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| 02:56.59 | SeRi | team, I have an odd issue that has been haunting me for a while... I am using Asterisk 1.8.32.3 and all works well EXCEPT when I call my voicemail from any phone other than my Polycom 670 the audio is choppy. IE: My Gigaset or my WiFi SIP Phone. Any ideas? This is all internal to me (LAN) |
| 03:13.34 | SeRi | by the way this is just to my vmail |
| 03:13.40 | SeRi | normal calls work just fine |
| 03:18.06 | Ellenor | WTF?! |
| 03:18.09 | Ellenor | That's weird. |
| 03:19.33 | Ellenor | SeRi: Either the phone's fuckered, the program itself is fuckered (but PLEASE, for the sake of Our Flying Spaghetti Monster, upgrade to the latest LTS) or you've fucked a configuration line somewhere. |
| 03:20.10 | SeRi | Ellenor: LOL, I saw that one comming from a mile away. I am working on the upgrade but I am not all there yet. |
| 03:20.38 | SeRi | Ellenor: Its odd as hell... I can make calls just fine |
| 03:20.57 | Ellenor | SeRi: did you try turning it off and on again? (semi-serious) |
| 03:21.06 | SeRi | is just calling the vmail that sounds choppy from this phones |
| 03:21.16 | SeRi | Ellenor: LOL, Yes many times. |
| 03:21.28 | SeRi | I can try soft phones with the same issue |
| 03:21.42 | SeRi | Ellenor: ^^ |
| 03:22.25 | Ellenor | SeRi: Tried reinstalling from scratch?\ |
| 03:23.05 | SeRi | Ellenor: Meh... Not worth it. I have a dev system with Asterisk 11 on it that I am migrating my dial plan too... |
| 03:23.15 | Ellenor | Okay |
| 03:23.49 | SeRi | Ellenor: Thanks though.... I just been lazy as shit to migrate... I am that one that thinks if it works... Why fuck with it... lol |
| 03:24.40 | Ellenor | SeRi: Hopefully this will put you in low gear and heave the trottle mode. |
| 03:24.43 | Ellenor | throttle* |
| 03:25.09 | SeRi | Ellenor: +1 |
| 03:25.17 | Ellenor | SeRi: eh? |
| 03:25.34 | Ellenor | SeRi: also stick around the IRC, it's much more fun when you chat AND give/take support. :P |
| 03:26.28 | SeRi | Ellenor: Oh I know all about that... ;) I am an all timers around here, Just work and Family are taking my time now days... |
| 03:26.39 | SeRi | old timers* |
| 03:26.44 | SeRi | for got the sed replace command |
| 03:26.47 | SeRi | LOL |
| 03:29.02 | SeRi | let me try tis |
| 03:29.18 | SeRi | s/tis/&this/g |
| 03:29.24 | SeRi | crap |
| 03:29.25 | SeRi | lol |
| 03:29.27 | SeRi | almsot |
| 03:29.50 | SeRi | s/almsot/almost/g |
| 03:29.53 | SeRi | nice |
| 03:30.12 | SeRi | github to the rescue... :) |
| 03:30.25 | Ellenor | s/cue/uce/ |
| 03:30.28 | Ellenor | woops |
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| 04:46.50 | carrar | bwhahah http://imgur.com/gallery/gGjMGGg |
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| 07:36.34 | TheKingOfChaos | Hi i am trying to config ISDN30 from TDC in denmark i am using TE133F i cant get it to change from T1 to E1 |
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| 09:19.51 | guest1009 | Howdy! How does Asterisk scale with more cores? We have four cores which seems to work pretty hard currently, can we increase them to 8 and asterisk will cope well? |
| 09:21.04 | guest1009 | we have about 60 active dialogs and work load about 3.5 - 4.0 on all four cores |
| 09:21.31 | guest1009 | source htop |
| 09:31.22 | guest1009 | I believe the h323 codec is using some cpu |
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| 09:52.50 | WIMPy | TheKingOfChaos: Either you chang the jumper ot there's a module parameter to override it. |
| 09:53.25 | TheKingOfChaos | therte is not any jumper on the card |
| 09:53.42 | WIMPy | guest1009: H.323 is not a CODEC. So it's something else. |
| 09:59.51 | guest1009 | Ok WIMPy, ofc you are correct. |
| 10:01.07 | guest1009 | When using Asterisk as b2bua can I pass a sip header from one leg to another? In this case P-Asserted Identity |
| 10:01.18 | guest1009 | Is there a smooth way of doing this? |
| 10:01.38 | WIMPy | If you enable them on both peers they would pass automagically. |
| 10:02.53 | guest1009 | same => n,sipaddheader(P-Asserted Identity: ${SIP_HEADER(P-Asserted Identity)}) |
| 10:02.56 | guest1009 | something like that? |
| 10:03.18 | WIMPy | Not neccessay. Just enable them in sip.conf. |
| 10:03.51 | guest1009 | I am unsure of what you are talking about, I mean I did not know you handle headers in sip.conf |
| 10:04.21 | WIMPy | You configure chan_sip there. |
| 10:04.37 | WIMPy | And it already does support PAI. |
| 10:05.14 | WIMPy | See 'trustrpid' and 'sendrpid'. |
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| 10:05.29 | guest1009 | if there is no PAI, will it be ignored? So we do not send an empty PAI |
| 10:06.02 | WIMPy | If it wasn't there, it would use the 'from'. |
| 10:06.02 | guest1009 | Asterisk does nothing when it receives a P-Asserted-Identity header. It ignores it totally no matter what settings you use for "trustrpid" or "sendrpid". It does not copy it from an inbound call leg to an outbound call leg for a bridged SIP-to-SIP call. |
| 10:06.05 | guest1009 | :( |
| 10:06.28 | WIMPy | Who says so? |
| 10:06.36 | guest1009 | voip-info, I know... they cannot be trusted |
| 10:06.44 | guest1009 | Can you confirm this is false statment? |
| 10:06.47 | WIMPy | Did you take a look at the values I gave you? |
| 10:07.28 | WIMPy | Look at sip.conf or wiki.asterisk.org. |
| 10:08.26 | WIMPy | You have to do it manually if you want it in addition to what's in the 'from'. |
| 10:08.47 | WIMPy | But PAI is definitely supported. |
| 10:09.08 | guest1009 | yeah we have different values in from and pai. From field show the actual number while PAI sends anonymous. |
| 10:09.33 | WIMPy | Funny combination. |
| 10:10.00 | guest1009 | Does that mean we need dialplan hacking? |
| 10:10.21 | WIMPy | Depends on what exactely you need. |
| 10:10.58 | guest1009 | We need the PAI to copy from incoming leg to outgoing, no matter what. |
| 10:11.55 | WIMPy | well, I guess your version might be better then. |
| 10:18.56 | guest1009 | yeah, we just need to check if the PAI header actually exists |
| 10:19.09 | guest1009 | before doing anything, or we will be sending an empty PAI header. I do not know about the consequences |
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| 10:48.26 | Lachezar | Is it possible to create SIP users in Asterisk that are not numeric (extensions)? |
| 10:49.28 | catphish | Lachezar: yes |
| 10:49.40 | catphish | Lachezar: i always name mine with people's usernames |
| 10:50.07 | catphish | there's no need to name extensions the same as their dialplan numbers |
| 10:50.54 | WIMPy | Using numbers will only make it easier for attackers. |
| 10:51.24 | Lachezar | catphish: Hm. So I suppose the requirement to have it all-numeric is somewhat Elastix-specific? |
| 10:51.41 | catphish | i don't know anything about elastix i'm afraid |
| 10:51.59 | WIMPy | There isn't even a need to have your extensions numeric. |
| 10:52.31 | catphish | but i use usernames for my SIP peers, then use numbers in the dialplan (although as WIMPy says technically those dont have to be numbers either) |
| 10:53.59 | Lachezar | catphish: OK. Let me restart: I have access to an Asterisk that has a trunk, and various internal extensions (connected to Hardware phones). I'd like to have a second SIP account for every extension, and send a Message to that account on incoming call. |
| 10:54.44 | catphish | Lachezar: you want 1 number to go to 2 SIP accounts |
| 10:55.06 | WIMPy | What kind of message? |
| 10:55.25 | catphish | probably best to avoid the word "extension" and talk about either SIP accounts or numbers, the 2 things are quite separate |
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| 10:56.28 | Lachezar | catphish: No, 1 number goes to one account, but when an incoming call comes Asterisk should send a SIP message to the companion account. |
| 10:56.38 | Lachezar | WIMPy: MessageSend(â¦) |
| 10:57.08 | Lachezar | catphish: Sorry. I'm pretty new at this. I still have no idea what an extension is. |
| 10:57.25 | WIMPy | Well, just do that. However you will probably have a hard time to find someone who knows how to do it with elastix. |
| 10:57.52 | WIMPy | "Extension" is something you can dial. |
| 10:58.00 | catphish | Lachezar: just change the dialplan to add the MessageSend() command to your number |
| 10:58.04 | catphish | i have no idea how to do that though |
| 10:58.42 | WIMPy | You can hang out in #elastix and hope someone else joins in. |
| 10:58.59 | WIMPy | But it might make more sense to scrap that. |
| 10:58.59 | Lachezar | I see. |
| 10:59.04 | WIMPy | ~elastix |
| 10:59.04 | infobot | hmm... elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 10:59.23 | WIMPy | Oh, no rant? |
| 10:59.44 | catphish | lol |
| 10:59.54 | WIMPy | If you want a GUI, go for the original FreePBX. |
| 10:59.56 | catphish | don't elastix themselves have support? |
| 11:00.21 | WIMPy | Don't know. Is it still alive? |
| 11:00.34 | catphish | i have no idea, i develop my own commercial asterisk gui |
| 11:01.00 | Lachezar | OK. Asterisk (hopefully) question: I have to use WebSocket+SIP to connect. However the internal Web Server is only set to handle HTTP, no HTTPS, which is why it fails when used from a https:// site. Is there any settings that would make the WebSocket endpoint handle HTTPS? |
| 11:01.37 | WIMPy | Probably tlsenable or something similar. |
| 11:01.54 | WIMPy | would never try to use Asterisk as a web server. |
| 11:02.43 | Lachezar | WIMPy: Asterisk is not the web-server, but WebSocket is the only way to connect to it from a Browser. |
| 11:03.15 | catphish | don't browsers have built in RTP? |
| 11:03.34 | catphish | "webrtp", i don't know if that supports SIP |
| 11:03.45 | WIMPy | It's called something more specific, IIRC. |
| 11:03.47 | Lachezar | catphish: RTP is what handles AUDIO/VIDEO. WebSocket is what handles SIP |
| 11:03.54 | WIMPy | That's the one. |
| 11:04.04 | catphish | Lachezar: that makes sense then, cool |
| 11:04.33 | catphish | i assumed webrtp had its own setup protocol, or used SIP |
| 11:04.45 | catphish | but i guess using websocket makes sense too |
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| 11:05.28 | Lachezar | WIMPy: thank you! There is a tlsenable, but was in another config file. Now I can see what I have to look for. |
| 11:05.33 | guest1009 | WIMPy: you know if there is a way to check if a SIP HEADER has a value? We was talking about the PAI example earlier, I was thinking that you could check its length with LEN and then do a simple if statement |
| 11:05.38 | guest1009 | ${SIP_HEADER(P-Asserted-Identity)} |
| 11:05.38 | WIMPy | Not sure if ebsockets, the way they are, are usefull for anything. |
| 11:05.53 | guest1009 | I was wondering if there is a better way |
| 11:06.57 | WIMPy | guest1009: Not unless SIP_HEADER returns somehting that isn't documented. |
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| 13:23.13 | Lachezar | Hm. I'm feeling quite queezy. Tried copy-pasting part in sip.conf for an extension XXX trying to add a CTRL-XXX as SIP account⦠Failing miserably( |
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| 13:43.48 | Lachezar | slaps his forehead hard: transport must have ws and wss at the same time. |
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| 14:28.45 | catphish | i wish people would kindly stop trying to brute force my PBX hosts :) |
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| 14:44.16 | igcewieling | catphish: let them in, send them to a special context |
| 14:45.03 | catphish | i'd just send all their calls to my desk |
| 14:45.13 | catphish | unfortunately they're invariably not humans |
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| 14:47.14 | igcewieling | exactly |
| 14:48.39 | *** join/#asterisk GHerbstman (171f2361@gateway/web/freenode/ip.23.31.35.97) |
| 14:51.32 | GHerbstman | Hello all, I am having trouble with queues in Astersik 13.5. Both my static and dynamic phones show a status of invalid. Can anyone help me to start determining the cause? |
| 14:52.54 | [TK]D-Fender | Show us |
| 14:52.55 | [TK]D-Fender | ~pb |
| 14:52.58 | 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. |
| 14:52.59 | [TK]D-Fender | ^^^ |
| 14:55.48 | GHerbstman | What info would be helpful? show queue all members are like this: ******* (Local/1076@from-queue/n from ********) (ringinuse enabled) (Invalid) has taken no calls yet |
| 14:56.24 | GHerbstman | Both static and dynamic members are always invalid. |
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| 14:57.41 | [TK]D-Fender | That is not a "phone" |
| 14:57.48 | [TK]D-Fender | that is a queue memeber with a lcaol channel device |
| 14:58.21 | [TK]D-Fender | And is usually an issue if you are targeting a dialplan exetnsion that doesn't eexist or if chan_local.so wasn't pre-loaded before app_queue |
| 14:59.13 | igcewieling | from-queue? sounds GUIish |
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| 15:02.05 | GHerbstman | This is an up to date recent install of the FreePBX Distro 10.13.66. I have confirmed chan_local.so is loaded. The associated extension 1076 does exist and is otherwise fully functional. |
| 15:02.05 | [TK]D-Fender | Make sure your modules.conf has a "preload => chan_local.so" |
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| 15:02.15 | [TK]D-Fender | PREloaded |
| 15:02.16 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 15:02.34 | [TK]D-Fender | and take this into their channel as things get over-written in GUYI-land |
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| 15:03.45 | GHerbstman | preload = chan_local.so is the second line in the modules section of modules.conf |
| 15:04.18 | Lachezar | Why my SIP clients are receiving messages when connected via SIP/WS, but not when connected via SIP/WSS ?!? DAmN! |
| 15:05.23 | GHerbstman | thanks D-Fender, are you suggesting I run this by the FreePBX people? |
| 15:08.40 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 15:10.06 | Lachezar | Meh. Setting-up WSS in Asterisk has adverse effect. Having external SSL tunnel over the WS works better⦠|
| 15:11.16 | *** part/#asterisk lumasepa (~gestoip@193.145.124.30.local.ull.es) |
| 15:13.12 | *** join/#asterisk CyberJacob (~CyberJaco@bouncer.bluesapphiremedia.co.uk) |
| 15:13.17 | CyberJacob | Is there a way to have a queue that doesn't get answered untill an agent picks up? |
| 15:16.16 | [TK]D-Fender | GHerbstman: "module reload app_queue.so" |
| 15:16.39 | [TK]D-Fender | CyberJacob: "core show application queue" <- |
| 15:17.48 | GHerbstman | Module 'app_queue.so' reloaded successfully. All members still invalid, no error messages. |
| 15:18.23 | *** join/#asterisk onebree (d1bf0df7@gateway/web/freenode/ip.209.191.13.247) |
| 15:18.26 | [TK]D-Fender | unload |
| 15:18.27 | onebree | Hello? |
| 15:18.28 | [TK]D-Fender | then load |
| 15:18.49 | onebree | Testing. Is this working? |
| 15:19.03 | CyberJacob | onebree: yes? |
| 15:19.11 | onebree | Thank you |
| 15:19.24 | onebree | I have a question on whether something is possible with Asterisk 13 |
| 15:19.46 | onebree | Is it possible to do 1:N video conferencing with Asterisk 13? |
| 15:20.06 | [TK]D-Fender | * only does "follow the speaker" |
| 15:20.13 | [TK]D-Fender | onebree: ^ |
| 15:20.45 | onebree | That means, the speaker will be displayed to all, correct? |
| 15:20.51 | [TK]D-Fender | yes |
| 15:20.55 | onebree | Thank you. |
| 15:21.26 | onebree | What I should say is -- video conferencing with 2+ participants. |
| 15:22.19 | onebree | We are looking to incorperate WebRTC as well, through sipML5. But the tutorial provided by asterisk is audio conferencing only. We want audio + video |
| 15:22.59 | [TK]D-Fender | onebree: Which I've just answered |
| 15:23.06 | *** join/#asterisk mlhess (~mlhess@drupal.org/user/102818/view) |
| 15:24.02 | onebree | D-Fender I appreciate your answers. But how would I go about setting up video conferencing? I cannot find tutorials on it |
| 15:24.11 | onebree | [TK]D-Fender I appreciate your answers. But how would I go about setting up video conferencing? I cannot find tutorials on it |
| 15:24.36 | [TK]D-Fender | Allow the codecs on all parties. use a conferencing feature |
| 15:25.05 | [TK]D-Fender | There shouldn't be a guide for video "conferencing" since conferencing isn't the key aspect |
| 15:25.07 | [TK]D-Fender | it's just video |
| 15:25.08 | onebree | I am new to asterisk and have been assigned this project for work. Forgive my novice-ness |
| 15:25.11 | [TK]D-Fender | there is no "mixing" of it. |
| 15:27.20 | onebree | So I do not need external programs like "app conference", as recommended on voip-info ? |
| 15:28.56 | GHerbstman | load Queue, I received a couple messages: Member has an invalid ringinuse value, using [the queue] value. Second, Error parsing persistent member string for '8001' (penalty), Then: Queue members successfully reloaded from database. |
| 15:29.00 | igcewieling | onebree: the voip-info is terribly out of date. Check the official docs or the book. |
| 15:29.43 | onebree | igcewieling the book covers only up to v11. I need v13 because we plan to incorperate webrtc |
| 15:30.46 | igcewieling | onebree: best of luck. |
| 15:30.58 | klictel | Hello all. Is there a channel specific to SIP in asterisk? |
| 15:31.00 | onebree | does that mean it is not possible? |
| 15:31.45 | onebree | igcewieling: does that mean it is not possible? |
| 15:31.55 | [TK]D-Fender | onebree: Link it exactly please |
| 15:32.16 | [TK]D-Fender | klictel: No. Just Asterisk. And you're already here |
| 15:32.33 | onebree | [TK]D-Fender: Link what? The voip-info page? Or something else? |
| 15:32.40 | [TK]D-Fender | yes |
| 15:33.11 | klictel | Thanks |
| 15:33.34 | onebree | http://www.voip-info.org/wiki/view/Asterisk+video |
| 15:34.11 | onebree | I cannot find many sources detailing how to enable video conferencing for v13. Again, I am new to Asterisk, and have been assigned this task at work |
| 15:34.21 | igcewieling | onebree: https://wiki.asterisk.org/wiki/display/AST/Video+Telephony |
| 15:34.27 | [TK]D-Fender | junk |
| 15:34.33 | igcewieling | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_app_confbridge |
| 15:35.31 | igcewieling | https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support and https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 |
| 15:35.55 | igcewieling | onebree: if you keep using voip-info you'll never get this working. |
| 15:37.01 | onebree | igcewieling: I HAVE seen the sipML5 tutorial. Right now audio webrtc conferencing works, but I did not set it up. (Supervisor did). |
| 15:37.17 | igcewieling | onebree: we are not here to do your job. |
| 15:37.33 | onebree | I was just asking if you were aware of tutorial links or whatnot. |
| 15:37.46 | onebree | As I said, I am new to Asterisk |
| 15:38.01 | [TK]D-Fender | nothing to "tutorial" for this |
| 15:38.06 | [TK]D-Fender | its TWO dumb settings |
| 15:38.21 | [TK]D-Fender | (maybe one or two more if pjsip is used) |
| 15:38.33 | onebree | Well, sorry for taking up your time then |
| 15:38.38 | igcewieling | [TK]D-Fender: I think he wants to use WebRTC. |
| 15:38.52 | [TK]D-Fender | Codecs are cordecs |
| 15:38.56 | [TK]D-Fender | codecs* |
| 15:39.02 | onebree | Yes, we want to use webrtc for video conf |
| 15:39.30 | [TK]D-Fender | So go allow general videosupport and the codecs in your peers |
| 15:40.27 | onebree | So: sip.conf > [general] videosupport=yes |
| 15:40.38 | onebree | Then just adding the codecs per user? |
| 15:41.50 | [TK]D-Fender | that's what I said |
| 15:42.03 | [TK]D-Fender | the SAME codec for video. |
| 15:42.12 | [TK]D-Fender | * will NOT transcode so they all ahve to support the same |
| 15:42.19 | onebree | I am just making sure. |
| 15:43.05 | onebree | Do I need to configure in Asterisk which codec to use? Say user A supports 3 and user B supports only 1 of those. |
| 15:43.33 | [TK]D-Fender | [10:42][TK]D-Fenderthe SAME codec for video. <----------------- |
| 15:43.35 | [TK]D-Fender | ONE |
| 15:43.43 | [TK]D-Fender | All the same. |
| 15:43.45 | [TK]D-Fender | Clear? |
| 15:43.46 | igcewieling | onebree: if the devices use different codecs they can't conference. |
| 15:44.01 | klictel | onebree try to move away from ambiguity in your first attempt... just use one codec, the same |
| 15:44.24 | onebree | igcewieling: That is good to know, thank you |
| 15:44.52 | igcewieling | onebree: you have an interesting few weeks ahead. |
| 15:45.08 | onebree | I ask this because the docs list 3 available codecs. How do I know which ONE I want? |
| 15:46.05 | onebree | igcewieling: I know I do. As I said before, we are going to incorperate sipML5 client-side, and do the CRUD actions for a conference in Rails. |
| 15:46.53 | igcewieling | heh, you have an interesting few months ahead |
| 15:47.25 | onebree | igcewieling: Why did you change that to months? Anything I should be aware of? |
| 15:48.43 | [TK]D-Fender | onebreeI ask this because the docs list 3 available codecs. How do I know which ONE I want? <- the best one your bandwidth can afford |
| 15:49.21 | onebree | [TK]D-Fender: Thank you, I will talk to my supervisor about that decision then. |
| 15:52.44 | GHerbstman | D-Fender: is there anywhere I have more detailed information on the cause of the invalid status of the queue member? |
| 15:56.08 | [TK]D-Fender | I just gave you the only 2 things |
| 15:56.20 | [TK]D-Fender | remove the members and try to re-add them |
| 16:01.41 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-hfxlmiklfjmkmjlj) |
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| 16:45.54 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 16:51.55 | GHerbstman | Thanks D-Fender for your help! |
| 16:54.01 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net) |
| 16:56.24 | [TK]D-Fender | Working now? |
| 17:06.43 | stefan27 | 42 days since last segmentation fault on any of the 150 asterisk installations, it's a new record! |
| 17:08.51 | pjensen00 | 2 days for me. But it was our B2BUA's fault. :( |
| 17:08.53 | stefan27 | looks like I made about 10 jira tickets this year, maybe in 2016 I dont have to make any if everything runs this smooth |
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| 17:11.57 | karelk | I am logging cdr into mysql |
| 17:12.18 | karelk | there are lots of usefull information about call, such as clid, duration, etc |
| 17:12.37 | karelk | but I would like to log the IP address as well |
| 17:12.48 | karelk | from which the call came |
| 17:12.52 | karelk | is that possible ? |
| 17:26.54 | *** join/#asterisk samneo (~samneo@115.184.112.232) |
| 17:32.17 | igcewieling | yes |
| 17:35.53 | [TK]D-Fender | https://botbot.me/freenode/asterisk/search/?q=cdr |
| 18:08.24 | SeRi | I have an odd issue that has been haunting me for a while... I am using Asterisk 1.8.32.3 and all works well EXCEPT when I call my voicemail from any phone other than my Polycom 670 the audio is choppy. IE: My Gigaset or my WiFi SIP Phone. Any ideas? This is all internal to me (LAN) |
| 18:08.29 | SeRi | By the way all calls to the outside world work just fine from this wireless phones the issue is only calling my vmail.. |
| 18:08.32 | SeRi | I have an odd issue that has been haunting me for a while... I am using Asterisk 1.8.32.3 and all works well EXCEPT when I call my voicemail from any phone other than my Polycom 670 the audio is choppy. IE: My Gigaset or my WiFi SIP Phone. Any ideas? This is all internal to me (LAN) |
| 18:08.37 | SeRi | By the way all calls to the outside world work just fine from this wireless phones the issue is only calling my vmail.. |
| 18:09.02 | SeRi | ops |
| 18:09.04 | SeRi | sorry about that |
| 18:09.37 | SeRi | I had the sceen up and didnt see the paste :P |
| 18:10.21 | lvlinux | Have you tried upgrading? 1.8 is gone the way of the dodo... |
| 18:11.45 | lvlinux | what sort of hardware are you running * on? |
| 18:12.18 | lvlinux | If it's underpowered, and your codecs are mismatched, it could be transcoding the voicemail files and stressing the CPU? Just a possiblity, but you didn't say what you were running it on. |
| 18:12.49 | igcewieling | SeRi: you're not doing something silly like allowing more than one codec for your phones? |
| 18:12.56 | lvlinux | That probably wouldn't apply unless you are using some sort of low power embedded system though. |
| 18:13.45 | igcewieling | lvlinux: IIRC 1.8 had some issues with transcoding to/from Siren, that could cause choppyness. |
| 18:14.03 | lvlinux | yep sure could |
| 18:15.37 | igcewieling | 1.8 was EOL'd 2015-10-21 so it isn't THAT out of date. |
| 18:15.37 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:16.30 | igcewieling | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 18:16.50 | [TK]D-Fender | EOL is a year after the cutoff for "no more bugfixes" |
| 18:17.14 | [TK]D-Fender | So nothing that is "broken" has been fixed for over a year |
| 18:17.21 | [TK]D-Fender | Ditch 1.8 |
| 18:17.23 | [TK]D-Fender | No excuses |
| 18:17.36 | [TK]D-Fender | but, but, but, = BS |
| 18:42.31 | SeRi | igcewieling: I am allowing G722 and g711 |
| 18:43.59 | igcewieling | try only allowing ulaw |
| 18:44.18 | igcewieling | or, if you are outside usa/canada, alaw |
| 18:44.22 | SeRi | igcewieling: whats odd is that I can make calls just fine is just dialing my vmail from this specific phones. |
| 18:44.32 | SeRi | igcewieling: I did. Is the same issue |
| 18:44.49 | SeRi | igcewieling: I only allowed g722 yesterday for testing but previously was only g711 |
| 18:52.45 | *** join/#asterisk vinrock (~vin@unaffiliated/vinrock) |
| 18:54.13 | igcewieling | gets revenge on a customer by giving them exactly what they asked for |
| 18:54.41 | robmal | He said 'fuck me' and you did? |
| 18:56.10 | igcewieling | No, I asked them to confirm the MAC of the polycom phone they needed a change to. I asked them confirm the MAC, since the MAC they gave me did not match the extension they gave me. They confirmed it was the right MAC, so I made the change. I knew it wasn't the MAC, but I didn'tfeel like arguing with them. 10 mins later they e-mail saying oh, yes, here is the correct MAC. |
| 18:57.23 | robmal | Well, not my kind of revenge i see. |
| 19:06.08 | MaliutaLap | igcewieling: nice move |
| 19:06.27 | MaliutaLap | igcewieling: now just to burn them with fire for being idiots |
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| 20:02.14 | Ellenor | SeRi: Upgrade now, is what is being said here. |
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| 20:48.34 | Cuzner | -= 7527 extensions (32214 priorities) in 4642 contexts. =- |
| 20:48.42 | Cuzner | i think it's time to clean up the dialplan LOL |
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| 21:05.37 | SeRi | Ellenor: I understand that I need to upgrade but thats not the solution. |
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| 21:32.19 | Ellenor | SeRi: It is. :\ |
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| 21:46.28 | _-Jon-_ | Hello all! |
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| 22:39.53 | Ellenor | Hello all. |
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| 23:12.04 | wyoung | hi Ellenor, how are you? |
| 23:12.22 | Ellenor | wyoung: Not bad. I wonder what the highest posted speed limit is in your country. |
| 23:12.29 | Ellenor | Whatever country that may be. |
| 23:12.36 | Ellenor | Well, highest you've observed. |
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| 23:15.24 | wyoung | Ellenor: Why do you wonder this for? |
| 23:15.38 | Ellenor | wyoung: Curiosity. |
| 23:15.41 | rrittgarn | For a while Montana (United States) didn't have speed limits - due to too many stupid people from other states they now have a limit of 80 on some roads. Texas currently holds the highest of 85MPH in the US |
| 23:16.01 | rrittgarn | personally 70 around chicago/illinois for highest observed |
| 23:16.50 | wyoung | Ellenor: ok, well when your curiosity becomes on topic I will answer you :) |
| 23:17.48 | Ellenor | wyoung: meh |
| 23:18.03 | Ellenor | rrittgarn: If the US were to metricate the roads, the speed limit on Texas' 85 roads would go down from 85mph to 135km/h, a reduction of just over 1 mile legally travellable per hour, if they wanted to keep signs practical, or it would stay at 136.794km/h as it is now to the nearest 3 dec places. |
| 23:18.05 | Ellenor | Ow. |
| 23:18.07 | Ellenor | He left. |
| 23:31.09 | wyoung | Ellenor: so, how about that asterisl hey |
| 23:31.13 | wyoung | asterisk* |
| 23:31.26 | Ellenor | mah |
| 23:31.30 | Ellenor | I gave up on * long ago |
| 23:31.35 | Ellenor | hack on top of hack tbh |
| 23:32.26 | wyoung | Ellenor: ok, so lets move to another channel then :) what are you interested in now? |
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