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| 02:45.25 | zr0 | is it possible to call the telephone company to give me two "channels" on a pots line? would this essentially cost as much as two pots lines? |
| 02:46.38 | leifmadsen | you can't have multiple lines on the same cable |
| 02:46.50 | leifmadsen | i.e. multiple calls -- it isn't VoIP |
| 02:47.08 | leifmadsen | each pair == 1 simultaneous call |
| 02:48.45 | zr0 | but i want the same number for both "lines" |
| 02:50.53 | jaytee | then you would request mulitple lines with the same DID number, it's usually referred to as a hunt group |
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| 02:54.15 | zr0 | i assume the cost = (cost of one pots line) * (number of lines in hunt group)? |
| 02:54.31 | zr0 | plus some bs fee |
| 02:54.59 | jaytee | pretty much, yeah |
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| 03:35.25 | leifmadsen | zr0: or if you have a decent internet connection, then getting a good SIP provider will prove to be cheaper |
| 03:42.19 | carrar | 00p |
| 03:42.21 | carrar | w00p! |
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| 03:49.59 | zr0 | leifmadsen: i thought i had a good SIP provider, but my DID termination is pretty flaky |
| 03:52.00 | leifmadsen | I like Unlimitel.ca, or bandwidth.com has been pretty decent |
| 03:53.08 | zr0 | i've been using Vitelity |
| 03:54.24 | zr0 | i guess you get what you pay for.. |
| 03:54.32 | leifmadsen | yep |
| 03:54.38 | leifmadsen | most SIP providers suck |
| 03:55.22 | howie | whats the best provider? |
| 03:57.26 | zr0 | leifmadsen: you use unlimitel.ca for DID termination? |
| 03:57.46 | leifmadsen | yeppers |
| 03:57.49 | leifmadsen | I love unlimitel.ca |
| 03:57.52 | leifmadsen | amazing service |
| 03:57.59 | leifmadsen | ~bestquestions |
| 03:58.04 | leifmadsen | ~thebestquestions |
| 03:58.05 | infobot | it has been said that thebestquestions is Whenever you ask a "what is the best..." or "who is the best..." type questions, you're asking for trouble, and possibly may be called a troll. These types of questions do not have answers. Your best bet is to rephrase the question as, "What kind of experience do people have with..." or "Who has experience with...". |
| 03:58.41 | leifmadsen | jbot bestquestions is "see ~thebestquestions" |
| 03:58.51 | zr0 | leifmadsen: seems pretty cheap |
| 03:59.04 | leifmadsen | infobot: bestquestions is "see ~thebestquestions" |
| 03:59.04 | infobot | okay, leifmadsen |
| 03:59.31 | leifmadsen | zr0: yep, very reasonably priced. I have had absolutely zero issues, and amazing call quality with them |
| 03:59.39 | leifmadsen | only have Canadian DIDs though |
| 04:00.00 | zr0 | ah, bumer |
| 04:00.04 | leifmadsen | but that's because they have physical connections in Canada, and don't resell Global Crossing and Level 3 like most providers |
| 04:00.17 | leifmadsen | that's bascially what you're getting -- just resold termination from some other carrier |
| 04:00.27 | leifmadsen | (from most providers, unlimitel is different) |
| 04:00.39 | leifmadsen | however, bandwidth.com seems to have their shit together |
| 04:00.49 | leifmadsen | although they just resell Level 3 like most providers |
| 04:01.10 | leifmadsen | you can buy directly from Level 3, but you need to have a contract for like a million minutes a month or something |
| 04:01.37 | zr0 | heh |
| 04:02.20 | zr0 | i think i'm probably going to stick with copper for termination |
| 04:02.27 | leifmadsen | so ya.... my experience is bandwidth.com for US, unlimitel for incoming Canadian, or all outgoing North America |
| 04:02.47 | leifmadsen | although voiptrunk.com has been pretty decent lately (have a free test account with them) |
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| 04:03.15 | zr0 | is there a simple way to keep track of your DIDs uptime? |
| 04:03.18 | leifmadsen | copper is sometimes safest :) although analog termination is more finicky than SIP termination in my experience |
| 04:03.35 | zr0 | other than calling it? :) |
| 04:03.37 | leifmadsen | zr0: not really... unless you're not getting calls, hard to tell if you're not getting calls :) |
| 04:03.56 | leifmadsen | I can't even think of a complex way of tracking that |
| 04:04.56 | zr0 | i really need the uptime for voicemail |
| 04:05.08 | zr0 | i fwd my cell to my pbx |
| 04:05.26 | leifmadsen | really, a PRI line is the best bet |
| 04:05.40 | leifmadsen | depending how many lines you need |
| 04:05.58 | leifmadsen | anything from 8-10 lines typically justify a partial T1 |
| 04:06.32 | leifmadsen | with analog there is just voltage changes to create the signal, which can sometimes confuse hardware. With digital circuits, you have better reliability |
| 04:28.55 | psykon | i thought on analog there were actually tones of varying freqs that would change and on digital there were 2 voltage levels that changed |
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| 04:46.08 | trymi1 | hi sirs |
| 04:46.18 | trymi1 | pls help me something |
| 04:47.09 | trymi1 | my login to asterisk is delay |
| 04:47.14 | trymi1 | I can login |
| 04:47.23 | trymi1 | but login process is very long time |
| 04:47.27 | trymi1 | why? |
| 04:47.34 | trymi1 | can you fix for me? |
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| 05:04.28 | psykon | i before i start googlin this: Anyone know of a nice howto compile asterisk from scratch guide? I wanna stay away from trixbox and pbiaf. |
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| 05:19.14 | [TK]D-Fender | psykon: instructions in the tarball |
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| 05:43.12 | leifmadsen | [TK]D-Fender: weirdo |
| 05:46.05 | carrar | mmmm tarballs |
| 05:46.15 | carrar | covered in BBQ sauce |
| 05:47.44 | leifmadsen | mmmm |
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| 05:54.50 | coppice | sounds like something evil to put with noodles |
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| 06:01.07 | [TK]D-Fender | checkout time, later all |
| 06:02.07 | carrar | THERE IS NO CHECKING OUT |
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| 08:20.09 | zebedeee | Hi all. Has anyone had any success compiling asterisk with the ilbc codec using the script in the contrib folder of the asterisk source? I can pull the source in, but am having major problems from there on. |
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| 09:39.49 | stix | Which action from the AMI should I use to transfer a call? |
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| 11:30.26 | Orbixx | Can Asterisk be used as a flexible IVR? |
| 11:30.41 | Orbixx | I'm looking for IVR, with a sort of "out of office hours, leave a message" thing, too. |
| 11:31.34 | Orbixx | Not too fussed about voice recognition, just keypad tone recognition |
| 11:40.09 | sfire | Orbixx, yes.. you can setup time conditions |
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| 12:11.50 | Orbixx | sfire: Awesome, just curious before I actually dive into learning Asterisk. |
| 12:13.16 | sfire | see asterisknow |
| 12:14.03 | Orbixx | Thanks. |
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| 12:15.32 | Orbixx | Wow, that looks pretty awesome. |
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| 12:56.20 | Daviey | ls |
| 13:01.40 | Pan3D | drwxr-x---+ 5 Daviey Group 170 Apr 28 22:04 File |
| 13:02.00 | Daviey | Pan3D: :) |
| 13:02.03 | Pan3D | ;) |
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| 13:29.03 | stix | Which action from the AMI should I use to transfer a call? |
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| 14:17.52 | a1_ | Anyone know how to modify LIDB data? I heard i have to use SS7 |
| 14:29.07 | a1_ | How do i change the TNID? |
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| 14:37.42 | rue_mohr | so, can a tdm800 do fxo message waiting? right now its detecting the message waiting as a ring signal, its driving the receptionist crazy |
| 14:42.10 | rue_mohr | I think ringdebounce takes effect after a hangup, a delay before it will accept ringing again |
| 14:42.15 | rue_mohr | yesno? |
| 14:43.06 | [TK]D-Fender | dlyrueNo. |
| 14:43.09 | [TK]D-Fender | rue_mohr: No. |
| 14:43.43 | rue_mohr | k... can ringdebouce make it distinguish the MWI signalling fromringing? |
| 14:44.04 | [TK]D-Fender | rue_mohr: Unsure |
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| 14:45.10 | dwery | Why, with mwimonitor=no I get "chan_dahdi.c:7669 ss_thread: MWI: Channel 1 message waiting!" ? |
| 14:45.56 | rue_mohr | where is mwimonitor? dahdi.cfg? |
| 14:46.27 | dwery | rue_mohr: in chan_dahdi.conf. it says the default values is =no and I haven't changed it |
| 14:46.27 | rue_mohr | er, duh, chan_dahdi |
| 14:46.46 | rue_mohr | hmm |
| 14:47.08 | rue_mohr | mwi works by placing a steady ring voltage on the line |
| 14:47.09 | dwery | I'm interfacing * with an LG LDK pbx which is supposes to send the caller id as FSK, maybe its misdetected as MWI? |
| 14:47.33 | dwery | mm then not. |
| 14:47.39 | rue_mohr | if you take a multimeter and measure your line voltage it might tell, 48 to 50V is normal, 90V is mwi |
| 14:48.01 | dwery | but with mwimonitor=no shouldn't the detection be disabled? |
| 14:48.12 | rue_mohr | good question |
| 14:48.18 | rue_mohr | next question? |
| 14:48.32 | dwery | how can I debug the caller id? :D |
| 14:48.52 | rue_mohr | you can do a NoOp with the caller id variable in there |
| 14:49.12 | dwery | I need mor low level debugging, I'm not sure that f*** pbx is actually generating it |
| 14:49.22 | rue_mohr | there are all sorts of articles from google about cid not working properly in prescence of mwi |
| 14:49.26 | dwery | hates LG pbxs |
| 14:49.50 | rue_mohr | deal with your mwi and your cid might be fixed |
| 14:50.01 | dwery | I don't think the pbx has MWI enabled |
| 14:52.34 | dwery | I checked, seems disabled |
| 14:55.03 | leifmadsen | hmmm darn -- looks like I can't use this broadcom 4503 USB adapter with CentOS 5.3 |
| 14:55.07 | leifmadsen | (not a new enough kernel) |
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| 15:11.13 | stope | ast_rtp_senddigit_begin: Don't know how to represent 'f' shows up on the console then I get one way audio... * 1.4.25.1, wanpipe 3.3.15.20, zaptel 1.4.12.1, libpri 1.4.7.... any suggestions? |
| 15:11.41 | stope | and it happens at random |
| 15:12.43 | leifmadsen | I think someone just mentioned that on the mailing list the other day.... |
| 15:13.04 | leifmadsen | and I forget what they changed to fix it |
| 15:13.25 | leifmadsen | oh right! you have to turn off DTMF detection in wanpipe |
| 15:13.31 | stope | I upgraded from 1.4.11 and upgraded the kernel..... |
| 15:13.34 | leifmadsen | it is falsely picking up a tone and thinking it is a fax |
| 15:13.42 | leifmadsen | it is a wanpipe issue |
| 15:13.43 | leifmadsen | not asterisk |
| 15:13.47 | stope | nice! I will turn it off |
| 15:13.58 | stope | tx leifmadsen!! |
| 15:14.03 | leifmadsen | np |
| 15:14.12 | leifmadsen | stope: let me know for sure if that fixes it |
| 15:14.38 | stope | one more ques... I will... about 1.6, I like the features in it, what version do you recommend for production? |
| 15:14.53 | [TK]D-Fender | stope: latest 1.6.0 |
| 15:15.04 | [TK]D-Fender | ~asteriskversioning |
| 15:15.05 | infobot | i heard asteriskversioning is Information about the new Asterisk versioning method with the 1.6.x series is available here: http://www.asterisk.org/node/48602 |
| 15:15.22 | stope | I would consider upgrading this same box today..... and going down the 1.6 path |
| 15:15.30 | stope | k, tx |
| 15:16.36 | leifmadsen | one of these days I'll upgrade my home box from 1.4 to trunk :) |
| 15:16.49 | leifmadsen | it is my production box right now, and I don't have the cycles today to make sure it goes smooth |
| 15:16.58 | leifmadsen | although I don't think I'm really using that many features.... it may "just work" |
| 15:17.18 | leifmadsen | but I think I'd like to go through my configuration and do a proper conversion, so it'll probably take an hour or so just to rebuilt the configs |
| 15:17.22 | leifmadsen | rebuild* |
| 15:17.26 | stope | I'm guessing that in wanpipe1.conf the line to change to no is: TDMV_HW_DTMF = YES |
| 15:19.00 | *** join/#asterisk UQlev (n=yuriy@87.228.199.125) |
| 15:19.01 | leifmadsen | ummm.... try it, and let me know. You may have to recompile wanpipe because there is an option during the setup to enable DTMF detection |
| 15:19.15 | leifmadsen | however I'm not sure if you can just disable it with the configuration -- I'd like to know for sure |
| 15:19.35 | stope | ok, I'll just do that, I'd rather recompile rather than fiddle with stuff |
| 15:19.46 | leifmadsen | I don't run wanpipe, but I know when I compiled it for a client there was a DTMF detection question in the wancfg setup |
| 15:20.13 | leifmadsen | coolio -- keep me posted! (I'd like to try and give as much info as possible on my blog about it since I just made a quick little post about it) |
| 15:20.22 | stope | is there an alternative to wanpipe when using a sangoma card? |
| 15:20.39 | leifmadsen | nope |
| 15:20.50 | leifmadsen | (my comment stemmed from the fact I don't use sangoma hardware) |
| 15:21.03 | stope | kinda figured :) |
| 15:22.39 | stope | when's the next taug event? website always shows the old stuff... |
| 15:23.00 | leifmadsen | really? thought simon updated it pretty regularily |
| 15:23.14 | leifmadsen | oh right! |
| 15:23.18 | leifmadsen | we can't have a meeting until the strike is over |
| 15:23.48 | stope | I've never gone but the topics are ones I would not want to miss out on |
| 15:23.57 | stope | alternate meeting place? |
| 15:24.14 | leifmadsen | not that I'm aware of |
| 15:24.23 | leifmadsen | the mailing list is probably the best place to keep abreast of meetings |
| 15:25.04 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.119.63) |
| 15:25.09 | DelphiWorld | hello |
| 15:25.58 | leifmadsen | howdy |
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| 16:04.49 | stope | the telco sends calls initially to server A and will automatically fail over to server B, is there something in the dialplan I can put so that the call automatically gets re-routed from the telco to server B while still keeping wanpipe and * up on server A? I have to do some testing on server B and want to run on the backup for a while.... |
| 16:04.52 | stope | I want all phone numbers to be answered by server B except for the one that I am testing with on server A |
| 16:07.12 | leifmadsen | stope: try Congestion() |
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| 16:40.27 | pif | hi, is there a graphical tool to analyse CDR data? |
| 16:40.41 | pif | to produce reports |
| 16:43.04 | leifmadsen | pif: asterisk-stat potentially |
| 16:43.25 | pif | ok |
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| 17:22.47 | dshap | can someone help me figure something out? at a specific, known point in one of my extensions, I need to detect whether there is any audio on the channel and if it lasts for 3 or more seconds |
| 17:23.14 | dshap | i've used BackgroundDetect() before but only in situations where the audio turns to silence at some point |
| 17:23.29 | dshap | in this case, the audio could be indefinite which means BackgroundDetect() won't trigger |
| 17:23.31 | dshap | any ideas? |
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| 20:47.41 | hesco | What does this mean, in response to a dialplan reload: 'pbx.c:2055 pbx_find_extension: Maximum PBX stack exceeded', and what can I do about it? |
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| 21:07.24 | MrTelephone | is there any talk about support for forwarding registration requests to a failover asterisk server? |
| 21:10.04 | kmem | the DLink VTA is a real piece of crap |
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| 21:31.17 | karleeto | can i get some opinions of what would be a "working" (doesnt have to be a super computer linux mahcine), but just what kind of resources would my server ideally need to be running to run a (just for learning) hosted asterisk system off of it? |
| 21:32.09 | karleeto | right now i have a virtual machine running linux (gentoo) |
| 21:32.09 | karleeto | and let me see what hw i have |
| 21:32.09 | karleeto | Linux server 2.6.25-gentoo-r7 #1 SMP Wed Sep 3 20:20:39 PDT 2008 i686 Intel(R) Pentium(R) D CPU 2.80GHz GenuineIntel GNU/Linux |
| 21:32.15 | Dovid | i think u can use a basic p4 for playing around with |
| 21:32.19 | Dovid | 500 megs will do |
| 21:32.43 | karleeto | i think i have like 512 ram and plently of hd space |
| 21:33.21 | mbrevda | waves: ous geshlufen? |
| 21:33.29 | karleeto | pluse since its a virtual machine, i would only be getting 1 10th of that processor power |
| 21:33.34 | Dovid | nein |
| 21:33.56 | Dovid | waves back: Zeh sof hashavuah |
| 21:34.26 | karleeto | Dovid: worse case cenario, it COULD be that low, basically. So 10 virts are running on that pentium D 2.8 |
| 21:34.42 | karleeto | Dovid: and i am garaunteed the ram |
| 21:35.36 | Dovid | i used to test on VM's easy to format and install |
| 21:35.40 | Dovid | back when i was with 1.x |
| 21:35.53 | Dovid | i used provps.com |
| 21:37.10 | karleeto | i'm using: |
| 21:37.13 | karleeto | http://www.hostvirtual.com/vps-hosting-plans/plans-and-pricing/ |
| 21:37.55 | karleeto | i have the xv1, do think its would run a hosted asterisk pbx on there for me to play around with? |
| 21:38.11 | Dovid | hmm |
| 21:38.14 | Dovid | url isnt loading for me |
| 21:38.20 | karleeto | and maybe prepare an image from that virtualhost to employ many of them later |
| 21:38.27 | Dovid | provps has images |
| 21:38.59 | karleeto | so does these guys, anyway i just already have this server from them |
| 21:39.04 | karleeto | i've had it for a while |
| 21:39.10 | Dovid | if they have images so u can set up a new site real fast |
| 21:39.28 | karleeto | they do, they even had a gentoo image |
| 21:39.29 | Dovid | then u can create an install script so if u scre up u can start frech < 30 minutes |
| 21:39.54 | karleeto | and see, i can modify it instal a hosted asterisk box based off of gentoo, and save and image of that |
| 21:39.59 | karleeto | thats what my plan here is |
| 21:40.37 | karleeto | my question was though, hehe, do i need to upgrad to one of the higher accounts, or should mine do just fine? |
| 21:40.41 | karleeto | http://www.hostvirtual.com/vps-hosting-plans/plans-and-pricing/ |
| 21:40.47 | karleeto | try again |
| 21:41.01 | Dovid | just did |
| 21:41.48 | karleeto | still nothings? |
| 21:41.56 | karleeto | well here is the cpu: |
| 21:42.00 | Dovid | ah ok |
| 21:42.02 | Dovid | they have bad dbs |
| 21:42.04 | Dovid | dns* |
| 21:42.07 | Dovid | with www it wont work |
| 21:42.18 | karleeto | oh, did it work wo it? |
| 21:42.22 | karleeto | can you see it now? |
| 21:42.30 | Dovid | yea |
| 21:42.52 | Dovid | for basic basic vx1 would work |
| 21:42.55 | Dovid | i would get more ram |
| 21:43.25 | Dovid | i hear people r using amazon |
| 21:43.31 | Dovid | u pay for the time that machine is on |
| 21:43.37 | Dovid | i think 10 cents an hour |
| 21:43.38 | Dovid | http://aws.amazon.com/ec2/ |
| 21:44.38 | karleeto | ok, so i can use my vx1 that i already have just to make into a perfet vm image, and then i can use that image as the base for my voip packages i'm about to start seling |
| 21:44.42 | WindowsUser | 744 hours in a month tho |
| 21:45.01 | Dovid | well ur not working all the time on testing r u ? ;) |
| 21:45.16 | *** part/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 21:45.24 | Dovid | 40 hours a week * 4 weeks in a month is $16.00 |
| 21:45.25 | karleeto | Dovid: i'm going to start doing this in spare time, getting things setup to start a busines |
| 21:45.34 | Dovid | for testing its worth it |
| 21:45.36 | WindowsUser | oh i wasn't following the whole conversation, i thought you were suggesting production on an ec2 |
| 21:45.44 | Dovid | nah. testing |
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| 21:45.53 | Dovid | with production i wouldnt use a VM |
| 21:46.13 | Dovid | u want a ded hosting in a good dc that caters to voice |
| 21:46.17 | karleeto | Dovid: then i am going to get an entire dedicated server for production setups |
| 21:46.26 | Dovid | for production |
| 21:46.34 | Dovid | but for testing u can use amazon |
| 21:46.52 | karleeto | or the one i already have at hostvirtual |
| 21:46.54 | karleeto | lol |
| 21:47.31 | Dovid | ah ok |
| 21:47.48 | Dovid | good thing about amazon is u can expand real easy |
| 21:48.03 | Dovid | so i hear. i never used it. other people i know use it and love it |
| 21:49.53 | MrTelephone | virtual machines are terrible |
| 21:50.29 | MrTelephone | nothing beats an actual server for an actual job |
| 21:50.51 | MrTelephone | what are people doing for failover I wonder |
| 21:52.31 | Dovid | 2 vm's on the same serer ;) |
| 21:52.36 | Dovid | i use vm's for testing |
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| 22:13.04 | karleeto | Dovid: exactly, its an invaluable resource! epcially since you can get something how you like it, then save an image and deploy it our on tons of mahcines |
| 22:29.37 | toughmarketing | Hey guys, I have a question. I am trying to use phpagi and I have a script that i am trying to make work.. Basically script.php?call=1xxxxxxxxxx&from=1xxxxxxxxxx&text=hello world, I am confused on how to make swift read hello world after connecting the call. |
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| 22:39.36 | [TK]D-Fender | toughmarketing: What does AGI have to do with connecting a call? |
| 22:40.41 | toughmarketing | [TK]D-Fender: Basically I have a checker in php on my site and if something goes wrong I want to make a phone call come in and read a message to me telling me what is wrong. |
| 22:41.30 | [TK]D-Fender | toughmarketing: No need for AGI for this.... just have it issue a "call file" or AMI Originate. |
| 22:41.32 | dshap | gotta use AMI |
| 22:41.39 | dshap | o ya |
| 22:41.41 | dshap | call file is the eziest |
| 22:41.51 | [TK]D-Fender | dshap: Or CLI originate for that matter |
| 22:41.51 | toughmarketing | if i issue a call file how could i make swift play my message ? |
| 22:42.13 | [TK]D-Fender | toughmarketing: You call swift the same way you would anywhere else... this is DIALPLAN. |
| 22:42.15 | dshap | [TK]D-Fender: u mean just like calling a system command w/PHP to have asterisk use Originate? |
| 22:42.41 | [TK]D-Fender | dshap: Sure : asterisk -rx "originate blah...." |
| 22:42.46 | dshap | oh ya |
| 22:43.02 | toughmarketing | okay im going to play with it |
| 22:43.03 | toughmarketing | thanks |
| 22:43.59 | dshap | [TK]D-Fender: i dunno if u remember, but earlier i was asking about how to detect audio so i could figure out when my SendDTMF failed to interact with an IVR...i got it working now, but it's ridiculous. it fails like often 3 times in a row |
| 22:44.12 | dshap | [TK]D-Fender: have you had similar unreliability with SendDTMF? |
| 22:44.46 | [TK]D-Fender | dshap: Your goal will be sujected to the random shit nature of X # of VM messages an BS. Verbal cadence with screw you. |
| 22:44.50 | [TK]D-Fender | dshap: WITHOUT the K-Y |
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| 22:45.40 | dshap | [TK]D-Fender: what do you mean K-Y? what im talkin about now has nothing to do with # of VM messages...im talking about SendDTMF failing to get its message across to the called party |
| 22:45.49 | [TK]D-Fender | dshap: Noble goal, well attempted but largely destined to fail because the system can only be so smart. That is no competition for the vastness of human stupidity. |
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| 22:46.19 | dshap | [TK]D-Fender: clearly i am not destined to fail, as I have already seen the existence of a "flawless" system |
| 22:46.20 | [TK]D-Fender | dshap: pastebin where you're at now.. |
| 22:46.28 | dshap | [TK]D-Fender: ok |
| 22:46.40 | *** join/#asterisk karleeto (n=karl@server.nashvilleproweb.com) |
| 22:49.00 | dshap | ok here |
| 22:49.02 | dshap | check this out |
| 22:49.04 | dshap | it's very simple really |
| 22:49.08 | dshap | [TK]D-Fender: http://pastebin.com/m6bd8b22d |
| 22:50.18 | dshap | i have a call file that calls this number: 2537094040 (AT&T voicemail system) |
| 22:50.22 | dshap | and links it to that extension |
| 22:50.31 | dshap | the SendDTMF() sometimes gets across, and sometimes it doesn't |
| 22:50.43 | dshap | it's totally unreliable |
| 22:50.56 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 22:51.04 | dshap | and i have not come up with any way to explain what causes it to fail or succeeed |
| 22:52.17 | dshap | it's driving me crazy |
| 22:56.06 | [TK]D-Fender | dshap: wHAT ARE YOU CALLING IN ON? |
| 22:56.46 | dshap | [TK]D-Fender: it's a "call file" that places an outgoing call from * to AT&T VM System |
| 22:56.56 | [TK]D-Fender | dshap: wHAT ARE YOU CALLING IN ON? <--- |
| 22:57.26 | dshap | [TK]D-Fender: i don't understand your question |
| 22:57.32 | [TK]D-Fender | dshap: ........ |
| 22:57.33 | Pan3D | :) |
| 22:57.33 | dshap | [TK]D-Fender: it's a SIP channel |
| 22:57.40 | dshap | .... |
| 22:57.41 | [TK]D-Fender | dshap: What are are you using to CALL THEM? |
| 22:58.10 | dshap | [TK]D-Fender: I am using my Asterisk server to place a call via Voip to my SIP provider to the PSTN |
| 22:58.14 | dshap | i don't know how to be any more clear than that |
| 22:58.32 | dshap | my asterisk server is just playing an audio file that it has stored locally |
| 22:58.33 | [TK]D-Fender | dshap: Holy shit. what fucking technology?> What service? Codec? |
| 22:58.45 | dshap | voip.ms SIP |
| 22:58.47 | dshap | not sure of the codec |
| 22:58.51 | dshap | pretty much all are supported |
| 22:59.03 | dshap | how would i find out what codec is being used? |
| 22:59.09 | [TK]D-Fender | dshap: Maybe when you know what you're doing you might know what might fail about it |
| 22:59.18 | [TK]D-Fender | dshap:YOU CHOOSE the codec. |
| 22:59.30 | [TK]D-Fender | dshap: So what are you doing? |
| 23:00.02 | dshap | [TK]D-Fender: well i've got allow=ulaw, allow=g729 in my sip.conf |
| 23:00.09 | dshap | [TK]D-Fender: does that help? |
| 23:00.31 | [TK]D-Fender | dshap: Pick one codec & verify it. Also verify the DTMF mode |
| 23:00.47 | dshap | [TK]D-Fender: dtmf mode is rfc2833 |
| 23:01.04 | dshap | so i should only have a single "allow" statement in my entire sip.conf? |
| 23:01.16 | [TK]D-Fender | dshap: For the PEER you are calling out of... |
| 23:01.27 | dshap | [TK]D-Fender: okay what about under general? |
| 23:01.31 | [TK]D-Fender | dshap: Stop treating your system like it can only do 1 thing for everything |
| 23:01.41 | dshap | ok i will, sorry |
| 23:01.46 | dshap | what about under [general] |
| 23:01.48 | [TK]D-Fender | dshap: What does [general] have to do with the one you use in your PEER? |
| 23:01.51 | dshap | should that have any codec specifications? |
| 23:02.06 | [TK]D-Fender | dshap: [general] is what gets used when you DON'T set codecs per peer |
| 23:02.54 | dshap | [TK]D-Fender: got it...here's what i have right now http://pastebin.com/m50e40656 |
| 23:03.34 | dshap | [TK]D-Fender: looks like i am only using ulaw for the [voipms] peer |
| 23:04.04 | [TK]D-Fender | dshap: Ok, well DTMF timing is a non-issue. RFC2833 says its passed OOB which means it'd be voip.ms' fault if it was unreliable |
| 23:04.23 | dshap | what does 00B mean? |
| 23:04.31 | [TK]D-Fender | dshap: Out Of Band |
| 23:04.39 | [TK]D-Fender | dshap: Not sent as audio |
| 23:04.50 | dshap | the timeout & duration parameters don't have an affect if it's not sent inband? |
| 23:05.06 | dshap | i feel like they've slightly changed the performance |
| 23:05.09 | dshap | but maybe it was just coincidence |
| 23:05.38 | [TK]D-Fender | dshap: Placebo effect for having lerned something new (yet not applicable) |
| 23:06.09 | dshap | uhh |
| 23:06.15 | dshap | so ur saying definitely it was a coincidence? |
| 23:06.16 | dshap | lol |
| 23:12.46 | dshap | [TK]D-Fender: when you said it would be voip.ms' fault, that got me thinking |
| 23:12.49 | dshap | and now i'm trying it with flowroute |
| 23:12.52 | dshap | and it's working every time |
| 23:12.55 | dshap | *knocks on wood* |
| 23:13.00 | dshap | that's crazy |
| 23:13.36 | [TK]D-Fender | dshap: yes.. crazy..... |
| 23:14.06 | dshap | [TK]D-Fender: flowroute works every time, voip.ms fails 75% of the time |
| 23:14.40 | dshap | [TK]D-Fender: that's outrageous...think if I tell them that they can fix it? |
| 23:15.56 | [TK]D-Fender | dshap: Feel free to try to make a difference. I suggest the "Death by a thousand pin-pricks" approach |
| 23:16.01 | *** join/#asterisk Selveste1_ (n=Selveste@88.83.86.20) |
| 23:16.15 | dshap | [TK]D-Fender: more like "death by losing a customer" |
| 23:16.25 | dshap | [TK]D-Fender: i can't use my service with voip.ms if they can't reliably pass DTMF |
| 23:18.07 | [TK]D-Fender | dshap: So if they lose you as a customer then go on to fix it.... they fixed someone ELSE's problem, not yours :) |
| 23:23.11 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 23:29.33 | *** join/#asterisk brut- (n=brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 23:50.16 | *** join/#asterisk P0RN5T4R (n=leet@213.63.210.46) |
| 23:50.19 | P0RN5T4R | hello |
| 23:51.04 | P0RN5T4R | someone online? i have a few questions :P |
| 23:51.32 | dshap | i'm online |
| 23:51.41 | dshap | although i'm not as qualified in answering questions as other people are |
| 23:51.44 | dshap | i know more than nothing |
| 23:52.23 | P0RN5T4R | i've been talking with a frien that as s VoiP company |
| 23:52.31 | P0RN5T4R | and i got interested in this matter |
| 23:52.50 | P0RN5T4R | he talked me about Asterisk |
| 23:53.40 | P0RN5T4R | can u gve me a heads-up on what do i need to do if iwanted to setup a voip server (dunno if this is how you call it) with asterisk |
| 23:53.57 | dshap | hah |
| 23:53.59 | dshap | yea |
| 23:54.01 | P0RN5T4R | need a machine and a link obviously |
| 23:54.03 | dshap | i was in your shoes literally 2 months ago |
| 23:54.10 | jaytee | ~book |
| 23:54.11 | 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 |
| 23:54.12 | dshap | and now i have a fully functioning asterisk PBX |
| 23:54.18 | dshap | ya the book helped me a lot |
| 23:54.22 | dshap | and the ppl in this channel as well |
| 23:54.33 | jaytee | and check out the wiki at www.voip-info.org |
| 23:54.34 | dshap | do u have a linux box? |
| 23:55.35 | P0RN5T4R | now now but i now i need it |
| 23:55.44 | dshap | yea |
| 23:55.44 | P0RN5T4R | but lets assume i do have a linux machine |
| 23:55.45 | [TK]D-Fender | P0RN5T4R: You install *. Go read that book to understand what * is about |
| 23:56.11 | dshap | if u have a linux machine, basically u install asterisk on it and then figure out how ur gonna connect to the phone network |
| 23:57.09 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 23:57.21 | jaytee | "If Bob has 3 oranges and Larry has 2 apples, how long would it take Phil to walk to Cleveland? Choose your answer: a) 3 hours b) eggs c) the question makes no sense." |
| 23:58.19 | [TK]D-Fender | jaytee: An apple a day is woth 2 in the bush! </mixedmetaphor> |
| 23:58.33 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 23:59.02 | jaytee | [TK]D-Fender passes and graduates Summa Cum Laude |