| 00:00.40 | ModFather | WIMPy what is the difference between : contactdeny= and deny= |
| 00:03.02 | WIMPy | The sample says that the contact* versions are only for registrations. |
| 00:07.17 | ModFather | when i add to the client [101] deny=0.0.0.0 |
| 00:07.29 | ModFather | is for registrations and all the rest? |
| 00:19.54 | ModFather | WIMPy can i unload queue rules without unload queue ? |
| 00:20.23 | WIMPy | unload? |
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| 00:22.18 | ModFather | i am getting some notices |
| 00:22.28 | ModFather | app_queue.c:8551 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. |
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| 00:23.40 | WIMPy | Asterisk always checks the mtime of config files to avoid re-reading any that haven't changed. |
| 00:24.30 | ModFather | so what do you suggest to remove that notice? |
| 00:24.57 | WIMPy | Don't do a reload if you haven't changed anything. |
| 00:25.16 | ModFather | i am changing other stuff that needs reload |
| 00:26.40 | WIMPy | Then take the notice as a chance to thank whoever implemented that optimization for saving resources. |
| 00:26.53 | lvlinux | Hmmm, I wonder how come chan_sip is not auto loading with autoload=yes in modules.conf? |
| 00:27.33 | lvlinux | is there somewhere else that messes with module loading? this is 13.4 with pjsip and chan_sip, but I only want to use chan_sip for now. |
| 00:33.46 | ModFather | WIMPy can somehow disable/unload dundi and iax2 ? |
| 00:34.20 | lvlinux | ModFather: modules.conf is where you do that. Or possibly just remove the config files for them. |
| 00:34.36 | ModFather | lvlinux i have already add it to module.conf |
| 00:34.39 | ModFather | but still exist |
| 00:34.40 | lvlinux | I think my problem is somethign wrong in my sip.conf... |
| 00:34.50 | lvlinux | you have added it to noload? |
| 00:34.52 | WIMPy | There's at least one word missing. Do you want to know how? |
| 00:35.17 | ModFather | noload => iax2.so |
| 00:35.17 | ModFather | noload => dundi.so |
| 00:35.50 | WIMPy | You do realize that it is only used on startup? |
| 00:36.16 | ModFather | yes and ? |
| 00:36.50 | WIMPy | ooops. Those are not existing module names. |
| 00:37.17 | lvlinux | pbx_dundi.so |
| 00:37.34 | lvlinux | chan_iax2.so |
| 00:37.45 | ModFather | thanks man |
| 00:37.50 | ModFather | i was reading inside lib |
| 00:37.50 | lvlinux | or rather "noload => chan_iax2.so" etc. |
| 00:38.07 | ModFather | thanks |
| 00:38.31 | WIMPy | You don't need the ".so". |
| 00:38.40 | lvlinux | ah ok good to know |
| 00:39.08 | lvlinux | i knew you didn't on the commandline when doing module load/unload but didn't know it could be ommited in the config files too. |
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| 02:14.46 | wyoung | WIMPy!!!!!!!!11 |
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| 08:51.05 | bulkorok | anyone setup homer5 ?! |
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| 10:33.08 | rl1 | a client of ours has asked wether they can hide their callerid. I wonder how do other SIP providers provide this functionallity. Do they call a number with a prefix or do they just INVITE from anonymous? How will asterisk authenticate a peer with anonymous from and contact headers? |
| 10:42.06 | rl1 | I suppose that will be like: INVITE - 401 - INVITE with authorization - call. Every time which doesn't seem to be convinient |
| 10:44.17 | rl1 | or with a prefix, like *99*18005557788 which looks like old POTS shit |
| 10:44.55 | rl1 | duh |
| 10:47.28 | YasT | You could use the callerid paramater in sip.conf |
| 10:48.57 | rl1 | We provide trunks with registration from asterisk to our users who want to hide their callerid |
| 10:49.02 | rl1 | that is the problem |
| 10:49.12 | rl1 | it's how we shound auth them |
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| 10:54.39 | mjamilfarooq | hey |
| 10:55.08 | mjamilfarooq | anybody can tell me the difference between command IAXpeers and IAXpeerlist |
| 10:55.12 | mjamilfarooq | ?? |
| 11:02.28 | mjamilfarooq | anyone?? |
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| 11:50.21 | cIuFuLiCi | hi |
| 11:50.31 | cIuFuLiCi | anyone on !? |
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| 13:27.03 | ModFather | hi [TK]D-Fender |
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| 13:49.01 | ModFather | Something is going wrong with my queue |
| 13:49.44 | rl1 | what is it |
| 13:50.01 | WIMPy | It works? |
| 13:50.40 | rl1 | LOL |
| 13:52.14 | ModFather | its infinity |
| 13:52.19 | ModFather | i have set retry and timeout |
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| 13:53.38 | bipolar | Is anyone using an R800 analog failover appliance to switch fxs ports going to internal phones instead of the usual case of using it to swich incoming PSTN lines? I have a redundant asterisk install and internal equpipment I'd like to swich automaticaly during failover. |
| 13:56.42 | [TK]D-Fender | ModFather: Fix your description of what's wrong |
| 13:57.30 | ModFather | [TK]D-Fender when i call my external number, the call is infinity until i drop it manually, i have set on queue.conf ( retry=2 ) and ( timout=30 ) and it doesnt drop the line |
| 13:57.54 | [TK]D-Fender | You didn't read what those MEAN |
| 13:58.18 | [TK]D-Fender | that is the delay BETWEEN agent call-outs, and the DURATION of the ringing to them |
| 13:58.37 | [TK]D-Fender | That is NOT the amount of time you can sit in the queue itself |
| 13:58.38 | rl1 | what is 'call is infinity' define that |
| 13:58.55 | [TK]D-Fender | rl1: Just poor english. He's expecting to QUIT the queue after a timeout |
| 13:59.16 | [TK]D-Fender | ModFather: "core show application queue" <------ |
| 14:00.53 | ModFather | yes forgive me for my poor english |
| 14:01.28 | rl1 | [TK]D-Fender, maybe you can help me. Do you know how do sane VoIP providers provide callerid hiding for their customers? Is it a prefix they dial with or just send INVITEs from anonymous? |
| 14:02.00 | [TK]D-Fender | ModFather: I got what you were asking. Read the app's instructions |
| 14:02.39 | [TK]D-Fender | rl1: I've seen both. There isn't a standard. I've also seen the more normal "privacy" headers used in presentation. |
| 14:02.41 | ModFather | timeout |
| 14:02.41 | ModFather | <PROTECTED> |
| 14:02.42 | ModFather | <PROTECTED> |
| 14:02.48 | [TK]D-Fender | setcallerpres |
| 14:02.53 | [TK]D-Fender | Or whatever the new function is |
| 14:04.51 | WIMPy | CALLERID(num-pres) |
| 14:05.07 | jeffspeff | anyone have experience connecting a NEC UX5000 to an asterisk box via SIP? i'm just wanting to do exten to exten dialing. I know how to configure this in asterisk, but no clue how to configure the NEC box. |
| 14:05.23 | rl1 | with num-pres you should send rpid which I don't use |
| 14:05.34 | ModFather | [TK]D-Fender i dont see anything related to core show application queue |
| 14:05.43 | ModFather | [TK]D-Fender i dont see anything related to what i need on core show application queue |
| 14:05.47 | WIMPy | Who uses rpid? |
| 14:05.47 | [TK]D-Fender | ModFather: read it AGAIN |
| 14:05.57 | ModFather | [TK]D-Fender okey buddy |
| 14:06.14 | [TK]D-Fender | WIMPy: Many ITSP's use that for passing the CID where they expect the username in the Fron: |
| 14:06.28 | rl1 | WIMPy, uhh. Where do you think number pres. is defined in an INVITE message |
| 14:06.44 | WIMPy | What about PAI? |
| 14:07.00 | rl1 | who uses PAI? |
| 14:07.03 | WIMPy | hasn't seen RPID for ages. |
| 14:07.20 | rl1 | and asterisk doesn't have support for p-asserted |
| 14:07.25 | WIMPy | Everyone I talk to. |
| 14:07.32 | WIMPy | It sure has. |
| 14:07.46 | rl1 | and you have to manually add it with SipAddHeader |
| 14:07.58 | rl1 | it does not |
| 14:08.06 | WIMPy | Read the docs again. |
| 14:08.11 | rl1 | link? |
| 14:08.17 | WIMPy | Or are you using 1.2? |
| 14:08.22 | WIMPy | sip.conf.sample |
| 14:08.39 | rl1 | 11.8 actually |
| 14:08.58 | rl1 | ;sendrpid = pai ; Use the "P-Asserted-Identity" header |
| 14:09.03 | rl1 | hmm. you were right |
| 14:09.09 | rl1 | okay sorry. |
| 14:09.27 | WIMPy | You should check for new features every few years :-)) |
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| 14:18.02 | ModFather | [TK]D-Fender n: No retries on the timeout; will exit this application and go to |
| 14:18.02 | ModFather | <PROTECTED> |
| 14:18.29 | [TK]D-Fender | ModFather: TIMEOUT <------------- |
| 14:19.26 | ModFather | yes after some seconds, it will timeout and go to the next step |
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| 14:21.33 | [TK]D-Fender | correct\ |
| 14:36.33 | ModFather | [TK]D-Fender fixed ;) |
| 14:38.54 | rl1 | SIP has so many crap headers for one fucking purpose |
| 14:39.08 | rl1 | like FROM, Contact, rpid, PAI goddamn it |
| 14:39.13 | Phil-Work | heh |
| 14:39.24 | Phil-Work | this has been my life for the past 2 weeks |
| 14:39.36 | rl1 | all those for one simple thing... cgpn. |
| 14:40.11 | rl1 | That's why I think H323 is more convinient for VoIP |
| 14:40.15 | Phil-Work | and it seems our carriers have no consistency about which of those they use for the CLI that gets presented when the call ends up on the PSTN |
| 14:40.56 | rl1 | or SIP-T |
| 14:41.38 | rl1 | I think sip is great for CO-end user signaling not for backbone SSP-SSP signaling |
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| 14:46.48 | paws | when asterisk emails me a voicemail i receive it from "Asterisk PBX voicemail@company.com", how can i change that name, Asterisk PBX ? |
| 14:47.02 | paws | I am using asterisk 13 on freepbx |
| 14:48.23 | rl1 | <PROTECTED> |
| 14:48.23 | rl1 | <PROTECTED> |
| 14:48.43 | Phil-Work | paws, I think it's fromstring |
| 14:48.57 | Phil-Work | see http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf |
| 14:52.26 | paws | do i need the " " |
| 14:52.40 | paws | fromstring=Company Voicemail |
| 14:53.13 | Phil-Work | I'd think not |
| 14:53.16 | Phil-Work | but try it and see |
| 14:54.08 | paws | nice |
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| 15:06.10 | WIMPy | r11: If it only was that header... SIP has lots incompatible methods for most features. |
| 15:25.49 | rl1 | WIMPy, yeah. I don't understand why do people love SIP though |
| 15:26.14 | rl1 | it's crap. It has a p2p nature. It shouldn't be used in anything than skype or crap like that |
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| 15:29.20 | rl1 | it lacks so many ISUP feauters like whether the called party number is national or international (you have to figure that out yourself by CC prefixes) |
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| 15:38.11 | [TK]D-Fender | SIP is routeable, multi-domain, etc |
| 15:38.45 | [TK]D-Fender | Not sure of SLA advantage. What does H.323 offer on that front? |
| 15:40.32 | mjordan | prefers not to fight the world |
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| 15:58.57 | orn | At some point, adding queue members that are SIP trunk members stopped working for me (likely an Asterisk update). Does anyone know if this is still possible? If so, how? |
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| 16:01.02 | [TK]D-Fender | Nothing changed, and we won't know what's wrong with your setup until you show us |
| 16:01.08 | [TK]D-Fender | Pastebin is your friend. |
| 16:01.10 | [TK]D-Fender | ~pb |
| 16:01.11 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:01.12 | [TK]D-Fender | ^^^ |
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| 16:06.27 | millsu2 | I have an issue where asterisk is showing the wrong state for some devices. How does astersik get the device presence? Does it send a request for the presence, or does it only change presence of a device if thedevice sends a change? |
| 16:06.50 | [TK]D-Fender | It has nothing to do with what a device thinks |
| 16:06.55 | paws | why is it that when i call my own number instead of my IVR turning on i get "all circuits are busy now" ? |
| 16:07.01 | [TK]D-Fender | * presence is based on the calls it actually has in progress, etc |
| 16:07.21 | paws | when i call from my pbx to my main number i get all circuits are busy... but when i call from outside it works fine |
| 16:07.41 | [TK]D-Fender | paws: that message means nothing, and what happens when you call... well we'd have to look at what that call was doing. |
| 16:07.57 | [TK]D-Fender | paws: Nothing say any IVR is involved anywhere yet. |
| 16:09.04 | millsu2 | ok, so sip show inuse for an extension shows all 0's and there doesn't seem to be any channels connected to the device in sip show channels, but the device state is still inuse. Is there anything else to check to find out why it is in use? |
| 16:09.16 | paws | http://pastebin.ca/3208487 |
| 16:09.31 | paws | thats what asterisk is showing me |
| 16:10.56 | [TK]D-Fender | paws: that is FreePBX which is not supported here. Please resume in #freepbx |
| 16:11.11 | [TK]D-Fender | millsu2: Show us |
| 16:11.22 | [TK]D-Fender | millsu2: hint dump & channel dump |
| 16:12.08 | paws | ok sorry |
| 16:12.14 | millsu2 | as in core show hints and core show channels? |
| 16:12.34 | [TK]D-Fender | yes |
| 16:15.19 | millsu2 | ok. it will need to find an extension that isn't actually in use, so it will take me some time. |
| 16:16.45 | orn | [TK]D-Fender: Here's the excerpt: http://pastebin.com/6dnzuZH4 -- basically the extension never starts ringing (I see no attempts from Asterisk) |
| 16:17.45 | *** join/#asterisk BBone_ (~Thunderbi@s75-157-233-55.bc.hsia.telus.net) |
| 16:18.25 | [TK]D-Fender | orn: check your STATE for the peer |
| 16:18.34 | [TK]D-Fender | odds aree it is (or was) unreachable at that time |
| 16:19.27 | orn | it's not being monitored. is qualify a requirement? (the behavior is consistently like this, and the trunk is currently reachable) |
| 16:20.52 | [TK]D-Fender | do a reload on the queue |
| 16:21.39 | orn | already tried that, as well as a pbx restart |
| 16:22.10 | orn | the queue is joined, but no attempts to call out are made |
| 16:22.54 | *** join/#asterisk znf (~ibm86@unu.card-sharing.eu) |
| 16:23.19 | [TK]D-Fender | hrm |
| 16:23.33 | [TK]D-Fender | I'd double check spelling, etc as I'm pretty sure those aren't the real names |
| 16:35.38 | ModFather | [TK]D-Fender is master on asterisk |
| 16:36.41 | [TK]D-Fender | ~[TK]D-Fender |
| 16:36.41 | infobot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 16:37.01 | craigify | a google proxy eh? |
| 16:37.31 | orn | spelling quadruple checked... :/ |
| 16:37.57 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 16:38.31 | *** part/#asterisk d-skull (~digifiv5e@unaffiliated/daynaskully) |
| 16:38.40 | ModFather | ~ModFather |
| 16:39.04 | ModFather | poor english, make TK angry all the time, and spamming all the time. |
| 16:39.07 | [TK]D-Fender | ~infobot |
| 16:39.07 | infobot | [infobot] A program on the IRC that helps users, ask it to do something by putting a ~ and then say a command! |
| 16:39.18 | [TK]D-Fender | MY BITCH |
| 16:39.23 | ModFather | hahahahahaha |
| 16:40.41 | orn | any ideas on further debugging steps Zen Master of the blatantly obvious? ;-) |
| 16:41.26 | ModFather | he is native italian also |
| 16:41.34 | orn | Is he? |
| 16:41.51 | ModFather | no man, i am just kidding |
| 16:42.05 | ModFather | he is not from this planet |
| 16:42.10 | orn | lol |
| 16:47.53 | orn | [TK]D-Fender: If I put any other member in the queue, it starts working |
| 16:49.07 | orn | it also works if I use a Local channel instead of SIP/siptrunk, so I guess I'll just use this as a workaround |
| 16:49.51 | [TK]D-Fender | I'd recommend that regardless |
| 16:50.27 | WIMPy | Yes, allows for many evil features. |
| 16:50.46 | orn | haha |
| 16:51.00 | orn | care to elaborate, as I am a fan of evil features? ;) |
| 16:51.56 | WIMPy | E.g. calling different ()number of) phones depending on the amount of callers in the queue. |
| 16:52.22 | orn | interesting :D |
| 16:52.43 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 16:52.48 | WIMPy | Or maximum waiting time. Adding a big firealarm bell usually makes people more motivated to take calls :-) |
| 16:54.01 | orn | haha i like your thinking |
| 16:59.20 | millsu2 | Here is my dump of hints and channels: http://pastebin.ca/3208562. The device 42971 is not being used, but shows as InUse inthe hints. |
| 17:00.36 | [TK]D-Fender | 42971@ext-local : SIP/42971&Custom:DND State:InUse Watchers 0 |
| 17:00.47 | [TK]D-Fender | that is a COMPOSITE hint including an AstDB value |
| 17:00.53 | [TK]D-Fender | Which you should clearly be looking at... |
| 17:01.04 | millsu2 | the Custom:DND stat is NOT_INUSE inastdb: /CustomDevstate/DND42971 : NOT_INUSE |
| 17:02.40 | [TK]D-Fender | nTechnially that could be a desync as the hint custom state... isn't necessarily an AstDB valuer |
| 17:02.56 | [TK]D-Fender | they may be held in sync, but might have fallen out do to bug, etc |
| 17:03.02 | [TK]D-Fender | not sure how to check those states |
| 17:03.34 | [TK]D-Fender | I'd try making 2 separate hints for the 2 components and see what they report |
| 17:06.10 | *** join/#asterisk millsu2 (~bgardner@69-195-66-124.unifiedlayer.com) |
| 17:06.49 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 17:06.51 | millsu2 | NoOp(${DEVICE_STATE(SIP/42971)}) produces NoOp("SIP/45910-000fc826", "INUSE") in new stack, so it appears that it's not the custom hint that is reporting as inuse |
| 17:07.59 | millsu2 | does the DEVICE_STATE command use the composite hint? |
| 17:09.12 | WIMPy | Device state ist exactely that, i.e. the raw thing as in 'sip show inuse'. |
| 17:09.14 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 17:09.17 | millsu2 | this is the hint definition in the dialplan: 42971,hint,SIP/42971&Custom:DND42971,CustomPresence:42971 |
| 17:09.53 | WIMPy | ... aaaand a 3rd value. |
| 17:09.57 | ModFather | [TK]D-Fender i need your help again , http://pastebin.ca/3208585 |
| 17:10.13 | [TK]D-Fender | millsu2: THAT ... is wierd |
| 17:10.22 | [TK]D-Fender | millsu2: and when you create a regulr hint? |
| 17:10.31 | ModFather | the queues working fine, after 15 seconds it change queue group, but after that it doesnt prompt me to enter on voicemail, it just saying goodbye and hangup |
| 17:10.33 | millsu2 | sip show inuse for that device is 0/0/0 |
| 17:11.05 | millsu2 | most of the devices are working, but a few of them seem to get stuck as inuse |
| 17:11.24 | [TK]D-Fender | millsu2: make a regular hit and see what it says, |
| 17:11.27 | [TK]D-Fender | hint |
| 17:12.34 | millsu2 | 42971@from-internal-custom: SIP/42971 State:InUse |
| 17:12.49 | *** join/#asterisk Kaian (~kaian@6.62-99-78.static.clientes.euskaltel.es) |
| 17:14.00 | millsu2 | so i'm wondering what else in asterisk marks a device as inuse if sip show inuse and sip show channels show nothing for this device |
| 17:14.36 | WIMPy | Good question. I would have assuemed they use the same data. |
| 17:15.54 | millsu2 | it sometimes fixes itself after a few hours and sometimes after a day or so. i wonder if I have hit a bug in asterisk then. |
| 17:16.22 | WIMPy | Likely. |
| 17:17.17 | ModFather | WIMPy do you know why it doesnt parse to voicemail? http://pastebin.ca/3208585 |
| 17:17.46 | ModFather | it goes to sales-queue, after 15seconds goes to cs-queue, and then it says GoodBye and hangup |
| 17:18.01 | [TK]D-Fender | I don't see the cALL |
| 17:18.12 | [TK]D-Fender | You can show me dialplan.. doesn't mean it's executing... |
| 17:18.22 | WIMPy | Yes. Look at what's actually happening. |
| 17:19.48 | WIMPy | Although I find AMI much more usefull than the *CLI as you will also see what variables the appications set on return. |
| 17:22.47 | ModFather | SIP/2.0 405 Method Not Allowed |
| 17:23.03 | ModFather | [Oct 20 17:22:30] WARNING[10256][C-00000001]: app_voicemail.c:6497 leave_voicemail: No entry in voicemail config file for 'mainvm' |
| 17:25.00 | [TK]D-Fender | Well that looks pretty clear.... |
| 17:25.44 | [TK]D-Fender | exten => 5555,4,Voicemail(mainvm) <- last I heard you cannot use non-numeric box values |
| 17:26.28 | ModFather | http://pastebin.ca/3208615 voicemail.conf |
| 17:26.45 | [TK]D-Fender | mainvm is NOT a BOX NUMBER |
| 17:26.56 | [TK]D-Fender | [13:25][TK]D-Fenderexten => 5555,4,Voicemail(mainvm) <- last I heard you cannot use non-numeric box values |
| 17:27.10 | ModFather | in the voicemail i had it as [mainvm] |
| 17:27.26 | [TK]D-Fender | And you aren't using [default] and have failed to mention the VM context name as well |
| 17:27.39 | [TK]D-Fender | ModFatherin the voicemail i had it as [mainvm] <- that is a VM Context, not a box # |
| 17:27.49 | [TK]D-Fender | "core show application voicemail" <- |
| 17:27.58 | [TK]D-Fender | You need to serioulsy learn to read your apps instructions.... |
| 17:28.01 | ModFather | i have box inside context |
| 17:28.10 | ModFather | dont act like this man, its so rude |
| 17:28.28 | [TK]D-Fender | exten => 5555,4,Voicemail(mainvm) <- <- I don't see a box # here. You are supposed to be telling it which box |
| 17:28.37 | [TK]D-Fender | That doesn't say "100" |
| 17:28.49 | ModFather | yes but on the BOOK it has an example like this not a number |
| 17:29.07 | [TK]D-Fender | Show me the page |
| 17:29.18 | [TK]D-Fender | I'm curious to see if they did something silly |
| 17:29.34 | [TK]D-Fender | But regardless... always read the app instructions direct and not just copy a sample |
| 17:29.38 | ModFather | sorry it has text and number |
| 17:29.39 | [TK]D-Fender | otherwise you have no idea what you're passing |
| 17:29.40 | *** join/#asterisk ryanxm (~ryan@unaffiliated/ryanxm) |
| 17:30.00 | [TK]D-Fender | Go re-read the page.... and more specifically the apps instructions |
| 17:30.04 | [TK]D-Fender | app =king |
| 17:30.41 | WIMPy | [TK]D-Fender: i would have sworn I already told you that mailbox names don't have to be numeric. |
| 17:30.42 | ModFather | http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue |
| 17:30.58 | ModFather | in this example it has s1234 |
| 17:31.02 | ModFather | instead of numbers |
| 17:31.22 | WIMPy | You just shouldn't call VoiceMailMain without the name as it will be pretty hard to enter a non-numeric name when it prompts you. |
| 17:31.46 | [TK]D-Fender | that is not "the book" |
| 17:32.00 | [TK]D-Fender | That is an ANCIENT WIKI describing syntax that's 10 years old |
| 17:32.06 | WIMPy | (For the non-source code hackers "pretty hard" = "impossible") |
| 17:32.22 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 17:32.36 | [TK]D-Fender | exten => 129,2,Queue(example_queue|tT|||300) ;dont set n option until really needed |
| 17:32.44 | [TK]D-Fender | | isn't a valid delimiter either since 1.6+ |
| 17:32.51 | [TK]D-Fender | it SHOULD ahve been removed in 1.4 as well.. |
| 17:33.02 | [TK]D-Fender | So be VERY careful what you take from there |
| 17:33.17 | [TK]D-Fender | As I said... read what the app says to use |
| 17:33.22 | [TK]D-Fender | not just some sample you find there |
| 17:33.31 | ryanxm | Any one here have exp in migrating from one box to another? I've run into a snag and hope someone can help |
| 17:33.32 | [TK]D-Fender | Things change |
| 17:33.39 | WIMPy | And the really good news is that the length limit for mailbox names seems to have been increased some time ago. I remember not being able to fit a full phone number there. |
| 17:33.51 | [TK]D-Fender | ryanshow us what you've got... |
| 17:33.58 | [TK]D-Fender | ryanxm: show us what you've got... |
| 17:34.13 | ryanxm | thank [TK]D-Fender, sec while i write |
| 17:38.14 | *** join/#asterisk superscrat (asanders@nat/digium/x-nnfyiclrvukffbvx) |
| 17:39.49 | ModFather | https://wiki.asterisk.org/wiki/display/AST/Configuring+Voice+Mail+Boxes |
| 17:40.30 | ryanxm | Ok I am migrating from Box1 to Box2. On box1 I did a full backup via the freepbx/asterisk interface and then on box2 did an import. All data looks to be correct and imported correctly, mysql tables, voicemails, user settings, etc. However I am not able to register a device (phone or software) to a migrated extension (say 4002). I will get a bunch of 'Added contact 4002 to AOR in the log, but no 'Endpoint 4002 is reachable' The strange thing is, whe |
| 17:40.53 | ryanxm | --sorry may be a newbie question or something dumb I missed |
| 17:41.24 | WIMPy | Your message was cut off. |
| 17:41.39 | WIMPy | But that looks like it's a question for #freepbx anyway. |
| 17:42.38 | ryanxm | well it looks like in the asterisk log it is registering (getting a successfullauth) but no endpoint becomes reachable |
| 17:42.49 | ryanxm | but if you think i should move this to #freepbx I will |
| 17:43.17 | WIMPy | If you use theyr backup function and it doesn't work, that would seem to make sense. |
| 17:43.55 | ryanxm | true, I will try with them then, thanks for the help |
| 17:45.31 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-kqalgcicsbgaetwe) |
| 17:59.17 | *** join/#asterisk Zogot (~Adium@90-145-117-39.bbserv.nl) |
| 18:01.40 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:02.00 | *** join/#asterisk cIuFuLiCi (~caius@dyn-86.106.35.88.tm.upcnet.ro) |
| 18:06.50 | cIuFuLiCi | anyone on ? |
| 18:08.12 | craigify | maybe |
| 18:08.17 | cIuFuLiCi | :) |
| 18:08.17 | craigify | :) |
| 18:08.30 | cIuFuLiCi | well i need some help with my asterisk :) |
| 18:08.38 | cIuFuLiCi | by the way i am a n00b |
| 18:08.52 | cIuFuLiCi | so if you wana help me be gentle with me |
| 18:09.23 | cIuFuLiCi | i have a problem with asterisk-guy |
| 18:09.35 | cIuFuLiCi | i can`t access the web page |
| 18:09.41 | cIuFuLiCi | i get a 404 error |
| 18:09.43 | cIuFuLiCi | :-/ |
| 18:10.26 | craigify | I'm not even sure I know what asterisk-guy is |
| 18:10.50 | cIuFuLiCi | :) |
| 18:11.00 | cIuFuLiCi | asterisk web interface |
| 18:11.01 | cIuFuLiCi | :D |
| 18:11.38 | cIuFuLiCi | http://www.asteriskguru.com/tutorials/asterisk_gui.html |
| 18:11.45 | cIuFuLiCi | this is asterisk-gui |
| 18:11.48 | robmal | I've never heard of it either. |
| 18:12.09 | robmal | "Asterisk GUI is no longer maintained and should not be used" this one? |
| 18:12.23 | cIuFuLiCi | yep that one :) |
| 18:12.40 | cIuFuLiCi | ok then and what should i use insted ? |
| 18:13.00 | *** join/#asterisk kritzikratzi (~kritzikra@cpe90-146-150-86.liwest.at) |
| 18:13.03 | craigify | I think their replacement product offering is asteriskNOW |
| 18:13.15 | cIuFuLiCi | i heard |
| 18:13.20 | cIuFuLiCi | but i use ubuntu |
| 18:13.20 | craigify | if you're completely new to asterisk, you might check out freepbx |
| 18:13.34 | cIuFuLiCi | :-/ |
| 18:13.54 | cIuFuLiCi | freebxp does have a web interface ? |
| 18:13.55 | craigify | I mean, it depends on what you want |
| 18:13.59 | craigify | you want to experiment? |
| 18:14.01 | cIuFuLiCi | it for me |
| 18:14.04 | cIuFuLiCi | to learn |
| 18:14.08 | craigify | or do you want to deploy asterisk for a practical purpose only |
| 18:14.09 | cIuFuLiCi | to call a friend |
| 18:14.22 | cIuFuLiCi | mostly to learn |
| 18:14.45 | robmal | So skip the gui. |
| 18:14.54 | robmal | You'll learn only that you hate it. |
| 18:15.03 | cIuFuLiCi | :) |
| 18:15.12 | cIuFuLiCi | yea but how can i create a user |
| 18:15.20 | cIuFuLiCi | ? |
| 18:15.32 | cIuFuLiCi | i know from there i can create users |
| 18:16.07 | craigify | http://www.amazon.com/Asterisk-Definitive-Guide-Russell-Bryant/dp/1449332420 |
| 18:16.21 | craigify | that will help you drill down into asterisk from scratch |
| 18:16.28 | craigify | and learn how it works |
| 18:17.32 | cIuFuLiCi | :) |
| 18:17.59 | craigify | If you want to get a functional PBX up and running now, my personal opinion is to install freepbx |
| 18:18.15 | craigify | freepbx does things the freepbx way, but it works |
| 18:19.02 | cIuFuLiCi | for me it is not a problem what i use |
| 18:19.21 | cIuFuLiCi | I instaled asterisk because i know about him from a friend |
| 18:19.49 | cIuFuLiCi | i am new in this dont know another program :D |
| 18:20.12 | cIuFuLiCi | i just wanna learn |
| 18:20.24 | craigify | I too share an interest in learning new technology, and I had an opportunity to learn Asterisk a number of years ago. I can tell you that you must get a good foundation first, and that book is a good place to start. |
| 18:22.41 | cIuFuLiCi | 10x |
| 18:24.18 | craigify | I am not sure if there are any translations into other languages other than English |
| 18:24.37 | craigify | but the author walks you through with examples you can create in a dialplan and test out |
| 18:24.51 | WIMPy | I don't think so. There used to be a german book, however. |
| 18:25.00 | cIuFuLiCi | i am romanian :) |
| 18:25.06 | cIuFuLiCi | english is ok |
| 18:25.09 | cIuFuLiCi | i hope |
| 18:25.09 | cIuFuLiCi | :D |
| 18:25.31 | *** join/#asterisk freddo (~chris@cpc7-sgyl32-2-0-cust689.18-2.cable.virginm.net) |
| 18:26.05 | craigify | Romanian is a highly inflected language, yes? |
| 18:26.15 | *** join/#asterisk chadxz (Adium@nat/digium/x-vunpcqcewmyobqgz) |
| 18:26.24 | craigify | and did it not preserve many of the tenses of Latin? |
| 18:26.51 | freddo | hi, i have been trying to find a few uk provides of SRTP enabled sip trunking - but not having great success, why is that or is there an alternative approach that I should be looking at? |
| 18:27.31 | WIMPy | Like it doesn't make sens if everyone listens in at the ITSP anyway? |
| 18:27.40 | craigify | Those were my thoughts |
| 18:27.40 | craigify | heh |
| 18:28.12 | craigify | encryption is only useful unless it's fully encrypted end to end |
| 18:28.30 | craigify | unfortunately.... |
| 18:28.37 | freddo | so from pstn->trunk provider-----> asterisk box |
| 18:29.45 | freddo | but should I not endeavour to encrypt the traffic as a matter of course? reducing the scope of the issue? |
| 18:30.22 | WIMPy | As craigify explained, that doesn't make sense. It only wastes resources, if it's not end-to-end. |
| 18:30.22 | robmal | Just set up an ipsec tunnel to the itsp. |
| 18:30.41 | robmal | Or l2 transmission. |
| 18:30.56 | WIMPy | That's the same. |
| 18:31.05 | robmal | Yes, but easier for the provider. |
| 18:31.35 | craigify | The problem is that the telephone network is not a secure network by any means |
| 18:31.56 | freddo | are there many providers doing this (tunnel) or srtp? or is everyone just not worrying? :) |
| 18:32.08 | ModFather | where asterisk stores the saved voicemails ? |
| 18:32.16 | craigify | so just because you encrypt a connection to a provider, which then hands off to the PSTN, you would not have achieved any extra security at all |
| 18:32.26 | Chainsaw | ModFather: Whereever you configure it to. |
| 18:32.29 | robmal | ModFather: /var/lib/asterisk/something |
| 18:32.30 | craigify | unless you want to connect one endpoint to another directly, not using the telephone network |
| 18:32.49 | WIMPy | freddo: Sipgate offers VPN. |
| 18:32.51 | freddo | but we would have done what we could to secure the connection and that's also important |
| 18:33.05 | craigify | freddo: I would disagree with you on that point. |
| 18:33.06 | WIMPy | And for those who really care, there's zrtp. |
| 18:33.31 | ModFather | Chainsaw i didnt specify any directory, i assume it uses default |
| 18:33.33 | robmal | freddo: Is this your concern or your clients? |
| 18:33.40 | *** join/#asterisk [Outcast] (~outcast@173.38.117.70) |
| 18:33.45 | robmal | ModFather: You did in /etc/asterisk/asterisk.conf |
| 18:34.18 | freddo | it's 'my' concern. a company i work with have customer calls coming over the pstn unencrypted and that makes me a bit worried :) |
| 18:34.54 | WIMPy | The PSTN is not encrypted. So as long as you use that, there's little point. |
| 18:34.58 | freddo | i'm wondering whats the best solution - sounds like there is no perfect one at this time, but that i can at least secure the leg from the trunk provider to the asterisk box |
| 18:35.00 | craigify | I'm not sure what else we could say to tell you that it is a false security measure |
| 18:35.23 | WIMPy | The solution is zRTP. |
| 18:35.46 | freddo | it's not a perfect measure, that much makes good sense. but it feels like it would still be worthwhile |
| 18:35.53 | WIMPy | And just securing some part of a transmission may help for marketing. ONLY. |
| 18:36.01 | craigify | ok |
| 18:36.09 | robmal | freddo: Go with Twilio. |
| 18:36.28 | freddo | in the UK does it help with PCI or ISO, surely we would have an obligation to try? (i'm guessing) |
| 18:36.34 | *** join/#asterisk [Outcast] (~outcast@173.38.117.70) |
| 18:36.50 | freddo | i moved out of voip development many years ago - so just interested in reality |
| 18:37.20 | WIMPy | Well, legal requirements often make no sense at all, but that's a completely different matter. |
| 18:37.35 | freddo | thanks robmal and thanks all :) it's an interesting one |
| 18:40.23 | *** join/#asterisk ericNL (eric@2a01:7e00:e000:69::c0ff:ee) |
| 18:43.51 | ModFather | robmal , my asterisk.conf by default doesnt have this value |
| 18:50.46 | robmal | Sooo /var/spool/asterisk maybe. |
| 19:03.04 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 19:12.58 | *** join/#asterisk TazzNZ (~TazzNZ@mail.insync.za.net) |
| 19:13.24 | *** join/#asterisk [Outcast] (~outcast@173.38.117.70) |
| 19:25.02 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 19:36.48 | *** join/#asterisk [Outcast] (~outcast@173.38.117.70) |
| 19:47.45 | ModFather | is it possible to store all voicemail's to the mysql database? |
| 19:48.03 | robmal | Everything is possible if you're brave enough. |
| 19:48.24 | ModFather | i am brave as hell :p |
| 19:48.34 | robmal | So good luck :-) |
| 19:48.38 | ModFather | awesome! |
| 19:52.06 | *** join/#asterisk theron (~theron@2620:10d:c091:200::b:f033) |
| 19:54.05 | *** join/#asterisk [Outcast] (~outcast@173.38.117.70) |
| 20:05.55 | *** join/#asterisk nny (4239d20d@gateway/web/freenode/ip.66.57.210.13) |
| 20:06.02 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 20:07.21 | nny | How do I check that an echo cancler is functioning normally (hardware) this is a Rhino card |
| 20:07.51 | nny | seems it was "working last night" but not today, suspect something is breaking during the day |
| 20:15.25 | *** join/#asterisk tmkerr (uid47880@gateway/web/irccloud.com/x-ladoohwesqnxvwpg) |
| 20:16.36 | tmkerr | does 'queue show queuenumber' show calls for that day? |
| 20:16.46 | tmkerr | what is the timeframe? |
| 20:21.40 | WIMPy | Since you reloaded the queue IIRC. |
| 20:27.25 | tmkerr | ok |
| 20:28.49 | nny | need a wall to bounce this off of. I have been battling these telco lines since install, and some of it I feel may be their hardware. They have an older Rhino RCB8FXX. I have adjusted the gain (see pb below) and here's the order of issues. Crosstalk introduced (low) after gain adjustment. Echo comes back occasionaly, dahdi restart fixes it. I am updating the rhino firmware but this card is horribly documented and their support non exist |
| 20:29.32 | nny | also it took a lot of gain adjustment to hit 14500, I used the proper method (dial multiple Milliwatt numbers, test. then call inwards from outbound, test, adjust tx) |
| 20:29.59 | WIMPy | Analog is evil! |
| 20:30.04 | nny | ^ agree |
| 20:30.22 | nny | normally I don't have these issues, I use Sangoma cards and they seem to just work when it comes to EC |
| 20:30.32 | nny | no offense to digium cards |
| 20:30.50 | *** join/#asterisk [Outcast] (~outcast@173.38.117.70) |
| 20:31.43 | nny | well, that was odd, thanks pastebin. It got removed http://pastebin.com/E8jw6RNa |
| 20:32.02 | nny | I considered adjusting taps etc but this seems like the hardware EC is quitting/dying/failing |
| 20:32.08 | nny | so i considered OSLEC |
| 20:32.32 | nny | but looking for advice really. I think I can find a way to disable the HWEC if OSLEC is better |
| 20:33.29 | tmkerr | WIMPy: what do you mean reloaded the queue? |
| 20:33.52 | WIMPy | For a few channels I wouldn't even consider HWEC. Totally overpriced and a modern PC will handle quite some channels in software. Plus you get the choice. |
| 20:34.23 | WIMPy | tmkerr: doing a 'queue reload ...'. |
| 20:38.18 | nny | welp, firmware update seemed needed. Time to hope for PFM |
| 20:38.32 | nny | if this keeps up I am moving them to a hybrid voip setup for a bit |
| 20:38.49 | nny | ha er well, hybrid on the network side |
| 20:42.17 | *** join/#asterisk notanoldman (~notanoldm@cpe-45-37-17-178.nc.res.rr.com) |
| 20:42.25 | notanoldman | What do you folks use for tts? |
| 20:42.59 | notanoldman | I was using googletts but it appears it is rate limited. |
| 20:43.10 | robmal | 50 per day, ye. |
| 20:43.31 | robmal | There's one service i hadn't had a chance to test... One sec. |
| 20:43.32 | WIMPy | espeak? |
| 20:43.42 | robmal | https://wit.ai/ |
| 20:44.07 | robmal | It's like learning skynet to understand 'please don't kill me', but it's free. |
| 20:44.58 | robmal | Also, you can cache google tts, so 50 new recordings a day is a lot. |
| 20:45.36 | robmal | Of course after testing 'sudo make me a sandwitch' in 20 languages and 'fuck' in 29. |
| 20:45.49 | WIMPy | LOL |
| 20:46.06 | notanoldman | lol |
| 20:46.09 | robmal | Been there, done that. |
| 20:48.29 | notanoldman | Has anyone tried this? http://www.voicerss.org/ |
| 20:50.10 | *** part/#asterisk WangDang (~wangdang@24.140.224.98) |
| 20:50.28 | robmal | Not that bad in polish. |
| 20:50.55 | robmal | It ignores punctuation but it might be the demo limit. |
| 21:05.31 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 21:20.42 | ModFather | robmal |
| 21:21.10 | ModFather | http://pastebin.ca/3208953 |
| 21:21.21 | ModFather | seems asterisk can connect to my database right? |
| 21:30.16 | robmal | Ye. |
| 21:30.28 | robmal | The date is odd. |
| 21:30.52 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:33.46 | ModFather | yes i didnt added time somewhere manually |
| 21:33.51 | ModFather | the server has correct time |
| 21:40.15 | ModFather | robmal i made a php script that runs very frequently and inserting into the database the voicemail content , all i need now is to setup another script to upload the .wav files to amazon s3 bucket |
| 21:43.12 | ModFather | i am thinking if asterisk can automatically insert the voicemails inside the database, instead of running a php script with cronjob |
| 21:43.32 | ModFather | [TK]D-Fender do you think its possible? |
| 21:44.47 | WIMPy | Just use the script that can be started to send the VM as e-mails. |
| 21:45.21 | robmal | Isn't there voicemail odbc? |
| 21:45.26 | [TK]D-Fender | There is |
| 21:45.29 | robmal | Ok. |
| 21:45.37 | robmal | ModFather: Have fun reinventing the wheel :-) |
| 21:45.39 | ModFather | WIMPy can you help me on this bro? i mean which script i need to start to send the vm as emails? |
| 21:45.49 | [TK]D-Fender | Also running a script "frequently" is a waste |
| 21:45.58 | [TK]D-Fender | considering * can call a script after a VM is left. |
| 21:45.59 | WIMPy | The one you will write. |
| 21:46.02 | [TK]D-Fender | so that it's not running for nothing |
| 21:46.08 | ModFather | [TK]D-Fender i agree but i didnt find somewhere where i can automatically send the VM to odbc |
| 21:46.22 | [TK]D-Fender | There is an enbtire voicmail module for this |
| 21:46.35 | [TK]D-Fender | https://www.google.ca/#q=asterisk+voicemail+odbc |
| 21:46.42 | [TK]D-Fender | Funny I see a LOT saying how right there |
| 21:46.49 | [TK]D-Fender | Doesn't look like you looked... |
| 21:46.59 | ModFather | :) [TK]D-Fender funny, i was there some hours before |
| 21:47.18 | WIMPy | looks if someone is looking... |
| 21:47.33 | ModFather | WIMPy ... https://www.google.ca/url?sa=t&rct=j&q=&esrc=s&source=web&cd=1&cad=rja&uact=8&ved=0CBwQFjAAahUKEwiml7STg9LIAhVBwxoKHSALAzU&url=http%3A%2F%2Fwww.voip-info.org%2Fwiki%2Fview%2FAsterisk%2BVoicemail%2BODBC%2Bstorage&usg=AFQjCNHmlsIM43s3aZbT4TmemMz8H9_38A |
| 21:47.36 | ModFather | this was the first result |
| 21:47.55 | ModFather | tell me.. do you see anywhere there how i can make my VM to automatically stored to ODBC? |
| 21:49.02 | ModFather | [TK]D-Fender http://pastebin.ca/3208995 |
| 21:49.41 | ModFather | [TK]D-Fender you are smart, with much knowledge, but bro.. your attitude is like a ..... ! try dont be that rude, i wont disturb you again |
| 21:49.58 | [TK]D-Fender | What part of "that WIKI is ancient garbage" have you NOT learned from eariler today? |
| 21:50.05 | [TK]D-Fender | And why are you stopping on the FIRST link there? |
| 21:50.23 | ModFather | [TK]D-Fender this wiki ancient garbage is first result on your <[TK]D-Fender>https://www.google.ca/#q=asterisk+voicemail+odbc |
| 21:50.26 | [TK]D-Fender | <PROTECTED> |
| 21:50.31 | robmal | https://db.tt/87BbkSCX |
| 21:50.32 | ModFather | i didnt stopped on the first link |
| 21:50.32 | [TK]D-Fender | And it talks about the res configs for this |
| 21:50.54 | ModFather | [TK]D-Fender already did that |
| 21:51.01 | [TK]D-Fender | I see all sorts of instructions there for preparing the DB as well |
| 21:51.20 | robmal | I'll try again. |
| 21:51.22 | robmal | https://db.tt/87BbkSCX |
| 21:51.26 | ModFather | thanks robmal |
| 21:51.30 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/ODBC+Voicemail+Storage |
| 21:51.44 | [TK]D-Fender | Follow the official WIKI |
| 21:52.04 | ModFather | robmal i have done that already |
| 21:52.30 | robmal | So what's your problem now? |
| 21:52.40 | ModFather | i have named it voicemails the table instead of voicemessages |
| 21:53.14 | ModFather | i also added: odbctable=voicemails inside voicemail.conf |
| 21:54.29 | ModFather | and: odbc show has ( Connected: Yes ) |
| 21:54.45 | ModFather | i have done all this steps |
| 21:55.00 | ModFather | that [TK]D-Fender says that i need to look at |
| 21:55.22 | robmal | Good job. |
| 21:55.24 | ModFather | robmal i am missing maybe a setting inside extensions that send voicemails to odbc instead of local storage |
| 21:55.31 | [TK]D-Fender | no |
| 21:55.36 | robmal | It's not configured in extensions. |
| 21:55.36 | [TK]D-Fender | You need to have that MODULE loaded |
| 21:55.46 | ModFather | I HAVE IT LOADED |
| 21:55.48 | [TK]D-Fender | because there are MULTIPLE app_voicemail modules |
| 21:55.56 | [TK]D-Fender | you have the WRONG ONE loaded |
| 21:55.59 | robmal | YOU ARE BOTH CORRECT |
| 21:56.03 | robmal | ;-) |
| 21:56.11 | robmal | I'm enjoying this a bit too much. |
| 21:56.18 | [TK]D-Fender | Also true |
| 21:56.23 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
| 21:56.35 | ModFather | i am going for a chigar... |
| 21:57.00 | robmal | I'm going for moar beer because i won't be sleeping this night again :-) |
| 21:58.22 | ModFather | robmal do you have any link to Asterisk Doc that explain which "module" ( the right one ) which need to be enabled? |
| 21:59.03 | [TK]D-Fender | Look in your modules folder |
| 21:59.26 | [TK]D-Fender | Also at the point where you compiled in the first place... proving you even had the requirements installed and that it got built |
| 21:59.42 | ModFather | i did odbc install ebfore compile.. |
| 21:59.50 | ModFather | before* compile asterisk |
| 22:00.44 | ModFather | res_odbc.so |
| 22:00.53 | [TK]D-Fender | So far that is not telling me "Yes I see it in menconfig" |
| 22:01.04 | [TK]D-Fender | That is NOT the voicemail app |
| 22:01.07 | [TK]D-Fender | that is ONE part |
| 22:01.13 | ModFather | yes i see it on make menuselect |
| 22:01.25 | [TK]D-Fender | and what's the module called? |
| 22:01.31 | [TK]D-Fender | You get a name there.... |
| 22:01.48 | ModFather | i cant remember that |
| 22:01.49 | ModFather | app_db.so |
| 22:02.36 | [TK]D-Fender | Remember? Go look NOW |
| 22:03.00 | ModFather | func_odbc.so |
| 22:04.40 | ModFather | <PROTECTED> |
| 22:04.40 | ModFather | <PROTECTED> |
| 22:06.23 | ModFather | do you mean that mr [TK]D-Fender ? |
| 22:09.43 | ModFather | i had selected File_Storage instead of ODBC_STORAGE how i can enable it now ? |
| 22:11.13 | *** join/#asterisk davidbowlby (~textual@cpe-71-79-82-152.columbus.res.rr.com) |
| 22:12.06 | ModFather | robmal are you still there mate? |
| 22:12.55 | davidbowlby | Hi, I have two asterisk servers behind the same firewall listening on different ports (5160, 5060). When I send an invite from the 5160 box, the domain resolves to the other peer that is associated to the same IP. The invite comes in showing it came from 5160. Is port not considered part of the call path lookup? |
| 22:15.52 | davidbowlby | it says looking for <extension> in <wrong context> (domain <common IP address>) |
| 22:16.21 | davidbowlby | I tried setting up my 5160 server to register, to see if that would help |
| 22:16.25 | davidbowlby | hasn't though |
| 22:16.36 | [TK]D-Fender | RECOMPILE |
| 22:17.16 | robmal | ModFather: Ye. |
| 22:17.19 | ModFather | I WILL RECOMPILE BUT I WANT TO ASK robmal if i will lost all the settings i have done so far.. or the recompile will only add the new modules without touch anything else i have already edit |
| 22:17.28 | robmal | NO YOU WON'T |
| 22:17.30 | robmal | :-) |
| 22:17.32 | ModFather | thanks BRO |
| 22:17.34 | robmal | This is fun! |
| 22:17.41 | ModFather | yes it is with this native canadian guy |
| 22:17.46 | ModFather | he isnt italian |
| 22:18.50 | ModFather | robmal so if i choose odbc_storage, and do make make install i will not lost configs and settings right? |
| 22:19.07 | robmal | As long as you DON'T MAKE SAMPLES. |
| 22:19.23 | ModFather | i will not, i promise that :p |
| 22:20.10 | ModFather | also robmal , you can choose only 1 option from ( ODBC, IMAP, FILE ) |
| 22:20.23 | robmal | IMPOSSIBLE. |
| 22:20.26 | ModFather | why it doesnt let you to choose all of them and then configure which one to use |
| 22:20.31 | robmal | DOES NOT COMPUTE. |
| 22:21.08 | ModFather | regarding the weird time on odbc show |
| 22:21.15 | ModFather | do you know where i should check ? |
| 22:21.20 | robmal | I'm too lazy now to check menuconfig but YOU NEED RES_ODBC AND CORRECT VOICEMAIL.CONF ONLY |
| 22:21.40 | ModFather | yes i will do that right now |
| 22:23.13 | ModFather | res_odbc.so i have that already on my modules, but i will do make menuselect to Enable option ODBC_STORAGE and then make make install |
| 22:23.38 | ModFather | i will adjust voicemail.conf and i will pray to success |
| 22:24.05 | robmal | Praying lead us to dark ages. SCIENCE. |
| 22:24.23 | ModFather | i am a Jew i need to pray |
| 22:24.55 | ModFather | but you can pray too, ( scientology ) |
| 22:24.57 | ModFather | rolf |
| 22:25.09 | robmal | Too lazy. |
| 22:25.33 | ModFather | at least you got nice attitude :) |
| 22:25.48 | robmal | https://scontent-waw1-1.xx.fbcdn.net/hphotos-xta1/v/t1.0-9/s720x720/12141763_858580920916561_6916471687420495839_n.jpg?oh=6f6827753fa19140818e507a379ad7c8&oe=56C42DA3 |
| 22:25.48 | davidbowlby | bleh, nm, newb move on my part |
| 22:26.15 | davidbowlby | called the context directly, which didn't have the entry |
| 22:27.58 | ModFather | robmal hahaahhaha |
| 22:28.05 | ModFather | thats an amazing image |
| 22:28.16 | robmal | Also true. |
| 22:29.55 | ModFather | robmal : Trust Me, I'm an "Engineer" |
| 22:29.58 | ModFather | hahahaha |
| 22:30.12 | robmal | That's a good fanpage. |
| 22:30.13 | ModFather | www.youtube.com/watch?v=MGJdlOQ-3QI |
| 22:30.39 | robmal | I'll find a better one, which i'll put in my signature when i'm old enough. |
| 22:31.07 | ModFather | how much old you are? |
| 22:31.54 | ModFather | i mean: "How old are you" :p |
| 22:34.40 | robmal | 1E |
| 22:35.18 | robmal | I can has lost the image. It went along like so: 'Working in IT is like being a rocket, you get shit done only when your ass is on fire' |
| 22:35.47 | robmal | Well, maybe s/shit/things/ |
| 22:35.55 | robmal | I seem to be saying 'shit' a lot recently. |
| 22:57.46 | *** join/#asterisk kritzikratzi (~kritzikra@cpe90-146-150-86.liwest.at) |
| 23:05.06 | *** join/#asterisk debianlv (~jouhary@208.80.156.182) |
| 23:05.41 | debianlv | Hi everyone, we are using AGI an would like to limit call length. any ideas how? |
| 23:05.59 | robmal | Yes. |
| 23:06.19 | WIMPy | wonders how those two things are related. |
| 23:06.36 | WIMPy | I drive an Audi and would like to plant an orange tree. |
| 23:06.44 | WIMPy | Well. |
| 23:07.10 | WIMPy | 'core show function TIMEOUT' and 'core show appication dial' would be some obvious ones. |
| 23:08.18 | debianlv | as far as i know limiting calls durations is done in extention.conf |
| 23:08.36 | debianlv | and we are using AGI so the normal function to use Timout doesn't work |
| 23:08.43 | WIMPy | Most likely. |
| 23:09.00 | WIMPy | IGI *IS* dialplan. |
| 23:09.05 | WIMPy | AGi |
| 23:10.21 | debianlv | AGI is an interface for adding functionality to Asterisk with many different programming languages |
| 23:10.32 | robmal | Oh, no, not this again. |
| 23:10.43 | robmal | debianlv: You have to set variable timeout for the call you're handling. |
| 23:10.50 | WIMPy | Called from the dialplan and executing dialplan applications. |
| 23:14.38 | ModFather | robmal i did that: http://pastebin.ca/3209094 |
| 23:14.44 | robmal | Yay! |
| 23:14.48 | ModFather | ;) |
| 23:14.54 | ModFather | also i adjusted voicemail.conf |
| 23:15.02 | robmal | Yay! |
| 23:15.09 | ModFather | odbcstorage=asterisk |
| 23:15.09 | ModFather | odbctable=voicemails |
| 23:15.21 | lvlinux | Hey, how do I debug DTMF on the console?? |
| 23:15.41 | robmal | logger.conf console=blahblah,dtmf |
| 23:15.47 | lvlinux | I have a Multitech MultiVoIP unit and it's not doing dtmf stuff right. |
| 23:16.03 | lvlinux | robmal: can't I do it temporarily without messing in a config file? |
| 23:16.04 | ModFather | robmal : also my res_odbc.conf http://pastebin.ca/3209095 |
| 23:16.27 | ModFather | do am i missing anything else? |
| 23:16.41 | robmal | lvlinux: No. |
| 23:16.49 | robmal | ModFather: I don't know. Does it work? |
| 23:16.50 | lvlinux | k thx |
| 23:16.55 | ModFather | from CLI odbc show says its connected |
| 23:17.01 | ModFather | am going to leave a voice message |
| 23:17.03 | ModFather | and check db |
| 23:17.06 | robmal | lvlinux: Edit logger.conf and reload the module, no need for restart. |
| 23:23.16 | ModFather | robmal i want to cry |
| 23:23.19 | ModFather | it worked!!!!! |
| 23:23.43 | ModFather | [TK]D-Fender cannadian friend, it worked !!!! |
| 23:24.05 | robmal | Well, guys at Digium did the work there, thank them. |
| 23:24.17 | ModFather | Digium made that odbc ? |
| 23:24.38 | ModFather | thanks robmal |
| 23:24.48 | robmal | Check the sources, they're signed. |
| 23:25.05 | ModFather | thanks them and thanks you too then |
| 23:25.16 | robmal | GL&HF |
| 23:25.21 | ModFather | gg |
| 23:28.12 | ModFather | robmal i am trying to play the file from "blob" |
| 23:28.16 | ModFather | but its empty :p |
| 23:28.18 | ModFather | any ideas? |
| 23:28.40 | robmal | Something went terribly wrong. |
| 23:28.58 | ModFather | any clue to what to look for? |
| 23:29.04 | ModFather | would be much appreciated |
| 23:29.08 | robmal | How big is the file? |
| 23:29.16 | ModFather | 2-3 seconds |
| 23:29.32 | robmal | And in kilobytes? |
| 23:29.37 | ModFather | let me see |
| 23:31.16 | ModFather | 135kb |
| 23:31.46 | robmal | So, you set the wrong format or didn't set it at all and it went all g711.a |
| 23:32.17 | ModFather | hmm i guess yes |
| 23:32.22 | ModFather | can you let me know where i can fix that? |
| 23:32.31 | ModFather | its the only remaining issue at the moment |
| 23:32.49 | robmal | There is a format= variable in voicemail.conf |
| 23:32.53 | robmal | Try wav |
| 23:32.54 | ModFather | aah |
| 23:32.56 | ModFather | yes 1 sec |
| 23:33.08 | *** part/#asterisk kharwell (kharwell@nat/digium/x-uvykpeaqpgcwxqmh) |
| 23:33.12 | ModFather | robmal format=wav49|wav |
| 23:33.14 | ModFather | i had this |
| 23:33.24 | ModFather | have* |
| 23:33.30 | robmal | I have no idea what wav49 is and i have no intention to find out now. |
| 23:33.40 | ModFather | then i am removing it |
| 23:33.41 | ModFather | 1 sec |
| 23:33.46 | lvlinux | what is the correct DTMF rfc2833 payload type? 96 or 101? I have phones that have 101 as default and this multitech has 96. |
| 23:34.08 | ModFather | i set it to wav |
| 23:34.11 | ModFather | and i am retrying |
| 23:34.13 | robmal | lvlinux: 101 should be ok. |
| 23:34.47 | lvlinux | so I should change the multitech to 101 to match the phones? |
| 23:35.04 | robmal | If there's an issue - guess ;-) |
| 23:35.45 | robmal | If you've got dtmf logging enabled you'll see how if asterisk picked the tone and how long it lasted. |
| 23:36.35 | lvlinux | well asterisk was receiving the tones and showing events from the multitech. looked fine. But when I do a test call from the multitech (w an FXS phone), if i press the buttons on the phone, just a very short amount of one or two every once in a while will play through the IP phone. |
| 23:37.04 | robmal | So change the payload. |
| 23:37.37 | lvlinux | k i'll try it |
| 23:39.34 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 23:44.10 | lvlinux | if i can get this thing to respond... serial connection and windoze only config software... ugh |
| 23:47.22 | robmal | You can try changing its dtmfmode on the asterisk side. |
| 23:47.49 | ModFather | robmal do you know any php script that export the blob content from asterisk database? |
| 23:48.14 | ModFather | am trying in google but no luck so far |
| 23:48.19 | robmal | Meh. |
| 23:49.07 | ModFather | what i am trying to do is, to play voicemails from browser ( web based ) |
| 23:49.16 | robmal | mysql -u asdf -p qwe ewqoi -e select recording from whatever >> file.wav |
| 23:49.23 | robmal | file file.wav |
| 23:49.26 | ModFather | ah |
| 23:49.29 | ModFather | that easy? |
| 23:49.38 | robmal | It could be easier, but yes. |
| 23:49.56 | ModFather | going to try thanks man |
| 23:50.52 | robmal | If you're going for browsers i'd go mp3 on the fly. |
| 23:50.59 | ModFather | hmmm |
| 23:51.10 | ModFather | mp3 on the fly? |
| 23:51.11 | ModFather | how? |
| 23:51.32 | ModFather | when i compile asterisk i checked mp3 support |
| 23:51.39 | ModFather | if i change format=wav to mp3 |
| 23:51.46 | ModFather | it will store it as mp3? |
| 23:51.51 | robmal | I have no idea. |
| 23:51.53 | robmal | It might. |
| 23:52.10 | ModFather | yes am going for browsers, i already done with the most part of the CRM in laravel |
| 23:52.12 | robmal | app_voicemail can do things once it stops storing the files. |
| 23:52.23 | robmal | Oh, +1 for laravel. |
| 23:52.38 | ModFather | yes i fell in love with it |
| 23:52.43 | *** join/#asterisk italorossi (~Adium@177.193.104.232) |
| 23:52.50 | ModFather | laravel 5 rocks |
| 23:53.00 | robmal | Well, maybe not fell in love but i like it. |
| 23:53.16 | ModFather | i was too much years in coldfusion.. |
| 23:53.28 | ModFather | laravel for me is now a paradise |
| 23:53.54 | ModFather | can you give me an example about your idea? with mp3 on the fly? |
| 23:54.04 | ModFather | its sounds interesting |
| 23:56.30 | robmal | http://bernaerts.dyndns.org/linux/179-asterisk-voicemail-mp3 |
| 23:57.32 | robmal | But i'd go with on-request conversion so once the user clicks play some loader starts spinning and a mp3 is made on the fly from a wav in the db. |
| 23:58.50 | robmal | I think that's more kosher. |
| 23:58.54 | ModFather | hahaahaha |
| 23:58.58 | ModFather | you know kosher? |
| 23:59.18 | ModFather | i suggest to you to eat only kosher products |
| 23:59.38 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 23:59.49 | robmal | I see no gain in restricting my food intake. |
