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| 01:34.49 | nsgn | goodevening. i've got need for a solution supporting 5 PoTS lines and 10 internal VoIP extensions. the budget is low and i'm very interested in asterisk. i'm intending to use the open source package installed on a whitebox. any hardware recommendations or advice for a first timer with asterisk would be much appreciated |
| 01:36.00 | Qwell | nsgn: 5 lines? something like a Digium TDM800 with 2 quad FXO modules. That'll give you 8 ports, if you ever need to expand. |
| 01:38.17 | nsgn | Qwell: thanks for the answer. i've been looking at that exact combination, actually. seems wonderful but the equipment is still pretty expensive. is using digium's PCI cards and modules a requirement? the pricing isnt that much greater than other entry level PBX solutions when you add in the whitebox |
| 01:38.38 | Qwell | requirement, no |
| 01:39.21 | jaytee | nsgn, yeah the TDM800 is a good card. For phones I prefer Polycoms, they're not as cheap as some other brands but the audio quality, performance and features are excellent. Avoid Grandstream phones at all costs. Cheap junk that will have you pulling your hair out. |
| 01:39.47 | Qwell | ~cheap |
| 01:39.47 | infobot | [cheap] a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 01:39.47 | Qwell | heh |
| 01:40.21 | Qwell | there's a huge difference between cheap and inexpensive. unfortunately, in telephony, cost usually does matter |
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| 01:40.32 | Qwell | (read: if it's cheap, it's cheap for a reason.) |
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| 01:40.54 | nsgn | jaytee: thanks for the polycom recommendation. i've been looking at them |
| 01:41.08 | nsgn | advice on more affordable FXO interfacing? |
| 01:41.26 | Qwell | you could do 5 FXO ATAs, but...ick |
| 01:41.51 | nsgn | Qwell: haha, sounds amusing. tell me how such a setup would work? what sort of ATAs are supported? |
| 01:41.54 | jaytee | nsgn, there are cheaper cards out there to connect analog POTS lines or digital PRI lines but they aren't as reliable. Digium and Sangoma make the best cards, highly reliable for the prices. Digium actually contributes to the open source version of Asterisk, it's CEO is the creator of the software originally. Sangoma makes good hardware but they don't really contribute to the open source effort. |
| 01:42.02 | Qwell | is that Linksys that [TK] recommends an 8-port FXO? |
| 01:42.09 | Qwell | is it linksys? I don't even know, heh |
| 01:42.20 | jaytee | no, it's FXS |
| 01:42.27 | jaytee | the SPA8000? I've got 4 of em. |
| 01:42.28 | Qwell | jaytee: s/don't really/don't at all/ :) |
| 01:42.35 | Qwell | jaytee: yeah that |
| 01:43.10 | nsgn | yeah, googled SPA8000. its FSX. i need to get lines into the system |
| 01:43.13 | nsgn | so FXO |
| 01:43.19 | nsgn | *FXS |
| 01:43.29 | jaytee | actually when it comes to pricing the TDM800 isn't as pricey as some of the rack mount 8 port FXO/SIP media gateways out there. It's usually several hundred dollars cheaper |
| 01:44.17 | stope | is there an easy way to disable voicemail and call waiting? |
| 01:44.20 | nsgn | jaytee: yeah. it's an option i'm seriously considering. i just wanted to eyeball any less expensive ways to do it and weigh my decision |
| 01:44.24 | Qwell | nsgn: and to answer your earlier question.. anything SIP, basically |
| 01:44.36 | jaytee | nsgn, FXO is to connect POTS lines to Asterisk. FXS ports connect analog phones to asterisk if you want to use analog phones. I'd recommend going with SIP phones if you're going to move to VOIP you might as well do it right. |
| 01:45.10 | nsgn | jaytee: yes, i'm just needing an FXO interface to get my 5 analog lines into asterisk. from there it's voip on the desks |
| 01:45.21 | jaytee | nsgn, if you get the TDM800 or even if you get a Sangoma card, I highly recommend you get the add on hardware echo cancellation module. |
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| 01:47.10 | jaytee | nsgn, you might go with a TDM410 card that supports 4 FXO modules to start and get a LInksys SPA 3102 ATA. It has 1 FXS and 1 FXO port on it and will connect to * via SIP over ethernet. Bit of a pain to configure compared to Digium cards in the server though. |
| 01:47.40 | nsgn | Qwell: ok. seems like i'd looked into asterisk a few years ago and had finally dug up some sort of compatible SIP FXO. no clue what it was at this point. anything recommended to get a few lines in cheap? |
| 01:48.01 | Qwell | nsgn: no - and there's that word. :) |
| 01:48.04 | nsgn | jaytee: if i'm getting the TDM4100 why do i need the ATA? |
| 01:48.23 | nsgn | *inexpensive ;) |
| 01:48.38 | Qwell | nsgn: the TDM410 is a 4-port card. I assume he meant the ATA for the 5th |
| 01:48.40 | jaytee | cheap? the X100 clone cards on Ebay. Usually the people that like those cards also like whips and chains and being sodomized though. |
| 01:49.03 | jaytee | yep, I meant the ATA as the 5th |
| 01:49.07 | nsgn | jaytee: haha, i remember discussing the x100 cards before. almost bought one at one point |
| 01:49.31 | nsgn | issue is that stacking 5 of them in a PC isnt much fun :) |
| 01:49.40 | nsgn | if you can even find a board that will let you |
| 01:49.44 | Qwell | stacking 1 isn't fun :p |
| 01:50.15 | jaytee | nsgn, I used them in initial testing then bought TDM410's. The X100 clones are cheap crap and I would often have to do a cold reboot of the server when one of the lines would "lock" |
| 01:50.29 | nsgn | so if i go digium it will just pop in and configure a breeze? for a small office never having played with this much ourselves is combining one decent digium card and AsteriskNOW a good game plan? |
| 01:51.41 | Qwell | AsteriskNOW... somebody should buy the guy who wrote that a beer. |
| 01:51.52 | Qwell | several beers even! |
| 01:52.03 | jaytee | AsteriskNOW 1.5 is pretty solid if you really need a GUI and don't need anything out of the ordinary in terms of custom applications. LIke most gui based derivitives of Asterisk it tends to limit what you can do in the dialplan. |
| 01:52.05 | Qwell | wonders if that'll work |
| 01:53.16 | nsgn | jaytee: we just need some pretty basic functionality. bring calls in from PoTS to ring all extensions. transfer calls between extensions internally, forward calls, voicemail |
| 01:53.21 | jaytee | Qwell, you should just start a web page with a PayPal donate button on it with a message saying, "Help support AsteriskNOW, buy the developer a beer...or two!" |
| 01:53.35 | Qwell | jaytee: heh |
| 01:53.50 | Qwell | jaytee: I can't take bribes^Wdonations. :) |
| 01:54.15 | jaytee | AsteriskNOW will do all that. it's when you want fancy and granular control or the ability to do "outside the box" type of work or customization that the gui versions aren't as "flexible" |
| 01:54.38 | jaytee | of the GUI versions though, AsteriskNOW is better than the rest IMHO |
| 01:54.47 | nsgn | jaytee: can you think of something a small office currently using a low end NEC phone system + voicemail system would miss using AsteriskNOW? |
| 01:56.51 | jaytee | nsgn, no not really. It has voicemail, call parking, conference bridging as well as Call Detail Recording all built in. Fairly rich in features. |
| 01:56.52 | nsgn | yeah, i figured it would kick the butt of any crappy standalone solution |
| 01:57.15 | nsgn | ok, so i need to get a PCI 1x card and two modules it seems |
| 01:57.22 | nsgn | how critical is echo cancelation? |
| 01:58.13 | jaytee | as an example of where I wouldn't want to use AsteriskNOW over standard Asterisk would be in a call center that does mostly outbound calling and uses a CRM database to pull customer numbers and initiate calls that get dumped in a queue for the agents with some customized connector for screenpops. Scenarios like that. |
| 01:58.51 | nsgn | yeah, nothing like that here |
| 01:59.06 | nsgn | this is pick up the phone, dial, talk, park/transfer, voicemail |
| 01:59.19 | jaytee | echo cancellation is very critical. Asterisk through the DAHDI (formerly zaptel) interface has software echo cancellation but it's not a solid as using hardware. |
| 01:59.29 | nsgn | oh, i'm seeing these single channel modules i didnt spot before. i can go with a 4 channel and a 1 channel FXO |
| 02:00.04 | nsgn | jaytee: so for the small office setup i'm describing is it worth $250 for the hardware cancelation or can we get by with software? |
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| 02:01.06 | Qwell | nsgn: you can do that, if you don't plan on expanding. the quad modules end up cheaper (since you can only put 2 singles in the space of a quad slot) |
| 02:01.28 | jaytee | nsgn, I'd go with the hardware module. Even with the best tuned software setup I still would notice a slight echo at the beginning of each call as it "trains itself" and it varies from call to call depending on the connection quality from the telco. |
| 02:01.43 | nsgn | ok |
| 02:01.56 | Qwell | jaytee: HPEC should be about as good, if you've got the CPU to spare |
| 02:02.10 | Qwell | (non-free, of course) |
| 02:02.41 | jaytee | Qwell, I never played with it. It wasn't ready for prime time by the time I moved everything to PRI using the TE212P cards with HWEC module. |
| 02:02.45 | nsgn | oh nice am i correct in seeing that we can get a discount by purchasing modules pre attached to cards? |
| 02:02.58 | Qwell | nsgn: possibly - depends on the reseller I guess |
| 02:03.06 | nsgn | http://store.digium.com/productview.php?product_code=1TDM808EF |
| 02:03.10 | nsgn | how does that look? |
| 02:03.33 | nsgn | thats 8 FXO ports + hardware echo canceling |
| 02:04.13 | jaytee | nsgn, yep. There are several outlets that sell Digium cards. www.telephonydepot.com, www.voipsupply.com and others. Their prices tend to be lower than Digium's website because they buy in high volume and discount heavily. |
| 02:04.15 | Qwell | looks right - just make sure you've got standard PCI |
| 02:04.42 | Qwell | that card also comes in a PCI-E variant |
| 02:04.45 | nsgn | Qwell: i'm building the whitebox so i'll be sure i have a free PCI slot |
| 02:04.47 | nsgn | ok |
| 02:04.53 | nsgn | jaytee: thanks, checking price there |
| 02:06.18 | jaytee | nsgn, for the PCI cards they make different models. I had to buy the TDM04B model for the 3.3v PCI slots in Dell 2650 (shitty server). Then I moved them to a 2950 before I ended up switching them out for TE212P cards for T1 PRI spans. |
| 02:06.48 | Qwell | 410/800 are 3.3v/5v |
| 02:07.06 | Qwell | I think only the T1 cards are one or the other |
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| 02:07.20 | nsgn | yeah, the card does both |
| 02:07.22 | jaytee | ah, that's right. it was the earlier card, the TDM400? |
| 02:07.33 | Qwell | jaytee: I think it's only been that way ever for the T1 cards |
| 02:07.40 | jaytee | that had two different "sub models" |
| 02:07.52 | Qwell | the "sub models" on the 400 were the modules |
| 02:07.58 | Qwell | that's what the B means - bundle |
| 02:08.22 | Qwell | I never remember which is which, but TDM04B is either 4 FXO or 4 FXS modules, bundled with the base card |
| 02:08.40 | jaytee | o4B was 4 FXO |
| 02:08.42 | Qwell | it's confusing, heh |
| 02:08.52 | Qwell | like the model he linked above - TDM808EF |
| 02:09.30 | jaytee | yeah, but I swear to God there was an earlier rev 4 port analog card that came in one PCI spec or the other. Maybe I was just confused. Things were a bit rushed back then. |
| 02:09.56 | nsgn | telephonydepot doesnt have the same bundle as i linked at digium. checking voipsupply |
| 02:10.12 | Qwell | nsgn: I think telephonydepot lets you choose models as part of the order |
| 02:10.19 | Qwell | choose modules* |
| 02:10.37 | nsgn | http://www.voipsupply.com/aex808e |
| 02:10.42 | nsgn | would that be the exact item i linked at digi? |
| 02:10.47 | jaytee | they do, if you select a specific card it then gives you options for the mods |
| 02:10.50 | Qwell | aex is the pci-e version |
| 02:10.56 | nsgn | ah, ok |
| 02:11.04 | nsgn | i can really do either |
| 02:11.09 | jaytee | those are sweet cards if you've got the bucks |
| 02:11.09 | nsgn | but it seems the pci version is a bit cheaper |
| 02:11.37 | Qwell | if you do end up getting the modules separately, you want the X400Ms |
| 02:11.43 | nsgn | jaytee: what other options are there? yall kindof lead me above to believe i need to go digium and need the echo |
| 02:12.33 | nsgn | and i really dont want to put up with something acting stupid |
| 02:12.46 | nsgn | is it cheaper getting the parts individually? |
| 02:13.12 | Qwell | nsgn: probably not |
| 02:13.23 | Qwell | I don't know though - it really is up to the reseller |
| 02:13.36 | nsgn | wait...looks like it is at telephonydepot? |
| 02:13.39 | nsgn | http://www.telephonydepot.com/Catalog/Digium-TDM800P/Digium-TDM800P-Blank-Board |
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| 02:13.44 | nsgn | thats the page i'm on |
| 02:14.03 | nsgn | says the card is $255, the modules are $175 each and the echo module is $235 |
| 02:14.11 | nsgn | that makes 840 |
| 02:14.19 | nsgn | am i missing something? |
| 02:15.00 | Qwell | nope, it's not uncommon for a reseller to be less |
| 02:15.21 | nsgn | i'm just surprised telephonydepot individually seems to beat voip supply selling it as a bundle |
| 02:15.24 | nsgn | but sweet |
| 02:15.33 | Qwell | apparently voipsupply doesn't carry that exact combo. |
| 02:15.35 | nsgn | $840 for all that sweet hardware is not bad at all |
| 02:15.58 | nsgn | because other than phones and a whitebox to drop it all in, it's all i need, correct? |
| 02:16.10 | jaytee | yes |
| 02:16.15 | nsgn | i'm liking this :) |
| 02:16.24 | nsgn | time to find some good phones on the same site |
| 02:16.25 | jaytee | consider other things like power and cabling |
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| 02:17.13 | Qwell | jaytee: get a load of this combo. why would anybody buy this? http://www.voipsupply.com/digium-tdm8s4e |
| 02:17.15 | jaytee | I went with the Polycom IP 330 phones except for executive admins that needed more than 2 lines and for that I went with the IP 550 models. |
| 02:17.21 | nsgn | jaytee: if by power you mean operational costs, of course |
| 02:17.45 | lesterc | guys - which of the following is right: |
| 02:17.47 | lesterc | exten => s,n,GotoIf($[${LEN($(CALLERID(num)))}=3 & ${CALLERID(num)}>900]? |
| 02:17.47 | lesterc | or |
| 02:17.51 | nsgn | jaytee: and by cabling i'm hoping to have the phone and data on the same (gigabit) network. any reason not to combine them in a small office? |
| 02:18.00 | lesterc | exten => s,n,GotoIf($[${LEN($(CALLERID(num)))}=3] & $[${CALLERID(num)}>900]?foo:bar) |
| 02:18.03 | jaytee | nsgn, no I mean is there an ac outlet close enough to the phone or would you need PoE |
| 02:18.14 | nsgn | jaytee: oh, yeah. no everything has power nearby |
| 02:18.21 | nsgn | i'm not gonna bother with poe for this setup |
| 02:18.44 | jaytee | and also do you plan on using a single CAT5 cable run for both phone and computer or building a separate voice network? |
| 02:19.10 | Qwell | if you do a single run, you aren't going to get gbit at the desktop (the passthrough ports on most phones are 100mbit) |
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| 02:19.41 | Qwell | although, that is one thing Cisco has gotten right. I believe they have a few gbit models now |
| 02:19.42 | nsgn | jaytee: some locations may be single run, but i'll be trying for dual wherever possible |
| 02:19.52 | jaytee | I work for a zoo and we're non-profit so to keep costs down we went with the Polycom 330's and 550's because they have a second ethernet jack in them to connect the computer up to so you only need one network cable run for both. |
| 02:19.54 | nsgn | i dont mind passing through a phone being 10/100 |
| 02:20.02 | nsgn | i just want the flexibility |
| 02:20.13 | nsgn | it's minimal extra config to run the two side by side i'd presume? |
| 02:20.25 | nsgn | disable DHCP if asteriskNOW has it by default, etc? |
| 02:21.00 | jaytee | Polycoms manage the traffic very well. I've yet to experience jitter on them. Even with our Cisco switches configured with QoS the Grandstream cheap phones I tested originally were absolutely horrible. |
| 02:21.16 | nsgn | http://www.telephonydepot.com/Catalog/Polycom-Phones/Polycom-Soundpoint-IP-320 |
| 02:21.26 | nsgn | basic polycom phone. actually supports PoE. i'm impressed for the price |
| 02:21.38 | Qwell | nsgn: IF you need passthrough, you want the 330 instead of the 320 |
| 02:21.47 | Qwell | that's the only difference between the two, so keep that in mind |
| 02:21.48 | nsgn | Qwell: ah, found it. thanks |
| 02:22.04 | nsgn | only issue i have with these is that the screen doesn't seem to have clear park functionality |
| 02:22.05 | jaytee | thanks Qwell, was about to point that out |
| 02:22.09 | nsgn | as much as other makes/models |
| 02:22.13 | nsgn | which i find really valuable |
| 02:22.34 | jaytee | parking is done using feature codes in * with a parking lot extension |
| 02:23.00 | nsgn | jaytee: is there no support for the phone's screen visually aiding in this? |
| 02:23.11 | nsgn | i've seen some mid level NEC systems that do it so slick |
| 02:23.27 | jaytee | not on the phones but the Polycoms have softkeys below the display. When you're on a call they give you 3 way conference, transfer etc |
| 02:23.50 | jaytee | and it's on the display |
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| 02:23.55 | nsgn | so if i'm on a call and wish to park it? |
| 02:24.02 | nsgn | i transfer it to a park extension? |
| 02:24.09 | Qwell | pretty much |
| 02:24.09 | jaytee | yep |
| 02:24.10 | nsgn | weird, but ok. |
| 02:24.27 | Qwell | the whole parking concept is a bit of a hack, if you think about it |
| 02:24.28 | nsgn | so if i was on a different phone and wanted to grab the call from park what would i do? |
| 02:24.39 | Qwell | dial an extension to pickup a parked call |
| 02:24.41 | nsgn | yeah it is. some systems just make it super slick |
| 02:24.51 | nsgn | so you just dial the park's extension, the park picks up and hands you the call? |
| 02:24.51 | Qwell | so, by default, 700 is to park, and it'll park on 701, 702, 703, or whatever |
| 02:24.52 | jaytee | when you park a call Asterisk tells you the "parking slot" so you dial that |
| 02:25.00 | VaGoNeTaS | guys |
| 02:25.00 | jaytee | what he said |
| 02:25.07 | jaytee | oh shit |
| 02:25.19 | VaGoNeTaS | finally i got the Redfone Quad working |
| 02:25.24 | VaGoNeTaS | but, still got a prob |
| 02:25.29 | VaGoNeTaS | i can make local calls |
| 02:25.33 | VaGoNeTaS | i can receive calls |
| 02:25.37 | nsgn | jaytee: so a voice says "you've parked this on 703" or something? |
| 02:25.41 | VaGoNeTaS | i can make national long distance calls |
| 02:25.44 | Qwell | nsgn: exactly |
| 02:25.48 | VaGoNeTaS | but i cant call cel phones |
| 02:25.48 | jaytee | nsgn, yes |
| 02:25.53 | VaGoNeTaS | http://pastebin.ca/1423304 |
| 02:26.10 | nsgn | Qwell: ok, thanks. in an office this size i imagine ittl be more transferring than parking. the advantage of park is that the call can wait without ringing the phone off the hook, correct? |
| 02:26.15 | VaGoNeTaS | somebody can help me out with this? |
| 02:26.38 | VaGoNeTaS | i've tried with dahdi, now i've downgrade from dahdi to zaptel |
| 02:26.44 | Qwell | I worked for a few days in an office with call parking, and I didn't really understand the reason for it.. |
| 02:26.56 | VaGoNeTaS | still says CHANUNAVAIL |
| 02:26.59 | Qwell | I'm sure it's useful on some occasions |
| 02:27.08 | nsgn | Qwell: is it what i said above? to allow a caller to wait without ringing the person they're trying to get to constantly until they answer? |
| 02:27.14 | nsgn | thats how i've seen it used |
| 02:27.14 | Qwell | but, blind and attended transfers will cover 99% of it |
| 02:27.22 | lesterc | in case anyone want so to know - the correct syntax is exten => s,n,GotoIf($[ $[${LEN(${CALLERID(num)})}=3] & $[${CALLERID(num)}>900]]?foo:bar) |
| 02:27.35 | VaGoNeTaS | today came the telco technician and he just tested the E1 and the call went through |
| 02:27.36 | jaytee | VaGoNeTaS, you made a successful call to the console or from it? |
| 02:27.42 | VaGoNeTaS | from it |
| 02:27.43 | Qwell | nsgn: yeah, that would be one reason I guess |
| 02:27.56 | nsgn | secretary answers, parks the call, notifies the intended recipient that they've got a call on park 1, and the recipient may make the caller wait 4 minutes before picking up, during which his phone is not having to ring and ring |
| 02:28.17 | VaGoNeTaS | im able to receive calls , make local and national long distance calls |
| 02:28.28 | VaGoNeTaS | but im not able to make calls to cel phones |
| 02:28.36 | VaGoNeTaS | we use a prefix here in chile |
| 02:28.40 | VaGoNeTaS | im located in Chile btw |
| 02:28.44 | Qwell | nsgn: assuming the recipient isn't on the phone, you could attended transfer. that's where the receptionist starts a transfer, dials their number, and if they answer, they complete the transfer. if they don't, they keep the call |
| 02:29.06 | Qwell | In the case that they don't answer, I guess that would be when you could park? dunno |
| 02:29.21 | Qwell | I just write code. I don't use phones. :p |
| 02:29.21 | jaytee | VaGoNeTaS, what about using a softphone or a real SIP phone to test instead of calling from the console? your console DSP keeps going AWOL and I'm betting it's got nothing to do with either DAHDI or ZAPTEL |
| 02:29.45 | VaGoNeTaS | jaytee i did |
| 02:29.49 | VaGoNeTaS | i've tried with Twinkle |
| 02:29.56 | VaGoNeTaS | and the call didnt go through |
| 02:30.03 | VaGoNeTaS | same error on the CLI |
| 02:30.09 | VaGoNeTaS | CHANUNAVAIL |
| 02:30.13 | VaGoNeTaS | today came the technician |
| 02:30.21 | nsgn | Qwell: so is that 330 the phone i want, or is there anything else affordable you think may be of my interest? |
| 02:30.22 | VaGoNeTaS | and he made a test call to a cel from his device |
| 02:30.31 | VaGoNeTaS | and the call went through |
| 02:30.37 | jaytee | bypassing Asterisk? |
| 02:30.39 | VaGoNeTaS | so is not a Line block |
| 02:30.44 | VaGoNeTaS | yes |
| 02:30.46 | VaGoNeTaS | bypassing it |
| 02:30.49 | Qwell | nsgn: I have very little experience with anything else. I couldn't really give any recommendations |
| 02:30.52 | VaGoNeTaS | with his device |
| 02:31.03 | VaGoNeTaS | is like a tester with a screen |
| 02:31.07 | Qwell | I see some people recommending Linksys phones, but really, for the difference in price, it's well worth it |
| 02:31.08 | *** join/#asterisk EdLin (n=joshua61@securabit/listener/edlin) |
| 02:31.13 | VaGoNeTaS | and is also a telephone |
| 02:31.17 | jaytee | ok, so you know that your E1 is ok. You need to figure out why your /dev/dsp is locked up all the time. |
| 02:31.31 | VaGoNeTaS | yes |
| 02:31.37 | VaGoNeTaS | the E1 is ok, we know that |
| 02:31.41 | *** part/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 02:31.50 | nsgn | Qwell: ah, ok. thanks for all the help you provided. i'm gonna add up my pricing and present it tomorrow |
| 02:31.56 | VaGoNeTaS | and we are able to call local with no problem at all |
| 02:32.02 | nsgn | gotta get the boss people to like it, but i think this will be an easy sell |
| 02:32.08 | VaGoNeTaS | national long distance through a carrier |
| 02:32.09 | *** join/#asterisk securevoip (n=securevo@c-76-123-20-170.hsd1.va.comcast.net) |
| 02:32.13 | VaGoNeTaS | receiving calls |
| 02:32.15 | jaytee | VaGoNeTaS, do you have any SIP hardware based phones? |
| 02:32.22 | VaGoNeTaS | yes i have |
| 02:32.31 | jaytee | have you tested with those? |
| 02:32.32 | VaGoNeTaS | but not right here |
| 02:32.38 | VaGoNeTaS | no i havent |
| 02:32.44 | VaGoNeTaS | look |
| 02:32.46 | jaytee | I'd start with that. |
| 02:32.49 | VaGoNeTaS | we have the same provider |
| 02:32.56 | VaGoNeTaS | in a different location |
| 02:33.00 | VaGoNeTaS | just the same provider |
| 02:33.06 | VaGoNeTaS | here in Santiago de Chile |
| 02:33.21 | VaGoNeTaS | and we have the exact same dialplan |
| 02:33.30 | VaGoNeTaS | on the extensions.conf |
| 02:33.33 | Qwell | Who the heck is _mwoodj_, and why does he have a Digium sponsor tag from freenode? O.o |
| 02:33.39 | *** part/#asterisk EdLin (n=joshua61@securabit/listener/edlin) |
| 02:33.43 | VaGoNeTaS | and i can make calls to cel phones from the other place |
| 02:34.12 | jaytee | VaGoNeTaS, and when you call from the "other place" what are you using as a phone? softphone, console? |
| 02:34.24 | VaGoNeTaS | softphone and console |
| 02:34.29 | VaGoNeTaS | just like here |
| 02:34.31 | Qwell | file: any idea? O.o |
| 02:34.55 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-54e153af6b860fd1) |
| 02:35.14 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 02:35.20 | jaytee | VaGoNeTaS, pastebin the output of ls -ao /dev |
| 02:35.28 | VaGoNeTaS | k just a min |
| 02:35.49 | VaGoNeTaS | the whole /dev ? |
| 02:36.08 | jaytee | yeah |
| 02:36.32 | VaGoNeTaS | it a lot of |
| 02:36.40 | VaGoNeTaS | its* |
| 02:36.58 | securevoip | VaGoNeTaS, line 37 of your pastebin seems to indicate ISDN Cause Code 127 for cell calls. ISDN Cause Code info is here: http://isdn.modemhelp.net/causecodes/causecodelist.shtml |
| 02:37.24 | nsgn | ok, let me ask this. what is a ballpark figure that some company in town would charge if we told them we needed a PBX to support 5 incoming lines and 10 internal VoIP phones + voicemail? i'm trying to see if my grand total will look good to the boss |
| 02:37.33 | nsgn | since i've never called another phone company for a bid |
| 02:37.43 | securevoip | I hope I am wrong cause that is a most vague Cause Code |
| 02:38.18 | jaytee | yeah, I had that cause code come up last week on my system. the distant end was answering and putting the phone on hold then hanging up. |
| 02:38.42 | securevoip | nsgn, are you looking for numbers on an Avaya/Cisco/etc solution from an integrator OR numbers on an * solution from an integrator? |
| 02:38.58 | nsgn | securevoip: NEC/avaya |
| 02:39.26 | securevoip | about $750 per station with everything... |
| 02:40.07 | securevoip | of course, that is going to vary by locality. NY city is going to be more $ than Idaho probably... |
| 02:40.44 | Qwell | securevoip: he's in Austin, so probably not too far off |
| 02:40.53 | jaytee | VaGoNeTaS, why are you using OSS instead of ALSA? |
| 02:41.15 | VaGoNeTaS | i dont know |
| 02:41.22 | VaGoNeTaS | i've just formatted the server |
| 02:41.29 | VaGoNeTaS | and installed ubuntu 8.04 server on it |
| 02:41.33 | VaGoNeTaS | and then the packages |
| 02:41.48 | jaytee | did you install the ALSA packages? |
| 02:41.51 | VaGoNeTaS | libpri, libfb, fonulator, zaptel, and asterisk 1.4.22-rc5 |
| 02:41.54 | VaGoNeTaS | no, i didnt |
| 02:42.36 | VaGoNeTaS | do you think that could be a reason for me to not be able to make cel phone calls? |
| 02:42.39 | jaytee | do that and configure alsa.conf and search the WIKI at voip-info.org for console channel, there's also a short piece in the book about it. |
| 02:42.45 | jaytee | VaGoNeTaS, yes |
| 02:43.06 | jaytee | were you testing with Twinkle from the same server or another computer? |
| 02:43.07 | VaGoNeTaS | for being CHANUNAVAIL |
| 02:43.51 | VaGoNeTaS | but what i cant understand is why im still able to make local calls and receive calls from cel phones, local and longdistance |
| 02:44.07 | VaGoNeTaS | still unable* |
| 02:44.14 | jaytee | VaGoNeTaS, to use the console channel you need a sound card in the server and either OSS (obsolete pretty much) or ALSA for sound which is why the /dev/dsp isn't opening properly. |
| 02:44.48 | VaGoNeTaS | yes, that's correct but, what about when im trying to reach the cel from a Softphone? |
| 02:45.13 | jaytee | Softphone on the server? |
| 02:45.22 | VaGoNeTaS | nope |
| 02:45.24 | *** join/#asterisk securevoip (n=securevo@c-76-123-20-170.hsd1.va.comcast.net) |
| 02:45.25 | VaGoNeTaS | different |
| 02:45.27 | VaGoNeTaS | with a SIP account |
| 02:45.30 | jaytee | hmmm |
| 02:45.33 | jaytee | dunno |
| 02:45.51 | jaytee | you get chanunavail from that call attempt too? |
| 02:46.30 | VaGoNeTaS | yes Sr |
| 02:47.05 | VaGoNeTaS | tried through vpn and local net with different client pc |
| 02:47.09 | VaGoNeTaS | and softphone |
| 02:47.34 | jaytee | what codecs did you allow for the softphone in your sip.conf? |
| 02:47.36 | VaGoNeTaS | with a sip acc |
| 02:47.54 | VaGoNeTaS | k, letme take a look, just a sec |
| 02:48.15 | VaGoNeTaS | i dont have one setted up |
| 02:48.42 | VaGoNeTaS | http://pastebin.ca/1423320 |
| 02:48.47 | VaGoNeTaS | thats one of my sips accounts |
| 02:51.03 | carrar | try pasting the whole sip.conf |
| 02:51.40 | VaGoNeTaS | that's the whole sip.conf |
| 02:51.51 | carrar | heh |
| 02:52.05 | carrar | might check out the book |
| 02:52.06 | carrar | ~book |
| 02:52.07 | 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 |
| 02:52.08 | carrar | it's free |
| 02:52.29 | VaGoNeTaS | you were looking for a sip acc |
| 02:52.30 | VaGoNeTaS | xD |
| 02:52.34 | jaytee | VaGoNeTaS, yes. You need to have a [general] section at the top of your sip.conf file |
| 02:52.57 | VaGoNeTaS | like what? |
| 02:53.00 | VaGoNeTaS | do you have an ex? |
| 02:53.12 | jaytee | VaGoNeTaS, and I'd modify that sip account to look like this: http://pastebin.ca/1423324 |
| 02:53.35 | carrar | gsm!! |
| 02:53.46 | jaytee | but you may need to change the allow=alaw to allow=ulaw. I'm not sure which version of G.711 they use where you are. |
| 02:53.50 | carrar | what, no ulaw |
| 02:54.38 | jaytee | he's in Chile or Brazil. do they use alaw or ulaw? |
| 02:54.58 | Qwell | umm |
| 02:55.07 | jaytee | and he's routing over an E1 |
| 02:55.10 | Qwell | I should know this |
| 02:55.10 | carrar | know anyone on the brazilian womans soccer team? |
| 02:55.13 | Qwell | if it's E1, it's alaw |
| 02:55.37 | jaytee | that's what I thought at first but I wasn't 100% certain |
| 02:56.05 | jaytee | I use T1 cuz I'm 'Merikun |
| 02:58.35 | jaytee | VaGoNeTaS, but you'll still need to make sure there is a sound card in the server and install the ALSA packages so you'll actually have a /dev/dsp and then you'll need to configure your alsa.conf file. |
| 02:59.45 | VaGoNeTaS | k |
| 02:59.51 | VaGoNeTaS | so let me find alsa |
| 03:00.04 | VaGoNeTaS | download it to the server and then im gonna configure it |
| 03:00.56 | jaytee | sudo apt-get install alsa |
| 03:01.19 | Qwell | hands jaytee a shovel for the worms that are spilling out all over the floor |
| 03:01.20 | jaytee | and maybe sudo apt-get alsa-utils. it's been awhile |
| 03:01.32 | jaytee | nooooooo!!!! |
| 03:01.37 | Deeewayne | hands Qwell a fishing pole for the worms |
| 03:01.39 | VaGoNeTaS | fuck |
| 03:01.42 | VaGoNeTaS | i just did it |
| 03:01.58 | VaGoNeTaS | <PROTECTED> |
| 03:01.58 | jaytee | what? you just fucked? |
| 03:02.01 | VaGoNeTaS | no |
| 03:02.02 | VaGoNeTaS | hahaha |
| 03:02.19 | VaGoNeTaS | i've just installed alsa-base and linux-sound-base |
| 03:02.21 | Qwell | goes fishing for Cupcakes |
| 03:02.25 | jaytee | I hope it was better for you than it was for me cuz I'm feelin nuthin here! |
| 03:02.26 | VaGoNeTaS | right after u told me to |
| 03:02.47 | jaytee | so if you do ls /dev/dsp do you get anything? |
| 03:03.10 | VaGoNeTaS | no such file or directory |
| 03:03.28 | jaytee | and once again, is there a sound card (HARDWARE) in the server? |
| 03:03.44 | VaGoNeTaS | yes |
| 03:03.46 | VaGoNeTaS | the integrated |
| 03:03.57 | VaGoNeTaS | on the motherboard |
| 03:04.01 | jaytee | ok |
| 03:04.12 | jplank | is there anything wrong with this? (polycom digitmap) 9,011x.t |
| 03:04.16 | jaytee | do a reboot |
| 03:04.33 | jaytee | jplank, you asked that last nite |
| 03:04.51 | jplank | I didn't see anyone answer |
| 03:04.57 | jplank | and I'm still struggling with it |
| 03:06.05 | jaytee | the T is for Timeout and needs to be a capital T |
| 03:06.18 | VaGoNeTaS | i cant reboot the machine |
| 03:06.21 | VaGoNeTaS | im remotelly |
| 03:06.30 | VaGoNeTaS | :S |
| 03:07.26 | jaytee | are you using SSH? |
| 03:07.33 | VaGoNeTaS | yep |
| 03:07.53 | jaytee | can't you just type sudo shutdown -r now at the command line? |
| 03:08.07 | jaytee | and then reconnect after a few minutes? |
| 03:10.06 | jaytee | VaGoNeTaS, quick question. Were you able to call local numbers on the softphone? did that only fail on cell calls? |
| 03:10.16 | VaGoNeTaS | yes, that's right |
| 03:10.33 | VaGoNeTaS | in the begining |
| 03:10.39 | VaGoNeTaS | on "the other place" |
| 03:10.44 | VaGoNeTaS | we had problem to call local |
| 03:10.47 | jaytee | then at least you can fix your sip.conf, do a sip reload and retest a cell call with the gsm codec allowed |
| 03:10.53 | VaGoNeTaS | but it was coz of the pridialplan was setted up to national |
| 03:11.03 | VaGoNeTaS | so i've change it to unknown |
| 03:11.08 | VaGoNeTaS | and problem solved |
| 03:11.23 | VaGoNeTaS | but now we have the same configuration and is not letting us make calls to cell |
| 03:11.40 | VaGoNeTaS | i've tried changing it to national and it wont let me call nowhere |
| 03:11.44 | VaGoNeTaS | not even local |
| 03:12.06 | jaytee | obviously the configuration is not identical or it would work |
| 03:12.16 | jaytee | something is missing on that system |
| 03:12.29 | VaGoNeTaS | yes |
| 03:12.41 | VaGoNeTaS | and the E1 is ok |
| 03:12.46 | VaGoNeTaS | coz we tested it |
| 03:12.51 | jaytee | and since you didn't have a general section in the sip.conf you should at least fix that part before messing around with the rest. |
| 03:13.10 | jaytee | yes, we've already established the fact that the E1 is working. |
| 03:13.24 | VaGoNeTaS | what should be below [general] on the sip.conf file? |
| 03:13.46 | *** part/#asterisk tanner (n=tanner@unaffiliated/tanner) |
| 03:13.51 | jaytee | look at the sample sip.conf file in the /configs directory of the tarball |
| 03:13.58 | jplank | errr |
| 03:14.10 | jplank | thanks jaytee, that was pretty stupid of me |
| 03:14.38 | jaytee | jplank, what? |
| 03:15.38 | jplank | you were right t should of been capital, I should of realized that |
| 03:15.50 | jaytee | easy mistake |
| 03:16.09 | jaytee | if you want some real fun try tweaking the dialplan digitmap of a Linksys ATA |
| 03:16.19 | VaGoNeTaS | k, ive just change it |
| 03:16.49 | jaytee | but the Polycom SIP Admin Guide has a bunch of stuff for the digitmap and timeout tweaking in it. |
| 03:17.08 | jplank | yea, I thats why I can't believe I missed that, I have it up |
| 03:17.12 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:17.12 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 03:17.50 | jaytee | as well as a bunch of goodies you can only take advantage of using FTP provisioning because they can't be changed in the web gui management interface |
| 03:18.08 | KyleK | digitmap? as in swapping 1 and 2? |
| 03:19.06 | jaytee | KyleK, nope, Polycoms have their own built in "dialplan" and they use the digitmap for filtering valid number patterns. |
| 03:19.24 | jaytee | very similar to pattern matching in Asterisk |
| 03:20.24 | VaGoNeTaS | jaytee |
| 03:20.32 | jaytee | VaGoNeTaS |
| 03:20.42 | VaGoNeTaS | and my dialplan is very simple, is just one line under the context [default] |
| 03:20.48 | VaGoNeTaS | exten => _X.,1,Dial(Zap/g0/${EXTEN},,T) |
| 03:20.49 | VaGoNeTaS | that one |
| 03:21.06 | jaytee | wow, that is simple! |
| 03:21.09 | *** join/#asterisk killfill (n=killfill@200.63.96.244) |
| 03:21.14 | killfill | hi |
| 03:21.23 | VaGoNeTaS | yes i know |
| 03:22.30 | killfill | im in a busybox linux, and i have set /etc/TZ to GMT+5. Asterisk seem to not care about the time zone, becouse its clock is 4 hours later. |
| 03:22.52 | VaGoNeTaS | becouse |
| 03:22.54 | VaGoNeTaS | xD |
| 03:23.33 | jaytee | killfill, your system clock is using GMT |
| 03:24.03 | killfill | hm.. |
| 03:24.14 | killfill | is that an afirmation or question?.. :P |
| 03:24.22 | jaytee | kind of both :-) |
| 03:24.51 | killfill | well "Thu May 14 23:24:24 GMT 2009" ... (correct date) |
| 03:25.02 | carrar | Fri May 15 03:25:02 UTC 2009 |
| 03:25.21 | killfill | yah, asterisk is telling UTC time.. |
| 03:25.25 | killfill | i need mine tho.. |
| 03:25.51 | jaytee | what's the clock set to in the BIOS? |
| 03:26.14 | killfill | not sure how to check this.its an embeded board |
| 03:26.20 | jaytee | oh |
| 03:27.26 | carrar | make sure you have ntpd running and synced |
| 03:28.03 | killfill | yup, it is. |
| 03:28.18 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 03:28.27 | carrar | clock tells you your bios time |
| 03:28.47 | killfill | but i.e. i can play with "TZ=GMT+5 date" changing the 5 number and date will change its report. |
| 03:28.54 | killfill | not like asterisk |
| 03:29.05 | carrar | clock -w sets your bias clock to your current time |
| 03:29.24 | carrar | or hwclock |
| 03:29.28 | killfill | hm.. busybox doesnt seem to have 'clock' |
| 03:30.48 | carrar | what time zone are you in anyways |
| 03:31.13 | killfill | well im in GMT-4 actually |
| 03:31.34 | killfill | dont know why i have to put gmt+5 to get the right date.. :P |
| 03:32.02 | carrar | what city |
| 03:32.07 | carrar | country |
| 03:32.27 | killfill | Santiago of Chile |
| 03:32.44 | carrar | Continental or EasterIsland |
| 03:32.57 | killfill | continental |
| 03:33.12 | carrar | ln -sf /usr/share/zoneinfo/Chile/Continental /etc/localtime |
| 03:33.13 | jaytee | killfill, do you know VaGoNeTaS ? |
| 03:34.50 | killfill | carrar, well actually i thought doing something like that, but this doesnt has /usr/share/zoneinfo or /etc/localtime.. but ive really not test it. maybe i should get the file from a PC and just try... |
| 03:35.06 | killfill | jaytee: VaGoNeTaS?.. not that i remember |
| 03:35.20 | killfill | :) |
| 03:35.31 | jaytee | think he said he's from Santiago de Chile too |
| 03:35.41 | killfill | ah.. cool. |
| 03:35.42 | carrar | why are you running busybox? |
| 03:36.25 | jaytee | that question entered my mind but I didn't ask it. sometimes it's better not to ask :-) |
| 03:36.31 | carrar | heh |
| 03:38.17 | jaytee | but then I don't have your razor sharp wit and I'm not quite at the level of linux guru like you where I can just spit out a command line statement to create a symlink for a timezone setting. |
| 03:38.30 | carrar | haha |
| 03:39.12 | jaytee | I'm more at the stage of "hmmm, where's that damn book? lemme see here, ah! here it is." |
| 03:40.02 | killfill | hm.. putting the etc/localtime seems to help asterisk |
| 03:40.25 | killfill | its not practical tho, i think it could just get TZ from the enviroment.. :P anyway.. |
| 03:40.49 | carrar | Sounds like busybox isn't practical |
| 03:40.55 | killfill | carrar: well its an embeded system. ubuntu will not fit.. :P |
| 03:41.03 | carrar | Look I'm running with 8k of memory |
| 03:41.08 | carrar | shit is broke but it works! |
| 03:41.14 | killfill | heh |
| 03:41.18 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 03:41.24 | killfill | could bee seen that way too. |
| 03:44.10 | KyleK | killfill: what are you running this busybox stuffs on? |
| 03:45.49 | killfill | wired.. when i start ast from command line date is correct, but not when it starts from the init scripts :S |
| 03:46.14 | killfill | its a ucAsterisk |
| 03:46.29 | killfill | Blackfin |
| 03:46.40 | carrar | cat out your TZ file |
| 03:46.59 | carrar | is it just GMT5 |
| 03:47.07 | killfill | "GMT+4" |
| 03:47.16 | carrar | try GMT4 |
| 03:47.32 | carrar | I've never used busybox so |
| 03:47.54 | carrar | Closest thing was a openmoko probably |
| 03:49.38 | killfill | it doesnt change things. |
| 03:49.57 | carrar | TZ=Chile/Continental date |
| 03:49.57 | carrar | Thu May 14 23:49:51 CLT 2009 |
| 03:50.03 | carrar | TZ=GMT4 date |
| 03:50.04 | carrar | Thu May 14 23:48:34 GMT 2009 |
| 03:50.14 | carrar | seems like the right thing |
| 03:50.24 | Qwell | carrar: busybox is just a "multi-use" binary that does "core" system things |
| 03:52.30 | killfill | yup, actually putting etc/localtime doesnt help. i think the asterisk is staring before the /etc/TZ is read, that why its ok when i start it from bash. |
| 03:55.33 | jameswf | lmao lookie what I found.... http://www.trixboxce.org |
| 04:00.02 | jaytee | learnt? is that a real word? |
| 04:00.18 | jaytee | did everyone have a "save" journey home? :-) |
| 04:03.04 | killfill | damn.. someone is starting asterisk as soon as i kill it. what could it be?. its not in inittab |
| 04:03.05 | carrar | jaytee: http://www.askoxford.com/asktheexperts/faq/aboutspelling/learnt |
| 04:03.13 | drmessano | HAAHHAH |
| 04:03.14 | Qwell | jameswf: ahh, ftocc... |
| 04:03.18 | drmessano | welcome Administrator's |
| 04:03.20 | carrar | Who is that someone |
| 04:03.25 | carrar | you've been owned! |
| 04:03.28 | drmessano | Yes, that's not bad grammar |
| 04:03.32 | Qwell | Having read the slides...all I can say is... |
| 04:03.35 | Qwell | HA |
| 04:03.47 | jaytee | I stand corrected |
| 04:03.52 | jaytee | damn brits! |
| 04:05.06 | jaytee | there's still a lot of crap there though even if learnt is an acceptable form |
| 04:05.50 | jaytee | not that I really care because I don't run Trixbox |
| 04:05.52 | *** part/#asterisk killfill (n=killfill@200.63.96.244) |
| 04:14.40 | *** part/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-54e153af6b860fd1) |
| 04:17.27 | *** join/#asterisk haryv (n=lanny@S010600a0c93f6f7e.vs.shawcable.net) |
| 04:21.17 | chendy | i have register 3 account from host a.b.c.d port 5060, 1 for sip trunk , 2 for sip end point. if there is a inbound call from sip trunk account with auth info, asterisk reports found peer which is a end point account, then return 403. |
| 04:21.22 | chendy | what 's up? |
| 04:21.37 | chendy | yup, i not the auth info is wrong |
| 04:22.10 | chendy | but how come asterisk match peer by address and port, not the From: header? |
| 04:23.15 | *** join/#asterisk mellow-yellow (n=mellow-y@64.122.192.190) |
| 04:28.13 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 04:43.41 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 04:45.38 | joako | Asterisk Business edition is the same code as Asterisk GPL? |
| 04:45.52 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) |
| 04:46.05 | joako | XO is telling my customer their SIP trunking works only with ABE, not opensource Asterisk (and same thing for fonality) |
| 04:46.19 | jql | funny |
| 04:46.56 | lanning | "We only support a supported version." :) |
| 04:47.03 | joako | Im sorry they say Digium Switchvox SMB |
| 04:47.21 | joako | What is the diffrence between Switchvox SIP and Asterisk GPL SIP? |
| 04:47.26 | lanning | same SIP stack |
| 04:47.43 | jql | they're just being snarky |
| 04:47.44 | lanning | though GPL gets modified faster |
| 04:48.27 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-67-247-172-185.rochester.res.rr.com) |
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| 05:02.59 | haryv | I wonder if my wife would get mad if I included her as 0 the operator in the ivr :) |
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| 05:36.11 | jameswf | they see me trollin |
| 06:00.47 | Nugget | trollin' in my five point oh, ragtop down so my hair can blo' |
| 06:07.29 | joako | Looking at the XO SIP documentation it looks horrible anyways, their diagram shows their Cisco IAD conneccted to a HUB with all the IP phones with public IP. And they assign one SIP Account per DID... WTF |
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| 06:30.19 | securevoip | did you get your problem fixed? |
| 06:30.48 | securevoip | oops... meant to say, VaGoNeTas, did you get your problem fixed? |
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| 07:55.52 | Faustov | hi, i can see something stuck in my ivr, how can i boot that call? |
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| 07:56.24 | kapr | soft hangup <channelname> |
| 07:56.51 | Faustov | awesome, thanks |
| 07:56.57 | Faustov | damn phone spambots |
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| 08:05.03 | *** join/#asterisk kyoshi (i=60f6990c@gateway/web/ajax/mibbit.com/x-f2d617f907276443) |
| 08:06.55 | kyoshi | trixbox and freepbx has a mechanism on outbound routes to modify the dialed string, for example, if i dial 9+NPANXX trixbox/freepbx can remove the 9 and replace it with 1 so 9+NPANXX becomes 1+NPANXX. Without going thru a hell of a macro, what would be the easiest way in my extensions file to do this? |
| 08:09.52 | kapr | for removing u can say 9|N... |
| 08:10.15 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 08:11.19 | kyoshi | but what about replacing? |
| 08:11.26 | ectospasm | kyoshi: if you have a string Dial(TECH/CHAN/${EXTEN}), you can do this... |
| 08:11.33 | kyoshi | I want 9NXXXXXX to become 1NXXXXXX |
| 08:11.50 | kyoshi | ecto: thats exactly what i am currently doing |
| 08:11.57 | ectospasm | Dial(TECH/CHAN/1${EXTEN:1}) |
| 08:12.10 | kyoshi | but i need to modify the the dialstring if it starts with something |
| 08:12.18 | kyoshi | so i need to remove the 9 and replace it with 1 |
| 08:12.22 | ectospasm | the ${EXTEN:1} will strip the 9, and the 1 will be prepended |
| 08:18.34 | ectospasm | hmmm, I'm looking at the way FreePBX does it (on AsteriskNOW), and I don't see it immediately. |
| 08:19.25 | kyoshi | so if i do Dial(TECH/CHAN/0110${EXTEN:3}) and i dial 011NXXXXXXXX it will replace the 011 with 0110 ? |
| 08:19.54 | xrmx__ | does anybody have experience on how much gain in latency we have to switch to a not even preempt to an preempt rt kernel on an asterisk 1.2 installation? |
| 08:20.21 | ectospasm | kyoshi: yes |
| 08:20.48 | kyoshi | nice |
| 08:20.52 | kyoshi | thanks, trying now |
| 08:21.52 | ectospasm | I see how FreePBX does it... when dialing out the trunk it uses ${OUTPREFIX_${DIAL_TRUNK}} in the dial string. |
| 08:23.42 | ectospasm | kyoshi: did it work? |
| 08:25.59 | *** join/#asterisk matsk (n=matkar@193-44-227-98.customer.telia.com) |
| 08:27.43 | kyoshi | nopes i think i done brokeded it |
| 08:27.59 | kyoshi | i even went back to the way it was. seems good n broke now |
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| 08:40.29 | ectospasm | kyoshi: you must have done something wrong then |
| 08:40.48 | ectospasm | are you sure you've got the proper number of parentheses and curly braces? |
| 08:41.18 | ectospasm | What does the full line look like? |
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| 08:46.03 | grEvenX | [May 15 10:31:35] ERROR[2689] func_odbc.c: No such DSN registered (or out of connections): oyatel_realtime (check res_odbc.conf) |
| 08:47.42 | grEvenX | I have set the limit to 5 in the odbc config, and enabled pooling |
| 08:47.56 | grEvenX | why would that lead to an error at any time? |
| 08:48.19 | grEvenX | when it reaches the limit of 5, shouldn't it recycle? |
| 08:48.30 | grEvenX | or queue it |
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| 08:51.19 | wdoekes | that is indeed odd.. I do not use pooling, and I assume that requests are queued |
| 08:51.38 | grEvenX | yeah, without pooling enabled I've never seen that error before |
| 08:51.50 | grEvenX | it started when chaning to use pooling |
| 08:53.29 | wdoekes | look at ast_odbc_request_obj in res_odbc.c |
| 08:53.54 | *** join/#asterisk ck_28 (n=CK@212.98.141.199) |
| 08:53.57 | ck_28 | hi ppl |
| 08:54.13 | ck_28 | i have a problem with asterisk freefax |
| 08:54.23 | ck_28 | T38(0/0/0) discard T30D from VoPP by no sess |
| 08:54.29 | ck_28 | any one can help me ? |
| 08:56.09 | wdoekes | grEvenX: it looks like you need to set the limit to minimal the amount of modules that use odbc |
| 08:56.48 | wdoekes | (at least, according to the comment "multiple modules can use the same connection" on non-pooled odbc) |
| 08:57.23 | grEvenX | wdoekes: hm, what is seen as "module" using odbc? I don't think we have more than 5 different modules using odbc |
| 08:59.48 | wdoekes | hmyes.. I would think so too :-/ and I'm not sure what AST_LIST_TRAVERSE does exactly |
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| 09:01.35 | ck_28 | where i can ask to solve the asterisk fax t38 problem ? |
| 09:01.45 | joobie | dont fax with asterisk |
| 09:01.48 | joobie | problem solved |
| 09:01.56 | joobie | dont fax with voip |
| 09:02.23 | ck_28 | joobie fax with what please advice |
| 09:02.35 | ck_28 | i want to send a file to fax |
| 09:02.46 | ck_28 | joobie using t38 protocol |
| 09:04.00 | mvanbaak | wdoekes: AST_LIST_TRAVERSE traverses the list you feed it |
| 09:07.52 | wdoekes | mvanbaak: haha, sounds plausible. but yes, I see that it returns NULL if it finishes the traversal without a break |
| 09:08.06 | wdoekes | s/returns/sets obj to/ |
| 09:08.14 | wdoekes | infobot-- |
| 09:09.40 | mvanbaak | only if the obj was set to NULL before |
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| 09:10.15 | mvanbaak | you can look at AST_LIST_TRAVERSE as a 'for (i = 0; i <= sizeof(struct); i++) {' |
| 09:11.17 | wdoekes | not entirely.. in the for, obj would either be the last element, or not set at all |
| 09:12.35 | mvanbaak | ah, you mean that. yeah |
| 09:12.59 | grEvenX | looking at bugs reported etc. it seems that the pooling uses at max 1 active query on any given connection |
| 09:13.08 | mvanbaak | well, normally when you have AST_LIST_TRAVERSE(some_list, obj_name, list) { |
| 09:13.26 | mvanbaak | you would like to do some 'if (obj_name->something .....' |
| 09:13.36 | grEvenX | so I guess the problem might come if you e.g. have a queue call that spawns a call to e.g. 10 agents, that uses sql queries as well |
| 09:13.54 | grEvenX | then you have 10 sql queries hitting the odbc connections at almost the same time |
| 09:14.24 | grEvenX | when you get to query #7, 8 or something, all the connections are busy in a query, and then gives that message |
| 09:14.25 | wdoekes | (mmyes, looking at linkedlists.h now) |
| 09:14.49 | mvanbaak | wdoekes: that's your best shot indeed. it has all the docs you need (I think) |
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| 09:16.36 | wdoekes | but, back to the question at hand: sorry grEvenX, can't help ;) I don't know why non-pooled connections are returned regardless of being "used" or not and pooled connections aren't |
| 09:17.25 | grEvenX | wdoekes: it seems pooling is a "fix" for e.g. MySQL, that allows only 1 active sql query at a time per connection. Thus it makes sense |
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| 09:17.59 | mvanbaak | wdoekes: for a nice formatted doc about the list macro's: http://www.asterisk.org/doxygen/trunk/linkedlists_8h.html |
| 09:18.33 | grEvenX | the issue is that I don't really want to use pooling because of the limits introduced on concurrent queries, but I was told to do so in one of my bugreports. (send in a faulty SQL from asterisk and have another query being executed at the same time = segfault) |
| 09:19.25 | wdoekes | nice, I didn't know doxygen did include-file-graphing |
| 09:19.25 | grEvenX | I think I will fix the issue myself instead, as asterisk devs won't fix it, and I think it's an abvious flaw in the logic |
| 09:20.03 | ectospasm | if you fix it yourself, please submit a patch for the bug you filed |
| 09:20.14 | grEvenX | I will, it's a one-liner fix |
| 09:25.49 | grEvenX | the bug is closed though, so I won't be able to do that: https://issues.asterisk.org/view.php?id=14748 |
| 09:26.59 | *** join/#asterisk af_ (n=getsmart@88-149-240-168.dynamic.ngi.it) |
| 09:32.22 | grEvenX | anyone here able to open it again? |
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| 09:39.00 | mvanbaak | grEvenX: sure, hang on |
| 09:40.01 | mvanbaak | there, it's open again |
| 09:43.34 | grEvenX | mvanbaak: thanks, I'm testing my patch now to see that it's ok |
| 09:44.48 | *** join/#asterisk zafar_ (n=IceChat7@116.71.217.196) |
| 09:45.36 | zafar_ | hey guys i have problem with my extension-custom.conf |
| 09:45.46 | zafar_ | this thing is not work |
| 09:45.47 | zafar_ | [ext-local-custom] |
| 09:45.47 | zafar_ | exten => _12x,1,dial(IAX2/sbcpbx/${EXTEN}) |
| 09:45.54 | zafar_ | any idea? |
| 09:46.36 | mvanbaak | zafar_: without CLI output etc we cant say nothing about it |
| 09:46.45 | mvanbaak | define 'this thing is not work' |
| 09:46.53 | zafar_ | ok hold on |
| 09:46.55 | mvanbaak | what goes wrong? what is the error on the console? |
| 09:48.24 | zafar_ | when i am ivr and i dial 121 it is going to the ivr option where i have probide 1 |
| 09:48.39 | zafar_ | means going for the first option in the ivr |
| 09:49.19 | zafar_ | if i dial 512 then its going to the 5 ivr element and not dialing ext 512 |
| 09:51.20 | mvanbaak | that dialplan snippet you posted there is not an IVR |
| 09:52.23 | zafar_ | yes its in extension_custom |
| 09:53.09 | zafar_ | _12x is on another server |
| 09:53.36 | mvanbaak | you making it us very difficult to help |
| 09:53.50 | *** join/#asterisk festr_ (n=festr@ns.hiro.cz) |
| 09:54.11 | zafar_ | ok here is the senario |
| 09:54.57 | zafar_ | i have two server one has extension like 12x and 22x and the second server has extion in the range of 13x and 3xx |
| 09:55.03 | festr_ | hi, I'm solving interesting problem :) asterisk 1.4. I'm calling from A to B (SIP) and I want to know who is calling to B by getting some variable from B channel |
| 09:55.36 | festr_ | but it seems there is no information about callerid in the B channel |
| 09:55.58 | festr_ | (while in ring mode) |
| 09:56.03 | zafar_ | when i dial the 2nd server and ivr is playing i should be able to reach the extensions on server 1 |
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| 09:57.38 | mvanbaak | zafar_: so far so good |
| 09:58.46 | zafar_ | when i m in the ivr i can dial any ext on server 2 similarly i want to server 1 ext to be available to me on server 2 |
| 09:59.30 | zafar_ | i did this succuess fully on earlier servers but its not working on this machine |
| 09:59.57 | mvanbaak | just make an exten on server2 pointing to server1 |
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| 10:00.42 | zafar_ | mvanbaak - any ext? |
| 10:01.26 | mvanbaak | zafar_: sure. I would create exten 12X on server2 to point to server1 |
| 10:01.49 | mvanbaak | zafar_: that way it does not matter what server the call is on, 12X will point to exten 12X on server1 |
| 10:03.27 | *** join/#asterisk ultrav1olet (n=ultrav1o@188.17.64.103) |
| 10:03.55 | ultrav1olet | I've already pronounced this problem but no one has told me anything so here it is again: |
| 10:04.10 | zafar_ | but i have around 60 ext that to need to go to the other server |
| 10:04.29 | mvanbaak | zafar_: you can use wildcards |
| 10:04.50 | zafar_ | like? |
| 10:04.53 | mvanbaak | _12X |
| 10:05.08 | ultrav1olet | We have Wildcard TDM400P REV E/F Board 5 board and when someone calls somewhere there's >50% chance to hit a wrong number |
| 10:05.32 | ultrav1olet | When we was running Asterisk 1.4.x and Zapata there was no such problem |
| 10:05.49 | ultrav1olet | Now we are running asterisk 1.6.x and DAHDI and this problem drives me mad |
| 10:05.57 | zafar_ | and thats what i am doing in extensions_custom.conf |
| 10:06.14 | ultrav1olet | Right now I've called one cell number four times and four times I've reached wrong numbers |
| 10:06.20 | zafar_ | am i making a mistake? |
| 10:07.03 | ultrav1olet | We have connected a normal phone to the same phone line and I tried called from that phone ... and everything works just fine |
| 10:07.56 | zafar_ | [ext-local-custom] anything in this block should be considered as local i believe and i am telling it where to find the extensions in exten => _12x,1,dial(IAX2/sbcpbx/${EXTEN}) |
| 10:08.25 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) [NETSPLIT VICTIM] |
| 10:08.34 | mvanbaak | zafar_: again, show us the CLI log of a failed call |
| 10:08.42 | zafar_ | hold on |
| 10:09.37 | mvanbaak | ultrav1olet: we will need CLI logs of the calls that go to the wrong number |
| 10:10.29 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1096591586.dsl.bell.ca) |
| 10:17.50 | zafar_ | mvanbaak : here is the CLI output http://pastebin.com/m67742186 |
| 10:20.50 | ultrav1olet | mvanbaak: dialplan has NOT changed since asterisk 1.4 |
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| 10:21.00 | ultrav1olet | and logs are perfectly sane |
| 10:23.08 | xrmx__ | is there something to read about configuring linux kernel for asterisk? like a hint for hz and preempt options, and numbers maybe |
| 10:23.51 | mvanbaak | zafar_: it only accepts the 5 because that extension exists |
| 10:24.03 | Flyser_ | I guess 1000 Hz with preemption would be the best for realtime applications like asterisk |
| 10:24.05 | mvanbaak | you hit 5, and asterisk finds that exten, so goes to it |
| 10:24.22 | zafar_ | yes but i have 4 digit exts in the ivr as well |
| 10:24.34 | mvanbaak | zafar_: dont mix them then ;) |
| 10:25.07 | zafar_ | but the same thing is working fine on another server but i build that like 4 months ago |
| 10:25.54 | zafar_ | thanx man anyway i ll find a fix for this soon |
| 10:25.57 | zafar_ | :) |
| 10:27.55 | xrmx__ | Flyser_, thanks |
| 10:31.03 | mvanbaak | ultrav1olet: if you say so. Sorry, cant help you without more info |
| 10:32.50 | ultrav1olet | please wait |
| 10:39.31 | *** join/#asterisk Gopaul (n=Miranda@61.17.185.118) |
| 10:39.48 | ultrav1olet | http://pastebin.ca/1423546 |
| 10:40.35 | ultrav1olet | and one more time I hit the wrong number |
| 10:40.43 | ultrav1olet | it happens pretty randomly |
| 10:42.07 | ultrav1olet | one time asterisk calls the right number, another three times I hit the wrong number |
| 10:42.18 | ultrav1olet | the same user, the same dialplan, absolutely the same log |
| 10:42.58 | ultrav1olet | I've given way to dispair |
| 10:53.02 | ck_28 | <PROTECTED> |
| 10:53.09 | ck_28 | T38(0/0/0) discard T30D from VoPP by no sess |
| 10:53.14 | ck_28 | any one can help me ? |
| 10:59.13 | ultrav1olet | mvanbaak: ran out of ideas? |
| 11:02.35 | grEvenX | ok, patch coming up |
| 11:03.20 | grEvenX | hm |
| 11:03.37 | grEvenX | any guidelines on patches to bugs/issues should be created ? |
| 11:10.33 | *** join/#asterisk Marquel (n=Flinx@port-12669.pppoe.wtnet.de) |
| 11:10.40 | Marquel | morning |
| 11:10.46 | grEvenX | found it |
| 11:11.58 | Marquel | is it possible to have a sip-phone dial a number, then have another sip-user just executing "off-hook" and that sip-user executes "Dial(<whatever the sip-phone dialed>)"? |
| 11:13.22 | mvanbaak | ultrav1olet: what's with the three w's ? |
| 11:14.06 | ultrav1olet | I thought it would help |
| 11:14.29 | ultrav1olet | someone from #asterisk told me to try |
| 11:15.16 | *** join/#asterisk saftsack (n=saftsack@p5792476A.dip.t-dialin.net) |
| 11:17.15 | grEvenX | mvanbaak: thanks for opening the ticket again, patch now submitted: https://issues.asterisk.org/view.php?id=14748 |
| 11:19.54 | ultrav1olet | ok, let's wait for developers |
| 11:24.48 | *** join/#asterisk anjaana_ladka (i=abhinav1@117.96.125.53) |
| 11:25.14 | anjaana_ladka | hello asterisk users... |
| 11:25.30 | anjaana_ladka | can anybody please help me out setting up my asterisk box... |
| 11:25.47 | anjaana_ladka | i have installed the asterisk box..... and created two sip users... |
| 11:25.59 | anjaana_ladka | clients are not able to register |
| 11:26.04 | *** join/#asterisk dunccfflail (n=dunc@2001:960:7bd:1194:0:0:0:1) |
| 11:26.17 | anjaana_ladka | i also have to set extensions.. |
| 11:26.31 | anjaana_ladka | the asterisk box contains tdm808 card... |
| 11:26.53 | anjaana_ladka | can any body please help me in getting a simple config to enable me to call from teh computer |
| 11:26.54 | anjaana_ladka | ???????? |
| 11:27.53 | dunccfflail | hi folks, i realise you probably get people turning up here all the time, but could anyone point me at a troubleshooting guide anywhere? |
| 11:29.16 | dunccfflail | i've done a lot of reading, i have a tdm400p with one FXO and one FXS module, which I'm pretty sure I have configured OK, and have got set to use the correct tone for my country, i can run zttool, and when I either ring the incoming line, or lift the handset on my analog phone, shows my channels becoming active |
| 11:29.49 | *** join/#asterisk HenrikBe (n=zapphir@h204n4fls32o954.telia.com) |
| 11:29.53 | dunccfflail | however i never get a dial tone on the phone, nor does asterisx notice (with -vvvvvvv) either the incoming call, or the phone going off the hook |
| 11:30.16 | dunccfflail | anybody help? |
| 11:30.57 | *** join/#asterisk desdesdesdes (n=f@196.211.34.2) |
| 11:31.49 | desdesdesdes | hi hter i am stuck on mp3 files for moh is there a program u guys recomend for converting file to be able to wotk wit *? |
| 11:34.12 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 11:40.28 | anjaana_ladka | hi folks... |
| 11:40.54 | anjaana_ladka | waiting for a response... can anybody please guide me and tell me how to proceed.??????? |
| 11:41.47 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 11:41.51 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-b6cd6eec9ca0eb09) |
| 11:43.00 | gr0mit | anjaana_ladka, i think we need a bit more info! |
| 11:43.14 | gr0mit | like pastebin of your sip.conf files |
| 11:43.36 | HenrikBe | is there a php function that processes the xml response from ajam? |
| 11:45.34 | dunccfflail | "what's the minimal config to get a dialtone on my phone connected to a FXS module" I guess would be a nice place to start, and how to troubleshoot it if it doesn't work |
| 11:48.28 | *** join/#asterisk thomasrr (n=scroogey@cp811981-a.mill1.nb.home.nl) |
| 11:48.29 | thomasrr | hello |
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| 11:53.48 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:53.48 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 12:30.39 | *** join/#asterisk ariel_ (i=3fd6ec96@gateway/web/ajax/mibbit.com/x-2be5281734eb702e) |
| 12:31.00 | ariel_ | Hello folks |
| 12:33.43 | desdesdesdes | hi |
| 12:37.04 | Marquel | bbl |
| 12:40.16 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 12:43.12 | *** join/#asterisk bOOinK (n=lulu@0x5359c272.cpe.ge-0-1-0-1110.vbynqu1.customer.tele.dk) |
| 12:43.16 | bOOinK | hello all |
| 12:43.39 | bOOinK | anyone running 1.4 ?? |
| 12:43.48 | wdoekes | sure |
| 12:44.24 | bOOinK | are you experiencing extreme load spikes from time to time aswell ? |
| 12:46.00 | wdoekes | not that I'm aware of. there's one machine on which the cpu load seems to increase over the course of days, but I haven't had enough time to dive into it deeply |
| 12:46.21 | wdoekes | (+ that particular version is a bit old) |
| 12:47.01 | bOOinK | hmm |
| 12:47.52 | bOOinK | because I see the spikes happening lige a few times every hour, and randomly between the 3 servers I have running |
| 12:50.36 | thomasrr | hmm, i have been trying to get voipbuster incoming calls going |
| 12:50.42 | thomasrr | it works everywhere but on my asterisk box |
| 12:50.57 | thomasrr | anyone happen to have experience with it and can have a look at what i am doing wrong? |
| 12:51.56 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
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| 13:19.37 | ghenry | nice http://www.prweb.com/releases/2009/05/prweb2419104.htm |
| 13:19.47 | ghenry | just seen this in a google alert for asterisk |
| 13:20.22 | *** join/#asterisk Flyser (n=Flyser@unaffiliated/flyser) |
| 13:21.53 | jaytee | that was in the News section of voip-info.org yesterday |
| 13:22.50 | Pan3D | hahaha, better than... what's that one cheezy product that came around last year? the guy with the ads featuring his daughter. |
| 13:23.02 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:23.10 | jdblack | Huh. |
| 13:23.11 | Pan3D | This looks nice |
| 13:23.43 | jdblack | Why bother with the pci slot. |
| 13:23.52 | jaytee | I can think of a billion cheezy products but no ad like that that triggers my memory |
| 13:24.12 | Pan3D | hold on, when I find it, you'll be like "Oh yeah". IT was the USB thing. |
| 13:24.29 | Pan3D | "MagicJack" |
| 13:24.35 | jaytee | jdblack, some people still want to interface to the PSTN over copper, either analog or PRI |
| 13:24.58 | jaytee | ah, vague memories of MagicJack |
| 13:25.04 | jdblack | Well, sure. but that's a different question than this. |
| 13:25.06 | Pan3D | yeah, I think PCI is fine |
| 13:25.17 | jdblack | This is essentially "a seperate computer the lives on a pci slot" |
| 13:25.39 | Pan3D | jdblack: how would you rather have it? |
| 13:25.42 | Pan3D | a dongle? or? |
| 13:25.44 | jdblack | so, replace the pci hookup with a power hookup, you have the same thing. |
| 13:26.07 | jdblack | basically, replace two dollars of pci connector for two dollars of plastic. |
| 13:27.31 | jdblack | The thing is basically an entire computer that draws current from the pci bus. See my point? |
| 13:27.32 | Pan3D | well, it's easy to come up with reasons why it's not exactly what you want, but I suspect a lot of folks are going to dig the product. |
| 13:28.24 | jdblack | I think you're right. A lot of people are going to love it. |
| 13:30.12 | jdblack | Yay! For the first time in almost twenty years, I have gone more than a week without a single smoke. I'm cured! |
| 13:30.32 | *** part/#asterisk boch (n=fran@200.61.191.9) |
| 13:30.56 | mmlj4 | excellent |
| 13:31.00 | russellb | jdblack: congratulations :-) |
| 13:31.05 | jdblack | Thanks. :) |
| 13:31.09 | telnettech | congrats jdblack!!!!! |
| 13:31.35 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 13:32.00 | Pan3D | jdblack: that's awesome. congrats |
| 13:32.07 | jdblack | And the most amazing part... Nobody died. That's like.. significant. |
| 13:32.12 | Pan3D | haha |
| 13:32.14 | *** join/#asterisk eliteSchaf (n=d3xter@81-89-105-51.blue.kundencontroller.de) |
| 13:32.16 | telnettech | looks as though you still have to work on the part where everything irritates you :) |
| 13:32.21 | Pan3D | I bet they are happy about that as well |
| 13:32.35 | jdblack | That's the thing. I haven't really been too irritated. |
| 13:32.43 | jdblack | Or irritating, for that matter. |
| 13:32.52 | telnettech | the PCI thing seemed to irritate you |
| 13:32.52 | jdblack | Well, other than normal, at least. |
| 13:33.10 | rob0 | Oh come on, what's the loss of an addiction without at least one homicide? |
| 13:33.25 | jdblack | seriously! We should have that right |
| 13:33.32 | rob0 | Pick one who needs it, do it! |
| 13:34.31 | jdblack | Society would improve in a heartbeat. "Congrats for quitting Smoking/Cocaine/Drinking/WoW abuse. Please accept this free get out of Capital Crime card. Choose your target wisely and society will thank you" |
| 13:34.37 | *** part/#asterisk ck_28 (n=CK@212.98.141.199) |
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| 13:40.24 | telnettech | anybody know of any telco settings on an MPLS circuit that the telco needs to configure when you have phones at 1 location registering back to the * server? |
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| 13:56.57 | therealcircut | morning all |
| 13:58.00 | telnettech | anybody know of any telco settings on an MPLS circuit that the telco needs to configure when you have phones at 1 location registering back to the * server? |
| 13:58.15 | rue_mohr | mpls? |
| 13:58.29 | rue_mohr | pots? |
| 13:58.54 | rue_mohr | pri? |
| 13:59.29 | telnettech | mpls |
| 13:59.45 | rue_mohr | whats that? |
| 14:00.02 | telnettech | having ptoblems with voice quality |
| 14:00.15 | rue_mohr | is that a digital service? |
| 14:00.19 | telnettech | yes |
| 14:00.26 | rue_mohr | dsl? |
| 14:01.01 | telnettech | http://en.wikipedia.org/wiki/MPLS |
| 14:01.02 | rue_mohr | what bandwidth, delay, and jitter does it have? |
| 14:01.41 | *** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com) |
| 14:02.13 | dunccfflail | mpls is just IP |
| 14:02.28 | rue_mohr | so you asking about routing settings |
| 14:02.38 | therealcircut | telnettech: i dont think you should have any special settings |
| 14:02.57 | therealcircut | when the provider throws you into their mpls cloud, its kindof up to them to make sure things are flowing as they should |
| 14:03.29 | therealcircut | u might ask them to prioritize your traffic to the asterisk server, but i dont know if they would be very accomodating with it |
| 14:03.41 | stope | I know of some telco's that charge for the mpls component... what a rip! |
| 14:03.44 | telnettech | thats what i was thinking but i havent dealt with a MPLS circuit so I wanted to see if there is anything that i needed to look out for |
| 14:03.53 | therealcircut | not usually |
| 14:04.15 | therealcircut | and if issue do come up its usually the burden of the mpls provider to fix any issues you will be having |
| 14:05.04 | therealcircut | they might ask what codecs or w/e your using and try to tell u thats the problem, but unless their running parts of their cloud on a 14.4baud modem in zaire i dont think it should be an issue |
| 14:05.21 | telnettech | I turned on tos=0xB8 so that the packets are marked for the diffserv as EF so Im just curious if that is what they wanted or if there was anything i could tell them to do on their side |
| 14:06.07 | therealcircut | whose the provider? |
| 14:06.15 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.138.206) |
| 14:06.23 | dunccfflail | can't remember if it will honour the TOS bit or not, it's probably up to the provider |
| 14:06.35 | dunccfflail | is only just starting to roll out our mpls |
| 14:07.22 | therealcircut | yea |
| 14:07.33 | therealcircut | 90% of mpls really falls on the provider to get your stuff running / fixed |
| 14:08.24 | telnettech | PAETEC is provider |
| 14:08.48 | *** join/#asterisk hi365 (n=hi365@94.159.176.248) |
| 14:09.05 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:09.10 | therealcircut | never heard of them |
| 14:09.16 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:10.06 | telnettech | I have worked with them in various locations around the US but with T-1 and PRI circuits |
| 14:10.34 | telnettech | never between locations and liek I said, I just know what I read about MPLS service |
| 14:11.12 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 14:11.31 | ayeso | Anyone know the major differences between ver 1.4 and 1.6 ? |
| 14:11.46 | therealcircut | yea, asterisk & your phones really shouldn't care when network you transport your voip over |
| 14:12.02 | therealcircut | if you have issues you should go to the provider first, as they are most likely the culprit |
| 14:12.12 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 14:13.56 | *** join/#asterisk jon_farmer (n=chatzill@195.74.96.119) |
| 14:15.10 | thomasrr | anyone here know what i should do to get incoming calls from voipbuster working in asterisk 1.6? |
| 14:16.05 | *** join/#asterisk ck_28 (n=CK@212.98.141.199) |
| 14:18.43 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
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| 14:26.59 | therealcircut | ThoMe: a good start would be to give us more info |
| 14:27.22 | ThoMe | therealcircut: jo. i think you mean thomasrr :P |
| 14:27.26 | *** join/#asterisk thomasrr (n=scroogey@195-240-213-212.ip.telfort.nl) |
| 14:27.28 | thomasrr | hello |
| 14:27.44 | therealcircut | ayeso: 1.4 is almost a culmination of 1.2 & 1.6 |
| 14:27.53 | therealcircut | im doing an upgrade from 1.2 -> 1.4 here |
| 14:27.57 | thomasrr | I am using the following config with my asetrisk box but i cant get the voipbuster voip-in number to work (to get incoming calls): http://pastie.org/479129 |
| 14:28.06 | mmlj4 | thomasrr: seems to me that voipbuster would have a page telling you how to do that |
| 14:28.09 | thomasrr | anyone know what i might be doing wrong? I am able to make outgoing calls |
| 14:29.05 | thomasrr | i couldnt find it |
| 14:29.16 | thomasrr | voipbuster isn't sharing much info imho |
| 14:29.23 | mmlj4 | can you receive calls? |
| 14:29.51 | mmlj4 | anyhow, what about extensions.conf? |
| 14:29.52 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 14:30.09 | mmlj4 | and what are you using? freebbx? trixbox? what? |
| 14:31.03 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 14:31.08 | ck_28 | hi |
| 14:31.15 | ck_28 | i am using asterisk free fax |
| 14:31.22 | ck_28 | i am recieving rtp.c:1739 ast_rtp_read: Unknown RTP codec 100 received from '80.239.172.145' |
| 14:31.34 | angryuser | hello, what was the model of linksys spa with 2 FXS and 2 Ethernet ports ? |
| 14:31.52 | ck_28 | SIP/msx-091c3cb0 2 G.711 send Active /tmp/fax1.tif |
| 14:32.03 | ck_28 | how to make t38 acceptable ? |
| 14:32.18 | ck_28 | always send g711 |
| 14:32.48 | thomasrr | mmlj4: i cant receive calls but i can do outgoing calls |
| 14:32.54 | thomasrr | asterisk 1.6 |
| 14:33.12 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 14:34.08 | mmlj4 | do you have an extension set up for your VoIP number? |
| 14:34.22 | mmlj4 | that needs to be in the right context |
| 14:34.23 | ck_28 | guy any have experience in asterisk fax |
| 14:34.57 | thomasrr | yes 31318XXXXXX |
| 14:35.14 | thomasrr | but still i dont given get a sip notify thing or something |
| 14:35.46 | mmlj4 | have you asked their tech support to trace things from there end? |
| 14:36.01 | *** join/#asterisk shareenergy (n=go@host-87-74-7-50.dslgb.com) |
| 14:36.06 | shareenergy | hello ppl |
| 14:36.23 | thomasrr | mmlj4: they dont respond |
| 14:36.29 | ck_28 | is there a channel i can find there a help on asterisk fax |
| 14:36.39 | mmlj4 | are you married to their service? |
| 14:36.39 | *** join/#asterisk InfoNutz (n=what@204.50.209.225) |
| 14:36.40 | *** join/#asterisk JenniferAkemi (n=Jennifer@76-10-182-237.dsl.teksavvy.com) |
| 14:36.53 | shareenergy | anybody knows how to avoid rtp on a video call to pass asterisk? |
| 14:37.10 | shareenergy | I get the traffic always cut, when I use video on asterisk |
| 14:37.19 | shareenergy | while opensips work fine |
| 14:37.19 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 14:38.40 | thomasrr | married??? |
| 14:39.05 | thomasrr | sip show peers |
| 14:39.06 | thomasrr | says |
| 14:39.12 | eppigy | u trippin |
| 14:39.20 | thomasrr | voipbuster/phone 194.120.0.198 N 5060 Unmonitored |
| 14:39.24 | InfoNutz | anyone know a good link on how to configure IVR for linux based asterisk? |
| 14:41.36 | leifmadsen | InfoNutz: really IVR is just a make up of standard asterisk dialplan |
| 14:41.41 | mmlj4 | thomasrr: I'm asking about extensions.conf |
| 14:41.47 | mmlj4 | extensions.conf |
| 14:41.53 | mmlj4 | not sip.conf |
| 14:41.56 | leifmadsen | (although the next version of Asterisk:TFoT needs to delve into creating them) |
| 14:45.17 | thomasrr | yes |
| 14:45.26 | InfoNutz | @leifmadsen: dialplan is the term for IVR? I just need to append a message to our clients saying there has been a change to our voicemail system. |
| 14:45.35 | thomasrr | posted the config file |
| 14:46.09 | cyford | hello everyone, how can i make asterisk use a calling card for international calls? |
| 14:46.18 | thomasrr | i made 3185XXXXXXXX => { } extension in the outgoing-voipbuster context |
| 14:46.36 | thomasrr | and in sip.conf i specified that context for the voipbuster block |
| 14:47.47 | mmlj4 | thomasrr: what is your VoIP numbef, from voipbuster? |
| 14:47.50 | *** join/#asterisk [netman] (n=netman@175.Red-79-145-182.dynamicIP.rima-tde.net) |
| 14:48.12 | mmlj4 | number |
| 14:48.59 | shareenergy | is there any way of doing a connection directly between 2 devices |
| 14:49.07 | shareenergy | <PROTECTED> |
| 14:49.22 | thomasrr | 318578506XX |
| 14:49.47 | InfoNutz | Hello everyone, i'm looking to change the voicemail prompts on asterisk. Is there a good link on configuring and manageing them? |
| 14:49.51 | mmlj4 | thomasrr: you need that number in extensions.conf if you want to receive calls for that number |
| 14:50.25 | mmlj4 | shareenergy: I don't understand your question, probably because it's too simple |
| 14:51.07 | mmlj4 | explain a bit more, please |
| 14:51.17 | shareenergy | sure |
| 14:51.24 | shareenergy | i have 2 sip video phones |
| 14:51.37 | shareenergy | when I use asterisk the quality is really bad |
| 14:51.43 | shareenergy | all codecs enable etc |
| 14:51.55 | shareenergy | if I use opensips quality is wonderfull |
| 14:51.57 | thomasrr | mmlj4: well i got an other number and extension |
| 14:52.01 | thomasrr | and in that context i do: |
| 14:52.19 | mmlj4 | opensips, never heard of it |
| 14:52.29 | shareenergy | so I wonder if there is any way to bypss video on asterisk |
| 14:52.40 | *** join/#asterisk ariel_ (i=3fd6ec96@gateway/web/ajax/mibbit.com/x-9367910033dd5859) |
| 14:52.41 | mmlj4 | are you meaning you only want SIP to be handled by *, but not the media channel? |
| 14:52.49 | shareenergy | so that they can connect rtp without passing on asterisk |
| 14:52.57 | shareenergy | :) exactly |
| 14:53.00 | thomasrr | goto from-voipbuster,31318XXXXXX,1; |
| 14:53.00 | mmlj4 | ok, right |
| 14:53.03 | thomasrr | and that works just dine |
| 14:53.13 | thomasrr | the context from-voipbuster appears to work |
| 14:53.57 | mmlj4 | I don't know, shareenergy... I haven't read much about how * does that... it may force everything to pass through *, or not... ask someone else |
| 14:54.18 | thomasrr | like this: context from-freenumber { 31XXXXXXXX => { goto from-voipbuster,3185XXXXXXX,1; }; }; |
| 14:54.37 | thomasrr | and then it recognizes it rings all the phones |
| 14:55.39 | ayeso | What is the best text to speech engine for asterisk? |
| 14:56.15 | *** join/#asterisk Alborracho (n=chatzill@190.25.135.1) |
| 14:59.07 | *** join/#asterisk ks3 (n=ks3@74.203.195.1) |
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| 15:01.24 | mmlj4 | ayeso: festival? |
| 15:01.53 | ayeso | mmlj4: Thats the only one i have tried, doesnt sound so good. |
| 15:02.05 | mmlj4 | there are non-free ones out there, some very good |
| 15:02.38 | mort_gib | ayeso: But loads of people HATE speaking machines ;-) |
| 15:03.17 | ayeso | mort_gib: I agree, Im one of them |
| 15:03.26 | mort_gib | :-) |
| 15:04.09 | thomasrr | how can i enable debug mode in asterisk 1.6? |
| 15:04.18 | *** join/#asterisk bbryant (n=bbryant@c-68-59-20-153.hsd1.sc.comcast.net) |
| 15:05.12 | *** join/#asterisk g-a-m-e-r-x (n=domenic@58.165.189.76) |
| 15:05.24 | Alborracho | thomasrr: core set debug on |
| 15:06.08 | thomasrr | doesnt work |
| 15:06.26 | g-a-m-e-r-x | heyy, just recieveing this error when i do "/sbin/ztcfg -vv" - line 0: Unable to open master device '/dev/zap/ctl' - can anyone here help me? |
| 15:07.00 | ck_28 | when i send a fax using t38 always i have an error 488 not acceptable here why ? |
| 15:07.49 | telnettech | gamerx: what version of asterisk? |
| 15:07.58 | g-a-m-e-r-x | 1.4 |
| 15:08.10 | g-a-m-e-r-x | 1.4.21.2 |
| 15:08.31 | Qwell | g-a-m-e-r-x: do you have the kernel modules loaded? |
| 15:08.51 | g-a-m-e-r-x | err, im not sure.. |
| 15:08.56 | g-a-m-e-r-x | how cani find out? |
| 15:09.07 | phl4kx | hi all |
| 15:09.11 | phl4kx | anyone here from PERU? |
| 15:09.22 | g-a-m-e-r-x | im not |
| 15:10.42 | thomasrr | do i need to make a user in sip.conf for the voip-in number associated with my voipbuster account? |
| 15:11.20 | g-a-m-e-r-x | any ideas guys? |
| 15:11.33 | *** join/#asterisk ScriptFanix (i=vincent@Tuluk.riquer.fr) |
| 15:11.39 | Katty | peru |
| 15:11.41 | telnettech | gamerx: are you sure that you have zaptel and not dahdi |
| 15:11.45 | ayeso | phl4kx: isnt machu pichu there? |
| 15:11.54 | *** join/#asterisk bsilberman (n=bsilberm@65.213.221.252) |
| 15:11.59 | ScriptFanix | Hi |
| 15:12.26 | Katty | hi |
| 15:12.40 | g-a-m-e-r-x | i installed zaptel "sudo apt-get install zaptel" |
| 15:12.47 | phl4kx | jaja |
| 15:13.12 | phl4kx | I like to know a good digium card with no problems with polarity in peru with telefonica |
| 15:13.38 | mort_gib | phl4kx: I use Telefonica in Spain |
| 15:13.49 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
| 15:14.07 | phl4kx | yes but us not the same in peru or yes? |
| 15:14.31 | mort_gib | :-) Don't know, but I would be tempted to think so |
| 15:14.57 | mort_gib | do you need analogue or ISDN |
| 15:15.03 | mort_gib | or PRI?? |
| 15:15.11 | phl4kx | analogue |
| 15:15.32 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 15:15.34 | mort_gib | That should be STRAIGHT forward |
| 15:15.44 | phl4kx | but the last card dont detect the hangup signal |
| 15:15.46 | phl4kx | problems with polarity |
| 15:16.21 | shareenergy | phl4kx that seems like a misconfiguration |
| 15:16.29 | phl4kx | in my zaptel.conf? |
| 15:16.41 | shareenergy | phl4kx we implement in costa rica, madrid, portugal and no problems |
| 15:16.44 | mort_gib | Huh, over my head I use Sangoma cards in Gibraltar, Spain, Denmark and UK |
| 15:16.55 | ScriptFanix | I'm trying app_swift.so, but I am not able to change the voice as indicated on http://www.voip-info.org/wiki/view/Asterisk+cmd+swift |
| 15:16.55 | phl4kx | juazz |
| 15:17.03 | phl4kx | ok thanks |
| 15:17.08 | mort_gib | I'm NOT going to say no problems, but nothing serious |
| 15:17.19 | phl4kx | ok |
| 15:17.46 | *** join/#asterisk g-a-m-e-r-x (n=Domenic@58.165.189.76) |
| 15:17.47 | phl4kx | can you recommend me a analog card with 2 FXO ports? |
| 15:17.50 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:18.04 | g-a-m-e-r-x | sorry, did anyone have any ideas |
| 15:18.15 | tzafrir_laptop | g-a-m-e-r-x, install zaptel-source |
| 15:18.24 | tzafrir_laptop | run (as root) m-a a-i zaptel |
| 15:18.37 | ScriptFanix | exten => 123,n,Swift('Hello, my name is Lawrence. I hope you're enjoying the way my voice sounds!') |
| 15:18.39 | ScriptFanix | exten => 123,n,Swift("Isabelle^Bonjour, je m'appelle Isabelle. J'espère que vous aimez le son de ma voi"!) |
| 15:18.46 | thomasrr | dtmfmode=rfc2283 |
| 15:18.48 | tzafrir_laptop | and this is something you should have read about in /usr/share/doc/zaptel/README.Debian |
| 15:18.49 | thomasrr | doesnt exist? |
| 15:19.09 | thomasrr | or 83 33 |
| 15:19.11 | ScriptFanix | Lawrence, an english voice, tries to say "Isabelle^Bonjour, je m'appelle Isabelle. J'espère que vous aimez le son de ma voi" |
| 15:19.13 | g-a-m-e-r-x | im installing it now |
| 15:19.18 | ScriptFanix | Isabelle being a french voice |
| 15:19.20 | lesouvage | I have build a simpe autodial routine in Asterisk using Meetme(). When the agent want to end the call he has to use * option 3 and kick the called party out of the conference. Problem is that not only the sip channel is disconnected but also the local channel that started the routine and was launched from a callfile. I'm using a dialstatement with the g parameter so execution of the context... |
| 15:19.22 | lesouvage | ...contnues and the M to start a macro to enter the conferenceroom. |
| 15:20.17 | lesouvage | Is there a way to really hang up the sip channel without disconnecting the original local channel. I already erased the /n parameter form the callfile line. |
| 15:20.23 | VaGoNeTaS | guys |
| 15:20.31 | VaGoNeTaS | i've configured an SIP Telephone |
| 15:20.36 | thomasrr | damn s hould never have rebooted the asterisk box :( :( |
| 15:20.36 | VaGoNeTaS | Grandstream BT-200 |
| 15:20.55 | VaGoNeTaS | but when i call or receive calls, i we can hear each other |
| 15:20.58 | VaGoNeTaS | but with statics |
| 15:21.05 | VaGoNeTaS | what can it be, the audo codecs? |
| 15:21.29 | VaGoNeTaS | what could it be* |
| 15:21.43 | g-a-m-e-r-x | please stop spamming |
| 15:21.49 | g-a-m-e-r-x | hehe |
| 15:21.53 | VaGoNeTaS | what? |
| 15:21.58 | VaGoNeTaS | who's spamming |
| 15:22.02 | VaGoNeTaS | u r |
| 15:22.07 | bmoraca | VaGoNeTaS: Grandstream phones are called "BudgetTone" for a reason...they're like the Yugo of VoIP phones |
| 15:22.15 | g-a-m-e-r-x | umm no, i think everybody would agree you were |
| 15:22.22 | thomasrr | why do i keep getting |
| 15:22.24 | thomasrr | [May 15 17:12:39] WARNING[3371]: chan_sip.c:21169 set_insecure_flags: Unknown insecure mode 'very' on line 36 ? |
| 15:22.35 | VaGoNeTaS | bmoraca well i know but, with the softphone, i get the same |
| 15:22.43 | VaGoNeTaS | statics |
| 15:22.48 | VaGoNeTaS | i can here, but with statics |
| 15:22.54 | VaGoNeTaS | i can listen* |
| 15:22.58 | Nugget | my brain keeps trying to turn "g-a-m-e-r-x" into a file permissions mask. I can see it's read/execute for others, but I can't figure out the rest. |
| 15:23.01 | bmoraca | VaGoNeTaS: you get static because you're talking to a grandstream. what if you call softphone to softphone? |
| 15:23.06 | VaGoNeTaS | nop |
| 15:23.13 | lesouvage | Does anybody knows where the kick of the conference routine is in the source code. I checked app_meetme but I culdn't find it. |
| 15:23.17 | VaGoNeTaS | i tried to call softphone to a local phone |
| 15:23.20 | VaGoNeTaS | and i got the same |
| 15:23.20 | VaGoNeTaS | statics |
| 15:23.22 | g-a-m-e-r-x | nugget, what do you mena? |
| 15:23.33 | Nugget | do an "ls -la" on any unix box. |
| 15:23.33 | VaGoNeTaS | i mean, i was able to speak and listen but with statics |
| 15:23.35 | g-a-m-e-r-x | niggets, hehehe |
| 15:23.42 | g-a-m-e-r-x | i get it xD |
| 15:23.43 | jaytee | VaGoNeTaS, make sure silence suppression is turned off on the phone and if by "static" you mean clicking that's probably jitter caused by the fact that Grandstreams tend to suck and all the ones I've used are famous for it. |
| 15:24.30 | thomasrr | hmm 0 SIP registrations. when calling sip show registry doesnt sound good |
| 15:24.48 | g-a-m-e-r-x | <tzafrir_laptop>, i did the m-a a-i zaptel as sudo but it said failed |
| 15:24.54 | InfoNutz | Hello everyone, i'm looking to change the voicemail prompts on asterisk. Is there a good link on configuring and manageing them? |
| 15:25.14 | tzafrir_laptop | g-a-m-e-r-x, what error? |
| 15:25.25 | g-a-m-e-r-x | oviously after i "sudo apt-get install zaptel-source" |
| 15:25.44 | leifmadsen | InfoNutz: not really, they are just audio files though, and should all be prefixed with vm- I believe |
| 15:26.07 | g-a-m-e-r-x | umm, http://dpaste.com/44426/ |
| 15:26.09 | leifmadsen | the voicemail application will automatically use the files as-is though -- you can't change the way voicemail works on asterisk |
| 15:26.13 | leifmadsen | (without code changes) |
| 15:27.05 | g-a-m-e-r-x | <tzafrir_laptop> http://dpaste.com/44426/ |
| 15:27.35 | tzafrir_laptop | g-a-m-e-r-x, try: m-a -t a-i zaptel |
| 15:27.36 | *** join/#asterisk MrISDN_ (n=kkeil@p5497F57D.dip.t-dialin.net) |
| 15:27.41 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 15:27.53 | tzafrir_laptop | this will run it without this box |
| 15:28.10 | tzafrir_laptop | alternatively, "view" the build log |
| 15:28.28 | jaytee | thomasrr, unless you're asterisk server is registering to another system 0 isn't all that bad. sip show peers shows what devices are registered TO you, sip show registry shows what hosts your * server is registered to |
| 15:29.06 | g-a-m-e-r-x | now its just http://dpaste.com/44430/ |
| 15:31.33 | *** join/#asterisk hi365 (n=hi365@94.159.176.251) |
| 15:32.12 | thomasrr | trying to register to voipbuster |
| 15:32.44 | thomasrr | and the inbound number doesnt want to work |
| 15:32.47 | jaytee | thomasrr, then it should not be 0 which means your sip registration failed |
| 15:33.26 | thomasrr | how can i see what goes wrong? |
| 15:33.29 | VaGoNeTaS | jaytee yes |
| 15:33.32 | tzafrir_laptop | hmm... Ubuntu maintainers seem to have missed this fix |
| 15:33.40 | VaGoNeTaS | but what about when i call from softphone to softphone |
| 15:33.41 | VaGoNeTaS | ? |
| 15:33.54 | thomasrr | jaytee: this is my config atm: http://pastie.org/pastes/479129 |
| 15:34.01 | jaytee | thomasrr, if this is a new install and you're just starting with Asterisk I'd recommend reading the book and check pages 97-101 for registering to a sip provider. |
| 15:34.06 | jaytee | ~book |
| 15:34.07 | 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 |
| 15:34.14 | thomasrr | well it worked boefre :/ |
| 15:34.20 | thomasrr | but not after the reboot |
| 15:34.28 | thomasrr | ip address hasnt changed :/ |
| 15:34.28 | tzafrir_laptop | g-a-m-e-r-x, It's something I fixed in Debian long ago. A trivial patch. I have no idea hwy it's not in Ubuntu yet |
| 15:34.58 | g-a-m-e-r-x | ohh, so its not easly fixable? |
| 15:35.21 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 15:35.21 | thomasrr | maybe its the dns :P |
| 15:35.39 | thomasrr | nope that aint it |
| 15:35.57 | jaytee | thomasrr, this line should be before your user accounts and after the [general] section |
| 15:36.11 | ck_28 | --------------------------do any one install the Fax For Asterisk module? ------------------- |
| 15:36.13 | jaytee | register => XXX@ XXX@sip1.voipbuster.com |
| 15:36.23 | g-a-m-e-r-x | what linux distro would work the best with asterisk |
| 15:36.48 | thomasrr | it shows voipbuster/username Monitored:1 online] though |
| 15:36.53 | Nugget | as far as asterisk is concerned they're all the same. Just pick the distro you hate the least and run with it. |
| 15:36.54 | jaytee | g-a-m-e-r-x, opinions vary. I prefer CentOS or RHEL but some like Debian. |
| 15:37.23 | jaytee | waits for someone to shout out "Slackware FTW!!!" |
| 15:37.34 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
| 15:37.36 | g-a-m-e-r-x | i might put centOs on my other computer then, thanks |
| 15:37.37 | Katty | hugs jaytee |
| 15:37.39 | Nugget | hey, two of my three asterisk servers are slackware. :) |
| 15:37.47 | Nugget | it's the least linuxy linux, imho. |
| 15:38.16 | BlargMaN00 | Nugget: doesn't that defeat the purpose of... well, linux?? |
| 15:38.29 | thomasrr | but it is jaytee after general and before the user accounts :| |
| 15:38.31 | jaytee | g-a-m-e-r-x, of most of the distros CentOS seems to have fewer problems than most and on the wiki there tends to be more how-to's available but there are alot of resources for Debian also. |
| 15:38.32 | telnettech | ALL of our systems are slackware then...We are just now going to be installing CentOS and have used only RHEL |
| 15:38.50 | VaGoNeTaS | maybe is the fucking codec |
| 15:38.55 | tzafrir_laptop | jaytee, the ubuntu package is unmaintained |
| 15:38.56 | jaytee | thomasrr, not in your pastebin it isn't it's after the [voipbuster] account. |
| 15:38.59 | tzafrir_laptop | the debian one is |
| 15:39.15 | tzafrir_laptop | and actually works |
| 15:39.40 | therealcircut | ugh |
| 15:40.24 | jaytee | tzafrir_laptop, I've talked with people running * fine on Ubuntu too. I'm not a distro bigot, just prefer RHEL or CentOS because it's what I ended up using * on and it's what I've grown used to. |
| 15:40.34 | jaytee | I still run Ubuntu as a desktop |
| 15:41.00 | jaytee | hugs Katty |
| 15:41.04 | g-a-m-e-r-x | <tzafrir_laptop>, is the error related to my having my modem being on /dev/ttySM0'? |
| 15:41.04 | therealcircut | can someone explain to me how this scenario might work, A calls B, B answers and wants to transfer to C, B dials a special extension which records A's name and announces it to C, then A is sent to ring C's phone |
| 15:41.08 | jaytee | what's for lunch? |
| 15:41.10 | tzafrir_laptop | jaytee, building Zaptel / DAHDI on those (Centos et. al.) is way more complicated |
| 15:41.26 | therealcircut | jaytee: u use the rpms or build from src? |
| 15:42.29 | g-a-m-e-r-x | i just want to get it set up so it might be able to routa a DID from where my family lives throught voi, just you know to learn a bit about it.. |
| 15:43.29 | jaytee | therealcircut, I always compile, I never use packages on standard Asterisk |
| 15:43.42 | g-a-m-e-r-x | anyway, ill see you all later i have to goto bed, its nearly 2am :( *yawn* |
| 15:44.04 | tzafrir_laptop | g-a-m-e-r-x, see if zaptel-source from http://packages.debian.org/sid/zaptel-source helps you |
| 15:44.09 | g-a-m-e-r-x | chances are ill be back before you know it xD |
| 15:44.26 | g-a-m-e-r-x | okay ill quickly try it now |
| 15:44.53 | jaytee | tzafrir_laptop, I took the advanced * class back in November and Jared had us install * 1.6 with DAHDI and it took about 20 minutes to compile libpri, dahdi, asterisk and the add-ons. not a tough chore at all. |
| 15:45.04 | therealcircut | jaytee: agreeed |
| 15:45.25 | therealcircut | after u get all the gcc / compiler stuff installed via yum |
| 15:45.28 | tzafrir_laptop | jaytee, but what about maintaining such a system? |
| 15:45.34 | therealcircut | the ./configure && make && make install is a breeze |
| 15:45.36 | g-a-m-e-r-x | just got adsl2+ approved today but i have to wait for them to provision it now :(~ |
| 15:45.39 | jaytee | and there's a good how-to on the wiki for CentOS with * 1.4 that can be easily adapted to 1.6 with DAHDI if the person using it has at least half a brain. |
| 15:45.57 | tzafrir_laptop | jaytee, and also: it was jared. I'm pretty sure other gurus know how to avoid the problems. newbs don't |
| 15:46.05 | g-a-m-e-r-x | thanks jaytee, you mean the centos wiki i gues? |
| 15:46.23 | g-a-m-e-r-x | or the * one? |
| 15:46.43 | tzafrir_laptop | the proper fix for that, of course, is (a) to get dahdi into mainline kernel and (b) reduce the dependency on it. (b) is implemented in later versions |
| 15:46.54 | tzafrir_laptop | g-a-m-e-r-x, centos wiki? ha |
| 15:46.58 | thomasrr | its connected |
| 15:47.18 | g-a-m-e-r-x | ha, i know im pryitty much a noob to most other distros |
| 15:47.18 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:47.29 | jaytee | g-a-m-e-r-x, I mean this one on voip-info.org http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation |
| 15:47.41 | tzafrir_laptop | g-a-m-e-r-x, please use 'reportbug' to report this bug to ubuntu |
| 15:47.51 | tzafrir_laptop | for starters. They should fix their broken packages |
| 15:48.09 | g-a-m-e-r-x | yeah |
| 15:48.19 | mmlj4 | they don't call it screwbuntu for nothing... |
| 15:48.20 | tzafrir_laptop | and I have already pointed to you to a place to get a later zaptel-source dpkg that should fix that |
| 15:48.29 | tzafrir_laptop | if it doesn't, it's a bug I should fix |
| 15:48.41 | Nugget | follows tzafrir's guide to git-svn and asterisk |
| 15:49.08 | thomasrr | hmm |
| 15:49.14 | thomasrr | May 15 17:39:27] WARNING[3371]: chan_sip.c:3075 retrans_pkt: Maximum retries exceeded on transmission 4075617656@192_168_100_37 for seqno 3 (Critical Response) -- See doc/sip-retransmit.txt. |
| 15:49.18 | thomasrr | not good right? |
| 15:49.50 | Katty | consumes burbon chicken |
| 15:49.54 | jaytee | tzafrir_laptop, regarding maintaining the system....well, that's a whole nuther ballgame and anyone who wants to run or administer * should at least know basic telecom principles but more importantly know IP networking and linux administration or they're up you know what creek |
| 15:50.15 | Katty | eppigy: season1, episode 7! What are little girls made of. |
| 15:50.29 | jaytee | mmmm, burbon chicken! is that anything like bourbon chicken? |
| 15:50.36 | g-a-m-e-r-x | tzafrir_laptop, says that that package is allready installed and is a newer version |
| 15:50.53 | g-a-m-e-r-x | okay all i havta go now, night! |
| 15:50.59 | jaytee | day! |
| 15:51.11 | g-a-m-e-r-x | lol morning i guess :P |
| 15:51.16 | g-a-m-e-r-x | 2 am :P |
| 15:51.23 | g-a-m-e-r-x | good morning xD |
| 15:51.26 | Katty | jaytee: probably |
| 15:51.26 | g-a-m-e-r-x | bya |
| 15:51.30 | Katty | jaytee: would you like the recipe? |
| 15:51.31 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:51.32 | jaytee | do you know where your children are? |
| 15:52.05 | jaytee | Katty, you're in Missouri, they have to put bourbon in everything there, it's state law! Swear to God it is. :-) |
| 15:54.00 | rbd | hi guys...wondering what voice codecs can be used if one is doing speech analytics... g711 definitely, but I've heard that g729 can't be used ...what about g726, etc? having a hard time finding this info online |
| 15:54.11 | thomasrr | anyone able to tell me how to activate debug mode? |
| 15:54.17 | thomasrr | core debug on, debug on dont work |
| 15:54.36 | Katty | jaytee: http://42ndgeekstreet.blogspot.com/2009/05/bourbon-chicken.html |
| 15:55.24 | *** join/#asterisk mykhyggz (n=mykhyggz@evolone.org) |
| 15:56.06 | *** join/#asterisk bsilberman (n=bsilberm@65.213.221.252) |
| 15:56.43 | Katty | jaytee: a little cornstarch and it'd be a sticky glaze (= |
| 15:56.53 | Katty | jaytee: optionally you can use baked chicken nuggets. |
| 15:57.23 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 15:57.28 | Katty | jaytee: gives it more of a typical chinese resturant taste. |
| 15:57.38 | VaGoNeTaS | how could i change my Audio Codec ? |
| 15:57.52 | Nugget | -- |
| 15:57.52 | Nugget | # Because you use the right editor: |
| 15:57.53 | Nugget | .*.swp |
| 15:57.53 | Nugget | -- |
| 15:57.53 | VaGoNeTaS | coz the phone calls are still getting static |
| 15:57.54 | Nugget | lol |
| 15:58.19 | [TK]D-Fender | VaGoNeTaS: Codecs don't cause static |
| 15:58.24 | ck_28 | rtp.c:1739 ast_rtp_read: Unknown RTP codec 100 received fr |
| 15:58.48 | ck_28 | any one know why the fax t38 fails only send g711 |
| 15:58.53 | ck_28 | [TK]D-Fender hi |
| 15:59.06 | VaGoNeTaS | [TK]D-Fender what does? |
| 15:59.25 | VaGoNeTaS | im able to make and receive calls, i've just setup an Grandstream BT-200 phone |
| 15:59.40 | ck_28 | [TK]D-Fender can you help me in asterisk fax module |
| 15:59.41 | VaGoNeTaS | im making calls and in can listen but with statics |
| 15:59.50 | jaytee | Katty, yeah I've used cornstarch to make a sticky glaze with a ginger/brownsugar/soysauce recipe I have for stir fry green beans |
| 15:59.54 | ck_28 | @russellb hi can you help me in asterisk fax modue |
| 15:59.56 | ck_28 | module |
| 16:00.56 | thomasrr | :( :( |
| 16:00.59 | [TK]D-Fender | VaGoNeTaS: Your description is incomplete and I don't have any impression of the kid of tests you're running |
| 16:01.33 | jaytee | Katty: http://www.youtube.com/watch?v=w2mTX09cTHg |
| 16:01.54 | *** join/#asterisk kfife (n=Miranda@home.chicagoventure.com) |
| 16:02.06 | *** part/#asterisk kfife (n=Miranda@home.chicagoventure.com) |
| 16:02.15 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:02.48 | [TK]D-Fender | jaytee: Old fave |
| 16:03.17 | jaytee | You go home! You too fat!! All you can eat doan me foevah! Son of a bitch! |
| 16:04.05 | jaytee | I love the routine he does about when he was at the French Riviera and kept craving Italian food. |
| 16:04.08 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 16:04.10 | thomasrr | i think i will just hire to some guy to fix this :P |
| 16:04.19 | VaGoNeTaS | of the what? |
| 16:04.25 | VaGoNeTaS | shit |
| 16:06.23 | thomasrr | core debug on doesnt work |
| 16:06.26 | thomasrr | how should i do it now? |
| 16:06.38 | jaytee | thomasrr, did you try sip set debug on? |
| 16:06.50 | thomasrr | hmm |
| 16:06.52 | thomasrr | that works :D |
| 16:07.30 | therealcircut | poor [TK]D-Fender |
| 16:07.36 | therealcircut | hes like a rockstar |
| 16:07.38 | thomasrr | wow |
| 16:07.39 | *** join/#asterisk trentcreek (n=kvirc@200.94.227.117) |
| 16:07.57 | Pan3D | hehe |
| 16:08.00 | jaytee | might want to try typing help all by itself sometime at the CLI just to see what's available. or do asterisk -rx "help">cli_commands.txt so you have a permanent file handy |
| 16:08.23 | trentcreek | who knows what could be a problem when dialing extensions fail, but dialing out works fine? |
| 16:08.36 | jaytee | ok, time to get some lunch, all that chinese buffet stuff is making me hungry. |
| 16:08.38 | jaytee | bbiab |
| 16:08.57 | [TK]D-Fender | trentcreek: everything you dial is an extension. |
| 16:09.05 | Pan3D | [TK]D-Fender has the patient of the Buddha. It is amazing. |
| 16:09.13 | Pan3D | patience* |
| 16:09.19 | [TK]D-Fender | trentcreek: We do not support FreePBX here. You know this. |
| 16:09.22 | trentcreek | okay then I should rephrase... |
| 16:09.31 | trentcreek | Yes, but I am asking ina Asterisk sense |
| 16:09.32 | [TK]D-Fender | Pan3D: That used to be true... |
| 16:09.41 | [TK]D-Fender | trentcreek: everything you dial is an extension. <----------- |
| 16:09.41 | jaytee | they tried to crucify Buddha too but he was so fat and heavy they couldn't lift him onto the cross so they said, "Just sit there!" |
| 16:09.42 | Pan3D | you still rock :) |
| 16:09.49 | Pan3D | haha |
| 16:09.56 | trentcreek | okay..rephrase...unable to dial internal external, but can external |
| 16:09.59 | [TK]D-Fender | trentcreek: And coming in asking that without providing pastebinned backup.... |
| 16:10.06 | Pan3D | jaytee: btw, you mentioned a recipe which sounds quite tasty. You should share that with us. |
| 16:10.12 | [TK]D-Fender | trentcreek: "internal" doesn't mean anything. |
| 16:10.21 | thomasrr | exitSIP/2.0 400 Bad request |
| 16:10.23 | thomasrr | SIP/2.0 400 Bad request |
| 16:10.27 | Pan3D | <3s greenbeans |
| 16:10.29 | thomasrr | hmm, that cant be good right? |
| 16:10.42 | [TK]D-Fender | thomasrr: Depends what it is in RESPONSE to |
| 16:14.40 | *** join/#asterisk ingenius (n=alektro@netsolution.com.ar) |
| 16:14.53 | *** join/#asterisk jtodd (n=jtodd@47.sub-75-252-63.myvzw.com) |
| 16:14.53 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:17.42 | thomasrr | INVITE sip:31857850XXX@sip1.voipbuster.com SIP/2.0 |
| 16:17.50 | thomasrr | is that something i am receiving or sending? |
| 16:18.09 | thomasrr | Via: SIP/2.0/UDP 195.XXX.XXX.XXX:5060;branch=z9hG4bK6003af0e;rport |
| 16:18.11 | thomasrr | is on the line below |
| 16:19.24 | thomasrr | i will just rebuild the system :P |
| 16:20.36 | Pan3D | thomasrr: whoa, back up |
| 16:21.25 | Pan3D | thomasrr: if you're going to use asterisk in any type of real setting, you should learn the SIP protocol and learn to debug. |
| 16:21.42 | Pan3D | in other words, understand what that means. |
| 16:21.54 | Talkradio | Pan3D do you have alink to a good sig debug tutorial? |
| 16:23.01 | Pan3D | The best way to do this is to have some SIP documentation (as in the protocol) open and initiate some calls just see watch the process from start to end. There is a very clear process in SIP once you get past the big blocks of text. It's a conversation with very specific rules. |
| 16:24.24 | Pan3D | Talkradio: Ok, I know this document may look scary, but it has some good explanation and examples "straight from the horses mouth" http://www.ietf.org/rfc/rfc3261.txt |
| 16:25.13 | [TK]D-Fender | thomasrr: What will "rebuilding" do? |
| 16:25.29 | [TK]D-Fender | thomasrr: Won't make you any more competant at reading SIP debug |
| 16:25.57 | [TK]D-Fender | thomYou clearly missed the header line at the start of the packet that says if you're READING or SNDING |
| 16:26.11 | *** part/#asterisk JenniferAkemi (n=Jennifer@76-10-182-237.dsl.teksavvy.com) |
| 16:26.23 | telnettech | thomasrr: read SIP Demystified |
| 16:27.20 | trentcreek | [TK]D-Fender: ALl trunks, extensions, etc. are on port 5060 except one of the extensions which is on port 1024...that make a difference? |
| 16:28.37 | [TK]D-Fender | trentcreek: A very hollow description, and still no debug |
| 16:29.06 | trentcreek | [TK]D-Fender: okay..here is SIP DEBUG ON http://www.pastebin.ca/1423826 |
| 16:29.26 | telnettech | thomasrr: this book was written by someone that was involved with the writing of the sip protocol and it breaks it down pretty good that even a moron like myself could understand |
| 16:29.28 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 16:31.30 | [TK]D-Fender | trentcreek: Looks like you've got 2 phones behind the same NAT |
| 16:31.45 | trentcreek | yes..correct |
| 16:31.56 | trentcreek | one will be remove later |
| 16:32.03 | [TK]D-Fender | trentcreek: I also don't see your SIP configs. |
| 16:32.12 | trentcreek | okay..working |
| 16:32.22 | [TK]D-Fender | trentcreek: And did you test something OTHER than these 2 devices behind the same NAT? |
| 16:32.31 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:32.39 | shareenergy | guys any way to define video resolution in asterisk? |
| 16:33.27 | trentcreek | [TK]D-Fender: I can dial out.. I also have tried setting up one of the IPKall numbers to one of the extensions (100), but only get a busy signal |
| 16:33.55 | [TK]D-Fender | trentcreek: IPKAL does not support OUTBOUND |
| 16:34.43 | trentcreek | I have it setup on inbound |
| 16:34.53 | *** join/#asterisk jjg (n=jjg@12.40.200.74) |
| 16:35.19 | [TK]D-Fender | trentcreek: and "only get busy signal"..... OK you really aren't looking at anything... |
| 16:35.44 | trentcreek | yeah...I am getting there... |
| 16:35.52 | [TK]D-Fender | moves on to something productives |
| 16:35.58 | trentcreek | let me have someone dial it again |
| 16:37.40 | jasonwoot | any quick CLI command to see if an ext is forwarded? |
| 16:39.03 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 16:39.45 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 16:41.27 | *** join/#asterisk a9k (i=a9k@block-66.135.80.2.montanasat.net) |
| 16:42.54 | [TK]D-Fender | jasonwoot: "extensions" has no concept of "forwarding" |
| 16:43.19 | *** join/#asterisk Micc (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 16:44.07 | jasonwoot | yeah, its the stupid Polycom 501 with the "forward" button right next to the "new call" button... what genius designed that? |
| 16:44.36 | jasonwoot | rasterisk | grep forward found it |
| 16:44.46 | Katty | jasonwoot: tell me about it |
| 16:44.52 | Katty | jasonwoot: the boss constantly puts his phone on forward |
| 16:44.57 | jasonwoot | I found the XML to remove that button, but the damn things reappear at random |
| 16:45.02 | Katty | jasonwoot: then whines to me that no one can call him |
| 16:45.17 | Katty | eppigy: Dagger of the Mind |
| 16:45.20 | jaytee | Pan3D, ask and ye shall receive! and for anyone else interested. Spicy Ginger Green Bean Stir fry: http://pastebin.ca/1423849 |
| 16:45.29 | [TK]D-Fender | jasonwoot: Do share... |
| 16:46.39 | Micc | I've got a problem with dtmf with one of my customers using an spa8000. We thought it was working yesterday with dtmfmode=info, but now when they call into their voicemail it doesn't see the dtmf tones. |
| 16:46.41 | jasonwoot | not enough to pastebin: <divert divert.fwd.1.enabled = "0"/> |
| 16:46.53 | jasonwoot | anywhere in the <reg> container |
| 16:48.29 | [TK]D-Fender | jasonwoot: I means what the rasterisk dumps showed you |
| 16:48.58 | a9k | I'm running asterisk 1.6.0.6 & wanpipe 3.3.15 with Sangoma A200. Everything works except outbound analog dial. It dials _one_ digit and not always the first digit. http://pastebin.ca/1423837 |
| 16:49.00 | Pan3D | jaytee: awesome, thanks! |
| 16:49.27 | jaytee | Pan3D, bon appetite |
| 16:49.41 | jasonwoot | [May 15 11:49:17] -- Now forwarding SIP/6430-b786bc78 to 'Local/20@INT-LOC-TOLL-BYPASS' (thanks to SIP/6379-0a12ba20) |
| 16:52.00 | telnettech | i love working from home |
| 16:52.02 | rbd | hi guys... is there a way for asterisk (in sip.conf) to preserve the codec in use ...current codec negiotiation seems kind of inflexible ..e.g. if I have allow=g711 and allow=g729, and a call comes in as g729, I want to keep it as G729 even if it isn't my #1 preferred codec...e.g. I always want to go with what the call originator prefers to use |
| 16:52.04 | [TK]D-Fender | jasonwoot: thats live debug, not a logged event though |
| 16:52.06 | rbd | is this possible? |
| 16:52.08 | telnettech | telecommuting is awesome!!!! |
| 16:52.22 | [TK]D-Fender | jasonwoot: as in not a "state" so much as an action. |
| 16:52.36 | [TK]D-Fender | telnettech: I love NOT working from home. |
| 16:52.43 | [TK]D-Fender | has the day OFF |
| 16:52.54 | voxter | I hate to ask cause its such a clusterfuck, but is anyone recently familiar with upgrading cisco 7941's to sip? I'm stuck in a loop of tftp requesting CTLSEP/SEP files - tried creating 0 byte CTLSEP and it just sticks looping on that one. |
| 16:53.19 | telnettech | I havent worked from home before but if I can get more done here at home then sitting in the office, Im all for it!!!! |
| 16:54.06 | telnettech | I have cleared my backlog of support tickets and only have the 2 that i need to concentrate on |
| 16:54.55 | telnettech | I started today with 23 tickets |
| 16:55.08 | telnettech | mgmt will be happy with the production |
| 16:55.23 | telnettech | or i should say productivity |
| 16:55.40 | stope | no matter what you do, they're never happy and always want more |
| 16:55.49 | Katty | gives [TK]D-Fender cheetos. |
| 16:56.00 | Katty | [TK]D-Fender: they're baked! |
| 16:56.05 | Katty | [TK]D-Fender: so you can keep your girlish....errr |
| 16:56.11 | Katty | [TK]D-Fender: [TK]D-Fenderisher figure. |
| 16:56.24 | [TK]D-Fender | Katty: So's anyone who believes that makes them "healthy" :p |
| 16:56.34 | jaytee | telnettech, no they won't. First rule of business is no matter what you do, it's not enough and one thing I've learned in life at 55 years of age is that management is never happy or if they are it's only after they've fired your ass. |
| 16:56.41 | Katty | [TK]D-Fender: live a little ;) |
| 16:57.25 | [TK]D-Fender | Katty: I'm looking at skydiving, hang-gliding and wind-surfing this year.... good enough? |
| 16:57.26 | telnettech | well they will prolly be doing that next since they will be moving support to the "NOC" in manila around the 25th of May |
| 16:57.32 | Katty | [TK]D-Fender: no |
| 16:57.36 | Katty | [TK]D-Fender: not good enough |
| 16:58.17 | jaytee | skydiving is like freebasing cocaine |
| 16:58.24 | telnettech | and they are trying to make it where "anyone" that can put a CD into a PC can install and "configure" asterisk as a PBX |
| 16:58.27 | *** join/#asterisk a9k (i=a9k@block-66.135.80.2.montanasat.net) |
| 16:58.43 | [TK]D-Fender | telnettech: Oh... you mean Trixbox? :p |
| 16:58.55 | Katty | you've not lived until you've had cheetos and ice cream in the same hour. |
| 16:59.05 | telnettech | TK: no they are trying to get it even simpler |
| 16:59.07 | Katty | probably /after/ the skydiving would be preferable. |
| 16:59.07 | [TK]D-Fender | Katty: I may well have already... |
| 16:59.45 | Katty | two lines a bump later... |
| 17:00.39 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-132-169.telkomadsl.co.za) |
| 17:00.57 | jaytee | telnettech, whenever someone comes up with something "idiot proof" God just invents a better idiot. |
| 17:01.41 | telnettech | jaytee: I agree but until then what should the rest of us idiots do to? |
| 17:01.42 | jaytee | or nature evolves a better idiot through adaptation |
| 17:02.08 | jaytee | telnettech, there's always Professional Bowling or Barber College |
| 17:02.38 | Kobaz | do you guys know of any providers other than broadbox that offer tier1 level sip termination/origination |
| 17:03.10 | telnettech | jaytee: I do have some experience cutting hair.<looks at himself in mirror> |
| 17:04.53 | jaytee | telnettech, when you can do a fancy combover job ala The Donald then you know you've made it to the big time :-) |
| 17:05.09 | [TK]D-Fender | telnettech: You should realize that some people are too dumb to be doing certain jobs. |
| 17:05.19 | Katty | [TK]D-Fender: yeah |
| 17:05.21 | Katty | [TK]D-Fender: you should retire |
| 17:05.26 | Katty | oh wait |
| 17:05.28 | Katty | did i say that outloud |
| 17:05.33 | Katty | oops. |
| 17:05.36 | Katty | goes back to star trek |
| 17:05.37 | rbd | anyone had any luck with speech recognition (e.g. lumenvox, nuance) over g729? I've heard g729 doesn't cut it for accurate speech recogn |
| 17:05.40 | telnettech | TK: tell mgmt that!!! |
| 17:05.46 | [TK]D-Fender | Katty: I'm learning how to "retire" people already :) I'm quite good at it. |
| 17:05.53 | [TK]D-Fender | is now 3rd Kyu |
| 17:06.09 | [TK]D-Fender | rbd: It doesn't |
| 17:06.22 | [TK]D-Fender | rbd: Too compressed for accuracy |
| 17:06.36 | [TK]D-Fender | rbd: Might help if you are limiting your dictionary |
| 17:06.38 | rbd | [TK]D-Fender, what about g726-32? or are we basically looking at 711 only? |
| 17:07.05 | [TK]D-Fender | rbd: G726-32 is going to have way better odds than anything else on the PSTN than G711 |
| 17:07.05 | jaytee | almost all of my problems with Lumenvox are with calls from cell phones |
| 17:07.45 | rbd | also, I was wondering people's opinions of lumenvox compaired to other (more expensive) solutions like nuance |
| 17:08.02 | rbd | I tried their demo out and it was OK, but it didn't respond to background noise very well |
| 17:08.24 | rbd | jaytee, yeah I saw that...it sucked when I was on my cellphone in a restaurant |
| 17:09.22 | jaytee | I have a submenu on my IVR for obtaining driving directions, 4 choices: North, South, East and West. Lumenvox hates South for some reason. |
| 17:10.55 | rbd | was looking at the MOS scores for the various codecs...I would think g726 would have a much higher MOS than g729 but that doesn't appear to be the case |
| 17:11.55 | [TK]D-Fender | jaytee: N1H1 !!!!!! |
| 17:15.31 | jaytee | [TK]D-Fender, sorry, but I must be having a senior moment. I'm not following. |
| 17:15.50 | telnettech | thats the flu virus |
| 17:15.52 | [TK]D-Fender | jaytee: Think about good reasons not to want to go "South" |
| 17:15.58 | jaytee | oh yeah! duh! |
| 17:16.21 | [TK]D-Fender | hands jaytee a file.... |
| 17:17.38 | jaytee | See! this is a perfect example of what happens to people when they do Windows support for a living for too long. I've reached an advanced state of MSD, Microsoft Systems Dementia |
| 17:18.44 | jaytee | and of course with my 401K pretty much evaporated I'll have to work till I'm 90. I can just see it now. |
| 17:19.18 | Katty | i really wish i hadn't read that |
| 17:19.18 | jaytee | Boss: "Hey, we're upgrading to Windows 11 next week!!!" "Oh, joy! Just shoot me now and be done with it" |
| 17:19.30 | Katty | eww. |
| 17:20.22 | Katty | oh! shiny! |
| 17:20.24 | Katty | gets distracted |
| 17:20.43 | *** join/#asterisk duckz (n=duckz@81-31-157-49.vm.dnshosting.it) |
| 17:20.53 | jaytee | my ex got laid off from Microsoft on the 5th. her and her fellow ex-employees are calling the day Cinco De Firo |
| 17:21.44 | jaytee | shiny is such a cool word. It has special meaning to a die hard Firefly fan like me :-) |
| 17:22.03 | [TK]D-Fender | jaytee: Same with "Darn" |
| 17:22.09 | jaytee | hehe |
| 17:22.18 | [TK]D-Fender | boots his co-workers into jet turbines |
| 17:22.20 | jaytee | when he kicks him into the engine intake? love that scent |
| 17:22.24 | jaytee | scene |
| 17:22.40 | [TK]D-Fender | jaytee: Yup.... totally bad-ass |
| 17:22.44 | *** join/#asterisk vasundhar (n=vasundha@122.169.130.112) |
| 17:23.12 | jaytee | "if you take sexual advantage of that woman, you'll go to a special hell, a hell usually reserved for child molesters and people that talk at the theater." |
| 17:23.49 | Katty | :> |
| 17:23.59 | vasundhar | I am getting segmentation error while trying to play mp4 on sip through asterisk any suggestions ? |
| 17:24.12 | jaytee | don't play mp4s |
| 17:24.17 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:24.29 | jaytee | mp4 is usually video not audio |
| 17:24.34 | [TK]D-Fender | jaytee: Doctor, Doctor it hurts when I ..... awwwwww fukkit |
| 17:24.51 | jaytee | haha |
| 17:25.04 | vasundhar | Yes I know Mp4 is video ... I want to play streaming video on call |
| 17:25.29 | jaytee | `wglwat |
| 17:25.39 | jaytee | ~wglwat |
| 17:25.39 | infobot | rumour has it, wglwat is well, good luck with all that |
| 17:25.42 | vasundhar | which is supported by asterisk with another file app_mp4.c located under apps dir |
| 17:26.02 | jaytee | it is? when did they come out with that? |
| 17:26.31 | vasundhar | well go to sip.fontventa.org :) |
| 17:26.57 | jaytee | nah, I'd rather just hang out here and goof off :-) |
| 17:27.55 | telnettech | i actually have time to read more about asterisk and sip |
| 17:28.07 | telnettech | with all my work caught up today :) |
| 17:28.44 | telnettech | that way i have more intelligent questions to ask TK about :) |
| 17:29.13 | vasundhar | O:-) |
| 17:30.32 | Katty | pst, vasundhar |
| 17:30.36 | Katty | price tag is still on the halo |
| 17:30.39 | Katty | hands vasundhar scissors |
| 17:31.42 | [TK]D-Fender | 's native name translates as "Runs With Scissors" |
| 17:33.34 | jaytee | telnettech, I got all caught up on work just so I could ask [TK]D-Fender to explain about quantum indeterminancy which always confused me and he just answered with "Yes, the cat is alive or no, the cat is dead. Take your pick" |
| 17:33.43 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 17:35.11 | eppigy | hello Katty ^___________^ |
| 17:35.22 | jaytee | he is Dave! |
| 17:35.34 | [TK]D-Fender | telnettech: He also asked me if I was still indecisive and and dodgy. I told him I'm not sure and I'llget back to him on it... |
| 17:35.46 | jaytee | hehehe |
| 17:36.01 | *** join/#asterisk sHoZaIb (i=rOfLz@216.131.64.27) |
| 17:37.43 | *** join/#asterisk dundel (n=dundel@200.2.161.143) |
| 17:37.56 | vasundhar | So far I thought only programming confuses me ... Now I realised I can't make out even chat .. |
| 17:38.28 | dundel | hi i'm using the D-Link DPH 140S, but my forward button is not working does anybody have experience with this type of voip phone? |
| 17:39.45 | Katty | HAI DAVE |
| 17:39.46 | Katty | let's hug. |
| 17:40.25 | *** part/#asterisk vasundhar (n=vasundha@122.169.130.112) |
| 17:41.27 | sHoZaIb | I am having problem when calling out there is no voice after call is been establish and rtp log do not show me any rtp packet from sip proxy |
| 17:43.18 | Katty | what i wouldn't give for a couple margaritas and some good company |
| 17:50.05 | jaytee | D-Link makes Voip phones? hmmmm |
| 17:50.11 | Katty | skeery |
| 17:52.25 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 17:52.58 | *** join/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 17:53.09 | *** part/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 17:54.56 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 17:55.18 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:58.15 | *** join/#asterisk kiall (i=mit@unaffiliated/kiall) |
| 17:58.15 | *** join/#asterisk WHYS (n=drumm@137.28.94.209) |
| 17:58.17 | telnettech | just when i could catch my breath, i get more tickets.....Like 6 new tickets.....i think the supervisor is hoarding tickets....:( |
| 17:59.27 | WHYS | Where can I find the recommended minimum specifications for * Business Edition? |
| 18:00.17 | kiall | Hey all - looking for some advice on IP Phone brands - What brands to people recommend, and why? Voice Quality is obv my biggest concern ... thanks... |
| 18:00.27 | jaytee | WHYS, have you looked on Digium's website? |
| 18:00.45 | WHYS | Yes, but where? I just can't find it. Been looking for 20 minutes |
| 18:01.40 | WHYS | kiall: check out Polycom, Aastra, and cisco - Aastra seem to be priced the best and has lots of features if you like that. |
| 18:02.27 | kiall | WHYS: cool .. I have a few Cisco 7941's already - love them, but there pricey ;) |
| 18:02.49 | *** join/#asterisk joelsolanki (i=joelsola@124.125.151.78) |
| 18:02.51 | joelsolanki | Hi all. |
| 18:03.14 | WHYS | yep. same here. I've been tasked to look at option, and Aastra seem like a good one, although I don't have one inhand. |
| 18:03.18 | [TK]D-Fender | kiall: Polycom > All |
| 18:03.27 | [TK]D-Fender | WHYS: Go call Digium and ask. |
| 18:03.43 | [TK]D-Fender | WHYS: And its no different than * OSE |
| 18:04.12 | kiall | [TK]D-Fender: any reason you prefer them? |
| 18:04.55 | joelsolanki | please see this pastebin. it is a debug of an incoming call where it says codecs not compatible codecs. But codecs are installed and licenses are also there. if i dial from eyebeam with g729 codec it works but if i dial DID --> VPS --> Asterisk then it says no compatiblle codecs. |
| 18:05.14 | kiall | WHYS: a recommended system spec is very hard to give since its *very* dependant on the number of concurrent calls and the codecs used... |
| 18:05.16 | joelsolanki | please see this pastebin . any hints ? http://pastebin.ca/1423921 |
| 18:05.26 | WHYS | I want to test and get ABE working on lower hardware, and THEN purchase a production server at the end of summer. the initial hardware is not going to be anything new. |
| 18:05.27 | [TK]D-Fender | kiall: Reliability, audio quality, massive configurability, competitive North American pricing, strong SIP support and growing feature set |
| 18:05.40 | [TK]D-Fender | WHYS: How about actual details... |
| 18:06.22 | WHYS | :) - ummm, not sure. an old dell desktop. 3Ghz - 512M, 80GB... |
| 18:06.24 | kiall | WHYS: yup ... how many concurrent calls are you planning on? + have you considered which codecs will be used? |
| 18:06.42 | [TK]D-Fender | WHYS: ... no comment. Stop worrying and just f-ing install it already :) |
| 18:06.57 | WHYS | Ulaw, just for testing, no more than ten. Got gold support. |
| 18:07.11 | [TK]D-Fender | WHYS: A P3 could do that |
| 18:07.13 | kiall | put it this way ... when i evaled * on a P2 300Mhz and had 5 concurrent calls no hassle... |
| 18:07.41 | WHYS | I have installed many time, but want to show support so had my boss spend some $ |
| 18:07.52 | [TK]D-Fender | joelsolanki: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x0 (nothing) <-- what part of this is not painfully apparent? |
| 18:07.55 | kiall | i literally just grabbed the only spare PC i had (we normally VM everything but I;ve heard * doesnt deal with VMs very well) |
| 18:08.02 | [TK]D-Fender | joelsolanki: No compatible codecs <- |
| 18:08.24 | WHYS | I just want to make sure digium support won't laugh me off when I call. Also want to be sure I can move the license. |
| 18:08.26 | joelsolanki | yes it says no compatible codecs |
| 18:08.30 | joelsolanki | but g729 is installed |
| 18:08.53 | [TK]D-Fender | joelsolanki: and you did not configure * to OFER IT |
| 18:08.57 | [TK]D-Fender | OFFER* |
| 18:08.58 | jaytee | yeah, I ran mine on a P3 Coppermine with 512MB of ram, 8 FXO ports and 20 phones before upgrading the hardware and only did that to prepare for a major migration of 200 phones |
| 18:09.02 | [TK]D-Fender | joelsolanki: Look at the "US" |
| 18:09.22 | [TK]D-Fender | joelsolanki: Go change your configs, you are not allowing G.729. |
| 18:09.44 | joelsolanki | let me see. else i will paste the config |
| 18:10.11 | [TK]D-Fender | joelsolanki: What's to paste? You don';t know how to put an "allow" statement to permit a codec? |
| 18:10.49 | joelsolanki | i have allow=all |
| 18:11.00 | WHYS | I have a Fax server setup as a VM. Works pretty good. It's on a really speedy server though. Still my desktop VMs worked well. not production quality though. |
| 18:11.00 | joelsolanki | let me try to disallow all and then allow g729 |
| 18:11.50 | [TK]D-Fender | joelsolanki: You do NOT have "allow=all" for the section that is processing the call. Go look at your own debug |
| 18:12.08 | [TK]D-Fender | joelsolanki: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x0 (nothing) <-- you are clearly restricting to 4 codecs there |
| 18:13.32 | joelsolanki | ok checking |
| 18:19.00 | *** join/#asterisk saftsack (n=saftsack@p5792476A.dip.t-dialin.net) |
| 18:25.33 | yang | is anyone willing to send a trial FAX ? |
| 18:26.50 | KavanS | yang, google "hp fax back" or "hp fax me" line |
| 18:26.59 | KavanS | yang, free fax service to test with :) |
| 18:27.17 | yang | i tried several allready |
| 18:29.03 | WHYS | sure. send my your fax number. |
| 18:31.21 | *** join/#asterisk simprix (n=simprix@cosmas.supportdept.com) |
| 18:32.43 | yang | thanks WHYS do you see query? |
| 18:32.56 | WHYS | nope |
| 18:33.01 | yang | oh |
| 18:34.39 | *** join/#asterisk gego (n=gego@dyndsl-085-016-098-004.ewe-ip-backbone.de) |
| 18:40.24 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 18:41.48 | a9k | Anyone understand the chan_dahdi.c code? I'm trying to find where it breaks the dial string down into digits. |
| 18:42.44 | [TK]D-Fender | a9k: Plug an analog phone in parallel and listen to it dial |
| 18:44.09 | a9k | [TK]D-Fender: I've done that and have log of the misbehaviour. see http://pastebin.ca/1423837 - only one digit sounds |
| 18:47.55 | *** join/#asterisk nephfl (n=none@wsip-98-175-64-147.ga.at.cox.net) |
| 18:48.06 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:48.10 | nephfl | my call files arent being picked up and i have no idea why |
| 18:48.39 | [TK]D-Fender | a9k: whats with the "T" in front? |
| 18:49.26 | a9k | [TK]D-Fender: i don't add that - I believe its for Tone dialing |
| 18:49.43 | [TK]D-Fender | a9k: What ver of wanpipe, & dahdi? |
| 18:53.08 | a9k | wanpipe-3.3.15 , asterisk-1.6.0.6 hmmm dadhi was dadhi-linux-current on Mar 8 2009. Is there a way go get the dadhi version number? |
| 18:53.31 | [TK]D-Fender | a9k: install over it |
| 18:54.14 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 18:54.32 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 18:55.10 | a9k | [TK]D-Fender: you mean make install? |
| 18:55.52 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:55.54 | nephfl | anybody know why my call file may not be processed? it has the correct owner/group |
| 18:57.17 | *** join/#asterisk hfb (n=hfb@pool-96-247-49-46.lsanca.dsl-w.verizon.net) |
| 18:58.34 | a9k | [TK]D-Fender: only sign of a date is "Firmware dahdi-fw-oct6114-064.bin" ... version 1.05.01 and other firmware versions |
| 19:01.51 | a9k | [TK]D-Fender: I'll try latest dadhi. Had to patch asterisk 1.6 with 14577 patch to get this far. |
| 19:03.33 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 19:03.33 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:06.39 | *** join/#asterisk Dovid (n=annon@174-147-189-134.pools.spcsdns.net) |
| 19:08.10 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 19:08.10 | *** mode/#asterisk [+o file] by ChanServ |
| 19:10.27 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 19:10.33 | guax | ~nat |
| 19:10.34 | infobot | rumour has it, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 19:10.46 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 19:10.53 | guax | there wasnt a bot shortcut for nat problems? |
| 19:11.04 | bmoraca | ~sipnat |
| 19:11.05 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:11.11 | guax | bmoraca, thank you |
| 19:12.28 | therealcircut | cepstral tech support is nice |
| 19:13.53 | Anth8708 | hey guys, for 1.6, is it still SIPAddHeader(Alert-Info:<ringtone name>) for polycom phones? |
| 19:14.31 | therealcircut | Anth8708: hold |
| 19:14.34 | therealcircut | let me paste u mine |
| 19:14.49 | Anth8708 | therealcircut: thanks:) |
| 19:14.58 | therealcircut | SIPAddHeader(Alert-Info: Ring Answer) |
| 19:14.59 | [TK]D-Fender | Anth8708: yes |
| 19:15.02 | therealcircut | were running 601's |
| 19:15.17 | Anth8708 | thanks guys |
| 19:15.37 | *** join/#asterisk uehueh (n=email@evdomip-1-21.iusacell.net) |
| 19:16.29 | uehueh | hello, I have a question, I am calling into my asterisk box, i have it routed to DISA, then I have the outbound Dial string with ,30,tT. I am trying to do blind transfer after I call a number and I am in progess with that call, I need to then transfer that call to a new person, any idea? |
| 19:16.36 | uehueh | blindxfer over DISA possible? |
| 19:17.05 | *** join/#asterisk ZenBSDi (n=ZenBSDi@unaffiliated/ZenBSDi) |
| 19:17.22 | ZenBSDi | yawns |
| 19:17.26 | ZenBSDi | Sup Room |
| 19:17.34 | uehueh | no ones here:\ |
| 19:17.43 | [TK]D-Fender | uehueh: yes |
| 19:17.43 | uehueh | you know anything about blindxfer over disa? |
| 19:17.46 | a9k | nothing but bots |
| 19:17.50 | ZenBSDi | lol |
| 19:17.56 | [TK]D-Fender | segfaults |
| 19:17.56 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 19:17.58 | uehueh | :x |
| 19:18.13 | uehueh | its possible Fender? |
| 19:18.23 | ZenBSDi | I don't use DISA .. |
| 19:18.25 | [TK]D-Fender | uehueh: yes |
| 19:18.52 | uehueh | hmm I am not having luck with it |
| 19:19.43 | ZenBSDi | I created a context called dialoutcenter.. I just dial 2350, it does a authenticate and then read(num) and then dial it :p |
| 19:19.46 | bmoraca | uehueh: you'd get more answers with a pastebin of your dialplan and a pastebin of your error logs |
| 19:20.12 | uehueh | i have blindxfer => *1 enabled in features.conf, disa is enabled, I call in, I get the dialtone, I call a new number, I am in progres with that number, I dial *1 and it does not do anything |
| 19:20.25 | uehueh | bmoraca, Its simply 2 lines.. |
| 19:20.46 | ZenBSDi | uehueh, ahh.. you want to be able to xfer calls too .. |
| 19:20.50 | ZenBSDi | I never thought of that :p |
| 19:20.51 | uehueh | yes |
| 19:21.03 | uehueh | i want to be in disa with a person, and then transfer him to someone else |
| 19:21.04 | uehueh | :D |
| 19:21.40 | [TK]D-Fender | uehueh: WITH a person. Sounds like a 3-way call to me. |
| 19:22.01 | [TK]D-Fender | uehueh: Doesn't make much sense |
| 19:22.14 | uehueh | Fender |
| 19:22.48 | [TK]D-Fender | uehueh: And I gues you'd better PASTEBIN a failed call attempt along with your confis |
| 19:22.50 | uehueh | what if you DISA in, you all someone from the corporate network... and then your explaining something, and then you want to transfer them to customer care |
| 19:22.51 | [TK]D-Fender | ~pb |
| 19:22.52 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 19:22.58 | uehueh | there is no Fail call.. |
| 19:23.17 | [TK]D-Fender | uehueh: Show me the call. |
| 19:23.19 | uehueh | it just does not reconize the *1 from features.conf |
| 19:23.27 | uehueh | OK |
| 19:23.29 | uehueh | just a second |
| 19:24.15 | ZenBSDi | I prefer pastebin.ca |
| 19:24.21 | ZenBSDi | it has a search feature :p |
| 19:26.06 | [TK]D-Fender | I prefer .com it lads a hell of a lot faster |
| 19:26.11 | [TK]D-Fender | loads* |
| 19:26.53 | Qwell | .ca supports ipv6 |
| 19:27.21 | uehueh | http://www.pastebin.ca/1424008 |
| 19:28.44 | [TK]D-Fender | uehueh: what happens when you try # for transfer? |
| 19:28.49 | uehueh | and features.conf is the default features.conf but I took out the comment |
| 19:28.55 | uehueh | one moment |
| 19:29.15 | uehueh | just call the disa and dial new number, when answered Ill press # |
| 19:29.21 | [TK]D-Fender | uehueh: Also right now your inbound channel's context has nowhere you can productively transfer callers to <- |
| 19:30.04 | ZenBSDi | I just looked in the features.conf and it's #1 |
| 19:30.23 | uehueh | yes Its #1 but i made it *1 because you cant dial # number from DISA |
| 19:30.28 | uehueh | but Im running 1.6x |
| 19:30.38 | ZenBSDi | ewwww... |
| 19:30.57 | uehueh | im only running 1.6x because it also fails in 1.4x |
| 19:31.21 | uehueh | :< |
| 19:31.34 | uehueh | what do you mean my inbound channels context has no where I can transfer calls to? |
| 19:31.47 | uehueh | my inbound context is simply answer, go to disa |
| 19:32.08 | uehueh | i want to transfer calls to , real phone numbers |
| 19:32.16 | uehueh | no external extensions |
| 19:32.52 | Beave | yawns |
| 19:33.01 | eppigy | TRABAJO |
| 19:33.27 | uehueh | estas trabjando? |
| 19:33.43 | uehueh | yo quiero morir por eso |
| 19:34.03 | seanbright | donde esta la biblioteca? |
| 19:34.11 | [TK]D-Fender | uehueh: What do you mean can't dial "3" + number? |
| 19:34.20 | *** part/#asterisk MrISDN (n=kkeil@p5497F57D.dip.t-dialin.net) |
| 19:34.27 | [TK]D-Fender | # |
| 19:34.52 | uehueh | well if it dial #+number it just stays on the same phone call, nothing happens.. |
| 19:35.41 | [TK]D-Fender | uehueh: IT? |
| 19:35.50 | uehueh | I |
| 19:35.56 | [TK]D-Fender | uehueh: You should hear something immediately on pressing # |
| 19:36.01 | uehueh | I just say on the current phone call |
| 19:36.13 | [TK]D-Fender | uehueh: What happens if instead of doing DISA you just dial out normally? |
| 19:36.14 | uehueh | I understand but I do not hear anything, nothing actually happens |
| 19:36.25 | uehueh | yes it works when I dialout normally.. |
| 19:36.39 | uehueh | but the thing is.. I want to do it over DISA |
| 19:36.40 | [TK]D-Fender | uehueh: Then ditch DISA and just make this a basic IVR. |
| 19:37.07 | uehueh | but I need to call all sorts of numbers for the company |
| 19:37.18 | uehueh | and Im limited with ivr creation experience.. |
| 19:37.24 | [TK]D-Fender | uehueh: No difference |
| 19:37.28 | uehueh | but Ill check it out on google |
| 19:37.43 | uehueh | I have another question.. if I am going to do it with IVR |
| 19:38.10 | uehueh | is there a way I can just dial, a certain sequence and it will transfer the current call to a predetermined number? |
| 19:39.22 | uehueh | its more like, Ok ms smith let me transfer you to my manager, but I want the people to be able to dial in from out of company numbers, but the predetermined number never changes |
| 19:41.01 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 19:41.30 | Dovid | how is G722 supported in asterisk 1.4.x ? |
| 19:42.04 | Dovid | if I use a local channel will that break the pass through ? |
| 19:45.51 | Katty | i like how they look up to the sky when trying to reach the enterprise in the earlier episodes. |
| 19:47.27 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 19:48.36 | a9k | Just great. latest dahdi changed struct dahdi_span so wanpipe latest from sangoma doesn't compile. Soooo frustrated. |
| 19:52.10 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 19:52.33 | *** join/#asterisk ks3 (n=ks3@74.203.195.1) |
| 19:53.34 | keith4 | does IAX have a better change of not being hosed by a restrictive firewall, compared with SIP? |
| 19:55.05 | bmoraca | keith4: technically, yes. however, a properly configured firewall should have no problem with either |
| 19:55.23 | Dovid | will using a local channel in Asterisk break g722 ? |
| 19:55.33 | Corydon76-dig | Better, yes, but Sonicwall is known to fuck up both protocols |
| 19:55.40 | [TK]D-Fender | Dovid: passthrough only |
| 19:55.53 | keith4 | bmoraca: i have a user who is in Prague for a month. using SIP, nobody can hear her |
| 19:55.56 | Dovid | TK: and Local will break that apart |
| 19:55.57 | Dovid | ? |
| 19:55.59 | [TK]D-Fender | keith4: Depends how restrictive. |
| 19:56.12 | [TK]D-Fender | Dovid: Channel type is not importan, your USE of it is |
| 19:56.14 | keith4 | thought i might give her an IAX softphone to try |
| 19:56.24 | bmoraca | keith4: what model firewall on both ends and what's your SIP.conf look like? |
| 19:56.39 | Corydon76-dig | If she sent a call through the forest, and nobody heard her, did she make a sound? |
| 19:57.07 | Dovid | TK: Ok. trying to figure out why g722 is failing when both ends are using it. I know using Local broke T.38 when it should have been pass through |
| 19:57.13 | Dovid | lol |
| 19:57.15 | Corydon76-dig | keith4: you're better off just installing openvpn and running the call over that |
| 19:57.16 | keith4 | I have no idea what the network is like on her end. just that she's somehow NAT'd, because the traffic comes from 195.113.65.8, but she says her computer is 10.8.76.210 |
| 19:57.30 | Dovid | TK: Have you ever tried any of the patches for G722 ? |
| 19:57.39 | Dovid | like: http://carlton.oriley.net/drupal/node/12 ? |
| 19:57.44 | keith4 | I've even moved her to a non-NAT'd asterisk box, on our end |
| 19:58.00 | Corydon76-dig | keith4: double and triple NAT usually does a fine job of screwing SIP to the wall |
| 19:58.07 | bmoraca | keith4: what's your sip.conf look like and what kind of router do you have? and, what kind of phone does she have? |
| 19:58.11 | Corydon76-dig | but IAX2 has no problem with either |
| 19:58.28 | bmoraca | sonicwalls are garbage :) |
| 19:58.38 | keith4 | bmoraca: the asterisk server is in a datacenter in california |
| 19:58.41 | Dovid | sonicwall will give u lots of white hairs for nothing |
| 19:58.58 | Dovid | and also drop packets just because they r bored when they r told not to |
| 19:59.02 | keith4 | yah... think I might try an IAX softphone |
| 19:59.03 | a9k | sonicwalls are high priced garbage. |
| 19:59.17 | bmoraca | keith4: that's all well and good...i've got asterisk servers in a datacenter in California, too...but it's not what I asked... |
| 19:59.22 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 19:59.33 | Katty | has anyone recorded on hold adverts/music before? and if so, how did you go about doing it |
| 19:59.46 | keith4 | bmoraca: just saying... it's not like it's behind some crap DSL router |
| 20:00.16 | keith4 | i'm fairly sure the problem is on the Prague end, as nobody else has any trouble, NAT'd or not, softphones or hardphones |
| 20:00.39 | bmoraca | keith4: i didn't say it was. but there are certain things you need to do on all routers to make sure that SIP passthrough is properly working...or potentially your NAT was screwed up (maybe you're using a PAT instead?)...there's a hundred ways this could get screwed up |
| 20:01.08 | keith4 | bmoraca: that's great. what I'm saying is: I don't have control over any of the network equipment |
| 20:01.10 | bmoraca | keith4: without knowing what kind of router she has there (can she even make changes to it?) you will never know |
| 20:01.15 | [TK]D-Fender | Katty: Microphone and a tape recorder :) |
| 20:01.16 | bmoraca | ahh |
| 20:01.19 | bmoraca | that's a problem, then |
| 20:01.26 | keith4 | I believe she's staying at a university dorm |
| 20:01.47 | keith4 | VPN is probably out of the question, too |
| 20:02.05 | ruben23 | hi nayone can interpret this error log on my asterisk server=>http://pastebin.com/m58582b9f |
| 20:02.25 | Katty | [TK]D-Fender: any other way that might be better quality |
| 20:02.25 | bmoraca | why do you say that? SSL VPN requires no passthrough at all |
| 20:03.04 | bmoraca | Katty: I've always just recorded them straight into Asterisk as whatever codec file i'm using |
| 20:03.13 | [TK]D-Fender | Katty: a mic and some piece of software for recording audio at maybe some kind of decent quality... |
| 20:03.22 | Katty | [TK]D-Fender: software suggestions? |
| 20:03.32 | [TK]D-Fender | Katty: Audacity |
| 20:03.36 | bmoraca | alternatively...Nero Wave Editor works well to save as an MP3 if that's the format your MOH directory is in |
| 20:03.37 | Katty | [TK]D-Fender: thank you |
| 20:03.54 | [TK]D-Fender | Katty: its on sourceforge for *NIX & Windows |
| 20:10.15 | keith4 | and OS X |
| 20:11.44 | eppigy | THE AUDACTOTY |
| 20:11.49 | eppigy | AUDACITY |
| 20:12.00 | [TK]D-Fender | eppigy: hukt on fonix werkt 4 u! |
| 20:12.07 | eppigy | 8[] |
| 20:12.19 | eppigy | i am semi-illiterate |
| 20:12.42 | tzafrir_laptop | which also means semi-literate |
| 20:12.49 | eppigy | YES |
| 20:13.18 | uehueh | lol |
| 20:13.54 | Anth8708 | I even hate to ask, but I'm having issues with Alert-Info and distinctive rings with polycom phones: http://pastebin.com/d52c1e9b |
| 20:17.12 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 20:21.53 | Katty | hi dave |
| 20:23.34 | *** join/#asterisk telecos (n=sergio@219.166.219.87.dynamic.jazztel.es) |
| 20:26.30 | *** join/#asterisk phl4kx (n=supervis@webmailserver.nisira.com.pe) |
| 20:26.38 | *** join/#asterisk BreezBl0k (n=BreezBl0@5acd71c7.bb.sky.com) |
| 20:27.28 | BreezBl0k | any one know how to dissable the on hold music in parking lot or have just ring instead? |
| 20:27.58 | [TK]D-Fender | BreezBl0k: change your MoH class to one with nothing to play |
| 20:30.09 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 20:31.15 | [TK]D-Fender | BRB |
| 20:32.09 | keith4 | IAX2 is just tcp/4569, right? |
| 20:32.30 | keith4 | for the record, IAX is working through this crazy-ass multi-NAT setup |
| 20:34.21 | SuPrSluG | udp |
| 20:36.43 | keith4 | ah, thanks |
| 20:37.17 | eppigy | hi Katty |
| 20:40.03 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:40.47 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 20:41.29 | *** join/#asterisk MindTheGap (n=MindTheG@201.80.82.57) |
| 20:43.03 | *** join/#asterisk outtolunc (n=me@c-71-198-197-201.hsd1.ca.comcast.net) |
| 20:56.13 | ks3 | Any thoughts why ReceiveFAX on 1.6 doesn't switch to T38? It uses audio fax mode, which works fine for small faxes, but breaks on larger documents. |
| 20:58.08 | BreezBl0k | <[TK]D-Fender> do you know how i can get it to ring instead then? |
| 21:00.02 | [TK]D-Fender | BreezBl0k: Make a sound file with ringing and use it in an MoH class |
| 21:00.27 | BreezBl0k | <[TK]D-Fender> Thanks |
| 21:10.21 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 21:26.18 | Katty | eppigy: omg. |
| 21:26.23 | Katty | eppigy: kirk just kissed a GIRL |
| 21:26.27 | eppigy | :D |
| 21:26.31 | eppigy | GO KIRK |
| 21:26.33 | Katty | eppigy: can you believe it took until episode 13? |
| 21:26.46 | eppigy | that was suspense back then |
| 21:26.52 | Katty | i guess. |
| 21:27.07 | *** join/#asterisk af_ (n=getsmart@88-149-240-168.dynamic.ngi.it) |
| 21:30.08 | [TK]D-Fender | 3 words : Green Alien Sex |
| 21:31.17 | eppigy | HOT |
| 21:31.43 | Katty | eww. |
| 21:32.07 | eppigy | slithering intimacy |
| 21:32.12 | Katty | EWW |
| 21:32.14 | Katty | dave! |
| 21:32.17 | eppigy | it doesnt get any better |
| 21:32.23 | eppigy | !! |
| 21:34.47 | *** join/#asterisk n00m (n=n00m@c-67-167-200-89.hsd1.il.comcast.net) |
| 21:34.52 | uehueh | fender |
| 21:35.26 | uehueh | transfers not working via ivr either :< |
| 21:35.32 | uehueh | only if i dialout from a SIP phone |
| 21:35.42 | uehueh | -- Executing [#@callthrough:1] Dial("SIP/64.2.142.30-b77006b0", "SIP/18002446227@vitel-outbound|30|tT") in new stack |
| 21:35.47 | uehueh | i have tT also |
| 21:36.03 | *** join/#asterisk malveo (n=malveo@79.143.115.144) |
| 21:36.11 | uehueh | nothing happens with you press # nor #1 |
| 21:36.44 | uehueh | do I need to set a context in features.conf or something? |
| 21:38.19 | [TK]D-Fender | uehueh: TRANSFERCONTEXT channel var |
| 21:38.59 | uehueh | channel var? |
| 21:39.14 | eppigy | FOOD TIME |
| 21:39.31 | eppigy | Katty: i am so tired of fast food in my area :[ |
| 21:39.44 | uehueh | TRANSFER_CONTEXT |
| 21:40.01 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 21:40.19 | *** join/#asterisk ingenius (n=alektro@netsolution.com.ar) |
| 21:40.43 | BreezBl0k | <[TK]D-Fender> where would the moh class be set for parking lot as ive tried parkedmusicclass=none in features_general_additional.conf and that still plays the default moh |
| 21:40.53 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 21:41.10 | [TK]D-Fender | BreezBl0k: the class is based on the channel. |
| 21:41.33 | [TK]D-Fender | BreezBl0k: Whatever device the call comes in on set the MoH chall for the channel. You can change that of course |
| 21:42.38 | BreezBl0k | <[TK]D-Fender> ok thanks, i dont think ill be able do what i want to do |
| 21:42.55 | [TK]D-Fender | BreezBl0k: And why not? |
| 21:44.30 | BreezBl0k | <[TK]D-Fender> i only want no music on hold or ringing when they are in the parking lot but if the calls on hold like normal i would like moh |
| 21:44.32 | Katty | eppigy: stop eating it then |
| 21:44.35 | Katty | eppigy: make yourself something! |
| 21:44.41 | *** join/#asterisk neuro9 (i=neuromat@c-67-167-49-106.hsd1.il.comcast.net) |
| 21:44.56 | neuro9 | hey hey |
| 21:45.06 | eppigy | Katty: D: |
| 21:45.08 | neuro9 | whats MeetMe need dahdi for? |
| 21:45.10 | [TK]D-Fender | BreezBl0k: Make another exten to do your parking which set the MoH class right before you park the call |
| 21:45.16 | [TK]D-Fender | neuro9: Mixing |
| 21:45.24 | dunccfflail | I've managed to get it so i can accept incoming calls, and then dial my IP phone, but when I answer the IP phone, asterisk CLI notices, but it doesn't "answer" the incoming call, it just keeps on ringing |
| 21:45.26 | neuro9 | ah okay |
| 21:45.31 | dunccfflail | anyone know what the problem there is? |
| 21:45.40 | neuro9 | whats the simplist way to install dahdi in gentoo? |
| 21:45.44 | neuro9 | emerging or building from svn |
| 21:45.45 | BreezBl0k | <[TK]D-Fender> cool thanks |
| 21:46.27 | [TK]D-Fender | neuro9: How did you install *? |
| 21:46.42 | neuro9 | TK: from source |
| 21:47.22 | neuro9 | tarball build with some custom patches |
| 21:49.17 | uehueh | -- User hit '*' to disconnect call. |
| 21:49.23 | uehueh | thats not in features |
| 21:50.33 | [TK]D-Fender | neuro9: then install DAHDI the same way |
| 21:50.46 | neuro9 | k |
| 21:50.46 | *** join/#asterisk h3x (n=Hex@64.192.116.17) |
| 21:51.00 | uehueh | im going throw up |
| 21:51.06 | uehueh | this asterisk stuff is completely unexpected.. |
| 21:51.16 | eppigy | haha what |
| 21:51.19 | Katty | ohohoh |
| 21:51.20 | Katty | it's almost5 |
| 21:51.22 | uehueh | i mean how do you do *2 to attended transfer |
| 21:51.25 | eppigy | yesh it is |
| 21:51.26 | Katty | points at clock |
| 21:51.33 | eppigy | crap it is way opast five here |
| 21:51.36 | uehueh | when you press the * it hangs up |
| 21:51.36 | eppigy | what am i doing |
| 21:51.40 | eppigy | WHAT HAVE I DONE |
| 21:51.45 | Katty | not eaten |
| 21:51.48 | Katty | apparently |
| 21:51.57 | eppigy | yes |
| 21:51.57 | Katty | how tragic |
| 21:51.59 | eppigy | :< |
| 21:52.57 | Katty | so |
| 21:53.01 | Katty | i'm gonna go hav edinner now. |
| 21:53.02 | Katty | and stuff. |
| 21:53.34 | Katty | the weekend is shiny and new :> |
| 21:53.38 | Katty | ctrlad |
| 21:53.41 | eppigy | YES |
| 21:53.45 | eppigy | HVAE FUNNLE |
| 21:54.42 | neuro9 | anyone got to play around with 1.6.2 beta yet? |
| 21:59.00 | bmoraca | i imagine someone probably has |
| 22:01.53 | mmlj4 | I'm having a problem passing variables via AGI... it's not consistent as to which variable gets passed to the script, if any get passed at all... ideas? |
| 22:05.20 | *** part/#asterisk a9k (i=a9k@block-66.135.80.2.montanasat.net) |
| 22:05.57 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 22:06.02 | [TK]D-Fender | mmlj4: Show us something useful |
| 22:07.44 | uehueh | [TK]D-Fender: thanks for your help today |
| 22:07.58 | uehueh | I got it to work |
| 22:08.31 | uehueh | and with GOTO_ON_BLINDXFR it transfers without entering the number |
| 22:08.37 | uehueh | :D |
| 22:08.53 | *** join/#asterisk ziram19 (n=chatzill@41.226.54.63) |
| 22:39.20 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 22:40.31 | *** join/#asterisk rue_mohr (n=backdoor@h24-207-90-17.cst.dccnet.com) |
| 22:40.54 | rue_mohr | so my sip phones lack a flash button, ideas how I can fix that? |
| 22:41.46 | rue_mohr | I need to be able to send flash to the zaptel channel I'm connected to |
| 22:42.05 | rue_mohr | I dont need it for anything else |
| 22:42.23 | *** join/#asterisk dthomas (n=darkness@linode.caliginous.net) |
| 22:44.21 | rue_mohr | hello? |
| 22:44.27 | defsdoor | curios |
| 22:44.40 | defsdoor | I just got a call on one of my extensions from sip:0 |
| 22:44.43 | *** join/#asterisk voxter (n=voxter@190.241.16.138) |
| 22:44.57 | defsdoor | logs just show something odd - [May 15 23:37:00] NOTICE[18299] pbx.c: Error in extension logic (missing '}') |
| 22:45.10 | defsdoor | [May 15 23:37:00] ERROR[18299] func_callerid.c: Unknown callerid data type 'nu' |
| 22:45.22 | rue_mohr | you extensions.conf is broken |
| 22:45.53 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:47.15 | *** part/#asterisk dthomas (n=darkness@linode.caliginous.net) |
| 22:47.29 | rue_mohr | you prolly have ( } in it |
| 22:47.43 | defsdoor | I'm thinking it's a cli module I added |
| 22:47.56 | defsdoor | set callerID |
| 22:48.00 | rue_mohr | how do I make it so sip users can flash teh zaptel channel |
| 22:48.31 | rue_mohr | defsdoor if you like, post your extensonds.conf to pastebin and have someone look at it |
| 22:48.40 | rue_mohr | I cant, I'm on a text console |
| 22:48.56 | defsdoor | it's ok - it is this setcallerid thing |
| 22:50.12 | defsdoor | what odd is how it trigger a call |
| 22:51.31 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 22:54.06 | [TK]D-Fender | defsdoor: "CLI module"? pardon? |
| 22:54.21 | defsdoor | [TK]D-Fender: using freepbx here ;) |
| 22:54.25 | [TK]D-Fender | rue_mohr: features.conf + Flash |
| 22:54.41 | [TK]D-Fender | defsdoor: move along then.... |
| 23:06.14 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 23:11.55 | rue_mohr | exit |
| 23:12.00 | rue_mohr | oh, hehe |
| 23:15.02 | *** join/#asterisk shinao1 (n=shinao1@41.219.221.167) |
| 23:22.15 | *** join/#asterisk Braxus (n=braxus@netblock-68-183-230-56.dslextreme.com) |
| 23:26.49 | *** join/#asterisk n00m (n=n00m@c-67-167-200-89.hsd1.il.comcast.net) |
| 23:39.39 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 23:41.07 | *** join/#asterisk propellerhead (n=yogurt2u@host1.190-30-31.telecom.net.ar) |
| 23:43.09 | stope | I've set up SLA and my caller id keeps on showing 'asterisk' even though I set the sip header with the correct cid, any hints? |
| 23:43.30 | *** join/#asterisk trentcreek (n=kvirc@200.94.227.117) |
| 23:51.14 | carrar | w00t! |
| 23:51.27 | carrar | stope, does a set callerid name/number work? |
| 23:52.19 | carrar | Set(CALLERID(number)=8675309) |
| 23:54.21 | stope | no |
| 23:54.30 | stope | still comes across as 'asterisk' |
| 23:57.11 | carrar | so thats caller id name |
| 23:57.14 | carrar | not number |
| 23:57.15 | carrar | ? |
| 23:57.27 | carrar | Set(CALLERID(name)=Jenny) |
| 23:57.36 | Nugget | heh |
| 23:57.51 | mmlj4 | carrar++ |
| 23:58.03 | carrar | heh |
| 23:58.27 | voxter | I hate you, cisco 79x1 series. |
| 23:58.29 | voxter | hate! |
| 23:58.40 | carrar | I love my 79x1 phones |
| 23:58.48 | carrar | using them with Asterisk |
| 23:58.53 | voxter | they are a bitch to get working, especially over the WAN behind nat |
| 23:59.07 | carrar | haha yeah behind nat, good luck |
| 23:59.23 | voxter | is it true that even the most recent versions, you still must specify qualify=never and nat=no ? |
| 23:59.34 | Nugget | you *have* to use SIP-TCP on the newer cisco phones (79x2 and 79x5) if you need to traverse NAT. |
| 23:59.44 | mmlj4 | I used to work in a video game room back in 1982... we had a red rug-lined room... so I used to write "867-5309: ask for Jenny" on the blank wall, in the shag |
| 23:59.45 | carrar | let me look, I have one working with asterisk and 1 with switchvox |
| 23:59.50 | voxter | Ive mangled it enough now that it can accept a call, but cant test audio or outbound dialing (phone is remote) |
| 23:59.51 | Nugget | dunno about the 79x1 phones. do those use the SEPnnnn.xml files? |