano | ha |
| 02:48.34 | jaytee | I loaded trixbox on one computer and it logged itself into one of my other computers and ate my homework |
| 02:48.35 | Qwell | Bill Miller told them the night before that I was behind AsteriskNOW 1.5 |
| 02:48.44 | Qwell | Andrew was clearly pissed. |
| 02:48.45 | drmessano | Awww that sucks |
| 02:48.46 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 02:48.49 | drmessano | Totally ruined the fun |
| 02:48.57 | Qwell | drmessano: it added to it |
| 02:49.07 | voxter | Qwell: hahaha |
| 02:49.10 | Qwell | also, xchat thinks "awesomeness" is a word |
| 02:49.11 | drmessano | For a different reason, no doubt |
| 02:49.23 | voxter | Qwell: I saw them both at astricon, and they very much kept to themselves this year. |
| 02:49.24 | russellb | Qwell: and then they're all like, "i can has your packages?" |
| 02:49.30 | russellb | we want to even _less_ actual work |
| 02:49.31 | voxter | Qwell: also, there was no trixbox booth :) |
| 02:49.33 | Qwell | russellb: dude, you don't even know |
| 02:49.41 | Qwell | YES. That happened. |
| 02:49.49 | russellb | nods :-D |
| 02:49.53 | voxter | rofl's |
| 02:49.55 | Carlos_PHX | The Trixbox "party" wasn't all that exciting. |
| 02:50.01 | voxter | there was a party? |
| 02:50.09 | voxter | clearly it was a hit |
| 02:50.10 | Qwell | he wants my 1.6 packages.. somebody asked them a couple days ago if they were planning on adding 1.6 :) |
| 02:50.15 | Carlos_PHX | Yeah, ninth floor, since they didn't pay for booth space. |
| 02:50.23 | Qwell | (I don't actually have 1.6 packages...he just thinks I do, I guess) |
| 02:50.30 | voxter | Carlos_PHX: lame. |
| 02:50.34 | Carlos_PHX | Quite. |
| 02:50.40 | drmessano | For some unknown reason, I pulled the Gizmo5 module from the Trixbox repo and started minor edits to make it work native in FreePBX... just so I can piss in their kool-aid |
| 02:50.41 | voxter | Carlos_PHX: I was drinking with digum instead. :P |
| 02:50.43 | Carlos_PHX | And I've been to Fonality parties before they were good. |
| 02:50.44 | Qwell | "he thinks I do"...why does that set of my grammar-detector? |
| 02:50.53 | Carlos_PHX | voxter: I did both. |
| 02:50.58 | Carlos_PHX | I'm cheap and easy. |
| 02:51.12 | voxter | Carlos_PHX: not in that particular order necessarily, of course, right? |
| 02:51.15 | Qwell | Carlos_PHX: wait, they didn't have a booth? |
| 02:51.23 | russellb | Qwell: corect |
| 02:51.25 | Carlos_PHX | Trixbox? |
| 02:51.26 | Qwell | they had a huge booth at itexpo the week before |
| 02:51.27 | Carlos_PHX | Right |
| 02:51.38 | russellb | instead they passed out postcards advertising their party in a room on the 9th floor |
| 02:51.38 | Qwell | right at the door too |
| 02:51.41 | Carlos_PHX | They just got a room and had a party, passed out fliers and stuff. |
| 02:52.05 | Carlos_PHX | The party was not only lame, the sleaze factor of the whole thing was revolting. |
| 02:52.05 | Qwell | I had to walk by whats-her-face every time... |
| 02:52.05 | drmessano | The Trixbox-Gizmo5 integration is gonna shake the world of open sores |
| 02:52.12 | Qwell | Carlos_PHX: I can imagine |
| 02:52.24 | russellb | drmessano: NO F'ING WAY ... they support SIP now? |
| 02:52.26 | Qwell | russellb: do you have any idea who she is? |
| 02:52.37 | Qwell | she always gives me such dirty looks |
| 02:52.43 | russellb | no clue |
| 02:52.47 | russellb | but i know who you're talking about |
| 02:52.48 | Qwell | know who I'm talking about? |
| 02:53.03 | Qwell | the one in that awesome picture Lauren took :D |
| 02:53.09 | russellb | nods |
| 02:53.20 | russellb | the one where their booth was empty and ours was full? ^_^ |
| 02:53.24 | Qwell | that's one of my favorite pictures EVER |
| 02:53.29 | Carlos_PHX | Link? |
| 02:53.42 | Qwell | Carlos_PHX: I really should bookmark it.. it always takes me forever to find |
| 02:54.02 | Qwell | BUT, I'll look, because it's that great |
| 02:54.13 | Carlos_PHX | I'll make sure it gets around some. |
| 02:54.24 | drmessano | I'll post it on the trixbox forums.. |
| 02:54.34 | drmessano | Boring news day |
| 02:54.46 | voxter | heh i got a good lashing when i posted the remote root exploit code on the trixbox forums a month or two ago |
| 02:54.52 | drmessano | rofl |
| 02:55.08 | drmessano | They are such communists.. and not in a good way |
| 02:56.04 | drmessano | Surely that kind of shenanigans is against the gpllaw |
| 02:56.06 | drmessano | O.o |
| 02:56.18 | Qwell | http://flickr.com/photos/laurensanderson/1470812020/ |
| 02:56.41 | drmessano | HA |
| 02:56.54 | drmessano | "hey guys, we just upgraded hudlite..... guys...." |
| 02:58.01 | russellb | that picture is classic |
| 02:58.09 | Qwell | http://farm2.static.flickr.com/1147/1470812020_0c8116f15a_b.jpg |
| 02:58.09 | Carlos_PHX | Truly awesome. |
| 02:58.11 | Qwell | larger |
| 02:58.17 | voxter | bahaha |
| 02:58.31 | voxter | your green glowing box is not that amazing dudes. |
| 02:58.31 | Qwell | that look on her face is priceless |
| 02:58.39 | drmessano | HAW |
| 02:58.57 | Qwell | ANYWAYS, yeah.. she always gives me dirty looks |
| 02:58.58 | Carlos_PHX | I was there, but couldn't see the situation through all the people at the Digium booth. |
| 02:59.00 | drmessano | What would have truly pwned.... "hey guys, can we borrow some of these chairs?" |
| 02:59.16 | Carlos_PHX | "you're not going to need them" |
| 02:59.22 | drmessano | Exactly |
| 02:59.24 | Qwell | AND |
| 02:59.28 | Qwell | if you look in the background... |
| 02:59.33 | Qwell | you can see Andrew at our booth :P |
| 02:59.38 | voxter | I was just gonna say.. |
| 02:59.51 | Qwell | he's right in front of the door |
| 02:59.53 | voxter | hes in front of the door |
| 02:59.55 | voxter | yea |
| 03:00.00 | Carlos_PHX | I'm pretty sure that cow skull is looking at their booth though. |
| 03:00.12 | Carlos_PHX | Maybe Trixbox has super cow powers. |
| 03:01.06 | drmessano | I love how the trixbox appliance has a built-in blue screen of death |
| 03:01.36 | Carlos_PHX | The switchvox woman makes a much better booth babe too. |
| 03:01.43 | russellb | gpllaw might not appreciate your use of the microsoft trademarked term BSoD |
| 03:02.11 | russellb | Carlos_PHX: but the great thing is she's not a booth babe. She helped _make_ the stuff |
| 03:02.15 | drmessano | HAW |
| 03:02.19 | Carlos_PHX | Oh yeah, I know that. |
| 03:02.28 | Carlos_PHX | That's even better. |
| 03:02.40 | russellb | the switchvox crew rox0rz |
| 03:02.49 | voxter | yah those guys have a stellar product |
| 03:02.56 | voxter | I wish i had done half of what they did |
| 03:03.05 | voxter | now if they'd just include bulk extension provisioning.... :) |
| 03:03.50 | drmessano | I remember when the trixbox appliance first came out.. people rushing to be the first to get one.. I was thinking there were probably quite a few cases of "hey boss, remember that trixbox I was telling you about? I ordered their new PBX appliance, here look" "What the hell is that green thing?" "Thats IT, boss." "Johnson, you effing idiot" |
| 03:03.59 | *** join/#asterisk n3hxs (n=HAMming@75.145.100.230) |
| 03:04.20 | Qwell | russellb: did she get like no pictures of Sokol standing on the table? |
| 03:04.23 | russellb | clearly your phone system in the rack in the closet needs to glow |
| 03:04.28 | Qwell | THAT was epic, |
| 03:04.30 | russellb | Qwell: *shrugs* |
| 03:04.40 | Qwell | that man can yell |
| 03:05.22 | drmessano | Clearly your phone system needs to be a full rack deep 2U box that uses so much power that calling it "green" is just a sad, tragic irony |
| 03:05.57 | Carlos_PHX | I was wondering what to do with all the extra power and cooling capacity in the server room, so the lights really help with that. |
| 03:06.33 | Qwell | http://flickr.com/photos/laurensanderson/1745822729/in/set-72157602226887812/ |
| 03:06.37 | Qwell | yeah, she did :D |
| 03:06.59 | Qwell | He managed to get the attention of every single person in the exhibit hall. |
| 03:07.04 | voxter | cant you buy switchvox without hardware? |
| 03:08.12 | russellb | i don't think end users can ... but i *think* that resellers and such can ... |
| 03:08.13 | drmessano | I think a nice trixbox appliance with an unlimited G729 license would make an awesome octoberween gift |
| 03:08.15 | russellb | don't quote me on that. |
| 03:08.33 | voxter | i should talk to them about reselling it, see what they say |
| 03:08.38 | Carlos_PHX | Don't they all include unlimited 729? |
| 03:08.55 | Carlos_PHX | I mean, our Fonality PBXtra did. |
| 03:09.03 | Qwell | O.o |
| 03:09.12 | Carlos_PHX | Yeah, you don't wanna know more. |
| 03:09.13 | russellb | oh man, don't get me started on the PBXtra mods to asterisk |
| 03:09.22 | voxter | oh no, do, im curious :) |
| 03:09.22 | drmessano | Carlos_PHX: Only in former Soviet Republics with unnecessary vowels |
| 03:09.27 | Qwell | Carlos_PHX: sure I do |
| 03:09.28 | voxter | did they distribute the intel codec? |
| 03:09.38 | Qwell | if they're violating copyright and patent law... |
| 03:09.58 | drmessano | Hang on. |
| 03:10.00 | Carlos_PHX | I can't remember all the details, but when Kevin Fleming worked on it (before he worked at Digium), he freaked. |
| 03:10.08 | drmessano | Lets give them the benefit of the doubt |
| 03:10.12 | Qwell | hmm |
| 03:10.22 | Carlos_PHX | I think it was supposed to be the Intel one and was not after all. |
| 03:10.23 | drmessano | It may not be "unlimited", but a "fuggedaboutit" license |
| 03:10.24 | voxter | why do we never have these conversations at astricon? :) heh |
| 03:10.25 | drmessano | That, is ok |
| 03:10.29 | Carlos_PHX | Oh yeah, it included non-free code. |
| 03:10.50 | Carlos_PHX | This was many years ago, I know they cleaned up since. |
| 03:10.59 | Carlos_PHX | Asterisk 0.something |
| 03:11.14 | Qwell | The trainer at the thing I was at, tried to say that Digium owns the patents on g729 |
| 03:11.19 | Qwell | I VERY quickly corrected him. |
| 03:11.23 | drmessano | ha |
| 03:11.43 | Carlos_PHX | fuggedaboutit license...ROFL, yeah, that was it. |
| 03:12.17 | russellb | g'night folks |
| 03:12.28 | voxter | im outta here too |
| 03:12.28 | drmessano | "How many G729 licenses does this come with?" "hey, yo, a whole-a-bunch.. " |
| 03:12.35 | voxter | enough excitement for one day |
| 03:12.47 | Carlos_PHX | drmessano: Said in the Simpsons mobster voice of course. |
| 03:13.40 | drmessano | Stop looking a gift horse in the uh, you know, a, his, a nose or sometin.. hey yo.. |
| 03:14.26 | drmessano | I think "Unlimited license" was the topper for the night. I can do no better |
| 03:14.47 | Carlos_PHX | We never did use them, Kevin wanted no part of it. We were going 70 concurrents, woulda been a lot of licenses. |
| 03:14.55 | Carlos_PHX | drmessano: Yeah |
| 03:14.58 | Carlos_PHX | No kidding. |
| 03:15.35 | drmessano | BRB, need to go ask for help with my CS4 keygen on the Adobe forums |
| 03:16.29 | Carlos_PHX | I'm guessing most people come here expecting script kiddies and hax0rs. |
| 03:17.23 | *** join/#asterisk chendo[work] (n=chendo@employ19.lnk.telstra.net) |
| 03:18.00 | chendo[work] | It should be possible to get asterisk to change the incoming caller ID based of information from a CRM, yeah? |
| 03:20.32 | Qwell | chendo[work]: sure |
| 03:20.44 | chendo[work] | So nothing will break if it might take a while for the data to come back? |
| 03:21.03 | chendo[work] | Like will it delay the phone ringing if the request takes a while? |
| 03:21.23 | chendo[work] | I actually know almost nothing about asterisk because the other guy set it up, but I want to see if it's possible before bringing it up and sounding stupid |
| 03:28.14 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 03:28.24 | *** join/#asterisk oilinki (n=oil@ppp-124-120-1-200.revip2.asianet.co.th) |
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| 03:32.33 | *** join/#asterisk ManxPower (n=manxpowe@c-76-105-105-153.hsd1.ga.comcast.net) |
| 03:33.12 | *** join/#asterisk newmember (n=chatzill@S010600036d1139bb.cg.shawcable.net) |
| 03:34.59 | *** join/#asterisk jameswf-home (n=james@68.2.99.240) |
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| 03:41.24 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:41.24 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 03:42.46 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
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| 03:44.30 | *** part/#asterisk Transience39 (n=Leo@resnet161-21.resnet.buffalo.edu) |
| 03:46.51 | ReDNeQ | evening |
| 03:52.33 | jblack | Wallstreet got bailed out, and all I got was my shirt taken off my back. |
| 03:53.53 | jameswf-home | jblack: not yet those A-Holes in the house have to vote |
| 03:57.46 | *** join/#asterisk fogo (n=Paul@rs-69-169-132-35-0003.broadweave.net) |
| 03:58.31 | *** part/#asterisk Deeewayne (i=dwayne@76.29.245.9) |
| 04:00.22 | jameswf-home | I dunno if this is cool or ceepy http://www.impinj.com/WorkArea/showcontent.aspx?id=2517 |
| 04:00.23 | jblack | I promised all three of my representatives that I'll do everything I can to make sure nobody forgets. I already checked adrates, and I can do weekly advertisements for a year for about 1200 bucks. |
| 04:00.50 | jameswf-home | you could use asterisk to cold call |
| 04:01.02 | jameswf-home | play a quick recoding |
| 04:01.17 | jblack | There is an exception for political calls, no? |
| 04:01.57 | jameswf-home | i get political ecordings often... you think that politicians would make something illegal for them |
| 04:02.04 | jblack | That is not creepy. It's just for a race. |
| 04:02.43 | implicit | some people are aparnoid |
| 04:02.43 | jblack | actually, 1.25, that's 96K calls. Seems the paper would be cheaper. |
| 04:03.06 | jblack | creepy is when they try to stick 'em on your kids, or your boss on you. |
| 04:03.34 | jameswf-home | yeah we did not show the boss the blackbery gps tracking software |
| 04:05.42 | jblack | gee. if the bailout is such a great idea, why is futures down 91 points? |
| 04:06.24 | jameswf-home | if you look in to the future you will see the house vote NO |
| 04:06.48 | jblack | I hope it does. |
| 04:07.44 | jameswf-home | so are thee vegas odds on the next bank to fail |
| 04:08.12 | jameswf-home | are there any small large banks left? |
| 04:08.27 | jameswf-home | $30 on US Bank |
| 04:09.36 | jblack | I heard something about national city? |
| 04:10.26 | jblack | Looks like Downey savings and first federal are about to push daisies too. |
| 04:11.16 | jblack | What we should keep an eye on is Bank of America and Citigroup. Those two are massive in size. |
| 04:11.52 | jameswf-home | NCC (Common) NYSE$2.89 + 1.1465.14%$3.14$1.92588,364,055$2.13 << that could be corect this is what wamu looked like 3 days before they went |
| 04:13.13 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 04:14.44 | jblack | Yeah. Go bailout. |
| 04:15.09 | jameswf-home | get it straight "RES-Q-PLAN |
| 04:15.13 | jameswf-home | gah |
| 04:15.35 | jblack | There's 50 ways to solve it. How does congress manage to pick the one that's most expensive and most likely to work? |
| 04:15.35 | jameswf-home | bail out is so last week |
| 04:15.54 | jblack | pardon, least likely to work |
| 04:17.13 | jameswf-home | they need to release this http://www.wireless.att.com/businesscenter/blackberry9000 |
| 04:17.56 | jblack | What I want are a handful of those pandoras. http://www.dcemu.co.uk/pandora-the-preorders-are-open-today-160374.html |
| 04:18.38 | jameswf-home | does it run linux? |
| 04:18.51 | jblack | It does everything, and it's open. |
| 04:19.00 | jameswf-home | im not a gamer |
| 04:19.20 | jblack | don't think of it as a DS. Think of it as a micro-laptop with 12 hour battery life and linux. |
| 04:20.16 | jameswf-home | mobile PBX... |
| 04:20.25 | jameswf-home | chan_mobile |
| 04:21.13 | *** join/#asterisk fogo (n=Paul@rs-69-169-132-35-0003.broadweave.net) |
| 04:31.45 | jblack | Down 115 now. I wonder how the dollar is doing |
| 04:32.30 | jblack | Up. |
| 04:34.20 | *** join/#asterisk oilinki (n=oil@ppp-124-120-1-200.revip2.asianet.co.th) |
| 04:37.44 | hardwire | hi |
| 04:46.16 | *** join/#asterisk saint_ (n=saint@cpe-75-83-219-24.socal.res.rr.com) |
| 04:46.28 | saint_ | Hi all...! |
| 04:47.04 | saint_ | Could someone help me troubleshot a SIP trace please ? I have it at http://imagebin.ca/view/Lgz7C1.html ... I'd like to know WTF is this P-Alcatel-CSBU in the INVITE ..? |
| 04:47.34 | saint_ | I can't find anything about "bypass=xx; fb=notransfer; ..." in any RFC |
| 04:49.23 | drmessano | A Screencap of wireshark? |
| 04:49.26 | drmessano | Geez |
| 04:49.51 | saint_ | well.. yeah... |
| 04:50.17 | drmessano | sip debug pastebin'ed would be much more useful |
| 04:50.42 | saint_ | all I have is a .CAP here ... |
| 04:51.19 | saint_ | Is there any RFC that allows to send proprietary informations, like the one in this header ..? |
| 04:51.22 | drmessano | You cant generate debug? |
| 04:52.33 | saint_ | well.. I could, but i m not at the office anymore ... so for now, that's all i have ... |
| 04:54.02 | drmessano | ok |
| 04:54.30 | hardwire | don't you think it's a bit fantastic that we have so many users in here but nothing going on? |
| 04:55.09 | drmessano | Sorry, I don't understand. Can you make a screencap for me and imagebin it? |
| 04:55.34 | hardwire | no |
| 04:56.38 | hardwire | can you screencap your question and tinypic it for me? |
| 04:57.41 | drmessano | Oh, you mothe... |
| 04:57.42 | drmessano | I mean |
| 04:57.45 | drmessano | yeah, brb |
| 04:57.49 | jblack | hardwire: Nah, that happens all the time. |
| 04:58.17 | jblack | sometimes 3-4 hours at a time with _nothing_ |
| 04:59.35 | drmessano | http://i16.tinypic.com/6t1excn.jpg |
| 05:00.12 | hardwire | drmessano: .. I haven't clicked yet |
| 05:00.22 | hardwire | before I do.. do you swear on your life that I won't regret it? |
| 05:00.29 | hardwire | ahahaha |
| 05:00.51 | drmessano | No, but they did |
| 05:01.44 | [TK]D-Fender | bed time, checking out. Later all |
| 05:02.08 | drmessano | Goatse, let me show you my |
| 05:02.59 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 05:04.54 | *** join/#asterisk Levonk (n=lk@adsl-75-62-136-107.dsl.lsan03.sbcglobal.net) |
| 05:23.43 | *** join/#asterisk [gnubie] (n=[gnubie]@cm141.omega112.maxonline.com.sg) |
| 05:33.26 | jameswf-home | omfg |
| 05:34.40 | jameswf-home | do NOT google goatse |
| 05:36.45 | [gnubie] | waves to all.. |
| 05:36.48 | implicit | y0y0 |
| 05:37.58 | [gnubie] | i am running asterisk-1.4.21.2 with a digium tdm card |
| 05:38.25 | [gnubie] | i am experiencing a one way audio when placing an outbound call... |
| 05:38.45 | mosty | outbound on what? |
| 05:39.15 | [gnubie] | caller is using sip phone located inside the lan and the callee is an analog telephone connected to a pots |
| 05:40.03 | [gnubie] | sip_phone ==lan==> asterisk ==pots_fxo==> analog_telephone |
| 05:41.03 | [gnubie] | that's the scenario |
| 05:41.17 | mosty | the problem is most likely between asterisk and the sip phone |
| 05:41.21 | [gnubie] | the callee cannot hear the caller but the caller hears the callee |
| 05:41.58 | [gnubie] | but when i captured the communication using tcpdump and decode/play the session using wireshark, i can hear them both |
| 05:42.38 | mosty | do you have NAT involved anywhere? |
| 05:42.38 | [gnubie] | mosty: i don't have any problem with an inbound call as well as extension to extension sip calls |
| 05:42.53 | [gnubie] | mosty: nope. |
| 05:43.26 | jameswf-home | whats the CLI output |
| 05:43.54 | [gnubie] | let me check the /var/log/asterisk/messages first |
| 05:47.17 | [gnubie] | if i understand this correctly based on the sip messages, it says that: |
| 05:48.32 | [gnubie] | sip phone sent an INVITE to the callee's telephone number |
| 05:50.02 | [gnubie] | then, 407 proxy authentication required from the sip phone to the callee's telephone number |
| 05:51.10 | [gnubie] | then, the callee acknowledged it with ACK |
| 05:51.40 | [gnubie] | then, sent an INVITE again to the callee's number |
| 05:52.08 | [gnubie] | 100 Trying |
| 05:53.00 | [gnubie] | then CANCEL sip:callee's_telephone_number |
| 05:53.18 | [gnubie] | SIP/2.0 487 Request Terminated |
| 05:53.32 | [gnubie] | SIP/2.0 200 OK |
| 05:53.54 | [gnubie] | ACK sip:callee's_telephone_number |
| 05:54.34 | [gnubie] | INVITE sip:callee's_telephone_number |
| 05:55.04 | [gnubie] | SIP/2.0 407 Proxy Authentication Required |
| 05:55.23 | [gnubie] | ACK sip:callee's_telephone_number |
| 05:55.35 | [gnubie] | INVITE sip:callee's_telephone_number |
| 05:56.59 | [gnubie] | SIP/2.0 100 Trying |
| 05:57.12 | [gnubie] | SIP/2.0 183 Session Progress |
| 05:59.22 | [gnubie] | CANCEL sip:callee's_telephone_number |
| 05:59.38 | [gnubie] | SIP/2.0 487 Request Terminated |
| 05:59.50 | [gnubie] | SIP/2.0 200 OK |
| 05:59.55 | Strom_C | hint hint |
| 05:59.58 | Strom_C | ~pb |
| 05:59.59 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 06:00.38 | [gnubie] | ACK sip:callee's_telephone_number |
| 06:00.44 | [gnubie] | Steve_J-obs: sorry, i should use it |
| 06:00.59 | [gnubie] | Strom_C: sorry, i should use it.. |
| 06:01.41 | [gnubie] | but i think it was done |
| 06:03.02 | [gnubie] | jameswf-home: if you want, i will paste the entire /var/log/asterisk/messages at the pastebin |
| 06:05.11 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 06:05.56 | *** join/#asterisk steliosk (n=Stelios@91.140.125.25) |
| 06:14.45 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:15.08 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:22.29 | *** join/#asterisk [netman] (n=netman@171.Red-83-45-38.dynamicIP.rima-tde.net) |
| 06:26.39 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
| 06:29.53 | [gnubie] | jameswf-home: kindly check this out => http://paste.debian.net/18372/ |
| 06:30.06 | [gnubie] | mosty: kindly check this out => http://paste.debian.net/18372/ |
| 06:32.47 | mosty | does a packet trace on your asterisk box to the sip phone show rtp traffic going in both directions? |
| 06:36.58 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:41.36 | [gnubie] | mosty: it looks like, yes |
| 06:42.01 | mosty | then i don't know why your phone isn't giving you the sound. what kind of phone is it? |
| 06:42.37 | [gnubie] | mosty: snom 300 |
| 06:43.10 | mosty | do a pcap trace on the phone. you can do that through the phone's web interface |
| 06:43.11 | [gnubie] | mosty: i don't have any problem with an inbound call with the same route.. |
| 06:43.18 | mosty | confirm that the rtp is arriving at the phone |
| 06:43.51 | [gnubie] | mosty: it's the callee (analog telephone) that cannot hear the snom 300 |
| 06:44.30 | mosty | ok, then a packet trace at the asterisk end should suffice |
| 06:45.21 | [gnubie] | mosty: that's what i have here.. i got a capture file from a tcpdump command |
| 06:46.26 | [gnubie] | mosty: that's how i learned that both of them sends their voice because when i decode/play it from a wireshark, i can hear both of them |
| 06:49.26 | [gnubie] | i was just wondering why the callee cannot hear the voice from the caller.. |
| 06:50.11 | mosty | usually when i see problems like this, it's because there is NAT in there somewhere, and the rtp doesn't go to the correct place |
| 06:52.50 | [gnubie] | mosty: the snom 300 is at 192.168.1.106 and the asterisk box is at 192.168.1.21.. since the callee is at the pots side (analog telephone) and the one that cannot hear the caller's voice, does it mean that the rtp from the caller's side must go to the 192.168.1.21, the asterisk's ip address? |
| 06:54.19 | mosty | yes |
| 06:54.37 | mosty | since the asterisk machine is the only thing connected to the analogue phone |
| 07:02.14 | [gnubie] | mosty: what should i do with this? it seems that the rtp goes to the asterisk box |
| 07:05.07 | mosty | what version of asterisk? what kind of tdm card? |
| 07:06.56 | *** join/#asterisk ayrjola (n=ayrjola@support.ccxtech.fi) |
| 07:07.10 | tzafrir_laptop | [gnubie], can you get a dump of the SIP/rtp session with wireshark / tcpdump? |
| 07:07.30 | [gnubie] | tzafrir_laptop: i already have one |
| 07:07.44 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 07:07.51 | Kernel_Core | hi all |
| 07:07.53 | tzafrir_laptop | that's on the phone |
| 07:07.56 | tzafrir_laptop | hi |
| 07:08.07 | [gnubie] | tzafrir_laptop: that's where i was able to hear both the caller and the callee's voice but the callee cannot hear the caller's voice |
| 07:08.35 | [gnubie] | tzafrir_laptop: kindly take a look at this => http://paste.debian.net/18372/ |
| 07:09.04 | [gnubie] | tzafrir_laptop: i created it using wireshark, from a tcpdump capture file |
| 07:09.56 | [gnubie] | 192.168.1.106 is the ip phone and the 192.168.1.21 is the asterisk box |
| 07:10.47 | tzafrir_laptop | [gnubie], that's a dump on the phone, right? can you get a dump on the asterisk server? |
| 07:10.55 | tzafrir_laptop | you can use, e.g. tcpdump |
| 07:11.12 | [gnubie] | tzafrir_laptop: that's on the asterisk server while having the session |
| 07:11.41 | ayrjola | help with oneway audio problem 1.4.21.2 sip trunk? most calls have oneway audio, no pattern when or which way. |
| 07:11.43 | tzafrir_laptop | e.g.: tcpdump -w |
| 07:12.32 | tzafrir_laptop | ok. So there are hardly any RTP packets, right? |
| 07:15.09 | ayrjola | sorry, bad discription. oneway audio problem only if incoming Call in dialplan goes to Dial() and destination is out from sip trunk where call came in. |
| 07:15.21 | [gnubie] | tzafrir_laptop: i open the tcpdump capture file on wireshark and i was able to hear the voices of the caller and the callee |
| 07:15.45 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 07:16.12 | tzafrir_laptop | [gnubie], which capture file? |
| 07:17.13 | [gnubie] | tzafrir_laptop: before i started calling the analog telephone from the pots via the tdm fxo port of the asterisk box, i executed this command: tcpdump -s 0 -i eth0 -w /tmp/one_way_audio.cap |
| 07:18.15 | [gnubie] | tzafrir_laptop: then i performed the call until such time the callee hangup the call because he cannot hear anything from me.. but i can hear him |
| 07:19.00 | [gnubie] | tzafrir_laptop: after that, i open the one_way_audio.cap on my wireshark and play the voip_calls and from there i was able to hear both the caller's and the callee's voices |
| 07:21.47 | [gnubie] | tzafrir_laptop: now, since i was able to hear both voices, how come that the callee cannot hear the caller's voice? |
| 07:22.26 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:22.32 | *** join/#asterisk InsolentDreams (n=Insolent@cpe-76-175-181-103.socal.res.rr.com) |
| 07:23.06 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 07:23.51 | [gnubie] | but with an inbound call, both can hear each other |
| 07:23.55 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 07:28.10 | *** join/#asterisk blogbasti (n=blogbast@calypso.planet-ic.de) |
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| 07:37.03 | write_erase | Hi.... I'd like to open my asterisk to the internet, so employees can use their softphones from home like there were in the company. Are there known secutity issues or best practices for this ? |
| 07:38.36 | mosty | avoid NAT issues by using a public IP |
| 07:38.55 | mosty | and firewall everything besides SIP/RTP traffic |
| 07:39.06 | [gnubie] | brb |
| 07:40.20 | write_erase | What about sip proxies that do the 'NAT' job ? |
| 07:40.44 | Maliuta | write_erase: it's not just about SIP |
| 07:41.12 | Maliuta | write_erase: I have yet to meet one that deals with the following rtp packets |
| 07:41.15 | mosty | write_erase, they can work, but it's simpler to just get a public ip address |
| 07:42.20 | write_erase | I understand |
| 07:42.37 | Rico29 | hi |
| 07:43.03 | write_erase | Salut Rico29 |
| 07:44.16 | Rico29 | ah, un fr |
| 07:45.53 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-1-200.revip2.asianet.co.th) |
| 07:52.10 | Rico29 | i need help for using the "Read" cmd in a perl AGI |
| 07:52.21 | Rico29 | 've tried many ways to use it |
| 07:52.28 | Rico29 | but it doesn't work |
| 07:52.32 | Rico29 | :( |
| 07:54.17 | *** join/#asterisk Juggie (n=Juggie@99.224.81.113) |
| 07:54.44 | tzafrir_laptop | Rico29, "doesn't work" as in? What does it actually do? |
| 07:55.03 | Rico29 | i'm pastebining it |
| 07:56.38 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:01.28 | Rico29 | tzafrir_laptop > http://debian.pastebin.com/m620e31d3 |
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| 08:06.25 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:10.42 | Rico29 | tzafrir_laptop > any idea ? |
| 08:17.44 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:32.45 | Rico29 | http://www.asterisk-france.net/community/showthread.php?p=31014 if somebody wants to take a look |
| 08:35.26 | *** join/#asterisk kissand (n=kissand@asterix.ucnet.uoc.gr) |
| 08:36.52 | kissand | hello |
| 08:38.22 | kissand | anyone from Digium INC? |
| 08:40.24 | *** join/#asterisk oilinki3 (n=oil@ppp-124-121-250-170.revip2.asianet.co.th) |
| 08:44.20 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
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| 08:56.07 | kissand | anyone from Digium INC? |
| 08:56.21 | Rico29 | http://www.asterisk-france.net/community/showthread.php?p=31014 if somebody wants to take a look |
| 08:56.59 | kissand | its in french |
| 08:57.30 | Rico29 | yes but easily understandable |
| 08:58.16 | kissand | not for m |
| 09:00.49 | tzafrir_laptop | kissand, They'll probably be here (and on the phone support line) in a few hours) |
| 09:01.08 | kissand | hmm right i forgot the time difference |
| 09:02.15 | kissand | will they be interested on helping me develop a voip network for greek schools based on asterisk? |
| 09:02.24 | kissand | for free of course :> |
| 09:03.19 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 09:10.46 | Rico29 | kissand > http://www.asterisk-france.net/community/showthread.php?t=6363 |
| 09:10.51 | Rico29 | with english translation |
| 09:11.31 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-0-46.revip2.asianet.co.th) |
| 09:13.30 | *** join/#asterisk baxtas (n=buzukeli@ip-195-14-171-9.bnk.lt) |
| 09:14.08 | baxtas | what will hapen if asterisk and ATA have diferent jitter parameters |
| 09:15.27 | baxtas | ? |
| 09:15.53 | baxtas | Pagautas kaip tu manai?? |
| 09:31.19 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 09:34.28 | mort_gib | kissand: what do you need |
| 09:34.59 | mort_gib | kissand: I'm NOT Digium, but would be glad to help... |
| 09:37.47 | tzafrir_laptop | ~docs |
| 09:37.48 | jbot | i guess docs is for basic documentation of Asterisk ask see http://voip-info.org/ (~voip-info) and TheBook (~book) |
| 09:39.22 | mvanbaak | TheBook is the best reference I think |
| 09:47.21 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 09:50.22 | InsolentDreams | ~book |
| 09:50.23 | jbot | from memory, book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 09:55.24 | InsolentDreams | I'm having problems debugging a iax connection, I have dundi setup between two servers, and they advertise and lookup properly, but calls seem to only go one way, and iax debugging in the asterisk terminal appears less than helpful. Is there a command I'm missing to debug iax connections? |
| 09:58.25 | *** join/#asterisk martijn_ (n=martijn@i181100.upc-i.chello.nl) |
| 10:01.54 | *** join/#asterisk stewbaby (n=stewart@ip-217-204-65-78.easynet.co.uk) |
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| 10:30.05 | *** join/#asterisk pputman (n=centrex@nat/digium/x-650c42fc0d4d5b7a) |
| 10:30.43 | pputman | does anyone know how to resolve a problem with asterisk no playing audio files with zaptel started? |
| 10:30.47 | pputman | s/no/not |
| 10:34.47 | *** join/#asterisk yang (i=yang@CAcert/Assurer/yang) |
| 10:41.23 | tzafrir_laptop | pputman, what device? What version of zaptel? |
| 10:42.35 | *** join/#asterisk mdalby (n=Mark@blinet-gw.lmh.zimo.co.uk) |
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| 10:44.16 | mdalby | Morning, Can anybody tell me how i'd go about checking the status of a single E1, I have 4 e'1's in a card but im concerned no calls are going through one of them |
| 10:44.28 | *** join/#asterisk krokodilerian (n=vasil@gw-sf.securax.net) |
| 10:46.39 | pputman | tzafrir_laptop, latest 1.4, and it's a te110p. I have heard in the past a problem where zaptel being loaded causes files to play with no audio I just can't remember what the solution was. |
| 10:46.50 | pputman | I'm thinking it might have something to do with timing? |
| 10:50.03 | pputman | actually yeah, his zaptel.conf is blank so I'm guessing once it's configured the sound files will start playing |
| 10:50.35 | tzafrir_laptop | pputman, try running zttest . does it give output (of close to 100%)? |
| 10:51.26 | pputman | tzafrir_laptop, I'll try that, I don't have full access to the system |
| 10:52.38 | pputman | but now I do remember jcolp telling me one time that an unconfigured card can cause this problem |
| 10:53.14 | eric2 | why is it that with * 1.4.21, rxfax causes it to crash? |
| 10:53.24 | eric2 | everything was fine with 1.4.11 |
| 10:54.31 | tzafrir_laptop | pputman, all the same. zttest can show that there is (no) functioning timing source. The lack of functining timing source is that cause you were talking about |
| 10:54.58 | *** join/#asterisk [gnubie] (n=[gnubie]@cm141.omega112.maxonline.com.sg) |
| 10:55.00 | pputman | tzafrir_laptop, okay, that's a place to start, thanks. |
| 10:55.06 | tzafrir_laptop | eric2, have you upgraded anything else? e.g.: libtiff? |
| 10:55.35 | eric2 | what does lib tiff have to do with it? I upgraded asterisk-plugings-fax as well |
| 10:55.39 | eric2 | hmm |
| 10:55.43 | eric2 | maybe you're right |
| 10:55.47 | eric2 | I'll check the version |
| 10:56.41 | eric2 | lib64tiff3 version 3.8.2 |
| 10:57.18 | eric2 | asterisk 1.4.21.2 and asterisk-plugins-fax 1.4.21.2 |
| 10:57.49 | eric2 | everything works great, but when the sending fax machine hangs up, asterisk crash's but I do get the tiff image on the server nicely |
| 10:59.34 | eric2 | this is what I have for my fax code: http://pastebin.ca/1216725 |
| 11:00.07 | eric2 | the noop(4444444) never gets executed as asterisk craps out on the rxfax line |
| 11:00.24 | tzafrir_laptop | eric2, what distro is it? |
| 11:00.35 | eric2 | mandriva |
| 11:00.40 | eric2 | and I"m using the rpm's |
| 11:00.55 | tzafrir_laptop | are you sure spandsp hasn't changed? |
| 11:01.29 | eric2 | with 1.4.21.2, it wants spandsp 0.05 |
| 11:02.03 | tzafrir_laptop | and what spandsp do you have? |
| 11:02.43 | eric2 | with 1.4.21.2 I have spandsp 0.05 but on my box with 1.4.1.11 it's using 0.00.4 |
| 11:03.20 | tzafrir_laptop | I guess you should upgrade spandsp as well |
| 11:03.46 | eric2 | hmm, spandsp 0.05 might be the culprit |
| 11:04.06 | tzafrir_laptop | I'm surprised urpmi/rpm let you upgrade this |
| 11:04.39 | tzafrir_laptop | senses the use of brute --force |
| 11:04.51 | *** join/#asterisk The-Bat (n=The-Bat@203.199.114.33) |
| 11:05.06 | eric2 | 0.0.4 might have to be the task or brute forcing |
| 11:05.17 | eric2 | *of |
| 11:08.07 | Blackvel | anyone of you used myvariable:0:LEN(myvariable)-2 yet? Would I have to set LEN of myvariable-2 before or can I use it with the substitution like myvariable:0:${LEN(myvariable)}-2? |
| 11:08.33 | Blackvel | how does that have to work exactly? |
| 11:09.39 | kaldemar | $[${LEN(myvariable)}-2] |
| 11:10.58 | kaldemar | always use $[] around arithmetic operations. |
| 11:12.19 | kaldemar | oh, sorry. LEN takes a string as an argument, not a variable name. so it would be $[${LEN(${myvariable})}-2] |
| 11:14.12 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 11:16.03 | Blackvel | something like that: |
| 11:16.08 | *** join/#asterisk CVirus (n=cvirus@41.196.142.40) |
| 11:16.11 | Blackvel | exten => s,102,Set(var_lencalleridnum=${LEN(${CALLERID(num)})}) |
| 11:17.26 | *** part/#asterisk CVirus (n=cvirus@41.196.142.40) |
| 11:18.21 | Blackvel | exten => s,103,LDAPget(CALLERID(name)=ldapconfig_cidmapping/${CALLERID(num):0:$[${var_lencalleridnum}-2]}) |
| 11:20.04 | kaldemar | looks fine with my tired eyes. but you can of course do that with just one line. decreases the readability of the extension though. |
| 11:21.57 | Blackvel | I think I am going to use even a 3rd ldapget call. so should be fine to compute LEN only one time |
| 11:22.06 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 11:22.18 | *** join/#asterisk propellerhead (n=yogurt2u@host113.190-136-116.telecom.net.ar) |
| 11:22.35 | Blackvel | what was the syntax? variablename:0:5? so its ${variablename:0:5} or was it ${variablename}:0:5? |
| 11:23.54 | Blackvel | where would I put the "0" if I have to? e.g number xxx-12 -> callerid(num)0:3 + 0? xxx-0 |
| 11:24.50 | Blackvel | exten => s,103,LDAPget(CALLERID(name)=ldapconfig_cidmapping/${CALLERID(num):0:$[${var_lencalleridnum}-2]}0) |
| 11:24.51 | Blackvel | ? |
| 11:26.51 | kaldemar | it's always ${variablename:0:5}. ${variablename}:0:5 would reference to "123:0:5" if the variable had 123 in it. |
| 11:27.24 | Blackvel | so, that would not make much sense :) |
| 11:27.26 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 11:27.36 | hi365 | if I call a macro from dial - should it have a problem passing dmtf? |
| 11:27.58 | Blackvel | ${CALLERID(num):0:3}0 |
| 11:28.04 | kaldemar | actually ${CALLERID(num):0:$[${var_lencalleridnum}-2]}0 makes very much sence if you want to suffix n first digits with a 0. |
| 11:28.07 | Blackvel | that would refer to 1230 , wouldn't it? |
| 11:28.13 | kaldemar | yes |
| 11:28.45 | Blackvel | thats cool |
| 11:28.47 | kaldemar | you don't need anything special for concatenation, just put the digits after the variable reference. |
| 11:29.09 | Blackvel | i am stil a bit unclear how to do it with the ldap thing from the technical side |
| 11:29.23 | Blackvel | i just have all these >1000 company names in my db |
| 11:29.36 | Blackvel | but usually with the direct extensions and not the -0 |
| 11:29.53 | Blackvel | when a new call comes in, it would be the same company, but not a well known person |
| 11:30.26 | defswork | I need some cheesey voice prompts that I can dynamically play to tell the time - but in proper voiceover man style "it's nearly 5 to 4" etc.. anyone know of anything ? |
| 11:30.28 | kaldemar | these variable handlings are very easy to test by trying it out in a test dialplan. i suggest playing around with these. |
| 11:30.52 | hi365 | im calling a macro from dial that includes a playback/background. while the file is being played, if I press a digit, the macro "crashes" and the dmtf doesnt get sent |
| 11:31.02 | Blackvel | is it best to use LDAPGet to match against len-2 + "0" and add extra rows in ldap for company name with "0" exten or adding 2nd "0" business number to each of the contacts? |
| 11:31.46 | kaldemar | uh. that's for you to decide. :) |
| 11:32.26 | Blackvel | I guess its not possible to find ldap phone -len 2 ....so exluding ldap business phone match on the last 2-3 digits |
| 11:32.37 | Blackvel | how do you guys do it in your business? |
| 11:33.16 | Blackvel | i am just alone (it contractor). so I can decide whats the best from the architecture spoken (and maybe least efforts) |
| 11:38.23 | hi365 | hmm, seems like this is a limitation of macro in dial. so let me ask this another way: |
| 11:38.39 | hi365 | how can i play more than one audio file to a callee? |
| 11:38.49 | hi365 | (and then have the call bridged) |
| 11:41.10 | hi365 | (i.e. use the A option with more than one file) |
| 11:44.18 | *** join/#asterisk Nathariel (n=boyan__@91.92.154.116) |
| 11:49.06 | gewuerzwiesel | hi all, is it possible to set different codecs for outgoing and internal calls? |
| 11:49.26 | gewuerzwiesel | using gsm: outging calls with bad quality, calls to voicemail with good quality |
| 11:49.43 | gewuerzwiesel | using alaw: outgoing calls with good quality, calls to ovicemail with bad quality |
| 11:53.15 | *** join/#asterisk n3hxs (n=HAMming@75.145.100.230) |
| 11:54.27 | *** join/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 12:15.09 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:15.11 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
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| 12:16.08 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:18.11 | angryuser | gewuerzwiesel : hm, maybe set the ${SIP_CODEC} before call |
| 12:19.30 | angryuser | : gewuerzwiesel net sure if it is still used, do a noop on the call |
| 12:20.35 | gewuerzwiesel | i'll try it, thx |
| 12:22.06 | mort_gib | gewuerzwiesel: You could set the codec on the channel, so internal channels defaults to alaw, and your voip provider only does G729 and so forth... |
| 12:22.28 | *** join/#asterisk Segnale007 (n=Segnale0@host251-13-dynamic.12-79-r.retail.telecomitalia.it) |
| 12:24.19 | angryuser | mort_gib : maybe his voicemail is external ? |
| 12:25.07 | angryuser | gewuerzwiesel : you are using external or * voicemail ? |
| 12:26.12 | gewuerzwiesel | angryuser: * |
| 12:28.00 | angryuser | gewuerzwiesel : so, set alaw to your outgoing peers as you only one codec, and gsm for you phones as only one,(or first on the list) and let * do the transcoding |
| 12:28.08 | angryuser | your* |
| 12:29.45 | gewuerzwiesel | ok, thx :) |
| 12:30.09 | mort_gib | angryuser: Which is what I meant to say :-) |
| 12:30.33 | angryuser | but gsm, is not the best choice anyway, and i dont understand why voicemail with alaw has voice problems |
| 12:30.35 | gewuerzwiesel | :) |
| 12:30.41 | angryuser | mort_gib : yes |
| 12:31.17 | gewuerzwiesel | angryuser: any sound from asterisk sound like played with 5$ speakers with alaw :) |
| 12:31.48 | coppice | $5 is quite a lot for a speaker. it should sound good |
| 12:31.50 | angryuser | gewuerzwiesel : i am using ulaw, but they sound clear and nice |
| 12:32.01 | angryuser | depends on phone too |
| 12:32.03 | mort_gib | gewuerzwiesel: Timing issue??? |
| 12:33.00 | mort_gib | I don't recognize your problem, playing with codec's is good, but you have another problem |
| 12:33.05 | gewuerzwiesel | coppice: ok, 5$ home cinema speakers :) |
| 12:33.31 | coppice | home cinema speakers generally cost less than $5 each |
| 12:34.23 | hi365 | is it not posible to send dtmf while in dial (to the caller)? |
| 12:35.41 | [TK]D-Fender | hi365: No. And all those times you navigated through IVR's on outbound calls was just a dream. |
| 12:38.31 | hi365 | [TK]D-Fender: TO the caller, not from a caller. let me explain. I have an intercom, which askes the caller for his name and palyes it back (to who ever answer the call). Now while the caller is listening to the promts (priv-callfrom&/tmp/door1) - how can they press the "door buzz" key? i tired using a mcaro to do the playback of the files, but although it "seems" to send dtmf - the door doesnt get buzzed |
| 12:38.36 | hi365 | let me show you my dialplan |
| 12:39.35 | [TK]D-Fender | hi365: That would be wise... |
| 12:39.51 | hi365 | [TK]D-Fender: its a bit messy as im trying different things.... http://pastebin.ca/1216788 |
| 12:40.10 | hi365 | the intercom dials 601 |
| 12:40.29 | *** join/#asterisk n3hxs (n=HAMming@adsl-70-128-62-214.dsl.ltrkar.swbell.net) |
| 12:41.43 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 12:42.02 | *** part/#asterisk gongoputch (n=kseel@74.95.184.161) |
| 12:42.25 | [TK]D-Fender | hi365: line 16 looks like tis calling itself. |
| 12:42.34 | [TK]D-Fender | hi365: and that is a complete mess. |
| 12:43.01 | hi365 | its not - one is in the context and one is local/601@from-internal |
| 12:43.04 | [TK]D-Fender | hi365: You should probably reword what you'd like to do and maybe we can suggest a different approach |
| 12:43.27 | hi365 | [TK]D-Fender: fixed speeling on line 18 - and now asterisk claims it works "Executing [5@door:1] SendDTMF("SIP/220-08365f08", "www5") in new stack" |
| 12:43.37 | hi365 | but the door isnt reciving the dmtf... |
| 12:43.39 | [TK]D-Fender | hi365: was hard to tell because of other spelling erros, etc... hard to trust whats "testing", vs "accident", and "just plain wrong" |
| 12:43.49 | hi365 | [TK]D-Fender: let me rephrase |
| 12:44.05 | [TK]D-Fender | hi365: from the beginning please and tell me exactly what devices are being used |
| 12:44.39 | hi365 | whne the user presses the call (pancode door pannel (sip)) button it promts him for his name. |
| 12:45.08 | hi365 | i then wnat the system to call a ring group (local/601@from@internal) and play back his name |
| 12:45.48 | hi365 | obviously, I want the panel to continue to play ring until ive finished listening to the name promt |
| 12:45.57 | hi365 | how would you go about it? |
| 12:46.33 | [TK]D-Fender | hi365: M() |
| 12:47.37 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 12:48.28 | hi365 | [TK]D-Fender: ok. seem though that the macro isnt passing the dtmf back to the door (although its getting played) |
| 12:49.10 | [TK]D-Fender | hi365: then backtrack and just have your SIP door phone call ONE phone and prove it is even set up right for the proper DTMF mode. |
| 12:49.27 | [TK]D-Fender | hi365: Because that's always the first things to suspect |
| 12:49.48 | hi365 | [TK]D-Fender: IF i wait till all the promts are finished playing and press 5 it buzzez the door jsut fine |
| 12:50.14 | hi365 | currently, if i press 5 it terminates the palyback - and then I need to press 5 again to get the door to buzz |
| 12:50.27 | [TK]D-Fender | hi365: Oh so your only issue is that you want to be able to do it WHILE the name is playing as well and not wait for the bridge? |
| 12:50.35 | hi365 | yup |
| 12:50.48 | hi365 | er... |
| 12:51.01 | [TK]D-Fender | hi365: I don't see how that is going to be possible. |
| 12:51.14 | [TK]D-Fender | hi365: only time DTMF will make it over is upon bridging |
| 12:51.18 | hi365 | now that you worded it like that - I see the problem |
| 12:51.41 | [TK]D-Fender | hi365: You can't keep him "out of the loop" and then expect to pass him DTMF.... |
| 12:51.58 | [TK]D-Fender | hi365: there is no "kinda bridged, only not" |
| 12:51.58 | *** join/#asterisk scardinal (n=supreme@90.184.100.170) |
| 12:52.17 | [TK]D-Fender | hi365: So your users will just have to live with it. |
| 12:52.26 | hi365 | right, i see that. hmm, anything that CAN be done? is there another way to keep him on hold while i listen to the playback? |
| 12:52.52 | [TK]D-Fender | hi365: Why bother recording the name anyway? If they are going to get bridged right after, whats the point? |
| 12:53.20 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:53.21 | [TK]D-Fender | hi365: Your guy at the door has a second or so to relaize what he has to say and could have been just passed on to say it himself. |
| 12:54.03 | *** join/#asterisk shriven (n=shriven@rdu.crosscomm.net) |
| 12:54.04 | hi365 | it seems more sophisticated , and it enables the user to open the food withou the akward "who is it?" |
| 12:54.14 | hi365 | s/food/door |
| 12:54.30 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 12:54.45 | shriven | hello. Has anyone here had success getting 1.6-rc6 to compile cleanly on debian? |
| 12:56.06 | [TK]D-Fender | hi365: From what I saw the user has to hear the name (3s max) and respond WHILE its playing otherwise this "advantage" is lost. Sounds like a total waste to me... |
| 12:56.47 | hi365 | [TK]D-Fender: im not saying i would do this at home. but this is what the client wants... |
| 12:57.15 | hi365 | [TK]D-Fender: heck your right. i give up! |
| 12:57.20 | [TK]D-Fender | hi365: So far I don't see any way to pass your DTMF without a bridge which means they'll hear the callee.. |
| 13:01.17 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:01.36 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:05.32 | stimpie | is it possible to get the ip address a sip session is coming from in the dialplan? |
| 13:07.49 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:08.55 | mort_gib | stimpie: SIPPEER(<SIPNAME>|ip) |
| 13:10.44 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:10.58 | stimpie | mort_gib, thanks thats looks to be it |
| 13:11.07 | shriven | ok, so I'm compiling 1.6 and get this error: "app_voicemail.c:169: error: âPTHREAD_RECURSIVE_MUTEX_INITIALIZER_NPâ undeclared here (not in a function)" |
| 13:11.10 | shriven | is that a big deal? |
| 13:11.32 | shriven | or would it be safe to turn on --ignore-errors and keep going |
| 13:13.36 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 13:17.52 | Katty | pamples things |
| 13:18.49 | jaytee | what is pampling? |
| 13:19.00 | eric2 | anyone having faxing isssues with 1.4.21.2?? Soon at rxfax is called in my dial plan, asterisk gets killed. Any ideas? I'm using spandsp 0.0.5 |
| 13:19.46 | tzafrir_laptop | eric2, this is not an issue with faxing |
| 13:19.46 | eric2 | d'oh, I thought you had gone to sleep :) |
| 13:19.46 | Katty | jaytee: http://www.youtube.com/watch?v=YHtTFJUnr_g |
| 13:20.00 | tzafrir_laptop | it's 16:19 here |
| 13:20.06 | eric2 | ok, so I'll force spandsp 0.0.4 |
| 13:20.41 | tzafrir_laptop | luke^Weric2, don't use the --force |
| 13:21.05 | eric2 | I'll see what I can do, but forcing is bad imo |
| 13:21.31 | tzafrir_laptop | for starters, are those two packages from the same repository? |
| 13:22.41 | eric2 | I think so, but I'll check to verify |
| 13:23.04 | eric2 | checks repository libs |
| 13:23.43 | Katty | jaytee: understand now? (= |
| 13:24.27 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 13:29.57 | coppice | eric2: probably things crash because you have built with one version of spandsp, but are trying to run with another |
| 13:30.56 | eric2 | I just used the rpms for my distro, I have a hunch that I should not use the latest spandsp but the prior version but still have to test this |
| 13:32.21 | coppice | most of the distros have ancient versions of spandsp |
| 13:35.07 | eric2 | I'll check the latest, 1.2.21.2 is listed.. but anyhow, I'll look for the latest and greatest version of spandsp |
| 13:35.28 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 13:35.44 | *** join/#asterisk r3zon8 (n=r3zon8@97.66.119.194) |
| 13:36.00 | coppice | the thing that causes most people trouble is having two versions on their machine in different directories |
| 13:37.38 | eric2 | the latest spandsp and asterisk is available for my distro, but I think I'll revert to an older version of both (even though its probably not recommended) |
| 13:37.45 | eric2 | I'll test with the older stuff anyhow |
| 13:39.11 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 13:39.25 | coppice | if you get crashes it will almost certainly be because you have multiple versions on your machine. randomly changing versions doesn't help with that |
| 13:39.46 | eric2 | I'll delete all and restart from scratch |
| 13:40.00 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 13:40.09 | Qwell | JerJer: get my email? |
| 13:41.41 | coppice | Qwell: do you know if the deal with Cisco is that they are dumping SIP or dumping SCCP? |
| 13:41.50 | Qwell | got me |
| 13:44.29 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:44.29 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:44.54 | Katty | lmadsen: ohai |
| 13:44.58 | Katty | hugs lmadsen |
| 13:45.12 | creativx | O HAISZ |
| 13:45.16 | lmadsen | Katty: ohai2u2! |
| 13:45.22 | lmadsen | I haz hugz?! |
| 13:45.24 | lmadsen | I do! |
| 13:45.27 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 13:45.32 | lmadsen | hugs katty |
| 13:45.42 | lmadsen | I'm much too cheerful for having just woke up |
| 13:45.57 | Katty | :> |
| 13:46.03 | lmadsen | today is clustering day! |
| 13:46.12 | seanbright | i thought it was thursday |
| 13:46.17 | Katty | still struggling with that wakeup bit :< |
| 13:46.20 | russellb | clustering is hawt |
| 13:46.21 | lmadsen | aka clustering day |
| 13:46.23 | Katty | prods soda |
| 13:46.29 | lmadsen | you mean pop |
| 13:46.31 | Katty | dear soda, plz to wake me up |
| 13:46.52 | Katty | russellb: yer hawt. |
| 13:46.56 | lmadsen | have a customer I'm upgrading tonight |
| 13:46.59 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:47.05 | Katty | oh? |
| 13:47.08 | Katty | hugs shriven |
| 13:47.09 | Katty | oh |
| 13:47.13 | *** join/#asterisk mog (n=mog@nat/digium/x-66323ab03169a5d8) |
| 13:47.13 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:47.14 | Katty | hugs _ShrikE |
| 13:47.16 | Katty | hugs mog |
| 13:47.19 | lmadsen | lol |
| 13:47.23 | shriven | oh hai |
| 13:47.24 | russellb | Katty: ooh ... <3 |
| 13:47.31 | Katty | shriven: sorry about the unsoilicited drive-by-hugging |
| 13:47.37 | seanbright | hmmm |
| 13:47.38 | shriven | lol |
| 13:47.41 | seanbright | tries something... |
| 13:47.43 | *** part/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 13:47.45 | _ShrikE | Hey Katty! How's the pooch doing? |
| 13:47.46 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 13:47.57 | Katty | _ShrikE: he's just peachy :> |
| 13:47.57 | seanbright | oh well. |
| 13:48.10 | Katty | _ShrikE: i let him nap in the bed until 2am last night |
| 13:50.22 | Katty | seanbright: :< |
| 13:50.30 | seanbright | eh? |
| 13:51.54 | Katty | _ShrikE: riddick likes ringtones. |
| 13:52.12 | Katty | _ShrikE: he does the lil baroo headtilt thing |
| 13:52.37 | angryuser | Got 200 OK on REGISTER that isn't a register hmmm, what is that ? |
| 13:52.51 | Katty | _ShrikE: he walks on a lead pretty well, except he often gets distracted and just sits there for a minute. he's learning his name too.... |
| 13:52.52 | russellb | that is chan_sip going "wtf?" |
| 13:53.03 | *** join/#asterisk jasonwoot (n=jasonrot@69.73.89.233) |
| 13:53.28 | seanbright | you need to install chan_pineapple |
| 13:53.43 | Katty | chan_pineapple goes with chan_vodka very well |
| 13:53.45 | seanbright | download it here -> http://127.0.0.1/chan_pineapple.c |
| 13:53.49 | Katty | especially if you hollow it out, and then freeze it |
| 13:54.05 | Katty | pineapple vodka slushie |
| 13:54.08 | angryuser | chan_vodka is nice when you know how to use it |
| 13:54.25 | Katty | vodka++ |
| 13:54.29 | Katty | bacardi > vodka :> |
| 13:54.30 | _ShrikE | Katty: I just adopted a 3 month old black lab pup. |
| 13:54.34 | krokodilerian | mm, vodka |
| 13:54.36 | Katty | _ShrikE: !!! |
| 13:54.42 | Katty | _ShrikE: post gifs! |
| 13:54.50 | seanbright | gifs? |
| 13:54.54 | seanbright | welcome to 1997 |
| 13:54.59 | Katty | seanbright: your irc pop culture is lacking. |
| 13:55.08 | seanbright | compuserve gifs, hot. |
| 13:55.11 | seanbright | Katty: apparently |
| 13:55.16 | russellb | Katty: cut him some slack, he's from windows land |
| 13:55.17 | *** join/#asterisk khronos (n=khronos@c-76-110-120-247.hsd1.fl.comcast.net) |
| 13:55.21 | Katty | oh right. |
| 13:55.23 | Katty | seanbright: i sorry. |
| 13:55.33 | _ShrikE | gets his memory stick |
| 13:55.33 | russellb | :-p |
| 13:55.41 | seanbright | russellb: there goes your christmas present, judas. |
| 13:55.45 | seanbright | heh |
| 13:55.52 | russellb | awww |
| 13:55.59 | jasonwoot | why does it say paper jam when there is no paper jam? |
| 13:56.05 | Katty | oh hey, christmas IS soon, isn't it. |
| 13:56.13 | russellb | sort of |
| 13:56.19 | Katty | jasonwoot: sometimes a copier can still think it has a paper jam until it's been reset. |
| 13:56.19 | seanbright | only a quarter year away |
| 13:56.20 | russellb | seanbright: it's all in good fun! |
| 13:56.32 | Katty | jasonwoot: regardless of whether the paper has been taken out or not |
| 13:56.34 | seanbright | russellb: i know |
| 13:56.40 | Katty | jasonwoot: i'd double check the doc feeder, and the finisher |
| 13:56.42 | seanbright | russellb: i'm not really getting you a christmas present |
| 13:56.52 | seanbright | russellb: that would be creepy. |
| 13:56.55 | russellb | indeed |
| 13:56.55 | Katty | jasonwoot: stuff can get caught around the fuser too |
| 13:57.18 | jasonwoot | wow, now who's lacking on their pop culture references? |
| 13:57.20 | Katty | what if it was a remote controlled roflcopter tho? |
| 13:57.21 | seanbright | russellb: i, however, will be expecting digium swag in my mailbox on december 24th. |
| 13:57.32 | russellb | ha, you might get a mouse pad |
| 13:57.41 | seanbright | t-shirt |
| 13:57.49 | seanbright | job offer |
| 13:57.49 | russellb | pfft, good luck |
| 13:57.50 | seanbright | heh |
| 13:57.59 | Katty | hmm. digium swag. |
| 13:58.06 | russellb | let me know when you have moved to HSV, then we'll talk ^_^ |
| 13:58.13 | seanbright | bah |
| 13:58.23 | Katty | i can be in HSV before lunch |
| 13:58.34 | jasonwoot | why does CDR report duration as 1 second when a call is transferred? |
| 13:58.44 | seanbright | i'm agnostic... i step into the bible belt and my hair would catch fire |
| 13:58.50 | seanbright | not good for business. |
| 13:58.54 | russellb | seanbright: you'd be fine. |
| 13:58.55 | Katty | i doubt that |
| 13:59.04 | Katty | everyone knows i'm a liberal athiest, no one says anything |
| 13:59.18 | Katty | my boss is a right winged bible thumper (= |
| 13:59.21 | russellb | because they're afraid that you might start spitting out fire |
| 13:59.37 | seanbright | ugh... liberal? well you had me at atheist. |
| 13:59.37 | Katty | they're afraid i won't fix their phone system for them |
| 13:59.52 | Katty | no phones FOR YOU |
| 14:01.00 | seanbright | i think i'm too old for spacecamp, otherwise i'd move down in a heartbeat. |
| 14:01.11 | seanbright | i just want to meet jinx |
| 14:01.12 | Katty | you're never too old for spacecamp. |
| 14:01.21 | Katty | i'd go. |
| 14:01.31 | seanbright | and leah thompson... and kelly preston |
| 14:01.32 | seanbright | mmmmm |
| 14:01.40 | lmadsen | it already takes me 3 hours to go see my parents... moving to HSV was not an option for me |
| 14:01.44 | jaytee | jinx and max.......friends foreverrrrrrrrrr |
| 14:01.57 | seanbright | jaytee: for.ev.errrrrrrrrr |
| 14:01.57 | Katty | lmadsen: yeah that's why i didn't move to HSV too |
| 14:02.10 | Katty | disappears for awhile |
| 14:02.23 | lmadsen | luckily nearly everything I do doesn't require me to be on-site |
| 14:02.59 | lmadsen | ok, off to work |
| 14:03.00 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:09.13 | shriven | hmmm now that we're all so talkative.... |
| 14:09.26 | shriven | does anyone know what this means? error: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here (not in a function) |
| 14:09.49 | shriven | it sounds like possibly my arch doesn't support recursive mutexes? (powerpc) |
| 14:11.24 | jasonwoot | jinx put max in space, jinx can get max back |
| 14:13.38 | seanbright | max = leaf phoenix |
| 14:13.55 | Katty | haha. i read that as meatloaf phoenix at first. |
| 14:14.03 | seanbright | mmmm, meatloaf. |
| 14:14.09 | seanbright | he would do anything for love, but he won't do that. |
| 14:14.20 | seanbright | man... i'm old. |
| 14:15.23 | Katty | sometimes i feel pretty old. |
| 14:17.17 | jasonwoot | at least you're not a gay soccer referee |
| 14:18.34 | *** join/#asterisk elfguy516 (n=elfguy51@96.56.103.35) |
| 14:18.50 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 14:20.56 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 14:21.27 | elfguy516 | how can someone query asterisk to tell if a conference is locked (other than dialing in to see what message you get) there must be a bit that get flipped somewhere or some kind of flag that goes up |
| 14:21.37 | *** join/#asterisk fred-tmft (n=fred-tea@c-98-243-174-131.hsd1.mi.comcast.net) |
| 14:21.44 | elfguy516 | a conference in meetme that is |
| 14:25.07 | seanbright | meetme list concise |
| 14:25.08 | seanbright | ? |
| 14:26.07 | seanbright | no, that's a lie. |
| 14:27.53 | Katty | what was the name of that lil fuzzy nice thing from Gremlins? |
| 14:28.11 | Katty | the one that goes Bright Light! Bright Light! |
| 14:29.37 | elfguy516 | gizmo? |
| 14:29.56 | Katty | gizmo! that's the one. |
| 14:32.55 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:34.34 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 14:35.07 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 14:35.28 | *** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com) |
| 14:38.02 | [gnubie] | waves to all.. gtg now.. |
| 14:40.35 | *** join/#asterisk nny_2 (n=scott@64.203.244.146) |
| 14:41.29 | Blackvel | anyone of you with a tool to export whole oulook addressbook to openldap (e.g over the java connector api JLDAP or JDBC-LDAP)? |
| 14:42.36 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:45.03 | ayrjola | pattern matching problem... my sip trunk provider sends numbers on international format with + sign included, is there other way than _.X? |
| 14:45.48 | Blackvel | can't get it replaced by 00 and do _00XX. matching? |
| 14:46.49 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 14:47.11 | *** join/#asterisk synchris (n=synchris@athedsl-4391256.home.otenet.gr) |
| 14:47.33 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 14:47.35 | ayrjola | its stupid that they send + sign but they are not going to change that just for us |
| 14:47.43 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 14:48.11 | khronos | Are you trying to stip the + off the front? |
| 14:48.48 | khronos | Then after you strip it sedn the rest though your Asterisk for processing? |
| 14:49.57 | ayrjola | ok, just make context that changes number + to 00. that sound like an idea :) |
| 14:50.03 | ayrjola | thanks |
| 14:51.08 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 14:51.17 | Kobaz | man this DYANMIC_FEATURES still is killing me |
| 14:51.30 | Kobaz | dynamic... |
| 14:51.38 | Blackvel | is there no zapata.conf internationalprefix=00 trick for international sip providers? |
| 14:51.59 | Blackvel | can't be to have to reinvent the wheel? |
| 14:52.06 | nosbig | ayrjola: That is the standard E.164 format... |
| 14:52.40 | Blackvel | didn't try it...but...can't there there a + extension mapping? |
| 14:52.46 | Blackvel | there be a |
| 14:52.50 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-f80fe3f3f144a01e) |
| 14:52.50 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:52.51 | nosbig | ayrjola: You might use a pattern (for US numbers) like: _+1NXXNXXXXXX |
| 14:52.53 | *** join/#asterisk zerko (i=zerko@noc.dls.tx.serverzone.net) |
| 14:53.16 | zerko | Hello, has anyone integrated asterisk and nagios for verbal phone notifications? |
| 14:53.21 | *** join/#asterisk netsurf (n=netsurf@99.135.242.178) |
| 14:53.51 | nny_2 | stupid question: on an rj45 connector (from right to left, 1-8) for a T1 wire, which is "tip" and which is "ring" |
| 14:53.59 | netsurf | so is this a good help channel for asterisk or just general talk? |
| 14:54.25 | russellb | both :) |
| 14:54.28 | nny_2 | netsurf: both |
| 14:54.29 | zerko | netsurf, it's a help channel and chat |
| 14:54.30 | nny_2 | heh |
| 14:54.38 | zerko | everyone seems to be sleeping right now, or just getting into the office :) |
| 14:54.43 | nny_2 | i am doing both |
| 14:55.04 | nny_2 | is wiring up a 255d shelf and Tellabs echo canceller to hit an echo fly with a sledgehammer |
| 14:55.07 | netsurf | ok... i have a question about voicemail then... we got a lot of "hangup" voicemails... is there a decent way to get rid of those? |
| 14:55.20 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 14:55.36 | [TK]D-Fender | nny_2: Correct... it isn't a bright question. T&R are ANALOG terms |
| 14:55.43 | netsurf | i have a vm helper script running under externalnotify in the voicemail config... i can detect a "too small" file and delete the wav and txt file fine... but the e-mail/page notifications still go out |
| 14:55.48 | nny_2 | netsurf: in voicemail.conf there is an option for the number of seconds of silence to be considered a vm afaik |
| 14:55.52 | nny_2 | [TK]D-Fender: yeah i know heh |
| 14:55.54 | gr0mit | nny_2, have you tried Oslec echo canceller ? |
| 14:56.03 | anonymouz666 | Corydon76-dig: do you ever used ODBC with Oracle? I wonder if it works fine... I just discover that page: http://home.fnal.gov/~dbox/oracle/odbc/ |
| 14:56.04 | zerko | russelb, In version 1.4.21.2.. is the -rx removed? |
| 14:56.10 | nny_2 | gr0mit: is that software or hardware? |
| 14:56.14 | *** join/#asterisk stewbaby (n=stewart@ip-217-204-65-78.easynet.co.uk) |
| 14:56.18 | nny_2 | gr0mit: using HPEC hardware atm |
| 14:56.18 | netsurf | nny_2: any idea on the options? |
| 14:56.21 | gr0mit | software for asterisk |
| 14:56.23 | tzafrir_laptop | zerko, no. What makes you think so? |
| 14:56.27 | nny_2 | netsurf: lemme look at the sample config one sec |
| 14:56.32 | nny_2 | netsurf: actually you can do that as well |
| 14:56.33 | netsurf | k |
| 14:56.38 | zerko | im typing asterisk -rx in command line and its saying command not found |
| 14:56.39 | netsurf | indeedy |
| 14:56.44 | nny_2 | netsurf: look in /asterisk_source_dir/configs |
| 14:56.51 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-58677f4dfe79ad64) |
| 14:56.51 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:56.54 | nny_2 | netsurf: nice sample configs with the options explained |
| 14:56.57 | [TK]D-Fender | zerko: pastebin your complete attempt |
| 14:56.59 | [TK]D-Fender | ~pb |
| 14:56.59 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:57.00 | [TK]D-Fender | ^^^^ |
| 14:57.04 | tzafrir_laptop | zerko, "command not found" refers to "asterisk" |
| 14:57.08 | zerko | can someone refresh my memory, I want to do a test call from command line |
| 14:57.12 | netsurf | you know... i probably should've looked their first lol |
| 14:57.20 | zerko | sorry, I just noticed its just asking for more syntax |
| 14:57.20 | tzafrir_laptop | zerko, echo $PATH |
| 14:57.27 | netsurf | but i had to overcomplicate it... minmessage=x |
| 14:57.28 | tzafrir_laptop | is /usr/sbin there? |
| 14:57.37 | zerko | yes tza |
| 14:57.40 | nny_2 | netsurf: np heh I have asked similar questions and had the same epiphany |
| 14:57.45 | zerko | thats not the problem, I made a mistake |
| 14:57.49 | zerko | I typed asterisk -rx |
| 14:57.58 | zerko | it is just asking for more arguments |
| 14:58.05 | zerko | What is the command to run a test call from command line? |
| 14:58.05 | nny_2 | zerko: um yeah |
| 14:58.09 | netsurf | well... this leads to my other problem... simple hangup messages will often times show up as 10 secons long when they're really 0 or 1 seconds... so those will still slip through |
| 14:58.21 | nny_2 | zerko: x indicates you want to pass a console command to * |
| 14:58.28 | tzafrir_laptop | zerko, no it's not (/me also wants an argument) |
| 14:58.43 | zerko | well, I remember using the -rx command to place a phone call |
| 14:58.44 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 14:58.46 | zerko | and it actually worked |
| 14:58.48 | nny_2 | netsurf: wondering if the channel is staying open for some reason |
| 14:58.57 | zerko | I just had to specify the SIP etc. |
| 14:59.02 | zerko | I can't remember in what format it was though |
| 14:59.06 | netsurf | could be... but the recording is short |
| 14:59.06 | nny_2 | zerko: indeed, you can also do that from the console |
| 14:59.21 | zerko | yes, what was the command for that do you know? |
| 14:59.23 | netsurf | nny_2: i'm trunking calls from a cisco call manager so i wouldn't be shocked |
| 14:59.27 | nny_2 | just do asterisk -r, (console) command foo blah |
| 14:59.58 | zerko | right, what is the command :) |
| 15:00.06 | zerko | that's what I need to know |
| 15:00.12 | jameswf-home | I dont get any calls to my telemarketer deal so I decided to take one of my DID's and go sign up for all the contest and insurance quotes etc... maybe a little bait to suck em in |
| 15:00.25 | nny_2 | netsurf: make a test call and watch for when asterisk hangsup vs you do |
| 15:00.29 | Corydon76-dig | anonymouz666: depends on which Oracle libraries you install. If you install InstantClient, there's a godawful resource leak in those libraries |
| 15:00.36 | zerko | james, lol |
| 15:00.38 | Corydon76-dig | but it works |
| 15:00.50 | nny_2 | jameswf: hahah telemarketer honey pot! |
| 15:01.00 | nny_2 | jameswf: you're my new hero |
| 15:01.02 | jameswf | I am waiting fir the university of phoenix people |
| 15:01.12 | anonymouz666 | Corydon76-dig: ok, thanks. |
| 15:01.22 | zerko | ok can someone please show me the command to place a call from the console :) |
| 15:01.40 | nny_2 | considers a honeypot with fake menu options to determine the skill of the telemarketer |
| 15:01.42 | netsurf | oh and a bit off topic from my current thing... starting asterisk from /etc/init.d/asterisk start always fails... i'm running gentoo on this box fyi |
| 15:01.44 | Corydon76-dig | zerko: "console dial" |
| 15:01.56 | nny_2 | could make a game out of it.. if they reach the fake CEO, they get super blacklisted |
| 15:02.03 | netsurf | nny_2: yeah i'll have to watch that... seems like all outside calls 10 seconds too long |
| 15:02.26 | jameswf | I have it directed at astycrapper... so they get to talk to a robot |
| 15:02.51 | nny_2 | netsurf: yeah just kick verbosity up to 5 in console and follow the dp visually vs audibly |
| 15:03.11 | nny_2 | jameswf: astycrapper.. need to check that out |
| 15:04.25 | zerko | http://www.linuxsystems.com.au/astycrapper/astycrapper1.mp3 |
| 15:04.26 | zerko | heh |
| 15:06.24 | netsurf | nny_2: yeah i'll take a look at it... thanks for the help |
| 15:06.28 | nny_2 | omg that is awesome |
| 15:06.33 | nny_2 | sounds like 90% of my clientel |
| 15:06.41 | zerko | heh |
| 15:06.51 | zerko | yes that is pretty awsome |
| 15:08.27 | netsurf | well i'm a tard... regarding the not starting |
| 15:08.36 | nny_2 | omg she doesnt give up |
| 15:08.40 | nny_2 | still going |
| 15:08.43 | netsurf | never configured the user it was supposed to run as... ;) |
| 15:08.51 | nny_2 | "whas dat now yur sayin der?" |
| 15:08.54 | Kobaz | http://pastebin.ca/1216937 i'm having some trouble with features.conf [applicationmap] and setting DYNAMIC_FEATURES in the dialplan |
| 15:08.55 | nny_2 | "Hello?" |
| 15:09.55 | nny_2 | "dem chitrin better seen den herd" |
| 15:09.56 | *** join/#asterisk UnixDawg_ (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 15:10.06 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:11.12 | jameswf | the second recoding or so the guy figures it out 5 min in then starts bringing people over |
| 15:12.19 | *** part/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 15:12.30 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 15:14.02 | *** join/#asterisk stewbaby (n=stewart@ip-217-204-65-78.easynet.co.uk) |
| 15:16.01 | *** join/#asterisk YoYo (n=chatzill@12.196.144.39) |
| 15:16.44 | zerko | anyone here using nagios + asterisk for phone notifications? |
| 15:16.56 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:17.11 | nny_2 | zerko: i started too a while ago |
| 15:17.11 | krokodilerian | zerko , i have |
| 15:17.21 | YoYo | anyone wanna make a quick $50 configuring a sangoma pri card and a x100p clone? |
| 15:17.24 | nny_2 | zerko: had issues with asterisk-snmp, but it was fairly new at the time |
| 15:17.26 | krokodilerian | zerko , it's pretty easy |
| 15:17.37 | zerko | nice |
| 15:17.43 | krokodilerian | zerko , please use the channel :) |
| 15:17.49 | zerko | I had made a program a few months ago, and ended up trashing it |
| 15:17.58 | zerko | I wish I had known there was something like this already |
| 15:18.14 | zerko | krokodilerian are there any detailed docs out there on setting it up? |
| 15:18.30 | *** join/#asterisk seanmh (i=HydraIRC@216.31.101.24) |
| 15:18.31 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 15:18.36 | krokodilerian | zerko , no, what I did was a simple over-ssh execution of a script |
| 15:18.50 | krokodilerian | that created/copied a call file that dialed out and played a message |
| 15:18.59 | waverly360 | Hey guys, anyone here ever run into a problem where a call gets stuck in a queue, and no other calls get distributed to agents? |
| 15:19.32 | zerko | krokodilerian hrm, could you give me the link to that script? |
| 15:19.50 | krokodilerian | zerko , i can give you sometihng similar :) |
| 15:19.50 | waverly360 | I tried to do a soft hangup on channel that got stuck, but asterisk claims that channel didn't exist. |
| 15:19.51 | krokodilerian | sec. |
| 15:20.02 | zerko | sure :) |
| 15:20.18 | waverly360 | My only resort was to restart asterisk which ended up dropping all of the calls. |
| 15:20.48 | *** join/#asterisk easycrypt (n=savek@ip-186.emscb.ruhr-uni-bochum.de) |
| 15:20.53 | easycrypt | hi all |
| 15:21.19 | *** join/#asterisk synchris (n=synchris@athedsl-4391256.home.otenet.gr) |
| 15:21.27 | easycrypt | is there a way to handle a case where the caller hangs up during the Dial() App with in the dialplan? |
| 15:21.36 | jameswf | easycrypt: h |
| 15:22.01 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 15:22.08 | eric2 | waverly360 - sounds like a logic issue in your dial plan |
| 15:22.12 | krokodilerian | zerko http://pastebin.com/m22d7a8bb |
| 15:22.32 | easycrypt | like "exten => 112,h,Noop(${DIALSTATUS}) ? |
| 15:22.40 | waverly360 | eric2: I really don't think that's the case. |
| 15:22.45 | easycrypt | because i don't get any output using it like that |
| 15:23.08 | waverly360 | eric2: How can a dialplan issue cause a call in a queue to get "hung"? |
| 15:23.41 | zerko | thank you, will check it after im done with this install |
| 15:23.55 | krokodilerian | zerko , it's really rudimentary |
| 15:24.05 | krokodilerian | but then with it you can pass parameters on where to call, which file to play,etc. |
| 15:24.16 | eric2 | I'm not sure about the queue thing, I have yet to implement it, but I did have a cyclical error in my dial plan then the timeout time limit was hit... until I fixed it |
| 15:24.17 | krokodilerian | simple shell scripting |
| 15:24.30 | eric2 | *whne |
| 15:24.32 | eric2 | *when |
| 15:24.56 | nny_2 | [TK]D-Fender: so back to the whole tip/ring thing. as far as 1 and 2 in the pair, you think tip =1 and ring =2? |
| 15:24.59 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 15:25.33 | Kobaz | heh |
| 15:25.36 | nny_2 | [TK]D-Fender: funny thing is the manual uses that terminology. I suspect it's because a good portion of telephony folks know squat about data |
| 15:25.36 | Kobaz | poor fender |
| 15:25.43 | [TK]D-Fender | nny_2: http://www.juniper.net/techpubs/hardware/m40/m40-hwguide/html/pinout4.html |
| 15:25.52 | waverly360 | eric2: I've had loops in my dialplan before...and problems with people forwarding phones to each other..that happens from time to time..but you can see it in the console. |
| 15:25.55 | Kobaz | [TK]D-Fender: we need to clone you |
| 15:25.57 | nny_2 | [TK]D-Fender: <3 |
| 15:26.00 | nny_2 | [TK]D-Fender: thanks |
| 15:26.05 | YoYo | when installing wanpipe stuff, where are the startup scripts, config files, 'n stuff placed? (I'm a linux nub) |
| 15:26.24 | jameswf | clones of [TK]D-Fender availible at google.com |
| 15:26.33 | eric2 | YoYo, sangoma? |
| 15:26.59 | YoYo | yeah |
| 15:27.08 | [TK]D-Fender | Kobaz: If you've ever read "Boys from Brazil", I'm the real reason against cloning ;) |
| 15:27.09 | eric2 | the script will do it all automatically for you |
| 15:27.19 | Kobaz | heh, i haven't |
| 15:27.30 | YoYo | I understand... and after running wancfg_zaptel, it's all working |
| 15:27.49 | Kobaz | but anyways |
| 15:27.49 | [TK]D-Fender | was born and the mold smashed itself, screaming "Never again! Never again!!!!!" |
| 15:27.50 | YoYo | but, I need to also load and configure wcfxs for a x100p |
| 15:27.54 | Kobaz | hehe |
| 15:27.58 | eric2 | that I have not done |
| 15:28.04 | YoYo | don't see anything in /etc/modprobe |
| 15:28.08 | Kobaz | [TK]D-Fender: http://pastebin.ca/1216937 :) |
| 15:28.11 | tzafrir_laptop | YoYo, x100p: wcfxo |
| 15:28.13 | Kobaz | i have a pastebin this time |
| 15:28.25 | YoYo | yeah... that |
| 15:28.28 | eric2 | I have a pastebin too :) |
| 15:28.32 | YoYo | (5 years later, and I still can't keep it straight) |
| 15:28.39 | tzafrir_laptop | YoYo, what distribution do you use? |
| 15:28.40 | Kobaz | [TK]D-Fender: you should set up a donate link |
| 15:28.46 | YoYo | centos 5 |
| 15:29.12 | tzafrir_laptop | it's either /etc/modprobe.conf or (better) a file under /etc/modprobe.d |
| 15:29.16 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 15:30.42 | jameswf | sexual harrasment violates company policy.... thankfully no chicks work here that anyone would wanna do anything sexual with much less hurass |
| 15:31.56 | [TK]D-Fender | Kobaz: I'm a very casual consultant.... |
| 15:32.05 | [TK]D-Fender | Kobaz: I try not to sit around with my hand out. |
| 15:32.09 | Kobaz | yeah |
| 15:34.49 | YoYo | GRRR... got it. when loading wcfxo, it pushes the PRI from span 1 to span 2 |
| 15:35.53 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:35.53 | YoYo | now to reconstruct my twisted and convoluted dialplan |
| 15:39.15 | jameswf | score university of phoenix called |
| 15:42.26 | Kobaz | do de do |
| 15:42.38 | Kobaz | [TK]D-Fender: any idea? |
| 15:43.59 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-238-140.balt.east.verizon.net) |
| 15:46.09 | *** join/#asterisk sircco (n=sircco@dh207-103-196.xnet.hr) |
| 15:46.49 | theleon | if i have fxo card, is that for connecting to isdn connector or plain even older pstn? |
| 15:47.56 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 15:48.10 | mgdm | theleon: PSTN |
| 15:48.29 | theleon | and what if i want to connect to isdn? |
| 15:49.12 | Blackvel | you need isdn card or external isdn gateway |
| 15:49.23 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:49.39 | theleon | is that called somehow.. like fxo or fxs |
| 15:51.15 | Blackvel | different options...yes |
| 15:51.19 | *** part/#asterisk jsmith (n=njsmith@72.21.36.138) |
| 15:51.20 | Blackvel | bristuff / zap |
| 15:51.33 | Blackvel | my isdn gw just connects with SIP to SIP peer |
| 15:51.33 | theleon | misdn? |
| 15:51.37 | Blackvel | e.g |
| 15:51.42 | theleon | i understand now |
| 15:51.50 | theleon | tnx :) |
| 15:51.52 | Blackvel | and you will find different opinions what is good or crap |
| 15:51.59 | theleon | digium is good i guess :) |
| 15:52.07 | theleon | is there some cheap fxo i can buy in store? |
| 15:52.13 | Blackvel | would be misdn then |
| 15:52.24 | Blackvel | there is one really cheap solution |
| 15:52.25 | theleon | like planet isdn card for cheap isdn |
| 15:52.41 | Blackvel | but that depends on your requirements (and maybe I could not recommend it) |
| 15:52.54 | Blackvel | private use or business? with voip phones or without? |
| 15:52.56 | theleon | well it's good for playing at home i guess |
| 15:52.57 | theleon | private |
| 15:53.03 | theleon | with voip phones |
| 15:53.08 | theleon | i'd like to connect asterisk to it |
| 15:53.12 | theleon | and play on my pstn |
| 15:53.14 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 15:53.44 | Blackvel | you can *try* the billion isdn/hfc-s (cheap) way with bristuff/junghanns.net |
| 15:53.58 | Blackvel | it worked fine with my isdn pbx (connect to asterisk) but not isdn pstn |
| 15:54.01 | Blackvel | had echo problems |
| 15:54.12 | tzafrir_laptop | theleon, an ISDN->analog convertor and then connecting it to a digital PBX like Asterisk is plain dumb |
| 15:54.14 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-176.usadatanet.com) |
| 15:54.14 | Blackvel | they may appear or no |
| 15:54.14 | theleon | eh yes billion ..but i dont have isdn at home |
| 15:54.19 | theleon | i have pstn |
| 15:54.28 | Blackvel | oh...analog |
| 15:54.33 | Blackvel | didn't you ask for isdn? |
| 15:54.34 | theleon | yes i need something for analog |
| 15:54.35 | theleon | nope |
| 15:54.35 | Blackvel | sorry |
| 15:54.45 | tzafrir_laptop | oh, analog . oh well... |
| 15:54.50 | theleon | eh.. :) |
| 15:55.11 | Blackvel | forget everything I said |
| 15:55.12 | Blackvel | :) |
| 15:55.19 | theleon | how about that grandstream handytone ? |
| 15:55.26 | theleon | its 30$ |
| 15:55.37 | tzafrir_laptop | that's the FXS adapter |
| 15:55.55 | theleon | hum |
| 15:56.02 | Blackvel | not sure about anlog |
| 15:56.13 | tzafrir_laptop | you need e.g. their 488, or an SPA 3102 |
| 15:56.14 | Blackvel | analog...but you can expect echos (if no hw echo cancellation) |
| 15:56.27 | Blackvel | with normal cards.... (no hardware box) |
| 15:56.28 | tzafrir_laptop | or decent software EC |
| 15:56.30 | theleon | maybe i can |
| 15:56.34 | theleon | use asterisk ec |
| 15:56.36 | theleon | that kernel module |
| 15:56.42 | Blackvel | I wish |
| 15:56.52 | theleon | im not sure if that would work |
| 15:56.57 | Blackvel | run even on isdn into echo problem with voip phone |
| 15:56.58 | tzafrir_laptop | Actually for me at home MG2 is good enough |
| 15:57.07 | theleon | mg2? |
| 15:57.09 | Blackvel | it was a mess with mg2 |
| 15:57.09 | theleon | that ec? |
| 15:57.17 | Blackvel | ppl recommend olsec |
| 15:57.21 | tzafrir_laptop | Blackvel, the voip phone should cancel its echo |
| 15:57.25 | Blackvel | I bought the other one |
| 15:57.26 | theleon | yeah olsec i used that on isdn a bit |
| 15:57.33 | Blackvel | tzafrir_laptop: the other party had echo (pstn side) |
| 15:57.36 | theleon | and it works very well |
| 15:57.53 | Blackvel | it MAY depend on the pc server * runs on |
| 15:57.59 | Blackvel | epia via nemia 1 gig |
| 15:58.07 | Blackvel | cpu speed...forget all |
| 15:58.18 | Blackvel | (up to 15%..but not working) |
| 15:58.25 | theleon | hey so if i want to connect to pstn i need ata? |
| 15:58.27 | theleon | is that it? |
| 15:58.31 | Blackvel | ou can try...or digium hpec...or what was the other...? |
| 15:58.48 | Blackvel | octasic |
| 15:58.49 | Blackvel | bought that |
| 15:58.52 | Blackvel | didn't help too |
| 15:59.07 | Blackvel | not sure how good EC works with SW or FXO cards really |
| 15:59.18 | Blackvel | I am happy that I bought 380 euro isdn gateway |
| 15:59.20 | Blackvel | no echos anymore |
| 15:59.27 | Blackvel | probably the same for fxo |
| 15:59.47 | theleon | what is pstn pass through? |
| 15:59.48 | Blackvel | theleon: most atas have fxs (to connect phone to) |
| 15:59.58 | Blackvel | some could have fxo as well as |
| 16:00.04 | theleon | im looking at handytone 486 |
| 16:00.11 | theleon | it says pstn passthrough |
| 16:00.20 | Blackvel | probalby for the connecting 2nd fxs port |
| 16:00.42 | Blackvel | not sure how asterisk could connect with fxo port to pstn |
| 16:00.46 | Blackvel | never tried that |
| 16:01.02 | theleon | well it says there is ethernet there.. |
| 16:01.06 | theleon | maybe handytone can offer me sip |
| 16:01.11 | theleon | like a sip trunk |
| 16:01.18 | theleon | i should read docs :) |
| 16:01.47 | Blackvel | hm |
| 16:02.31 | theleon | http://www.grandstream.com/ht486.html |
| 16:02.33 | theleon | could this do it? |
| 16:03.17 | theleon | damn..no fxo port |
| 16:03.20 | Blackvel | I want a sweet openldap tool ... I hate playing around with LDIF and ldappadd commands |
| 16:03.36 | Blackvel | there is jldap (java to ldap) |
| 16:03.47 | Blackvel | but I dont want to start a new programming project |
| 16:05.57 | theleon | Blackvel: u have linux? |
| 16:06.51 | theleon | you can use phpldapadmin |
| 16:06.55 | theleon | its web based and it works |
| 16:07.29 | theleon | for linux i used gq for years |
| 16:07.32 | theleon | it rarely crashes |
| 16:07.43 | *** join/#asterisk Nasra (n=maxshipp@CPE001217b1920e-CM00159a010eda.cpe.net.cable.rogers.com) |
| 16:09.12 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 16:11.54 | jameswf | I need to figure out how to get more telemarketers |
| 16:13.31 | UnixDawg | put yourself on the call list to be called everyday |
| 16:14.23 | jameswf | I am signing up for vacation contests |
| 16:14.27 | jameswf | :) |
| 16:14.46 | jameswf | Universit of phoenix was gold a call in under 12 hours |
| 16:16.07 | Blackvel | theleon: yes linux |
| 16:16.22 | theleon | Blackvel: gq is good |
| 16:16.43 | zerko | krokodilerian hey. where do I put this script? |
| 16:16.45 | Blackvel | but how do I get the data from windows laptop (outlook) into openldap? no export, no manual file modification, no umlaut modifications...etc |
| 16:17.09 | Blackvel | gq-project.org? |
| 16:17.15 | theleon | freshmeat.net |
| 16:17.32 | Blackvel | telemarketers? for what? |
| 16:18.43 | jameswf | my little honeypot |
| 16:19.08 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 16:19.57 | Carlos_PHX | Any of you guys know anything about Vitality (service provider). |
| 16:21.22 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 16:23.11 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 16:26.34 | Blackvel | what can I do with snom 370 and LDAP (e.g openldap) |
| 16:26.41 | Blackvel | dail the number from the address book? |
| 16:30.14 | *** join/#asterisk shaw22dog (n=shaw@pacman.oaklandcorp.com) |
| 16:30.59 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 16:33.24 | gewuerzwiesel | how can I set up different outgoing MSNs for different sip phones? f.e I have the sipphone 31, on outgoing calls It should have the MSN 952453, and the 32 should usw 952454? |
| 16:33.40 | krokodilerian | zerko , just run that from nagios from a notify command |
| 16:33.52 | krokodilerian | anyway, g'night, still fighting jetlag |
| 16:35.08 | Blackvel | gewuerzwiesel: works for me with fromuser=MSN in sip.conf |
| 16:35.44 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 16:36.00 | Blackvel | alternatively just make extensions.conf with exten => X.,1,Set(CALLERID(num)=952453) |
| 16:36.06 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 16:36.18 | Blackvel | you can define different contexts for sipphone 31 and 32 |
| 16:40.23 | gewuerzwiesel | Blackvel: ok thx, i'll try it |
| 16:41.56 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 16:42.29 | shaw22dog | Anyone bored enough to lend some advice to a guy who is having periodic audio issues on outbound calls? |
| 16:46.44 | jameswf | ok car waranties, schools MLM scams some of these have to generate calls |
| 16:46.54 | jameswf | I need to get on the free satalite list |
| 16:48.00 | Carlos_PHX | shaw22dog: Maybe, describe the problem, codec and protocol, and who the PSTN provider is. |
| 16:49.26 | *** join/#asterisk soulfreshner (n=derick@dsl-243-4-106.telkomadsl.co.za) |
| 16:51.31 | soulfreshner | I have a client that tells me his phone keeps disconnecting after a while... could it have something to do with busydetect=yes? |
| 16:52.16 | soulfreshner | it looks like it only happens with calls from a ZAP channel |
| 16:53.19 | gewuerzwiesel | Blackvel: great, it works :) thx |
| 16:53.30 | mav3rick | hi all |
| 16:54.36 | mav3rick | is it possible to have one channel doing a Playback and a Dial simultaneously ? the Playback would stop while bridging the call |
| 16:54.43 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:55.15 | jameswf | the dial is a 1 second or so process its pretty fast |
| 16:55.32 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 16:55.36 | mav3rick | the workaround I found is to create a callout file to Dial, and then using ChannelRedirect that redirects both channel to a MeetMe room |
| 16:56.32 | mav3rick | jameswf: I want someone to hear a music (Playback, I don't want moh) while the Dial is ringing |
| 16:56.53 | jameswf | what music do you want? |
| 16:57.08 | mav3rick | specific files I have |
| 16:57.15 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 16:57.16 | mort_gib | What is the difference between MOH and music?? |
| 16:57.19 | jameswf | put those files in a moh context |
| 16:57.36 | mav3rick | I want a specific file |
| 16:57.44 | mav3rick | I have like 4000 files |
| 16:57.56 | mort_gib | Then create a MOH for the file(s) |
| 16:58.01 | mav3rick | and I want a specific one (based on callerid) |
| 16:58.14 | jameswf | so put them in 4000 conrext and set context based on CID |
| 16:58.29 | mav3rick | how can I select one in the context ? |
| 16:58.37 | mort_gib | Or pick up from a DB which user is supposed to hear what |
| 17:00.13 | jameswf | m(context) |
| 17:00.21 | mav3rick | if I put my 4000 files in a MoH context, I can't choose a specific file in, can I ? |
| 17:00.22 | jameswf | http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 17:00.23 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:01.28 | mav3rick | and I want the files to be played from the beginning |
| 17:01.58 | mav3rick | I think MoH "shares" read position between all channels using the same context |
| 17:02.50 | thehar | go at&t.. everyone in slc just lost service |
| 17:03.01 | thehar | still down. |
| 17:04.11 | *** join/#asterisk Xentac (n=xentac@archlinux/developer/Xentac) |
| 17:04.37 | Xentac | ok, my freepbx+trixbox+asterisk problem has been debugged down to an asterisk problem |
| 17:04.54 | zerko | ok |
| 17:04.56 | zerko | guys |
| 17:04.57 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 17:05.04 | Xentac | if I set up a dialplan that calls 8 lines (SIP/400&SIP/401&...) only the first 4 ring |
| 17:05.12 | zerko | http://pastebin.com/m22d7a8bb |
| 17:05.17 | zerko | how do I get this to play a file? |
| 17:05.18 | [TK]D-Fender | Xentac: pastebin your CLI output. |
| 17:05.24 | Xentac | no matter which ones are the first |
| 17:05.36 | zerko | I know it should go somewhere in the SIP line |
| 17:05.41 | zerko | but dont know exactly what to add |
| 17:05.44 | [TK]D-Fender | Xentac: Along with peer dumps for all members prior to calling |
| 17:06.12 | Xentac | [TK]D-Fender: do I get a peer dump from the asterisk console? |
| 17:06.21 | *** join/#asterisk chazz (n=chazz@nat/digium/x-9b057f3168eb5039) |
| 17:06.37 | Xentac | is that like a "sip show peers"? |
| 17:06.52 | [TK]D-Fender | Xentac: "sip show peer [peerwithoutbraces]" |
| 17:07.04 | Xentac | alrighty |
| 17:07.09 | Xentac | sets that stuff up |
| 17:07.11 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:09.52 | Xentac | [TK]D-Fender: http://pastebin.com/m526d0436 |
| 17:10.05 | zerko | http://pastebin.com/m22d7a8bb - can someone tell me where/what I need to add in order for the call to play a file when I pick up please? |
| 17:14.14 | zerko | anyone? |
| 17:14.23 | Xentac | [TK]D-Fender: anything else you can think of that'll help? |
| 17:18.24 | [TK]D-Fender | Xentac: -- Executing [667@from-internal:1] Dial("SIP/400-09fb5118", ""SIP/500&SIP/501&SIP/502&SIP/503&SIP/400&SIP/401&SIP/402&SIP/403"") in new stack |
| 17:18.25 | jameswf | ok so I am going to do podcasts of the telemarketer calls to the honeypot.... |
| 17:18.32 | [TK]D-Fender | Xentac: its encased in quotes. First irregularity |
| 17:18.40 | Xentac | [TK]D-Fender: alrighty |
| 17:18.52 | [TK]D-Fender | Xentac: Next you did not follow through with letting someone answer and I have no sense of exactly how long you let it ring for. |
| 17:19.09 | [TK]D-Fender | Xentac: 2 situations that incur automatic distrust |
| 17:19.35 | Xentac | ok, I'll get someone to answer |
| 17:20.18 | [TK]D-Fender | Xentac: 10s wait please |
| 17:20.18 | Xentac | I don't know how you'll know how long it rings for though |
| 17:20.18 | Xentac | sure |
| 17:20.18 | [TK]D-Fender | Xentac: And make sure it doesn't double-quote |
| 17:20.18 | Xentac | yup, I got that |
| 17:20.50 | Xentac | hmmm... |
| 17:20.54 | [TK]D-Fender | Xentac: And while you're at it, "sip show peers" (not full dump) |
| 17:21.01 | Xentac | ok, I got rid of the double quotes |
| 17:21.04 | Xentac | and reloaded |
| 17:21.10 | Xentac | and now it only called two extensions |
| 17:21.17 | Xentac | er, ringed |
| 17:21.47 | zerko | http://pastebin.com/m22d7a8bb - does anyone know where this script came from? |
| 17:23.19 | Xentac | [TK]D-Fender: http://pastebin.com/m74c85726 |
| 17:23.50 | Xentac | I don't know if I can add timestamps or something to this CLI output |
| 17:24.34 | Xentac | hmmm, I tried again and it dialed the first 4 extensions, like the original problem |
| 17:25.12 | Xentac | maybe I dialed too soon after the reload and that's why it only rang two extensions instead of four? |
| 17:26.39 | [TK]D-Fender | Xentac: Thing to remember : trixbox is running custom code for * & FreePBX and has channel limit mechanisms. This may be why. Especailly since nobody has ever come in here with this kind of problem. |
| 17:26.56 | Xentac | nods. |
| 17:27.20 | Xentac | so it might still be a trixbox or freepbx problem :( |
| 17:27.20 | Xentac | alrighty |
| 17:27.23 | [TK]D-Fender | HIGH likelyhood |
| 17:27.29 | Xentac | I'll see if I can find anything about the channel limit stuff |
| 17:27.30 | Xentac | thanks |
| 17:27.47 | [TK]D-Fender | I'm going to try to be civil for a moment.... |
| 17:27.49 | [TK]D-Fender | *ahem* |
| 17:27.51 | [TK]D-Fender | *cough* |
| 17:28.32 | jameswf | astycrapper: 1st victim http://astycrapper.podbean.com/2008/10/02/university-of-phoenix/ |
| 17:28.33 | *** join/#asterisk tsopis64_ (n=Aris@athedsl-404972.home.otenet.gr) |
| 17:28.55 | [TK]D-Fender | trixbox is an hot steamy pile of manure whose methane emissions are then set on FIRE to bake the turd to a cinder |
| 17:29.10 | Xentac | [TK]D-Fender: I get that impression |
| 17:29.35 | Xentac | I'm just too stupid of an asterisk user to be able to comprehend and administer much else |
| 17:29.51 | Xentac | or have other people administer it |
| 17:30.02 | [TK]D-Fender | Xentac: And pretty much the only thing we'd accept as a non-* related bug ais if you showed us a core SIP stack problem, etc. |
| 17:30.21 | [TK]D-Fender | Xentac: EVERYTHING else is trixbox / orther GUI/interface/whatever BS |
| 17:30.42 | Xentac | nods. |
| 17:31.06 | [TK]D-Fender | "JoetrixUser : Hey my calls in Trixbox don't work, it must be an * problem, you have to help me" = BS |
| 17:31.43 | [TK]D-Fender | Xentac: I hope you find some nice solid backup for your specific case of course so you can take it to the PTB's |
| 17:32.13 | Xentac | PTB's? |
| 17:32.22 | [TK]D-Fender | "Powers That Be" |
| 17:32.36 | *** join/#asterisk moy (n=moy@nat/ibm/x-7c142c6a12d03454) |
| 17:32.37 | Xentac | ah |
| 17:32.53 | Xentac | ok, well, thanks a lot for looking at it |
| 17:32.59 | Xentac | and for being civil ;) |
| 17:34.05 | *** join/#asterisk propellerhead (n=yogurt2u@host113.190-136-116.telecom.net.ar) |
| 17:34.59 | *** join/#asterisk plaerzen (n=camthomp@vip2.tundraeng.com) |
| 17:35.07 | Katty | fender? civil? |
| 17:35.09 | Katty | that's a first. |
| 17:36.02 | [TK]D-Fender | Xentac: I managed to hold back from using any actual swear words (exempt "bs") or anti-homosexual type words (never meant as a slight against them) either. |
| 17:36.34 | [TK]D-Fender | Katty: I'd slipped in the past 2 months, but have come around lately with some due wake-up calls... |
| 17:36.59 | tzanger | heh |
| 17:37.20 | Blackvel | g'd evening. bye |
| 17:37.23 | Katty | pats [TK]D-Fender |
| 17:38.05 | Katty | http://fantasticcontraption.com/ <- fun. |
| 17:38.26 | tzanger | can't play :-( |
| 17:38.34 | tzanger | need updated flash for leenooks |
| 17:38.49 | Katty | :< |
| 17:39.00 | plaerzen | question: has any of you ever ran into the problem of direct dial #'s not being routed to voicemail properly while internal extensions work fine? |
| 17:39.03 | Xentac | Katty: I know the guy who made that, I used to work with him |
| 17:39.14 | Katty | Xentac: send my compliments. |
| 17:39.15 | plaerzen | the phone rings twice and then drops the call if it's from an outside line |
| 17:39.47 | *** join/#asterisk chigital (n=chigital@tmo-115-1.customers.d1-online.com) |
| 17:41.56 | [TK]D-Fender | plaerzen: Pastebin your complete call attempt along with the associated dialplan. |
| 17:41.58 | [TK]D-Fender | ~pb |
| 17:41.58 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:42.00 | [TK]D-Fender | ^^^^^^^^^^ |
| 17:42.26 | *** join/#asterisk soulfreshner (n=derick@dsl-243-4-106.telkomadsl.co.za) |
| 17:43.51 | *** join/#asterisk darkskiez (n=mbryars@ip04.contempt.adsl.gxn.net) |
| 17:44.29 | *** join/#asterisk chigital (n=chigital@tmo-115-1.customers.d1-online.com) |
| 17:44.40 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:44.40 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 17:44.44 | Katty | lmadsen: YOU |
| 17:44.45 | soulfreshner | I'm trying to get FOP to work - but I keep getting flashing red/green lights |
| 17:44.46 | Katty | lmadsen: GET OUT |
| 17:45.06 | Katty | soulfreshner: is manager connected? |
| 17:45.12 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:45.15 | soulfreshner | I've upgraded to v.0.29... |
| 17:45.36 | soulfreshner | Katty, it looks like it |
| 17:45.56 | soulfreshner | it says Manager 'user' connected in the CLI |
| 17:46.05 | Katty | is that the one you setup for FOP? |
| 17:46.25 | soulfreshner | yep |
| 17:47.15 | Katty | does your html/flash bits match the stuff from v.0.29 in the config folder? |
| 17:47.44 | soulfreshner | googling pointed me to a problem with the latest flash player - but the new version is supposed to fix that (http://www.asternic.org/) |
| 17:48.01 | soulfreshner | well - I just replaced the older versions |
| 17:48.08 | soulfreshner | of the swf file |
| 17:48.16 | Katty | i'd start over. |
| 17:48.25 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 17:48.25 | Katty | most likely you have a version mismatch of something. |
| 17:48.42 | Katty | just backup your buttons.cfg, op_style and stuff |
| 17:48.54 | Katty | dump the rest, start over. |
| 17:49.02 | soulfreshner | it;s a fresh install |
| 17:49.25 | soulfreshner | no backups needed, I mean |
| 17:52.35 | *** join/#asterisk chigital (n=chigital@tmo-115-1.customers.d1-online.com) |
| 17:53.47 | *** join/#asterisk chigital (n=chigital@tmo-115-1.customers.d1-online.com) |
| 17:54.11 | *** join/#asterisk i3inary (n=shaman@cpe-76-88-94-149.san.res.rr.com) |
| 18:03.22 | tristanbob_ | any recommendations on a voip gateway? (other than asterisk) |
| 18:03.30 | tristanbob_ | for 2 PRIs |
| 18:05.00 | plaerzen | [TK]D-Fender: haha, figured it out (somewhat). It's only the 3g iPhones that are giving us that problem. Getting a hangup request after 2 rings. The console is saying span1 |
| 18:05.21 | plaerzen | the 2g iPhone works fine too :P |
| 18:05.49 | [TK]D-Fender | tristanbob_: AudioCodes Mediant 2000 |
| 18:06.06 | tristanbob_ | [TK]D-Fender, do you use those? |
| 18:06.29 | [TK]D-Fender | tristanbob_: Once a long time ago. |
| 18:08.27 | *** join/#asterisk synchris (n=synchris@athedsl-161100.home.otenet.gr) |
| 18:11.23 | plaerzen | has anyone had that problem with the iPhone 3g before? |
| 18:12.26 | jameswf | I hear new i phones are crap |
| 18:16.16 | tzanger | I like 'em |
| 18:16.19 | tzanger | mine's pretty good |
| 18:16.35 | tzanger | although it's refusing to move to the canadian itunes store (itunes on the PC is correct) |
| 18:17.58 | zerko | anyone here use festival? |
| 18:19.19 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-44197b7a4cf492ae) |
| 18:19.19 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 18:20.09 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-44197b7a4cf492ae) |
| 18:23.39 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:23.46 | *** join/#asterisk dklima (n=daniel@border.positivo.com.br) |
| 18:23.50 | *** part/#asterisk dklima (n=daniel@border.positivo.com.br) |
| 18:26.20 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-d03f06097d658497) |
| 18:26.20 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 18:26.35 | *** join/#asterisk darkskiez (n=mbryars@ip03.contempt.adsl.gxn.net) |
| 18:26.43 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 18:27.07 | zerko | ? |
| 18:28.00 | Katty | yawns |
| 18:28.33 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 18:29.07 | *** join/#asterisk Cutlass (n=chatzill@c-67-176-208-15.hsd1.il.comcast.net) |
| 18:30.17 | Cutlass | is there a way to configure asterisk so that it is never in the media path? I know setting canreinvite=yes allows media to be shuffled off once the call is established, but I don't want the media to go through asterisk to begin with |
| 18:30.59 | [TK]D-Fender | Cutlass: Media should not be established until the call is sent audio. |
| 18:31.08 | [TK]D-Fender | Cutlass: Aside from that, * is not a proxy. |
| 18:33.18 | Cutlass | understood...so regarding your first response, you're saying that the actual media will not flow until the call is established...but I'm want the SDP to be negotiated such that only the endpoints are involved...I guess you're saying that's not possible since it's a B2BUA, correct? |
| 18:33.45 | [TK]D-Fender | Cutlass: Probably so. |
| 18:34.47 | Cutlass | hummm...ok..is there a way to control SDP parameters from the dialplan? |
| 18:35.24 | *** join/#asterisk NovceGuru (n=NovceGur@rrcs-70-62-198-142.central.biz.rr.com) |
| 18:35.26 | [TK]D-Fender | Cutlass: Nope, the idea is pretty much dead in the water as * is a B2BUA |
| 18:35.29 | *** join/#asterisk wtsexton (n=tim@potatosalad.worldspice.net) |
| 18:35.47 | *** join/#asterisk Laureano (n=Laureano@190.245.108.2) |
| 18:38.15 | wtsexton | has anyone had the issue of when using the background function calls are dropping with chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission, this is only when playing back to the call, calls in and out work fine. Just the drops when using background sounds in menus |
| 18:40.43 | *** join/#asterisk zionvier (n=pete@c-67-166-22-106.hsd1.co.comcast.net) |
| 18:42.01 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0, 1.4.22 (2008/10/02), *-Addons 1.6.0 (2008/10/02), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0+2.0.0 (2008/10/02), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 18:42.41 | Qwell | ooo, Zaptel gone? |
| 18:43.24 | russellb | Qwell: yeah, it's old news |
| 18:43.25 | russellb | :) |
| 18:43.29 | *** join/#asterisk darkskiez (n=mbryars@ip03.contempt.adsl.gxn.net) |
| 18:43.42 | *** join/#asterisk stegbth (n=stegbth@mail.a-fk.de) |
| 18:44.10 | stegbth | hello everybody |
| 18:45.03 | Cutlass | [TK]D-Fender..thanks for the info |
| 18:45.21 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-238-140.balt.east.verizon.net) |
| 18:45.38 | stegbth | may i ask trixbox question's with sangoma here? |
| 18:45.41 | SteveTotaro | can the asterisk verison of a switchvox box be upgraded by grabbing the newest releases? |
| 18:45.53 | Qwell | SteveTotaro: Do you want to keep your support? |
| 18:46.11 | SteveTotaro | the customer has come to me since they are not getting support |
| 18:46.18 | russellb | no, it can not. |
| 18:46.23 | russellb | Switchvox contains custom modifications. |
| 18:47.01 | Xentac | [TK]D-Fender: I'm an idiot. just so you don't worry about this anymore, I didn't have a large enough rtpstart and rtpend range |
| 18:47.05 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 18:47.15 | Xentac | feel free to not be civil towards me now ;) |
| 18:47.19 | SteveTotaro | http://bugs.digium.com/view.php?id=4101 |
| 18:47.31 | wtsexton | any pointers in the right direction? |
| 18:47.34 | SteveTotaro | the log is filled with dropping frame changed to slin |
| 18:47.46 | SteveTotaro | and they have choppy audio |
| 18:48.00 | SteveTotaro | even pri to voicemail is choppy |
| 18:48.13 | *** join/#asterisk funxion (n=x@63.214.236.169) |
| 18:49.54 | SteveTotaro | Qwell, the support question makes me wonder if i can not really upgrade like russellb claims |
| 18:50.06 | Qwell | if you want it to work, no, you can't |
| 18:50.20 | SteveTotaro | it doesn't really work now |
| 18:50.22 | x86 | is it possible to use DAHDI with Wanpipe? |
| 18:50.23 | russellb | you can't upgrade it. the answer is that simple. |
| 18:50.27 | Qwell | x86: don't know |
| 18:50.33 | russellb | SteveTotaro: and there is nothing you can do other than work through support. |
| 18:50.38 | x86 | or, preferrably, in place of wanpipe on Sangoma hardware? |
| 18:50.53 | Qwell | x86: I haven't seen anything that suggests it can. They've known about the switch for a while. |
| 18:51.03 | SteveTotaro | these guys are going to file suit against digium |
| 18:51.14 | SteveTotaro | they have two machines, one cold swap |
| 18:51.15 | x86 | Qwell: yeah I haven't either... |
| 18:51.19 | funxion | doesis it possible to control the g729 payload size in asterisk /1-4 |
| 18:51.24 | russellb | threats on IRC are not going to help |
| 18:51.32 | SteveTotaro | it's not a threat |
| 18:51.36 | x86 | Qwell: i like the whole thing about ability to control echo canceller on a per-channel basis |
| 18:51.49 | russellb | well if you want to help the customer, go call the right people |
| 18:51.52 | SteveTotaro | obviously i can boot to single user mode, creat a user in the root group |
| 18:52.07 | russellb | but if you try to update it, you will _completely_ break it |
| 18:52.08 | SteveTotaro | change ssh back to 22 |
| 18:52.53 | SteveTotaro | http://bugs.digium.com/view.php?id=4101 so how to i fix this bug in switchvox? |
| 18:53.03 | Qwell | SteveTotaro: call support |
| 18:53.14 | SteveTotaro | do i download the latest free version and then install tokens or something? |
| 18:53.51 | russellb | sighs |
| 18:54.14 | SteveTotaro | man switchvox really went downhill |
| 18:54.17 | waverly360 | Wait...I can't use DAHDI with Sangoma's Wanpipe drivers? |
| 18:54.25 | SteveTotaro | it was great a couple of years ago |
| 18:54.26 | Qwell | waverly360: You'll have to ask Sangoma. |
| 18:54.30 | wtsexton | this is odd calls only drop when getting to background menu |
| 18:54.51 | SteveTotaro | answer them first sexton |
| 18:55.04 | jaytee | wtsexton, check what format your sound files are and what the preferred codecs on the phones are. |
| 18:55.45 | wtsexton | should be g711 ulaw |
| 18:55.53 | wtsexton | I'll start there |
| 18:55.55 | jaytee | wtsexton, and what SteveTotaro said about using Answer() |
| 18:56.06 | SteveTotaro | if you are using switchvox you are probably dropping frames |
| 18:56.36 | funxion | does anyone know if the packetization option in sip.conf works or not? |
| 18:56.43 | russellb | SteveTotaro: are you charging a customer for your time ranting in here about it? |
| 18:56.44 | russellb | heh |
| 18:57.08 | SteveTotaro | i am attempting to fix what digium cannot |
| 18:57.19 | hardwire | SteveTotaro: lies |
| 18:57.28 | wtsexton | ah I don't see an answer() before the menu plays |
| 18:57.38 | jaytee | wtsexton, then add one |
| 18:57.40 | hardwire | You're attempting to fix what Digium can't focus on. |
| 18:57.46 | Qwell | SteveTotaro: I'm going to say this one more time - and only one more time. Call support. |
| 18:57.47 | SteveTotaro | ok, then why would a customer with lotsa tokens contact me |
| 18:57.49 | wtsexton | I shall, thanks |
| 18:59.58 | funxion | has anyone seen my question? |
| 19:00.01 | funxion | does anyone know if the packetization option in sip.conf works or not? |
| 19:00.20 | wtsexton | still drops even with an answer |
| 19:00.54 | jaytee | wtsexton, and never plays the file? |
| 19:01.07 | wtsexton | no it plays the file, half way then drops |
| 19:01.26 | *** join/#asterisk dmz (n=dmz@12.25.86.34) |
| 19:01.28 | jaytee | any error on the CLI when it does that? |
| 19:01.33 | wtsexton | yes |
| 19:01.51 | jaytee | oh, right, the packet retransmit errors |
| 19:01.52 | waverly360 | Qwell: I just sent an email to the techdesk there. They're usually pretty good about responding quickly, so I should have an answer soon... |
| 19:02.00 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:02.00 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 19:02.07 | wtsexton | yes, oddly it only does it on the menu, not on calls that are picked up |
| 19:02.21 | Qwell | waverly360: I'd be interested in hearing the answer.. I know we're going to get that question a lot in the near future |
| 19:02.34 | waverly360 | Qwell: I'll let you know. |
| 19:03.48 | waverly360 | On another note, I'm curious to hear how some people handle dialing restrictions for certain users to prevent them from dialing long distance or international. In order to keep things simple, I planned to have some logic that prevented users from dialing numbers that contained more than X number of digits. Does anyone see any problems with this? |
| 19:04.15 | Carlos_PHX | We have multiple outbound contexts. |
| 19:04.27 | Carlos_PHX | We do an include for each context we want to enable. |
| 19:04.48 | Carlos_PHX | include => pstn-9-toll |
| 19:04.53 | Carlos_PHX | include => pstn-9-international |
| 19:05.51 | Carlos_PHX | Calling features are in yet another context. Premium features in another, so they don't get them if they don't pay. |
| 19:06.15 | thehar | yay russellb ! |
| 19:06.29 | russellb | :) |
| 19:06.29 | Kobaz | exten => _9XXXXXXXXXX.... |
| 19:06.34 | Kobaz | match 9 and then 10 digits |
| 19:06.40 | Kobaz | extra |
| 19:06.45 | thehar | wait. asterisk.org still says 1.6.0-rc6 not 1.6.0 |
| 19:06.48 | Kobaz | and make more rules for international and whatnot |
| 19:07.15 | waverly360 | Carlos_PHX: So how are you allowing one user to dial long distance, but not another? |
| 19:07.16 | jaytee | Carlos_PHX, that's basically how I'm handling mine. Users get local outbound by default but everything else is an add in option based on user/department |
| 19:07.16 | Qwell | looks at russellb |
| 19:07.21 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:07.25 | russellb | Qwell: hm? |
| 19:07.31 | Qwell | asterisk.org :D |
| 19:07.34 | thehar | looks at russellb |
| 19:07.36 | russellb | working on it! |
| 19:07.37 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 19:07.42 | thehar | 1.6! |
| 19:07.52 | Kobaz | 1.500 |
| 19:07.53 | Kobaz | ! |
| 19:08.01 | Kobaz | oh yeah |
| 19:08.04 | thehar | released :) |
| 19:08.07 | Kobaz | so is there a "what's new in 1.6" article yet |
| 19:08.11 | *** join/#asterisk jetlagmk2 (i=jetlag@70.17.37.238) |
| 19:08.25 | thehar | uhm. Changelog? |
| 19:08.31 | Kobaz | uhh |
| 19:08.39 | thehar | or upgrade.txt |
| 19:08.43 | anonymouz666 | waiting for [TK]D-Fender to install the Asterisk 1.6 |
| 19:08.45 | Kobaz | yeah, i'll go read through 20 1meg changelogs |
| 19:08.48 | russellb | Kobaz: I have some things on russellbryant.net ... but also read the CHANGES file |
| 19:09.01 | russellb | the CHANGES file is a list of new features |
| 19:09.04 | Kobaz | k |
| 19:09.11 | russellb | read the announcement :-p |
| 19:09.48 | Kobaz | yeah i didnt even look at the home page when i asked |
| 19:09.48 | Kobaz | heh |
| 19:09.49 | jaytee | wait! what? new features? in a friggin release candidate? anybody ever heard of feature freezes? |
| 19:10.08 | Kobaz | jaytee: i meant new features in 1.6 overall, i haven't really kept track of anything |
| 19:10.09 | russellb | release candidates did not get new features ... |
| 19:10.22 | jaytee | whew! |
| 19:11.07 | voxter | hey congrats russell, and everyone else at digium :) |
| 19:11.28 | Qwell | voxter: now, send us a million dollars. |
| 19:11.29 | Qwell | each |
| 19:11.38 | russellb | voxter: thanks much :) |
| 19:11.41 | jaytee | cuz I know there are some companies where the poor bastard doing all the programming has some vapid, leggy blonde that can't type come up to him and say "Um, I know you're almost done and getting ready to ship your thingy but the boss just called and said he wants "Floating Menus". |
| 19:11.57 | russellb | 1.6.0 is almost 2 years of new development ... lots of new features, and _lots_ of good stuff under the hood |
| 19:11.58 | russellb | <3 |
| 19:12.11 | voxter | Qwell: one MIL-lion dollars. muuwahahaha |
| 19:13.37 | thehar | contemplates 1.6 for production and snickers. |
| 19:14.14 | russellb | thehar: depends on what "production" means to you :) |
| 19:14.40 | thehar | uhm 83 phones and 4 pri mostly maxed 100% of the time. |
| 19:14.52 | Qwell | thehar: yeah, it'll work fine for that ;) |
| 19:14.52 | russellb | should be fine! |
| 19:14.56 | thehar | haha |
| 19:15.02 | thehar | and INTENSE queueing |
| 19:15.11 | russellb | pfft, just update overnight |
| 19:15.12 | russellb | it'll be fine |
| 19:15.18 | Qwell | overnight? |
| 19:15.19 | zoid_99 | hah.. INTENSE queueing |
| 19:15.23 | Qwell | that's what "restart now" was added for |
| 19:15.25 | thehar | snickers |
| 19:15.28 | x86 | w00t w00t... 1.6.0 is stable now? |
| 19:15.38 | russellb | x86: it is released anyway :) |
| 19:15.46 | x86 | hehe |
| 19:15.56 | russellb | it's to the point where we are happy for people to start using it |
| 19:15.56 | x86 | should i wait for a while before deploying it into production? |
| 19:16.05 | funxion | does anyone know the proper characters to send when trying to telnet to manager api to login? |
| 19:16.05 | Nugget | telnet is eeeeeeevil! |
| 19:16.11 | x86 | oh wait, i can't deploy it anyway... sangoma cards don't work with dahdi yet |
| 19:16.18 | russellb | x86: pwnt |
| 19:16.30 | Qwell | x86: waverly360 is looking into it |
| 19:16.39 | x86 | Qwell: who's that? |
| 19:16.42 | x86 | russellb: hah |
| 19:16.43 | Qwell | dunno |
| 19:16.46 | thehar | russellb: cd /usr/src && wget http://downloads.digium.com/pub/asterisk/asterisk-1.6.0.tar.gz && tar zxvf asterisk-1.6.0.tar.gz && cd asterisk-1.6.0.tar.gz && ./configure && make && make install |
| 19:16.47 | x86 | Qwell: lol |
| 19:16.49 | Qwell | but he's looking into it :p |
| 19:16.50 | thehar | done and done |
| 19:17.00 | Qwell | thehar: && asterisk -rx "restart now" |
| 19:17.05 | thehar | hehe |
| 19:17.14 | russellb | you might want to read UPGRADE.txt, too |
| 19:17.18 | russellb | but ... that might be too logical |
| 19:17.21 | thehar | 1.0 to 1.6 done and done |
| 19:17.37 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 19:17.51 | x86 | 1.0 -> 1.6.... wow ;) |
| 19:18.10 | thehar | oh hai x86 |
| 19:18.13 | x86 | that's like days of dialplan fixes, sip.conf fixes, iax.conf fixes, and god only knows what else :P |
| 19:18.16 | x86 | HAI! |
| 19:18.21 | russellb | UPGRADE-1.2.txt, UPGRADE-1.4.txt, and UPGRADE.txt, then |
| 19:18.31 | thehar | nonsense. |
| 19:18.41 | Qwell | just do it. it'll totally work without any problems |
| 19:18.48 | x86 | I remember a bunch of hassle going from 1.0 to 1.2 |
| 19:18.50 | thehar | haha |
| 19:18.52 | x86 | Qwell: hahaha |
| 19:18.52 | lmadsen | doesn't the release notice ummm... state to read the UPGRADE.txt? |
| 19:19.04 | thehar | lief again non-sense |
| 19:19.24 | thehar | s/lief/leif/ |
| 19:19.55 | funxion | Im trying to login to manager api via telnet and am unable to authenticate I'm sending Action: login\r |
| 19:19.55 | funxion | Username: mark\r |
| 19:19.55 | funxion | Secret: 4st3r1sk\r |
| 19:20.17 | Qwell | \r\n |
| 19:20.37 | funxion | thnx |
| 19:20.50 | funxion | I still get Action: login\r\n |
| 19:20.50 | funxion | Username: mark\r\n |
| 19:20.51 | funxion | Secret: 4st3r1sk\r\n |
| 19:20.54 | funxion | sery |
| 19:20.58 | funxion | Response: Error |
| 19:20.58 | funxion | Message: Authentication Required |
| 19:20.59 | Qwell | and another \r\n at the end |
| 19:21.07 | NovceGuru | is there a ~commercial asterisk appliances with support command ? :P |
| 19:21.31 | *** join/#asterisk hubguruJR (n=hubguruJ@ntegratedoffice.ntegratedsolutions.com) |
| 19:21.43 | funxion | Qwell i sent |
| 19:21.43 | funxion | Action: login\r\n\r\n |
| 19:21.43 | funxion | Username: mark\r\n\r\n |
| 19:21.43 | funxion | Secret: 4st3r1sk\r\n\r\n |
| 19:21.51 | Qwell | I said at the end |
| 19:21.54 | funxion | i got Response: Error |
| 19:21.55 | funxion | Message: Authentication Required |
| 19:21.55 | lmadsen | only one CR after each line |
| 19:21.59 | lmadsen | then 2 CR after the last command |
| 19:22.05 | Qwell | CRLF* |
| 19:22.12 | lmadsen | CRLFFU |
| 19:22.16 | thehar | oh DAHDI |
| 19:22.16 | waverly360 | NovceGuru: Are you asking if there are asterisk appliances out there you can buy? |
| 19:22.25 | funxion | sry misunderstood |
| 19:22.37 | voxter | hmmm, te110p's are kinda crappy arent they |
| 19:22.48 | NovceGuru | waverly360: looking for a list of them with some intial/small reviews |
| 19:23.02 | *** join/#asterisk LemensTS (n=matthew@adsl-70-238-160-24.dsl.stlsmo.sbcglobal.net) |
| 19:23.08 | NovceGuru | I know there are tons, as google has turned up, but wondered of the opinions of the channel |
| 19:23.28 | LemensTS | what wireless headset do you guys reccomend? |
| 19:24.17 | *** join/#asterisk hfb (n=hfb@pool-96-247-116-5.lsanca.dsl-w.verizon.net) |
| 19:25.26 | funxion | omg |
| 19:25.35 | funxion | Qwell check this |
| 19:25.36 | funxion | Action: login\r\n |
| 19:25.36 | funxion | Username: mark\r\n |
| 19:25.36 | funxion | Secret: 4st3r1sk\r\n\r\n |
| 19:25.36 | funxion | Response: Error |
| 19:25.36 | funxion | Message: Authentication Required |
| 19:26.03 | LemensTS | Nice password |
| 19:26.07 | *** join/#asterisk YoYo (n=chatzill@12.196.144.39) |
| 19:26.19 | funxion | im testing with default |
| 19:26.28 | Qwell | there is no default |
| 19:26.36 | YoYo | any pointers on why a 7940 won't register with asterisk 1.2? says icmp unreachable, but I can ping the * box all day |
| 19:26.38 | LemensTS | u set it up in manager.conf right? |
| 19:26.43 | funxion | yes |
| 19:27.03 | funxion | it was an old sample config file that I uncommented |
| 19:27.16 | Qwell | and where did you get that password from? |
| 19:27.26 | funxion | astmantest will connect to it using those credentials |
| 19:27.35 | funxion | but I cannot telnet for some reason |
| 19:27.54 | *** join/#asterisk Siya (n=djerk@194.60.207.239) |
| 19:27.59 | *** part/#asterisk Siya (n=djerk@194.60.207.239) |
| 19:27.59 | funxion | just want to login to monitor and parse events |
| 19:29.07 | funxion | am I passing the credentials wrong? |
| 19:29.27 | funxion | I did this like 3 years ago and have forgotten the correct syntax |
| 19:30.14 | thehar | google? |
| 19:30.28 | funxion | um yeah |
| 19:30.30 | funxion | did that |
| 19:30.52 | funxion | followed what I found and it doesnt werk |
| 19:31.02 | *** join/#asterisk GlobeTrotter (n=eric@196.40.26.99) |
| 19:31.05 | Damin | Hmm.. |
| 19:31.14 | Damin | Anyone ever done any work w/ Cisco CUBE? |
| 19:31.20 | funxion | I have |
| 19:31.25 | funxion | I have 8 of them |
| 19:31.28 | *** join/#asterisk mike345 (n=mike_sim@64.74.198.10) |
| 19:31.30 | funxion | gonna get 5 more |
| 19:31.46 | Damin | funxion: So.. you wanna help debug an issue? |
| 19:32.07 | Damin | funxion: Client has 1 way audio.. I am sending RTP back to the Cube, but it isn't passing it through to the phones.. |
| 19:32.40 | funxion | ur using it as call manager? |
| 19:32.51 | Damin | funxion: Asterisk 1.2 on my side.. Cisco 3845 on his side.. running IOS 12.4.20T |
| 19:33.09 | funxion | wihch ios exactly |
| 19:33.11 | Damin | funxion: Separate Call manager box sitting behind CUBE.. |
| 19:33.19 | funxion | ok easier |
| 19:33.57 | funxion | so ur sending call from phone to * to cube to CMEcorrect? |
| 19:34.01 | *** join/#asterisk Eduardo_Assis (n=Eduardo_@201-13-199-187.dial-up.telesp.net.br) |
| 19:34.15 | Damin | funxion: No.. SCCP Phone to Call Manager -> CUBE -> Asterisk |
| 19:34.42 | funxion | is it sccp all the way through? |
| 19:34.44 | Damin | funxion: Call setup works great, but audio (RTP) never seems to be passing through the Cube back to the phone / call mangler.. |
| 19:35.03 | Damin | funxion: Nope.. |
| 19:35.06 | funxion | sip? |
| 19:35.17 | Damin | funxion: CUBE is registering as a SIP endpoint to my * box.. |
| 19:35.19 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 19:35.27 | funxion | do you have ip routing enabled? |
| 19:35.36 | Damin | funxion: I'll ask.. :) |
| 19:35.44 | [TK]D-Fender | Yup... so we've finally hit * 1.6.0 full release... |
| 19:35.45 | Damin | funxion: "I believe so, yes" |
| 19:36.02 | jaytee | it's official? whoohooo!!!! |
| 19:36.10 | Damin | ROCK! |
| 19:36.15 | Damin | 1.6.1 shortly to follow! |
| 19:36.31 | funxion | while it does not appear in the config have him isse "enable ip routing" in config mode |
| 19:36.50 | Damin | funxion: Will do.. |
| 19:36.53 | *** part/#asterisk Eduardo_Assis (n=Eduardo_@201-13-199-187.dial-up.telesp.net.br) |
| 19:37.14 | funxion | if its already enabled its no biggie but if not it shouuld start working |
| 19:37.56 | funxion | [TK]D-Fender do you know the correct syntax to login to * manger api by telnetting to port 5038? |
| 19:38.13 | funxion | I'm trying |
| 19:38.13 | funxion | Action: login\r\n |
| 19:38.14 | funxion | Username: mark\r\n |
| 19:38.14 | funxion | Secret: 4st3r1sk\r\n\r\n |
| 19:38.20 | [TK]D-Fender | funxion: Its in the book, WIKI, and a few other places../.. |
| 19:38.42 | funxion | just tell me if my syntax is wrong |
| 19:39.06 | [TK]D-Fender | funxion: I don't know it by heart |
| 19:40.08 | Damin | funxion: ip routing is enabled. They have tons of other routes being used.. |
| 19:40.19 | Damin | funxion: show ip route returns a bunch of crap.. |
| 19:40.43 | funxion | are they using it as an ip to ip gateway with other gateways? |
| 19:40.50 | Damin | funxion: They are using EIGRP internally for their other routers.. |
| 19:41.03 | wtsexton | looking at the debug log the only time I'm getting the transmission error is during playback using background |
| 19:41.04 | Damin | funxion: You mean SIP gateways? Or IP routing gateways? |
| 19:41.10 | funxion | SIP |
| 19:41.24 | jameswf | I should put an Ad on craigslist... call my grandpa hes lonely |
| 19:41.31 | thehar | snickers |
| 19:41.34 | Damin | funxion: The only thing might be Meeting Place Express.. |
| 19:42.44 | funxion | ? |
| 19:42.57 | Damin | funxion: SIP based meetme crap I believe... |
| 19:43.12 | Damin | funxion: It's a separate server in the Cisco fashion.. |
| 19:43.25 | Damin | funxion: but it is on the same subnet, so it's not really routed... |
| 19:44.17 | funxion | what codec are you using? |
| 19:44.22 | wtsexton | jaytee, sorry I had to step away, did you have an idea where I should lock? |
| 19:44.25 | wtsexton | look I mean |
| 19:45.40 | funxion | Damin? |
| 19:46.46 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 19:47.52 | Assid | are the logs in asterisk (cdr-csv) default for UTC? |
| 19:49.18 | wtsexton | hold up, I'm not issuing an answer |
| 19:49.29 | jaytee | wtsexton, you're using Background()? |
| 19:49.33 | wtsexton | yea |
| 19:50.00 | wtsexton | I moved the Answer(), I think I may have fixed it |
| 19:50.27 | wtsexton | I was trying to ring a few sip phones and if they didn't pick up I'd answer and send to the menu |
| 19:50.38 | wtsexton | but I changed it to answer then ring, then transfer to menu if they didn't pick up |
| 19:51.43 | zerko | anyonee here familiar with festival? |
| 19:51.49 | wtsexton | does that make sense? |
| 19:52.09 | jaytee | yep |
| 19:52.15 | Kobaz | Added a new dialplan application, Bridge, which allows you to bridge the calling channel to any other active channel on the system. |
| 19:52.22 | Kobaz | wow that's a really cool new feature in 1.6 |
| 19:52.33 | Nugget | that's slick |
| 19:52.45 | jaytee | Kobaz, really? |
| 19:53.04 | jaytee | russellb!!!!!!!! is he for real? |
| 19:53.28 | Kobaz | fo reels yo |
| 19:53.36 | Nugget | yeah, there it is. |
| 19:53.43 | Nugget | Usage: Bridge(channel[,options]) |
| 19:53.49 | Nugget | Allows the ability to bridge two channels via the dialplan. |
| 19:54.10 | wtsexton | I've got a good feeling that fixed it |
| 19:57.19 | Kobaz | whew, finished reading the changes file |
| 19:57.22 | Kobaz | that took a while |
| 19:58.33 | [TK]D-Fender | I'll be upgrading at home over the weekend |
| 19:58.50 | Kobaz | should be fun |
| 19:59.06 | *** join/#asterisk Xaviertoor (i=Meu@189-015-136-146.xd-dynamic.ctbcnetsuper.com.br) |
| 19:59.28 | Xaviertoor | anybody use dahdi_hardware |
| 19:59.29 | Xaviertoor | pci:0000:00:09.0 wcfxo+ e159:0001 Wildcard X101P |
| 20:00.11 | jaytee | Bridge might solve an issue I have where I want to add recorded audio or MOH to a Page() call and Page() doesn't support MOH or playing a sound file |
| 20:00.27 | Kobaz | yeah. all kinds of stuff |
| 20:00.38 | Kobaz | i wonder if you can just keep bridging |
| 20:00.47 | Kobaz | like bridge a and b, and then tack on c |
| 20:00.50 | Kobaz | without the need for a meetme |
| 20:01.30 | Kobaz | because with meetme, if you're the only person left, you're still on the call |
| 20:01.46 | Kobaz | if it's all bridged calls, the second to last channel hangs up, and then the last one will get booted as well |
| 20:03.10 | wtsexton | jaytee, thank you for the help |
| 20:03.29 | wtsexton | gotta answer the phone before I start shoving audio at the metaswitch |
| 20:03.59 | thehar | oooh metaswitch/ |
| 20:04.04 | thehar | wtsexton: 3510? |
| 20:04.57 | wtsexton | yea |
| 20:05.13 | thehar | mmmm we're puchasing one next year |
| 20:05.40 | thehar | 5000 cards or 4500 ? |
| 20:06.00 | *** join/#asterisk teh_recon (n=scdds@mail.imprinters.com) |
| 20:07.51 | *** join/#asterisk tkbeat (n=tk@p54B9778B.dip.t-dialin.net) |
| 20:07.57 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:08.17 | wtsexton | no clue, I don't manage it :) |
| 20:08.33 | jaytee | we're getting one of these next year: Uber Ethernet-Paketvermittlung-Fräser Schnitzelhersteller MK-101 |
| 20:09.45 | wtsexton | sounds exciting, it come with quad sarcasm multiplixers? |
| 20:09.51 | jaytee | yep |
| 20:09.55 | wtsexton | nice |
| 20:10.28 | jaytee | ya gotta hand it to them German engineers. If anybody can make a high quality packet switch/router/schnitzel making machine then they can |
| 20:11.49 | Carlos_PHX | Mmmm...Asterisk-controlled schnitzel machine. |
| 20:12.35 | waverly360 | So I've found a site that gives me a list of all of the dialing codes of the world: country code, idd, and ndd here: http://www.kropla.com/dialcode.htm but I was wondering if there's a more official site where I can grab this information a bit more systematically so that I can integrate it into my dialplan whenever a country code changes. My followup question to that is, do these codes change often enough that I should even be worried about it? |
| 20:12.39 | jaytee | it may sound absurd but it works way better than that damn Trixbox donut machine that came out last year |
| 20:12.43 | wtsexton | press one now for wiener schnitzel |
| 20:12.59 | tzanger | 1111111!!!!1111!111ONE111!!111111 |
| 20:13.20 | tzanger | country codes change?! |
| 20:13.39 | waverly360 | tzanger: well..that's my question |
| 20:13.51 | unpaidbill | oh my goooood 1.6!!! |
| 20:13.59 | unpaidbill | high five dudes! thanks! |
| 20:14.06 | Carlos_PHX | waverly360: You should use the lists provided by your termination provider. |
| 20:15.29 | waverly360 | Carlos_PHX: Well, I have a lot of customers using different carriers...are you saying that the lists can be different from one provider to the next? |
| 20:19.43 | Carlos_PHX | I guess it depends on what you're trying to accomplish. |
| 20:20.13 | Carlos_PHX | We don't actually change the dialplan, we use the rate centers to price calls for billing. |
| 20:20.25 | Katty | hmm. |
| 20:20.30 | Katty | i feel like mexican for dinner. |
| 20:20.38 | Katty | guacamole. fajitas.... |
| 20:20.53 | Carlos_PHX | I was hoping you weren't talking about your gardener. |
| 20:20.58 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:21.00 | jameswf | heh |
| 20:21.05 | *** join/#asterisk arpu (n=arpu@chello084114184079.13.15.vie.surfer.at) |
| 20:21.07 | Katty | my 'gardener' is at work right now. |
| 20:21.14 | jameswf | in Arizona all food is prepaired by mexicans |
| 20:21.18 | Katty | he doesn't cook mexican. |
| 20:21.27 | Katty | going /out/ for mexican |
| 20:21.39 | jameswf | to home depot? |
| 20:21.42 | jameswf | :)) |
| 20:21.42 | Carlos_PHX | jameswf: Strange to see a Mexican sushi chef, isn't it? |
| 20:21.44 | jameswf | sorry |
| 20:21.57 | Katty | for all you know, i'm partially mexican. |
| 20:22.01 | Katty | and you've just insulted me. |
| 20:22.06 | Katty | now how do you feel? |
| 20:22.11 | jameswf | I always see mexicans cooking chinese never see chinese people cooking mexican |
| 20:22.24 | anonymouz666 | hi Katty |
| 20:22.28 | Katty | hihi |
| 20:22.30 | jaytee | tiene usted una tarjeta del verde? |
| 20:22.37 | Carlos_PHX | We have a Chinese-Mexican restaurant here. Makes an interesting combo. |
| 20:23.07 | anonymouz666 | Katty: do you like tacos and burritos? |
| 20:23.11 | Katty | sounds interesting. |
| 20:23.12 | wtsexton | I went to a Hibachi steak house and the cook was mexican, it was entertaining |
| 20:23.18 | Carlos_PHX | jaytee: You should omit the "del" from that. Then again, sounds funnier that way. |
| 20:23.19 | Katty | anonymouz666: yep (= |
| 20:23.28 | Katty | anonymouz666: refried beans are the best. |
| 20:23.31 | Katty | anonymouz666: i could eat them plain. |
| 20:23.34 | citywok | in a dialplan i Dial() a phone number, and then i want to Monitor() it, but it never gets to monitor because its sitting on dial. how cna i fix this, so that it records the call that it dials? |
| 20:23.57 | Carlos_PHX | citywok: Record before dialing. |
| 20:24.23 | jaytee | makes confused Scooby sounds |
| 20:24.32 | Carlos_PHX | Me too |
| 20:24.41 | citywok | i was considering that, but wasnt sure, and i dont want to break everything lol |
| 20:24.46 | SteveTotaro | wow switchvox tech support is so quick to say it is the network or the t1 |
| 20:24.53 | Katty | if it breaks, just put it back the way it was ;) |
| 20:25.09 | Carlos_PHX | Any time you modify your dialplan you're on the verge of breaking everything no matter what, so test on a non-production system. |
| 20:25.48 | citywok | i'll just take the live system out of production for an hour and try it, gotta wait for 20 calls to end first though. |
| 20:26.05 | Katty | not all of us have that luxury (= |
| 20:26.11 | Carlos_PHX | If you unload Asterisk, then you won't have to wait for the calls to end. |
| 20:26.22 | citywok | haha, then i'd have a call center full of pissed off people :-) |
| 20:26.31 | jaytee | given the current state of technology today it is very possible that had he existed in the present then all the King's horses and all the King's men could have put Humpty Dumpty back together again. |
| 20:26.35 | wtsexton | no, you'd have a call center full of happy people |
| 20:26.35 | Katty | indeed you would ;) |
| 20:27.23 | wtsexton | I'm sure if I unplugged the pbx and told tech support, no calls for a while, they'd be happy :) |
| 20:28.28 | jaytee | just route your help desk calls to Dell |
| 20:29.12 | jaytee | use the 800 number for their Vostro line of equipment and shunt the calls to the Phillipines |
| 20:29.13 | wtsexton | we want to stay in business |
| 20:29.42 | jaytee | yeah, that could be a sticking point with the customer |
| 20:30.23 | Katty | plots placing an order for delivery here at work. |
| 20:31.02 | jaytee | wow, if I worked someplace where I had to plot placing an order I'd find another job |
| 20:31.21 | Katty | that just means i'm thinking about it |
| 20:31.28 | Katty | am hungry :< |
| 20:32.24 | wtsexton | I just had nasty bell, had to eat something |
| 20:32.27 | jaytee | I'm afflicted yet again with the "enigmatic hankering" |
| 20:33.00 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 20:33.17 | Katty | i crave quacamole regularly for unknown reasons. |
| 20:33.19 | Carlos_PHX | She's going to write a dialplan to place the order. |
| 20:33.30 | Carlos_PHX | That way you can avoid speaking to humans. |
| 20:33.56 | Carlos_PHX | Mmmm...thinks about the packages of guacamole in the fridge a few feet away. |
| 20:34.13 | Katty | maybe i lack vitamin k |
| 20:34.25 | *** part/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 20:34.37 | *** part/#asterisk hubguruJR (n=hubguruJ@ntegratedoffice.ntegratedsolutions.com) |
| 20:36.30 | Katty | ah HA! |
| 20:36.45 | Katty | i have conned the boyfriend into taking me out to dinner at the mexican place i love ^_^ |
| 20:37.18 | *** join/#asterisk hi365_m (n=hi365@213.151.59.254) |
| 20:37.33 | wtsexton | ah |
| 20:37.49 | *** join/#asterisk soulfreshner (n=derick@dsl-243-4-106.telkomadsl.co.za) |
| 20:38.31 | angryuser | with the latest 1.4.22 do i need to rename all Zap to dahdi to call them ? ;) |
| 20:40.10 | Carlos_PHX | Wow, the Nigerians are getting cheap. I just got the usual scam letter but only asking for $99. |
| 20:40.42 | wtsexton | they know we're broke |
| 20:40.49 | Carlos_PHX | < Adds their phone number to the "call randomly in the middle of the night" list. |
| 20:41.10 | Carlos_PHX | wtsexton: Not sure whether to laugh or cry. |
| 20:41.13 | YoYo | does anyone know of, or have, a patch that will convert voicemail to MP3 format before emailing? |
| 20:41.59 | wtsexton | well, I don't have $99, I can ask my wife tho |
| 20:42.04 | tzafrir_laptop | angryuser, no. it can be used with both zap and dahdi |
| 20:42.26 | angryuser | tzafrir_laptop : thanks reading it now |
| 20:42.41 | Carlos_PHX | YoYo: You can use Sox and a script, don't know if there's a quicker way. |
| 20:43.08 | soulfreshner | why would a zap channel automatically disconnect after a time? |
| 20:44.28 | SteveTotaro | because it was not answered |
| 20:45.05 | *** join/#asterisk whymarkwh (n=dsfsdfsd@196.211.34.2) |
| 20:45.09 | *** part/#asterisk whymarkwh (n=dsfsdfsd@196.211.34.2) |
| 20:45.19 | Carlos_PHX | Ah, it's always interesting when people first learn that you can run voice on an unanswered channel. |
| 20:45.20 | angryuser | tzafrir_laptop : i cant find asterisk_safe script, it is not packet with sources ? |
| 20:45.24 | *** join/#asterisk darkskiez (n=mbryars@ip03.contempt.adsl.gxn.net) |
| 20:45.29 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
| 20:45.33 | angryuser | packed* |
| 20:45.35 | tzafrir_laptop | angryuser, safe_asterisk ? |
| 20:45.47 | angryuser | tzafrir_laptop : or safe_asterisk |
| 20:46.00 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 20:46.30 | apocn | is there a way to change the menu of the VoiceMailMain? |
| 20:46.32 | tzafrir_laptop | in contrib/scripts |
| 20:46.35 | angryuser | tzafrir_laptop : ah it's in contrib |
| 20:46.46 | hi365_m | tzafrir_laptop: sup? |
| 20:47.02 | tzafrir_laptop | apocn, one way is to give it up and use minivm |
| 20:47.06 | hi365_m | are there any varibales set that tell you how long a call was holdong/talking in a queue? |
| 20:47.07 | tzafrir_laptop | hi365_m, hi |
| 20:47.11 | *** join/#asterisk WimpMan (n=wimpy@gw.fl.yeti.dk) |
| 20:47.17 | hi365_m | shana tova! |
| 20:47.28 | WimpMan | Salvete! |
| 20:47.52 | angryuser | tzafrir_laptop : so basicly it restarts asterisk if it crashes, no more ? |
| 20:48.28 | tzafrir_laptop | hi365_m, ${CDR(duration)} ? |
| 20:48.43 | tzafrir_laptop | angryuser, basically |
| 20:48.44 | hi365_m | hmm, bit doesnt that include the hold/wait time? |
| 20:49.30 | hi365_m | *but |
| 20:49.40 | codefreeze-lap | hi365_m: how about duration minus billsec ? |
| 20:49.47 | apocn | tzafrir_laptop: do you have a good link for reading about it? |
| 20:50.31 | hi365_m | codefreeze-lap: maybe just bill seconds (i want the talk time) |
| 20:50.50 | tzafrir_laptop | apocn, I think that there's a doc about it in the doc/ directory |
| 20:50.55 | codefreeze-lap | hi365_m: nothing specific about hold time, and billsec will include hold time, I'm sure. |
| 20:51.19 | hi365_m | DOES NOT want hold time - just talk time |
| 20:53.22 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 20:55.43 | apocn | tzafrir_laptop: which domain? asterisk.org/doc ? |
| 20:56.37 | *** join/#asterisk bradleyprice861 (n=bradleyp@fw.datafax.net) |
| 20:56.42 | tzafrir_laptop | http://svn.digium.com/svn/asterisk/tags/1.4.21.2/doc |
| 20:57.02 | tzafrir_laptop | (or the specific version you use) |
| 21:00.05 | *** join/#asterisk dlynes (n=dlynes@S01060016b68219f1.vs.shawcable.net) |
| 21:00.49 | apocn | tzafrir_laptop: thanks a LOT |
| 21:02.03 | *** join/#asterisk stimpie (n=stimpie@84-104-5-14.cable.quicknet.nl) |
| 21:07.36 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:11.09 | *** join/#asterisk flush (n=SYN_ACK@ip216-239-68-3.vif.net) |
| 21:12.44 | dlynes | So, what exactly has changed between 1.4 and 1.6 as far as the release cycle goes? |
| 21:12.52 | dlynes | 1.4 wasn't feature frozen, either |
| 21:13.58 | dlynes | With 1.6, did the SIP engine get rewritten yet? |
| 21:15.26 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 21:16.10 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 21:18.44 | jmacz | Greetings, I'm trying to use * 1.6.0 + TLS but when starting or reloading, the CLI shows an "SSL cert error" message. Cert is self-signed (PEM file) generated with openssl (rsa 2048 bits). |
| 21:18.55 | jmacz | Are there some conditions the cert must meet in order for ASterisk to use it? |
| 21:21.09 | bkw_ | don't thinkyou'll want 2048 bit for one |
| 21:21.28 | bkw_ | we use 1024 on our certs |
| 21:23.15 | *** join/#asterisk n3hxs (n=HAMming@adsl-70-128-62-214.dsl.ltrkar.swbell.net) |
| 21:24.33 | jmacz | bkw_, so it should work after generating a 1024 bit cert? |
| 21:37.59 | *** join/#asterisk pirulo (n=pirulo@70.56.223.76) |
| 21:38.46 | jmacz | bkw_, no luck with 1024 cert ("openssl genrsa -out asterisk.key 1024", then "openssl rsa -in asterisk.key -out asterisk.pem"). Any other idea? |
| 21:39.42 | bkw_ | jmacz: we don't gen our cert like that |
| 21:41.07 | bkw_ | jmacz: http://svn.freeswitch.org/svn/freeswitch/trunk/scripts/gentls_cert.in |
| 21:41.09 | bkw_ | I use that |
| 21:41.15 | bkw_ | it should be the same way for Asterisk |
| 21:41.25 | jmacz | bkw_, thank you very much |
| 21:41.38 | jmacz | I'll try that :) |
| 21:41.56 | bkw_ | you might wanna see i fyou can whip up a script from that for other Asterisk folks |
| 21:42.02 | bkw_ | I'm sure setting that stuff isn't the easiest |
| 21:42.08 | bkw_ | if you don't have anything to go on in the first place |
| 21:44.29 | angryuser | tzafrir_laptop : hm do i need to launch asterisk safe script in some kind of special way ? when launched manually from console it's ok, but from script on boot, the script launches (asterisk_safe), but does nothing, just sitting there |
| 21:45.52 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 21:47.24 | LemensTS | you dont need zaptel if you use wanpipe right? asterisk will get the timing from the sangoma card ? |
| 21:48.55 | tzafrir_laptop | If you want timing from wanpipe it has to go through zaptel |
| 21:49.35 | LemensTS | oh i see that. wanpipe is a driver for the zaptel module, right? |
| 21:58.36 | angryuser | found it, error in pid dir |
| 22:02.50 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 22:11.45 | *** join/#asterisk darkskiez (n=mbryars@ip03.contempt.adsl.gxn.net) |
| 22:15.44 | LemensTS | . |
| 22:25.33 | *** part/#asterisk stegbth (n=stegbth@mail.a-fk.de) |
| 22:25.33 | *** join/#asterisk sucituanbo (n=john@c-24-21-121-148.hsd1.wa.comcast.net) |
| 22:29.33 | *** join/#asterisk Shotygun (n=thorn@213.31.43.3) |
| 22:29.59 | sah-work | anyone know if asterisknow support sangoma cards |
| 22:32.03 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:59.36 | sah-work | is there any way to fake a zap card so i can setup a test system |
| 22:59.46 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 23:04.14 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-d03f06097d658497) |
| 23:08.19 | *** join/#asterisk Tebi (n=tebi@gw.aller.fi) |
| 23:10.36 | riddlebox | sah-work, nope, but you can get a cheap ATA |
| 23:10.46 | sah-work | what about ztdummy |
| 23:11.07 | sah-work | ah, it is a timer |
| 23:11.11 | sah-work | ok. |
| 23:11.33 | sah-work | i am building a replacement system and want to simulate it before i cut over. |
| 23:11.38 | sah-work | doing in live would be stupid |
| 23:13.41 | riddlebox | it will replace the system you currently have running correct |
| 23:13.51 | *** join/#asterisk mackes (n=root@cpe-76-180-145-138.buffalo.res.rr.com) |
| 23:14.06 | riddlebox | why not just copy all files over, then you will just have to move the zap cards |
| 23:14.43 | *** join/#asterisk stencil (n=stencil@pdpc/supporter/student/stencil) |
| 23:15.00 | mackes | Hey Everyone, |
| 23:15.06 | riddlebox | hey |
| 23:15.25 | mackes | Does anyone know if the AstDB can be moved from 1.2 to 1.4 by just copying it over? |
| 23:16.09 | mackes | I would like to upgrade from 1.2 to 1.4 (on different hardware over a few minutes and not loose my sip registrations |
| 23:22.44 | *** join/#asterisk LiNeTuX_Home (n=LiNeTuX@253.238.95.24.cfl.res.rr.com) |
| 23:30.48 | dlynes | Is anyone running 1.6 yet? |
| 23:31.00 | dlynes | Is it anymore stable than 1.4.0 was? |
| 23:32.06 | *** join/#asterisk StephenF (n=stephen@c-67-188-58-4.hsd1.ca.comcast.net) |
| 23:32.16 | vader-- | 1.2.27 was stable as shit |
| 23:32.25 | vader-- | i am using it now for about 3 years |
| 23:32.29 | mackes | Yeah. 1.2 rocks. |
| 23:32.54 | mackes | I think my org better upgrade before we fall to far behind |
| 23:33.05 | vader-- | you running 1.2 too? |
| 23:33.15 | mackes | I was somewhat mocked at Bootcamp for still running 1.2 |
| 23:33.16 | mackes | yep |
| 23:33.19 | mackes | Love it |
| 23:33.23 | vader-- | ya same here |
| 23:33.29 | vader-- | on debian sarge 3.1 |
| 23:33.30 | vader-- | hehe |
| 23:33.39 | mackes | The Debug commands are better in 1.4 |
| 23:33.52 | mackes | We are running it on CentOS 5.2 |
| 23:33.57 | vader-- | only issue i have with 1.2 is sometimes it screws up the voicemail boxes |
| 23:34.08 | Carlos_PHX | dlynes: I'm running 1.2, 1.4, and 1.6 without stability problems. |
| 23:34.09 | mackes | really, how so? |
| 23:34.09 | vader-- | and leaves the txt files in the boxes and deletes the wavs and gsm files |
| 23:34.14 | *** join/#asterisk sircco (n=sircco@dh207-103-196.xnet.hr) |
| 23:34.16 | StephenF | do most people remove the vertical features and dial plan on the PAP2T when used with asterisk? |
| 23:34.24 | mackes | Hmmmm |
| 23:34.27 | Carlos_PHX | 1.6 is only in limited test though. |
| 23:34.31 | vader-- | so the MWI on the phone is on saying there is a message |
| 23:34.33 | StephenF | it seems like the ATA wants to handle some of the dial plan logic itself |
| 23:34.42 | *** join/#asterisk MindTheGap (n=MindTheG@189.59.202.173) |
| 23:34.46 | sircco | can i emulate deadagi and get duration of call with queue(1000|||||blah.php) ? |
| 23:34.49 | vader-- | when they go to check the message it tries to play it and then boots them out of the voicemail system |
| 23:34.51 | dlynes | Carlos_PHX: do you know if it's using oej's new sip stack, or not? |
| 23:34.55 | mackes | Yeah... do your dialplan in Asterisk--- just have the ATA pass you the call |
| 23:35.09 | mackes | weird |
| 23:35.16 | vader-- | ya |
| 23:35.19 | StephenF | ok, it has all these features setup like *71 for DND and so on |
| 23:35.29 | vader-- | it occurs when someone deletes a voicemail from the system |
| 23:35.34 | mackes | We had not had that issue. |
| 23:35.38 | mackes | Hmm |
| 23:35.40 | StephenF | just gotta figure out how to tell the ATA to pass all calls directly to asterisk |
| 23:35.43 | vader-- | so i have to go into the mailbox every so often and delete the left over txt file |
| 23:35.44 | vader-- | and it's fine |
| 23:35.52 | mackes | We have about 100 mailboxes.. no issues. |
| 23:35.53 | vader-- | haven't figured out what causes it |
| 23:36.06 | vader-- | any of you guys using juniper ssg firewalls? |
| 23:36.23 | mackes | We are about to buy a few in the next few weeks |
| 23:36.27 | mackes | Cisco right now |
| 23:36.30 | vader-- | ssg's? |
| 23:36.42 | *** join/#asterisk n3hxs (n=HAMming@adsl-70-128-62-214.dsl.ltrkar.swbell.net) |
| 23:36.55 | mackes | The large models with intrusion protection, vpn, et |
| 23:37.05 | vader-- | which model do you know? |
| 23:37.09 | mackes | Hmmm |
| 23:37.12 | mackes | One sec |
| 23:37.23 | vader-- | i just bought a cisco asa 5520 |
| 23:37.26 | vader-- | and im not happy with it |
| 23:37.39 | vader-- | im currently using a netscreen 50 |
| 23:38.07 | vader-- | and it has way more features than the asa |
| 23:38.24 | angryuser | is it difficult to manage cisco routers?, i never had a change to work with, frebsd pfsense zeoshell... |
| 23:38.39 | angryuser | freebsd* |
| 23:38.53 | sircco | angryuser: asa is quirky and illogical sometimes, different things from model to model |
| 23:39.05 | mackes | Juniper Networks SSG 300 Series |
| 23:39.25 | angryuser | juniper use webstuff ? |
| 23:39.36 | angryuser | to configure ? |
| 23:39.46 | mackes | Yeah... We run PIX and Cisco VPN concentrators. We are moving to Juniper |
| 23:40.00 | vader-- | mackes are you doing an ssl vpn or all ipsec? |
| 23:40.05 | mackes | Both. |
| 23:40.16 | vader-- | i have a cisco vpn concentrator now 3005 |
| 23:40.23 | vader-- | but it doesn't have a vista ssl client |
| 23:40.26 | mackes | SSL for some Webmail and Web Interanet access, and IPSEC for full access |
| 23:40.27 | vader-- | so we need to move to something else |
| 23:40.42 | mackes | We have 3005's as well |
| 23:40.42 | vader-- | are you going with a juniper ssl vpn? |
| 23:40.54 | angryuser | vader-- : or move vista ;) |
| 23:41.20 | mackes | For some small stuff..... Most users will use IPSEC |
| 23:41.22 | vader-- | mackes are you getting juniper vpn device? |
| 23:42.01 | mackes | CDW is pushing the SSG 300 Series with several modules |
| 23:42.25 | vader-- | what type of connection and how many users do you have behind these firewalls? |
| 23:42.33 | mackes | We need it to protect our credit card environment for PCI compliance |
| 23:42.45 | vader-- | we are pushing a 40/40 Mbps metro ethernet connection with 400 users growing to probably 1000 users in 2-3 years |
| 23:43.02 | mackes | We have a 20 MB Fractional DS3 for Internet Bound connections |
| 23:43.35 | mackes | Our vendors will us the SSG 300 with RSA tokens for access to a protected segment of our network |
| 23:44.50 | mackes | Our normal staff will continue to use Cisco 3002 's to access our normal network segment. |
| 23:45.13 | vader-- | have you ordered these ssg's yet? |
| 23:46.18 | *** join/#asterisk CrazyTux (n=brandon@adsl-75-43-206-81.dsl.lsan03.sbcglobal.net) |
| 23:46.41 | mackes | Not yet... We just got the quote today. End of next week I think we will order |
| 23:47.39 | *** part/#asterisk sircco (n=sircco@dh207-103-196.xnet.hr) |
| 23:47.43 | vader-- | check msg |
| 23:50.06 | mackes | Is your metro Ethernet a Fiber connection with an Ethernet port on the terminating device? |
| 23:51.22 | LiNeTuX_Home | We've got a metro-e w/ethernet port... it's nice. |
| 23:51.35 | LiNeTuX_Home | I can even do TDMoE over it |
| 23:52.57 | *** join/#asterisk Corydon76-dig (i=ten@pdpc/supporter/bronze/Corydon76-home) |
| 23:52.57 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 23:56.28 | vader-- | mackes right now it's going from fiber into a converter then out as copper to a layer 3 switch which does some simple routing right now then into our netscreen 50 |
| 23:56.49 | vader-- | when we get the ssg 520 we are going to take out the switch and the netscreen 50 |
| 23:56.53 | vader-- | go directly into the ssg |
| 23:57.40 | mackes | Neat. |
| 23:58.05 | mackes | We have a similar setup |
| 23:58.41 | vader-- | i have to talk to juniper and make sure everything will work for this setup |
| 23:59.08 | vader-- | right now our metro ethernet comes in on a vlan 199 with a x.x.22.52/30 address |
| 23:59.15 | *** join/#asterisk logicwrath (n=no@c-68-42-253-39.hsd1.mi.comcast.net) |
| 23:59.22 | vader-- | the layer 3 switch does a static route for that |
| 23:59.28 | logicwrath | ~stun |
| 23:59.29 | jbot | rumour has it, stun is that feeling you get when you realise your SIP call actually got through!. Simple Traversal of UDP over NATs, or a client side method to cater to crappy sip servers, or a phaser setting |
| 23:59.37 | vader-- | the switch is x.x.88.1/24 |
| 23:59.52 | vader-- | and our firewall is x.x.88.3/24 |
