| 19:45.08 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 19:45.08 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.7.0 (2012/07/30), 1.8.15.0 (2012/07/30), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 19:45.42 | becca_r | Hello |
| 19:45.57 | Hive | <PROTECTED> |
| 19:45.58 | jeffspeff | Hi |
| 19:46.14 | Hive | i'll try with en/queue-perio.... again for safe measure |
| 19:46.40 | becca_r | Does anyone know if there is a variable or an easy way to reference the number of calls holding in a queue? For example, I'd like to play a message if calls holding is greater than x callers. |
| 19:47.02 | jeffspeff | Hive, i know for sure that when using Background() or Playback() you don't have to specify the en dir |
| 19:47.11 | Hive | becca_r: try QUEUE_WAITING_COUNT |
| 19:47.28 | becca_r | Hive: Thank you. |
| 19:47.31 | Hive | jeffspeff: yeah, i'm really confused as to why this isnt working :[ |
| 19:48.40 | b0ot | Does asterisk support any sort of decentralized distributed configuration? |
| 19:48.57 | jeffspeff | Hive, are you using a SIP provider/trunk? |
| 19:50.00 | jeffspeff | also, are you using SIP phones? |
| 19:50.15 | Hive | yes at both |
| 19:50.29 | Hive | (answeing jeffspeff) |
| 19:51.29 | jeffspeff | what codec are you setting for your sip phones, and what codec are you setting for your provider? |
| 19:54.08 | Hive | inbound calls are ular |
| 19:54.20 | Hive | ulaw, peers are ular or gsm |
| 19:54.41 | Hive | though i dont know if that really should affect what the caller hears |
| 19:54.49 | Hive | err ulaw not ular |
| 19:55.03 | jeffspeff | most phones are set for ulaw |
| 19:55.09 | jeffspeff | i would add/change |
| 19:55.13 | jeffspeff | deny=all |
| 19:55.15 | jeffspeff | allow=ulaw |
| 19:55.20 | jeffspeff | for all peers and providers |
| 19:55.35 | Hive | they have disallow=all |
| 19:56.12 | jeffspeff | what order is your allow statement? |
| 19:56.18 | gg608f | hi all, anyone have installed LUA module in asterisk? |
| 19:56.22 | gg608f | I have this error |
| 19:56.23 | gg608f | http://pastebin.com/wNAspAze |
| 19:56.40 | gg608f | usr/bin/ld: cannot find -llua |
| 19:57.22 | Hive | disallow -> allow ulaw for provider |
| 19:57.32 | Hive | disallow -> allow ulaw -> allow gsm for peers |
| 19:58.52 | jeffspeff | Hive, you're throwing me off with the -> |
| 19:59.01 | jeffspeff | please tell me you don't have -> in your sip.conf |
| 19:59.09 | Hive | sorry |
| 19:59.11 | Hive | they are new lines |
| 19:59.19 | Hive | disallow=all |
| 19:59.22 | Hive | allow=ulaw |
| 19:59.24 | Hive | allow=gsm |
| 20:00.22 | jeffspeff | and there's your problem |
| 20:00.40 | jeffspeff | just a sec, and i'll tell ya |
| 20:00.43 | jeffspeff | on the phone |
| 20:01.07 | *** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12) |
| 20:06.30 | *** join/#asterisk wonderworld (~ww@dsdf-4db5c562.pool.mediaWays.net) |
| 20:06.56 | jeffspeff | Hive, allow=ulaw,gsm |
| 20:08.11 | *** join/#asterisk doolittlework (~doolittle@41-134-22-14.dsl.mweb.co.za) |
| 20:08.17 | jeffspeff | Hive, i take that back |
| 20:08.24 | jeffspeff | I think i just told you wrong |
| 20:08.27 | doolittlework | CAN SOME ON PLEASE TEL ME WHAT Requested transfer capability: 0x10 - 3K1AUDIO |
| 20:08.38 | doolittlework | sorrry for yelling |
| 20:08.50 | doolittlework | ment to be the other way around |
| 20:09.20 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 20:09.27 | Hive | aww bummer :[ |
| 20:09.29 | doolittlework | i am routing a incomming call from provider to a pabx using te4xxp and i get the above |
| 20:09.46 | jeffspeff | Hive, just for giggles, try doing allow=all |
| 20:09.55 | jeffspeff | that would rule out a codec problem |
| 20:12.12 | _Corey_ | anyone seen a Cisco SIP phone dropping calls with a 400 error whenever it gets an RPID update from Asterisk? |
| 20:12.38 | jeffspeff | _Corey_, mine never did |
| 20:13.01 | _Corey_ | jeffspeff: Any idea what firmware rev you had? I have a feeling this may just be too old |
| 20:13.24 | doolittlework | strange this that if i phone in on the pri from my mobile phone it works like a charm call goes throught witout any hickups, but when i phone from a land line i just get a dead silence with the above error |
| 20:13.35 | _Corey_ | jeffspeff: (I think it's running 8.8) |
| 20:13.40 | jeffspeff | oh, it's been over a year... they were SPA504g's and had the latest firmware that wasn't part of the unified communications framework |
| 20:13.52 | _Corey_ | yeah, those are Sipura firmware |
| 20:13.56 | _Corey_ | I should have clarified |
| 20:14.01 | jeffspeff | oh |
| 20:14.06 | _Corey_ | I'm talking 79XX series Cisco SIP FW |
| 20:14.10 | jeffspeff | oh |
| 20:14.13 | jeffspeff | :S |
| 20:14.19 | _Corey_ | yeah, vintage junk |
| 20:14.21 | jeffspeff | those use the UC firmwares don't they |
| 20:14.30 | jeffspeff | ?? maybe not |
| 20:14.37 | jeffspeff | it's been forever |
| 20:14.40 | Hive | jeffspeff, you think i should allow all on the provider? |
| 20:14.52 | jeffspeff | i've since moved on to a pure polycom solution |
| 20:14.59 | jeffspeff | Hive, it's not a security problem or anything |
| 20:15.09 | jeffspeff | it's just codec filtering |
| 20:15.45 | jeffspeff | this way * will use the first compatible codec sent from the peer/provider |
| 20:16.34 | jeffspeff | Hive, also, are you getting bi-directional audio? like are the calls working fine, just not able to hear the file play? |
| 20:18.16 | sm7syx | Anyone here with experience from Phonzo as provider? |
| 20:18.24 | drmessano | Phonzo? |
| 20:18.39 | drmessano | That's a great name. Damnit |
| 20:18.45 | sm7syx | yes a Norwegian / Swedish one |
| 20:18.47 | _Corey_ | too much like Bozo |
| 20:19.00 | sm7syx | agreed ;) |
| 20:19.04 | drmessano | I guess I will need to go with my second choice.. Gonzoip |
| 20:19.22 | sm7syx | Hahahah love it |
| 20:20.10 | sm7syx | Just that I do have a problem calling out ;( |
| 20:20.35 | drmessano | GonzoIP.. you're crazy wacky looney ITSP |
| 20:20.36 | jeffspeff | lol |
| 20:20.38 | drmessano | your* |
| 20:20.42 | jeffspeff | love it |
| 20:21.53 | drmessano | GonzoIP.. we're serious about VoIP.. Seriously LOONEY |
| 20:23.15 | sm7syx | KermIT stop it ;) |
| 20:23.22 | *** join/#asterisk Sean-Der (~sean@38.117.156.114) |
| 20:23.32 | drmessano | "We support GSM, G729, G722, ULAW, ALAW, SILK, and Siren7/Siren14... But only one per day, per week.. and it's completely random" |
| 20:23.33 | doolittlework | what does the SetTransferCapability('SPEECH') do? |
| 20:24.21 | Hive | jeffspeff: Yes, the call works fine, it's as if the file isnt even trying to play though... no errors or warnings or anything. Nothing in the CLI that says playing file either T_T |
| 20:24.57 | jeffspeff | that's odd, but i have neve messed with queues at all |
| 20:25.07 | drmessano | Typical support message: "I can not make calls with G729 today. It worked yesterday. What happened?" "It's not Tuesday anymore" |
| 20:25.28 | jeffspeff | lmao |
| 20:25.41 | jeffspeff | drmessano, where do you get that message from? |
| 20:26.12 | drmessano | I made it up. I expect it will be a common problem with GonzoIP's codec rotation |
| 20:27.08 | drmessano | Maybe update via twitter with the codec of the day? "Happy Wednesday! We're supporting GSM today. Don't miss out!" |
| 20:27.19 | doolittlework | drmessano: can you please nudge me in the right direction error: Requested transfer capability: 0x10 - 3K1AUDIO |
| 20:28.44 | drmessano | doolittlework: One error? No description or debug? Allow me to nudge you with a ClueBat(TM) and ask for more info |
| 20:29.28 | *** join/#asterisk danfromuk (~IceChat77@2.27.27.15) |
| 20:30.11 | doolittlework | i am making a call from my mobile phone into a pri to my sip phone and it works, from a landline i get the above error |
| 20:31.06 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 20:31.17 | *** join/#asterisk twanny796 (~twanny@85.232.216.83) |
| 20:31.17 | WIMPy | doolittlework: It means it is voice and not some other form of audio, like a modem. |
| 20:31.20 | danfromuk | Hi guys. I know this sounds like a rookie question but here goes.... If I want to set a channel variable before a Dial(LOCAL/ command, what do I need to do for the channel variable to be passed through to the LOCAL/ extension? |
| 20:31.30 | danfromuk | Is it one _ or two __? |
| 20:32.27 | WIMPy | doolittlework: There must be something else going on. Asterisk doesn't care for BC on imcomming calls. |
| 20:34.35 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 20:34.54 | Hive | I'm having trouble getting a periodic announcement to play for my CALLER. Is there something wrong with the queue formatting or the way I call the queue? http://pastebin.com/bMq8Twbb |
| 20:36.51 | *** join/#asterisk iulhk (iulhk@119.154.71.121) |
| 20:37.58 | mjordan | Hive: you shouldn't need the 'en/'. You should only need the name of the sound file. |
| 20:38.10 | mjordan | (at least from the documentation in queues.conf) |
| 20:39.45 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 20:42.53 | iulhk | can i use perl agi and php agi at the same time at the same server? as well can i call php agi from perl agi? as normally we call from one agi to other agi |
| 20:43.37 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 20:45.58 | *** join/#asterisk Ralcala (~demo@201.127.8.248) |
| 20:48.12 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 20:50.47 | Hive | mjordan: oops, i forgot to take that out. I have tested it without the en/ as well without luck |
| 20:50.56 | Hive | but I'm guessing you don't see anything wrong with the formatting either |
| 20:53.04 | Ralcala | hi dudes |
| 20:53.26 | Ralcala | i got a issue with * 1.18 on centos 5.4 |
| 20:54.24 | mjordan | Hive: do you see the verbose message "Playing periodic announcement" |
| 20:54.31 | Ralcala | 1 old micronet sp5050 fxo gw and 3 grand stream fxs gw |
| 20:54.59 | Ralcala | the next code was for ivr |
| 20:55.02 | Ralcala | http://pastebin.com/mmEvzVZS |
| 20:55.19 | Ralcala | fxo gw got 6 lines |
| 20:56.24 | Ralcala | when ivr receives a second call and one operator is busy with a current call |
| 20:56.34 | Hive | mjordan: no, i dont see anything like that |
| 20:56.55 | mjordan | kk. When the periodic announcement is played, you should get a verbose level 3 message indicating that its playing the periodic announcement |
| 20:57.08 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 20:57.19 | Ralcala | the call just interrupts for 20 seconds and then continues..... |
| 20:57.31 | danfromuk | Why would the CLI say that asterisk was recording but no file created? |
| 20:57.31 | Ralcala | any suggestions |
| 20:57.46 | mjordan | try setting periodic-announce-frequency=60 |
| 20:57.52 | mjordan | no spaces between the = and the number |
| 21:00.04 | *** join/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210) |
| 21:01.28 | Hive | mjordan: i changed frequency to 60 and timeout to 120, listened for the full 2 minutes and nothing played |
| 21:01.32 | Hive | no hit on the cli either |
| 21:01.35 | mjordan | hm |
| 21:01.47 | Hive | also it's not playing my musicclass either :[ |
| 21:01.49 | mjordan | what type of channel is in the queue? |
| 21:01.52 | Hive | so confused |
| 21:01.52 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 21:02.05 | mjordan | hm. Well, that's probably the root cause of the problem, not app_queue. It sounds like it can't find your sound files. |
| 21:02.45 | danfromuk | Actually, looking deeper, it seems that during the recording, the wav file is created, then when i hang up, its deleted. |
| 21:02.51 | danfromuk | What would cause that to happen? |
| 21:02.54 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.37) |
| 21:03.12 | Hive | mjordan: sound files are located at /var/lib/asterisk/sounds/en/queue-periodic-announce.gsm (.g729 and .ulaw are also present) |
| 21:03.45 | Hive | which is the default i think, so i dont know why it wouldnt be able to find those |
| 21:05.56 | mjordan | Hive: what is your astdatadir? |
| 21:06.10 | mjordan | in asterisk.conf |
| 21:06.45 | mjordan | Hive: and what version of Asterisk are you using? |
| 21:06.46 | Hive | <PROTECTED> |
| 21:06.56 | Hive | Asterisk 10.4.0 |
| 21:07.08 | mjordan | are you on a 32-bit machine by any chance? |
| 21:07.35 | danfromuk | Anyone know if there was there a bug in asterisk 1.8.12 that causes recordings to get deleted at the end of a call? |
| 21:07.41 | Hive | 64 bit |
| 21:09.13 | mjordan | hm |
| 21:09.28 | gg608f | hi all, anyone have installed LUA module in asterisk? |
| 21:09.28 | gg608f | <PROTECTED> |
| 21:09.28 | gg608f | <PROTECTED> |
| 21:09.28 | gg608f | <PROTECTED> |
| 21:09.31 | mjordan | what channel technology is in the queue? |
| 21:09.41 | mjordan | danfromuk: no |
| 21:09.43 | Hive | how can I tell that? |
| 21:10.01 | mjordan | well, what channel is sitting in the queue when you're testing? |
| 21:10.09 | Hive | ahh |
| 21:10.39 | mjordan | danfromuk: and you haven't said how you're recording the message. |
| 21:10.40 | danfromuk | Any way to get asterisk to output more information on what its doing? IE why its creating the .wav file during recording and deleting it at hangup? verbose and debug dont show anything. |
| 21:10.49 | Hive | SIP/<channel info> |
| 21:11.07 | danfromuk | mjordan: One line.... cmd Record(/path/to/file.wav) |
| 21:11.37 | mjordan | You need the 'k' option. |
| 21:11.44 | mjordan | by default it does not keep the recording on hangup. |
| 21:11.50 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Application_Record |
| 21:12.24 | Hive | mjordan: type sip i believe |
| 21:12.25 | mjordan | last sentence in the description of the application. |
| 21:12.27 | danfromuk | I wasnt aware of the k option. What would be the benefit of recording without the K option? |
| 21:12.28 | mjordan | Hive: you are correct |
| 21:12.31 | iulhk | can i use perl agi & php agi files at the same time at the same server for different services ? |
| 21:12.47 | danfromuk | Must be new. Recently upgraded from 1.4 where k wasnt needed. |
| 21:12.49 | mjordan | Hive: I'm not entirely sure why its not playing. Let me lab it up locally. |
| 21:13.47 | Hive | mjordan: thank you very much |
| 21:18.38 | mjordan | Hive: k, I think I might know what's going on here |
| 21:18.45 | mjordan | Hive: are you testing with a single caller in the queue? |
| 21:19.16 | Hive | yes |
| 21:19.22 | Hive | bleh |
| 21:19.29 | Hive | ill try a double 1s |
| 21:19.33 | mjordan | Hive: :-) The periodic-announce is only for people who are in the queue waiting for an agent to become free |
| 21:19.39 | *** join/#asterisk cyborg-one (1000@212-178-5-139.broadband.tenet.odessa.ua) |
| 21:19.53 | mjordan | Hive: So if you have a single caller in the queue and the agents are ringing, then it won't play. The agents aren't busy. |
| 21:20.15 | mjordan | Hive: But if you have all of your agents busy and none of them are being rung, then the periodic announcement logic will kick in for whoever else is sitting in the queue |
| 21:20.39 | mjordan | try having a single agent and calling into the queue. Have the agent answer. |
| 21:20.50 | mjordan | Then take another phone and call into the queue. You should start getting your periodic announcements. |
| 21:22.50 | *** join/#asterisk pyther (~pyther@unaffiliated/pyther) |
| 21:22.50 | mjordan | Hive: http://pastebin.com/NavsSKKb |
| 21:23.04 | Hive | okay this is deff getting somewhere, I can see the following in my CLI: Playing 'queue-periodic-announce.ulaw' (language 'en') |
| 21:23.18 | Hive | though I can't hear it, as well as not being able to hear musicclass |
| 21:24.08 | Hive | Thanks for helping me get here mjordan |
| 21:24.49 | mjordan | Hive: np :-) |
| 21:27.54 | Hive | Mjordan: in that pastebin, yours is playing .gsm |
| 21:28.09 | mjordan | that's all I have installed |
| 21:28.12 | Hive | ahh okay |
| 21:31.37 | danfromuk | Hi, whats the extension wildcard for ANYTHING? Like _X. is for any number. |
| 21:33.04 | _Corey_ | danfromuk: Well, "anything" would be _! I believe |
| 21:33.05 | *** join/#asterisk lorsungcu_ (~anonymous@12.40.178.133) |
| 21:33.29 | danfromuk | What about anything except for nothing? |
| 21:33.39 | danfromuk | _.? |
| 21:33.47 | danfromuk | Which comes up with a warning in the CLI. |
| 21:33.59 | _Corey_ | danfromuk: _. is one or more chars, _! is zero or more |
| 21:33.59 | Hive | yes because it accepts everything :P |
| 21:34.23 | danfromuk | Thats fine. Its not exposed to users. |
| 21:34.26 | _Corey_ | danfromuk: lemme try with one underscore: _. |
| 21:34.35 | _Corey_ | that's one or more |
| 21:34.44 | _Corey_ | (of any char) |
| 21:35.06 | _Corey_ | I'd stay away from the _. or ! matches if you can... they get dangerous |
| 21:35.40 | Hive | mjordan: in queues.conf, musicclass is used to have music play instead of ringing right? or is that a separate option |
| 21:36.02 | danfromuk | ok, thanks for your help again. |
| 21:41.34 | mjordan | Hive: are you getting the " -- Music class default requested but no musiconhold loaded." message? |
| 21:51.52 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 22:00.14 | navaismo | How much memory needs tha AGI cmd in order to run? Im getting this --> [Aug 6 17:58:52] WARNING[1111] res_agi.c: Failed to fork(): Cannot allocate memory |
| 22:08.34 | Carko | how much free memory do you have |
| 22:10.02 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-xsmymwktnexvmfoy) |
| 22:13.30 | navaismo | 159 MB |
| 22:16.22 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 22:20.39 | navaismo | ? |
| 22:22.10 | Ralcala | hi fellas |
| 22:22.35 | Ralcala | how i can avoid interrupting a current call with another coming? |
| 22:23.09 | *** join/#asterisk tamiel (~tamiel@c-69-181-254-90.hsd1.ca.comcast.net) |
| 22:27.57 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.37) |
| 22:32.12 | ChannelZ | Ralcala: turn off call waiting on the phone |
| 22:32.33 | ChannelZ | Or program your dialplan to look if the channel is in use first and then act accordingly |
| 22:32.36 | ChannelZ | Or use call counters |
| 22:33.32 | *** join/#asterisk cisco (~cisco@201.20.110.36) |
| 22:37.32 | *** join/#asterisk N2Li (4445a96f@gateway/web/freenode/ip.68.69.169.111) |
| 22:42.00 | navaismo | I disable 10 unused modules and still getting the --> WARNING[1111] res_agi.c: Failed to fork(): Cannot allocate memory |
| 22:42.23 | N2Li | Any U.S. Based Asterisk Developers looking for hourly work developing creative and innovative asterisk applications, and work from home doing it? (2 to 4 months work available, maybe more after depending how things go) |
| 22:42.36 | Digweed | anyone an idea on how to strip contact uri user part from an outgoing invite? |
| 22:42.47 | Digweed | host part should stay but i need to drop the user part |
| 22:42.49 | Digweed | asterisk 1.6 |
| 22:42.53 | *** join/#asterisk brainiac (~Brainiac@cpe-70-112-100-117.austin.res.rr.com) |
| 22:45.52 | ChannelZ | navaismo: you have some other issue... is asterisk running under 'nice' or something and constraining it's memory usage? |
| 22:46.12 | ChannelZ | s/it's/its/ |
| 22:46.37 | N2Li | Any Asterisk Developers looking for hourly work developing creative and innovative asterisk applications, and work from home doing it? (2 to 4 months work available, maybe more after depending how things go) - (U.S. Developers would be more appealing) (All 'under-the-table' work, no reporting, no taxes, weekly pay, greendot or paypal are fine, check or money orders fine to, will use whatever method works best for you, for your payment |
| 22:46.41 | ChannelZ | Or could be a red herring and the fork is failing for some other reason |
| 22:48.42 | navaismo | ChannelZ, is running in the RaspberryPi |
| 22:49.39 | ChannelZ | hmm can't help you there |
| 22:50.30 | navaismo | :'( |
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| 23:22.35 | navaismo | ok another one about DPMA: inter-vlan use of dpma is possibile and if so how do we go about it |
| 23:26.40 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
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| 23:40.24 | navaismo | how can I execute a system command and assign the result to one variable in the dialplan |
| 23:40.31 | riwarren | can anyone take a look at this pastebin of my CLI log? I can't get outgoing calls through my trunk to work and i've tried just about everything. http://pastebin.com/1vnKVrJh |
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| 23:56.01 | Dar1us | navaismo: you can't afaik |
| 23:56.13 | Dar1us | navaismo: I ended up writing an AGI script which did that though |
| 23:57.36 | navaismo | yes, i cant use the AGI cmd i dont have enogh memory to do that |