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| 01:01.11 | g-maurizi | how can I place a call from a terminal? I need to write a script on an external machine that issues a command over SSH to place an outgoing call? It seems the command should be "console dial" but this doesnt work? |
| 01:02.54 | g-maurizi | 'originate' gives a command not found. |
| 01:25.10 | g-maurizi | "channel originate SIP/VoIPms/15554443333 application playback hello-world" should work but does not, any help is much appreciated. |
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| 01:40.09 | ChannelZ | define "doesn't work" |
| 01:45.17 | g-maurizi | I'm trying to place a call from my asterisk/freepbx machine to an outbound number with a specific CID, I don't care about the audio at all. |
| 01:45.24 | g-maurizi | this is an alarm system trigger. |
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| 01:46.42 | g-maurizi | I'd prefer to just simple make the call using asterisk -rx "something", I don't want to have to fudge around and create an application or dialplan or extension if possible. but it seems some of the stuff needed to do that has been removed from asterisk. |
| 01:47.09 | g-maurizi | I'm looking into placing a .call file in /var/spool/asterisk/outgoing now. |
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| 01:57.43 | TazzNZ | g-maurizi: why don't you just make a call file ? |
| 01:57.51 | TazzNZ | oh - yeah - what you said |
| 01:59.52 | g-maurizi | TazzNZ: as far as I can tell a call file first calls the extension in the 'context' and then when that extension is answered, proceeds to connect the call using "channel", I don't want an internal extension to ring nor do I want to create a dialplan or appliation with "extension XXX" that I put in the call file.. I just want to use asterisk -rx "something" to directly dial an outside number |
| 01:59.52 | g-maurizi | with a specific CID |
| 02:00.56 | g-maurizi | I'm not learned enough to create an application living at "extension" that runs from the call file when the outbound number is dialed. |
| 02:01.02 | TazzNZ | nope - you can make it "dial" the number directly |
| 02:01.08 | TazzNZ | let me dig up an example |
| 02:01.12 | g-maurizi | really? Ty so much. |
| 02:01.23 | TazzNZ | give me 10 mins thou |
| 02:01.26 | g-maurizi | no prob |
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| 02:27.33 | TazzNZ | g-maurizi: back now |
| 02:29.04 | TazzNZ | g-maurizi: shall I PM you the file ? |
| 02:29.17 | TazzNZ | I might trigger the channel flood limits |
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| 02:39.48 | g-maurizi | Yes please! |
| 02:40.06 | g-maurizi | I'm now working on a custom app, but I could really use an example. :) |
| 02:49.50 | [TK]D-Fender | g-maurizi> TazzNZ: as far as I can tell a call file first calls the extension in the 'context' and then when that extension is answered, proceeds to connect the call using "channel", I don't want an internal extension to ring nor do I want to create a dialplan or appliation with "extension XXX" that I put in the call file.. I just want to use asterisk -rx "something" to directly dial an outside number |
| 02:49.54 | [TK]D-Fender | g-maurizi, that's backwards |
| 02:50.29 | [TK]D-Fender | g-maurizi, * calls the Channel: and when it answers it gets dumped into the dialplan where you tell it to (unless you specify "Application" |
| 02:56.06 | g-maurizi | Well, I figured out a way to get it to work either way -- I have the call file set with the extension of an application that plays a wav file that says "a security event has been detected", and the channel is SIP/trunkname/externalnumber |
| 02:56.18 | g-maurizi | if there's a more elegant way to do this I'm very open to the help! |
| 02:57.26 | [TK]D-Fender | how much more elegant do you need? |
| 02:57.32 | [TK]D-Fender | Single command that does just that..... |
| 02:58.28 | g-maurizi | Sorry I am confused as that's not a single command, that's a file I have to copy to a directory that gets executed within a timespan like a cron-job.. did you mean that * is itself an asterisk cli command? |
| 02:59.41 | [TK]D-Fender | there are like 4 different ways to do this |
| 02:59.43 | [TK]D-Fender | call file |
| 02:59.56 | [TK]D-Fender | AMI / CLI originate |
| 03:00.33 | [TK]D-Fender | The whatever that AJAM method was |
| 03:00.45 | [TK]D-Fender | (which is still kinda AMI) |
| 03:02.53 | g-maurizi | Ty! it sounds like cli originate could work but is pretty involved to get the correct syntax and understand it's operation. I may go that route. I think this isn't a bad route as I can further program the extension/custom app to require keypress confirmation from the called party. :) |
| 03:04.05 | [TK]D-Fender | involved to get the syntax? |
| 03:04.07 | [TK]D-Fender | * TELLS you |
| 03:04.22 | [TK]D-Fender | "core show help channel originate" |
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| 03:05.18 | g-maurizi | Originate is meant for a channel in progress based on core show help channel originate. |
| 03:06.04 | [TK]D-Fender | no |
| 03:06.12 | Katty | Fender bender |
| 03:07.23 | [TK]D-Fender | Katty, Mew. |
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| 03:24.06 | phix | Afternoon! |
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| 04:28.48 | Drumitar | followed this tutorail to setup corosync https://wiki.asterisk.org/wiki/display/AST/Corosync, i notice though when in the asterisk commandline if i do module show like, there is no res_corosync.so module , any help would be great ! |
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| 05:03.37 | fling | I'm about to install asterisk into an lxc container. Unfortunately it will be running behind the nat in this case (for some sip peers). Are there much cons for this design? |
| 05:03.50 | fling | I'm also planning to use usb dongles⦠|
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| 10:14.08 | jkroon | WIMPy, btw, have you managed to figure out the recording format for voicemail files in odbc? |
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| 13:04.25 | WIMPy | jkroon: I wouldn't dare to add additional databases. |
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| 13:05.57 | jkroon | haha, no, what i mean is i want to download the prompts and convert them to mp3, but there does not seem to be any indication as to the recorded format ... |
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| 13:40.31 | kannan | hello, i request any recommendation for a reliable T1 PRI provider in Detroit, MI, USA area |
| 13:44.35 | maddawg2 | hey all, kinda new to the whole asterisk thing.. and kinda new to setting up my own PBX from scratch. I'm working at a company that currently uses an ancient AVAYA system with a PRI for service. We recently moved from using T1 lines for data to using Comcast Cable. Our business needs didn't warrant the expense of a T1 line for data |
| 13:44.46 | maddawg2 | the only thing it provided us was an SLA that we can do without |
| 13:45.10 | maddawg2 | so we've upgraded to a 150mbps down and 20mbps up from the GIANT comcast corporation |
| 13:45.31 | maddawg2 | and i am hoping to set up an asterisk box that would replace our current phone system and want to use VoIP |
| 13:45.45 | maddawg2 | however I am seeing so much information regarding SIP trunking |
| 13:45.48 | maddawg2 | and VoIP |
| 13:45.55 | maddawg2 | and kinda confused |
| 13:46.09 | maddawg2 | because I've found carriers that sell VoIP service as well as SIP Trunkings |
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| 13:46.19 | maddawg2 | and Im a bit puzzled as to the difference |
| 13:46.36 | maddawg2 | since i thought SIP was a protocol while VoIP was really just a method of connecting |
| 13:46.46 | [TK]D-Fender | kannan: Nothing about PRI is supposed to be "unreliable" |
| 13:46.52 | [TK]D-Fender | kannan: It's TDM.... |
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| 13:47.43 | kannan | i meant a company thats well reputed and not going to go out of business one day soon, type of reliable , [TK]D-fender |
| 13:47.46 | [TK]D-Fender | maddawg2because I've found carriers that sell VoIP service as well as SIP Trunkings <- SIP is a VoIP PROTOCOL. |
| 13:47.55 | [TK]D-Fender | maddawg2: So taht is redundent |
| 13:48.38 | maddawg2 | i know |
| 13:48.41 | maddawg2 | hence the confusion |
| 13:49.18 | kannan | any one has experience with Clear Rate Communications for T1 PRIs? |
| 13:50.20 | [TK]D-Fender | VoIP is a concept. SIP is an implementation. |
| 13:50.43 | maddawg2 | so basically what I am looking for a is a SIP provider |
| 13:50.57 | [TK]D-Fender | That's the usualy protocol of choice |
| 13:51.00 | [TK]D-Fender | ~itsp |
| 13:51.07 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 13:51.08 | [TK]D-Fender | ^^^ |
| 13:51.11 | maddawg2 | i was just confused why they have VoIP services and a seperate thing for SIP services |
| 13:51.20 | maddawg2 | I think maybe VoIP must be a hosted thing |
| 13:51.35 | [TK]D-Fender | I'd probably re-read whatever you saw there again... |
| 13:52.09 | maddawg2 | ~itsplist-us |
| 13:52.09 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.broadvoice.com, http://www.jnctn.com, http://www.sipstation.com, http://vitelity.net, http://voip.ms and http://flowroute.com |
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| 13:55.31 | maddawg2 | i was looking at SIP.US as well seems pretty cool... tho trying to understand the difference between SIP Trunks and SIP Channels |
| 13:55.40 | maddawg2 | most sites i've seen just sell "Trunks" |
| 13:55.58 | file | nothing is different, except billing probably |
| 13:56.32 | file | some providers sell channels where it's actually X number of simultaneous channels, unlimited calling (to an area) for each |
| 13:56.40 | [TK]D-Fender | The terms are used largely interchangeably |
| 13:56.49 | [TK]D-Fender | Look at what the service actually includes. |
| 13:56.56 | file | yeah, marketing marketing marketing |
| 14:06.34 | jzu_ | hmm, seems like OnePipe forwards all calls, hence all incoming calls show as forwarded |
| 14:06.41 | jzu_ | wonder if there's anything I could do about it |
| 14:09.17 | maddawg2 | also one oter question.... if a DiD calls to an outside phone it always shows as coming from the main number |
| 14:09.28 | maddawg2 | what would i need to do to get the DiD to pass through on caller ID |
| 14:09.36 | maddawg2 | is that a feature my SIP provider needs to do ? |
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| 14:11.01 | avb | thats weird asterisk MuteAudio seems not working :-/ |
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| 15:45.06 | jzu_ | [2015-04-13 17:24:53] NOTICE[16131][C-0000015a] channel.c: Dropping incompatible voice frame on SIP/didww.sip.1pipe.com-0000016e of format g722 since our native format has changed to (ulaw) |
| 15:45.10 | jzu_ | hmm |
| 15:46.17 | jzu_ | mehh, seems like I have way too much trouble getting OnePipe to work |
| 15:48.25 | [TK]D-Fender | or perhaps you should just settle on ulaw |
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| 15:51.26 | jzu_ | hehe, yeah :) |
| 15:51.40 | jzu_ | wonder why OnePipe sends CallerID with two ++'s: "CallerIDNum: ++358407007600" |
| 15:51.54 | jzu_ | and every call they deliver is forwarded |
| 15:55.24 | [TK]D-Fender | Go strip them |
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| 16:29.40 | aruntomar | when using asterisk 11.0+dahdi+pri, we used to receive messages like âplease check the number youâve dialedâ. After upgrading to asterisk 13, we get only ringtone. hence, it becomes very difficult to identify whether itâs a wrong number or that someone is not picking up the call. has anyone else faced this issue. just fyi, the config files are the same. |
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| 16:32.36 | RPerre | Hi. How can I add SIP URI vars on/before Dial command on dialplan? thanks |
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| 16:35.58 | Xadof | aruntomar: I'm unaware of what config changes may have been made to cause this or if there's an existing bug open for that, since I am still on Asterisk 11, however make sure that when upgrading between major versions (11 to 12, 12 to 13, 11 to 13) that you read the UPGRADE-X.txt files in the source root (I think they're online somewhere). They should list all changes that are not backwards |
| 16:35.58 | Xadof | compatible. So, you would need to look at those files for changes to your channel driver's (DAHDI's) config file regarding early media. It would probably say that some setting you use is depreciated in favor of another setting, that the default setting has changed (ie default yes becoming default no, and you don't have it set) or that the required value has changed |
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| 16:39.40 | aruntomar | Xadof: before upgrading, i had looked at the release and upgrade txt files. we were quiet happy with 11.x series. but then we wanted to use ARI, hence, had to upgrade it. |
| 16:41.36 | Xadof | aruntomar: I've no knowledge of Dahdi, I was just referencing those files in case you were unaware of them. But just to make sure you know, you would have to read both UPGRADE-12.txt and UPGRADE-13.txt, one for each bump up. They aren't redundant |
| 16:41.50 | aruntomar | Also, ther is another weird thing we noticed. when i call a phone. and the endpoint, disconnects/rejects the call, ideally, we should get an immediate hangup event on the server as well as caller. but it keeps ringing for another 15-20 sec and then gives a 408 status. |
| 16:43.56 | Xadof | aruntomar: the ringing for 15-20 seconds is going to happen while the called end is playing back early media to you. The termination will only hit you after early media playback is complete. Chances are that you're receiving early media and that asterisk is converting it to ringing, for example if you use the "r" option in Dial |
| 16:45.26 | aruntomar | Xadof: we donât use ârâ option in dial. |
| 16:45.28 | Xadof | aruntomar: just to be clear, when I say that the termination will only hit after early media playback is complete, I mean that is the general functionality of the endpoint that is sending you early media, not that asterisk is doing that. The only problem seeems to be that the early media is not being played to your local endpoints (ie agents, customers), and so they can not determine that |
| 16:45.28 | Xadof | the call-in-progress has already failed from the message you are used to hearing |
| 16:45.59 | Xadof | aruntomar: I said for example. Because I already referenced that I don't use/know dahdi, and that I don't use/know asterisk 12 or 13. |
| 16:46.29 | Xadof | aruntomar: note that I'm not an op in this channel. I'm also not an asterisk dev or contributor. I'm only trying to assist as best as I can. |
| 16:46.35 | aruntomar | Xadof: thatâs ok, i was just clarifying. i appreciate the help and brainstorming. |
| 16:47.16 | aruntomar | Xadof: sometimes a different perspective is required, when we are too deep inside our own setup, itâs difficult to see things differently. |
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| 16:49.07 | aruntomar | i think, what i need to do is pass the telco messages directly to the endpoint. |
| 16:49.57 | aruntomar | and from the links on the internet, i read that setting âpriindication=outofbandâ in chan_dahdi should do that. but it doesnât work. |
| 16:56.33 | Xadof | aruntomar: from briefly reading about it, it seems to me that it's not referencing whether or not to pass the indications between the remote and local endpoints, but rather whether to do so out of band or inband. Although I don't know the channel tech, my initial thoughts are that you would want it to be in band, so that your local endpoints can hear the audio, but it may have nothing to |
| 16:56.33 | Xadof | do with audio. My asterisk 11 chan_dahdi.conf.sample says that that setting can't be changed without a full core restart, but I see a resolved bug report suggesting that that may be fixed (or maybe they just made it persist, since the bug report was that the setting was being dropped upon reload). So if you haven't restarted since making changes to that setting, it may be worth a try if |
| 16:56.34 | Xadof | you've no channels active. |
| 16:57.03 | Xadof | aruntomar: I'm going away until this evening, but good luck |
| 16:57.47 | aruntomar | Xadof: thanks for the help. |
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| 17:24.45 | twitchnln | Morning, anyone ever seen delayed voicemails? I have a box that intermittently is not having VMs show up in VMB when called until days later. |
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| 17:41.04 | TazzNZ | twitchnln: Asterisk version ? |
| 17:41.22 | TazzNZ | also some box stats would be great, along with more details on your setup |
| 17:41.24 | TazzNZ | like load etc |
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| 17:46.33 | pankid | Can I setup writing my cdr to a network mysql along with the local sql without setting up odbc? |
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| 18:00.49 | maddawg2 | question: if i plan to use my asterisk server over the internet without a T1 card installed do I need DAHDI and LibPRI? |
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| 18:01.16 | [TK]D-Fender | MeetMe still requires it. |
| 18:01.24 | [TK]D-Fender | That's about all as of current versions of * |
| 18:01.38 | [TK]D-Fender | Then again ConfBridge as evolved past it at this point. |
| 18:02.28 | maddawg2 | was that directed to me? |
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| 18:09.32 | [TK]D-Fender | yes |
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| 18:11.14 | maddawg2 | so is ConfBridge a replacement for MeetMe then? |
| 18:12.00 | file | yes |
| 18:12.32 | sweets_ | i have an * 11 application that is mainly an IVR, which will eventually transfer the caller via DIAL() to a final destination....i would like to keep track of why the call terminated inside the IVR (Before the DIAL() is issued)...i've tried using HANGUPCAUSE() but it doesn't seem to always work |
| 18:12.56 | sweets_ | any thoughts? is the h extension the best approach? or using the new hangup handlers interface? |
| 18:13.17 | [TK]D-Fender | sweets_: if you are going to hit a Dial ... the call hasn't terminated. |
| 18:13.33 | [TK]D-Fender | sweets_: Or are you simply referring to things that aren't going to happen? |
| 18:13.34 | sweets_ | [TK]D-Fender: but the call has been answered inside the IVR |
| 18:13.48 | sweets_ | it just hasn't been bridged to an external user yet |
| 18:14.03 | [TK]D-Fender | h works fine. Typically you run IVR's in their own context so that works.... |
| 18:14.29 | [TK]D-Fender | Hangup handlers also work... but not sure if you can CLEAR one if it DOES continue past a certain point |
| 18:14.39 | [TK]D-Fender | "h" sounds to be the safest |
| 18:15.16 | sweets_ | thanks, i'll stick with the 'h' approach |
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| 19:17.31 | eppigy | Hello |
| 19:17.33 | eppigy | I am Dave |
| 19:18.03 | tuxd00d | Hello Dave. |
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| 19:20.33 | robmal | Affirmative Dave, I read you. |
| 19:23.55 | eppigy | Roger that |
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| 19:47.08 | djgerm | Hello! I am trying to recompile asterisk from the ubuntu source (to get ilbc support), and I am getting an issue where pjproject library isn't found. Unknown value '@PBX_PJPROJECT@' found in build_tools/menuselect-deps for PJPROJECT |
| 19:47.08 | djgerm | menuselect/menuselect --check-deps menuselect.makeopts |
| 19:47.09 | djgerm | Now, I am under the impression that pjproject is already included in the source, so I am a bit confused. Does anybody have any ideas of where to go from here? |
| 19:48.07 | [TK]D-Fender | it isn't |
| 19:48.11 | [TK]D-Fender | the library is separate |
| 19:49.18 | djgerm | hmm on here it says otherwise https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject |
| 19:49.41 | djgerm | but I trust you :) so I have installed the dependency from apt |
| 19:50.24 | [TK]D-Fender | Funny that page show rather specific places to download it from... |
| 19:50.33 | [TK]D-Fender | Wouldn't need to do that if it were included, would you? |
| 19:50.48 | djgerm | I meant "Asterisk 11 uses an embedded pjproject for the ICE, STUN and TURN libraries in its RTP engine for WebSockets support. Therefore you do not need to follow the instructions here for Asterisk 11." |
| 19:51.28 | [TK]D-Fender | And * 11 doesn't support chan_pjsip |
| 19:51.33 | [TK]D-Fender | that came in 12 |
| 19:52.09 | djgerm | hmm, i think i need pjproject for res-rtp-asterisk |
| 19:52.11 | djgerm | I don |
| 19:52.17 | djgerm | t think I need chan_pjsip. |
| 19:52.41 | [TK]D-Fender | The libs are still required because of how they split up chan_sip for RTP, etc |
| 19:53.07 | [TK]D-Fender | Don't know the best terminology for the fine print but it's a dependency for parts of the code even if you're not using the channel driver |
| 19:53.25 | djgerm | ah. so if I just specify /usr/lib/libpj-x86_64-pc-linux-gnu.a as the lib, it should⦠work |
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| 19:57.46 | file | no |
| 19:57.49 | file | you can not use .a |
| 19:58.04 | djgerm | oh |
| 19:58.05 | file | stuff will not work in weird ways without shared libraries |
| 19:59.19 | djgerm | oh uhâ¦. any suggestions? |
| 19:59.46 | djgerm | <PROTECTED> |
| 19:59.49 | djgerm | ? |
| 20:01.14 | maddawg2 | ok a bit confused when exactly do I need DAHDI... it says if i'm not intergrating any traditional telephony equipment |
| 20:01.28 | maddawg2 | I'm guessing that me using IP based phones (Linksys SPA942) |
| 20:01.32 | maddawg2 | doesnt qualify |
| 20:02.47 | malcolmd | DAHDI's required for app_meetme. otherwise, you won't need it. app_confbridge is a capable replacement for app_meetme that doesn't depend on DAHDI |
| 20:03.39 | maddawg2 | ok good to know |
| 20:03.52 | maddawg2 | also is there a GuI recommended for use with asterisk? |
| 20:03.56 | maddawg2 | Asterisk 13 that is |
| 20:04.19 | robmal | malcolmd: So if i'm moving all my conf stuff to confbridge i can dump dahdi? It's not needed anywhere else? |
| 20:04.21 | [TK]D-Fender | Not sure if any are "recommended". BUt if you want one, FreePBX is the best free one around |
| 20:04.27 | [TK]D-Fender | maddawg2: ^ |
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| 20:04.53 | [TK]D-Fender | maddawg2: Runner-ups are largely commercial |
| 20:05.01 | malcolmd | robmal: correct. modern versions of asterisk can provide their own timing, independent of dahdi. DAHDI is a prerequisite for app_meetme though. |
| 20:05.12 | maddawg2 | ok i did see freepbx that is a sperate Liux distro that has the tools built in correct? |
| 20:05.15 | robmal | Great news! |
| 20:05.30 | [TK]D-Fender | malcolmd: Was app_page switched off of MeetMe as a core? |
| 20:05.37 | malcolmd | [TK]D-Fender: yes |
| 20:05.46 | [TK]D-Fender | \o/ |
| 20:05.59 | malcolmd | iirc the switch for app_page happened in 12. |
| 20:06.02 | [TK]D-Fender | So basically.... MeetMe can just DIAF alone now :) |
| 20:06.27 | [TK]D-Fender | maddawg2: bundles the OS, Asterisk, FreePB, and a few extra goodies |
| 20:06.49 | malcolmd | pretty much. the old SLAStation and SLATrunk dial plan apps use app_meetme, but i don't think anyone uses those. |
| 20:07.06 | [TK]D-Fender | malcolmd: We can use those as KINDLING ;) |
| 20:07.21 | malcolmd | indeed :D |
| 20:07.25 | maddawg2 | bleh |
| 20:07.36 | maddawg2 | i was hoping i could just throw it on my existing setup lol |
| 20:08.16 | [TK]D-Fender | maddawg2: Pretty much nothing lets you do that. No GUI should be expected to work around your manual "everything" |
| 20:08.30 | [TK]D-Fender | maddawg2: Or were you refeering to your OS? |
| 20:08.59 | maddawg2 | my OS |
| 20:09.12 | [TK]D-Fender | maddawg2: FreePBX is a simple tarball to to that... |
| 20:09.23 | [TK]D-Fender | maddawg2: they ALSO make an all-in-one Distro |
| 20:09.38 | [TK]D-Fender | maddawg2: You'll still basically trash your * setup though |
| 20:10.19 | maddawg2 | ok well i dont have a set up so that's ok |
| 20:10.24 | maddawg2 | i havent even compiled it yet |
| 20:10.37 | maddawg2 | i just already intalled ubuntu server 14.04LTS |
| 20:10.51 | [TK]D-Fender | FreePBX has guides for it |
| 20:20.54 | djgerm | dang. specifying `make PJPROJECT_LIB=/usr/lib/libpj.so` didn't give me any errors, but also left me with an asterisk that won't start. |
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| 20:28.50 | ectospasm | I have an issue with VoiceMail(), voicemail.conf, and Asterisk 13.1-cert1. Even though several mailboxes are set with delete=yes, the emails are not deleted, and the system default of 200 messages is reached. |
| 20:29.50 | ectospasm | To my knowledge to voicemail to email feature is working |
| 20:30.40 | [TK]D-Fender | heads home... |
| 20:38.18 | ectospasm | Yeah, voicemail to email is working fine. |
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| 21:18.45 | jodavis21 | Hello, I previously had Asterisk working with WebRTC. However, about a month ago the audio stopped working without throwing any errors. I was advised to update to the current Asterisk, so I have done that (11.17.1). I now see the error "dtls_srtp_setup: Could not set policies when setting up DTLS-SRTP on '0x7f0e34023110' RTP Read error: Unspecified. Hanging Up." I recreated the certs but no go. Any suggestions? Thanks. |
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| 21:25.56 | basicer | With ExtenSpy() is there a way to whisper into the other side of the bridge? |
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| 23:37.58 | djgerm | so⦠I had asterisk 11.7 running very smoothly and efficiently able to place thousands of near concurrent calls based on the stock apt-get install asterisk on ubuntu. However I needed to get ilbc support, and after some wrangling, I was able to compile 11.17.1 from source. Now, when I place a lot of calls (not even 500!) I am getting load upwards of 290, and calls start failing left and right. |
| 23:38.45 | djgerm | Does anybody have any ideas as to what to tune? It was quite efficient "out of the box" when installed from package after I just up'd my file limits. |
| 23:40.30 | ctpdump | have you checked the codecs before and after? |
| 23:40.38 | ctpdump | maybe you're using some codecs now that are more cpu hungry |
| 23:40.57 | ctpdump | also, the load of 290 can be related to i/o wait, not necessarily high cpu usage |
| 23:41.02 | djgerm | i am using only ilbc. |
| 23:41.17 | ctpdump | ah, and before what codecs were you using? |
| 23:41.25 | djgerm | alaw/ulaw |
| 23:41.33 | ctpdump | have a look to see how much cpu ilbc requires |
| 23:42.03 | ctpdump | a/ulaw are reasonably uncompressed if I recall correctly so you weren't using much cpu prior |
| 23:42.12 | djgerm | core show translation? |
| 23:42.47 | ctpdump | from: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-UnderstandingVoIP-SECT-3.html |
| 23:42.50 | ctpdump | G.711 imposes minimal (almost zero) load on the CPU. |
| 23:42.57 | ctpdump | Because iLBC uses complex algorithms to achieve its high levels of compression, it has a fairly high CPU cost in Asterisk. |
| 23:43.44 | ctpdump | I haven't used ilbc so I can't advise either way |
| 23:43.49 | djgerm | hmmm |
| 23:44.03 | djgerm | fascinating! thanks! |
| 23:44.05 | ctpdump | but I would think you may now have enough cpu power |
| 23:44.20 | ctpdump | have a dig around and see if you find any calculators |
| 23:44.31 | ctpdump | like x channels require x amount of cpu |
| 23:44.41 | djgerm | k |
| 23:44.57 | djgerm | yeah⦠being CPU bound nowadaysâ¦. well that's rare for me! |
| 23:45.13 | ctpdump | I said the same thing initially :) |
| 23:45.23 | ctpdump | cpus are so underused these days |
| 23:45.51 | ctpdump | I have to leave, good luck |
| 23:45.58 | djgerm | thanks!! |
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| 23:51.18 | basicer | With ExtenSpy() is there a way to whisper into the other side of the bridge? |
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