| 00:00.10 | jblack | The eternal question. I think console administration is superior. |
| 00:00.12 | bcrisp | well its an easier install |
| 00:00.35 | bcrisp | i think the idea is to not allow the OS / settings be an issue so they package it all together |
| 00:00.43 | etfonhomey | I'm not sure what the target audience for AsteriskNOW is. |
| 00:00.49 | Davedan | bcrisp: on ubuntu it's 'apt-get install asterisk' that's it |
| 00:01.17 | bcrisp | Davedan: i mean... asterisknow has a package containing the OS, gui, and asterisk all in one |
| 00:01.19 | bcrisp | for quick startup |
| 00:01.21 | Davedan | etfonhomey: probably people that don't want asterisk to work :) |
| 00:01.40 | bcrisp | i, for one, am a new linux user |
| 00:01.40 | etfonhomey | Davedan, LOL! I believe it's supposed to make it easier! |
| 00:01.53 | etfonhomey | Davedan, did you turn off iptables? |
| 00:02.12 | Davedan | I didn't trun iptables off. just did a restart |
| 00:02.31 | Davedan | I'm a windows users for many years and that's the only thing I learned from it |
| 00:03.36 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 00:04.30 | Davedan | is it possible to set a voip echo test in the web admin? |
| 00:05.24 | etfonhomey | If you can edit the extensions.conf directly from the GUI, then yes. :) |
| 00:05.57 | Davedan | ok. this gui is useless |
| 00:06.05 | etfonhomey | I have no clue how the GUI works. GUI's scare me. When one click can change multiple lines in multiple config files, I don't like that loss of control. |
| 00:06.30 | Davedan | is there a way to create a dummy call to a device (not using the gui) |
| 00:08.01 | etfonhomey | Fire up X-Lite on another machine and configure a second line. |
| 00:08.02 | bcrisp | davedan just set up another device |
| 00:08.14 | bcrisp | call yourself on xlite :) |
| 00:09.26 | Davedan | It's embrassing to call myself |
| 00:09.46 | Davedan | I hope nobody is looking |
| 00:10.19 | etfonhomey | If FWD was still free, you could have setup their test number on there. |
| 00:10.49 | bcrisp | Davedan.. u might try zoiper .. i like it better than x-lite |
| 00:10.58 | jblack | ipkall is still free |
| 00:11.25 | Kobaz | i like twinkle even better |
| 00:11.28 | Kobaz | but it doesnt do iax |
| 00:11.45 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 00:11.57 | Davedan | is Trixbox the same as asterisknow? |
| 00:12.14 | Kobaz | what do you think? |
| 00:12.21 | bcrisp | davedan.. what are you looking for ? |
| 00:12.23 | jblack | If asterisknow is codeine, trixbox is cocaine. |
| 00:12.23 | Kobaz | is pepsi the same thing as coke |
| 00:12.29 | bcrisp | i mean what are your administrative needs? |
| 00:12.38 | bcrisp | Kobaz, i prefer kiet doke |
| 00:12.44 | Kobaz | heh |
| 00:12.47 | Kobaz | i don't drink soda at all |
| 00:12.54 | bcrisp | i like espresso |
| 00:12.55 | Kobaz | haven't in 8+ years |
| 00:12.57 | bcrisp | with a drop of cream |
| 00:13.05 | Kobaz | soda is really bad for you |
| 00:13.11 | ChannelZ | so is breathing |
| 00:13.22 | bcrisp | every breath you take you are oxidizing away |
| 00:13.33 | Kobaz | there's been several research studies recently |
| 00:13.42 | Kobaz | it's conclusive, and unanamous |
| 00:13.48 | bcrisp | lots of sugar and lots of caffeine .. its kind of a no-brainer |
| 00:13.48 | Kobaz | life is the leading cause of death |
| 00:13.52 | etfonhomey | Davedan, run away from Trixbox! |
| 00:14.05 | ChannelZ | Life is a sexually transmitted disease with a 100% mortality rate |
| 00:14.06 | bcrisp | news alert: Scientists shown to cause cancer in laboratory rats |
| 00:14.13 | Davedan | bcrisp: I need to experience with a java/c++ clien. I just want the simples thing |
| 00:14.20 | Davedan | ok. I'll stick with CLI and conf files |
| 00:14.42 | bcrisp | davedan, you could communicate with * via the management interface |
| 00:14.42 | *** join/#asterisk rdahlin_1 (n=rdahlin_@78-73-17-198-no168.tbcn.telia.com) |
| 00:15.21 | Davedan | bcrisp: by management interface you mean the command line? |
| 00:15.33 | ChannelZ | no much worse |
| 00:15.48 | Davedan | ChannelZ:? |
| 00:15.50 | bcrisp | the AMI |
| 00:16.06 | Davedan | bcrisp: what's the difference between AMI and CLI? |
| 00:16.07 | bcrisp | socket based communication with the asterisk server .. u can trigger actions, read events etc |
| 00:16.30 | bcrisp | i.e. you could write a C# client to remotely administer or monitor the server |
| 00:16.41 | bcrisp | or java yatta yatta |
| 00:16.56 | ChannelZ | or you could poke a sharp stick in your eye |
| 00:16.59 | Davedan | this is behind my needs. |
| 00:17.00 | bcrisp | indeed |
| 00:17.26 | bcrisp | for instance, i wrote a simple app to gather information about call queues for realtime stats |
| 00:18.32 | *** join/#asterisk tzafrir__laptop (n=tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 00:19.57 | bcrisp | Davedan: did you read this? |
| 00:19.59 | bcrisp | ~book |
| 00:20.00 | 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 |
| 00:20.49 | Davedan | bcrisp: I've read half of it and made good progress. then I thought that the web gui might save me from learning all of asterisk but I guess I was wrong |
| 00:20.55 | Davedan | I'll go back and read this book again |
| 00:21.07 | bcrisp | it talks about the mgmt interface there also |
| 00:21.24 | Davedan | mgmt? |
| 00:22.25 | bcrisp | management |
| 00:22.35 | bcrisp | asterisk management interface (AMI) |
| 00:23.30 | Davedan | k |
| 00:23.47 | Davedan | I think I'll use the CLI and not the AMI. I don't have time to write a GUI myself |
| 00:23.49 | bcrisp | davedan what is your endgoal? |
| 00:24.06 | bcrisp | hmm |
| 00:24.08 | Katty | hi |
| 00:24.17 | bcrisp | CLI isnt comparable to AMI |
| 00:24.37 | bcrisp | CLI = command line interface |
| 00:24.50 | bcrisp | hi katty, neck better? |
| 00:25.00 | Katty | yes, thank you (= |
| 00:25.03 | Davedan | bcrisp: my end goal is to setup asterisk for development of a sip client |
| 00:25.18 | bcrisp | davedan.. oh ok, ya then you dont really nead a gui i would imagine |
| 00:25.25 | bcrisp | just need to learn the sip protocol :) |
| 00:25.46 | Davedan | actually I need to write a gui for the client |
| 00:25.55 | Davedan | the sip code is already written |
| 00:26.02 | bcrisp | well ya.. i meant * gui ... (like monitoring call centers yatta yatta) |
| 00:26.20 | Davedan | yes |
| 00:26.40 | bcrisp | i wouldnt mind writing a softphone |
| 00:27.26 | Davedan | nobody is stopping you :) |
| 00:28.17 | etfonhomey | bcrisp, just do a sip debug in the CLI and stare at it all day. :) |
| 00:30.51 | bcrisp | nah time consuming :) |
| 00:30.56 | Davedan | what is sip debug? |
| 00:31.19 | bcrisp | i had to write code to manually encode tcp/ip packets that was so fuun |
| 00:31.42 | voipmonk | its a command you enter at the asterisk command line interface to show you information about sip conversations occuring on your system in realtime |
| 00:31.45 | bcrisp | davedan: it tells * to output debug messages from the sip module |
| 00:31.46 | etfonhomey | Davedan, do a sip set debug at the CLI. |
| 00:32.07 | bcrisp | i.e. CLI: sip set debug on |
| 00:32.08 | etfonhomey | You see all the messages going back and forth between * and the sip peers. |
| 00:32.52 | Davedan | cool |
| 00:33.12 | Davedan | is it better then wireshark for debuging sip? |
| 00:33.12 | bcrisp | davedan, you can also go into logger.conf |
| 00:33.21 | bcrisp | and change what shows up on console |
| 00:34.09 | bcrisp | u can also start asterisk with debug / verbose options: asterisk -rvvvd |
| 00:34.46 | bcrisp | im such a newb |
| 00:35.49 | bcrisp | ~roulette |
| 00:35.50 | infobot | ACTION watches bcrisp pull the trigger: Click! |
| 00:36.32 | voipmonk | you're not doing that bad, bcrisp |
| 00:36.55 | voipmonk | I've seen much worse |
| 00:36.58 | bcrisp | still can't resolve the queue priority issue |
| 00:37.17 | bcrisp | it just sits and retries all day on a member that doesnt answer |
| 00:37.22 | bcrisp | id like it to then pass on to the next priority |
| 00:37.51 | bcrisp | member => SIP/userwhowontanswer,1 |
| 00:38.04 | bcrisp | member => Local/blablabla@context/n,2 |
| 00:38.49 | ChannelZ | isn't that what leavewhenempty is for? delete the agent from the queue if they don't answer their phone |
| 00:39.28 | bcrisp | hmm, well i dont want to remove them if they fail to answer once.. id just like it to try another queue member |
| 00:39.40 | bcrisp | (they are the same person) first entry is their desk, 2nd is their cell |
| 00:39.53 | ChannelZ | round-robbin |
| 00:40.05 | bcrisp | ok |
| 00:40.08 | bcrisp | ill try that |
| 00:41.30 | bcrisp | imm i dont see that as an option in 1.6.1.11 |
| 00:41.33 | bcrisp | queues.conf.sample |
| 00:42.34 | ChannelZ | hmm looks like they changed it - it's called 'linear' now it looks like |
| 00:42.51 | ChannelZ | (sort of anyway) |
| 00:43.32 | bcrisp | i guess that will work but its not really what i want |
| 00:43.44 | bcrisp | if i have 5 members in priority 1 i want it to ring all of them |
| 00:43.49 | bcrisp | if nobody answers, move to priority 2 |
| 00:43.52 | bcrisp | but it doesnt do that |
| 00:44.21 | bcrisp | it will only ring priority 2 if they physically reject the call in priority 1 or disconnect |
| 00:49.47 | *** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
| 00:53.32 | *** join/#asterisk Eataix (n=Eataix@124-168-216-217.dyn.iinet.net.au) |
| 00:53.47 | *** part/#asterisk Eataix (n=Eataix@124-168-216-217.dyn.iinet.net.au) |
| 00:55.28 | bcrisp | because im good enough, im smart enough, and gosh darn it, people like me |
| 00:57.14 | dlynes | bcrisp: now just wait a cotton pickin' minute... |
| 00:57.25 | dlynes | bcrisp: where's your sources? |
| 00:57.50 | bcrisp | haha |
| 00:57.57 | bcrisp | hiya |
| 00:58.03 | dlynes | hey :) |
| 00:58.35 | dlynes | bcrisp: so are you an expert in queues yet? |
| 00:58.40 | bcrisp | getting there |
| 00:58.47 | bcrisp | still have issues hehe |
| 00:59.00 | dlynes | yeah...same here |
| 00:59.11 | dlynes | but i really haven't spent much time on solving my issues, either |
| 01:00.09 | bcrisp | i have Local/ instances set up in higher penalty |
| 01:00.22 | bcrisp | but if a user is available in priority 1 it keeps retrying them |
| 01:00.27 | bcrisp | rather than moving on to the next priority |
| 01:00.40 | dlynes | bcrisp: well, of course |
| 01:00.45 | dlynes | bcrisp: they're available |
| 01:00.54 | dlynes | bcrisp: so they should answer the damned phone! |
| 01:01.13 | dlynes | bcrisp: asterisk can't fire your employees |
| 01:01.35 | dlynes | bcrisp: it can only treat your staff the way you tell it to treat them |
| 01:01.38 | bcrisp | well i want the priority to have one chance to answer |
| 01:01.47 | bcrisp | otherwise members of the 2nd priority have the opportunity |
| 01:02.45 | dlynes | bcrisp: are you using the 'timeout=' option? |
| 01:02.53 | dlynes | bcrisp: and the 'retry=' option? |
| 01:03.03 | bcrisp | retry = sets the time before retry |
| 01:03.08 | bcrisp | correct? |
| 01:03.10 | dlynes | bcrisp: i know |
| 01:03.14 | bcrisp | yes timeout = 10, retry = 5 |
| 01:03.19 | dlynes | hrm |
| 01:03.32 | dlynes | and it's still acting that way? i.e. never trying the second member of the queue? |
| 01:03.45 | bcrisp | no, it will try all available members repeatedly |
| 01:03.48 | bcrisp | rather than shifting priority |
| 01:04.00 | bcrisp | UNLESS the call is physically rejected from the available member in priority 1 |
| 01:04.02 | *** part/#asterisk Davedan (n=me@CBL217-132-75-171.bb.netvision.net.il) |
| 01:04.04 | dlynes | bcrisp: you mean putting the idiots that don't answer into a different priority? |
| 01:04.29 | bcrisp | well really its 3 people in priority 1 - desk soft phones |
| 01:04.35 | bcrisp | if they are on the road they arent at their desk |
| 01:04.40 | bcrisp | and id like it to then try cell phones |
| 01:04.43 | bcrisp | (priority 2) |
| 01:05.04 | bcrisp | and they absolutely will forget to logout of their softphones :) |
| 01:05.20 | dlynes | bcrisp: why not implement the queue member as a local channel, then? |
| 01:05.21 | ChannelZ | Why let the phones at their desk ring a bunch if they are on the road? Remove them from the queue |
| 01:05.23 | bcrisp | so ideally, anyone at their desk should be the first responder |
| 01:06.03 | bcrisp | can i pastebin it for clarity? |
| 01:06.05 | dlynes | bcrisp: then you have full flexibility on how to handle it |
| 01:06.15 | dlynes | bcrisp: go ahead, if you think it'll help |
| 01:06.28 | dlynes | bcrisp: keep in mind, you're probably more advanced on queues than I am |
| 01:06.36 | dlynes | bcrisp: considering you've spent more time on it than i have |
| 01:06.54 | dlynes | bcrisp: I've been using asterisk for about 4 or 5 years now, but just started looking at queues |
| 01:07.58 | bcrisp | http://pastebin.ca/1712724 |
| 01:08.45 | bcrisp | first 2 entries are the desk softphones |
| 01:09.15 | bcrisp | they should have first chance to answer, if neither answers, id like it to move to priority 2 |
| 01:12.09 | ChannelZ | Why don't you use the 'cascading queues' setup |
| 01:12.47 | ChannelZ | Put all the desk phones in one queue, and the mobiles in another. Send callers to the first with a timeout on the queue.. then send them to the second |
| 01:14.16 | bcrisp | didnt know i could do that |
| 01:14.58 | ChannelZ | well that might not actually do what you want... because the timeout is based on the age of the call |
| 01:15.11 | bcrisp | ya.. i dont mind leaving them in the queue if the queue members are busy |
| 01:15.19 | bcrisp | but if they are not answering i need it to move along |
| 01:15.36 | bcrisp | i.e. ben and john are busy taking queue calls in priority 1 |
| 01:15.44 | bcrisp | i dont want their cell # to start ringing too hehe |
| 01:15.48 | dlynes | bcrisp: also, do you want to try the queue members one by one? |
| 01:15.59 | ChannelZ | well it really seems like your agents should be logging in and out from where they are available instead of spending all this time trying to call people who aren't answering |
| 01:16.04 | bcrisp | dlynes: not really, i could use linear strategy right? |
| 01:16.05 | dlynes | bcrisp: or do you want to try both desk phones first, and then try both cell phones? |
| 01:16.21 | bcrisp | dlynes, only if they arent answering |
| 01:16.32 | bcrisp | if all agents are busy i dont want to call the cell #s |
| 01:16.39 | bcrisp | i guess logging them out is the best choice |
| 01:16.54 | dlynes | bcrisp: ah....so you do have call-limit=1 on both sip peers? |
| 01:17.02 | bcrisp | yep |
| 01:17.23 | dlynes | bcrisp: and do you have limitonpeer=yes in your sip general section? |
| 01:17.24 | Katty | mmmmmmmmmmmmmm |
| 01:17.28 | Katty | sugar cookie candle |
| 01:17.30 | Katty | omnomnomnomnom |
| 01:17.38 | bcrisp | limitonpeer let me see |
| 01:17.41 | dlynes | bcrisp: and are you using sip type=friend? |
| 01:18.17 | dlynes | bcrisp: unless your answer to all of those is yes, it won't know if all your sip phones are busy, or not |
| 01:18.55 | drmessano | bcrisp: I fail to see what this has to do with Windows |
| 01:19.54 | bcrisp | huh? |
| 01:20.12 | dlynes | drmessano: he never mentioned windows |
| 01:20.13 | bcrisp | dlynes: sip type=friend yes |
| 01:20.29 | bcrisp | dont see limitonpeer setting |
| 01:20.44 | dlynes | bcrisp: you'll want to set it, then |
| 01:20.58 | dlynes | bcrisp: otherwise call-limit=1 won't do anything |
| 01:21.43 | bcrisp | its not in the sip.conf.sample for 1.6.1.11 |
| 01:22.11 | bcrisp | maybe deprecated? |
| 01:22.22 | bcrisp | the call-limit=1 does work |
| 01:22.55 | bcrisp | for an interface name |
| 01:22.58 | bcrisp | sip/blabla |
| 01:23.14 | bcrisp | but it has no idea that sip/blabla and local/blabla is gonna ring the same person :/ |
| 01:23.39 | bcrisp | fk it , ill just tell them to logout of their softphones |
| 01:23.47 | bcrisp | or configure agent callback |
| 01:24.51 | dlynes | bcrisp: maybe that issue's fixed in the 1.6.1 series then |
| 01:24.57 | dlynes | bcrisp: i remember it was still an issue in 1.4 |
| 01:25.08 | dlynes | bcrisp: and 1.2, for that matter |
| 01:25.15 | dlynes | bcrisp: especially if you were wanting to do blf |
| 01:25.26 | bcrisp | dlynes: ya i guess the queue behavior that im looking for is weird |
| 01:25.58 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:26.41 | bcrisp | the other problem is that if the queue members are busy i dont want it to call the damn cell phones haha |
| 01:27.24 | bcrisp | i guess the answer that makes the most sense is login from the device ur currently using |
| 01:27.35 | ChannelZ | so make a 'hotkey' extension they can call when they are at their desk or leaving their desk and CHANGE THE AGENTS IN THE QUEUE ACCORDINGLY! |
| 01:27.59 | bcrisp | yeah |
| 01:28.06 | ChannelZ | You're trying to make this as complex as possible just so a couple of people can be as lazy as possible |
| 01:29.18 | ChannelZ | shit, just have them forward their extension to their cell phone when they want to leave |
| 01:30.08 | dlynes | bcrisp: like i said...implement it all as local channels, so that you can perform the complex behavior inside the local channels |
| 01:30.39 | dlynes | bcrisp: i.e. one local channel for both sip devices, and another local channel for the cell phones |
| 01:31.42 | dlynes | bcrisp: erm actually |
| 01:31.46 | dlynes | bcrisp: that won't work, either |
| 01:31.48 | dlynes | bcrisp: also |
| 01:31.58 | dlynes | bcrisp: what kind of queue strategy are you using? |
| 01:32.42 | dlynes | bcrisp: is it ring all, or round robin, or round robin with a memory? |
| 01:35.07 | bcrisp | ring all |
| 01:35.55 | *** part/#asterisk etfonhomey (n=etfonhom@74-131-159-160.dhcp.insightbb.com) |
| 01:36.04 | bcrisp | ChannelZ: the happier they are, the more $$ i make |
| 01:36.12 | bcrisp | :D |
| 01:36.43 | bcrisp | i like the idea of a hotkey |
| 01:37.10 | dlynes | bcrisp: how about no hotkey, and just letting them have their way? |
| 01:37.28 | bcrisp | ya dlynes :) |
| 01:39.07 | bcrisp | i originally thought i could use the membermacro |
| 01:39.13 | bcrisp | but that only fires once it is connected |
| 01:39.41 | dlynes | bcrisp: In a local channel, try: exten => s,1,Dial(SIP/blahblah&SIP/blahblah2,${timeout}) ; exten => s,n,Goto(s-${DIALSTATUS},1) ; blahblahblah ; exten => s-BUSY,1,Dial(Dahdi/1/cellphone1&Dahdi/1/cellphone2) ; .... ... |
| 01:40.11 | dlynes | bcrisp: erm actually....s-NOANSWER, not s-BUSY |
| 01:40.22 | bcrisp | ahhh i see |
| 01:40.31 | dlynes | bcrisp: in s-BUSY, you'd do exten => s-BUSY,1,Busy |
| 01:40.32 | bcrisp | so a queue member points to a local channel with an s extension |
| 01:40.50 | dlynes | bcrisp: No...to whatever you decide to point to in that local channel |
| 01:41.00 | bcrisp | i guess channels kind of confuse me |
| 01:41.05 | dlynes | bcrisp: I just chose 's' for lack of a better value to put there |
| 01:41.27 | dlynes | bcrisp: asterisk is pretty much limited to your imagination |
| 01:41.39 | bcrisp | ya.. just understanding the pieces is a little tricky at first |
| 01:41.42 | bcrisp | i could use something like |
| 01:41.43 | dlynes | bcrisp: your users can have everything as simple as they want |
| 01:41.57 | dlynes | bcrisp: but generally the more simple they want it, sometimes it makes your life more complicated |
| 01:42.08 | dlynes | bcrisp: but that doesn't mean it can't be done |
| 01:42.14 | dlynes | bcrisp: you just have to think creatively |
| 01:42.15 | bcrisp | could you give me an example of the local channel thing? |
| 01:42.36 | dlynes | bcrisp: [mynewdialplancontext] exten => s,1,Noop(This is where I start) |
| 01:42.52 | dlynes | bcrisp: Local/s@mynewdialplancontext |
| 01:42.57 | bcrisp | oooooh |
| 01:43.07 | bcrisp | nice |
| 01:43.19 | dlynes | bcrisp: I think that's how it goes, anyways |
| 01:43.19 | bcrisp | that way i dont need priorities in the queue at all |
| 01:43.33 | dlynes | bcrisp: It's been a while since I've had to do any Local channel code |
| 01:43.56 | bcrisp | i like that |
| 01:44.11 | dlynes | bcrisp: yeah...that's how you do it |
| 01:44.13 | dlynes | bcrisp: http://www.voip-info.org/wiki/view/Asterisk+Local+channels |
| 01:45.14 | bcrisp | that is beautiful |
| 01:45.37 | dlynes | bcrisp: asterisk is like the swiss army knife of phone systems and not to mention automation |
| 01:45.53 | dlynes | bcrisp: you can even use it to completely automate your home security system, if you wanted to |
| 01:45.56 | bcrisp | ya im trying to learn bit by bit and studying all the conf file samples |
| 01:46.26 | dlynes | bcrisp: some of it you can only learn by playing |
| 01:46.31 | bcrisp | ya |
| 01:46.34 | heliosj | Asterisk's limitation tends to be the imagiantion of the person using it. |
| 01:46.44 | bcrisp | are you saying im not imaginitive? |
| 01:46.48 | dlynes | you just need to be able to bust yourself out of the box |
| 01:46.50 | heliosj | No? |
| 01:47.17 | dlynes | i.e. think outside the box |
| 01:47.20 | bcrisp | lol |
| 01:47.25 | dlynes | Asterisk isn't a phone system |
| 01:47.27 | dlynes | It's a scripting platform |
| 01:47.42 | heliosj | Toolkit. |
| 01:47.46 | bcrisp | i understand.. its just a matter of understanding the bits, then arranging them in new ways |
| 01:47.47 | bcrisp | i get that |
| 01:47.50 | bcrisp | (im learning the bits) |
| 01:48.07 | bcrisp | off to play with local channels thx for the info guys |
| 01:48.23 | dlynes | bcrisp: I'm currently working on extending the ami library someone else wrote for php, too |
| 01:48.36 | bcrisp | dlynes: im doing the same for c# |
| 01:48.48 | dlynes | bcrisp: i found it quite limiting for what I wanted to do, so I fixed the bugs, and now I'm adding more functionality to it |
| 01:49.05 | dlynes | bcrisp: when I'm finished it, I'll probably post it to my website |
| 01:49.27 | bcrisp | im writing an AMI events parser as a sep component |
| 01:49.37 | bcrisp | because of the weird way that some responses are sent as events etc from AMI |
| 01:52.32 | dlynes | bcrisp: yeah...that's just as a separate function in the php class |
| 01:52.42 | dlynes | bcrisp: I'm adding in additional functions for the other ami events and queries |
| 01:52.51 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 01:53.03 | dlynes | bcrisp: the original author pretty much only had logon, logoff, get key, and put key |
| 01:53.08 | dlynes | bcrisp: and nothing else |
| 01:53.11 | bcrisp | yikes |
| 01:53.36 | bcrisp | Queues() is a fun one:) |
| 01:53.43 | bcrisp | output is garbage |
| 01:55.06 | *** join/#asterisk etfonhomey (n=etfonhom@74-143-192-74.static.insightbb.com) |
| 01:59.31 | dlynes | bcrisp: how so? |
| 02:01.01 | bcrisp | i mena the Action: Queues |
| 02:01.03 | bcrisp | mean |
| 02:01.14 | Katty | hi |
| 02:01.29 | dlynes | bcrisp: what kind of output does it spit out? |
| 02:01.40 | dlynes | Katty: Hey katty...how's squirreldom? |
| 02:01.40 | bcrisp | human readable |
| 02:01.50 | dlynes | bcrisp: thought you said it was garbage? |
| 02:02.02 | bcrisp | i mean .. its not easy to parse |
| 02:02.09 | Katty | i'm sure they're all asleep, dlynes |
| 02:02.09 | dlynes | bcrisp: why not? |
| 02:02.38 | dlynes | bcrisp: anything's parsable |
| 02:03.07 | Katty | yawns |
| 02:03.19 | dlynes | bcrisp: I've even written a parsing engine for the electronic filings on Edgar, and those you could describe as being unparsable, I suppose |
| 02:03.21 | bcrisp | dlynes: its not a huge deal |
| 02:03.34 | bcrisp | dlynes: just not what i would expect |
| 02:03.35 | Katty | you know what IS a huge deal |
| 02:03.51 | dlynes | Katty: christmas? |
| 02:03.56 | Katty | no |
| 02:03.59 | dlynes | oh |
| 02:03.59 | Katty | that's not a big deal |
| 02:04.05 | dlynes | It's a huge deal |
| 02:04.13 | Katty | that's just an excuse for commercial places to make money |
| 02:04.15 | dlynes | It's the most important birthday of the year |
| 02:05.17 | Katty | if only it was warmer |
| 02:05.36 | dlynes | If it's warm enough for squirrels, it's warm enough for humans |
| 02:05.48 | dlynes | they've got less fat to keep them warm |
| 02:06.45 | Katty | it's not nearly warm enough for me |
| 02:06.54 | Katty | 60F would start feeling nice |
| 02:07.06 | Katty | 75F would be ideal |
| 02:07.25 | bcrisp | uh oh |
| 02:07.36 | bcrisp | Registration for '17772899142@callcentric.com' timed out, trying again.... |
| 02:07.53 | Katty | if you can /feel/ the temperature around you, it's not the right temperature |
| 02:08.29 | dlynes | Katty: 75F, you could feel the burning around you...is that what you mean? |
| 02:08.34 | bcrisp | it was in the 100s in october here |
| 02:08.40 | Katty | 75F is just perfect |
| 02:08.51 | dlynes | bcrisp: ewww |
| 02:08.59 | dlynes | bcrisp: that would suck |
| 02:09.14 | dlynes | bcrisp: any temperature that's so hot you need aircon is too hot for me |
| 02:09.15 | bcrisp | i remember driving in rush hour traffic when it was 118 |
| 02:10.27 | dlynes | Yeah...well...you're in the Philippines...way too hot for this guy |
| 02:10.56 | dlynes | it'd be like northern australia...pretty damned hot there, too |
| 02:11.11 | bcrisp | in in phoenix |
| 02:11.57 | bcrisp | where'd u get the philippines from? |
| 02:13.00 | voipmonk | lol |
| 02:13.33 | Katty | i figured voipmonk would show up after that comment |
| 02:13.44 | voipmonk | goes back to his rock |
| 02:13.47 | Katty | :< |
| 02:14.02 | Katty | deposits hot cocoa near rock |
| 02:14.09 | Katty | :> |
| 02:14.24 | dlynes | bcrisp: ph.ph.cox.net? |
| 02:14.28 | Katty | :< |
| 02:14.29 | voipmonk | mmmm... ( i actually mix it with espresso ) |
| 02:14.35 | Katty | :>>> |
| 02:14.44 | Katty | brb |
| 02:16.08 | Katty | returns |
| 02:16.34 | bcrisp | dlynes im getting register timeouts with my outgoing sip provider |
| 02:16.51 | bcrisp | the local channel thing appears to be working until it dials out |
| 02:17.00 | bcrisp | then it immediately comes back with failed to answer in 10 sec |
| 02:17.02 | bcrisp | (timeout) |
| 02:17.54 | Katty | the universe is involved in some conpsiracy against us |
| 02:18.04 | Katty | winter is all a big Joke |
| 02:24.04 | bcrisp | dlynes: here is what i get from CLI http://www.pastebin.ca/1712780 |
| 02:25.42 | bcrisp | dlynes: here is the context i set up, with member => Local/353@queueforwarding/n : http://www.pastebin.ca/1712782 |
| 02:26.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:37.12 | b14ck | Hey, in sip.conf, in regards to hash table sizes. The documentation says: "for maximum efficiency, adjust the following valuesw to be slightly larger than the maximum number of in-memory objects (devices)". |
| 02:37.19 | jblack | dlynes: that dcc is based on your concept |
| 02:37.28 | b14ck | What if the asterisk install has no endpoints, and only has incoming calls going to an IVR or something? |
| 02:37.40 | b14ck | Would you reduce those hash variables to 1? |
| 02:37.45 | jblack | b14ck: Sounds like a typical customer service setup to me. :) |
| 02:37.59 | b14ck | what? |
| 02:38.00 | b14ck | =p |
| 02:40.53 | b14ck | http://pastie.org/740965 <-- snippet from sip.conf |
| 02:41.27 | b14ck | basically, my PBX setup has no endpoints, all calls that come in or go out are going straight through a SIP trunk. should these settings be set down lower to 1? |
| 02:44.18 | eppigy | Katty: hi |
| 02:51.05 | drfreeze | Anyone know how * chooses which moh to play? |
| 02:51.19 | drfreeze | Is it a random based on the machine? |
| 02:51.57 | ChannelZ | there are a couple of schemas for that in musiconhold.conf |
| 02:52.16 | *** join/#asterisk simplydrew (n=simplydr@ool-44c2ab91.dyn.optonline.net) |
| 02:53.09 | ChannelZ | the files are played 'in normal sorting order' of the directory, and then there is a 'random' setting |
| 02:53.19 | *** part/#asterisk etfonhomey (n=etfonhom@74-143-192-74.static.insightbb.com) |
| 02:54.17 | ChannelZ | 1.6 actually has a 'sort' option |
| 02:56.57 | Katty | eppigy: hello sir |
| 03:01.15 | Katty | eppigy: i got a lovely candle at lowes earlier. it's called sweet icing sugar cookies |
| 03:05.34 | x86 | can someone send me a test fax to 425-998-1930? |
| 03:06.03 | eppigy | Katty: sounds nice :> |
| 03:09.23 | drfreeze | I have installed the same version of * on 3 different servers, and all default to a different song |
| 03:09.55 | drfreeze | Ok, We are having Polycom provisioning problems in a remote office connected by a T1 |
| 03:10.13 | drfreeze | The same type of phones in the same office as the asterisk server provisioned just fine |
| 03:10.30 | drfreeze | The other office phones (all but one) give the error Error: file does not contain a compatible image |
| 03:12.27 | *** join/#asterisk gbr_ (n=gbr_@189.90.51.240) |
| 03:13.55 | voipmonk | back |
| 03:14.07 | voipmonk | with munchings and crunchings |
| 03:18.40 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 03:20.21 | Katty | riddick just attacked sammy |
| 03:20.38 | drfreeze | Ok, looks like I need to back off one release |
| 03:21.15 | drfreeze | Anyone have suggestions on how to load different sip.ld files for different phoen models? |
| 03:21.40 | *** join/#asterisk coppice (n=chatzill@50.131.92.116.dyn.pacific.net.hk) |
| 03:22.45 | Katty | i think sammy is okay, mostly in shock |
| 03:23.18 | Katty | but he's not moving much |
| 03:23.22 | Katty | won't eat a treat |
| 03:25.49 | x86 | drfreeze: put the version you want for XXX phones, reboot all XXX phones, after they flash and boot up proper, rm sip.ld... repeat process for every YYY ZZZ etc etc phone type you have |
| 03:25.58 | x86 | PITA, but that's how it goes |
| 03:26.10 | x86 | then new XXX or YYY phones you will have to manually flash |
| 03:30.35 | *** join/#asterisk ticoit (n=ticoit@201.191.190.250) |
| 03:30.58 | drfreeze | :( |
| 03:31.14 | dlynes | jblack: pardon? |
| 03:31.59 | drfreeze | It looks like the 0000000000.cfg file tells each phone which file it should load |
| 03:32.33 | *** part/#asterisk gbr (n=gbr_@189.90.51.240) |
| 03:33.20 | dlynes | jblack: that svg (which got cancelled) is based on what concept? |
| 03:34.34 | dlynes | hrm...I guess he's awol, now |
| 03:34.49 | dlynes | bcrisp: you're making my head hurt |
| 03:35.08 | dlynes | bcrisp: too spammy :( |
| 03:36.15 | bcrisp | whahaha |
| 03:37.16 | bcrisp | dlynes: spammy? |
| 03:38.41 | dlynes | bcrisp: extremely (first pastebin) |
| 03:39.09 | bcrisp | dlynes: ah sorry |
| 03:40.10 | bcrisp | the notable part was the: -- Registration for '17772899142@callcentric.com' timed out, trying again (Attempt #1) |
| 03:40.26 | bcrisp | that happens as soon as the dialplan tries to call out |
| 03:40.38 | dlynes | bcrisp: That's not notable |
| 03:40.42 | dlynes | bcrisp: that's just coincidence |
| 03:40.56 | bcrisp | it happens every time |
| 03:40.58 | dlynes | bcrisp: it was completely unrelated |
| 03:41.03 | dlynes | bcrisp: different thread |
| 03:41.20 | bcrisp | ok |
| 03:41.27 | dlynes | bcrisp: that's a registration |
| 03:41.50 | bcrisp | it spits that message out only when this queue event occurs |
| 03:42.05 | dlynes | bcrisp: from one of your 'register => blahblah:blahblah@blahblah:5060' lines in your sip.conf file |
| 03:42.15 | dlynes | bcrisp: it's just purely coincidental |
| 03:42.55 | dlynes | bcrisp: you're probably spacing your calls out to that number at the same frequency as the registration timeouts |
| 03:43.39 | p3nguin | CallCentric registration annoys me. It always interjects during debug. |
| 03:43.41 | bcrisp | notice how it immediately says nobody answers? |
| 03:43.43 | dlynes | bcrisp: if you don't do any queues at all, and make no calls |
| 03:43.51 | dlynes | bcrisp: it didn't |
| 03:43.59 | bcrisp | ya it does tho.. |
| 03:44.01 | dlynes | bcrisp: it said nobody answered after 10s of waiting |
| 03:44.03 | bcrisp | it may say 10 sec |
| 03:44.07 | bcrisp | thats not the truth |
| 03:44.10 | bcrisp | it immediately spits that out |
| 03:44.25 | dlynes | bcrisp: not according to what I see in your log |
| 03:44.25 | bcrisp | as soon as the SIP/bcrisp ends |
| 03:44.39 | dlynes | bcrisp: are you logging this to a log with timestamps? |
| 03:44.48 | *** join/#asterisk Ta^3 (n=tacvbo@189.136.32.249) |
| 03:44.49 | dlynes | bcrisp: if so, repastebin it, with timestamps |
| 03:44.51 | bcrisp | i think so |
| 03:44.56 | bcrisp | whats the default loc for those? |
| 03:45.03 | dlynes | bcrisp: keep in mind, timestamps aren't enabled by default |
| 03:45.05 | Katty | breathes |
| 03:45.10 | Katty | i think sammy is okay |
| 03:45.14 | dlynes | bcrisp: /etc/asterisk/logger.conf |
| 03:45.20 | bcrisp | k |
| 03:45.31 | dlynes | bcrisp: btw |
| 03:45.32 | p3nguin | /var/log/asterisk/* |
| 03:46.10 | dlynes | bcrisp: another way you can see that it's obvious the registration timeout and your call end are not related is because your call used callcentric, but your registration used callcentric.com |
| 03:46.30 | bcrisp | ok, its just weird that it always does that |
| 03:46.37 | bcrisp | i've run it probably 15 times |
| 03:46.38 | p3nguin | (2143.39) <p3nguin> CallCentric registration annoys me. It always interjects during debug. |
| 03:46.53 | dlynes | p3nguin: i seen that, but bcrisp obviously didn't |
| 03:46.54 | bcrisp | k |
| 03:47.05 | dlynes | p3nguin: you're just telling him what I told him :) |
| 03:47.07 | bcrisp | ok, the phone never rings |
| 03:47.27 | p3nguin | dlynes: I'm hoping that I can reinforce what you've said, and maybe he'll give up. |
| 03:47.29 | dlynes | bcrisp: it might never ring...that might be true, but asterisk is still saying that it's ringing it for 10s |
| 03:47.29 | bcrisp | swigs some mylanta |
| 03:47.44 | dlynes | whatever mylanta is |
| 03:48.05 | dlynes | must be an american thing? |
| 03:48.17 | p3nguin | You don't have Mylanta in Canadia? |
| 03:48.23 | dlynes | no idea what it is |
| 03:48.34 | dlynes | if i knew what it was, i might be able to answer that question |
| 03:48.45 | bcrisp | its not importan |
| 03:48.48 | p3nguin | http://www.walgreens.com/store/catalog/Stomach-Remedies/Maximum-Strength-Antacid-Anti-Gas/ID=prod2658&navCount=1&navAction=push-product?V=G&ec=frgl_630105&ci_src=14110944&ci_sku=sku302658 |
| 03:48.57 | dlynes | oh...that shit |
| 03:49.05 | dlynes | I think i've heard the name before |
| 03:49.17 | dlynes | don't know if it was on an american or a canadian tv channel, though |
| 03:49.40 | dlynes | nah...pretty sure we don't have that stuff here |
| 03:49.49 | bcrisp | ok so if * is saying its ringing the phone for 10 secs and it never rings.. |
| 03:49.59 | dlynes | pepto bismol is pretty popular though |
| 03:50.02 | p3nguin | Sounds like a device issue. |
| 03:50.16 | dlynes | bcrisp: what p3nguin said |
| 03:50.20 | bcrisp | a device issue |
| 03:50.35 | dlynes | bcrisp: probably remote end is acting like it's receiving it, but just completely ignoring it |
| 03:50.42 | bcrisp | hmm |
| 03:50.48 | p3nguin | I don't really know what's going on, since I just got back a little bit ago. |
| 03:51.01 | dlynes | bcrisp: iow, probably a callcentric issue |
| 03:51.07 | bcrisp | i can dial in to employees context and use callcentric no problem |
| 03:51.09 | dlynes | bcrisp: if you're timing out registering to it |
| 03:51.24 | dlynes | bcrisp: i would imagine you probably can't send unauthenticated calls to it, either |
| 03:52.01 | bcrisp | im only timing out when im in the damn queue |
| 03:52.11 | p3nguin | "dial in to employees context" means dial the numbers on your phone and there is an exten match in a context called [employees]? |
| 03:52.19 | bcrisp | fk |
| 03:52.23 | drfreeze | Ugh |
| 03:52.30 | *** join/#asterisk tacubo (n=tacvbo@189.136.32.249) |
| 03:52.36 | dlynes | bcrisp: did you try making any outbound calls on callcentric since you started getting these registration timeouts? |
| 03:52.48 | bcrisp | i dont get them |
| 03:53.00 | dlynes | bcrisp: then I must be hallucinating |
| 03:53.05 | bcrisp | only .. when... the queue member is called |
| 03:53.07 | bcrisp | does it happen |
| 03:53.30 | dlynes | bcrisp: can we see your complete dialplan? |
| 03:53.35 | p3nguin | I'm SOOOOO glad that the only problem I have is that Transfer() doesn't work when I expect it to work. |
| 03:53.35 | bcrisp | i have a sip device set up that defaults to "employees" context |
| 03:53.40 | bcrisp | within that context there is pattern matching |
| 03:53.42 | dlynes | bcrisp: maybe there's a tight loop in there somewhere |
| 03:53.45 | bcrisp | that then sends to callcentric |
| 03:53.50 | bcrisp | and it works every single time no issus |
| 03:54.10 | dlynes | bcrisp: scrub your passwords first |
| 03:54.38 | bcrisp | hmm |
| 03:56.23 | bcrisp | ya im watching from cli .. it rings SIP/bcrisp twice, then enters the callcentric and immediately returns "nobody picked up in 10 sec" |
| 03:57.12 | dlynes | bcrisp: dialplan |
| 03:57.26 | dlynes | bcrisp: and if you want, also pastebin a log with timestamps |
| 03:57.39 | dlynes | bcrisp: but no point repastebinning a log that doesn't have timestamps |
| 03:58.03 | dlynes | s/dialplan/complete dialplan/ |
| 03:58.54 | bcrisp | you need the complete dialplan? |
| 03:59.01 | bcrisp | k |
| 04:00.11 | dlynes | yes |
| 04:00.18 | dlynes | but like I said....scrub your passwords first |
| 04:00.35 | p3nguin | People put passwords in their dialplans? |
| 04:02.25 | bcrisp | http://pastebin.ca/1712848 |
| 04:02.37 | bcrisp | i dont think i have passwords in my dialplan |
| 04:03.32 | dlynes | p3nguin: yes...believe it or not |
| 04:03.56 | bcrisp | in my prior pastebins for sip.conf i just wrote a bogus password in |
| 04:03.57 | dlynes | p3nguin: Dial(SIP/username:password@siphost/exten) |
| 04:04.37 | bcrisp | dlynes: if my dialplan is a mess i apologize - learning |
| 04:04.53 | p3nguin | It's terrible. Your dialplan logic needs serious work. |
| 04:05.07 | dlynes | bcrisp: you realize if you have autofallthrough=yes set, that asterisk will perform default behaviour on any calls that you haven't specifically handled, right? |
| 04:05.44 | dlynes | bcrisp: it's probably better not to use that when you're first starting out, so that you're not getting weird behaviour that you don't understand why it's doing it |
| 04:06.04 | bcrisp | dlynes: ok |
| 04:06.25 | bcrisp | the support and sales contexts aren't used btw |
| 04:06.38 | *** join/#asterisk etfonhomey (n=etfonhom@74-131-159-160.dhcp.insightbb.com) |
| 04:06.53 | dlynes | bcrisp: now, 'callcentric' has been defined in sip.conf? |
| 04:07.42 | *** join/#asterisk moy (n=moy@189.162.193.102) |
| 04:08.05 | bcrisp | yes |
| 04:08.31 | bcrisp | when a device with default context of employees dials out it works no probs |
| 04:09.11 | bcrisp | does my queueforwarding context dialplan loo kwrong? |
| 04:09.19 | bcrisp | "loo kwrong" awesome |
| 04:09.21 | p3nguin | So then you know that CallCentric isn't the problem. |
| 04:10.19 | dlynes | bcrisp: Change the line 'exten => 353-NOANSWER,1,Dial(SIP/14807172182@callcentric,20)' so that it's 'exten => 353-NOANSWER,1,Dial(SIP/14807172182@callcentric)' |
| 04:10.27 | dlynes | bcrisp: then it's the same as the employees context |
| 04:10.32 | dlynes | bcrisp: Let's see how that goes? |
| 04:10.37 | p3nguin | I'm not seeing any exten matching 353-${DIALSTATUS},1 within the queueforwarding context. |
| 04:10.53 | dlynes | slaps p3nguin . |
| 04:10.55 | bcrisp | 353-NOANSWER |
| 04:11.05 | bcrisp | dlynes i tried removing the 20 |
| 04:11.16 | dlynes | bcrisp: and it still fails right away? |
| 04:11.42 | bcrisp | ya immediately |
| 04:11.52 | dlynes | bcrisp: btw...you're not handling DIALSTATUS after calling Dial(SIP/14807172182@callcentric,20) |
| 04:12.03 | *** join/#asterisk eXcAliBuR (n=awww@207.134.8.34) |
| 04:12.18 | dlynes | bcrisp: Handle it with a Noop(DIALSTATUS=${DIALSTATUS}) so we can see what happens |
| 04:12.26 | dlynes | bcrisp: and then pastebin the resulting log |
| 04:12.29 | bcrisp | after the line? |
| 04:12.35 | eXcAliBuR | ok, here is my stupid question for the night... can asterisk do phone chains... like for school closures? |
| 04:12.43 | dlynes | bcrisp: after the NOANSWER line, and before the BUSY line |
| 04:12.52 | dlynes | eXcAliBuR: phone chains? |
| 04:12.56 | bcrisp | before i had exten => 353-NOANSWER,n,Hangup() |
| 04:13.14 | eXcAliBuR | you know, dial a list of people 1 after the other |
| 04:13.18 | dlynes | bcrisp: yeah....leave that in there, and just before it, do the noop |
| 04:13.24 | bcrisp | dlynes: ok |
| 04:13.27 | dlynes | eXcAliBuR: yes, of course it can |
| 04:14.14 | bcrisp | dlynes, would that line look like: 353-NOANSWER,n,Noop(..... ? |
| 04:14.19 | dlynes | eXcAliBuR: and then play back a pre-recorded message to them? |
| 04:14.21 | dlynes | bcrisp: yes |
| 04:14.23 | bcrisp | k |
| 04:14.44 | dlynes | bcrisp: make sure there's no ';' 's in there |
| 04:15.08 | eXcAliBuR | yup |
| 04:15.27 | dlynes | eXcAliBuR: script it using AGI, or AEL |
| 04:16.11 | bcrisp | like this right? : http://pastebin.ca/1712858 |
| 04:16.35 | dlynes | bcrisp: exactly |
| 04:16.39 | bcrisp | k ill try it now |
| 04:16.42 | dlynes | bcrisp: so let's see the pastebinned log |
| 04:17.38 | dlynes | eXcAliBuR: another way you can do it, too is just to use a perl script or something that creates a bunch of call files |
| 04:17.44 | dlynes | eXcAliBuR: that's probably the best way to do it |
| 04:18.04 | eXcAliBuR | i'm guessing AGI since that's in my handbook |
| 04:18.10 | eXcAliBuR | AEL isn't |
| 04:18.11 | eXcAliBuR | ;( |
| 04:18.15 | dlynes | eXcAliBuR: although...not sure how you would track success or failure of each call though |
| 04:18.24 | dlynes | eXcAliBuR: using call files that is |
| 04:21.01 | dlynes | bcrisp: I'm guessing you saw something that told you why it's failing, and that's why you haven't pastebinned the log? |
| 04:22.43 | bcrisp | dlynes that was my complete dialplan |
| 04:22.49 | bcrisp | oops sorry |
| 04:22.53 | bcrisp | irc window was scrolled up |
| 04:23.14 | bcrisp | no dlynes: im looking at the log and its not telling me anything.. i probably dont have logger configed right |
| 04:23.18 | bcrisp | let me clear the log and retry |
| 04:23.35 | dlynes | bcrisp: after you've retried, just pastebin what you get |
| 04:23.42 | bcrisp | ok |
| 04:24.00 | dlynes | bcrisp: even if you think it's not telling you anything...let me make that determination |
| 04:24.07 | bcrisp | ok |
| 04:24.29 | bcrisp | is there a cmd to clear a file without deleteing it.. just clear contents? |
| 04:24.57 | bcrisp | i thought it was "touch" |
| 04:25.11 | dlynes | bcrisp: rm $file && touch $file |
| 04:25.35 | dlynes | bcrisp: then logger restart from the asterisk cli |
| 04:26.19 | bcrisp | k |
| 04:27.33 | p3nguin | bcrisp: "> file" |
| 04:28.03 | p3nguin | That will turn it into an empty file. |
| 04:28.44 | bcrisp | http://pastebin.ca/1712869 |
| 04:30.21 | bcrisp | i hung up after it went and starting dialing SIP/bcrisp again |
| 04:32.08 | dlynes | bcrisp: wtf is that? |
| 04:32.26 | dlynes | bcrisp: that's not your log |
| 04:32.31 | bcrisp | queue_log ? |
| 04:32.41 | dlynes | bcrisp: i never asked you to show me queue_log |
| 04:32.46 | bcrisp | ... |
| 04:32.47 | dlynes | bcrisp: show me the full log |
| 04:32.53 | bcrisp | ok sorry |
| 04:33.06 | dlynes | bcrisp: i.e. your normal /var/log/asterisk/full (but where you made the call...not the whole log) |
| 04:33.15 | bcrisp | ok ill just clear it |
| 04:35.11 | p3nguin | wonders if bcrisp will use two commands, or only one |
| 04:35.27 | ChannelZ | doesn't |
| 04:36.02 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 04:36.42 | bcrisp | http://pastebin.ca/1712876 |
| 04:36.46 | dlynes | p3nguin: your method still required two commands |
| 04:36.55 | dlynes | p3nguin: cause you'd still need to do logger restart |
| 04:37.01 | dlynes | p3nguin: logger rotate does it in one |
| 04:38.02 | bcrisp | ok did i do it right? |
| 04:38.06 | dlynes | bcrisp: wtf!!! |
| 04:38.10 | bcrisp | lol |
| 04:38.13 | dlynes | bcrisp: you didn't do dialplan reload after you made your changes |
| 04:38.20 | bcrisp | huh? |
| 04:38.24 | dlynes | bcrisp: yeah |
| 04:38.32 | bcrisp | dlynes: ya i did |
| 04:38.45 | dlynes | bcrisp: well, i don't see it executing that line |
| 04:39.40 | dlynes | bcrisp: erm |
| 04:39.43 | dlynes | bcrisp: wait |
| 04:39.57 | dlynes | bcrisp: yeah |
| 04:40.26 | bcrisp | :< |
| 04:40.32 | bcrisp | do you see where it says timeout 10000 |
| 04:40.40 | bcrisp | and the timestamp is 1 sec? |
| 04:40.59 | dlynes | bcrisp: dialplan show queueforwarding, and pastebin it |
| 04:41.05 | dlynes | bcrisp: from the asterisk cli |
| 04:41.11 | bcrisp | k |
| 04:41.53 | bcrisp | http://pastebin.ca/1712877 |
| 04:41.54 | dlynes | bcrisp: actually...yeah...nobody picked up in 10000 ms, but only 1s elapsed |
| 04:42.09 | bcrisp | (fkn liars lol) |
| 04:42.37 | dlynes | bcrisp: Try pastebinnning the result of this: 'ps auxffww | grep asterisk' |
| 04:43.32 | bcrisp | whats that? |
| 04:43.46 | dlynes | bcrisp: it'll tell me how many asterisk processes you have running |
| 04:43.58 | bcrisp | http://pastebin.ca/1712879 |
| 04:44.16 | dlynes | bcrisp: ok...that's normal |
| 04:44.45 | dlynes | bcrisp: ok...there's something fubar in your dialplan |
| 04:44.52 | bcrisp | k |
| 04:44.54 | dlynes | bcrisp: have you turned off autofallthrough yet? |
| 04:44.59 | bcrisp | let me do that now |
| 04:45.14 | dlynes | bcrisp: after you've done that, redo your call, and pastebin the log |
| 04:45.28 | bcrisp | autofallthrough=no right? |
| 04:45.33 | dlynes | bcrisp: correct |
| 04:45.35 | bcrisp | ok |
| 04:45.42 | dlynes | bcrisp: i suspect that's probably what's causing the odd behaviour |
| 04:46.00 | dlynes | bcrisp: makes it very hard to troubleshoot issues when that's enabled |
| 04:46.25 | dlynes | bcrisp: you don't have priorityjumping enabled as well, do you? |
| 04:46.44 | bcrisp | no |
| 04:47.28 | dlynes | cool...i like this new asterisk 1.6.1 |
| 04:47.34 | dlynes | core show threads is definitely cool |
| 04:48.11 | bcrisp | http://pastebin.ca/1712883 |
| 04:48.43 | dlynes | bcrisp: did you do a dialplan reload |
| 04:48.46 | dlynes | bcrisp: ? |
| 04:49.06 | bcrisp | yes |
| 04:49.23 | dlynes | bcrisp: ok...restart asterisk, and repastebin the log after you've restarted and rerun the call then |
| 04:49.33 | dlynes | bcrisp: i guess the autofallthrough doesn't pick up on a reload |
| 04:49.36 | bcrisp | whats the preferred way to restart asterisk |
| 04:49.43 | bcrisp | i actually restarted the service, then the dialplan |
| 04:49.57 | dlynes | bcrisp: /etc/init.d/asterisk restart for this time |
| 04:50.05 | bcrisp | service asterisk restart doesnt work? |
| 04:50.10 | dlynes | bcrisp: but normally from the cli, restart when convenient |
| 04:50.17 | dlynes | bcrisp: or service asterisk restart |
| 04:50.25 | bcrisp | ya i already did that before running this call |
| 04:50.26 | dlynes | bcrisp: it does the same thing as /etc/init.d/asterisk restart |
| 04:50.35 | dlynes | bcrisp: you did? |
| 04:50.38 | bcrisp | ya |
| 04:50.58 | bcrisp | ill do it once more to make sure |
| 04:51.03 | dlynes | bcrisp: ok...pastebin your logger.conf file |
| 04:51.15 | dlynes | bcrisp: You must be missing something in your logging statement |
| 04:51.24 | bcrisp | my logger.conf is a real mess |
| 04:51.26 | bcrisp | its from the sample |
| 04:51.57 | dlynes | bcrisp: cat logger.conf | grep -v "^[<SPACE><TAB>]*$" |
| 04:52.08 | dlynes | bcrisp: and replace space and tab with the actual characters |
| 04:52.14 | dlynes | bcrisp: erm |
| 04:52.28 | bcrisp | hm |
| 04:52.30 | dlynes | bcrisp: cat logger.conf | grep -v "^[<SPACE><TAB>]*;.*$" |
| 04:52.34 | bcrisp | k |
| 04:53.28 | bcrisp | http://pastebin.ca/1712887 |
| 04:54.03 | bcrisp | my entire irc window is red |
| 04:54.09 | dlynes | bcrisp: damn...wonder why you're only logging verbose, then |
| 04:54.23 | bcrisp | hm? |
| 04:54.41 | dlynes | bcrisp: every single log entry i see in your log is at 'VERBOSE' level |
| 04:55.01 | dlynes | bcrisp: you have no 'NOTICE' or 'DEBUG' levels |
| 04:55.02 | bcrisp | is this being overwridden elsewhere? |
| 04:55.11 | dlynes | bcrisp: the Noop() will show up at NOTICE level |
| 04:55.58 | dlynes | bcrisp: nvm...it's VERBOSE level, too |
| 04:56.24 | dlynes | bcrisp: oh...you know what it probably is |
| 04:56.37 | dlynes | bcrisp: change your timeout for your queue to 20s, instead of 10s |
| 04:56.45 | dlynes | bcrisp: i.e. in queues.conf |
| 04:56.52 | bcrisp | ok |
| 04:57.31 | dlynes | bcrisp: and then restart asterisk (I don't know how to reload queues) |
| 04:57.42 | bcrisp | module reload app_queue |
| 04:57.46 | bcrisp | lol |
| 04:57.51 | dlynes | bcrisp: well, i suppose that would work ;) |
| 04:58.43 | bcrisp | yes! |
| 04:58.58 | bcrisp | it worked |
| 04:59.09 | bcrisp | timeout in queues.conf to 20 instead of 10 |
| 04:59.36 | bcrisp | ah because its combining the time |
| 04:59.39 | bcrisp | duh |
| 05:00.02 | bcrisp | 10 seconds rining sip/ben, (queue timeout) |
| 05:00.15 | bcrisp | dlynes: you da ma |
| 05:00.16 | bcrisp | man |
| 05:00.33 | bcrisp | i actually may need it to be closer to 30 |
| 05:00.50 | bcrisp | it rings once on my phone then stops heh |
| 05:01.25 | bcrisp | im going to get a 6 pack of beer and pour one out on your behalf |
| 05:01.26 | dlynes | bcrisp: ok....cool |
| 05:01.34 | dlynes | bcrisp: i guess we figured out what the issue was, then |
| 05:01.41 | bcrisp | bejesus |
| 05:01.44 | dlynes | bcrisp: good to know, in case i run into that issue, too |
| 05:01.54 | bcrisp | it makes since |
| 05:02.01 | bcrisp | it treats that call to local as a single thing |
| 05:02.04 | bcrisp | sense |
| 05:02.12 | dlynes | bcrisp: basically what was happening was that the queue was cutting off the call after 10s |
| 05:02.22 | bcrisp | yep |
| 05:02.23 | dlynes | bcrisp: but your initial call to the sip channel already lasted 10s |
| 05:02.27 | bcrisp | yep |
| 05:02.35 | dlynes | bcrisp: so that's why it was cutting it off prematurely on the second call |
| 05:02.58 | dlynes | bcrisp: so it had nothing to do with local channels...it was the queue |
| 05:03.12 | bcrisp | i learn more about * when there's a problem then when its workin |
| 05:03.23 | bcrisp | thanks for ur time |
| 05:03.26 | dlynes | bcrisp: yeah...you learn more about anything when there's a problem |
| 05:03.36 | dlynes | bcrisp: because then you work harder at the problem to solve it |
| 05:03.47 | bcrisp | true |
| 05:04.08 | bcrisp | i really like that channel idea |
| 05:04.14 | bcrisp | local channel |
| 05:04.23 | dlynes | bcrisp: I've got some pretty elaborate macros just so I can avoid having to rewrite the code again, down the road |
| 05:04.39 | dlynes | bcrisp: and some pretty elaborate dialplan contexts |
| 05:04.49 | bcrisp | id like to see em sometime |
| 05:05.06 | bcrisp | we're gonna have some interesting scenarios once asia is online |
| 05:05.08 | dlynes | bcrisp: well, there's one dialplan i'm probably going to convert to a database-based macro soon |
| 05:05.23 | dlynes | bcrisp: because i'm getting tired of writing 30 or 40 gotoif's |
| 05:05.28 | bcrisp | rofl |
| 05:05.34 | bcrisp | you dont like writing BASIC? |
| 05:05.59 | bcrisp | can you use ael? |
| 05:06.11 | dlynes | bcrisp: no...it's more like if(...) then .... else if (...) then .... else if( .... ) then ... else ... |
| 05:06.24 | bcrisp | oh, so its like QBASIC? |
| 05:06.25 | bcrisp | hehe |
| 05:06.33 | dlynes | bcrisp: nah..it's like ael |
| 05:06.37 | dlynes | bcrisp: but different |
| 05:06.57 | dlynes | bcrisp: or perl or php or any other structured language, for that matter |
| 05:07.25 | bcrisp | ya i havent gotten that far yet |
| 05:07.29 | bcrisp | only a few weeks in now |
| 05:07.33 | dlynes | bcrisp: gotoif really doesn't have anything to do with goto |
| 05:08.14 | bcrisp | i like the idea of AGI |
| 05:08.23 | dlynes | bcrisp: on my main server, i currently ahve about 3,814 lines of dialplan code |
| 05:08.33 | bcrisp | :) |
| 05:08.49 | bcrisp | you should see my credit card gateway library .. you'd scream |
| 05:08.51 | dlynes | bcrisp: but, I also maintain several other satellite phone systems |
| 05:09.13 | dlynes | so total lines of code would probably be closer to about 5000 |
| 05:09.22 | dlynes | but that's just asterisk dialplan code |
| 05:09.24 | bcrisp | ya.. im a total newb when it comes to telecomm |
| 05:09.28 | dlynes | no ael, no ami, no agi |
| 05:09.56 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 05:10.00 | dlynes | I'm currently not using any of those three for asterisk scripting |
| 05:10.21 | dlynes | Only for interfaces for other things |
| 05:10.27 | bcrisp | i think it must take years to remember all the little flags and switches in * |
| 05:10.45 | dlynes | bcrisp: nobody ever knows all of them |
| 05:10.59 | dlynes | bcrisp: because nobody ever works with everything in asterisk |
| 05:11.17 | dlynes | bcrisp: I've only worked with sip, iax and zap |
| 05:11.19 | bcrisp | ya some of the conf files are a little daunting |
| 05:11.24 | dlynes | bcrisp: never worked with isdn or skinny |
| 05:11.31 | dlynes | bcrisp: or some of the other more esoteric stuff |
| 05:11.36 | bcrisp | ya |
| 05:11.41 | bcrisp | i havent worked with iax |
| 05:12.08 | dlynes | well..one company i subcontract for i work with iax2 and sip |
| 05:12.21 | dlynes | another one I work with iax2, sip and zap (pri's and analog lines) |
| 05:12.46 | dlynes | both companies i need to interface with analog extensions and analog lines |
| 05:13.02 | dlynes | it's just i usually use a gateway to interface with them at the one job, and the other I use pci cards |
| 05:13.05 | bcrisp | interesting |
| 05:13.33 | bcrisp | well i sure appreciate your time |
| 05:13.38 | dlynes | not a problem |
| 05:13.43 | dlynes | heading to bed now, anyways |
| 05:13.46 | bcrisp | is it beer:30 ? |
| 05:13.56 | dlynes | but glad I was able to figure out that issue |
| 05:14.05 | bcrisp | yeah, i was puzzled |
| 05:14.14 | bcrisp | i think the message is a little misleading |
| 05:14.21 | dlynes | don't really drink much....got a teetotaller for a wife :0 |
| 05:14.27 | bcrisp | heh |
| 05:14.31 | bcrisp | i dont either |
| 05:14.38 | bcrisp | but its been a hell of a decade... |
| 05:14.49 | bcrisp | gnite |
| 05:14.57 | dlynes | yeah...anyways |
| 05:15.06 | dlynes | don't get stuck in a rut, thinking about things one way |
| 05:15.07 | bcrisp | and thank you too p3nguin .. true power animal |
| 05:15.09 | dlynes | expand your mind |
| 05:15.15 | bcrisp | are you asking me to smoke pot? |
| 05:15.18 | bcrisp | (hopes not) |
| 05:15.26 | bcrisp | jk |
| 05:15.27 | dlynes | nah....just break out of the box |
| 05:15.32 | dlynes | think outside the box |
| 05:15.35 | bcrisp | well once i understand how this fits together |
| 05:15.36 | bcrisp | ya |
| 05:15.48 | bcrisp | its like a giant lego kit |
| 05:15.53 | dlynes | remember...asterisk is NOT a phone system |
| 05:16.05 | bcrisp | its like a PLC |
| 05:16.18 | bcrisp | (programmable logic controller) |
| 05:16.47 | bcrisp | ya im excited .. |
| 05:16.49 | bcrisp | nite! |
| 05:16.53 | *** part/#asterisk bcrisp (i=bcrisp@ip72-222-167-229.ph.ph.cox.net) |
| 05:17.10 | dlynes | Yeah...I work with those every once in a while, too |
| 05:17.19 | dlynes | and the Panelmate (touch screen for a plc) |
| 05:25.43 | *** join/#asterisk shams (n=chatzill@c58-107-200-17.thoms2.vic.optusnet.com.au) |
| 05:26.23 | shams | Hi anyone here can help me on minor configuration of Extensions.conf |
| 05:26.37 | ChannelZ | perhaps |
| 05:26.54 | *** join/#asterisk simplydrew (n=simplydr@ool-44c2ab91.dyn.optonline.net) |
| 05:31.29 | ChannelZ | ...or perhaps not... |
| 05:33.42 | shams | hi I need to strip the 3 digits from an exsiting dnid and passed it on to next priority as extension |
| 05:34.07 | ChannelZ | ${EXTEN:3} |
| 05:35.17 | shams | yes I did that , but on the next prioty should I type exten => _X., or exten => ${EXTEN},n,Dialxxxxxxx |
| 05:36.06 | *** join/#asterisk bcrisp (i=bcrisp@ip72-222-167-229.ph.ph.cox.net) |
| 05:36.30 | ChannelZ | well what are you trying to do? You're not really "stripping" the extension, ${EXTEN:3} just represents the extension minus the first 3 numbers |
| 05:37.32 | ChannelZ | So if you're trying to dial that number minus the first 3, it's all one line.. you just do exten => _X.,1,Dial(SIP/${EXTEN:3}) or whatever it is you're dialing |
| 05:38.11 | shams | ok this is what i want to achive, I have customers who send me tech prefix , for example 888+number, I want to strip off these 888 and send the call to next priroty so a2billing.php can executed |
| 05:39.03 | ChannelZ | ok but what are you DOING with the number after removing the 888? |
| 05:39.09 | shams | in next priorty where a2billing.agi is executed only need to send number without 888 |
| 05:39.44 | shams | a2billing.php will dialed the number as dnid and send the call to provider |
| 05:40.29 | shams | may be it is helpful if i put the exten commands here to make it sense |
| 05:40.51 | ChannelZ | yes I don't know shit about a2billing.php except that everyone hates it because it's a mess |
| 05:40.55 | shams | step one call is coming from a customer as 8884421094958589@ip |
| 05:42.02 | shams | step 2 extenstions.conf file should strip 888 and send 4421094958589 to different context , where a2billing.php will executed and send the call to our provider as specified in a2billing |
| 05:42.59 | shams | i have managed to do all other bits , just when trying to passed the number without 888 to next context is not working, I tried the Set(Callerid(dnid)=${EXTEN:3}) but did not work |
| 05:45.57 | ChannelZ | is that what a2billing is pulling from? CALLERID(dnid) ? |
| 05:46.45 | *** join/#asterisk bcrisp (i=bcrisp@ip72-222-167-229.ph.ph.cox.net) |
| 05:47.17 | shams | a2billing is pulling the the full number with 888 |
| 05:48.15 | shams | yes this callerid(dnid) is pulled by agi_dnid, |
| 05:48.23 | shams | but when trying to dialed the number it pick up somehow 888 + number |
| 05:49.11 | *** join/#asterisk xmitter (n=xmitter@c-24-21-212-187.hsd1.or.comcast.net) |
| 05:49.15 | shams | I guess i m making a mistake on the this part exten => <what should be here > ,2,DeadAGI(a2billing.php |4) |
| 05:50.32 | *** join/#asterisk simplydrew (n=simplydr@ool-44c2ab91.dyn.optonline.net) |
| 05:50.49 | ChannelZ | no you're not because you can't just change that unless you jump somewhere else with Goto |
| 05:51.41 | ChannelZ | do Set(CALLERID(dnid)=${EXTEN:3} |
| 05:52.55 | bcrisp | channelz you crazyman |
| 05:53.09 | ChannelZ | pulls his pants up |
| 05:54.16 | bcrisp | ~roulette |
| 05:54.17 | infobot | ACTION watches bcrisp pull the trigger: Click! |
| 05:54.40 | bcrisp | i like trivia bots |
| 05:55.12 | bcrisp | ~trivia |
| 05:56.05 | bcrisp | channelz what do u do for a living? |
| 05:57.15 | ChannelZ | I'm an editor, graphic designer, animator.. I do motion graphics and edit for TV, etc. Post-production |
| 05:57.22 | bcrisp | NICE |
| 05:57.23 | shams | I runs ok |
| 05:57.27 | bcrisp | sorry caps |
| 05:57.53 | shams | I runs IT consultantecny |
| 05:58.29 | bcrisp | im a developer, consultant, <insert job role> |
| 05:59.01 | ChannelZ | exactly |
| 05:59.29 | bcrisp | i dont think someone can use * if they don't love to learn |
| 05:59.49 | shams | yes I do Goto , and it goes to next context |
| 06:00.12 | ChannelZ | well what did you put in the Goto? |
| 06:00.33 | shams | just a minute let me type all teh command one by one |
| 06:01.25 | ChannelZ | I mean if you want to do it that way, you would Goto(${EXTEN:3},1) and then have to have an exten line that would match it (which your existing one would which will cause all kinds of problems), or jump to a different context to keep it separated like Goto(foo,${EXTEN:3},1) |
| 06:01.36 | ChannelZ | shams: pastebin.ca |
| 06:01.47 | ChannelZ | http://pastebin.ca I should say |
| 06:03.43 | shams | ok this is one command exten => _786.,1,Goto(a2bgold,_X.,1) |
| 06:04.03 | shams | exten => _X.,1,Wait,1 |
| 06:04.04 | shams | exten => _X.,2,Set(CALLERID(DNID)=${EXTEN:3}) |
| 06:04.30 | ChannelZ | uhhh |
| 06:04.38 | shams | [a2bgold] |
| 06:04.40 | shams | exten => _X.,1,Wait,1 |
| 06:04.42 | shams | exten => _X.,2,Set(CALLERID(DNID)=${EXTEN:3}) |
| 06:04.44 | shams | exten => _X.,3,DeadAGI(a2billing.php|4) |
| 06:04.45 | shams | exten => _X.,4,Wait,2 |
| 06:04.47 | shams | exten => _X.,5,Hangup |
| 06:04.49 | ChannelZ | stop doing that |
| 06:05.10 | ChannelZ | ok you have a mess |
| 06:05.17 | shams | yes i think so |
| 06:05.25 | shams | any suggesstion ? |
| 06:05.43 | ChannelZ | you said things enter your dialplan with 888 on the front - is that literal, or it's any 3 random digits? |
| 06:06.07 | shams | sorry it is literal 888 or 786 |
| 06:06.20 | ChannelZ | ok. |
| 06:06.36 | shams | 888 go to differnt context [a2bslvr]and 786 goes to [a2bgold] |
| 06:08.02 | ChannelZ | ... |
| 06:08.34 | shams | this one is for 786 |
| 06:09.27 | ChannelZ | http://pastebin.ca/1712927 |
| 06:11.50 | shams | ok i will try ,thanks a lots channelZ |
| 06:12.49 | ChannelZ | IF a2billing is REALLY pulling CALLERID(dnid) then you could also just do http://pastebin.ca/1712929 |
| 06:13.40 | *** join/#asterisk jasonwert (n=jasonwer@97-83-97-13.dhcp.trcy.mi.charter.com) |
| 06:16.12 | Corydon76-dig | I'm still in shock over winning a Polycom 335. I so rarely win anything. |
| 06:16.22 | ChannelZ | Did you have to show your boobs? |
| 06:16.32 | *** join/#asterisk hakr (i=bryan@element.techlive.tv) |
| 06:17.01 | Corydon76-dig | They aren't that big. Nobody but my husband wants to see my chest |
| 06:17.28 | ChannelZ | Well even better win then! |
| 06:17.48 | ChannelZ | Where'd you win? |
| 06:18.02 | Corydon76-dig | On the voip users conference |
| 06:18.05 | shams | it is working now channelZ thanks a lots |
| 06:18.09 | *** join/#asterisk bbt (n=sam@samuels.id.au) |
| 06:18.09 | Corydon76-dig | e4 |
| 06:18.13 | ChannelZ | ah nice |
| 06:18.17 | ChannelZ | shams: cools |
| 06:18.27 | shams | the second post was right |
| 06:18.50 | ChannelZ | well it's simpler if it worked |
| 06:19.22 | shams | because a2billing will always dialed dnid , without asking for another number |
| 06:19.28 | shams | so dnid has to be right |
| 06:26.04 | *** join/#asterisk simplydrew (n=simplydr@ool-44c2ab91.dyn.optonline.net) |
| 06:29.28 | ChannelZ | wanders off to watch Coraline |
| 06:31.04 | TJNII | closes eBay before he spends any more money |
| 06:32.41 | coppice | TJNII: watch the Weird Al E-Bay video, and bring back your sense of perspective |
| 06:34.04 | TJNII | I've had good luck with industrial and commercial stuff. eBay thinks I'm a buisness. |
| 06:37.50 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:0:0:0:2) |
| 07:08.55 | *** join/#asterisk psykon (n=joel@221-116.96-97.tampabay.res.rr.com) |
| 07:16.22 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
| 07:23.50 | *** join/#asterisk gooph (n=gooph@pool-71-96-244-205.dfw.dsl-w.verizon.net) |
| 07:33.28 | *** join/#asterisk vitaminx (n=vitaminx@89.130.31.1) |
| 07:41.32 | *** join/#asterisk Tech_Travis (n=Administ@cpe-76-87-9-130.socal.res.rr.com) |
| 07:50.31 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 08:09.18 | *** part/#asterisk Tech_Travis (n=Administ@cpe-76-87-9-130.socal.res.rr.com) |
| 08:14.34 | *** join/#asterisk Dovid (n=annon@213.8.121.90) |
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| 08:21.21 | *** join/#asterisk verywiseman (n=khaled@unaffiliated/verywiseman) |
| 08:22.06 | TheGGiant | I don't suppose there's anybody online that is a master of SIP configuration for Asterisk 1.6.11? |
| 08:22.43 | TheGGiant | (And/Or a correct place to go looking for help?) |
| 08:25.52 | ChannelZ | what's the problem? I'm a master by no means, but can be crafty |
| 08:26.45 | TheGGiant | Well the short version is: I've been setting up a home PBX to basically run as a SIP proxy for my Vonage - So far registration seems to work well and incoming calls works great |
| 08:27.04 | TheGGiant | But outgoing .... I just can't seem to get it working |
| 08:27.58 | TheGGiant | I have it set up as a peer - but there's just so many options to mess with in terms of trunking and what not - I think I'm just getting myself confused somewhere. |
| 08:28.50 | ChannelZ | so your little Vonage box is registering to your * box and you're handling incoming calls through it already? |
| 08:29.22 | TheGGiant | Well i have the little box unplugged as of this moment - until I have it completely working - then I'll worry about making it an extension. I'm using a softphone for testing right now |
| 08:29.38 | TheGGiant | But yeah, incoming works great |
| 08:29.58 | TheGGiant | I even figured out how to have it dial multiple extensions once it's in |
| 08:30.04 | TheGGiant | So that part seems to be okay |
| 08:30.13 | TheGGiant | But when I try to dial out with the softphone |
| 08:30.38 | TheGGiant | The debugging shows the connection going in, and it even shows ring |
| 08:30.50 | TheGGiant | But it just eventually fails |
| 08:31.59 | ChannelZ | hmm.. and the dialed number never actually rings in real life? |
| 08:32.15 | TheGGiant | *CLI> == Using SIP RTP CoS mark 5 |
| 08:32.16 | TheGGiant | <PROTECTED> |
| 08:32.18 | TheGGiant | <PROTECTED> |
| 08:32.19 | TheGGiant | <PROTECTED> |
| 08:32.21 | TheGGiant | [Dec 13 03:11:00] ERROR[29379]: tcptls.c:344 ast_tcptls_client_start: Unable to connect SIP socket to X.X.X.X:10000: Connection refused |
| 08:32.40 | TheGGiant | (I guess I didn't have to blank the IP from i.voncp.com - oops) |
| 08:32.56 | TheGGiant | yeah |
| 08:33.01 | ChannelZ | ok so you are getting an error |
| 08:33.26 | TheGGiant | Yeah - but I've been banging through the configuration files commenting things |
| 08:33.38 | TheGGiant | THe original error I got was 'congestion' |
| 08:33.48 | ChannelZ | well I'm wondering why it's saying port 10000 |
| 08:33.58 | TheGGiant | That's the port Vonage wants me to use - |
| 08:34.18 | ChannelZ | oh.. well OK.. is your * box a or behind a firewall? |
| 08:34.39 | TheGGiant | Right now it is - a MASQing unix firewall |
| 08:35.19 | ChannelZ | and actually that's a strange error message, are you using TCP SIP? |
| 08:35.22 | TheGGiant | (Eventually this box goes back into place - I'm replacing that unit with this one) |
| 08:35.34 | TheGGiant | Umm... Apparently? |
| 08:35.44 | TheGGiant | I just recently added 'Transport=tcp,udp because it wasn't working |
| 08:35.57 | TheGGiant | (I've been banging my head on this a while now) |
| 08:36.13 | TheGGiant | Here's the peer section: |
| 08:36.15 | ChannelZ | Vonage may or may not support TCP |
| 08:36.42 | ChannelZ | in either case is the firewall allowing traffic out of that port, tcp and udp? (I'd remove TCP by the way) |
| 08:36.53 | TheGGiant | [vonage] |
| 08:36.54 | TheGGiant | type=peer |
| 08:36.56 | TheGGiant | ;auth=md5 |
| 08:36.58 | TheGGiant | ;auth=USER:PASS@i.voncp.com |
| 08:36.59 | TheGGiant | nat=yes |
| 08:37.01 | TheGGiant | qualify=no |
| 08:37.03 | TheGGiant | host=i.voncp.com |
| 08:37.04 | TheGGiant | bindport=10000 |
| 08:37.06 | TheGGiant | port=10000 |
| 08:37.07 | TheGGiant | ;username=USER |
| 08:37.09 | TheGGiant | defaultuser=USER |
| 08:37.10 | TheGGiant | fromuser=USER |
| 08:37.12 | TheGGiant | authname=USER |
| 08:37.14 | TheGGiant | fromdomain=i.voncp.com |
| 08:37.15 | TheGGiant | secret=PASS |
| 08:37.17 | TheGGiant | context=internal |
| 08:37.17 | mchou | tf |
| 08:37.18 | TheGGiant | canreinvite=no |
| 08:37.20 | TheGGiant | dtmfmode=rfc2833 |
| 08:37.21 | TheGGiant | srvlookup=no |
| 08:37.23 | TheGGiant | transport=tcp,udp |
| 08:37.25 | TheGGiant | Yeah, it's set to allow and MASQ any traffic from the inside |
| 08:37.29 | mchou | use pastebin |
| 08:37.34 | TheGGiant | Oops sorry! |
| 08:38.20 | mchou | you have voage business acct? |
| 08:38.29 | mchou | vonage* |
| 08:38.30 | ChannelZ | I was just going to ask that |
| 08:38.33 | TheGGiant | OKay, I put it back to UDP only - and now I"m getting 'Status CONGESTION' |
| 08:38.43 | TheGGiant | That's a long story |
| 08:39.00 | mchou | a yes/no will suffice |
| 08:39.02 | ChannelZ | well the short answer is they probably don't allow you to bypass their little ATA unless you do |
| 08:39.08 | TheGGiant | I was a vonage customer WAAAAAAAAAAAAAY back when they used to actually give you your SIP cred when you asked for it - so no. |
| 08:39.31 | ChannelZ | as ypiu |
| 08:39.33 | ChannelZ | oops |
| 08:40.04 | mchou | TheGGiant: how did you figure vonage wanted you to connect on port 10000? |
| 08:40.05 | TheGGiant | The most annoying part is that incoming works great. |
| 08:40.10 | ChannelZ | as you're setup now, the congestion is probably coming from them.. which could possibly mean it's being blocked on their end, or it's expecting a wierd dialstring, or who knows |
| 08:41.07 | TheGGiant | Yeah I think it may have to do with the dialstring - I had it working once a while back using a wierd dialstring, but I could only dial one number because I couldn't get the extension to squeeze into it |
| 08:41.29 | TheGGiant | It's in my router's setting panel |
| 08:41.43 | ChannelZ | well do you know what it's _supposed_ to look like? |
| 08:42.14 | mchou | TheGGiant: your router setting panel? |
| 08:42.37 | mchou | TheGGiant: how do you know that doesnt refer to RTP? |
| 08:42.39 | TheGGiant | Yeah - they gave me my cred and taught me where to look for the settings on my router - this was forever ago |
| 08:43.19 | TheGGiant | Not sure what it should look like - just was one of my guessing attempts |
| 08:43.29 | TheGGiant | RTP? |
| 08:43.37 | mchou | google |
| 08:44.01 | TheGGiant | Oh geez |
| 08:44.13 | TheGGiant | So you're telling me that that port is only for registry and not for peering outgoing? |
| 08:44.25 | mchou | nope |
| 08:44.38 | mchou | your google fu failed you |
| 08:44.46 | *** join/#asterisk Tim_Toady (n=moi@188.4.65.219.dsl.dyn.forthnet.gr) |
| 08:44.53 | TheGGiant | Okay okay, hold on |
| 08:45.22 | ChannelZ | what he's saying is that port 10000 is (sort of) a 'standard' port number that RTP uses (the actual media stream) |
| 08:45.31 | ChannelZ | SIP is typically 5060 |
| 08:45.47 | TheGGiant | Is there any reason that incoming would work great on that port and not outgoing? |
| 08:46.36 | ChannelZ | hard to say.. you actually have port=10000 in your sip.conf in your working incoming config? |
| 08:46.55 | TheGGiant | yah |
| 08:46.57 | TheGGiant | one sec |
| 08:47.15 | TheGGiant | register => USER:PASS:USER@i.voncp.com:10000/vonage |
| 08:47.25 | mchou | lol |
| 08:47.34 | TheGGiant | ? |
| 08:48.34 | TheGGiant | I feel as though I've missed something super simple here? |
| 08:48.50 | mchou | yeah |
| 08:48.58 | TheGGiant | :( |
| 08:48.58 | mchou | like how SIP works |
| 08:49.09 | TheGGiant | That would be a fair assumption |
| 08:49.12 | TheGGiant | I've really new to it |
| 08:49.25 | TheGGiant | *I'm |
| 08:49.28 | TheGGiant | So - what did I miss? |
| 08:49.33 | mchou | ~book |
| 08:49.34 | 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 |
| 08:50.12 | mchou | basic sip concepts, for one |
| 08:50.51 | mchou | second, an actual debug would be way more helpful than you silly guesses |
| 08:50.57 | mchou | your* |
| 08:52.17 | TheGGiant | Well I appreciate any help I can get - |
| 08:52.22 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:54.10 | TheGGiant | What kind of debug would help? |
| 08:54.31 | ChannelZ | and your * actually registers with vonage with that line successfully? |
| 08:54.32 | *** join/#asterisk Dovid[Laptop] (n=annon@213.8.121.90) |
| 08:54.40 | TheGGiant | Yup |
| 08:54.44 | mchou | lol |
| 08:54.57 | ChannelZ | well without knowing their setup it could well be |
| 08:55.03 | mchou | that would be INCREDIBLE |
| 08:55.14 | ChannelZ | why |
| 08:55.18 | TheGGiant | Yeah... |
| 08:55.20 | TheGGiant | I'm curious too |
| 08:55.59 | mchou | cause the syntax is rather messed up |
| 08:56.20 | TheGGiant | *CLI> sip show registry |
| 08:56.22 | TheGGiant | Host dnsmgr Username Refresh State Reg.Time |
| 08:56.23 | TheGGiant | i.voncp.com:10000 Y USER 15 Request Sent Sun, 13 Dec 2009 03:5 |
| 08:56.38 | TheGGiant | I used the syntax from the 'example' config |
| 08:56.52 | TheGGiant | ; Format for the register statement is: |
| 08:56.54 | TheGGiant | ; register => [transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] |
| 08:57.01 | ChannelZ | thats not registered |
| 08:57.14 | mchou | lol |
| 08:57.32 | TheGGiant | That might be a problem |
| 08:57.36 | TheGGiant | So why does it ring in? |
| 08:57.44 | TheGGiant | Is it because i'm usign the port direct? |
| 08:57.54 | mchou | go read the book |
| 08:58.19 | TheGGiant | Hmm |
| 09:02.12 | mchou | there's no point to using scattergun "Hail Mary" approach |
| 09:02.37 | TheGGiant | Ugh. Now I still don't know how to fix it, and also feel like a moron because it sounds from mchou's reaction that I'm totally missing something simple... |
| 09:03.10 | mchou | you fix it by understanding what you're doing |
| 09:03.21 | mchou | not trying shit at random |
| 09:04.22 | ChannelZ | Don't worry, mchou will pretty much never help you in any useful way |
| 09:05.37 | mchou | ChannelZ: yeah, you the shining example of enlightenment |
| 09:05.43 | TheGGiant | :p thanks - but he IS right about it. I enjoy learning by doing when it's for me - and the 'hail mary' approach gets me in a lot faster. Learning from a book would be lest frustrating. |
| 09:06.04 | TheGGiant | I appreciate the help you guys have already given me |
| 09:06.13 | TheGGiant | at least I know what to attack next |
| 09:06.40 | ChannelZ | the biggest problem is shooting in the dark on the Vonage side |
| 09:06.58 | TheGGiant | ChannelZ: from your response I've done something wrong with SIP registration, and from mchou's responses I obviously have something wrong in my understanding of HOW it works |
| 09:07.04 | TheGGiant | Yeah it's true |
| 09:07.10 | TheGGiant | But it's a lot of fun to work around it |
| 09:07.14 | TheGGiant | :) |
| 09:07.40 | TheGGiant | Obviously my sip show registry shouldn't be showing request sent if it was working, no? |
| 09:08.50 | ChannelZ | yeah, it means it hasn't gotten a reply |
| 09:09.14 | ChannelZ | interestingly if I try to register with them on port 10000 I do get a response |
| 09:10.25 | ChannelZ | it seems more like the port is being blocked by your firewall, which you never answered |
| 09:10.57 | TheGGiant | Oh |
| 09:10.59 | TheGGiant | Wait a minute |
| 09:11.09 | TheGGiant | When you register, it connects back, no? |
| 09:11.30 | ChannelZ | well there's a return packet |
| 09:11.34 | TheGGiant | Hmm |
| 09:11.46 | TheGGiant | Naw that would masq back -and it wouldn't explain why the old router can be behind a nat too |
| 09:12.04 | ChannelZ | but by the nature of UDP yes it sort of connects back |
| 09:12.07 | TheGGiant | He's right, I need to stop guessing and figure out how it works before I start poking at it. |
| 09:12.46 | TheGGiant | Let me check the firewall log and see if anything is being eaten |
| 09:12.49 | ChannelZ | Either your outgoing request is being tossed to the floor, or the return is not making it back in. |
| 09:13.40 | TheGGiant | when I turn on Debug |
| 09:13.47 | TheGGiant | I see the header transmitting |
| 09:14.38 | TheGGiant | CSeq: 171 REGISTER |
| 09:14.47 | TheGGiant | SIP/2.0 200 OK |
| 09:18.02 | TheGGiant | And when I call the number I see the INVITE request from them... so why doesn't it show a state other than 'Request Sent'... |
| 09:18.33 | ChannelZ | what you just pasted was out of context |
| 09:18.38 | TheGGiant | OOps sorry |
| 09:19.53 | TheGGiant | Basically when I turn on debugging, I get 'REGISTER' requests all over the place. When I call the number, I get a SIP/2.0 180 Ringing -> CSeq: INVITE. but I also get an error - Unable to create channel of type SIP... |
| 09:21.40 | ChannelZ | you get register requests but do you ever get a response? we can't read your screen |
| 09:21.41 | TheGGiant | I wish Vonage didn't use your damn number as the username - |
| 09:22.04 | TheGGiant | Makes trying to pass on debug information annoyingly difficult as it contains my phone number everywhere! :) |
| 09:22.25 | TheGGiant | Yeah, it says: |
| 09:22.30 | ChannelZ | DONT PASTE IT HERE |
| 09:22.31 | mchou | ever heard of sed? |
| 09:22.40 | ChannelZ | http://pastebin.ca |
| 09:23.13 | TheGGiant | mchou - yeah of course, but it's just scary to share info in case I mess up |
| 09:23.52 | TheGGiant | OOh Pastebin is nifty |
| 09:23.57 | TheGGiant | http://pastebin.ca/1713003 |
| 09:24.23 | TheGGiant | One sec, I'll see if I can't throw a full debug up |
| 09:33.04 | ChannelZ | I'm going to bed - in short I believe your first problem is a firewall problem, and the second is lack of the proper info from Vonage on how they handle inbound and outbound so you'd have a prayer of getting it configged right |
| 09:33.40 | ChannelZ | your registration problem is just a small piece of the puzzle |
| 09:34.05 | TheGGiant | http://pastebin.ca/1713015 |
| 09:34.15 | TheGGiant | Ahh |
| 09:34.20 | TheGGiant | Thanks for the help |
| 09:36.39 | ChannelZ | oof that debug is waaaay too much |
| 09:36.52 | TheGGiant | Sorry - the interesting stuff happens at 950 |
| 09:36.54 | TheGGiant | that's when I called in |
| 09:37.07 | *** join/#asterisk Omorika (n=krash@78-0-243-56.adsl.net.t-com.hr) |
| 09:37.07 | TheGGiant | Answered it with the softphone |
| 09:37.10 | TheGGiant | Then hung up |
| 09:38.55 | TheGGiant | Anyway thanks for your help. I think mchou is right - if I have any hope of getting this up I need to know the protocol inside and out |
| 09:44.31 | TheGGiant | Well here's a question I can ask that maybe someone knows the answer: |
| 09:44.34 | TheGGiant | "SIP/2.0 407 Proxy Authentication Required" |
| 09:44.50 | TheGGiant | Any idea what can be done if my outbound requests are halted by this error? |
| 09:46.22 | mchou | umm |
| 09:46.40 | mchou | you need to authenticate |
| 09:46.58 | mchou | i.e. submit the proper credentials |
| 09:47.04 | mchou | wtf |
| 09:47.08 | mchou | go READ |
| 09:48.27 | TheGGiant | Alright. Thanks for you help |
| 09:48.34 | TheGGiant | Have a good one |
| 09:50.25 | *** join/#asterisk b14ck (n=comradeb@cpe-24-24-136-239.socal.res.rr.com) |
| 09:53.28 | *** join/#asterisk yoshx (n=yoshx@88-138-188-188.adslgp.cegetel.net) |
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| 10:28.07 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 10:40.25 | *** join/#asterisk HaMF (n=hannes@nat-wh.rz.uni-karlsruhe.de) |
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| 10:48.24 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
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| 11:00.05 | *** join/#asterisk elliot98 (n=elliot@unaffiliated/elliot98) |
| 11:00.10 | elliot98 | greetings! |
| 11:00.36 | elliot98 | Asterisk is sending an "Unauthorized" response when a certain phones tries to register |
| 11:00.40 | elliot98 | why is this? |
| 11:01.55 | drmessano | Because it's not authorized |
| 11:02.25 | elliot98 | what does that mean? |
| 11:02.41 | elliot98 | the account is dynamic, all passwords etc. is in order |
| 11:03.05 | mchou | elliot98: are these phones on the local lan? |
| 11:03.22 | elliot98 | they are connecting remotely |
| 11:03.30 | elliot98 | through a public i[ |
| 11:03.32 | elliot98 | ip |
| 11:03.34 | mchou | "fromdomain" |
| 11:04.08 | elliot98 | but they are connecting from a dynapic ip |
| 11:04.25 | mchou | so? |
| 11:04.45 | mchou | fromdomain is something YOU as ast admin specify |
| 11:05.11 | mchou | RTFM |
| 11:05.29 | elliot98 | isn't fromdomain used when asterisk itself needs to connect to another server? |
| 11:06.04 | mchou | yes, that's one of it's uses |
| 11:06.52 | elliot98 | what else is it used for? |
| 11:07.31 | mchou | to substitute the domain name |
| 11:09.32 | elliot98 | so I should replace it with the name of the domain name? |
| 11:14.33 | elliot98 | I see something here... |
| 11:15.12 | elliot98 | apparently, when phones first connect, Asterisk responds that with an "unauthorized" packet containing md5 hashing info, nonce, etc. |
| 11:15.29 | elliot98 | other phones then again send the REGISTER packet with the hashing info |
| 11:15.42 | elliot98 | this particular phone just gives up after the first time |
| 11:17.19 | elliot98 | and doesn't try to REGISTER with any sort of md5 digest |
| 11:19.40 | elliot98 | any idea why this phone doesn't want to reREGISTER itself? |
| 11:22.59 | mchou | is it a grandstream? :) |
| 11:27.34 | *** join/#asterisk kannan (n=kann@121.246.242.95) |
| 11:28.17 | elliot98 | not sure what kind of phone it is...doesn't look like a brandname to me...it was working though |
| 11:28.26 | elliot98 | that is what is funny |
| 11:42.21 | *** join/#asterisk tamiel (n=tamiel@ip-1.net-81-220-19.versailles.rev.numericable.fr) |
| 12:02.27 | elliot98 | I am wondering if another issue has anything to do with this: |
| 12:02.55 | elliot98 | if I try to run an internet speed test from one of those internest speed sites, I always get a latency error |
| 12:03.11 | elliot98 | with that PPP connection |
| 12:03.25 | elliot98 | I wonder if the ISP is having some odd routes that mess things up |
| 12:04.33 | *** join/#asterisk Dovid (n=annon@213.8.121.90) |
| 12:34.06 | *** part/#asterisk bbt (n=sam@samuels.id.au) |
| 12:35.59 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 12:48.05 | kannan | i am upgrading from asterisk 1.4.21.2 to latest 1.4.27.1. Where to get addons (i am using areski stats, so need mysql addons) |
| 12:49.41 | kannan | cannot find any addons pkg on website? |
| 12:53.30 | Tim_Toady | you can get it from http://downloads.asterisk.org/pub/telephony/asterisk/ kannan |
| 12:54.08 | kannan | thank you |
| 12:55.39 | *** join/#asterisk wierdo (n=jimmy@77.78.3.197) |
| 12:56.30 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 13:24.38 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 13:29.57 | *** join/#asterisk shams (n=chatzill@c58-107-200-17.thoms2.vic.optusnet.com.au) |
| 13:30.15 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 13:43.35 | dlynes | elliot98: it's a phone you've specified a peer name for? |
| 13:44.22 | dlynes | elliot98: or is it a phone that's trying to send unauthenticated traffic to your phone system? |
| 13:44.33 | dlynes | elliot98: i.e. a guest |
| 13:44.36 | dlynes | elliot98: ? |
| 13:50.13 | Katty | well sammy seems okay this morning. he right arm is red, but he's walking on it. |
| 13:50.18 | Katty | thinking perhaps his arm is just bruised. |
| 13:50.35 | Katty | he's also eating okay |
| 14:10.07 | *** join/#asterisk jmacz (n=jmacz@190.25.7.149) |
| 14:11.28 | *** join/#asterisk Dovid (n=annon@213.8.121.90) |
| 14:17.24 | *** join/#asterisk Buklov (n=buklov@213.138.71.254) |
| 14:22.38 | *** part/#asterisk etfonhomey (n=etfonhom@74-131-159-160.dhcp.insightbb.com) |
| 14:28.36 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:29.25 | *** join/#asterisk [netman] (n=netman@223.Red-88-19-164.staticIP.rima-tde.net) |
| 14:46.49 | *** join/#asterisk TimToady_ (n=moi@212.251.120.242.dsl.dyn.forthnet.gr) |
| 14:47.57 | *** join/#asterisk CheBuzz (n=CheBuzz@81.21.46.199) |
| 14:49.59 | CheBuzz | This may be an obvious question, but I couldn't find the answer anywhere. Is there any way to use IAX between two * boxes without compiling the ztdummy driver? Ie, like just using apt-get install ztdummy in Ubuntu (that doesn't work) |
| 14:50.29 | CheBuzz | Or maybe using something else for timing besides ztdummy that doesn't require modifying the kernel/building from source. |
| 15:02.13 | [TK]D-Fender | CheBuzz: They have a DAHDI package, and you don't require it for IAX2 except for trunk mode |
| 15:03.27 | CheBuzz | Right, I understand that. Trunking is what I'm going for. |
| 15:05.43 | [TK]D-Fender | then install DAHDI |
| 15:07.00 | CheBuzz | Thanks, I'll do some reading on DAHDI |
| 15:07.09 | *** join/#asterisk puzzled_ (n=foobar@puzzled.xs4all.nl) |
| 15:17.51 | *** join/#asterisk voipmonk (n=voipmonk@69.172.114.221) |
| 15:18.35 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 15:32.15 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 15:44.40 | *** join/#asterisk ivan_paes (n=paes@201-67-161-42.fnsce703.dsl.brasiltelecom.net.br) |
| 15:48.39 | elliot98 | dlynes: the phone has its own extension num and password |
| 15:49.32 | elliot98 | dlynes: but if first needs the nonce and digest stuff from the server, so it sends a REGISTER packet |
| 15:49.40 | elliot98 | dlynes: and * responds with the info |
| 15:49.59 | elliot98 | dlynes: usually, the phone then resends the REGISTER with the hashed password |
| 15:49.59 | *** join/#asterisk levity (n=levity@unaffiliated/canuck) |
| 15:50.09 | elliot98 | dlynes: but this phone stopped doing that |
| 15:50.32 | dlynes | elliot98: if you do sip show peer peername, you should be able to see what kind of phone it is |
| 15:50.49 | dlynes | elliot98: it'll show the agent string at the bottom of the sip show peer info |
| 15:53.15 | cusco | hi |
| 15:55.14 | elliot98 | dlynes: it says - VOB820-PHONE |
| 15:55.35 | dlynes | elliot98: there ya go, so now you know it's not a grandsucks |
| 15:55.41 | dlynes | anyways...gotta run |
| 15:57.45 | elliot98 | so is that good or bad? |
| 15:58.03 | elliot98 | thanks! |
| 16:08.11 | *** join/#asterisk mokmeister (n=mokmeist@109.78.25.192) |
| 16:12.49 | *** join/#asterisk bcrisp (i=bcrisp@ip72-222-167-229.ph.ph.cox.net) |
| 16:18.46 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
| 16:25.18 | *** join/#asterisk Alagar (n=Administ@122.164.38.25) |
| 16:52.23 | *** part/#asterisk CheBuzz (n=CheBuzz@81.21.46.199) |
| 16:58.08 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:01.13 | *** join/#asterisk af_ (n=getsmart@88-149-230-89.dynamic.ngi.it) |
| 17:03.04 | *** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
| 17:05.38 | *** join/#asterisk voipmonk (n=voipmonk@69.172.114.221) |
| 17:08.21 | dlynes | no idea |
| 17:08.32 | dlynes | never heard of it |
| 17:09.45 | *** join/#asterisk danj1980 (n=dan@91.108.0.113) |
| 17:18.43 | bcrisp | shiga who? |
| 17:25.01 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 17:25.30 | *** join/#asterisk farkus (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 17:27.29 | HaMF | Hi, I use asterisk with two digium TE110-wildcards to connect a Nixdorf 8818 pbx with the PSTN. It works. Usually. But sometimes (randomly) asterisk freezes (takes 100% CPU) and you have to restart asterisk. Can anyone help me track the problem? |
| 17:27.40 | HaMF | this is my config: zaptel.conf: http://pastebin.com/d3adbf2ac ; zapata.conf: http://pastebin.com/d5070c5cd extensions.conf: http://pastebin.com/d65ea880a |
| 17:28.06 | HaMF | and: asterisk |
| 17:28.14 | voipmonk | what version of asterisk and dahdi? |
| 17:28.17 | HaMF | Asterisk 1.4.21.2~dfsg-3 on an debian lenny-system |
| 17:28.20 | voipmonk | sorry |
| 17:28.21 | voipmonk | zaptel |
| 17:28.23 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 17:29.33 | HaMF | zaptel 1.4.11 |
| 17:29.58 | *** join/#asterisk neurosys (n=neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 17:30.34 | cusco | hi |
| 17:30.39 | HaMF | this is what ztscan says: http://pastebin.com/d4dd897f4 and this is lspci on the cards: http://pastebin.com/d3d96282d |
| 17:30.41 | cusco | when a extension is circuit-busy |
| 17:30.43 | cusco | what does it mean? |
| 17:31.17 | dlynes | bcrisp: you shagged who? |
| 17:32.23 | dlynes | cusco: it means it's either not accepting any calls at this time (all of its available channels are used up), or you sent it a codec that it can't handle |
| 17:32.51 | dlynes | cusco: when you get a circuit-busy, it'll usually tell you a SIP status value as well |
| 17:33.06 | cusco | status is CHANANAVAILB |
| 17:33.13 | dlynes | cusco: Such as SIP/500 Internal error |
| 17:33.31 | dlynes | cusco: CHANUNAVAIL means that the channel isn't even up |
| 17:33.59 | dlynes | cusco: iow, there's no route to the peer (usually) |
| 17:34.20 | dlynes | cusco: or it's just not running a sip or iax2 service |
| 17:34.51 | cusco | :/ |
| 17:35.05 | cusco | I can call it directly |
| 17:35.10 | cusco | the extension |
| 17:35.15 | cusco | he is using x-lite |
| 17:35.17 | dlynes | HaMF: what's the last thing you see on the console when it locks up? |
| 17:35.31 | cusco | but asterisk queue can't reach him with chananavailb |
| 17:35.35 | dlynes | cusco: what do you mean by 'I can call it directly'? |
| 17:35.48 | dlynes | cusco: and how is it different from how you're calling it with the queue? |
| 17:35.57 | dlynes | cusco: you're not making any sense |
| 17:36.05 | cusco | ... |
| 17:36.08 | dlynes | cusco: please pastebin your queues.conf and your extensions.conf files |
| 17:36.15 | cusco | I can dial his extension |
| 17:36.17 | cusco | (611) |
| 17:36.20 | dlynes | cusco: both in their entirety...and scrub any passwords |
| 17:37.09 | dlynes | cusco: and also pastebin a log of where it's working from a direct call and where it's not working from a queue call |
| 17:37.14 | cusco | ok |
| 17:37.43 | cusco | Im grepping full for 611: |
| 17:38.29 | dlynes | cusco: show me the entire context of the call |
| 17:38.39 | dlynes | cusco: not just the lines that have the phrase '611' in them |
| 17:39.45 | HaMF | dlynes, thats a difficult question... I only once had the chance to watch how the process freezes and at this time there was no pri debug enabled. and everything seemed fine. |
| 17:39.58 | cusco | dlynes: ok hold |
| 17:41.19 | HaMF | dlynes, I already tried logging every output pri debug generates to a file, but when I was doing so, the server did not crash for about 6 days. |
| 17:42.08 | dlynes | HaMF: I don't need pri debug just yet |
| 17:42.22 | dlynes | HaMF: I was just hoping for normal output at the time it crashed |
| 17:42.45 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 17:43.45 | cusco | dlynes: this one I can reach 611: http://paste.debian.net/53888/ |
| 17:44.06 | voipmonk | HaMF what does dmesg say, whats in the /var/log/messages dir on those dates when the system crashed? |
| 17:44.17 | voipmonk | HaMF do you use splunk? |
| 17:44.25 | *** part/#asterisk LemensTS (n=customgt@adsl-70-238-155-79.dsl.stlsmo.sbcglobal.net) |
| 17:45.18 | cusco | dlynes: now the queue also reached 611... weird |
| 17:45.29 | drmessano | <PROTECTED> |
| 17:45.31 | cusco | dlynes: tho, he can't hear me |
| 17:45.36 | drmessano | and ploonk |
| 17:45.37 | cusco | I can hear him |
| 17:45.41 | drmessano | But not splunk |
| 17:45.45 | cusco | but he can't hear me only on that extension |
| 17:45.49 | cusco | could that be somehow relate |
| 17:45.51 | cusco | dto asterisk? |
| 17:46.47 | cusco | dlynes: thanks for helping, nevermind |
| 17:46.48 | cusco | :/ |
| 17:46.53 | dlynes | cusco: you figured it out? |
| 17:47.04 | HaMF | dlynes, nope sorry. The only thing I recorded (and the only strange thing I noticed) was: http://pastebin.com/d2459d9b9 (note the missing channel Zap/32 in this list). Well when I notice, the service crashed, it already did crash. |
| 17:47.06 | cusco | nope, its just working now |
| 17:47.16 | HaMF | (usually happens once a day) |
| 17:48.06 | dlynes | HaMF: did you try voipmonk's suggestions? Also, where's the console dump, if you have one (without pri debug info)? |
| 17:49.31 | dlynes | HaMF: also, depending on what Linux distro you're on, you might want to check your /var/log/syslog around the time when it crashed, too |
| 17:49.33 | drmessano | You have 32 channels configured in Zaptel? |
| 17:50.01 | dlynes | drmessano: he's got two te110 cards |
| 17:50.15 | drmessano | oh, thank goodness |
| 17:50.21 | dlynes | drmessano: so, assuming he's in europe, an e-1 would be 31 channels |
| 17:50.28 | drmessano | Yes, i know |
| 17:51.51 | HaMF | voipmonk, dmesg/syslog did not output any usefull information (nothing except the usuall). and asterisk messages usually looks like http://pastebin.com/d7a0cfd8 |
| 17:52.03 | cusco | dlynes: actually, he can't hear me on that extension, but if he changes the extension he can |
| 17:52.11 | cusco | so its not a computer-problem |
| 17:52.26 | HaMF | I did not yet figure out what creates the unable to forward voice frame warnungs. |
| 17:52.28 | cusco | how do I figure how what is wrong with extension 611 |
| 17:52.44 | HaMF | no, i dont use splunk |
| 17:54.18 | HaMF | voipmonk, but if this can help in solving the problem, tell me how to use it. |
| 17:54.39 | dlynes | HaMF: btw...have you considered upgrading to something that uses dahdi, instead of zaptel? |
| 17:55.10 | dlynes | HaMF: Or are you bent on using this particular version just so you don't have to do a build cycle? |
| 17:56.19 | voipmonk | splunk helps to organize and search through your logs - if you had it installed you could search using the time of your crash without having to grep your whole drive :) |
| 17:56.22 | dlynes | cusco: well, i've asked at least once now for both logs and both config files |
| 17:56.45 | voipmonk | HaMF we need debug & logs for the times your system spiked and crashed |
| 17:56.46 | dlynes | cusco: and i've only seen one log posted, and no config files |
| 17:56.48 | HaMF | yes... (to both questions), but I set up a gentoo-system and installed asterisk but dahdi does not provide a (v)zaphfc-module which I need for the third span |
| 17:57.15 | dlynes | HaMF: i see...ok |
| 17:57.15 | HaMF | so before switching to another version of asterisk (1.6) I have to either buy a new card or work around this problem. |
| 17:57.30 | dlynes | HaMF: what's the card you're using that requires it? |
| 17:57.42 | cusco | dlynes: sorry |
| 17:58.04 | cusco | dlynes: what configuration files would you need |
| 17:58.12 | dlynes | cusco: queues.conf and extensions.conf |
| 17:58.12 | cusco | the ael we use to dial another extension? |
| 17:58.24 | cusco | erm |
| 17:58.25 | dlynes | cusco: or your ael, if you're not using extensions.conf |
| 17:58.37 | dlynes | cusco: or both, if that's what you're doing |
| 17:58.39 | cusco | we use mysql |
| 17:58.43 | cusco | ... |
| 17:58.50 | cusco | queues are stored in mysql |
| 17:58.55 | dlynes | cusco: oh...realtime asterisk extensions? |
| 17:58.58 | cusco | yes |
| 17:59.09 | dlynes | ah...then I'm lost |
| 17:59.15 | dlynes | never used the real time extensions |
| 17:59.22 | cusco | its the same really.. |
| 17:59.40 | cusco | just we have other apps to decide if the extension is with any queue |
| 17:59.41 | cusco | or not |
| 17:59.54 | dlynes | cusco: btw |
| 18:00.09 | HaMF | dlynes, have a look at http://pastebin.com/d4dd897f4 (ztscan) its the third card which is an "normal" BRI ISDN card with a Cologne Chip (HFC) |
| 18:00.12 | dlynes | cusco: you have noticed that you're trying to play a file to the caller(?), that doesn't exist, right? |
| 18:00.19 | cusco | yes dlynes |
| 18:00.27 | dlynes | HaMF: nvm...no need to explain...you're using BRI |
| 18:01.14 | *** join/#asterisk cpoulson (n=chris@pdpc/supporter/active/cpoulson) |
| 18:01.26 | dlynes | cusco: can you try fixing that problem, to see if you start to get two-way audio? |
| 18:01.37 | cusco | lol |
| 18:01.39 | dlynes | cusco: just in case it's a codec translation issue that's preventing two way audio |
| 18:01.56 | cusco | that problem has been there for a long time, maybe you can take a look at, that: let me paste ael |
| 18:02.10 | dlynes | cusco: either that, or just get rid of the code that tries to play the followme file |
| 18:02.50 | madsara | This is odd... packet catures indicate taht between my sip phone and my asterisk, only the sip phone is sending RTP, nothing is coming from the asterisk to the phone |
| 18:03.08 | cusco | dlynes: no weneed that, but It should go into the IF |
| 18:03.13 | cusco | dlynes: http://paste.debian.net/53894/ |
| 18:03.18 | cusco | _XXX |
| 18:03.38 | cusco | we have a: Set(NewCallMsg=followme/${PARTNER}); |
| 18:03.45 | *** part/#asterisk levity (n=levity@unaffiliated/canuck) |
| 18:03.50 | cusco | but sometimes ${PARTNER} is empty |
| 18:04.02 | cusco | (like dialing directly to an extension) |
| 18:04.23 | cusco | shouldn't it fall under: if (${NewCallMsg} = "followme/") |
| 18:04.29 | cusco | (line 31 at the paste website= |
| 18:04.30 | cusco | ) |
| 18:04.31 | HaMF | voipmonk, what exact logs do you need? Unfortunately I can't provide a console-dump (which most likely is what we need). As said when I was logging the console output the error did not occur. But I can set up splunk if it's of any use. |
| 18:05.20 | dlynes | HaMF: /var/log/asterisk/full and /var/log/messages and /var/log/syslog (if it exists), and a 'dmesg' dump at the time of the crash (if one exists) |
| 18:08.02 | dlynes | cusco: I don't see any issues there... |
| 18:09.28 | cusco | dlynes: right, nor do I. so the followme should be directed to followme/no-follow |
| 18:09.45 | cusco | instead, it just prints "followme/" no such file bla bla |
| 18:09.51 | cusco | and doesn't print any of the NoOp |
| 18:10.47 | *** join/#asterisk haryv (i=lanny@174.1.114.16) |
| 18:11.20 | dlynes | cusco: ${PARTNER} == no-follow? |
| 18:11.26 | haryv | unbelievable! Nortel Corporate Greed at its best http://news.ca.msn.com/top-stories/cbc-article.aspx?cp-documentid=22732123 |
| 18:14.47 | drmessano | Who cares? |
| 18:14.53 | drmessano | Fsck Nortek |
| 18:14.55 | drmessano | Fsck Nortel too |
| 18:16.04 | bcrisp | making enemies of those who make a lot of money is a step towards socialism |
| 18:16.32 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:16.35 | bcrisp | regardless of bankruptcy protection... u either pay them market value or they leave |
| 18:16.41 | bcrisp | quite simple |
| 18:17.57 | dlynes | bcrisp: the problem is management getting paid, and the staff getting screwed |
| 18:18.09 | *** join/#asterisk simplydrew (n=simplydr@ool-44c2ab91.dyn.optonline.net) |
| 18:18.30 | bcrisp | govt bailing out companies is just dumb |
| 18:18.31 | dlynes | bcrisp: it's the second time now that nortel's done that bs while in bankruptcy court |
| 18:18.43 | bcrisp | right |
| 18:18.44 | dlynes | bcrisp: nobody's bailing out nortel, except other companies |
| 18:18.50 | madsara | http://www.fark.com/cgi/vidplayer.pl?IDLink=4845526 |
| 18:19.01 | dlynes | bcrisp: the government hasn't done squat for them |
| 18:19.05 | bcrisp | ah |
| 18:19.54 | dlynes | bcrisp: that being said, nortel would've been a better company to bail out than gm or chrysler |
| 18:19.59 | *** join/#asterisk puzzled_ (n=foobar@puzzled.xs4all.nl) |
| 18:20.10 | bcrisp | i agree |
| 18:20.11 | dlynes | bcrisp: gm and chrysler's business methods have gone the way of the dinosaur |
| 18:20.23 | dlynes | bcrisp: and they need to change their ways, or they're dead |
| 18:20.33 | bcrisp | you don't give tax payer money to companies that produce products that are not in demand |
| 18:20.34 | dlynes | bcrisp: no amount of government bailout is going to help them |
| 18:20.57 | dlynes | it's all just going to get pissed down the toilet, anyways |
| 18:21.12 | dlynes | nobody wants to buy a car that breaks down 6 months after they buy it |
| 18:21.19 | bcrisp | right.. and the "cash for clunkers" - artificial.. just redistribution of tax funds to support a failing business |
| 18:21.24 | thehar | snowwww |
| 18:21.36 | dlynes | that being said |
| 18:21.41 | thehar | so much snowwww |
| 18:21.45 | dlynes | ford and gm trucks are still very much in demand |
| 18:22.20 | dlynes | but the cars are horrible |
| 18:22.41 | bcrisp | right, but the funds go towards supporting ridiculous union agreements |
| 18:22.51 | dlynes | bcrisp: yep |
| 18:23.18 | dlynes | and hwat's the union doing for them now, that they're unemployed? |
| 18:23.20 | dlynes | nothing |
| 18:23.37 | bcrisp | the profit margin on vehicles is actually negative for many |
| 18:24.17 | dlynes | bcrisp: the new city golf apparently had an profit margin that was barely above zero |
| 18:24.45 | dlynes | bcrisp: so vw realized with the amount they were producing, they had to change their assembly line drastically |
| 18:25.02 | dlynes | bcrisp: the new gti has a much healthier profit margin |
| 18:25.10 | bcrisp | ya.. thats vw for u |
| 18:25.24 | dlynes | and the new gti is freaking awesom |
| 18:25.31 | dlynes | s/awesom/awesome/ |
| 18:25.48 | dlynes | I've always been a sucker for vw :) |
| 18:25.53 | prgmrchris | ew |
| 18:26.07 | dlynes | Every car I've owned has been a vw |
| 18:26.21 | HaMF | dlynes, I just had a look through the log files, but could not find any hints (at the times when asterisk crashed). I configured monit so it will dump the logs to extra files as soon as asterisk crashes again and I turned on every debug possibility, asterisk provides (except pri debug)) so let's wait.. |
| 18:26.22 | dlynes | well..and they've all been golfs :) |
| 18:26.27 | bcrisp | i owned a jetta and a passat |
| 18:26.49 | dlynes | the gl and the gti were both pretty slick |
| 18:26.56 | bcrisp | passat was nice but it was a turbo that liked to cake up the oil |
| 18:27.05 | dlynes | as long as you're not forced to try driving an automatic |
| 18:27.20 | dlynes | those jettas with an automatic tranny are pretty boring |
| 18:27.25 | bcrisp | own a mazda now.. been great |
| 18:27.47 | dlynes | HaMF: btw...if you like pri debug |
| 18:27.53 | dlynes | HaMF: there's also pri intense debug |
| 18:28.01 | HaMF | I know |
| 18:28.09 | dlynes | HaMF: but i suspect your issue has nothing to do with your pri |
| 18:28.21 | dlynes | HaMF: i suspect it's either a driver or a dialplan issue |
| 18:28.32 | dlynes | HaMF: does it lock up the entire machine, or only asterisk? |
| 18:29.40 | HaMF | dlynes, I had some fun hours using pri debug and pri intense debug. My hope is that I at least can figure out what leads to the problem. (And this has to be an action on one of the cards.. hopefully) |
| 18:30.02 | dlynes | HaMF: is it the entire machine that locks up though, or only asterisk? |
| 18:30.27 | HaMF | dlynes, well, asterisk uses 100%cpu, so you cant use the machine anymore. but if you kill asterisk, everything's fine |
| 18:30.51 | dlynes | HaMF: ok, so it could be either then...either the driver or the dialplan |
| 18:31.16 | HaMF | did you have a look at the dialplan I posted earlier? |
| 18:31.40 | dlynes | HaMF: no...but i suspect that's probably the least of your worries, unless you've got a loop in it |
| 18:31.54 | dlynes | HaMF: if there's a loop, that could be the reason, though |
| 18:32.46 | HaMF | dlynes, yeah but the dialplan is rather simple |
| 18:32.57 | dlynes | HaMF: can you pastebin it again? |
| 18:33.12 | HaMF | sure |
| 18:34.57 | *** join/#asterisk voipmonk (n=voipmonk@69.172.114.221) |
| 18:35.09 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:38.09 | HaMF | here it is http://pastebin.com/d206535b3 (I deleted the comments ;... they describe the strange signalling in germany and why we're dialing what whe're dialing) |
| 18:40.23 | HaMF | (hihi the dialplan is 300 lines. 50 lines for extensions, 250 for the comments...) |
| 18:42.28 | dlynes | HaMF: I don't see anything out of the ordinary there |
| 18:42.43 | dlynes | HaMF: However, you do have autofallthrough=yes set |
| 18:42.56 | dlynes | HaMF: which might make debugging issues with your dialplan a little erratic |
| 18:43.16 | dlynes | HaMF: but, your dialplan is so simple, i don't think it woudl be an issue |
| 18:44.08 | HaMF | dlynes, yes autofallthrough is on but i already turned if off for testing and asterisk froze anyways. |
| 18:44.53 | dlynes | HaMF: i wasn't implying that it would cause crashing (it probably would never cause crashing) |
| 18:45.22 | dlynes | HaMF: it's what it would be running instead that might crash your system...but there's nothing for it to run instead, except the default: Hangup |
| 18:45.28 | voipmonk | stop asterisk |
| 18:45.32 | voipmonk | run it with -vvvvgcd |
| 18:45.38 | voipmonk | then when it dies it will dump a core file |
| 18:45.48 | voipmonk | then we can debug from the core file :) |
| 18:45.48 | _Raptor_ | does anyone here know about russian voip providers who offer a sip flatrate tarife? |
| 18:45.58 | voipmonk | in the mean time - test with dahdi on your test server |
| 18:46.02 | voipmonk | let it run for a week |
| 18:46.09 | dlynes | voipmonk: He's running a binary distribution that probably has all the debug information removed |
| 18:46.14 | voipmonk | and then if u dont have any trouble , migrate to it :) |
| 18:46.27 | voipmonk | thats his fault |
| 18:46.51 | HaMF | debian provides debugging symbols in a seperate package |
| 18:47.05 | dlynes | voipmonk: and he's can't run dahdi...he needs the drivers for the hfc for bri |
| 18:47.09 | *** join/#asterisk hakr (i=bryan@element.techlive.tv) |
| 18:47.12 | voipmonk | asterisk is a toolkit not a solution |
| 18:47.33 | *** join/#asterisk Davedan (n=me@CBL217-132-75-171.bb.netvision.net.il) |
| 18:47.48 | HaMF | the problem is that asterisk doesnt die, it "just freezes". |
| 18:47.59 | Davedan | can you recommend a sip client for ubuntu for testing? |
| 18:48.08 | HaMF | I already tried to detatch to asterisk using gdb but thats kind of ugly |
| 18:48.13 | dlynes | Davedan: ekiga? |
| 18:48.16 | voipmonk | set your environment up for debugging and then try to reproduce the freezing |
| 18:48.31 | HaMF | working on an system that takes one minute to accept a character |
| 18:48.32 | Davedan | dlynes: on karmic ekiga isn't installed by default anymore. Is it the best? |
| 18:48.51 | dlynes | Davedan: no idea...I use a sip hardphone |
| 18:49.05 | dlynes | Davedan: and all my clients use xten or something similar on windows |
| 18:49.17 | voipmonk | HaMF, why would you run a telephony app on such a system? |
| 18:49.23 | voipmonk | nevermind, dont answer that :) |
| 18:49.27 | *** join/#asterisk JonMR (n=jon@67-207-128-103.slicehost.net) |
| 18:49.30 | HaMF | *hust* |
| 18:49.34 | Davedan | dlynes: thanks |
| 18:49.37 | dlynes | laughs. |
| 18:49.41 | voipmonk | http://www.voip-info.org/wiki/view/Asterisk+debugging |
| 18:50.52 | dlynes | Davedan: apt-get install ekiga, no? |
| 18:51.11 | dlynes | Davedan: a lot of people on here seem to be using sjphone as well |
| 18:52.47 | drmessano | I love Windows Messenger so much with Asterisk I run XP in a VM to use it |
| 18:55.21 | HaMF | voipmonk, if you could predict when the system will crash again, you could do the things mentioned at "Debugging asterisk". but let's wait for clean logs :) |
| 18:58.39 | bcrisp | drmessano seriously? |
| 18:59.20 | bcrisp | messenger is the devil |
| 18:59.50 | drmessano | No, but it does work well as a softphone |
| 19:01.56 | bcrisp | i just use skype |
| 19:02.59 | drmessano | Using their well hidden SIP functions? |
| 19:05.42 | *** join/#asterisk Tim_Toady (n=moi@77.49.183.230.dsl.dyn.forthnet.gr) |
| 19:20.32 | *** join/#asterisk xmitter (n=xmitter@c-24-21-212-187.hsd1.or.comcast.net) |
| 19:24.48 | eppigy | Katty: :< |
| 19:25.58 | drmessano | wonders if xmitter has one way audio |
| 19:26.44 | xmitter | heh, heh. Yeah. |
| 19:28.23 | drmessano | That was a bad one, even for me |
| 19:29.08 | *** join/#asterisk TommyBotten (n=noosjent@217-14-12-26-dhcp-osl.bbse.no) |
| 19:29.42 | *** join/#asterisk ViniciusFontes (n=vinicius@189.7.198.85) |
| 19:31.46 | *** part/#asterisk ViniciusFontes (n=vinicius@189.7.198.85) |
| 19:34.11 | Davedan | I've changed extensions.confg and used 'dialplan reload' but 'dial plan who' gives me 28 extensions and 62 priorities |
| 19:36.52 | *** join/#asterisk wierdo (n=jimmy@77.78.3.197) |
| 19:42.17 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 19:49.38 | bcrisp | wb Davedan |
| 19:50.58 | p3nguin | I've never even heard of this "dial plan who" thing before. |
| 19:51.22 | ChannelZ | either |
| 19:51.51 | Obeliks | why can't I use format=ogg_vorbis for voicemail? Am I missing something? It shows up in core show file formats |
| 19:53.36 | dlynes | p3nguin: that's because Davedan's making stuff up |
| 19:53.48 | p3nguin | ah |
| 19:54.06 | dlynes | p3nguin: there's no such file as extensions.confg, either |
| 19:54.17 | p3nguin | I took that as a typo. |
| 19:54.24 | dlynes | p3nguin: I think he meant dialplan show |
| 19:54.40 | dlynes | p3nguin: but he's seriously dyslexic |
| 19:54.50 | *** join/#asterisk freddyk (n=f@host218-83-dynamic.3-79-r.retail.telecomitalia.it) |
| 19:54.52 | bcrisp | loves the new dialplan whosyodaddy command |
| 19:55.05 | *** join/#asterisk xmitter (n=xmitter@c-24-21-212-187.hsd1.or.comcast.net) |
| 19:55.17 | ChannelZ | 'core just work goddamnit' is better |
| 19:56.14 | bcrisp | ya i like that |
| 19:56.21 | bcrisp | core fixmyshit |
| 19:57.23 | Davedan | ? |
| 19:58.02 | Davedan | I meant dialplan show |
| 19:58.41 | ChannelZ | Davedan: what was the actual question? |
| 19:59.12 | Davedan | I'm following the asterisk book and trying to setup a test sip account |
| 19:59.27 | Davedan | the book says I should see 1 extension with the example but I'm getting 28 |
| 19:59.53 | bcrisp | hmm |
| 19:59.58 | Davedan | this is what I'm using for extensions.conf http://dpaste.com/132894/ |
| 20:00.28 | bcrisp | you need to add an extension |
| 20:00.36 | ChannelZ | well you should have 0 extensions, I see none. |
| 20:00.50 | ChannelZ | have you reloaded the dialplan since changing it? |
| 20:00.58 | Davedan | maybe asterisk takes config from somewhere else? |
| 20:01.10 | Davedan | in the CLI I used: 'dialplan reload' |
| 20:01.32 | bcrisp | Davedan whats the name of your sip device u want to test with? |
| 20:01.40 | bcrisp | SIP/Davedan ? |
| 20:02.03 | ChannelZ | When you did the reload, did it say something like " == Parsing '/etc/asterisk/extensions.conf': Found" ? |
| 20:02.17 | Davedan | this is the result of 'dialplan reload' http://dpaste.com/132898/ |
| 20:03.04 | bcrisp | ok davedan, looks like you have contexts but no extensions |
| 20:03.24 | bcrisp | in your sip.conf, where you added your device, what is the default context set as? |
| 20:03.45 | bcrisp | oh my bad |
| 20:03.47 | bcrisp | 500 |
| 20:03.50 | Davedan | http://dpaste.com/132903/ |
| 20:04.23 | bcrisp | ok davedan in your extensions.conf, under the [phones] definition |
| 20:04.25 | bcrisp | add this |
| 20:04.27 | ChannelZ | ah it looks like it's loading extensions.ael also |
| 20:04.34 | bcrisp | oh |
| 20:05.22 | Davedan | I'm using the ubuntu asterisk package. the only thing I changed was extensions.conf and sip.conf like the book says |
| 20:05.25 | ChannelZ | Davedan: remove /etc/asterisk/extensions.ael (or rename it something like _extensions.ael) |
| 20:06.06 | bcrisp | where is extension 500 defined? |
| 20:06.27 | Davedan | I've changed extensions.ael to extensions.ael.sample |
| 20:06.32 | ChannelZ | bcrisp: all that is coming from extensions.ael I think |
| 20:06.34 | Davedan | dialplan reload |
| 20:06.39 | Davedan | dialplan show |
| 20:06.42 | Davedan | still 30 extensions |
| 20:06.52 | bcrisp | ya pastebin extensions.ael |
| 20:07.05 | ChannelZ | we don't need to see it, he just doesn't want to load it |
| 20:07.10 | bcrisp | ah |
| 20:07.27 | bcrisp | davedan, you have extension 1000 in your sip.conf for context phones |
| 20:07.31 | Davedan | yes |
| 20:07.33 | ChannelZ | Davedan: wasn't it 28 before? it's gotten BIGGER? |
| 20:07.36 | bcrisp | so in your extensions.conf, under [phones] |
| 20:07.37 | bcrisp | add |
| 20:07.51 | bcrisp | exten => 500,1,Dial(SIP/1000,10) |
| 20:08.16 | Davedan | what is this? |
| 20:08.18 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.134.46.dyn.user.ono.com) |
| 20:08.36 | bcrisp | that says, when in the context "phones", 500 should dial SIP/1000 and timeout after 10 sec |
| 20:09.20 | bcrisp | then do a dialplan reload |
| 20:09.41 | Davedan | I don't understand. will it remove all the other extensions? |
| 20:09.56 | bcrisp | no |
| 20:10.18 | Davedan | I don't want to add things right now. just to understand why asterisk adds all the extensions |
| 20:10.22 | Davedan | thanks |
| 20:11.13 | bcrisp | oh |
| 20:11.41 | bcrisp | ok davedan |
| 20:11.44 | bcrisp | open up modules.conf |
| 20:11.53 | ChannelZ | I think you are not editing the file you think you are editing |
| 20:11.59 | bcrisp | noload => pbx_ael.so |
| 20:12.10 | bcrisp | then restart asterisk |
| 20:12.29 | *** join/#asterisk DocAwesome (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:12.29 | *** mode/#asterisk [+o DocAwesome] by ChanServ |
| 20:13.06 | Davedan | ChannelZ: When I change extensions.conf I see a different number of extensions. maybe bcrisp is right and there is a module that loads something |
| 20:13.12 | Davedan | maybe I can just ignore it |
| 20:13.36 | TJNII | check for users.conf |
| 20:13.41 | ChannelZ | well if you renamed the other extension files it shouldn't be loading anything |
| 20:13.42 | TJNII | I saw that in one of your pasetbins |
| 20:14.27 | bcrisp | ChannelZ: did he restart asterisk or just reloaded the dialplan? |
| 20:14.45 | ChannelZ | reloaded I think |
| 20:15.00 | Davedan | I did both |
| 20:15.14 | Davedan | this is the standard asterisk package for ubuntu. I didn't do special things |
| 20:15.26 | bcrisp | hmf |
| 20:15.33 | ChannelZ | well who knows what they've hacked up in that distro |
| 20:15.37 | ChannelZ | pastebin your users.conf |
| 20:15.39 | voipmonk | :) |
| 20:15.46 | ChannelZ | because something odd is going on |
| 20:15.57 | bcrisp | there's an angry gremlin somewhere... |
| 20:16.05 | voipmonk | u guys are sugar coating what this guy needs to hear |
| 20:16.10 | Davedan | users.conf http://dpaste.com/132913/ |
| 20:16.14 | ChannelZ | "build it yourself" |
| 20:16.32 | voipmonk | hides back under his rock |
| 20:16.35 | Davedan | ChannelZ: using a package shouldn't be a bad thing |
| 20:16.44 | ChannelZ | No, it shouldn't. |
| 20:16.47 | voipmonk | a lot of things are labelled under shouldnt |
| 20:16.53 | bcrisp | i dont think users.conf is the problem |
| 20:17.00 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 20:17.05 | bcrisp | Davedan did you try the noload? |
| 20:17.13 | ChannelZ | ubuntu likes to take standard config files and then include others that they intend YOU to configure in |
| 20:17.57 | ChannelZ | but based on one of your other pastes I don't see what is going on -- on line 17 of your reload for instance it says 'parsing users.conf' but then proceeds to add tons of AEL contexts and things |
| 20:19.05 | Davedan | <PROTECTED> |
| 20:19.07 | bcrisp | i think if you tell * to not load ael that would be prevented... |
| 20:19.14 | bcrisp | Davedan: open modules.conf |
| 20:19.35 | ChannelZ | but he supposedly renamed the ael extensions file so even if the module WAS loading, it shouldn't be able to find the config. |
| 20:19.50 | bcrisp | wonders if this package has a lot of other stuff in it |
| 20:20.00 | ChannelZ | I'm sure it does |
| 20:20.01 | voipmonk | whispers symbolic links maybe? - |
| 20:20.15 | Davedan | it's probably doesn't matters |
| 20:20.24 | Davedan | I don't mind the extensions to be there |
| 20:20.27 | bcrisp | davedan patebin your modules.conf |
| 20:21.37 | ChannelZ | You will mind when you start trying to configure your own and it doesn't do what you're telling it to do |
| 20:21.49 | bcrisp | ya something is plugging in to the mix |
| 20:21.58 | bcrisp | generating automagic contexts and extensions |
| 20:23.01 | voipmonk | nice |
| 20:23.04 | Davedan | so I'll build from source as you suggested |
| 20:23.09 | voipmonk | :) |
| 20:23.30 | ChannelZ | remove the package including it's config first |
| 20:23.33 | bcrisp | ~roulette |
| 20:23.34 | infobot | ACTION watches bcrisp pull the trigger: BANG! |
| 20:23.39 | bcrisp | dies |
| 20:23.42 | jblack | Natalie Portman is going to be in pride and prejudice and zombies? |
| 20:23.58 | bcrisp | jblack: not sure, but she'll be in my dreams :/ |
| 20:24.02 | TJNII | ~roulette |
| 20:24.03 | infobot | ACTION watches tjnii pull the trigger: Click! |
| 20:24.41 | TJNII | thinks infobot should kick whoever loses ~roulette. |
| 20:25.17 | jblack | http://www.imdb.com/title/tt1374989/ |
| 20:25.23 | ChannelZ | infobot needs ops first |
| 20:25.44 | TJNII | True. |
| 20:30.14 | dlynes | jblack: so what was that about last night? |
| 20:31.39 | *** join/#asterisk shinao1 (n=shinao1@41.219.239.202) |
| 20:33.36 | cusco | 18:10 < dlynes> cusco: ${PARTNER} == no-follow? --- no, that variable has other functionalities. but it seems fine.. |
| 20:33.42 | cusco | by the way, I have another question |
| 20:34.29 | *** join/#asterisk shinao1 (n=shinao1@41.219.239.202) |
| 20:34.59 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:36.18 | cusco | http://paste.debian.net/53903/ - audio file not found for playback |
| 20:36.58 | cusco | root@perfpbxr:/var/lib/asterisk/sounds# ls -lia audio/ZON/opcaoinvalida.ulaw |
| 20:36.59 | cusco | 9125914 -rwxrwxrwx 1 root root 19760 2009-12-13 15:46 audio/ZON/opcaoinvalida.ulaw |
| 20:38.21 | dlynes | cusco: it's probably looking for /var/lib/asterisk/sounds/audio/ZON/opcaoinvalida.ulaw.ulaw |
| 20:38.38 | dlynes | cusco: don't specify the extension...asterisk will automatically choose it |
| 20:38.44 | ChannelZ | Davedan: What version of ubuntu are you running |
| 20:39.26 | dlynes | jblack: nvm...seems you just say a whole bunch of random stuff |
| 20:39.35 | cusco | dlynes: ahh! ok |
| 20:40.09 | dlynes | cusco: i guess this is your first time using hte playback() app? |
| 20:40.23 | Davedan | ChannelZ: karmic |
| 20:40.28 | cusco | no... |
| 20:40.30 | cusco | I forgot |
| 20:40.31 | cusco | lol |
| 20:40.41 | dlynes | oh....guess you just never specified the extension before |
| 20:40.46 | dlynes | and this time you decided to do it |
| 20:40.59 | cusco | I just copied the filename yes |
| 20:41.02 | ChannelZ | Davedan: is that asterisk 1.6? |
| 20:41.38 | Obeliks | any idea how I can set the language on a chan_lcr channel so my voicemail prompts are in german? Set(CHANNEL(language)=de) throws an error |
| 20:41.38 | Davedan | I think so |
| 20:42.02 | cusco | cheers dlynes |
| 20:42.27 | dlynes | cusco: I think everyone's run into that problem before, when they first started using asterisk |
| 20:42.34 | dlynes | cusco: so you definitely won't be the last one :) |
| 20:43.45 | cusco | well I am trying to get round asterisk... my boss set it up, but wants me to maitain |
| 20:43.48 | cusco | maintain it |
| 20:43.55 | cusco | I really know nothing about asterisk :/ |
| 20:44.12 | dlynes | cusco: ah...have you read 'the book' yet? |
| 20:44.18 | Obeliks | sorry. I'm an idiot ;) had a typo in "language" |
| 20:44.55 | dlynes | Obeliks: You've got a typo in your nick, too ;) |
| 20:44.57 | cusco | dlynes: time to time I read a bit |
| 20:45.05 | Obeliks | dlynes, :P |
| 20:45.23 | dlynes | I would think it was supposed to be Obelisk, not Obeliks |
| 20:45.36 | cusco | obelix ? |
| 20:45.45 | Obeliks | nope, actually it's been Obeliks for 10-15 years or so ;) |
| 20:45.47 | dlynes | cusco: obelisk is a big rock |
| 20:46.03 | cusco | wikipedia says obelisk |
| 20:46.30 | dlynes | Obeliks: obeliks means something in deutsch? |
| 20:47.11 | Obeliks | no. But I guess you "pronounce" it the same way as Obelix |
| 20:48.09 | dlynes | Obeliks: oh...nvm...I thought the dude in the Asterix comic strips was Obelisk as well, cause he always carries around an obelisk, but his name was Obeliks |
| 20:48.48 | dlynes | erm..nvm....his name is Obeliks in some other language I guess |
| 20:48.49 | Obeliks | I guess that's the trick behind it ;) |
| 20:48.58 | dlynes | Looks like Polish |
| 20:49.02 | Obeliks | dlynes, yeah, I figured that out way after I got the nick ;) |
| 20:49.32 | dlynes | so there ya go |
| 20:49.41 | dlynes | obelisk in polish is obeliks |
| 20:49.58 | cusco | the fat dude from Asterix, yea I was thinking that too |
| 20:50.10 | cusco | they mix up everyones names to make it funny, so it could as well be |
| 20:50.14 | *** part/#asterisk Davedan (n=me@CBL217-132-75-171.bb.netvision.net.il) |
| 20:53.13 | *** join/#asterisk Wildy (n=simba@83.149.10.63) |
| 20:59.47 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 21:11.47 | ChannelZ | ...neeed...more...bandwidth.. |
| 21:12.15 | HaMF | hihi |
| 21:12.40 | jblack | Yeah. I' starting to feel cheated by 3.0mb/768kb for 45 a month. |
| 21:12.44 | *** join/#asterisk Davedan (n=me@DSL217-132-63-74.bb.netvision.net.il) |
| 21:12.51 | HaMF | ui |
| 21:14.24 | *** join/#asterisk grabes222 (n=Miranda@72.20.207.237) |
| 21:14.46 | ChannelZ | is stuck with shitty DSL |
| 21:15.09 | ChannelZ | I kicked Comcast to the curb for all of their intrusiveness |
| 21:16.21 | Davedan | I'm trying to call with x-lite to the echo test. on the CLI I see http://dpaste.com/132933/ but I don't hear anything on x-lite and it shows calling... |
| 21:16.33 | grabes222 | Can someone take a look at this backtrace, I can repeat this on anything higher than 1.6.0.14. Its after I receive a fax and call System http://pastebin.ca/1713594 |
| 21:18.34 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 21:19.10 | ChannelZ | Davedan: does it show connected? there's nothing on that paste to indicate anything is wrong (besides the syntax warning) |
| 21:19.45 | grabes222 | Davedan : yeah, my guess is you may be having NAT/Firewall related issues because you can't hear audio |
| 21:20.26 | Davedan | grabes222: I'm using 3 virtual box computers all behind the same router using bridged network so I don't think NAT should be a problem |
| 21:20.28 | JAMMAN2110 | Gah |
| 21:20.33 | Davedan | unless I need to turn NAT off |
| 21:20.35 | JAMMAN2110 | The countries biggest telephone network is down |
| 21:20.57 | grabes222 | Davedan : No that shouldn't be an issue then. |
| 21:21.42 | Davedan | ChannelZ: x-lite says registering... and then ready |
| 21:21.53 | ChannelZ | after you dial I mean |
| 21:22.28 | Davedan | ChannelZ: after I dial it doesn't say 'connected' just 'calling...' |
| 21:23.07 | grabes222 | Davedan : Do you have a codec incompatability issue? |
| 21:23.50 | ChannelZ | Do a Playback(demo-echotest) for instance in your dialplan, are you getting audio FROM * ? |
| 21:23.57 | Davedan | grabes222: I don't know |
| 21:24.16 | Davedan | ChannelZ: I don't understand. write this in the CLI? |
| 21:24.24 | ChannelZ | no in your extensions.conf |
| 21:24.27 | grabes222 | In xlite, what codecs do you have enabled, and in your sip.conf what do you have for allow |
| 21:24.53 | ChannelZ | right now it looks like you're doing a NoOp and then an Echo.. shove a Playback step in there so * plays something, see if you have audio going at least one way |
| 21:25.04 | grabes222 | x-lite: right click -> options -> advanced -> Audio Codecs |
| 21:25.08 | *** join/#asterisk xmitter (n=xmitter@c-24-21-212-187.hsd1.or.comcast.net) |
| 21:25.43 | Davedan | sip.conf http://dpaste.com/132941/ |
| 21:26.02 | Davedan | extensions.conf http://dpaste.com/132942/ |
| 21:29.20 | ChannelZ | insert "exten => 500,n,Playback(demo-echotest)" after your Verbose line, before the Echo |
| 21:30.19 | ChannelZ | I don't think Echo will work as the first application |
| 21:30.39 | Davedan | ChannelZ: work :) |
| 21:30.43 | Davedan | what did that do? |
| 21:31.09 | ChannelZ | well it's playing back a sound file first which sets up the media stream - I don't think Echo triggers the right things internally |
| 21:31.44 | Davedan | it's weird that the book doesn't explain to add your line but maybe it does later |
| 21:31.44 | Davedan | thanks |
| 21:32.05 | ChannelZ | It might be a bug, that very well might work in older versions |
| 21:32.09 | ChannelZ | in fact.. |
| 21:32.48 | ChannelZ | it might just be that you need to Answer() first |
| 21:33.31 | ChannelZ | Playback() automatically triggers an Answer if the channel hasn't been already. Echo apparently doesn't |
| 21:33.39 | Davedan | I need to answer in the command line? |
| 21:33.46 | ChannelZ | no |
| 21:33.49 | ChannelZ | in extensions.conf |
| 21:41.39 | HaMF | dlynes, the asterisk process is freezing right now |
| 21:42.25 | HaMF | I did a gdb backtrace, anything else? |
| 21:43.07 | *** join/#asterisk voipmonk (n=voipmonk@69.172.114.221) |
| 21:43.13 | HaMF | voipmonk :) |
| 21:43.25 | voipmonk | hey |
| 21:43.41 | HaMF | I have a neary accesible system with a "frozen" asterisk |
| 21:44.01 | HaMF | anything I can do now to do provide more debugging info? |
| 21:44.37 | drmessano | Which version of Asterisk is this? |
| 21:44.41 | voipmonk | proc? |
| 21:44.46 | voipmonk | what processor? |
| 21:45.31 | HaMF | 1.4.21 |
| 21:45.37 | jblack | drmessano: In the unlikelyhood that I don't see you before then, happy tuesday! |
| 21:45.54 | drmessano | lol.. |
| 21:46.12 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 21:46.20 | drmessano | HaMF: Zaptel version? |
| 21:46.48 | jaytee | just a few more years and he'll be able to join AARP |
| 21:47.01 | drmessano | Yeah, the big 33 |
| 21:47.07 | drmessano | Halfway there! |
| 21:47.23 | jblack | That's it? |
| 21:47.23 | *** join/#asterisk Tech_Travis (n=Administ@cpe-76-87-9-130.socal.res.rr.com) |
| 21:47.59 | jblack | Heh. I constantly get invitations from AARP. |
| 21:47.59 | jblack | A lot of the time I put "retired" for occupation, so when they buy their lists, they junk mail me. |
| 21:47.59 | drmessano | I liked being 22.. Called the doubel deuce.. now i'm just old |
| 21:48.32 | bcrisp | fk |
| 21:48.34 | drmessano | Running to the store.. bbiaf |
| 21:48.43 | HaMF | drmessano, 1.4.11 |
| 21:48.46 | bcrisp | some people riding quads in the neighborhood drove through my yard and killed my tree |
| 21:49.59 | ChannelZ | did they kill themselves too? |
| 21:50.25 | jblack | I met my ex-wife on august 15. I got married on august 15. my daughter was conceived on august 15. My ex-wife notified me of seperatation on august 15. We almost got divorced on august 15 (paperwork delayed it until aug 30). |
| 21:51.16 | bcrisp | ChannelZ unfortunately no |
| 21:51.29 | bcrisp | they denied everything of course |
| 21:51.35 | ChannelZ | damn. It must not have been a big tree |
| 21:51.38 | bcrisp | checking out survellance cameras |
| 21:51.52 | bcrisp | they side swiped it.. nah it was a small palm |
| 21:52.06 | bcrisp | drove all through my lawn (which im reseeding) |
| 21:52.28 | bcrisp | aggravating |
| 21:52.42 | ChannelZ | hope you got them on cam, call the po-po! |
| 21:52.57 | bcrisp | ill get em next time |
| 21:53.01 | bcrisp | im shopping for the cams now |
| 21:53.32 | ChannelZ | oh.. I thought you meant you had them and were checking out what they captured :) |
| 21:53.42 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:0:0:0:2) [NETSPLIT VICTIM] |
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| 21:54.23 | jblack | anyone seen katty today? |
| 21:57.35 | TommyBotten | jblack: /whowas katty |
| 21:59.10 | TJNII | !seen Katty |
| 21:59.14 | TJNII | ~seen Katty |
| 21:59.17 | infobot | katty <n=asterisk@mail.copi-rite.com> was last seen on IRC in channel #asterisk, 8h 8m 42s ago, saying: 'he's also eating okay'. |
| 22:04.30 | jblack | tjnii: Wow. Has the bot been upgraded to track emails, phone calls and instant messaging too? |
| 22:09.22 | grabes222 | Actually this appears to be specific to app ReceiveFax, and only when it destroys the channel. Any BT gurus here http://pastebin.ca/1713594 |
| 22:10.48 | TommyBotten | grabes222: Which problem are you having? |
| 22:11.31 | grabes222 | TommyBotten : Ast <= 1.6.0.14 faxing is fine, Faxing on any version higher I get a core dump after a successful fax |
| 22:13.03 | grabes222 | Last thing I see on the console is a double free corruption in libc |
| 22:13.37 | dlynes | jblack: actually, it's had the ~seen command for at least 5 years now |
| 22:14.25 | dlynes | grabes222: have you tried it in asterisk 1.6.1.8 or higher? |
| 22:14.54 | grabes222 | Yes, that back trace is from 1.6.1.11 |
| 22:15.09 | jblack | dlynes: I'm aware of seen commands. I'm not aware that seen has been updated to include contacts outside of an irc channel. |
| 22:15.21 | jblack | such as phone calls, emails, private messages, im, and such. |
| 22:15.35 | dlynes | jblack: oh..it hasn't |
| 22:16.00 | dlynes | jblack: /whowas is an irc command |
| 22:16.19 | jblack | drops the politeness for a moment |
| 22:16.48 | jblack | There was an emergency last night, and I'm checking to see if anyone has heard from katty at all via any method. Not just in a public channel. |
| 22:17.16 | dlynes | jblack: ah |
| 22:17.19 | jblack | So your help, while kind and very nice, at this particular moment is useless. But thanks anyways. =) |
| 22:19.24 | jblack | Ironically, her last public comment answered my question anyways. How's her pet ferret. =) |
| 22:19.25 | *** join/#asterisk levity (n=levity@unaffiliated/canuck) |
| 22:20.46 | jblack | I like cheese |
| 22:20.46 | dlynes | grabes222: odd...I've got it working just fine (both digium and soft-switch.org versions) |
| 22:21.58 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 22:22.15 | grabes222 | dlynes : It works fine on asterisk 1.6.0.14, but nothing higher |
| 22:22.38 | grabes222 | dlynes : Its strnage |
| 22:23.19 | dlynes | grabes222: yeah...works fine for me on asterisk 1.6.1.8 |
| 22:24.19 | dlynes | grabes222: however, i'm checking the results of your bt |
| 22:27.06 | dlynes | HaMF: have you been able to figure out what it's done just before it appears to freeze yet? |
| 22:27.32 | dlynes | HaMF: or can you pastebin a dmesg? |
| 22:28.34 | HaMF | dlynes, you wanted logs: gdb log: http://pastebin.com/m6392889a ; relevant part of dmesg http://pastebin.com/m738f2666 ; syslog: http://pastebin.com/m739d0086 ; console output (already frozen) http://pastebin.com/m55a254bb ; full log (lines repeating 10^6 times removed): http://pastebin.com/m61b8b87d |
| 22:29.16 | HaMF | I've just cleaned the full log (there were way too much repeating messages of "Avoiding inital deadlock"...) |
| 22:29.29 | dlynes | grabes222: btw...your bt isn't from 1.6.1.11 |
| 22:29.36 | dlynes | grabes222: it's from a beta version of 1.6.1.11 |
| 22:32.06 | Davedan | do I need to configure something other then adding the sip extension to let to soft phones talk? |
| 22:32.48 | dlynes | HaMF: Try following this email: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg84634.html |
| 22:34.16 | dlynes | HaMF: it seems you've got your card or your software misconfigured...i.e. it's configured as a t1, when it's an e1 |
| 22:34.21 | *** join/#asterisk m0e (n=eX@41.196.209.68) |
| 22:34.24 | Davedan | I can hear the echo test on both clients but when a client tries to call the other one I'm getting 'call failed, call not found' |
| 22:34.46 | Davedan | and no the CLI I'm getting: 'chan_sip.c:19546 handle_request_invite: Call from '1001' to extension '1000' rejected because extension not found.' |
| 22:35.01 | dlynes | Davedan: can you be more specific by actually giving us a log, instead of telling us in your own words? |
| 22:35.12 | HaMF | hmmm |
| 22:35.22 | dlynes | Davedan: and also pastebin a copy of your extensions.conf file |
| 22:35.41 | Davedan | http://dpaste.com/132941/ |
| 22:35.43 | dlynes | HaMF: you have noticed the multitude of warnings and errors in your log file, right? |
| 22:35.45 | Davedan | http://dpaste.com/132942/ |
| 22:35.59 | Davedan | dlynes: by log you mean the output of the CLI? |
| 22:36.01 | m0e | quick question.. what does the following mean "chan_dahdi.c:4668 handle_alarms: Detected alarm on channel 1: Red Alarm" |
| 22:36.20 | dlynes | Davedan: or /var/log/asterisk/full |
| 22:36.33 | dlynes | m0e: depends on what's on channel 1 |
| 22:36.38 | dlynes | m0e: is it a pri? |
| 22:36.41 | m0e | pstn line |
| 22:36.50 | m0e | nope.. just plain old pstn line |
| 22:37.16 | dlynes | m0e: probably means the pstn line's not connected, or you've got a pstn line connected into an fxs port (really bad), or something similar |
| 22:37.28 | Davedan | dlynes: under /var/log/asterisk I have event_log, queue_log and messages. what is relevant? |
| 22:37.40 | dlynes | Davedan: gimme the messages log, then |
| 22:37.40 | m0e | i have no fxs ports.. so atleast thats not it :) |
| 22:38.45 | m0e | hmm.. I'm getting this now "-- Reconfigured channel 1, FXS Kewlstart signalling" |
| 22:39.05 | dlynes | m0e: so then you're good to go |
| 22:39.20 | m0e | though in the chan_dahdi.conf I have this set "signalling=fxs_ks" |
| 22:39.24 | m0e | oh.. so no probs? |
| 22:39.32 | dlynes | m0e: that's all correct |
| 22:39.33 | HaMF | dlynes, what is the preceeding mail to the one you linked? |
| 22:39.42 | m0e | perfect, thanks a bunch :) |
| 22:39.45 | Davedan | dlynes: it's too long. I'm pasting the end of it http://dpaste.com/132972/ |
| 22:40.04 | dlynes | HaMF: This is the head email: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg84616.html |
| 22:40.05 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 22:40.23 | HaMF | thanks |
| 22:40.34 | dlynes | Davedan: Ok...I refuse to help |
| 22:40.43 | ChannelZ | Davedan: you don't have an extension 1000 defined |
| 22:40.44 | dlynes | Davedan: if you can't read error messages, you need some serious help |
| 22:41.00 | ChannelZ | or 1001 for that matter |
| 22:41.08 | dlynes | Davedan: and it ain't asterisk help you need....it's mental help |
| 22:41.46 | Davedan | ChannelZ: I need to define 1000 both on sip.conf and extensions.conf? |
| 22:41.48 | dlynes | Davedan: What does this line suggest to you? [Dec 13 17:38:01] WARNING[2353] pbx.c: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Verbose(1|Echo test application)) |
| 22:42.01 | ChannelZ | Davedan: Yes - the peers in sip.conf are NOT extensions |
| 22:42.12 | m0e | is there some document that describes what all the config files in /etc/asterisk are exactly? |
| 22:42.16 | ChannelZ | They are devices. You might have called it "1000" and "1001" but that means nothing |
| 22:42.25 | dlynes | m0e: *.conf-dist |
| 22:42.28 | jblack | dlynes: Are you the one that helped me with that logo? |
| 22:42.41 | m0e | thanks again :) |
| 22:42.41 | Davedan | ChannelZ: ok. I'll try to read about extensions |
| 22:42.41 | dlynes | jblack: which logo? |
| 22:42.41 | jblack | mow |
| 22:42.59 | dlynes | jblack: oh yeah...that's what you were asking me about last night...soemthing to do with mow.svg |
| 22:43.07 | dlynes | jblack: i had no freaking clue what you were talking about |
| 22:43.16 | dlynes | jblack: and i still don't :) |
| 22:43.27 | dlynes | jblack: but what about svg? |
| 22:43.38 | dlynes | jblack: You're just wanting to edit it? |
| 22:43.39 | *** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) [NETSPLIT VICTIM] |
| 22:43.59 | HaMF | well I guess I'll try removing crc4 although the PSTN states that they do support crc4 |
| 22:44.03 | jblack | No, just wated to show you what I ended up with |
| 22:44.11 | jblack | mow.merconline.com |
| 22:44.20 | dlynes | jblack: oh...yeah...it was obviously not me you were talking to, though |
| 22:44.53 | *** join/#asterisk brut- (n=brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) [NETSPLIT VICTIM] |
| 22:44.58 | jblack | Oh, ok |
| 22:44.58 | dlynes | jblack: that being said, the logo looks good enough |
| 22:45.15 | dlynes | jblack: but your username and password fields extend all the way over into the content pane |
| 22:45.32 | jblack | yeah. thanks |
| 22:45.33 | dlynes | jblack: they overlap 'Once' logged in |
| 22:46.02 | dlynes | jblack: didn't know if you could see that or not...figured it might look ok in IE, but not in Firefox |
| 22:46.38 | dlynes | jblack: I guess you're using Inkscape? |
| 22:46.48 | jblack | yeah. I don't have IE. |
| 22:46.59 | ChannelZ | hurray! |
| 22:47.05 | dlynes | jblack: Inkscape's a pretty cool svg editor |
| 22:47.16 | dlynes | jblack: well...pretty cool for dtp in general |
| 22:48.09 | dlynes | Davedan: You are seeing all those errors and warnings, right??? |
| 22:48.41 | jblack | Yeah. I like inkscape. THat's what I use. |
| 22:49.09 | dlynes | jblack: yeah...I use gimp and Pixel for the graphics |
| 22:49.25 | dlynes | jblack: but the author of Pixel is pissing me off....hasn't updated the project in over a year |
| 22:49.51 | dlynes | jblack: Paid him for the program, and he hasn't updated it since |
| 22:49.59 | Davedan | dlynes: right |
| 22:50.26 | dlynes | Davedan: it means all those parts in your dialplan where you see a '|', you need to replace with a ',' |
| 22:50.46 | TommyBotten | grabes222: I've been seeing the same thing. But not using T.38 though |
| 22:50.50 | dlynes | Davedan: because you're using a 1.2 or an old 1.4 dialplan in a new 1.4 or 1.6 |
| 22:50.56 | HaMF | dlynes, I'll continue tomorrow. Thanks for your support! |
| 22:51.12 | dlynes | HaMF: you're welcome...so did that help you with your issue at all? |
| 22:51.30 | HaMF | I disabled crc4 on span 1. |
| 22:51.36 | Davedan | dlynes: it says 'warning' so I thought I can ignore this for now |
| 22:51.52 | dlynes | Davedan: that's labelled as a warning, but it's actually an error |
| 22:51.55 | HaMF | let's see if the problem occurs again... |
| 22:51.57 | ChannelZ | you can but , looks nicer than | anyway |
| 22:52.01 | grabes222 | TommyBotten : I am not using T.38 on this machine |
| 22:52.07 | Davedan | dlynes: ok. changed that |
| 22:52.11 | dlynes | Davedan: you should actually do all the replacements |
| 22:52.21 | HaMF | and tomorrow I'll have a closer look at which channel just died |
| 22:52.27 | dlynes | Davedan: some of the old code just won't work at all |
| 22:52.28 | TommyBotten | grabes222: Ok. Just thought you should know when tracking down the bug |
| 22:52.34 | dlynes | HaMF: ah...thought it was something more than that |
| 22:52.38 | TommyBotten | grabes222: I'm also having the same issue on .1.12-rc1 |
| 22:52.44 | dlynes | HaMF: like the jumpers were set wrong on the card, or something |
| 22:53.20 | HaMF | I did double check the jumpers before installing the cards so I'm pretty sure they are set correctly |
| 22:53.26 | Davedan | dlynes: I'm following the oficial book. had no idea it's outdated |
| 22:53.40 | dlynes | Davedan: it's been outdated as of about 1.4.26 |
| 22:53.49 | dlynes | Davedan: with respect to the ',' |
| 22:54.19 | dlynes | Davedan: the '|' has been deprecated for a while |
| 22:54.25 | voipmonk | back |
| 22:54.28 | dlynes | Davedan: it's just that it's obsoleted now |
| 22:54.46 | dlynes | Davedan: there's a switch you can enable now, though...but at the same time....why? |
| 22:55.44 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
| 22:57.13 | HaMF | so. good night :) |
| 22:57.22 | dlynes | HaMF: good luck |
| 22:57.35 | HaMF | thanks dlynes |
| 22:58.30 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 23:01.05 | Davedan | ChannelZ, dlynes: I've managed to make a call between two phones. now I need to understand how I did it :) |
| 23:01.06 | Davedan | thanks |
| 23:01.53 | ChannelZ | by adding extensions to extensions.conf that dial SIP/xxxx |
| 23:02.38 | grabes222 | TommyBotten : If you use 1.0.0.14 are you ok? |
| 23:02.49 | grabes222 | TommyBotten : Sorry 1.6.0.14 |
| 23:05.32 | drmessano | jblack: I made the mistake of getting married on Valentines day, 2 or 3 marriages back |
| 23:05.41 | drmessano | scratches head to remember |
| 23:06.15 | jblack | how many have you had? |
| 23:06.34 | drmessano | takes off his shoes |
| 23:06.37 | drmessano | Gimme a minute |
| 23:07.29 | drmessano | Carry the 2 |
| 23:07.29 | jblack | I want mcdonalds, but I don't want to get dressed, and I probably smell bad. |
| 23:07.29 | drmessano | Carry my TV out the door |
| 23:07.30 | drmessano | Umm |
| 23:07.30 | ChannelZ | Have you at least figured out that marriage apparently isn't your destiny? |
| 23:07.40 | ChannelZ | jblack: drivethrus |
| 23:07.54 | jblack | My driver side window doesn't work. :( |
| 23:08.13 | ChannelZ | back in |
| 23:08.26 | jblack | I can't drive from the passenger seat. |
| 23:08.42 | jblack | It's either impossible, or I just lack the skill. |
| 23:09.10 | jblack | also, it's cold out. |
| 23:10.04 | drmessano | ChannelZ: Yes, much like skateboarding |
| 23:10.51 | chuckf | jblack: its mcdonalds. you won't be the worst smelling thing there |
| 23:11.11 | jblack | True. those employees can get rank |
| 23:11.26 | chuckf | not to mention the 'food' itself |
| 23:11.31 | ChannelZ | just look at peopleofwalmart.com - how you dress seems to be not important |
| 23:12.10 | jblack | wtf?? http://www.peopleofwalmart.com/?p=7680 |
| 23:13.49 | grabes222 | TommyBotten : What kind of hardware are you using for PSTN side? |
| 23:14.52 | grabes222 | argh, I am going to fen lost it *** glibc detected *** /usr/sbin/asterisk: double free or corruption (!prev): 0x00000000025c9850 |
| 23:24.28 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:35.10 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 23:35.37 | *** join/#asterisk levity (n=levity@unaffiliated/canuck) |
| 23:41.08 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |