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| 00:13.39 | mushroomed | [TK]D-Fender: Thank you bro |
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| 00:44.31 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 01:03.12 | mushroomed | [TK]D-Fender: handle_response_invite: Received response: "Forbidden" from '"Anonymous" ... |
| 01:09.04 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 01:14.39 | mushroomed | Can someone help me with this (Dialplan Originate CMD) --> handle_response_invite: Received response: "Forbidden" from '"Anonymous" |
| 01:16.20 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 01:18.02 | WIMPy | Where is the relation between those two things? |
| 01:18.23 | WIMPy | You're calling something that doesn't want to accept calls from you. |
| 01:20.32 | mushroomed | WIMPy: I don't get what you mean |
| 01:23.15 | WIMPy | I just explained "Forbidden". |
| 01:24.31 | mushroomed | WIMPy: Originate doesn't work for me |
| 01:27.35 | WIMPy | Do you want to know why? |
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| 01:33.11 | mushroomed | WIMPy: Can you give me an example of Originate to do this: |
| 01:33.19 | mushroomed | For example, I call my internal *99 and it asks for a number so I enter 12345678 and then it says 'one-moment-please', hangs up and call 12345678. When it answers, asterisk plays 'one moment please' and transfer that call to my extension |
| 01:34.02 | WIMPy | I don't know what you tried to call, but did you see that "Forbidden"? It was obviousely not ok for whatever it was. |
| 01:34.28 | mushroomed | If I do this --> same=>n,Originate(SIP/trunk/${NUMBER},exten,internal,1440,1) |
| 01:34.53 | WIMPy | there is no generic concept of calling a number. You can call a peer an optionally give it a number to call for you. |
| 01:34.55 | mushroomed | Then it just call ${NUMBER} and when callee answers, my device rings and the call is connected |
| 01:35.53 | mushroomed | And that is not what I want according to what I wrote above |
| 01:35.53 | WIMPy | Is "trunk" the name of a configured peer? |
| 01:36.31 | mushroomed | WIMPy: It already works, now it calls (No more Forbidden Message) but I don't understand how-to do what I want |
| 01:37.53 | WIMPy | Originate () will wait for that originate to finish. If you want to end the call and then start a new call, you need to do it externally via a script. |
| 01:38.26 | WIMPy | Either by writing a call file, by using asterisc -x "channel originate ..." or via AMI originate. |
| 01:40.56 | mushroomed | WIMPy: That would Originate the call too, and then ... when is the one-moment-please played to the callee before he/she is connected to me? |
| 01:41.17 | WIMPy | Use the dialplan. |
| 01:41.52 | WIMPy | Don't send the call to the extension that calls your phone but to an extension that answers the call, plays the file and then calls your phone. |
| 01:42.30 | mushroomed | WIMPy: Ok, good, that was what I was thinking too, but now my last question is how-to do the "and then calls your phone"? That is a kinda transfer |
| 01:42.57 | WIMPy | No, just a simple Dial() as always. |
| 01:43.13 | mushroomed | Oh, mmkay |
| 01:43.18 | mushroomed | Let me try it |
| 01:43.26 | WIMPy | You could also Goto() anther extension that does so anyway. |
| 01:43.42 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
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| 02:06.38 | ruel | hey raspberrypifan |
| 02:06.43 | raspberrypifan | how r u |
| 02:07.24 | lvlinux | doin fine-- u made any progress finding a sip provider for down there (or an ata)? |
| 02:08.08 | raspberrypifan | i found an ata |
| 02:08.10 | raspberrypifan | for cheap |
| 02:08.13 | raspberrypifan | but its been a bit of a mess |
| 02:08.39 | lvlinux | with f |
| 02:08.53 | lvlinux | an fxo port u mean? |
| 02:08.56 | raspberrypifan | yup |
| 02:09.00 | lvlinux | cool |
| 02:09.07 | lvlinux | which model is it? |
| 02:09.24 | raspberrypifan | http://www.ebay.com/itm/131052305172?ssPageName=STRK:MEWNX:IT&_trksid=p3984.m1497.l2649 |
| 02:09.28 | raspberrypifan | techincally we agreed on 10 total |
| 02:09.30 | raspberrypifan | but then it says 5 |
| 02:09.33 | raspberrypifan | but they wont answer |
| 02:12.00 | lvlinux | hmm |
| 02:12.28 | raspberrypifan | anywa when that gets resolved |
| 02:12.30 | raspberrypifan | itll be cheap |
| 02:13.51 | lvlinux | well good |
| 02:14.01 | raspberrypifan | yu |
| 02:14.06 | *** join/#asterisk makubi_ (~makubi@xdsl-81-173-236-35.netcologne.de) |
| 02:20.13 | *** join/#asterisk dw5304 (~dw5304@nas4free/dw5304) |
| 02:26.56 | igcewieling | I'm having an issue where hangup handlers exit prematurely. Dialplan, CLI output, and show dialplan info at http://pastie.org/9144425 Notice the timestamps in the hangup handler. |
| 02:34.51 | dw5304 | first time asterisks user setting up queues for the first time anyone help me out in not allowing a call to be placed to a person who is allready on the line? |
| 02:35.31 | [TK]D-Fender | dwplease rephrase your question.... |
| 02:35.36 | [TK]D-Fender | dw5304: please rephrase your question.... |
| 02:36.46 | dw5304 | ok have an extension setup to queue a call. im using a soft phone and it keeps calling the same extention while the other line is in use. |
| 02:37.05 | dw5304 | soft phone is xlite |
| 02:37.46 | [TK]D-Fender | How are these 2 lines registered? |
| 02:37.56 | [TK]D-Fender | And clarify your use of the term "line" |
| 02:38.30 | dw5304 | ok the "2" lines are registed by default... went to settings and entered the domain and user |
| 02:38.39 | dw5304 | in x-lite |
| 02:39.05 | [TK]D-Fender | What user for each? |
| 02:39.10 | [TK]D-Fender | You are still very thin on details... |
| 02:39.12 | dw5304 | only one account |
| 02:39.17 | dw5304 | preferances |
| 02:39.25 | dw5304 | user and pass only once |
| 02:39.39 | [TK]D-Fender | Ok, then we'll refrain from calling it "2 lines". |
| 02:39.44 | dw5304 | ok |
| 02:39.46 | dw5304 | fair enough |
| 02:39.57 | [TK]D-Fender | Show us your queue3 status the the caller already on a call and getting another call. |
| 02:40.22 | [TK]D-Fender | "queue show", and the queue configs. |
| 02:40.26 | dw5304 | ok |
| 02:42.06 | dw5304 | http://pastebin.com/Wnxnf7sw |
| 02:44.08 | dw5304 | http://i.snag.gy/sbnlV.jpg |
| 02:44.09 | [TK]D-Fender | (ringinuse enabled) <---- |
| 02:44.16 | [TK]D-Fender | you allow it to be called... while in use |
| 02:44.47 | dw5304 | enabled? |
| 02:44.57 | [TK]D-Fender | that's what it says |
| 02:45.00 | dw5304 | ah |
| 02:45.00 | dw5304 | ok |
| 02:45.03 | dw5304 | lol |
| 02:45.07 | dw5304 | how do i kill that then :) |
| 02:45.08 | dw5304 | my bad |
| 02:45.22 | [TK]D-Fender | give a read to the sample config that * comes with.... |
| 02:46.18 | KNERD | on the initial "make menuselect" how can we perform that action the first time without it loading the menu interface, and instead just exit once completed? |
| 02:51.31 | raspberrypifan | asterisk has a gui now? |
| 02:52.17 | [TK]D-Fender | No. |
| 02:52.22 | [TK]D-Fender | All of them are bolt ons |
| 02:52.38 | [TK]D-Fender | There are GUI's for Asterisk, but Asterisk does not inherently have one |
| 02:53.35 | raspberrypifan | why did momma asterisk not wanan give her riches away? |
| 03:05.52 | raspberrypifan | asterisk has a marketing manager? |
| 03:06.16 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 03:06.40 | [TK]D-Fender | Not so sure... |
| 03:07.01 | *** join/#asterisk Lee- (~Lee@unaffiliated/lee-) |
| 03:07.08 | raspberrypifan | https://www.youtube.com/watch?v=PfSL-kekuDE |
| 03:07.10 | raspberrypifan | this guy says he is |
| 03:07.33 | [TK]D-Fender | Guess so.... |
| 03:07.53 | [TK]D-Fender | Which is off because ABE no longer exists, so Asterisk isn't "sold" really... |
| 03:07.57 | [TK]D-Fender | it sells other stuff |
| 03:08.02 | [TK]D-Fender | like Digium's cards :) |
| 03:08.18 | raspberrypifan | whats ABE |
| 03:09.02 | [TK]D-Fender | Asterisk Business Edition |
| 03:09.18 | raspberrypifan | is that thing is an hour long |
| 03:09.20 | [TK]D-Fender | Visit the Smithsonian for some background on it... |
| 03:09.23 | raspberrypifan | got any shorter videos |
| 03:09.26 | raspberrypifan | on asterisk |
| 03:10.12 | [TK]D-Fender | keep searching. |
| 03:10.58 | raspberrypifan | sigh |
| 03:11.02 | raspberrypifan | no easy guides |
| 03:11.04 | raspberrypifan | i see |
| 03:11.12 | WIMPy | ~book |
| 03:11.12 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 03:11.22 | WIMPy | That's the easiest guide. |
| 03:12.10 | raspberrypifan | but it doesnt have moving pictures |
| 03:12.33 | WIMPy | Neither does Asterisk. |
| 03:12.57 | [TK]D-Fender | It can transport them however :) |
| 03:13.36 | WIMPy | Without knowing what it is. |
| 03:13.39 | raspberrypifan | does it have that crazy apple codec? |
| 03:13.48 | [TK]D-Fender | No |
| 03:13.56 | [TK]D-Fender | * does standards... not "crazy" |
| 03:14.00 | raspberrypifan | h smoething for another |
| 03:17.46 | Lee- | Can anyone recommend a vendor who specializes in school districts or offers pricing discounts for K-12 schools? My district is looking to upgrade their old system to a VoIP system and I would like to see them embrace an open source solution, but I have no working experience with asterisk and am not familiar with the companies involved. Any pointers appreciated. Thanks. |
| 03:18.12 | raspberrypifan | is ur your district needing a mom to do spanish parent faciliating, i have a mother who needs a new job |
| 03:24.22 | [TK]D-Fender | I'd be guessing you'd need someone LOCAL as well since with noone having any experience... I wouldn't expect them to understand wiring, install a physical box, etc... |
| 03:24.59 | [TK]D-Fender | VoIP can come into play in multiple places... and isn't always the answer |
| 03:32.26 | Lee- | The IT department can certainly manage running cables and installing physical boxes. What they would likely need is support getting it set up. Basically they're looking at a cisco based system now and being that our district is broke, I want to advocate an asterisk based solution, but without knowing a company that can provide reliable installation and post installation support, I can't advocate for asterisk. Lots of companies claim they offer commercial s |
| 03:32.26 | Lee- | upport, but I don't know which are trustworthy. |
| 03:33.40 | raspberrypifan | what SIP? |
| 03:33.47 | raspberrypifan | ddoes it come with a straw |
| 03:34.42 | [TK]D-Fender | I installed DAHDI, but HE DIDN'T COME BACK!!!! |
| 03:34.45 | [TK]D-Fender | #issues |
| 03:35.37 | raspberrypifan | also whats a dial plan |
| 03:36.12 | [TK]D-Fender | I usually forgo plans and just wing it... |
| 03:36.28 | raspberrypifan | this guy is talking |
| 03:36.34 | raspberrypifan | who arent u experts in this stuff |
| 04:02.52 | raspberrypifan | y'all were just mentioned in the video |
| 04:02.53 | raspberrypifan | say hi ppl |
| 04:06.50 | [TK]D-Fender | "y'all"? |
| 04:06.55 | [TK]D-Fender | He said "y'all"? |
| 04:07.02 | [TK]D-Fender | That's so ratchet.... |
| 04:13.20 | raspberrypifan | lol |
| 04:13.26 | raspberrypifan | im from NY but live in the southern |
| 04:13.40 | raspberrypifan | whats a sip tho [TK]D-Fender |
| 04:14.32 | [TK]D-Fender | It's the precursor to chan_gulp |
| 04:15.52 | raspberrypifan | whats that even mean |
| 04:18.27 | [TK]D-Fender | YES |
| 04:18.59 | raspberrypifan | can asterisk do call termination |
| 04:19.45 | [TK]D-Fender | Asterisk terminates all my calls. Some with extreme prejudice... |
| 04:21.56 | raspberrypifan | [TK]D-Fender huh should there be an asterisk of topic channel.... |
| 04:26.14 | ChannelZ | YouTube Has Died Of Dysentery |
| 04:27.34 | raspberrypifan | ? |
| 04:31.40 | *** join/#asterisk bulkorok (~Adium@053d9234.dynamic.tele-ag.de) |
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| 05:01.28 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-26-100.pn.at.cox.net) |
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| 06:49.37 | KNERD | wow..asterisk really hates virtual machines |
| 06:49.58 | r00f | why? |
| 06:50.08 | KNERD | it just does not work correctly |
| 06:50.31 | r00f | i am running 4 asterisks in vms, including a2billing, and no issues so far |
| 06:50.44 | KNERD | which versions and what VMS? |
| 06:51.00 | r00f | asterisk 11, under Hyper-V HA cluster |
| 06:51.04 | r00f | hyper-v 2012 |
| 06:52.15 | *** join/#asterisk CeBe (~CeBe@port-92-206-126-182.dynamic.qsc.de) |
| 06:52.26 | KNERD | I am on KVM now...just utter fail |
| 06:52.36 | KNERD | I am can get 1.8.x to work |
| 06:52.50 | r00f | i have nothing to say. not familiar with kvm |
| 06:54.20 | KNERD | it's even worse under OpenVZ |
| 06:54.29 | KNERD | cannot use anything past 1.8.12 |
| 06:55.11 | [TK]D-Fender | KNERD: Apparently ESXi works well and has for quite a long while. |
| 06:55.28 | [TK]D-Fender | it's the go-to from what I hear |
| 06:56.04 | KNERD | I may have to give that a look then |
| 06:56.22 | KNERD | thanks for that tip |
| 06:57.47 | KNERD | I just tried 11.9 on KVM/CentOS 6.5 x64 and it won't even load the modules...even with astmoddir => /usr/lib64/asterisk/modules in asterisk.conf |
| 06:58.03 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 06:58.13 | [TK]D-Fender | show us |
| 06:58.26 | KNERD | i just did ;-) |
| 06:58.38 | KNERD | > sip show peers No such command 'sip show peers' (type 'core show help sip show' for other possible commands) |
| 06:58.46 | [TK]D-Fender | NO, pastebin the entire damn file <- |
| 06:59.09 | [TK]D-Fender | Explicit enough? |
| 06:59.20 | [TK]D-Fender | Apparently I hadn't been today. |
| 06:59.33 | r00f | why would you |
| 06:59.36 | [TK]D-Fender | figures he'd balance one last thing out. |
| 06:59.36 | KNERD | http://pastebin.com/0R7ZMC8T |
| 07:00.05 | [TK]D-Fender | ls -la /usr/lib64/asterisk/modules |
| 07:00.12 | [TK]D-Fender | and then manually load one. |
| 07:00.27 | [TK]D-Fender | And show us how you're starting * so that it looks at that asterisk.conf |
| 07:01.07 | KNERD | oh..those modules are there..I can see them |
| 07:01.19 | [TK]D-Fender | pastebin it all |
| 07:01.54 | KNERD | I started it with /usr/sbin/safe_asterisk |
| 07:02.09 | *** join/#asterisk mirela666 (~mirko.bra@188.93.81.142) |
| 07:02.21 | [TK]D-Fender | [02:58]KNERD> sip show peers No such command 'sip show peers' (type 'core show help sip show' for other possible commands) <- this by itself doesn't tell us that chan_sip/pjsip doesn't have it's own problems and fails to load |
| 07:02.22 | Penguin | [TK]D-Fender: its |
| 07:02.40 | KNERD | yeah let me try now |
| 07:03.27 | *** join/#asterisk yaaago (~kresp0@gateway/tor-sasl/kresp0) |
| 07:04.08 | KNERD | module load chan_sip.so Unable to load module chan_sip.so Command 'module load chan_sip.so' failed. SIP channel loading... |
| 07:04.44 | KNERD | and again: sip show peers No such command '> sip show peers' (type 'core show help > sip' for other possible commands) |
| 07:05.04 | KNERD | i guess I am going back to asterisk 1.8.x |
| 07:05.05 | [TK]D-Fender | http://www.slideshare.net/saghul/running-asterisk-on-virtualized-environments |
| 07:06.28 | [TK]D-Fender | Anyway, nothing left to commont on here.... |
| 07:06.35 | [TK]D-Fender | heads off for the night. |
| 07:07.01 | KNERD | that wa sa bit dated |
| 07:09.31 | r00f | readelf -h /usr/lib64/asterisk/modules/chan_sip.so |
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| 07:23.01 | KNERD | it's therte |
| 07:23.03 | KNERD | there |
| 07:24.00 | KNERD | r00f: http://pastebin.com/LGnm7jyj |
| 07:31.36 | r00f | asterisk was also compiled for x64? |
| 07:31.59 | r00f | selinux is not interfering? |
| 07:32.31 | r00f | afaik, selinux should have some settings if using kvm |
| 07:33.14 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 07:34.36 | KNERD | selinux is always diabled |
| 07:34.45 | KNERD | yes it was compile dwith x64 |
| 07:35.17 | r00f | then i haz no ideas why wouldn't it load |
| 07:35.37 | r00f | try esxi, as suggested |
| 07:36.12 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:8d68:d33c:e1b3:ba9c) |
| 07:38.19 | KNERD | hah |
| 07:38.37 | KNERD | this is a VPS rental..I will switch down to 1.8.12 |
| 07:39.25 | KNERD | even used : ./configure --libdir=/usr/lib64 |
| 07:43.47 | r00f | i have run asterisk on hosted vps with like... idk.. 5 different hosters. asterisk from 1.6 up to 11 |
| 07:44.02 | r00f | never had any issues. seems like your hoster bug |
| 07:44.56 | KNERD | no |
| 07:45.07 | KNERD | i have had this issue many times |
| 07:45.21 | r00f | so i was lucky then |
| 07:45.25 | KNERD | theer is even a bug reported with Digium |
| 07:45.33 | KNERD | still open after 1.5 years |
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| 10:05.12 | KNERD | yeah I think that provoder is borked |
| 10:05.44 | r00f | still no luck? |
| 10:05.48 | KNERD | no |
| 10:06.06 | KNERD | i dropped to 1.8.12 anmd same results which tells me it is their service |
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| 10:09.27 | KNERD | 1.8.12 was the last version which would always function on any virtual system without problems |
| 10:10.20 | KNERD | Even the freeswitch people admit they have problems with KVM someones on certain CPUs |
| 10:10.33 | KNERD | time to canel service |
| 10:12.20 | mjt | what problems pops up in a virtual environment? |
| 10:13.15 | r00f | his asterisk cannot load modules on hosted vps |
| 10:13.56 | mjt | this is not an inheret problem of virtualization |
| 10:15.12 | r00f | i could not say alot on this topic. just stated that i never had any problems with vps |
| 10:15.41 | r00f | if there is someone who can help online now, it will be insteresting to see the solution |
| 10:16.44 | mjt | "unable to load modules" smells like a broken install - like missing libraries, or library version (runtime vs compile-time) mismatch, that sort of things |
| 10:17.57 | mjt | reads a slideshare.net comment... |
| 10:18.14 | KNERD | but its not |
| 10:18.30 | KNERD | i have built asterisk on variosu system probably over 100 times |
| 10:19.20 | KNERD | i have a sheet I follow I have put together over the years |
| 10:19.38 | KNERD | and if the build goes wrong, it;s not me |
| 10:19.54 | mjt | but it's not about virtualization either :) |
| 10:20.05 | KNERD | oh yes it is |
| 10:20.16 | mjt | heh |
| 10:20.29 | mjt | which virt solution they're using? |
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| 10:20.37 | r00f | slideshare.net comment is interesting, but slightly irrelevant |
| 10:20.43 | KNERD | i was using KVM |
| 10:20.49 | mjt | ok, so kvm |
| 10:21.07 | mjt | kvm emulates whole _hardware_ (cpu, memory, hdds, network cards etc) |
| 10:21.19 | mjt | the rest is a regular OS install (linux/windows/whatever) |
| 10:21.41 | mjt | when compile/build process fails somehow, it is not a _hardware_ problem |
| 10:21.44 | mjt | it is an OS problem |
| 10:22.13 | *** join/#asterisk Corey84 (~Corey84@50.242.210.195) |
| 10:22.18 | mjt | like, again, wrong/missing libs, broken compiler, etc |
| 10:22.41 | mjt | hardware (virtual or not) just can not break compilation this way |
| 10:22.54 | KNERD | um..I been building asterisk on virtual systems for 4 years now |
| 10:23.47 | mjt | it doesn't really matter if it is a virtual system or not. for _build_ process anyway |
| 10:24.05 | KNERD | yes it does matter |
| 10:24.09 | eirirs | <-- no problems with vmware or hyper-v |
| 10:24.15 | mjt | virtual part comes to play when you talk about some performance characteristics, especially in realtime priorty |
| 10:24.18 | mjt | heh |
| 10:24.21 | mjt | it does how, exactly? |
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| 10:25.03 | KNERD | because there is a big report which has been open with digium for nealryl 2 years dealing with virtual environments |
| 10:25.07 | KNERD | *bug |
| 10:25.10 | mjt | it looks more like you don't understand how build process works if you say that. It is a regular system (say, linux), with the same kernel, libraries, compiler and stuff. |
| 10:25.32 | mjt | there might be a bugreport, but not about _building_ |
| 10:25.38 | KNERD | lo |
| 10:25.55 | mjt | care to show it to me please? |
| 10:26.13 | KNERD | i am looking for it now |
| 10:26.51 | mjt | it is sort of like saying that intel cpus are broken because asterisk has problems building on intel machines, while amd cpu works fine |
| 10:28.42 | mjt | well, i can imagine a _build_ time issue too. I remember a fun issue in gmp (gnu multi-precision math lib): they had a built-in list of CPU types and, depending on cpuid result, went to one or another branch, instead of looking at cpu capability bits. As a result, gmp had issues running in kvm because gmp just didn't know "qemu" cpu. |
| 10:29.55 | mjt | so, an incorrect optimization like this can lead to probs in a virtual environments, but it is not because virt environments are broken or even _different_, but because of a wrong optimization. |
| 10:31.58 | mjt | (actually that gmp issue was a bit more twisted, but the root cause is still this: wrong approach taken by gmp) |
| 10:36.40 | KNERD | mjt: https://issues.asterisk.org/jira/browse/ASTERISK-20128 |
| 10:36.55 | mjt | looking... |
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| 10:39.11 | mjt | this smells like a good combination of several issues |
| 10:39.31 | KNERD | yes, and gcc is one of them |
| 10:40.57 | mjt | the linked gcc bug shows it is not virt-specific problem |
| 10:41.13 | mjt | and what i _suspect_ here is something similar to what gmp did, really |
| 10:41.41 | mjt | judjing from that this issue is - apparently - happens more often in virt environment than native |
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| 10:42.32 | KNERD | yes, as I have had the same think happen on a Pentium 4 system I was building...i had to put asterisk 1.8.12 for it function properly |
| 10:42.43 | mjt | something like it tries to determine the "best" instruction set to use at runtime, not finding a "known" cpu, falls back to a generic code which is less tested than others, and hits the bug there |
| 10:43.11 | r00f | mjt: still there are alot of people who never run into these kind of problems. which means that either the gmp-related bug is very very selective in software involved, or it is not only software fault |
| 10:43.17 | KNERD | and all that mtune, etc doe snot help either |
| 10:43.19 | mjt | just wild guessing ofcourse, it is impossible to say more without good analying |
| 10:44.14 | KNERD | a lot of peopel I know has been held at atserisk 1.8.x |
| 10:44.33 | mjt | r00f: this is redhat kvm at least, and i know their qemu relatively well, it is a definitive "invalid instruction" generated by the real cpu from the code used by the guest |
| 10:44.41 | r00f | personally I was running 1.6, 1.8 and 11 in kvm |
| 10:44.47 | r00f | 5 different host providers |
| 10:44.50 | r00f | no problems |
| 10:44.53 | KNERD | though..i was able to build a asterisk 12 system on KVM a month ago |
| 10:46.11 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 10:46.54 | mjt | kvm runs a regular OS inside, which is 100% identical to non-virtualized one. It may not emulate some more exotic CPU instructions, but that's basically about it. When there's a bug in there, in 99.9% times it manifests by whole guest stalling or qemu crashing. |
| 10:47.05 | KNERD | though I see digium is working on it...11.9 now is able to dected a vitual environment and the "Build_native" option is XXX |
| 10:47.27 | mjt | wonders how distributions are building asterisk... |
| 10:48.12 | KNERD | schoomze is build their own asterisk now and centos...it works great on KVM wihtout issue |
| 10:48.27 | KNERD | and on OpenVZ |
| 10:48.46 | KNERD | i would have to ask exactly what they are doing |
| 10:49.01 | mjt | https://buildd.debian.org/status/fetch.php?pkg=asterisk&arch=i386&ver=1%3A11.8.1~dfsg-1&stamp=1394523605 |
| 10:49.09 | mjt | one of the examples of a build log |
| 10:49.18 | KNERD | because as a test I tried the PEL build of asterisk, and it fails also |
| 10:49.22 | KNERD | *EPEL |
| 10:49.55 | mjt | so debian builds it with explicit CFLAGS |
| 10:49.57 | mjt | CFLAGS="-g -O2 -fstack-protector --param=ssp-buffer-size=4 -Wformat -Werror=format-security -D_FORTIFY_SOURCE=2" |
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| 10:52.32 | KNERD | i guess I can try that |
| 10:52.33 | mjt | KNERD: tried to build it with debugging (-g) and see where it fails? |
| 10:53.08 | mjt | ie, run it under gdb, it will catch the signal and you can see where it is, by asking a backtrace (bt command) |
| 10:53.14 | KNERD | too late..i cancelled th service |
| 10:53.38 | KNERD | i will go with another provider I used before where I know asterisk was working great |
| 10:53.49 | mjt | oh. but you had another issue, with modules being unable to load |
| 10:53.59 | KNERD | it is similar |
| 10:54.00 | mjt | which is very likely somethig else |
| 10:54.05 | KNERD | no |
| 10:54.08 | mjt | um? |
| 10:54.09 | KNERD | it is virtual related |
| 10:54.17 | mjt | heh |
| 10:54.26 | KNERD | as I said, if my guide fails.. then it is not my doing |
| 10:54.34 | KNERD | i have this down to a T |
| 10:54.42 | mjt | it just can _not_ be virt-related, that's what i'm telling you since the beginning :) |
| 10:55.05 | KNERD | yes, but is occurs mor eon virtual |
| 10:55.33 | KNERD | another provider I just mentoned, asterisk worked first try |
| 10:56.14 | KNERD | i dropped them because they have an application scanner..it would not allow rsyslog to run |
| 10:56.38 | r00f | btsync is better anyways ;) |
| 10:57.29 | mjt | we had a bug in busybox. Which never occured until some completely unrelated change went in. That change was resizing one buffer by one (because it was one byte too small). As a result, other variables moved one byte down too. And as a result, in another place which accessed another, unrelated, variable, which just happened to come after the one which we resized, we started getting an unaligned access exception. |
| 10:58.23 | mjt | now one can say that this resizing made busybox buggy.. but it isn't :) |
| 10:58.44 | KNERD | btsync? |
| 10:58.59 | KNERD | that does logging? |
| 10:59.09 | r00f | oh my |
| 10:59.16 | r00f | i've read it as rsync |
| 10:59.17 | r00f | nvm |
| 10:59.25 | r00f | is out for coffee |
| 10:59.27 | KNERD | hah |
| 11:00.16 | KNERD | so busybox had terrible memory managementr |
| 11:00.53 | mjt | well it's not a memory management, but yes, those access methods were written without thinking about unaligned access implications |
| 11:01.32 | mjt | and the buffers were explicitly declared PACKED to be placed at any, even odd, addresses, to save space. |
| 11:02.22 | r00f | size does matter, after all |
| 11:02.38 | mjt | that's premature optimization, classic |
| 11:04.26 | KNERD | oh wait...the service is still u..i can try with CFLAGS="-g -O2 -fstack-protector --param=ssp-buffer-size=4 -Wformat -Werror=format-security -D_FORTIFY_SOURCE=2" |
| 11:04.36 | mjt | but anyway. the gmp example actually can be treated as an example of why virt environments are "worse" than non-virtual: this buggy code in gmp was old, and it was used many many times on all regular CPUs out there, and worked fine. Because the devs added all interesting CPU families and vendors to there. But they didn't add qemu cpu, maybe because qemu even didn't exist at the time |
| 11:04.49 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 11:05.33 | mjt | so, qemu (aka a virt environment) can be treated here as an example of "new" cpu vendor, which isn't necessary worse, but which just isn't _known_ to existing software, and existig software fails in there. |
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| 11:06.46 | mjt | it might be a much better cpu than intel (from some point of view, say, energy efficient), but for a potential customer it is worse because his existing software does not work on it |
| 11:11.49 | KNERD | no change |
| 11:11.57 | mjt | ? |
| 11:12.02 | KNERD | CFLAGS="-g -O2 -fstack-protector --param=ssp-buffer-size=4 -Wformat -Werror=format-security -D_FORTIFY_SOURCE=2" |
| 11:12.13 | mjt | yeah, and what did you expect it to change? |
| 11:12.17 | r00f | try gdb-ing it if you have -g now |
| 11:12.21 | mjt | what prob you're trying to solve? |
| 11:12.29 | KNERD | nothing |
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| 11:12.31 | KNERD | I gave up |
| 11:12.40 | KNERD | time to move on to another provider |
| 11:13.01 | mjt | you said you already cancelled contract, no? |
| 11:13.34 | KNERD | yes, but it has not been cancelled yet by them so the server is still up |
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| 11:18.03 | KNERD | 2 years ago i was using OpenVZ form a provider without a problem, but I was getting too much latency due to the location was too far away |
| 11:18.33 | KNERD | then they said htey have a new location..so i justed to that one...i get better results |
| 11:19.03 | KNERD | then later they announced a newer location which was even closer...still using OpenVZ |
| 11:19.14 | KNERD | now all of a sudden Asterisk is broke |
| 11:19.44 | KNERD | turns out they were using a much newer version of OpenVZ at this new location |
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| 11:27.41 | r00f | gents, are there any "IVR best practices" to know? |
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| 11:35.47 | iulhk | $AGI->get_data($filename, $timeout, $maxdigits), unable to get dtmf by using this agi command in asterisk-10.8.0. can anybody guide? |
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| 12:05.59 | andycol | Hi All |
| 12:06.29 | andycol | can anyone assist me to change the inbound cid to display a random number and then log it as that number in the cdr table? |
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| 12:07.38 | r00f | hiding calls from wife? |
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| 12:09.42 | davlefou | I have reboot my server since my sip have message : 404, server not found |
| 12:10.00 | *** join/#asterisk huorpalantir (~atrus@D57DD0BA.static.ziggozakelijk.nl) |
| 12:10.09 | davlefou | i whose login in with ssh. |
| 12:11.06 | *** part/#asterisk huorpalantir (~atrus@D57DD0BA.static.ziggozakelijk.nl) |
| 12:11.30 | andycol | lol no for my office that want to take that uniqueid and use it as a reference for customers that called in |
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| 12:14.16 | r00f | random numbers are not so good for reference, for they are pseudorandom. if you dislike asterisk's uniqueid, you could append unixtime to the cid, for example |
| 12:15.05 | davlefou | it works in iax but not in sip |
| 12:15.06 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 12:15.37 | andycol | is it possible then to look up into a mysql database and use a reference from there? |
| 12:16.00 | andycol | how do i do the unixtime to the cid? |
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| 12:18.06 | r00f | look at strftime func |
| 12:19.09 | iulhk | $AGI->get_data($filename, $timeout, $maxdigits), unable to get dtmf by using this agi command in asterisk-10.8.0. can anybody guide? |
| 12:19.32 | andycol | ok and to look up a number from the db? |
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| 12:59.32 | Katty | morning |
| 13:03.22 | Chainsaw | ltns Katty :) |
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| 13:32.24 | davlefou | Hi, since i have reboot my server sip don't work, network seems block it but iax is ok |
| 13:42.29 | [TK]D-Fender | Then unblock your network |
| 13:43.04 | *** join/#asterisk bhavikpatel6842 (~root@122.169.36.229) |
| 13:44.10 | bhavikpatel6842 | Hi All |
| 13:45.25 | *** join/#asterisk raspberrypifan (~textual@71-22-220-224.gar.clearwire-wmx.net) |
| 13:46.47 | raspberrypifan | who can dial a sip number for me |
| 13:48.47 | *** part/#asterisk armvoip (~arpit@122.169.36.229) |
| 13:48.54 | bhavikpatel6842 | I have an issue with get_variable php AGI function. |
| 13:50.36 | bhavikpatel6842 | Currently I am using phpagi.php,v 2.14 but not getting proper value in that function.c |
| 13:50.57 | davlefou | [TK]D-Fender, how? my iptable seems ok |
| 13:51.22 | [TK]D-Fender | davlefou: You said "seems", but haven't shown us anything |
| 13:51.55 | bhavikpatel6842 | can any one idea about PHP-AGI get_variable function. |
| 13:53.10 | [TK]D-Fender | bhavikpatel6842: Show us your actual code. So far tons of projects use it just fine and AGI's spec hasn't changed |
| 13:53.23 | [TK]D-Fender | bhavikpatel6842: And show us the failed call |
| 13:53.27 | *** part/#asterisk bhavikpatel6842 (~root@122.169.36.229) |
| 13:53.32 | davlefou | http://pastebin.com/j21AnpdW |
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| 13:53.42 | raspberrypifan | [TK]D-Fender cna you call a sip number for me, i need to know if it still work |
| 13:53.48 | *** join/#asterisk bhavikpatel6842 (~root@122.169.36.229) |
| 13:55.04 | bhavikpatel6842 | I am just set variable using callfile and showing in Asterisk CLI with same value. |
| 13:55.16 | [TK]D-Fender | bhavikpatel6842: Show us something we can debug. |
| 13:55.44 | bhavikpatel6842 | but when trying to get that variable like $agi->get_variable('VAR123'); |
| 13:55.56 | [TK]D-Fender | bhavikpatel6842: Actual full code, and the failed call and supporting bits |
| 13:56.45 | bhavikpatel6842 | if I set 2 varaible in callfile and generate call |
| 13:56.45 | bhavikpatel6842 | $str .= "Set:__WBCLID=$callerid\n" ; |
| 13:56.46 | bhavikpatel6842 | $str .= "Set:__DIALEDNUMBER=$phonenumber\n" ; |
| 13:57.01 | bhavikpatel6842 | then in answer context I called one AGI file |
| 13:57.04 | [TK]D-Fender | bhavikpatel6842: Full... not tiny snippets like thta. |
| 13:57.06 | [TK]D-Fender | ~pb |
| 13:57.06 | 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. |
| 13:57.08 | [TK]D-Fender | ^^^^^ |
| 13:57.21 | bhavikpatel6842 | okey 1 min please |
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| 14:00.19 | davlefou | [TK]D-Fender, some ideas? |
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| 14:01.20 | [TK]D-Fender | davlefou: I see "serious network error" on every packet... and no firewall dump... |
| 14:01.52 | [TK]D-Fender | davlefou: And no confirmation of the IAX2 part you mentioned. |
| 14:02.17 | slykens | Hi all - any known issues with dahdi and ubuntu kernels > 3.11.0-19? Performed an upgrade over the weekend and my te410p just doesn't seem to want to talk to my pri provider anymore. Roll back does not appear to resolve it either :? |
| 14:04.11 | davlefou | i have made test call with iax2 |
| 14:05.22 | [TK]D-Fender | slykens: Have you recompiled DAHDI for your new kernel? |
| 14:05.32 | bhavikpatel6842 | [TK]D-Fender = here is my log and code--http://pastebin.com/eAYR3V7W |
| 14:05.50 | bhavikpatel6842 | Please help if you found any issue ? |
| 14:06.05 | davlefou | [TK]D-Fender, iax sound test in debug mode http://pastebin.com/7hTHPA4V |
| 14:06.13 | bhavikpatel6842 | if you want AGi file then also I can provide you in pastebin. |
| 14:07.44 | slykens | d-fender - Yes, libpri, then dahdi, then asterisk. Have tried rolling back the kernel and asterisk as well to no avail. Right now dahdi just doesn't seem like it's talking to the card - won't detect internal or external loops. Other times it's a ton of various MDL errors and broken half functionality. |
| 14:07.53 | [TK]D-Fender | I don't see that variable being set, and I see output I don't see in the code |
| 14:08.13 | [TK]D-Fender | bhavikpatel6842: ^^ |
| 14:08.56 | bhavikpatel6842 | As you can see variable set in SIPCALLID=NzY3MTdmMTVhN2YyMzFkYmMwZGNjYTljOTQzMTBlOWI. which get in another variable  $wbtoken=$this->obj_agi->get_variable('WBCLID'); |
| 14:09.07 | bhavikpatel6842 | and I just print that variable |
| 14:09.33 | [TK]D-Fender | WBCLID <- I don't see it exist |
| 14:09.47 | bhavikpatel6842 | like I have issue if I want SIPCALLID then get that variable in SIPDOMAIN.you can check in pastebin. |
| 14:09.56 | [TK]D-Fender | demo/demo.php: ******************** WELCOME TO IVR ********************** <- I don't see why this is printed at all. |
| 14:10.14 | bhavikpatel6842 | I just print that variable in asterisk CLI |
| 14:10.15 | [TK]D-Fender | bhavikpatel6842: You have called for a variable that I don't see being set anywhere |
| 14:10.29 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 14:10.49 | bhavikpatel6842 | giving you new pastebin. |
| 14:12.28 | [TK]D-Fender | bhavikpatel6842: http://pastebin.com/eAYR3V7W <- line 56 shows a result there... |
| 14:13.05 | [TK]D-Fender | bhavikpatel6842: How it got that value I don't know yet, but I see it there. |
| 14:14.12 | bhavikpatel6842 | Yeah that was the issue please check my new logs. |
| 14:14.27 | [TK]D-Fender | What new log? You need to give us the link... |
| 14:15.42 | bhavikpatel6842 | http://pastebin.com/fpPZs3Zv |
| 14:15.50 | bhavikpatel6842 | yes please check. |
| 14:16.01 | bhavikpatel6842 | all variable getting in different variables. |
| 14:16.26 | [TK]D-Fender | <PROTECTED> |
| 14:17.02 | bhavikpatel6842 | TOken getting in TRUNKNAME ,outboundflag getting in DIALEDNUMBER etc.please check. |
| 14:17.15 | bhavikpatel6842 | <PROTECTED> |
| 14:17.15 | bhavikpatel6842 | <PROTECTED> |
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| 14:17.55 | bhavikpatel6842 | Getting my issue ? |
| 14:18.57 | [TK]D-Fender | IS that not the Show me where you are setting all of these, and comment out the verbose and actuall enable AGI DEBUG. |
| 14:19.11 | [TK]D-Fender | You aren't looking from the right place to start |
| 14:20.02 | *** join/#asterisk makubi (~makubi@xdsl-81-173-238-140.netcologne.de) |
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| 14:22.40 | *** part/#asterisk bhavikpatel6842 (~root@122.169.36.229) |
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| 14:50.45 | workingcats | KNERD, bit late, but i am also running asterisk in a VM, v8 on top of vmware esxi (planning to migrate to kvm "when i have time") |
| 14:51.05 | workingcats | it's not very high load though |
| 14:52.22 | workingcats | when i set it up i made a quick test with 100 parallel SIP calls (99 asterisk to sipgate, plus 1 softphone to asterisk to sipgate to check if quality is good) |
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| 15:34.17 | *** part/#asterisk Alex_Bkash (~atique@124.6.236.252) |
| 15:34.59 | leifmadsen | KNERD: didn't see the initial message, but we run hundreds of Asterisk boxes of various capacities on VMware ESXi |
| 15:35.06 | leifmadsen | Asterisk 1.4 and 1.8 |
| 15:35.08 | leifmadsen | fyi |
| 15:35.51 | leifmadsen | what ends up happening is that we actually tune it to use less CPU and RAM and not more. I think they are around 1GB RAM and 2 vCPU right now (down from 3GB and 4 vCPU) |
| 15:36.02 | leifmadsen | I'm not in the infrastucture team though |
| 15:37.19 | *** join/#asterisk nix8n82 (~AndChat27@24.143.11.65) |
| 15:38.25 | KNERD | Hmmm...thanks.. |
| 15:39.15 | KNERD | I am supposwe to get a server maybe in 2 weeks with Xeomn Processors..guess I will give this ESXi a try..that is thrid person to say they had god success |
| 15:39.38 | Chainsaw | KNERD: ESXi is nice, but not cheap if you want to use it properly. |
| 15:41.04 | KNERD | they do have afree version if I am not mistaken |
| 15:45.01 | KNERD | I have built asterisk 12 on Virtual box with sucess, but I dont think there is a server version |
| 15:45.03 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 15:45.05 | Chainsaw | KNERD: Yes. And none of the features you care about for virtualisation will work. |
| 15:45.14 | Chainsaw | KNERD: Like migrating from one box to the other. |
| 15:47.08 | KNERD | i personally dont need that at the moment |
| 15:57.14 | marceloamorim | Hey guys, I`m trying to use the cel instead the cdr, but I opened the master.csv on the openoffice calc and I didn`t understand very well |
| 15:57.48 | marceloamorim | there is any program that can analyse those files for me? |
| 16:00.00 | KNERD | looks like VMware ESXi is toast |
| 16:00.23 | KNERD | and being replaced with VMware vSphere Hypervisor |
| 16:00.47 | KNERD | marceloamorim: cel? |
| 16:00.54 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:01.22 | BCS-Satori | Hey all, is it possible to schedule a cron job or a job within asterisk to execute a "sip notify polycom-check-cfg XXX" each night? We are running into an issue with Polycom where the 650 phones have some memory leak with BLF and crash after 6-7 days of running. Polycom is working on a fix but I am trying to find a temporary solution to bypass the crashing as reboot the phones seems to solve the issue. |
| 16:01.44 | marceloamorim | yeah, channel event logging ( CEL ) |
| 16:05.37 | BCS-Satori | Nevermind; figured it out can use "asterisk -rx "sip notify polycom-check-cfg 6125"" from SSH so I can create a cron job in linux |
| 16:06.08 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 16:08.24 | KNERD | marceloamorim: this may help...it's give an example and the corresponding columns: https://wiki.asterisk.org/wiki/display/AST/CEL+Design+Goals |
| 16:12.37 | *** part/#asterisk tdonahue (~tdonahue@vonmail.vonworldwide.com) |
| 16:17.05 | marceloamorim | I`ll set events=all to see whats going on |
| 16:18.13 | marceloamorim | but I`ll still need some analyser to check the calls with the cel |
| 16:18.30 | marceloamorim | I`ll take a break for lunch, brb |
| 16:30.07 | davlefou | [TK]D-Fender, i have found the problem |
| 16:33.02 | [TK]D-Fender | davlefou: Ok.... |
| 16:33.39 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 16:34.41 | davlefou | [TK]D-Fender, thanks for you help |
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| 16:37.12 | [TK]D-Fender | davlefou: Can't really take any credit... You hadn't shown anything, so I had nothing to comment on. |
| 16:40.59 | *** join/#asterisk retentiveboy (~retentive@c-50-150-62-29.hsd1.ga.comcast.net) |
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| 16:51.57 | davlefou | [TK]D-Fender, You have trie to help me, that is the more important |
| 16:52.39 | [TK]D-Fender | davlefou: I suppose there's that... |
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| 16:53.26 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 17:55.30 | jwwwww_ | good evening people. |
| 18:02.50 | ruel | good evening jwwwww_ |
| 18:18.21 | *** join/#asterisk armvoip (~arpit@115.242.79.137) |
| 18:18.47 | armvoip | Hi Everyone :) |
| 18:19.22 | armvoip | I am trying to install Asterisk 11.9.0 on my VPS with OS centos 6.5 |
| 18:19.46 | armvoip | everything goes well when i do ./configure |
| 18:20.02 | armvoip | but while doing make, it throws some errors in channel modules. |
| 18:21.46 | [TK]D-Fender | ~pb |
| 18:21.46 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 18:21.48 | [TK]D-Fender | ^^^^^ |
| 18:23.00 | armvoip | sure |
| 18:23.00 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 18:26.55 | armvoip | http://pastebin.com/qEGJxFXp |
| 18:29.39 | *** join/#asterisk Penguin (~xwQ5kwYl6@20264.odci.gov.united-states.rltk.us) |
| 18:29.47 | [TK]D-Fender | I'd recommend trashing your extracted folder, and repeat the whole process and pastebin the whole thing. |
| 18:30.00 | jwwwww_ | I'm looking for a documentation about how to setup a basic asterisk with a diaplan made with ael. does someone have a suggestion ? the server run debian wheezy. |
| 18:30.25 | [TK]D-Fender | And since you're on a VPS odds are you won't be able to compile DAHDI anyway... you could use menuselt to disable it before running make |
| 18:30.26 | armvoip | okay sure, let me try again. |
| 18:30.37 | [TK]D-Fender | ~asteriskwiki |
| 18:30.37 | infobot | i guess asteriskwiki is http://wiki.asterisk.org |
| 18:30.39 | [TK]D-Fender | jwwwww_: ^^^^^ |
| 18:30.50 | armvoip | yeah i tried disabling dahdi, but same error came in chan_sip.c |
| 18:30.55 | armvoip | let me try again |
| 18:31.03 | [TK]D-Fender | jwwwww_: Almost nobody here use AEL and help will be hard come by... |
| 18:31.24 | [TK]D-Fender | armvoip: don't forget to pastebin it from scratch |
| 18:31.54 | armvoip | sure, thanks |
| 18:32.22 | jwwwww_ | [TK]D-Fender: I found a bit of documentation in the wiki, but not too much how to setup the dialplan. |
| 18:32.42 | [TK]D-Fender | That is ALL that AEL is... dialplan |
| 18:33.58 | jwwwww_ | [TK]D-Fender: umm I just get a new task, I must add some functionality to an existing asterisk server. I know the very basic of asterisk, but nothing about ael.infortunatly the dialplan is made with ael. |
| 18:34.21 | [TK]D-Fender | https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=4620445 |
| 18:34.25 | [TK]D-Fender | The syntax is all there... |
| 18:34.35 | [TK]D-Fender | Everything else is dialplan apps and each of those is documented too |
| 18:35.32 | jwwwww_ | I go read those docs again, I must have missed stuff. |
| 18:36.09 | jwwwww_ | thanks ! |
| 18:36.42 | [TK]D-Fender | You're welcome |
| 18:46.44 | *** join/#asterisk woopstar (~woopstar@unaffiliated/woopstar) |
| 18:47.22 | armvoip | D-Fender: That worked, i deleted that extracted source and extract again and tried same steps again and it got worked. Install went fine but while starting asterisk its throwing below errors: |
| 18:47.24 | armvoip | http://pastebin.com/qmgTYvwG |
| 18:47.28 | woopstar | Hi. We're trying to use the webrtc2sip to enable a web client to call. But when connecting it suddenly just says Wrong Password. But we hare 200% sure that we type the correct password. Any clues ? |
| 18:49.04 | ruel | woopstar: run a sip debug and pastebin |
| 18:49.13 | woopstar | ruel: Give me 2 seconds. |
| 18:50.49 | woopstar | ruel: http://pastebin.com/CX7WtRAK |
| 18:52.02 | woopstar | I'll have them verify the md5secret password in the database again, just to be more than sure.. |
| 18:53.09 | ruel | yeah i would do that and also check that it is inserted properly into the right places in the client config---sometimes it's tricky there. |
| 18:53.51 | woopstar | And md5secret is computed by: MD5(name:realm:password) - right? |
| 18:56.13 | ruel | hmm, not positive about that exactly---it may use another parameter too but I'm not sure. |
| 18:57.12 | ruel | does your realm match on both sides though? |
| 18:57.50 | woopstar | lol |
| 18:57.57 | woopstar | im so gonna fire a person tomorrow |
| 18:58.05 | ruel | if u have ur realm name set in sip.conf, but your client is connecting to an ip addr it can bugger it up. |
| 18:58.08 | woopstar | "Yes I've checked the MD5 has ten times" |
| 18:58.11 | woopstar | .... |
| 18:58.13 | woopstar | it was wrong |
| 18:58.34 | ruel | lol well i guess that straightens it out then. |
| 18:59.01 | [TK]D-Fender | armvoip: Start * manually, not with the script |
| 18:59.28 | [TK]D-Fender | armvoip: "asterisk -U yourasteriskuser -G yourasteriskgroup -gvvvvvvvvvvvvc" |
| 19:00.04 | armvoip | okay as its default install, user and group should be "asterisk" right? |
| 19:00.13 | *** join/#asterisk navaismo (~navaismo@200-52-45-221.dynamic.axtel.net) |
| 19:00.32 | *** join/#asterisk hehol (~Adium@ip-78-94-76-253.unitymediagroup.de) |
| 19:01.07 | ruel | armvoip: they can be whatever u want as long as they exist on the system i thot. |
| 19:01.14 | armvoip | Says "Illegal instruction" |
| 19:01.33 | navaismo | anyone familiar with the busy/congestion tones from nortel |
| 19:02.15 | navaismo | cant detect the hangup |
| 19:02.29 | [TK]D-Fender | armvoip: Show us the whole attempt |
| 19:03.24 | armvoip | sure |
| 19:05.46 | armvoip | http://pastebin.com/hFYR3DUq |
| 19:05.51 | armvoip | Please see this from starting point. |
| 19:05.57 | armvoip | installation was perfect. |
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| 19:40.31 | [TK]D-Fender | armvoip: Yup, no idea on this one... |
| 19:41.10 | armvoip | no problem, thanks. i am trying something, I will let you know. |
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| 19:54.45 | armvoip | D-Fender: http://stackoverflow.com/questions/19607378/illegal-instruction-error-comes-when-i-start-asterisk-1-8-22 |
| 19:54.59 | armvoip | found solution here and recompiled it. It WORKS now !!! |
| 19:55.35 | armvoip | Solution: enter in "make menuselect" -> "Compiler flags" and disable "BUILD_NATIVE" option; then recompile Asterisk |
| 19:55.51 | armvoip | i hope it wont affect anything. |
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| 19:56.41 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 19:56.58 | [TK]D-Fender | armvoip: I have seen that referenced before but never had to deal with it myself |
| 19:57.32 | armvoip | oh okay, anyways asterisk is started now. I will check more and lets see how it goes. |
| 19:59.26 | *** part/#asterisk armvoip (~arpit@115.242.79.137) |
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| 20:13.13 | *** join/#asterisk BCrookAtRA (~bcrook@2001:470:1f11:942::443) |
| 20:15.21 | BCrookAtRA | I'm trying to play a recording to my callers that contains all ten touchtones (the goal is to trick autodialer robots into thinking we pressed whatever digit they say to press to be removed). But the touchtones always come out sounding clipped as if they were too loud, no matter how quiet I make them, or howlong i play them. Does asterisk 'intercept' touchtones somehow? |
| 20:16.07 | BCrookAtRA | CIT tones play normally as expected |
| 20:16.16 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 20:16.16 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 20:16.18 | BCrookAtRA | and they are much louder even |
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| 20:24.27 | navaismo | heeeeeeeeeeeeeeeeelp |
| 20:24.46 | BCrookAtRA | navaismo: thats going to be quite effective, I'm sure |
| 20:25.34 | navaismo | while(true){print "help please"} |
| 20:26.11 | BCrookAtRA | sorry. I feel like you're missing a semicolon |
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| 20:27.36 | *** join/#asterisk shuggans (~shuggans@ucomnetworks-209-203-168-138.ucom.net) |
| 20:27.36 | shuggans | Hi all - I have an issue I'm not sure how to ask about. I've set up a pbxinaflash box and ordered a grandstream GXP2140 to test out. I set up tagged interfaces for vlans, but on the freepbx status page my voice interface only shows traffic received, not transmitted. |
| 20:28.28 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 20:35.19 | navaismo | :'( :'( |
| 20:36.25 | navaismo | please help asap |
| 20:36.36 | *** join/#asterisk [sr] (~kvirc@pal-213-228-163-73.netvisao.pt) |
| 20:36.37 | litn | what's up navaismo |
| 20:36.52 | [sr] | guys |
| 20:37.06 | navaismo | no hangup between nortel opt11c and GXW4108 |
| 20:37.07 | [sr] | it should be released a new version of libpri |
| 20:38.59 | *** join/#asterisk MrJoshGeddes (storm3y@unaffiliated/storm3y) |
| 20:44.23 | navaismo | :( :'( |
| 20:44.36 | navaismo | cant fix it please fix my setup |
| 20:45.16 | navaismo | Katty, fix asap required or a cookie |
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| 20:51.31 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 20:55.12 | navaismo | no cookies :'( |
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| 22:14.53 | swiss__ | ahoy hoy all |
| 22:15.04 | swiss__ | quick very nooby question |
| 22:20.04 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 22:20.38 | newtonr | ~ask |
| 22:20.38 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 22:20.42 | swiss__ | im trying to get this example: http://www.voip-info.org/wiki/index.php?page_id=2905 to run but no luck |
| 22:20.58 | swiss__ | didnt want to ask a room full of afk folks |
| 22:21.43 | swiss__ | i can dial 9999+mycell and get a call but it does not do the follow me part |
| 22:21.56 | swiss__ | numbers have been edited to reflect my test numbers |
| 22:22.24 | swiss__ | sorry, i should specify im trying the second example on that page |
| 22:22.25 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:23.54 | *** join/#asterisk g_r_eek (~g_r_eek@176.92.76.43) |
| 22:23.56 | newtonr | swiss__, you'll need to pastebin an asterisk log with verbose so we can see where it is failing. |
| 22:24.11 | swiss__ | will do |
| 22:24.39 | newtonr | swiss__, https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 22:24.58 | newtonr | probably dont need the 'debug' channel type |
| 22:27.28 | newtonr | swiss__, note there is also app_followme, https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_FollowMe which doesn't appear to be used in the example you are following |
| 22:27.55 | newtonr | just randomly adding other options for you |
| 22:29.21 | swiss__ | haha, can always use more options |
| 22:29.35 | swiss__ | wow that debug info is fairly extensive |
| 22:30.07 | newtonr | yeah the debug channel can make it real noisy if you are not looking for something low level |
| 22:31.37 | swiss__ | i cant really pass this along, there is far too much info considering this is a work setup |
| 22:32.02 | newtonr | Then you are on your own, unless you can scrub it :) |
| 22:32.34 | swiss__ | hmmm, tempting as it would take a while AND look like work |
| 22:33.03 | newtonr | you could try reproducing the problem on a test system with two phones unrelated to your production system |
| 22:33.34 | newtonr | either way, if you get some sanitized debug post it here and someone may be able to help. i'm about to be gone in just a bit |
| 22:34.31 | swiss__ | fiar enough ill see what i can do |
| 22:34.43 | swiss__ | would the verbose cli output be of any use? |
| 22:34.51 | swiss__ | that will be far easier to sanitize |
| 22:35.42 | newtonr | yup, i said you don't really need the debug channel. just notice,warning,error,verbose are fine, but make sure verbose is turned up to 5 or higher |
| 22:36.01 | swiss__ | ah, misssed that part, doh |
| 22:36.07 | swiss__ | ill shoot that along in a second |
| 22:36.09 | newtonr | in the log file you should see actual messages prefixed with VERBOSE to be sure you are getting the output |
| 22:36.17 | newtonr | on the console it won't say VERBOSE |
| 22:36.38 | swiss__ | i see |
| 22:37.03 | newtonr | i say that, because logger.conf lets you direct log channels to both locations (console and files) |
| 22:37.07 | newtonr | any way :) |
| 22:37.57 | *** join/#asterisk Cubber (~ronny@mail.adirondackitsolutions.com) |
| 22:39.41 | swiss__ | http://pastebin.com/axzUScg9 |
| 22:40.40 | [TK]D-Fender | <PROTECTED> |
| 22:40.52 | swiss__ | ? |
| 22:40.59 | [TK]D-Fender | You are never ever supposed to do an IVR like that in a macro from Dial |
| 22:41.06 | navaismo | fender help! |
| 22:41.08 | [TK]D-Fender | You should only use Read() |
| 22:41.36 | swiss__ | im not sure where i set that? |
| 22:42.16 | [TK]D-Fender | this is the DIALPLAN.... |
| 22:42.19 | [TK]D-Fender | Clearly extensions.conf |
| 22:44.52 | swiss__ | http://pastebin.com/gbMsXUzN |
| 22:44.58 | swiss__ | theres my extensions.conf |
| 22:45.03 | swiss__ | no idea what you mean |
| 22:45.31 | [TK]D-Fender | line 13 |
| 22:45.39 | [TK]D-Fender | waitexten = bad |
| 22:45.43 | swiss__ | ah |
| 22:45.46 | [TK]D-Fender | do a read instead |
| 22:46.50 | swiss__ | so instead it will be: exten => s,n,Read(5) |
| 22:46.54 | swiss__ | do i have that right? |
| 22:46.58 | [TK]D-Fender | no |
| 22:47.13 | [TK]D-Fender | you should go read its instructions.... |
| 22:50.05 | swiss__ | exten => s,n,Read(digit|1||1|5) |
| 22:50.10 | swiss__ | something along those lines then |
| 22:50.46 | swiss__ | ? |
| 22:51.07 | navaismo | please he-elp |
| 22:51.31 | [TK]D-Fender | swiss__: | is not a valid delimiter since 1.2 |
| 22:51.55 | swiss__ | exten => s,n,Read(digit,1,,1,5) |
| 22:52.08 | [TK]D-Fender | better... |
| 22:54.37 | swiss__ | but still missing... |
| 22:54.41 | swiss__ | ? |
| 22:55.25 | [TK]D-Fender | I don't see an actual question there... |
| 22:55.59 | swiss__ | what am i missing, im staring at the wiki file for read and not seeing a problem |
| 22:56.43 | [TK]D-Fender | So maybe there isn't one |
| 22:57.39 | swiss__ | then why reply with an elipsies |
| 22:59.14 | [TK]D-Fender | I didn't |
| 22:59.38 | swiss__ | "<swiss__> exten => s,n,Read(digit,1,,1,5) |
| 22:59.38 | swiss__ | <[TK]D-Fender> better..." |
| 22:59.57 | [TK]D-Fender | that is your IRC client recognizing your nick |
| 23:00.06 | [TK]D-Fender | I didn't type them |
| 23:00.11 | swiss__ | hahaha fiar enough |
| 23:01.05 | swiss__ | new irc client to me |
| 23:02.25 | navaismo | ? |
| 23:03.21 | swiss__ | welp tried that and reloaded and still no follow up call regardless if i accept or deny it |
| 23:04.07 | [TK]D-Fender | swiss__: I haven't seen what else you di based on the read |
| 23:04.24 | [TK]D-Fender | And what is this about a "follow-up call"? |
| 23:05.20 | swiss__ | heres what im trying to make work, specifically the second example: http://www.voip-info.org/wiki/index.php?page_id=2905 |
| 23:05.58 | swiss__ | basically i want to all 1 extension and then have asterisk call multiple cell phones until someone answers |
| 23:06.16 | [TK]D-Fender | that wiki is full of buggy code samples |
| 23:06.53 | [TK]D-Fender | So go show us your new code and your new call attempt |
| 23:06.57 | navaismo | and someone are awesome |
| 23:07.32 | navaismo | when asterisk wiki fail voip-info has the answer(mostly old answers) |
| 23:09.09 | swiss__ | http://pastebin.com/Qsv5z2SG |
| 23:09.31 | swiss__ | note: i know the audio file isnt playing and doesnt work because its teh wrong format, thats fine |
| 23:09.38 | swiss__ | for now, that i cna manage |
| 23:09.59 | [TK]D-Fender | <PROTECTED> |
| 23:10.05 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 23:10.08 | swiss__ | good to know |
| 23:10.12 | [TK]D-Fender | And that should be a playback, not Background() |
| 23:11.33 | [TK]D-Fender | <PROTECTED> |
| 23:11.35 | [TK]D-Fender | [May 6 16:01:53] WARNING[19951]: file.c:663 ast_openstream_full: File 1 does not exist in any format |
| 23:12.00 | [TK]D-Fender | your read syntax was also incorrect as it was told to playback a sound file which you don't have |
| 23:14.26 | *** part/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 23:24.10 | swiss__ | alright fixed the read() command |
| 23:25.17 | swiss__ | http://pastebin.com/VJFZG2aQ |
| 23:25.24 | swiss__ | ND GET THIS NO MATTER HOW HARD I MASH BUTTONS |
| 23:25.31 | swiss__ | sorry missed the a there |
| 23:27.20 | [TK]D-Fender | Maybe your DTMF mode is wrong for your peer <- |
| 23:34.51 | *** join/#asterisk raspberrypifan (~textual@71-22-220-224.gar.clearwire-wmx.net) |
| 23:35.45 | raspberrypifan | hello |
| 23:37.09 | snadge | mornin |
| 23:37.55 | raspberrypifan | do you have a sip number ,i need to try to see if this number i have still connects |
| 23:40.17 | navaismo | 12345678@pbxdm.ath.cx |
| 23:40.32 | raspberrypifan | ? |
| 23:41.07 | navaismo | dont you ask for a sip number? |
| 23:41.19 | swiss__ | alright so that all seems to work, but i still cant get it to call the next phone number on the list |
| 23:42.17 | [TK]D-Fender | swiShow us the full config & call.... |
| 23:42.37 | raspberrypifan | i want you to call one for me |
| 23:42.39 | raspberrypifan | not to give me one |
| 23:43.18 | raspberrypifan | Direct SIP 728996 |
| 23:43.34 | navaismo | too bad tomorrow will die my domain i just need one last voice |
| 23:44.13 | navaismo | 728996 at wich domain? |
| 23:44.16 | raspberrypifan | hm |
| 23:44.17 | raspberrypifan | idk |
| 23:44.21 | raspberrypifan | http://www.voiprotel.net/voiprotel/index-4.html |
| 23:44.24 | raspberrypifan | its at the very bottom |
| 23:44.30 | raspberrypifan | the only place in Ecuador that might do SIP trunking |
| 23:44.31 | raspberrypifan | possibly |
| 23:44.35 | raspberrypifan | but they might be dead |
| 23:48.30 | navaismo | http://pastebin.com/xnqXLLAd |
| 23:48.42 | navaismo | try the pstn phones |
| 23:54.31 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 23:56.16 | raspberrypifan | so they might be dead |
| 23:58.57 | raspberrypifan | are there any other sip/voip irc channels |
