| 00:04.35 | cusco_ | or console dial ext@context |
| 00:04.56 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:04.56 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:05.51 | WIMPy | If you have a console channel. |
| 00:12.21 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 00:18.26 | xoveruk | and whats does the (S) mean in iax2 show peers? |
| 00:23.36 | *** join/#asterisk gusto (~gusto@ppp-62-216-201-80.dynamic.mnet-online.de) |
| 00:31.05 | ChannelZ | static |
| 00:39.03 | xoveruk | thanks |
| 00:42.02 | xoveruk | does console dial ext@context dial and then hangup? |
| 00:43.10 | xoveruk | can i increase verbosity to check if the authentication is working across the iax2 channel? |
| 00:58.49 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
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| 01:11.41 | the_eye_ | Could you help me I got error file.c: File record/test does not exist in any format in voice menu |
| 01:15.15 | the_eye_ | asterisk 1.8 at Ubuntu 12.04 |
| 01:25.20 | ChannelZ | Think it pretty much says what it means |
| 01:25.43 | ChannelZ | What is 'test'? Where is it? How did you make it? |
| 01:30.56 | the_eye_ | test is a wav file in /var/lib/asterisk/sounds/record |
| 01:32.18 | ChannelZ | that's not the normal location things like Background or Playback look in, so unless you're specifying the entire path to the file... |
| 01:38.23 | the_eye_ | what is the corect location ? I try to copy to |
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| 01:39.09 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:41.24 | ChannelZ | /var/lib/asterisk/sounds/XX generally, where XX is the language you're running |
| 01:46.31 | *** join/#asterisk Katty (~derik@96-35-244-101.dhcp.stls.mo.charter.com) |
| 01:46.36 | Katty | HAI LADS. |
| 01:46.59 | Katty | this irc client is weird. it's all colorful n stuff. n has a gui. |
| 01:48.10 | the_eye_ | ChannelZ, thanks I use full path and its ok |
| 01:56.13 | carrar | woah |
| 01:56.16 | carrar | Katty |
| 01:56.27 | Katty | i know, right, i'm never here. |
| 01:56.29 | carrar | SMACK DAB in the middle of the weekend and here you ARE |
| 01:56.42 | Katty | yes. it's scary. waiting for company tho. |
| 01:56.44 | carrar | Must be HOT outside |
| 01:56.49 | carrar | What company |
| 01:57.06 | Katty | i am just killing time till company comes ove.r |
| 01:57.20 | carrar | You having a knitting party? |
| 01:58.56 | carrar | A tupperware like party for Digium Hardware? |
| 01:59.01 | ChannelZ | no it's a PAR-TAAAY |
| 01:59.20 | carrar | Point the CAMS INSIDE |
| 01:59.26 | carrar | BROADCAST IT |
| 01:59.48 | carrar | or perhaps it's a IRC PARTAY |
| 02:00.32 | ChannelZ | It's hot in here. |
| 02:00.49 | *** part/#asterisk fulcan (~brads@li503-84.members.linode.com) |
| 02:02.07 | ChannelZ | Hmm. How come there are no functions/apps to change your away status on jabber/gtalk? |
| 02:02.19 | Katty | meep, no. |
| 02:02.30 | Katty | it is a beer and can or movie party. |
| 02:02.39 | Katty | not much of a party, i admit, but i'm getting old(er) |
| 02:02.53 | ChannelZ | 'can or movie'? |
| 02:03.06 | carrar | Which movie? |
| 02:03.12 | carrar | Grease? |
| 02:03.16 | carrar | heh |
| 02:03.20 | ChannelZ | Magic Mike |
| 02:05.44 | Katty | err cah or a movie |
| 02:05.50 | Katty | cards against humanity. |
| 02:06.15 | Katty | idk what movie. we watched memento last time |
| 02:06.17 | carrar | You making Cards and talking about humanity |
| 02:06.21 | carrar | nice |
| 02:06.32 | Katty | :P |
| 02:07.59 | ChannelZ | Run Lola Run. Turn it up real loud |
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| 07:35.45 | Preston | Anybody awake? |
| 07:35.50 | Preston | I have a weird idea. |
| 07:36.53 | Preston | I was wondering if a sip client could point multiple sip sessions to the same RTP stream, so that multiple recipients could hear the same audio without having to be broadcasted multiple times. |
| 07:37.06 | Preston | I'm not awake of one that exists, but am considering writing one, however this is just in the idea phase. |
| 07:37.33 | Preston | This is of course assuming that the SIP call is being placed to the same server. |
| 07:38.11 | Preston | IAX2 supports multiplexing. |
| 07:38.12 | Preston | hmm |
| 07:38.33 | Preston | Does that mean what I think it means? |
| 07:45.43 | [TK]D-Fender | <PROTECTED> |
| 07:46.14 | [TK]D-Fender | An RTP steam goes to ONE client. Nothing more. SIP is not a "braondcast" tool |
| 07:46.23 | [TK]D-Fender | Nor is IAX2 |
| 07:46.56 | Preston | http://www.voip-info.org/wiki/view/Asterisk+MulticastRTP+channels |
| 07:47.05 | Preston | So that isn't what I think it is? |
| 07:47.18 | [TK]D-Fender | IAX2 trunk mode is the multiplexing they are referring to. This bonds multiple calls to a SINGLE endpoint into a shared packetization to save on overhead |
| 07:49.01 | [TK]D-Fender | news to me... |
| 07:49.28 | Preston | Seems like a SIP packet could say, "Listen to this RTP stream on this IP:port" to multiple clients. |
| 07:50.04 | [TK]D-Fender | that isn't just listen. * has to SEND |
| 07:50.08 | Preston | so that if multiple clients wanted to hear the same thing, they wouldn't have to broadcast it between the endpoint servers N number of times. |
| 07:50.18 | [TK]D-Fender | it is still N. |
| 07:50.19 | Preston | exactly |
| 07:50.21 | [TK]D-Fender | * has to send to each |
| 07:50.31 | Preston | Asterisk would send a call to multiple clients. |
| 07:50.34 | Preston | who would pick up |
| 07:50.36 | Preston | and hear the same message |
| 07:50.51 | Preston | "There is a severe thunderstorm warning for this area. There is a severe thunderstorm warning for this area. There is a severe thunderstorm warning for this area." etc |
| 07:51.23 | Preston | All through one RTP audio stream. |
| 07:51.36 | Preston | At least that's what I'm guessing that MulticastRTP is. |
| 07:52.07 | [TK]D-Fender | Looks vagues and your end goal seems to be real PSTN numbers... not sure what service could work with that |
| 07:52.29 | Preston | Yeah |
| 07:52.33 | Preston | I could be working into a dead end. |
| 07:55.11 | Preston | http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom |
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| 09:31.00 | _zoom_ | hi, any recommended integrator for sugarcrm ce? |
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| 09:49.46 | _zoom_ | hi, need good sugarcrm connector? |
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| 10:17.34 | henningb | Does anyone have experience with a non working MusicOnHold is not working in queues, when its works anywhere else? |
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| 14:18.44 | Nebukadneza | heho |
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| 14:29.23 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:31.13 | Nebukadneza | say, for a scenario where i have a couple of sip softphones and i'd like to connect them to each other with support for some kind of nat traversal / firewall punchholing (whatever it may actually be) and the users should be able to self-register themselves ... can asterisk help me somehow for that scenario (without excessive use of external tools)? |
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| 14:35.05 | Freeaqingme | does anybody have any opinions on Elastix? |
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| 14:57.39 | [TK]D-Fender | Freeaqingme: Uses an older forked & embedded version of FreePBX. Most non-suers will not want to bother having to try to support you on it the second those diffrences start being problematic |
| 14:58.15 | [TK]D-Fender | Freeaqingme: If you seriously want a GUI, pick one of the standard installers like AsteriskNOW, the FreePBX direct ISO, or roll your own. |
| 14:58.20 | Freeaqingme | hmmz, I never realized it used freepbx |
| 14:58.53 | Freeaqingme | I dont know if i want a GUI (I'd rather not in fact), but replicating the functionality elastix provides would take a serious investment ;) |
| 15:01.26 | [TK]D-Fender | what do you really NEED? |
| 15:02.51 | Freeaqingme | uhm, costbased trunk selection, I gotta have some follow me set up, tie it together with gtalk, have voicemails sent to email, some conferencing stuff, and a little video over sip |
| 15:02.54 | Freeaqingme | @ [TK]D-Fender |
| 15:03.01 | Freeaqingme | and yes, I can do that just fine with plain asterisk ;) |
| 15:03.44 | [TK]D-Fender | the LCR stuff I wonder what Elastix offers for that... |
| 15:04.22 | Freeaqingme | [TK]D-Fender, I hadnt looked at it seriously yet, I just noticed it was also part of its feature list |
| 15:04.37 | Freeaqingme | I had to know the disadvantages first. The fact that it uses freepbx makes it a nogo for me |
| 15:04.57 | *** join/#asterisk hehol (~Adium@ip-178-202-243-147.unitymediagroup.de) |
| 15:05.13 | [TK]D-Fender | because? |
| 15:05.50 | Freeaqingme | because I've had some really bad experiences with freepbx, although that's an ancient (1.4) install |
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| 16:28.07 | Elanar | hey guys :) |
| 16:28.36 | Elanar | i hope you can help me... but please excuse my bad english :D |
| 16:29.55 | Elanar | i connect 2 softphones with my asterisk server. and all is working fine... BUT... i got no sound. If one side press "hold" in x-lite and press it again, then you can hear the side which has pressed hold before... |
| 16:30.46 | Elanar | its mysterious :D |
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| 16:57.19 | Elanar | no one who knows my problem and know, how to fix it? |
| 17:03.48 | devdvd | Elanar, |
| 17:04.19 | devdvd | are the 2 phones on the same network as the * server? |
| 17:04.24 | devdvd | i.e no nat? |
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| 17:40.44 | Elanar | devdvd: no they are not... but with rtT at the end of the exten all works fine :) |
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| 18:40.07 | danfromuk | Anyone know if the Nortel i2007 NTDU96 works on Asterisk? |
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| 18:48.32 | ChannelZ | danfromuk: if they have SIP firmware for it, probably |
| 18:49.00 | danfromuk | looks like they dont |
| 18:49.02 | danfromuk | http://www.voip-info.org/wiki/view/Asterisk+UNISTIM+channels |
| 18:49.44 | danfromuk | Anyone used chan_unistim? |
| 18:53.16 | [TK]D-Fender | danfromuk: Do ou already own the phones? |
| 18:54.03 | danfromuk | No. They looked quite good, so wanted to test them out. |
| 18:56.21 | [TK]D-Fender | danNortel is DEAD, and Unistim was a horrble mistake. |
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| 18:57.03 | danfromuk | Explains why there is one on ebay for £20 |
| 18:59.39 | danfromuk | ok, thanks for the pointer. |
| 18:59.41 | danfromuk | i'll avoid |
| 18:59.59 | danfromuk | what phones are people successfully using with asterisk, and would recommend? |
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| 19:00.07 | danfromuk | we currently focus on polycom |
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| 19:06.09 | WIMPy | Not sure I can recommend a SIP phone, but the old Snoms have been the best I've seen so far. |
| 19:06.21 | mariusno | i have tried a policom against asterisk, worked good, but was a hassle to configure it. |
| 19:07.16 | mariusno | dont remember why... but we used the web interface after first destroying the firmware on one, while uploading via tftp |
| 19:08.32 | danfromuk | i love the polycoms. just dont think they look very professional. i believe people like to see lots of buttons. like the yealink phones. |
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| 19:09.43 | WIMPy | Yes. Yealink looks interesting, but they seem to be as rare as Polycom. |
| 19:10.01 | ChannelZ | Seems like they have lots of buttons to me. The phones are just shaped stupid |
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| 19:11.37 | [TK]D-Fender | Most people dont need "lots of buttons" |
| 19:12.37 | [TK]D-Fender | Last I checked for the UK, Linksys was the best value phone around. |
| 19:15.30 | carrar | install lots of buttons on the wall |
| 19:15.33 | carrar | and flashing lights |
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| 19:18.49 | danfromuk | Ive tried linksys. The users didnt like them. |
| 19:19.13 | danfromuk | plus, the ones we tried enter 'end of support' almost straight away |
| 19:22.31 | [TK]D-Fender | Well as long as that is a nicely qualified answer.... |
| 19:22.46 | [TK]D-Fender | That helps you actually determine the points to aim for... |
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| 19:24.19 | danfromuk | We tried the spa921 i think. There were problems with the firmware and transfering calls when behind a nat. |
| 19:24.46 | danfromuk | Never had a problem with the polycoms, so we forgot about the linksys phones altogether |
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| 19:35.58 | aruntomar | i've a weird problem, dahdi channels show up in cat /proc/dahdi/2 but no spans are showing up in dahdi_tool. even inside asterisk console 'dahdi show status' gives error command not found. this server was earlier running things perfectly. is it that my dahdi modules have got screwed up? |
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| 19:42.08 | [TK]D-Fender | It means chan_dahdi isn't loaded |
| 19:42.18 | [TK]D-Fender | could mean you didn't have DAHDI installed first |
| 19:42.43 | [TK]D-Fender | Or you have a modules.conf SNAFU |
| 19:42.50 | [TK]D-Fender | or the channel configs are screwed up |
| 19:45.32 | aruntomar | [TK]D-Fender: i check the module "module show like dahdi" and it's there, reloaded it also. |
| 19:46.00 | [TK]D-Fender | PB all of it |
| 19:49.37 | aruntomar | [TK]D-Fender: i've a redfone appliance and this server was running for the past 2 yrs. got the appliance replaced after a voltage fluctuation blew it. now facing this problem. config: centos 5.5, asterisk 1.4, dahdi 2.3.0, libpri 1.4. should i upgrade all of them to the latest stable versions? |
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| 21:17.28 | hesco | None of the ITSP's I've researched seem to have the DIDs I need in rural communities. Where do the ITSPs get their DIDs and how difficult would it be for me to go directly to their source? |
| 21:20.21 | ChrisInSydney | hesco: Which country. US i assume |
| 21:20.28 | ChrisInSydney | ? |
| 21:22.28 | hesco | Yes, US, Georgia specifically |
| 21:22.47 | hesco | though if this plays out we might want to later expand from there |
| 21:24.20 | ChrisInSydney | I'm in Aus, so not a lot of help..but...I vaguely remember DIDX has pretty good SIP coverage for DIDs |
| 21:25.36 | ChrisInSydney | depends on how many you need, and what you are doing, maybe set up a PRI with a conventional wholesaler. Termiante to SIP there |
| 21:26.01 | ChrisInSydney | otherwise, there is the googles |
| 21:27.06 | hesco | yes, I've been searching google quite a bit. but turning up nothing useful. |
| 21:27.42 | ChrisInSydney | I have a wholesaler here who can pretty much get anywhere, but I have to take a 100 block. If there is no coverage I have to commit to a 10000 block :-( |
| 21:27.53 | hesco | one provider I found has coverage in many (not all) of the communities I want, but a limited supply of them. |
| 21:27.56 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 21:28.54 | hesco | I could imagine buying a 100 block for a market at a time if I had to. if the prices were right. |
| 21:28.58 | ChrisInSydney | if you are in that niche market, you may have to use a T1 or so to terminate inbound and overstamp the DID for outbound with your SIP wholesaler |
| 21:29.35 | hesco | I can imagine buying a T1, but if I do, where do the DIDs come from? |
| 21:30.13 | ChrisInSydney | The wholesaler you buy off should be able to give you a conventional number block |
| 21:31.49 | ChrisInSydney | If I was doing that here, I would fiind a wholesaler / retailer in the area, grab a local DID block, set up an on network exchange based divert for that number block to an E1 in a metro data centre and route calls via SIP from there |
| 21:33.04 | ChrisInSydney | on net diverts shouldn't cost anythng once set up except they charge per month or per annum per number |
| 21:34.04 | ChrisInSydney | I would be asking your current provider how they would go about it. My guys are pretty helpful here |
| 21:34.27 | hesco | thats an idea |
| 21:34.55 | hesco | I thought they'd resent the competition and my idea to buy these w/o their involvement. |
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| 21:35.43 | WIMPy | Just don't ask experts. |
| 21:36.12 | hesco | so in addition to the T! and number block, I'd face only bandwidth charges and SIP termination fees? |
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| 21:36.58 | ChrisInSydney | pretty much, |
| 21:37.44 | ChrisInSydney | depending on the provider, if you are getting DIDs out of their area, why whould they care ? especially if they are getting the tolling for outbound |
| 21:38.29 | hesco | right |
| 21:38.40 | hesco | well thanks, this has been helpful. |
| 21:39.06 | ChrisInSydney | so depending on how "easy" it is to get the numbers forwarded or what infrastructure is in place in the community, and any required SLAs will dictate what you will do |
| 21:40.46 | ChrisInSydney | you can always grab a Cisco 28xx router with the required PVDMs and a dual E1/T1 card and install it in a servced office or at a local ISP in the community |
| 21:41.51 | ChrisInSydney | or an Asterisk box with a card in it. Thats what we are here for :-) |
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| 21:43.34 | WIMPy | What are you guys planning? |
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| 21:45.31 | ChrisInSydney | World domination |
| 21:45.54 | ChrisInSydney | while the girls are off planning the wedding |
| 21:45.57 | WIMPy | That's a german monopoly. |
| 21:45.57 | ChrisInSydney | :D |
| 21:46.11 | WIMPy | And it seems to work quite well again. |
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