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| 00:48.40 | R | booo |
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| 01:14.39 | dieno | can any 1 tell me how to install iLBC Codecs |
| 01:14.40 | dieno | ? |
| 01:15.45 | *** join/#asterisk MarkMoss (n=mmoss@65.183.175.226) |
| 01:17.09 | bkw_ | they install by default |
| 01:19.16 | dieno | ok but asterisk is not using it |
| 01:19.26 | dieno | i allow it |
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| 02:45.26 | _DAW | does the switch statement only work with IAX channels? |
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| 03:12.45 | Un1x | hmm anyone try that FX-300 GSM Call Director? |
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| 03:34.23 | *** join/#asterisk duderock (n=natrix@CPE-72-131-35-138.wi.res.rr.com) |
| 03:36.03 | duderock | Need some help, not sure if it's an asterisk problem. I'm running asterisk on Ubuntu Server on an AMD X2. Tried both SMP & non-SMP kernels and the system keeps hanging. If anyone has time to check out my logfile - most recent system failure was 17:17 - http://natrix.freeshell.org/syslogsept01.txt |
| 03:39.49 | CunningPike | duderock: Took a look - doesn't really tell me anything |
| 03:40.02 | duderock | yeah, i didn't see anything either.. |
| 03:40.07 | CunningPike | duderock: What hardware do you have in the machine? |
| 03:40.59 | duderock | Pike: AMD X2, Gigabyte 551SLI MOBO, 800mhz DDR2, 2x SATA (in software raid) |
| 03:41.25 | benjk | looks like php or db related to me |
| 03:42.11 | duderock | any clues from this: |
| 03:42.42 | duderock | Sep 1 20:14:09 talk kernel: Symbols match kernel version 2.6.15. |
| 03:42.42 | duderock | Sep 1 20:14:09 talk kernel: No module symbols loaded - kernel modules not enabled.Sep 1 20:14:09 talk kernel: Symbols match kernel version 2.6.15. |
| 03:42.47 | findlay | how would I strip the leading '1' US country code from a variable in extensions.conf? |
| 03:43.02 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 03:43.08 | benjk | ${VARIABLE:1} |
| 03:43.30 | findlay | benjk: documentation? |
| 03:43.33 | benjk | if it's always 1 digit |
| 03:44.32 | benjk | if you want to strip *any* country code, you will need to have a database that tells you how many digits the country code is |
| 03:44.44 | findlay | ok |
| 03:45.00 | benjk | then do ${VARIABLE:${NUM_OF_DIGITS}} |
| 03:45.41 | benjk | I have a country code database for 228 countries and territories |
| 03:46.33 | benjk | http://www.sunrise-tel.com/STSDialingCodeDBAndLibrary.html |
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| 03:50.52 | Ryushin | Is there any particular reason why it takes almost an hour for the polycom ip430 phone to finish booting? |
| 03:52.53 | benjk | slow network? |
| 03:53.12 | benjk | broken phone? |
| 03:53.30 | Ryushin | Not sure. All the ip430's are taking this long. |
| 03:53.37 | coppice | gigabytes of code, because they write with the efficiency of the MS office developers? :-) |
| 03:53.41 | Ryushin | Network is fast, so that is not it. |
| 03:53.57 | benjk | if you say so |
| 03:54.07 | findlay | could be a/several timeout(s) of some kind? |
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| 03:55.14 | benjk | have you set up a separate network with a tftp server and a single phone, direct ethernet link? |
| 03:55.21 | Ryushin | I'm seeing this in the boot log of the phone: CryptoConfigFileEncrypted: Could not read file /ffs0/local/0004f2060f51-phone_cfg |
| 03:55.53 | Ryushin | And that file and directory is not there. |
| 03:56.16 | benjk | so how come it boots after an hour? |
| 03:56.20 | duderock | benjk: what would you recommend as a good starting point to diagnose my problem? The system just locks up without warning.. |
| 03:56.49 | benjk | duderock, build a vanilla system without php and without database |
| 03:57.15 | duderock | benjk: to what end? |
| 03:57.23 | Ryushin | benjk: I guess because it gives up. How long is it supposed to take a ip430 to boot? |
| 03:57.24 | benjk | then gradually add the bits and pieces to see if and when the problem kicks in |
| 03:57.38 | duderock | benjk: okay then... |
| 03:58.01 | benjk | 430 hourse |
| 03:58.04 | benjk | hours |
| 03:58.05 | Ryushin | Would that the a local file to the phone itself? |
| 03:58.40 | benjk | I presume the phone is trying to dl the file from a tftp server |
| 03:59.04 | Ryushin | I'm also seeing this in the boot log: Could not get all 36 bytes of the header |
| 03:59.08 | Ryushin | I'm using ftp. |
| 03:59.10 | *** join/#asterisk flujan (n=flujan@201-27-89-159.dsl.telesp.net.br) |
| 03:59.20 | benjk | well, simplify your setup |
| 03:59.39 | benjk | build a network with 1 phone, 1 ethernet cable, 1 tftp server |
| 03:59.41 | benjk | see what that does |
| 04:00.13 | Ryushin | I'll change my phone1.cfg file back and see what that does. Thanks though. |
| 04:00.19 | benjk | first you may want to check if the configuration is ok, tftp server, files etc |
| 04:01.43 | duderock | benjk: so you think it's a problem with MYSQL or PHP4? The MYSQL errors were due to a hard reset.... |
| 04:01.59 | findlay | broadvoice makes me mad |
| 04:02.03 | benjk | it could be anything, hardware for example |
| 04:02.23 | benjk | scan all the log files you have |
| 04:02.44 | duderock | that's what I've been up to for the last 20 minutes or so... |
| 04:03.09 | duderock | Nothing like reading log files on a Friday night. ;) |
| 04:03.22 | duderock | Sure beats what I was doing 2 hours ago... (moving!) |
| 04:03.29 | benjk | you need to gradually exclude possible sources then |
| 04:03.36 | benjk | like building another system |
| 04:03.47 | benjk | removing components gradually |
| 04:03.59 | benjk | or starting with minimal setup and adding components gradually |
| 04:04.12 | flujan | hi guys... aare Sangoma cards good to use with asterisk? |
| 04:04.16 | duderock | This is my first build, I haven't installed anthing but the basics yet |
| 04:04.18 | benjk | of course |
| 04:04.25 | Ryushin | Okay, I'll start that. Thanks. |
| 04:04.30 | Ryushin | Time to head home and go to bed. |
| 04:05.20 | flujan | Dothe boards have it own processor for voice conversion or it uses the host CPU as Digium boards? |
| 04:05.21 | benjk | flujan, during US night time: Sangoma rocks |
| 04:05.32 | benjk | during US daytime: Digium rocks |
| 04:06.01 | flujan | benjk, sorry i dont got the idea... Sangoma is just good for using during the nights???? :D |
| 04:06.04 | benjk | as far as #asterisk is concerned |
| 04:06.27 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 04:06.29 | benjk | no that's the answers you will generally get when you ask this question in this channel |
| 04:06.57 | benjk | in other words, it can be a bit of a religious thing |
| 04:07.11 | benjk | both Digium and Sangoma cards work with Asterisk |
| 04:07.28 | benjk | some people like Digium and nothing but Digium |
| 04:07.38 | benjk | some other people like Sangoma and nothing but Sangoma |
| 04:08.16 | benjk | outside of the US many people prefer Sangoma because Sangoma care more about other countries |
| 04:08.23 | Un1x | Bah, i'm bored i cant debate on wich O/S to install FreeBSD or Slackware... |
| 04:08.32 | EyeCue | FreeBSD. |
| 04:08.37 | benjk | BSD of course |
| 04:08.49 | Un1x | this is for a Laptop people,. not server :) |
| 04:08.50 | EyeCue | no bias, i swear :D *parts #freebsd* |
| 04:08.56 | EyeCue | So? :) |
| 04:09.12 | benjk | even Linus said he had never started Linux if he had known about BSD386 at the time |
| 04:09.18 | Un1x | i dont know just dont like FreeBSD as desktop |
| 04:09.20 | Un1x | Haha |
| 04:09.44 | Un1x | Well if someoen can go back in time then show him BSD386 |
| 04:09.44 | Un1x | and we will se a end to Linux :) |
| 04:09.44 | benjk | hehe |
| 04:09.45 | EyeCue | linux is like a bratty little sister |
| 04:09.49 | EyeCue | thats all |
| 04:09.49 | EyeCue | :] |
| 04:10.12 | benjk | BSD has always been the secret real Unix |
| 04:10.14 | Un1x | haha yea. |
| 04:10.28 | *** join/#asterisk fugitivo (n=ajf@190.48.165.130) |
| 04:10.30 | fugitivo | hi |
| 04:10.34 | flujan | benjk, humm... but sangoma cards have it own processors right? ... or does it still use the host processor? |
| 04:10.36 | Un1x | benjk: i love FreeBSD it was my First ever *nix os i was on and ive always loved it |
| 04:10.44 | benjk | if it wasn't for BSD most of the Unixy things in Unix/Linux wouldnt be there |
| 04:10.47 | Un1x | i dont think i'll ever stop using it just debating on whether or not :P |
| 04:11.26 | benjk | flujan, Sangoma cards can be fitted with a daugther card that does EC on the board |
| 04:11.37 | coppice | benjk: I thought SCO was the real Unix :-) |
| 04:11.58 | coppice | benjk: they don't seem to sell the daughter cards separately, though |
| 04:12.05 | benjk | ah |
| 04:12.47 | coppice | i might be wrong, but sites listing all the other bits separately don't list the EC boards |
| 04:13.20 | flujan | benjk, only this? thought it has processors just like dialogic cards... so sangoma works like digium boards.... |
| 04:13.31 | benjk | Well, yes and no |
| 04:13.52 | benjk | all Sangoma cards have Xilinx PGAs that do *some* of the processing |
| 04:14.19 | benjk | same for Digium cards |
| 04:14.32 | coppice | the sangoma E1 cards have a processor, but it doesn't seem to do anything. they have anf FPGA which does HDLC, and a few other bits and pieces |
| 04:15.19 | benjk | flujan, TDM protocols are layered, you will find that cards generally do the lower layer processing on board, and the higher layer processing on the host |
| 04:15.41 | benjk | the differences are how far up the card will hand off processing to the host |
| 04:16.40 | coppice | The dialogoc boards have a very brain dead balance of on board and off board processing. It made sense in the 1980s when hosts were slow, but they never adapted |
| 04:16.44 | flujan | hum... good to now |
| 04:17.32 | flujan | i am asking because my boss bought a 406P and we had problems with ec and now with the board itself. |
| 04:17.46 | flujan | the board just hangs up any dial attempt. |
| 04:17.47 | benjk | in high call density environments it makes most sense to do echo canceling and some heavy duty codecs on the board |
| 04:18.15 | benjk | flujan, that may be the driver software though |
| 04:18.27 | coppice | not sure about the codecs, but on board EC is a definite candidate for a good design |
| 04:18.41 | flujan | the digium reseller in brazil said that 406 were bad and he will give us a 407P. |
| 04:18.54 | flujan | but my boss is angry with digium.. |
| 04:18.59 | coppice | flujan: the TE406P worked so well, they seem to have replaced it :-) |
| 04:19.04 | duderock | Any idea what this means in /var/log/messages: Sep 1 17:05:19 talk -- MARK -- |
| 04:19.55 | flujan | coppice, do you remeber me asking you question about unicall some time ago? So since them i am fighting with this board... |
| 04:20.18 | benjk | they'll replace it with a TE666P |
| 04:20.19 | flujan | coppice, When i put the board to work (removing the ec module) the card stop working... |
| 04:20.42 | flujan | benjk, lol... :D |
| 04:20.45 | coppice | the problems with 406 seems to be with the EC. If you take off the daughter board I have heard they work OK |
| 04:21.37 | benjk | what's the P for in TExxxP anyway? |
| 04:21.52 | coppice | PCI maybe? |
| 04:21.58 | benjk | ah ok |
| 04:22.15 | CunningPike | Pike |
| 04:22.51 | benjk | heh |
| 04:25.22 | JT | is there a simple way to increase the audio volume of a Playtones() tone? |
| 04:27.43 | flujan | here: http://pastebin.ca/158479 this is debug i have when I try to place a call. I was having the same problem with the EC module in a 406P |
| 04:28.13 | flujan | the board worked for some time and now... from nothing i got this error messages... |
| 04:28.44 | benjk | did you contact Digium support? |
| 04:29.17 | benjk | they should either be able to fix it (if it is software related) or replace the board (if it is hardware related) |
| 04:30.19 | flujan | benjk, i am from Brazil... No digium support here. Just the reseller which has no technical support... :( |
| 04:30.32 | benjk | you can always call Digium over IAX2 |
| 04:30.58 | benjk | IAX2/guest@misery.digium.com |
| 04:30.59 | flujan | benjk, really? |
| 04:31.44 | flujan | maybe I should try to send them a e-mail... :D |
| 04:31.56 | benjk | you can do that too, yes |
| 04:34.05 | flujan | just to relieve my consciousness..http://pastebin.ca/158483 |
| 04:34.16 | flujan | this zapata.conf look reasonable? |
| 04:38.19 | duderock | http://paste.ubuntu-nl.org/22298 |
| 04:38.23 | duderock | anyone ever see this? |
| 04:38.51 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/xbmodder) |
| 04:38.59 | Sargun | is there some sort of device that I can plug a T1/E1 into one side, and it will be a phone block on the other side? |
| 04:39.09 | Sargun | because I need to serve up ~700 line |
| 04:39.10 | Sargun | s |
| 04:39.12 | *** join/#asterisk ellisdee (i=ellisdee@cpe-70-116-118-236.houston.res.rr.com) |
| 04:39.19 | Sargun | with the minimum number of boxes |
| 04:39.26 | coppice | Sargun: its called a channel bank |
| 04:40.13 | Sargun | is there a (very) large channel bank that'll take in raw SIP? |
| 04:40.23 | Sargun | or I can hook via PCI? |
| 04:40.42 | benjk | yeah, its called a SIP gateway |
| 04:41.16 | Sargun | benjk, link to one that can take maybe -- |
| 04:41.24 | Sargun | 96 channels, or greater? |
| 04:42.08 | benjk | depends on how many T1 interfaces it has |
| 04:43.35 | benjk | Cisco AS5300 |
| 04:43.55 | benjk | pricey, but you seem to want a black box |
| 04:44.18 | benjk | otherwise, you could build one yourself |
| 04:44.38 | Sargun | what cards?> |
| 04:44.50 | Sargun | we do need a black box |
| 04:44.55 | benjk | T1 PCI cards |
| 04:45.17 | Sargun | and then how do you split the 8 conductor T1 interface into 48 conductors (24 lines)? |
| 04:45.31 | Sargun | A channel bank seems just as expensive as a SIP gateway |
| 04:45.34 | benjk | for example, you could build a bunch of Soekris boxes with Sangoma single or dual port T1 cards |
| 04:45.59 | benjk | if you build 10 of those, its still much much cheaper than a Cisco AS5300 |
| 04:46.19 | Sargun | and then how do you split the 8 conductor T1 interface into 48 conductors (24 lines)? |
| 04:46.32 | Sargun | the T1 cards into independent FXS ports |
| 04:46.45 | benjk | I thought you wanted SIP? |
| 04:46.45 | coppice | you use a channel bank |
| 04:47.01 | Sargun | ok |
| 04:47.03 | flujan | guys... time to go bed :) |
| 04:47.05 | Sargun | I have an asterisk box |
| 04:47.06 | benjk | you either have SIP or FXS |
| 04:47.07 | flujan | see you :) |
| 04:47.13 | Sargun | and about 700 analog phone clients |
| 04:47.17 | benjk | see ya |
| 04:47.22 | Sargun | how do I put 700 holes into my box? |
| 04:47.30 | Sargun | cheaply as possible? |
| 04:47.30 | benjk | then why do you want a SIP gateway? |
| 04:47.45 | Sargun | I don't know |
| 04:47.48 | Sargun | thats what you said? |
| 04:47.51 | benjk | no |
| 04:47.54 | benjk | you asked for SIP |
| 04:48.08 | Sargun | then what is the answer? |
| 04:48.24 | benjk | if you have a bunch of T1s coming in and you want to hook up 700 analog phones, you don't need SIP really |
| 04:48.57 | benjk | anyway, if you have T1s on one end and analog phones on the other, then you use channel banks |
| 04:49.41 | Sargun | What if I just have an asterisk box? |
| 04:49.52 | benjk | tough luck |
| 04:49.52 | Sargun | and wanna hook up 700 analog devices to it |
| 04:50.04 | Sargun | eh |
| 04:50.07 | Sargun | how do I do it though |
| 04:50.11 | benjk | you don't |
| 04:50.14 | Sargun | lol |
| 04:50.30 | Sargun | whats the maximum number of ports an asterisk box can handle? |
| 04:50.41 | benjk | what kind of ports? |
| 04:51.07 | Sargun | analog |
| 04:51.10 | Sargun | FXS |
| 04:51.25 | benjk | reliably or sucky? |
| 04:51.52 | Sargun | sucky |
| 04:51.57 | benjk | maybe 24 |
| 04:52.00 | Sargun | hook up a buncha ATAs? |
| 04:52.06 | Sargun | what about reliable? |
| 04:52.06 | Sargun | :-P |
| 04:52.19 | benjk | that's a bit pricey though when you have 700 phones |
| 04:52.28 | benjk | also, its a nightmare to manage at that number |
| 04:52.32 | tessier | Is there any way to transfer a call between two phones from the console? |
| 04:52.54 | tessier | Sargun: You need a channel bank. |
| 04:53.01 | tessier | Do not connect analog devices to an asterisk box. |
| 04:53.23 | tessier | A channel bank that does T-1 or some device that does analog to SIP. Like a Mediatrix or something. |
| 04:53.34 | benjk | reliably, about 4 or so with a good (expensive) analog card |
| 04:54.56 | benjk | for 700 phones there is no way around channel banks, though |
| 04:56.39 | tessier | Why have 700 analog phones anyhow? With channel banks involved it might be cheaper to go with SIP phones. |
| 04:56.50 | benjk | and you may need multiple Asterisk servers too |
| 04:57.18 | benjk | tessier, you obviously never worked for an organisation as big as to have 700 phones |
| 04:57.59 | benjk | if they have 700 analog business phones on their desks and they are not written off yet, then there is absolutely now way in the universe that will get you a budget for replacing them |
| 04:58.00 | tessier | No, I haven't |
| 04:58.00 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 04:58.08 | tessier | Oh, I do know that much |
| 04:58.11 | *** join/#asterisk duderock (n=natrix@CPE-72-131-35-138.wi.res.rr.com) |
| 04:58.15 | tessier | I was wondering if this is a new install or an existing one |
| 04:58.42 | benjk | well, he said he *has* 700 phones to connect |
| 04:59.32 | duderock | can someone take a look at this log? it's from startup => crash and it has happened twice now in the last two days... spin_lock - no zaptel cards installed!? |
| 05:00.03 | benjk | the best thing is to start off with channel banks and then gradually move to IP phones as they need more phones or replace phones in the future |
| 05:00.15 | duderock | http://paste.ubuntu-nl.org/22300 |
| 05:02.45 | benjk | and you start replacing phones by getting the most impressive most expensive Cisco on the desk of the top executives |
| 05:03.19 | benjk | but make sure that you have tested them (and your setup) very well before you put them on their desks |
| 05:03.55 | Sargun | what about a SIP Gateway |
| 05:04.00 | benjk | what for? |
| 05:04.05 | Sargun | will that terminate at analog lines? |
| 05:04.11 | benjk | no |
| 05:04.29 | benjk | T1 SIP gateway will gateway between T1 and SIP |
| 05:04.33 | benjk | surprise surprise |
| 05:04.34 | Sargun | ergh |
| 05:07.04 | duderock | big trouble in little Milwaukee... ;) |
| 05:07.20 | duderock | here's the problem: |
| 05:07.29 | duderock | http://paste.ubuntu-nl.org/22300 |
| 05:13.21 | duderock | any ideas? |
| 05:17.52 | Sargun | Is there a Digital interface card that will take a T3 |
| 05:19.49 | benjk | Sangoma's T3 card |
| 05:20.04 | benjk | but the software ins't channelised |
| 05:25.08 | Sargun | mhmm |
| 05:25.25 | tessier | Sangoma makes a T3 card? Neato. Their dual T-1 card works pretty well so far. |
| 05:25.27 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:25.27 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:25.39 | benjk | you don't want 672 channels on a single server anyway, trust me on that |
| 05:26.18 | benjk | tessier, its for data |
| 05:27.25 | coppice | tessier: The Sangoma T3 card will not currently interface to a T3 carrying voice |
| 05:28.24 | *** join/#asterisk HaMYaI (i=CKGLOB@125.25.129.43) |
| 05:29.28 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 05:31.06 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 05:35.19 | *** join/#asterisk natrixgli (n=natrixgl@CPE-72-131-35-138.wi.res.rr.com) |
| 05:45.10 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 05:52.16 | Sargun | I am thinking of implementing VoIP to the masses: |
| 05:52.20 | Sargun | http://pastebin.ca/158557 |
| 05:52.28 | Sargun | how would I go from COX to PHX |
| 05:52.31 | Sargun | NOC to COX would be a T1 (dry) |
| 05:52.35 | Sargun | but I need a cross connect |
| 05:52.38 | Sargun | So I'd terminate at a channel bank in the CO |
| 05:52.41 | Sargun | would I get a cross connect |
| 05:52.45 | Sargun | and how much do cross connects cost? |
| 05:53.34 | benjk | I think you confuse this channel for the sales hotline of your telco |
| 05:53.40 | Sargun | heheh |
| 05:53.48 | Sargun | do you know about how much they cost |
| 05:54.01 | benjk | here in JP yes |
| 05:54.02 | Sargun | I got the price of T1 termination |
| 05:54.10 | Sargun | how much in Japan? |
| 05:54.26 | benjk | between 300 and 500 USD per month |
| 05:54.40 | Sargun | holy-crap |
| 05:54.44 | Sargun | for a single line? |
| 05:54.51 | benjk | 23 channels |
| 05:55.20 | Sargun | no -- I don't mean a T1 line |
| 05:55.35 | Sargun | for a channel bank @ CO -- and then cross connects |
| 05:56.05 | benjk | you can get a T1 for data or for PRI, that's it |
| 05:56.20 | benjk | or SS7 |
| 05:57.14 | sx-wks | well... we get *paid* by the telco for all the traffic we receive :D |
| 05:57.23 | benjk | nice |
| 05:57.56 | Sargun | how? |
| 05:58.42 | benjk | if you are a telco, you get paid for terminating traffic for other telcos |
| 06:00.26 | *** join/#asterisk dalekurt (n=dalekurt@port0033-afw-adsl.cwjamaica.com) |
| 06:00.32 | dalekurt | Hey Strom_C |
| 06:01.14 | dalekurt | Anyone on IAXtel or FWD |
| 06:01.34 | sx-wks | Sargun: basically, we don't do outgoing calls |
| 06:01.53 | sx-wks | we get a few mils of a eur per minute or so |
| 06:02.03 | sx-wks | but we make millions of minutes |
| 06:02.23 | Sargun | ah |
| 06:02.26 | sx-wks | (over 10 E1 so far) |
| 06:02.53 | Sargun | sx-wks, basically I want to know how much it is for a last-mile cross-connect |
| 06:03.07 | sx-wks | Sargun: depends on the location, I guess |
| 06:03.18 | sx-wks | and the technology |
| 06:03.59 | sx-wks | we get our E1s over what are probably some E3 on fiber |
| 06:05.55 | sx-wks | Sargun: all I know is that the most expensive and PITA part is making holes in the ground to dump the cable in |
| 06:06.13 | Sargun | heh |
| 06:09.49 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 06:10.57 | sx-wks | lilo: heya |
| 06:13.23 | Sargun | why are you here? |
| 06:13.32 | Sargun | to lay down the law :-P |
| 06:13.55 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.228) |
| 06:13.59 | benjk | lying down on the lawn is always nice |
| 06:14.01 | sx-wks | gah, stupid chanspy crashes |
| 06:14.32 | sx-wks | getting layed with the lawyers assistant is nice too :D |
| 06:14.39 | benjk | heh |
| 06:15.15 | sx-wks | (sex in the city style, obviously) |
| 06:26.27 | *** join/#asterisk skozombie (n=hmsc@171.110.233.220.exetel.com.au) |
| 06:27.53 | snappy | oh man, lying down on peoples lawns were one of my pasttimes |
| 06:28.03 | skozombie | snappy: drunk? |
| 06:28.22 | snappy | Nah when I was a kid, I used to like lying down on good grass |
| 06:32.25 | skozombie | I'm recieving faxes thru asterisk, but it's only getting the top few inches, any suggestions? |
| 06:32.51 | *** join/#asterisk linlin (i=linlin@c-67-173-38-87.hsd1.il.comcast.net) |
| 06:33.29 | ido | skozombie: fax it upside down? :-o |
| 06:33.54 | ido | skozombie: is it getting disconnected by asterisk, or is the remote fax complaining that the line quality is too poor? |
| 06:34.09 | skozombie | it's reporting everything is fine |
| 06:34.13 | skozombie | which is what's confusing me |
| 06:34.23 | ido | the fax machine sending the document says everything is fine? |
| 06:34.30 | skozombie | yeah |
| 06:34.34 | ido | hmm |
| 06:34.51 | ido | odd! in that case, what are you using to receive the fax? |
| 06:35.00 | skozombie | rxfax |
| 06:36.02 | ido | i don't have any experience with rxfax, but if you haven't already looked for one, look for a verbose/debug reporting/logging option in rxfax. |
| 06:38.05 | sx-wks | skozombie: then you should try with iaxmodem and hylafax. it is said to work much better |
| 06:39.02 | skozombie | sx-wks: ok, i just want to save the fax to a file :/ |
| 06:39.20 | sx-wks | heh. aren't we all :D |
| 06:45.14 | skozombie | heh, getting the top inch of a doc is pretty unhelpful |
| 06:45.52 | skozombie | `060903-004408 WARNING[6689]: channel.c:2355 set_format: Unable to find a codec translation path from unknown to unknown |
| 06:45.55 | skozombie | 060903-004408 WARNING[6689]: app_rxfax.c:309 rxfax_exec: Unable to restore read format on 'Zap/2-1' |
| 06:45.58 | skozombie | could that be related? |
| 06:56.52 | *** join/#asterisk kerlnel20 (n=kerlnel2@210.238.181.187) |
| 06:57.07 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 06:57.17 | kerlnel20 | hi there |
| 06:57.38 | kerlnel20 | would it be possible in asterisk to have voice and video conferencing? |
| 07:00.01 | sx-wks | kerlnel20: it has voice conferencing. video is a work in progress... you haven't read the docs have you ? |
| 07:03.02 | Qwell | sx-wks: he's been coming in here for about a year. You'd think so |
| 07:03.10 | sx-wks | heh |
| 07:04.37 | sx-wks | Qwell: http://bugs.digium.com/view.php?id=7871 |
| 07:05.04 | kerlnel20 | opps sorry |
| 07:05.36 | sx-wks | lol |
| 07:27.45 | skozombie | engin-out/90064568 202.61.13.40 N 5060 Unmonitored |
| 07:27.54 | skozombie | does that mean my outgoing sip worked or not? |
| 07:28.08 | skozombie | i'm trying to setup sip for outgoing calls |
| 07:29.12 | sx-wks | Name/username Host Dyn Nat ACL Port Status |
| 07:29.13 | sx-wks | free_telecom/0870380122 212.27.52.5 5060 OK (42 ms) |
| 07:29.22 | skozombie | :/; |
| 07:29.23 | sx-wks | ^ that's what I have for a working setup |
| 07:29.33 | skozombie | fuck i hate this voip provider |
| 07:29.46 | skozombie | it's so crap |
| 07:29.50 | sx-wks | lol |
| 07:29.56 | benjk | Unmonitored means you dont have qualify=yes in your sip.conf |
| 07:31.05 | skozombie | if I had a dollar for every 'restart now' ... |
| 07:31.24 | skozombie | w00t! it *looks* to be up |
| 07:31.53 | skozombie | engin-out/0249131897 202.61.13.40 N 5060 OK (145 ms) |
| 07:32.03 | skozombie | tried using phone number instead of account number |
| 07:32.06 | skozombie | now to make it work |
| 07:32.53 | skozombie | <PROTECTED> |
| 07:32.53 | skozombie | <PROTECTED> |
| 07:32.53 | skozombie | <PROTECTED> |
| 07:32.57 | skozombie | suggestions? |
| 07:32.58 | sx-wks | heh |
| 07:33.19 | sx-wks | you forgot the number to call I guess :D |
| 07:34.06 | sx-wks | something like Dial(SIP/${EXTEN}@engin-out) |
| 07:34.26 | skozombie | can't i just use it without a number to get a dial tone? |
| 07:34.35 | sx-wks | nope |
| 07:34.38 | skozombie | ah ok |
| 07:35.17 | sx-wks | sip works like t1... you are to give the dial tone locally |
| 07:36.20 | skozombie | cool |
| 07:36.22 | skozombie | <PROTECTED> |
| 07:36.27 | skozombie | <PROTECTED> |
| 07:36.27 | skozombie | <PROTECTED> |
| 07:36.54 | sx-wks | hmmm |
| 07:37.19 | skozombie | any logs i can check to see if sip autenticatedproperly? |
| 07:37.37 | sx-wks | sip debug peer engin-out |
| 07:39.52 | skozombie | <-- SIP read from 202.61.13.40:5060: |
| 07:39.52 | skozombie | SIP/2.0 404 Not Found |
| 07:39.56 | skozombie | :/ |
| 07:40.10 | sx-wks | uh ho |
| 07:40.40 | skozombie | From: "asterisk" <sip:asterisk@192.168.0.10>;tag=as55ffd968 |
| 07:40.45 | skozombie | thats in the outgoing sip packet |
| 07:40.49 | skozombie | is that a prob? |
| 07:40.59 | skozombie | should it be my voip number instead? |
| 07:42.59 | sx-wks | skozombie: you need to setup nat properly on the asterisk box |
| 07:43.13 | skozombie | portforwarding etc? |
| 07:43.17 | sx-wks | namely, put your public (fixed) ip address |
| 07:43.54 | skozombie | forward ports? |
| 07:44.06 | sx-wks | as there's no way the other box will be able to find 192.168.0.10 from where it is (outside) |
| 07:44.40 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 07:44.48 | sx-wks | in sip.conf, fill the externip=<blah> line |
| 07:45.16 | *** join/#asterisk lionelp (n=lionel@ip-128.net-82-216-65.rev.numericable.fr) |
| 07:45.30 | sx-wks | skozombie: solution number 2 is to install linux 2.6.18 on your gateway with the sip masquerading module |
| 07:45.41 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 07:49.02 | skozombie | sx-wks: *nods* |
| 07:52.03 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:52.12 | skozombie | I'm trying to get the ghetto modem-router to forward ports |
| 07:54.22 | sx-wks | skozombie: give yourself a break and buy a soekris :D |
| 07:54.35 | benjk | yeah, I was just about to say the same |
| 07:54.41 | benjk | or a WRAP |
| 07:56.12 | sx-wks | or a routerboard, or a ... |
| 08:02.37 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.228) |
| 08:12.22 | *** join/#asterisk suma (n=suma@cm53.omega182.maxonline.com.sg) |
| 08:18.44 | *** join/#asterisk SkoZombie (n=hmsc@85.100.233.220.exetel.com.au) |
| 08:18.49 | SkoZombie | what ports need forwarding? |
| 08:19.43 | Strom_C | all of them |
| 08:19.45 | Strom_C | ;) |
| 08:19.51 | Strom_C | what protocol are you using? |
| 08:22.30 | SkoZombie | SIP |
| 08:23.28 | Strom_C | 5060 for signaling and ports 10000-20000 inclusive for media |
| 08:23.51 | [hC] | whats the latest opinions on iax jitterbuffer and trunking? |
| 08:23.55 | [hC] | together or not |
| 08:30.58 | suma | looks like there is a replacement for g729 codec which accepts g729 data and also can encode g729 data |
| 08:31.05 | suma | anyone familier with that ? |
| 08:48.02 | *** part/#asterisk natrixgli (n=natrixgl@CPE-72-131-35-138.wi.res.rr.com) |
| 08:56.03 | snappy | Is anyone familiar with panasonic phone systems (tda100/tva50). I'm new to POTS/PSTN/PBX systems, we have a adtran 612 with I believe 16 pairs that connects to it, which then connects to our TDA50, which then connects to our TVA100. However not much has been configured. However I'm wondering if we can replace parts (the tva50 voice procesing system) with asterisk. It appears that the company doesnt want to pay to get our system properly configured, s |
| 08:58.13 | Strom_C | tva is voicemail, correct? |
| 09:09.09 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 09:20.18 | *** join/#asterisk natrixgli (n=natrixgl@CPE-72-131-35-138.wi.res.rr.com) |
| 09:26.07 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 09:26.07 | natrixgli | hi there. if inbound/outbound calling is functioning properly, do I need to be concerned about this error? 'Could not parse /etc/asterisk/localprefixes.conf' |
| 09:34.07 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
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| 10:05.58 | *** join/#asterisk _fa_ (n=faceoff@devel4.net) |
| 10:06.16 | _fa_ | Where Can I found information about new language for asterisk dialplans? |
| 10:14.51 | *** join/#asterisk lukketto (n=lukketto@host84-158.pool8710.interbusiness.it) |
| 10:22.15 | *** part/#asterisk lukketto (n=lukketto@host84-158.pool8710.interbusiness.it) |
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| 13:00.15 | uwe | hello, how can i disable incoming calls on some extensions in asterisk? |
| 13:00.47 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 13:02.39 | *** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.208) |
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| 13:06.28 | *** join/#asterisk knobo (n=Knut@c85-196-83-87.static.sdsl.no) |
| 13:06.53 | knobo | what is the searchcontexts in voicemail.conf? |
| 13:11.04 | *** join/#asterisk saftsack (n=saftsack@p54A7D04E.dip.t-dialin.net) |
| 13:23.58 | *** join/#asterisk Here_And_There (n=Here_And@pool-68-238-244-21.phlapa.fios.verizon.net) |
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| 13:32.48 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 13:43.26 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:43.29 | blitzrage | morn |
| 13:45.33 | *** join/#asterisk abatista (n=Ariel@70-46-87-158.ftl.fdn.com) |
| 13:46.44 | *** join/#asterisk RoyK (n=roy@gprs-ggsn6-nat.mobil.telenor.no) |
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| 13:51.27 | dieno | can ne 1 tell me how to install g.729 |
| 13:53.46 | *** part/#asterisk dieno (n=Dieno@58.65.193.77) |
| 14:00.16 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 14:07.34 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 14:08.14 | EmleyMoor | Is there a good discussion of fax solutions that exist for asterisk anywhere? |
| 14:08.48 | l-fy | what fax solution |
| 14:09.35 | EmleyMoor | Well, for a start, could I use Asterisk to detect a fax on my normal PSTN number and send it to one person, and on my Callsign number send it to another? |
| 14:10.04 | EmleyMoor | Also, is there any fairly easy way to send/receive faxes without using PSTN? |
| 14:12.28 | EmleyMoor | How much REN does FXS support |
| 14:12.29 | EmleyMoor | ? |
| 14:20.14 | bkw_ | l-fy, what are you doing here? |
| 14:20.27 | l-fy | spending some quality time with my friends |
| 14:20.35 | Assid | hes lying |
| 14:20.42 | Assid | hes trying to find someone to get laid |
| 14:21.00 | l-fy | Assid > in english is "she" |
| 14:21.12 | Assid | oh |
| 14:21.54 | EmleyMoor | Well, short of using the Spivak "er", "he" is applicable for unknown gender <g> |
| 14:22.12 | bkw_ | l-fy, did you calm down from your psychotic rampage you went on the other day? |
| 14:22.35 | Assid | psychotic rampage.. oh no.. not you too |
| 14:22.48 | l-fy | bkw_ > still upset for being kick out? |
| 14:22.53 | bkw_ | na |
| 14:23.48 | bkw_ | why would I be upset you have like 8 people in your channel |
| 14:24.02 | l-fy | than please go and bother someone else |
| 14:24.06 | l-fy | NEXT!!!!!! |
| 14:24.14 | bkw_ | its "then" not "than" |
| 14:25.31 | *** join/#asterisk doolph (n=doolph@200.46.148.58) |
| 14:25.38 | doolph | hi |
| 14:25.45 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 14:26.10 | *** join/#asterisk ManxPower (n=ManxPwer@122.sub-70-219-47.myvzw.com) |
| 14:26.11 | bkw_ | docelmo, you here? |
| 14:26.34 | Assid | shh.. hes meditating |
| 14:26.52 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 14:27.05 | doolph | there's any free website that I can learn more about agi scripting with php? |
| 14:29.43 | SplasPood | Anyone here have res_snmp working? |
| 14:30.31 | Assid | router! |
| 14:31.19 | EmleyMoor | Assid: I am hoping when I get my new router it will make my existing one redundant - I can then consider it as my asterisk box, as it's good enough |
| 14:32.11 | Assid | explain good enough? |
| 14:32.16 | EmleyMoor | Well, it probably needs more memory, but it has a fast enough processor |
| 14:34.07 | Assid | never! no such thing as fast enough.. |
| 14:34.15 | EmleyMoor | OK then, not inadequately too slow |
| 14:34.27 | Assid | there we go... glad you see it my way |
| 14:35.20 | EmleyMoor | I don't suppose any of you know where the first telephone exchange was in Sheffield? |
| 14:35.20 | SplasPood | In the central office? :P |
| 14:35.52 | Assid | was it running asterisk? |
| 14:35.52 | EmleyMoor | Assid: Definitely not - it was manual! |
| 14:35.53 | Assid | oh okay.. thought asterisk docs might have it |
| 14:35.53 | Assid | :P |
| 14:36.58 | EmleyMoor | Considering whether I should rename that machine before I make it my asterisk box - and a name relating to Sheffield's first telephone exchange would fit my naming policy and be appropriate :-) |
| 14:38.38 | Assid | call it firstphone |
| 14:38.47 | Assid | oh.. oh.. call it "hello" |
| 14:38.56 | EmleyMoor | Though I suppose I could call it "central" - that is a constituency in Sheffield and several of my others are named after constituencies |
| 14:40.20 | EmleyMoor | (I have attercliffe, hallam and hillsborough already) |
| 14:40.52 | Assid | i call my boxes box1 box2 box3 |
| 14:41.05 | Assid | i find it pretty understandable |
| 14:43.36 | SplasPood | heh I now have two sip phones, and two analog FXS attached phones that all ring at the same time in a room 10x10 |
| 14:43.51 | Assid | and your doing this because???? |
| 14:43.59 | EmleyMoor | 10x10? Is that in feet? |
| 14:44.09 | SplasPood | yea, prolly a bit bigger than 10x10 |
| 14:44.10 | Assid | inches |
| 14:44.15 | SplasPood | feet.. |
| 14:44.39 | EmleyMoor | Blimey, that sounds sadder than what I am planning, but then there is a good reason for mine |
| 14:44.39 | SplasPood | I have 1 of each type of device I have deployed :P |
| 14:44.49 | Assid | okay and the idea of putting soo many phones in 1 room is cause you are eccentric? |
| 14:45.11 | SplasPood | nah testing |
| 14:45.17 | SplasPood | ringers are off on all but 1 |
| 14:45.27 | Strom_C | pffft, that's nothing - I have three SIP phones and two Zap phones on my desk alone |
| 14:45.27 | Assid | so how are you testing if it doesnt ring |
| 14:45.32 | SplasPood | When they all ring it sounds like a call center tho |
| 14:45.35 | Assid | what if the ringers arent working.. how would yu know |
| 14:45.50 | SplasPood | Assid: If a client goes "The polycom does this when I do that" I then go do this while I do that :P |
| 14:45.53 | Assid | dude.. put on the ringers!! you arent testing till they ring |
| 14:46.08 | Assid | man your lucky |
| 14:46.16 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 14:46.23 | SpaceBass | anyone using broadvoice having problems registering? |
| 14:46.28 | EmleyMoor | Talking of ringers, how many REN of Zap phones can I put on one FXS port? |
| 14:46.29 | *** part/#asterisk raggi (n=ra66i@skynet.cx) |
| 14:46.31 | Strom_C | SpaceBass: let me look |
| 14:46.34 | Assid | i had to deploy polycom based phones and i never held one in my hand before |
| 14:46.47 | SplasPood | Assid: luckily they're pretty solid |
| 14:46.49 | Strom_C | SpaceBass: mine is registering fine |
| 14:46.55 | Assid | yeah.. i got my 301 |
| 14:47.04 | SpaceBass | Strom_C, I've been at "request sent" for 24 hours |
| 14:47.11 | SpaceBass | rebooted, reloaded.... |
| 14:47.54 | Strom_C | Assid: i'm doing my big polycom install today; I had a 430 overnighted to me earlier this week for me to fuck around with |
| 14:47.54 | SplasPood | or rather, SVN HEAD |
| 14:48.03 | SplasPood | Strom_C: haha.. I read that as "I had 430 *phones* over-nighted to me |
| 14:48.20 | Strom_C | that would be nice |
| 14:48.27 | Strom_C | i could wallpaper my apartment with them |
| 14:48.34 | SplasPood | I'd plug them all in in this room, and make em all ring :P |
| 14:49.01 | Assid | well |
| 14:49.23 | Assid | if they all have the same sip configuration.. its pretty easy. just gotta make 1 other file for the phones registration |
| 14:49.35 | Assid | and voila.. deployed |
| 14:49.44 | Assid | PnP |
| 14:50.04 | Strom_C | right; the intimidating bit was that i had touched some polycom configs before, and if you don't know *exactly* what you're doing, then the XML files are pretty overwhelming |
| 14:50.29 | Assid | true |
| 14:50.30 | *** join/#asterisk Nebukadneza (n=daddel9@i3ED6F5EA.versanet.de) |
| 14:50.37 | Assid | you could make a config to just edit the parts you need to |
| 14:50.43 | Strom_C | so better to become comfortable with them while I'm not billing the client hourly :) |
| 14:50.47 | Assid | mostly the registration since thats all you really edit |
| 14:53.16 | SplasPood | Strom_C: Docs are pretty good tho |
| 14:53.30 | SplasPood | One thing I've never figured out, that I'd really like, is the ability to change that '456' password |
| 14:53.38 | Strom_C | Documentation still doesnt matter if the configuration files are difficult to read |
| 14:53.53 | SplasPood | Strom_C: thats what search is for ;) |
| 14:54.04 | Strom_C | and when you have single lines of text with three dozen options on them, that becomes a problem |
| 14:54.06 | Assid | thats easy |
| 14:54.10 | Assid | you can do it from the phone |
| 14:54.16 | Strom_C | yeah, but what if you need to maintain someone else's configuration? |
| 14:54.23 | SplasPood | Assid: what strom said |
| 14:54.28 | Assid | edit the phone.. and then give them |
| 14:54.29 | SplasPood | but I was actually unaware of that |
| 14:54.40 | SplasPood | Assid: what if you want to change em all tho |
| 14:54.42 | Assid | your the one deploying it anyways |
| 14:55.18 | Strom_C | if the file itself is near-unreadable, then it won't be immediately clear to you what's been changed, modified, or set by the other person |
| 14:55.44 | SplasPood | Strom_C: Well thats why you have the 1 big 'defaults' file, and then only the custom stuff goes in each phone's file |
| 14:55.55 | Strom_C | you're not listening to me :) |
| 14:56.00 | SplasPood | that and all my configs are in SVN |
| 14:56.20 | SplasPood | soon to be generated on the fly directly from DB via a custom tftp server |
| 14:56.32 | Strom_C | I'm talking about walking into a situation where you're asked to make changes on a system which was initially set up by someone else |
| 14:56.37 | SplasPood | Ahh |
| 14:56.42 | SplasPood | yea then yer mostly screwed |
| 14:56.57 | Strom_C | hence the huge disadvantage of polycom-style configuration files |
| 14:57.11 | SplasPood | Well I don't know that they're much worse than any other vendor in that respect |
| 14:57.16 | Strom_C | there are better ways to do XML |
| 14:58.33 | SplasPood | such as? |
| 14:58.37 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:58.57 | Strom_C | such as not having three dozen disparate configuration options on a single line of text |
| 14:59.33 | Strom_C | put each option on a separate line so that you can easily look at it and understand what's going on |
| 14:59.48 | SplasPood | oh, well I cleaned them up... |
| 14:59.53 | Strom_C | unfortunately, I tried that with my polycom config file, and it actually crashed the phone |
| 15:00.00 | SplasPood | hrm, works 4 me |
| 15:00.03 | Assid | i just keep 2 sets |
| 15:00.07 | SplasPood | I just put line breaks between options |
| 15:00.10 | Assid | 1 - phone config.. 1 sip config |
| 15:00.20 | Assid | and ofcourse the directory.. which is normal |
| 15:00.24 | SplasPood | yea |
| 15:00.27 | SplasPood | thats how I do it too |
| 15:00.34 | Strom_C | SplasPood: i tried that with the 430, and it wouldn't boot |
| 15:00.44 | SplasPood | doesn't change the fact that polycom crams a ton of options into one element |
| 15:00.59 | Assid | yeah.. |
| 15:02.13 | Assid | i wish the 301's supported more than just 729/ulaw/alaw |
| 15:02.14 | *** join/#asterisk sonium (n=Sonium@p5492A52B.dip0.t-ipconnect.de) |
| 15:02.15 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:02.53 | Strom_C | the polycoms aren't bad phones by any stretch, but I still prefer my cisco phones |
| 15:03.16 | Assid | nah.. polycoms...... :)~ |
| 15:04.50 | Strom_C | one of the things I prefer about the cisco is that the handset design makes it easier to cradle against your shoulder if you need two hands free for a second or two |
| 15:05.11 | Strom_C | the polycom has that stupid "transmitter end is tiny" problem |
| 15:06.45 | Strom_C | that, and on the cisco, I can actually /hear/ the t-carrier noise, which the polycom filters out :) |
| 15:07.06 | *** join/#asterisk ariel_ (n=Ariel@70-46-87-158.ftl.fdn.com) |
| 15:08.49 | Assid | t-carrier noise? |
| 15:09.19 | Strom_C | yep |
| 15:09.34 | Strom_C | assuming the call is going over t-carrier, of course |
| 15:09.49 | Assid | the world should move to voip |
| 15:09.58 | Assid | quality of calls would be soo much better |
| 15:09.59 | Strom_C | yeah, but for the time being, it isn't |
| 15:10.00 | Assid | and cheaper |
| 15:10.40 | Strom_C | good t-carrier sounds just as good as g.711u with small packet sizes and low latency |
| 15:11.01 | Strom_C | i'd rather talk on t-carrier than, say, a voip call using g.729 |
| 15:11.09 | Assid | really? |
| 15:11.23 | Strom_C | yes. the other end actually becomes intelligible! |
| 15:11.24 | Assid | ifind 729 pretty decnet |
| 15:11.33 | Strom_C | you must have no ears then :) |
| 15:11.35 | SplasPood | is g729 much more susceptible to network jitter? |
| 15:11.43 | Strom_C | g729 just sounds like ass |
| 15:13.24 | Strom_C | i mean, come on...you're vocoding, and transmitting instructions for reconstructing that on the other end. It's completely different from actually transmitting PCM |
| 15:13.53 | Strom_C | doesn't handle a lot of things well at all...people mumbling, background noise, music, etc |
| 15:14.01 | Assid | what if its 729-729 |
| 15:14.13 | Strom_C | still vocoding |
| 15:17.10 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:17.16 | Strom_C | g726 is actually quite nice, because it's just a different way of encoding PCM data, so you wind up with almost identical quality to g711 but in about half the bandwidth |
| 15:17.47 | Assid | 726 huh? |
| 15:18.02 | Assid | my stupid phone doesnt have 726 support |
| 15:18.15 | Strom_C | but, of course, only for situations where you don't have enough bandwidth to do g711 ;) |
| 15:18.23 | eKo1 | g726 is also adpcm right? |
| 15:18.31 | Strom_C | g726 /is/ adpcm |
| 15:18.40 | Strom_C | g711 is straight pcm |
| 15:18.48 | Strom_C | non-linear, of course, but still pcm |
| 15:18.49 | EmleyMoor | If I set up one of my Zap phones to have different dial rules, can that include "* prefix for internal calls", for example? |
| 15:18.59 | *** join/#asterisk Sedorox (i=sedorox@smartserv/cna/Sedorox) |
| 15:19.00 | Assid | why cant polycom give us a patch for 726 |
| 15:19.06 | eKo1 | and what would linear pcm be? |
| 15:19.12 | Strom_C | eKo1: slin |
| 15:19.22 | Strom_C | EmleyMoor: you could do that, though you might conflict with locally-assigned vertical service codes |
| 15:19.24 | Assid | err.. does 501 support 726 ? |
| 15:19.24 | eKo1 | Ah! |
| 15:19.39 | EmleyMoor | Strom_C: Like what for example? |
| 15:20.03 | Strom_C | Assid: do 711 over a LAN, and then use 726 for your trunking |
| 15:20.30 | eKo1 | Will faxing work with G.736? |
| 15:20.36 | eKo1 | err G.726. |
| 15:20.40 | Strom_C | eKo1: no |
| 15:21.21 | Strom_C | EmleyMoor: well, I don't know much about the numbering plan there, but here in the US, for example, we dial *67 to block caller ID presentation on a per-call basis |
| 15:21.43 | eKo1 | Well, that sucks. |
| 15:21.57 | Assid | err.. voicepulse supports 726 ? |
| 15:22.02 | EmleyMoor | Ah, I see... could always do *9* instead of * for such codes |
| 15:22.24 | Strom_C | Assid: most of the carriers should support it |
| 15:22.30 | Assid | free codec |
| 15:22.31 | Assid | ? |
| 15:22.33 | Strom_C | yes |
| 15:22.43 | Strom_C | and very lightweight to transcode on the asterisk box |
| 15:22.53 | Assid | lighter than 729 ? |
| 15:22.59 | Strom_C | by leaps and bounds |
| 15:23.15 | Assid | hrmm your right |
| 15:23.30 | Assid | im currently using gsm to voicepulse |
| 15:23.36 | Strom_C | oh blech |
| 15:23.43 | Strom_C | i wouldn't talk on that connection :) |
| 15:23.47 | Assid | tried ulaw.. didnt really get much quality difference |
| 15:24.38 | Strom_C | wow...im amazed you cant tell the difference |
| 15:24.45 | Strom_C | because I hear a /huge/ difference between the two |
| 15:25.20 | Strom_C | gsm adds a very grainy quality to the audio |
| 15:25.22 | Assid | so whats the allow.. g726 / 726 ? |
| 15:25.29 | Strom_C | g726, yes |
| 15:27.05 | Assid | well.. added 726|gsm |
| 15:28.53 | Strom_C | restrict it to g726 only and see how you like that |
| 15:30.04 | Assid | well.. quality seems same |
| 15:30.20 | Strom_C | what codec are you using between the phone and the asterisk box? |
| 15:30.34 | Assid | i just called the asterisk box from skype |
| 15:30.34 | Strom_C | and, more importantly, what are you calling? |
| 15:30.53 | Strom_C | that's not a good indicator, because skype uses g729 for everything |
| 15:30.59 | Assid | hrmm |
| 15:31.01 | Strom_C | so all your skype calls will sound equally abysmal |
| 15:31.18 | Assid | well.. i will come to know when they come in on monday or so |
| 15:31.26 | eKo1 | I thought Skype used a proprietary codec. |
| 15:31.37 | Strom_C | skype uses g729 for PSTN calls |
| 15:31.44 | eKo1 | Interesting. |
| 15:32.00 | Strom_C | anyway, I have to take a shower |
| 15:32.57 | Assid | alrite |
| 15:33.39 | *** join/#asterisk WildPikachu (n=WildPika@unaffiliated/wildpikachu) |
| 15:34.01 | WildPikachu | if I don't include an Answer() and I just have a Dial(SIP/me) .... will it only answer if I pickup? |
| 15:35.00 | Assid | how will it |
| 15:35.03 | *** join/#asterisk lilalinux (i=e-trolle@deepthroat.deswahnsinns.de) |
| 15:35.07 | lilalinux | hey guys |
| 15:35.08 | Assid | you need to answer the phone before you transfer the line |
| 15:35.58 | EmleyMoor | WildPikachu: Are you on about answering an incoming line? |
| 15:36.23 | lilalinux | I have a problem with an agi: I simply want to do "SAY DIGITS 1234 #", but it hangs up before it said all digits. How do I wait for the command to be completed? |
| 15:36.43 | Assid | should jitterbuffer be on/off? |
| 15:38.16 | WildPikachu | hrmmm .... currently all normal pstns here ring for each incoming call ... I want to plug one of the pstn's into my pabx and route the call to a sip phone, but if it picks up the pstn the other pstn's won't ring ... so if i'm not here asterisk will be picking up each call |
| 15:38.21 | WildPikachu | doubt i'm making sense |
| 15:38.38 | Assid | can i set the codecpriority per context |
| 15:39.38 | EmleyMoor | WildPikachu: If you want asterisk to answer your PSTN, then yes |
| 15:39.59 | WildPikachu | so if I have s,1,Dial(SIP/me) and no Answer() ...... will my SIP ring? |
| 15:40.21 | EmleyMoor | I believe (though I seek confirmation of this), that the FXS card can support 5 REN. |
| 15:40.55 | EmleyMoor | Your SIP will ring, but what the effect would be on the incoming PSTN call, I can't say |
| 15:41.17 | WildPikachu | EmleyMoor, if I pickup the SIP, will it then answer the PSTN call? |
| 15:41.47 | EmleyMoor | That I couldn't tell you. I think, though, that an unanswered PSTN call is not, technically, in Asterisk |
| 15:43.21 | WildPikachu | hrmmmm |
| 15:43.59 | EmleyMoor | Anyone installed FXS in Britain? Does using an NTE5 socket provide for the ring capacitor and a way to connect extensions? |
| 15:48.46 | Strom_C | WildPikachu: if you don't do an answer() before dial(), the line will supervise once the endpoint sends its own ANSWER message |
| 15:49.09 | WildPikachu | supervise = pickup? |
| 15:49.12 | Strom_C | in fact, it's recommended that you /don't/ answer before dialing an endpoint which passes its own answer supervision |
| 15:49.12 | Strom_C | yes |
| 15:49.50 | WildPikachu | so it will work how I want it, only pickup if the SIP picks up? *starts to get excited* |
| 15:50.29 | Strom_C | yep |
| 15:55.12 | *** join/#asterisk RoyK (n=roy@ti211110a081-1433.bb.online.no) |
| 15:55.40 | RoyK | . |
| 16:02.32 | lilalinux | does anybody know, why this most simple agi doesnt work? http://rafb.net/paste/results/ZyH1ug97.html |
| 16:05.05 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:24.26 | *** join/#asterisk _DAW (n=_DAW@adsl-222-12-239.msy.bellsouth.net) |
| 16:24.35 | _DAW | Hello mates |
| 16:34.31 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 16:39.30 | eKo1 | Quick ?: Can I use the latest zaptel and libpri sources with Asterisk 1.0? |
| 16:40.03 | EmleyMoor | Strom_C: Does that mean I can process based on CLID and time before calling an endpoint? |
| 16:41.40 | EmleyMoor | (i.e. I need not answer first, until I have decided whether to dial phones or connect to voicemail?) |
| 16:43.14 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 16:47.56 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 16:51.19 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 16:51.32 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 16:52.18 | dlynes_laptop | Gooooooooooooooooood morning, asterisk! |
| 16:52.40 | file | This is K9.11, your daily #asterisk news radio station! |
| 16:52.48 | file | Coming up next - how to configure sip.conf for all your needs |
| 16:52.57 | file | But first a word from our sponsors, oh wait - we have none! |
| 16:53.08 | eKo1 | Digium? |
| 16:53.25 | eKo1 | FSF? |
| 16:53.31 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 17:00.18 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 17:00.27 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:01.01 | dlynes_laptop | buenas dias, paolob |
| 17:01.06 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 17:01.07 | paolob | hola dlynes_laptop |
| 17:01.26 | dlynes_laptop | como esta? |
| 17:03.10 | paolob | dlynes_laptop, bien, gracias |
| 17:03.32 | paolob | dlynes_laptop, se sabe nada de la 1.4? |
| 17:04.17 | dlynes_laptop | no idea what you just said...hehehe....I think maybe you said 1.4 still isn't out yet? |
| 17:04.52 | *** join/#asterisk saftsack (n=saftsack@p54A7D04E.dip.t-dialin.net) |
| 17:05.15 | dlynes_laptop | I comprendez las Francais and los Mandarindo |
| 17:05.16 | paolob | dlynes_laptop, yes, when should 1.4 go out? |
| 17:05.33 | dlynes_laptop | It was rumored to come out in June |
| 17:05.48 | dlynes_laptop | So, i'm guessing it'll come out when it comes out |
| 17:05.59 | EmleyMoor | June next year maybe <g> |
| 17:07.18 | Ryushin | Okay, from what I've been reading, people are having problems with the polycom ip430 phones. I've already upgraded the bootrom to the 3.1.x series, so I can't downgrade. What bootrooms and sip versions are people using with this phone? |
| 17:07.37 | benjk | you can always edit the version number yourself and then rebuild |
| 17:07.39 | dlynes_laptop | Polycoms can't be downgraded? |
| 17:07.41 | dlynes_laptop | that's lame |
| 17:09.49 | benjk | hi dlynes, back from holidays? |
| 17:10.28 | dlynes_laptop | Never was on holidays |
| 17:10.35 | benjk | ah |
| 17:10.38 | benjk | busy then |
| 17:10.41 | dlynes_laptop | Was just busy getting out billing system finished, and working on a side project |
| 17:10.45 | dlynes_laptop | s/out/our/ |
| 17:10.57 | dlynes_laptop | Well, and making time for a new gf :) |
| 17:11.20 | benjk | oh, that explains of course |
| 17:12.56 | eKo1 | I dislike billing systems... |
| 17:13.30 | dlynes_laptop | I dislike giving out free phone service |
| 17:14.01 | eKo1 | I like free phone service. |
| 17:14.53 | eKo1 | Gimme gimme gimme |
| 17:15.02 | Ryushin | Well, from what I've read, it can't be downgraded below 3.1.x |
| 17:15.16 | dlynes_laptop | eKo1: Skype has free phone service |
| 17:15.23 | dlynes_laptop | eKo1: otoh, their phone service sucks |
| 17:15.25 | Ryushin | But the 468* reset keys don't work on the 430's. I haven't found out what does work yet. |
| 17:15.26 | benjk | no |
| 17:15.40 | benjk | skype has free p2p phone calling |
| 17:15.45 | benjk | not free phone service |
| 17:15.50 | dlynes_laptop | benjk: Skype has free p2p and free North American calling atm |
| 17:16.02 | dlynes_laptop | benjk: It won't be free forever, but it's free for now |
| 17:16.51 | benjk | I still won't call theirs a service |
| 17:17.09 | dlynes_laptop | Neither would I, considering the quality |
| 17:17.12 | benjk | hijacking customer resources |
| 17:17.23 | benjk | not providing your own infrastructure |
| 17:17.25 | dlynes_laptop | My friend called me on Skype pstn service the other day |
| 17:17.30 | benjk | thats not service in my book |
| 17:17.38 | dlynes_laptop | It's horrible...some of the time I couldn't even understand what he was saying |
| 17:17.59 | dlynes_laptop | Hijacking customer resources? |
| 17:18.03 | mog | skype = lame.... |
| 17:19.01 | benjk | Skype use their customer's boxes to provide virtual servers |
| 17:19.09 | dlynes_laptop | oh |
| 17:19.28 | benjk | thats why skype totally bogs down people;s machines |
| 17:19.31 | dlynes_laptop | How does that help when someone completely unrelated is calling someone else completely unrelated? |
| 17:19.37 | benjk | it grabs all the resources it can |
| 17:19.56 | benjk | they may use your box to route the call |
| 17:20.22 | benjk | especially if you are not behind NAT and if you have good bandwidth |
| 17:20.23 | dlynes_laptop | Oh...you mean to try to reduce the latency? |
| 17:20.42 | benjk | not only that, they don't have any proxies/servers |
| 17:20.42 | dlynes_laptop | i.e. for international calls? |
| 17:20.50 | benjk | they have nothing |
| 17:20.55 | benjk | only their software |
| 17:21.06 | dlynes_laptop | ah |
| 17:21.07 | benjk | and the gateways for PSTN calls |
| 17:21.09 | dlynes_laptop | that is quite lame |
| 17:21.53 | benjk | so if you were to run a service, even a free service, like say Gizmo or FWD, then at least you'd provide one or more proxies |
| 17:22.00 | benjk | Skype are too cheap |
| 17:22.10 | benjk | they don't want to look after the infrastructure |
| 17:22.36 | dlynes_laptop | benjk: well, you have to admit...they're smart about one thing |
| 17:22.39 | benjk | so they wrote their software such that their customers' boxes are used to build virtual proxies |
| 17:22.50 | *** join/#asterisk yxa (i=lonari@cm121.gamma228.maxonline.com.sg) |
| 17:22.51 | dlynes_laptop | benjk: they never have to pay for bandwidth :) |
| 17:22.59 | benjk | yeah |
| 17:23.07 | benjk | nor server boxes |
| 17:23.17 | benjk | electricity |
| 17:23.25 | benjk | colo |
| 17:23.27 | dlynes_laptop | aircon |
| 17:24.56 | Qwell | space, maintenance, wiring |
| 17:25.10 | dlynes_laptop | rent |
| 17:25.25 | Qwell | advertising.. |
| 17:25.40 | dlynes_laptop | They advertise plenty |
| 17:25.43 | Qwell | really? |
| 17:25.48 | dlynes_laptop | Yeah |
| 17:25.54 | Qwell | I can't say I've ever seen a skype ad |
| 17:26.04 | dlynes_laptop | I've even seen a skype ad on tv |
| 17:26.05 | eKo1 | Me neither. |
| 17:26.09 | EmleyMoor | Skype could just as easily be run by one guy with a K6-2 box in Cleckhuddersfax as by a big firm |
| 17:26.33 | dlynes_laptop | And I'm sick of seeing Vonage ads |
| 17:26.41 | dlynes_laptop | They've got too many |
| 17:26.43 | file | VoIP with Vonage! |
| 17:26.47 | EmleyMoor | Vomage! |
| 17:27.10 | Ciber311 | anyone here have a spa942? |
| 17:27.11 | dlynes_laptop | Those bastards probably run on Digium hardware, too |
| 17:27.16 | dlynes_laptop | *blink* |
| 17:27.37 | *** join/#asterisk rbordeaux (i=hidden-u@80.169.196.234) |
| 17:27.50 | benjk | IIRC, Vonage are a Cisco shop |
| 17:28.08 | dlynes_laptop | How can they afford to be so cheap when they're using Cisco everything? |
| 17:28.22 | Luke-Jr | dlynes_laptop: TV is stupid |
| 17:28.23 | benjk | they have investment |
| 17:28.28 | Qwell | dlynes_laptop: note that vonage hasn't been profitable in...ever |
| 17:28.43 | benjk | they are not profitable yet |
| 17:28.51 | dlynes_laptop | So, iow, they have no plans to ever be profitable? |
| 17:28.53 | benjk | if they ever will be |
| 17:28.59 | benjk | their IPO was disappointing too |
| 17:29.05 | Qwell | benjk: extremely |
| 17:29.08 | dlynes_laptop | As long as they keep getting investment monies? |
| 17:29.20 | benjk | they didn't plan for it to turn out this way of course |
| 17:29.47 | benjk | but hey, remember that Stones song |
| 17:30.01 | benjk | you can't always get what you wa-ant |
| 17:30.02 | dlynes_laptop | Start me up? |
| 17:30.04 | dlynes_laptop | oh |
| 17:30.14 | dlynes_laptop | Thought you were talking about the Windows 95 theme song :) |
| 17:30.22 | benjk | ouch |
| 17:32.31 | Nugget | heh |
| 17:32.53 | suma | guys anyone got the g729 link ? as received in the mailing lists ? |
| 17:32.58 | suma | is that true? |
| 17:33.15 | *** join/#asterisk ToxaP (n=ToxaP@213.227.193.75) |
| 17:34.00 | benjk | I would have thought that a Stones song about Windows would definitely have been "Sympathy for the devil" |
| 17:34.18 | EmleyMoor | Or "Throw us at them"? |
| 17:35.31 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 17:35.32 | benjk | ah, no, "I can't get no ... satisfaction" |
| 17:35.39 | benjk | thats clearly about Windows |
| 17:35.51 | benjk | couldn't be anything else |
| 17:36.27 | suma | Anyone got the blackfin fxo board ? |
| 17:36.42 | suma | looks that is promising and cheap too |
| 17:37.45 | *** join/#asterisk sapo_original (n=zorra@201.47.22.20.adsl.gvt.net.br) |
| 17:38.02 | sapo_original | ei, algum brazuaca aqui? |
| 17:38.11 | sapo_original | anybody knows the digivoice hardware? |
| 17:38.52 | benjk | fxo board doesn't sound promising to begin with |
| 17:40.27 | *** join/#asterisk ToxaP (n=ToxaP@213.227.193.75) |
| 17:41.28 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 17:42.02 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
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| 17:42.46 | eKo1 | sapo_original: are you brazilian? |
| 17:42.52 | sapo_original | [eKo1]: yes |
| 17:42.59 | suma | fxs is bit complex in power requirements |
| 17:43.08 | suma | fxo is easier to get start with |
| 17:43.49 | sapo_original | eKo1 why? |
| 17:45.39 | eKo1 | brazilian = brazuaca? |
| 17:46.01 | caio1982 | brazuca |
| 17:46.08 | benjk | bazooka |
| 17:46.14 | caio1982 | that too |
| 17:46.35 | benjk | bra booster |
| 17:46.54 | benjk | bra zooker |
| 17:47.43 | caio1982 | you might have more fun with brazilian chicks instead with brazilian's nickname |
| 17:47.44 | caio1982 | :) |
| 17:48.08 | benjk | Oh well, I can't complain |
| 17:48.15 | benjk | after all, I live in JP |
| 17:48.36 | caio1982 | hehe |
| 17:49.44 | dlynes_laptop | I'd much rather have a japanese girl than a brazilian girl, personally |
| 17:50.19 | dlynes_laptop | Man...there was a lot of super cute japanese girls at this sushi restaurant i was at last night :) |
| 17:50.21 | benjk | you can always go for a Japanese-Brazilian if you can't make up your mind |
| 17:50.33 | Ciber311 | lol dlynes_laptop |
| 17:50.50 | dlynes_laptop | Ciber311: well, you know i'm a sucker for asian girls....especially chinese :0 |
| 17:51.03 | Ciber311 | yup |
| 17:51.13 | benjk | yellow fever, heh |
| 17:51.22 | Ciber311 | haha |
| 17:51.30 | dlynes_laptop | Nah...I just love Chinese culture |
| 17:51.37 | dlynes_laptop | So it naturally extends to Chinese girls :) |
| 17:51.50 | benjk | nah, you are showing symptoms of yellow fever |
| 17:51.52 | coppice | dkynes_laptop: sucking chinese girls is a pleasant passtime :-) |
| 17:52.04 | dlynes_laptop | I worked in a Chinese company for about five years, so I kinda osmosed the culture |
| 17:52.07 | caio1982 | rofl |
| 17:52.19 | benjk | trust me, I am talking from first hand experience here |
| 17:52.31 | dlynes_laptop | Or entertaining chinese girls that suck is a pleasant pastime, too :) |
| 17:52.36 | RoyK | rofp |
| 17:53.12 | benjk | RoyK, rofp? rolling on the floor puking? |
| 17:53.18 | dlynes_laptop | Man, Brooke Shields is starting to become irritating now |
| 17:53.54 | coppice | how come? I haven't heard that name for years |
| 17:54.21 | dlynes_laptop | Oh...her and her "heartfelt apology" from Tom Cruise |
| 17:54.25 | dlynes_laptop | like gimme a break already |
| 17:54.30 | dlynes_laptop | keep your private life private |
| 17:55.22 | dlynes_laptop | He had criticized her for using antidepressant drugs instead of vitamins and exercise |
| 17:56.15 | dlynes_laptop | I think i've heard her and her jay leno appearance about ten times this morning already |
| 17:56.32 | coppice | any normal person would be depressed if they woke up and found they were brook shields |
| 17:56.46 | benjk | you should perhaps change your TV channels |
| 17:56.46 | dlynes_laptop | no kidding |
| 17:56.50 | dlynes_laptop | she looks so ugly now |
| 17:57.09 | dlynes_laptop | It's like she's not properly eating...her cheeks are all sunken in and everything |
| 17:57.29 | eKo1 | Could she be anorexic? |
| 17:57.35 | dlynes_laptop | no idea |
| 17:57.42 | dlynes_laptop | but she doesn't look as nice as she used to |
| 17:57.43 | coppice | sounds even worse than the last time i saw her. she was starting to look like candice bergman |
| 17:57.45 | dlynes_laptop | that's for sure |
| 17:58.02 | dlynes_laptop | I remember when she was about 5 or 6 years younger than me |
| 17:58.13 | dlynes_laptop | Now she's about ten or fifteen years older |
| 17:58.19 | eKo1 | hahaha |
| 17:58.23 | benjk | time warp |
| 17:58.33 | Ciber311 | dlynes_laptop: ever use a linksys spa 942> |
| 17:58.34 | Ciber311 | ? |
| 17:58.37 | dlynes_laptop | I mean she looks soooooooooooo old |
| 17:58.41 | coppice | there was a time when candice bergman was so cute, and look what happened |
| 17:58.44 | caio1982 | damn you guys, you made me googled for recent pictures of her |
| 17:58.58 | dlynes_laptop | Ciber311 been thinking about trying them...my regular wholesaler started stocking them |
| 17:59.19 | Ciber311 | i think i'm gonna order one now |
| 17:59.26 | coppice | there's a taiwanese singer who 5 years ago looked adorable. now she looks like a worn out hooker |
| 17:59.32 | dlynes_laptop | Ciber311: do they have a tiltable display |
| 17:59.35 | dlynes_laptop | Ciber311: ? |
| 17:59.38 | Ciber311 | no |
| 17:59.46 | dlynes_laptop | Then I guess I won't be trying them |
| 17:59.48 | Ciber311 | i demoed a polycom 501 to a client yesterday |
| 18:00.04 | dlynes_laptop | kinda useless without a tiltable display |
| 18:00.10 | Ciber311 | and he was bugging me about it not having a backlit screen and brighter lights/more lights lol |
| 18:00.22 | dlynes_laptop | It doesn't have a backlit screen? |
| 18:00.26 | Ciber311 | no |
| 18:00.27 | benjk | maybe you can tilt the entire phone |
| 18:00.37 | dlynes_laptop | that's lame |
| 18:00.40 | Ciber311 | yup |
| 18:00.53 | Ciber311 | and i understand why he was annoyed about it |
| 18:00.56 | Ciber311 | it's a bar heh |
| 18:01.03 | Ciber311 | well bar/restaurant |
| 18:01.03 | dlynes_laptop | Aastra's a cheaper quality phone, and it has an adjustable display, and a backlit display |
| 18:01.14 | Ciber311 | so they won't be able to see crap on the polycom's at night |
| 18:01.44 | Ciber311 | so it looks like i'll have to get the spa942's for this guy if they don't suck |
| 18:02.05 | dlynes_laptop | So polycoms would suck in a warehouse, too |
| 18:02.23 | Ciber311 | anywhere other than a well-lit desk :P |
| 18:03.14 | dlynes_laptop | the polycoms have a tiltable display though, right? |
| 18:03.18 | Ciber311 | no |
| 18:03.26 | dlynes_laptop | wtf? |
| 18:03.35 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 18:03.35 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:03.49 | dlynes_laptop | I thought they were supposed to be a business leader? |
| 18:03.54 | sapo_original | anybody know any bug in the agentcallback application? |
| 18:03.57 | Ciber311 | they are lol |
| 18:04.15 | benjk | not at the expensive of the more expenisve Cisco phones though ;) |
| 18:04.17 | Qwell | sapo_original: the whole application is a big bug |
| 18:04.19 | dlynes_laptop | I don't know of any of my customers that would use a phone that didn't have a tiltable display |
| 18:04.30 | Qwell | dlynes_laptop: bah, put bricks under it |
| 18:04.40 | dlynes_laptop | Qwell: power bricks? |
| 18:04.46 | Qwell | just bricks |
| 18:04.56 | Ciber311 | dlynes_laptop: the polycom's sit at a good angle |
| 18:05.08 | Ciber311 | so you can read the screen just fine while sitting or standing |
| 18:05.49 | sapo_original | if my agents stay logged using loginagent works, but, if i using the callbacklogin, the call is answered but, the agent and the customer don't hear anything.... |
| 18:06.10 | Qwell | sapo_original: That sounds like a NAT issue |
| 18:06.14 | benjk | come on guys, you are suffering from obesity already, so you can use the extra exercise getting up each time you get a call so you can see the display and shed a few calories in the process |
| 18:06.26 | sapo_original | [Qwell]: local network |
| 18:06.34 | *** join/#asterisk tzafrir_laptop (n=tzafrir@80.178.4.6.adsl.012.net.il) |
| 18:08.15 | sapo_original | Qwell i need any application appers to ballbacklogin... any sugestion? |
| 18:09.18 | sapo_original | i have any agents, but, hers dont stay in the telephone all the time.. i need the phone rings |
| 18:16.31 | dlynes_laptop | benjk: speak for yourself |
| 18:16.44 | dlynes_laptop | benjk: I've got the opposite problem |
| 18:17.25 | benjk | nah, I'm fine, I am neither skinny nor overweight |
| 18:17.57 | dlynes_laptop | yeah...i'm skinnier than skinny :0 |
| 18:18.03 | benjk | living in Japan is like being on diet all the time ;) |
| 18:19.09 | dlynes_laptop | benjk: nah...when you've got that many asian girls around, you couldn't possibly be on an asian-free diet |
| 18:19.33 | Qwell | free-asian diet maybe? |
| 18:27.53 | *** join/#asterisk nnp_ (n=nnp@85.134.141.69) |
| 18:28.42 | nnp_ | hey guys, does anyone know of a decent program/script that can create SIP headers? I want to simulate a connection |
| 18:28.53 | Qwell | sipp |
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| 18:34.08 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
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| 18:37.03 | *** join/#asterisk hellor (n=50ccb724@c-67-160-249-95.hsd1.ca.comcast.net) |
| 18:37.12 | hellor | Hi folks...Good morning |
| 18:37.43 | hellor | I'm trying to use the match function ${MATH(${ITEM-NUMBER}+ ${ID}*1000000) |
| 18:38.22 | Qwell | hmm |
| 18:38.41 | hellor | which results 178797.0000 |
| 18:38.58 | hellor | is there a way to chomp the decimals out (or make the output to be an integer?) |
| 18:39.36 | file | hellor: type "show function MATH" on the asterisk CLI |
| 18:40.08 | hellor | gotcha... |
| 18:40.32 | hellor | by the way, I am doing 2 calculations and it seems to be working 1.2.9 |
| 18:40.48 | hellor | according to the specs, it shouldn't be working :-) |
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| 18:54.55 | *** join/#asterisk tld (n=terje@115.80-203-96.nextgentel.com) |
| 19:03.10 | Assid | iom bored |
| 19:06.18 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 19:06.34 | Ciber311 | hmm |
| 19:06.56 | Ciber311 | anyone have intercom/paging working on polycom 501's? |
| 19:07.29 | SwK | yes |
| 19:07.51 | Ciber311 | does it work well? |
| 19:07.52 | Assid | paging??? |
| 19:07.52 | SwK | you have to set up an alert type on the polycom like ring-answer |
| 19:07.53 | *** join/#asterisk puowvip (i=ircuser@thegrey.diamond.org) |
| 19:08.00 | SwK | it works just fine |
| 19:08.02 | Assid | whats that |
| 19:08.13 | Ciber311 | SwK: where the heck do you set that? |
| 19:08.57 | Assid | whats a paging system? |
| 19:09.03 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.119) |
| 19:09.07 | SwK | ciber311: you set the _ALERT_TYPE channel variable before you call dial() in the dial plan to something that matches the alter types you have setup in the polycom's configs |
| 19:09.23 | SwK | if you use the old school 4 file system its in ipmid |
| 19:09.36 | SwK | assid: its liek auto answer on speaker phone on the polycoms |
| 19:09.42 | *** part/#asterisk puowvip (i=ircuser@thegrey.diamond.org) |
| 19:09.51 | Assid | why would anyone want that? |
| 19:10.13 | Ciber311 | i have a client that wants it in his restaurant heh |
| 19:10.25 | Ciber311 | apparently it's very convenient to not have to touch the phone |
| 19:10.58 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.119) |
| 19:11.24 | SwK | assid: you never worked in an office with overhead paging? or in an office where the secretary answers the phone for the exec and "intercoms" him hands free to tell him he has a call? |
| 19:11.37 | SwK | ciber311 it is... especially for paging |
| 19:12.16 | Ciber311 | SwK: have you tried it with a linksys spa 942? |
| 19:12.41 | Assid | well.. you could always call in like a normal phone.. and the exec can answer the call |
| 19:12.56 | SwK | ciber311: nope I dont recommend cheap phones ;) |
| 19:12.57 | Ciber311 | i won't be able to use the polycom's since he needs backlit screens |
| 19:13.14 | Ciber311 | SwK: they basically cost the same as the 501's :P |
| 19:13.17 | SwK | assid: you've never been an exec w/ a secretary have you? |
| 19:13.28 | SwK | yeah but its still a cheap phone |
| 19:13.40 | Ciber311 | have you actually used it? |
| 19:13.49 | SwK | yes |
| 19:13.57 | SwK | I have 2 941's and a stack of 841s |
| 19:13.57 | Ciber311 | what's wrong with it? |
| 19:14.06 | SwK | they are cheap feeling, the speakphone sucks |
| 19:14.16 | Assid | either which way .. i wouldnt want someone speaking out of nowhere .. specially if im busy working or i am in a meet |
| 19:14.27 | SwK | and for a few bucks more I can use a polycom that "just works" |
| 19:14.39 | *** join/#asterisk AsteriskAlbania (i=Asterisk@80.91.113.253) |
| 19:14.51 | Ciber311 | but the polycom's have some annoying issues |
| 19:14.54 | SwK | assid: well secretaries usually know what their execs are doing and when they dont want to be disturbed |
| 19:14.57 | SwK | like? |
| 19:14.59 | Ciber311 | mostly the no backlit display |
| 19:15.05 | Ciber311 | and no leds |
| 19:15.15 | *** join/#asterisk Nivex (n=kjotte@user-0c8hq6g.cable.mindspring.com) |
| 19:15.15 | SwK | you dont need LEDs |
| 19:15.22 | SwK | backlit display I will agreewith |
| 19:15.32 | Ciber311 | that damn screen is a pain to notice thins |
| 19:15.34 | Ciber311 | things |
| 19:15.45 | Assid | anwyays.. how do you do it.. i dont mind learning anwyasy |
| 19:16.35 | SwK | assid: check out voip-info theres a page on autoanswer with polycoms |
| 19:16.41 | AsteriskAlbania | The quality of voice has fallen totally at the asterisk , and the grandstream get frozen during the call and without a power cycle it never thaws, what can go so terriblly wrong , any help is appriciated |
| 19:16.44 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 19:16.44 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 19:16.46 | tld | So far I've mostly been using VoIPJet for US termination, but at 0.013 it's not extremely cheap, and I want another route, either as new primary, or as backup. Any good suggestions? |
| 19:16.53 | SwK | and just keep in mind that post 1.2 its _ALTER_TYPE not ALTER_TYPE as the asterisk channel variable |
| 19:17.07 | SwK | tld: ho many minutes? |
| 19:17.17 | Ciber311 | tld: try http://www.voxee.com/ ? |
| 19:17.26 | tld | SwK: Not too many, it's a community thing. |
| 19:17.31 | SwK | .013 is not that bad a rate |
| 19:17.33 | tld | Ciber311: thanks |
| 19:17.41 | SwK | unless you are getting a stack of minutes |
| 19:17.44 | Assid | i thought voxee was having some routing issues? |
| 19:17.58 | Ciber311 | no idea |
| 19:18.12 | Ciber311 | i do need a reliable sip/iax provider though |
| 19:18.20 | tld | SwK: .013 isn't bad, but doesn't seem great either. Then again, I'm Norwegian, and not too up to date on US teleco costs. |
| 19:18.21 | Ciber311 | business level quality |
| 19:18.24 | AsteriskAlbania | The quality of voice has fallen totally at the asterisk , and the grandstream get frozen during the call and without a power cycle it never thaws, what can go so terriblly wrong , any help is appriciated |
| 19:18.51 | SwK | tld: if you want cheaper then that you'll have to be talking 1Mil minutes |
| 19:18.58 | SwK | 0.02USD is average |
| 19:19.03 | SwK | for good termination |
| 19:19.04 | tld | SwK: I can live with .013. |
| 19:19.21 | Assid | move providers have LCR routes |
| 19:19.26 | tld | Just seemed steep as it's what you'd normally pay the incumbent teleco in Norway 10 years ago. |
| 19:19.30 | Assid | which start at .005 or so and onwards |
| 19:19.37 | tld | probably me just not being up to date. |
| 19:19.41 | Assid | voicepulse, plainvoip, etc give that |
| 19:19.52 | SwK | voicepulse sucks |
| 19:20.02 | Assid | hehe.. quality issies with you too? |
| 19:20.03 | tld | LCR is "non cooked", so you pay different rates for different numbers? |
| 19:20.07 | SwK | yes |
| 19:20.16 | Assid | incomin/outgoing issues? |
| 19:20.29 | SwK | incoming fails all the time with no explanation |
| 19:20.40 | SwK | the MOS is crap |
| 19:20.44 | Assid | nah for me.. call quality is like wayy whacked |
| 19:20.48 | Assid | MOS ??? |
| 19:20.55 | SwK | call audio quality |
| 19:21.09 | tld | If I rate quality over price, and want a good quality, stable provider, supporting SIP termination, any names that jump out? |
| 19:21.21 | tld | Assid: Mean Opinion Score |
| 19:21.36 | Assid | you know any other originator ? |
| 19:21.43 | Assid | with 4 channels? |
| 19:21.45 | SwK | assid: I work for several |
| 19:22.20 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 19:23.28 | Assid | any of them any good? |
| 19:23.56 | docelmo | try www.plainvoip.com |
| 19:24.09 | Assid | originator |
| 19:24.31 | Assid | not terminator |
| 19:25.16 | SwK | yeah I head plainvoip sucked :) |
| 19:25.28 | SwK | assid: yeah some of them heh |
| 19:25.53 | Assid | okay like? |
| 19:25.58 | SwK | cxmusa.com |
| 19:26.06 | Assid | coz these guys are getting a bit tired of bad quality |
| 19:26.09 | SwK | depends on what you want |
| 19:26.40 | SwK | voicecondiuts seems good, plainvoip is ok (i know people that work at both of those places) |
| 19:29.19 | docelmo | shutup ken |
| 19:29.23 | docelmo | it works just fine thank you |
| 19:29.25 | docelmo | :P |
| 19:30.17 | file | did you forget to give SwK some cash or something? :P |
| 19:30.23 | Assid | acutally.. verizon based network it messes with |
| 19:30.38 | SwK | heh |
| 19:30.45 | SwK | whats up file |
| 19:31.00 | file | not much, relaxin' on a lazy Saturday |
| 19:31.01 | file | you? |
| 19:31.06 | SwK | docelmo: Oh I didnt see you sitting there before I started saying ow bad what you did is ;) |
| 19:31.17 | AsteriskAlbania | The quality of voice has fallen totally at the asterisk , and the grandstream get frozen during the call and without a power cycle it never thaws, what can go so terriblly wrong , any help is appriciated |
| 19:31.33 | SwK | just got home from watching my kid's football team beat the crap outta their opponents this mroning |
| 19:31.41 | file | lol |
| 19:31.46 | SwK | 27-0 |
| 19:32.05 | SwK | any they are only allowed to run up the middle after they are ahead 20 points |
| 19:32.26 | file | 27-0? ha |
| 19:35.58 | docelmo | how bad my DID? bitch I just added 7 area codes and 15 or 20 rate centers |
| 19:36.37 | Ryushin | I figure I'll keep asking here every couple of hours or so, hoping someone knows how to reset the ip430's. 468* doesn't work. |
| 19:39.01 | Assid | docelmo: some issues bro.. |
| 19:39.11 | Assid | people complaining of outage during call |
| 19:39.35 | Assid | like all of a sudden it breaks up |
| 19:40.32 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
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| 19:44.40 | *** part/#asterisk nnp_ (n=nnp@85.134.141.69) |
| 19:45.48 | *** join/#asterisk ComputerWarm (n=dan@209.29.158.199) |
| 19:45.55 | ComputerWarm | Hello |
| 19:46.01 | ComputerWarm | anyone here use A2Billing? |
| 19:49.12 | *** join/#asterisk hfern (n=hfern@h-64-105-50-65.dllatx37.dynamic.covad.net) |
| 19:50.18 | ComputerWarm | anyone at all |
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| 19:56.14 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.169) |
| 19:59.16 | ComputerWarm | does anyone know how to fix this Channel SIP/did.voip.les.net-0821f438 was never answered ? its not even calling |
| 19:59.52 | *** join/#asterisk dhanes (i=dhanes@hubble.switched-it.com) |
| 20:00.34 | *** join/#asterisk ToyMan (n=stuq@pool-70-19-192-209.bos.east.verizon.net) |
| 20:10.22 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 20:21.04 | hfern | hello, can anyone tell me why the pattern '_011' is matched when I dial 01131880000047 when I also have a pattern '_01131ZX.' defined...? |
| 20:21.05 | *** join/#asterisk Here_And_There (n=Here_And@pool-68-238-244-21.phlapa.fios.verizon.net) |
| 20:21.51 | hfern | if I remove the '_011.' pattern then it does match the '_01131ZX.' pattern |
| 20:22.16 | hfern | i thought that the most specific pattern is always matched first |
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| 20:50.43 | *** join/#asterisk rene- (n=rene1@gea-gye-internet.telconet.net) |
| 20:50.50 | rene- | hey |
| 20:51.59 | rene- | i am seeing my D-channel going up and down each 5 and a half hours for around 25 minutes. the telco suggested looping back asterisk and looping back the E1.. to see if the d-channel goes down or what.. i would like to hear other people opinons on this 'test' the carrier wants me to do |
| 20:53.27 | *** join/#asterisk Strom_M (n=strom@dsl001-174-194.lax1.dsl.speakeasy.net) |
| 20:58.08 | *** join/#asterisk Merlin2006 (n=me@host-87-74-5-98.bulldogdsl.com) |
| 20:58.19 | Merlin2006 | hello all |
| 20:58.54 | Merlin2006 | just wondering if there is anyone in here that could help me out with the install of a b410p card? |
| 21:07.12 | tzafrir_laptop | Merlin2006, not me. But try asking a more specific Q |
| 21:13.11 | Merlin2006 | fair enough... i've installed trixbox and the mISDN-b410p adn the mISDN-user drivers and i believe the card has come to life. i currently hav eone port blugged into a trend so that i can monitor the line without upsetting bt with constant restarts. Anyway, the trend is reporting that layer 1 is up and layer 2 is down. im just not sure where to go from here, there is no mISDN.conf in /etc/asterisk/ and im not sure iv got zaptel.conf configu |
| 21:13.30 | Merlin2006 | please excuse the appaling spelling :) |
| 21:17.09 | *** join/#asterisk marv (n=ilovekim@c-71-228-189-127.hsd1.al.comcast.net) |
| 21:19.20 | *** join/#asterisk MarkMoss (n=mmoss@65.183.175.226) |
| 21:22.50 | dflow | Merlin2006: when you do misdn show stacks, you have L2Link UP / L1Link UP ? |
| 21:23.29 | Merlin2006 | 1 sec |
| 21:23.37 | *** join/#asterisk mocker (n=ks@mocker.org) |
| 21:23.50 | mocker | Is anyone having problems dialing iaxtel numbers? |
| 21:25.22 | mocker | For some reason I'm getting a 'Call Rejected by 69.73.19.178: No such context/extension |
| 21:25.34 | file | what are you trying to dial? |
| 21:26.07 | mocker | The digium number for a test. ;) |
| 21:26.14 | mocker | 700.428.6000 |
| 21:26.31 | file | you have to put a 1 in front |
| 21:26.36 | mocker | Right... |
| 21:26.47 | mocker | I was just pasting from their site. |
| 21:26.48 | file | okay, try to dial right now |
| 21:26.52 | Merlin2006 | dflow, im getting a command not found error. but my trend is showing layer 1 up and layer 2 down |
| 21:28.31 | mocker | file: Just did.. |
| 21:28.33 | mocker | Same result. |
| 21:28.40 | file | I see nothing on IAXtel |
| 21:28.44 | file | besides tons of registrations |
| 21:28.49 | mocker | Hmm. |
| 21:28.55 | file | what host are you using? |
| 21:29.44 | mocker | Just iaxtel.com? |
| 21:29.56 | mocker | That's what the 69.73.19.178 resolves to. |
| 21:29.59 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 21:30.01 | *** join/#asterisk kusznir (n=kusznir@66-233-98-93.lew.clearwire-dns.net) |
| 21:30.13 | file | yeah I saw nothing about a call attempt |
| 21:30.24 | mocker | Huh. |
| 21:30.58 | mocker | file: More than likely something on my end then. :) |
| 21:31.08 | file | and I just tested it |
| 21:31.34 | file | quite funky |
| 21:31.49 | mocker | This is a new setup, so I probably have just left something out. |
| 21:31.55 | mocker | Especially if you can call it. |
| 21:32.12 | mocker | I was just thrown by No such context/extension from that IP. |
| 21:32.15 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 21:32.16 | file | did you forget to specify a context in the dial line? |
| 21:32.49 | file | Dial(guest@iaxtel.com/17004286000@iaxtel) |
| 21:33.50 | mocker | exten => _1700XXXXXXX,1,Dial(IAX2/myuser:mypass@iaxtel.com/${EXTEN:1}@iaxtel) |
| 21:33.59 | mocker | That's my dial line.. |
| 21:34.20 | Strom_M | why are you stripping the 1 off the front of the number? |
| 21:34.22 | file | it should be ${EXTEN} |
| 21:34.26 | file | yeah, don't strip the 1 off |
| 21:34.39 | mocker | Oh, I have to pass the one.. |
| 21:34.44 | mocker | That would be my problem. |
| 21:34.45 | file | yes |
| 21:34.49 | mocker | Thanks. |
| 21:35.18 | *** join/#asterisk tsurk0 (n=tsurko@digsys226-159.pip.digsys.bg) |
| 21:36.32 | *** join/#asterisk j0 (i=dan@CABLE-72-53-45-212.cia.com) |
| 21:36.40 | mocker | file: Works like a charm now. |
| 21:36.56 | file | excellent |
| 21:38.00 | Strom_M | hmm, is there a way to get this polycom phone to pay attention to tftp information from the dhcp server? |
| 21:42.47 | Strom_M | er, nevermind |
| 21:47.48 | *** join/#asterisk redondos_ (n=redondos@190.48.8.124) |
| 21:48.17 | Ryushin | I figure I'll keep asking here every couple of hours or so, hoping someone knows how to reset the ip430's. 468* doesn't work. |
| 21:48.32 | Strom_M | Ryushin, you reset it through the menu |
| 21:49.09 | Ryushin | There is a setting in the menu to format the file system, but nothing to reset back to factory config. |
| 21:49.19 | Ryushin | Plus the phone won't boot at this point. |
| 21:49.42 | Ryushin | It complains about not finding the application. The IP601 boots fine. |
| 21:50.12 | Ryushin | Strom_M: My dhcp works fine for the polycom phones. |
| 21:50.23 | Ryushin | I use vstfp and it pulls it from that. |
| 21:50.40 | Ryushin | I can give you a copy of my dhcpd.conf file. |
| 21:51.02 | Strom_M | ive got the dhcpd figured out now, hence the "nevermind" |
| 21:51.52 | Ryushin | Strom_M: Do you have any ip403 phones? Which firmware and sip are you using? |
| 21:52.01 | Strom_M | i have an ip430 at home |
| 21:52.06 | Strom_M | but im not at home right now |
| 21:52.10 | Ryushin | k |
| 21:53.10 | Strom_M | right now I'm attempting to figure out why this stupid 301 won't read the config file I have sitting in /tftpboot for it |
| 21:53.19 | Strom_M | atftpd is serving the file |
| 21:57.41 | *** join/#asterisk sb_mx (n=sb_mx@189.135.75.127) |
| 21:59.28 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 21:59.57 | Strom_M | Ryushin, why do you want to reset the phone back to factory config? |
| 22:00.28 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 22:00.48 | *** join/#asterisk Altair256 (n=Altair25@mail.clccorp.com) |
| 22:01.15 | dflow | Merlin2006: uh, you have chan_misdn compiled and loaded to asterisk? |
| 22:01.43 | dflow | something like this : |
| 22:01.45 | dflow | erlang*CLI> show modules like misdn |
| 22:01.53 | dflow | chan_misdn.so Channel driver for mISDN Support (Bri/Pr 0 |
| 22:03.38 | Merlin2006 | hmm, no |
| 22:03.45 | Merlin2006 | ill look int o doing that now |
| 22:03.48 | Merlin2006 | thanks |
| 22:04.48 | *** join/#asterisk scubaSteve (n=steve@ns1.misel.com) |
| 22:05.05 | scubaSteve | Anyone have docs for the VoicePulse API? |
| 22:05.10 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 22:05.21 | TripleFFFF | wahts callerid specs ? name = 14 chars .. and number ? |
| 22:06.11 | *** part/#asterisk scubaSteve (n=steve@ns1.misel.com) |
| 22:07.00 | Merlin2006 | 6 chars in uk typically |
| 22:07.43 | TripleFFFF | ?? |
| 22:07.49 | TripleFFFF | well.. number cant be 6 numbers |
| 22:07.49 | TripleFFFF | lol |
| 22:07.56 | TripleFFFF | north ameria = 10 digits |
| 22:08.00 | TripleFFFF | so europe > then that |
| 22:10.12 | Strom_M | god, i hope i havent spent a half hour yelling at a phone because i had my permissions set wrong :) |
| 22:10.47 | Strom_M | hahahahahaha, it was a permissions issue |
| 22:10.56 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 22:12.22 | Strom_M | actually, total nub^3 |
| 22:17.00 | j0 | i have plenty of bandwidth, but still have bad outgoing voice quality.. it seems that my packet loss of 5% outbound is causing problems.. using different codecs like gsm or ulaq arn't helping at all... is there any way i can make it more resilient to packet loss? |
| 22:17.03 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 22:17.32 | j0 | maybe i shouldn't say packet loss... testyourvoip.com says "packet discards: 5%, packet loss: 0.4%" .. whatever that means |
| 22:27.26 | Strom_M | j0, what does your path to the network look like? |
| 22:29.52 | ellisdee | sound like bad routes |
| 22:30.27 | ellisdee | perform a couple of tracert's from the pbx |
| 22:30.39 | ellisdee | see what your latency, and pl look like going out |
| 22:31.03 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 22:31.03 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:31.07 | ellisdee | or to another server on the lane |
| 22:31.08 | ellisdee | lan |
| 22:31.25 | j0 | ellisdee, Strom_M: 17 hops.. about 100ms |
| 22:31.52 | Strom_M | jitter? loss? |
| 22:32.01 | ellisdee | yea what strom asked |
| 22:32.10 | ellisdee | like 20ms then a sudden 100ms spike? |
| 22:32.22 | j0 | how do i measure jitter accurately? |
| 22:32.32 | j0 | whats that program that runs a continuous traceroute |
| 22:32.39 | Strom_M | mtr |
| 22:32.50 | j0 | thank goodness.. been trying to remember that all week :) |
| 22:33.02 | ellisdee | tracert |
| 22:33.22 | ellisdee | or mtr |
| 22:33.27 | Strom_M | jitter is also measured well if you run a continuous ping for about 100-300 pings, and then report the variation in round trip times back to us |
| 22:33.46 | j0 | some hops cycle between 80 and then 120ms |
| 22:33.55 | *** join/#asterisk hohum (n=dcorbe@jomama.interceltelecoms.net) |
| 22:33.56 | j0 | k |
| 22:33.56 | Strom_M | 40ms variation?! |
| 22:33.58 | Strom_M | ouch |
| 22:34.10 | hohum | 40ms variation?! |
| 22:34.11 | hohum | ouch |
| 22:34.23 | ellisdee | shouldn't bee too bad |
| 22:34.28 | ellisdee | if packet loss is minimal |
| 22:34.33 | ellisdee | but the variation is 40ms |
| 22:34.41 | ellisdee | the quality should not be effected too bad |
| 22:34.50 | ellisdee | or if there is no packet loss |
| 22:35.02 | j0 | ah.. now its varying much more.. like 100ms every few seconds |
| 22:35.17 | ellisdee | ugh |
| 22:35.20 | Strom_M | yech |
| 22:35.23 | j0 | from 30ms to 130ms |
| 22:35.31 | Strom_M | ooh, that's icky |
| 22:35.41 | ellisdee | what kind of provider you dealing with? |
| 22:35.42 | j0 | and this is still within my isp |
| 22:35.51 | j0 | 3web, through shaw cable in canada |
| 22:36.04 | ellisdee | can you paste to me |
| 22:36.05 | ellisdee | the tracert |
| 22:36.13 | ellisdee | only first 6 hops |
| 22:36.16 | j0 | those are the bastards trying to charge $10 voip tax for qos |
| 22:36.57 | ellisdee | 13.567 ms 62.541 ms 18.884 ms |
| 22:36.58 | ellisdee | damn |
| 22:37.00 | ellisdee | =\ |
| 22:37.04 | ellisdee | thats not looking good man |
| 22:37.09 | j0 | new isp? |
| 22:37.17 | ellisdee | performance like this occur quite often? |
| 22:37.24 | ellisdee | have you attempted to contact them in regards to this? |
| 22:37.48 | j0 | "i'm sorry, we cannot connect your call, all reps. are busy with OTHER customers, goodbye <click>" |
| 22:37.55 | j0 | wth kinda isp hangs up on you when their lines are "full"? |
| 22:38.01 | ellisdee | lol |
| 22:38.19 | j0 | or.. at the 15min waiting mark.. "i'm sorry were busy, goodbye" |
| 22:38.20 | ellisdee | i'd submit tickets via email |
| 22:38.32 | j0 | yeah i'm trying to get a response that way.. either that or switch isps |
| 22:38.33 | ellisdee | and continue to contact them through telephone. |
| 22:38.40 | ellisdee | do your homework |
| 22:38.41 | j0 | so this is *clearly* why my voip is crap? |
| 22:38.57 | Strom_M | yup |
| 22:39.07 | j0 | alright.. gonna get a quote from the competition |
| 22:39.18 | Strom_M | telus dsl ftw |
| 22:39.31 | Merlin2006 | dflow: am i right in thnking that the zaptel config is going to have notihgn to do with this setup? |
| 22:39.32 | ellisdee | looks good |
| 22:39.37 | Strom_M | or agt or bell canada or telebec or whoever your telco is :) |
| 22:39.46 | j0 | yeah, i'm gonna try telus |
| 22:39.56 | j0 | even though their throughput is significantly lower than shaw/3web |
| 22:40.05 | j0 | heck i'll just use both isps |
| 22:41.31 | Strom_M | "I walked across the street to Inglewood" sounds more like the title of a song than what I had to do to get a sandwich ten minutes ago |
| 22:41.51 | dflow | Merlin2006: sure , misdn.conf is what you are looking for |
| 22:42.04 | Merlin2006 | yeah |
| 22:42.06 | jbalcomb | Anyone using a web interface to the asterisk Record command? |
| 22:42.10 | *** join/#asterisk duderock (n=natrix@host-216-153-249-214.mil.choiceone.net) |
| 22:42.21 | Merlin2006 | iv just broken asterisk installing the misdn drivers :( |
| 22:42.37 | Strom_M | Merlin2006, step 1 is to not use trixbox |
| 22:42.46 | Merlin2006 | lol |
| 22:42.50 | Merlin2006 | im thinking that |
| 22:43.10 | duderock | check out the error at the end of this log - does it have anything to do with zaptel modules? (no cards installed, cannot modprobe 'ztdummy') http://paste.ubuntu-nl-org/22365 |
| 22:43.14 | jbalcomb | I'd like to let my account reps record the queue announcement, the periodic message, busy, unavail, and temp greetings for each of thier clients |
| 22:43.58 | Strom_M | duderock, that url doesn't make sense |
| 22:44.10 | Merlin2006 | asterisk falls over whils booting chan_misdn.so i mustve done soethign wrong inthe misdn.conf |
| 22:44.24 | duderock | oops |
| 22:44.32 | duderock | paste.ubuntu-nl.org/22365 |
| 22:44.46 | kusznir | Hi all: I'm trying to get trixbox working in my config, but am having difficulty with asterisk, and I don't know how to turn on any further debugging info. Currently, asterisk is rejecting any incomming iax-trunked calls (such as through iax.fwdnet.net). |
| 22:45.17 | kusznir | wireshark shows "no authority found", but the asterisk console (at verbose 7) shows no activity whatsoever. |
| 22:45.34 | Strom_M | kusznir, step 1 is to not use trixbox |
| 22:45.37 | j0 | alright.. i'm visiting someones house with telus.. heh |
| 22:46.00 | Strom_M | j0, hehe |
| 22:46.17 | Merlin2006 | lol, Strom_M, u really dont like trixbox do you? |
| 22:46.25 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 22:46.28 | Strom_M | it's junk |
| 22:46.37 | kusznir | I don't know if this is related or a seperate issue, but trying to make outbound calls through that same trunk: asterisk console shows the expected Dial() line followed immediately by "all parties are busy". Network dumps show no traffic on iax. |
| 22:46.39 | duderock | I agree with Strom - CentOS uses a ridiculously old Kernel... |
| 22:46.46 | Merlin2006 | hmm |
| 22:46.57 | kusznir | Storm_M duderock: what would you recommend? |
| 22:47.17 | duderock | Well, don't ask me quite yet... |
| 22:47.18 | Merlin2006 | i tried installing asterisk onto fc5 last night, but didnt get this far.... |
| 22:47.21 | Strom_M | kusznir, my asterisk installs are all asterisk svn release branch running on debian sarge |
| 22:47.30 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 22:47.33 | kusznir | (My old system was asterisk 1.0.8 on OpenWRT / Linksys WRT54GS...its not doing so hot..... ) |
| 22:47.36 | duderock | I'm running * with Freepbx on Ubuntu Server |
| 22:47.49 | duderock | (from asterisk SVN) |
| 22:48.12 | kusznir | Is freepbx a decent config/management frontend? |
| 22:48.19 | Strom_M | freepbx is also junk |
| 22:48.24 | kusznir | (or is there something else I should be using) |
| 22:48.31 | Strom_M | you should be using vi :) |
| 22:48.40 | dflow | yeah vi r0x |
| 22:48.49 | duderock | but I'm having issues right now... (see http://paste.ubuntu-nl.org/22365 |
| 22:49.01 | kusznir | I tried the "vi the files" method, but don't seem to know enough about how to set stuff up...I was spending hours configuring it to get a very basic system..... |
| 22:49.03 | duderock | I agree, vi rox, but end users need GUIs... |
| 22:49.16 | duderock | Really the configs are very straight forward. |
| 22:49.20 | Strom_M | kusznir, you should read the book |
| 22:49.21 | Strom_M | ~book |
| 22:49.28 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:49.29 | kusznir | (I administer a bunch of servers, and prefer vi for them :) ) |
| 22:49.31 | duderock | kusznir - get the oreilly book |
| 22:49.38 | Strom_M | duderock, anyone who needs a GUI has no business administering a production PBX |
| 22:49.42 | Merlin2006 | that book really is the nuts |
| 22:49.53 | Merlin2006 | but i lef tit at the fofice :( |
| 22:50.00 | Strom_M | Merlin2006, download it for freeeeeeeeeeee |
| 22:50.04 | Strom_M | see above url |
| 22:50.11 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 22:50.14 | ellisdee | <Strom_M> kusznir, step 1 is to not use trixbox |
| 22:50.15 | ellisdee | lol |
| 22:50.19 | Merlin2006 | thanking you kindly |
| 22:50.22 | hmmhesays | words are all I have so I write them |
| 22:50.31 | duderock | Strom_M: any ideas on that logfile? |
| 22:51.07 | Merlin2006 | i still think iim gonna crack on and try and get this workign witheh trixbox install im halfway thru |
| 22:51.30 | Strom_M | duderock, i have no clue what the hell you pasted |
| 22:51.53 | kusznir | One more question: Any suggestions/recommendations on running asterisk in a VM? (I don't need any zap interfaces, purely IAX and SIP) |
| 22:52.16 | duderock | It's a complete boot cycle, followed by the errors at the end just before locking up.. |
| 22:52.25 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 22:52.25 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:52.31 | dflow | <PROTECTED> |
| 22:53.00 | duderock | The last 25 or so lines are the problem, but I thought it might be helpful to see the startup messages, in case it's hardware. |
| 22:53.00 | kusznir | (I don't have a physical box to dedicate to it, nor do I want to pay for power for a dedicated box to just run 2 phones :). I have a myth system that's up 24/7, and I'd like to just add asterisk to it, but protect its installs from myth upgrades) |
| 22:53.16 | Strom_M | kusznir: trixbox...freepbx....VMs....you just like pain, don't you? |
| 22:53.17 | Merlin2006 | Strom_M: i love you for that link, i was lost without my bible, lol |
| 22:53.24 | Strom_M | :) |
| 22:54.02 | *** join/#asterisk casterman (n=Administ@kasadsyed.net8.nerim.net) |
| 22:54.17 | dflow | hehe |
| 22:54.50 | kusznir | Strom_M: Yea...well, I'm a linux sysadmin :) (run ~200 systems + servers in a university dept with only 20hrs/week and no other help....pain is a way of life now.....) |
| 22:55.24 | Altair256 | I realize this isn't the place for it, but I have a Trixbox update script question. Anyone knowledgeable and willing to answer a quick question?\ |
| 22:56.17 | dflow | Altair256: As Storm_M said , step 1 is to not use trixbox |
| 22:56.35 | Merlin2006 | ahhh yes, i remember now, theres bugger all in tehre about bri configing |
| 22:56.58 | Altair256 | thanks for the help, dflow. Anyone else? |
| 22:58.44 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:01.48 | Strom_M | why does everyone keep calling me Storm? |
| 23:01.55 | Strom_M | I'm not a weather system |
| 23:03.30 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 23:04.08 | dflow | Strom_M: sorry :>> |
| 23:08.05 | shodan-away | I'm getting "unable to request echo training on channel 1" when I try to make an outbound call with a x100p , I set echotraining=4000 in my zapata.conf and I'm using mark2 echo canceller with the more aggressive option |
| 23:08.56 | pyrom | any "cheap" usb isdn "adapters" which works with asterisk, as a isdn "terminal" e.g to connect a isdn phone. |
| 23:16.04 | mocker | Can anyone recommend a good ITSP with E911 support? Or a good place to find reviews? |
| 23:18.20 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:18.20 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:19.29 | rene- | i am seeing my D-channel going up and down each 5 and a half hours for around 25 minutes. the telco suggested looping back asterisk and looping back the E1.. to see if the d-channel goes down or what.. i would like to hear other people opinons on this 'test' the carrier wants me to do |
| 23:27.56 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:38.33 | pyrom | <PROTECTED> |
| 23:42.40 | *** part/#asterisk Merlin2006 (n=me@host-87-74-5-98.bulldogdsl.com) |
| 23:42.47 | *** join/#asterisk Merlin2006 (n=me@host-87-74-5-98.bulldogdsl.com) |
| 23:42.52 | *** join/#asterisk dusan2 (i=dusan@209-223-47-160-static.oplink.net) |
| 23:45.36 | rene- | <PROTECTED> |
| 23:48.33 | *** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
| 23:49.24 | jbalcomb | I want to take a variable from a web page and make it available to my Asterisk dial plan |
| 23:49.46 | Qwell | jbalcomb: func_curl |
| 23:50.00 | jbalcomb | does it make more sense to have my PHP code use the AMI to set the variable is the astdb or use func_odbc to have my dial plan fetch is from MySQL? |
| 23:52.15 | *** join/#asterisk chode (n=chode@p54B012FC.dip0.t-ipconnect.de) |
| 23:53.00 | docelmo | say can someone point me in the direction of how to setup a dialplan that can answer alpha numeric extensions? for instance something like exten => brian-100,1,Dial.... |
| 23:53.35 | Qwell | docelmo: That'd work |
| 23:54.13 | docelmo | ok but what about doing matching like if I wanted to match something along the same linex with _X.,1,Dial.... |
| 23:54.26 | docelmo | I tried that chan_sip keeps kicking back it cant find the exten |
| 23:54.36 | Qwell | well, X only matches 0-9 |
| 23:54.56 | Qwell | though, you could probably do like |
| 23:55.01 | docelmo | ya I know.. is there something I can use to match anything including alpha? |
| 23:55.03 | Qwell | _[0-9a-z]. |
| 23:55.07 | jbalcomb | Qwell: developer documentation is beyond me. What does that function do with the URL you pass it? |
| 23:55.09 | Qwell | I'm pretty sure that works |
| 23:55.14 | Qwell | jbalcomb: whatever you want |
| 23:55.24 | docelmo | thanks Q I will give that a whirl |
| 23:55.26 | Qwell | jbalcomb: use the function as a variable if you want |
| 23:55.51 | jbalcomb | Qwell: is it just looking at the URL or is it parsing the page the url refers to? |
| 23:56.13 | Qwell | jbalcomb: it returns the page, just like curl would |
| 23:57.17 | Qwell | webpages don't HAVE to be html |
| 23:57.26 | jbalcomb | hrmm.. ok, ive never heard of curl. i don't if thats quite what i'm looking for. i guess i shouldn't have used 'take'. |
| 23:57.33 | Merlin2006 | should i be expecting the level 2 to come up on this card with the connection to asterisk? ie level 1 will come up as long as mISDN modules are laded? |
| 23:58.31 | jbalcomb | Qwell i'm thinking more like "what is your exten?" <4416> [submit]; dial($exten submitted from web page); |
| 23:59.30 | jbalcomb | does it make more sense to have my PHP code use the AMI to set the variable in the astdb or use func_odbc to have my dial plan fetch it from MySQL? |