| 01:04.48 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:04.48 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.6, 1.6.1.18, 1.6.0.26 (2010/03/12), 1.4.30 (2010/03/12), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 01:05.01 | p3nguin | infobot: Glad you could join us. |
| 01:05.26 | ariel_ | is hungry |
| 01:05.38 | p3nguin | just had pizza |
| 01:05.44 | Katty | makes ariel_ dinner |
| 01:05.49 | ariel_ | t/y |
| 01:06.24 | ariel_ | But I am looking at a box of honey nut Cheerios that just looks good... |
| 01:06.59 | Katty | that doesn't sound very good to me, but that's okay |
| 01:07.10 | Katty | at least we won't fight over who gets what breakfast cereal |
| 01:09.05 | Katty | more of a cinnamon toast crunch person |
| 01:09.43 | sawgood | So, in a nutshell ... why is it that when I have these three remote phones on a LAN (pointing to an IP PBX with a static live public IP) ... the phones register fine, but they cannot 'call each other' ... |
| 01:10.09 | ariel_ | nat |
| 01:10.15 | p3nguin | ~sipnat |
| 01:10.16 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 01:10.46 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:11.12 | Katty | basically what's happening is they're registering as publicipfoo on your asterisk box |
| 01:11.29 | Katty | so when 1 extension goes to call another, it's just sending the call to publicipfoo |
| 01:11.36 | Katty | cause that's where it's registered at |
| 01:11.41 | sawgood | right ... |
| 01:11.53 | p3nguin | Extensions cannot call another anything. extensions are not phones. |
| 01:11.55 | Katty | unlike internal stuff registering as 168, 178, etc |
| 01:12.43 | Katty | p3nguin: your beating a dead horse |
| 01:12.49 | p3nguin | My beating? |
| 01:13.18 | p3nguin | OH! You meant, "You're beating a dead horse." |
| 01:13.31 | Katty | sawgood: does that make sense now? |
| 01:13.34 | p3nguin | The possessive threw me. |
| 01:13.36 | sawgood | 100% |
| 01:13.42 | Katty | sawgood: good. |
| 01:13.49 | Katty | p3nguin: i'm not in the mood for your sarcastic comments. |
| 01:13.56 | p3nguin | Learn English. |
| 01:14.46 | sawgood | So, when a person has more than one remote phone on a LAN ... they need a good NAT router |
| 01:15.07 | p3nguin | sawgood: Regardless of what some people try to portray, phones are not extensions and extensions are not phones. Remember that in the future. |
| 01:15.49 | p3nguin | sawgood: First thing I would make sure of is that the peer definitions of the phones are configured 100% correctly. |
| 01:15.51 | sawgood | p3nguin: I knew that as well. But, for convernsation skate (90% of the time) most people would call an extension a 'phone' using the situation I gave |
| 01:16.20 | sawgood | p3nguin: looking at that now ... thank you |
| 01:16.55 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-64-227-77.mia.bellsouth.net) |
| 01:17.18 | p3nguin | sawgood: If you still have the problem after verification, try running phones on different ports. The phone ports will need to be configured in the corresponding peer definition for asterisk's part... I don't know what to tell you about the phone's configuration. |
| 01:17.52 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 01:18.22 | p3nguin | Example: phone1 could use the standard port 5060, phone2 could use 5061. |
| 01:18.59 | p3nguin | I can't guarantee it will help, but I would test it if I were in your situation. |
| 01:19.42 | *** join/#asterisk darkavanger (~Darkavang@41.225.137.138) |
| 01:19.58 | darkavanger | hi every one |
| 01:20.06 | darkavanger | ive got a problem with RTP |
| 01:20.32 | darkavanger | calls from inbound DID come on one way |
| 01:20.42 | darkavanger | here is my rtp log |
| 01:20.48 | p3nguin | Did you forward the ports at your firewall? |
| 01:21.14 | darkavanger | yes on my iptables |
| 01:21.19 | darkavanger | i have no firewall |
| 01:21.28 | darkavanger | i m on a dedicated server |
| 01:21.30 | darkavanger | iptables -A INPUT -p udp --dport 20000 -j ACCEPT |
| 01:21.34 | p3nguin | iptables creates the firewall, so yeah you do. |
| 01:21.35 | darkavanger | iptables -A INPUT -p udp --dport 10000 -j ACCEPT |
| 01:21.47 | p3nguin | That accepts two ports. |
| 01:21.51 | darkavanger | that's what i did |
| 01:21.56 | ariel_ | remove it |
| 01:22.03 | p3nguin | You needed --dport 10000:20000 |
| 01:22.08 | darkavanger | ah ok |
| 01:22.11 | p3nguin | range |
| 01:22.18 | darkavanger | how can i remove it |
| 01:22.34 | p3nguin | iptables -D INPUT -p udp --dport 20000 -j ACCEPT |
| 01:22.38 | p3nguin | same for the other |
| 01:22.48 | darkavanger | ok thanks |
| 01:24.26 | jsjc | I get this when I try to call myself... Call from '1234' to extension '1234' rejected because extension not found. |
| 01:24.28 | jsjc | I get this when I call to the SIP account from outside Call from '' to extension '1231' rejected because extension not found. |
| 01:24.29 | jsjc | Anybody has an idea, I been told about naming devices... but I have changed names around and nothing, contexdt is sip and I have all properly in the dialplan (at least i am pretty convinced...) |
| 01:25.04 | p3nguin | jsjc: The extensions don't exist within the context(s) where the calls are being sent. |
| 01:25.06 | darkavanger | i ve set the firewall but the problem persists |
| 01:25.11 | darkavanger | <PROTECTED> |
| 01:25.18 | darkavanger | <PROTECTED> |
| 01:25.34 | p3nguin | darkavanger: More info is going to be needed. |
| 01:25.42 | jsjc | p3nguin: I have all in context called sip. and the this within exten => 1234,blahblah |
| 01:25.46 | jsjc | isnt that extension? |
| 01:26.02 | p3nguin | That shows that 1234 is an extension, yes. |
| 01:26.30 | jsjc | well that is exactly how i have it within [sip] context |
| 01:26.34 | p3nguin | I would probably not call a context "sip", though. I would want it to be more descriptive. |
| 01:26.42 | jsjc | and in sip.conf I have everything in sip context |
| 01:26.47 | p3nguin | You'll have to provide more info. |
| 01:26.49 | darkavanger | p3nguin: what kind of infos ?? i'll tell you all : asterisk 1.4.30 i686 |
| 01:26.54 | darkavanger | freepbx |
| 01:27.10 | jsjc | p3nguin: let me get a pastebin together |
| 01:27.20 | *** join/#asterisk blaines (~blaines@wsip-70-184-115-122.ph.ph.cox.net) |
| 01:27.23 | darkavanger | p3nguin: can you help me set my sip_nat.conf |
| 01:27.38 | p3nguin | darkavanger: If the problem resides on your configurations, there is nothing we can do about it here. We don't do FreePBX here. |
| 01:27.45 | darkavanger | i do not have an internal ip it is the same as the external |
| 01:28.11 | darkavanger | it think it is a sip_nat;conf problem |
| 01:28.22 | p3nguin | Maybe. |
| 01:28.34 | p3nguin | doesn't know jack about sip_nat.conf, though |
| 01:28.46 | darkavanger | ok |
| 01:28.51 | p3nguin | But then again, since we don't do FreePBX here, why should I? |
| 01:28.54 | darkavanger | thanks!! |
| 01:31.13 | *** join/#asterisk jtodd (upuj3hacwk@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 01:31.13 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 01:33.10 | jsjc | p3nguin: Here are the sip.conf and extensions.conf http://pastebin.com/UXpeWvnK |
| 01:34.08 | p3nguin | jsjc: Where is the call coming FROM when you get the "extension not found" error? |
| 01:34.11 | jsjc | as you can see everything is within sip context |
| 01:34.24 | jsjc | I can either call from 1234 to 1234 and get that |
| 01:34.46 | p3nguin | When you say "from 1234" do you mean SIP/1234? |
| 01:34.58 | jsjc | yes |
| 01:35.14 | jsjc | and I can try to call SIP/telefonica and the same issue |
| 01:35.34 | p3nguin | 1234's definition has context specified 2 times. |
| 01:35.39 | jsjc | I have a register=>blah blah that has /1231 at the end |
| 01:35.44 | *** join/#asterisk adelas (~adela@pool-71-108-37-251.lsanca.dsl-w.verizon.net) |
| 01:35.50 | *** join/#asterisk siera04 (~chatzilla@113.18.8.82) |
| 01:35.59 | jsjc | p3nguin: true! |
| 01:36.04 | p3nguin | I need to see a sip debug. |
| 01:36.04 | jsjc | lets see... I been killing myself all night after wokring yesterday |
| 01:36.10 | jsjc | if is that i will shoot myself! |
| 01:37.09 | jsjc | hmnm was not that |
| 01:37.14 | jsjc | let me get you a sip debug |
| 01:37.21 | jsjc | where you want the debug from? |
| 01:37.23 | p3nguin | Ideally, you'll create some better context names and change your general context to be called something else. |
| 01:37.31 | jsjc | 1234 to 1234? |
| 01:37.31 | jsjc | or any will be fine? |
| 01:37.34 | p3nguin | "sip set debug on" |
| 01:37.41 | p3nguin | Then make a call that ends in failure. |
| 01:37.48 | jsjc | ok |
| 01:37.48 | p3nguin | Then "sip set debug off" |
| 01:37.57 | p3nguin | paste the ENTIRE thing in pastebin.com |
| 01:38.14 | p3nguin | It could be large, so make sure you get all of it. |
| 01:39.49 | p3nguin | I'm also concerned about the allow=alaw,ulaw,gsm line. I didn't know that was valid. |
| 01:41.30 | jsjc | p3nguin: all added into pastebin! |
| 01:41.39 | p3nguin | need URL |
| 01:41.44 | jsjc | i dont know if is valid I am new in this I been reading and pretty much copy paste from here and ther |
| 01:41.50 | jsjc | p3nguin: same url |
| 01:41.55 | p3nguin | can't be |
| 01:41.58 | jsjc | yes |
| 01:42.02 | jsjc | ammended |
| 01:42.06 | p3nguin | I'll wait. |
| 01:42.13 | jsjc | http://pastebin.com/ZhGQEdEP |
| 01:43.39 | p3nguin | Hmm. Did you remember to save your extensions.conf and then run "dialplan reload" afterward? |
| 01:44.34 | jsjc | yes |
| 01:44.37 | jsjc | i restarted asterisk |
| 01:44.50 | jsjc | because I do not have dialplan reload (dont know why |
| 01:45.04 | p3nguin | Can you show me what "dialplan show 1234@sip" says? |
| 01:45.05 | ariel_ | what version of asterisk |
| 01:45.10 | jsjc | just dialplan debug,set,show |
| 01:45.24 | ariel_ | use extension reload |
| 01:45.33 | jsjc | There is no existence of 'sip' context |
| 01:45.34 | jsjc | Command 'dialplan show 1234@sip' failed. |
| 01:45.44 | jsjc | ohh so the problem asterisk is not seen this bloody dialplan!? |
| 01:46.19 | p3nguin | I don't see anything wrong with your dialplan. It's VERY simplistic. |
| 01:46.56 | jsjc | that is why is driving nuts! I am almost to throw towel and get a pro to config my simplistic setup hehe |
| 01:47.14 | jsjc | but i really want to do it myself so i learn basics to be able to administer it in futre |
| 01:47.23 | sun28 | moin |
| 01:47.51 | p3nguin | Without knowing why your system is missing commands and doesn't recognize a valid extension@context, I don't know what else to do. |
| 01:48.12 | jsjc | ok i will reinstall this asterisk thing and test again |
| 01:48.19 | p3nguin | Which version are you using? |
| 01:48.25 | jsjc | be back in some time with more simple questions |
| 01:48.28 | jsjc | latest 1.6.2.6 |
| 01:48.33 | ariel_ | do you have all the config files |
| 01:48.45 | jsjc | ariel_: what you mean by all config files? |
| 01:48.51 | ariel_ | what os and how did you install it |
| 01:49.16 | jsjc | ubuntu apt and config files I have just essential for my system |
| 01:49.17 | ariel_ | ... " /etc/asterisk directory should have lots of files |
| 01:49.22 | jsjc | modules I have noloaded lots of them |
| 01:49.25 | ariel_ | oh |
| 01:49.34 | jsjc | ariel_: I took away all the ones I didnt needed |
| 01:50.01 | jsjc | so just have about 14 or so |
| 01:50.15 | jsjc | but as of now just really need 5-6 |
| 01:50.44 | ariel_ | really? |
| 01:51.25 | ariel_ | basic, modules.conf asterisk.conf extension.conf features.conf wait many more are needed for the basic |
| 01:51.39 | jsjc | well of what i have larned in the last couple of days a system with asterisk.conf,extensions.conf,sip.conf its ready to go |
| 01:51.43 | ariel_ | even rtp.conf |
| 01:51.51 | jsjc | rtp.conf it is needed? |
| 01:51.58 | ariel_ | yes |
| 01:52.02 | p3nguin | A lot of the configs will just override the built-in defaults. |
| 01:52.19 | p3nguin | Let me remove my rtp.conf and see what happens. |
| 01:52.24 | jsjc | so everyone told me that the samples are not properly working configs |
| 01:52.31 | jsjc | so what I should keep in my system? |
| 01:52.56 | jsjc | I am interested in knowing how to setup the most minimal sterisk setup without anything added so I can go adding as I need |
| 01:53.22 | jsjc | less things can go wrong and as I get knowing my system I can start with more things |
| 01:53.36 | ariel_ | yes but no |
| 01:53.51 | jsjc | ok so sould I get all those samples an keep them there? |
| 01:54.41 | ariel_ | for now |
| 01:54.44 | p3nguin | Sample configs are not working configs. They are for reference only. |
| 01:55.14 | ariel_ | yes but the one I feel he has issues are more like the modules.conf and asterisk.conf |
| 01:55.37 | p3nguin | What module deals with rtp.conf? sip reload doesn't reload rtp.conf. |
| 01:56.37 | jsjc | i know that dahdi does |
| 01:56.58 | jsjc | because when you make a call trough dahdi and not have rtp module then freaks out |
| 01:57.00 | jsjc | hehe |
| 01:57.04 | jsjc | I found out las tnight |
| 01:57.18 | p3nguin | Doesn't seem to. |
| 01:57.32 | jsjc | but I am away from dahdi til I can get first IRQ's in my system sorted so now I am with SIP for the moment to learn how this works and understand dialplan basics |
| 01:57.54 | p3nguin | dahdi reload only parses chan_dahdi.conf and users.conf. |
| 01:58.10 | ariel_ | users.conf ? |
| 01:59.14 | jsjc | I have users.conf without info tho... |
| 01:59.32 | ariel_ | users.conf is only needed if you have the asterisk-gui |
| 01:59.37 | p3nguin | I just wanted to test what happened when there was no rtp.conf. |
| 01:59.42 | p3nguin | That's not accurate. |
| 01:59.50 | ChannelZ | You lose your hearing |
| 01:59.59 | jsjc | I have a new version let me put my config in place. |
| 02:00.00 | p3nguin | You can use users.conf without asterisk-gui. |
| 02:01.12 | p3nguin | I speculate that without rtp.conf, your rtp port range will be the default range of 5000-31000. |
| 02:01.39 | p3nguin | I just wanted to make sure asterisk didn't choke if the file was missing entirely. |
| 02:01.51 | ariel_ | it does |
| 02:02.13 | p3nguin | Without finding that module will re-parse the file, I can't test. |
| 02:02.20 | p3nguin | s/that/what/ |
| 02:02.20 | ariel_ | brb, my child just got up and she needs to be put back to bed. |
| 02:11.52 | *** join/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 02:11.56 | jsjc | I just realized with default config files this thign does not even start! |
| 02:12.11 | p3nguin | good |
| 02:12.24 | jsjc | hehe I been from bad into worst... |
| 02:12.34 | jsjc | but hopefully will make me learn how this system works |
| 02:12.46 | jsjc | hehe lets try.. |
| 02:12.48 | p3nguin | Let's see what manxpower-work thinks about this. |
| 02:12.49 | ManxPower-work | Asterisk is not a PBX. It is a telephony toolkit that helps you build a PBX. Much like a library helps you write a program. |
| 02:12.59 | p3nguin | manxpower-work: http://pastebin.com/ZhGQEdEP |
| 02:13.45 | ManxPower-work | what am I looking for? |
| 02:13.54 | jsjc | Never nothing is easy! but it is good at least for learning. |
| 02:14.13 | p3nguin | manxpower-work: Call from SIP/1234 to extension 1234@sip fails with extension not found. |
| 02:15.18 | ManxPower-work | set allowguest=no in sip.conf [general] do a module reload chan_sip.so and retest |
| 02:15.26 | *** join/#asterisk blaines (~blaines@ip68-106-24-21.ph.ph.cox.net) |
| 02:15.28 | jsjc | is there anywhere where could find a set of .conf that will the system "work" at least in that sort of conf? |
| 02:15.34 | p3nguin | no |
| 02:15.37 | ManxPower-work | It still won't work, but you should get a more helpful erro message. |
| 02:15.45 | jsjc | well cannot retest now because need to get back everything as it was hehe |
| 02:15.51 | jsjc | give me just few minutes |
| 02:16.05 | p3nguin | jsjc: Well, I take that back. I provided a working example... let me get it for you. |
| 02:16.34 | p3nguin | jsjc: http://pastebin.com/m59d17875 |
| 02:17.06 | p3nguin | This working example has a couple static peers, a dynamic peer with register statement, and a phone. |
| 02:17.29 | jsjc | ok let me work within that one... |
| 02:17.39 | jsjc | let me get all system to same situation and brb |
| 02:18.35 | p3nguin | I'm still seriously worried that "dialplan reload" doesn't exist for you. |
| 02:20.24 | jsjc | no does not |
| 02:20.28 | jsjc | i was worried with it as well |
| 02:20.29 | ManxPower-work | p3nguin, usually caused by lack of general of globals or crap before them in sip.conf |
| 02:20.38 | ManxPower-work | general or globals |
| 02:21.34 | p3nguin | His paste does reflect both of those. I'm kinda stumped -- everything looked in order to me. It was so basic, there isn't much that can be wrong. |
| 02:22.34 | ariel_ | you know that asterisk starts up without config files... just tested it. but does not load even dialplan or nothing just is a shell |
| 02:25.20 | jsjc | ok so the not having config is not the issue... |
| 02:25.56 | ariel_ | yes it is |
| 02:25.59 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:26.28 | ariel_ | it starts in a shell but does nothing |
| 02:29.37 | jsjc | but i have the essential config and modules for what i want. |
| 02:29.58 | jsjc | ManxPower-work: I have tried with allowguest=no under general but still same... |
| 02:31.18 | jsjc | extension not found... |
| 02:31.23 | jsjc | let me try with that sip.conf |
| 02:33.28 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 02:37.16 | ManxPower-work | huh? I said put it in the [general] section of sip.conf |
| 02:37.29 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 02:37.39 | ManxPower-work | also pastebin your extensions.conf pastebin.va |
| 02:37.53 | ManxPower-work | or sans-typoe pastebin.ca or pastebin.com |
| 02:40.31 | jsjc | I think the issue is that asterisk is not reading the dialplan.. |
| 02:41.07 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 02:41.10 | jsjc | extensions is in the same pastebin as I put before just a bit underneath... |
| 02:43.16 | ManxPower-work | which pastebin is that? |
| 02:44.23 | ManxPower-work | you are correct. extensions.conf is not being read by Asterisk. |
| 02:44.31 | ManxPower-work | once I see the extensions.conf I can say more. |
| 02:44.44 | *** join/#asterisk tengulre (~tengulre@125.71.208.16) |
| 02:46.19 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 02:46.31 | ChannelZ | Ah great, Comcast has started routing things to me ipv6 and it's totally not working |
| 02:48.01 | *** join/#asterisk jhirley (~jhirley@h69-21-54-248.ldlwvt.dsl.dynamic.tds.net) |
| 02:48.06 | *** join/#asterisk jhirley_ (~jhirley@h69-21-54-248.ldlwvt.dsl.dynamic.tds.net) |
| 02:48.50 | jsjc | it is extensions.conf mentioned somewhere to asterisk? so asterisk knows? |
| 02:49.01 | jsjc | is there anymodule that need to be loaded so extensions conf it is read? |
| 02:49.14 | jsjc | maybe in my no-load frenzy i selected some important modules |
| 02:49.27 | *** part/#asterisk jhirley_ (~jhirley@h69-21-54-248.ldlwvt.dsl.dynamic.tds.net) |
| 02:51.08 | ChannelZ | extensions.conf is automatically used by the pbx core |
| 02:51.50 | jsjc | so that might not be the issue... |
| 02:52.34 | ChannelZ | pbx_config |
| 02:53.28 | *** join/#asterisk sourcode (~code@ppp-58-8-121-192.revip2.asianet.co.th) |
| 02:55.48 | jsjc | haha let me see i think i did disabled... |
| 02:56.06 | jsjc | haha it is ... |
| 02:56.14 | jsjc | so out of the pbx_ modules |
| 02:56.25 | jsjc | config it is important what about loopback and realtime? they sound could be |
| 02:57.16 | ChannelZ | leave them all |
| 02:58.03 | ChannelZ | you probably don't need dundi or ael unless you're using either of those |
| 02:58.18 | jsjc | ok! let see I remember one of them gave me the shit.. yes thats the one! dundi and ael |
| 02:58.19 | jsjc | ehhe |
| 02:58.32 | jsjc | so I thought well if they have pbcx in front and they give me the crap screww all of them! |
| 02:59.09 | jsjc | hehe have dialplan reload! |
| 02:59.13 | jsjc | dis si looking better |
| 02:59.18 | *** join/#asterisk wwalker (~wwalker@mailbox.eclipsing.com) |
| 02:59.26 | ChannelZ | no, pbx is pretty much what asterisk is |
| 02:59.34 | jsjc | hehe this is looking good!!! let me see |
| 02:59.38 | wwalker | anyone used timeout with originate and manager? |
| 02:59.53 | wwalker | I keep setting timeout but it is ignored |
| 03:00.20 | jsjc | ohh this is awsome! at least I get new errors |
| 03:00.22 | jsjc | hehe |
| 03:00.22 | wwalker | is there a debug option that will have manager print out the request it receives |
| 03:02.14 | *** join/#asterisk sourcode (~code@ppp-58-8-121-192.revip2.asianet.co.th) |
| 03:03.30 | jsjc | yeah!! love you ChannelZ ! you solved my biggest headache of the day! |
| 03:03.53 | jsjc | how sexy is having the first call in asterisk in my life... ohh feels good! |
| 03:07.46 | jsjc | when i load asterisk I get this two lines Cannot find variable 'SIPPEER' in tree 'description' Cannot find variable 'SIPCHANINFO' in tree 'description' why this could be? |
| 03:08.01 | ManxPower-work | jsjc, perhaps you should stop disabling stuff until you know what you are doing |
| 03:08.26 | jsjc | ManxPower-work: but beofre my asterisk had 27million errors on load |
| 03:08.37 | jsjc | now after disabling whatever gave errors (if I do not needed) now it is clean. |
| 03:08.48 | jsjc | no major errors just warnings |
| 03:08.59 | ManxPower-work | except for hours and hours of wasted time |
| 03:09.04 | jsjc | because the truth is that I do not need to waste resources if I do not need a module why loading it? |
| 03:09.25 | jsjc | ManxPower-work: that is totally true... over a day of headache |
| 03:10.03 | jsjc | but i am learning and learning by mistake is the best way to understand the things (at least I think so) |
| 03:14.11 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 03:16.29 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 03:19.19 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 03:21.36 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 03:25.18 | *** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 03:25.19 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 03:27.18 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 03:28.57 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 03:37.03 | *** join/#asterisk diegomad (mad@186.0.4.182) |
| 03:41.02 | ManxPower-work | you are also wasting many people's time. |
| 03:43.25 | magicblaze007 | [TK]D-Fender: I need some help. Are you there? |
| 03:43.39 | [TK]D-Fender | magicblaze007: What is it? |
| 03:44.05 | [TK]D-Fender | magicblaze007: As you seem to feel the need to target individuals for support.... |
| 03:44.21 | magicblaze007 | Sorry to disturb you. |
| 03:44.30 | magicblaze007 | if you are busy, I can ask later. |
| 03:45.47 | [TK]D-Fender | magicblaze007: just ask already.... |
| 03:45.50 | magicblaze007 | I've a SPA 3102 which wont talk my asterisk server. I've tried everything I could think of. I've no clue how to check if it is emitting ethernet packets that my router is stopping. I tried wireshark, but perhaps I dont know how to use it...cant see any packets there. |
| 03:46.21 | [TK]D-Fender | magicblaze007: Where is it relative to your * server? Do you have SIP debug enabeld on it? |
| 03:46.52 | magicblaze007 | yes. sip debug is enabled. My ata is behind a router... My asterisk has a public static ip. XLite works from behind my router. |
| 03:47.23 | magicblaze007 | Scenario 1 : http://www.bestit.net.au/customer-service/71-voip/183-how-do-i-setup-and-connect-my-spa-3102-.html?showall=1 |
| 03:47.29 | magicblaze007 | Without the "Filter" Box. |
| 03:48.09 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 03:49.01 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 03:49.01 | magicblaze007 | my spa-3102 has software version 5.1.10. and i configured its line 1 to talk to my asterisk box. |
| 03:49.20 | [TK]D-Fender | magicblaze007: What port on your 3102 do you have connected to your other router? |
| 03:49.36 | magicblaze007 | "Internet" |
| 03:50.00 | [TK]D-Fender | magicblaze007: imagebin screen-shots of its config and status pages. |
| 03:50.08 | magicblaze007 | k |
| 03:50.20 | [TK]D-Fender | magicblaze007: And show me that * is listening on the interface its supposed to be targeting and that SIP debug is enabeld |
| 03:51.07 | magicblaze007 | k |
| 03:51.14 | magicblaze007 | lemme do that |
| 04:11.26 | *** join/#asterisk blaines (~blaines@ip68-106-24-21.ph.ph.cox.net) |
| 04:11.28 | hardwire | hows magicblaze007 coming along? |
| 04:12.28 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 04:12.42 | [TK]D-Fender | hardwire: slowly |
| 04:12.46 | carrar | perhaps more blazing then progress |
| 04:14.32 | hardwire | magicblaze007: pick up the pace! |
| 04:14.33 | hardwire | :P |
| 04:14.45 | hardwire | grabs the 2x4 of knowledge. |
| 04:18.01 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
| 04:18.35 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 04:27.28 | ChannelZ | all that 'magic blaze' makes your brain slow |
| 04:33.38 | *** join/#asterisk blaines (~blaines@ip68-106-24-21.ph.ph.cox.net) |
| 04:39.25 | sawgood | any tips for this boggle: I have three remote phones (sitting on a NAT LAN using 192.168.1.x addresses) ... all three phones register just fine (as remote phones) (even though all three are using port 5060) |
| 04:39.41 | sawgood | On the Asterisk box (sip show peers) shows all three registered |
| 04:40.20 | sawgood | phone 1 on port 5062 ... phone 2 on 26499 and phone 3 on 5060 |
| 04:40.36 | sawgood | I guess Asterisk auto-configured the ports ... |
| 04:40.54 | sawgood | I have no idea why phone 2 got UDP port 26499 (5060 is set on the phone and in Asterisk) |
| 04:42.27 | *** join/#asterisk zzlane33333 (~zzlane333@c-98-223-211-192.hsd1.in.comcast.net) |
| 04:43.15 | zzlane33333 | Is anyone on here an Asterisk consultant? |
| 04:43.22 | sawgood | When looking at sip show peers (what does the D stand for in dyn) |
| 04:43.32 | sawgood | zzlane33333: its easier to ask your question |
| 04:43.46 | *** join/#asterisk blaines (~blaines@ip68-106-24-21.ph.ph.cox.net) |
| 04:45.26 | zzlane33333 | I have a mixed environment of an Asterisk 1.6 softswitch with Asterisk/FreePBX 1.4 pbx's that I am having some configuration issues with. Most notable, the dial plan is not allowing a 1 to be in front of the 9 digit number, also caller ID is not passing from the 1.4 systems to the 1.6 softswitch. |
| 04:46.18 | sawgood | zzlane33333: do you have the ability to grab Ethernet frames with either Wireshark and a HUB and/or with tcpdump? |
| 04:46.25 | zzlane33333 | absolutely |
| 04:46.38 | zzlane33333 | the firewall between the pstn equipment at our datacenter and my equipment is running pfsense |
| 04:46.44 | zzlane33333 | I can do dumps from that interface |
| 04:47.07 | sawgood | So, you are actually using "Asterisk" as a voice softswitch? |
| 04:47.28 | zzlane33333 | its configured more as a "proxy" than a "softswitch" |
| 04:47.31 | sawgood | As in a SBC for carrier services? |
| 04:47.41 | sawgood | proxy for SIP registrations? |
| 04:47.49 | zzlane33333 | all the trunks are IAX2 |
| 04:47.56 | zzlane33333 | SIP is only used for handsets |
| 04:47.57 | sawgood | oh .. sorry ... |
| 04:48.01 | zzlane33333 | np |
| 04:48.04 | zzlane33333 | I forgot to mention that |
| 04:48.17 | sawgood | SIP in your case ... SIP can be used for trunks all day/night |
| 04:48.38 | sawgood | Why do you want to use IAX instead of a SIP trunk? |
| 04:49.25 | zzlane33333 | Because SIP uses RTP ports for calls and IAX2 uses the same port it setup for the persistent connection |
| 04:49.42 | sawgood | right ... right .... |
| 04:50.58 | zzlane33333 | Primarily thats it |
| 04:51.24 | zzlane33333 | It's all local trunking not going out over the web |
| 04:51.34 | zzlane33333 | The PSTN equipment is all at the datacenter |
| 04:52.03 | sawgood | nice ... so you have 'a private IAX circuit' from your datacenter to the PSTN? |
| 04:52.26 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-64-223-140.mia.bellsouth.net) |
| 04:52.27 | zzlane33333 | yes |
| 04:52.42 | sawgood | oh ... a IAD at the site (using a IAX connection) ... to your datacenter which has probably a SS7 type connection to the PSTN |
| 04:52.46 | sawgood | very nice indeed! |
| 04:52.49 | sawgood | congratulations |
| 04:53.19 | sawgood | Isn't the cost for a private IAD using a 3rd party carrier's network to your datacenter rather 'expensive'? |
| 04:53.27 | zzlane33333 | thanks, the guy who set it all up for them was quite clever |
| 04:53.38 | sawgood | I see ... |
| 04:53.49 | zzlane33333 | I'm getting a good deal |
| 04:53.49 | sawgood | This is definitely not the norm ... |
| 04:53.51 | zzlane33333 | nope |
| 04:54.11 | sawgood | Most people cram it all over DSL or a cable connection and 'complain' at slightest bit of 'jitter' |
| 04:54.27 | zzlane33333 | We are going to do this over DS1 links back to the private circuit |
| 04:54.31 | zzlane33333 | no commodity for our customers |
| 04:54.45 | sawgood | amazing ... |
| 04:55.00 | zzlane33333 | yeah |
| 04:55.05 | zzlane33333 | if I can get these things peering |
| 04:55.05 | contrabanda | please need help with sangoma card. i can see hat hardware is existing from dahdi_hardware but from dahdi_scan i can only see digium card |
| 04:55.06 | zzlane33333 | :-D |
| 04:55.12 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 04:57.02 | joako | I think HP support is outsourced to china now? Anyone encounter this? |
| 04:57.10 | *** join/#asterisk andres833 (~andres833@190.156.198.153) |
| 05:00.38 | sawgood | Anyone know why remote phones behind a NAT will not call each other? |
| 05:01.04 | sawgood | no audio ... also ... I'm not 100% sure if they are registered |
| 05:01.54 | ChannelZ | misconfigured |
| 05:02.21 | sawgood | ChannelZ: there are lots and lots of sub-variables to this case ... |
| 05:02.30 | *** join/#asterisk nix8n82 (~nate@mo-65-41-196-62.sta.embarqhsd.net) |
| 05:02.43 | sawgood | I have three phones (from the phone side ... they are all set to register on 5060) .. all three |
| 05:02.47 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 05:02.54 | sawgood | The Asterisk side is set to 5060 on all three phones |
| 05:05.58 | sawgood | http://pastebin.com/d4pgJ0M7 |
| 05:06.24 | sawgood | wondering if someone could look at this to tell me where my boggle is |
| 05:07.27 | ChannelZ | Two are registered, one is not |
| 05:07.59 | sawgood | right ... but do you see the PORTS they registered on (both the phone and Asterisk are set to 5060) |
| 05:08.11 | sawgood | How did they 'pick' a non 5060 port |
| 05:08.13 | contrabanda | please need help with sangoma card. i can see hat hardware is existing from dahdi_hardware but from dahdi_scan i can only see digium card |
| 05:08.50 | ChannelZ | Those are the remote ports |
| 05:09.55 | joako | sawgood, You have a defective NAT device |
| 05:10.13 | sawgood | What does the DYN stand for? |
| 05:10.23 | sawgood | Does the N stand for "No" NAT? |
| 05:10.46 | *** join/#asterisk s0lid (~s0lid@112.201.240.34) |
| 05:11.27 | s0lid | hi im using asterisk 1.4.29.1 im trying to load the cdr_addon_mysql module but i got this error |
| 05:11.29 | s0lid | WARNING[29813]: loader.c:433 load_dynamic_module: Error loading module 'cdr_addon_mysql.so': /usr/lib/asterisk/modules/cdr_addon_mysql.so: undefined symbol: mysql_options |
| 05:11.39 | s0lid | im not quite sure what it is |
| 05:11.40 | sawgood | joako: My NAT device is defintiely not defective ... it is a higer end grade router ... just not set up for RTP/SIP at the time |
| 05:12.09 | joako | sawgood, I just saw your pastebin.... can you ever get 703 to register? |
| 05:12.25 | sawgood | joako: yes, 703 registers ... then 'stops' at some point |
| 05:12.35 | sawgood | If I look at the phone, it says registered ... |
| 05:12.56 | ChannelZ | N under Nat means yes that device is natted |
| 05:13.08 | joako | sawgood, If you leave 701 & 702 off does asterisk indicate 703 is registerd for say 12 hours? |
| 05:13.32 | sawgood | I'll try right now ... I'll unplug 701 and 702 (both are Yealink phones) |
| 05:13.39 | sawgood | the 703 is a Linksys |
| 05:13.54 | hardwire | hey sawgood |
| 05:14.04 | sawgood | hi hardwire! |
| 05:14.12 | joako | sawgood, But based on your sip show peers it seems like the NAT is beeing handled correctly |
| 05:14.42 | sawgood | joake: does the "N" in the NAT column stand for 'NAT' or "NO" |
| 05:16.17 | sawgood | x701 and x702 are un-plugged ... x703 has been rebooted (on the phone it says registered) .. but sip show peers shows nothing registered |
| 05:17.02 | ChannelZ | I told you N under Nat means 'yes the device is natted' |
| 05:17.45 | sawgood | Ok ... so with only x703 plugged in ... x703 is registered (according to the phone), and I can dial *65 (and I hear, "your extension is 703") |
| 05:17.57 | sawgood | So, one remote phone works ... how do I get the other two to work? |
| 05:18.02 | sawgood | Or, at least one more of them |
| 05:18.11 | hardwire | *65? |
| 05:18.33 | ChannelZ | hardwire: that's a linksys service code I think |
| 05:18.37 | hardwire | oooh ok |
| 05:18.38 | ChannelZ | Is the console showing the registration attempts? |
| 05:18.54 | sawgood | *65 is a FreePBX feature code to 'extension playback' |
| 05:19.05 | ChannelZ | oh. snap |
| 05:19.07 | sawgood | let me paste a sip show peers with the phone working |
| 05:19.25 | joako | sawgood, N stands for NAT |
| 05:19.51 | sawgood | http://pastebin.com/ZdVitEeS |
| 05:19.59 | joako | sawgood, but does 703 show registered in Asterisk... you can place a call when the phone isn't registered |
| 05:20.01 | sawgood | What does the D stand for in the dyn column |
| 05:20.17 | ChannelZ | That it's a dynamic IP |
| 05:20.33 | ChannelZ | (host=dynamic in sip.conf) |
| 05:20.39 | sawgood | joako: the only way I know to 'tell' if a phone is registered is the sip show peers command |
| 05:21.03 | ChannelZ | So that device is registered fine.. |
| 05:21.11 | ChannelZ | Plug one of the others in, then what happens |
| 05:21.52 | sawgood | http://pastebin.com/A8D83CnK |
| 05:22.10 | sawgood | now, x703 shows unreachable ... but it still allows me to call *65 |
| 05:22.47 | ChannelZ | Did it show 703 unregister in the console? |
| 05:23.03 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-npjfwrtnhldxcxpj) |
| 05:23.20 | sawgood | do I do this by issuing a sip show peers |
| 05:23.40 | ChannelZ | no it would have just said it, if you have some verbosity on |
| 05:24.21 | hardwire | ChannelZ: oh snap :P |
| 05:24.45 | sawgood | status = unreachable (does this mean not registered) |
| 05:25.01 | sawgood | I think the phone is registered (even though sip show peers says unreachable) |
| 05:25.26 | magicblaze007 | hardwire: still coundt debug the spa 3102. |
| 05:25.28 | hardwire | sawgood: turn qualify=off for that peer |
| 05:25.29 | ChannelZ | is qualify turned on for it in sip.conf ? |
| 05:25.35 | hardwire | magicblaze007: suck |
| 05:25.53 | hardwire | ChannelZ: you mean in menu x submenu y peer 22 |
| 05:26.14 | ChannelZ | you probably want qualify ON for NAT devices else the firewall will forget about the association. |
| 05:26.31 | hardwire | probably. but I think he has one way SIP to begin with |
| 05:26.43 | sawgood | I get two way audio ... |
| 05:26.43 | ChannelZ | But that it comes up 'unreachable' means either the phone isn't replying to the OPTIONS from asterisk or the replies are disappearing |
| 05:26.53 | hardwire | sawgood: one good way of telling if the peer is registered is "sip show peer xyz" |
| 05:26.54 | sawgood | I turned quailfy to NO |
| 05:27.04 | ChannelZ | That won't help you really |
| 05:27.08 | hardwire | if you see an IP address anywhere and a user agent.. it registered |
| 05:27.22 | ChannelZ | First.. the phones are on a machine behind a firewall, yes? Is Asterisk behind one too? |
| 05:27.54 | sawgood | the three remote phones (two are unplugged) are behind a Linksys RV082 router with 192.168.1.x IP addresses |
| 05:28.05 | sawgood | there is NO port forwarding rules to help these phones on the router right now |
| 05:28.25 | magicblaze007 | Anyone has a SPA 3102 here? |
| 05:28.27 | ChannelZ | and that router has a public IP of 173.13.158.17 |
| 05:28.31 | sawgood | yes |
| 05:28.37 | ChannelZ | Ok so what about the Asterisk side |
| 05:28.42 | magicblaze007 | I'm getting a bad auth from it when i connect to my asterisk server... |
| 05:28.57 | magicblaze007 | have tried almost everything i could |
| 05:28.59 | sawgood | The Asterisk box is on another totally 100% different circuit with a live public static IP |
| 05:29.10 | hardwire | magicblaze007: freepbx? |
| 05:29.23 | magicblaze007 | yes |
| 05:29.24 | ChannelZ | magicblaze007: http://forums.digium.com/viewtopic.php?t=19261 |
| 05:29.33 | hardwire | magicblaze007: what debugging logs are you looking at? |
| 05:29.34 | sawgood | the phones and the Asterisk box have nothing in common (I have two separate LANS/WANS in my LAB) to have the abliity to do nice LAB work |
| 05:29.37 | ChannelZ | sawgood: and no firewall? |
| 05:29.46 | hardwire | sawgood: oh.. lol |
| 05:29.51 | sawgood | the * box has NO firewall between it and the ISP |
| 05:29.53 | hardwire | is asterisk bound to a specific WAN? |
| 05:30.06 | sawgood | iptables is off |
| 05:30.14 | hardwire | are the phones coming in one wan.. and is asterisk using the default gateway for the other? |
| 05:30.19 | ChannelZ | ok. So that more than likely means the firewall on the phone's side is confused |
| 05:30.36 | sawgood | There is only ONE NIC active in the * box (static public IP) |
| 05:30.41 | hardwire | ok |
| 05:31.08 | hardwire | sawgood: answer the question real quick.. the dual wan router.. is it routing packets from * out a different interface than what your peers are coming into? |
| 05:31.18 | sawgood | no ... |
| 05:31.20 | hardwire | with UDP it's very important that you have a clear return path. |
| 05:31.36 | sawgood | the RV082 only uses the Internet port with a static IP address |
| 05:31.41 | sawgood | very simple and clean |
| 05:32.09 | hardwire | no.. in your lab |
| 05:32.15 | sawgood | I agree this is a NAT concern ... but it is a concern which I face with many clients ... (remote phones on a LAN other than the IP PBX_ |
| 05:32.26 | *** join/#asterisk imcdona (imcdona@173.160.189.69) |
| 05:32.30 | sawgood | in the LAB the RV082 has one static IP address on the one interface |
| 05:32.36 | magicblaze007 | ChannelZ: I am trying to follow your directions |
| 05:32.37 | sawgood | the DMZ or 2nd interface has NO cable |
| 05:32.39 | hardwire | sawgood: the other day you didn't know what asterisk was.. so you played me a little? |
| 05:32.54 | magicblaze007 | the lan setup has to be "Bridge" and not "nat", why so? |
| 05:33.02 | sawgood | hardwire: no sure what you mean ... I'm sorry if I did that |
| 05:33.20 | sawgood | I appreciate your time and help ... so I'm glad you mentioned this ... so we will have no boggles |
| 05:34.12 | hardwire | sawgood: whats your asterisk boxes static IP? |
| 05:34.17 | sawgood | with the sip show peer 703 command (how do I determine for sure 100% if the phone is registered)? |
| 05:34.33 | sawgood | asterisksbc.dyndns.org |
| 05:35.16 | hardwire | User-Agent: Asterisk PBX 1.6.0.26 |
| 05:35.19 | hardwire | ok.. SIP works |
| 05:35.37 | hardwire | one of the gotchas is : Via: SIP/2.0/UDP 127.0.1.1:54581;branch=z9hG4bK.2c45a90b;alias;received=66.58.172.5;rport=35616 |
| 05:35.41 | sawgood | <PROTECTED> |
| 05:35.56 | hardwire | To: sip:nobody@asterisksbc.dyndns.org;tag=as4ac2bce3 |
| 05:35.58 | hardwire | err |
| 05:36.06 | hardwire | From: sip:sipsak@127.0.1.1:54581;tag=3f74120e |
| 05:36.18 | sawgood | hardwire ... are you trying to register a phone or something? |
| 05:36.18 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 05:36.19 | hardwire | thankfully the contact is set.. |
| 05:36.28 | hardwire | sawgood: just bouncing a packet off your router |
| 05:36.34 | sawgood | right on ... cool |
| 05:36.50 | sawgood | wonder why is gives the loopback address |
| 05:36.53 | hardwire | oh.. nm.. that 127 bull is my problem |
| 05:36.56 | hardwire | I hate sipsak |
| 05:36.59 | hardwire | it throws me for a loop |
| 05:37.10 | sawgood | sipsak is on my to-do list .. should I leave it alone? |
| 05:37.13 | hardwire | no |
| 05:37.34 | hardwire | but I screwed up.. I didn't set the binding interface so it did some wonky BS |
| 05:37.43 | sawgood | ok ... cool |
| 05:37.50 | hardwire | anyways.. yer sip works |
| 05:38.08 | sawgood | I think this concern is a return problem of SIP and RTP to come next |
| 05:38.27 | hardwire | are you behind the remote router at all? |
| 05:38.31 | sawgood | I need to know how to have more than one phone working on the SAME LAN as remote phones to an Asterisk box in another location |
| 05:38.41 | sawgood | I have full access to all the equipment now |
| 05:38.43 | sawgood | its my lab |
| 05:38.56 | sawgood | I put a lot of time and money in this beautiful setup |
| 05:39.01 | hardwire | so it's linksys <-> linksys? |
| 05:39.02 | sawgood | 19" rack full of equipment |
| 05:39.17 | sawgood | Linksys RV082 with the latest 2.x firmware for the router |
| 05:39.20 | magicblaze007 | still cant debug my spa 3102 |
| 05:39.20 | sawgood | is that what you meant? |
| 05:39.40 | sawgood | The Asterisk box has only a DSL 'modem' in front of it |
| 05:39.49 | hardwire | magicblaze007: you are not forgotten |
| 05:39.57 | sawgood | I have not upgraded to a block of 5 static IP addresses for the DSL connection I have |
| 05:39.57 | magicblaze007 | hardwire: thanks |
| 05:40.13 | sawgood | The RV082 is a Comcast cable static block of 16 |
| 05:40.35 | sawgood | Soon, I am going to upgrade to a block of 30 IP addresses ... |
| 05:40.47 | hardwire | magicblaze007: what debugging logs are you looking at? |
| 05:41.03 | hardwire | ok. |
| 05:41.06 | magicblaze007 | SIP/2.0 401 Unauthorized |
| 05:41.10 | hardwire | magicblaze007: what debugging logs are you looking at? |
| 05:41.36 | magicblaze007 | sometimes i get bad auth...with some settings...with x-lite same settings works perfectly. |
| 05:41.43 | hardwire | magicblaze007: how are you debugging this? |
| 05:41.51 | magicblaze007 | sip debug on |
| 05:41.58 | magicblaze007 | looking at packets |
| 05:42.09 | magicblaze007 | when i reboot my spa 3102 |
| 05:42.11 | hardwire | pastebin the packets. if you already did what is the link? |
| 05:42.32 | sawgood | How do I know for sure (100%) a phone is registered with the sip show peer 703 command? |
| 05:42.32 | hardwire | sawgood: you should probably turn sip debug on as well if you haven't already. |
| 05:42.42 | hardwire | sawgood: you would see the user agent |
| 05:42.58 | magicblaze007 | here is the problem: http://pastebin.com/CMvvPJyY |
| 05:43.19 | sawgood | sip debug on is not a vaild command |
| 05:43.38 | hardwire | magicblaze007: that's not a lot of info |
| 05:43.42 | magicblaze007 | nope here: http://pastebin.com/ww1T9xaY |
| 05:43.48 | sawgood | <PROTECTED> |
| 05:43.48 | magicblaze007 | I think I now know the problem...at last... |
| 05:43.49 | ChannelZ | sawgood: it's sip set debug on |
| 05:43.54 | hardwire | sawgood: so it registered |
| 05:43.58 | hardwire | at some point |
| 05:44.04 | sawgood | 100% it is registered? |
| 05:44.08 | magicblaze007 | My asterisk box is sitting behind a firewall... |
| 05:44.09 | sawgood | thank you for that tip! |
| 05:44.17 | magicblaze007 | the outside static ip is not the inside ip of that box... |
| 05:44.27 | hardwire | magicblaze007: quit picking and choosing packets.. show me the entire debug |
| 05:44.35 | sawgood | sip set debug on (is working now) |
| 05:44.40 | magicblaze007 | hardwire: the problem is in those two packets |
| 05:44.56 | magicblaze007 | Unauthrorized. |
| 05:44.56 | hardwire | magicblaze007: I highly doubt the problem is due to the packets |
| 05:45.04 | magicblaze007 | no its not the packets... |
| 05:45.21 | magicblaze007 | its in there...staring i think...see the sip:11@192.168.122.45 |
| 05:45.26 | hardwire | magicblaze007: what is extension 11's password? |
| 05:45.26 | ChannelZ | sawgood: THE PHONE IS REGISTERED. That it becomes 'unreachable' means it's not responding to OPTIONS packets (with qualify=yes), either because they're not making them back to the phone, or the phone's replies are not making it back to * |
| 05:45.32 | magicblaze007 | thats not my asterisk server... |
| 05:45.40 | hardwire | magicblaze007: whos is it? |
| 05:45.54 | sawgood | ChannelZ: Well, when I do a sip show peer 701 (for the unplugged Yealink phone |
| 05:45.55 | sawgood | ) |
| 05:46.01 | magicblaze007 | hardwire: its mine. And the password is the same as the x-lite one...and the x-lite works... |
| 05:46.06 | sawgood | it has a useragent statement too (and it is not even plugged in) |
| 05:46.24 | magicblaze007 | hardwire: do i make any sense? |
| 05:46.26 | imcdona | What is the name of the SIP provider that begins with f and is capable of doing sip re-invites to connect two enedpoints? |
| 05:46.33 | sawgood | ChannelZ: nice answer thank you! |
| 05:46.42 | ChannelZ | The only thing 'registered' means it that the device logged in to asterisk successfully and thus asterisk now knows it's IP address. That's it. |
| 05:46.58 | hardwire | magicblaze007: where is the asterisk box in relation to your x-lite? |
| 05:47.25 | magicblaze007 | k...here is my problem... |
| 05:47.37 | magicblaze007 | hardwire: Lemme explain the setup |
| 05:47.49 | sawgood | ChannelZ: so sip show peer 702 can have false positives then? |
| 05:47.51 | magicblaze007 | My router at home has 2 machines + an ata adaptor |
| 05:48.00 | magicblaze007 | the x-lite is on one of these machines. |
| 05:48.05 | ChannelZ | With qualify=yes Asterisk sends the device what amount to "hello" packets every once in awhile to make sure it can still reach it. This is important on NAT connections because if the system is idle, the NAT box will time-out the connection and break the port translation mapping. |
| 05:48.21 | magicblaze007 | the asterisk box has a public ip...but is inside a firewall... |
| 05:48.32 | sawgood | ChannelZ: I love that answer ... very nice! |
| 05:48.38 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 05:48.49 | magicblaze007 | when i use x-lite to register an id, i see <sip:10@asterisk.server.edu> |
| 05:49.01 | magicblaze007 | when i use ata, i see <sip:11@192.xxxx> |
| 05:49.08 | magicblaze007 | I've no clue why this is happening. |
| 05:49.12 | hardwire | magicblaze007: where is the asterisk server? |
| 05:49.20 | magicblaze007 | outside ...on the internet |
| 05:49.31 | hardwire | whats it's IP? |
| 05:49.33 | magicblaze007 | has a public ip, behind a firewall... |
| 05:49.47 | ChannelZ | If it's behind a firewall, it doesn't have a public IP |
| 05:49.58 | sawgood | oh ... I see the problem (possibly) ... now that i have set sip debug on (when SIP messages arrive from 703 ... in the options field they pass the 192.168.1.x address), so * has no route back to the phone! |
| 05:49.59 | magicblaze007 | ChannelZ: It has a wierd setup... |
| 05:50.09 | magicblaze007 | it has a public ip, and hence xlite can talk to it |
| 05:50.15 | hardwire | magicblaze007: is nat turned on on the spa? |
| 05:50.19 | magicblaze007 | yes |
| 05:50.28 | hardwire | does the spa use the same account as xlite? |
| 05:50.28 | magicblaze007 | you can ssh to my asterisk server, no problem |
| 05:50.31 | magicblaze007 | yes |
| 05:50.36 | hardwire | magicblaze007: what is the IP? |
| 05:51.06 | ChannelZ | sawgood: that's what "nat=yes" is for in sip.conf, it tells * to ignore the IP it's getting in the SIP headers and send it back to the actual IP things came from. |
| 05:51.09 | magicblaze007 | hardwire: i would refrain from saying it here...if i can. |
| 05:51.12 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 05:51.22 | hardwire | <PROTECTED> |
| 05:51.31 | sawgood | ChannelZ: nat=yes is in my sip.conf file |
| 05:51.37 | magicblaze007 | hardwire: why do you need to know? |
| 05:51.44 | hardwire | so I can see if you're insane. |
| 05:51.51 | hardwire | it's part of the standard diagnostic procedure. |
| 05:52.02 | hardwire | 1.) is end user bat-sh*t crazy |
| 05:52.10 | hardwire | 2.) are they attempting to use freepbx? |
| 05:52.13 | magicblaze007 | lol |
| 05:52.21 | ChannelZ | sawgood: you should see something like "reliably transmitting via NAT to x.x.x.x" in the debug which should show * sending replies back to the right IP |
| 05:52.23 | magicblaze007 | I am using freepbx. |
| 05:52.40 | sawgood | I'll look for that now ... brb |
| 05:53.01 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 05:53.18 | sawgood | Retransmitting #4 (NAT) to 173.13.158.17:5060: |
| 05:53.42 | sawgood | I have that statement in one for the SIP messages which arrived from the phone to the * box |
| 05:53.47 | ChannelZ | ok the 'retransmitting' is a bother.. which means the phone ain't replying |
| 05:53.59 | magicblaze007 | I've somekind of a firewall setup for the server...so when i do a nslookup "192.168x.xxxx" it does give me my asterisk server name...but when i do nslookup localhost it gives me the correct ip |
| 05:54.05 | ChannelZ | Now whether it's because it's not getting the packets to begin with, or it's replies are going astray, I can't tell you |
| 05:54.16 | sawgood | OPTIONS sip:703@192.168.7.105:5060 SIP/2.0 |
| 05:54.19 | hardwire | magicblaze007: is externip set in sip.conf? |
| 05:54.26 | magicblaze007 | nope |
| 05:54.32 | hardwire | set it |
| 05:54.44 | hardwire | sawgood: is externip set in sip.conf? |
| 05:54.56 | sawgood | yes ... for sure 100% |
| 05:55.05 | sawgood | I have the three needed lines in sip.conf for sure! |
| 05:56.02 | ChannelZ | is 192.168.7.105 the LAN address of the phone in question? |
| 05:56.21 | magicblaze007 | hardwire: what i dont understand is how come x-lite is working but spa is not... |
| 05:57.28 | hardwire | magicblaze007: humor me |
| 05:57.43 | sawgood | ChannelZ: yes 192.168.7.105 is the Linksys phone behind the RV082 router |
| 05:58.07 | hardwire | sawgood: what is the "Contact" portion of that packet you referenced. |
| 05:58.09 | ChannelZ | ok that's normal then |
| 05:58.10 | sawgood | I'm sure I could open ports on the router to forward ... but what if I havem more than one phone (and most of the time client phones do not have static IP addresses) |
| 05:58.35 | sawgood | Contact: <sip:Unknown@69.109.164.23> |
| 05:58.38 | ChannelZ | Which one is doing the NAT, the Linksys or the RV082? |
| 05:58.57 | sawgood | ChannelZ: the RV082 handles the NAT ... |
| 05:59.06 | sawgood | the phone does nothing but act as a DHCP client |
| 05:59.11 | magicblaze007 | ok, my asterisk server is wierd...i dont understand how its network works...if i "nslookup localhost" I get an ip 129.186.x.y. When I "nslookup name.of.my.server" I get 129.186.z.w. When I "nslookup 192.168....." on my server, i get back the name of my server...anyone understands whats going on? I think i was told there is a firewall in front of it. |
| 05:59.16 | ChannelZ | oh I misread that you had two routers |
| 05:59.23 | hardwire | sawgood: what is the Contact: portion of the incoming packets |
| 05:59.29 | sawgood | I pasted it |
| 05:59.38 | hardwire | no.. that's for outgoung |
| 05:59.39 | sawgood | Contact: <sip:Unknown@69.109.164.23> |
| 05:59.42 | hardwire | 69.109.164.23 = our asterisk box |
| 05:59.44 | hardwire | your |
| 05:59.53 | sawgood | yes |
| 05:59.55 | hardwire | hmm. |
| 06:00.03 | hardwire | paste the whole packet to pastebin |
| 06:00.20 | ChannelZ | It really sounds like your router is stupified about the incoming traffic |
| 06:00.40 | sawgood | http://pastebin.com/CQxnDAzV |
| 06:00.44 | ChannelZ | it's just dumping them on the floor |
| 06:00.57 | sawgood | I'm sure the RV082 needs port forwarding ...(it is a solid NAT device) .... |
| 06:01.11 | sawgood | one phone works (I can dial out from the phone and get two way audio) |
| 06:01.14 | hardwire | sawgood: transmitting means outbound :P |
| 06:01.18 | ChannelZ | yes, it does |
| 06:01.22 | hardwire | re-transmitting means outbound too. |
| 06:01.34 | sawgood | but as soon as I try two or three phones, all two or three stop working |
| 06:01.34 | hardwire | reboot the phone and capture the inbound packet. |
| 06:01.39 | sawgood | they cannot call each other |
| 06:01.49 | hardwire | sawgood: aww.. that's a pretty common issue with routers that can't do connection tracking |
| 06:02.05 | hardwire | set phone one to use port 5060, phone two to use 5070, phone three to use 5080 |
| 06:02.16 | hardwire | in the Line configuration you can set which SIP port they listen on |
| 06:02.20 | sawgood | hardwire: so, I need a more SIP ready edge device to have multiple remote phones? |
| 06:02.27 | hardwire | if your router is too stupid to figure out how to translate.. you're fighting a massive battle |
| 06:02.36 | hardwire | sawgood: do this on your phones.. not your * |
| 06:02.58 | ChannelZ | sawgood: no.. all your phones contact * on port 5060, but they need to send FROM different port numbers.. so you can static port-forward that back to them. |
| 06:03.08 | sawgood | ok ... I'll make one work on port 5060, one on 5061, and one on 5062 (is that ok) |
| 06:03.16 | hardwire | ChannelZ: most of the time you don't need the static forwarding back in |
| 06:03.22 | hardwire | sawgood: no.. they are multi line |
| 06:03.24 | sawgood | from the phone side (the * side will stay 5060 on all three phones) |
| 06:03.25 | hardwire | use 5060/5070/5080 |
| 06:03.35 | hardwire | that way phone 1 line 2 can be 5061 |
| 06:03.46 | sawgood | hardwire: excellent point! |
| 06:03.53 | hardwire | sawgood: you can get around this by using an stun server as well. |
| 06:04.07 | hardwire | sicne your firewall is on my hit list now |
| 06:04.17 | sawgood | I'll leave the Asterisk setting for ports at 5060 for all three phones (I'll change the port settings on the phone web gui) |
| 06:04.21 | sawgood | does this sound right? |
| 06:04.23 | hardwire | yes |
| 06:04.39 | hardwire | ChannelZ: sorry.. I had caffeine |
| 06:04.44 | hardwire | I have to work it out |
| 06:04.58 | ChannelZ | sawgood: if you only leave one phone connected does it stay registered and reachable? Or does it eventually come up as 'unreachable'? I thought you said earlier it did. |
| 06:05.00 | *** join/#asterisk blaines (~blaines@ip68-106-24-21.ph.ph.cox.net) |
| 06:05.56 | *** join/#asterisk ktwilight[m] (~ktwilight@91.180.42.203) |
| 06:06.01 | sawgood | ChannelZ: the phone (one phone by itself) Linksys SPA 942 will eventually say "unreachable" with the sip show peers command |
| 06:06.33 | sawgood | I have all three on now, and I am changing the ports at the phone side (leaving * set to 5060) |
| 06:07.15 | ChannelZ | So I'm not sure what you're really changing here because at least in one of your previous pastes the phones were already coming in with different source ports |
| 06:07.16 | sawgood | the 'thought' of a STUN/ICE type server is starting to sound really cool (isn't there public STUN servers) for this exact reason (mulitple RFC1918 phones using NAT on a LAN)? |
| 06:08.09 | magicblaze007 | hardwire: I found my problem! |
| 06:08.11 | hardwire | I'm just going over NAT traversal options |
| 06:08.12 | sawgood | ChannelZ: I know ... I know ... that is basically what started this thread (I did not know HOW they did this) they were set to 5060 on both the phone and the * side BUT YET THEY automatically picked different ports on their own |
| 06:08.18 | hardwire | magicblaze007: woot |
| 06:08.39 | hardwire | sawgood: the phones don't pick |
| 06:08.41 | hardwire | the firewall does. |
| 06:09.13 | hardwire | it maps 5060 to a phone if it's source port is 5060 |
| 06:09.19 | hardwire | it maps 1025 to a phone if 5060 is taken |
| 06:09.21 | hardwire | the phone has no idea |
| 06:09.30 | hardwire | stun lets the phone know what to set rport to |
| 06:09.43 | hardwire | that way it can advertise correctly if there is a problem with the registration server, etc.. |
| 06:09.57 | sawgood | right now, I have three phones (two Yealink T28P) 5070 and 5080 and a Linksys SPA 942 at 5060 (on the phone side) |
| 06:10.06 | sawgood | all three have rebooted ... and are attempting to register |
| 06:10.26 | hardwire | sawgood: halt asterisk before you register them and restart it to flush out previous registrations |
| 06:10.27 | sawgood | sip set debug on (is running) |
| 06:10.29 | hardwire | then turn on the phones |
| 06:10.44 | sawgood | amportal stop (or just exit out of the CLI) and come back in again? |
| 06:10.47 | hardwire | well.. I guess that doesn't work since astdb |
| 06:10.55 | hardwire | sawgood: nevermind |
| 06:11.27 | hardwire | I'm probably hurting more than helping right now. I just don't like it when stuff doesn't just work. |
| 06:11.43 | hardwire | magicblaze007: what was the problem? |
| 06:11.50 | sawgood | I'm sure this is a NAT problem on the RV082 side (where the phones are behind) |
| 06:11.51 | hardwire | so that we can either fishslap you or congratulate you |
| 06:11.58 | sawgood | the Asterisk box is live on the Internet with no firewall |
| 06:12.28 | sawgood | Is there a known good 'public' proxy I can pass the registration through? |
| 06:12.41 | sawgood | or some work around (for cases with more than one remote phone on a LAN) |
| 06:12.44 | hardwire | sawgood: pastebin 'ip route show table all' on your asterisk box |
| 06:13.29 | sawgood | http://pastebin.com/fsvfs879 |
| 06:13.58 | hardwire | you're using the same 192.168.7 subnet behind both routers eh? |
| 06:14.02 | hardwire | that should be fine I guess |
| 06:14.10 | sawgood | Do you think this might be a concern ... |
| 06:14.22 | hardwire | I thought you said asterisk was plugged into the internet directly |
| 06:14.26 | sawgood | The Asterisk box has a 2nd NIC with no cable plugged in (it is on the same subnet as the phones) |
| 06:14.30 | sawgood | sorry!!!!!!!!!!!! |
| 06:14.47 | hardwire | ifconfig eth1 0.0.0.0 |
| 06:14.55 | hardwire | try again |
| 06:14.56 | sawgood | there might be a route statement in the Asterisk box which is stale ... but failing because * tries to route to that NIC |
| 06:15.06 | sawgood | I can turn off eth1 and reboot |
| 06:15.38 | sawgood | eth1 is onboot=yes ... but it has no CAT-5e cable in the jack |
| 06:15.51 | sawgood | let me set eth1 to onboot=no and reboot the box |
| 06:15.55 | sawgood | no false positives |
| 06:16.05 | sawgood | this way, I'll have no 'stale' route on the * box ... |
| 06:16.17 | sawgood | this is worth a shot, right? |
| 06:17.30 | ChannelZ | * is sending the packets back to the right IP. So long as the route that packet is taking is the same way it came in and it's not making it's way back to the firewall router on a different IP/interface |
| 06:17.38 | sawgood | The Asterisk box is rebooting now (and there will be no eth1 when it starts back up again) |
| 06:17.52 | sawgood | ok .. I was wrong |
| 06:18.13 | sawgood | well, anyhow ... eth1 will not work when the box is turned back on (thus there will be no false route in the route table) |
| 06:18.17 | sawgood | out of good measure! |
| 06:18.21 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 06:19.49 | *** join/#asterisk khussein78 (~khussein7@188.225.192.238) |
| 06:19.57 | khussein78 | hello all |
| 06:20.06 | sawgood | very strange ... the box booted back up ... I ssh to it ... but I get an error when I try to start asterisk -rvvvv |
| 06:20.24 | sawgood | well, forget that ... it worked now |
| 06:20.29 | sawgood | maybe the service had not started yet |
| 06:21.07 | sawgood | awww ... darn it ... same NAT boggle ... |
| 06:21.20 | sawgood | I think the return packets are not making it back to the RV082 box ... |
| 06:21.51 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-cjeyxckbrmkpgtje) |
| 06:21.51 | khussein78 | when i ring on my asterisk box from outside and then choose group xxx and disconnect from my side, i see that the phones on this group still ringing for more than 5 times ? |
| 06:22.16 | ChannelZ | Does it have an inordinately short NAT timeout? |
| 06:22.18 | sawgood | one interesting thing now is ... (the two Yealink phones (one set for 5070 and one set for 5080) never even try to register) |
| 06:22.32 | sawgood | I changed the port on the phone side and not the * side |
| 06:22.42 | ChannelZ | khussein78: what is 'the outside' in this case? |
| 06:23.14 | ChannelZ | sawgood: I don't know anything about the yealink phones, they might be trying to send TO asterisk on 5070/5080 |
| 06:23.31 | ChannelZ | That being the case, you'd need to set that in sip.conf so * listens.. |
| 06:24.09 | sawgood | ChannelZ: how do I 'tell' sip.conf to listen for a range from 5060-5089 |
| 06:24.37 | sawgood | 5060-5069 for the LInksys 5070-5079 for Yealink 1 and 5080-5089 for Yealink 3? |
| 06:24.41 | khussein78 | ChannelZ, from my mobile or from any phone line outside my compnay |
| 06:25.04 | ChannelZ | khussein78: but how does it come into your * box - analog, PRI, VoIP... |
| 06:25.23 | sawgood | ChannelZ: Do you think STUN is the quicker fix here? |
| 06:25.45 | khussein78 | FXO cards (openvox) |
| 06:26.14 | sawgood | 'most' of these clients which will have the 'need' for multiple remote phones (will have a cheap Linksys or Netgear router) as the edge device, so I'll have to face this over and over and over |
| 06:26.21 | ChannelZ | so you just don't have disconnect supervision on those lines then |
| 06:26.51 | ChannelZ | sawgood: it might be since your router is baffled but I have no idea really |
| 06:27.03 | khussein78 | ChannelZ, FXO cards |
| 06:28.40 | sawgood | ChannelZ: the thing is ... the Linksys RV082 is as about a 'high' end as a non commerical router goes (so if anything can do NAT effectively) it should be this router |
| 06:28.48 | ChannelZ | khussein78: you don't have disconnect supervision on those lines from the telco.. or maybe the openvox just isn't detecting it. |
| 06:28.51 | sawgood | It is almost a $300 dollar edge device |
| 06:28.51 | ChannelZ | khussein78: what country? |
| 06:30.20 | khussein78 | ChannelZ, Palestine |
| 06:31.12 | ChannelZ | sawgood: Lets sstart over. disconnect all phones. turn on sip debug. Plug back in just one, like the Linksys.. set to default 5060. Capture the SIP dialog of it registering, and * replying.. so wait 20 seconds or so. Pastebin. |
| 06:31.42 | ChannelZ | khussein78: hmm I dunno anything about foreign telcos. Are you using kewlstart, or loop start, etc.. ? |
| 06:32.09 | khussein78 | ChannelZ, i don't know really about these ! |
| 06:32.30 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 06:32.56 | ChannelZ | khussein78: how is it configured in dahdi/system.conf |
| 06:34.07 | sawgood | http://pastebin.com/Hhfh010c |
| 06:34.41 | *** part/#asterisk aceio (~c2cbd7fe@gateway/web/freenode/x-ndftptnibstygsph) |
| 06:35.09 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 06:35.19 | ChannelZ | ok that's not at all what I wanted to see |
| 06:36.19 | sawgood | ok ... so should I unplug the phone ... *while sip debug is running" and wait 30 seconds |
| 06:36.20 | ChannelZ | But, it does seem to imply it got an OPTIONS packet from * in the first place, so that's close enough I guess. |
| 06:36.23 | sawgood | before pastebin? |
| 06:36.51 | ChannelZ | no, now just let it sit there. Do a sip show peers again. |
| 06:36.56 | sawgood | Its like the RV082 is 'closing NAT' after a short window |
| 06:37.12 | sawgood | 701 is now unreachable |
| 06:37.22 | sawgood | it was "OK" at first ... |
| 06:37.24 | ChannelZ | that's what I'm trying to determine, the nat timeout is probably ridiculously low |
| 06:37.34 | ChannelZ | ok so that was less than 30 secs. |
| 06:37.35 | sawgood | let me look in the RV082 |
| 06:37.59 | ChannelZ | So now try this. edit sip.conf and put "qualifyfreq=10" up in the general section |
| 06:38.02 | ChannelZ | sip reload |
| 06:38.12 | ChannelZ | then reboot the phone so it re-registers |
| 06:38.40 | sawgood | problem is this box runs AsteriskNOW 1.5, so the sip.conf is not to be edited ... I'm not sure what 'sub file' this statement should go in |
| 06:38.41 | sawgood | sorry! |
| 06:39.03 | ChannelZ | You should toss that thing out the window |
| 06:39.07 | sawgood | ha! |
| 06:39.13 | sawgood | I don't know how to build from src |
| 06:39.18 | sawgood | it never works for me |
| 06:39.27 | ChannelZ | your sip peers are probably in a database.. where the general config is, no idea |
| 06:39.31 | sawgood | they say, ./configure ... then make then make install |
| 06:39.37 | sawgood | but it never comes through like that for me |
| 06:39.57 | ChannelZ | that's all you need.. you just need the developer tools (gcc et al) |
| 06:40.12 | sawgood | oh! |
| 06:40.22 | sawgood | I'll try that tomorrow on a lab box |
| 06:40.25 | sawgood | for sure!!! |
| 06:40.37 | sawgood | I'm asking for help in FreePBX now ... for the sub-file to add the statement to |
| 06:40.45 | ChannelZ | I'm booting *now in a virtual box to try and look |
| 06:42.47 | ChannelZ | sip_registrations_custom.conf it looks like |
| 06:43.26 | sawgood | not a single NAT setting option in the GUI for the RV082 ... very strange |
| 06:43.37 | sawgood | ok .. thanks ... I'll do that syntax right now |
| 06:44.16 | kaldemar | sawgood: my one and only experience with an RV model router was horrific. |
| 06:45.00 | sawgood | wow ... what edge device do you use? |
| 06:45.17 | sawgood | I have an Intertex surfin bird (which is a SIP known good edge device) |
| 06:45.29 | sawgood | I put the syntax in the file, and I issued a sip reload command |
| 06:45.38 | kaldemar | the poor bastard dropped TCP flows that tried to to window scaling (this was fixed in a firmware release though) and just broke after a couple of years. an RV042. |
| 06:45.41 | sawgood | is it possible to see if that statement is in memory now? |
| 06:45.59 | sawgood | RV042 is a much different router than the RV082 |
| 06:46.03 | sawgood | almost night and day difference |
| 06:46.13 | sawgood | different chipset, CPU, and memory |
| 06:46.21 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 06:47.23 | ChannelZ | sip show settings might |
| 06:47.47 | ChannelZ | 'Reg. min duration' probably |
| 06:48.05 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 06:48.33 | sawgood | http://pastebin.com/69iKD8B8 |
| 06:48.42 | sawgood | looking through it now |
| 06:48.56 | ChannelZ | 65 |
| 06:49.17 | ChannelZ | (I was wrong) |
| 06:49.35 | sawgood | oh ... its not in the sip show settings output? |
| 06:49.41 | ChannelZ | it is - line 65 |
| 06:49.42 | sawgood | or 65 should be the setting |
| 06:49.49 | ChannelZ | in your paste |
| 06:49.58 | sawgood | cool |
| 06:50.02 | sawgood | it is set to 10 like you asked |
| 06:50.08 | sawgood | cool? |
| 06:50.14 | ChannelZ | so reboot the phone so it reregs |
| 06:50.39 | sawgood | look at line 67 |
| 06:50.55 | sawgood | do you think that is a false positive (there is a 2nd NIC in the * box with no CAT-5e cable) |
| 06:50.57 | ChannelZ | yea I dunno why you have that one showing up twice but it doesn't matter at this point |
| 06:51.02 | sawgood | ok |
| 06:52.32 | sawgood | keep getting SIP 200 OK messages over and over |
| 06:52.55 | ChannelZ | ok |
| 06:52.59 | sawgood | no more 'retyiing' NAT messages (so far) |
| 06:53.04 | ChannelZ | and sip show peers still showing it qualified? |
| 06:53.17 | sawgood | yes |
| 06:53.20 | sawgood | OK (80ms) |
| 06:53.48 | ChannelZ | ok so that's the answer then, your router is just tearing down the NAT connections too quick |
| 06:53.58 | sawgood | amazing! |
| 06:54.05 | sawgood | where did you 'pull that setting' from? |
| 06:54.11 | sawgood | how did you know, man? |
| 06:55.14 | ChannelZ | because we knew the phone was communicating bi-directionally with Asterisk, but after an amount of time it stopped responding to 'pings' |
| 06:55.17 | sawgood | I am able to get two way audio |
| 06:55.26 | sawgood | nice job ... |
| 06:55.41 | sawgood | So, should I move the port back to 5060 on the two Yealink phones (try one at a time) |
| 06:55.59 | ChannelZ | Well that's possibly a different issue still |
| 06:56.10 | sawgood | yeah ... more than one remote phone on a LAN, right? |
| 06:56.15 | ChannelZ | With things set as they are now, put one back at 5060 and then connect it and let it register and see what happens |
| 06:56.25 | *** join/#asterisk smooth_penguin (~smoove@59.95.13.86) |
| 06:56.26 | sawgood | ok |
| 06:56.57 | sawgood | phone has to boot up using 5070 first (it is a PoE phone) ... |
| 06:57.08 | sawgood | I'll change the web gui back to 5060 if it fails to register |
| 06:57.18 | sawgood | who knows what the Yealkink phone does ... |
| 06:57.56 | ChannelZ | it probably won't register |
| 06:58.08 | sawgood | no it won't register ... |
| 06:58.14 | sawgood | I'll change it to 5060 and reboot |
| 06:58.58 | sawgood | registered very fast! |
| 06:59.14 | sawgood | sip show peers shows it coming in on port 5062 |
| 06:59.34 | sawgood | I see 200 OK messages going back to it on port 5062 |
| 06:59.38 | sawgood | so that is a good thing, right? |
| 07:00.17 | sawgood | I'll try to call between phones |
| 07:00.43 | ChannelZ | to clarify, it's coming in on 5060 but going out on 5062 |
| 07:00.52 | sawgood | calls between phones is fine (two way audio) |
| 07:00.58 | ChannelZ | (IE the source port being munged by the router is 5062) |
| 07:01.00 | sawgood | yes, your statement is correct! |
| 07:01.14 | sawgood | I'll plug in the 3rd phone now |
| 07:02.08 | ChannelZ | so the trick is to try and set the qualifyfreq as high as possible but below whatever the timeout on the router happens to be. |
| 07:02.24 | sawgood | very nice tatic! |
| 07:02.34 | sawgood | are you in CO? |
| 07:02.49 | ChannelZ | yes |
| 07:03.06 | sawgood | can I private /msg you ? |
| 07:03.11 | ChannelZ | yeah |
| 07:03.50 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 07:04.18 | *** join/#asterisk Da-Geek (~Da-Geek@213-193-154-130.perm.easynet.be) |
| 07:05.48 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 07:08.15 | *** join/#asterisk ecto (~weechat@24.42.221.223) |
| 07:09.35 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:12.37 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 07:13.19 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 07:15.22 | *** join/#asterisk Ad-Hoc (~nimbus@62.169.216.185) |
| 07:15.40 | khussein78 | ChannelZ, i am using asterisk 1.4.21.2-2 with trixbox, it still use zaptel and zapata not dahdi |
| 07:16.27 | kaldemar | khussein78: then you need to look at /etc/zaptel.conf and /etc/asterisk/zapata.conf |
| 07:16.58 | *** join/#asterisk felipe_ (~felipe@my.nada.kth.se) |
| 07:17.18 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 07:19.20 | *** join/#asterisk Tim_Toady (~moi@188.4.45.188.dsl.dyn.forthnet.gr) |
| 07:22.07 | khussein78 | kaldemar, here is my zapata.conf http://paste.ubuntu.com/406380/ |
| 07:24.52 | ChannelZ | so you're using kewlstart.. |
| 07:24.57 | kaldemar | khussein78: you're using ks signalling, which is good. busydetect is also enabled, but your busycount is rather high. it is 3 by default. 15 requires quite some time to detect a busy. try setting it to 5 for example, restart asterisk and try again. |
| 07:25.21 | kaldemar | trixbox might screw up any manual modifications though |
| 07:26.12 | khussein78 | i will change it and check it again, thanks to all |
| 07:26.16 | ChannelZ | I'd turn off busydetect (notoriously unreliable) and see if "hanguponpolarityswitch=yes" does anything for you |
| 07:26.25 | kaldemar | busypattern may also be useful if tuning busycount won't do it. |
| 07:26.32 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:28.17 | kaldemar | polarityswitch is definitely better if the telco provides it. |
| 07:29.08 | ChannelZ | and make sure your zaptel.conf is set to fxsks matching zapata |
| 07:29.08 | *** join/#asterisk Da-Geek (~Da-Geek@213-193-154-130.perm.easynet.be) |
| 07:30.02 | kaldemar | asterisk wouldn't start if it was something else. |
| 07:30.15 | khussein78 | yes it is |
| 07:30.22 | ChannelZ | ah |
| 07:30.46 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 07:30.51 | khussein78 | also i see busydetect added twice ? is that meaning something |
| 07:32.58 | ChannelZ | mistake |
| 07:33.12 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 07:34.03 | khussein78 | ok |
| 07:44.25 | ChannelZ | bed time |
| 07:50.07 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
| 07:50.25 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 07:54.34 | carrar | almost dinner time |
| 07:59.36 | *** join/#asterisk ltd (~z@pat.transact.net.au) |
| 08:01.38 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 08:07.06 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 08:09.24 | tengulre | what's mean adsi=yes |
| 08:10.13 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 08:16.28 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 08:18.18 | blaines | what's your favorite asterisk distro? asterisknow/trixbox/custom? |
| 08:22.26 | zamba | just asterisk, i guess? |
| 08:22.30 | zamba | or is that just stupid? |
| 08:22.58 | blaines | well I'm trying to figure out why people like asterisk vs asterisknow vs trixbox |
| 08:23.06 | blaines | I can't tell the difference |
| 08:25.50 | zamba | i have never used either asterisknow or trixbox |
| 08:26.11 | blaines | ah |
| 08:26.39 | blaines | I'm just trying to setup an asterisk box 'right' just don't want regrets later |
| 08:29.01 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 08:29.46 | *** join/#asterisk nicknick (~administr@host213-123-201-13.in-addr.btopenworld.com) |
| 08:35.28 | Katty | i think ihave a sinus infection |
| 08:35.34 | Katty | this head pressure is unreal |
| 08:37.12 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 08:39.03 | blaines | katty: quick! what's your fav asterisk? |
| 08:39.12 | *** join/#asterisk diegomad (mad@186.0.4.182) |
| 08:40.24 | Katty | uhmm |
| 08:40.38 | Katty | i dno't get the question |
| 08:43.15 | blaines | katty: trixbox/freepbx/asterisknow, etc |
| 08:43.26 | Katty | oh |
| 08:43.29 | Katty | just plain ole asterisk |
| 08:43.59 | blaines | ah cool - how do you manage it? webadmin or command line? |
| 08:44.07 | Katty | i download the tarballs from digium's site, and compile them on a debian platform |
| 08:44.31 | Katty | the only 'web' stuff i have running is a php wrap around to query the call logs.... asterisk-stat |
| 08:46.06 | carrar | <PROTECTED> |
| 08:46.11 | carrar | can't sleep? |
| 08:46.14 | Katty | no |
| 08:46.21 | carrar | Asterisk questions bothing you? |
| 08:46.29 | Katty | i have this 'head pressure' and i'm trying to keep from having a panic attack over it |
| 08:46.38 | carrar | POT |
| 08:46.45 | carrar | should solve that |
| 08:46.51 | Katty | i don't want pot |
| 08:46.54 | Katty | i wanna see my doctor |
| 08:46.55 | carrar | heh |
| 08:46.57 | Katty | but it's 4am |
| 08:47.05 | carrar | 5PM here |
| 08:47.21 | Katty | they need a regular doctor during the night |
| 08:47.32 | carrar | 24 hour care |
| 08:47.35 | Katty | someone who takes my insurance cards, but doesn't charge ER prices |
| 08:47.35 | carrar | some do |
| 08:47.50 | carrar | we have those in Seattle |
| 08:48.07 | Katty | yeah..we have like 30k people here |
| 08:48.32 | carrar | You need to move out of that dariy farming community ;) |
| 08:48.38 | carrar | dairy |
| 08:49.28 | Katty | no i just need to move into the looney bin |
| 08:49.32 | Katty | they have doctors 24/7 |
| 08:49.36 | carrar | heh |
| 08:49.53 | *** join/#asterisk mpe_ (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 08:50.16 | plundra | What's a suitable codec for the "master" (best quality) audio files? adpcm or slin? (Which will then be converted with "file convert ...") |
| 08:50.50 | plundra | adpcm sounds "fancy" with it's adaptiveness, so maybe slin is the way to go? |
| 08:51.01 | Katty | carrar: i won't get to go to my usual doc until 8am |
| 08:51.07 | Katty | carrar: that's another 4 hours :< |
| 08:51.53 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 08:53.20 | carrar | You need some relaxation technics? (sp) |
| 08:53.39 | Katty | i'm using relaxation techniques. |
| 08:53.47 | Katty | distraction |
| 08:53.52 | carrar | IRC!! |
| 08:53.54 | blaines | carrar: fav asterisk? |
| 08:54.18 | carrar | Please use foll sentences |
| 08:54.21 | carrar | full |
| 08:54.21 | Katty | carrar: MY ARM JUST TWITCHED |
| 08:54.25 | Katty | carrar: I"MG OING TO DIE |
| 08:54.27 | Katty | carrar: etc |
| 08:54.28 | blaines | carrar: what is your fav asterisk? |
| 08:54.30 | blaines | :) |
| 08:54.36 | carrar | heh Katty, I'm here for you!! |
| 08:54.43 | carrar | Asterisk is my FAV asterisk!! |
| 08:55.07 | Katty | carrar: you know how when you're drunk, you turn your head and it takes the room a minute to catch up? |
| 08:55.09 | blaines | blasphemy! freepbx/asterisknow/trixbox/custom? |
| 08:55.26 | carrar | Katty, haha I haven't been that drunk in years |
| 08:55.46 | carrar | Thats usually puking drunk |
| 08:56.07 | carrar | blaines, then thats not really "asterisk" |
| 08:56.08 | Katty | yeah |
| 08:56.12 | carrar | thats freebpx |
| 08:56.14 | Katty | well that's how i feel when i turn my head |
| 08:56.21 | carrar | I don't use that crap |
| 08:56.26 | carrar | I use Asterisk Source |
| 08:56.28 | carrar | the BEST |
| 08:56.31 | Katty | and my neck is really really stiff |
| 08:56.47 | Katty | the aches from the flu are gone, and the fever is gone..checked that just a bit ago |
| 08:57.00 | blaines | carrar: uh asterisk distro? hehe - so custom |
| 08:57.10 | blaines | do you use a webadmin or command line? |
| 08:57.16 | carrar | Asterisk from Source |
| 08:57.21 | carrar | the original |
| 08:57.24 | carrar | and only |
| 08:57.48 | blaines | hardcore |
| 08:57.53 | carrar | webadmin is for punks |
| 08:57.56 | carrar | and crap |
| 08:58.02 | Katty | your mom's a punk |
| 08:58.12 | carrar | with a mohawk |
| 08:58.20 | Katty | a PURPLE mohawk |
| 08:59.10 | blaines | oo here's a good Q |
| 08:59.20 | carrar | Are you sure? |
| 08:59.33 | blaines | carrar, Katty: landline or voip? |
| 08:59.48 | blaines | like do you get landline to pbx or voip? |
| 08:59.59 | blaines | sip w/e |
| 09:00.04 | Katty | i use a pri |
| 09:00.10 | carrar | PRI/SIP |
| 09:00.13 | Katty | but i've also setup pots lines and sip accounts before |
| 09:00.58 | blaines | does the PRI get a full t1 input? |
| 09:01.08 | blaines | and each t1 channel == ph line? |
| 09:01.15 | Katty | well. |
| 09:01.20 | Katty | you can get a pri or a t1 |
| 09:01.24 | Katty | or a partial t1 |
| 09:01.25 | carrar | read up on PRI's Blaine |
| 09:01.33 | Katty | pris have 23 channels, and 1 data channel |
| 09:01.38 | Katty | t1s have 24 channels |
| 09:01.39 | carrar | PRI uses a T1 |
| 09:01.48 | carrar | typically |
| 09:01.54 | Katty | and partial t1s are....basically however many channels you require |
| 09:01.54 | blaines | err yea bad grammar today |
| 09:02.07 | carrar | FULL T1 PRI is 23 channels |
| 09:02.14 | Katty | carrar: yeah ours comes in, physically, on the same hardware a t1 uses |
| 09:02.38 | carrar | PRI's have faster setup |
| 09:02.39 | Katty | i can't say i honestly understand the difference between using a pri vs a t1 |
| 09:02.54 | blaines | okay so they take that T1 to an adtran or direct to pbx? |
| 09:03.09 | Katty | adtran, usually |
| 09:03.17 | Katty | ours goes to an adtran anyway |
| 09:04.02 | blaines | huh, how do you access the phone channels through an adtran? |
| 09:04.15 | carrar | magic |
| 09:04.30 | blaines | ooooo |
| 09:05.13 | carrar | pure telcom magic |
| 09:05.22 | carrar | blaines: Whats with all the questions? |
| 09:05.26 | blaines | no, seriously, I'm not sure how the last leg from provider to client works |
| 09:05.28 | carrar | What are youbuilding |
| 09:05.40 | blaines | carrar: already built, replacing |
| 09:05.43 | carrar | grab asterisk from source and start building |
| 09:05.49 | carrar | use nothing else |
| 09:06.18 | carrar | and use postgres |
| 09:06.27 | blaines | yea that's cool I've used asterisk - someone setup freepbx(?) and I aquired the client, I'm to re-asterisk it |
| 09:06.44 | carrar | migrate them off that crap |
| 09:06.59 | blaines | yea, the web admin is severly broken |
| 09:07.08 | blaines | *severely |
| 09:07.31 | carrar | Katty: You still with us? |
| 09:08.02 | *** join/#asterisk Tulga (~chatzilla@203.91.113.10) |
| 09:08.15 | blaines | I'm familiar with linux/asterisk but didn't know why someone would choose freepbx(?) or how ppl are connecting their asterisk pbx to the phone network |
| 09:08.20 | Tulga | what it mean? |
| 09:08.22 | Tulga | [Mar 30 17:05:38] ERROR[4147]: chan_dahdi.c:10036 mkintf: Signalling requested on channel 47 is ISDN PRI but line is in Unknown signalling 896 signalling |
| 09:08.23 | Tulga | [Mar 30 17:05:38] ERROR[4147]: chan_dahdi.c:15972 build_channels: Unable to register channel '33-47,49-63' |
| 09:08.27 | blaines | ow |
| 09:08.30 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 09:08.36 | blaines | Tulga: pastie.org |
| 09:08.47 | Tulga | sorry. just 2 lines :) |
| 09:09.11 | carrar | blaines, for stand alone customers PRI's are the first choice but have a higher cost |
| 09:10.04 | blaines | carrar: ~$300-$500/mo right? |
| 09:10.39 | carrar | depends wher you are and if it's a local T1 or LD T1 |
| 09:11.22 | blaines | ah |
| 09:11.47 | blaines | now what if I put an asterisk pbx in a datacenter? how's that work? |
| 09:12.08 | carrar | link it to your SBC |
| 09:12.17 | carrar | and serve multiple clients off your SBC |
| 09:12.28 | blaines | sorry, SBC? |
| 09:12.36 | carrar | Session border controller |
| 09:12.48 | carrar | behind that you can have SIP or PRI connections |
| 09:13.14 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 09:13.35 | blaines | hm, an SBC may be to costly - just thinking of helping a small office telecommute |
| 09:13.46 | carrar | or a SBC can be free |
| 09:13.56 | carrar | open source is your friend |
| 09:14.20 | blaines | oh, I'll have to look into that |
| 09:14.36 | carrar | kamailio & opensips are opensource |
| 09:15.06 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 09:15.27 | carrar | later, dinner time here |
| 09:15.28 | blaines | carrar: thanks tons man |
| 09:15.31 | pinoyskull | hi, how do i force disconnect a bridge? |
| 09:15.56 | *** join/#asterisk Akiraaa (~Akiraaaa@79.112.30.234) |
| 09:16.51 | Katty | returns |
| 09:17.54 | blaines | Katty: how do you access the phone channels through an adtran? |
| 09:18.16 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:18.24 | Katty | blaines: no idea. |
| 09:18.28 | Katty | blaines: telco set it all up |
| 09:18.29 | blaines | oh |
| 09:18.32 | blaines | ahhhhh |
| 09:18.34 | blaines | cool |
| 09:18.50 | Katty | actually |
| 09:19.02 | Katty | now that i think about it, we may not be using the adtran anymore |
| 09:19.11 | Katty | i think we used that when we were using a partial t1 |
| 09:19.12 | blaines | yea that last leg is what I'm most unfamiliar with - and there doesn't seem to be much published info |
| 09:19.25 | Katty | which took the t1 down to pots lines and reserved some for data/bandwidth |
| 09:19.44 | blaines | right, I'm familiar with partial t1 |
| 09:19.48 | blaines | interesting |
| 09:20.41 | blaines | wish I had a 'spare' pri and some equipment I could learn and play with this stuff on |
| 09:20.43 | blaines | haha |
| 09:21.10 | Katty | don't we all |
| 09:22.16 | pinoyskull | hi, how do i force disconnect a bridge? |
| 09:23.56 | *** join/#asterisk Z_God (~julius@wlan234098.mobiel.utwente.nl) |
| 09:29.12 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
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| 09:44.36 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 09:47.18 | Katty | hmm. |
| 09:47.30 | Katty | almost 5. that means 3 hours until dr office opens :/ |
| 09:48.09 | Katty | dear dr. please fix head pressure problems. also, please rearrange neck until it stops hurting. love, Katty |
| 09:50.32 | creativx | i have a spare drill.. |
| 09:55.20 | Katty | is that for the head pressure or rearranging the neck? |
| 09:58.39 | *** join/#asterisk kervel (~quassel@d5152FE02.static.telenet.be) |
| 09:59.20 | kervel | hello, i'm debugging a non-working IAX2 connection. i did iax2 enable debug and now i have a debug log, but i wonder if somebody can help me how to interprete the log |
| 09:59.40 | kervel | i see all kinds of ACK/ANSWER/HANGUP things but i don't know what they mean (and what the "reason code" means |
| 10:00.18 | kervel | the iax2 connection is over a openvpn link, and its flakey (works sometimes, sometimes not). but even when it doesn't work i see traffic between the servers |
| 10:03.52 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-etlgqukrezztlaoq) |
| 10:04.02 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 10:05.27 | carrar | mm groovey dinner |
| 10:05.34 | Katty | carrar: hi |
| 10:05.40 | carrar | Katty: hi |
| 10:05.49 | Katty | carrar: i realized my sister is going in for her laminectomy this morning |
| 10:06.02 | Katty | carrar: it makes my sinus infection...laughable in comparison |
| 10:06.02 | carrar | what is that? |
| 10:06.20 | Katty | it's back surgery |
| 10:06.37 | Katty | basically they take bits of bone off your spine to relieve pressure being put on the chord |
| 10:06.50 | carrar | ouch |
| 10:07.03 | Katty | well you're not awake for it |
| 10:07.07 | Katty | but it's a 5 hour surgery |
| 10:07.21 | carrar | sounds dangerous |
| 10:07.34 | Katty | yeah there are some risks |
| 10:07.37 | carrar | any spinal surgery sounds crazy |
| 10:07.46 | Katty | but the doctor doing the surgery has 20 years of experience doing it |
| 10:07.59 | carrar | or so you hope |
| 10:08.09 | Katty | but the risks of them NOT doing the surgery are...yeah |
| 10:08.31 | Katty | pressure on the spinal chord is dangerious |
| 10:08.43 | carrar | I had pressure on mine today |
| 10:08.59 | carrar | walking around for 2.5 hours with a crap load of stuff on my back |
| 10:09.15 | Katty | nods |
| 10:09.17 | carrar | but I'm used to it |
| 10:09.49 | Katty | i'm not sure your spinal chord ever gets used to it :P |
| 10:10.15 | Katty | but it's not like you're squeezing it for months and months at a time straight |
| 10:10.28 | Katty | if she doesn't have the surgery, she could end up paralyzed |
| 10:11.21 | Katty | hopefully everything turns out okay |
| 10:11.24 | carrar | yeah thats never good |
| 10:12.50 | Katty | thinkin i'm gonna trying to sleep again |
| 10:12.57 | carrar | OK THEN! |
| 10:13.00 | carrar | I'll be here |
| 10:13.11 | Katty | yeah if i don't pass out in 15min i'll be back :P |
| 10:13.20 | carrar | heh |
| 10:13.25 | carrar | dream of kittens |
| 10:13.31 | carrar | and ferrets |
| 10:13.35 | Katty | that'd be awesome |
| 10:13.47 | carrar | flying dreams are the best |
| 10:16.36 | *** join/#asterisk codefreeze-lap (~murf@mail.parsetree.com) |
| 10:38.31 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 10:41.46 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-awzdibsnpzhtkkbr) |
| 10:42.42 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-xrzdaigxwdbzcfyo) |
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| 11:17.49 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
| 11:21.00 | *** join/#asterisk lbarth (~lbarth@62.216.165.71) |
| 11:21.04 | lbarth | hello |
| 11:21.11 | carrar | harro |
| 11:24.46 | *** join/#asterisk hmodes (~hmodes@mr.anderson.matrix.gs) |
| 11:27.11 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 11:27.51 | suprstar | anyone have any luck dialing avaya feature access codes from asterisk? ie *30[extension] to turn on the vm light on the phone.. |
| 11:27.56 | suprstar | ast console says -- Requested transfer capability: 0x00 - SPEECH |
| 11:28.21 | suprstar | when u dial *30 with a phone u get dialtone again, then it waits for the extension |
| 11:34.26 | sawgood | suprstar: are you trying to 'turn on a light' on one of the programmable buttons? |
| 11:35.25 | suprstar | no, we have avaya 2402's, 2410's and 2420's. regular digital phones, physical red light on the top |
| 11:35.50 | sawgood | So, you want to use a digital phone on an Asterisk box (what type of gateway do you have to do this)? |
| 11:36.52 | suprstar | actually, we have an avaya phone switch, my asterisk server is connected via multiple E1's. the phones are connected to avaya, if it goes to cover, it gets forwarded to asterisk voicemail |
| 11:37.32 | suprstar | so i have an mwi script that creates the outgoing call file in /var/spool/asterisk/outgoing |
| 11:37.34 | sawgood | oh .. the Asterisk box is just a voicemail box then .... |
| 11:37.39 | suprstar | ya, |
| 11:37.41 | sawgood | nice! |
| 11:37.50 | sawgood | Why not use IP Office? |
| 11:37.53 | sawgood | very expensive! |
| 11:37.55 | suprstar | the mwi script runs, asterisk says call completed.. |
| 11:38.17 | suprstar | it's a huge phone network, 8000 phones in 200 bldg's. |
| 11:38.22 | sawgood | wow! |
| 11:38.27 | sawgood | Enterprise at that |
| 11:38.30 | sawgood | amazing! |
| 11:38.32 | suprstar | avaya wants 400k for vm, |
| 11:39.05 | suprstar | i have everything working perfectly EXCEPT this light..... |
| 11:40.10 | suprstar | i think asterisk hears the dial tone, then switches to speech, then avaya doesnt 'hear' the dtmf tones for the extensino |
| 11:41.21 | suprstar | idk what transfer capability the call as in before it switches to speech, or to force it to stay in that mode, |
| 11:41.36 | suprstar | or if that's even gonna solve it.. but i g2 try something.. |
| 11:41.57 | sawgood | superstar: I know Digium will tell you for $595 dollars (a whole lot less than 400k) |
| 11:42.10 | pentanol | hello, anybody here did communitate users in one room distributed on few asterisk servers? |
| 11:42.13 | sawgood | maybe later on more people will be in the room |
| 11:43.07 | suprstar | yeah im sure someone knows something, it'll get figured out.. im not on a big time crunch |
| 11:43.50 | sawgood | cool |
| 11:44.01 | suprstar | we have an existing asterisk system, ver 0.9 on 7yo hardware, looking thru the configs on there.. i dont have much left to look at and compare :D |
| 11:44.24 | sawgood | you are running a pre 1.x release of Asterisk? |
| 11:44.39 | *** join/#asterisk rare1980_ (~rare1980@12.25.228.67) |
| 11:44.43 | suprstar | ya in full production! been rock solid for 6-7 years |
| 11:45.03 | rare1980_ | i have download g729 codec file from http://asterisk.hosting.lv/ |
| 11:45.12 | suprstar | so a lot of the config syntax is all different |
| 11:45.15 | sawgood | Since 2003-04 ... amazing ... that is almost pre SIP 2.0 days |
| 11:45.19 | rare1980_ | now plz any one can guide mein how to install g729? |
| 11:45.31 | suprstar | they do sip too, |
| 11:45.56 | suprstar | i didnt set em up tho, im just tasked with replacing em |
| 11:53.47 | *** join/#asterisk smooth_penguin (~smoove@59.95.35.55) |
| 11:55.15 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 11:56.27 | *** join/#asterisk jtodd (rxebuuj768@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 11:56.27 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 12:00.38 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 12:06.08 | kruemeltee | how do I use prefixes within the dialled number? As I've got a range of numbers from my sip-provider I have to handle then separately ... (at first I have to know what number was dialled) :-) |
| 12:07.45 | kruemeltee | currently I just get the prefix, whether I dialled the prefix-00 or the prefix-99 number ... :-/ |
| 12:08.08 | suprstar | can you put like exten => _286X. in your dialplan for 286 prefix? |
| 12:08.32 | carrar | rare1980_, You're not gonna get support in here for illegally using that g729 codec |
| 12:08.51 | kruemeltee | sure ... but if I do this and have no _286 within my dialplay, an error occured ... I do not get the "X" part of this number |
| 12:09.22 | suprstar | X is a wildcard, |
| 12:09.34 | suprstar | _286X. means 286-anything |
| 12:09.36 | kruemeltee | yea ... I know ... |
| 12:10.17 | kruemeltee | yea ... but if I dialled 2861 the call goes to the Extension _286, not to the _286X |
| 12:10.32 | kaldemar | _286X. means anything that starts with 286 and is at least 5 digits. "." is one or more. |
| 12:10.39 | kruemeltee | and if _286 is missing asterisk tells me "no extension found" |
| 12:10.47 | contrabanda | helloooo |
| 12:11.33 | kruemeltee | right ... I've just tried it with _286. too ... the same thing is happening ... seems as if the dialled number is everytime the same ... |
| 12:11.44 | contrabanda | please need help. i have configured /etc/dahdi/system.conf and indicated dchan=16 but srom the satatus i see that 16 PRI HDLCFCS RED |
| 12:11.47 | kruemeltee | 5 digits? |
| 12:11.56 | contrabanda | how can i fix it |
| 12:11.57 | kruemeltee | not 4? |
| 12:11.57 | contrabanda | ? |
| 12:12.19 | *** join/#asterisk farkus (~chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 12:12.45 | kruemeltee | _286X means 2860, 2861, ... 2869, doesn't it? |
| 12:12.46 | kaldemar | kruemeltee: X is one digit, "." is one or more. with 286 in front, it's 5 or more. |
| 12:13.03 | *** join/#asterisk Skeeter- (skeeter@c216.218.2-65.clta.globetrotter.net) |
| 12:13.04 | suprstar | yeah thats true, sry.. do u have 3 digit AND 4 digit extensinos in your system? |
| 12:13.37 | kruemeltee | yes I have ... |
| 12:13.41 | contrabanda | hellooooo |
| 12:15.18 | kruemeltee | our global number is 896911 ... after that we have two following digits ... so we can dial 896911-00 up to 896911-99 ... but whatever I dialled, the shown number within asterisk is always the same ... it 896911 ... nothing more |
| 12:16.01 | ManxPower-work | contrabanda, Almost nobody uses SS7 here. |
| 12:16.37 | ManxPower-work | contrabanda, Looks to me like your card is in T-1 mode, not E-1 mode |
| 12:16.55 | kruemeltee | somebody told me to register more than one times at the provider ... every time used with a different number and a different context ... within the contexts I'm able to set the "dialled number" correctly and return to the default ... is that the way I have to do this? |
| 12:23.06 | kruemeltee | currently I'm reading more manuals about our contract with the sip-provider ... the keyword here is "pbx-trunking" ... but I didn't find something about this issue within my asterisk book(s) (hve two of them) |
| 12:24.37 | ManxPower-work | kruemeltee, that's because there is no such thing as a SIP trunk. The term was made up by marketing people to make customers feel better about VoIP. |
| 12:27.03 | kruemeltee | another term is "prefix-binding" ... this seems to me like I have to register multiple times ... but I don't know how ... that's why I ask how to deal with these kind of "bindings" |
| 12:28.21 | contrabanda | ManxPower-work: no its in E1 mode. Now i am trying to setup PRI signalling isdn |
| 12:28.33 | ManxPower-work | kruemeltee, how many times your register or even if you register multiple times is TOTALLY UP TO YOUR CARRIER. |
| 12:29.24 | contrabanda | anybody is experienced with sangoma cards? |
| 12:29.31 | kruemeltee | okay ...so I have to ask them specifically how to do this :-) |
| 12:29.36 | ManxPower-work | contrabanda, RED alarm means "no signal received from far end" |
| 12:29.41 | kruemeltee | thought it's an * issue :-) |
| 12:30.11 | contrabanda | ManxPower-work: yes but its shows HDLC protocol |
| 12:30.16 | ManxPower-work | you really should be using hardhdlc as documented by Sangoma |
| 12:30.17 | contrabanda | thats my main problem |
| 12:30.40 | ManxPower-work | no, your main problem is that Asterisk does not see a line plugged into that port. |
| 12:30.52 | *** join/#asterisk brezular (~brezular@bband-dyn165.178-40-18.t-com.sk) |
| 12:31.01 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:31.02 | *** join/#asterisk |amadeus| (~amadeus@75.130.147.152) |
| 12:31.04 | ManxPower-work | Either that or you did something reall stupid like setting the wrong signalling mode. |
| 12:31.30 | ManxPower-work | contrabanda, does dahdi_tool show that port in RED alarm? |
| 12:32.30 | *** join/#asterisk pentanol (~pentanol@77-35-13-226.pppoe.primorye.net.ru) |
| 12:33.01 | ariel_ | morning |
| 12:33.10 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 12:33.15 | contrabanda | ManxPower-work: no i have just deleted everything and want to install once again. |
| 12:33.45 | ManxPower-work | contrabanda, I wish you the BEST of luck. I'm glad that very few peopel have so much trouble with setting up a card. |
| 12:33.59 | *** join/#asterisk mag-samara (~mezhuev@81.211.87.226) |
| 12:34.02 | ManxPower-work | It's pretty much dirt simple when you follow the Sangoma Wiki instructions |
| 12:36.49 | contrabanda | ManxPower-work: should i have to choose 1. Zaptel/Dahdi - PRI CPE option? |
| 12:37.06 | contrabanda | or 11. Sangoma SMG/sangoma_prid- PRI CPE |
| 12:37.13 | ManxPower-work | 1 |
| 12:37.56 | mag-samara | Hello to ALL! |
| 12:37.56 | mag-samara | Does anybody knows how to adjust Jitter Buffer for app_meetme for incoming calls from SIP? Now I'm using local channel in dialplan, but I don't know how to adjust JB in this case. Anybody help! |
| 12:38.12 | ManxPower-work | mag-samara, using 1.6.x or 1.4.x? |
| 12:38.22 | mag-samara | 1.6.0.6 |
| 12:39.08 | ManxPower-work | mag-samara, did you read the UPGRADE*.txt files or the jitterbuffer* files in the Asterisk source? in 1.4 you can't de-jitter a local channel. I think that was changed in 1.6.mumble |
| 12:40.18 | ManxPower-work | mag-samara, REMEMEBR you can only de-jitter INCOMING audio, not OUTGOING audio |
| 12:40.30 | mag-samara | ManxPower-work: I didn't read. And what is it mean *mumble*? |
| 12:40.58 | ManxPower-work | mag-samara, it means I don't remember if that features was added to 1.6.0.x, 1.6.1.x or 1.6.2.x |
| 12:40.58 | contrabanda | ManxPower-work: now from service dahdi status i see 16 PRI Hardware-assisted HDLC |
| 12:41.27 | ManxPower-work | mag-samara, you will continue to have issues if you don't read the upgrade files, since much of the docs you will see will be for 1.4 |
| 12:42.06 | ManxPower-work | contrabanda, amazing what happens when you follow the instructions... |
| 12:42.13 | mag-samara | ManxPower-work: Thank you. I'm going to search that file "UPGRADE....txt" |
| 12:42.47 | ManxPower-work | mag-samara, look for the jitterbuffer info as well. should be in /path/to/src/asterisk/doc and the UPGRADE*.txt should be in /path/to/src/asterisk |
| 12:42.47 | contrabanda | ManxPower-work: i followed but what does it chande i still see HDLC i need dchannel there |
| 12:43.05 | ManxPower-work | contrabanda, that is the D-channel |
| 12:43.40 | ManxPower-work | contrabanda, I suspect you still have a RED alarm |
| 12:43.53 | ManxPower-work | But that would be because you don't have the line plugged in. |
| 12:45.03 | contrabanda | ManxPower-work: no its in use 16 PRI Hardware-assisted HDLC (In use) |
| 12:45.26 | ManxPower-work | contrabanda, That is CORRECT! |
| 12:45.32 | ManxPower-work | It is in use by asterisk |
| 12:46.07 | *** part/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 12:46.09 | *** join/#asterisk digilink (~digilink@tn-76-5-159-171.sta.embarqhsd.net) |
| 12:46.23 | Skeeter- | is there anything similar as shopbot.ca and pricebat.ca that search online shops in the US? |
| 12:46.36 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 12:47.47 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 12:53.54 | Skeeter- | btw, anyone tried that new TP-Link gigabit wireless N router? |
| 12:55.22 | Skeeter- | TL-WR1043ND |
| 12:55.45 | Skeeter- | It has some good spec, and would be pretty nice with asterisk in it |
| 12:57.55 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 12:59.58 | *** join/#asterisk socain (~socain00@74.255.249.66) |
| 13:02.41 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:08.42 | *** join/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 13:21.17 | *** join/#asterisk jhirley (~jhirley@h69-21-54-248.ldlwvt.dsl.dynamic.tds.net) |
| 13:24.35 | *** join/#asterisk Meaw (~dino@188.161.160.93) |
| 13:31.27 | *** join/#asterisk beefpastry (~tmr@74-129-198-56.dhcp.insightbb.com) |
| 13:33.12 | *** join/#asterisk shader (~user@nom27317d.nomadic.ncsu.edu) |
| 13:34.26 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 13:34.59 | *** join/#asterisk Slashman (~Slash@ariane.fimasys.com) |
| 13:43.09 | *** join/#asterisk sejo (~SeJo@exherbo/developer/sejo) |
| 13:43.34 | sejo | hey all can someone point me to the documentation on how to create the IVR interface? |
| 13:45.22 | suprstar | idk about any docs, but it's real ez to do |
| 13:45.27 | suprstar | exten => 5413,1,Ringing() |
| 13:45.27 | suprstar | exten => 5413,2,Wait(1) |
| 13:45.27 | suprstar | exten => 5413,3,Goto(menu5413,s,1); |
| 13:45.46 | sejo | thanks |
| 13:46.04 | suprstar | then make a context for it |
| 13:46.08 | ManxPower-work | you should not put Ringing unless you need iut. |
| 13:46.30 | ManxPower-work | Asterisk will play ringing sound by default |
| 13:46.34 | suprstar | i just do that to make it pretty for the caller |
| 13:46.36 | suprstar | o |
| 13:46.54 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:46.55 | suprstar | it didn't in the old version i have in production now |
| 13:46.57 | beefpastry | Anybody willing to answer hopefully a quick question on feature code availability for picked up parked calls? |
| 13:47.50 | ManxPower-work | beefpastry, as long as you are not using the non-standard feature codes in FreePBX/Trixbox, then ask your question |
| 13:49.02 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
| 13:49.25 | beefpastry | Well, I am, but *I think* the questions supercedes that...I can't in-call blind after picking up a parked call (my polycom efk can't 're-park'). Am I missing something? |
| 13:50.00 | ManxPower-work | beefpastry, No matter how much you wish it was not so, virtually every question and problem you have will end up being "a FreePBX/Trixbox thing". |
| 13:50.23 | ManxPower-work | beefpastry, what version of Asterisk? |
| 13:50.39 | *** join/#asterisk moy (~chatzilla@74.12.123.160) |
| 13:50.43 | beefpastry | 1.6.0.26 |
| 13:50.54 | *** join/#asterisk jtodd (wsnzz0049i@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 13:50.54 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 13:51.22 | ManxPower-work | My guess would be an issue between freePBX and 1.6.x |
| 13:52.05 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 13:52.14 | ManxPower-work | beefpastry, I'd ask you for CLI output, but it would not do me any good. FreePBX/Trixbox has complex dialplan and scripts |
| 13:52.14 | beefpastry | Could be...just upgraded this past weekend, and I don't recall having the issue in 1.4.24 (but I also don't recall making the attempt). |
| 13:52.23 | ManxPower-work | rollback then. |
| 13:52.45 | beefpastry | Appreciate the attempt. |
| 13:53.33 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 13:54.20 | mag-samara | ManxPower-work: I found only in CHANGES file information concerning added jitterbuffer support for chan_local. But there is no information how to adjust it (change default parameters). I need to reduce JB. But how? |
| 13:54.32 | beefpastry | It's not really a big enough deal to roll back...the advantages of 1.6 outweigh this small thing...can still use the regular transfer. I'd re-do the efk, but then I can't hear the parked extension. |
| 13:55.01 | ManxPower-work | mag-samara, The Changelog and that sort are way too detailed. The UPGRADE*.txt tells you changes that are important |
| 13:55.57 | ManxPower-work | beefpastry, I suspect you'll have other 1.6.x issues. FreePBX was pretty much written for Asterisk 1.2, adapted to work with 1.4, and even then you get messages like "don't use X, it is old" in the CLI. |
| 13:59.11 | mag-samara | ManxPower-work: There is completely nothing in UPGRADE* files (-1.4.txt, 1.6.txt). Also in Changelog there is a couple of words about adding /nj in chan_local. ? |
| 14:00.52 | beefpastry | Really other than this one thing, it's working well. My stuff may not be all that complicated, though. I tried the upgrade to see if the fax would work out of the box...and it did (Couldn1.4 couldn't handle it, and my SIP provider does t.38) |
| 14:00.54 | mag-samara | ManxPower-work: Also, I'm trying to grep buffer in ./doc/* folder. Still nothing =( |
| 14:01.31 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:01.31 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:01.56 | ManxPower-work | well in 1.4 in /path/to/src/asterisk/doc/jitterbuffer.txt talks all about the jitterbuffer. |
| 14:02.28 | ManxPower-work | I don't currently use 1.6, but I do read the changelog and upgrade*.txt for it. |
| 14:04.18 | ManxPower-work | "Verizon tech tagged and located the numbers but did not tag and locate number that is down" |
| 14:04.22 | ManxPower-work | gotta love VZ |
| 14:05.32 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 14:16.04 | *** join/#asterisk drfreeze (~Jim@207.191.114.82) |
| 14:16.31 | drfreeze | Hello |
| 14:17.47 | drfreeze | I am setting up a polycom 335 for a client where the phone is indenpendently connected to their network and registered with 'some' VoIP service. |
| 14:18.05 | Naikrovek | ok |
| 14:18.33 | drfreeze | They plan on using the phone for about 2000 min/mo and I'm wondering if anyone has a suggestion for a good and inexpensive voip provider |
| 14:18.52 | jblack | 2k min a month? There's a zillion to choose from |
| 14:19.14 | jblack | even an expensive plan will only run him $40 a month. |
| 14:19.31 | jblack | (that's at 2 cents a minute, many providers are far cheaper) |
| 14:20.00 | Naikrovek | i pay $44 (including taxes and fees) and get unlimited incoming and free local and longdistance |
| 14:20.23 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 14:20.25 | drfreeze | This phone will primarily be used for outgoing calls |
| 14:25.32 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 14:26.26 | drfreeze | Ugh: 3.9¢ after first 200 minutes |
| 14:27.06 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:27.50 | phonic | try callcentric.com |
| 14:28.01 | phonic | they have pretty good/cheap plans. |
| 14:28.33 | phonic | I think (off the top of my head) that unlimited is $19.95/month, and pay-as-you-go is $0.019/min |
| 14:34.31 | drfreeze | voicepulse looked good, until I saw that they do not support polycom as the primary phone |
| 14:41.11 | drfreeze | Looks like Vonage may work |
| 14:41.19 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 14:41.22 | Naikrovek | vonage? |
| 14:41.27 | wcselby | o/ |
| 14:41.30 | Naikrovek | with a polycom phone? |
| 14:42.21 | suprstar | anyone have experience dialing avaya feature access codes from asterisk? ie *30[extension] to turn the voicemail light on the phone on |
| 14:44.05 | Naikrovek | i love that my bosses think IRC is some unix system admin thing |
| 14:44.23 | wcselby | you want to turn on MWI on a phone connected to avaya, or connected to asterisk? |
| 14:44.27 | Naikrovek | Xchat full screen, boss rolls by: "you're always working hard, nice to see that." |
| 14:44.29 | Naikrovek | pfft. |
| 14:44.33 | wcselby | Naikrovek - lol |
| 14:44.35 | suprstar | avaya, |
| 14:45.01 | suprstar | *30, then i get dial tone again, then i dial 4 digit ext, then i get error tone |
| 14:45.14 | *** join/#asterisk coppice (~chatzilla@202.62.81.147) |
| 14:45.21 | wcselby | can't you just setup an exten in asterisk "exten => *30X.,1,Dial(SIP/AvayaPeer/${EXTEN})" ? |
| 14:45.36 | wcselby | hmmm |
| 14:45.40 | suprstar | yeah i have that already, |
| 14:45.47 | wcselby | not sure about the dial *30 then dial tone again deal |
| 14:45.57 | suprstar | i can see the call being made on asterisk and the switch |
| 14:46.05 | suprstar | this is over an e1 not sip |
| 14:46.23 | wcselby | okay, change the tech from SIP to DAHDI/G1/ |
| 14:46.29 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 14:46.33 | wcselby | or whatever you have configured |
| 14:46.38 | wcselby | same principle though |
| 14:46.40 | suprstar | prolly an avaya prob. |
| 14:46.42 | suprstar | ya it is, i know |
| 14:47.07 | suprstar | exten => _*X.,1,NoOp(${EXTEN} dialed via * catchall) |
| 14:47.07 | suprstar | exten => _*X.,n,Dial(DAHDI/g1/${EXTEN}) |
| 14:47.07 | suprstar | exten => _*X.,n,Hangup |
| 14:48.48 | wcselby | what happens when you do this? |
| 14:48.59 | ManxPower-work | he dials the number with a * in it. |
| 14:49.39 | wcselby | vi /home/phones/polycom/ip601/0004f204ecec.cfg |
| 14:49.43 | wcselby | erm |
| 14:49.45 | wcselby | sorry, wrong window |
| 14:49.47 | Naikrovek | heh |
| 14:49.48 | suprstar | <PROTECTED> |
| 14:49.48 | suprstar | <PROTECTED> |
| 14:49.48 | suprstar | <PROTECTED> |
| 14:50.29 | ManxPower-work | suprstar, that is a really screwed up dial line! |
| 14:50.43 | suprstar | ya i tried a million things, the 'w' is a pause right? |
| 14:50.53 | ManxPower-work | D() is NOT part of the dial string. |
| 14:51.11 | suprstar | ya, i tried it with and without, im kinda grasping at straws here |
| 14:51.18 | wcselby | yeah where is that D(2029) coming from? |
| 14:51.19 | suprstar | thats in my mwi script, if i take it out: |
| 14:51.25 | wcselby | s/where/what |
| 14:51.31 | ManxPower-work | Dial(DADHI/g1/*30w,,D(2029)) |
| 14:51.35 | wcselby | bleh, nvm |
| 14:52.04 | wcselby | i should focus on one thing at a time this morning it seems like |
| 14:52.05 | ManxPower-work | you might want more than ine w, but other than that it should work like I showed |
| 14:52.59 | suprstar | ok let me try that once, |
| 14:54.59 | suprstar | <PROTECTED> |
| 14:54.59 | suprstar | <PROTECTED> |
| 14:54.59 | suprstar | <PROTECTED> |
| 14:54.59 | suprstar | <PROTECTED> |
| 14:54.59 | suprstar | <PROTECTED> |
| 14:54.59 | suprstar | <PROTECTED> |
| 14:55.00 | suprstar | [Mar 30 09:45:12] NOTICE[9760]: pbx_spool.c:349 attempt_thread: Call completed to DAHDI/g1/*30ww,,D(2029) |
| 14:55.02 | *** join/#asterisk geneticx_wrk (~geneticx_@host-208-88-126-198.biznesshosting.net) |
| 14:55.51 | suprstar | whats with the "-- Requested transfer capability: 0x00 - SPEECH" - that doesnt happen on my old system. Is that because asterisk is hearing the dialtone after *30? |
| 14:56.11 | [TK]D-Fender | suprstar: PASTEBIN. Please stop flooding the channel |
| 14:56.29 | ManxPower-work | suprstar, pastebin that part of your dialplan, it does not look right. |
| 14:56.54 | [TK]D-Fender | [10:54]<suprstar>[Mar 30 09:45:12] NOTICE[9760]: pbx_spool.c:349 attempt_thread: Call completed to DAHDI/g1/*30ww,,D(2029) <--and you can't put "w"'s alongside the number for the delay. |
| 14:57.05 | [TK]D-Fender | suprstar: You put the w's inside your D() |
| 14:57.13 | ManxPower-work | suprstar, do you have a PRI to your PBX? |
| 14:58.04 | wcselby | he said it was an E1 connection to his avaya |
| 14:58.18 | ManxPower-work | wcselby, That means nothing. |
| 14:58.29 | ManxPower-work | I suspect it is E-1 PRI, rather than E-1 CAS |
| 14:58.34 | ManxPower-work | But it's good to be sure. |
| 14:58.45 | wcselby | ahh, i see what you're getting at |
| 14:59.01 | ManxPower-work | Yes, with PRI you would not put the w as part of the destination number, only on CAS |
| 15:00.20 | Katty | drags in |
| 15:00.50 | suprstar | im pretty sure they're E1 pri's |
| 15:02.08 | suprstar | signalling = pri_cpe |
| 15:03.07 | Skeeter- | ManxPower-work, do you have a park sofkey that work with polycoms? |
| 15:03.33 | ManxPower-work | Skeeter-, parking is for pussies |
| 15:03.42 | Skeeter- | ManxPower-work, explain |
| 15:06.12 | smooth_penguin | hey Katty ! |
| 15:06.24 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:07.11 | Katty | hey smooth_penguin |
| 15:08.53 | smooth_penguin | :> |
| 15:12.02 | *** join/#asterisk TimeRider (steve@5ac7b3c1.bb.sky.com) |
| 15:13.19 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:13.19 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 15:17.49 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
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| 15:21.22 | suprstar | http://pastebin.com/46h340VU |
| 15:21.29 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 15:21.33 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 15:21.40 | suprstar | is my mwi script and the relevant extensinos.conf |
| 15:22.00 | *** join/#asterisk Netgeeks (~chris@173.11.68.155) |
| 15:22.49 | suprstar | without the w's the call doesn't complete, I get Call failed to go through, reason (1) Hangup |
| 15:22.49 | suprstar | <PROTECTED> |
| 15:23.26 | bmoraca_work | Cause No. l - Unallocated (unassigned) number. |
| 15:23.46 | bmoraca_work | PBX isn't getting all the digits or getting the number of digits it expects |
| 15:24.41 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:25.23 | suprstar | if i do a 'list trace tac 0049' on the avaya gui, it shows 'dial *302029' which is correct |
| 15:25.49 | suprstar | i dont know avaya well enough to troubleshoot it there tho.. |
| 15:26.06 | suprstar | learning tho :P |
| 15:26.38 | suprstar | if i do it from a sip phone registered on asterisk, i get avaya error tone |
| 15:26.38 | *** join/#asterisk afink (~chatzilla@204.26.87.226) |
| 15:26.49 | bmoraca_work | the avaya is telling asterisk that the number you're sending it isn't correct. additionally, w's in your dial string should have 0 effect on a PRI as they're only used for analog tone or pulse dialing |
| 15:27.22 | suprstar | without the w's the call doesn't complete, I get Call failed to go through, reason (1) Hangup |
| 15:27.35 | suprstar | i think it just dials too fast |
| 15:27.44 | ManxPower-work | suprstar, you cannot use "w" in the dial string on a PRI |
| 15:27.54 | ManxPower-work | you can only use in the D() part. |
| 15:28.08 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.79) |
| 15:28.15 | bmoraca_work | suprstar: that's the EXACT SAME error you're getting now. the avaya doesn't understand the numbers you're sending it |
| 15:28.49 | suprstar | with the w's I get Call completed to DAHDI/g1/*30ww2029ww |
| 15:28.59 | *** join/#asterisk moy (~chatzilla@74.12.131.229) |
| 15:29.05 | bmoraca_work | and then the avaya throws you back a call code 1 |
| 15:29.10 | bmoraca_work | it's the same damn thing |
| 15:29.18 | bmoraca_work | s/call/cause/ |
| 15:29.33 | *** join/#asterisk Polysics (~Luca@host83-67-dynamic.30-79-r.retail.telecomitalia.it) |
| 15:29.35 | Polysics | hello |
| 15:29.53 | Polysics | how do i perform more than 1 command with FastAGI? |
| 15:29.56 | bmoraca_work | suprstar: but that's besides the point, because w's do nothing for a PRI. |
| 15:30.02 | wcselby | can FFA receive faxes over PRI? |
| 15:30.15 | Polysics | i have a simple test with SayDigits 123 then SayDigits 456, but it only says 123 |
| 15:30.19 | ManxPower-work | suprstar, I suspect Asterisk ignores everything after the w since W IS NOT A VALID DIALED DIGIT ON A PRI |
| 15:30.33 | [TK]D-Fender | wcselby: Yes |
| 15:30.39 | ManxPower-work | Polysics, be sure to read from stdin |
| 15:30.42 | bmoraca_work | wcselby: i would imagine so, yes. however, it seems pointless to spend the money on that for a PRI. |
| 15:31.28 | suprstar | so what would the proper way to send that call be? DAHDI/g1/*30,,D(2029) ? |
| 15:31.29 | ManxPower-work | Polysics, or on FastAGI be sure to read whatever asterisk sends you |
| 15:31.45 | ManxPower-work | suprstar, Didn't I give you that dial string about 20 mins ago? |
| 15:31.50 | suprstar | yep |
| 15:32.04 | Polysics | so i need to wait for 200/OK? |
| 15:32.10 | *** join/#asterisk Z_God (~julius@wlan224038.mobiel.utwente.nl) |
| 15:32.21 | ManxPower-work | but maybe you should just start by dialing as DAHDI/g1/*30 If that doesn't work then nothing else will |
| 15:32.28 | ManxPower-work | Polysics, I think you need to read the AGI spec |
| 15:32.52 | Polysics | what is a good resource? wiki isn't a good help |
| 15:33.12 | wcselby | ManxPower-work - it's interesting how many example AGI scripts are out there that so obviously failed to understand how to do AGI properly. |
| 15:33.28 | Polysics | wcselby, i am just starting out .-) |
| 15:33.51 | ManxPower-work | Polysics, why not just use an AGI library for your language and let that take care of the dirty work? |
| 15:34.06 | wcselby | Polysics - no worries, so was I once. I kept getting errors in my cli when AGI's would run, and they didn't work the way I wanted. I was using example AGI scripts I found online. |
| 15:34.18 | Polysics | because there actually are no usable and not-dead implementations in Ruby :-( |
| 15:34.27 | ManxPower-work | I thought there was AGI docs in Asterisk, but apparently not |
| 15:34.35 | wcselby | I ended up reading up on AGI on the wiki and in the book (mostly in the book), and I learned a lot! |
| 15:34.49 | *** join/#asterisk jcape (~jcape@209.120.251.81) |
| 15:35.01 | wcselby | there's decent info in the book |
| 15:35.02 | wcselby | ~book |
| 15:35.03 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 15:35.08 | jcape | Any pointers on doing followme in odbc? |
| 15:43.59 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 15:51.27 | *** join/#asterisk dinesh___ (~dinesh@84-73-120-175.dclient.hispeed.ch) |
| 15:52.12 | dinesh___ | hi all, I've got a kind of stupid question, but I suscribed to localphone.com for an incoming number, and when i'm unavailable it will always divert to the voicemail, even when i disable it |
| 15:52.33 | dinesh___ | so there there numbers that will always be unavailable, so that I just forward to that number when i'm unavailable ? |
| 15:52.38 | dinesh___ | are there* |
| 16:01.15 | Gugge | dinesh___: huh? |
| 16:01.47 | mag-samara | ManxPower-work: I have read asterisk-1.4.*/doc/jitterbuffer.txt. Now I can see that asterisk 1.4 has more than one JB implementations. BUT what is "the best practice" for enable JB for incoming SIP calls to app_meetme? And another question: is it possible to change JB paremeters for Local/.../nj construction? |
| 16:04.02 | hardwire | ChannelZ: how did sawgood turn out? |
| 16:09.22 | mag-samara | Anybody help me with JitterBuffer and app_meetme please... |
| 16:12.10 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 16:14.46 | bmoraca_work | oidview.com is the coolest site ever |
| 16:16.14 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:16.17 | Naikrovek | ever? |
| 16:18.08 | bmoraca_work | yes, ever |
| 16:19.21 | bmoraca_work | snmp is way cool |
| 16:19.58 | *** part/#asterisk jcape (~jcape@209.120.251.81) |
| 16:24.19 | bmoraca_work | dude...i can, via SNMP, monitor the individual DS0s on my AS5400 and all active calls. that's WAY cool! |
| 16:26.14 | *** join/#asterisk oc-bill (~none@static-63-238-185-39.t1.cavtel.net) |
| 16:28.10 | oc-bill | Please don't blast me - as I'm new, but I need some help. I'm going to convert one of my offices over to a full asterisk solution. I'm looking at phones, and it seems like there are a ton of choices. Can anyone make a recommendation on a nice, reliable phone set that also suppports the MWI light from asterisk vm? |
| 16:28.27 | Naikrovek | oc-bill: polycom |
| 16:28.34 | Naikrovek | polycom polycom polycom polycom |
| 16:28.42 | Naikrovek | what's your per-phone budget |
| 16:28.59 | *** join/#asterisk btsteve (~tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 16:29.01 | oc-bill | VERY good question :) Probably under $150 |
| 16:29.13 | Naikrovek | oc-bill: soundpoint ip-335 |
| 16:29.23 | oc-bill | only looking at 20 phones, and 2 or 3 "manager" phones |
| 16:30.00 | wcselby | the polycom's are nice, so are the cisco SPA phones (the new line of them, the SPA 5xx series). |
| 16:30.01 | Naikrovek | ip335 for the majority of phones, soundpoint ip450 or better for "manager" phones. don't know why managers need better phones but whatever |
| 16:30.21 | wcselby | Naikrovek - because they're better than the little people, and their phones should be too. :P |
| 16:30.32 | Naikrovek | wcselby: recipe for disaster |
| 16:30.38 | oc-bill | typically "better" means more speed dials ... they call them "side-cars" - for the operator, etc |
| 16:30.53 | Naikrovek | more buttons |
| 16:30.59 | oc-bill | yep |
| 16:31.02 | Naikrovek | even the ip335 has 99 speed dial slots |
| 16:31.08 | wcselby | oc-bill - if you go with polycoms, get at least one ip550 or 650 for reception |
| 16:31.13 | Naikrovek | yeah |
| 16:31.18 | coppice | Naikrovek: the managers will be OK with an IP335 that has an "I'm very important" sticker on them |
| 16:31.31 | oc-bill | ha |
| 16:31.35 | Naikrovek | it's true |
| 16:31.41 | wcselby | you can put up to three side cars on the 550 / 650, each side car supports an additional 13 speed dials, I think |
| 16:31.45 | Naikrovek | just tell them their phones have the better firmware and they'll be happy |
| 16:31.56 | Naikrovek | wcselby: 14 as i recall |
| 16:32.01 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 16:32.02 | wcselby | Naikrovek - probably |
| 16:32.14 | wcselby | i know it's up to 19 speed dials for an ip601 with one side car |
| 16:32.20 | oc-bill | do the side cars also have a light indicating whether the person is on the phone or not? |
| 16:32.34 | wcselby | oc-bill - yes, if you set it up properly |
| 16:32.35 | Naikrovek | oc-bill: not a light, but an icon next to their name |
| 16:32.42 | Naikrovek | oh light, really |
| 16:33.02 | wcselby | Naikrovek - yeah, at least on our 601's they do. i don't kknow why they'd remove functionality on a 650 model |
| 16:33.03 | oc-bill | awesome.. thanks everyone |
| 16:33.09 | oc-bill | one last question |
| 16:33.23 | oc-bill | i see they're offered by a few vendors --- any one in particular I should support? |
| 16:33.35 | wcselby | oc-bill - also, the cisco SPA 5xx lines should do most of the same things as the polycom's we've mentioned |
| 16:33.47 | bmoraca_work | i suspect that the "managers" want BLF and those kinds of things |
| 16:34.08 | wcselby | oc-bill, also to consider, are you going to use POE or will each phone need a power adapter |
| 16:34.37 | wcselby | oc-bill - and also, will you need to run two network drops to each desk, or will you buy a phone with a switch in it so you only need one drop, etc |
| 16:35.12 | oc-bill | good points... looks like the IP330 is a switch based , and offers the option of AC or POE |
| 16:35.12 | mag-samara | Anybody help me with JitterBuffer and app_meetme please... |
| 16:35.16 | wcselby | i tend to use voipsupply, but that's just because they're who I've used. Katty and Naikrovek use another one, the name slips my mind at the moment |
| 16:35.20 | *** join/#asterisk flapjacks (~flapjacks@wsip-70-166-201-90.ph.ph.cox.net) |
| 16:35.25 | bmoraca_work | hrm |
| 16:35.40 | bmoraca_work | whatsupgold sucks...i need a new (preferably windows-based) SNMP monitor |
| 16:36.05 | *** join/#asterisk methodvon (~methodvon@pool-71-191-177-105.washdc.fios.verizon.net) |
| 16:36.14 | bmoraca_work | wcselby: i like VOIPLINK. telephonydepot has good prices, but they're shitty to work with |
| 16:36.33 | bmoraca_work | for polycom phones, though, i buy them from ingram micro or techdata because i'm polycom certified like that |
| 16:36.44 | oc-bill | bmoraca_work: I know it's not windows based, but Nagios rocks |
| 16:37.09 | wcselby | nagios is good, cactus is okay, zenoss is fun when you get it properly setup. all *nix based though |
| 16:37.09 | oc-bill | bmoraca_work: simple to setup and manage |
| 16:37.43 | bmoraca_work | oc-bill: can it monitor a particular OID group and provide me a report on all instances under it? i have an application that writes a new instance all the time, and i'd like to monitor them all (the instance IDs constantly change) |
| 16:37.44 | oc-bill | bmoraca_work: we use Nagios for alerting paired with Cacti (and a weathermap plugin) for historical/trends |
| 16:38.46 | oc-bill | bmoraca_work: good question. I dont believe that would be a cakewalk. You'd have to write a custom script to do it, and since you weren't keen on going the *nix route, it's going to be a pushup |
| 16:38.53 | bmoraca_work | (AS5400 writes a new instance for each and every call that goes through. i'd like to monitor currently active calls, complete with duration, calling party and called party) |
| 16:39.12 | [TK]D-Fender | IP320/330 = outdated. Get the new 321/331 or 335 instead |
| 16:39.14 | wcselby | bmoraca_work - I did that with zenoss once, but it was a pain to setup and I don't remember how I did it. but that was also a custom app that didn't already have a zenoss plugin. if you're talking a cisco device, there'e probably already a plugin that does was you want (plugins are called zenpacks with zenoss, I think) |
| 16:39.14 | *** join/#asterisk Ursadon (~kvirc@95.79.94.74) |
| 16:42.35 | bmoraca_work | this looks like waaaaay too much work |
| 16:43.24 | bmoraca_work | i don't have time to learn an entirely new architecture |
| 16:45.16 | Naikrovek | telephonydepot.com is where i get mine |
| 16:49.18 | *** join/#asterisk methodvon (~methodvon@pool-71-191-177-105.washdc.fios.verizon.net) |
| 16:49.32 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 16:49.43 | *** join/#asterisk methodvon (~methodvon@pool-71-191-177-105.washdc.fios.verizon.net) |
| 16:49.57 | wcselby | holy crap |
| 16:50.09 | seanbright | you can say that again |
| 16:50.12 | wcselby | windows 7, shift-right click on program links to get the menu I'm used to |
| 16:50.47 | *** join/#asterisk soman (~somnath@dsl-jklbrasgw1-fe19fb00-113.dhcp.inet.fi) |
| 16:53.09 | ManxPower-work | I still use XP |
| 16:53.18 | hardwire | any idea on how to Record native to the negotiated codec? |
| 16:53.48 | ManxPower-work | hardwire, did you check channelvariables.txt? |
| 16:54.05 | hardwire | ManxPower-work: orly.. and it won't attempt to make an sln path? |
| 16:54.13 | hardwire | will just use the channel codec then |
| 16:54.14 | [TK]D-Fender | hardwire: No |
| 16:54.14 | ManxPower-work | IIRC the CHANNEL() variable or maybe SIP() variable has the current codec. |
| 16:54.39 | ManxPower-work | hardwire, now what about calls that get transcoded? |
| 16:54.53 | hardwire | ManxPower-work: no point in retranscoding them.. that's why I want to do this |
| 16:54.56 | ManxPower-work | hardwire, anyway, stop being a cheapskate and buy some G729 licenses 8-)_ |
| 16:55.03 | methodvon | I'm looking for information on best practices for implementing speech-to-text in Asterisk. Any push in the right direction would be greatly appreciated. |
| 16:55.03 | hardwire | ManxPower-work: I haz em |
| 16:55.12 | mag-samara | ManxPower-work: I have read asterisk-1.4.*/doc/jitterbuffer.txt. Now I can see that asterisk 1.4 has more than one JB implementations. BUT what is "the best practice" for enable JB for incoming SIP calls to app_meetme? And another question: is it possible to change JB paremeters for Local/.../nj construction? |
| 16:55.38 | ManxPower-work | mag-samara, you now know everything I know about the Asterisk jitterbuffer. |
| 16:55.41 | wcselby | mag-samara - if you're not getting a response in here, you can try the mailing list |
| 16:55.53 | ManxPower-work | ~mailinglist |
| 16:55.54 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 16:56.22 | mag-samara | =( I see. Thans you anyway. |
| 16:57.25 | hardwire | there.. I now have it set up so that evil things are afoot. |
| 16:57.29 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:59.18 | *** join/#asterisk smooth_penguin (~smoove@59.95.35.55) |
| 16:59.56 | zzlane33333 | I am having trouble with Audio passing through from my handset to the PBX, I have had this before because of my NAT settings but I'm not exactly sure how to setup NAT in FreePBX 2.7 |
| 17:00.08 | zzlane33333 | I see Asterisk SIP Settings |
| 17:00.21 | zzlane33333 | I see the option to set it up* in Asterisk SIP Settings |
| 17:00.36 | zzlane33333 | however it doesn't appear to be taking affect |
| 17:00.37 | *** join/#asterisk Ryushin (proxy@windwalker.openinnovations.com) |
| 17:00.52 | Naikrovek | zzlane33333: freepbx stuff goes in #freepbx |
| 17:01.11 | zzlane33333 | woops |
| 17:01.14 | zzlane33333 | sorry wrong channel |
| 17:03.26 | *** join/#asterisk Skeeter- (skeeter@c216.218.2-65.clta.globetrotter.net) |
| 17:06.26 | *** join/#asterisk Meaw (~dino@188.161.160.93) |
| 17:07.17 | Skeeter- | The webinar from Asterisk + Lumenvox is starting for those who are interested |
| 17:10.09 | bmoraca_work | what's it regarding? |
| 17:10.35 | *** join/#asterisk methodvon (~methodvon@pool-71-191-177-105.washdc.fios.verizon.net) |
| 17:12.52 | [TK]D-Fender | Is that on the Discovery Channel? |
| 17:13.00 | outtolunc | gotomeeting <G> |
| 17:13.26 | Skeeter- | <PROTECTED> |
| 17:13.54 | outtolunc | yeah, slow system |
| 17:14.31 | oc-bill | one last question - paging and moh |
| 17:14.43 | oc-bill | just an fxs card? |
| 17:15.22 | Naikrovek | yes paging and moh are possible on asterisk |
| 17:15.32 | Naikrovek | not sure what you're asking re: fxs card |
| 17:15.53 | oc-bill | sorry :) |
| 17:16.11 | oc-bill | just wondering what hardware i'd need. we have a paging module now that is just an analog station |
| 17:16.19 | oc-bill | so that seems like an FXS card/port |
| 17:16.28 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 17:16.33 | oc-bill | not sure about what i'd need for music on hold though |
| 17:16.40 | wcselby | moh is built into the system |
| 17:17.01 | wcselby | you would need to create .wav files of the music you want, or use the (crappy) freeplay files that come with asterisk |
| 17:17.31 | wcselby | paging can be done, whether it will work with your existing system depends on a lot of factors. do you have a model number for your paging module? |
| 17:17.41 | Naikrovek | oc-bill: polycom phones can be paged as well. intercom as well |
| 17:18.09 | wcselby | anyone have a list of what the different polycom ringers are? |
| 17:18.18 | ManxPower-work | oc-bill, Music on Hold is the music some hears when they are placed on hold. I don't know what the heck you are talking about for a "paging module" |
| 17:18.21 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 17:18.48 | oc-bill | :) sorry guys. let me get the name of it |
| 17:18.52 | ManxPower-work | oc-bill, maybe you are just confused and mean "overhead paging", but we can't (usually) readminds. |
| 17:18.58 | bmoraca_work | oc-bill: best paging I've found is a Valcom VIP201. thing works freaking great. |
| 17:19.42 | bmoraca_work | Cyberdata makes some interesting ceiling-tile speakers, but the Valcom will integrate with your current speakers and supports background-music, which the Cyberdatas don't unless you're using MGCP |
| 17:19.48 | oc-bill | that's what we have.. valcom - not sure of the model |
| 17:20.26 | bmoraca_work | oc-bill: even easier. the VIP series are SIP-enabled paging gateways. the 201 allows for 1 sip zone, but they scale up to 8. it'll interface well with your environment |
| 17:21.38 | *** join/#asterisk flapjacks (~flapjacks@wsip-70-166-201-90.ph.ph.cox.net) |
| 17:21.59 | *** join/#asterisk theron (~theron@ip244.scolloc.lh.net) |
| 17:22.33 | *** join/#asterisk bakermd (~bakermd@38.104.0.102) |
| 17:24.13 | bakermd | When a call fails (originated from a .call file) I can see in the CLI that an error is generated such as "Got SIP response 484 "Address Incomplete" back", however I do not see how I can get a handle on this data (i.e. for saving in a DB). I have found that ${REASON} is available to me, but it simply is set to '1' which relates to 'HANGUP' ... not very useful. Ideas? |
| 17:26.07 | ManxPower-work | bakermd, send the call to Local/ext@context, then handle it as a regular Dial with all the info available like HANGUPCAUSE and DIALSTATUS |
| 17:26.53 | bakermd | ManxPower-work: This call is going out, not to a local ext - does this still apply? |
| 17:27.15 | RobH | bah, i lost my bookmarks |
| 17:27.17 | RobH | ~book |
| 17:27.18 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:27.19 | ManxPower-work | yes, you use Local/ to send the call out via the dialplan rather than directly to the channel |
| 17:27.36 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 17:29.07 | ManxPower-work | bakermd, read up on /path/to/src/asterisk/doc/channelvariables.tex Complain to Digium if you don't like that it's not in plaintext |
| 17:29.31 | bakermd | How do I read a .tex? |
| 17:29.37 | *** join/#asterisk dennisG (~dennisG@2002:541e:88d0:0:213:2ff:fe56:e380) |
| 17:29.38 | ManxPower-work | cat will do it. |
| 17:29.44 | ManxPower-work | just ignore all the formatting crap. |
| 17:29.44 | bakermd | cool - thanks |
| 17:30.05 | ManxPower-work | otherwise you can install LaTeX, which is like 100MB of stuff |
| 17:30.38 | Naikrovek | pdflatex asterisk.tex; pdfview asterisk.pdf <---- something like that |
| 17:30.51 | ManxPower-work | Naikrovek, you run a GUI on your Asterisk server?? |
| 17:31.17 | Naikrovek | no |
| 17:31.26 | ManxPower-work | then how are you going to view that via SSH? |
| 17:31.35 | Naikrovek | i downloaded the source to my desktop machine and did it there |
| 17:31.45 | Naikrovek | ManxPower-work: x-server is how you do it over ssh |
| 17:32.15 | ManxPower-work | Seems to be a lot of work and time just to look at a file that should be in plaintext |
| 17:32.17 | p3nguin | Most people don't have Xorg installed on their servers. |
| 17:32.25 | p3nguin | At least normal server admins don't. |
| 17:32.29 | Naikrovek | you don't need an XServer on the asterisk server |
| 17:32.31 | ManxPower-work | p3nguin, don't worry. We'll never win this arguement. |
| 17:32.36 | Naikrovek | you need x clients on the asterisk server |
| 17:32.43 | ManxPower-work | Naikrovek, you just need one on your desktop. |
| 17:32.45 | Naikrovek | X is a lost art apparently |
| 17:33.16 | ManxPower-work | Naikrovek, yes, it is. |
| 17:33.49 | ManxPower-work | If they want a barrier to entry release Asterisk in RAR format, don't make the docs hard to read |
| 17:33.58 | Naikrovek | heh |
| 17:34.11 | ManxPower-work | That is ALL the conversion to TeX is in my mind, a barrier to entry. |
| 17:34.37 | Naikrovek | practically, i think you're right. technically it's no worse than info |
| 17:34.50 | Naikrovek | part of the build process should compile those docs into a pdf |
| 17:35.28 | ManxPower-work | Naikrovek, it does for some of them, but still have to download them to a machine with a GUI to read them. |
| 17:36.05 | Naikrovek | yeah |
| 17:36.11 | Naikrovek | there's no reason they couldn't be plaintext |
| 17:36.21 | Naikrovek | other than someone has a latex fetish at digium |
| 17:36.54 | ManxPower-work | if they MUST be in TeX, at least convert to plaintext during the build process |
| 17:37.42 | *** join/#asterisk zomenox (~brock@254120.firewall.winthrop.edu) |
| 17:40.10 | bakermd | this tex crap is really annoying |
| 17:40.14 | wcselby | i always wondered about the reasoning behind that |
| 17:40.30 | bakermd | No way am I installing a GUI on my server |
| 17:40.30 | Naikrovek | latex fetish at digium <-- there's your explanation |
| 17:40.36 | Naikrovek | you don't NEED A gui on your server |
| 17:40.46 | Naikrovek | to view PDFs that live on the server |
| 17:40.59 | bakermd | hehe - yeah, if they were PDF |
| 17:41.02 | Nugget | latex and home-made jewelery made from T10i dongles |
| 17:41.06 | p3nguin | I can't see a way to get around it if you're going to view the files while they are still on the server. |
| 17:41.08 | bakermd | Maybe there is a latex viewer for the mac... |
| 17:41.19 | Naikrovek | with X, the server lives on the desktop, the client lives on the server |
| 17:41.50 | bakermd | True - getting the x client working well on OpenVZ containers is not always easy though |
| 17:41.57 | ManxPower-work | you are assuming people are running X on their desktop |
| 17:42.05 | ManxPower-work | Most people do not. |
| 17:42.08 | bakermd | I have a Mac, so I do |
| 17:42.08 | p3nguin | I don't think I can install any apps to view the files without at least libx11. |
| 17:42.19 | Naikrovek | so, if you can't work out FTP or SCP or whatever, then you install ONLY a PDF viewer on the server, and you install an X server on your desktop, and magically when you view the PDF on the server, *gasp* it displays on your desktop |
| 17:42.33 | Naikrovek | X is a fundamental unix thing; how is this lost already |
| 17:43.01 | p3nguin | What viewer do you think I can install without x11 being needed as a dependency? |
| 17:43.03 | bakermd | I'd rather SCP the files over and read them on my machine / print them |
| 17:43.04 | ManxPower-work | "install an X server on your desktop" Dude, that's like saying "just stop using so much energy to stop global warming" |
| 17:43.19 | Naikrovek | ManxPower-work: if you run linux you already have one |
| 17:43.28 | ManxPower-work | p3nguin, that's a big difference between libX11 and the X server. |
| 17:43.35 | ManxPower-work | Naikrovek, but if you run Windows you do not. |
| 17:43.56 | ManxPower-work | As much as I hate windows it's elitest to expect people to install an X server just to read a documentation file. |
| 17:43.59 | wcselby | ManxPower-work - they don't use windows boxes at Digium |
| 17:44.14 | Naikrovek | ManxPower-work: xming is a 2.1mb download |
| 17:44.15 | ManxPower-work | wcselby, They don't do a lot of stuff that the real world does. |
| 17:44.27 | wcselby | ManxPower-work - i had an interesting discussion with some of the digium guys about that at astricon last year. |
| 17:44.29 | [TK]D-Fender | Dumb text FTW |
| 17:44.31 | ManxPower-work | Heck they don't even run the latest releases of ASTERISK on their OWN PBX . |
| 17:44.36 | wcselby | lunch, bbl |
| 17:45.27 | ManxPower-work | wcselby, What it "screw the windows users, if they can't figure it out, they don't deserve to use Asterisk"? |
| 17:46.39 | bakermd | I'm not a fan of WIndows, but they certainly deserve to use Asterisk... they're just misguided |
| 17:46.52 | Naikrovek | here we go |
| 17:47.14 | bakermd | "And that's all I have to say about that" |
| 17:47.17 | Naikrovek | no it's not |
| 17:47.21 | bakermd | Wow - 35MB for a latex viewer |
| 17:47.22 | ManxPower-work | bakermd, I'm pretty Elitist, but this goes too far even for me. 8-| |
| 17:47.35 | Naikrovek | you don't say that windows users are misguided, then say that's all there is |
| 17:47.44 | Naikrovek | there's a ton of anger in there |
| 17:47.44 | methodvon | impressive 10 minute tangent on how to view a PDF |
| 17:47.48 | Naikrovek | indeed |
| 17:48.11 | bakermd | lol - calm down - everyone is entitled to an opinion, and I think my statement was clearly opinionj |
| 17:48.12 | Naikrovek | i still don't understand how so many people don't understand how X works |
| 17:48.14 | [TK]D-Fender | PLAIN TEXT. My "cat" can read it |
| 17:48.29 | [TK]D-Fender | [13:48]<bakermd>lol - calm down - everyone is entitled to an opinion, and I think my statement was clearly opinionj <- That's what YOU think... |
| 17:48.29 | ManxPower-work | [TK]D-Fender, channelvariables.tex? |
| 17:48.30 | Naikrovek | plain text is the most portable document format there is |
| 17:48.40 | bakermd | [TK]D-Fender: Correct, though the formatting is difficult to read around |
| 17:48.54 | coppice | plain test is only portable for English |
| 17:49.05 | Naikrovek | coppice, meet unicode |
| 17:49.10 | ManxPower-work | bakermd, chances are the info is in asterisk.pdf |
| 17:49.14 | Naikrovek | unicode, meet coppice |
| 17:49.41 | coppice | unicode and I are old sparring partners. |
| 17:50.18 | *** join/#asterisk makrohard (~makrohard@141.40.138.10) |
| 17:50.20 | makrohard | hi |
| 17:52.11 | bakermd | ManxPower-work: LOL! You are correct.. the docs I need are in there... thanks |
| 17:52.21 | bakermd | 'course I can read the .tex now as well ;) |
| 17:54.33 | ManxPower-work | Hey, they should convert the UPGRADE*.txt to .tex format! |
| 17:54.54 | makrohard | after the successful basic install of asterist and freepbx I still get those errors http://nopaste.info/bd0f3cdce8.html (connection attemt to manager failed). I doublechecked the amportal.conf, manager.conf and op_server.cfg. All passwords match. Which service could try conecting unsuccsessfulle, still? Which config files to check? Is there a way to determine the service, doing the unsuccessful connects? |
| 17:54.56 | Katty | so |
| 17:55.00 | ManxPower-work | ~freepbx |
| 17:55.01 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 17:55.02 | Katty | i have confirmed flu and sinus infection |
| 17:55.14 | bmoraca_work | it's freakin hilarious when customers screw up their networks. |
| 17:55.16 | ManxPower-work | makrohard, FreePBX is not supported here. |
| 17:56.23 | makrohard | That's what the bot told me. I'll thy there, if there will be no asterisk -specific answers in here :) I'm not sure if freepbx causes the problem |
| 17:56.34 | makrohard | but thanks for the advice |
| 17:57.13 | *** join/#asterisk tamiel (~tamiel@ip-253.net-81-220-87.toulouse.rev.numericable.fr) |
| 17:57.15 | Naikrovek | bmoraca_work: yes. what's funnier is when the 'tards in your india office buy a SAN capable of storing 6TB act surprised when they discover it's a lot less when they use RAID, despite about 6 months of time to research this |
| 17:57.59 | bmoraca_work | Naikrovek: <obligatory snide comment about "outsourcing"> |
| 17:58.08 | Naikrovek | bmoraca_work: yes |
| 17:58.17 | Naikrovek | i've made the comment to myself a thousand times today |
| 17:58.34 | *** join/#asterisk Docteh (~mage@24.86.183.239) |
| 17:58.56 | bmoraca_work | some moron put a "permit ip any any" in the inbound ACL of the WAN interface of this router. WTF! |
| 17:59.23 | Naikrovek | bmoraca_work: funnier part is that they didn't do any kind of usage survey to find out what they actually needed before they settled on 6Tb. they FIGURED they'd need 6TB, ordered 6TB, discover it's only 3, then do a survey and discover they need about 8. |
| 17:59.26 | bmoraca_work | SDM has a freakin WIZARD for configuring firewalls for crying out loud |
| 17:59.32 | Naikrovek | could not have came out worse |
| 17:59.47 | bmoraca_work | that's excellent |
| 18:00.03 | Naikrovek | excellent doesn't even begin to describe |
| 18:00.21 | Naikrovek | totally fucking retarded is how i would describe it |
| 18:00.29 | Naikrovek | and their skill levels |
| 18:01.13 | makrohard | The log, pasted above is the asterist log btw |
| 18:01.24 | Naikrovek | once in a while, i'll have a conversation with one of them, where they talk intelligently. then i think to myself "finally, i'll let them make this decision and see how it turns out" then it turns out like this |
| 18:01.49 | ManxPower-work | bmoraca_work, A GUI firewall configuration, welcome to the days of "customer empowerment" |
| 18:01.51 | Katty | Naikrovek: well... |
| 18:01.59 | Katty | Naikrovek: sometimes it isn't always the tech's choice too |
| 18:02.18 | Katty | Naikrovek: you've got those pesky management and sales folk that like to get in the way when you outsource things |
| 18:02.40 | bmoraca_work | ManxPower-work: indeed...but the GUI WIZARD automatically puts a "deny ip any any" rule at the bottom. this had to have been done manually. |
| 18:02.56 | Naikrovek | Katty: yes i know, but they somehow formulated a proposal, with data showing that this is exactly what they needed and why, presented it to me, i OKd it, i presented to CEO, and purchase was authorized, and made as the techs over there requested |
| 18:02.58 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.132.134.dsl.dyn.forthnet.gr) |
| 18:03.27 | Katty | Naikrovek: proposal screams sales. |
| 18:03.28 | Naikrovek | Katty: and they're not outsourced, they just happen to be in india. the owner of the company I work for owns the company in india as well, and their clients are all in india |
| 18:03.34 | makrohard | yea, you were absolutely right. shutting down amportal and starting asterisk only leaves blank logfiles (no further attempts), I'll have to focus in #freepbx |
| 18:03.48 | bmoraca_work | wow, this firewall is fubared. |
| 18:04.07 | Naikrovek | Katty: there were no sales people involved here. trust me. and if there were, they should have sold us more than 6tb if there were worth their weight in air |
| 18:04.17 | Katty | Naikrovek: well, luckily, if you're not happy and it doesn't go against the proposal they're obligated to fix it (= |
| 18:04.19 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 18:05.32 | Naikrovek | Katty: this is a purely internal proposal, our CEO wants to know that the people ordering hardware know what they need. except it turns out that they invented all the numbers in the proposal, except price |
| 18:05.47 | Katty | joy. |
| 18:05.50 | bmoraca_work | lol |
| 18:05.58 | Katty | how about i go visit and cough on them |
| 18:06.04 | Naikrovek | could you? |
| 18:06.09 | Naikrovek | these people are almost saboteurs |
| 18:06.20 | Katty | well not really |
| 18:06.30 | Katty | the doc doesn't want me to going anywhere until my fever is gone |
| 18:07.34 | bmoraca_work | Katty: swineflu? |
| 18:07.48 | Katty | dunno |
| 18:07.53 | Katty | i don't really care |
| 18:10.36 | Katty | unlike 90% of this country, i'm not getting my panties in a wad over swine flu. |
| 18:12.53 | ManxPower-work | rabbies? |
| 18:14.13 | *** join/#asterisk luke-jr_ (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d) |
| 18:14.26 | luke-jr_ | why would Asterisk send "407 Proxy Authentication Required" when SIP has guest enabled? :( |
| 18:15.36 | ManxPower-work | luke-jr_, with allowguest=yes before any register lines in sip.conf [general]? |
| 18:16.25 | ManxPower-work | luke-jr_, And the username claimed by the client is not in sip.conf at all? |
| 18:16.38 | luke-jr_ | there should be no username claimed I think |
| 18:16.55 | ManxPower-work | sip debug is your friend |
| 18:16.57 | luke-jr_ | alloweguest is supposed to be default, right? |
| 18:17.02 | Katty | ManxPower-work: yes. rabbies. |
| 18:17.03 | luke-jr_ | sip debug denies any packet :/ |
| 18:17.13 | ManxPower-work | sip debug does not deny any packets |
| 18:17.43 | luke-jr_ | it doesn't show the INVITE nor 407 |
| 18:17.47 | luke-jr_ | yet tcpdump does |
| 18:18.05 | ManxPower-work | luke-jr_, that makes no sense at all. |
| 18:18.55 | luke-jr_ | :( |
| 18:19.56 | luke-jr_ | meh |
| 18:19.58 | luke-jr_ | sip.conf lies :( |
| 18:20.06 | luke-jr_ | it says allowguest is on by default, but it isn't |
| 18:20.21 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 18:21.23 | p3nguin | ;allowguest=no ; Allow or reject guest calls (default is yes) |
| 18:22.35 | p3nguin | It's commented out in my config, and I get anonymous calls. Seems truthful to me. |
| 18:23.13 | Skeeter- | they made the webinar crash |
| 18:23.14 | Skeeter- | nice |
| 18:24.34 | ManxPower-work | p3nguin, I doubt he actually has an allowguest issue since sip debug does not even show the packets |
| 18:24.56 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 18:25.34 | [TK]D-Fender | No packets = issue not involving * is involved |
| 18:25.47 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:28.35 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 18:28.50 | Naikrovek | wow, raden_work is back |
| 18:28.58 | Naikrovek | ain't seen you for a while |
| 18:29.59 | raden_work | Yea been on the road |
| 18:30.02 | raden_work | :( :( |
| 18:30.20 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 18:36.50 | *** join/#asterisk circut (~circut@c-71-57-110-244.hsd1.il.comcast.net) |
| 18:37.12 | circut | hey all, i just ported my config from zapatel to dahdi, and it appears my trunk groups are not working |
| 18:37.51 | circut | I changed: Dial(Zap/G1/${num}) to Dial(DAHDI/G1/${num}) but i get a 603 |
| 18:38.02 | circut | the asterisk console says it cannot create channel of type dahdi |
| 18:38.25 | circut | but using Dial(DAHDI/1/${num}) works, of course the problem with that is only 1 channel is ever used |
| 18:42.02 | bmoraca_work | circut: pastebinning system.conf and chan_dahdi.conf would be INFINITELY helpful |
| 18:42.06 | bmoraca_work | ~pb |
| 18:42.07 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 18:43.23 | circut | system.conf: http://pastebin.com/e6mHrArG |
| 18:44.09 | circut | chan_dahdi.conf: http://pastebin.com/kQi3QXTf |
| 18:44.25 | [TK]D-Fender | [14:37]<circut>the asterisk console says it cannot create channel of type dahdi <- maybe you should show us the attempt so we can see the CAUSE of it |
| 18:45.24 | bmoraca_work | circut: let's see dahdi-channels.conf now |
| 18:46.13 | circut | dahdi-channels.conf: http://pastebin.com/95zqPmkJ |
| 18:46.49 | circut | [TK]D-Fender: when i do: Dial(DAHDI/3/${num}) the first call gets out, a second call the channel i assume is busy. i see this on the console |
| 18:46.52 | circut | [Mar 30 13:44:58] WARNING[15579]: app_dial.c:1518 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 18:47.07 | bmoraca_work | well, you don't have a group 1 configured, so of course dialing DAHDI/G1/${num} is going to fail |
| 18:47.22 | circut | boom |
| 18:47.26 | circut | where do i define a group at? |
| 18:47.29 | bmoraca_work | circut: both of your T1 spans are configured as group 0 |
| 18:47.39 | bmoraca_work | er...the "group" setting in dahdi-channels.conf |
| 18:47.56 | Katty | i'm not feeling so well all of a sudden |
| 18:48.26 | bmoraca_work | Katty: if you're going to puke, do it ------------------> that way |
| 18:48.39 | circut | i thought those were formated: 'group=start_chan,end_chan' |
| 18:48.53 | circut | my bad ;/ |
| 18:50.45 | bmoraca_work | it's group=<groupnum> |
| 18:50.46 | bmoraca_work | simple |
| 18:51.34 | [TK]D-Fender | [14:50]<bmoraca_work>it's group=<groupnum> <- group=<groupnum>,<anothergroupnum>,<anothergroupnum>,<anothergroupnum>, etc |
| 18:51.59 | bmoraca_work | yeah, yeah. why you'd need your channels in multiple groups is beyond me, though |
| 18:52.01 | bmoraca_work | but meh |
| 18:54.17 | *** join/#asterisk RobH (~robh@2620:0:860:2:21e:c2ff:fe03:2465) |
| 18:56.31 | Katty | 465=1~3~4yu |
| 18:57.17 | ChannelZ | Don't you mean :O~~~~~~~~~~ |
| 18:57.31 | Katty | t71~1u~~333333333333333333333333333333333333333333333334444444444444444444444444444444444rrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrdddddddddddddddddddddddddddddddddddddddddddddddddddddrrrrrrrrrrrrrrrrrrrsz22222dxww uil.,\~7845 |
| 18:57.49 | ChannelZ | suspects ferrets |
| 18:59.10 | thehar | lol |
| 18:59.13 | *** join/#asterisk rare1980_ (~rare1980@119.152.121.250) |
| 18:59.19 | smooth_penguin | odd key jumping ferret |
| 18:59.30 | ChannelZ | more than one |
| 19:00.31 | Katty | ,961WOAH |
| 19:00.33 | jaytee | arglebarglewhoosh |
| 19:00.33 | Katty | sorry |
| 19:00.44 | Katty | that would have been merry |
| 19:00.58 | circut | i just did rm -rf /etc/asterisk |
| 19:01.04 | circut | and now my config files are gone? |
| 19:01.18 | jaytee | congratulations!!! Tell him what he's won, Bob! |
| 19:01.24 | circut | :) |
| 19:01.31 | thehar | laff |
| 19:01.58 | jaytee | backup early, backup often but always look in your rearview mirror when backing up! |
| 19:01.59 | circut | anything to shift focusfrom my goofy ass dahdi question |
| 19:02.11 | flapjacks | lol |
| 19:02.32 | *** join/#asterisk chazzam (~chazz@173-24-238-25.client.mchsi.com) |
| 19:04.54 | suprstar | you were supposed to alias rm='tar -zcvf' ... |
| 19:11.06 | ManxPower-work | There are reasons to have channels in multiple groups |
| 19:11.46 | circut | suprstar: nice |
| 19:12.25 | ManxPower-work | for example, we want users to be able to use all 23 channels (g1) of a PRI, but when someone leaves a group voicemail the system will only will only use 12 channels (g2) for calling everyone's cell phone and telling them they have a voicemail (don't ask the users were nutcases) |
| 19:13.26 | *** join/#asterisk Lantizia (~user@212.183.140.20) |
| 19:14.16 | *** join/#asterisk Unseen (shani@2001:5c0:1501:2b00:20c:29ff:feed:beef) |
| 19:14.17 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 19:14.24 | Unseen | Hello everyone. |
| 19:14.44 | Docteh | lol |
| 19:14.49 | Unseen | can any one suggest me any cheap and good ip phones that support sip protocls ? |
| 19:15.03 | thehar | not grandstream |
| 19:15.09 | Docteh | ManxPower-work: i would be so tempted to leave a group voice mail of "omg spam" |
| 19:15.10 | Nugget | not cisco |
| 19:15.52 | ManxPower-work | Docteh, Having the system try to push 50 calls thru a 23 channel PRI was...interesting. |
| 19:17.12 | ChannelZ | Unseen: define 'cheap' |
| 19:17.29 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:17.42 | ManxPower-work | ~phones |
| 19:17.43 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
| 19:17.49 | ChannelZ | I have a dozen Linksys SPA922's which I like well enough and are < $100/ea |
| 19:18.21 | [TK]D-Fender | Unseen: Polycom IP 321 = $80 |
| 19:18.37 | Naikrovek | I second [TK]D-Fender |
| 19:18.44 | ChannelZ | Actually the 922's are ~100-110 but the 921's (not PoE) are $85-ish |
| 19:18.45 | Naikrovek | polycom ip321 |
| 19:19.12 | Docteh | whats wrong with cisco? the difficulty in getting support? |
| 19:19.18 | Naikrovek | expensive |
| 19:19.20 | Naikrovek | very expensive |
| 19:19.33 | Naikrovek | except for the SPAs which are actually linksys as I understand it |
| 19:20.00 | Docteh | used to be Sipura |
| 19:20.20 | [TK]D-Fender | Cisco= expensive, older ones are Cisco-standard PoE, not 802.3af, licensing issues, SIP firmware functionality & stability issues, etc |
| 19:20.29 | Naikrovek | ah yes licensing |
| 19:20.43 | Naikrovek | in addition to the phone, you must buy a $100 license to USE the phone |
| 19:20.53 | Naikrovek | cisco = prohibitively expensive |
| 19:21.26 | hardwire | ok so I have a call file that is calling a Local/... extension that dials a phone and sending it to an extension that immediately answers and executes Record |
| 19:21.47 | hardwire | however as soon as the calls are bridged the Dial() to the phone ends unexpectidly according to asterisk. |
| 19:22.07 | wcselby | the newest line of cisco spa phones are completely new and work well with asterisk |
| 19:22.07 | *** join/#asterisk Alagar (~Administr@122.164.33.94) |
| 19:22.14 | hardwire | which executes the h extension which should only be called when Dial is complete |
| 19:22.30 | hardwire | so I'm guessing there's a gotcha here that I don't quite understand |
| 19:22.36 | Naikrovek | wcselby: yes, but still more expensive than polycom for the same featureset |
| 19:22.37 | wcselby | the cisco spa 5xx series |
| 19:22.51 | [TK]D-Fender | wcselby: Not completely new... the chassis is still the Linksys variety... newer handset from what I hear, and I haevn't seen the web configs on them, but the linksys ones were straight Sipura Heritage |
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| 19:23.01 | wcselby | Naikrovek - probably, I haven't checked prices on the lwoer end ones |
| 19:23.18 | wcselby | the handset looks and feels more like a cisco 79xx phone |
| 19:23.30 | wcselby | the rest of the unit, except the 525, looks like the old sipura units |
| 19:23.43 | wcselby | the 525 looks like a fancy 7945 or so |
| 19:24.20 | wcselby | it's wifi enabled, and comes with a usb port for a flash drive full of mp3 you can use as ringtones. |
| 19:24.31 | wcselby | of course, i've heard some people say there are some pretty serious bugs with the 525 right now |
| 19:26.53 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 19:27.06 | wcselby | i dunno, i guess i just like the "feel" of cisco phones. I like everything else about the polycoms, so I dunno. the few aastra's I've used have been nice and they're actually easy to setup via the web interface |
| 19:28.46 | Naikrovek | never used cisco phones or aastra myself |
| 19:28.49 | Naikrovek | but i love my cisco phones |
| 19:28.57 | Naikrovek | about to order a heap more |
| 19:29.01 | wcselby | of course, you mean polycom |
| 19:29.02 | Naikrovek | aaak |
| 19:29.03 | Naikrovek | yes |
| 19:29.04 | wcselby | :P |
| 19:29.04 | Naikrovek | polycom |
| 19:29.12 | wcselby | what are you ordering? |
| 19:29.18 | Naikrovek | 335s |
| 19:29.22 | wcselby | i'd really love to have a 550 on my desk at home |
| 19:29.29 | wcselby | or a 650 |
| 19:29.32 | wcselby | i just don't want to pay fo rit |
| 19:29.38 | Naikrovek | yeah i'm ordering a 650 + sidecar for reception |
| 19:29.40 | [TK]D-Fender | IP 550 = waste |
| 19:29.53 | ManxPower-work | hugs his 550 |
| 19:29.55 | wcselby | I thought you said it was the 450 was a waste? |
| 19:30.07 | rare1980_ | any one who is expert on a2billing? i know this is asterisk chat room but still... if any one can help? |
| 19:30.39 | [TK]D-Fender | wcselby: No, the 450 is slightly questionable, the 550 is "way out" |
| 19:31.10 | [TK]D-Fender | wcselby: IP 335 is backlit+HD. This makes the 450 questionable. the 550 costs a lot more and adds litte for the difference from the 450. |
| 19:32.42 | wcselby | does trhe 450 look more like a 550 or a 335 (I say as I pull up google) |
| 19:32.54 | [TK]D-Fender | 450 is more like a 430 |
| 19:34.03 | wcselby | so the 450 is a 3 button phone, the 550 is a 4 button phone, and the 650 is a 6 button phone? |
| 19:34.28 | [TK]D-Fender | wcselby: Yes... |
| 19:36.06 | wcselby | so what's the 670? |
| 19:36.16 | [TK]D-Fender | wcselby: massive overkill |
| 19:36.19 | wcselby | lol |
| 19:36.21 | Naikrovek | color 650 as i understand it |
| 19:36.28 | [TK]D-Fender | Naikrovek: No, 670 = colour |
| 19:36.37 | [TK]D-Fender | Naikrovek: 650 = backligh greyscale |
| 19:36.39 | Naikrovek | 670 = colored 650 |
| 19:37.01 | Naikrovek | [TK]D-Fender: that's what i meant |
| 19:37.24 | wcselby | 670 has gige also I guess |
| 19:37.27 | [TK]D-Fender | Naikrovek: And Rciky Martin is straight. |
| 19:37.28 | rare1980_ | guys do u know why voipmonk is not coming online on mirc??? |
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| 19:37.44 | [TK]D-Fender | And by "striaght" I mean "so very comlpetely not straight" |
| 19:37.46 | wcselby | is voipmonk a user in here? |
| 19:37.46 | Naikrovek | rare1980_: don't know. hit his website and call him |
| 19:38.05 | [TK]D-Fender | Damn, typing desync today... |
| 19:38.09 | rare1980_ | i paid him for some services but he is not answring his fone |
| 19:38.18 | rare1980_ | i don't knw wht to do |
| 19:38.28 | Naikrovek | rare1980_: that dude is usually super responsible; suspect family emergency or something |
| 19:38.29 | [TK]D-Fender | rare1980_: Hire someone else |
| 19:39.08 | rare1980_ | [TK]D-Fender: i could hire someone else |
| 19:39.23 | rare1980_ | but i have already paid himm.. :( |
| 19:40.10 | rare1980_ | who else is expert on a2billing |
| 19:40.28 | ManxPower-work | rare1980_, voipmonk is a reliable person. |
| 19:40.35 | rare1980_ | humm ok |
| 19:40.59 | ManxPower-work | rare1980_, I bet some emergency happened. |
| 19:41.09 | Naikrovek | that's my feeling also |
| 19:41.12 | rare1980_ | may be |
| 19:41.27 | rare1980_ | thanks guys.. let me e-mail him |
| 19:41.27 | ManxPower-work | He's been around since like 2002. |
| 19:41.35 | rare1980_ | humm long time |
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| 19:52.05 | bmoraca_work | being unreliable is so much more fun than being reliable |
| 19:52.47 | p3nguin | Keeps people on their toes, right? |
| 19:53.06 | bmoraca_work | and people expect less from you |
| 19:54.17 | bmoraca_work | if i tell them "maybe" and "in a couple days" and then deliver more quickly 4 times out of 5, they love me for it and aren't too upset on the few times i can't get something done quickly enough |
| 19:57.16 | ChannelZ | It worked for Scotty |
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| 20:20.57 | doctorray | Is AMI the best way to have a script get a list of parked calls, or is there a magical other way? |
| 20:21.30 | bmoraca_work | doctorray: you pretty much have to keep track of that in real time with a 3rd party application |
| 20:22.42 | wcselby | i like the idea of using magic |
| 20:23.42 | doctorray | but, a 3rd party app that connects to the manager interface and parses the stream of data, right? |
| 20:24.31 | doctorray | i have a bit of experience in expect and think i could create something stable enough to pull a list |
| 20:25.16 | ChannelZ | you go girl! |
| 20:25.25 | doctorray | I just wonder if I could use a combination of devstate, or possibly inserting a call into the parking hints contect that updated a database, so that I don't have to go through ami |
| 20:29.32 | ChannelZ | I guess you could 'hotwire' it all yourself if you do it outside of the 'features' - Set something in AST DB or whatnot before parking a call, and make your own extensions that pick up those calls and remove it from the DB (rather than using 'parkedcalls') |
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| 20:32.08 | doctorray | hmm |
| 20:32.16 | raden_work | why is firefox so freakin slow ? |
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| 20:33.03 | wcselby | raden_work - 'cuz |
| 20:33.51 | superbeef | has anybody pulled off doing a resporg to xfer a mobile number to a VOIP provider? |
| 20:34.35 | doctorray | superbeef: resporg = tollfree? |
| 20:34.39 | superbeef | oh |
| 20:34.44 | superbeef | i guess not resporg then |
| 20:34.58 | doctorray | I've only done LNP from landlines to voip |
| 20:35.14 | doctorray | but it's supposed to work mobile to voip as well |
| 20:35.14 | bmoraca_work | doctorray: why reinvent the wheel? iSymphony and HUDlite both already monitor parked calls |
| 20:36.28 | bmoraca_work | superbeef: you should be able to LNP any number to any provider. FCC says so. |
| 20:36.42 | superbeef | bmoraca_work: Hmmmm... my brain is brewing |
| 20:36.43 | doctorray | bmoraca_work: looking to use microbrowser on polycom phones. iSymphony is clunky from my tests and HUDlite is a no-go on 1.6.x (or was last I checked) |
| 20:36.44 | bmoraca_work | some losing carriers are way worse to deal with than others, though |
| 20:37.22 | superbeef | I thinki talked work into giving me an android, so I'm thinking about retiring my iphone, but i want to keep my number alive |
| 20:38.23 | bmoraca_work | superbeef: if your android isn't on AT&T, then you'll need to tell your new cell carrier to port the number over. or is your new phone going to have a different number on purpose? |
| 20:38.53 | superbeef | bmoraca_work: its a work phone.... so i'll keep the work #, ATT is my personal phone |
| 20:39.04 | bmoraca_work | ahhh |
| 20:39.10 | doctorray | superbeef: I hope you're not thinking about using sipdroid unless you're on wifi all the time |
| 20:39.15 | bmoraca_work | you should be able to port that AT&T number anywhere |
| 20:39.58 | superbeef | doctorray: nope.. just wanting to send it to an asterisk box or something where I can do follow-me, etc |
| 20:42.39 | superbeef | superbeef: maybe tie it to an autotuner VST plugin |
| 20:42.57 | superbeef | wow i just talked to myself in IRC |
| 20:42.57 | doctorray | that would be.. awesome |
| 20:46.28 | dinesh___ | does asterisk log the console somewhere ? |
| 20:47.07 | superbeef | dinesh___: /var/log/full |
| 20:47.13 | superbeef | dinesh___: /var/log/asterisk/fll |
| 20:47.14 | superbeef | full |
| 20:47.28 | dinesh___ | okay i don't have that one, so I guess i didn't enable that |
| 20:47.52 | bakermd | dinesh___: /etc/asterisk/logging.conf |
| 20:48.01 | dinesh___ | ok thanks |
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| 20:55.33 | nbash | whats the command from the cli to enable and view sip debugging? |
| 20:56.29 | wcselby | sip set debug ip xxx.xxx.xxx.xxx or sip set debug peer PEERNAME |
| 20:56.38 | wcselby | and then sip set debug off |
| 20:57.37 | bmoraca_work | they're testing the fire alarm...wow that's annoying |
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| 21:20.31 | nbash | wcselby: Thanks |
| 21:20.36 | nbash | exit |
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| 21:34.47 | wcselby | does anyone know if queuemetrics will work with MS SQL? |
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| 21:49.20 | Slugs_ | guys what would explain a system appearing to be running normal and then getting alot of this error, diff extensions of course |
| 21:49.23 | Slugs_ | [Mar 30 17:00:26] NOTICE[3252] chan_sip.c: Call from 'desk-8589' to extension '8888759370' rejected because extension not found |
| 21:49.32 | Slugs_ | lots of em |
| 21:49.58 | [TK]D-Fender | Slugs_: well... do you accept that a call is placed matching that peer, dialing that number? |
| 21:50.51 | Slugs_ | yes |
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| 21:53.14 | hardwire | search and replace would be an added bonus today |
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| 21:54.24 | [TK]D-Fender | Slugs_: Wel then you'd better look at the context its pointed to because you have no extension to match that # |
| 21:57.39 | Slugs_ | ~pb |
| 21:57.40 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 21:58.14 | Slugs_ | [TK]D-Fender, http://pastebin.com/C5ifSpeY |
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| 21:59.26 | Slugs_ | Chainsaw, it's late for you, no? |
| 21:59.46 | Chainsaw | Slugs_: That is correct, it is 11pm local time. |
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| 22:04.32 | [TK]D-Fender | Slugs_: I also don't see your failed call with SIP DEBUG enabled... |
| 22:05.56 | Slugs_ | true one sec |
| 22:12.15 | Slugs_ | [TK]D-Fender, if i get hundreds of these, is that ok ? |
| 22:12.19 | Slugs_ | [Mar 30 18:00:03] WARNING[18152] pbx.c: Unable to register extension '8579', priority 1 in 'stations', already in use |
| 22:13.18 | [TK]D-Fender | Slugs_: that has nothing to do with this |
| 22:13.29 | [TK]D-Fender | Slugs_: show me the failed call |
| 22:14.10 | Slugs_ | when i turn sip debug on, its flys thru like a crazy monkey on cli, there are 500 endpoints all using sip |
| 22:14.46 | doctorray | can I have both the ability to one step park (#72) and pass DTMF pound sign to the other side, or is that asking too much? |
| 22:15.30 | Katty | herroes |
| 22:16.26 | bmoraca_work | doctorray: generally best to stay away from using # in telephony applications |
| 22:16.58 | [TK]D-Fender | Slugs_: PASTEBIN |
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| 22:53.30 | jsjc | I wonder where is the default directory for files such us tt-weasel monkeys and rest of prompts in asterisk |
| 22:54.22 | Katty | i believe it's /var/lib/asterisk/sounds/en |
| 22:54.48 | Katty | but you can simply locate tt-weasel |
| 22:59.28 | bmoraca_work | jsjc: all of your paths are in asterisk.conf |
| 22:59.37 | jsjc | ok thanks.. |
| 22:59.53 | jsjc | because cannot find either one or in asterisk.conf and tt-monkeys still plays so I wonder.. |
| 23:00.03 | jsjc | I will look harder |
| 23:00.48 | jsjc | oh in usr share |
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| 23:13.20 | carrar | Katy |
| 23:13.22 | carrar | Katty |
| 23:14.37 | carrar | ogenki desu ka |
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| 23:44.01 | Katty | hi |
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| 23:45.52 | Katty | jaytee: herroes. |
| 23:46.15 | captain2 | err ohh err ;) |
| 23:46.20 | jaytee | hi Katty |
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| 23:46.58 | captain2 | quick question... i tried googling but didn't get any "confident" answers... If my asterisk box is using the ulaw codec with all providers and devices, what's the best format to monitor calls with (for performance, not quality)? ulaw or wav? |
| 23:48.12 | wwalker | captain2: good question, I've meant to ask that myself before. |
| 23:48.17 | captain2 | currently i'm recording in wav, and i'm getting choppiness under wayyyyy less load than without recording... |
| 23:48.43 | Katty | what happens when you record in ulaw or gsm? |
| 23:48.52 | captain2 | haven't tried |
| 23:49.05 | captain2 | also... i don't have a very good testing methodology :( |
| 23:49.22 | captain2 | so i hoped i could get a quick answer without having to spend the time to discover it myself |
| 23:49.45 | Katty | well i have no other answer than by the process of elimination |
| 23:49.50 | Katty | but perhaps someone else here will |
| 23:50.50 | captain2 | woah woah... no need to get all Katty with me! :P |
| 23:50.59 | jaytee | if the both ends of the conversation are using ulaw then use ulaw for recording, that way you won't burden the CPU with transcoding |
| 23:51.12 | Katty | erm, i don't get katty with people |
| 23:51.28 | captain2 | Katty: i was kidding... just a little pun ;) |
| 23:51.38 | Katty | k |
| 23:51.43 | captain2 | jaytee: hmm... |
| 23:52.06 | jaytee | things that make ya go "hmmmmmm" |
| 23:52.17 | rene- | most carriers are now enforcing higher rates for toll free origination from canada, :S teliax started billing me for this just now, my bill just went a lot higher, did it bite somebody else? |
| 23:52.20 | Katty | happy meals make me go hmmmm |
| 23:52.42 | Katty | why are they called Happy Meals? |
| 23:53.07 | rene- | or does anybody know of something a good deal lower than 0.07/min TF from canada |
| 23:53.08 | jaytee | cuz it makes the CEO and the franchise owner very, very Happy |
| 23:53.28 | Katty | that sounds very likely |
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| 23:54.31 | Katty | hi generalhan |
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| 23:54.41 | generalhan | hi Katty |
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| 23:57.29 | captain2 | jaytee, Katty: haha wow what a throwback ;) |
| 23:57.39 | captain2 | i thought probably whatever format asterisk uses internally in its buffers would be the best bet... i'd imagine ulaw would be more difficult to do any kind of mixing, etc, so I thought maybe wav would be a safer bet... but then again, i don't know what i'm talking about, so maybe not :P |
| 23:58.09 | jaytee | I don't get it? |
| 23:58.36 | generalhan | ok, im having some issues with my globals context ... this was working in 1.4.18, but after upgrading to 1.4.26.1 on a new build it seems to work differently. i am trying to set a date variable as DATE=${STRFTIME(${EPOCH}||%m%d%Y)}... that used to update everytime i called it to the actual date. now it seems to be just tossing in the DATE from the last day i reloaded extensions.conf. |
| 23:59.06 | zonyl | Hi All. I just upgraded to 1.6 and I have a dialplan issue: This is failing to work now in incoming: exten => s/5559627224,n,Set(CALLERID(name)="Frank") Anyone have any ideas or a better way to debug my dialplan? (Is there a way to turn on more info to console about what is going on) |
| 23:59.07 | generalhan | is that behaving like it should, and i was just lucky that it was working the way i wanted it before ? or is ther another way that i can do this now ? |