| 00:02.54 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 00:02.54 | *** mode/#asterisk [+o angler] by ChanServ |
| 00:12.50 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 00:12.50 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 00:50.13 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 00:50.13 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 01:02.58 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
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| 01:32.26 | Kilo`byte | okay so if i use blink, i get issues with RTP "We are requesting SRTP for audio, but they responded without it!" |
| 01:32.58 | Kilo`byte | how would i fix that, best case server side (blink works with another SIP server and linphone works with both fine) |
| 01:33.26 | [TK]D-Fender | That message means exactly what it says |
| 01:33.51 | [TK]D-Fender | You are not configuring both sides in a way in which they agree |
| 01:34.02 | [TK]D-Fender | So go fix that |
| 01:34.20 | Kilo`byte | [TK]D-Fender: well, is there a way where the server can automatically detect weather to zuse RTP or not? |
| 01:34.27 | Kilo`byte | can't type tonight |
| 01:34.32 | [TK]D-Fender | No., |
| 01:34.39 | [TK]D-Fender | You tell a peer to use SRTP... and it will. |
| 01:34.44 | [TK]D-Fender | Tell it not to .. and it won't |
| 01:34.53 | Kilo`byte | good time to dig up blink documentation \o/ |
| 01:35.04 | [TK]D-Fender | Tell it not to .. and then try asking it to and it will fail. |
| 01:36.46 | Kilo`byte | [TK]D-Fender: wait, huh? |
| 01:37.01 | *** part/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 01:38.19 | Kilo`byte | to me it looks like the server wants to use RTP, but the client doesn't |
| 01:39.12 | *** join/#asterisk rudilee (~rudilee@116.50.28.75) |
| 01:43.50 | Kilo`byte | [TK]D-Fender: or am i wrong with that assumption |
| 01:44.05 | [TK]D-Fender | try loking at the actual call in full |
| 01:44.09 | [TK]D-Fender | that means SIP DEBUG enabled |
| 01:44.27 | [TK]D-Fender | ~assumptions |
| 01:44.27 | infobot | assumptions is, like, now unless otherwise stated, most support here is based on the assumptions that: 1. You're using the box you're trying to fix now, not some other box (or OS), 2. You're using Debian, 3. You have physical access to the system (it's not hours' drive, or flight, away), 4. You understand English, 5. You will check commands given against online documentation (man, info, /usr/share/doc, HOWTOS, Google) (more in assumptions2) |
| 01:44.42 | [TK]D-Fender | hrm... |
| 01:44.46 | [TK]D-Fender | that's new |
| 01:44.58 | [TK]D-Fender | ~assumption |
| 01:44.58 | infobot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 01:45.04 | [TK]D-Fender | There you are!' |
| 01:45.18 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 01:45.40 | *** join/#asterisk frek818 (~frek818@172.56.16.99) |
| 01:46.55 | Kilo`byte | [TK]D-Fender: not fulfilling the second one, other than that its correct (the box is downstairs and i am connected via ssh) |
| 01:47.32 | Kilo`byte | regarding link... |
| 01:48.12 | Kilo`byte | ah lol |
| 01:48.15 | Kilo`byte | xD |
| 01:48.40 | Kilo`byte | [TK]D-Fender: what would i look for in log, as it output quite a bit of data and i am not that familar with SIP internals |
| 01:51.17 | Kilo`byte | eh i gotta get some sleep |
| 01:51.26 | Kilo`byte | will figure this out tomorrow |
| 01:51.39 | Kilo`byte | linphone is just too crashy for my taste |
| 01:51.45 | Kilo`byte | thanks anyways though |
| 01:52.50 | *** join/#asterisk frek818 (~frek818@172.56.16.99) |
| 02:06.32 | *** join/#asterisk dvorkbjel (~viskestel@li607-220.members.linode.com) |
| 02:14.51 | *** join/#asterisk justinmrkva (~justinmrk@unaffiliated/justinmrkva) |
| 02:15.58 | justinmrkva | Hey, I'm wondering if I can get a little help setting up Asterisk as a VoIP server. |
| 02:16.18 | WIMPy | ~ask |
| 02:16.18 | 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. |
| 02:18.02 | justinmrkva | I want to use it with the Zoiper app, but I'm a little confused. First, I found this tutorial http://www.asteriskguru.com/tutorials/zoiper.html but it looks like the user has to set up the account in two places, first to register the account (under prerequisites) and second to define (?) the account (under Creating a SIP account). |
| 02:19.21 | justinmrkva | Am I correct in my assumption that the line "register => user:password@domain.com" will register that user properly? |
| 02:22.59 | WIMPy | register lines are for outbound registrations. |
| 02:23.13 | WIMPy | I.e. if you Asterisk is to register itself somewhere. |
| 02:23.34 | justinmrkva | Okay, that makes sense. (I literally have no idea what I'm doing :D) |
| 02:23.37 | WIMPy | your |
| 02:23.55 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 02:24.40 | justinmrkva | Most of the guides are chock full of telephony jargon and extra stuff for configuring hardware that I don't care about, so it's extremely confusing at first. |
| 02:25.56 | WIMPy | Unfortunately Asterisk terminology is often quite different from other telephony terminology. |
| 02:26.08 | WIMPy | ~primer |
| 02:26.09 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
| 02:26.12 | justinmrkva | Which in itself can be confusing anyway :) |
| 02:26.13 | WIMPy | ~book |
| 02:26.13 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 02:38.44 | *** join/#asterisk frek818 (~frek818@172.56.16.99) |
| 03:09.46 | *** join/#asterisk frek818 (~frek818@172.56.16.99) |
| 03:10.12 | justinmrkva | The burner.com page was a much saner introduction than most that I found :) |
| 03:10.57 | *** join/#asterisk foulou (~foulou@gw-tech.lagoon.nc) |
| 03:11.06 | justinmrkva | A quick question as well. Is there any problem with having two SIP clients behind a single router? (i.e. with a single external IP address) assuming both are set up with host=dynamic? |
| 03:12.18 | justinmrkva | I had two devices, each with a different account. Calling one from the other didn't work consistently. I was able to get them to connect once and everything looked good, but I wasn't able to get them to connect again |
| 03:15.57 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:18.52 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 03:19.00 | justinmrkva | Okay. I have a friend testing. We're getting extremely inconsistent... everything. Sometimes (on my client) I get zero incoming packets but a lot of outgoing packets. Sometimes I get nothing incoming or outgoing. What should I be looking for here? |
| 03:20.28 | justinmrkva | The simple echo program seems to be working fine, we're just not getting any audio between two clients/users. The calls are automatically dropped after a few seconds. |
| 03:21.55 | justinmrkva | This is what the relevant section of extensions.conf looks like: http://hastebin.com/rizatofuye.coffee |
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| 03:44.54 | *** join/#asterisk bazman (~army@59-100-195-234.mel.static-ipl.aapt.com.au) |
| 03:50.16 | justinmrkva | So for some reason, zoiper works fine but linphone doesn't connect at all. No idea why. It just gives two beeps and I get no audio. |
| 03:50.17 | ChannelZ | Extensions doesn't really help for audio problems. Are you behind a firewall? Is the remote friend behind one? |
| 03:51.26 | *** join/#asterisk frek818 (~frek818@172.56.16.99) |
| 03:54.47 | bazman | Has anyone a suggestion on how I can outsource the development of ECMA-312 specifically for ATS-QSIG which means 4 channels (3B+D) muxed into a single E1 64kbit slot. Thanks |
| 03:58.30 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 04:03.50 | justinmrkva | So Zoiper works fine on my WiFi network but Linphone won't work unless it's on 3G (indicating some sort of firewall issue). My friend is also on 3G using Vimphone and he can't get anything, even the echo test. |
| 04:05.03 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 04:15.16 | *** join/#asterisk frek818 (~frek818@172.56.16.99) |
| 04:23.13 | ChannelZ | Is the machine Asterisk is running on behind NAT/afirewall? |
| 04:23.40 | justinmrkva | Nope, no firewall on the server and it's a dedicated IP, no NAT |
| 04:24.30 | ChannelZ | well it's entirely possible his carrier purposely blocks "rogue" voip |
| 04:26.26 | ChannelZ | beyond that it would be more instructive to see a SIP debug of a test call |
| 04:27.55 | justinmrkva | Another weirdness. I have two phones, A and B. A is running Linphone, B is running Vimphone. (They're essentially the same). Both are on WiFi behind a router. I can call A from B and B from A. However, neither A nor B can call the echo test number. But if I put A on 3G, it can call the echo number just fine and A and B can still call each other, but they lose all audio. |
| 04:28.54 | *** join/#asterisk timahvo1 (~rogue@mail.cickenya.com) |
| 04:35.12 | justinmrkva | So it looks like two devices inside the router can call each other, but not the test extension. On 3G/LTE, they can call the test extension, but not each other. |
| 04:46.18 | *** join/#asterisk u0m3_ (~u0m3@92.80.89.9) |
| 04:50.35 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 05:09.06 | justinmrkva | So except for the iPhone on LTE, none of the devices are able to get the test server to respond at all. I even went as far as connecting to the SIP server via VPN, and still not a single device can get a response. |
| 05:09.35 | justinmrkva | I'm calling it a bust. I can't rely on something that's so picky about the network. |
| 05:16.51 | [TK]D-Fender | justinmrkva, you haven't really looked at the call yet from what I've seen here |
| 05:17.02 | [TK]D-Fender | justinmrkva, You showed some dialplan but that has no impact on audio |
| 05:17.36 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 05:20.39 | justinmrkva | I checked the log and it's trying to do NAT traversal and just keeps retransmitting the INVITE. Similar retransmits happen when calling the test number as well. |
| 05:20.48 | justinmrkva | s/log/debug |
| 05:21.07 | [TK]D-Fender | "sip set debug on" |
| 05:21.13 | [TK]D-Fender | pastebin the entire attempt end to end |
| 05:21.25 | [TK]D-Fender | let's see what's really going on |
| 05:27.38 | justinmrkva | Here it is: http://pastebin.com/yeBmwhQ2 I replaced the IP addresses. The phone on the WiFi network is using justinmrkva-test and is on [[IP1]] and the LTE phone is on [[IP2]] using justinmrkva. The server is [[SRVIP]] |
| 05:28.00 | *** join/#asterisk frek818 (~frek818@172.56.16.99) |
| 05:29.00 | [TK]D-Fender | I need to see actual IP's to track this. |
| 05:29.07 | [TK]D-Fender | PM me a PW'd PB if you have to |
| 05:29.50 | [TK]D-Fender | Also restart the SIP client on the device if you can during that timeframe |
| 05:30.29 | justinmrkva | There's no reason to need real IPs |
| 05:30.38 | [TK]D-Fender | From what I see you also don't have any NAT keep-alive's so taht could cause the NAT mapping hole to close up behind you |
| 05:30.55 | [TK]D-Fender | Your peers should all have "wualify=yes", "directmedia=no" |
| 05:31.00 | [TK]D-Fender | Fix these and retest |
| 05:31.07 | [TK]D-Fender | (after resting the SIP clients |
| 05:31.12 | [TK]D-Fender | restarting* |
| 05:32.41 | justinmrkva | s/wualify/qualify I assume :) |
| 05:33.08 | justinmrkva | Also can those go in the [general] block or do they have to be repeated for each user? |
| 05:33.14 | [TK]D-Fender | qualify* |
| 05:33.27 | [TK]D-Fender | I recommend doing it explicit. |
| 05:33.43 | [TK]D-Fender | make settings like you mean them |
| 05:35.37 | justinmrkva | Actually it's better to write once than multiple. Reduces mistakes when making broad changes and/or adding new entries. |
| 05:35.47 | justinmrkva | In any case, I did 'sip reload' - testing now |
| 05:36.23 | [TK]D-Fender | Or you end up making a global assumption, and never thing about certain values until the defaults screw you. |
| 05:36.50 | *** join/#asterisk gryphon_ (~gryphon@82.140.120.164) |
| 05:36.57 | [TK]D-Fender | I prefer the my way, first time, every time, always right, no guessing approach. |
| 05:37.07 | [TK]D-Fender | To each their own (as long as it works) |
| 05:38.14 | *** join/#asterisk gkuhnert (~gkuhnert@60.225.168.171) |
| 05:38.58 | gkuhnert | hello all - Anyone here got a few mins to debug inbound IAX calls failing on inbound trunk route? |
| 05:39.19 | justinmrkva | After making those changes, neither A nor B can call each other (even on the same network). I'm trying the one-on-WiFi, one-on-LTE approach again and I'll paste the new debug. |
| 05:40.20 | [TK]D-Fender | gkuhnert, PASTEBIN is your friend |
| 05:40.21 | [TK]D-Fender | ~pb |
| 05:40.21 | 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. |
| 05:40.22 | [TK]D-Fender | ^^^ |
| 05:41.00 | gkuhnert | Who needs that when you have your own web server :) |
| 05:41.07 | gkuhnert | but thanks for the tip about not flooding here :) |
| 05:42.06 | gkuhnert | standby - will replicate issue and get current log |
| 05:43.33 | *** join/#asterisk cargill (~ondra@host86-177-94-163.range86-177.btcentralplus.com) |
| 05:43.50 | gkuhnert | Actually - this one won't be anything big in terms of debug... |
| 05:44.00 | gkuhnert | Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT |
| 05:44.01 | gkuhnert | <PROTECTED> |
| 05:44.39 | gkuhnert | No reason for reject message showing - trying to proceed beyond this point.....but without cause code, its a bit hard |
| 05:44.59 | gkuhnert | Is there any debugging settings that will show more data? |
| 05:45.13 | justinmrkva | [TK]D-Fender New paste with qualify=yes and directmedia=no (same IP replacements as before) |
| 05:45.16 | justinmrkva | http://pastebin.com/qtdH7nVc |
| 05:46.31 | *** join/#asterisk LiuYan (~hola@unaffiliated/liuyan) |
| 05:47.40 | [TK]D-Fender | Peer audio RTP is at port 10.0.1.10:7078 |
| 05:47.42 | [TK]D-Fender | there we go. |
| 05:48.02 | [TK]D-Fender | bad media IP is still being accepted |
| 05:48.27 | [TK]D-Fender | ok, for your peers : nat=force_rport,comedia |
| 05:50.45 | *** join/#asterisk stasdizzi (~stasdizzi@159.224.69.205) |
| 05:52.02 | gkuhnert | http://www.gregkuhnert.com/asterisk |
| 05:52.10 | gkuhnert | config is in that directory for anyone that has a moment... |
| 05:57.38 | justinmrkva | Okay, using nat=force_rport,comedia still doesn't work. Actually, the iPhone (on LTE) receives audio, and the Android phone (NAT) doesn't. Hanging up from the Android phone works, but hanging up from the iPhone leaves the Android phone "connected". Debug: http://pastebin.com/8QDRaFSg |
| 05:59.41 | [TK]D-Fender | try with jsut comedia |
| 06:04.38 | justinmrkva | Here's the result of that: http://pastebin.com/PqBVvp9R |
| 06:05.36 | justinmrkva | Still no audio on the Android phone, but when I hung up the iPhone, the Android phone appeared to respond. |
| 06:05.43 | [TK]D-Fender | justinmrkva, Ok, seems Peer audio RTP is at port 10.0.1.10:7078 |
| 06:05.44 | [TK]D-Fender | gah |
| 06:05.53 | [TK]D-Fender | I'm just missing something easy on this... |
| 06:06.18 | [TK]D-Fender | this is just a SIP NAT otions and I'm mixed mixed up on the one to get it to properly ignore the RTP IP |
| 06:06.40 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 06:08.07 | *** join/#asterisk stasdizzi (~stasdizzi@159.224.69.205) |
| 06:09.50 | gkuhnert | for whoever came to get my conf file - sorry - wrong permission - it is available now - together with log files.... |
| 06:15.26 | gkuhnert | anyone with any ideas here? |
| 06:15.27 | gkuhnert | bueler? |
| 06:15.45 | gkuhnert | http://www.gregkuhnert.com/asterisk |
| 06:15.48 | [TK]D-Fender | We'd have to see the receiving end |
| 06:17.04 | gkuhnert | This is the receiving end |
| 06:17.39 | gkuhnert | Call successfully leaves originating end - hits me - no nat issues - but call rejected by IAX without any cause code |
| 06:17.58 | gkuhnert | Context is correct... |
| 06:18.06 | [TK]D-Fender | Need to see both ends.... |
| 06:19.43 | gkuhnert | Config file from other end? |
| 06:19.48 | gkuhnert | Let me see if I can get that. |
| 06:20.47 | [TK]D-Fender | and DEBUG |
| 06:20.55 | [TK]D-Fender | also need to knwo precisely what version on each side |
| 06:21.02 | gkuhnert | What specific debug would you like? |
| 06:21.11 | gkuhnert | Nothing else is printed with core debug 999 |
| 06:21.16 | gkuhnert | and core verbose 99 |
| 06:21.19 | gkuhnert | 999 |
| 06:22.40 | gkuhnert | source = 4.2 |
| 06:22.50 | gkuhnert | source = 1.4.2 |
| 06:23.44 | *** join/#asterisk LiuYan (~hola@unaffiliated/liuyan) |
| 06:24.41 | [TK]D-Fender | theres the problem |
| 06:24.51 | [TK]D-Fender | IAX2 changed in 1.6.X |
| 06:24.51 | gkuhnert | Yes, its old - but stable.... |
| 06:25.05 | [TK]D-Fender | "you need "requirecalltoken=no" in your newer peer |
| 06:25.09 | gkuhnert | I know there were changes - which is why I have some of the options you see in my config. |
| 06:25.27 | gkuhnert | Already have that.... in my config |
| 06:25.29 | [TK]D-Fender | I don't see it in you PEER |
| 06:25.38 | [TK]D-Fender | and that's what you use to call |
| 06:25.59 | gkuhnert | Damm you.... |
| 06:26.08 | gkuhnert | I know I had that in there once before |
| 06:26.14 | gkuhnert | let me put it back and re test. |
| 06:26.35 | gkuhnert | If thats it - I am going to be doing a major face palm |
| 06:27.42 | gkuhnert | Call still failed. |
| 06:27.47 | gkuhnert | (I had that in there once before) |
| 06:29.38 | gkuhnert | Any other takers? |
| 06:29.39 | [TK]D-Fender | Well I don't see the other side and I'm out of time. |
| 06:29.50 | [TK]D-Fender | maybe someone else will pick this up |
| 06:29.59 | [TK]D-Fender | heads to bed |
| 06:30.06 | *** part/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 06:31.27 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 06:33.45 | gkuhnert | damm |
| 06:33.52 | gkuhnert | anyone else out there - source config now posted. |
| 06:36.50 | gkuhnert | http://www.gregkuhnert.com/asterisk |
| 06:36.57 | gkuhnert | (one more time for any new joiners) |
| 06:41.34 | *** join/#asterisk CeBe (~CeBe@port-92-200-117-227.dynamic.qsc.de) |
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| 07:08.47 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 07:10.31 | doop | gkuhnert: where's the rest of iax.conf |
| 07:10.50 | gkuhnert | that is just the additional file - but give me a sec - may have just found something |
| 07:11.45 | gkuhnert | Nope - was not it. |
| 07:12.00 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 07:12.15 | doop | not sure what you mean by "additional file" -- can you post the rest of iax.conf please |
| 07:12.17 | gkuhnert | The rest of the iax.conf file are includes... |
| 07:12.34 | gkuhnert | i.e. = freepbx built config |
| 07:12.52 | doop | you'd probably be better off asking in freepbx's channel |
| 07:12.57 | gkuhnert | just posted the main iax.conf |
| 07:13.38 | doop | 403 forbidden on iax.conf gkuhnert |
| 07:13.47 | gkuhnert | damm |
| 07:13.48 | gkuhnert | 1 sec. |
| 07:14.15 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 07:14.22 | gkuhnert | renamed. |
| 07:14.32 | gkuhnert | My server blocks conf files |
| 07:14.40 | doop | teriffic |
| 07:14.51 | doop | ok so here's the thing |
| 07:15.00 | gkuhnert | It is actually - stops hackers from getting data they shouldn't see. |
| 07:15.19 | doop | without knowing all the crap that's wheelbarrowed into the other iax.conf files i can't help you |
| 07:15.30 | gkuhnert | Not much to speak of.... Just looking at em |
| 07:15.45 | *** join/#asterisk zblk (~andrey@92.53.115.234) |
| 07:15.45 | doop | well that "not much to speak of" might be quite crucial |
| 07:16.35 | gkuhnert | Let me add the relevant ones to the server... |
| 07:16.44 | doop | thanks |
| 07:16.50 | doop | add them all |
| 07:18.18 | gkuhnert | Thats it |
| 07:18.23 | gkuhnert | The only missing file is a registration file... |
| 07:18.29 | gkuhnert | just left it out for security reasons |
| 07:18.44 | gkuhnert | The others are empty files. |
| 07:19.32 | gkuhnert | Put the relevant one there anyway for luck... |
| 07:19.37 | gkuhnert | just tweaked content. |
| 07:19.48 | doop | just for sanity checking here, are you using the same codec on both ends |
| 07:20.11 | gkuhnert | I just did that manually for testing purposes a minute ago.... hard wired it into the block - no issue |
| 07:20.21 | gkuhnert | Other strange piece - Outbound calls work - just inbound fails. |
| 07:20.28 | gkuhnert | and no data in log files with high debug. |
| 07:20.43 | doop | what do you mean "hard wired it into the block" |
| 07:20.47 | doop | codec is not a thing with wires |
| 07:21.00 | gkuhnert | Just edited the block for the peer.... put it in there |
| 07:21.16 | gkuhnert | i.e. - cut and pasted codecs allowed from sending config |
| 07:21.46 | doop | and when you say "no issue" does that mean the call completed successfully? |
| 07:22.14 | gkuhnert | Outbound works |
| 07:22.16 | gkuhnert | inbound fails |
| 07:22.21 | gkuhnert | outbound completes perfectly |
| 07:22.39 | doop | when inbound fails, do you get an error message |
| 07:22.58 | gkuhnert | see full |
| 07:23.08 | gkuhnert | its the log file - rejected - no cause code listed |
| 07:23.23 | doop | the log file from which end |
| 07:23.29 | gkuhnert | receiving end |
| 07:23.38 | doop | what error do you get on the calling end |
| 07:23.48 | gkuhnert | Unfortunately, not my server - |
| 07:24.02 | gkuhnert | we debug'd ages ago - and couldn't find anything relevant ..... |
| 07:24.11 | gkuhnert | server owner now in transit.... so can't get right now |
| 07:24.16 | doop | ok |
| 07:24.45 | doop | and 0212345678 is a valid number on the called server? |
| 07:25.07 | gkuhnert | That is edited only in what I display to you... |
| 07:25.10 | gkuhnert | but it matches in real life |
| 07:25.35 | gkuhnert | The number on called server should be matching an inbound route with a wildcard match..... |
| 07:25.45 | doop | and you're absolutely one hundred thousand million per cent sure that you've not misspelled a context name or something, yes? |
| 07:25.50 | gkuhnert | I have other peers matching inbound wildcard match - that should send to a group. |
| 07:26.06 | gkuhnert | I even went as far as to packed capture the IAX packet - and compare to config. |
| 07:26.23 | gkuhnert | What really gets me - is that there is nothing in the log - if it left a clue, we'd be OK.... |
| 07:26.24 | gkuhnert | but it hasnt |
| 07:26.33 | doop | ok. forgive me in trying to get up to speed on this issue |
| 07:26.37 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 07:26.51 | gkuhnert | One thing that was puzzling to me.... |
| 07:27.00 | gkuhnert | sending config file says context = username |
| 07:27.11 | doop | does this happen only with this one server you're trying to receive calls from or? |
| 07:27.12 | gkuhnert | but that is not what I saw in receiving packets last time I did a trace... |
| 07:27.18 | gkuhnert | Correct. |
| 07:27.27 | gkuhnert | Only one IAX peer - other peers are SIP |
| 07:28.20 | doop | i think that context= is only for receiving calls |
| 07:28.25 | doop | note the type=friend |
| 07:28.39 | *** join/#asterisk CriminalMinds (~hola@unaffiliated/liuyan) |
| 07:29.18 | gkuhnert | Yep |
| 07:29.20 | gkuhnert | that is his end. |
| 07:29.22 | doop | just for shits and giggles, what happens if you make a type=friend context in lieu of your peer and user contexts in iax.conf on your end |
| 07:29.28 | gkuhnert | I just ttried that for my end already :) |
| 07:29.35 | doop | and? |
| 07:29.38 | gkuhnert | see file with NEW on my end |
| 07:29.40 | gkuhnert | in the filename |
| 07:29.44 | gkuhnert | FAIL |
| 07:30.12 | doop | what is this aLaw=ulaw crap |
| 07:30.20 | doop | there's no such setting |
| 07:31.15 | doop | there's no codec setting in your user and peer definitions -- have you tried putting such settings there |
| 07:31.20 | doop | disallow=all |
| 07:31.22 | doop | allow=ulaw |
| 07:31.25 | doop | allow=alaw |
| 07:32.17 | gkuhnert | yeah |
| 07:32.19 | gkuhnert | i just saw that. |
| 07:32.24 | gkuhnert | 1 sec.... |
| 07:32.29 | gkuhnert | bad cut and paste... |
| 07:32.51 | doop | i should hope you're reading what you're cutting and pasting into your config files |
| 07:34.06 | gkuhnert | When I pasted it a moment ago - there was a character set issue during pasting from the email of his config..... Yes, I read it - and I made a mistake when correcting..... |
| 07:34.37 | gkuhnert | My gui (freepbx) changed it to.... |
| 07:34.39 | gkuhnert | allow=alaw&ulaw&ilbc |
| 07:34.46 | gkuhnert | is that legal syntax? |
| 07:35.14 | doop | how about let's just bring it down to allow=alaw and forget the other two for now |
| 07:35.28 | doop | unless you're into ulaw in which case choose ulaw instead of alaw |
| 07:36.03 | gkuhnert | FAIL |
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| 07:36.15 | doop | on both ends? |
| 07:37.36 | gkuhnert | I can still dial out |
| 07:37.55 | gkuhnert | My view = not codec issue |
| 07:38.16 | doop | your choices now are to try just plain vanilla asterisk, to try SIP instead of IAX, and/or to ask for support in the freepbx channel |
| 07:39.44 | gkuhnert | To be honest, freepbx channel won't help if there is no debug messages to find... That is the frustrating bit.... |
| 07:39.48 | gkuhnert | the config LOOKS right.... |
| 07:40.55 | doop | i dont know what to tell you beyond freepbx is generally problematic when it comes to these kinds of issues |
| 07:41.05 | gkuhnert | To be honest - I think it would be considered a bug to reject a call without a cause code. |
| 07:41.27 | gkuhnert | and that aint gonna be freepbx. |
| 07:41.48 | doop | well, try vanilla asterisk and see if you can reproduce the bug |
| 07:41.51 | gkuhnert | Funny story - the guy that I am peering with - he's actually built what I think is the world's largest asterisk cluster..... servicing tens of thousands of users.... |
| 07:42.05 | gkuhnert | and this is a smaller older asterisk that I peer with him on.... |
| 07:42.17 | doop | what about trying a newer asterisk |
| 07:42.36 | gkuhnert | And doop - with respect, freepbx is just a front end.... It shouldn't be this hard - and asterisk should give cause codes. |
| 07:42.44 | gkuhnert | Anyway.... I will go back to him again..... |
| 07:42.52 | doop | i realize and appreciare your frustration man |
| 07:43.13 | gkuhnert | Was trying to self-serve on public forums.... He's hard to pin down.... |
| 07:43.31 | gkuhnert | Thanks anyway. |
| 07:43.33 | doop | have you tried a newer version of asterisk |
| 07:43.40 | gkuhnert | For me = fully patched... |
| 07:43.42 | doop | sorry i'm not able to be of help |
| 07:43.46 | gkuhnert | His is a little older / custom / bespoke.... |
| 07:43.50 | gkuhnert | but VERY stable. |
| 07:44.03 | doop | perhaps someone else here can help you then |
| 07:44.13 | gkuhnert | and works with MANY iax peers of various flavours |
| 07:44.19 | gkuhnert | including other freepbx peers |
| 07:44.56 | gkuhnert | Anyway. Enough for now. Nearly dinner time. |
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| 08:03.10 | rue_bed | can anyone tell me how to check the "audio levels" on a voip channel, in real time? |
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| 14:56.31 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.1.0 (2014/12/15), 11.15.0 (2014/12/15), 1.8.32.1 (2014/11/20); Standard: 12.8.0 (2014/12/15); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 15:31.24 | pigpen | I have an odd thing happening. Not necessarly Asterisk directly though. Polycom Soundpoint 550's. 3.2.7 I think. Work fine, network provisioned, then out of the blue, picking up the handset does not answer an incoming call. |
| 15:31.50 | pigpen | Unplugging the handset and plugging it back in resolve the issue temporarly. |
| 15:32.03 | pigpen | Happening to 20+ phones. |
| 15:32.10 | pigpen | ideas? |
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| 15:50.48 | Johnny5_ | pigpen: Common denominators. PoE switch perhaps? Would advise following up with Polycom support. |
| 15:52.19 | pigpen | yeah. that is what I was thinking (poly support) |
| 15:52.41 | pigpen | I doubt it is the poe switches, as we have two vendors, happening across the board. |
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| 16:29.38 | litn | hello, is there any way for me to check who hung up a call? I mean, which end? |
| 16:29.47 | litn | the asterisk logs don't say |
| 16:30.40 | [TK]D-Fender | If you're talking about a call in the past.. what you have is what you have. There is nothing more. |
| 16:30.53 | [TK]D-Fender | Going forward.. you'd have to be debugging the channel-types in question |
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| 16:33.48 | litn | this is from our sip trunk (flowroute) to a local ext. |
| 16:34.36 | [TK]D-Fender | So SIP debug it is... |
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| 18:19.11 | babak | Could some one point me where to study for making big queue with multiple Asterisk ? |
| 18:19.46 | [TK]D-Fender | "big queue"? |
| 18:20.37 | doop | in spanish: queue grande |
| 18:22.22 | babak | I mean a queue with 500 agent and more than that waitng |
| 18:25.23 | [TK]D-Fender | Low odds that anyone has written a "study" about it. |
| 18:26.23 | [TK]D-Fender | The only people who do things like that are companies that need to do it and they aren't in the business of writing "how to duplicate our hard-earned successes" studies. |
| 18:28.18 | *** join/#asterisk Kunsi (kunsmannf@unaffiliated/kunsi) |
| 18:28.37 | newtonr | babak, have you already been through the Asterisk Definitive Guide? |
| 18:28.45 | newtonr | ~book |
| 18:28.45 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:29.03 | babak | [TK]D-Fender: I should write a RFP and evaluate some contact center solution for our company , so need to know which methods are available and which method is better |
| 18:29.24 | newtonr | babak, it isn't specifically what you asked for, but it does provide you lots of information about Administering Asterisk, building queues and clustering systems |
| 18:29.42 | babak | newtonr: yes according to book max queue size is limited by one server capacity |
| 18:30.22 | Kunsi | hello, i've got a problem not directly related to asterisk, but maybe soneone in here is able to help me: i've got two snom 360 phones, both connected to my network. they get an IP via DHCP, but they don't react to ping or http, also they don't attempt a connection to my asterisk server |
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| 18:30.50 | pabelanger | babak: have a site with 1500 queue members |
| 18:30.52 | pabelanger | over 3 boxes |
| 18:31.16 | Kunsi | ( i have tried resetting phones and different firmware versions, yes) |
| 18:31.48 | babak | pabelanger: 1500 members in one queue ? |
| 18:32.20 | pabelanger | across multiple queues |
| 18:32.38 | pabelanger | why do you want a single queue? |
| 18:33.27 | babak | our company need 3000 agent so my guess is one queue may be big |
| 18:34.18 | pabelanger | Ya, I'd try and reduce the agents into smaller queues |
| 18:34.24 | pabelanger | just for the purpose of managing |
| 18:35.17 | babak | pabelanger: what is your max agent or member per queue ? |
| 18:35.55 | pabelanger | 200 - 250 |
| 18:36.10 | pabelanger | mostly because we break them down into zones |
| 18:36.14 | pabelanger | or markets |
| 18:36.30 | babak | we now have multiple contact centers but consultants told that we should have one big contact center |
| 18:36.51 | pabelanger | why? |
| 18:36.59 | pabelanger | is multiple working for you today? |
| 18:37.24 | pabelanger | one big contact centre = single point of failure |
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| 18:37.31 | babak | better CRM, WFM.... |
| 18:37.42 | pabelanger | Ya, I can see that |
| 18:37.55 | pabelanger | asterisk down fall with app_queue is multi box |
| 18:38.00 | pabelanger | so, it is harder to manage |
| 18:38.29 | pabelanger | we have another client split over 7 physical locations |
| 18:38.37 | pabelanger | each site runs their own instances of everything |
| 18:38.44 | pabelanger | little harder to manage |
| 18:38.47 | pabelanger | but do able |
| 18:39.14 | babak | so begger queue solution based on Asterisk should based on AGI or ARI ? |
| 18:39.37 | pabelanger | well, for us, we are using ARI |
| 18:39.46 | pabelanger | but ARI means you need to build a queue from scratch |
| 18:40.00 | pabelanger | you can use AGI too |
| 18:40.10 | pabelanger | Still know people that built queues on asterisk using it |
| 18:41.11 | [TK]D-Fender | AGI isn't really fit for a "queue". It's missing control over other channels in there. |
| 18:41.26 | [TK]D-Fender | You'd need something globally overseeing them like an App can. |
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| 18:41.57 | pabelanger | right |
| 18:42.08 | pabelanger | other issues like creating bridges too |
| 18:42.17 | pabelanger | so you'd have to use app_meetme or app_confbridge |
| 18:42.35 | [TK]D-Fender | ew! |
| 18:42.54 | [TK]D-Fender | Or 'more fun ways to run out of headroom fast!" |
| 18:46.09 | babak | theoretically is it possible to bypass media from Asterisk queue box to other Asterisk boxes ? |
| 18:46.28 | babak | to increase capacity |
| 18:46.34 | [TK]D-Fender | No |
| 18:46.48 | [TK]D-Fender | App queue doesn't rebridge channels |
| 18:46.57 | pabelanger | Ya, not with current app_queue |
| 18:47.05 | [TK]D-Fender | AFAIK it always just sits there in the middle. |
| 18:47.17 | [TK]D-Fender | And media isn't the issue |
| 18:48.32 | pabelanger | Ya, ideally your not transcoding media on the queue box to start with |
| 18:48.51 | [TK]D-Fender | That's already suicide |
| 18:56.11 | WangDang | Hi, noob here. I'm reading throught the O'reilly book, 3rd Ed. I don't get extension state. Seems like device state; a way to know if someone is available or not. I'm sure I'm missing some minor detail, but don't really understand what it does that device state doesn't do. Any pointers? |
| 19:02.12 | [TK]D-Fender | WangDang: like it says it is based on DIALPLAN EXTENSION |
| 19:02.55 | [TK]D-Fender | WangDang: Perhaps a particular set of dialplan will conditionally call a number of devices. You can't track them all.. but you can see if the EXTNSION is being used and is in a particular state based on calls executing it from the dilaplan |
| 19:04.36 | WangDang | Why can't you track them all? Won't they all have a device state that can be known? |
| 19:05.05 | WangDang | or more accurately, won't they EACH have a device state |
| 19:06.00 | file | each device has a device state, an extension state is a composite of potential multiple and serves as a mapping |
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| 19:10.50 | WangDang | ya, OK. Seeing it as a mapping of 1 ext to potentially multiple end points, makes it a bit clearer, but why wouldn't * just consult the device directly, rather than trying to get it through a mapping? |
| 19:11.17 | file | because you don't subscribe a phone to a device - you subscribe it to an extension |
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| 19:18.56 | WangDang | so it's the sip phones that are asking for this info about extension state? |
| 19:20.26 | file | many can, yes |
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| 19:36.37 | WangDang | file: thanks, I think I get it now. I'm gonna have another crack at the code samples in that section of the book, now. Thanks |
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| 19:40.37 | [TK]D-Fender | WangDang: Suppose you want to know if there any callers sitting in a dialplan loop you have processing around like for a survey. you could monitor that for example. |
| 19:40.53 | [TK]D-Fender | WangDang: it's as valuable as you find reason to classify them as such. |
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| 20:16.02 | WangDang | [TK]D-Fender: I understand your first comment about monitoring a dialplan loop for callers, but I don't follow your second. It's valuable to classify what, how? I'm not sure what your referring back to by the "them" and the "as such". That one classifies a dialplan loop as being "occupied" by a caller? |
| 20:16.57 | [TK]D-Fender | Suppose you have some dialplan your callers go to that is acting as a survey. You could see if you HAVE any callers filling them in by their channel currently executing said extension |
| 20:17.24 | [TK]D-Fender | It's up to YOU to think of places it might be valuable to use EXTENSION_STATE for |
| 20:18.45 | [TK]D-Fender | WangDang: Perhaps you are using a certain extension for voicemail... then you could tell if they are in the process of leaving a voicemail. |
| 20:19.22 | [TK]D-Fender | WangDang: At this point it isn't the "who" that you are tracking, it's the "where" |
| 20:20.37 | WangDang | so you're tracking where a caller is in the dialplan? I thought I was getting it up to there |
| 20:24.19 | WangDang | has to slip out of the learning loop to make a couple calls to DO_ERRAND |
| 20:30.29 | [TK]D-Fender | WangDang: If "a" caller is executing dialplan. This involves states as well like ringing. |
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| 21:29.59 | [TK]D-Fender | checkout time, BBIAB |
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| 22:20.12 | tdonahue | hi all |
| 22:25.19 | tdonahue | does anyone know if features.conf will honor a #include to an external file like sip.conf or extensions.conf will do? |
| 22:26.15 | newtonr | it should |
| 22:27.34 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Using+The+include+and+exec+Constructs only mentions it not working in asterisk.conf and modules.conf |
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| 22:32.35 | pabelanger | tdonahue, yes |
| 22:38.43 | tdonahue | newtonr and pabelanger, thanks |
| 22:40.23 | tdonahue | i was trying to find it on the documentation for features.conf, never thought to search for documentation only on #include |
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