| 00:04.37 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 00:04.52 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:04.53 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:17.32 | KNERD | I recompiled asterisk after installing SpanDSP and checking the menu FAX and FAX_SPANDSP..now I am getting a warning after MAKE INSTALL http://pastebin.ca/2164379 |
| 00:19.23 | jpsharp | You're mixing files that were created the first time you built asterisk and when you rebuilt it with fax support. |
| 00:20.03 | KNERD | I still don't get it... |
| 00:20.11 | KNERD | why can't it just write over them |
| 00:21.20 | KNERD | I first built it with 1.8.7.1 |
| 00:21.26 | KNERD | should I go back to that? |
| 00:21.57 | jpsharp | Or delete /usr/sbin/asterisk and /usr/lib/asterisk/modules and redo your make install. |
| 00:22.08 | jpsharp | delete or rename, whichever you prefer. |
| 00:23.25 | KNERD | ouch..won't my voice mail, etc be gone? |
| 00:24.41 | jpsharp | Nope, those are stored in other directories. |
| 00:24.58 | jpsharp | voicemail is stored in /var/spool/asterisk, configs in /etc/asterisk |
| 00:25.02 | KNERD | thank you |
| 00:25.27 | jpsharp | As long as you don't touch those dirs, you'll be good. And "make install" doesn't stomp on those directories if they exist. |
| 00:40.29 | KNERD | yeah., it worked |
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| 00:59.15 | jpsharp | Excellent. |
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| 02:39.20 | KNERD | jpsharp: is Span DSP used in the same way as Asterisk FAx? |
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| 02:41.08 | *** join/#asterisk Yourname` (~whocares@unaffiliated/yourname/x-837320) |
| 02:41.36 | jpsharp | The same way as Asterisk Fax from Digium? |
| 02:41.45 | KNERD | correct |
| 02:42.00 | KNERD | the "fax" command in the CLI is not there |
| 02:42.11 | jpsharp | No, they are different. |
| 02:42.40 | jpsharp | Different dialplan commands and different semantics/syntaxes in those commands. |
| 02:44.36 | KNERD | jpsharp: oh, okay because the SpanDSP docs are basically non-existant |
| 02:47.15 | jpsharp | Here's what I use for sending Faxes using SpanDSP |
| 02:47.18 | jpsharp | http://pastebin.ca/2164407 |
| 02:48.32 | jpsharp | And for receiving |
| 02:48.33 | jpsharp | http://pastebin.ca/2164408 |
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| 02:59.55 | KNERD | thanks, I will have to fix my sending faxed through browser |
| 03:22.42 | KNERD | back later |
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| 03:45.54 | *** join/#asterisk sparkymarkd (~mark@187.157.145.215) |
| 03:47.03 | sparkymarkd | hi.. could someone help me with getting Cisco 7920 wifi phones working with chan_sccp? |
| 03:50.29 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 03:50.59 | jpsharp | im rusty at it, but whats going on? |
| 03:52.16 | sparkymarkd | I've compiled it and installed, port 2000 is open, but I can't seem to get the phones connecting at all |
| 03:53.01 | sparkymarkd | are the firmwares required on the tftp server? |
| 03:53.24 | linocisco | hi. I have one Land line incoming CO line and I want to share it among 3 analog extensions with analog desktop phone like (http://www.headsetzone.com/kx-ts105w.html). what kind of PCI cards I should buy to test with asterisk? |
| 03:54.08 | jpsharp | As long as the firmware is on the phone, it doesn't need to be on the tftp server. |
| 03:55.32 | jpsharp | linocisco: If you're set on using PCI cards, get this one: http://store.digium.com/products.php?category_id=45 |
| 03:55.34 | sparkymarkd | would you know if 7920.40-03-01 firmware would work with sccp?? I bought these phones from ebay, not doing enough research, now I have to do my 'research' the hard way |
| 03:55.56 | jpsharp | linocisco: With one FXO module and 3 FXS modules. |
| 03:56.08 | sparkymarkd | same I'd say.. *if you're set on pci.. I prefer VOIP gateways |
| 03:56.19 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 03:57.02 | jpsharp | I don't know about specific version numbers. I know I had a 7920 working with Asterisk a few years ago. |
| 03:57.05 | p3nguin | sparkymarkd: You need to have the SCCP files on the tftp server. |
| 03:57.05 | sparkymarkd | but I have not been in this for too long, so not sure if I'm just personally leaning toward it |
| 03:57.39 | jpsharp | Ohyeah. Its that damn universal loader stuff. |
| 03:57.39 | sparkymarkd | the SEP<mac>.conf.xml right? |
| 03:57.46 | *** join/#asterisk viro (~viro101@cpe-71-72-174-83.woh.res.rr.com) |
| 03:57.53 | p3nguin | You need the software and the configs. |
| 03:58.11 | jpsharp | That's the config file. You need the software on there to so the universal loader can get it. |
| 03:58.24 | sparkymarkd | would you mind looking at a config file? wil take me a min to paste on pastebin |
| 03:58.44 | sparkymarkd | the cisco software?? |
| 03:58.50 | sparkymarkd | a bit confused there |
| 03:59.18 | p3nguin | When you power up the phone, it will begin looking for the software. |
| 03:59.30 | p3nguin | It will also look for its configs. |
| 03:59.58 | jpsharp | It'll grab a basic config & then load the appropriate firmware off the server. |
| 04:00.56 | jpsharp | Though it really can only load one set since there's only SCCP support on that phone. |
| 04:01.15 | p3nguin | I know it doesn't have SIP, but no MGCP either? |
| 04:01.40 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 04:02.45 | jpsharp | Nope. SCCP only. |
| 04:02.48 | sparkymarkd | I really don't know if it has MGCP?? is there better support in asterisk for that?? just kinda read that cisco has that, but don't remember asterisk supporting it or not |
| 04:03.09 | p3nguin | SIP is the best supported one, but there's no SIP for that phone. |
| 04:03.28 | p3nguin | And apparently there isn't MGCP either, but that's okay... no one uses it with Asterisk anyway. |
| 04:03.37 | jpsharp | A client bought a dozen of them, then dropped them in my lap and said "Make it work". |
| 04:03.41 | sparkymarkd | dang, now my phone says asterisk connected.. |
| 04:03.53 | p3nguin | It's fairly easy to make them work. |
| 04:04.12 | jpsharp | This was back when SCCP was fairly new code. |
| 04:04.12 | sparkymarkd | I've tried so much, and could not get it, so just now I booted up my server, started up the phone and it looks like it might work |
| 04:04.44 | jpsharp | It was me. I scare VOIP systems into working. |
| 04:04.50 | jpsharp | *raaaaar* |
| 04:05.09 | sparkymarkd | LOL.. same thing I tell others when I do tech support |
| 04:05.24 | sparkymarkd | but now when I try to call, it says no manager found, then reconnects.. |
| 04:05.59 | jpsharp | Are you seeing errors on the Asterisk console? |
| 04:06.05 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:06.29 | sparkymarkd | loggin in now |
| 04:07.17 | sparkymarkd | asterisks crashes.. or at least kicks me out |
| 04:08.05 | sparkymarkd | http://pastebin.com/UgH3zbK7 |
| 04:08.17 | sparkymarkd | but it comes right back up |
| 04:11.12 | sparkymarkd | http://pastebin.com/S9358H2j -- that's the start of the asterisk log file - full |
| 04:14.38 | jpsharp | ERROR[3410] sccp_channel.c: SCCP: Can't allocate SCCP channel if a line is not defined |
| 04:14.59 | jpsharp | You're missing some config. |
| 04:15.01 | sparkymarkd | yes, googled it.. can't find others with same error.. |
| 04:15.45 | jpsharp | You have to define a line, then attach that line to the phone. |
| 04:16.06 | p3nguin | It sounds like you defined a devive but not any lines for it to use. |
| 04:18.21 | sparkymarkd | I created the sep<mac>.conf.xml, then created a custom extension in FreePBX with SCCP/121 as the dial |
| 04:18.41 | sparkymarkd | heres my config file: http://pastebin.com/VpCb5Qqk |
| 04:18.41 | p3nguin | sccp.conf <----------- |
| 04:19.02 | sparkymarkd | ^ is the sep000000000.conf.xml |
| 04:19.16 | p3nguin | negative |
| 04:19.19 | p3nguin | oh |
| 04:20.35 | jpsharp | dunno about using SCCP with FreePBX. |
| 04:20.38 | sparkymarkd | this is my sccp.conf http://pastebin.com/yJWDTtbk |
| 04:22.40 | p3nguin | A 7920 addon? |
| 04:22.44 | p3nguin | Never heard of it. |
| 04:22.50 | sparkymarkd | ok, maybe that's my problem.. I thought I had found a site where the directions where to do a custom extension, then make it dial SCCP/{ext} |
| 04:23.27 | p3nguin | We don't know anything about FreePBX here. |
| 04:23.53 | sparkymarkd | ah, yes, I remember changing that.. after nothing worked, I started change other things.. |
| 04:24.37 | sparkymarkd | ok.. well, I'm happy I at least got somewhere.. up till now I was in the darkness about it.. |
| 04:24.43 | jpsharp | Huh. That skinny.conf doesn't look anything like the generic Asterisk skinny.conf |
| 04:25.05 | p3nguin | chan_sccp and chan_skinny have totally different syntax. |
| 04:25.06 | jpsharp | Ohwait. Nevermind. |
| 04:25.20 | sparkymarkd | but I'd like to get back to the software you mentioned before.. what is the it?? |
| 04:25.30 | jpsharp | Yes, I just realized my mistake. |
| 04:25.32 | sparkymarkd | yeah it's not skinny.. I'm using Chan_sccp |
| 04:26.24 | sparkymarkd | does skinny support enough functions to where I should use it?? this will be for a very simple install.. not many functions needed |
| 04:26.49 | jpsharp | That's why I was confused. I used skinny for my 7920. |
| 04:26.59 | p3nguin | Unless it has been completely redone, it doesn't even work at all. |
| 04:27.46 | sparkymarkd | ok.. for chan_sccp I had to revert to 1.6.. some are using it on 1.8, but it's still very buggy, have to manually code some to get it to work |
| 04:27.51 | sparkymarkd | I don't have the time for that |
| 04:28.00 | p3nguin | I use it on 1.8. |
| 04:28.11 | sparkymarkd | 4.1?? |
| 04:28.13 | p3nguin | No coding required. |
| 04:28.21 | sparkymarkd | svn branch? |
| 04:28.27 | p3nguin | Let me check the version I have. |
| 04:29.08 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 04:29.16 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-ukitlfnwdthmoizh) |
| 04:29.45 | p3nguin | Skinny Client Control Protocol (SCCP). Release: 4.0.0 DEV - 3206 (built by 'rob' on '2012-01-17 20:52:23 UTC') |
| 04:31.16 | sparkymarkd | ok... I should have checked here before.. reading online you don't see anybody saying that it's working, but I just yesterday joined the chan list, so not read much ther eyet |
| 04:31.40 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 04:32.01 | sparkymarkd | so if I'm this far, do I need any other files on tftp server? |
| 04:32.05 | jpsharp | Not too many people use it. Everyone's converted their SIP capable phones from SCCP to SIP. |
| 04:32.22 | p3nguin | I've been using chan_sccp without too much trouble for at least a couple years. |
| 04:33.08 | jpsharp | Sounds like your phone is trying to talk to Asterisk. Asterisk is just misbehaving, so I'd focus on that side now. |
| 04:33.09 | sparkymarkd | yeah, if you read on the list and some other forums, there's guys saying it works much betrer than switching your phone to SIP |
| 04:33.57 | p3nguin | I actually had to wait for them to make chan_sccp work with asterisk 1.8 so I could upgrade from 1.4 and still be able to use my phones. SIP on Cisco 7900 series phones sucks. |
| 04:36.47 | p3nguin | But chan_sccp isn't perfect, either. Pick your poison, I guess. |
| 04:39.37 | sparkymarkd | yeah, I guess nothing is perfect.. |
| 04:39.37 | sparkymarkd | thanks p3nguin and jpsharp.. I'll mess around with my config file now to see if I can get this to work properly |
| 04:40.12 | jpsharp | Now I want to see if I can find a cheap 7920 to play with. |
| 04:40.39 | jpsharp | Just for the sheer masochistic value of it. |
| 04:40.55 | sparkymarkd | I bought 5 on ebay.. total was less than 300 I think.. been a few weeks |
| 04:41.53 | sparkymarkd | but most of them have a broken top cover, the bottom 2 corners seem to be their weakest link |
| 04:42.20 | jpsharp | Found one for $34.99. I think maybe I'll get it. |
| 04:42.43 | sparkymarkd | change of topic... are there any good WIFI phones for less than 200US? |
| 04:43.29 | sparkymarkd | SIP on android or Iphone just does not seem to cut it.. |
| 04:43.43 | jpsharp | Everyone I've used, including the 7920, all suck in one way or another. |
| 04:44.09 | jpsharp | Crappy battery life, crappy WiFi radios, and/or crappy audio. |
| 04:44.34 | jpsharp | Mostly crappy wifi radio. |
| 04:44.36 | sparkymarkd | I had a QuickPhone working fairly good, but only with a MikroTik wifi router.. it SUCKS on a cheapo linksys router |
| 04:44.54 | p3nguin | Good and WiFi do not go along with VoIP. You can have good or you can have WiFi. |
| 04:45.20 | jpsharp | There's lots of router teats you have to make to get wifi voip to work remotely good. |
| 04:45.26 | jpsharp | router tweaks, too. |
| 04:45.33 | jpsharp | I don't know of routers that have teats. |
| 04:45.49 | sparkymarkd | that's what it seems like.. might also have to have a dedicated router for VOIP only - should help at least |
| 04:45.57 | sparkymarkd | LOL |
| 04:46.23 | jpsharp | Yes, you'll want a dedicated AP for your voip traffic. |
| 04:46.42 | p3nguin | You don't need a dedicated router for it, though. |
| 04:46.51 | p3nguin | VLANs do marvelous things. |
| 04:46.56 | sparkymarkd | I have a Mikrotik in a fairly large warehouse mounted on the roof, and the QuickPhone had better coverage there than any of the other DECT analog phones thye had |
| 04:47.45 | sparkymarkd | I'm sure mikrotik has all the prioritizations available too to get it working good.. |
| 04:48.21 | p3nguin | VLANs and QoS... should be good enough, especially if you have a good AP for the phones. |
| 05:20.48 | *** join/#asterisk din3sh (~din3sh@41.212.200.184) |
| 05:20.58 | *** join/#asterisk Russ (~russ@ip68-4-185-173.oc.oc.cox.net) |
| 05:21.06 | din3sh | mrning all |
| 05:21.11 | iscsi | moin |
| 05:24.54 | volga629 | what can cause app voicemail no load imap settings ? |
| 05:24.57 | volga629 | not |
| 05:35.07 | din3sh | is it possible to play an annoucement when using one-touch-monitor (automon) ? |
| 05:36.17 | dijib | morning |
| 05:36.35 | *** join/#asterisk KNERD (~KNERD@cpe-72-178-173-150.rgv.res.rr.com) |
| 05:37.25 | din3sh | mrning dijib |
| 05:38.42 | dijib | morning |
| 05:40.38 | dijib | p3nguin: did you call it a night? |
| 05:42.48 | *** join/#asterisk Russ (~russ@ip68-4-185-173.oc.oc.cox.net) |
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| 06:41.00 | schmidts | good morning |
| 06:42.03 | KNERD | I have no idea why 1.8.11 would NOT include voice mail as part of the default build. Rebult and it's COMEDIAN voice mail????? |
| 06:42.33 | KNERD | app_voicemail.so Comedian Mail (Voicemail System) |
| 06:43.31 | drmessano | It's always been comedian mail |
| 06:43.37 | KNERD | oh..okay |
| 06:43.43 | KNERD | strange |
| 06:43.45 | KNERD | but okay |
| 06:44.01 | drmessano | Did you install 1.8.11 certified? |
| 06:44.05 | KNERD | yes |
| 06:44.18 | drmessano | Thats why things weren't included |
| 06:44.43 | KNERD | what else is there do download? |
| 06:45.13 | drmessano | A non-certified build.. 1.8.13 is current |
| 06:45.37 | KNERD | okay...i just downloaded what came up first on the download site |
| 06:48.01 | din3sh | what is a non-certified build? |
| 06:48.03 | jpsharp | comedian mail was a poke at Nortel's Meridian Mail voicemail system. |
| 06:48.25 | KNERD | I am not laughing |
| 06:49.38 | drmessano | non-certified is standard asterisk |
| 06:50.55 | drmessano | certified are builds supported by Digium under SLA.. non-core modules, which aren't covered by SLA, are disabled by default.. It's also the only way to get the Digium phone module under 1.8. It's basically "Asterisk Business Edition", but GPLv2 |
| 06:57.30 | din3sh | the one from public download section is the non-certified one then ? |
| 06:57.51 | KNERD | oh |
| 06:58.05 | KNERD | din3sh: http://downloads.asterisk.org/pub/telephony/asterisk/ |
| 06:58.53 | *** join/#asterisk Russ (~russ@ip68-4-185-173.oc.oc.cox.net) |
| 06:59.33 | din3sh | aha thats the non-certified one |
| 07:09.30 | linocisco | jpsharp, Do i need to buy echo cancellation module on each card or Asterisk software itself could do for that small amount of 3 extensions FXS ? |
| 07:09.49 | *** join/#asterisk Neptu (~Neptu@41.143.18.163) |
| 07:12.29 | ChannelZ | Unless your computer is a 386 software EC should run just fine |
| 07:13.11 | ChannelZ | Unless your lines are particularly problematic hardware EC might do a better job, but you can always go down that road if you need to |
| 07:13.51 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85f8.bcn.adamo.es) |
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| 07:16.40 | [TK]D-Fender | PCI cards for FXS is a waste most of the time |
| 07:17.01 | [TK]D-Fender | Actually.. just about the only time you'd want it you DONT want EC anyway |
| 07:17.17 | *** join/#asterisk shadebob (~shadebob@41.251.182.216) |
| 07:18.29 | linocisco | [TK]D-Fender, but for home use to test asterisk, it is the only minimum |
| 07:18.50 | [TK]D-Fender | I woulndt touch a card likwe that for home use at all |
| 07:19.07 | [TK]D-Fender | Over-priced, less flexible, more PITA to manage |
| 07:20.29 | linocisco | [TK]D-Fender, so , IF you want and need similiar situation like me, what will you choose to minimize the cost with the same functionality? |
| 07:20.58 | [TK]D-Fender | 3 FXS? get 2 x Linksys PAP2T-NA's |
| 07:21.13 | KNERD | what is needed for that "format mp3" app? |
| 07:23.14 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:b4d0:358c:88c4:3364) |
| 07:28.40 | linocisco | [TK]D-Fender, I dont understand which product? |
| 07:28.58 | linocisco | [TK]D-Fender, I have exsiting analog desktop phones |
| 07:30.25 | [TK]D-Fender | LinI just gave you the MANUFACTURER and the MODEL NUMBER |
| 07:30.36 | [TK]D-Fender | What is there to "not understand"? |
| 07:31.59 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 07:32.41 | KNERD | I just built and installed 1.8.13.0 and upon start I am getting this scrolling non stop.... MOH: exec failed: No such file or directory |
| 07:36.57 | linocisco | [TK]D-Fender, can I tweak with asterisk command using that linksys device? |
| 07:37.16 | [TK]D-Fender | Linhuh? |
| 07:37.32 | [TK]D-Fender | linocisco: huh? |
| 07:39.13 | linocisco | [TK]D-Fender, I said we dont need RJ45 port , VOIP. We just need one FXO RJ11 port and 3 FXS RJ11 ports for analog extension. The linksys device doesn't match my requirement |
| 07:40.13 | [TK]D-Fender | linocisco: I just had to scroll back several pages to see your reference to FXO at all |
| 07:40.31 | [TK]D-Fender | linocisco: At which point you'd get 1 x PAP2T-NA, and 1 x SPA-3102 |
| 07:40.39 | [TK]D-Fender | linocisco: That would cover yuor exact requirement |
| 07:40.45 | [TK]D-Fender | your* |
| 07:49.12 | linocisco | [TK]D-Fender, how can those connect each other and make it 3 FXS? |
| 07:49.43 | [TK]D-Fender | linocisco: PAP2T = 2 FXS. SPA-3102 = 1 FXS, 1 FXO. |
| 07:49.48 | [TK]D-Fender | Do the math |
| 07:50.03 | KNERD | naught times naught = naught naught |
| 07:50.33 | linocisco | [TK]D-Fender, how can I join those two device together ? using which interfaces to which ? |
| 07:51.40 | [TK]D-Fender | ~ata |
| 07:51.40 | infobot | ata is, like, Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 07:51.51 | [TK]D-Fender | They talk SIP to * |
| 07:53.28 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:55.00 | linocisco | [TK]D-Fender, thanks for your explanation. What I mean is how to physically connect from which port of which device to which port of which device using which cable.? |
| 07:55.33 | linocisco | [TK]D-Fender, in addition, there is no VOIP network here in my home |
| 07:55.40 | [TK]D-Fender | You plug PHONES right into the FXS ports and your telco LINE into the FXO jack. |
| 07:55.57 | [TK]D-Fender | And both of those devices connect to your NETWORK SWITCH to talk to * |
| 07:56.25 | KNERD | linocisco: you need to watch a intro video..hang on |
| 07:57.06 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:57.08 | KNERD | [TK]D-Fender: http://www.youtube.com/watch?v=UP9b_FEZuUE |
| 07:57.11 | linocisco | [TK]D-Fender, there is no ethernet switch in my home. and no need to put that switch |
| 07:57.13 | KNERD | [TK]D-Fender: http://www.youtube.com/watch?v=UP9b_FEZuUE |
| 07:57.14 | KNERD | oops |
| 07:57.20 | KNERD | linocisco: http://www.youtube.com/watch?v=UP9b_FEZuUE |
| 07:58.07 | linocisco | [TK]D-Fender, my home has no vOIP network. just to share one FXO line to three FXS analog phone. no computers with LAN. Meaning No ethernet LAN at home. no switch |
| 07:59.31 | KNERD | linocisco: WATCH that video |
| 07:59.46 | linocisco | KNERD, watching last link |
| 08:00.42 | [TK]D-Fender | linocisco: Get a switch / router then |
| 08:01.12 | linocisco | [TK]D-Fender, that will be extra cost equal to digium PCI analog card |
| 08:01.20 | linocisco | [TK]D-Fender, I mean all in total |
| 08:01.28 | [TK]D-Fender | Will it? How much is the PCI solution? |
| 08:02.39 | linocisco | [TK]D-Fender, 634.49 USD according to digium card selector . but it is not from openvox, china made card as I dont know equivalent model . |
| 08:03.09 | linocisco | [TK]D-Fender, without echo can, $375.99 |
| 08:05.40 | [TK]D-Fender | PAP2T = $50. SPA-3102=$70. Cheap router=$25. Total=$145 |
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| 08:15.14 | linocisco | [TK]D-Fender, ebay price? |
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| 08:21.18 | KNERD | linocisco: http://www.ebay.com |
| 08:21.38 | linocisco | KNERD, there is no ebay here in my country |
| 08:21.50 | KNERD | what country? |
| 08:23.45 | KNERD | ohg yes...none |
| 08:24.09 | [TK]D-Fender | USA |
| 08:24.22 | [TK]D-Fender | Almost the same in Canada. |
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| 09:28.46 | jol | Is there some way to stop the current AGI command, making it end early, and let the AGI continue? Such as STREAM FILE on a long sound file, and you want to stop the playback and let the AGI continue. |
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| 09:33.14 | linocisco | who wants to read asterisk - The definite guide for FREE ? |
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| 09:47.31 | linocisco | http://astbook.asteriskdocs.org/ |
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| 10:38.45 | Alex_Bkash | any one have VPN experience for deploying VoIP |
| 10:41.03 | WIMPy | ~polls |
| 10:41.03 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
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| 10:58.21 | jero | :) |
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| 12:27.50 | danfromuk | Hi. I've got a server on a public ip, and some phones behing a nat device in an office. Sometimes, when calls are transferred, the call is lost. Its probably a nat issue but ive not had any problems before. It just started happening. |
| 12:28.37 | danfromuk | How can I diagnose the issue? |
| 12:30.58 | leifmadsen | watch the sip traffic during the transfer and determine what is going wrong |
| 12:31.18 | leifmadsen | either do it with something like tshark and analyze it in wireshark or use "sip set debug on" |
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| 12:32.56 | WIMPy | Are you using directmedia? |
| 12:33.13 | danfromuk | media all goes through the server |
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| 12:34.23 | danfromuk | ok, ill run tcpdump and see whats I can spot |
| 12:34.25 | danfromuk | what* |
| 12:36.55 | din3sh | how can i prevent a call arriving in a context from being transfered via sip refer? |
| 12:37.39 | jol | danfromuk, wireshark has nice built-in voip support |
| 12:38.25 | jol | danfromuk, You can even listen to media in a pcap file |
| 12:38.43 | danfromuk | ok, thanks. i'll give it a go to see if i can find the problem. |
| 12:40.00 | leifmadsen | din3sh: there is a flag in Dial() to ignore forwarding requests |
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| 12:49.01 | din3sh | what about ${TRANSFER_CONTEXT}, would it be of any help? |
| 12:50.54 | leifmadsen | that's for which context to use when performing a transfer from the phone |
| 12:50.57 | leifmadsen | so maybe |
| 12:54.53 | din3sh | i have calls coming from a SIP provider in the * box, I want the call to remain within * SIP extensions, and prevent the users to transfer on PRI line |
| 12:54.57 | din3sh | is that even possible? |
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| 12:57.29 | leifmadsen | yes |
| 12:57.52 | leifmadsen | use that channel variable you have and it will allow you to control which context is used for pattern match lookups when dialed |
| 12:59.11 | din3sh | ${TRANSFER_CONTEXT} this one? |
| 12:59.25 | leifmadsen | yes |
| 12:59.59 | din3sh | ok will try that |
| 13:00.05 | din3sh | thanks |
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| 13:52.39 | McBoingBo | in Asterisk 1.2, how can I test against where someone is calling FROM, I figured out "exten => s,5,GotoIf($["${CALLERIDNUM}"="<somne number>"]?s,10)" |
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| 13:55.18 | kaldemar | McBoingBo: 1.2 already has func CALLERID, it is preferred instead of the caller id variables. |
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| 14:01.30 | McBoingBo | kaldemar, but I need what number they dialed not where they are coming from |
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| 14:09.23 | leifmadsen | McBoingBo: ${EXTEN} |
| 14:09.55 | leifmadsen | it should be the request that came in |
| 14:10.02 | McBoingBo | leifmadsen, even from external? just makes me think internally only |
| 14:10.45 | leifmadsen | might be the DNID but that depends on if that was sent across or not |
| 14:10.56 | WIMPy | Asterisk doesn;t have a concept of internal and external. |
| 14:11.39 | WIMPy | But maybe you should put that question in to a context to make sense. |
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| 14:13.33 | McBoingBo | yeah EXTEN will not work |
| 14:14.00 | pzn | I need to add 500 entries in sip.conf, with all similar parameters. only changes [nnnn] and username=nnnn. I don't use passwords. is there an easy way to make it in asterisk? should I make a shell-script to append the 500 configurations to sip.conf? |
| 14:14.36 | WIMPy | You can use templates for common parameters. |
| 14:15.36 | leifmadsen | templates is the answer ya |
| 14:16.06 | McBoingBo | leifmadsen, so with EXTEN/DNID not working for me, is it because of Asterisk 1.2 or perhaps my provider is not passing through this info? |
| 14:16.09 | leifmadsen | and you could also use #exec in there to generate the headers and defaultuser= parameter |
| 14:16.16 | leifmadsen | McBoingBo: no idea |
| 14:16.22 | pzn | leifmadsen, WIMPy: I mean this in extensions.conf I do this "exten => _4XXX" then it is valid for extensions 4000 to 4999... is there a similar thing in sip.conf? |
| 14:16.25 | leifmadsen | the answer might be yes to both questions |
| 14:16.34 | leifmadsen | pzn: no |
| 14:17.04 | leifmadsen | you'll need a for or while loop in a script to generate the dynamic data |
| 14:17.09 | WIMPy | McBoingBo: What kind of provider? |
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| 14:17.38 | McBoingBo | WIMPy, we are using Babytel |
| 14:17.47 | WIMPy | What is that? |
| 14:18.04 | WIMPy | Are you using POTS? Or an ITSP? |
| 14:19.29 | luckman212 | Is it bad practice to run Asterisk 1.8 on an x86_64 OS? I built a new Atom server and it has 4GB RAM, so naturally I wanted to install a 64bit OS so I'd be able to use all 4GB. Does anyone know? |
| 14:19.59 | pzn | leifmadsen, then make the parameters default, and do some loop with the variable information. thanks! |
| 14:20.06 | leifmadsen | luckman212: bad? no |
| 14:20.17 | leifmadsen | pzn: ya the defaults apply to the template |
| 14:20.36 | leifmadsen | [my_user_name](template_to_apply) |
| 14:20.42 | leifmadsen | [template_to_apply] |
| 14:20.46 | leifmadsen | callerid=foobar |
| 14:20.51 | leifmadsen | oops |
| 14:20.56 | leifmadsen | [template_to_appy](!) |
| 14:20.58 | leifmadsen | that |
| 14:21.04 | leifmadsen | (!) marks it as a template |
| 14:21.06 | luckman212 | leifmadsen: ok well full disclosure the box is already built & running. Just wanted to see if there were any major known issues before I push forward with creating dialplan etc. |
| 14:21.17 | leifmadsen | I'm not sure why there would be |
| 14:21.22 | McBoingBo | WIMPy, sorry do not know the hardware very well, using a provider (middle mad) and the dialed number I am trying to pull from thin air is an alias to our DID, so not sure if we could gather that info anyway |
| 14:21.26 | leifmadsen | do you have other issues with 64-bit software? |
| 14:22.19 | luckman212 | leifmadsen: no - but I am remembering lots of pain when I first started running 64bit Windows versions. I know it's apples & oranges, but... |
| 14:22.44 | drmessano | Asterisk compiles and runs fine on 64-bit. Have run it that way for a number of years, and have a dozen deployments that are 64-bit. No issues. |
| 14:22.46 | luckman212 | I have very little experience running 64bit linux OSes |
| 14:23.01 | leifmadsen | it's not a concern |
| 14:23.09 | leifmadsen | your application of experience with 64-bit windows is not applicable here |
| 14:23.09 | luckman212 | ok, that's good to hear |
| 14:23.31 | drmessano | 64-bit Linux has been stable for a long time |
| 14:23.37 | leifmadsen | you can run with more then 4GB of RAM on a 32-bit kernel if it has PAE extensions |
| 14:24.43 | luckman212 | leifmadsen: hmm. I will need to research that |
| 14:25.03 | drmessano | That's a lot RAM unless your deployment is HUGE |
| 14:25.08 | drmessano | Even then.. |
| 14:26.41 | luckman212 | The extra RAM is not all for Asterisk -- and I was thinking of using a RAMdisk to store recordings, prompts etc to minimize disk delays |
| 14:27.20 | drmessano | ok |
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| 14:56.31 | wcselby | o/ |
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| 15:03.30 | wcselby | anyone here using google voice for inbound calls with gtalk.conf and jabber.conf? I am, and I've noticed I had to add an "Answer()" before I try to Dial() my SIP endpoint if I want to use the google voicemail method as described on wiki.asterisk.org |
| 15:03.38 | wcselby | has anyone else experienced this? |
| 15:04.02 | wcselby | otherwise the call just continues to ring, even after you pickup the SIP endpoint |
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| 15:12.40 | McBoingBo | I want to setup a different ring for a particular extension, not sure where to start, looking through book I must be missing it, any help appreciated, thanks |
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| 15:13.10 | wcselby | McBoingBo it depends on the phone you're using |
| 15:13.22 | ChannelZ | well doesn't Google Voice require you to answer the call, and press 1 or whatever to accept it? |
| 15:13.39 | wcselby | McBoingBo you can do this pretty easily with polycom phones using SipAddHeader().... |
| 15:13.43 | McBoingBo | wcselby, yeah figured that much, I am using Polycom IP430 |
| 15:14.20 | din3sh | is it possible to set static IP in polycom ip 550 provisioning file? |
| 15:14.22 | McBoingBo | wcselby, yeah the only article I came across was using SipAddHeader(), will read up thanks! |
| 15:14.39 | wcselby | McBoingBo it's SipAddHeader(Alert-Info: ... ) |
| 15:16.34 | wcselby | ChannelZ yeah, but isn't the Dial() app supposed to answer the line? I was followng the "Using Google's voicemail" section (and the followup "Filtering callerid" section) from this page -> https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 15:16.45 | wcselby | basically, it used to work without requiring the Answer() |
| 15:16.58 | wcselby | Now, you need the Answer() in order for it to work |
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| 15:17.31 | ChannelZ | So what? |
| 15:17.47 | wcselby | it's not a complaint, just seeing if other people are seeing the same thing |
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| 15:20.49 | ChannelZ | It's possible there needs to be a delay in there |
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| 15:21.53 | ChannelZ | I'm not sure if 'w' works in that case |
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| 15:35.02 | Scapal | what is the recommended approach for simple voicemail? I'm using parts of MiniVM but it is filled with crashing bugs |
| 15:35.30 | wcselby | why not just use app_voicemail? |
| 15:35.57 | wcselby | i guess I should rephrase my question - what do you mean by "simple voicemail"? |
| 15:37.49 | Scapal | well, I completely rewrote the dialplan in lua to manage and record voicemails. For example, I got rid of server stored emails, I only forward them to emails. |
| 15:38.12 | wcselby | that doesn't sound very simple |
| 15:38.18 | Scapal | so I only needed some voicemail building blocks: record the message and send it with a mail template |
| 15:38.43 | wcselby | so why not use the normal built-in voicemail app? |
| 15:38.48 | Scapal | simple blocks to build great things vs big monolithic voicemail |
| 15:39.31 | wcselby | as far as I know, and I could be wrong, but the app_voicemail that ships with * can do exactly what you're asking for if you configure it to |
| 15:39.48 | wcselby | just add delete=yes to each option, and it doesn't do any vm sotrage |
| 15:40.07 | wcselby | of course, you'll want to have valid email addresses in each option |
| 15:40.20 | wcselby | and you can control what mail command app_voicemail uses in voicemail.conf |
| 15:40.25 | Scapal | what if you enter voicemailmain? it still talks about folders.... |
| 15:40.27 | wcselby | i believe it's the mailcmd= option |
| 15:40.45 | wcselby | if you've built your own vm system, why would you enter VMM? |
| 15:41.01 | wcselby | i'm saying, you can use app_voicemail to do simple record and forward |
| 15:41.17 | [TK]D-Fender | [11:37]Scapalwell, I completely rewrote the dialplan in lua to manage and record voicemails. For example, I got rid of server stored emails, I only forward them to emails. <- there is a single voicemail.conf parameter to do this |
| 15:41.19 | Scapal | yes, I think that is what I'm going to do |
| 15:42.54 | Scapal | I was just hesitating between that and doing it myself with app Record and sending the mail with sendEmail, like I already do for faxes |
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| 15:51.20 | Scapal | [TK]D-Fender: it is more the VoiceMailMain I was unhappy with. The only thing left I'm using it to actually record and send the caller's message. I went using MiniVM, only needing a couple of voicemail building block. Now I have to forget that path since MiniVM has too many bugs. But I think indeed that I'll just use the app_voicemail for that part |
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| 15:51.53 | *** part/#asterisk clintc (~clintc@n128-227-69-165.xlate.ufl.edu) |
| 15:55.58 | volga629 | Hello EveryOne, VoiceMail imap storage not delivery into specific folder, just to inbox |
| 15:57.21 | wcselby | volga629 is there a question in there somewhere, or are you just stating a fact? |
| 15:57.24 | volga629 | I looked into debugs I see on mail server it process and doing some Indexing and after I delivery through lmtp with sendmail -t ? |
| 15:58.23 | volga629 | This is question where the start point to debug this issue ? |
| 16:00.46 | wcselby | unfortunately i've not worked with imap storage for voicemail, so I'm not sure |
| 16:00.55 | wcselby | hopefully someone else in the channel has and can be of more help |
| 16:01.23 | volga629 | Let just do small test and fpaste |
| 16:03.09 | mjordan | volga629: a DEBUG log isn't going to show much. If you don't mind editing the source, you can log to debug the full string being used to send the message to the IMAP server in imap_mailbox_name. You'd want to dump out the spec buffer. |
| 16:03.44 | mjordan | that being said, if you've properly configured voicemail using the imapfolder option, it should attempt to send the message to that folder |
| 16:05.32 | wcselby | mjordan what file would that be configured in? voicemail.conf? |
| 16:05.37 | volga629 | All configuration done according to asterisk documentation |
| 16:06.04 | wcselby | perhaps if he shared the config file in a pastebin, we could see if it's configured properly? |
| 16:07.04 | volga629 | on the way |
| 16:10.31 | volga629 | http://fpaste.org/3c4n/ |
| 16:10.43 | volga629 | this what I see on mail server side |
| 16:10.50 | Qwell | and your config? |
| 16:11.44 | volga629 | on the way |
| 16:12.37 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 16:13.08 | mjordan | wcselby: the imapfolder option is configured in voicemail.conf |
| 16:14.47 | volga629 | http://fpaste.org/8KYU/ |
| 16:16.46 | mjordan | volga629: is that your entire configuration? |
| 16:16.51 | mjordan | verbatim? |
| 16:16.56 | mjordan | you appear to be missing contexts |
| 16:18.19 | mjordan | volga629: that aside, I'm going to lean on this being your mail server, for two reasons |
| 16:18.37 | *** join/#asterisk Russ (~russ@ip68-4-185-173.oc.oc.cox.net) |
| 16:18.53 | mjordan | 1. Your log actually shows a search being performed by Asterisk, and it occurs in the VoiceMail folder, which is what you specified in your configuration. That implies Asterisk is using that folder name to look for voicemails from your mail server |
| 16:19.54 | mjordan | 2. When the mail server stores the message, it does so in folder "Inbox". Asterisk always refers to that folder as "INBOX" - if it was Asterisk that was using the default new message folder instead of the one that was overridden, I would expect to see "INBOX" in that INFO statement |
| 16:20.26 | volga629 | no that all what I am using for right now, just want make basic happy and after look what next |
| 16:22.43 | volga629 | I think that the way zimbra write logs, because I did small test and created sub folder under INBOX and specified like INBOX/VoiceMail and I so it recognized no issue |
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| 16:39.38 | *** join/#asterisk KNERD (~KNERD@cpe-72-178-173-150.rgv.res.rr.com) |
| 16:40.33 | KNERD | I just built and installed 1.8.13.0 and upon start I am getting this scrolling, non-stop...." MOH: exec failed: No such file or directory" |
| 16:41.19 | volga629 | I went though the log and found this [2012-06-25 12:07:24] WARNING[24966] app_voicemail.c: msgnum 0, mailbox message 0 is zero |
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| 16:43.02 | mjordan | volga629: when you use IMAP voicemail, messages - including warnings and errors - are reported from the IMAP c-client library. This can be all sorts of stuff, depending on the interactions between Asterisk and whatever IMAP server is involved. Sometimes they're benign, other times they're worth exploring, but its a case by case basis |
| 16:43.37 | [TK]D-Fender | KNERD: Have you considered actuallyl ooking at your MOH configs and showing us backup? I haven't seen any over the days you've been asking that same question with as much detail.... |
| 16:44.43 | KNERD | days? I just built it a few hours ago |
| 16:45.47 | [TK]D-Fender | I remember seeing this yesterday or the day before... |
| 16:45.51 | din3sh | KNERD: i just built and installed 1.8.13.0 earlier today, smoothly , no problems |
| 16:45.54 | [TK]D-Fender | Either way : backup |
| 16:45.55 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:46.17 | KNERD | yeah..i will just backup to 1.8.8.7 again |
| 16:46.57 | [TK]D-Fender | No.. provide BACKUP so we have something to actually DEBUG |
| 16:50.04 | din3sh | KNERD: your logs should show what is triggering that message |
| 16:55.18 | volga629 | I see, just trying understand what right what not |
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| 17:07.41 | McBoingBo | when I change Polycom ringtones with "SipAddHeader(Alert-Info: <NAME>)" I understand that the NAME is a value set, but how do I tie the Polycom default ringtones with the class? |
| 17:07.46 | McBoingBo | clear as mud? |
| 17:09.19 | KNERD | [TK]D-Fender: maybe I got it now...script was not downloading the sound files before...now it is |
| 17:09.32 | KNERD | make script |
| 17:11.35 | [TK]D-Fender | McBoingBo: You should probably be looking in your Polycom configs for the ALERTINFO tag .... just like the admin guide says |
| 17:17.27 | McBoingBo | [TK]D-Fender, I have referenced the admin guide, I know I should be using the alertinfo tag, but do I simply reference one of my rings from the <RINGER> tag like " <alertInfo voIpProt.SIP.alertInfo.1.value="Highest Double Trill" voIpProt.SIP.alertInfo.1.class="9"/>" ? |
| 17:17.47 | McBoingBo | because I tried that and it did not work |
| 17:19.03 | drmessano | Those need to be multiple lines |
| 17:19.31 | drmessano | Thats what I have here for autoanswer, and it works fine |
| 17:19.58 | McBoingBo | shoot I didnt reboot phone >< |
| 17:21.33 | [TK]D-Fender | McBoingBo: it's in the RINGTYPE you set for that class |
| 17:21.38 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 17:22.55 | McBoingBo | ok so I think it is taking header info but ring didnt make a sound, half working ! |
| 17:23.22 | McBoingBo | [TK]D-Fender, not sure what you mean by that, I am trying to understand how the default rings and the alertinfo tie together |
| 17:24.07 | [TK]D-Fender | I just gave you a TAG to look for |
| 17:25.54 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 17:26.05 | McBoingBo | [TK]D-Fender, no such tag RINGTYPE, first thing I did |
| 17:27.03 | [TK]D-Fender | There is an has been for far longer than half a decade |
| 17:27.15 | [TK]D-Fender | Its in the guide and it's in the sample configs |
| 17:27.17 | McBoingBo | RINGER is what seems to hold the phone's default set of ringtones |
| 17:27.34 | [TK]D-Fender | .... |
| 17:27.39 | [TK]D-Fender | and that isn't what you change |
| 17:28.19 | McBoingBo | [TK]D-Fender, sorry I took you literally, grepped and there is ringType |
| 17:29.14 | *** join/#asterisk HyperNerdV2 (~HyperNerd@173.225.78.46) |
| 17:31.11 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 17:31.45 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 17:31.47 | McBoingBo | [TK]D-Fender, ok sweet, so voIpProt.SIP.alertInfo.1.value= is only descriptive but voIpProt.SIP.alertInfo.1.class= has to match se.rt.10.name="Custom 3" (which would be class=10?) |
| 17:32.33 | Katty | hello my asterisk does not work at all how to fix plz??? |
| 17:33.19 | McBoingBo | har har |
| 17:33.22 | [TK]D-Fender | McBoingBo: Name has to match, number as well |
| 17:36.29 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 17:36.58 | McBoingBo | [TK]D-Fender, still not working I put "<voIpProt.SIP.alertInfo.1.value="Custom 3" voIpProt.SIP.alertInfo.1.class="10"/>" for ringType "<CUSTOM_3 se.rt.10.name="Custom 3" se.rt.10.type="ring" se.rt.10.ringer="9" se.rt.10.callWait="7" se.rt.10.mod="1"/>" |
| 17:37.56 | [TK]D-Fender | Where you put those tags is very important |
| 17:38.03 | *** join/#asterisk Naikrovek (~uname@unaffiliated/naikrovek) |
| 17:38.09 | [TK]D-Fender | Would be nice to see the COMPLETE product |
| 17:38.47 | Katty | waves at Naikrovek |
| 17:39.00 | Naikrovek | waves at Katty |
| 17:39.02 | Naikrovek | how you doing |
| 17:39.25 | Katty | a bit warm... |
| 17:39.34 | Katty | nearly 100F over here today. how's it out your way? |
| 17:40.04 | McBoingBo | [TK]D-Fender, http://pastebin.com/dcjLFxfS thanks |
| 17:40.25 | Katty | Naikrovek: how're you dear? |
| 17:40.45 | Naikrovek | good |
| 17:40.56 | Naikrovek | not sure how hot, but it'll be warm today here, as well |
| 17:41.05 | Katty | and the wifie? |
| 17:41.27 | McBoingBo | I am starting to think Katty's Asterisk setup is working just fine ;P |
| 17:41.55 | Katty | of course it is, don't be silly. |
| 17:42.55 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 17:43.13 | drmessano | If it wasn't working, she wouldn't be making money |
| 17:43.31 | Katty | oh idk about that. |
| 17:43.39 | Katty | i'd be making money and fixing it, then going back to reddit. |
| 17:43.47 | drmessano | Oh, don't be so humble. |
| 17:43.48 | McBoingBo | yeah I was poking her back for mocking the helpless ones |
| 17:44.09 | drmessano | Katty has one of the largest Asterisk based 1-900 businesses in the world |
| 17:44.13 | [TK]D-Fender | <alertInfo voIpProt.SIP.alertInfo.1. <-- set to 10 as well |
| 17:44.23 | Katty | i do?! |
| 17:44.26 | Katty | fancy!!!! |
| 17:44.29 | Katty | i should give myself a raise. |
| 17:44.48 | McBoingBo | [TK]D-Fender, oh christ! I thought that was like rule 1 of the alertinfo tasks.....I really did try to read up on this! |
| 17:44.49 | drmessano | You can thank her for video support. It was added to support one of her "endeavors" |
| 17:45.25 | drmessano | Katty, we should thank you for giving us a raise |
| 17:45.36 | drmessano | ;) Wink Wink Nudge Nudge |
| 17:46.31 | McBoingBo | [TK]D-Fender, ok adding 10 to alertinfo and rebooting phone |
| 17:47.42 | *** join/#asterisk shadebob (~shadebob@ll81-2-182-156-192-81.ll81-2.iam.net.ma) |
| 17:47.58 | Katty | raises her glass in drmessano's direction |
| 17:48.23 | *** join/#asterisk ulogic (421fc7ab@gateway/web/freenode/ip.66.31.199.171) |
| 17:48.52 | drmessano | lol |
| 17:49.13 | McBoingBo | [TK]D-Fender, ok well 10 had no sound coming from phone again will try another #, I am assuming that to find out which ring it is you need to check se.rt.10.ringer="9" and look for what ringer is 9....right? |
| 17:50.32 | McBoingBo | [TK]D-Fender, so line se.rt.10.ringer="9" refers to se.pat.ringer.9.name="Highest Double Trill" is how I have been setting it up |
| 17:52.31 | wcselby | o/ |
| 17:52.56 | Katty | glomps wcselby |
| 17:53.09 | wcselby | ahh, glomps :) |
| 17:53.13 | wcselby | o/ Katty |
| 17:53.29 | Katty | waves |
| 17:53.40 | wcselby | glomps are still good right? they haven't hcanged in the last 6 months or so? |
| 17:53.53 | drmessano | So Katty, how are you doll? |
| 17:54.46 | Katty | wcselby: no glomps are still lovely. |
| 17:54.56 | wcselby | that's good |
| 17:54.56 | wcselby | :) |
| 17:55.07 | Katty | drmessano: feeling a bit better since we last talked. still have a sinus infection tho. |
| 17:55.13 | Katty | drmessano: bought a bbq grill, which you probably saw. |
| 17:56.00 | drmessano | Yes I did. Well done. |
| 17:56.41 | *** join/#asterisk bobb_WU (~bobb_WU@206.74.211.13) |
| 17:56.58 | Katty | i need to get a chimney for it. none of my local stores carry the weber brand. |
| 17:57.04 | Katty | will probably have to get it off amazon. |
| 17:58.59 | bobb_WU | i'm having an issue with an asterisk server running a dahdi analog card. it won't dial out any more and i'm confused as to why |
| 17:59.03 | drmessano | Ah.. We had talked about a grill, but cooking for the last few months has been hell |
| 17:59.14 | Katty | why's that? |
| 17:59.17 | wcselby | bobb_WU please show us the CLI output of a failed call |
| 17:59.24 | wcselby | using pastebin |
| 17:59.25 | wcselby | ~pb |
| 17:59.26 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 18:00.22 | drmessano | Katty, She had a job for a few months where she was working either super early, or super late, and I had some projects at work.. So we were almost never home for dinner and hungry at the same time. Ended up making a lot of individual meals |
| 18:00.39 | bobb_WU | its not much |
| 18:00.41 | bobb_WU | http://pastebin.com/tynbDWcV |
| 18:00.49 | bobb_WU | as soon as i hit the first digit, it hangs up on me |
| 18:00.58 | Katty | drmessano: ah yes, but she started her new job today, right? |
| 18:01.11 | Katty | drmessano: perhaps back to a normal schedule soon? |
| 18:01.18 | drmessano | Didnt make sense to put any effort in to it. Hell, I haven't made one of my pizzas in MONTHS.. Whats the point if shes always reheating it? May as well be frozen |
| 18:01.21 | bobb_WU | it should be looking at chan_dahdi to get one of two contexts (analog or hotline) that determines how it calls out |
| 18:01.23 | drmessano | Yes she did |
| 18:01.32 | drmessano | So we should be right back on |
| 18:01.39 | Katty | cheers! |
| 18:01.49 | wcselby | bobb_WU are you using analog phones connected to FXS ports, or are you using SIP phones that are dialing out on FXO ports hooked up to analog phone lines? |
| 18:01.51 | Katty | i know how it is just cooking for yourself. why bother lol |
| 18:01.55 | drmessano | A grill is high up the list.. Want to grill veggies very badly |
| 18:02.04 | Katty | i would rather eat a can of tuna, some triscuits, and mircowaved veggies |
| 18:02.22 | bobb_WU | fxo- analog phones are hooked up to this node |
| 18:02.31 | McBoingBo | ok thanks for the help today, I will try to figure this out I have enough info to hurt myself now |
| 18:02.36 | Katty | yeah jaytee suggested getting one of those little flat grill things with holes in it to grill veggies |
| 18:02.39 | wcselby | bobb_WU have you tried swapping out the phones? |
| 18:02.49 | drmessano | Mmmmm.. Grilled corn |
| 18:02.52 | Katty | drmessano: i'm just not sure if i can find one to fit this itty bitty grill. |
| 18:02.58 | bobb_WU | yeah i brought the box back to the office for troubleshooting, and it seems to be happening on all ports |
| 18:03.04 | Katty | drmessano: ah, yes. i'm making grilled corn this week (wish me luck!) |
| 18:03.10 | bobb_WU | it does have a dial tone though, and i made sure the dahdi module is loaded |
| 18:03.11 | wcselby | sounds like a bad card, or maybe bad ports |
| 18:03.16 | drmessano | Katty, woohooo.. Jealous |
| 18:03.18 | Katty | drmessano: do you think i should cook it in the husk? |
| 18:03.21 | wcselby | is it a digium card? |
| 18:03.27 | bobb_WU | it is |
| 18:03.30 | Katty | drmessano: seems like everyone i know soaks it in water for awhile, then grills it in the husk to steam it |
| 18:03.45 | wcselby | bobb_WU you may still have warranty on it then, contact digium technical support |
| 18:03.55 | bobb_WU | thanks |
| 18:04.22 | drmessano | Katty, depends on what you want. In the husk you get sort of a roasted feel to it. If you grill it, you get some char and that FIRE taste |
| 18:04.23 | bobb_WU | real quick- it should find the context automatically from the chan_dahdi file, yes? |
| 18:04.40 | wcselby | yeah....? |
| 18:04.51 | ulogic | @sruffel: dahdi_cfg supports rxgain for individual channels when parsing /etc/system.conf but the only reference to DAHDI_RADPAR_RXGAIN in any of the drivers is in wctdm24xxp. I know I can set gain in chan_dahdi.conf, but I want the gain to occur before it gets to asterisk and possibly before it could be read by dahdi_monitor. Do the TDM400 cards have the capability to support this feature? |
| 18:04.56 | wcselby | assuming youv'e set that up properly |
| 18:05.19 | Katty | drmessano: well how do you get it to cook all the way through if you don't steam it in the husk? |
| 18:05.38 | Katty | drmessano: can you cook it directly on the grill with the lid on, and expect it to cook all the way through? |
| 18:05.39 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:05.45 | bobb_WU | it was working before so maybe it is actually a bad port. the port next to it just rang through. i will call digium. thanks for your help wcselby! |
| 18:05.55 | wcselby | no problem |
| 18:05.56 | Katty | drmessano: i guess that's a silly question, considering that's how you cook a steak lol |
| 18:06.17 | wcselby | Katty - don't leave corn on the grill too long though |
| 18:06.26 | Katty | long is all relative! |
| 18:06.29 | wcselby | lol |
| 18:06.29 | Katty | i don't know how to grill yet. |
| 18:06.39 | Katty | i need a proper grilling recipe. |
| 18:06.40 | drmessano | Katty, I have grilled corn over flame and not had a problem.. Have to turn it a few times and not burn it |
| 18:06.47 | Katty | light charcoal! wait x ammount of minutes |
| 18:07.14 | drmessano | Wait until its glowing red hot, with little flame.. |
| 18:07.24 | drmessano | 20 mins? |
| 18:07.29 | Katty | and i had another person that said wait until the charcoal was white, then wait a few more mintues |
| 18:07.34 | Katty | this all seems too tricky! |
| 18:08.17 | drmessano | Well, the charcoal will be white and glowing red from the inside.. it gets white rather quick.. Have to wait for the glow |
| 18:08.28 | Katty | takes notes |
| 18:08.34 | Katty | light charcoal, wait for white glow |
| 18:08.47 | Katty | i forsee post it notes all over my patio |
| 18:08.49 | drmessano | You'll be able to tell.. the bricks will be glowing like they have Red LED's inside them.. |
| 18:08.53 | drmessano | lol |
| 18:09.05 | *** join/#asterisk ulogic_ (421fc7ab@gateway/web/freenode/ip.66.31.199.171) |
| 18:10.07 | wcselby | how big of a grill did you get Katty ? |
| 18:10.15 | Katty | it's just a little smokey joe silver. |
| 18:10.17 | Katty | so about 15" i believe |
| 18:10.29 | wcselby | is it pretty deep? |
| 18:10.36 | Katty | i guess? idk |
| 18:10.51 | ulogic_ | Just came back after losing my connection...I hope i did not mess a reply during that interval regarding rxgain for individual channels in /etc/dahdi/system.conf |
| 18:10.52 | Katty | i don't know how to compare it to anything else |
| 18:10.56 | KNERD | din3sh: the log is reporting WARNING[1990] res_musiconhold.c: poll() failed: Interrupted system call |
| 18:11.11 | wcselby | important part of old smokey's is don't forget to open the vent on the bottom as well as the one on top of the lid |
| 18:11.21 | Katty | i was wondering about that |
| 18:11.24 | Katty | should the vents always be open? |
| 18:11.28 | Katty | for your typical burger and corn cooking |
| 18:11.29 | wcselby | when you want to cook, yes |
| 18:11.37 | Katty | when should you close them? |
| 18:11.41 | wcselby | you can close the top one a bit if you want to get more of a smokey flavor |
| 18:11.46 | KNERD | [TK]D-Fender: Define "backup" in as to your meaning in reference to my question |
| 18:11.53 | Katty | wcselby: and the bottom? what about that? |
| 18:11.56 | wcselby | but the bottom one always stays open until you're ready to put the fire out |
| 18:12.00 | *** join/#asterisk EmleyMoor (~phil@topdeck.tinsleyviaduct.com) |
| 18:12.08 | wcselby | otherwise the fire doesn't get enough oxygen and dies out |
| 18:12.09 | Katty | wcselby: so it's used to smother the fire? |
| 18:12.14 | Katty | ah HA! |
| 18:12.19 | Katty | i feel like i've learned something useful finally! |
| 18:12.23 | wcselby | :) |
| 18:12.34 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:12.34 | EmleyMoor | On a TDM card (e.g. AEX410), which pin of the RJ11 is the "A" wire? |
| 18:12.38 | Katty | hi tony |
| 18:12.40 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:13.00 | Qwell | EmleyMoor: It shouldn't be any different from any RJ11 jack. |
| 18:14.07 | wcselby | is 129.99 a good price for an archos 10.1" 8gb android tablet? |
| 18:14.15 | EmleyMoor | Qwell: ... so you suggest I compare a lead with RJ11 one end and 431A the other and work it out? |
| 18:14.47 | Katty | suggests chocolate chip cookies and vanilla almond milk |
| 18:14.49 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 18:16.10 | EmleyMoor | Well, my leads are red A, green B |
| 18:16.44 | Qwell | EmleyMoor: I'm saying that RJ11 is a standard. It's not going to change implementation to implementation. If you have a specific thing you need to do, then it's going to be no different than any other RJ11 interface. |
| 18:16.48 | *** join/#asterisk HyperNerdV2 (~HyperNerd@173.225.78.46) |
| 18:17.15 | *** join/#asterisk n0tk (~n0tk@gateway/tor-sasl/n0tk) |
| 18:19.46 | EmleyMoor | Right... but here, 431A is used in most places where you'd see RJ11 |
| 18:20.33 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:21.02 | drmessano | Katty, I wish I could drink almond milk. Highly allergic lol |
| 18:21.35 | Qwell | EmleyMoor: Them I'm sure there exist adapters. |
| 18:22.02 | EmleyMoor | Qwell: Easiest to make my ow |
| 18:22.05 | EmleyMoor | own |
| 18:25.45 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:28.14 | wcselby | so, i haven't played any with CEL yet. i was experimenting earlier with google voice calls, and I'm getting a weird error on hangup related to CEL. anyone have any advice what it means to me or what I should be doing about it, if anything? |
| 18:28.15 | wcselby | http://pastebin.com/yE9RZC8Q |
| 18:28.17 | *** join/#asterisk HyperNerdV2_ (~HyperNerd@173.225.78.46) |
| 18:28.57 | wcselby | a quick google of the error message didn't give me any useful results that I noticed... |
| 18:29.13 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
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| 18:31.52 | dandate2 | is VAD/silence suppression working yet in asterisk? |
| 18:33.31 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
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| 18:39.11 | *** join/#asterisk navaismo (~navaismo@189.144.194.18) |
| 18:39.11 | ruunix | hi there |
| 18:39.20 | ruunix | how can i see if my dahdi os working? |
| 18:39.28 | ruunix | *is |
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| 18:44.02 | wcselby | ruunix that's a bit vague |
| 18:44.16 | ulogic_ | ruunix: from the asterisk CLI, issue the command: dahdi show channels |
| 18:44.44 | wcselby | do you have hardware of some sort? |
| 18:44.54 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:45.13 | ruunix | i have updated my centos .. so dahdi is not working. . i have updated asterisk and dahdi .. all looks find .. but the calls are not workinkg .. |
| 18:45.23 | EmleyMoor | <PROTECTED> |
| 18:45.41 | ulogic_ | ruunis: You have to recompile the drivers after a kernel update |
| 18:45.43 | wcselby | ruunix "dahdi is not working" isn't helping. "calls are not working" isn't helping. |
| 18:46.14 | wcselby | how are they "not working", there are several different ways they could be "not working" |
| 18:46.17 | *** join/#asterisk cjk (~cjk@vodsl-10979.vo.lu) |
| 18:46.38 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:46.39 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:46.54 | wcselby | is it just inbound? just outbound? both directions? are callers being presented with any error messages? do the calls just fail? is there any error message in the CLI output of asterisk? |
| 18:46.55 | wcselby | etc |
| 18:47.41 | ruunix | telefonia*CLI> dahdi |
| 18:47.41 | ruunix | No such command 'dahdi' (type 'core show help dahdi' for other possible commands) |
| 18:47.44 | cjk | hi, I need some help with dtmfs. I see the dtmfs in the network stream but my last asterisk in the chain, skips or does not recognize dtmfs sometimes. any idea? PRI <-- asterisk1--> SIP <--asterisk2--> |
| 18:50.03 | EmleyMoor | If I get voltage but no dialtone, could it be that I need to recompile modules? |
| 18:50.19 | ulogic_ | ruunix: have you installed DAHDI from source or a package? |
| 18:50.33 | ruunix | source |
| 18:50.35 | ruunix | Command 'module load chan_dahdi.so' failed. |
| 18:51.01 | ruunix | this link |
| 18:51.01 | ruunix | http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/releases/dahdi-linux-complete-2.6.1+2.6.1.tar.gz |
| 18:51.08 | ruunix | i get it here |
| 18:51.43 | ulogic_ | ruunix do a "make clean;make;make install" then "service dahdi start", then restart asterisk |
| 18:52.01 | Qwell | ulogic_: You missed the ./configure. |
| 18:52.18 | ruunix | this package have not configure |
| 18:52.23 | ruunix | just makefile |
| 18:52.58 | ulogic_ | probably dont need the ./configure since I presume it was previously done |
| 18:53.25 | Qwell | You need to rebuild Asterisk after installing DAHDI. |
| 18:53.40 | ulogic_ | if you have the source, then configure will be in the source directory |
| 18:54.02 | ruunix | dahdi after asterisk or dahdi before asterisk ? |
| 18:54.11 | ulogic_ | dahdi first |
| 18:54.25 | ruunix | ok .. maybe this is the problem |
| 18:55.03 | EmleyMoor | Hmmm... asterisk recognises my dahdi devices but does not see them go off-hook, it seems |
| 18:56.45 | EmleyMoor | Hmmm... probing with my multimeter seems to help (!) |
| 18:57.23 | ulogic_ | EmleyMoor: I had that problem on certain motherboards if I had a vga= parameter in my boot kernel command line |
| 18:57.54 | EmleyMoor | ulogic_: Hmmm... odd! |
| 18:58.28 | wcselby | ugh |
| 18:58.42 | wcselby | i quit my old job last week, the one that was driving me insane. they've not taken it well |
| 18:58.58 | wcselby | i am constantly getting text messages form them now |
| 18:59.08 | wcselby | always with a hint of a threat |
| 18:59.23 | wcselby | asking for where things are, what passwords are, etc. |
| 18:59.38 | wcselby | i left plenty of documentation, a password database that they all had access to, etc. |
| 18:59.38 | EmleyMoor | This looks more like a connector problem |
| 19:04.07 | navaismo | wcselby, tell they need to pay for your support LOL |
| 19:05.03 | wcselby | navaismo i don't want to have anything to do with these people ever again |
| 19:05.06 | ruunix | this is really not workinkg |
| 19:05.29 | wcselby | they're litiguous to the point of rediculousness. if i offered to fix their stuff for a fee, they'd call it extortion or blackmail and take me to court |
| 19:05.43 | jpsharp | Sounds like the people I used work for. Didn't work for a company called Agiosat, by chance? |
| 19:05.54 | ruunix | why some times i think that i m the only one in the world with a specific problem... is something like you look for the problem in google .. and no body with the same problem .. |
| 19:07.20 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:07.23 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 19:07.24 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:07.55 | navaismo | wcselby, seriously? what a mess |
| 19:08.07 | KNERD | [TK]D-Fender: din3sh: i found the issue...it was a MOH streaming issue |
| 19:08.43 | KNERD | I guess deleting some things broke it |
| 19:09.19 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 19:10.18 | Qwell | ruunix: Did you re-run ./configure and make install in Asterisk after installing DAHDI? |
| 19:11.17 | *** join/#asterisk Neptu (~Neptu@41.251.159.114) |
| 19:12.39 | ulogic_ | ruunix: what kind of output does "service dahdi status" give you? |
| 19:14.28 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 19:14.34 | ruunix | ### Span 1: WCTDM/0 "Wildcard TDM800P" |
| 19:14.35 | ruunix | <PROTECTED> |
| 19:14.35 | ruunix | <PROTECTED> |
| 19:15.12 | ulogic_ | ruunix: well at least we know DAHDI is working |
| 19:16.01 | ruunix | Failed to open /dev/dahdi/transcode: No such file or directory |
| 19:16.04 | ruunix | maybe this ? |
| 19:16.06 | ulogic_ | do you should have these channels set up in /etc/asterisk/chan_dahdi.conf? |
| 19:17.27 | ruunix | yes i have the channels .. |
| 19:17.43 | ruunix | but asterisk even recognize a command dahdi |
| 19:18.35 | ulogic_ | I have two machines in my test lab, and neither one of them has /dev/dahdi/transcode -- where are you seeing that error message |
| 19:18.41 | [TK]D-Fender | That doesn't mean chan_dahdi.so is built or loaded |
| 19:18.56 | [TK]D-Fender | Try loading it by hand |
| 19:19.46 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 19:20.26 | ulogic_ | if you try to define a non-existent channel in chan_dahdi.conf, that will cause chan_dahdi.so to fail |
| 19:21.10 | [TK]D-Fender | If you're missing the config file altogether it will fail. |
| 19:21.31 | [TK]D-Fender | if the .so wasn't even there because you compiled/installed in the wrong order it will fail... |
| 19:21.48 | ruunix | damn .. i say that sometimes i think that i m the only one in the world with a specific problem .. i got this /usr/share/dahdi/waitfor_xpds: Missing dahdi_hardware , and this message even exist in google .. |
| 19:22.07 | ruunix | lets remove all .. and install all again . |
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| 19:22.38 | ulogic_ | if you issue from the * CLI "module load chan_dahdi.so" you might see some error messages |
| 19:22.47 | ruunix | yes |
| 19:23.05 | ruunix | is not a chan_dahdi.so module .. |
| 19:23.21 | ruunix | some asterisk version recomendation ? |
| 19:23.32 | ruunix | i have downloaded certified-asterisk-1.8.11-cert2 |
| 19:24.04 | ruunix | after that i have used 1.6.2 |
| 19:24.16 | Qwell | I ask again... |
| 19:24.17 | Qwell | ruunix: Did you re-run ./configure and make install in Asterisk after installing DAHDI? |
| 19:24.22 | ulogic_ | make sure to run ./configure in the asterisk source directory after DAHDI was installed |
| 19:24.35 | ruunix | yes .. but dahdi have not a configure |
| 19:24.39 | ruunix | just a makefilke |
| 19:25.22 | ulogic_ | That's is true, but you do need to ./configure asterisk after resolving a missing dependency. |
| 19:25.59 | [TK]D-Fender | ^ |
| 19:26.11 | [TK]D-Fender | you need to ./confuger ASTERISK |
| 19:28.11 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 19:29.52 | ruunix | ok .. dahdi > make clean > make > make install > make config |
| 19:30.09 | ruunix | now asterisk > make clean > configure > make > make install |
| 19:31.36 | ariel_ | OK folks, other then AEL which I really dislike and don't want to use, how can we do remote agent login in asterisk 1.8.X ??? |
| 19:31.42 | [TK]D-Fender | I'd recommend trashing and re-extracting Asterisk for compile.. |
| 19:32.10 | [TK]D-Fender | ariel_: AEL has nothing to do with "login", and you haven't told us what kind of "login" youre referring to |
| 19:32.32 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 19:32.33 | ariel_ | agent logins to more then 3 or 4 different queues |
| 19:32.40 | p3nguin | more than |
| 19:32.47 | p3nguin | thAn |
| 19:33.08 | ariel_ | not via extensions but via agent numbers as they share phones |
| 19:33.25 | ariel_ | in asterisk 1.6 remote_agent_login was removed |
| 19:33.26 | wcselby | ariel_ is there a specific app in the dialplan that handles this "login" ? |
| 19:33.30 | [TK]D-Fender | ariel_: Dialplan is dialplan... what's wrong with extensions.conf? |
| 19:33.54 | ruunix | something here is very wrong.. i think that i gonna reinstall all the operative system .. |
| 19:34.02 | [TK]D-Fender | AddQueueMember() <- |
| 19:36.00 | ariel_ | not that simple |
| 19:36.22 | wcselby | ariel_ - are you trying to replicate agentcallbacklogin? |
| 19:36.30 | [TK]D-Fender | ariel_: If you have special requirements you should probably start by telling us those... |
| 19:36.37 | ariel_ | wcselby: yes |
| 19:36.48 | [TK]D-Fender | AddQueueMember() <- |
| 19:37.23 | ariel_ | [TK]D-Fender: I need agents to login via a agent ID, so they can then be joined to many different queues. Once a call comes in, they will then be able to answer the call. |
| 19:37.43 | [TK]D-Fender | ariel_: That's what that app is for.... |
| 19:38.07 | wcselby | ariel_ i understand what you're getting at |
| 19:38.15 | wcselby | it's a pain, but it's the best app for the job |
| 19:38.24 | wcselby | i'll see if I can find some of my sample dialplans I used for it |
| 19:38.34 | ariel_ | [TK]D-Fender: it's not that simple, as they might not be at the same device |
| 19:38.34 | wcselby | you may want to look at handling some of the process in agi |
| 19:38.54 | ariel_ | wcselby: that would be fine, as well as able to set different passcode for the agent |
| 19:39.05 | wcselby | if you're wanting to login to multiple queues with one login, based on an agent id or something |
| 19:39.07 | [TK]D-Fender | ariel_: .... and that app can be used to add whatever you want to the queue |
| 19:39.23 | ariel_ | yes |
| 19:40.42 | ariel_ | so which one is willing to help with this dial plan setup and get some payments for the work? |
| 19:40.52 | wcselby | i'm typing up some sample now |
| 19:41.10 | wcselby | to point you in the right direction |
| 19:41.34 | ariel_ | wcselby: you want to get paid for helping us out with setting this up? |
| 19:41.40 | Katty | hi. |
| 19:41.46 | ariel_ | Katty: hello |
| 19:41.48 | wcselby | ariel_ I'm always open to the option |
| 19:41.55 | ariel_ | pm ok? |
| 19:41.56 | Katty | hugs ariel_ |
| 19:41.57 | wcselby | sure |
| 19:41.58 | Katty | how are you dear |
| 19:42.10 | ariel_ | Katty: fine, just behind on work load |
| 19:47.04 | ulogic_ | ruunix: Make sure there are no parse errors in chan_dahdi.conf |
| 19:48.13 | ulogic_ | ruunix: in the asterisk CLI, issue "dahdi showchannels" if it is an invalid command, then chan_dahdi.so did not load |
| 19:48.30 | ruunix | well .. |
| 19:48.45 | ulogic_ | ruunix: to see the errors, issue "module load chan_dahdi.so" and look for any error messages |
| 19:48.58 | ruunix | asterisk 1.8.lastversion have not compiled chan_dahdi .. |
| 19:49.07 | ruunix | now i have downloaded 1.6.24 |
| 19:49.19 | [TK]D-Fender | runnGo look in make menuconfig <-------- |
| 19:49.46 | ruunix | and now chan_dahdi is compiled .. maybe a version problem.. |
| 19:50.49 | *** join/#asterisk _pll (~anonymous@200.110.25.33) |
| 19:51.46 | _pll | Hello, I am trying to enable the ringinuse=yes option in queues.conf but it doesn't seem to be working. |
| 19:52.03 | ulogic_ | a few lines earlier, it should read "dahdi show channels" (I left out a space previously) |
| 19:52.27 | _pll | I am reading the source and apparently I have to set per member a ignorebusy=0 for ringinuse=yes to work. |
| 19:52.42 | _pll | The problem is, I don't see a way to set it for static members. |
| 19:53.19 | _pll | Any work around? |
| 19:53.52 | *** part/#asterisk Bullmoose (~Bullmoose@71-33-0-168.bois.qwest.net) |
| 19:58.49 | _pll | Apparently I could use the QUEUE_MEMBER function to change it. |
| 19:59.54 | _pll | Hmm, documentation is lacking on how to change member information. |
| 20:00.23 | ruunix | Dahdi working now .. |
| 20:00.26 | ruunix | damn . . |
| 20:00.44 | _pll | hmm, another way would be to set the default to 0 and recompile app_queues.c |
| 20:02.21 | _pll | Looks like I only have to change one line. |
| 20:02.40 | _pll | Let's try... |
| 20:03.17 | *** join/#asterisk shadebob (~shadebob@mail.caracteres.ma) |
| 20:04.00 | *** join/#asterisk viro (~viro101@cpe-71-72-174-83.woh.res.rr.com) |
| 20:07.43 | _pll | hmm, for some reason, after unloading app_queues then copying the new app_queues crashed asterisk. |
| 20:11.25 | _pll | I had a similar experience with chan_agent.so, after unloading, I couldn't load it afterwards because the "agent" tech was still registered. |
| 20:12.16 | ruunix | but ... after update sip users cannot register :( |
| 20:13.49 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 20:13.55 | ruunix | or after downgrade .. |
| 20:17.57 | Qwell | Why would you downgrade? |
| 20:18.10 | Qwell | You've been given the answer several times, and been asked several questions which you've ignored. |
| 20:18.20 | Chainsaw | Qwell: Oh ouch, not one of those :( |
| 20:30.37 | dandate2 | is silence suppression working with g729? |
| 20:32.50 | ruunix | i dont know if that was with me .. but i have not ignored nothing .. |
| 20:32.59 | ruunix | i made all that everybody here says |
| 20:33.14 | ruunix | and nothing works .. just the downgrade works .. |
| 20:34.26 | ruunix | certified-asterisk-1.8.11-cert2 dont compile chan_dahdi . |
| 20:34.56 | _pll | Hmm, looks like I am hitting this https://issues.asterisk.org/jira/browse/ASTERISK-18754 |
| 20:35.17 | _pll | And recompiling app_queues to set the ignorebusy default to 0 isn't working. |
| 20:35.24 | ruunix | asterisk-1.6.2.24 works perfect. . anyway thanks everybody for the help |
| 20:35.48 | _pll | I must be missing something in the source. |
| 20:37.16 | _pll | Oh, Asterisk 10.5.1 has some changes regarding ringinuse and ignorebusy |
| 20:37.26 | _pll | I got to update then. |
| 20:37.51 | _pll | At least in documentation... |
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| 20:40.32 | _pll | hmm, doesn't seem to affect the way to set ignorebusy. |
| 20:42.40 | _pll | It's still a vague when it says I have to use the queue_member function to set the ignore busy variable. http://www.asterisk.org/docs/asterisk/trunk/functions/queue_member |
| 20:43.21 | *** join/#asterisk danfromuk (~IceChat77@2.30.232.56) |
| 20:43.39 | _pll | Maybe something like QUEUE_MEMBER(myqueue/9999,ignorebusy)=yes |
| 20:43.50 | _pll | I would have to set it for each member I guess. |
| 20:43.59 | _pll | I meant =no |
| 20:46.30 | _pll | lol wut "Unknown option ignorebusy provided to QUEUE_MEMBER, valid values are: logged, free, ready, count, penalty, paused, ignorebusy" |
| 20:51.10 | _pll | Looks like penalty, paused and ignorebusy don't work. |
| 20:52.45 | _pll | Oh, the format is QUEUE_MEMBER(myqueue,ignorebusy,sip/9999) |
| 20:52.51 | _pll | And oddly, the result is 0. |
| 20:53.12 | _pll | Which means, that isn't what is preventing ringall=yes from working. |
| 20:53.56 | _pll | I mean, ringinuse=yes |
| 20:56.22 | _pll | This makes matters more complicated. I am now aimless. |
| 20:57.34 | _pll | Another solution would be to revert to a previous version from Asterisk 1.8.7.1 https://issues.asterisk.org/jira/browse/ASTERISK-18945?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel#issue-tabs |
| 20:58.54 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 21:00.38 | _pll | eeeeh |
| 21:00.45 | _pll | setting it to 1 makes it work. |
| 21:00.59 | _pll | Something odd is going on here. |
| 21:02.11 | _pll | It's backwards... |
| 21:04.41 | _pll | yep, setting ignorebusy to 1 makes it work. |
| 21:05.04 | _pll | But isn't the default 1? I recompiled the default to be 0. |
| 21:05.05 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 21:05.09 | _pll | I am going to revert the change. |
| 21:05.59 | _pll | Oh... I see where the issue is. |
| 21:07.22 | _pll | Let's hope it doesn't crash again. |
| 21:08.51 | _pll | welp, it crashed. |
| 21:09.09 | _pll | Something must be lingering around app_queues.so for Asterisk to crash when the file is replaced. |
| 21:09.26 | _pll | I did an unload just fine, odd. |
| 21:10.30 | _pll | This silence is unsettling, maybe I need to be voiced or something. I wonder if people are even reading me or I am just talking to myself. Which is fine too because I am solving the issue. |
| 21:12.56 | _pll | vanilla app_queues.so has the value of ignorebusy in 0. So my change to the default didn't even change anything. |
| 21:13.34 | _pll | So ast_variable_retrieve must be returning 0 somehow. |
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| 21:30.33 | _pll | Is this the proper way to get the member=${CUT(CHANNEL(name),-,1)} |
| 21:30.46 | _pll | ? |
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| 21:41.56 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-iflxechtuysajwyy) |
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| 22:03.16 | Greenlight | Howdy folks. Hoping someone can help where Google can't! My SIP provider sents some useful headers in the "OK" message. How do I go about retreiving these, seems ${SIP_HEADER(HeaderName)} doesn't return them (because they're not in the invite?) ? |
| 22:03.27 | *** join/#asterisk gusto (~gusto@ip-109-84-0-46.web.vodafone.de) |
| 22:03.35 | gusto | hi |
| 22:03.47 | gusto | how is it going? |
| 22:04.40 | gusto | kleszcz ? :-D |
| 22:04.56 | *** join/#asterisk Osage_ (~Osage@71-81-57-248.dhcp.slid.la.charter.com) |
| 22:05.13 | [TK]D-Fender | Greenlight: There is no option |
| 22:05.30 | Greenlight | Hmm that sucks ;( |
| 22:05.35 | gusto | well, in europe we say: 'there is no alternative' |
| 22:05.37 | gusto | :D |
| 22:06.18 | Greenlight | Just seemed it would be really good to capture the rate and tariff that my provider send, and put them in the CDR's |
| 22:06.59 | WIMPy | Oh, nice. What does that look like? |
| 22:09.10 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 22:09.59 | gusto | btw. i did not notice that macro() seems to be obsolete, is it? |
| 22:11.41 | Greenlight | DOnt think so |
| 22:11.49 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.40.134) |
| 22:12.15 | gusto | yes, but last time i built my super-complicated dialplan, asterisk was bullshitting against me that i should rather use sub()routines |
| 22:13.23 | WIMPy | Im not exactely sure about the situation. I think the use of macros in applications has been deprecated, but I'm not sure that means macros themselves are deprecated. |
| 22:13.52 | gusto | yes, something like that |
| 22:14.55 | Chainsaw | My dialplan still has macros in it. And that works in 10.5.1 |
| 22:15.31 | Greenlight | < same |
| 22:16.05 | gusto | yes |
| 22:16.28 | Greenlight | So are there any workarounds to get sip-headers that aren't in the invite, or do folks just have to live with it? |
| 22:19.18 | gusto | it wasnt an error, just like a warning/advice |
| 22:20.34 | gusto | so it must work, of course, i did not question that |
| 22:20.34 | gusto | i started to like macros |
| 22:20.34 | gusto | and goto and so on |
| 22:20.34 | Greenlight | So they're removing macros in favour of subs then ? |
| 22:20.34 | WIMPy | also uses macros. |
| 22:20.34 | WIMPy | Or actually... no, it's only one. |
| 22:20.34 | Chainsaw | At least three. |
| 22:20.34 | WIMPy | But that's the most important part. |
| 22:20.35 | pabelanger | well, start changing them out, when Asterisk 11 come out, Macros will be depreciated |
| 22:20.35 | pabelanger | comes* |
| 22:20.35 | p3nguin | They will be stripped of all their monetary value? |
| 22:20.35 | pabelanger | auto correct fail |
| 22:20.39 | gusto | what? |
| 22:20.47 | WIMPy | somehow dislikes the idea od using Gosum without any intention of a return. |
| 22:20.53 | gusto | first things first ... what is meant by INVITE? |
| 22:20.59 | WIMPy | s/od/of/ |
| 22:21.14 | Greenlight | INVITE is the initial SIP message |
| 22:21.23 | gusto | ah |
| 22:21.34 | gusto | i thought invite is the signal to start a call |
| 22:21.42 | Greenlight | It is |
| 22:21.52 | p3nguin | I've tried using the sub routine in a Dial(), and it fails. There isn't enough documentation on it to tell me why. I tried the sub routing with and without a return. |
| 22:22.24 | Greenlight | Doesn't the current Dial() have option of calling macros ... |
| 22:22.56 | Greenlight | I use |
| 22:22.58 | Greenlight | exten => _X.,n,Dial(SIP/aql-sip-wholesale/${EXTEN},90,M(trunk-ok,1)) |
| 22:23.11 | Greenlight | So how would we change that to a sub? |
| 22:23.56 | gusto | that trunk-ok is a macro? |
| 22:24.15 | Greenlight | mmhmm |
| 22:24.18 | gusto | ts |
| 22:24.20 | p3nguin | If macros are deprecated, M() would be deprecated. A sub routine option has been added to Dial(), but I can't get it to work correctly. Because it doesn't work, I had to go back to the macro method. |
| 22:24.31 | gusto | i did not know, that it can ba called just by M() |
| 22:24.38 | *** join/#asterisk shadebob (~shadebob@mail.caracteres.ma) |
| 22:24.45 | Greenlight | M() calls the macro on connect |
| 22:24.57 | Greenlight | I use the docs off voip-info, perhaps they're out of date |
| 22:25.22 | p3nguin | U() is the alternative. |
| 22:25.26 | gusto | i have no idea i have the printed book |
| 22:25.48 | gusto | the latest printed one 3rd version |
| 22:26.12 | gusto | ~book |
| 22:26.13 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:26.14 | Greenlight | Ahh yea U see U now |
| 22:27.05 | gusto | i do not like that documentation on voip-info, when you do not know what's going on it's useless |
| 22:27.19 | gusto | it's only good for ppl who know what they are doing |
| 22:27.33 | Greenlight | I've noticed of late it seems somewhat outdated |
| 22:27.56 | Greenlight | But it always been high up on Google when I search for things, which is probably why I tend to use it |
| 22:28.12 | gusto | well, it does have a broad variety of things documented, so it's nearly impossible to keep that current all time |
| 22:28.19 | Greenlight | Ofc |
| 22:28.21 | p3nguin | I think the majority of the stuff is still geared for 1.2. |
| 22:28.26 | Greenlight | Indeed |
| 22:28.30 | WIMPy | It was soewat outdated when I began some 5 years ago. |
| 22:28.42 | WIMPy | But it still _can_ be useful. |
| 22:28.44 | Greenlight | Heh, so quite far behind then :) |
| 22:28.59 | gusto | yes, because ppl search on google when they want to have some info fast ... but it's not a good advice for beginners/good practice |
| 22:29.13 | Greenlight | Still fairly accurate in most cases but it's just the little things and updates that are missing |
| 22:29.43 | gusto | well |
| 22:30.39 | Greenlight | For example voipinfo's ChanSpy page doesnt mention the "B" option which I needed just earlier today |
| 22:30.55 | gusto | documentation is work too, they would need some kind of management system that would automatically suggest part to be reviewed on relevance/updates |
| 22:31.03 | Greenlight | Was about to start looking at redirecting to conferences before I checked other sources and seen there was such an option |
| 22:31.29 | gusto | i can not imagine chanspy to be usefull at all |
| 22:31.41 | Greenlight | In a callcentre context it is :) |
| 22:31.53 | Greenlight | Our customers use it all the time |
| 22:31.54 | *** join/#asterisk catphish (~charlie@2001:9d8:2005:12::3) |
| 22:31.57 | gusto | and what do you want to spy on with it? |
| 22:32.09 | Greenlight | Team leaders and managers listening to agents |
| 22:32.16 | gusto | ah |
| 22:32.22 | Greenlight | That kidna thing |
| 22:32.28 | gusto | so you can really listen to an open channel with it? |
| 22:32.31 | Greenlight | mmhmm |
| 22:32.39 | Greenlight | The "whisper" option is good too |
| 22:32.40 | gusto | well, that sounds like NSA thing |
| 22:32.54 | Greenlight | The agent can hear the team leader, but the customer can't |
| 22:33.07 | gusto | hahahahahahhahahahahahahhahahahaa |
| 22:33.09 | Greenlight | Once they get over having a laugh with it, its very hadny for training |
| 22:33.34 | Greenlight | But yes, whenever I explain the feature they just take the piss for a good hour or two :) |
| 22:33.41 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 22:34.04 | gusto | what does it mean 'taking a piss'? |
| 22:34.12 | Greenlight | umm - having a laugh |
| 22:34.28 | WIMPy | "bepisst vor lachen" |
| 22:34.34 | Greenlight | I had one time where an agent wasn't aware of this "feature" and were convinced they were hearing voices ;) |
| 22:34.35 | gusto | Greenlight: where are you from? |
| 22:34.51 | Greenlight | Scotland, UK |
| 22:34.57 | gusto | Greenlight: that's cool |
| 22:35.10 | gusto | Greenlight: my name is a scottishman |
| 22:35.13 | catphish | is Set(CALLERID(num-pres)=prohib) the correct way to withold caller id send to a public network |
| 22:35.17 | p3nguin | To some, take a piss means to urinate. |
| 22:35.18 | gusto | or how do you call yourself, ha? |
| 22:35.28 | Greenlight | Scotsman :) |
| 22:35.31 | Greenlight | Or Scottish |
| 22:35.35 | gusto | ok, scotsman |
| 22:35.37 | gusto | well |
| 22:35.41 | gusto | schotte in germany |
| 22:35.44 | p3nguin | Scottish Man |
| 22:35.50 | Greenlight | p3nguin: Yes, that's the literal meaning, i sometimes forget I'm talking to internation folks |
| 22:35.51 | WIMPy | catphish: Yes |
| 22:36.03 | gusto | but i never have been there |
| 22:36.06 | catphish | p3nguin: there's a great difference between a piss and /the/ piss |
| 22:36.24 | catphish | at least in my neck of the woods |
| 22:36.41 | WIMPy | Or going in a pis[s|h]? |
| 22:36.43 | Greenlight | "Neck of the woods" there's another one |
| 22:36.45 | WIMPy | s/in/on/ |
| 22:36.47 | p3nguin | In the USA, get pissed usually means to be angered. In the UK, getting pissed seems to mean to become drunken. |
| 22:36.59 | WIMPy | oh, dang. "going" |
| 22:37.00 | gusto | well |
| 22:37.11 | catphish | in the uk it now means both, which is really confusing |
| 22:37.13 | Greenlight | p3nguinL: Depends on the context, can mean both |
| 22:37.36 | Greenlight | We like to be nice and confusing here in the UK |
| 22:37.47 | gusto | well |
| 22:37.54 | gusto | but you wanted to separate, right? |
| 22:38.08 | Greenlight | I guess |
| 22:38.10 | gusto | you separatists :-D |
| 22:38.38 | gusto | i like the UK, but i would blow up the queen |
| 22:38.45 | Greenlight | 'S |
| 22:38.47 | Greenlight | ;S |
| 22:38.53 | catphish | gusto: i dont think you're allowed to say that |
| 22:38.59 | Greenlight | Likely not |
| 22:39.11 | gusto | where? |
| 22:39.17 | gusto | i am not inside the UK, i can |
| 22:39.17 | Greenlight | The Queen's not all that bad |
| 22:39.19 | catphish | thats like threatening to blow up robin hood airport |
| 22:39.28 | WIMPy | I know who is not going to watch the olympic games live :-) |
| 22:39.31 | Greenlight | catphish: Perhaps not as extreme |
| 22:39.36 | catphish | :) |
| 22:39.49 | gusto | why? who? |
| 22:39.52 | Greenlight | Pfft Olympic games |
| 22:40.06 | WIMPy | gusto: You. |
| 22:40.12 | gusto | btw. the czechs are putting it on the 3rd MUX FTA in HD !!!! |
| 22:40.30 | catphish | is Set(CALLERID(num-pres)=prohib) the correct way to withold caller id send to a public network |
| 22:40.32 | Greenlight | 3rd MUX FTA - Whats that like freeview? |
| 22:40.34 | gusto | and meaning by HD = fullHD |
| 22:40.43 | catphish | is on topic is noone else is :) |
| 22:40.46 | Greenlight | catphish: I beleive it depends on your provider |
| 22:40.47 | WIMPy | catphish: Yes (still) |
| 22:40.47 | catphish | *if |
| 22:41.02 | catphish | WIMPy: sorry, totally missed that! |
| 22:41.14 | catphish | thanks |
| 22:41.36 | p3nguin | I did some extensive testing with the presentation thing recently, and I determined that the provider is definitely deciding what is the right way to do it. |
| 22:41.46 | Greenlight | agrees |
| 22:41.55 | gusto | yes |
| 22:42.01 | gusto | he can set it anyway |
| 22:42.02 | catphish | what other options exist? |
| 22:42.08 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 22:42.11 | Greenlight | Annoymous and blank afaik |
| 22:42.14 | p3nguin | My primary provider ignores whatever type of pres I send, giving the call recipient my number anyway. |
| 22:42.15 | catphish | apart from sending it, which im not allowed to do |
| 22:42.20 | catphish | *not sending it |
| 22:42.33 | gusto | p3nguin: that's what i said as well |
| 22:42.34 | Greenlight | I've "type 5 cli" they'll forward whatever number I send :) |
| 22:42.39 | p3nguin | I decided to just send a totally random callerid number. |
| 22:43.01 | Greenlight | Althoght I guess I'd get in trouble pretty quick with some numbers |
| 22:43.14 | catphish | i can send whatever i want, but i need to send a real number but mark it witheld for correctness |
| 22:43.15 | p3nguin | Basically, "If you won't hide my number when I say to hide it, I'll give you a bogus number to go ahead and share." |
| 22:43.19 | gusto | Greenlight: what do you not understand on that czech dvb-t? |
| 22:43.38 | WIMPy | Depending on your seetings for RPID and PAI, setting the presentation should send "anonymous". |
| 22:43.43 | Greenlight | I've forgot to configure CLI before and the extension number (say 200) gets sent through |
| 22:43.53 | catphish | i try the pres i mentioned before |
| 22:44.01 | catphish | hopefully they'll respect that |
| 22:44.15 | catphish | i believe my provider uses cisco call routers |
| 22:44.16 | Greenlight | Only way to know for sure is to test |
| 22:44.32 | Greenlight | Cisco are as bad as MS for making their own way up |
| 22:44.37 | catphish | lol |
| 22:44.46 | gusto | catphish: you mean cisco PBX software? |
| 22:45.01 | Greenlight | We had to change the DTMF payload type in the code for our SIP provider to accept it |
| 22:45.22 | catphish | gusto: i thought cisco just did callmanager on their routers |
| 22:45.33 | gusto | catphish: i have no idea |
| 22:45.37 | p3nguin | A provider wouldn't really be using a PBX. |
| 22:45.45 | catphish | indeed |
| 22:45.48 | Greenlight | More of a SIP gateway |
| 22:45.55 | p3nguin | switch? |
| 22:46.03 | gusto | whatever you call it |
| 22:46.17 | gusto | it's just something like asterisk, but not opensource |
| 22:46.30 | catphish | not really |
| 22:46.33 | carrar | hey could use asterisk |
| 22:46.36 | carrar | they |
| 22:46.40 | Greenlight | Mind Asterisk can be setup in much the same way, to act very much like a SIP gateway |
| 22:46.43 | catphish | a call router / switch is much more limited |
| 22:46.59 | catphish | asterisk can do it, not sure what the overhead is like |
| 22:47.11 | WIMPy | I'd rather say: It does other things. |
| 22:47.15 | gusto | yes, but you can have it on a specialized HW that does also DAHDI and so on |
| 22:47.22 | catphish | my customers keep telling me asterisk wont scale like their proprietary call routers |
| 22:47.25 | gusto | i mean the equivalent of DAHDI |
| 22:47.26 | Greenlight | We've a Asterisk box setup just as a "call gateway" and no RTP or media touches it, just the SIP traffic |
| 22:47.28 | catphish | but i see it scale quite well |
| 22:47.33 | WIMPy | Sipgate used to use Asterisk for some products. |
| 22:47.38 | Greenlight | 500K calls a day without breaking a sweap |
| 22:47.41 | Greenlight | *sweat |
| 22:48.15 | p3nguin | Why wouldn't you use something like kamailio for that instead of asterisk? |
| 22:48.19 | catphish | even with rtp and no transcoding, asterisk seems to breeze through 1k simultaneous calls |
| 22:48.27 | Greenlight | mmhmm |
| 22:48.36 | Greenlight | I know Asterisk, so I used that |
| 22:48.47 | Greenlight | Rather than learn somethign new |
| 22:48.54 | gusto | someone uses asterisk professionally on different OS'es from linux? |
| 22:49.08 | catphish | in my tests the cpu gave up at 1k calls on a core2duo iirc |
| 22:49.10 | Greenlight | Works on Solaris and BSD doesnt it ? |
| 22:49.16 | gusto | yes, whatever |
| 22:49.18 | p3nguin | It works pretty good for me on FreeBSD. |
| 22:49.29 | Greenlight | catphish: WIth RTP as well? |
| 22:49.31 | gusto | i have also one installation on freebsd |
| 22:49.36 | catphish | Greenlight: yep |
| 22:49.44 | gusto | and netbsd and openbsd in a virtualbox here on my laptop |
| 22:49.44 | Greenlight | Nice, was that 10.X ? |
| 22:49.57 | Greenlight | Or 1.8.X ? |
| 22:50.05 | catphish | 1.8.7 |
| 22:50.20 | Greenlight | Yea, since 1.8 it's seemed much better at high load than 1.4 and 1.6 |
| 22:50.23 | gusto | 1.8 is cool |
| 22:50.56 | Greenlight | I'd be somewhat scared to push 1k sim calls on a single production box with that spec though :) |
| 22:51.19 | catphish | someone reported 13,000 calls on asterisk 10 on 12 core xeon |
| 22:51.22 | gusto | what specification? |
| 22:51.27 | _pll | Hey, hello again. |
| 22:51.43 | catphish | i run mine on a 24 core xeon, so im not at all worried about workload :) |
| 22:51.43 | gusto | isn't that more a question of bandwidht? |
| 22:51.52 | gusto | bandwidth? |
| 22:52.05 | Greenlight | We do a load of conferences and transcoding and I try to keep <500 sim channels on this spec of servers: http://www.ovh.co.uk/dedicated_servers/mg_best_of.xml |
| 22:52.15 | catphish | 100k per call = 100k calls on gigabit? |
| 22:52.35 | catphish | no, 10k calls on gigabit |
| 22:52.44 | gusto | yes |
| 22:52.48 | catphish | so you max out everything at roughly the 10k call mark |
| 22:52.49 | Greenlight | It's more of a questions of pps than bandwidth at high call volume |
| 22:52.58 | Greenlight | pps = packets per second |
| 22:53.04 | catphish | yes, i'd never push it that far |
| 22:53.14 | catphish | in fact, i only have a 100M transit port |
| 22:53.26 | gusto | hahahahahahhaa |
| 22:53.31 | gusto | i have no transit port :-D |
| 22:53.38 | gusto | only a dsl connection |
| 22:53.40 | Greenlight | We've 10Gb but never even come close |
| 22:54.03 | catphish | my itsp is tiny :) |
| 22:54.05 | Greenlight | Guess it's nice to know we can survive a limited DOS attack though |
| 22:54.12 | gusto | but i am not professional, i do it for fun and maybe someday some more friends join, i like it sofar |
| 22:54.44 | Greenlight | Asterisk is great, althought I now use it professionally and it's at the core of our business, I've really enjoyed learning and using it |
| 22:55.10 | Greenlight | I'll still often discover new things from just tinkering around and trying stuff with it |
| 22:55.23 | catphish | so you should :) |
| 22:55.42 | Greenlight | Then a month later is like "how did I manage without that new feature" |
| 22:56.10 | gusto | yes |
| 22:56.13 | gusto | maybe |
| 22:57.08 | gusto | but a DOS attack can be prevented someway |
| 22:57.22 | p3nguin | Depends on the type of DoS. |
| 22:57.29 | Greenlight | A DDOS can't |
| 22:57.40 | gusto | no |
| 22:57.43 | gusto | it can be |
| 22:57.47 | catphish | it can be stopped in some ways |
| 22:57.50 | p3nguin | DDoS can be mitigated. |
| 22:57.54 | catphish | if you're on the ball |
| 22:58.06 | Greenlight | If you get flooded, and you're ISP won't help, you're fooked |
| 22:58.10 | p3nguin | Upstream can help. |
| 22:58.14 | WIMPy | The best way to stop a (D)DOS is not to be connected to the internet. |
| 22:58.15 | gusto | because you have a couple of routes that attacks come from and you can selectively turn them off |
| 22:58.26 | catphish | if your upstream will firewall source IPs, you're good |
| 22:58.52 | gusto | well, that's what i am talking about |
| 22:58.57 | Greenlight | catphish: Yea if you ISP or data centre will play ball |
| 22:58.59 | catphish | if its a random source attack, you're more screwed, but you can minimize its impact on the application layer by filtering |
| 22:59.16 | catphish | i only ever got DoS'd once |
| 22:59.24 | catphish | took out my entire ISP |
| 22:59.33 | Greenlight | Ouch |
| 22:59.38 | p3nguin | Been there. |
| 22:59.42 | catphish | i was able to fix it by talking to the criminals faster than the isp could filter it |
| 22:59.52 | gusto | even i had a DOS attack on me!!! |
| 23:00.06 | Greenlight | My ISP have put in place some protection measures but still they see DDOS attacks sustained over 100Gbs ... |
| 23:00.14 | gusto | it took down my DSL connection :-D it was originated from bejing/china |
| 23:00.14 | WIMPy | That's the good thing if you've got a lot of bandwidth, so you provider gets in trouble as well. That makes it more likely for them to help :-) |
| 23:00.46 | catphish | the amount of bandwidth you have isnt necessarily important |
| 23:00.55 | catphish | since the excess will hit your upstream anyway |
| 23:00.59 | Greenlight | Course it is, or well the bandwith the provider has |
| 23:01.11 | WIMPy | Yes |
| 23:01.30 | Greenlight | If your provider can firewall it, and they have plenty spare pipes then great |
| 23:01.36 | Greenlight | Otherwise you're screwed |
| 23:01.45 | catphish | having more bandwidth is ultimately necessary to mitigate and filter the attack |
| 23:01.48 | gusto | isn't that also a problem on ipv6? |
| 23:01.57 | catphish | its no different |
| 23:02.01 | Greenlight | ipv4 or 6 doesnt matter |
| 23:02.14 | Greenlight | Addresses will likely be spoofed anyways |
| 23:02.14 | gusto | because ipv6 has some more multicast functionality, maybe they can misuse that |
| 23:02.30 | catphish | unlikely, and trivial to filter |
| 23:02.51 | gusto | yes |
| 23:02.52 | gusto | but ISP's are idiots |
| 23:02.53 | carrar | Arbor networks has a excellet product to mitigate DDOS |
| 23:02.55 | Greenlight | The problem for SIP or VOIP is that most attacks are UDP based |
| 23:02.57 | carrar | http://www.arbornetworks.com/tms |
| 23:02.58 | gusto | so it is not that trivial to them |
| 23:03.05 | carrar | it however is not cheap |
| 23:03.28 | carrar | You still however need the BW to accept the traffic |
| 23:03.43 | catphish | someone somewhere needs the bw |
| 23:03.47 | catphish | to accept and filter |
| 23:03.52 | Greenlight | Attacks can be extreme these days >100Gbs easy |
| 23:03.57 | carrar | NSP's typically use that product |
| 23:04.00 | catphish | anycast is great for that |
| 23:04.03 | catphish | as you can filter locally |
| 23:04.38 | catphish | where bandwidth is cheap |
| 23:04.52 | gusto | hm |
| 23:05.03 | gusto | only filtering at the routers makes sence |
| 23:05.08 | gusto | to stop it at the source |
| 23:05.17 | carrar | blow up your routers that way |
| 23:05.23 | Greenlight | That's what my ISP does |
| 23:05.26 | gusto | ??? |
| 23:05.37 | catphish | if you have anycast, you can accept traffic on a local 10G link, filter it, then send valid connections to your central servers |
| 23:05.51 | catphish | normally akamai do that for people |
| 23:05.55 | Greenlight | And if they flood the 10G? |
| 23:06.18 | catphish | 1) they normally don't 2) it doesn't matter, only that location will lose connectivity |
| 23:06.22 | *** join/#asterisk fulcan (~brads@li186-148.members.linode.com) |
| 23:06.26 | gusto | yes |
| 23:06.29 | catphish | everywhere else will still be able to connect to you |
| 23:06.30 | Greenlight | Ahh I see |
| 23:06.36 | gusto | regionalisation is the answer to everything! |
| 23:06.43 | Greenlight | So they route traffic to a "hidden" ip, via their own servers |
| 23:06.58 | Greenlight | So you're never exposed directly ? |
| 23:07.00 | catphish | ie if I DoS facebook.com from here, only facebook in the uk will be impacted |
| 23:07.13 | gusto | or you ISP |
| 23:07.16 | catphish | ok, think about 8.8.8.8 |
| 23:07.27 | carrar | thinks about hawaii instead |
| 23:07.28 | Greenlight | That has Geo servers |
| 23:07.29 | catphish | it's a single ip, but it exists in every country |
| 23:07.32 | Greenlight | yea |
| 23:07.42 | catphish | if you send traffic to it, it only goes to your local node |
| 23:07.44 | gusto | yes, like whitehouse.gov |
| 23:07.45 | gusto | -D |
| 23:07.57 | Greenlight | Still, taking down servers for all of a country is still a big hit |
| 23:08.13 | catphish | whitehouse.gov is akamai :) |
| 23:08.21 | Greenlight | And even 100G isn't a lot these days |
| 23:08.21 | gusto | i have no idea |
| 23:08.23 | gusto | maybe |
| 23:08.28 | catphish | it is. |
| 23:08.31 | gusto | ok |
| 23:08.38 | gusto | i did not question that |
| 23:09.01 | carrar | This is why you always get 10 x 100gbps links |
| 23:09.01 | gusto | i only noticed some time ago, that it responds quickly and it does not even go through the exchange |
| 23:09.02 | Greenlight | Naa, imagine a hacker with a botnet of 10,000 servers |
| 23:09.05 | carrar | just to be safe |
| 23:09.21 | Greenlight | Or even 100 servers on 1Gb links |
| 23:09.23 | catphish | Greenlight: sometimes you're just fucked |
| 23:09.25 | *** join/#asterisk Neptu (~Neptu@41.251.159.114) |
| 23:09.30 | Greenlight | Yea ;) |
| 23:09.37 | Greenlight | That's life I guess |
| 23:09.50 | Greenlight | All we can do is try to minimise the impact of attacks |
| 23:09.59 | catphish | but often they'll only take down 1 global location |
| 23:10.02 | catphish | indeed |
| 23:10.29 | Greenlight | We had attacks from Palestine yesterday |
| 23:10.36 | Greenlight | Normally China or Russia though |
| 23:10.39 | gusto | yes |
| 23:10.47 | gusto | i had one from palestina as well |
| 23:10.48 | catphish | i dont know why they bother really |
| 23:11.08 | gusto | it wanted to call some numbers through my asterisk, i found it in the /asterisk/messages log |
| 23:11.10 | bluregard | Hi everyone. I have a quick question about indications.conf. The US info tone is defined as !950/330,!1400/330,!1800/330,0 I get most of this, but I'm a bit confused by the last 0. What's the point of defining a 0Hz tone with no duration on a tonelist that isn't repeated? |
| 23:11.30 | Greenlight | Its a delay? |
| 23:11.45 | Greenlight | (I'm just guessing - never used indications.conf) |
| 23:11.47 | bluregard | which I could understand if it was at the beginning |
| 23:11.55 | Greenlight | Isn't it repeated? |
| 23:12.01 | catphish | i got ddosd because i once redirected an illegal cedit card trade IRC network's dns to freenode |
| 23:12.09 | bluregard | no, ! tells it not to repeat. |
| 23:12.13 | jpsharp | gusto: That's what happened to my machine. Someone rooted it through that festering pile of crap called Wordpress, then snagged my * credentials to dial Ukraine and London on my dime. |
| 23:12.47 | catphish | wordpress + asterisk, hope you learned a valuable lesson :) |
| 23:12.48 | Greenlight | bluregard: Sorry, I don't know then |
| 23:13.00 | jpsharp | Yeah, I don't use wordpress any more. |
| 23:13.09 | Greenlight | I'd imagine ;) |
| 23:13.14 | catphish | asterisk is best served alone, as is wordpress |
| 23:13.16 | carrar | who puts asterisk and wordpress on the same machine? |
| 23:13.19 | catphish | for opposing reasons |
| 23:15.17 | gusto | how is it possible? |
| 23:16.11 | gusto | how did they get the credentials from asterisk over wordpress? did you have them in a database, or what? |
| 23:17.28 | p3nguin | catphish: jpsharp |
| 23:17.28 | jpsharp | I had one machine. |
| 23:17.28 | p3nguin | err |
| 23:17.28 | p3nguin | carrar: jpsharp does |
| 23:17.28 | catphish | i wouldnt share either with anything |
| 23:17.28 | bluregard | since the duration portion is optional does anyone know what the default duration for a tonelist element is? |
| 23:17.28 | carrar | heh |
| 23:17.28 | carrar | virtualize |
| 23:17.29 | carrar | xen baby xen |
| 23:17.29 | carrar | or KVM |
| 23:17.29 | Chainsaw | carrar: We're going to need a boat, eh, basket. |
| 23:17.29 | Chainsaw | +bigger |
| 23:17.30 | jpsharp | Twasn't beefy enough to run virtualizer. |
| 23:17.31 | WIMPy | Virtualisation has been proofed ineffective as security measurement. |
| 23:17.31 | carrar | keep enjoying the hacks then |
| 23:17.31 | WIMPy | So I wouldn't relay on that. |
| 23:17.31 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 23:17.31 | catphish | virt is a good part of a security policy |
| 23:17.31 | jpsharp | So I flushed that machine, bought a bigger machine I could run VMware on. |
| 23:17.31 | catphish | not bulletproof, but you're much less likely to mess it up then proper linux permissions |
| 23:17.32 | jpsharp | Nothing is bulletproof. |
| 23:17.32 | carrar | 100 yards of water |
| 23:17.32 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust1000.sgyl.cable.virginmedia.com) |
| 23:17.32 | carrar | is |
| 23:17.32 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 23:17.32 | jpsharp | People just bring bigger bullets. |
| 23:17.32 | WIMPy | If you only see it as replacement for propper permissions, you're fine, but the good old chroot would be just as good. |
| 23:17.32 | Greenlight | hmm damn connection... I miss cable, ADSL sicks |
| 23:17.32 | Greenlight | *sucks |
| 23:17.33 | *** join/#asterisk linguini (~user@c-67-170-164-151.hsd1.wa.comcast.net) |
| 23:17.33 | catphish | bigger bullet = more resistance in water |
| 23:17.33 | *** join/#asterisk Neptu (~Neptu@41.251.159.114) |
| 23:17.33 | carrar | also a "bullet" *IS* limited in size |
| 23:17.33 | carrar | otherwise it's no longer called a "bullet" |
| 23:17.33 | WIMPy | Motorised bullets have been in use for decades :-) |
| 23:17.34 | Greenlight | Missiles? |
| 23:17.34 | catphish | torpedo ftw |
| 23:17.41 | carrar | Missile != bullet |
| 23:17.52 | jpsharp | gusto: The machine was rooted through an exploit in Wordpress. |
| 23:18.05 | catphish | gusto: read /etc/asterisk/sip.conf usually does it |
| 23:18.18 | catphish | that is likely to be world readable |
| 23:18.42 | Greenlight | A lot of ITSP's use the technilogical equivelant of a "honey trap" - an easy to compromise server which lets them in, then bands that IP accross the network |
| 23:18.50 | Greenlight | *bans |
| 23:18.51 | catphish | and any one of many php exploits allow file reading |
| 23:19.23 | catphish | mmmm honey |
| 23:19.29 | Greenlight | ;) |
| 23:19.41 | WIMPy | It works! |
| 23:19.42 | WIMPy | :-) |
| 23:20.31 | Greenlight | Right folks, am off for some shut eye, it's been a pleasure as always, laters! |
| 23:20.57 | WIMPy | thanks for the reminder. Some of the newer config files in my /etc/asterisk are world readable. |
| 23:21.10 | catphish | :) |
| 23:21.53 | catphish | my pbxs only run asterisk, mysql and ssh, if either are compromised i'm screwed, so permissions seem pointless |
| 23:21.54 | gusto | yes, but apache should prevent /etc/asterisk and it's contents to be read |
| 23:21.57 | WIMPy | Well, confbridge.conf or ccss.conf probably aren't that interesting, but anyway. |
| 23:22.16 | WIMPy | Tell it so. |
| 23:22.35 | catphish | i wouldnt let apache anywhere near asterisk |
| 23:22.42 | WIMPy | If you allow your apache to read /, well... |
| 23:23.19 | catphish | i just go with the simple security of separate hardware where possible |
| 23:23.20 | gusto | hmm |
| 23:23.31 | gusto | so apache should be sandboxed then, like to openbsd! |
| 23:24.31 | catphish | indeed |
| 23:24.41 | gusto | so you say, better use two vservers than one big one |
| 23:25.01 | *** join/#asterisk tamiel (~tamiel@c-69-181-254-90.hsd1.ca.comcast.net) |
| 23:25.49 | catphish | better to grow your company and use 2 big ones ;) |
| 23:26.05 | gusto | i have no company |
| 23:26.05 | *** join/#asterisk jeffspeff (~Jeff@173.216.212.130) |
| 23:26.14 | jeffspeff | gusto, then get one! lol |
| 23:26.21 | gusto | well |
| 23:26.22 | catphish | make one, they'e fun |
| 23:26.39 | jeffspeff | i'm just kidding, i have no clue what anybody is talking about |
| 23:26.49 | gusto | no |
| 23:27.05 | jeffspeff | no what? |
| 23:27.08 | gusto | but that does not account only for asterisk, but for IMAP for example as well |
| 23:27.35 | gusto | so imagine my wordpress would be exploited someway, they could read my email as well |
| 23:27.47 | gusto | because it's all on the server |
| 23:27.57 | gusto | right? |
| 23:28.08 | catphish | yes but also consider that imap might be exploited and they might deface your website |
| 23:28.19 | catphish | all services should be protected in the same manner |
| 23:28.23 | p3nguin | Or flush your toilet. |
| 23:28.41 | catphish | p3nguin: [citation needed] |
| 23:28.42 | jeffspeff | i deface his/her mom! oh snap! |
| 23:28.47 | jeffspeff | sorry.. i'm done |
| 23:28.50 | jeffspeff | couldn't help it |
| 23:28.57 | p3nguin | If services run on the same host, exploiting one leads to access of the others. |
| 23:29.03 | p3nguin | (usually) |
| 23:29.12 | p3nguin | Jails help. |
| 23:29.15 | gusto | of course |
| 23:29.22 | gusto | well, when you have freebsd |
| 23:29.25 | catphish | indeed, but toilets are usually separated by default |
| 23:29.30 | WIMPy | Yes. Put all the haxx0rs there. |
| 23:29.36 | gusto | but the most providers only give Vservers |
| 23:29.40 | catphish | linux has chroot jails |
| 23:29.47 | gusto | but there are some products now with KVM |
| 23:29.51 | catphish | and openvz |
| 23:30.08 | catphish | i tend to go with kvm these days |
| 23:30.53 | gusto | so a virtualized freebsd in a KVM with jails inside would be a good option, right? |
| 23:31.14 | catphish | yes it would |
| 23:31.46 | gusto | so i have to learn jails sometime |
| 23:32.36 | gusto | sometime soon |
| 23:32.47 | catphish | if i want to create a prefix that sets a variable can i easily match the prefix, strip back the exten and start again from the top? |
| 23:32.53 | catphish | just with a goto? |
| 23:33.26 | WIMPy | catphish: Yes and no. |
| 23:34.04 | WIMPy | You can do it, but you have to be aware that the first match decides if you get in to the dialplan at all. |
| 23:34.04 | catphish | i want to create a local exten that matches a "withold my number" prefix, sets a variable, then removes the prefix and starts again |
| 23:34.16 | catphish | yeah that makes sense |
| 23:34.26 | WIMPy | So if you've got anything that uses overlap dialling, that breaks it. |
| 23:34.27 | catphish | so i'd lose the ability to reject the call |
| 23:34.28 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 23:35.30 | catphish | and am i free to modify exten? |
| 23:35.38 | catphish | or should i use local? |
| 23:35.57 | WIMPy | I haven't take it that far, yet, but I guess you could do some loopback zwitch trickery to make it work in all cases. |
| 23:36.20 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 23:36.27 | WIMPy | Modify extension=Goto() |
| 23:37.25 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 23:37.26 | WIMPy | Where do you want to send the calls? |
| 23:37.39 | WIMPy | Both local and remote? |
| 23:37.59 | catphish | well i have a local context that contains routes for outbound calls |
| 23:38.01 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 23:38.21 | catphish | i want to match a withhold my number prefix, then continue to route the call as if the prefix were missing |
| 23:38.36 | WIMPy | In that case I'd go and double them. |
| 23:38.53 | catphish | the only issue is that i want to strip the exten so it isn't logged in my cdr with the prefix |
| 23:39.47 | catphish | i can easily do Goto(${EXTEN:3},1) |
| 23:39.47 | WIMPy | Oh, CDR it a different matter. |
| 23:39.47 | catphish | but that won't change the exten variable will it? |
| 23:39.48 | catphish | or the cdr(exten_ |
| 23:39.54 | catphish | *) |
| 23:40.00 | WIMPy | No |
| 23:41.08 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 23:41.27 | catphish | i can fix it with a local |
| 23:41.38 | catphish | (if necessary) |
| 23:41.52 | catphish | but it seems like an unnecessary extra channel in this case |
| 23:43.39 | *** join/#asterisk din3sh (~din3sh@41.136.83.132) |
| 23:43.39 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 23:45.28 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 23:46.48 | catphish | oh, i think GOTO *does* change the exten variable |
| 23:46.55 | catphish | need to test though |
| 23:47.14 | carrar | PLEASE TEST AND REPORT BACK ASAP |
| 23:47.16 | catphish | but i'll need to modify the cdr separately which should be possible with the cdr command |
| 23:47.16 | [TK]D-Fender | ${EXTEN} is where you ARE |
| 23:47.28 | catphish | [TK]D-Fender: great :) |
| 23:47.39 | catphish | i should be ok then |
| 23:47.47 | catphish | i assume i can modify cdr(exten) |
| 23:47.52 | WIMPy | Modify extension=Goto()=modify extension... |
| 23:48.14 | carrar | This=is=getting=crazy== |
| 23:48.18 | catphish | yeah i was throw off by cdr(exten) |
| 23:48.28 | WIMPy | ... but here's my number... |
| 23:48.29 | catphish | but that can be set independently i hope |
| 23:48.43 | catphish | ...but its witheld... |
| 23:49.46 | catphish | http://cdn.smosh.com/sites/default/files/bloguploads/funny-call-me-maybe-nemo.jpg |
| 23:51.44 | [TK]D-Fender | Jut realized my D50 doesn't have a HOOK SWITCH. F-ing freaky... |
| 23:51.56 | catphish | weird |
| 23:52.21 | catphish | but how can you pulse dial l33t style? |
| 23:52.50 | [TK]D-Fender | U KANT. I R FAIL |
| 23:55.52 | catphish | <PROTECTED> |
| 23:55.52 | catphish | <PROTECTED> |
| 23:55.52 | catphish | <PROTECTED> |
| 23:55.59 | catphish | hopefully that'll do me |
| 23:57.06 | WIMPy | And if you do it for each route instead of a goto, you don't mess up any matching. |
| 23:57.48 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 23:57.50 | catphish | matching? |
| 23:57.59 | catphish | doesn't the goto handle that correctly? |
| 23:58.14 | WIMPy | The initial dialplan matching. |
| 23:58.30 | catphish | match-as-you-dial style? |
| 23:58.42 | WIMPy | Yes. |
| 23:58.50 | [TK]D-Fender | catphish: CDR() is READ-ONLY |
| 23:58.50 | catphish | it's not something i've ever depended on but i see the point |
| 23:58.58 | [TK]D-Fender | except for "userfield |
| 23:59.00 | catphish | is it :| |
| 23:59.00 | WIMPy | But it makes a difference in signalling as well. |
| 23:59.12 | gusto | is someone here using asterisk on a openwrt router? |
| 23:59.17 | catphish | i was sure i modified cdr before |
| 23:59.25 | catphish | maybe not built in fields |
| 23:59.38 | catphish | if i cant modify cdr i have a much more annoying problem |
| 23:59.41 | WIMPy | ~polls |
| 23:59.41 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 23:59.47 | WIMPy | gusto: ^^ |
