| 00:01.19 | pabelanger-lap | ~itsp |
| 00:01.20 | 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. |
| 00:01.57 | a1fa | ~itsplist-us |
| 00:01.58 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 00:02.09 | a1fa | lol |
| 00:02.16 | a1fa | i have a client who wants to use packet8 |
| 00:02.24 | a1fa | i've not heard good things about them |
| 00:03.14 | a1fa | besides, they want to sell him the astra phone.. lolZ |
| 00:03.27 | a1fa | VO IP Phone - 6755i ($199.99) 4x$99 |
| 00:05.36 | citywok | we pay a little over $200 per 6757i and love 'em, not sure how aastra is a bad thing. |
| 00:07.08 | a1fa | whats so good about them? |
| 00:07.16 | a1fa | besides the share line so you can tell when line is in use |
| 00:07.42 | citywok | the speakerphone is great, and on the 6 line displays we get the XML browser is very easy to work with |
| 00:08.05 | citywok | the speakerphone is crystal clear even if the speaker is 15 feet away from the phone and facing the other direction |
| 00:08.37 | a1fa | does it work with *? |
| 00:08.59 | citywok | why would i be in the * channel tlaking about it if it didnt? |
| 00:09.01 | a1fa | ~packet8 |
| 00:09.10 | a1fa | how do you configure shared line thing? |
| 00:09.16 | citywok | i dont use it |
| 00:09.23 | citywok | i use blf to see if somebody is on the line |
| 00:09.42 | a1fa | blfz? |
| 00:13.10 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 00:15.17 | a1fa | ~sipgate |
| 00:15.18 | infobot | well, sipgate is http://www.sipgate.de |
| 00:15.46 | a1fa | shush.. |
| 00:15.59 | a1fa | anyway, i am debating to tell this customer whether or not he should outsource his pbx |
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| 00:21.06 | rustyclarkson | Is there an easy way to have the MOH class used dependent on time of day? |
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| 00:31.23 | nezZario | Hm ok, few questions about the voicemail thing |
| 00:31.32 | pabelanger-lap | rustyclarkson: GoToIfTime |
| 00:31.39 | pabelanger-lap | ~ask |
| 00:31.40 | 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. |
| 00:31.49 | nezZario | For one, when I call Voicemail(blah@default) .. How can I make 1) the person not say "thank you!" at the end (I am passing 's' alreayd) |
| 00:31.57 | nezZario | and 2) have it to where they can press any key, not just #, to end recording |
| 00:35.24 | pabelanger-lap | nezZario: You need to make the changes in app_voicemail.c |
| 00:36.09 | nezZario | to the source? |
| 00:36.11 | nezZario | that's lame :/ |
| 00:37.06 | nezZario | also i keep getthing this message |
| 00:37.08 | nezZario | <PROTECTED> |
| 00:37.13 | nezZario | over, and over, and over, and over..... |
| 00:37.43 | nezZario | [Jun 24 00:22:47] WARNING[1451]: db.c:57 dbinit: Unable to open Asterisk database '/var/db/asterisk/astdb': No such file or directory |
| 00:39.11 | pabelanger-lap | nezZario: Does the file exist? |
| 00:39.17 | rustyclarkson | nezZario: then create the directory |
| 00:41.00 | nezZario | yea i figured that part out |
| 00:41.05 | nezZario | so there is no way but to modify the dang source?? |
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| 00:41.54 | pabelanger-lap | nezZario: Correct, make the change and submit your patch to https://issues.asterisk.org/ |
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| 00:59.00 | joobie | hey guys.. got a recording for an IVR that im trying to cleanup the background noise in.. it's got like a constant fuzz .. i tried playing with "noise reduction" features where you select just the noise for the sample profile and then apply that noise reduction across the whole audio.. worked ok, but the voice ends up being a bit robotic after.. any idea how to best resolve this or howt o best approach this? |
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| 01:06.59 | hansin | Does anyone notice sometimes that ekiga.net acts funny for SIP stuff? It worked fine yesterday. Then today I can register my softphone, I can even call in via a DID number and the softphone rings. But I can answer the call and there is no ringing on the DID side of the call (incoming). Seems strange. If I try and answer the call the softphone keeps ringing (yeah, that rings) yet Wireshark... |
| 01:07.01 | hansin | ...shows RTP traffic?? |
| 01:07.43 | hansin | I'm no VoIP expert so I could be missing something. But this just worked yesterday no problem (and many days before). |
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| 01:09.52 | Godfather_ | hi |
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| 01:23.48 | {Beowulf} | has anybody implemented Skype for Asterisk here? |
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| 01:32.07 | boodu | ++ |
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| 02:09.14 | ManxPower | ~answers |
| 02:09.20 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
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| 02:16.50 | nezZario | anyone ever screwed around in the asterisk c source? |
| 02:17.00 | nezZario | just trying to make some minor changes but i can't find documentation for any of these functions |
| 02:20.12 | WIMPy | Did you "make progdoc"? |
| 02:22.32 | nezZario | gmake: *** No rule to make target `progdoc'. Stop. |
| 02:22.46 | *** join/#asterisk s0lid (~s0lid@121.96.69.36) |
| 02:24.25 | nezZario | i'm just trying to make it so a person can press any key, instead of just pound, to finish recording a voicemail |
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| 02:27.12 | WIMPy | Sorry progdocs. |
| 02:28.17 | WIMPy | It told you when you made install. |
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| 02:32.14 | picard1421 | quick asterisk question.. when i have a call it will time out after like 6 seconds.. it basically dies wayy to quickly |
| 02:32.30 | picard1421 | Dial("Local/44number@from-internal-e797,2", "SIP/callwithus/011number|300|") in new stack |
| 02:32.45 | picard1421 | i want that to last like 20 seconds |
| 02:32.48 | picard1421 | before the call times out? how do i do that? |
| 02:33.39 | picard1421 | http://pastebin.com/NXP2GbuP |
| 02:33.45 | picard1421 | any help is appreciated thanks |
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| 02:43.45 | joobie | specify a timeout with your dial cmd picard1421 |
| 02:43.48 | joobie | look at the manual for dial() |
| 02:44.00 | joobie | it's like the last arg i think |
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| 03:49.14 | dandate2 | so do datacenters usually provide phone line service for digium card predictive dialers / dial up isps or do those have to be inhouse operations |
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| 04:04.55 | drclue | I'm looking for input as to what folks would like to see in an HTML5 SDK. I have developed an open source server side interface that can support ~200 concurrent HTML5 clients on even a cheap server. The client side uses an XML based interface that is Asterisk version agnostic |
| 04:07.05 | drclue | This is NOT an AJAX interface but rather utilizes real persistent socket connection between the HTML5 browser and the multi-process websocket based AMI interface |
| 04:08.49 | drclue | EVERY asterisk event is supported , and the interface supports both client instance specific events and broadcast general events , as well as custom commands and custom events on all and targeted channels |
| 04:13.03 | drclue | The user defined modules based command / response system is being implemented so as to share the same kind and type of interface afforded by my current open source FastAGI interface , in the hopes that many of the FastAGI plugins can be equally deployed in this context as well within the limitations of the contexts involved. |
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| 04:41.22 | dandate2 | to get a pbx hooked up to the pstn do I have to use a digium card or could it be done with COM ports? |
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| 04:57.26 | picard1421 | joobie: are you there? |
| 04:59.29 | picard1421 | where is the dial script locatd |
| 04:59.32 | picard1421 | which directory? |
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| 05:33.36 | picard1421 | anyone need some quick help |
| 05:33.38 | picard1421 | my calls are timing out |
| 05:33.41 | picard1421 | http://pastebin.com/4FWLNVjv |
| 05:34.19 | picard1421 | basically its dialing out but my cell phone take like 5 rings to respond to the call.. aka the call will time out before my cell phone gets the incoming call.. how can i make the dial command not timeout as fast? also where is that stored? what do i have to edit? thanks! |
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| 05:37.43 | Kyosh | picard: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 05:38.03 | Kyosh | Dial(type/identifier, timeout, options, URL) |
| 05:39.06 | Kyosh | ex: exten =>_1X.,1,Dial(SIP/${EXTEN},60) where the 60 is 60 sec timeout |
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| 05:44.54 | picard1421 | hmmm kk |
| 05:48.10 | picard1421 | hey |
| 05:48.12 | picard1421 | under |
| 05:48.14 | picard1421 | the |
| 05:48.30 | picard1421 | the line 8 |
| 05:48.36 | picard1421 | what is the |300 |
| 05:49.37 | *** join/#asterisk farkus (chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 05:50.30 | ChannelZ | that's the call timeout |
| 05:50.39 | ChannelZ | core show application dial |
| 05:59.48 | picard1421 | ok then what |
| 05:59.50 | picard1421 | basically |
| 05:59.52 | picard1421 | i want that to timeout |
| 05:59.56 | picard1421 | like 15 seconds later |
| 06:00.54 | picard1421 | basically i want it to try and connct the call for 15 seconds longer before it hagns it up.. what would i have to change? |
| 06:02.06 | ChannelZ | 15 |
| 06:02.22 | [TK]D-Fender | - SIP/callwithus-09d0b828 answered Local/19485527888@from-internal-0706,2 |
| 06:02.27 | [TK]D-Fender | ANSWERED |
| 06:02.38 | [TK]D-Fender | Whats this timeout story about then? |
| 06:02.50 | ChannelZ | I dunno I just got here |
| 06:03.01 | picard1421 | the problem is .. |
| 06:03.38 | [TK]D-Fender | picard1421: Problem is you're asking dialplan questions about FREEPBX. |
| 06:03.41 | [TK]D-Fender | ~freepbx |
| 06:03.52 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 06:03.52 | picard1421 | ok basically im using a callback script for my cell phone.. what happens is when it dials the cellphone back the dial command already times out before the call is connected. |
| 06:03.52 | [TK]D-Fender | picard1421: 2nd door to the left |
| 06:03.52 | picard1421 | but the concept should still be the same |
| 06:04.09 | ManxPower | FreePBX. now I remember why I stopped coming here. |
| 06:04.13 | *** part/#asterisk ManxPower (~manxpower@216.186.151.147) |
| 06:04.17 | ChannelZ | The concept is the same - where you tell FreePBX to do something is another matter |
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| 06:17.38 | ChannelZ | Hmm. ManxPower must have been headed to #freepbx then? |
| 06:17.54 | [TK]D-Fender | ChannelZ: Yeah.. that's it... that's the ticket... |
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| 06:22.30 | ChannelZ | :) |
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| 06:46.34 | Dovid | morning ev1 |
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| 06:51.35 | ChannelZ | hlo |
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| 07:05.57 | Godfather_ | hi |
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| 07:18.02 | *** join/#asterisk Keal (~jargon@unaffiliated/jargon) |
| 07:18.07 | ChannelZ | My right armpit is itching something fierce. |
| 07:18.18 | ChannelZ | Just thought you should know. |
| 07:18.27 | Keal | Anybody want to hop on the Timothy-based Python build of Asterisk? |
| 07:18.59 | Keal | Timothy is a military-grade memory core. |
| 07:20.01 | *** join/#asterisk s0lid (~s0lid@121.96.69.36) |
| 07:21.35 | Keal | This new build pretty much 100% of the Asterisk developer community's role in expertise. |
| 07:21.52 | Keal | This new build pretty much undermines* 100% of the Asterisk developer community's role in expertise. |
| 07:22.49 | Keal | Everyone will be newbs again with the new build. |
| 07:23.55 | ChannelZ | I do not understand the words you use |
| 07:23.58 | *** join/#asterisk farkus (chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 07:24.46 | Keal | This new rebuild of Asterisk is called Apteryx |
| 07:26.04 | Keal | It makes Google Voice and Goog411 look like caveman era telephony. |
| 07:26.49 | jargon | Now you know who I am. |
| 07:27.15 | jargon | ..right? |
| 07:27.22 | Kyosh | hah |
| 07:27.49 | Kyosh | dude |
| 07:27.51 | jargon | I designed the first contemporary Apache prototype back in 1997. |
| 07:28.21 | jargon | I saved the United States Secret Service mainframes from obsoletion. |
| 07:28.34 | jargon | I saved Landwarrior. |
| 07:28.55 | wdoekes2 | Princess Leia? |
| 07:29.10 | Kyosh | jargon |
| 07:29.14 | Kyosh | you're histerical |
| 07:29.31 | wdoekes2 | *hysterical</grammar_nazi> |
| 07:29.34 | wdoekes2 | sorry |
| 07:29.36 | jargon | I developed the founding philosphies of Augmented Reality and Tactical Creation. |
| 07:29.53 | Kyosh | ok look |
| 07:30.01 | jargon | I wrote the proof for the dark space and matter bomb. |
| 07:30.21 | jargon | I designed the foundations for quantum crack-proof encryption. |
| 07:30.21 | Kyosh | <Keal> Anybody want to hop on the Timothy-based Python build of Asterisk? |
| 07:30.23 | Kyosh | <PROTECTED> |
| 07:30.32 | Kyosh | <Keal> This new rebuild of Asterisk is called Apteryx |
| 07:30.38 | Kyosh | <Keal> It makes Google Voice and Goog411 look like caveman era telephony. |
| 07:30.45 | Kyosh | so now |
| 07:31.24 | Kyosh | it looks like you're blowing tons of bullshit up your own ass |
| 07:31.24 | Kyosh | so now i am curious |
| 07:31.24 | Tim_Toady | 'obvious troll is obvious' |
| 07:31.24 | Kyosh | are you using asterisk code, but rewritten in python, or are you writing an entirely new telephony engine? |
| 07:31.32 | jargon | I cracked Hand ogf God with two independent coinciding proofs with no common mathematical basis. |
| 07:31.40 | Kyosh | i see |
| 07:31.40 | jargon | Hand of God* |
| 07:31.46 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:31.46 | Kyosh | ok back to my videos |
| 07:31.52 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 07:32.50 | coppice | apteryx are very tasty |
| 07:33.49 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
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| 07:35.33 | Kyosh | bahit must be april 1st |
| 07:36.58 | ChannelZ | his calendar must be written in Python |
| 07:38.44 | *** join/#asterisk sulex (~sulex@93-35-8-50.ip52.fastwebnet.it) |
| 07:39.43 | jargon | I invented a Delaunay triangulation method simpler than O(n log log n) |
| 07:41.11 | jargon | Kyosh, entirely rebuild. Every line rewritten. |
| 07:41.32 | jargon | Even python is 100% rebuilt. |
| 07:42.43 | *** join/#asterisk s0lid (~s0lid@121.96.69.36) |
| 07:43.55 | jargon | Apteryx is a full kernel/scripting/telephony system rebuild. |
| 07:44.17 | jargon | You can't even refer to it as Unix anymore. |
| 07:44.25 | coppice | no its not. its a bird that can't fly |
| 07:44.44 | coppice | with a dark gamey meat, that is delicious |
| 07:44.56 | ChannelZ | sounds supar |
| 07:45.10 | jargon | It is what Hurd would be if Stallman actually knew his ass from a hole in the ground. |
| 07:47.18 | *** join/#asterisk BANSAL (~bansal@117.207.85.104) |
| 07:47.58 | *** join/#asterisk farkus (chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 07:49.11 | jargon | Unlike Stallman, I actually finish 100% of a project myself, then pass it on to people that will actually spent trillions continuing the development for reasons of global security rather than some halfass anarchy conditon. |
| 07:49.14 | jargon | condition |
| 07:49.27 | ChannelZ | yawns |
| 07:50.08 | jargon | Apteryx addresses into nearing infinite range using bitslice. |
| 07:51.08 | jargon | You can address higher than there are subatomic particles in the known universe with Apteryx. |
| 07:51.51 | dohd | and the most important question remains unanswered... |
| 07:52.00 | dohd | will it get you into ladies' panties at parties |
| 07:52.11 | dohd | if not, infinite improbability is way more useful |
| 07:53.44 | *** join/#asterisk s0lid (~s0lid@121.96.69.36) |
| 07:53.56 | puzzled | is this some kind of bot spewing bs? |
| 07:56.25 | Kyosh | jargon, so why compare it to asterisk? you may as well compare it to every telephony engine out there. asterisk, freeswitch, yate, etc |
| 07:57.10 | Kyosh | puzzled, seems that way |
| 07:57.37 | Kyosh | he designed a patton-f2 micron kernel controlled processor engine that vibrates his clit at 1k cycles per second |
| 07:58.01 | puzzled | yup, maybe that Apt... is something they want pushed up in Google ranks. if it works that way |
| 07:58.17 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:58.43 | Kyosh | and according to him his clit vibrator will cripple a billion dollar corporation like google and their product google voice |
| 07:58.55 | Kyosh | him and his $500 pc will cripple google |
| 07:59.04 | puzzled | lol |
| 07:59.15 | Kyosh | hey anything is possible |
| 07:59.37 | puzzled | yes just not for a zillion years :) |
| 07:59.46 | Kyosh | right |
| 07:59.56 | Kyosh | he is desinging the next level in communications |
| 08:00.04 | Kyosh | when his head speaks with his ass |
| 08:00.34 | puzzled | hope he gets the flux capacitor sorted out. that part is always a bitch to get right |
| 08:00.48 | Kyosh | oh yea i totally forgot about the flux! |
| 08:00.51 | Kyosh | shit |
| 08:03.00 | puzzled | hope the power supply in his $500 pc is capable of handling the 1.21 gigawatts required to get it going |
| 08:03.24 | Kyosh | 1.21 gigawatts? ONE POINT TWENTY ONE GIGAWATTS!??!?!? |
| 08:03.28 | *** join/#asterisk mloe (~ho@ip-70-160.dobrich.net) |
| 08:03.45 | puzzled | told ya it was a bitch to get right :) |
| 08:04.04 | mloe | guys, I'm using AsteriskNOW 1.7 and in /etc/asterisk don't have extensions.conf.. only extensions_custom.conf |
| 08:04.07 | Kyosh | yea i guess as long as its 5.0x10^-50 amps |
| 08:04.09 | mloe | why is that? |
| 08:04.14 | puzzled | but then it does more sip sessions than there are phones on the planet |
| 08:04.48 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 08:04.55 | Kyosh | mloe, go to #asterisknow ? |
| 08:05.01 | *** join/#asterisk sulex (~sulex@firewall.blindata.ch) |
| 08:05.01 | puzzled | mloe: no idea. don't use it |
| 08:05.05 | *** join/#asterisk Belgarath (~Belgarath@banda.pl) |
| 08:05.07 | mloe | ok |
| 08:05.11 | *** join/#asterisk troffasky (~r00t@no.ipdetails.info) |
| 08:05.22 | mloe | but I wanted to configure an extension |
| 08:05.22 | mloe | :)) |
| 08:05.27 | mloe | I'll ask there |
| 08:05.37 | Kyosh | then use extensions_custom.conf |
| 08:05.44 | Kyosh | but wait |
| 08:05.47 | mloe | :) |
| 08:05.52 | Kyosh | use the web interface since asterisknow has one |
| 08:06.01 | mloe | oh okay |
| 08:06.04 | Kyosh | duh |
| 08:06.07 | mloe | :D |
| 08:06.09 | Kyosh | thats the point of astnow |
| 08:06.14 | Kyosh | its has a web int |
| 08:06.24 | mloe | yes but |
| 08:06.28 | Kyosh | but? |
| 08:06.35 | Kyosh | no but |
| 08:06.45 | mloe | somehow the system doesn't allow other PCs to connect to the interface |
| 08:06.46 | mloe | :)) |
| 08:06.56 | mloe | I'm trying to reach the host from another computer |
| 08:07.02 | mloe | but - access forbidden |
| 08:07.11 | Kyosh | did you read the tutorial/getting started? |
| 08:07.20 | mloe | I'll search these allow/denny |
| 08:07.24 | mloe | yep, I'm starting now |
| 08:07.24 | mloe | :) |
| 08:07.25 | Kyosh | you may have to configure apache to allow 0.0.0.0 |
| 08:09.14 | *** join/#asterisk s0lid (~s0lid@121.96.69.36) |
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| 08:26.17 | *** join/#asterisk jony123 (jony123@unaffiliated/jony123) |
| 08:26.41 | jony123 | With freepbx can i still make changes to extentions.conf and other files in /etc/asterisk ? |
| 08:27.13 | Tim_Toady | jony123 u can, but for this better ask at #freepbx |
| 08:28.10 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
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| 08:52.46 | mort_gib | Is there ANY way of parsing the extensions.conf file before loading it?? |
| 08:54.26 | *** join/#asterisk ltd (~z@pat.transact.net.au) |
| 08:54.51 | *** join/#asterisk freeedrich| (~eeePC@hansaserver.de) |
| 08:55.29 | *** join/#asterisk freeedrich| (~eeePC@hansaserver.de) |
| 09:00.13 | jargon | cya |
| 09:06.17 | *** join/#asterisk sputnick (~sputnick@unaffiliated/sputnick) |
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| 09:13.14 | *** part/#asterisk gavimobile (~user@bzq-84-108-29-62.cablep.bezeqint.net) |
| 09:18.37 | mloe | hey guys I'm trying to change my asterisk password |
| 09:18.47 | mloe | I type asterisk -r |
| 09:18.56 | mloe | and then passwd but it shows me that there is no such command |
| 09:20.25 | tzafrir | mloe, what do you mean by "asterisk password"? Password to the manager interface? Voicemail? SIP user? |
| 09:20.42 | mloe | manager interface |
| 09:20.43 | mloe | :) |
| 09:20.59 | mloe | I have FreePBX, too |
| 09:25.29 | *** join/#asterisk freeedrich| (~eeePC@hansaserver.de) |
| 09:26.27 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 09:28.44 | tzafrir | mloe, edit /etc/asterisk/manager.conf ? |
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| 09:38.34 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
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| 10:21.22 | *** join/#asterisk js (~js@ip-109-91-185-129.unitymediagroup.de) |
| 10:22.04 | js | hi. I'm new to asterisk. When I start asterisk via the rc.d script, it terminates after a few seconds. I looked into event_log, queue_log and messages, but could only find warnings in messages. is there any other place to look or can the warnings be the reason for the shutdown? |
| 10:23.19 | kaldemar | start asterisk with asterisk -vvvvc and see where it crashes. |
| 10:24.29 | js | asterisk: Error detected by libpthread: Invalid condition variable. |
| 10:24.32 | js | ouch |
| 10:24.39 | js | that sounds like a bug in asterisk? |
| 10:25.53 | js | it seems exporting PTHREAD_DIAGASSERT=A helps |
| 10:26.12 | js | but that should not happen in the first place, I guess |
| 10:27.30 | *** join/#asterisk DelphiWorld (~Delphi@41.103.48.6) |
| 10:27.32 | DelphiWorld | hello all |
| 10:27.59 | DelphiWorld | any way to select channels to compile or thant make menuconfig? |
| 10:30.11 | DelphiWorld | hi Qwell |
| 10:30.22 | DelphiWorld | please could someone help me to select my needed channels |
| 10:31.40 | DelphiWorld | i am unable to select using enter |
| 10:31.44 | tzafrir | DelphiWorld, what do you want? |
| 10:32.09 | DelphiWorld | tzafrir: i want to select the needed channels for asterisk using make menuconfig but i am unable to togle selection using enter |
| 10:32.10 | tzafrir | You can't use menuselect? |
| 10:32.42 | DelphiWorld | tzafrir, yes |
| 10:33.15 | js | what is skinny.conf? it seems to bind something to TCP port 2000 |
| 10:33.33 | DelphiWorld | js, is for CISCO SCCP propritary protocol |
| 10:34.00 | js | is there some place where can I disable it? |
| 10:34.09 | js | I'm overwhelmed by more than 10k lines of config |
| 10:34.11 | tzafrir | hmm... it had a conflict on that port. With some service that later on moved to a different port, IIRC |
| 10:34.29 | js | is there some kind of an example config that does nothing so I can be sure asterisk does only the stuff I added? |
| 10:34.30 | tzafrir | js, grep -v '^;' is your friend |
| 10:35.03 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 10:35.13 | tzafrir | IIRC the sample file should do nothing unless you edit it |
| 10:35.14 | js | modules.conf sounds like what I'm looking for |
| 10:35.20 | js | tzafrir: well, it opened port 2000 here :/ |
| 10:35.56 | js | interesting, modules.conf does not list skinny, still, it seems to be loaded |
| 10:36.13 | tzafrir | DelphiWorld, unless someone comes up with a better suggestion, I'll point out menuselect/contrib/menuselect-dummy |
| 10:36.37 | tzafrir | Which is more command-line oriented and is configured from a simple config file |
| 10:37.06 | js | tzafrir: is something wrong with my setup if it loads skinny.conf and opens port 2000 by default? |
| 10:37.13 | tzafrir | But then again, if we take one step baackwords, channels are enabled by default, Unless some of their dependencies are missing |
| 10:37.50 | tzafrir | DelphiWorld, please run: ./menuselect/contrib/menuselect/menuselect-dummy -c |
| 10:37.54 | DelphiWorld | tzafrir,, what i need to do in this directory? |
| 10:38.06 | Martinblr | there is no audio after I enabled stun |
| 10:38.16 | tzafrir | and then print the output of: ./menuselect/contrib/menuselect/menuselect-dummy -m chan_whatever -v |
| 10:38.18 | DelphiWorld | tzafrir, you answered my question befaure i could ask it;) |
| 10:38.33 | tzafrir | replace chan_whatever with the name of the channel driver you want to build |
| 10:38.38 | Martinblr | I have remote extension registered in asterisk, after enabled stun server no voice is there |
| 10:38.45 | DelphiWorld | tzafrir, Failed to open makeopts. Aborting: No such file or directory |
| 10:39.21 | tzafrir | hmm... looks like ./configure was not run? Or failed to run? |
| 10:39.30 | DelphiWorld | tzafrir, and also something else is confusing me! |
| 10:39.43 | js | even if I comment out the stuff in skinny.conf, it still opens port 2000. why? |
| 10:39.56 | DelphiWorld | tzafrir, i have allready doing ./configure, make and make install for default config |
| 10:40.23 | DelphiWorld | tzafrir, i see chan_gingle, chan_gtalk, and chan_jabber! 3 diferent channels. what i should use to do jingle? |
| 10:40.39 | *** join/#asterisk farkus (chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 10:40.41 | DelphiWorld | js, yuninstall chan_skinny.so |
| 10:40.48 | *** join/#asterisk Akiraa (~Akira@92.81.173.253) |
| 10:41.10 | tzafrir | DelphiWorld, from what dommand did you get the error about makeopts? |
| 10:41.38 | DelphiWorld | tzafrir, using your proposed line menuselect_dumy -c or something else |
| 10:42.27 | tzafrir | makeopts is a file that is generated by configure. Are you sure 'make' was run successfully? |
| 10:42.53 | DelphiWorld | tzafrir, i am realy confused i am not sure |
| 10:42.59 | js | DelphiWorld: yuninstall? I added noload => chan_skinny.so now, seems to work |
| 10:43.34 | DelphiWorld | js, good/nice/awesome/clever;) |
| 10:44.54 | *** join/#asterisk Delido1983 (~Delido198@212.144.236.196) |
| 10:45.06 | DelphiWorld | tzafrir, do i need to move it to main directory? |
| 10:47.55 | tzafrir | DelphiWorld, I suggested to use it for debugging |
| 10:50.03 | DelphiWorld | tzafrir, how do i clean my server? |
| 10:50.07 | DelphiWorld | i want to remove asterisk binarys |
| 10:53.18 | DelphiWorld | a2billing don't work with 1.6.X? |
| 10:54.41 | DelphiWorld | dan |
| 10:54.44 | *** part/#asterisk DelphiWorld (~Delphi@41.103.48.6) |
| 10:59.31 | js | what's the effect of callduration in extensions.conf? |
| 10:59.32 | *** join/#asterisk ctaloi (~ctaloi@pool-71-176-70-70.syrcny.fios.verizon.net) |
| 10:59.41 | js | does it limit the length of a call? |
| 11:06.39 | Martinblr | where do we get all the countries pstn tone details. any reference..? |
| 11:08.49 | tzafrir | Martinblr, zonedata.c in dahdi-tools . But configs/indications.conf.sample should have the same data |
| 11:09.03 | coppice | The ITU E.180 spec is the most complete list of the world's tones, but its not the most up to date |
| 11:09.12 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 11:18.12 | *** join/#asterisk ruied (~ruied@89-180-247-219.net.novis.pt) |
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| 11:43.28 | x-demon | hi, how i can specify all extensions with @ |
| 11:43.30 | x-demon | ? |
| 11:43.38 | x-demon | something like user@domain |
| 11:44.14 | x-demon | tried _.@. but this gets overlwriten by _Z. , which is for pstn calls |
| 11:44.24 | x-demon | so i can call only user@domain |
| 11:44.29 | x-demon | but not number@domain |
| 11:44.52 | x-demon | something like 1000@domain.com gets routed via sip trunks |
| 11:52.16 | *** join/#asterisk Mw3 (mw3@mw3.hu) |
| 11:53.35 | *** join/#asterisk structz (~structz@gandalf.ai.unesp.br) |
| 11:56.09 | Martinblr | Cant able to find greece tones in ITU spec.. in zonedata.c also only some parameters are there |
| 11:57.55 | coppice | there is a long list of greek tones there. what is missing? |
| 11:59.38 | Martinblr | db parameter |
| 11:59.44 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 12:01.02 | coppice | that's not a tone |
| 12:02.33 | Martinblr | coppice: where I can get that frequencey DB value for greece |
| 12:03.15 | coppice | ITU E.180 |
| 12:03.40 | *** join/#asterisk felipe_ (~felipe@my.nada.kth.se) |
| 12:03.51 | Martinblr | I tried there not able to |
| 12:05.49 | coppice | i can see then all in E.180 |
| 12:06.36 | Martinblr | coppice: let me have a check again |
| 12:06.54 | *** join/#asterisk italorossi (~italoross@189.23.15.2) |
| 12:07.34 | sekil | http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf |
| 12:15.08 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
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| 12:41.49 | tzafrir | x-demon, anything after the '.' will practically be ignored |
| 12:42.15 | tzafrir | So '_.' is really a catch-all |
| 12:42.20 | x-demon | tzafrir, so how i can parse that? |
| 12:45.23 | x-demon | i used _0. and _[a-z]. |
| 12:45.33 | x-demon | but that's not good. |
| 12:49.31 | tzafrir | why? |
| 12:51.53 | Delido1983 | hello, i have a problem with the incoming callerid num from other locations. Some like Handy or analog i see the correct callerid num but 2 customers who called in i see only the basenumer "XXXXX0" where is the problem? |
| 12:52.58 | x-demon | tzafrir, at least because i should put 0 before every numerical sip id |
| 12:53.10 | pabelanger | Delido1983: What is being provided from your CO? |
| 12:53.30 | Delido1983 | what you mean whit CO? (sorry) |
| 12:53.53 | pabelanger | Delido1983: First off, how are your calls coming into Asterisk? What is the interface? |
| 12:54.39 | Delido1983 | okay we have an pri (31 channels) and Sip phones installed are Asterisk 1.6.2 whith the newest DAHDI |
| 12:56.23 | pabelanger | Delido1983: So, how is your telco (CO) sending the ANI across your PRIs? If you enable debug on your trunks is the ANI "XXXXX0"? |
| 12:56.45 | *** join/#asterisk Trixboxer (~Trixboxer@115.124.115.69) |
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| 13:00.27 | *** join/#asterisk ltd (~z@pat.transact.net.au) |
| 13:00.52 | Delido1983 | pabelanger: i must set pri debugging on ? |
| 13:02.43 | Delido1983 | pabelanger: without debugging i see only the basenumber, i check this aggain and enable debugging... thanks allot |
| 13:03.42 | pabelanger | Delido1983: It is one option, and other is to exten => 5551234,1,Verbose("CallerID: " ${CALLERID(all)}) for your incoming context |
| 13:04.02 | *** join/#asterisk boingolov (~boingolov@c-68-53-91-143.hsd1.tn.comcast.net) |
| 13:04.25 | pabelanger | Delido1983: Either way, your trying to find out if the Telco is the problem. I assume it is. |
| 13:04.55 | *** join/#asterisk viraptor (~viraptor@212.11.65.66) |
| 13:05.17 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 13:08.10 | viraptor | hi, I'm trying to start a connection via IAX, but I see no packets going out, iax debug shows correct frames with the correct ip, but there's nothing visible on tcpdump (on the same host) for example - what can I do to debug it further? (1.6.2.9) |
| 13:09.49 | pabelanger | viraptor: Firewall? |
| 13:10.00 | viraptor | not even loaded |
| 13:10.14 | Delido1983 | pabelanger: thanks allot i will test it, i have set the option in my incoming context. |
| 13:10.38 | pabelanger | viraptor: pb your iax.conf |
| 13:10.41 | pabelanger | Delido1983: np |
| 13:11.36 | Delido1983 | pabelanger: okay the customer has called me.. i the only the base number. the log show: Verbose("DAHDI/31-1", ""CallerID: " "" <00XXX900>") in new stack |
| 13:12.01 | viraptor | pabelanger: it's empty - I'm calling out by Dial(IAX2/ip_address/${EXTEN}) |
| 13:14.14 | structz | Hello all |
| 13:14.17 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:14.21 | structz | What's the best way to upgrade releases ?!!? |
| 13:15.30 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 13:16.03 | structz | anyone!? |
| 13:16.46 | pabelanger | Delido1983: Then the issue is with your telco |
| 13:16.46 | pabelanger | Delido1983: They are filtering your incoming ANI |
| 13:17.15 | *** join/#asterisk retentiveboy (~pdugas@atl.pra-corp.com) |
| 13:17.20 | pabelanger | viraptor: Not the best method to use IAX2. Setup your iax.conf with a peer, you can even register from your box to the IAX server |
| 13:17.46 | pabelanger | structz: backup, download latest release, extract, compile, install, restore backups |
| 13:18.04 | Delido1983 | pabelanger: it is an option in the /etc/asterisk/chan_dahdi.conf? or you mean vodafone filtering the incoming ANI? |
| 13:18.21 | structz | pabelanger, /etc/asterisk < bkp only confs? |
| 13:18.49 | Delido1983 | pabelanger: in my /etc/asterisk/chan_dahdi.conf i set the following options: callerid=asreceived , hidecallerid=no ,usecallerid=yes |
| 13:19.08 | *** join/#asterisk houms (~houms@wsip-70-167-244-115.dc.dc.cox.net) |
| 13:19.08 | viraptor | pabelanger: using that, because 1.4 cached the endpoint ip forever - not sure if that was fixed in 1.6 or not - anyways, Dial(ip) is not technically wrong and should work anyways, righ |
| 13:19.17 | viraptor | ht? |
| 13:19.45 | viraptor | aahhhhhh.... found it - PEBKAC - it works now :) |
| 13:20.14 | houms | can someone help me with figuring out why i cannot get desktop notifications on incoming calls? I have tried with outcall and spark with asterisk-im plugin. |
| 13:20.15 | pabelanger | viraptor: Correct it should work |
| 13:20.55 | pabelanger | houms: #freepbx |
| 13:22.05 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
| 13:22.45 | structz | pabelanger, One doubt the "-c"(color) issue about debian and asterisk has been solved on 1.6.2.9 ? |
| 13:24.32 | pabelanger | structz: Yes: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.9 |
| 13:24.49 | structz | :D thankz |
| 13:25.16 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 13:26.25 | *** join/#asterisk smooth_penguin (~smoove@59.95.3.221) |
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| 13:33.10 | *** join/#asterisk decklar (~decklar@196.31.81.190) |
| 13:33.46 | Hydrant_ | does anyone know a good sip client for the iphone? My web searching so far shows that a lot of them supposedly just forward your information to criminals... |
| 13:34.37 | *** join/#asterisk nixerpenguin (~crhodes@charm.precion.net) |
| 13:36.35 | *** join/#asterisk ccesario (~ccesario@189-29-61-213-ac.cpe.vivax.com.br) |
| 13:38.14 | *** join/#asterisk retentiveboy (~pdugas@atl.pra-corp.com) |
| 13:41.13 | Chainsaw | Hydrant_: The acrobits SIP phone works quite well for me. |
| 13:41.24 | Chainsaw | Hydrant_: Not over 3G, but I believe my telco has a hand in that. |
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| 13:49.38 | Hydrant_ | Chainsaw: what telco might I ask? |
| 13:49.45 | Hydrant_ | Chainsaw: it looks very good... thanks for the tip |
| 13:50.14 | Chainsaw | Hydrant_: O2 UK |
| 13:50.38 | Chainsaw | Hydrant_: The application offers a separate disable/enable for 3G, so if you find it doesn't work... you can just turn it off. |
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| 13:58.08 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:58.53 | *** join/#asterisk coppice (~chatzilla@202.64.176.19) |
| 14:02.59 | Katty | hi guys |
| 14:04.56 | Chainsaw | Morning Katty :) |
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| 14:08.37 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:09.22 | xheliox | What is it that you wonder? :P |
| 14:09.34 | The_Boy_Wonder | ha |
| 14:17.06 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 14:18.48 | beek | Mornin' Katty |
| 14:21.00 | structz | Hello all anyone here can give me a help on DUNDi ?! |
| 14:27.23 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:27.23 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:28.43 | xheliox | structz: Don't ask to ask, ask your question, if someone can answer it, they will. |
| 14:28.49 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 14:30.10 | *** join/#asterisk zamba (marius@flage.org) |
| 14:30.30 | structz | xheliox, ok |
| 14:32.10 | structz | Question: I Have 4 Servers "interconnected" via DUNDi each one anounces their numbers but some of then are in the same code area, i need to make another announce only for those in the same code area to call each other without using TELECOM OPERATORS!! |
| 14:33.39 | Katty | hugs Chainsaw |
| 14:33.41 | Katty | hugs beek |
| 14:33.58 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.73) |
| 14:34.12 | Katty | crittercam is up again, now that i have wireless functional again |
| 14:34.37 | Katty | for some reason the integrated wireless in the router was having spaz attacks everytime i enabled it |
| 14:35.33 | *** join/#asterisk mifadir (~ifadir@dynamic.casap1-189-30-137-41.wanamaroc.com) |
| 14:36.26 | Katty | infernix: crittercam |
| 14:36.28 | Katty | oh |
| 14:36.31 | Katty | infobot: crittercam |
| 14:36.31 | infobot | [crittercam] Katty's live broadcast of The Nut House at http://ustre.am/8H5d |
| 14:36.35 | Katty | infernix: disregard |
| 14:37.56 | beek | Katty: what does that note say that the little stuffed animal is holding? |
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| 14:40.58 | xheliox | the nut house describe the squirrels or the operator? :P |
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| 14:45.07 | Katty | beek: Diner Hours 6am - 8pm CST |
| 14:45.41 | Katty | beek: i gotta go get some more critterfoods today sometime |
| 14:46.18 | Katty | beek: ryan stole my logitech c600, so it's a bit fuzzier than usual |
| 14:46.26 | beek | The pan does look a bit empty. |
| 14:46.36 | Katty | beek: yes, a bit. |
| 14:46.39 | Katty | beek: :< |
| 14:47.10 | beek | The whole area looks so much more comfortable and cool with those leaves! |
| 14:47.48 | Katty | yes the front yard is lovely this time of year. |
| 14:47.55 | Katty | i even have a couple trees blooming |
| 14:49.07 | Katty | one of my friends suggested taking replacing the dark green rope with some clear fishing line |
| 14:49.39 | Katty | it's a brilliant idea i never thought of...and that heavy duty fishing line can easily support a squirrel i'm sure |
| 14:49.55 | *** join/#asterisk morphious (~morphium@93.186.202.3) |
| 14:50.02 | morphious | Hi |
| 14:50.15 | *** join/#asterisk mallchin (~mallchin@mail.dataproservices.co.uk) |
| 14:50.26 | mallchin | hi guys, can anyone tell me what this error means please: |
| 14:50.27 | mallchin | translate.c: plc_samples 160 format f |
| 14:50.39 | mallchin | [Jun 24 15:45:12] WARNING[25754] translate.c: plc_samples 160 format f |
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| 14:52.05 | morphious | Hi, seems like I don't understand the exten syntax. I did exten => _X.,1,Noop(foobar) - that works fine. for _X.,2,Noop... 3 and 4 this still works. but for _X.,5,Noop... my calls won't go through anymore... what am I doing wrong? |
| 14:54.57 | *** join/#asterisk mallchin (~mallchin@mail.dataproservices.co.uk) |
| 14:55.17 | mallchin | hi guys, sorry, someone rebooted me :-/ |
| 14:55.35 | mallchin | could anyone tell me what the following warnings mean please |
| 14:55.41 | mallchin | [Jun 24 15:53:39] WARNING[25968] utils.c: trying to reset empty pool |
| 14:55.46 | mallchin | [Jun 24 15:53:39] WARNING[25968] translate.c: plc_samples 160 format f |
| 14:55.51 | mallchin | thanks :) |
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| 15:01.25 | pabelanger | mallchin: Unless you are experiencing problems, you can ignore them |
| 15:01.55 | mallchin | pabelanger: great, thanks |
| 15:02.14 | mbrevda | pabelanger: ping |
| 15:03.21 | structz | Question: I Have 4 Servers "interconnected" via DUNDi each one anounces their numbers but some of then are in the same code area, i need to make another announce only for those in the same code area to call each other without using TELECOM OPERATORS!! |
| 15:04.33 | *** part/#asterisk mifadir (~ifadir@dynamic.casap1-189-30-137-41.wanamaroc.com) |
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| 15:05.50 | *** join/#asterisk wcselby (~wcselby@208.180.112.123) |
| 15:05.52 | wcselby | o/ |
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| 15:08.57 | structz | wcselby, heya |
| 15:08.59 | structz | all good? |
| 15:09.30 | wcselby | heya structz |
| 15:09.34 | wcselby | good enough |
| 15:09.50 | structz | :D |
| 15:10.29 | wcselby | i could do without all the oil spewing out in the gulf, but hey, it hasn't washed ashore near me yet |
| 15:10.37 | *** join/#asterisk nix8n82 (~nate@63.162.27.14) |
| 15:11.19 | wcselby | the moratorium could potentially put a lot of people I know out of work though... |
| 15:11.27 | wcselby | but otherwise, no big asterisk issues to report :) |
| 15:11.29 | *** join/#asterisk ruied (~ruied@89-180-247-219.net.novis.pt) |
| 15:12.16 | nix8n82 | what moratorium? |
| 15:13.15 | wcselby | aren't they putting a new moratorium in place since their first one got nix'ed by the judge? |
| 15:13.49 | nix8n82 | I came in late on the conversation I was wondering what issue you were talking about. |
| 15:14.21 | wcselby | oh, i was rambling about the oil spill. i live in the gulf coast |
| 15:14.32 | wcselby | no one else was really responding to me |
| 15:14.37 | wcselby | :) |
| 15:17.14 | nix8n82 | I'm quite ignorant of the science when it comes to life, but isn't having millions of gallons of toxic crude in the ocean going to screw over the world on a global scale, especially if we can't contain it? |
| 15:17.36 | Katty | hey wcselby :> |
| 15:17.38 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 15:17.54 | ChannelZ | It's just a stupid political move |
| 15:17.56 | wcselby | well, this one ranks at about number 6 or 7 on the all-time worst list of amount of oil spilled in oceans worldwide over the last 100 years or so |
| 15:18.06 | wcselby | so it's certainly not going to screw over the world |
| 15:18.10 | wcselby | but it's a bad one |
| 15:18.14 | xheliox | nix8n82: sure, but drill baby drill. being able to make the trip to Grandma's house on a cheap tank of gas is far more important than the future of the planet. |
| 15:18.54 | wcselby | and it will be contained by august at the latest, when the relief well is drilled and stablizes the pressure |
| 15:19.06 | wcselby | heya Katty! |
| 15:19.13 | xheliox | wcselby: you can't make a statement like that until they actually stop the leak. we all presume it'll be fixed in August, but add a hurricane into the mix or another technical failure and what if goes on until winter or even next year? |
| 15:19.29 | nix8n82 | What about hurricanes and ocean tides? |
| 15:19.32 | xheliox | and frankly, at this point, do you believe _anything_ BP says? |
| 15:20.25 | xheliox | oh, it's just 1000 barrels a day, oh it's just 5000, maybe it's 15000... oh perhaps 30000, sorry about ur oceans, forgive us plz? Frankly, I hope they choke on their crude. |
| 15:20.28 | nix8n82 | I don't believe anything that anyone in so call "authority" says is anywhere close to 100% accurate |
| 15:21.04 | wcselby | xheliox - following your logic, the world could also end in thermo-global-nuclear war before august, so why should we be worried about whether they can stop the leak by then? |
| 15:22.20 | wcselby | xheliox - also, your glass is apparently half-empty |
| 15:22.24 | wcselby | :) |
| 15:22.59 | xheliox | Yeah, because hurricanes in the Gulf Of Mexico is as common as nuclear war. My point was, you can't rank this disaster until the oil literally stops flowing. If it's August, great, but I certainly no great confidence that will happen at this point. |
| 15:23.31 | xheliox | have no* |
| 15:23.32 | mocker | If only we could use Asterisk to plug the hole. |
| 15:23.48 | *** join/#asterisk italorossi (~italoross@189.23.15.2) |
| 15:23.49 | ChannelZ | FreePBX might be able to |
| 15:23.50 | wcselby | i've got some servers we could throw down there |
| 15:23.53 | ChannelZ | it's a big enough turd |
| 15:23.56 | xheliox | we could round up all the x100p generics and shove them down the hole.. |
| 15:24.02 | wcselby | ChannelZ - I was going there........lol |
| 15:24.15 | wcselby | i was going to mention the bloat, but meh |
| 15:24.16 | wcselby | :) |
| 15:24.20 | wcselby | you beat me to it |
| 15:24.21 | ChannelZ | :) |
| 15:24.41 | xheliox | Plug it with FreePBX, aren't the oceans polluted enough, ffs? |
| 15:25.12 | ChannelZ | My personal choice would be to shove Nancy Pelosi in there, what with being made of all that nice stretchy plastic and all, but alas I must scamper off to work. |
| 15:25.29 | wcselby | i still say the moratorium is a bad idea, it's going to ruin a large percentage of the gulf coast economy |
| 15:25.55 | beek | wcselby: And this oil slick isn't? |
| 15:26.00 | wcselby | and considering that's where I live, I have a vested interest in not destroying that economy |
| 15:26.23 | wcselby | beek - not as much as a moratorium would |
| 15:26.58 | wcselby | beek - the situation is already bad, why would we go and make it worse? |
| 15:27.09 | xheliox | imho, we should have a moratorium and make BP pay all the displaced oil workers. |
| 15:27.11 | beek | wcselby: It seems to me that the industry needs to spend some time figuring out how to fix these problems when they occur before we allow them to make new messes. |
| 15:28.05 | beek | BP certainly has a nice test case on its hands with which to do R&D in that area. |
| 15:28.59 | wcselby | beek - so the 30-ish rigs that have been out there drilling deep water for years and have never had an accident, we need to stop those guys because of this one problem. plus, all hundreds (I think) of other rigs out there, not drilling deep water, they should be stopped too. |
| 15:31.00 | wcselby | and while we're at it, lets go ahead and force the loss of another 200,000 or so jobs |
| 15:31.21 | coppice | wcselby: no. what you need is a disaster plan, and the proper equipment in place ready |
| 15:32.04 | wcselby | coppice - you mean like the one the government worked out with the oil companies, but then never funded to get the proper equipment in place? |
| 15:32.27 | wcselby | or maybe the plan to burn off the oil that the oil companies took to the government, but the government told them no they couldn't do that? |
| 15:32.32 | coppice | wcselby: until 63 days ago a CEO who financed a disaster plan would have been thrown out for wasting valuable shareholder resources. in the last 63 days he would have been praised by the shareholders as a hero |
| 15:32.39 | mocker | Thank god we have Kevin Costner |
| 15:32.54 | beek | mocker: :D |
| 15:33.02 | wcselby | lol :) |
| 15:33.03 | coppice | mocker: can you plug the hole with him, then? |
| 15:33.20 | wcselby | coppice - the film reel from all of his movies should do the trick |
| 15:33.30 | wcselby | original film reel |
| 15:33.34 | wcselby | only thing that'd work |
| 15:33.36 | wcselby | :) |
| 15:34.02 | coppice | just the bad ones would be enough.... of, that's the same thing, isn't it |
| 15:34.16 | mocker | . o O ( Water World ) |
| 15:34.17 | wcselby | like i said.......lol :) |
| 15:34.46 | *** join/#asterisk s0lid (~s0lid@121.96.69.36) |
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| 15:43.18 | *** join/#asterisk chai_sangeen (~chai_sang@89.148.26.253) |
| 15:43.25 | wcselby | yay me, I get to parse cdr records now |
| 15:43.45 | mocker | wcselby: awk! |
| 15:51.39 | mallchin | what do I do if I want to pass a string to a macro that contains commas? the commas seem to break the string into different arguments to the macro |
| 15:52.09 | mocker | mallchin: Have you tried escaping them? |
| 15:52.58 | mallchin | mocker: yes, it would appear to work, but the esacpe sequence appears in the aruments inside the macro |
| 15:53.31 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 15:54.46 | mallchin | mocker: I would have though the macro would parse the escape sequence causing it to ignore the comma, but then strip the escape sequence from the string, but it simply passes it along |
| 15:55.23 | *** part/#asterisk geemee (~ocs@mailhost.exterity.com) |
| 15:55.30 | wcselby | mallchin - you could escape it, then use a ${FILTER()} function inside the macro to remove the escape characters maybe? |
| 15:56.07 | mallchin | wcselby: thanks for the suggestion, but, there must be a better way? |
| 15:56.33 | mallchin | or, is that the only way? |
| 15:56.42 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 15:58.43 | mallchin | oh well, hometime, another day :) |
| 15:58.46 | mallchin | bye for now |
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| 16:07.51 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 16:09.20 | Jumpie | . |
| 16:09.40 | wcselby | .. |
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| 16:22.06 | thehar | anyone worked with Mitel + asterisk sip peering? |
| 16:25.51 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 16:25.53 | troffasky | what's your question? |
| 16:33.28 | *** join/#asterisk Bladerunner05 (~macbookai@81-174-56-54.static.ngi.it) |
| 16:33.30 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:34.51 | Bladerunner05 | hello on asterisk 1.4.33 I configure asterisk-addons to use mysql for cdr, loaded module, configured cdr.conf but no entries in table cdr.. |
| 16:35.15 | *** join/#asterisk MrTelephone (~test@h697179-171.picriverisp.net) |
| 16:36.25 | MrTelephone | Can someone tell me how you get an ADIT 600 with fxs cards to work with a cisco AS5350? I have a ds0-channel group of 1-24 channels but there is no option for fxo signalling. Only fxs signalling. Is it possible to run a channel bank off a AS5350? |
| 16:37.50 | *** join/#asterisk jsgoecke (~Adium@c-98-207-5-162.hsd1.ca.comcast.net) |
| 16:37.54 | jsgoecke | http://labs.voxeo.com/2010/06/24/new-adhearsion-version-0-8-4-released/ |
| 16:37.59 | jsgoecke | Good morning everyone |
| 16:39.42 | *** join/#asterisk nightwalk (~nightwalk@daimon.vixel.org) |
| 16:41.39 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
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| 16:42.38 | Godfather_ | hi |
| 16:43.20 | Godfather_ | how can i enable the software echo canceller on my tdm410p? http://pastebin.com/RSNqTiNT |
| 16:43.51 | Godfather_ | it says Echo cancellation: Currently off |
| 16:45.16 | Tim_Toady | Godfather_ you add 'echocancel=yes' in ur chan_dahdi.conf |
| 16:45.17 | tzafrir | Currently off: no active echo cancellation at the moment |
| 16:45.22 | Tim_Toady | for each channel |
| 16:45.27 | tzafrir | Does it say if there's one configured? |
| 16:46.19 | Godfather_ | Tim_Toady, yes, i have it in my chan_dahdi.conf |
| 16:46.32 | Tim_Toady | then it sould be ok |
| 16:46.38 | *** join/#asterisk guilhermebr (~Guilherme@ns2.aser.com.br) |
| 16:46.44 | Godfather_ | tzafrir, ok, i will make a phone call to see it again |
| 16:46.48 | Godfather_ | thx |
| 16:47.46 | mweichert | Hi, I'm trying to receive SMS on IAX2 trunk. In my trunk I have "requirecalltoken=no" but everytime an SMS comes in I get: "chan_iax2.c: Call rejected, CallToken Support required. If unexpected, resolve by placing address 174.137.63.202 in the calltokenoptional list or setting user voipms requirecalltoken=no" |
| 16:51.54 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 16:57.49 | *** join/#asterisk s0lid (~s0lid@222.127.250.7) |
| 16:59.29 | Godfather_ | Tim_Toady, i can't test it, it says me channel congestion... :-? |
| 16:59.32 | Godfather_ | http://pastebin.com/BDMnTuCg |
| 17:00.25 | Tim_Toady | Godfather_ paste ur chan_dahdi.conf |
| 17:01.12 | Godfather_ | ok |
| 17:03.52 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 17:03.52 | *** join/#asterisk grinder13 (~grinder@146.176.165.57) |
| 17:04.15 | Godfather_ | Tim_Toady, http://pastebin.com/C1Vjgn8q |
| 17:04.35 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 17:04.50 | Godfather_ | i just put "#include dahdi-channels.conf" and the end of the file |
| 17:05.08 | *** join/#asterisk Draiven (~draiven@190.145.100.60) |
| 17:05.08 | Godfather_ | the other was generated by dahdi_genconf i supose. |
| 17:05.22 | Tim_Toady | then we need dahdi-chnnels.conf |
| 17:06.09 | Godfather_ | Tim_Toady, http://pastebin.com/Pc8QS789 |
| 17:08.57 | Godfather_ | Tim_Toady, |
| 17:09.08 | Godfather_ | I owned myself |
| 17:09.44 | Godfather_ | i changed the fxo connection from one side to another (from the left port to the right port) |
| 17:09.47 | Godfather_ | now is working |
| 17:10.59 | Godfather_ | but echo cancellation no |
| 17:11.00 | Godfather_ | http://pastebin.com/56egpaDs |
| 17:12.12 | Tim_Toady | is the echo canceller module loaded? |
| 17:13.14 | Godfather_ | http://pastebin.com/cVtjJ7UX |
| 17:13.23 | Godfather_ | i've no idea what module i'm searching for |
| 17:13.36 | Godfather_ | app_echo.so? |
| 17:13.41 | Tim_Toady | not in asterisk, the kenrel module |
| 17:13.51 | Tim_Toady | lsmod | grep echo |
| 17:14.04 | Godfather_ | Tim_Toady, nope |
| 17:14.07 | Godfather_ | it isnt |
| 17:14.23 | Tim_Toady | modprobe dahdi_echocan_mg2 |
| 17:14.42 | Godfather_ | not found |
| 17:17.03 | Tim_Toady | ls /lib/modules/`uname -r`/dahdi | grep echo |
| 17:17.04 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 17:17.15 | *** join/#asterisk devdvd (~myemail@173-31-171-48.client.mchsi.com) |
| 17:17.58 | Godfather_ | Tim_Toady, http://pastebin.com/z5vRrzVW |
| 17:18.03 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 17:18.21 | Tim_Toady | dahdi_echocan_mg2.ko |
| 17:18.24 | Tim_Toady | there it is |
| 17:18.41 | Godfather_ | insmod ? |
| 17:18.44 | Tim_Toady | run depmod -a and try loading it again |
| 17:18.59 | Tim_Toady | modprobe dahdi_echocan_mg2 |
| 17:19.03 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:19.33 | Godfather_ | lol |
| 17:19.47 | *** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn190.78-98-189.t-com.sk) |
| 17:19.57 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 17:20.07 | *** join/#asterisk fred1_ (~djc@65.209.147.101) |
| 17:20.34 | Godfather_ | http://pastebin.com/XdETE0y8 |
| 17:20.45 | Godfather_ | i test it, one sec. |
| 17:21.31 | structz | :D Got it!! |
| 17:22.02 | Godfather_ | Tim_Toady, works :) |
| 17:22.23 | Godfather_ | mg2 is the best on software method? |
| 17:22.30 | Godfather_ | should i use another? |
| 17:23.02 | Tim_Toady | its ok.... oslec is maybe better but in ur case it will be hard to install |
| 17:23.35 | tzafrir | hmm... something is wrong if you have to explicitly load the EC module |
| 17:23.58 | tzafrir | It should be modprobed automatically (by the kernel) when you run dahd_cfg |
| 17:24.20 | Godfather_ | tzafrir, i didnt run dadh_cfg |
| 17:24.40 | Tim_Toady | tzafrir there was a prob with his kernel modules depedencies |
| 17:24.51 | Godfather_ | i just run dahdi_genconf and modified dahdi-channels.. |
| 17:24.54 | Tim_Toady | depmod -a seems to fix it |
| 17:25.19 | Tim_Toady | prior to this he wasnt able to load it even manually |
| 17:26.27 | *** join/#asterisk BANSAL (~bansal@117.207.81.12) |
| 17:26.32 | Godfather_ | Tim_Toady, thx |
| 17:35.05 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.219.69.dsl.dyn.forthnet.gr) |
| 17:40.28 | Pimmetje | P!MMetje |
| 17:40.39 | Qwell | nice password |
| 17:40.44 | Tim_Toady | lol |
| 17:41.58 | MrTelephone | If you have a FXS channel bank using CAS singalling why do you specify fxo in zapata.conf? |
| 17:43.37 | MrTelephone | ASTERISK <-> T1 CAS (FXO) <-> ADIT 600 (FXS) <-> Handsets |
| 17:45.57 | *** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 17:47.53 | *** join/#asterisk cesar_CR (~cesar@201.196.220.82) |
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| 17:58.25 | *** join/#asterisk exothermc (~miles@74.85.89.146) |
| 18:02.28 | *** join/#asterisk italorossi (~italoross@201.76.154.130.intranet.digi.com.br) |
| 18:03.45 | *** join/#asterisk bsaxon_ (~bsaxon@12.107.149.61) |
| 18:04.13 | exothermc | when dialing on a PRI how do you send DTMFs after the call sets up? I have to input a code for LD is why. Can that be part of the dial string? |
| 18:06.50 | exothermc | I tried the D(digits) option on the dial string but it doesn't appear to do anything. |
| 18:07.30 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 18:08.09 | [TK]D-Fender | exothermc: Show us your failed attempt |
| 18:09.27 | rustyclarkson | Is there a variable(or some way) to determine the length of time a caller was on hold in a call? |
| 18:09.32 | exothermc | ahh ok so D() does work, well kinda, it only sends the dtmfs after the call completely sets up. Since asterisk doesn't think the call sets up when it comes time to enter digits (doing it manually works btw) |
| 18:10.28 | exothermc | [TK]D-Fender: Well there is no failure per say yet. I need to dial 11 digits out the dahdi channel, wait a until a tone is played back in media then send another 6 digits. |
| 18:10.55 | exothermc | [TK]D-Fender: I just don't know how to accomplish this out dahdi |
| 18:10.56 | [TK]D-Fender | exothermc: When means "just wait", not "wait for" |
| 18:11.14 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 18:12.10 | exothermc | [TK]D-Fender: I'm sorry but that last post didn't make any sense to me. |
| 18:12.24 | *** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 18:15.11 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-159-189.dsl.stlsmo.sbcglobal.net) |
| 18:15.22 | LemensTS | asterisk doesnt convert flv to wav does it? |
| 18:15.24 | LemensTS | (file convert) |
| 18:16.30 | Qwell | no.. |
| 18:16.49 | rustyclarkson | asterisk isn't a media converter |
| 18:17.43 | jsgoecke | rustclarkson it does transcoding all the time, no? |
| 18:17.49 | jsgoecke | rustyclakrson that is |
| 18:17.54 | rustyclarkson | :p |
| 18:17.59 | rustyclarkson | yes it can |
| 18:18.00 | drmessano | or rust <tab> |
| 18:18.44 | rustyclarkson | i just kinda think of flv as being a very specific format for asterisk to support |
| 18:19.02 | jsgoecke | Ah, I see |
| 18:19.29 | [TK]D-Fender | * is not a Youtube processing engine. |
| 18:19.51 | rustyclarkson | ^^ that's probably what I should have said |
| 18:19.57 | jsgoecke | Yes, much better vehicles for that |
| 18:20.03 | exothermc | Does anyone know how to, using the dial command, send an LD access code down a PRI controlled by dahdi after the 11 digits have been passed? |
| 18:21.15 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |
| 18:25.10 | *** part/#asterisk LemensTS (~LemensTS@adsl-70-238-159-189.dsl.stlsmo.sbcglobal.net) |
| 18:25.12 | [TK]D-Fender | exothermc: D() |
| 18:26.25 | rustyclarkson | exothermc: I think you're best of determining why asterisk doesn't think the call is set up when it comes time to enter digits, otherwise you're somehow going to have to code a DTMF delay |
| 18:26.41 | exothermc | [TK]D-Fender: Ya that doesn't work. We are expected to enter the code before the PRI returns that the call is setup. D() appears to only work once the call gets setup (aka answers on the other end, after ringing) |
| 18:26.54 | *** join/#asterisk cesar_CR (~cesar@201.196.220.82) |
| 18:27.24 | [TK]D-Fender | exothermc: If you're in "early media" mode... you might be screwed |
| 18:27.41 | exothermc | [TK]D-Fender: damn that is what I was thinking. |
| 18:28.25 | exothermc | [TK]D-Fender: Seem odd that D() waits until answer before sending. Kinda defeats the whole purpose of putting that in the dial command when you could just do it in the next priority. |
| 18:29.05 | [TK]D-Fender | exothermc: Pastebinthe complete call with PRI debug enabled |
| 18:29.24 | exothermc | [TK]D-Fender: ok |
| 18:30.03 | *** join/#asterisk tzanger (~tzanger@gromit.mixdown.ca) |
| 18:30.06 | tzanger | good afternoon |
| 18:30.20 | tzanger | Is DAHDI device state not in 1.4.x at all? |
| 18:30.30 | tzanger | is it a 1.6 thing only? |
| 18:33.50 | [TK]D-Fender | tzanger: Pardon? |
| 18:35.15 | tzanger | heh |
| 18:35.20 | tzanger | the device state (hinting) |
| 18:35.24 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 18:35.26 | tzanger | SIP/IAX/Local work |
| 18:35.35 | exothermc | [TK]D-Fender: http://pastebin.com/HE24SJ5Y |
| 18:35.35 | tzanger | I thought Zaptel/DAHDI were in 1.4 too in the later ones |
| 18:35.41 | exothermc | [TK]D-Fender: That is two calls |
| 18:35.46 | tzanger | but looking at the 1.4.33.1 changelog I don't see any mention of it |
| 18:36.33 | exothermc | The first I don't actually do anything except wait which ends with the carrier saying "You did not dial enough digits to complete your call" The second call I entered the pack code manually so you could see the D() executing late. |
| 18:38.20 | [TK]D-Fender | exothermc: Yeah I'm strongly suspecting the "receive ready" is early media indication... don't know how * can handle that. |
| 18:38.37 | exothermc | [TK]D-Fender: damn |
| 18:39.10 | [TK]D-Fender | exothermc: Get googleing |
| 18:39.16 | WIMPy | RR is nothing. |
| 18:39.17 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 18:40.14 | exothermc | [TK]D-Fender: ya been doing that, no luck yet. |
| 18:44.19 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 18:47.34 | *** join/#asterisk rare1980_ (~as@116.71.177.224) |
| 18:48.05 | rare1980_ | i m hving issue in configuring inbound trunk on my asterisk box |
| 18:48.45 | rare1980_ | exten => 12122310647,1,Goto(test,cc100,1) |
| 18:48.45 | rare1980_ | exten => 12122310647,2,Hangup |
| 18:48.54 | rare1980_ | i ve wrote this in extension.conf |
| 18:49.00 | rare1980_ | but call is not comming in |
| 18:49.29 | ChannelZ | the console tells all. What is the call coming in as instead? |
| 18:51.01 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 18:53.37 | [TK]D-Fender | rare1980_: Your 2nd priority should never get used even if a call DOES land on that exten |
| 18:53.55 | [TK]D-Fender | rare1980_: and CONTEXT matters. Who cares if the exten is the right pattern if it isn't in the right PLACE? |
| 18:55.03 | rare1980_ | [tk]D-fender: please can you write correct syntax |
| 18:56.52 | [TK]D-Fender | rare1980_: What part of "look at the CONTEXT the call is arriving in" are you not understanding? |
| 18:57.32 | [TK]D-Fender | rare1980_: Doesn't matter how well I describe the item I want you to find if I point you in the WRONG PLACE to look for it. |
| 18:57.53 | [TK]D-Fender | rare1980_: Now open you eyes and look at the CALL <- |
| 18:58.23 | rare1980_ | let me c |
| 19:00.26 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 19:06.16 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 19:18.45 | *** join/#asterisk DelphiWorld (~Delphi@41.200.7.26) |
| 19:18.49 | DelphiWorld | hi all! |
| 19:18.59 | *** part/#asterisk Delido1983 (~Delido198@212.144.236.196) |
| 19:19.10 | DelphiWorld | how do i reset my grandstream gxp1200 to default configuration? |
| 19:19.57 | grinder13 | hello! I 've installed Asterisk from SVN. I 'm experimenting with calls using SIP clients like sip-communicator and SFLphone and such. things work fine, apart from the speex codec. The error I am getting is this: "No compatible codecs, not accepting this offer" Obviously, I 've compiled Asterisk with the speex codec being enabled in the menuselect. What am I doing wrong? any ideas? |
| 19:22.33 | [TK]D-Fender | grinder13: Perhaps you aren't allowing it. Maybe you should look at the CALL SIP DEBUG and see what is actually being offered |
| 19:23.12 | grinder13 | the thing is that I am allowing the speex codec in sip.conf |
| 19:23.18 | grinder13 | let me check though |
| 19:24.03 | [TK]D-Fender | grinder13: You THINK you are. But thta isn't actually "looking" |
| 19:28.20 | grinder13 | hmm, let me see. the allow=speex can be either in the [general] context or in my [user] context, right? |
| 19:29.22 | [TK]D-Fender | grinder13: Look at the CALL <----------------------------------------- |
| 19:30.33 | grinder13 | initially, I had in my user's context. when I put it in the [general] context it worked fine |
| 19:30.47 | grinder13 | thanx |
| 19:31.23 | [TK]D-Fender | grinder13: That points to a more serious problem. |
| 19:31.25 | [TK]D-Fender | grinder13: Look at the CALL <----------------------------------------- |
| 19:31.37 | grinder13 | ok |
| 19:33.02 | *** part/#asterisk DelphiWorld (~Delphi@41.200.7.26) |
| 19:40.03 | rustyclarkson | Is there a variable(or some way) to determine the length of time a caller was on hold in a call? I know there is for Queues, but what about outside of a Queue? |
| 19:40.21 | *** join/#asterisk mifadir (29891ea3@gateway/web/freenode/ip.41.137.30.163) |
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| 19:48.31 | *** join/#asterisk lukhas (~lucas@bearstech/lukhas) |
| 19:48.39 | lukhas | hello |
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| 20:01.16 | timeshell | uhlo |
| 20:01.24 | lirakis | uhlo |
| 20:03.44 | lukhas | is there a way to make Asterisk play MP3 files depending on phone keypresses? |
| 20:04.13 | jblack | yes. That's very easy. Read up in the book about making dialplans to do it |
| 20:04.18 | lukhas | the soundfiles should be configured for every extension calling |
| 20:04.45 | lukhas | and I'm not sure I expressed my needs very well :) |
| 20:05.30 | jblack | yes. Again, you'll want to read the section about writing dialplans; there are variables that will be of interest to you |
| 20:05.38 | lukhas | ok |
| 20:07.33 | pabelanger | ~book |
| 20:07.34 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:07.37 | pabelanger | lukhas: ^^ |
| 20:07.48 | lukhas | ah, this book |
| 20:08.00 | lukhas | Google sent me to http://www.the-asterisk-book.com/unstable/ |
| 20:08.24 | lukhas | thanks :) |
| 20:09.00 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 20:11.24 | *** join/#asterisk jerome` (~jerome@atom.dedwen.info) |
| 20:17.52 | lukhas | hm, Playback() refuses to play my random MP3, presumably I'm missing some codec |
| 20:23.31 | *** join/#asterisk vader-- (~me@c-71-225-201-226.hsd1.nj.comcast.net) |
| 20:27.09 | [TK]D-Fender | lukhas: You need asterisk-addons for MP3 support |
| 20:27.22 | [TK]D-Fender | lukhas: Assuming you're actually calling Playback() right in the first place |
| 20:28.16 | lukhas | it plays ogg files with no problem |
| 20:28.19 | lukhas | in the same folder |
| 20:29.39 | lukhas | I'll test my stuff with ogg then, and see later for the addons :) |
| 20:34.11 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 20:35.05 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 20:37.46 | lukhas | ok, found out how to express my needs in English: during a conversation, I need the caller to be able to play sound files, these sound files can change for each call and for each caller |
| 20:38.16 | *** join/#asterisk tecnico (~tecnico@75.76.169.148) |
| 20:38.19 | lukhas | what I was thinking was to store the list of soundfiles in asterisk's database (via a command-line call, for instance) |
| 20:38.43 | lukhas | and then when the caller presses key 1 on his phone, play the soundfile number 1, etc. |
| 20:38.46 | [TK]D-Fender | lukhas: Why would there be sound playing in the background of a conversation? |
| 20:39.02 | lukhas | [TK]D-Fender: the call needs to have the callee listen to a soundfile |
| 20:39.09 | lukhas | music, advertisement, etc. |
| 20:39.20 | [TK]D-Fender | lukhas: Ads while you're talking to a PERSON?! |
| 20:39.46 | [TK]D-Fender | ": during a conversation, I need the caller t" <-- means while talking to another PERSON |
| 20:39.50 | lukhas | more like "here's an ad we want to test <plays>. did you like it?" |
| 20:40.15 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 20:40.18 | [TK]D-Fender | lukhas: Sounds like you want this to be a menu where the caller does the interaction and there is no other PERSON involved |
| 20:40.26 | boodu | hello |
| 20:40.31 | lukhas | there needs to be another person involved |
| 20:40.44 | lukhas | it's for polls |
| 20:40.48 | [TK]D-Fender | ........ |
| 20:41.03 | [TK]D-Fender | lukhas: Describe a COMPLETE call scenario |
| 20:41.44 | lukhas | the caller calls, asks few questions, plays a soundfile, that is most likely to be an ad, but may be just a music excerpt, then asks few other questions, then hangs up, or play another soundfile, etc. |
| 20:42.07 | lukhas | (please bear with me, I just understood the French version of the requirement 20 minutes ago) |
| 20:44.59 | [TK]D-Fender | lukhas: This is very odd to be speaking to someone and playing back ads interactive like that. |
| 20:45.06 | lukhas | indeed |
| 20:45.28 | lukhas | it's for a company doing phone polls |
| 20:45.35 | [TK]D-Fender | lukhas: You could make a more complex AGI you call via features.conf dynamic feature. Messy & complex, but possible |
| 20:45.42 | lukhas | mostly for marketing campaigns |
| 20:45.47 | pabelanger | you almost need to transfer your caller into a queue, play message, then asterisk calls back the original agent |
| 20:46.00 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 20:46.00 | lukhas | and oddly enough, some people actually answer to those polls :) |
| 20:46.08 | [TK]D-Fender | pabelanger: No, the caller seems to have to be in control to play another on demand |
| 20:46.27 | lukhas | yes, the caller needs to have control of the playback |
| 20:46.38 | lukhas | like stopping it before the end |
| 20:47.52 | pabelanger | then, place both caller and callee into a queue, when caller presses DTMF, jump out to context, place message, then reconnect into queue |
| 20:47.57 | [TK]D-Fender | lukhas: You just made it that much harder with that statement |
| 20:48.07 | lukhas | yeah, I was fearing that |
| 20:48.18 | lukhas | I thought it was complex, but not that hard |
| 20:48.22 | rustyclarkson | could features.conf somehow work for a connected call? |
| 20:49.01 | pabelanger | lukhas: modify a n-way conference for your needs |
| 20:49.03 | pabelanger | http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO |
| 20:49.30 | lukhas | and the fun part is that soundfiles can be different for every call, so I also need a way to change them dynamically |
| 20:49.53 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 20:49.55 | lukhas | or rather, for every call and every caller |
| 20:50.46 | ChannelZ | What number will they be calling from so I can pre-blacklist them? |
| 20:50.57 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-73-235.home.otenet.gr) |
| 20:51.46 | lukhas | :) |
| 20:52.17 | lukhas | I'm not sure I like the idea myself, but the technical challenge is, well, challenging |
| 20:52.58 | lukhas | pabelanger: having a look, thanks |
| 20:54.23 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 20:55.19 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 20:55.32 | [sr] | hi fellow's! |
| 20:56.19 | [sr] | it's almost monday again to start a new free week of work |
| 20:56.20 | [sr] | :P |
| 20:57.25 | lukhas | pabelanger: yeah, that could do the trick with some adaptations |
| 20:57.54 | lukhas | I'll re-read that page after a good night's sleep to be sure :) |
| 20:57.54 | pabelanger | lukhas: good luck |
| 20:57.58 | lukhas | yeah, thanks |
| 20:58.23 | lukhas | I was planning on something much easier, to be honest, this is starting to look like my .emacs |
| 20:58.43 | lukhas | anyway, thanks for the help, I'm off to bed now :) |
| 20:58.45 | [sr] | is in-lobe with VOIP |
| 21:01.20 | [sr] | WIMPy: you're the responsible :P |
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| 21:14.29 | p3nguin | wonders if he should also go to that magical in-lobe place |
| 21:15.54 | [TK]D-Fender | lukhas: Your best bet may be MeetMe + background IVR |
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| 21:28.09 | [sr] | p3nguin: lol |
| 21:28.55 | aphexer | anyone can help me with this?: before answering the call, i'd like to call the respondent, and give him some information through text to speech, after that, the caller is to be connected with the respondent... How can I do that? |
| 21:30.20 | p3nguin | Sounds like basic dialplan stuff to me. |
| 21:32.22 | [TK]D-Fender | aphexer: "core show application dial" <- |
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| 21:38.32 | aphexer | [TK]D-Fender: are you referring to the A(x) option? |
| 21:38.42 | aphexer | to play an announcement to the called party? |
| 21:39.15 | aphexer | p3nguin: I don't really consider it to be basic, you may ;) I have my dial plan which is working, I just want to give some information to the called party before connecting |
| 21:39.19 | aphexer | not sure how |
| 21:42.17 | [TK]D-Fender | aphexer: No, not A(). Unless you plan on building the TTS before the caller answers (would be efficient anyway actually). |
| 21:43.21 | *** join/#asterisk ccesario (~ccesario@189-29-61-213-ac.cpe.vivax.com.br) |
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| 21:44.36 | aphexer | [TK]D-Fender: well, I was hoping to get the data out of a database and then use text-to-speech. Not sure how easy that will be to implement. If not A(), then what option are you referring to? |
| 21:44.57 | aphexer | I was thinking that maybe Originate() could help me |
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| 21:46.15 | p3nguin | That's for making calls from the CLI. |
| 21:46.59 | p3nguin | I don't think there is an Originate() dialplan application. |
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| 21:47.53 | [TK]D-Fender | aphexer: Do it BEFORE calling the target. have that TTS output ready and waiting in a file and then pass that in A() |
| 21:48.06 | [TK]D-Fender | aphexer: Or use another more functional dial option. |
| 21:51.10 | aphexer | p3nguin: ah only CLI, ok, that explains some confusion :) |
| 21:51.47 | aphexer | [TK]D-Fender: ok, I'll try to the first option first, to have the TTS output ready in a file and use A(). But do you have another way in mind, what other more functional option are you referring to? |
| 21:53.06 | [TK]D-Fender | aphexer: Perhaps you should go back and read those DIAL OPTIONS again. |
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| 21:56.26 | aphexer | [TK]D-Fender: oh, you mean M()? |
| 21:56.38 | [TK]D-Fender | aphexer: Was it that obvious? :) |
| 21:57.07 | aphexer | Well... no ;) Still quite new to asterisk, don't know about all those options with much terminology I don't know yet ;) |
| 21:57.15 | aphexer | thanks for helping |
| 21:57.23 | aphexer | I'll read more about M() and A() |
| 21:58.42 | [TK]D-Fender | aphexer: M() clearly lets you do a lot more. A() is a lot cleaner and simpler for certain things though. |
| 21:59.00 | [TK]D-Fender | aphexer: If you can pre-build the audio you want to play A() will do it |
| 21:59.09 | [TK]D-Fender | aphexer: Otherwise M() should be very simple as well |
| 21:59.17 | [TK]D-Fender | aphexer: I'd bet on the M() approach personally. |
| 22:00.10 | aphexer | ok thanks :) |
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| 22:01.38 | rustyclarkson | In a queue, after a caller gets connected to a queue member, I would like to set the MOH to be played if the queue member puts the caller on hold. I tried the Macro in the Queue() function to Set(CHANNEL(musicclass)=...), however, this seems to only apply when the caller puts the queue member on hold, not the other way around... What are my options for doing what I want? Will sip.conf work on inbound calls? Thanks. |
| 22:02.49 | [TK]D-Fender | rustyclarkson: Set it before sending them into the queue |
| 22:03.04 | [TK]D-Fender | rustyclarkson: Or perhals the MoH class specified in the queue itself overrides this |
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| 22:16.47 | p3nguin | Why the hell should the caller be able to put the agent on hold? |
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| 22:30.19 | rustyclarkson | [TK]D-Fender: apparently i screwed up my initial testing and the macro does set the correct channel MOH |
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| 22:38.46 | joobie | hey guys.. got a recording for an IVR that im trying to cleanup the background noise in.. it's got like a constant fuzz .. i tried playing with "noise reduction" features where you select just the noise for the sample profile and then apply that noise reduction across the whole audio.. worked ok, but the voice ends up being a bit robotic after.. any idea how to best resolve this or howt o best approach this? |
| 22:43.28 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 22:43.30 | a1fa | yellow |
| 22:43.43 | a1fa | what do you guys think about Cisco SPA800 ? |
| 22:43.48 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:45.19 | a1fa | I have a customer who wants to keep his Analog PBX |
| 22:45.24 | a1fa | and switch to voip |
| 22:45.30 | a1fa | :X |
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| 22:52.25 | a1fa | Who do you guys recommend for bulk minutes? |
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| 23:03.15 | Godfather_ | i'm using cdr_mysql. When i call using DAHDI "disposition" always take "ANSWER", can i solve it? |
| 23:05.12 | [TK]D-Fender | Godfather_: Analog channels are considered answered the moment they are placed by default. You can change that by doing "callprogress=yes" but you risk your calls dropping at random. |
| 23:06.15 | Godfather_ | [TK]D-Fender, ok, thx |
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| 23:23.17 | daog | hello all, few week ago i bought a tdm410p card with 4 fxo port after configure i found a problem for the outgoing call. the problem is that always show that the call was answered |
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| 23:24.16 | daog | anyone know how i can fix it becouse the cdr show me wrong billsec value |
| 23:25.24 | [TK]D-Fender | [19:05]<[TK]D-Fender>Godfather_: Analog channels are considered answered the moment they are placed by default. You can change that by doing "callprogress=yes" but you risk your calls dropping at random. |
| 23:25.27 | [TK]D-Fender | daog: ^^^^^ |
| 23:25.34 | Godfather_ | loool |
| 23:25.37 | Godfather_ | :-) |
| 23:26.26 | Godfather_ | [TK]D-Fender, i'm trying to use oslec with dahdi |
| 23:26.38 | Godfather_ | [Jun 25 01:23:26] WARNING[12679]: chan_dahdi.c:2005 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) |
| 23:26.48 | Godfather_ | any ideas? |
| 23:26.56 | Godfather_ | here is some output http://pastebin.com/Y5XQk8Rp |
| 23:27.29 | ryanlin | all |
| 23:27.45 | ryanlin | our asterisk phones can dial out |
| 23:27.51 | ryanlin | but we can't dial in |
| 23:27.59 | ryanlin | it says sip channel unreachable cause code 20 |
| 23:28.06 | [TK]D-Fender | Godfather_: No idea |
| 23:28.06 | ryanlin | oops |
| 23:28.11 | ryanlin | can not create a sip channel |
| 23:28.27 | ryanlin | anyone? |
| 23:28.40 | Godfather_ | ok |
| 23:28.44 | [TK]D-Fender | ryanlin: PASTEBIN a failed call with SIP DEBUG enabled <- |
| 23:28.45 | [TK]D-Fender | ~pb |
| 23:28.46 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 23:28.48 | [TK]D-Fender | ^^^^^^ |
| 23:28.51 | ryanlin | sure one esc |
| 23:28.54 | ryanlin | sec*** |
| 23:30.46 | ryanlin | http://internetworkpro.org/pastebin/6942 |
| 23:30.49 | daog | d-dender y change the callprogress on the chan_dahdi.conf but steel happen the same all time show me answered |
| 23:31.13 | ryanlin | 172.30.253.196 is the callmanager |
| 23:31.44 | ryanlin | let me get the dial peer config on cme |
| 23:32.39 | [TK]D-Fender | ryanlin: NOT what I asked for. |
| 23:33.12 | ryanlin | oh |
| 23:33.13 | ryanlin | the debuhg |
| 23:33.15 | ryanlin | the debug |
| 23:35.44 | ryanlin | hold on |
| 23:36.02 | ryanlin | http://internetworkpro.org/pastebin/6943 |
| 23:36.28 | daog | slaps [TK]D-Fender around a bit with a large trout |
| 23:37.07 | [TK]D-Fender | ryanlin: And did you COMPLETELY restart *? |
| 23:37.15 | ryanlin | yep |
| 23:37.33 | ryanlin | do you see anything wrong with the config? |
| 23:37.43 | [TK]D-Fender | daog: Next, don't PM people like that. Its rude. So is most use of DCC period. |
| 23:37.48 | daog | TK : how i do you know how i can fix the problem eith the cdr to show the correct value on the billsec |
| 23:37.52 | ryanlin | the phone can dial out..but not receive any calls |
| 23:38.06 | [TK]D-Fender | daog: [19:37]<[TK]D-Fender>ryanlin: And did you COMPLETELY restart *? <- you rather |
| 23:38.40 | daog | sorry was a mistake i apologyze |
| 23:39.44 | [TK]D-Fender | ryanlin: [Jun 24 16:34:34] WARNING[3762]: chan_sip.c:4956 create_addr: No such host: 3288 <----- do "sip show peer 3288" and include a NEW pastebin where you're at verbose 10 as well |
| 23:39.54 | ryanlin | ok |
| 23:40.57 | ryanlin | http://internetworkpro.org/pastebin/6944 |
| 23:43.01 | *** join/#asterisk srbaker (~srbaker@S010600254b0794e4.cc.shawcable.net) |
| 23:43.12 | srbaker | heya folks. we'd like to deploy some multi-point video conferencing |
| 23:43.23 | srbaker | can someone link me to here i can find information on this? |
| 23:43.31 | russellb | asterisk isn't really going to help you. |
| 23:43.44 | [TK]D-Fender | srbaker: #elsewhere |
| 23:44.03 | srbaker | shame. i thought there was h.264 sip integration |
| 23:44.04 | [TK]D-Fender | srbaker: But not #stelsewhere . They are exclusively medical. |
| 23:44.26 | russellb | srbaker: there is some basic support for video calls, but not conferencing |
| 23:49.35 | daog | any idea in order to fix the billsec value for the false answered call |
| 23:50.06 | ryanlin | any ideas? |
| 23:50.48 | [TK]D-Fender | [19:38]<[TK]D-Fender>daog: [19:37] <[TK]D-Fender> ryanlin: And did you COMPLETELY restart *? <- you rather |
| 23:51.35 | daog | yes i restart the dahdi and asterisk but steel the same thing |
| 23:51.53 | [TK]D-Fender | daog: pastebin your configs and your call attempt |
| 23:53.54 | daog | <PROTECTED> |
| 23:53.55 | daog | <PROTECTED> |
| 23:53.55 | daog | <PROTECTED> |
| 23:53.55 | daog | <PROTECTED> |
| 23:53.55 | daog | <PROTECTED> |
| 23:53.55 | daog | <PROTECTED> |
| 23:53.55 | daog | <PROTECTED> |
| 23:53.56 | daog | <PROTECTED> |
| 23:54.10 | daog | this call wasn't answered |
| 23:54.19 | [TK]D-Fender | daog: PASTEBIN <-------- do NOT flood in here again |
| 23:54.20 | [TK]D-Fender | ~pb |
| 23:54.21 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 23:54.23 | [TK]D-Fender | ^^^^^^ |
| 23:55.42 | [TK]D-Fender | daog: And provide all of what I have requested |
| 23:59.37 | daog | http://pastebin.com/sVNvMuAL |
| 23:59.59 | a1fa | [TK]D-Fender : how configurable is SPA8000? Can I have 1 SIP = 4FXS? |