| 00:03.39 | Gary_B | id prefer something in call rather than User to User Signalling because that puts requiremnts on the telecom and phone system |
| 00:04.56 | Gary_B | also, it puts requiremnts on my detection equipment |
| 00:05.59 | Gary_B | its the quality of a standard phone call that is the limiting factor here i think |
| 00:07.23 | Gary_B | actually i need to go off and research my thoughts on this! |
| 00:07.54 | WIMPy | It would also save some lines, I guess. |
| 00:08.03 | Gary_B | lol |
| 00:08.23 | WIMPy | You said abov 4 lines and two phones. |
| 00:09.04 | WIMPy | Sounds like two more lines than could be used. |
| 00:09.53 | Alton35 | Seems like you could send some information across in the sip negotiation. |
| 00:15.13 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
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| 00:17.46 | *** part/#asterisk Gary_B (~IceChat7@85.211.198.89) |
| 00:58.28 | *** join/#asterisk cracoucas (cracoucas@dispo-82-248-169-48.adsl.proxad.net) |
| 00:58.34 | cracoucas | hi everybody |
| 00:59.01 | cracoucas | Is it possible with asterisk to make five people talking at the same time with a single SIP line? |
| 01:01.45 | rue_mohr | G729 needs a liscence to handle iirc, does g711? |
| 01:03.53 | *** join/#asterisk timeshell (~timeshell@206.248.136.108) |
| 01:03.55 | rue_mohr | 3cam is making voip sets too... |
| 01:07.02 | rue_mohr | how popular is MGCP? |
| 01:11.12 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 01:13.57 | *** part/#asterisk cracoucas (cracoucas@dispo-82-248-169-48.adsl.proxad.net) |
| 01:22.02 | b14ck | rue_mohr, g711 is free |
| 01:22.06 | b14ck | You don't need any licenses. |
| 01:22.13 | rue_mohr | ok |
| 01:22.14 | b14ck | g711 (ulaw) is the most popular VoIP codec |
| 01:22.20 | b14ck | It's basically the standard. |
| 01:22.24 | b14ck | everyone supports/uses it |
| 01:22.33 | rue_mohr | I didn't realize 711 was ulaw |
| 01:22.56 | rue_mohr | anyone ever worked with a panasonic IP phone with asterisk? |
| 01:23.26 | b14ck | not me |
| 01:24.14 | *** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net) |
| 01:24.40 | rue_mohr | VODAVI is a name comming up more and more |
| 01:25.52 | rue_mohr | http://salestores.com/panaso429.html <-- starting to itch to play with one, an ip phone my customers can relate to |
| 01:29.53 | b14ck | why not just get an aastra or polycom? |
| 01:30.00 | b14ck | all the ip phones basically do the exact same thing |
| 01:30.05 | b14ck | any customer knows how to use a phone |
| 01:30.09 | b14ck | you just dial the digits, lol |
| 01:30.33 | b14ck | furthermore, the polycom & aastras work great with SIP, and dont need proprietary conversion stuff in order to make it work |
| 01:34.40 | rue_mohr | yes, but aastra and polycom are both starved for programmable keys that users like for ___1___ touch stuff |
| 01:34.57 | rue_mohr | as opposed to 3 touches thru a menu |
| 01:35.39 | rue_mohr | needing to get to page 2 of the corperate directory when your trying to just transfer a call to joe is not something my customers appreciate |
| 01:52.12 | *** join/#asterisk radic (~radic@dslb-094-216-229-195.pools.arcor-ip.net) |
| 01:53.37 | WIMPy | There are several sip phones that support side cars, like Snom or Linksys. |
| 02:06.14 | *** join/#asterisk mattwj2002 (~matt@wikisource/pdpc.active.mattwj2002) |
| 02:06.59 | mattwj2002 | hi guys |
| 02:07.44 | mattwj2002 | is there a usb modem that will work with asterisk as an fxo and fax modem for ubuntu? |
| 02:07.58 | mattwj2002 | I would like to be able to send fax out it as well |
| 02:12.54 | mattwj2002 | anyone here? |
| 02:17.18 | *** join/#asterisk timeshell (~timeshell@206.248.136.108) |
| 02:17.44 | *** join/#asterisk [netman] (~netman@83.39.226.254) |
| 02:18.44 | mattwj2002 | hi timeshell |
| 02:19.09 | timeshell | Hello |
| 02:19.36 | mattwj2002 | do you know if they make an fxo usb version that works with ubuntu and asterisk? |
| 02:19.59 | timeshell | Sorry I don't know |
| 02:20.09 | mattwj2002 | okay |
| 02:20.43 | mattwj2002 | so do you know anything about the x100p? |
| 02:20.44 | mattwj2002 | :) |
| 02:20.49 | timeshell | Yes |
| 02:20.54 | mattwj2002 | nice |
| 02:21.00 | timeshell | I have one |
| 02:21.11 | mattwj2002 | where can I buy an authentic one? |
| 02:21.34 | mattwj2002 | I don't think digium still sells them |
| 02:21.46 | mattwj2002 | also does it support fax? |
| 02:21.47 | timeshell | I got mine off eBay |
| 02:21.51 | mattwj2002 | oh ok |
| 02:23.22 | timeshell | I don't know that it supports fax or not. I think that's more a function of asterisk. However I think it's basically a sort of voice modem |
| 02:24.05 | timeshell | I don't use mine anymore |
| 02:24.30 | timeshell | I went full sip last December |
| 02:24.45 | mattwj2002 | okay |
| 02:24.54 | mattwj2002 | well I have an ooma |
| 02:25.34 | mattwj2002 | it is a propriety home phone system |
| 02:25.39 | mattwj2002 | really cheap |
| 02:26.29 | mattwj2002 | I was thinking of trying to hook up an fxo to that :) |
| 02:34.15 | drmessano | An ooma really isn't a "phone system" |
| 02:34.27 | drmessano | it's a glorified ATA with buttons |
| 02:41.04 | mattwj2002 | true |
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| 05:12.09 | rue_mohr | ARG my LAst asterisk setup at home died when the bearings went on the hd, so I did raid on the new one, determined to go down, the ATA controller failed this time |
| 05:12.15 | rue_mohr | >:| |
| 05:13.09 | golikwid|mac | phones are over rated |
| 05:13.18 | golikwid|mac | just send an mail |
| 05:13.21 | golikwid|mac | emaik |
| 05:13.24 | golikwid|mac | email |
| 05:13.27 | golikwid|mac | omg i cant type |
| 05:14.35 | p3nguin | That sounds as bad as my issues with the lawn mower. |
| 05:15.06 | *** join/#asterisk knot (yiffstar66@unaffiliated/devemo) |
| 05:20.37 | rue_mohr | lawn mowers are easy, fuel, compression, spark, ya got those 3 and your ok |
| 05:22.01 | p3nguin | 1) The tires would not hold air long enough to mow the entire lawn. They were too split and dry rotted. |
| 05:22.48 | p3nguin | 2) I had to buy new tires. The new tires wouldn't stay inflated because the wheels/rims were too rusty to seal well. So I cleaned up the rust with a wire wheel and got the tires sealed. |
| 05:23.15 | p3nguin | 3) The engine didn't want to crank over. The battery was dead. I charged the battery. |
| 05:24.41 | p3nguin | 4) Something was wrong with the starter, because it didn't crank over the engine. It felt like it was seized, so I rotated the crank shaft by hand to free up the starter drive gear from the flywheel. |
| 05:26.27 | p3nguin | 5) I got it to start and mowed part of the lawn. I stopped to talk to my wife about getting me a drink because it was so hot. I went to start it again and the starter drive gear sheared clean off. |
| 05:29.24 | p3nguin | 6) I finally got a replacement starter gear and repaired the starter. I mowed for a while, but had to stop for supper. When I went back, it wouldn't crank over -- like the battery was disconnected -- but everything appeared to be normal. I took off the panel to inspect the started solenoid and discovered a fuse. The ignition fuse was blown, so I replaced it and was able to start the thing again. |
| 05:30.06 | *** join/#asterisk [netman] (~netman@83.54.223.93) |
| 05:30.06 | golikwid|mac | this story is really taking a turn for the worst |
| 05:30.39 | *** part/#asterisk fireman_biff (~biff@65.48.132.153) |
| 05:30.41 | p3nguin | It was literally one thing after another. It was like some sort of weird fate that I was not supposed to mow the friggin' grass. |
| 05:31.07 | golikwid|mac | fixing the mower - 12 hours, actual time mowing the lawn 15 mins, spending the day covered in grease and grass clippings - priceless |
| 05:31.37 | p3nguin | By the end of the day, I had had enough of that crap. |
| 05:31.39 | golikwid|mac | sounds like it might have been faster to just use a pair of scisors |
| 05:31.45 | p3nguin | haha |
| 05:32.21 | golikwid|mac | i got a new mower yesterday and paid the neighbor kid to mow the grass |
| 05:32.34 | golikwid|mac | he didnt feel i should get any discount because of his lack of mower... |
| 05:32.53 | golikwid|mac | but it is mostly shorter in almost all of the yard now |
| 05:33.31 | p3nguin | You weren't willing to pay him full price so he only cut part of the lawn? |
| 05:33.38 | golikwid|mac | no |
| 05:33.52 | golikwid|mac | i paid full price and bought the mower |
| 05:34.01 | p3nguin | and he did a crappy job? |
| 05:34.11 | golikwid|mac | he did ok |
| 05:34.24 | golikwid|mac | not great |
| 05:34.32 | p3nguin | I'm trying to figure out what will go wrong next on my mower. |
| 05:34.50 | golikwid|mac | blade might fall off and stab into your leg |
| 05:34.51 | golikwid|mac | ? |
| 05:35.02 | golikwid|mac | wear thick jeans |
| 05:35.42 | p3nguin | I've already had to put new belts on it, so there could be more torque applied to the blades now, leading to fatigue, resulting in failure... |
| 05:35.53 | p3nguin | You might be on the right track. |
| 05:36.07 | golikwid|mac | i hope not |
| 05:36.10 | golikwid|mac | wear a cup |
| 05:37.09 | p3nguin | Something has to go wrong, and I want to be prepared, so I should probably take your advice. |
| 05:38.49 | p3nguin | This could also be a good time to look for a neighbor kid who likes to mow grass. |
| 05:39.13 | golikwid|mac | yea |
| 05:41.26 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 05:41.26 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 05:42.26 | golikwid|mac | why is there no flight of the conchords season 3 |
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| 06:26.13 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 06:32.48 | rue_mohr | p3nguin, hmm ok, maybe get a push mower? |
| 06:32.49 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 06:33.04 | rue_mohr | I'll give you one for free, I pulled it out of a scrap metal bin |
| 06:33.16 | rue_mohr | it only needed a carb overhaul |
| 06:33.53 | rue_mohr | ok, my asterisk box is back up, new motherboard and new fans, whisper quiet |
| 06:34.03 | rue_mohr | this time I bet the motherboard goes |
| 06:34.14 | rue_mohr | as long as nothing kills my T100P card, I'm ok |
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| 06:41.24 | rue_mohr | ooo in all the chaos the raid1 came apart... |
| 06:41.29 | rue_mohr | applies gloo |
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| 07:48.49 | cosmo83 | Hi Guys .. need some help. I have a sangoma A104 card , one port of which is connected to the PSTN and other one is to my EPBX trying to use it as a channelbank |
| 07:49.54 | cosmo83 | The issue is that , when iam dialing from my SIP client to the dahdi extensions the callerid is not being transferred to my EPBX and the call is landing on my default number |
| 07:50.24 | cosmo83 | iam able to dial from dahdi extension to my SIP extension pretty easily |
| 07:50.59 | cosmo83 | any help would be appreciated |
| 07:52.17 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 07:53.23 | cosmo83 | anybody ??? |
| 07:54.35 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 07:59.04 | ChannelZ | The port your dialing out of to the other thing (EPBX?) is it an FXO or FXS? I don't know what this other thing is you speak of |
| 08:01.00 | cosmo83 | It is a Seimens HIPATH EPBX |
| 08:01.25 | cosmo83 | FXS |
| 08:02.02 | *** join/#asterisk exothermc (~miles@174.127.153.10) |
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| 08:02.51 | rue_mohr | for mgcp, |
| 08:03.17 | rue_mohr | [0040cb430630] <-- can the section be the mac address with host=dynamic ? |
| 08:05.07 | ChannelZ | cosmo83: try 'hidecallerid=no' in your /etc/asterisk/chan_dahdi.conf in the [channels] section before you define the channels |
| 08:05.44 | cosmo83 | its already set to no |
| 08:05.51 | ChannelZ | But I'm not positive of the Sangoma card can/will transmit caller ID, I'm pretty sure the Digium ones do though |
| 08:06.19 | ChannelZ | What does your dialplan look like when dialing from your SIP phone? |
| 08:06.52 | cosmo83 | Channelz , I am using trixbox for this simple setup. Span 1 is connected to PSTN . Span 2 is connected to the EPBX in pri_net mode |
| 08:07.40 | cosmo83 | Then in trixbox i have setup one SIP channel with extension 2309 |
| 08:08.01 | cosmo83 | and a Zaptel channel with 2310 extension |
| 08:08.10 | ChannelZ | oh wait these are PRI cards not analog |
| 08:08.16 | cosmo83 | nope |
| 08:08.37 | cosmo83 | these are PRI cards |
| 08:09.00 | ChannelZ | you said PSTN and I automatically assumed (and then you answered my FXO/FXS question :) |
| 08:09.22 | ChannelZ | in that case I'm not sure; Is the callerid set properly for the SIP channel? |
| 08:09.30 | cosmo83 | ya |
| 08:09.44 | cosmo83 | i have two SIP clients able to communicate well |
| 08:11.42 | cosmo83 | Iam calling DAHDI/32 (32 is the first channel number of the SPAN connected to EPBX ) in the inbound route |
| 08:12.47 | cosmo83 | the prob is that the EPBX should be able to handle the transfer to the right extension based on the callerid. It is always landing my default extension |
| 08:13.18 | ChannelZ | Do you get debug from the epbx? IE are you getting no CID at all, or something else? |
| 08:14.32 | cosmo83 | Channelz , lets take this problem to a simple level. Lets talk about the SIP Client |
| 08:15.26 | cosmo83 | Iam dialing out to my cellphone from my SIP Client and the number i receive on my cellphone is the pilot number |
| 08:15.55 | ChannelZ | the pilot number? |
| 08:16.26 | cosmo83 | generally here in India , we have something called the pilot number (which is the default extension for incoming) |
| 08:17.57 | ChannelZ | that doesn't make any more sense ot me |
| 08:18.47 | ChannelZ | You dial SIP from Asterisk to your cell phone.. over the same PRI? And your cell phone does or doesn't show the phone number you configured for the SIP client? |
| 08:24.40 | ChannelZ | (also do you have 'callerid=asreceived' in chan_dahdi.conf?) |
| 08:27.23 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 08:29.15 | ChannelZ | oooook, guess that's the end of the conversation. I'm going to bed |
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| 08:35.33 | cosmo83 | sorry , got disc !! |
| 08:35.47 | cosmo83 | ChannelZ, still there ? |
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| 09:04.31 | rue_mohr | anyone here familiar with VxWorks? I need to know how to set a devices IP |
| 09:06.20 | rue_mohr | ooo shoretells stuff looks coool |
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| 10:25.11 | Nynv20 | Hello i need help please |
| 10:25.27 | *** join/#asterisk rishikesh (~Manipur@117.242.156.25) |
| 10:25.30 | rishikesh | hi |
| 10:26.20 | rishikesh | i would like to setup extension 121 as musiconhold for broadcasting radio on my pbx |
| 10:26.26 | rishikesh | how to do this? |
| 10:27.05 | *** join/#asterisk [netman] (~netman@83.54.218.101) |
| 10:27.12 | rishikesh | plz help me |
| 10:27.13 | Nynv20 | how can i switch from text mode to graphical mode in Astresk now ? |
| 10:27.44 | Nynv20 | how can i switch from text mode to graphical mode in AstreskNOW ? |
| 10:27.52 | rishikesh | i would like to setup extension 121 as musiconhold for broadcasting radio on my pbx |
| 10:27.56 | rishikesh | plz help me |
| 10:28.17 | rishikesh | i use asterisk now 1.7 |
| 10:28.32 | Nynv20 | how do you switch to graphic mode rishikesh |
| 10:28.33 | Nynv20 | ? |
| 10:28.57 | rishikesh | you need web browser |
| 10:29.01 | Gugge | ive never used asterisknow, but wtf is graphic mode? :P |
| 10:29.14 | Nynv20 | i just installed it |
| 10:29.21 | Nynv20 | and when i run the virtual machine |
| 10:29.28 | Nynv20 | it logs into bash mode |
| 10:29.34 | rishikesh | point your browser to of your asterisk box |
| 10:29.47 | Nynv20 | what you mean? |
| 10:29.54 | rishikesh | i would like to setup extension 121 as musiconhold for broadcasting radio on my pbx |
| 10:29.58 | Nynv20 | by pointing my browser to of my astrisk box |
| 10:30.03 | rishikesh | anybody here to help me |
| 10:30.05 | rishikesh | yes |
| 10:30.42 | Gugge | rishikesh: http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf#Exampleusingasxmmswmvstreamsoranythingth |
| 10:31.01 | rishikesh | i dont understand reading that link Gugge |
| 10:31.16 | rishikesh | it is not clear and not working also with the guide |
| 10:31.24 | rishikesh | it says no. is not available |
| 10:31.37 | Gugge | rishikesh: sorry, if you dont understand the info at that link, what you are trying to do is not something you can do :/ |
| 10:32.04 | rishikesh | when i dial 121 it says no is not available |
| 10:32.04 | Gugge | you need to make a script outputting the audio stream |
| 10:32.25 | Gugge | and then you need to setup a musiconhold class using that stream |
| 10:32.27 | rishikesh | first i want my extension working |
| 10:32.32 | rishikesh | for musiconhold |
| 10:32.42 | Gugge | and then you need to setup an extension that plays that class |
| 10:32.42 | rishikesh | forget streaming for now |
| 10:32.51 | Gugge | setting up the extension is an asterisknow thing |
| 10:32.55 | Gugge | cant help you there |
| 10:32.59 | Gugge | this is #asterisk, not #asterisknow |
| 10:33.05 | rishikesh | ok |
| 10:33.11 | *** part/#asterisk rishikesh (~Manipur@117.242.156.25) |
| 10:33.25 | Nynv20 | how about me Gugge |
| 10:33.26 | Nynv20 | ? |
| 10:33.31 | Gugge | same thing |
| 10:33.33 | Gugge | this is #asterisk, not #asterisknow |
| 10:33.52 | Gugge | i have no idea how the webinterface on asterisknow works |
| 10:34.00 | Nynv20 | i understand |
| 10:34.11 | Nynv20 | im just a student |
| 10:34.18 | Nynv20 | i want to run the astrisk server.. any of them |
| 10:34.23 | Nynv20 | astriskNOW or astrisk |
| 10:34.24 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 10:34.32 | Nynv20 | any of them |
| 10:34.37 | Nynv20 | which is easier ? |
| 10:35.11 | Gugge | i guess asterisknow is more userfriendly, because it has a webinterface |
| 10:35.13 | Gugge | http://www.asterisk.org/AsteriskNOW-1.5-QuickStart |
| 10:35.17 | lost_soul | asterisknow would prolly be a bit more userfriendly |
| 10:35.46 | lost_soul | but as far as learning asterisk.. using asterisk itself without the gui would be better |
| 10:36.40 | Nynv20 | Thank you Gugge |
| 10:39.36 | *** join/#asterisk BANSAL (~bansal@117.199.116.87) |
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| 10:50.05 | tzafrir | How do I roll back a change on voip-info.org ? |
| 10:51.22 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 10:51.22 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 11:23.00 | *** join/#asterisk rishikesh (~Manipur@117.242.156.25) |
| 11:23.16 | rishikesh | how do i add skype integration with asterisk? |
| 11:23.28 | *** join/#asterisk nova911 (~Adium@59.162.86.164) |
| 11:24.02 | *** join/#asterisk ELLOMO (~ELLOMO@datastorm.nl) |
| 11:24.46 | ELLOMO | Anybody experience with Skype for Asterisk and setting the avalability by chan_skype.conf? |
| 11:25.06 | ELLOMO | [Aug 22 12:36:01] WARNING[7806] chan_skype.c: Unknown config option availability |
| 11:25.23 | ELLOMO | WARNING[7806] chan_skype.c: Unknown config option status |
| 11:25.46 | v1s | ELLOMO: I use siptosis |
| 11:26.24 | ELLOMO | I bought the skype for asterisk license... ;-) |
| 11:27.44 | v1s | how many channels u get ? |
| 11:28.07 | ELLOMO | just one |
| 11:28.43 | ELLOMO | I just tried to follow the docs... http://docs.digium.com/SFA/skype_for_asterisk_admin_manual.pdf |
| 11:29.38 | *** join/#asterisk nova911 (~Adium@122.182.0.38) |
| 11:31.31 | pabelanger | ~sfa |
| 11:31.52 | v1s | can u pb ur conf just take out any pass or personal info |
| 11:31.53 | pabelanger | feed infobot |
| 11:32.09 | pabelanger | infobot: :) |
| 11:32.10 | infobot | (: |
| 11:32.23 | pabelanger | ~skype |
| 11:32.24 | infobot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, with most solutions being complex, complicated, and hack-ish . Digium's SkypeForAsterisk (see ~SkypeForAsterisk) is a new solution that is a cleaner non-dependent option. |
| 11:32.37 | pabelanger | rishikesh: ^^ |
| 11:33.11 | pabelanger | ELLOMO: pb your chan_skype.conf |
| 11:35.15 | *** join/#asterisk nova911 (~Adium@59.162.86.164) |
| 11:35.34 | ELLOMO | [maarten.test] |
| 11:35.36 | ELLOMO | secret=funny |
| 11:35.37 | ELLOMO | context=incoming |
| 11:35.39 | ELLOMO | exten=2001 |
| 11:35.40 | ELLOMO | direction=both |
| 11:35.42 | ELLOMO | about=Dit is een test |
| 11:35.43 | ELLOMO | fullname=Who Knows |
| 11:35.45 | ELLOMO | country=Netherlands |
| 11:35.46 | ELLOMO | city=Oosterwolde |
| 11:35.48 | ELLOMO | gender=1 |
| 11:35.49 | ELLOMO | status=Logged In |
| 11:35.51 | ELLOMO | mood_text="Waiting for a call" |
| 11:35.52 | ELLOMO | phone_mobile=+31516522222 |
| 11:35.53 | fenrus | lol |
| 11:35.54 | ELLOMO | email=maarten@test.nl |
| 11:35.54 | fenrus | pastebin |
| 11:35.55 | ELLOMO | availability=Online |
| 11:36.00 | ELLOMO | oops |
| 11:36.00 | fenrus | goddamnit |
| 11:36.10 | ELLOMO | goddamnit? |
| 11:37.18 | ELLOMO | http://pastebin.com/E4AvSje9 |
| 11:40.22 | ELLOMO | more complete (whole file) http://pastebin.com/5zJgSym1 |
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| 11:44.09 | rishikesh | thats it? |
| 11:44.29 | rishikesh | with that incoming nd outgoing is possible? |
| 11:44.45 | ELLOMO | yes |
| 11:44.49 | rishikesh | personal skype account is also ok? |
| 11:45.17 | ELLOMO | in extensions.conf I have the exntension 2001 which rings all the phones |
| 11:45.48 | ELLOMO | nope, you need to add the account as a member to skype manager business account |
| 11:46.13 | ELLOMO | in other words, you cannot use the skype client anymore with this account |
| 11:46.25 | ELLOMO | unless you remve is from the member list |
| 11:46.57 | ELLOMO | outgoing is not yet tested... |
| 11:47.05 | *** join/#asterisk nova911 (~Adium@59.162.86.164) |
| 11:47.17 | rishikesh | and one thing |
| 11:47.58 | rishikesh | how do i setup musiconhold extension so that everyone on my sip network can call that extension and hear a particular sound |
| 11:48.28 | rishikesh | i want to setup extension 121 as musiconhold so that everyone can call and listen songs |
| 11:48.38 | ELLOMO | hehehe broadcasting music... |
| 11:49.14 | *** join/#asterisk mpe_ (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 11:49.42 | ELLOMO | Set(CHANNEL(musicclass)=whatever) |
| 11:49.48 | rishikesh | yes |
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| 11:50.10 | rishikesh | i have gone through so many sites about that but its not working |
| 11:50.22 | rishikesh | it says extension no is not available |
| 11:51.23 | rishikesh | how to setup musiconhold extension |
| 11:55.27 | v1s | rishikesh: is the moh not work or the extension not working |
| 11:55.38 | rishikesh | extension not working |
| 11:56.02 | rishikesh | moh is working by default |
| 11:56.10 | v1s | pb ur extensions file ill see if I can help. |
| 11:56.22 | rishikesh | ok |
| 11:58.18 | rishikesh | my server is in virtualbox |
| 11:58.30 | rishikesh | so i am not able to copy my config files |
| 11:58.52 | v1s | ssh to it? |
| 11:58.54 | rishikesh | in which section do i have to add extension declaration? |
| 11:59.24 | v1s | what ever context u have set for the use |
| 11:59.33 | v1s | u should have the extension u want to use in it. |
| 12:00.17 | rishikesh | example? |
| 12:01.17 | rishikesh | if want to add 121 then what will be like? |
| 12:02.02 | WIMPy | rishikesh: Maybe you should try this as a first step: |
| 12:02.05 | WIMPy | ~book |
| 12:02.06 | 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 |
| 12:02.48 | rishikesh | but tell me |
| 12:03.01 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 12:03.26 | v1s | rishikesh: think something like this should work: http://pastebin.com/GZCrE8Xq |
| 12:05.46 | rishikesh | i have seen the example |
| 12:05.52 | rishikesh | with that no. 8888 |
| 12:06.11 | rishikesh | it is not normal extension like user extension |
| 12:06.25 | rishikesh | it is like voical number or extension rite? |
| 12:06.40 | v1s | thats not an extension |
| 12:06.48 | v1s | that ur calling |
| 12:06.55 | v1s | its just the sip u ser account |
| 12:07.16 | v1s | i just put the number in there random to remove the info i had in it from my conf |
| 12:07.34 | rishikesh | ok |
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| 12:43.10 | phix | hey gang |
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| 15:41.16 | wikii | [TK]D-Fender :: please check http://pastebin.ca/1922156 |
| 15:42.25 | wikii | i have pasted Degub pri of both successful and unsuccessful calls |
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| 17:08.28 | Dovid | anyone know what version of dahdi wanpipe 3.4.9 needs ? |
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| 17:33.22 | vgserrat | hello. -dies anyone can tell me if there is a guide to configure tdm400p with dahdi? I cant find it . Thanks |
| 17:33.52 | vgserrat | hello. Does anyone can tell me if there is a guide to configure tdm400p with dahdi? I cant find it . Thanks |
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| 17:50.09 | *** join/#asterisk ELLOMO (~ELLOMO@datastorm.nl) |
| 17:53.40 | ELLOMO | Hi, using asterisk 1.4.35-rc1 with two codecs, g729(unlicensed) and ulaw is driving me nuts. in total 8 extensions but one of them is behind low speed internet so only calls from and to one extension should be in G729. All extensions are capable of G729 so no license and transcoding is needed, but how to setup this config? |
| 17:54.23 | drmessano | You need at least one license |
| 17:55.06 | ELLOMO | drmessano, when I config all extensions in G729 i dont have any problem evrything can talk |
| 17:56.06 | drmessano | Yes, obviously |
| 17:57.38 | drmessano | Spend the $10 and buy the one license.. Otherwise you're going to have problems with preferred codec on the phones, etc |
| 17:57.49 | ELLOMO | but why asterisk cannot see if one extenion has a disallow all and allow ulaw&g729 and the other ony g729 that it can just select g729 without transcoding or whatsoever? |
| 17:57.50 | drmessano | or force them all to G729 |
| 17:58.30 | drmessano | g729 is not optimal over uLAW.. Asterisk will not prefer a lesser quality codec |
| 17:58.52 | drmessano | There is no code to optimize for "G729 license avoidance" |
| 18:00.02 | ELLOMO | drmessano, so if I buy a license and then configure ext1 with Ulaw & G729 and ext2 with only G729, asterisk will just select G729? |
| 18:00.31 | drmessano | No sir, it's going to transcode.. which you need a license to do. |
| 18:01.15 | ELLOMO | drmessano, but why??? there is *no need* for transcoding because both ends can speak perfectly G729 |
| 18:01.29 | drmessano | So set them ALL to G729 |
| 18:01.35 | ELLOMO | this is already proven when I only allow G729 on all extensions |
| 18:01.41 | drmessano | So set them ALL to G729 |
| 18:02.26 | ELLOMO | drmessano, I am afraid that with the transcoding I get unacceptable delays due to the transcode process |
| 18:02.45 | drmessano | Delays? |
| 18:02.49 | drmessano | How slow is your system? |
| 18:02.51 | ELLOMO | Using G729 only, which I have done for a couple of years, |
| 18:03.21 | ELLOMO | causes me not to have DTMF on dialout lines |
| 18:03.52 | drmessano | Thats a configuration issue |
| 18:03.56 | ELLOMO | and some international calls dont work at all cuz they use other codecs |
| 18:04.11 | drmessano | ELLOMO: Yes, that is why you buy G729 licenses |
| 18:04.17 | drmessano | To Transcode to other codecs |
| 18:04.19 | ELLOMO | the system is AMD geode @500MHz |
| 18:04.30 | drmessano | Ok, that's just silly |
| 18:04.58 | ELLOMO | sille? |
| 18:05.03 | ELLOMO | in what way? |
| 18:05.23 | drmessano | How old is that system, 7 or 8 years? |
| 18:05.51 | ELLOMO | nope, it is an embedded box with Astboxlinux on CF |
| 18:06.08 | ELLOMO | works perfect (with G729) |
| 18:06.30 | drmessano | Your hardware specs do not currently meet basic requirements for your wishlist.. |
| 18:06.53 | drmessano | Your wishlist includes the need to transcode, and your hardware is not acceptable for the task |
| 18:07.33 | ELLOMO | I dont what to transcode, there is technically no need for it, all the enpoints can speak g729 |
| 18:08.41 | ELLOMO | just the enpoints that can ALSO speak ULAW should speak in ULAW to remote peers |
| 18:08.43 | drmessano | The scenario you presented requires transcoding either for the one device, or your trunks that you say "some require other codecs".. Sorry, but you can't just "not want" to transcode and have a need where transcoding is required |
| 18:09.14 | drmessano | Yes, and those WILL |
| 18:09.27 | ELLOMO | yes, that is what I see happening |
| 18:09.31 | drmessano | But they WONT "DUMB DOWN" to G729 magically for that one extension |
| 18:09.53 | ELLOMO | no negotiation? |
| 18:10.07 | ELLOMO | to find a matching codec? |
| 18:10.20 | ELLOMO | that is not part of the protocol? |
| 18:16.08 | ELLOMO | drmessano, What if I through out G729 and use ILBC instead in combination with Ulaw? |
| 18:16.51 | *** join/#asterisk rue_mohr (~rue@24.207.119.38) |
| 18:17.06 | rue_mohr | anyone here familiar with VxWorks? I need to know how to write the network config to disk |
| 18:18.30 | rue_mohr | someone must be, its used on all the voip gateways |
| 18:18.31 | drmessano | ELLOMO, DO NOT PRIVATE MESSAGE ME |
| 18:19.08 | drmessano | ELLOMO: You asked, I answered, and this is going around in huge circles. Google for asterisk+g729+codec+negotiation since the answers you've gotten here do not appear to satisfy you |
| 18:19.58 | ELLOMO | drmessano, no, I just want to know how things work. I can only learn when I ask. |
| 18:20.05 | rue_mohr | VxWroks, anyone... |
| 18:20.19 | drmessano | ELLOMO: Explained to you. |
| 18:20.33 | rue_mohr | please put up hands... or feet.... maybe wiggle your ear... |
| 18:20.56 | ELLOMO | drmessano, thank for that, replacing G729 by ilbc would that help? |
| 18:21.02 | drmessano | rue_mohr: Is your goal to annoy someone into answering your off-topic question? |
| 18:24.24 | carrar | rue_mohr, call VxWroks support |
| 18:24.28 | carrar | thanks |
| 18:24.32 | carrar | OR |
| 18:24.35 | carrar | Join #VxWroks |
| 18:24.39 | carrar | and ask there |
| 18:25.14 | ELLOMO | someone who knows if replacing g729 with ilbc will solve my issue? |
| 18:25.37 | carrar | Whats your problem |
| 18:25.48 | carrar | just use g.711 |
| 18:25.55 | carrar | problem solved |
| 18:25.57 | carrar | NEXT |
| 18:26.18 | ELLOMO | yes, but I have one extension that is on another continent with slow speed internet. |
| 18:26.34 | ELLOMO | now I use g729 on all extensions only for this one |
| 18:26.44 | carrar | purchase g729 codecs |
| 18:27.25 | carrar | If you show up at astricon they sometimes give free codecs away |
| 18:27.33 | carrar | which would be enough for 1 phone |
| 18:27.35 | ELLOMO | system runs at 500mhz, is low power device.. would replacing g729 by ilbc solve the issue? |
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| 18:28.48 | carrar | get a appropriate sized server for the task you are trying to accomplish |
| 18:29.29 | carrar | You will still need to transcode |
| 18:35.52 | ELLOMO | carrar, thanks, then I should start looking for something silent and more powerfull to replace my small box |
| 18:38.17 | rue_mohr | there's a channel.... imageine that.. |
| 18:38.50 | rue_mohr | liar |
| 18:39.25 | rue_mohr | so NObody has work with media gateways that use vxworks? |
| 18:39.27 | rue_mohr | wow |
| 18:39.39 | rue_mohr | from what I can tell, they almost all use it |
| 18:39.50 | rue_mohr | I have a VOG4000 I'm really close to getting to work |
| 18:47.18 | rue_mohr | [Nov 26 04:36:53] NOTICE[2319]: chan_mgcp.c:1753 find_subchannel_and_lock: Gateway 'VP-5-40' (and thus its endpoint 'aaln/4') does not exist |
| 18:47.28 | rue_mohr | is that a configu error on asterisk or the gateway? |
| 18:47.54 | rue_mohr | finding my way around vxworks is not easy, documentation on how to do real things is scarse |
| 18:50.25 | rue_mohr | aha its on the gateway |
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| 19:15.13 | hluesea | does anyone guide me about digium vs sangome cards ? i need 24 ports analog card and i don't decide which brand or are they have any difference (sangoma and digium). |
| 19:16.28 | drmessano | hluesea: A channel bank may be a better option |
| 19:17.12 | hluesea | can you suggest me a brand or model ? |
| 19:17.25 | drmessano | Mediatrix 4124 perhaps |
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| 19:32.36 | rue_mohr | hluesea, I used a digium card, terrible echo problems, even with hwec, try a sagnoma |
| 19:33.01 | rue_mohr | drmessano, is a channelbank of that size ~$8000? |
| 19:33.40 | rue_mohr | I was looking at some by Quintum AXM800 and AXE800, pretty costly, |
| 19:34.03 | rue_mohr | suspect it can be cheaper to get a T1 channelbank and card, not sure yet |
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| 19:36.41 | carrar | use a sound card, solidering iron |
| 19:36.45 | carrar | make yourself a ATA |
| 19:37.01 | rue_mohr | I can tell you some good motorola chips to use! |
| 19:37.12 | carrar | use the radioshack 555 |
| 19:37.13 | rue_mohr | ... were you being carcastic? |
| 19:37.19 | rue_mohr | no no |
| 19:37.27 | carrar | very much so |
| 19:37.34 | rue_mohr | there are some nice ulaw codec chips |
| 19:37.46 | rue_mohr | the datasheets come with application examples |
| 19:38.13 | rue_mohr | its shockingly easy to make a pots -> ulaw |
| 19:38.48 | carrar | and the SIP? |
| 19:39.17 | rue_mohr | I'v lean towards using one of the mux chips to bring togethor a T1 channel |
| 19:39.37 | rue_mohr | as their the right format anyhow |
| 19:39.44 | rue_mohr | same clock rate and coding |
| 19:40.22 | rue_mohr | are you interested and idle enough to page over a datasheet? |
| 19:40.49 | rue_mohr | its amazing how long T1 technology has been pre-boxed |
| 19:41.09 | rue_mohr | there should be people making channelbanks for like $40 |
| 19:41.24 | carrar | http://cgi.ebay.com/Adit-600-Carrier-Access-unit-CAC-/290466841229?pt=LH_DefaultDomain_0 |
| 19:41.27 | carrar | there you go |
| 19:41.30 | carrar | 16 FXS ports |
| 19:41.35 | carrar | to T1 |
| 19:41.44 | carrar | $49 |
| 19:41.48 | rue_mohr | FXO ports are more valueable |
| 19:42.00 | carrar | so slide in a FXO blade |
| 19:42.16 | carrar | who knows, those could be FXO blades |
| 19:42.29 | carrar | but I doubt it |
| 19:42.37 | carrar | as says fxs |
| 19:42.50 | rue_mohr | a 4 port T1 card?... odd |
| 19:43.02 | carrar | not a 4 port T1 |
| 19:43.37 | carrar | I think it's a bri card |
| 19:44.11 | carrar | the T1 ports you would use are on the back anyways |
| 19:44.15 | carrar | there are 2 of them |
| 19:44.23 | carrar | since the ADIT600 supports 48 ports |
| 19:44.29 | carrar | toss out that 4 port card |
| 19:44.44 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:44.51 | tzafrir | I''ll ask again: how do I recover http://voip-info.org/ to an earlier version? |
| 19:45.31 | rue_mohr | the whole site or a page? |
| 19:47.34 | rue_mohr | anyone know any remarkably nice hardware to run asterisk on for small (really small) offices? |
| 19:48.12 | rue_mohr | haha, from what I can tell a N900 is powerfull enough, lacks some interface tho.. |
| 19:49.43 | rue_mohr | can asterisk compile for an arm9? |
| 19:50.58 | Chainsaw | rue_mohr: You could compile it on a DEC Alpha if you were so inclined. |
| 19:51.04 | Chainsaw | rue_mohr: It's just C code. There is no pixie dust involved. |
| 19:51.41 | rue_mohr | it would be funny somehow to be running asterisk on a nokia phone |
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| 19:56.18 | rue_mohr | http://cgi.ebay.com/Avaya-G10-Quick-Edition-PSTN-Gateway-4-Port-FXO-w-power-/230513239752?pt=LH_DefaultDomain_0 |
| 19:56.25 | rue_mohr | ^^ is avaya stuff junk? |
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| 20:48.53 | carrar | rue_mohr, audiocodes is good |
| 20:51.47 | tzafrir | rue_mohr, sure it can |
| 20:52.06 | tzafrir | rue_mohr, if it doesn't build, it's a bug |
| 20:52.40 | tzafrir | Chainsaw, it won't build on HURD, for starters |
| 20:52.53 | Chainsaw | tzafrir: Oh noes :( |
| 20:52.59 | tzafrir | (hint: look for PATH_MAX and friends) |
| 20:53.25 | tzafrir | rue_mohr, there are existing packages of asterisk for the n900 |
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| 22:00.34 | voxter | http://www.grandstream.hk/product_GXV3140.shtml |
| 22:00.42 | voxter | Am i misled? does this actually look half decent, or am i insane? |
| 22:00.54 | voxter | please note, i did say HALF decent. |
| 22:01.02 | voxter | this has got to be ontop of some other OS |
| 22:18.33 | [TK]D-Fender | voxter: "full-featured shit" |
| 22:19.26 | voxter | [TK]D-Fender: hey, i never assumed it would work well, it clearly seems to do too much for what its supposed to already, it just seems unlikely that GS would have produced this on their own, maybe i am giving the pictures too much credit. :) |
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| 22:29.25 | drmessano | [15:33] <rue_mohr> drmessano, is a channelbank of that size ~$8000? <--- or $1100 ???? |
| 22:30.11 | drmessano | Not that details are important here |
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| 22:48.05 | mmlj4 | how horrible is Elastix? I have to support it next week, unless I can convince $client to let me do vanilla * |
| 23:02.54 | tzanger | mmlj4: I'm about to undo YAEI (yet another elastix install) |
| 23:03.13 | tzanger | it's fine, as was freepbx and all the others, so long as you do not want to ever do anything outside of what they already do |
| 23:06.19 | drmessano | mmlj4, You can work around the dialplan quite a bit.. The things you have a VERY hard time with are some the the more integrated routines, like the macros that that glue the modular dialplan together.. |
| 23:07.07 | drmessano | mmlj4: Elastix itself is a real pain due to the way they implement FreePBX and the host of other apps they load onto their kitchen sink of an install |
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| 23:12.36 | mmlj4 | I'd never touch any of those distros, if I wasn't being forced |
| 23:14.04 | mmlj4 | though their approach "we do voip, email, hylafax, etc. all for you" is interesting... only heavy hitter sysadmins could pull that all off |
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| 23:47.28 | rue_mohr | drmessano, I'm searching for away of making an almost-canned system that can do 8FXOx4FXS+24voip for < $2000 |
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| 23:48.08 | rue_mohr | if I can get 6FXOx16voip for under $1000 would be awesome |
| 23:48.34 | rue_mohr | :/ if only someone still made the 616 |
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| 23:53.01 | [TK]D-Fender | what is "16 voip"? |
| 23:53.09 | rue_mohr | voip sets |
| 23:53.15 | rue_mohr | :) |
| 23:53.40 | rue_mohr | so no ppp dialup network connections :) |
| 23:55.15 | rue_mohr | what is the "limit" for a 100mbit interface? 20 calls? |
| 23:55.25 | rue_mohr | 711 |
| 23:57.41 | Gugge | rue_mohr: you think a call takes 5Mbit ? |
| 23:58.11 | rue_mohr | I'm never measured it, apparently you cant stream video over 5.5mbit |
| 23:58.47 | Gugge | who told you that? |
| 23:58.50 | [TK]D-Fender | [19:58]<rue_mohr>I'm never measured it, apparently you cant stream video over 5.5mbit <- VerizonMath (tm) |
| 23:59.03 | rue_mohr | #videolan #mplayer, and #gstreamer |
| 23:59.22 | Gugge | you should try going to youtube, on a 5Mbit connection |
| 23:59.25 | rue_mohr | they all say its impossable to stream CIF video over 5.5mbits |
| 23:59.44 | rue_mohr | hey you talk to them, I'm just the guy trying to do it |
| 23:59.47 | [TK]D-Fender | rue_mohr: And how much BW does CIF represent? |