| 00:04.28 | *** join/#asterisk lorsungcu (~anonymous@50-78-230-69-static.hfc.comcastbusiness.net) |
| 00:09.37 | Kobaz | http://www.voipsupply.com/cisco-spa122 any good? |
| 00:09.55 | [TK]D-Fender | sure |
| 00:10.26 | Kobaz | need a little remote office setup for my brother with some cordless phones |
| 00:13.26 | Kobaz | what's a good cordless phone these days |
| 00:14.28 | Kobaz | http://www.newegg.com/Product/Product.aspx?Item=76-105-229&ParentOnly=1 |
| 00:14.33 | Kobaz | doesnt have to be fancy |
| 00:14.34 | *** join/#asterisk ruied (~ruied@po-217-129-254-134.netvisao.pt) |
| 00:16.21 | micmic- | Siemens Gigaset |
| 00:16.35 | micmic- | was working for me very well. |
| 00:16.43 | ruied | hello! I'm getting Sending fake auth rejection for device 1000<sip:1000@X.X.X.X X.X.X.X is my public IP, is this some bad configuration in asterisk or coul it be some kind of attack ? |
| 00:16.45 | Kobaz | k |
| 00:17.19 | micmic- | Kobaz: good price & decent quality - and you can pick from a mega simple AS180 to some more complex models with big displays etc. |
| 00:19.06 | *** join/#asterisk gusto (~gusto@ppp-93-104-64-229.dynamic.mnet-online.de) |
| 00:19.10 | [TK]D-Fender | ruied: No, it is SOP |
| 00:19.38 | [TK]D-Fender | ruied: As for being an attack... look at the IP it's coming FROM. |
| 00:20.24 | ruied | [TK]D-Fender, SOP? what is that? the strange thing to me is the IP is my router's public IP... |
| 00:20.44 | [TK]D-Fender | ruied: Show us the complete debug |
| 00:20.56 | [TK]D-Fender | Standard Operating Procedure. |
| 00:21.04 | micmic- | if someone feels powerful enough in 1.6.0.15 -> I have an issue with AMI Action: Bridge, pbx.c complains that it was sent to invalid extension, context 'extensions'. |
| 00:21.43 | [TK]D-Fender | micmic-: Show us complete debug and applicable configs |
| 00:21.59 | micmic- | [TK]D-Fender: on the way. |
| 00:22.02 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 00:22.12 | [TK]D-Fender | micmic-: And relaize that release if OLD within it's branch, and that branch AND the two the followed are both no longer supported at all |
| 00:22.37 | micmic- | WARNING[18430] pbx.c: Channel 'SIP/XXXXX' sent into invalid extension 'PHONE NUMBER' in context 'extensions', but no invalid handler. |
| 00:22.54 | micmic- | [TK]D-Fender: yes, I know that - but I cannot just upgrade - there's plenty of software written "around it" |
| 00:23.15 | [TK]D-Fender | ~pb |
| 00:23.15 | 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. |
| 00:23.17 | [TK]D-Fender | ^^^ |
| 00:23.50 | [TK]D-Fender | micmic-: and I'm pretty sure that error means exactly what it says... |
| 00:23.55 | ruied | [TK]D-Fender, http://pastebin.com/RQwB3p0G |
| 00:24.11 | [TK]D-Fender | ruied: that is not SIP DEBUG for the attempts |
| 00:24.17 | micmic- | [TK]D-Fender: yes, |
| 00:24.20 | [TK]D-Fender | ruied: Go to * CLI and start looking at it |
| 00:25.04 | ruied | ok |
| 00:25.59 | micmic- | [TK]D-Fender: but I am missing a link (logical) to make sure I get the point - if the PHONE NUMBER is not defined in extensions - because that PHONE NUMBER can be pretty much whatever |
| 00:26.14 | micmic- | [TK]D-Fender: how to proceed here? (a pointer or two would be great, I'd love to do more digging/reading) |
| 00:26.51 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 00:26.52 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 00:27.24 | [TK]D-Fender | micmic-: We have no idea what you're doing. We have no idea what numbers you are passing it. You've told us nothing. I could tell you how to bild a bridge and find out you wanted to bake a cake. |
| 00:28.00 | [TK]D-Fender | micmic-: you should show us actual config and the actual attempt and tell us what you are trying to DO before you ask us "how" |
| 00:29.02 | Kobaz | hehe |
| 00:29.10 | Kobaz | select count(*) from log_queue.agents; |
| 00:29.10 | Kobaz | <PROTECTED> |
| 00:29.44 | micmic- | [TK]D-Fender: extconfig.conf -> extensions => mysql,FD,Extensions |
| 00:29.51 | micmic- | all extensions are kept in a mysql table. |
| 00:29.58 | [TK]D-Fender | micmic-: Stop being generic about this. |
| 00:30.18 | [TK]D-Fender | micmic-: Show us what you're doing or we can't tell you where you went wrong |
| 00:30.27 | micmic- | [TK]D-Fender: yes yes, I am getting there |
| 00:30.33 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
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| 00:30.46 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 00:30.48 | [TK]D-Fender | micmic-: Go direct and skip "story time". |
| 00:30.52 | [TK]D-Fender | Do not pass "go" |
| 00:30.57 | [TK]D-Fender | Do not collect $200 |
| 00:31.06 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:31.06 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:31.13 | [TK]D-Fender | Do not eat the yellow snow |
| 00:31.23 | Kobaz | don't pee on the electric fence |
| 00:31.29 | Kobaz | or into the wind |
| 00:32.05 | Kobaz | so, would having 44 million records in my queue agents log be a good indication i should delete some old data |
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| 00:32.21 | micmic- | [TK]D-Fender: simple as that: the receptionist has a customer on the phone and calls another person on the phone (PHONE NUMBER). If another person says "OK", she connects these two. Software generates AMI message Action: Bridge with two channels. |
| 00:32.40 | micmic- | and then things go bananas, => that log line pbx.c |
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| 00:35.33 | [TK]D-Fender | micmic-: Awaiting debug.... |
| 00:36.20 | micmic- | sec. pastebin |
| 00:38.43 | micmic- | [TK]D-Fender: http://pastebin.ca/2315800 |
| 00:42.10 | micmic- | aaa |
| 00:42.13 | [TK]D-Fender | MicNow try again including AMI DEBUG so we can even see the request. |
| 00:42.13 | micmic- | I think I have something. |
| 00:42.22 | *** join/#asterisk tallest_red (~CNZ@ip98-169-207-41.dc.dc.cox.net) |
| 00:42.49 | [TK]D-Fender | micmic-: There are no details there. |
| 00:43.06 | micmic- | the request sent to AMI |
| 00:43.07 | Kobaz | detaaaaails |
| 00:43.30 | micmic- | Action: Bridge |
| 00:43.35 | micmic- | Channel1: SIP... |
| 00:43.40 | micmic- | Channel2: SIP... |
| 00:43.42 | micmic- | and that's it. |
| 00:44.00 | [TK]D-Fender | PASTEBIN <- |
| 00:44.42 | Kobaz | inpuuut |
| 00:46.15 | micmic- | as I am looking at the source code in that bit responsible for that |
| 00:46.35 | micmic- | <PROTECTED> |
| 00:47.00 | [TK]D-Fender | Guess you want to just waste time... best of luck with this. |
| 00:47.15 | micmic- | not really, I would like to solve the problem. |
| 00:47.18 | Kobaz | micmic-: seriously... provide more data |
| 00:47.27 | [TK]D-Fender | micmic-: You clearly don't |
| 00:47.49 | [TK]D-Fender | micmic-: You have bee playing "secret squirrel" from the start and provide nothing of value to look at. |
| 00:47.57 | micmic- | [TK]D-Fender: I am not an asterisk primer, pretty new to it |
| 00:48.07 | [TK]D-Fender | micmic-: This runaround is a waste of our time |
| 00:48.28 | [TK]D-Fender | micmic-: Real debug. Real configs. |
| 00:48.33 | micmic- | ok. |
| 00:48.46 | micmic- | [TK]D-Fender: can I then please ask you - which configs, to make things clear |
| 00:48.48 | Kobaz | micmic-: if you're new, then the best thing to do, is learn from the best, and follow their instructions... since the veterans do actually know what they are talking about |
| 00:49.00 | *** join/#asterisk poseidon (~joe@vps6967.inmotionhosting.com) |
| 00:49.11 | [TK]D-Fender | Show an actual call with actual dumps of the AMI being issued |
| 00:49.40 | poseidon | Hello. I'm interested in building some software to monitor calls in my call center. Does anyone have suggestions on where I can pull information about calls and queues? |
| 00:49.51 | [TK]D-Fender | poseidon: AMI |
| 00:50.00 | [TK]D-Fender | poseidon: Queue Logs |
| 00:50.08 | Kobaz | micmic-: if you have super secret data like a password, just put some X's in it, but otherwise paste the logs as-is |
| 00:50.22 | micmic- | [TK]D-Fender: ok, working on it. |
| 00:50.58 | micmic- | [TK]D-Fender: the thing is - I have only data I gathered when doing some debug - live dumps can be done earliest in ca. 10 hours |
| 00:51.01 | micmic- | [TK]D-Fender: is that ok? |
| 00:51.37 | Kobaz | you can only provide what data you have |
| 00:51.45 | Kobaz | and if it's not enough to diagnose the problem we'll let you know |
| 00:51.47 | [TK]D-Fender | micmic-: Come back when you've got something to show us. |
| 00:51.58 | [TK]D-Fender | micmic-: This is not a guessing game. |
| 00:52.13 | Kobaz | but being selective yourself about what data to provide is a recipe for lots of time wasted |
| 00:52.16 | micmic- | [TK]D-Fender: ok, good. We are about to reproduce this tomorrow (in 10 hrs) |
| 00:52.32 | micmic- | [TK]D-Fender: I'll get as much stuff as possible, live dumps, AMI requests etc. |
| 00:52.42 | micmic- | [TK]D-Fender: thanks for your time and sorry if you had an impression I was wasting your time |
| 00:52.44 | Kobaz | core set verbose 5 |
| 00:52.46 | micmic- | Kobaz: same to you :) |
| 00:53.36 | Kobaz | micmic-: there's just way too many people who come in here and say "help, it's broke, help me fix" and then provide some vague description of the problem and no hard evidence |
| 00:54.12 | micmic- | Kobaz: I can imagine, but now you set the rules and I got some instructions when I can come and ask for help. that's clear now ;) |
| 00:54.13 | Kobaz | asterisk is a VERY complex system and everything is interacting with everything else and without log data, you really can't diagnose anything at all |
| 00:54.22 | Kobaz | good good |
| 00:54.37 | [TK]D-Fender | You think X is happening. Y is actually happening. We can't see that. |
| 00:55.18 | [TK]D-Fender | Do I trust where anything is being pointed? Of course not. Samuel L. Jackson had wonderful things to say about assumptions..... |
| 00:55.27 | Kobaz | hehe |
| 00:55.30 | micmic- | rotfl ;) |
| 00:55.34 | [TK]D-Fender | So if you want to fix something you have to actually look at it. |
| 00:56.00 | *** part/#asterisk saysocomm (~dotcomm@75-144-224-89-Tallahassee.hfc.comcastbusiness.net) |
| 00:56.02 | micmic- | [TK]D-Fender: indeed, although believeme - as Kobaz said - there are sooo many things, I am really lost in the beginning ;) |
| 00:56.17 | Kobaz | it takes a good two years to really get into asterisk |
| 00:56.21 | Kobaz | enjoy the ride :P |
| 00:56.54 | Kobaz | but that's just about any complex skill really |
| 00:57.09 | Kobaz | you're only good at it until you've run into a bajillion and one problems and know how to fix them all |
| 01:02.51 | micmic- | ok, thanks a lot - I better hit the bed now |
| 01:02.57 | Kobaz | have fun |
| 01:04.59 | [TK]D-Fender | |<--ruied has left freenode (Ping timeout: 264 seconds) <-- not getting his debug either I guess |
| 01:05.14 | Kobaz | poor fella |
| 01:06.03 | *** part/#asterisk kresp0 (~kresp0@81.61.25.130.dyn.user.ono.com) |
| 01:08.18 | poseidon | [TK]D-Fender: Any suggestions for documentation on those/ |
| 01:08.36 | [TK]D-Fender | ~book |
| 01:08.37 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 01:08.38 | [TK]D-Fender | ^ |
| 01:08.53 | [TK]D-Fender | wiki,asterisk.org <- |
| 01:08.59 | [TK]D-Fender | s/,/./ |
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| 02:48.00 | poseidon | [TK]D-Fender: Do you know where I can find actions I can perform with AMI? |
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| 04:02.43 | jpsharp | Anyone have a working outbound fax using spandsp? I cannot for the life of me get mine back up and running. |
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| 04:06.07 | CrazyTux[m] | Hey guys - does anyone know if the manager.conf can be configured to use a FIFO instead of TCP |
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| 04:20.00 | batphone | anyone in here familiar with netsapiens soft switch? i cant seem to find a channel to ask in. |
| 04:23.34 | batphone | http://www.netsapiens.com/applications/sip-trunking/ |
| 04:23.55 | batphone | this is actually a really cool product, but the last time i used it we had to bolt on an LCR |
| 04:25.00 | batphone | its a linux box with a mysql back end with a web based front end that is more of a database development environment than a true GUI for an application |
| 04:25.53 | batphone | it has been several years since i used it, so i thought id ask around and see if anyone has been able to make it scale |
| 04:26.38 | batphone | my machines had four xeons and dozens of gigs of ram in them and could process about a million calls a day per box |
| 04:27.57 | batphone | ive seen it handle upwards of 200 CPS pretty easily, but i was able to write a benchmarking tool that found the upper limit of many aspects of the system |
| 04:29.29 | batphone | i was responsible for integrating with other SIP speaking carriers and out of the multitudes of SIP speaking gear out there I only found one or two that simply could not talk to it |
| 04:29.42 | batphone | one was a specific version of a cisco call manager |
| 04:30.02 | batphone | which that vendor, some tiny cell phone provider somewhere in mexico, could not upgrade |
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| 04:30.16 | batphone | hi wireddd |
| 04:30.23 | wireddd | hello |
| 04:30.27 | batphone | i have a question for you |
| 04:30.38 | batphone | have you ever used a netsapiens soft switch? |
| 04:30.50 | wireddd | nope |
| 04:31.09 | batphone | k. i have another question too. |
| 04:31.21 | batphone | have you ever built a GIGANTIC asterisk box? |
| 04:31.45 | wireddd | nope |
| 04:31.56 | batphone | dang dude |
| 04:32.14 | batphone | so, whats up with you tonight. have any questions for me/ |
| 04:32.17 | batphone | ? |
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| 04:32.59 | batphone | i have another question btw. |
| 04:33.16 | batphone | any of you ever use GCS as an LCR? |
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| 04:34.00 | wireddd | I can't get the second sagnoma a200 card to work in my asterisk box |
| 04:34.13 | batphone | whats it doing |
| 04:35.01 | wireddd | I can't figure out how to get asterisk to talk to it properly |
| 04:35.02 | WIMPy | What is GCS? |
| 04:35.04 | batphone | man i havent installed a sangoma in years. it was a pain in the ass to get it working. i had to use gentoo to make it happen. it was a t1 card, i might even have it laying around |
| 04:35.17 | batphone | GCS is an LCR based on OpenSER |
| 04:35.41 | batphone | asterisk shouldnt have a problem with the card |
| 04:35.50 | wireddd | it is a trixbox pro box as well and their support has been less than helpful |
| 04:36.05 | batphone | oh well shit, trixbox |
| 04:36.24 | batphone | i had to admin one of those too. just getting multiple phones to ring was a huge hassle |
| 04:36.33 | wireddd | yeah, wasn't my choice to go with them in the first place |
| 04:36.48 | batphone | why did you buy a sangoma? |
| 04:37.13 | batphone | is it for faxes or what |
| 04:37.22 | wireddd | no, analog lines |
| 04:38.12 | batphone | i never had to open up my trixbo |
| 04:38.13 | batphone | x |
| 04:38.28 | batphone | not sure what that process would be like, to get new hardware configured and all |
| 04:38.54 | wireddd | we just bought the software and support, it is running on a dell pc |
| 04:39.00 | batphone | WIMPy: http://globalconverge.com/page/load_page/i/Ng== |
| 04:39.10 | batphone | i built a big set up based on those |
| 04:39.14 | wireddd | I think they sold us the sagnoma cards too though |
| 04:39.20 | batphone | wireddd: you are kidding |
| 04:39.26 | batphone | wireddd: and they wont help you get them working? |
| 04:39.44 | wireddd | oh they have tried and failed several times |
| 04:40.02 | batphone | digium actually recommended some DECT phones to us once and then wouldnt support them when they didnt interoperate with trixbo |
| 04:40.05 | batphone | x |
| 04:40.26 | WIMPy | Noone supports Trixbox. |
| 04:40.29 | wireddd | getting 1 working is easy, everything is autodetected and just works, the second one is another story |
| 04:40.59 | batphone | ah, i was thinking of switchvox, derp sorry |
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| 04:41.06 | batphone | im a little zonked right now |
| 04:41.42 | batphone | gonna get some coffee. i have a 12 pound brisket going thats been smoking for about 5 hours now. |
| 04:42.14 | wireddd | WIMPy, yeah... not even fonality apparently |
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| 05:14.02 | mail323 | Does anyone know why a Polycom phone would refuse to store the SIP password? |
| 05:14.58 | batphone | mail323: maybe in the big config file it downloads there is an xml schema for it |
| 05:15.12 | batphone | <store pasword no> |
| 05:15.16 | batphone | something like that |
| 05:15.48 | ChannelZ | Or maybe it disapproves of your terrible password. |
| 05:16.09 | batphone | the letter "a". |
| 05:16.45 | mail323 | No. I don't know why suddenly these phones are giving me hell. I would always copy the {macaddress}-phone.cfg file and now I have tried three phones and they don't even attempt to download the file. |
| 05:17.15 | mail323 | So I Just said "screw this" and try to configure the phone manually. I save the settings but when the phone reboots it still can't register and when I go back the password is blank |
| 05:19.54 | ChannelZ | have they been haxx0r3d and are provisioning from someplace else? Or you have some syntax wrong in the config causing it not to parse correctly and fail? |
| 05:21.28 | mail323 | ChannelZ: No they are provisioning from the internal server. I was getting some unxplained configuration errors but I have the same configuration files for a while without any issues |
| 05:25.56 | mail323 | Can I just use a "secret = " for a null password? |
| 05:48.00 | batphone | mail323: there are some pretty detailed logs you can pull up with polycoms |
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| 05:48.24 | batphone | mail323: id have to dig for it but you can find a list of hotkeys you can press on the phone to have it upload its configuration files and debug logs i think |
| 05:48.49 | batphone | i was referencing sip.cfg |
| 05:49.03 | batphone | in my earlir comment about the xml config file |
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| 05:53.01 | batphone | win /7 |
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| 06:12.59 | mail323 | batphone: The phone is not downloading [macaddress]-phone.cfg, I even tried deleting it, so I don't understand why manually configured the phone will not save the password |
| 06:13.38 | batphone | mail323: create a mac address file with all 0' |
| 06:13.42 | batphone | all zeroes |
| 06:13.48 | batphone | it will do a TFTP get for that by default |
| 06:13.56 | mail323 | batphone: Yes that is there by default |
| 06:13.59 | batphone | do you have logging cranked up on your TFTP server? |
| 06:14.20 | mail323 | batphone: No. I use HTTP |
| 06:14.22 | batphone | is it that the phone is not attempting to download the file, or is it that your settings in the file are not taking effect? |
| 06:14.36 | batphone | ok, so your apache logs or whatever should tell you that much |
| 06:14.59 | batphone | you should be able to get a list of files that the phone attempts to download pretty easily. |
| 06:15.55 | batphone | tear them down until you get one to download. ive had to do that before but found it easier to just have the phone upload its current manual config, then edit that config. |
| 06:16.26 | batphone | that way you are starting with a clean slate from the phone's perspective. it sounds like you have syntax problems in your config files. |
| 06:16.33 | batphone | that or a networking problem. |
| 06:17.02 | mail323 | I just rebooted another phone and it also does not download [macaddress]-phone.cfg |
| 06:17.23 | ChannelZ | DNS? |
| 06:18.24 | batphone | mail323: can you pastebin your phone's config files? |
| 06:18.26 | mail323 | ChannelZ: Obviously the DNS is fine if the phone can download some files. |
| 06:19.22 | ChannelZ | I didn't realize it was downloading anything. |
| 06:21.39 | mail323 | http://pastebin.ca/2315890 |
| 06:28.16 | *** join/#asterisk gerhard7 (~gerhard7@82-169-24-72.ip.telfort.nl) |
| 06:32.47 | *** join/#asterisk kopilo (~kopilo@crm.anat.org.au) |
| 06:33.51 | mail323 | ChannelZ: You were right! Once I changed the server to FQDN the phone downloads [macaddress]-phone.cfg and everything works! Going to format it just to be sure |
| 06:36.14 | ChannelZ | so it was an IP before? |
| 06:37.06 | kopilo | hey I know this is a legacy issue and it has more to do with debian than asterisk but I was wondering if you could help. I'm having issues with voicemail being recorded, I'm using format=wav but it errors out claiming any recording is less than 2 seconds. When I did some googling apparently that is because the right codec is not installed, but I have libavcodecs installed. Help? |
| 06:38.19 | kopilo | is it possible to record voicemail using vorbis on 1.6x? |
| 06:42.19 | *** join/#asterisk Nobody08 (~chatzilla@d216-232-17-171.bchsia.telus.net) |
| 06:42.21 | kopilo | these are the warnings I am recieving:http://pastie.org/6234477 |
| 06:45.48 | mail323 | ChannelZ: It's a long story, it was always a hostname but the network was split into different vlans and the server was replaced with a different hostname |
| 06:47.32 | mail323 | kopilo: The path " /tmp/asterisk_recordings/_var_spool_asterisk_voicemail_ANAT_Voicemail_107_tmp_mFEk90.wav" seems rather odd. Are you able to leave voicemail with the default settings? |
| 06:47.39 | *** join/#asterisk Nobody08 (~chatzilla@d216-232-17-171.bchsia.telus.net) |
| 06:50.21 | mail323 | .wav format would be managed by format_wav.so, on my system the only dependencies for it are libpthread.so.0 and libc.so.6 |
| 06:51.24 | kopilo | thanks |
| 06:51.26 | mail323 | On this subject, anyone have issues playing .wav voicemail attachments on Android phones? |
| 06:51.49 | *** join/#asterisk Nobody08 (~chatzilla@d216-232-17-171.bchsia.telus.net) |
| 06:51.51 | kopilo | default settings, that would be nice, I didn't set this system up |
| 06:52.28 | mail323 | kopilo: Mainly you would be looking at format = in voicemail.conf |
| 06:52.33 | kopilo | I can verify that the folders exist with perms 777 |
| 06:52.36 | kopilo | ahh |
| 06:52.49 | kopilo | ;format=g723sf|wav49|wav ? |
| 06:53.44 | mail323 | kopilo: the semicolon at the beginning of that line indicates it is commented out. Any other instance of that line? |
| 06:54.19 | kopilo | the only other one I tried switching via uncommenting and it didn't work >.> |
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| 06:54.44 | *** part/#asterisk Nobody08 (~chatzilla@d216-232-17-171.bchsia.telus.net) |
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| 06:54.47 | *** part/#asterisk Nobody08 (~chatzilla@d216-232-17-171.bchsia.telus.net) |
| 06:54.47 | mail323 | Well a few things, is there any "include" line in voicemail.conf? Is the disk full? |
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| 06:54.53 | *** part/#asterisk Nobody08 (~chatzilla@d216-232-17-171.bchsia.telus.net) |
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| 06:54.58 | *** part/#asterisk Nobody08 (~chatzilla@d216-232-17-171.bchsia.telus.net) |
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| 06:55.08 | *** part/#asterisk Nobody08 (~chatzilla@d216-232-17-171.bchsia.telus.net) |
| 06:55.10 | *** join/#asterisk Nobody08 (~chatzilla@d216-232-17-171.bchsia.telus.net) |
| 06:55.11 | kopilo | disk is at 94% |
| 06:55.17 | *** part/#asterisk Nobody08 (~chatzilla@d216-232-17-171.bchsia.telus.net) |
| 06:55.19 | *** join/#asterisk Nobody08 (~chatzilla@d216-232-17-171.bchsia.telus.net) |
| 06:55.55 | kopilo | no includes inside voicemail.conf |
| 06:55.57 | mail323 | if you have no include line in voicemail.conf and no other format = line that is not commented out then I would try to add format = wav |
| 06:56.06 | *** join/#asterisk Nobody08 (~chatzilla@d216-232-17-171.bchsia.telus.net) |
| 06:56.28 | kopilo | yeah I tried that too :/ |
| 06:57.32 | mail323 | Do you have a /tmp/asterisk_recordings/ directory? |
| 06:57.45 | kopilo | no |
| 06:57.54 | mail323 | Create it! |
| 06:58.14 | mail323 | maybe someone/something rm -rf /tmp/* |
| 06:58.34 | kopilo | lol |
| 06:58.38 | kopilo | created it |
| 06:58.52 | kopilo | still exitied |
| 06:59.03 | mail323 | permissions? |
| 06:59.18 | kopilo | switching it back to wav and changing perms |
| 06:59.53 | mail323 | In asterisk.conf do you have something along the lines of record_cache_dir = /tmp/asterisk_recordings? |
| 07:00.24 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-219-40.red.bezeqint.net) |
| 07:01.46 | kopilo | no luck so far, checking |
| 07:02.02 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 07:02.17 | kopilo | record_cache_dir = /tmp/asterisk_recordings |
| 07:02.22 | *** join/#asterisk Rac-on (jasper@bambi.rac-on.nl) |
| 07:02.25 | kopilo | check |
| 07:02.39 | *** join/#asterisk Nobody08 (~chatzilla@d216-232-17-171.bchsia.telus.net) |
| 07:03.28 | mail323 | I'm convinced the issue is related to the error " WARNING[21663]: file.c:1148 ast_writefile: Unable to open file" |
| 07:03.34 | kopilo | me too |
| 07:04.03 | mail323 | By default in asterisk.conf that line is commented out. And it seems when it's commented out it tries to write the temp file to /var/spool/asterisk/voicemail/default/1234/tmp |
| 07:04.19 | mail323 | BTW you are reloading or restarting when you change the config? |
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| 07:04.24 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 07:04.32 | mail323 | I would try to comment out the record_cache_dir line |
| 07:04.48 | kopilo | reloading |
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| 07:07.48 | kopilo | no joy |
| 07:13.16 | mail323 | kopilo: What's the current error? Might want to try a restart if you edit asterisk.conf |
| 07:14.06 | kopilo | file.c:1148 ast_writefile: Unable to open file /tmp/asterisk_recordings/_var_spool_asterisk_voicemail_ANAT_Voicemail_107_tmp_vEk6JT.wav: No such file or directory |
| 07:14.06 | kopilo | <PROTECTED> |
| 07:14.18 | kopilo | people are working late makes it impossible to restart >.< |
| 07:14.38 | kopilo | phew they are leaving |
| 07:14.44 | mail323 | "restart when convenient" and then go watch youtube |
| 07:14.44 | kopilo | restart all the things! |
| 07:15.07 | mail323 | No, just restart the asterisk service |
| 07:15.54 | kopilo | #service restart asterisk worked! |
| 07:16.27 | kopilo | thank you so much |
| 07:17.04 | kopilo | I also enabled astctlowner = root |
| 07:17.20 | mail323 | I can honestly say I have no idea what that does |
| 07:19.17 | kopilo | nods |
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| 07:21.50 | kopilo | I still could kiss you mail323 |
| 07:21.59 | *** join/#asterisk bogrd__ (~sandeep@115.119.115.26) |
| 07:22.36 | kopilo | later I will have to work out how to store the messages in a differnt location but for now it is working |
| 07:24.22 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 07:24.25 | bogrd__ | hi, i'm trying to connect to the asterisk-11.2.1 AMI via HTTP by following the steps on the asterisk book at ( http://ofps.oreilly.com/titles/9781449332426/asterisk-AMI.html ). but when i try to access the AMI using wget / curl, I get a 404 error. Any help would be great! :) thanks in advance.. |
| 07:24.29 | v0lZy | lo |
| 07:24.37 | *** join/#asterisk radic (~radic@dslb-178-007-135-249.pools.arcor-ip.net) |
| 07:25.27 | kopilo | bogrd first things first: is port 80 open on your firewall? |
| 07:25.56 | kopilo | oh and are you curling from the same computer the asterisk server is on? |
| 07:26.09 | bogrd__ | kopilo: yes.. |
| 07:26.21 | mail323 | bogrd__: Did you set webenabled = yes in manager.conf? |
| 07:26.22 | bogrd__ | kopilo: and i'm curling on the same computer |
| 07:26.30 | bogrd__ | mail323: one sec.. checking that.. |
| 07:26.57 | kopilo | I gather bogrd did if those instructions were followed |
| 07:28.03 | bogrd__ | kopilo: mail323: enabled was no :P sorry.. its working fine now.. no idea how i missed it ! :) |
| 07:28.22 | kopilo | yay :D |
| 07:28.31 | kopilo | yay for simple fixes |
| 07:30.13 | kopilo | I have a yes or no question if anyone likes. If I have an asterisk server connected to the interwebs in office a and softphones with internet in office b, can the softphones in office b be peers on the asterisk server? |
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| 07:34.33 | volga629 | Hello Everyone, I am trying use t38modem, and I want about the following messages |
| 07:34.38 | volga629 | 2013/02/20 02:05:13.796Opal Liste...0x20cab700SIPReceived NOTIFY message-summary |
| 07:34.40 | volga629 | 2013/02/20 02:05:13.797Opal Liste...0x20cab700SIPCould not find a SUBSCRIBE corresponding to the NOTIFY message-summary |
| 07:34.42 | volga629 | 2013/02/20 02:05:13.797Opal Liste...0x20cab700SIPSending PDU 481 Call Leg/Transaction Does Not Exist (346 bytes) to: r |
| 07:36.32 | bulkorok | hi |
| 07:37.19 | kopilo | have no idea what this means: Could not find a SUBSCRIBE corresponding to the NOTIFY |
| 07:41.05 | volga629 | Let me see on debug |
| 07:44.23 | volga629 | that what I found https://fpaste.networklab.ca/SCtr/ |
| 07:45.14 | *** join/#asterisk madhatt3r (madhatt3r@62.117.203.84.dyn.user.ono.com) |
| 07:45.43 | kopilo | interesting: Scheduling destruction of SIP dialog 'd648b783-9879-e211-8908-000c29865b25@capbxsrv01.pbxclst.networklab.ca' in 32000 ms (Method: REGISTER) |
| 07:46.20 | eirirs | destructive |
| 07:47.00 | volga629 | what interesting that I be able sent fax no problem |
| 07:48.09 | kopilo | well the sip dialog/fax is being scheduled for destruction which explains why "Call Leg/Transaction Does Not Exist" or maybe I'm just guessing too much |
| 07:48.59 | *** join/#asterisk Nobody08 (~chatzilla@d216-232-17-171.bchsia.telus.net) |
| 07:49.42 | volga629 | I think might need disable subscription, because t38modem I guess don't have this option |
| 07:49.56 | kopilo | nods |
| 07:51.27 | mail323 | I'm still looking for a working example of T.38 in Asterisk. Never seen it work. |
| 07:51.48 | volga629 | I can send faxes no problem |
| 07:52.02 | kopilo | nice |
| 07:52.14 | volga629 | but this message hmmm |
| 07:52.17 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 07:52.25 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 07:53.51 | mail323 | volga629: If you are not having any issues with it, then it's safe to ignore. |
| 07:55.41 | volga629 | I think is safe for ignore, but I asked on forum t38modem more information about it. If I get this correctly it related to mailbox. |
| 07:56.06 | volga629 | which modem don't have it |
| 07:56.40 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 07:57.00 | mail323 | volga629: Yes that's possible. You should probably look at a full SIP debug. I'd love to look it it but it's late and I was about to leave. |
| 07:58.51 | kopilo | same |
| 07:58.56 | volga629 | yes, I will try check it tomorrow |
| 08:01.01 | volga629 | thank you Everyone. |
| 08:01.14 | *** part/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
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| 08:39.03 | yang | From your experience, which VoIP hardphones would you recommend for simple dialing without extension, I am looking into one with good sound quality ? Are Polycoms still having that kind of a reputation ? |
| 08:41.17 | ChannelZ | simple dialing "without extension"? |
| 08:41.31 | yang | without the extension set |
| 08:41.54 | ChannelZ | still scratching my head |
| 08:42.14 | yang | the additional extension hotkeys plugged to hardphone |
| 08:42.25 | ChannelZ | oh.. a sidecar |
| 08:43.06 | yang | Also I am looking for a phone in the 100 eur range |
| 08:43.38 | ChannelZ | in any event I have Linksys SPAs (now Cisco something-or-anothers) which I like well enough, they do enough without 500 buttons, aren't ugly... |
| 08:45.53 | yang | ah its the VoIP adapter |
| 08:45.57 | yang | well that isnt a hardphone |
| 08:46.14 | yang | I know those yeah |
| 08:47.53 | ChannelZ | eh? |
| 08:48.19 | ChannelZ | I'm talking about actual phones. I think they are Cisco SPA5xx like the SPA502G |
| 08:49.09 | yang | ok |
| 08:49.42 | yang | I thought you were referring to SPA3102 |
| 08:49.43 | ChannelZ | ~100 US, whatever that is in euros |
| 08:49.43 | infobot | I think you lost me on that one, ChannelZ |
| 08:49.52 | ChannelZ | hush bot |
| 08:50.18 | ChannelZ | Oh no, but I have one of those too :) |
| 08:51.31 | *** join/#asterisk studybot_ (~studybot_@gateway/tor-sasl/studybot/x-68286794) |
| 08:56.10 | yang | ChannelZ: are they good for SIP ? |
| 09:08.30 | ChannelZ | I haven't had any problems |
| 09:09.07 | ChannelZ | (again mine are Linksys, before Cisco bought them, and I don't know if they changed the firmware and Crisco-ized it) |
| 09:09.26 | *** join/#asterisk din3sh (2988ff5e@gateway/web/freenode/ip.41.136.255.94) |
| 09:09.50 | din3sh | Hello Ppl |
| 09:10.18 | yang | ChannelZ: thanks for the hint |
| 09:11.31 | ChannelZ | Sure. Just one humble opinion |
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| 09:32.47 | linocisco | hi all |
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| 09:46.36 | danfromuk | Hi, Ive got a client whose isp blocks sip calls. Custom ports dont seem to help. So i want to set up a vpn for that user to get around the problem. Once the vpn is set up, whats best? Give the asterisk box a private IP and then give all vpn users private ips? Or give the users a public ip which they can use to access asterisk using its public ip? |
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| 10:29.23 | linocisco | hi all |
| 10:29.25 | linocisco | I need ur help on asterisk |
| 10:29.25 | linocisco | My asterisk config on QNAP TS-259 Pro+ which is settop box. I also configured grandstream HT702 2 FXS ports and let them call each other. First thing is that i can only choose user extension starting from 6000 only. 2nd thing is big annoyance beacuse after pressing ext no. to call to another ext., i feel it is so long to hear ringing tone. it is big delay i heard nothing before it actually rang when i called another. What would be the most likely p |
| 10:29.25 | linocisco | roblem and how to correct? Actually, I am not familiar with asterisk commands. To make sure, I restarted both QNAP server and grandstream device but still the same when I dial. |
| 10:29.26 | linocisco | regards |
| 10:33.13 | kaldemar | the HT is what you need to look at, not asterisk. |
| 10:34.04 | kaldemar | it generates the ringing tone and defines when it dials after you have pressed keys. |
| 10:34.23 | kaldemar | also, asterisk does not limit what you can dial. |
| 10:36.21 | bogrd__ | hi, i'm trying to use the originate command in asterisk-11.2.1 CLI. It says "No such command...". Am i missing something here? even the originate command is not auto-completing in asterisk-CLI... |
| 10:36.27 | linocisco | kaldemar, so what parameters should I check on HT? |
| 10:37.16 | kaldemar | linocisco: no idea. check its manual. |
| 10:37.33 | kaldemar | bogrd__: what command are you trying to use, exactly? |
| 10:38.03 | R1ck | does _X. not match +31260000000 ? |
| 10:38.22 | kaldemar | R1ck: no. X is a digit, + is not. |
| 10:39.01 | bogrd__ | kaldemar: this >> originate SIP/hello application dial SIP/world |
| 10:40.20 | bogrd__ | kaldemar: basically i'm trying to bridge two calls using this command.. |
| 10:41.32 | kaldemar | bogrd__: the command is "channel originate" instead of "originate". |
| 10:42.59 | *** join/#asterisk ashd (~ashleyd@94-195-121-125.zone9.bethere.co.uk) |
| 10:43.11 | bogrd__ | kaldemar: oh.. thanks.. its working now! :) and this can be used to bridge calls right? |
| 10:46.02 | R1ck | kaldemar: if I replace _X. with _[0-9+]! I guess it would match, but is this usage.. frowned upon? |
| 10:48.27 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 10:51.09 | kaldemar | bogrd__: more like bridge endpoints. but yes. |
| 10:51.20 | kaldemar | R1ck: it's completely acceptable. |
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| 11:08.24 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 11:08.30 | bombev | hi all |
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| 11:18.26 | R1ck | kaldemar: oh, nice. |
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| 11:46.47 | x1user | Hello. Think that i have some problems with NAT and Asterisk. Some users when pick up the phone dont hear anything ;> |
| 11:48.40 | kaldemar | ~sipnat |
| 11:48.40 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
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| 12:18.47 | Ice_Strike | I have enabled /etc/xinetd.d/tftp but where are the log? I want to see if hardware phone have attempted to connect to tftp server |
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| 12:24.58 | Greenlight | I've got a really odd situation, and can't think how to move forward in diagnosing it. I've two channels, which start of bridged. Audio is fine both ways. I redirect them both into a ConfBrdige, again all is fine, audio both ways. I then bridge the channels again (Via a ManagerBridge), and now I have one way audio. |
| 12:25.55 | racho | i know the question is not very appropriate but if someone has used linphonesh can it enlighten me how to terminate a call without making the user drop from an asterisk queue? |
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| 12:33.59 | Greenlight | Would a manager bridge always honour the "directmedia=no" settings against the peer - or could it be trying to remotely bridge them ? If so, how could I avoid it doing that ? |
| 12:41.33 | file | the directmedia setting is enforced within chan_sip, no matter who tries to use it it will be enforced if set |
| 12:42.05 | Greenlight | Hmm well it's not that then. Thanks |
| 12:42.23 | Greenlight | Just don't see how suddenly the channels have one way audio |
| 12:42.43 | Greenlight | Something's happening to one/both of them I guess, but i've no idea where to look now |
| 12:46.08 | Greenlight | Even odder, the MixMonitor that's running on one of the channels, stops writing to the file. Audiohook inheritance is enabled. |
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| 13:15.54 | Greenlight | Whats the best way to see some sort of debugging output in regards channels being bridged ? |
| 13:17.38 | [TK]D-Fender | As in? |
| 13:18.34 | Greenlight | Bridging channels using a manager bridge, is stopping MixMonitor recording (audiohook inhertiance is enabled) and causing one way audio. There's got to be something odd happeing but I can't see what from the standard CLI output |
| 13:19.09 | Greenlight | So I need some further debugging output of some sort |
| 13:20.31 | Greenlight | It may be that the bridge thread is becoming deadlocked and never completing, I'm just not sure |
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| 13:26.39 | kaldemar | Greenlight: sip debug during the bridge command and rtp debug to see if there is any rtp going through asterisk. |
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| 13:43.13 | Greenlight | sip debug shows nothing at all during the bridge, so at least I know it's 100% not trying to remoetly bridge etc. rtp debug stops showing anything after the bridge |
| 13:44.20 | Greenlight | where abouts does "rtp debug" measure and report from ? |
| 13:50.12 | Greenlight | Like, the sip channel must still be receiving RTP, so if it's not being reported by "rtp debug" where abouts does that indicate the issue is? |
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| 14:12.01 | Greenlight | Are "#ifdef FOR_DEBUG" type debugging messaged enabled in menuselect? |
| 14:12.31 | Greenlight | I see something in the code that looks like it's existing if a zombie flag is set, and i'm thinking that could be what's happening here |
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| 14:23.55 | krotos | hi all guy |
| 14:24.09 | krotos | the "R" key on old phone, can be captured on asterisk to activate atxfer? |
| 14:25.40 | [TK]D-Fender | krotos, that is a question for whatever interface you have it plugged into |
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| 14:37.18 | R1ck | I want to firewall my asterisk server, but when I do, the asterisk server cant send audio to SIP phones - why is that or where can I find info on how to work around that? |
| 14:38.10 | R1ck | is that RTP? |
| 14:38.25 | [TK]D-Fender | R1ck, If it can't send where you want it to send then you are clearly blocking things you shouldn't be. Stop doing that |
| 14:38.38 | [TK]D-Fender | R1ck, And yes, the audio is RTP |
| 14:38.39 | R1ck | thats what I'm asking |
| 14:38.46 | R1ck | what do I need to unblock |
| 14:39.03 | R1ck | so I'll try RTP first |
| 14:39.53 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 14:41.38 | jmetro | rtp is the audio stream |
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| 14:48.09 | bitwize | I need help defining multiple blacklist filters for a AMI manager in manager.conf, the filter works fine if I define "eventfilter=!Event: RTCP*", the problem occurres when I'm trying to define multiple blacklist filters. When I define multiple "eventfilter"-rows no events beeing filtered. Am i supposed to define all filters on the same "eventfilter"-row? |
| 14:48.24 | bitwize | If so, how would this syntax look like? |
| 14:49.19 | bitwize | I cannot find this information on google nor in my o'reilly book |
| 14:49.45 | Greenlight | Oooh, I was looking for a similar filter, those damn RTCP events are bloody spammy |
| 14:51.24 | Greenlight | I thought you could define them on separate rows, I've webpage somewhere that I had open about it, 2 secs |
| 14:51.38 | Greenlight | Ahh here: http://www.fop2.com/blog/make-fop2-snappier-using-ami-eventfilter |
| 14:51.38 | bitwize | Indeed, there is alot of event-info that I do not need to receive :) |
| 14:51.41 | bitwize | ok, great |
| 14:53.24 | Greenlight | The only difference what what you said you tried is that your using a wildcard match for the RTCP, rather that defining them separetely, maybe try that ? |
| 14:53.57 | bitwize | Exactly, I'll give it a try, bbs :) |
| 14:57.06 | bitwize | The same problem occures without the wildcard, if I only define one eventfilter-row with one blacklist filter it wors fine, when I define multiple rows no events beeing filtered.... |
| 14:58.34 | Greenlight | Hmm, odd. I didn't get as far as actually trying it at this side, it's on my "todo" list. Lemme try and see, two secs |
| 14:58.47 | bitwize | Maybe this is related to my version of Asterisk... 1.8.4.4 |
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| 15:00.00 | Greenlight | I'm running 11.2.1 so will let you know in a momnety |
| 15:02.10 | coreyf1513 | bitwize: I don't have 1.8 anymore but you might try latest. at one point i tried multiple filters without success, then weeks later the filters worked (I assume due to an asterisk update) |
| 15:02.41 | Greenlight | Seems to be working a treat on 11.2.1; think it's your version |
| 15:02.51 | bitwize | OK, it looks like there is also possible to dynamically populate temporary filters via Action-commands through AMI. In worst case maybe I can populate theese filters from the client when connection has been established. |
| 15:03.48 | Greenlight | You could just upgrade to latest version of 1.8 ? |
| 15:04.11 | Greenlight | From what coreyf1513 said it's been fixed at some point |
| 15:05.02 | bitwize | ahh.. actually we are planning an upgrade from 1.8.4.4 to 11.2.1 in a couple of weeks so I can live with just the RTCP*-filter for now.. |
| 15:05.17 | bitwize | coreyf1513: great thanks for the input! |
| 15:05.26 | Greenlight | Well, it 100% works in 11.2.1 so you'll be sorted then |
| 15:05.49 | bitwize | Greenlight: Thank you for helping me out, I just wait for the migration to 11.2.1 |
| 15:06.07 | Greenlight | No worries, glad to help |
| 15:08.43 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-cjaijkntzojjydzk) |
| 15:11.36 | cali | Should localnet be set under [general] in sip.conf? |
| 15:12.07 | leifmadsen | yes |
| 15:12.13 | leifmadsen | along with externip |
| 15:12.35 | Faustov | I love ex turnips |
| 15:12.40 | cali | Is it discarded if it is not in [general]? |
| 15:13.06 | leifmadsen | yes |
| 15:13.19 | leifmadsen | it is a global setting |
| 15:13.28 | leifmadsen | points at asteriskdocs.org and sip.conf.sample |
| 15:13.46 | *** join/#asterisk madhatt3r (madhatt3r@62.117.203.84.dyn.user.ono.com) |
| 15:13.56 | cali | the same for externaddr? |
| 15:14.01 | leifmadsen | read the docs |
| 15:14.12 | leifmadsen | it is clear how they work together |
| 15:14.31 | cali | thank you |
| 15:14.38 | *** join/#asterisk [gabri] (~root@servidor01.servidor-online.com) |
| 15:14.52 | [gabri] | hi people i am triying phpagi and i have a problem |
| 15:15.37 | [gabri] | my simple script are in http://pastebin.com/SUFvzvhW |
| 15:15.59 | [gabri] | My error is " Call to undefined method AGI::text2wave" |
| 15:16.10 | [gabri] | i installed text2wave without problem |
| 15:16.40 | [gabri] | My call script are in http://pastebin.com/v8pc1AFb |
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| 15:26.17 | [gabri] | i can't fix the problem |
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| 15:32.59 | [gabri] | hi? |
| 15:35.09 | coreyf1513 | [gabri]: documentation mentions a procedure text2wav, not text2wave |
| 15:38.29 | [gabri] | coreyf1513: thaaaaaanks thanks thanksssssssssssssss |
| 15:38.38 | [gabri] | coreyf1513: you are god man!!!!!!!! |
| 15:39.59 | cali | Getting: http://pastebin.com/n8Gd0dVU looks like a NAT issue but I have set nat=yes in peer's configuration. |
| 15:41.19 | [TK]D-Fender | cali, That is a response to a QUILFY packet. What gives you the impression that this is anything negative? Go place an actual call. |
| 15:41.36 | [TK]D-Fender | QUALIFY |
| 15:42.16 | cali | Well, when placing a remote call only the recipient can hear something. |
| 15:43.40 | *** join/#asterisk navaismo (~navaismo@189.241.118.172) |
| 15:46.51 | Greenlight | Right, *something* is preventing RTP packets from passing when channels are bridged. I've narrowed it down to something inside ast_channel_bridge in channel.c - are there any specific places in here that I should focus my attentions on |
| 15:47.39 | Greenlight | This appears to only happen if the channels have previously already been brdiged or redirected, so am thinking something is getting set or left over from the bridge or redirect, causeing the final bridge to not work right |
| 15:48.04 | R1ck | what other firewall ports do I need to open, besides RTP and SIP, for a SIP trunk to function properly? right now if I start the firewall and I call the number, nothing happens in asterisk (even with very verbose logging and sip debug peer), if I turn off the firewall everything works as expected.. |
| 15:49.31 | *** join/#asterisk areski (~areski@81.184.35.151.dyn.user.ono.com) |
| 15:52.34 | [TK]D-Fender | R1ck, You keep talking about "firewall" generically. Show us your actual settings |
| 15:52.39 | [TK]D-Fender | ~pb |
| 15:52.39 | 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. |
| 15:52.40 | [TK]D-Fender | ^^^ |
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| 15:53.26 | [TK]D-Fender | R1ck, So far this is also with no implication of any other networking transform in the chain. |
| 15:55.17 | R1ck | [TK]D-Fender: by "show us your settings" you mean, my firewall script? |
| 15:56.48 | [TK]D-Fender | R1ck, Or at least what it has done |
| 15:58.00 | Greenlight | Why don't I see debug output in the CLI after doing "core set debug 9999" ? |
| 15:59.11 | [TK]D-Fender | perhaps you should pick a more reasonable number |
| 15:59.13 | R1ck | [TK]D-Fender: http://pastebin.ca/2316009 SIP rules from line 346 and RTP rules from 551 |
| 15:59.24 | [TK]D-Fender | And confirm things like calls actually taking place.... |
| 15:59.39 | Greenlight | I thought it was like verbose, and a highter number included everything ? |
| 16:00.51 | Greenlight | There's a load of "ast_debug(1," that it's hitting in the source code, along with extra ones I've added, yet none of them are showing in CLI |
| 16:03.22 | R1ck | what's funny is that I can make outgoing calls just fine with the firewall on |
| 16:04.13 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:04.13 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:04.56 | [TK]D-Fender | $IPTABLES -A OUTPUT -o $ETH -m udp -p udp --sport ${PORT} -j ACCEPT |
| 16:05.16 | [TK]D-Fender | not sure what the source is supposed to look like... |
| 16:05.25 | [TK]D-Fender | but thqat script is just way too much for me to wade through. |
| 16:05.55 | [TK]D-Fender | And you are restricting IP's on those sections as well and we have no details. |
| 16:06.56 | [gabri] | coreyf1513: can i ask other question? |
| 16:07.05 | R1ck | doesnt matter what source is supposed to look like, the bit that allows RTP is with --dport ${PORT} |
| 16:08.11 | R1ck | my trunk peer is 217.21.203.27 which is allowed on line 484 |
| 16:10.12 | [TK]D-Fender | r1Dport is whatever the dest asks for. |
| 16:10.16 | [TK]D-Fender | You have no control over that |
| 16:10.57 | [TK]D-Fender | You also haven't told us the rest of the networking involved in all of the pieces of this |
| 16:11.06 | *** join/#asterisk gauravp (~gaurav@c-68-80-206-60.hsd1.pa.comcast.net) |
| 16:12.07 | R1ck | the --dport bit doesnt matter, I can delete that rule, the bit that makes RTP work is with --sport |
| 16:12.56 | gauravp | Hi all, Just wanted to check if anyone else is having issues calling toll-free numbers from Asterisk via Google-Voice this morning? Checked online and saw reports on forums several months ago, so it does seem to happen from time to time |
| 16:13.29 | [gabri] | When i call to my asterisk i have this error: [Feb 20 17:13:08] NOTICE[27109][C-00000001]: chan_sip.c:25423 handle_request_invite: Call from '0034912692714' (91.121.129.20:5060) to extension 's' rejected because extension not found in context 'entrada-sip'. |
| 16:13.33 | [gabri] | why? |
| 16:14.04 | [TK]D-Fender | [gabri], Because precisely like it says, there is no match for "s" in [entrada-sip] |
| 16:14.10 | [TK]D-Fender | [gabri], What was unclear about that? |
| 16:14.25 | [gabri] | [entrada-sip] |
| 16:14.26 | [gabri] | exten => s,n,Answer |
| 16:14.26 | [gabri] | exten => s,n,Hangup |
| 16:14.32 | [gabri] | i have s in entrada-sip |
| 16:14.43 | [TK]D-Fender | [gabri], You have no 1 priority there |
| 16:15.00 | [TK]D-Fender | [gabri], You can't just use "n" without having a 1 first |
| 16:15.10 | [gabri] | ok thanks |
| 16:15.13 | [gabri] | thanks thanks !! |
| 16:17.21 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 16:25.56 | *** join/#asterisk Kraln (~kraln@69.169.90.240) |
| 16:28.24 | R1ck | is there some kind of benchmarking / stress testing tool for testing voip connections? We have an ADSL line here with 1mbit up and I want to find out how many concurrent outbound calls that will handle |
| 16:29.04 | Greenlight | It depends on the codec you're using, and how much headroom you want to leave |
| 16:29.21 | Rico29 | R1ck> do you know sipsak ? |
| 16:29.27 | *** join/#asterisk Kraln (~kraln@69.169.90.240) |
| 16:29.30 | WIMPy | Just ask aunt google for a voip bandwidth calculator and you will know. |
| 16:29.40 | R1ck | Rico29: no, but thanks |
| 16:29.59 | R1ck | WIMPy: I'm not interested in calculators.. I want to do real-world tests |
| 16:30.51 | Greenlight | Then make some phone calls .... |
| 16:31.03 | R1ck | yes, I think we'll do that |
| 16:31.10 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 16:31.14 | drmessano | That's going to be a big subjective |
| 16:31.26 | *** join/#asterisk gusto (~gusto@ppp-93-104-69-64.dynamic.mnet-online.de) |
| 16:31.27 | drmessano | Calculate it out, and subtract a couple for overhead |
| 16:31.31 | Greenlight | If you're using G711, and that 1mb is *dedicated* then personally I'd only go upto 8 or 9 |
| 16:31.59 | drmessano | Bandwidth calculators are not just magic... there is real math involved |
| 16:32.02 | Greenlight | Technically you can sequeeze 10 or 11 but I'd not push it that far |
| 16:32.33 | Greenlight | 1mb sounds like adsl, and it's likely contended, and subject to resync hihger or lower over time, so you need to leave some headroom |
| 16:32.59 | *** join/#asterisk nickfennell_ (~nickfenne@unaffiliated/nickfennell) |
| 16:33.01 | R1ck | oh, that G711 leads me to another question: my trunk provider stated they only allow alaw over the trunk. Does that mean if my phones use G711 to my asterisk server, they can't call out? or will Asterisk convert it to alaw automatically? |
| 16:33.14 | Greenlight | alaw is G711 |
| 16:33.22 | Greenlight | So you're fine |
| 16:33.48 | Rico29 | Greenlight> G711a != G711u |
| 16:33.57 | drmessano | Asterisk will transcode |
| 16:33.58 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-219-40.red.bezeqint.net) |
| 16:34.01 | drmessano | Not a problem |
| 16:34.07 | Greenlight | Rico29: I never said they were the same |
| 16:34.10 | R1ck | ah okay |
| 16:34.32 | igcewieling | it is better to use the same codec for the phones as for the provider, but there are many good reasons why you might not want to. |
| 16:35.10 | igcewieling | any provider which does not support g729 is a provider you want to stay away from |
| 16:35.21 | drmessano | R1ck: "G711" is missing the important "a" or "u".. If your phones can use g711a, you won't be transcoding |
| 16:35.42 | drmessano | [11:35] <igcewieling> any provider which does not support g729 is a provider you want to stay away from <---- ++ |
| 16:35.43 | *** join/#asterisk CrazyTux[m] (~Brandon@ip72-211-223-108.oc.oc.cox.net) |
| 16:36.15 | drmessano | If a provider is not willing to spend money on you, then you shouldnt spend money on them |
| 16:37.48 | igcewieling | exactly what drmessano, even if you don't actually want to use g729. |
| 16:41.00 | [gabri] | hi |
| 16:41.32 | [gabri] | i have this script http://pastebin.com/6PTmcPHH |
| 16:42.10 | [gabri] | when i execute this i haven't got errors but the result isn't good |
| 16:43.00 | carrar | moof |
| 16:44.36 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 16:48.58 | WIMPy | [gabri]: You have to implement the protocol. Just throwing data at AMI doesn't work. |
| 16:49.21 | pabelanger | ~collectdebug |
| 16:49.21 | infobot | collectdebug is, like, a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 16:50.36 | [gabri] | mm WIMPy and with Setvar? |
| 16:50.42 | [gabri] | http://voip-info.villaverde.nom.es/wiki/view/Asterisk+Manager+API+Action+SetVar.html |
| 16:51.10 | WIMPy | [gabri]: Doesn't matter. You have to wait for Asterisk to acknowledge your login befor sending any other actions. |
| 16:51.38 | Qwell | The script is also immediately logging off, without doing anything else. |
| 16:51.50 | Qwell | well, trying to |
| 16:52.02 | Qwell | not that it ever actually gets sent |
| 16:52.24 | [gabri] | mmm i need to do two actions? |
| 16:52.43 | Qwell | You need to actually send the actions that you've got. |
| 16:52.49 | WIMPy | You already have 3 actions. |
| 16:53.27 | [gabri] | mmm ok |
| 16:53.34 | WIMPy | AMI isn't a one way street. You have to listen for it's answers. |
| 16:54.25 | WIMPy | There are some libraries out there to alledgedly make it easier for you. |
| 16:54.31 | *** join/#asterisk k611 (~K610@94.139.41.21) |
| 16:55.11 | [gabri] | mmm great |
| 16:55.30 | [gabri] | can you give the name of the best library |
| 16:55.48 | eirirs | the library in Alexandria. |
| 16:55.53 | [TK]D-Fender | [gabri], phpagi alreday has this <- |
| 16:55.59 | igcewieling | [gabri]: what language |
| 16:56.00 | [TK]D-Fender | already* |
| 16:56.02 | Qwell | eirirs: I don't know you, but I love you. |
| 16:56.21 | eirirs | hehe |
| 16:56.35 | Greenlight | Am going through the source code here, and am just hitting a brick wall. Can't see anything that would cause a bridge, to suddenly stop any rtp at all from a channel. Anyone ever seen something like that happen before, or able to point in a direction of the function where rtp may be getting stopped, dropped, or not moved on ? |
| 16:58.27 | Greenlight | Whatever is happening seens to be occuring *before* it hits either the ast_generic_bridge, or the technology specific bridge method, as both have the same issue |
| 16:59.51 | WIMPy | Greenlight: I think wee need a description of what's going on first. |
| 17:02.17 | Greenlight | Basically, whats happening is that I've two SIP channels, which are bridged (via AMI). All works fine, audio flows both ways. I then dual-redirect to a ConfBridge. Again all is fine, audio both ways, things are good. I then Bridge the two channels again, and suddently I have one-way audio, and it's like the RTP packets are getting stopped somewhere. |
| 17:03.38 | Greenlight | Now, I've previously Bridge channels that are in a ConfBridge without any issue, so I *think* that the Redirect, is somehow setting a flag, or doing something, that's causing the following bridge to go a bit crazy |
| 17:03.46 | WIMPy | Have you tried to take a close look at whatever events are generated? Anything different there? |
| 17:03.55 | [gabri] | mmm igcewieling [TK]D-Fender i tried this: http://pastebin.com/gkPP33P1 |
| 17:04.14 | [gabri] | i can see that the mark users is logged, but i can't see any call |
| 17:04.54 | Greenlight | The events all look the same, so far the debug output looks the same. But at the point of the bridge function being called, rtp just stops |
| 17:05.09 | Greenlight | "rtp debug" goes from spamming the CLI to completely quiet |
| 17:05.42 | WIMPy | Greenlight: Completely quiet? Didn't you say oy get one way? |
| 17:05.47 | Greenlight | As I say, I've been stepping through the source code, but I'm not that familiar with it |
| 17:06.43 | Greenlight | WIMPy: Ahh I didn't have a mic plugged in on that particular test, so there was no audio being generated in that direction, but when I did have a mic, or called a differnt phone, the audio was one way |
| 17:06.55 | Greenlight | The other thing that's strange |
| 17:07.01 | Greenlight | You ever used xlite? |
| 17:07.16 | Greenlight | You know how it likes to spam RTP 126 packets to "keep-alive" ? |
| 17:07.22 | WIMPy | I treid to some years ago. |
| 17:07.48 | Greenlight | Like every minute or so, you get a unrecognised RTP notice in the CLI |
| 17:07.49 | WIMPy | Nope. |
| 17:08.30 | WIMPy | Have you found out the pattern which channel gets stopped? Is it always the first/second of the last bridge action? |
| 17:08.31 | Greenlight | Nothing to worry about, however, when the bridge is doing it's "rtp-blocking" these seem to build up in the background, and I've noticed when the bridge ends, I get them all through at once (the CLI noticed) |
| 17:09.03 | Greenlight | It's always the same order |
| 17:09.37 | WIMPy | Have you tried other combinations not involving ConfBridge in between? Like just 3 phones connected round robin? |
| 17:11.01 | Greenlight | I've another application that does hundreds of thousands of bridges out of ConfBridges every day without a problem, so I *think* it's that DualRedirect which is somehow doing it |
| 17:11.28 | Greenlight | I don't know if there's a channel flag, that can get set somewhere, or something like that |
| 17:12.23 | WIMPy | If you don't get the warning about theunknown RTP packets, either. It rather sounds like it stops receiving. |
| 17:13.15 | WIMPy | Like some thread isn't woken up, maybe. Or a receie task going missing. |
| 17:13.21 | Greenlight | Exactly |
| 17:13.37 | WIMPy | I don't know the internals of the RTP stuff. Just guessing. |
| 17:13.43 | Greenlight | That's my train of though, but unfortunately, so much of the asterisk code is alien to me |
| 17:14.01 | [gabri] | mmm igcewieling [TK]D-Fender i have the call http://pastebin.com/7uDL8JCQ but the Set command isn't work texto is empty |
| 17:14.08 | Greenlight | I was kinda hoping someone would read this and go "ah ha, this is what it is..." |
| 17:14.13 | Greenlight | Maybe wishful thinking :) |
| 17:14.23 | file | the RTP stack uses the PBX thread for reading, if the PBX thread is not servicing the channel it will not read RTP packets and they can build up |
| 17:14.27 | Qwell | [gabri]: Set: isn't a valid header. |
| 17:14.48 | file | I have no immediate comment as to why that would occur |
| 17:14.58 | [gabri] | mm Qwell what are the mistake, the ":" character? |
| 17:15.01 | Greenlight | Any where in the code to point my attention at ? |
| 17:15.20 | file | not really... that spans a lot |
| 17:15.39 | file | what version of Asterisk and what is the exact AMI command usage? |
| 17:15.44 | Greenlight | 11.2.1 |
| 17:15.45 | Greenlight | SO latest |
| 17:15.49 | [gabri] | mm Qwell this cal file are working, http://pastebin.com/zfKX3EqE |
| 17:15.53 | Greenlight | Two secs I'll paste bin |
| 17:16.15 | [TK]D-Fender | [gabri], 'Set:'=> 'texto=esto no mola absolutamente nada', <-- and the reason you put the : in there is? |
| 17:17.10 | [gabri] | mm [TK]D-Fender i removed the : and the results is the same it was a mistake |
| 17:18.02 | [gabri] | Set it's correct? i am thinking about Variables command |
| 17:18.39 | Qwell | [gabri]: What does the documentation say about it? |
| 17:18.43 | *** part/#asterisk Phoebus (~Phoebus@pdpc/supporter/active/phoebus) |
| 17:20.16 | [gabri] | mm Qwell i don't know i am looking for in http://www.eder.us/projects/phpagi/ |
| 17:20.32 | Qwell | [gabri]: manager show command Originate |
| 17:20.39 | Greenlight | http://pastebin.com/xLHW0LrB |
| 17:20.47 | [gabri] | http://phpagi.sourceforge.net/phpagi2/docs/phpAGI/AGI_AsteriskManager.html#methodOriginate |
| 17:21.05 | Greenlight | @file: http://pastebin.com/xLHW0LrB That's the exact command usage |
| 17:21.21 | [gabri] | mm Qwell the documentation says aboutVariable Channel variable to set, multiple Variable: |
| 17:21.35 | file | Greenlight, there was substantial work done to the dual redirect stuff in Asterisk 11 that hasn't yet hit a release - I'd be curious if it solved it... |
| 17:21.55 | Greenlight | Ahh that's interesting |
| 17:22.06 | Qwell | file: maybe it was implemented as a duel redirect, and only one channel can win? |
| 17:22.08 | [gabri] | Qwell: thanks thanks thanks |
| 17:22.19 | [gabri] | Qwell: [TK]D-Fender the correct way is with Variable |
| 17:22.19 | file | Qwell, !!! |
| 17:22.20 | [gabri] | thankssssssssssss |
| 17:22.24 | [gabri] | thanks thanks mens!!!! |
| 17:22.33 | file | Greenlight, can you try the 11 branch to see if that does it? |
| 17:22.38 | Greenlight | Sure |
| 17:22.51 | Greenlight | svn isn't it |
| 17:22.55 | file | yes |
| 17:22.59 | Greenlight | Lemme go set it up |
| 17:23.57 | mjordan | file: Greenlight: pretty sure that fix is in 11.3.0-rc1 |
| 17:24.33 | file | ooh it is |
| 17:24.34 | Greenlight | Any idea what specifically it fixed - is there an JIRA issue around ? |
| 17:24.36 | *** join/#asterisk bkw_ (~Adium@freeswitch/developer/bkw) |
| 17:24.40 | *** part/#asterisk bkw_ (~Adium@freeswitch/developer/bkw) |
| 17:24.51 | [gabri] | mm Qwell [TK]D-Fender thanks!!! |
| 17:25.10 | *** join/#asterisk Nobody08 (~chatzilla@d216-232-17-171.bchsia.telus.net) |
| 17:25.15 | file | ASTERISK-18975 and ASTERISK-19948 but because it was a race condition it wouldn't surprise me if things went a little wonky... |
| 17:25.49 | Greenlight | Well - I shall grab 11.3.0-rc1 and cross my fingers, toes, legs and arms |
| 17:25.56 | file | Greenlight, http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.3.0-rc1.tar.gz |
| 17:26.15 | file | if that doesn't solve it I'll take a gander further |
| 17:26.31 | Greenlight | It be wgetting it's way to my box already |
| 17:26.33 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
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| 17:38.30 | Greenlight | Ok, testing now, fingers crossed! |
| 17:39.27 | Greenlight | Damn. Still one way audio after that bridge |
| 17:39.34 | file | lame |
| 17:40.15 | Greenlight | I wonder if I can get away with two seperate redirects |
| 17:40.34 | Greenlight | *if* that's what's causing peculiar behaviour |
| 17:41.22 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 17:42.43 | file | nothing immediately stands out |
| 17:45.37 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:49.07 | Greenlight | Hmm, not as simple to change to single redirect's as I'd thought, need to ensure the other channel goes back into the dialplan and doesn't just get hungup |
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| 17:53.27 | Greenlight | Damn .. gonna need to re-work how some of this is done. Will pick this up again later or tomorrow, as need to leave the office now. Thanks for your help |
| 17:53.40 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-219-40.red.bezeqint.net) |
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| 17:56.29 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 18:21.19 | navaismo | anyone here have seen this error while compiling wanpipe: error: ‘struct sock’ has no member named ‘sk_sleep’ |
| 18:21.49 | Qwell | navaismo: You need to #undef sandals |
| 18:21.53 | Qwell | Otherwise, no socks allowed. |
| 18:22.11 | WIMPy | :) |
| 18:22.29 | navaismo | ok.. now im lost :S |
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| 19:12.37 | Kobaz | is there any way to reconfigure a single sip peer either in 11 or trunk |
| 19:12.42 | Kobaz | like a sip reload 1234 |
| 19:12.48 | Kobaz | rather than an all out sip reload |
| 19:13.30 | [TK]D-Fender | Nope. |
| 19:13.35 | Chainsaw | Kobaz: That's a good one. I don't think sip qualify peer 1234 goes deep enough for what you have in mind. |
| 19:13.43 | [TK]D-Fender | Reload is the module, not a partial parse of a config file |
| 19:13.56 | Kobaz | not module reload chan_sip |
| 19:13.59 | Chainsaw | Kobaz: So you get to just reload the whole shebang and like it I suppose. |
| 19:13.59 | Kobaz | a 'sip reload' |
| 19:14.30 | Kobaz | it would be really useful when using realtime to do a single peer |
| 19:14.51 | *** join/#asterisk jercos (jercos@babbage.subluminal.net) |
| 19:15.35 | jercos | Hullo. So when I attach to asterisk with -r, I'm informed of the current verbosity with the phrase "at least", does that imply that there are cases where the actual verbosity will be higher than the number given? |
| 19:18.26 | Kobaz | this is exciting |
| 19:18.35 | Kobaz | [TK]D-Fender: i started porting my local asterisk branch to 11 |
| 19:18.48 | Chainsaw | Kobaz: I'm on 11. It's been very good to me. |
| 19:18.58 | Kobaz | not too many crashes and deadlocks? |
| 19:19.08 | Chainsaw | Kobaz: System uptime: 3 weeks, 6 days, 4 hours, 35 minutes, 39 seconds |
| 19:19.31 | Chainsaw | Kobaz: Now it's just two offices, so it has done only ~3100 calls in that time. |
| 19:19.34 | Kobaz | ah |
| 19:19.35 | Kobaz | oh |
| 19:19.36 | Kobaz | hehe |
| 19:19.42 | drmessano | I've had some problems with 11 lately, but I don't want to be the "ZOMG 11 IS TEH BROKE" guy because I haven't spent much time on it |
| 19:19.44 | Chainsaw | Kobaz: But still, back in the 1.2 days 4 days was a stretch. |
| 19:19.47 | Kobaz | my servers do ~5000+ calls a day |
| 19:19.53 | Kobaz | i restart asterisk nightly |
| 19:20.00 | Kobaz | so i dont even know how long i would be able to keep it up |
| 19:20.02 | Chainsaw | Kobaz: Yeah, most people do a lot more. But I do SIP over TCP, I fiddle with the SSL options... |
| 19:20.05 | drmessano | But I have a box that manages to keep SIP going for about 24 hours before it stops responding |
| 19:20.17 | Chainsaw | Kobaz: So I'm quite good at breaking it. If there's something wrong with it, I will find it. |
| 19:20.21 | Kobaz | ah |
| 19:20.22 | Kobaz | yeah me too |
| 19:20.30 | mjordan | Kobaz: sorcery may potentially allow for that sort of thing |
| 19:20.35 | Kobaz | i was the one screaming zomg 1.8 is teh broke when it first came out |
| 19:20.38 | Chainsaw | Kobaz: But in the interest of full disclosure, that does have a distro patchset applied. |
| 19:20.44 | Kobaz | and then i put in three bug fixes into sip and then it stopped crashing |
| 19:20.47 | Chainsaw | Kobaz: Oh, the non-existent SIP peer problems? |
| 19:21.00 | Kobaz | it was like sip header parsing |
| 19:21.01 | Chainsaw | Kobaz: Those were infuriating. I had to rewrite my whole dial plan to work around it, because I was not believed. |
| 19:21.08 | Kobaz | i found a null deref in the header checking |
| 19:21.19 | Kobaz | and some other stuff |
| 19:21.23 | Kobaz | ref leaks in subscribes too |
| 19:21.28 | Kobaz | asterisk was using 2 gigs in 4 days |
| 19:21.30 | Kobaz | with the ref leaks |
| 19:21.55 | drmessano | Chainsaw: Had any issues with chan_sip becoming non-responsive after a day or so with TCP, TCPTLS? |
| 19:22.37 | Chainsaw | drmessano: Not anymore. The certificate chaining support in Asterisk is broken though. |
| 19:22.40 | Kobaz | but all those chan_sip fixes also went into trunk and also went into 10/11 |
| 19:22.45 | Kobaz | so i should be good on that aspect |
| 19:22.46 | Chainsaw | drmessano: So if you want to use a *real* certificate, you need to apply a patch. |
| 19:22.56 | drmessano | Oh really? |
| 19:23.01 | drmessano | Where is this patch? |
| 19:23.05 | Chainsaw | https://issues.asterisk.org/jira/browse/ASTERISK-17727 |
| 19:23.06 | Kobaz | orly forreels |
| 19:23.13 | Chainsaw | Sitting in the bug tracker unloved. Like most of the stuff in my distro patchset. |
| 19:23.54 | drmessano | When you say REAL, you mean, self-signed work, but anything involving a cert issued from some authority is wonky? |
| 19:24.06 | Chainsaw | drmessano: Confirmed. |
| 19:24.20 | drmessano | Son of a bitch |
| 19:24.23 | Chainsaw | drmessano: We have a wildcard cert for the entire organisation. Obviously I want to use that. |
| 19:24.23 | drmessano | Ok |
| 19:24.29 | drmessano | Me too |
| 19:24.30 | drmessano | !! |
| 19:24.41 | Kobaz | wildcard ftw |
| 19:24.46 | Chainsaw | drmessano: It won't work now. If you pay Digium, please shout at them. They don't listen to me. |
| 19:25.18 | drmessano | Chainsaw: The "hold my beer and watch this" moment in the logs was something along the lines of a file not found with the cert? |
| 19:25.34 | Chainsaw | drmessano: It'll generally pretend the cert isn't valid and refuse to load. |
| 19:25.36 | drmessano | Thats vague.. I am trying to remember the exact error |
| 19:25.38 | drmessano | Yes |
| 19:25.59 | Chainsaw | drmessano: Upstream bug. Ignored for a long time. |
| 19:26.41 | Chainsaw | drmessano: I have loads of these "obviously correct" ones that are lingering. https://issues.asterisk.org/view.php?id=18010 comes to mind. |
| 19:27.03 | Chainsaw | drmessano: I think your hang is this though: https://issues.asterisk.org/jira/browse/ASTERISK-18345 |
| 19:27.27 | Chainsaw | can tell that SSL is untested |
| 19:27.35 | Chainsaw | Because anyone that tried to use it in earnest would notice that. |
| 19:27.57 | drmessano | I am working on applying it |
| 19:28.03 | Chainsaw | And having scavenged those two SSL-related patches is the only thing that gets me those three weeks. |
| 19:28.13 | Chainsaw | drmessano: I have them ready-made here. |
| 19:28.58 | Chainsaw | drmessano: http://distfiles.gentoo.org/distfiles/gentoo-asterisk-patchset-3.2.tar.bz2 |
| 19:29.18 | Chainsaw | drmessano: It should all apply. Most if not all have headers to show origins. |
| 19:29.38 | Chainsaw | is assuming 11.2.1 here |
| 19:32.08 | *** join/#asterisk nightrid3r (~kvirc@62.205.64.13) |
| 19:32.59 | *** join/#asterisk areski (~areski@80.174.255.57.dyn.user.ono.com) |
| 19:34.21 | drmessano | Yep |
| 19:35.59 | drmessano | Im building now.. |
| 19:36.56 | Chainsaw | That one will SSL for you. Guaranteed. |
| 19:39.14 | *** part/#asterisk gauravp (~gaurav@c-68-80-206-60.hsd1.pa.comcast.net) |
| 19:39.53 | drmessano | I applied both... Much appreciated. I should know in about 24 hours. Maybe less. The sucky part is that I absolutely have to initiate a call on that box to break it. It will sit there like a Zombie before chan_sip dies to the point that monit sends an email. |
| 19:40.06 | drmessano | Defeats the purpose of proactive monitoring :) |
| 19:40.32 | drmessano | So I will make a test call every couple hours until it dies |
| 19:40.38 | drmessano | no, _IF_ |
| 19:40.41 | drmessano | :) |
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| 19:55.39 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 19:56.43 | autofsckk | hello, anybody using asterisk on pfsense? |
| 20:01.03 | *** join/#asterisk solmsted (~solmsted@pool-71-251-234-174.rcmdva.fios.verizon.net) |
| 20:06.15 | ChannelZ | isn't that just a firewall script? |
| 20:06.52 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-sqbmhthkhzujoekx) |
| 20:07.05 | ChannelZ | oh.. it's a whole FreeBSD distro.. |
| 20:07.32 | nightrid3r | pfsense is a firewall distro with a number of addons |
| 20:08.24 | leifmadsen | I just use astlinux |
| 20:08.33 | leifmadsen | I don't even use the asterisk part of it, just the firewall/routing part |
| 20:08.39 | leifmadsen | since it works so well |
| 20:09.16 | lorsungcu | i prefer dedicated hardware for firewalls |
| 20:09.35 | lorsungcu | still don't quite get the attraction of using big x86 machines as routers/firewalls |
| 20:15.42 | WIMPy | The advantage ist that you can configure them to do what you need. |
| 20:16.18 | solmsted | Hello, I've been writing a FastAGI application with Asterisk 1.8.20. Is this the right channel to ask questions? |
| 20:16.44 | WIMPy | If it's about the Asterisk part |
| 20:16.52 | WIMPy | ~ask |
| 20:16.52 | 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. |
| 20:17.20 | solmsted | My FastAGI application has been working great, but I ran into one problem: I have to pass a relatively large amount of data to an Asterisk command. |
| 20:17.20 | solmsted | My TCP server shows that all of it is getting written to the socket. The Asterisk console shows it only receives the first 2048 characters. My TCP server receives two '510' errors back from Asterisk. |
| 20:17.26 | solmsted | I'm guessing that the command is getting split into two packets, and Asterisk is acting on the data in the first packet without waiting for the newline character. When Asterisk receives the second packet, it looks like gibberish. |
| 20:17.31 | solmsted | Does this make sense? Anyway to fix it, other then by sending less data? |
| 20:17.59 | lorsungcu | WIMPy: not sure what you can do using something like pfsense that you couldn't do with some dedicated solution |
| 20:18.23 | WIMPy | What kind of packet is 2048 bytes in size? |
| 20:18.45 | *** join/#asterisk tech_travis (~Travis@174.46.237.60) |
| 20:18.53 | solmsted | idk, packet might not be the right word |
| 20:19.01 | WIMPy | lorsungcu: I have no idea what pfsense can do. |
| 20:20.27 | *** join/#asterisk jsjc (~Adium@86.Red-83-42-206.dynamicIP.rima-tde.net) |
| 20:20.35 | WIMPy | But I know people who changed over to using PCs as routers because of Cisco stupidities. |
| 20:21.18 | lorsungcu | yeah, I've used plenty of stuff like pfsense |
| 20:21.21 | lorsungcu | vyatta |
| 20:21.31 | lorsungcu | some others |
| 20:21.41 | lorsungcu | ended up using Routerboards |
| 20:22.30 | WIMPy | Yes, nice things. |
| 20:28.33 | autofsckk | hi, anybody using asterisk on a pfsense box? |
| 20:48.22 | Chainsaw | drmessano: Indeed, if. But let me know. Because if that works very well for you we should find a way to petition Digium to take this more seriously. |
| 20:48.38 | Chainsaw | drmessano: Normally I could talk to leifmadsen, but he's not a bug marshal anymore. I miss this. |
| 20:49.22 | Chainsaw | drmessano: Sorry for the delay, I had one of these famous "short phone calls" with our DBA. The man can talk. |
| 20:49.24 | leifmadsen | heh |
| 20:49.30 | leifmadsen | talk to rnewton |
| 20:49.39 | leifmadsen | he is the new primary bug marshal |
| 20:50.06 | Chainsaw | leifmadsen: Okay. As a welcome present I shall await drmessano's verdict before I bother in earnest. |
| 20:50.14 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-hwipzctgmpkhovbk) |
| 20:55.53 | newtonr | leifmadsen: newtonr :) |
| 20:56.00 | leifmadsen | doh |
| 20:56.02 | newtonr | leifmadsen: just looked over and saw |
| 20:56.09 | leifmadsen | ya, that guy |
| 20:56.56 | *** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 20:57.44 | newtonr | Chainsaw: i'll scroll up and read, but otherwise feel free to private message me anytime, or more appropriately bring it up in #asterisk-bugs |
| 20:58.14 | *** join/#asterisk Mon|A|rch (~SBean@72.29.180.35) |
| 20:58.18 | Mon|A|rch | so, having an odd issue |
| 20:58.29 | leifmadsen | even it up |
| 20:58.34 | Mon|A|rch | two offices are making calls through this extension. one office, no trouble |
| 20:58.40 | Mon|A|rch | other office, can't make outbound calls |
| 20:58.41 | Mon|A|rch | http://pastebin.com/mBT2QbJe |
| 20:58.44 | Mon|A|rch | there's the sip log |
| 20:58.58 | Mon|A|rch | first it dials their extension with originate, no problem, then it tries to dial out |
| 20:59.09 | Mon|A|rch | i can't find the error in the sip log |
| 20:59.18 | Mon|A|rch | any insight would be appreciated |
| 20:59.26 | Chainsaw | newtonr: I shall peruse the bug channel when drmessano reports back. Thank you. |
| 20:59.32 | Mon|A|rch | I'll paste the relevant dialplan extension in a sec, but it's pretty simple |
| 20:59.52 | nny | http://pastebin.com/HhXtdjKC I have an odd issue where a phone is disconnecting at 25 seconds. The pastebin shows a portion of the sip dialog. In the Via: section is shows an IP address of another phone in the network not involved int he test, this is asterisk 1.4.43. Thoughts? |
| 21:00.05 | nny | er it shows two IP address in Via: |
| 21:00.10 | nny | addresses* |
| 21:00.30 | nny | I have a full pastebin of the sip dialog if needed |
| 21:02.06 | nny | http://pastebin.com/cwEJyNZP full pastebin |
| 21:05.05 | *** join/#asterisk BrokenArrow (~BrokenArr@unaffiliated/brokenarrow) |
| 21:08.32 | nny | also the PBX has public IP, the remote phone is behind nat (and sip.conf has the proper config (nat=yes, alwaysauthreject=yes |
| 21:08.33 | nny | externip=98.101.28.XXX)) |
| 21:08.35 | *** join/#asterisk micdobro (~mic@0305ds4-vby.2.fullrate.dk) |
| 21:08.54 | micdobro | Kobaz: hello, ayt? |
| 21:08.55 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 21:09.15 | nny | It appears the 115 address in Via: is the phone's local IP address. Is this normal for both to appear in the sip dialog? |
| 21:11.29 | Mon|A|rch | is the problem I'm having the provider's fault? |
| 21:11.47 | Mon|A|rch | it looks like everything goes well, then the to becomes "anonymous@anonymous.invalid" |
| 21:15.35 | igcewieling | "anonymous@anonymous.invalid" normally means "callerid blocked" |
| 21:15.59 | Mon|A|rch | okay |
| 21:16.11 | igcewieling | though the proper way is with the Privacy: header, but many providers do not use that, especially with end user accounts. |
| 21:16.54 | Mon|A|rch | well, calls are only failing from one of the two locations |
| 21:21.10 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 21:21.42 | Mon|A|rch | but they're all using the same network |
| 21:21.44 | Mon|A|rch | : / |
| 21:21.51 | Mon|A|rch | although they're several states apart |
| 21:22.09 | Mon|A|rch | both offices dial out identically, by pressing 9 first |
| 21:22.42 | *** join/#asterisk imcdona (~imcdona@c-71-227-200-25.hsd1.wa.comcast.net) |
| 21:23.47 | Mon|A|rch | it happens immediately after "SIP<blabla> is making progress passing it to SIP/<blabla>" |
| 21:24.03 | Mon|A|rch | it reads a BYE immediately after |
| 21:24.12 | Mon|A|rch | I'm assuming that's where the trouble is happening |
| 21:24.41 | Mon|A|rch | is there any meaningful information that I could get out of the information after that? or do i need to check the sip server logs |
| 21:25.15 | Mon|A|rch | gateway i should say |
| 21:25.15 | *** join/#asterisk maetrik (maetrik@185.14.184.81) |
| 21:26.37 | maetrik | Asterisk 1.4.37 and a SIP provider with SRV records........ I can't get it to work. I use the same setup on several Asterisk 1.4.21 and .26 servers and it works everywhere except for a couple of 1.4.37 servers. |
| 21:26.44 | maetrik | It does not seem to do the SRV lookup |
| 21:27.00 | maetrik | CentOS 5 on all machines, with and without dnsmasq. Same problem. |
| 21:27.48 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 21:29.54 | leifmadsen | maetrik: srvlookup=yes in sip.conf? |
| 21:30.05 | maetrik | Sorry I did not mention that, yes it has that included :) |
| 21:31.19 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 21:31.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 21:31.42 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 21:31.42 | maetrik | The biggest problem of all is that I can't upgrade these servers, I have to work with the tools I've got. |
| 21:31.49 | maetrik | And I suspect there is an issue with SRV records and 1.4.37 |
| 21:32.44 | maetrik | WARNING[27369]: acl.c:401 ast_get_ip_or_srv: Unable to lookup srv.myvoipprovider.com |
| 21:33.36 | Kobaz | do de do |
| 21:34.42 | maetrik | I have tried dnsmgr enabled and disabled, same problem. |
| 21:38.04 | *** join/#asterisk tallest_red (~CNZ@ip98-169-207-41.dc.dc.cox.net) |
| 21:43.34 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 21:44.52 | jeffspeff | i have a digium AEX800, TDM800P and a TDM400P. do all of these cards work with a regular POTS line? |
| 21:45.22 | navaismo | ya |
| 21:45.32 | jeffspeff | i've found several of them at work not in use anymore as we're pure voip now, i'm wanting a card to play with and test, never messed with any of them before |
| 21:46.38 | jeffspeff | so, if i configure it correctly, i can plug in a regular old pots phone and make/receive calls ? |
| 21:52.07 | Mon|A|rch | what reasons are there that when I make a call the server would not make the call, but also not give me an error? |
| 21:52.43 | Mon|A|rch | getting BYE packets after attempting to pass a Dial()'d call to an originated call |
| 21:53.24 | Mon|A|rch | I'll repaste the SIP debug, if anyone has an answer, PM me so I'll notice |
| 21:53.26 | Mon|A|rch | exten => 800,1,Monitor(gsm,${STRFTIME(${EPOCH},,%d%m%Y)}-${PATCODE},m) |
| 21:53.26 | Mon|A|rch | <PROTECTED> |
| 21:53.26 | Mon|A|rch | <PROTECTED> |
| 21:53.26 | Mon|A|rch | <PROTECTED> |
| 21:53.26 | Mon|A|rch | <PROTECTED> |
| 21:53.27 | Mon|A|rch | <PROTECTED> |
| 21:53.31 | navaismo | jeffspeff, yes |
| 21:53.35 | Mon|A|rch | whoa, wrong paste, my bad |
| 21:53.44 | jeffspeff | navaismo, thanks |
| 21:53.58 | Mon|A|rch | http://pastebin.com/mBT2QbJe |
| 21:59.43 | *** join/#asterisk cheetahw26 (~chatzilla@cpe-24-92-62-218.nycap.res.rr.com) |
| 21:59.54 | navaismo | seems like your peer doest like anonymous calls |
| 21:59.56 | navaismo | From: <sip:313@10.3.1.1>;tag=CCC7CAF4-13D3 |
| 21:59.56 | navaismo | To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as020e2e40 |
| 22:01.36 | cheetahw26 | I've been asked to design a basic audio conferencing system, where users can dial in, enter some pass code and join a conference. It needs to support up to 50 users per conference and must be able to have at least two separate conferences running simultaneously... |
| 22:02.09 | Mon|A|rch | navaismo, okay |
| 22:02.24 | Mon|A|rch | does that mean I need to set ${CALLERID(name)}? |
| 22:02.31 | cheetahw26 | I've worked minimally with asterisk probably ~7 years ago, but haven't played with it since... and was curious if anyone could offer any recommendations on where to start, similar setups, step-by-step instructions with options? |
| 22:02.43 | Mon|A|rch | also, i appreciate you taking a look at that very much |
| 22:03.03 | *** join/#asterisk frozenfew (~ck@186.80.2.114) |
| 22:03.27 | Mon|A|rch | cheetahw26, there's the new "Asterisk: the definitive guide" online |
| 22:03.33 | Mon|A|rch | it's essentially a text book |
| 22:03.36 | Mon|A|rch | but it's very complete |
| 22:03.47 | leifmadsen | cheetahw26: asteriskdocs.org |
| 22:03.54 | cheetahw26 | thanks |
| 22:03.57 | leifmadsen | cheetahw26: for Asterisk 11, see ofps.oreilly.com |
| 22:04.00 | Mon|A|rch | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html |
| 22:04.13 | leifmadsen | 4th edition draft is on the OFPS site |
| 22:04.27 | leifmadsen | asteriskdocs.org contains the 3rd edition (Asterisk 1.8 based) |
| 22:04.41 | Mon|A|rch | listen to leif, he wrote the book |
| 22:05.03 | cheetahw26 | nice... and that covers audio conferencing setups I imagine |
| 22:05.24 | cheetahw26 | well... I suppose I can read it first and then ask later :) |
| 22:05.36 | leifmadsen | cheetahw26: it does |
| 22:05.45 | leifmadsen | cheetahw26: for that in particular, I suggest you read the ofps version |
| 22:05.56 | cheetahw26 | I'll probably need to know everything in it anyway, if I am going to set it up and configure it... |
| 22:05.59 | leifmadsen | that section was updated quite a bit |
| 22:07.48 | cheetahw26 | excellent thanks |
| 22:10.11 | *** join/#asterisk gusto (~gusto@ppp-62-216-209-138.dynamic.mnet-online.de) |
| 22:11.03 | Mon|A|rch | so, to change the caller id to avoid getting that anonymous@anonymous.invalid, am I going to need to change just ${CALLERID(name)} or (name) (dnid) and (num) |
| 22:14.25 | navaismo | you can try using SET & CALLERID(all) |
| 22:14.33 | leifmadsen | Mon|A|rch: use Verbose() or NoOp() to look at the output of the various CALLERID() methods and determine what you need to change |
| 22:16.36 | Mon|A|rch | leifmadsen, makes sense, thank you |
| 22:18.42 | Mon|A|rch | so, noop(${CALLERID(all)}) gave me """ <>" |
| 22:18.47 | Mon|A|rch | does that mean none of them are set? |
| 22:19.04 | *** join/#asterisk bitwize (~bitwize@c83-253-251-219.bredband.comhem.se) |
| 22:21.05 | malcolmd | leifmadsen: yay for ATFOT confbridge documentation :D |
| 22:21.18 | leifmadsen | malcolmd: hells ya |
| 22:21.28 | leifmadsen | s/TFOT/TDG/ |
| 22:21.38 | malcolmd | m'bad |
| 22:21.47 | leifmadsen | the future is now the past |
| 22:22.00 | malcolmd | sure was |
| 22:23.23 | navaismo | Mon|A|rch, yep empty, use SET to set it |
| 22:23.44 | leifmadsen | s/SET/Set/ |
| 22:23.53 | leifmadsen | SET implies a dialplan function (uppercase) |
| 22:24.11 | Mon|A|rch | well |
| 22:24.14 | leifmadsen | and now I go to dinner |
| 22:24.36 | Mon|A|rch | i tried Set($CALLERID(name)}=${SOMEVAR}), which worked |
| 22:24.54 | *** join/#asterisk vandyk (~quassel@177.41.175.185) |
| 22:24.55 | Mon|A|rch | but that noop(${CALLERID(all)}) still gives me the same result |
| 22:25.42 | ChannelZ | You set wrong |
| 22:25.47 | Mon|A|rch | do i want to just set ${CALLERID(all)} to some value? |
| 22:25.56 | ChannelZ | Set(CALLERID(name)=xxxxx) |
| 22:26.01 | Mon|A|rch | derp |
| 22:26.06 | Mon|A|rch | oops |
| 22:26.09 | Mon|A|rch | thanks ChannelZ |
| 22:26.16 | ChannelZ | Set is funny that way |
| 22:26.38 | Mon|A|rch | i forget that sometimes |
| 22:26.58 | ChannelZ | since its goal is to set a variable, it's kind of already escaped with ${} |
| 22:27.26 | Mon|A|rch | yeah, i knew that already, for some reason i just blanked and did it wrong |
| 22:30.20 | Mon|A|rch | alright, getting ""nameofid"" <>" |
| 22:30.26 | Mon|A|rch | guessing the <> is where the number should be |
| 22:31.19 | dr0ck | it is |
| 22:31.21 | navaismo | SET(CALLERID(all)="mmm"<1111>) |
| 22:33.51 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:36.41 | *** join/#asterisk bitwize (~bitwize@c83-253-251-219.bredband.comhem.se) |
| 22:41.25 | WIMPy | Don't use quotes. |
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| 22:49.28 | *** join/#asterisk vandyk (~quassel@189.27.216.202.dynamic.adsl.gvt.net.br) |
| 22:49.49 | Mon|A|rch | i wasn't using quotes |
| 22:54.38 | Mon|A|rch | hm, same issue |
| 22:54.50 | nny | : http://pastebin.com/HhXtdjKC I have an odd issue where a phone is disconnecting at 25 seconds. The pastebin shows a portion of the sip dialog. In the Via: section is shows an IP address of another phone in the network not involved int he test, this is asterisk 1.4.43. Thoughts? |
| 22:54.51 | Mon|A|rch | is there a sip debug log? |
| 22:54.57 | nny | http://pastebin.com/cwEJyNZP full pastebin |
| 22:55.22 | *** join/#asterisk Tagor (~Tagor@s55978a13.adsl.online.nl) |
| 22:55.28 | nny | the ip address in Via: seems to be the local and public address of the phone (115 is the phone's address on the local network) |
| 22:55.49 | Tagor | Is it possible to disable transcoding on outgoing calls? |
| 22:55.51 | vandyk | I was having problems with calls being dropped after 30 seconds |
| 22:55.58 | *** join/#asterisk jsjc (~Adium@86.Red-83-42-206.dynamicIP.rima-tde.net) |
| 22:55.58 | vandyk | it was NAT problems |
| 22:56.39 | navaismo | Mon|A|rch, only if you enable the full log |
| 22:56.43 | ChannelZ | Can we please have widespread IPv6 now? |
| 22:57.24 | navaismo | nny, retransmitting issues maybe means a network or nat issue, |
| 22:57.58 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 22:58.18 | nny | navaismo: i am thinking this is a remote issue, however I am trying to do due diligence. It started recently |
| 22:58.55 | nny | I have sip.conf relevant settings nat=yes xternip=98.101.28.XXX) |
| 22:59.15 | Mon|A|rch | navaismo, how do i do that? |
| 22:59.24 | Mon|A|rch | also, why the heck isn't the full log on automatically |
| 22:59.27 | jmetro | only takes 253 guesses to fill in the X's |
| 22:59.34 | navaismo | via logger.conf |
| 22:59.35 | vandyk | does your firewall has right nat rules? |
| 22:59.51 | Mon|A|rch | k |
| 22:59.54 | navaismo | Mon|A|rch, via logger.conf and its disabled because it grows like a hell |
| 23:00.03 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-gvgfpafvnqjbhjdr) |
| 23:00.06 | nny | vandyk: the pbx is on a public interface |
| 23:00.21 | nny | externip not xternip |
| 23:00.55 | vandyk | and between phone and pbx box does not have any firewall? |
| 23:01.21 | navaismo | nny, how is the nat setting on the extension that doesn't ack |
| 23:01.40 | nny | navaismo: nat=yes |
| 23:01.43 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:02.19 | nny | navaismo: my gut says remote router malarky. I may see if they can plug in directly to the modem if if's not a DSL |
| 23:02.48 | navaismo | Mon|A|rch, if you want only the sip for current session without enable the full log, use tee and save to a txt file, run from linux shell: asterisk -rnvvvvvvdd | tee mylog.txt |
| 23:03.21 | navaismo | Mon|A|rch, the do your stuff as usual and when you finish exit from asterisk cli and you will see the mylog.txt file with all the output |
| 23:03.33 | nny | damn it's DSL |
| 23:04.05 | Mon|A|rch | navaismo, k |
| 23:06.30 | Mon|A|rch | unfortunately my only avenue for testing today just left the office >< |
| 23:06.36 | Mon|A|rch | wonder if i can remote into their computers |
| 23:06.40 | nny | navaismo: going to have him test on another network |
| 23:08.45 | navaismo | Mon|A|rch, teamviewer, vnc, showmypc |
| 23:08.58 | Mon|A|rch | lol |
| 23:09.06 | Mon|A|rch | I've already remoted in |
| 23:13.24 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
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| 23:34.59 | Mon|A|rch | so it looks like the caller id wasn't the problem |
| 23:35.10 | Mon|A|rch | :( |
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| 23:53.18 | ChannelZ | What was the problem? |
| 23:56.00 | *** part/#asterisk tech_travis (~Travis@174.46.237.60) |