| 00:01.12 | valueape | Quick question: I'm trying to execute a script when a conference closes down (to compress and ship recorded conferences). I had been using MixMonitor's script execution to run before - but now I want to use CONFBRIDE record_file in my dialplan ("ext-meetme").I'm not sure where to start how to do this. |
| 00:01.58 | valueape | Basically I'd be using System () call - but I'm not sure where I need to put it - for when the conference closes not when a user leaves |
| 00:08.13 | SeRi | slav3_kitten: why it wouldnt work? |
| 00:08.20 | SeRi | is cisco por non standard? |
| 00:08.28 | SeRi | s/por/poe/ |
| 00:08.37 | [TK]D-Fender | It is its own standard |
| 00:08.44 | slav3_kitten | yep |
| 00:08.47 | [TK]D-Fender | Which virtually no other manufacturer cared about |
| 00:08.55 | [TK]D-Fender | it is NOT 802.3af |
| 00:09.03 | slav3_kitten | cisco poe is pre standard. out of it 802.3af was developed |
| 00:09.35 | slav3_kitten | which is why all modern cisco stuff uses 802.3af and can do cisco as well |
| 00:11.11 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
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| 00:27.54 | SeRi | Thats sucks for you |
| 00:28.42 | slav3_kitten | meh |
| 00:36.51 | ghost75 | does anyone know something about a lockfile for voicemail ? |
| 00:43.08 | Kobaz | so how does srtp work if you're trying to do call recording |
| 00:43.23 | Kobaz | according to polycom info, the phone will drop any call that's not srtp end-to-end |
| 00:44.04 | Kobaz | but using asterisk. you have two dialogues open, one from a phone to asterisk, and another from asterisk to another phone |
| 00:44.13 | Kobaz | so asterisk is sitting in the middle, is it the srtp endpoint? |
| 00:45.23 | [TK]D-Fender | Kobaz: * is a B2BUA. There is no such thing as end-to-end. You should know better |
| 00:45.35 | Kobaz | with a reinvite you have end-to-end |
| 00:45.45 | Kobaz | but then you cant record anyway with a reinvite |
| 00:45.50 | Kobaz | ...just making sure |
| 00:45.55 | [TK]D-Fender | If you're recording, you CAN'T reinvite |
| 00:46.00 | Kobaz | yeah |
| 00:46.02 | [TK]D-Fender | The audio HAS to pass through * |
| 00:46.05 | Kobaz | yeah |
| 00:46.13 | Kobaz | but i was saying for argument sake if you weren't recording |
| 00:46.16 | Kobaz | and did a reinvite |
| 00:46.23 | Kobaz | then it would be end to end |
| 00:46.27 | [TK]D-Fender | They might completely renegotiate |
| 00:46.44 | [TK]D-Fender | I could imagine anyway, not having attempted it personally |
| 00:47.36 | Kobaz | so i could have a one-off srtp endpoint sitting out there somewhere and everything else is just rtp |
| 00:47.40 | Kobaz | was what i was getting at |
| 00:51.13 | [TK]D-Fender | yes |
| 00:51.22 | Kobaz | cool cool |
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| 01:16.28 | qakhan | plz let me know cheap DID providers |
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| 01:36.03 | jpsharp | http://bit.ly/Vccbnw\ |
| 01:38.55 | g_r_eek | anyone know a good guide to configure asterisk 11 with real time? |
| 01:39.01 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 01:39.11 | g_r_eek | or is it the same like 1.8 ? |
| 01:43.30 | Kobaz | the same |
| 01:44.42 | g_r_eek | Kobaz: thans |
| 01:44.45 | g_r_eek | thanks |
| 01:46.06 | ghost75 | your /var/spool/asterisk/voicemail folders also have no write rights on group ? |
| 01:52.37 | Kobaz | right writes |
| 01:52.56 | Kobaz | right rite writes |
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| 01:53.57 | ghost75 | why i dont have? |
| 01:56.55 | Kobaz | kite flight writes right fight rite |
| 01:57.11 | ghost75 | shite |
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| 02:00.12 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:06.31 | g_r_eek | on a real time setup i am getting: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available -- do i need to install and add ons ? |
| 02:08.24 | dfgas-cr48 | `hmmmm |
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| 02:44.06 | g_r_eek | fixed it had to install add-on on the menu select |
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| 03:53.18 | infinity_ | does anyone know of a phone number that maps to enum that can be used to test with ? |
| 04:01.58 | ChannelZ | eh? |
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| 04:03.42 | jpsharp | Tried: +43 720 0101011 This number is reachable only via ENUM+SIP. After connect you will hear the word "ENUM" followed by some music. |
| 04:03.47 | jpsharp | ? |
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| 04:08.08 | infinity_ | do any companies support calling via enum? |
| 04:13.56 | drmessano | infinity_, wouldn't that take a public registrar that's accessible worldwide? |
| 04:14.36 | infinity_ | i thought there were companies suporting enum infrastructure. just no one is registered. |
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| 05:52.49 | SeRi | p3nguin: pkg still in transit. |
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| 06:42.20 | ChannelZ | is that a poop reference? |
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| 07:02.09 | dubzilla | i love asterisk |
| 07:02.18 | dubzilla | paired with a database you can do amazing things |
| 07:17.37 | slav3_kitten | where on earth is country code 97 |
| 07:18.15 | slav3_kitten | hmm 972 |
| 07:18.16 | slav3_kitten | isreal |
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| 09:43.32 | ectospasm | slav3_kitten: http://countrycode.org/ |
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| 09:56.07 | J4nus | I would like to configure Kamailio as a SIP registrar and have several trunks with different Asterisk (playing the role of a media gateway). Is somebody able to give me some documentation or some help ? |
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| 11:34.44 | weinerk | Please help, on asterisk 1.4 - in a macro if I do a lengthy operation (a few seconds), then other concurrent calls clober ARG1. |
| 11:34.46 | weinerk | Is that possible? Any ideas on how to address this. |
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| 13:46.58 | slav3_kitten | ectospasm, thanks :D |
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| 14:42.51 | BlackBishop | in ael .. is there any way I can make a variable _XXXXX expression to match both 2 to 5 char ? |
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| 14:50.16 | Kasper__ | hello |
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| 14:53.37 | mathi | hi |
| 14:55.24 | mathi | stupid question but... to configure asterisk you just need to update a few files (extensions.conf, sip.conf, ...). So when do you really need a GUI interface (FreeBBX, Elastix, ...) to manage your server? |
| 14:57.26 | slav3_kitten | never |
| 14:57.46 | slav3_kitten | all those gui projects are kinda dead iirc |
| 14:58.16 | slav3_kitten | the confs can get convoluted an such but that's what syntax hilighting is for |
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| 17:25.02 | sybren | hi there! |
| 17:25.42 | sybren | I'm trying to hook up Asterisk with Google Talk, so that I can do SIP <-> gtalk voice calls. |
| 17:26.30 | sybren | It works just fine, but *only* if I do a video-call from gtalk -> SIP. Then the audio works just fine. When I do an audio-only call gtalk->SIP or SIP->gtalk the call is connected, but both ends are silent. |
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| 17:48.23 | sybren | btw, it seems to work fine when I use GTalk within GMail on my desktop, but not with the GTalk client on my mobile phone (GTalk version 4.1.2 on Android 4.1) |
| 17:51.02 | Invader | So I have thought of a new project and woundered what you guys thought Ras Pi with arduino/GPRS sheild. Install asterisk with open bts |
| 17:53.35 | WIMPy | Where does trhe arduino fit in there? |
| 17:54.44 | Invader | Broadcast a gsm singal For your own gsm network |
| 17:55.31 | jpsharp | You need something more than a GPRS shield. |
| 17:55.59 | Invader | Your not thinking a USRP are you jpsharp ? |
| 17:56.48 | jpsharp | Something along those lines. The GPRS shield doesn't have the smarts to be a base station. |
| 17:56.59 | WIMPy | You need something that will do the radio part. |
| 17:57.47 | WIMPy | The calypso terminal based basestation was put on ice as far as I know. |
| 17:57.48 | Invader | jpsharp WIMPy that is kind of what I was thinking but was not 100% sure. I do have a few "cheap" usrps I can try that just work on UBS |
| 17:58.08 | WIMPy | But the topic might fit better to #osmocom. |
| 17:58.23 | Invader | I am talking to the openbts guys atm |
| 17:58.32 | WIMPy | ok |
| 17:59.05 | jpsharp | Asterisk + openbts has been done lots of times. There's just questions of financing the base station part of it and how legal you want it to be. |
| 17:59.06 | WIMPy | Or you get an old bas station and only use OpenBSC. |
| 17:59.11 | WIMPy | +e |
| 17:59.11 | Invader | I work at a gsm company and the thought of creating an open network its awesome to me |
| 17:59.49 | WIMPy | Want to take over your employer? ;-) |
| 18:00.03 | jpsharp | Spectrum licensing will be your proverbial Achilles heel. Radiowaves don't come cheap. |
| 18:00.08 | Invader | We are a small telco :) cant sorry |
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| 18:00.32 | Invader | jpsharp, yep dont remind me. |
| 18:00.44 | WIMPy | I guess that depends on where you are. |
| 18:01.02 | WIMPy | Doesn't seem to be an issue here. |
| 18:01.07 | Invader | There are unlicensined spectrum but none in the gsm 700-1900 range. |
| 18:01.14 | jpsharp | True. Not much spectrum regulation in outer botswannaland. |
| 18:01.43 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.230) |
| 18:01.52 | greenwolf | has anyone used the g.722 liscense in a production asterisk system? if so how was the quality of the call and is it worth purchasing? |
| 18:02.06 | Invader | the FCC sent us a fine open for 70K for being on the wrong band |
| 18:02.10 | jpsharp | g722? Doesn't need to be licensed. |
| 18:02.11 | WIMPy | G.722 is free |
| 18:02.19 | jpsharp | g729 is good quality, though. |
| 18:02.20 | WIMPy | Did you mean G.729? |
| 18:02.49 | greenwolf | yes sorry thats what i meant |
| 18:02.50 | greenwolf | lol |
| 18:02.54 | WIMPy | There are many versions of G.729 with different quality. |
| 18:03.05 | greenwolf | how is it against actually 64Kbps quality |
| 18:03.27 | greenwolf | well does it sound very close to actual analogue phone qulaity such as ulaw or alaw |
| 18:03.52 | jpsharp | I'm always scared of getting an FCC notice whenever I screw up and transmit wrong on my ham station. |
| 18:04.30 | greenwolf | yea its happened to me...i am a ham btw |
| 18:04.44 | greenwolf | they keep your liscense in a pending decision for a long time it sucks |
| 18:05.08 | greenwolf | ive been pending my liscense for over 2 years now...i dont think they are gunna pull it but i think they do it just to scare you |
| 18:05.46 | jpsharp | They probably lost the case file. |
| 18:06.33 | Invader | Me and the FCC swear at each other a lot. |
| 18:07.32 | Invader | just to get a basestation up and working with the paper work at the fcc is 50k |
| 18:07.33 | greenwolf | i just checked the ULS on FCC website and im still PA decisons |
| 18:07.38 | greenwolf | holy crap its been over 3 years now |
| 18:07.57 | jpsharp | My dad goes round and round with them on a regular basis licensing 220Mhz commercial stuff. |
| 18:08.16 | greenwolf | they wont even update my address or call sign to my new address...crazy it you ask me..gotta be careful on ham radio because theres always someone out there that will snitch on you and send a letter to the FCC about anything u did wrong |
| 18:08.26 | greenwolf | jpsharp: yea pretty kool |
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| 18:15.38 | greenwolf | always be careful of what your broadcasting on ham radio cuz theres always someone listening to rat you out to the FCC |
| 18:15.52 | greenwolf | its pretty sad now adays |
| 18:16.31 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.167.54.124) |
| 18:17.04 | greenwolf | has anyone seen a difference running asterisk on freebsd rather than linux? |
| 18:21.06 | wltjr | is there a way to match 1 or 2 digit extension, with the same expression? |
| 18:22.13 | [TK]D-Fender | wltjr: No |
| 18:22.31 | [TK]D-Fender | wltjr: You can do 1 or LONGER, but not 1 or 2. |
| 18:22.37 | wltjr | [TK]D-Fender: thanks didn't think so, was hoping to not duplicate stuff, but likley have to oh well :) |
| 18:23.22 | WIMPy | You can use Goto(). |
| 18:24.42 | wltjr | WIMPy: I was thinking I might be able to go that route, probably can |
| 18:31.27 | wltjr | goto worked fine less is more :) |
| 18:31.29 | greenwolf | wimpy: can u answer the question i asked |
| 18:32.23 | greenwolf | are their better results when running asterisk on FreeBSD as oppose to linux? |
| 18:32.41 | WIMPy | The bsd one? No. But apart drom drivers for hardware I wouldn't expect any differences. |
| 18:33.45 | greenwolf | so if i have digium cards will the hardware work just as fine with BSD? or are the drivers only for linuxs |
| 18:33.49 | greenwolf | linux* |
| 18:36.27 | WIMPy | I thought there was a BSD version, but I don't see it. So I guess that makes a no. |
| 18:38.33 | greenwolf | so i would have to run a whole VoIP system with asterisk to be able to get asterisk to work on BSD |
| 18:38.46 | greenwolf | with no digium or telephony cards |
| 18:39.54 | WIMPy | You should at least be able to use CAPI, I guess, but I really have no idea if there are any other options available. |
| 18:42.22 | p3nguin | jpsharp, greenwolf: What kinds of radio(s) do you have? |
| 18:43.35 | *** part/#asterisk sybren (~sybren@213.93.82.119) |
| 18:45.49 | greenwolf | i got 2 icom's and a yaesu |
| 18:46.15 | greenwolf | i used to have a bunch of mobile stations but got rid of them and moved everything to portable sattions |
| 18:46.20 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 18:46.20 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 18:46.25 | *** part/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 18:46.31 | p3nguin | I'm looking for model numbers. |
| 18:46.51 | p3nguin | Like a 2100 or a 3000. |
| 18:47.05 | greenwolf | so ur looking to buy? |
| 18:47.23 | greenwolf | my friend has a bunch of HT radios for sale and a really low price |
| 18:47.33 | p3nguin | No. Just wanted to know what your're running. |
| 18:47.36 | greenwolf | i bet you will get a really good deal if you go thru him |
| 18:47.42 | p3nguin | What does he have? |
| 18:47.50 | greenwolf | yeasu Ft-60r |
| 18:47.56 | greenwolf | thats mine |
| 18:48.08 | p3nguin | dual bander |
| 18:48.16 | p3nguin | I have an Icom 2100 mobile, but I don't use it currently. |
| 18:48.42 | greenwolf | yea the yeasu is a dual band and also has 6meter on it |
| 18:49.00 | greenwolf | i was going to have the crystal set modified so i can have 220mhz |
| 18:49.04 | greenwolf | but never got around to doing it |
| 18:49.11 | SeRi | <PROTECTED> |
| 18:50.06 | p3nguin | The FT-60R is a 2m/70cm radio... how do you have 6m on it? |
| 18:51.02 | greenwolf | sorry the 6m is a different one |
| 18:51.06 | greenwolf | its on the icom i have |
| 18:51.18 | greenwolf | yea the yeasu is a dual band 2m/440 |
| 18:51.30 | greenwolf | hey have u ever played around with packet radio or echolink? |
| 18:51.50 | greenwolf | i was going to create my own echolink node station and have it interlink to the internet |
| 18:52.00 | p3nguin | I use echolink as a client node, but not as a link node. |
| 18:52.03 | greenwolf | maybe even patch asterisk in it for calls |
| 18:52.19 | greenwolf | have u ever got asterisk to patch phone calls for ham radio"? |
| 18:52.23 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 18:52.26 | p3nguin | I never tried. |
| 18:52.42 | greenwolf | your pretty smart i think you could get that going no problem |
| 18:52.52 | greenwolf | i was thinkin of doing something like that |
| 18:53.13 | greenwolf | i'll let you know how it works out...i might play with that tonight now that were talking about ham radios lol :) |
| 18:53.22 | p3nguin | If it involves buying things, I probably won't get around to it. I have a wife, kid, and rent to spend my money on. I don't get to buy too many toys. |
| 18:53.52 | greenwolf | damn too bad...but i hear ya you got alot of responsibilites to take care of before you can Play :) |
| 18:53.53 | p3nguin | I'm really interested in IRLP, but I'd have to buy the interface card. |
| 18:54.13 | greenwolf | what is the difference between IRLP and echolink btw |
| 18:54.34 | p3nguin | It's very similar in how you use it, but it's a different technology underneath. |
| 18:56.12 | p3nguin | Echolink seems to be much easier for the user. You just select the node and use whatever frequency they are set up on. |
| 18:57.15 | p3nguin | I guess I need to see if there are any IRLP apps for Android. Then I can mess with the user end of that. |
| 18:58.42 | p3nguin | I was wanting to see if it was possible to run IRLP on HF. I thought it might be interesting to talk on something like 80m over the internet. |
| 18:59.06 | p3nguin | Every node I have seen so far is always 2m or 440. |
| 18:59.25 | p3nguin | or maybe 220, but I can't recall one specifically. |
| 19:00.09 | p3nguin | I think echolink only works with carrier squelch, so HF/SSB is out of the question, but I think IRLP is a little more flexible. |
| 19:00.33 | p3nguin | gah |
| 19:00.39 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 19:02.48 | SeRi | comcast seems to keep their node map very well hidden. |
| 19:03.15 | p3nguin | What are you trying to find on their system? |
| 19:03.35 | SeRi | I finally talk to the eng. He told me the issue was at the node in spring. |
| 19:03.52 | SeRi | so I am trying to see which one in this area. |
| 19:04.27 | p3nguin | What was your impression of the engineer vs. the other people you dealt with? |
| 19:05.15 | SeRi | curtious. He knew what he was talking about. he didnt try to talk me down as I dont know what the internet is. we where at level. |
| 19:06.00 | p3nguin | That's often how they are. Sometimes they are a little irritated when you ask technical questions, but they can give you the answers. |
| 19:07.02 | SeRi | Well he was very well educated with the history of my account so he knew I was the frustrated one. |
| 19:07.10 | SeRi | he started the converstaion the right way |
| 19:07.20 | SeRi | no bull shit. staright to the point |
| 19:08.23 | p3nguin | Good. I'm glad you had a pleasant experience with someone for a change. |
| 19:08.38 | SeRi | yeap. |
| 19:08.47 | SeRi | by the way the pkg still in route |
| 19:08.50 | SeRi | in MO today. |
| 19:09.26 | SeRi | I tested my own headset to see how far I can go with it. I went across the street with no issues. |
| 19:10.00 | SeRi | about 3 houses down and still clear. But my office is in the front of the house which sort of helps |
| 19:12.03 | *** join/#asterisk josefig (~josef@unaffiliated/josefig) |
| 19:20.39 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 19:21.24 | *** join/#asterisk minotaur01 (~minotaur@S01060018e7f9c7df.hm.shawcable.net) |
| 19:22.21 | josefig | hello guys, If I want to setup my asterisk as a bridge, It's something like this: SIP-SoftPhone--->Asterisk--->H323-Provider I'd like to terminate my calls via my H.323 provider someone can help me ? |
| 19:33.23 | leifmadsen | josefig: just use chan_sip on the one side for your sip softphone, then configure chan_ooh323 on the other |
| 19:33.29 | leifmadsen | Asterisk does the bridging automatically |
| 19:34.54 | josefig | leifmadsen: but I'd like to record the call details (CDR) on the asterisk and create the SIP users. I'm reading this guide http://www.callcentric.com/support/device/asterisk/1_8 just please let me know if I'm ok |
| 19:39.22 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 19:39.38 | [TK]D-Fender | josefig: Multiple small bugs with that "guide" |
| 19:40.22 | [TK]D-Fender | josefig: And it is for setting up CC's SIP service. This has nothing to do with using an H.323 ITSP |
| 19:40.33 | josefig | oh hello [TK]D-Fender :) nice to see you again |
| 19:40.40 | [TK]D-Fender | josefig: And doesn't show setting up a SIP phone type account either. |
| 19:40.59 | [TK]D-Fender | josefig: So far as I can see this doesn't actually demonstrate anything related to your stated goals |
| 19:43.32 | *** join/#asterisk Sicelo900 (~user@unaffiliated/sicelo) |
| 19:59.14 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 20:08.42 | p3nguin | zombie rabbit, or rabbit zombie? |
| 20:14.25 | ghost75 | is there any way to change rights on voicemail folders within asterisk? |
| 20:16.33 | [TK]D-Fender | ghost75: When? |
| 20:17.01 | [TK]D-Fender | ghost75: and no, not "directly". voicemail.conf dopes specify the perms it will use. |
| 20:18.06 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 20:20.31 | ghost75 | my voicemail folders have no write rights on group |
| 20:21.19 | ghost75 | but all files within have 777, very strange |
| 21:01.42 | *** join/#asterisk raden (~Jon@24-240-51-238.dhcp.stpt.wi.charter.com) |
| 21:03.27 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
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| 22:09.50 | *** join/#asterisk lukejt (~luke@149.241.199.189) |
| 22:38.31 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:38.52 | lukejt | might anyone know why I can't access most CDR() variables from the AGI when using the Dial command? |
| 22:39.23 | lukejt | not only that but CDR(start) returns what should be CDR(end) |
| 22:41.02 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:44.13 | [TK]D-Fender | lukejt: your timing is unclear. When are you checking this exactly? Where does the Dial come in? |
| 22:53.11 | lukejt | [TK]D-Fender: Simple script really: ANSWER, EXEC DIAL SIP/trunk/1234, user initiated hangup which sends a SIGHUP to my AGI. From my signal handler, I'm trying to access CDR(answer) but it's not available |
| 22:56.03 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.251) |
| 22:56.44 | lukejt | All CDR vars are populated when I just do an ANSWER, STREAM FILE file "", HANGUP |
| 22:58.21 | ghost75 | the cdr vars are going to be modified during the call, is that the use of them? |
| 23:02.37 | *** join/#asterisk josefig (~josef@unaffiliated/josefig) |
| 23:03.21 | lukejt | I'm just trying to access the answered time of my Dial command for billing purposes |
| 23:04.27 | ghost75 | ok |
| 23:04.39 | ghost75 | or just parse Master.csv :) |
| 23:09.27 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 23:16.02 | *** join/#asterisk minotaur01 (~minotaur@S01060018e7f9c7df.hm.shawcable.net) |
| 23:18.53 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 23:18.53 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 23:18.56 | *** part/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 23:25.57 | *** join/#asterisk Alex25 (~kvirc@bzq-79-183-209-222.red.bezeqint.net) |
| 23:26.16 | p3nguin | So... |
| 23:26.17 | p3nguin | I still need to know. |
| 23:26.22 | p3nguin | zombie rabbit, or rabbit zombie? |
| 23:28.23 | Alex25 | What is the right way to play music in early media while desstination is 'ringing'? which command/option should be used? |
| 23:30.10 | WIMPy | Playback(...,noanswer) |
| 23:31.01 | Alex25 | yes, but i need Dial to execute while playing the file. how to get this done? |
| 23:31.03 | WIMPy | Or Dial(...,,m) |
| 23:31.33 | p3nguin | I would Dial with the m option and make sure to not Answer() before that. |
| 23:33.21 | Alex25 | it seems the 'm' option only take moh classes, but not arbitrary files |
| 23:33.27 | Alex25 | is this correct? |
| 23:33.33 | p3nguin | Yes. |
| 23:33.34 | WIMPy | yes |
| 23:33.50 | p3nguin | If you want to play files, you'll have to use Playback with the noanswer option. |
| 23:34.25 | Alex25 | so is there a way to playback while Dial is executed parallel |
| 23:35.17 | p3nguin | I think a Dial macro should be able to do that. |
| 23:35.38 | p3nguin | Dial(...,,M(your-macro-here)) |
| 23:37.21 | Alex25 | for the M option I'm reading it 'Runs the macro x when the call is answered...' |
| 23:37.41 | p3nguin | Oh, crap. |
| 23:38.23 | Alex25 | 'answered' is not good :) |
| 23:39.16 | Alex25 | ok np |
| 23:39.46 | Alex25 | I'll set some MOH classes then |
| 23:40.19 | p3nguin | You realize that playing it in early media may mean that no one ever hears it, right? |
| 23:40.39 | WIMPy | What makes you think so? |
| 23:40.55 | Alex25 | it's only for a DID number |
| 23:41.04 | Alex25 | so caller should hear it |
| 23:41.20 | WIMPy | Depends on your provider. |
| 23:41.22 | ghost75 | moh reminds me every time of medal of honour |
| 23:41.25 | Alex25 | i know |
| 23:41.38 | p3nguin | I occasionally try doing things in early media, and it sometimes is never heard. Like when I test with my cell phone, I don't hear early media. |
| 23:42.00 | Alex25 | you should hear it |
| 23:42.31 | p3nguin | Anything before the channel goes "Up" isn't heard. |
| 23:42.53 | Alex25 | it should work in all developed countries |
| 23:42.54 | p3nguin | If I do a literal answer or a playback, then I hear the audio. |
| 23:44.00 | Alex25 | I actually have some apps which work in early media, so you can test |
| 23:44.06 | p3nguin | I'm not trying to tell you that it will not work. I just wanted you to know that YMMV. |
| 23:45.06 | p3nguin | If your results are satisfying, that's what matters. |
| 23:45.10 | Alex25 | I know that |
| 23:46.00 | Alex25 | usually bi directional early media is a problem in some countries, or using some providers |
| 23:46.27 | Alex25 | but one-way audio allways worked for me |
| 23:46.57 | p3nguin | I'll have to check it out again soon. |
| 23:46.59 | Alex25 | although I heard in some poor country it doesn't work properly |
| 23:47.32 | Alex25 | call +390142276081 |
| 23:47.44 | Alex25 | it will not charge u |
| 23:48.04 | Alex25 | and you'll be able to record a message |
| 23:52.08 | p3nguin | I can't right now; I'm in the middle of supper. |
| 23:52.24 | Alex25 | bon apetit |