| 00:00.55 | asterisk99 | twisla: BUDDA BING --- BUDDA BOOM !!! Tony Soprano is happy --- Asterisk compiled all the way thru |
| 00:01.09 | twisla | :) |
| 00:03.26 | *** join/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
| 00:03.39 | fugitivo | jenwa: i do |
| 00:03.52 | [av]bani | fugitivo uses asterisk in a boiler room |
| 00:04.06 | [av]bani | 1000 illegal immigrants selling herbal viagra |
| 00:04.16 | fugitivo | shhhh |
| 00:04.20 | [av]bani | powered by open source(tm) |
| 00:05.25 | fugitivo | that's not asterisk, that's the spam server |
| 00:07.17 | *** join/#asterisk octothorpe_ (n=octothor@c-67-186-207-234.hsd1.ut.comcast.net) |
| 00:07.56 | jenwa | fugutivo: how big is your call center? |
| 00:08.53 | terrapen | I wonder how Asterisk will run on a SGI Onyx |
| 00:09.14 | jenwa | os/2 |
| 00:09.16 | jenwa | ;-) |
| 00:09.37 | terrapen | screw that. I want to point to a massive, refridgerator-sized box and say, "That's my PBX." |
| 00:09.47 | sevard | Question: I have a MSQL problem. We have the exact same information in the mysql database as in the sip.conf but it seems to break when we use the database. |
| 00:09.58 | terrapen | 16 CPUs or something like that |
| 00:10.10 | Qwell[] | terrapen: Talk to Netgeeks |
| 00:10.16 | *** join/#asterisk forao (n=fasdfasd@pool-138-89-152-184.mad.east.verizon.net) |
| 00:10.20 | Qwell[] | he's been playing with like an E4500 |
| 00:10.40 | jenwa | lol |
| 00:10.50 | asterisk99 | terrapen: Just install Asterisk on IBM System/390 ... It'll easily run 400 Linux Virtual Machines (It's about the size of a refrigerator) |
| 00:11.12 | asterisk99 | terrapen: You could have 400 Asterisks running on one box!!! |
| 00:13.47 | terrapen | sweet |
| 00:13.52 | terrapen | but anyone can run linux |
| 00:13.58 | terrapen | not everone can run IRIX :P |
| 00:15.36 | terrapen | maybe VMS would be a better bet |
| 00:17.44 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
| 00:23.47 | [av]bani | terrapen: hp/ux |
| 00:23.55 | *** join/#asterisk jamalot (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 00:24.34 | jamalot | Hmm... anybody using AsteriskWin32? |
| 00:25.03 | [av]bani | actually my favorite os of all time is ultrix |
| 00:27.09 | Abydos313 | jamalot bite your tongue..heh |
| 00:30.46 | jamalot | Abydos313: Haha I know... I'm playing with it until I can get my old Linux box out of storage. |
| 00:31.26 | *** join/#asterisk jhnjwng (n=wj1918@pool-70-21-166-133.nwrk.east.verizon.net) |
| 00:31.49 | Abydos313 | how is it, i landed on the program page yesterday and actually was temped to try it |
| 00:31.50 | jamalot | I'm trying to figure out why when * tries to dial out on my VOIP provider, it doesn't transmit any codec capabilites (rather just a "telephone-event". |
| 00:32.00 | *** join/#asterisk |omni| (i=cathode@c-67-185-96-86.hsd1.wa.comcast.net) |
| 00:32.33 | jamalot | So I get a 488 "Not Acceptable Here" msg. |
| 00:32.43 | Abydos313 | no idea |
| 00:34.45 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 00:37.50 | [av]bani | jamalot: presumably because when you registered, they rejected all the codecs you offered |
| 00:42.06 | jamalot | av: Hmm... I've tried a tonne of different allow/disallow combos... but to no avail. I'm using AXVoice... found a config on Nerd Vittles, but it's not quite working out for me. |
| 00:43.09 | *** join/#asterisk reallost1 (n=reallost@12-215-208-184.client.mchsi.com) |
| 00:47.39 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
| 00:50.36 | *** join/#asterisk alephcom (n=darren@host75.net14.mcsnet.ca) |
| 00:52.18 | octothorpe_ | jamalot: pastebin your cli output |
| 00:52.23 | octothorpe_ | ~pb |
| 00:52.24 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 00:52.25 | xtrvd | Does anybody know of a Dialplan validator that could be lingering around the internet? |
| 00:56.49 | Zodiacal | anyone know of an option some where that will make asterisk never say the extention to the pstn user? |
| 00:57.00 | Zodiacal | i remember hearing it once a few weeks ago and i wanta kill it |
| 00:57.27 | Zodiacal | would voicemail(Uext) do it? |
| 00:57.33 | Zodiacal | and then place a playback() in front of that? |
| 00:58.02 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:00.36 | x86 | anyone use SJPhone or another SIP phone with URI support? |
| 01:00.53 | x86 | i want to test to see if i can get incoming SIP calls correctly |
| 01:01.04 | x86 | someone please try sip:bryce@shellshark.net |
| 01:03.41 | fugitivo | if it works, i don't have audio here |
| 01:03.46 | [av]bani | x86: any hot ph0n3 s3xx0r yet? |
| 01:04.13 | fugitivo | did it work? |
| 01:04.34 | fugitivo | wait, i'm not registered to my asterisk, wtf |
| 01:04.37 | fugitivo | :/ |
| 01:04.44 | fugitivo | that's not my asterisk! |
| 01:05.00 | fugitivo | i remember, it's a customer's asterisk |
| 01:05.34 | x86 | fugitivo: can you try direct without going through your local asterisk? |
| 01:06.00 | fugitivo | i think it was working that way |
| 01:10.47 | *** join/#asterisk Netgeeks (n=chris@68-185-24-8.static.mdfd.or.charter.com) |
| 01:18.02 | Mavvie | iax is dumb. |
| 01:18.09 | Mavvie | I say: you have ulaw and you have alaw. |
| 01:18.10 | |omni| | iax is neat. |
| 01:18.18 | Mavvie | it says: no compatible codec found. |
| 01:18.33 | Mavvie | I say: bandwidth=high and you have ulaw and you have alaw. |
| 01:18.50 | Mavvie | it says: oh look! is that what you mean? I take alaw. |
| 01:19.05 | Mavvie | I don't mind that it takes gsm when the option is there. |
| 01:19.15 | Mavvie | but if the only options are ulaw or alaw, then !()@#*)(!@*#) pick one! |
| 01:20.39 | *** join/#asterisk jamalot (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 01:20.41 | SibRphrek | anyone know how to make an extension directly dial multiple extentions voicemails? |
| 01:20.47 | SibRphrek | like i wanna leave a group voicemail for everyone |
| 01:21.54 | x86 | Mavvie: iirc, a-law wont work with bandwidth=low |
| 01:22.08 | Mavvie | x86: that's right. |
| 01:22.21 | x86 | Mavvie: i'm fairly sure that only GSM and G.729 work with bandwidth=low |
| 01:22.48 | x86 | even if you force allow=ulaw, etc... bandwidth=low will trump that and only allow certain codecs still |
| 01:23.09 | x86 | so just dont specify bandwidth=, or always put it bandwidth=high, and you'll be fine ;) |
| 01:23.24 | Mavvie | x86: I don't care if it preferes the lowest rate codec, I care that it rejects the higher ones. |
| 01:23.38 | jamalot | Now that I got my codec problem solved... how do I alter how often * registers with my VOIP provider? |
| 01:23.50 | x86 | bandwidth=high disallow=all allow=ulaw allow=alaw allow=gsm |
| 01:23.55 | x86 | then you'll be fine ;) |
| 01:25.56 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 01:26.17 | mphill | are there any other asterisk management software packsages besides AMP/freepbx? (vi is not an answer) |
| 01:26.46 | Faithful | anyone got experience with gnokii? |
| 01:27.07 | *** join/#asterisk GeneG (n=GeneG@toronto-HSE-ppp4164687.sympatico.ca) |
| 01:28.32 | x86 | mphill: notepad |
| 01:28.39 | GeneG | Hello all. I just added a few lines of code to app_voicemail.so so that voice-mail attachments are sent as MP3 files (encoded using Lame) rather than WAV attachments. I'm a little clueless as to the current state of the Asterisk art, so to speak. Is this patch something that others might find useful or does it already exist in recent versions of the system? |
| 01:29.31 | [av]bani | GeneG: afaik you can just run a script after voicemail exits which compresses any wav to mp3 |
| 01:29.41 | [av]bani | easier than hacking up app_voicemail :) |
| 01:30.32 | GeneG | Bani, how would you do that through the dialplan? As far as I can tell the entire record-e-mail cycle is handled by app_voicemail. |
| 01:36.33 | brockj49464 | anybody know if there is problems with * matching up to the correct peer when multiple accounts to the same provider? |
| 01:36.39 | bkw_ | yes |
| 01:36.40 | bkw_ | it can't |
| 01:36.44 | bkw_ | its retarded |
| 01:38.36 | brockj49464 | any work arounds? |
| 01:39.06 | *** join/#asterisk kemp (n=10330405@61.183.76.92) |
| 01:42.11 | *** join/#asterisk |omni| (i=rob@c-67-185-96-86.hsd1.wa.comcast.net) |
| 01:42.24 | brockj49464 | only one I have found is insecure=very but that does not put them into the correct channels just allows calls to complete |
| 01:43.53 | Snake-Eyes | mphill, those are the main free ones, there are others like thirdlane and scopserv if want to pay money |
| 01:47.54 | sevard | BAH, I need freaking help with msql+asterisk annybooddy |
| 01:49.48 | brockj49464 | sevard: I know a little of each but not together does that help? |
| 01:50.16 | alephcom | sevard: What's the problem? |
| 01:50.46 | sevard | Question: I have a MSQL problem. We have the exact same information in the mysql database as in the sip.conf but it seems to break when we use the database. |
| 01:51.11 | forao | sevard did you try using postgresql? |
| 01:51.20 | forao | i know this is not a solution, but just wondering if you ever tried |
| 01:51.27 | sevard | not with asterisk |
| 01:51.31 | _Sam-- | ive setup extensions in mysql table and used extconfig to load them before |
| 01:51.34 | sevard | i seriously doubt it's a database problem |
| 01:51.51 | sevard | i actually can't see *anything* wrong |
| 01:51.54 | forao | do you *know* mysql is better to work with asterisk than postgresql? |
| 01:51.59 | sevard | except it just doesn't damned work |
| 01:52.25 | sevard | forao: i don't have a particular affinity to either database, i just use what's installed |
| 01:53.20 | Abydos313 | oracle for rpm based distro's is free :)) xe version limited to 4gb and one tablespace |
| 01:54.08 | [av]bani | _Sam--: yes, but you're evil. most people here are good kindhearted people. |
| 01:54.33 | _Sam-- | how the hell did you know |
| 01:55.23 | sevard | it works fine when i load sip.conf from disk, but fails when i load sip.conf from the database |
| 01:55.30 | sevard | they're both the same data |
| 01:55.41 | sevard | i'm thinking about trying seperate tables next |
| 01:55.58 | Faithful | anyone got a mobile phone working as a GSM gateway? |
| 01:55.59 | sevard | even though that shouldn't be an issue |
| 01:56.18 | _Sam-- | sevard : there are logs someplace saying whats going on. |
| 01:56.33 | sevard | i've looked through the logs, zero errors, zero warnings |
| 01:56.49 | _Sam-- | dunno, worked for me with sip peers |
| 01:57.00 | sevard | the only observable difference is that one is loading from disk, and the other is loading from the database |
| 01:57.11 | sevard | extensions.conf doesn't matter, sip.conf from the db breaks it |
| 01:57.35 | sevard | 488 not allowed with sip.conf from the db |
| 01:57.55 | _Sam-- | i dont know where to tell you to look...when i set mine up it worked the first try |
| 01:58.08 | sevard | were you on redhat |
| 01:58.20 | _Sam-- | debian, using mysql. |
| 01:58.46 | sevard | if only debian didn't use pam. |
| 02:00.07 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 02:01.38 | *** join/#asterisk AsteriskNewbie (n=linux_ba@63.250.96.18) |
| 02:02.34 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 02:02.38 | *** join/#asterisk kratzers (n=kratzers@65.119.216.4) |
| 02:02.39 | AsteriskNewbie | Anyone has experience with TDM2400P?? I'm trying to determine if it fits my n eeds ... |
| 02:03.29 | Qwell | AsteriskNewbie: Do you need ~24 analog ports? |
| 02:03.34 | Qwell | if so, it does.. |
| 02:04.07 | AsteriskNewbie | Hmm ..no .. not 24 ... |
| 02:04.12 | *** join/#asterisk MstlyHrmls (n=mh@66.193.14.132) |
| 02:04.15 | AsteriskNewbie | I have about analog lines ... |
| 02:04.30 | sevard | syntax error: line count missing. |
| 02:04.33 | Qwell | between 5 and 24? |
| 02:04.37 | Qwell | sevard: indeed |
| 02:04.39 | AsteriskNewbie | Yes .. exactly .. |
| 02:04.47 | Qwell | AsteriskNewbie: it should work fine |
| 02:05.04 | AsteriskNewbie | Ok Qwell .. I was clear from the picture though |
| 02:05.12 | AsteriskNewbie | as to how the input would be plugged in .. |
| 02:05.18 | AsteriskNewbie | am currently plugged into a Norstar system .. |
| 02:05.22 | AsteriskNewbie | + |
| 02:06.10 | AsteriskNewbie | Qwell ... u there?? |
| 02:06.20 | Qwell | it's an amphenol connector |
| 02:06.57 | AsteriskNewbie | Ok .. so I guess I'd have to re-terminate my lines then ... |
| 02:07.36 | *** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com) |
| 02:07.37 | _Sam-- | you just plug the amphenol into a the breakout panel |
| 02:07.51 | Qwell | indeed |
| 02:07.58 | *** join/#asterisk bla (n=bla@netblock-66-218-41-231.dslextreme.com) |
| 02:08.49 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 02:08.51 | *** join/#asterisk b0xii (i=b0xii@cpe-70-116-68-157.houston.res.rr.com) |
| 02:08.52 | Ariel_ | hello folks |
| 02:08.57 | [av]bani | amphenol... sounds like an antacid |
| 02:09.05 | AsteriskNewbie | hehehehehe .. I'd say .. |
| 02:09.06 | _Sam-- | well, in his case i guess you would need 2 break out panels, unless you could plug the amphenol right into the 2400 |
| 02:09.35 | AsteriskNewbie | Hmm .... |
| 02:10.51 | _Sam-- | the amphenol that comes into his place that has his pots...could that plug right into the 2400? |
| 02:11.06 | _Sam-- | or that would go into a breakout panel, then you would run those into the breakout panel from the 2400 |
| 02:11.22 | justinu | if it's rj21x yes |
| 02:11.34 | Ariel_ | the tdm2400 can be plugged into an 66 block with the correct cable |
| 02:11.42 | AsteriskNewbie | Yeah ... that's exactly what I was wondering Sam ... |
| 02:11.59 | SibRphrek | hey [av]bani sup man |
| 02:12.30 | AsteriskNewbie | I believe my lines are all run into a breakout panel right now ... |
| 02:12.36 | justinu | rj21: http://www.stonewallcable.com/Assets/product_images/sc738201_conn_cW.jpg |
| 02:13.59 | AsteriskNewbie | Hmm .... |
| 02:14.00 | _Sam-- | justinu: one of those type cables is only good for 24 lines? |
| 02:14.07 | _Sam-- | i had a 50 pair connector like that once at my isp |
| 02:14.08 | _Sam-- | i thought |
| 02:14.25 | _Sam-- | hmm i guess 50 pair = 24 lines |
| 02:14.32 | AsteriskNewbie | :) |
| 02:14.51 | _Sam-- | i think they dragged 2 of those cables in to give me 50pots |
| 02:15.37 | AsteriskNewbie | Hmm ... |
| 02:15.52 | AsteriskNewbie | Should still be fine if you have < 24 lines ... |
| 02:15.56 | Qwell | amphenol is 25 pairs, I think |
| 02:16.01 | _Sam-- | correct |
| 02:16.09 | justinu | sam: 25 pair (50 wire) |
| 02:16.18 | MikeJ[Laptop] | _Sam-->, they have passthrough 66 blocks.. with the amphenol on each side.. |
| 02:16.28 | MikeJ[Laptop] | you probably want somthing like that |
| 02:16.38 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 02:16.44 | MikeJ[Laptop] | then you put the little metal brigdge things going all the way down |
| 02:16.53 | _Sam-- | yeah i know what you mean |
| 02:17.01 | MikeJ[Laptop] | that way you can still test lines with a test set |
| 02:17.04 | _Sam-- | its not for me, im just asking the questions |
| 02:17.09 | _Sam-- | but its goot to learn |
| 02:17.21 | AsteriskNewbie | so IncomingLine -->breakout_panel-->amphenolcable-->Tdm2400p ??? |
| 02:17.40 | Qwell | AsteriskNewbie: basically |
| 02:17.41 | MikeJ[Laptop] | I beleive so.. |
| 02:17.52 | MikeJ[Laptop] | I don't know exactly what connector is on there... |
| 02:17.57 | MikeJ[Laptop] | or.. I don't recall... |
| 02:18.11 | SibRphrek | anyone doing voicemail broadcasting? |
| 02:18.13 | AsteriskNewbie | is there any way I can keep my currrent system up and running while I'm playing with the cabling?? |
| 02:18.34 | Qwell | AsteriskNewbie: should be able to plug in cables while it's booted, just fine |
| 02:18.54 | Ariel_ | don't you like the metal things... clips or you can do the crossconnect wires or not they also have some plastic red jumpers |
| 02:19.06 | AsteriskNewbie | Hmm ... |
| 02:19.30 | _Sam-- | AsteriskNewbie : how many lines do you have? |
| 02:19.39 | AsteriskNewbie | _Sam: I have 10 lines ... |
| 02:19.55 | _Sam-- | i dont have personal experience with it... |
| 02:19.56 | *** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com) |
| 02:20.05 | _Sam-- | but maybe the sangoma is made in 10 lines |
| 02:20.14 | _Sam-- | iinstead of 12 for the digium |
| 02:20.19 | AsteriskNewbie | sangoma? |
| 02:20.39 | _Sam-- | you are the right track...sorry to steer you off course. |
| 02:20.49 | Ariel_ | sangoma I think it's also 4 ports each model or 3 so your still off the count |
| 02:20.52 | Qwell | _Sam--: 6x4 for the 2400p |
| 02:21.06 | _Sam-- | sangoma is 2 ports fxo i thought |
| 02:21.07 | AsteriskNewbie | Hmm ... |
| 02:21.08 | Qwell | or, 4x6...semantics |
| 02:21.34 | Ariel_ | argh mouse's battery is out..... |
| 02:21.56 | AsteriskNewbie | maybe I'm even approaching this the wrong way. The immediate problem to extend the phone system to a new location ...via asterisk, I hope .. |
| 02:22.14 | AsteriskNewbie | more like send a Norstar extension to asterisk ... |
| 02:22.27 | AsteriskNewbie | Tried using Nortel ATA .. but they don't do the job reliably .. |
| 02:22.37 | Qwell | AsteriskNewbie: ditch the norstar entirely :p |
| 02:22.44 | _Sam-- | AsteriskNewbie : do you have a data circuit between the two locations? |
| 02:22.53 | SibRphrek | no one is doing Voicemail broadcasting?! |
| 02:23.18 | AsteriskNewbie | No .. not yet .. |
| 02:23.26 | Ariel_ | SibRphrek, voicemail broadcasting? |
| 02:23.28 | AsteriskNewbie | And Qwell: .. that's exactly what I'm thinking .. |
| 02:23.29 | SibRphrek | yeah |
| 02:23.45 | SibRphrek | Ariel_: I wanna leave multiple extensions voicemails with 1 phone call |
| 02:23.49 | _Sam-- | AsteriskNewbie : for what 10 pots lines cost per month, you could get a data T1 maybe |
| 02:24.01 | _Sam-- | not that there is another type of t1...but a regular type t1 |
| 02:24.06 | SibRphrek | a data T1 costs from my company 579/monht |
| 02:24.16 | SibRphrek | and we're one of the highest in NYC |
| 02:24.18 | _Sam-- | and instead of 10 lines, you could have alot more than 10 |
| 02:24.33 | AsteriskNewbie | That's what I thought .. but management doesn't understand much of this .. as you can guess :p |
| 02:24.38 | Abydos313 | we pay 303 for each voice t1 and 475 for internet t1 |
| 02:25.31 | _Sam-- | AsteriskNewbie : the initial hardware costs would also be substantially less using a t1 |
| 02:25.38 | _Sam-- | well, if you dont include the cost of a t1 router |
| 02:25.47 | Qwell | Abydos313: $303? wtf? |
| 02:25.55 | Qwell | Verizon wouldn't budge past $600 |
| 02:25.56 | Abydos313 | :) paetech |
| 02:26.03 | AsteriskNewbie | Hmm .... |
| 02:26.05 | Abydos313 | we have 4 right now |
| 02:26.15 | Abydos313 | since 2001 |
| 02:26.35 | AsteriskNewbie | Anyway .. other than ditching the Norstar and current setup right away, which I'd like to do, believe me ... is there another good solution? |
| 02:26.51 | Qwell | AsteriskNewbie: the tdm2400 is a good solution |
| 02:27.14 | SibRphrek | t1 router is 599 |
| 02:27.19 | _Sam-- | maybe less even |
| 02:27.23 | SibRphrek | but if you already have a cisco 1700 we can reprogram it |
| 02:27.30 | AsteriskNewbie | Qwell: Ok. But I'd definitely have to ditch the Norstar to get a straightforward solutio with the tdm2400 ..?? |
| 02:27.31 | SibRphrek | WTF |
| 02:27.39 | _Sam-- | if you got a 2400 card, how would it interact with the norstar? |
| 02:27.40 | SibRphrek | this is straight forward shit yet it's still not working |
| 02:27.47 | Abydos313 | i have a personal 1700 kickin, just waitin to use it ;) |
| 02:27.57 | dja_ | help...sometimes when I dial an extension it doesn't ring (even though my phone is ringing)...looking in the logs, I see "chan_sip.c: sip_xmit...returned -1: Operation not permitted". Help? |
| 02:27.58 | Qwell | _Sam--: analogly? |
| 02:28.07 | Qwell | or get a T1 between them |
| 02:28.23 | _Sam-- | im not sure how the 2400 at the remote location would be part of the norstar system |
| 02:28.28 | _Sam-- | just because im not the expert |
| 02:28.40 | Ariel_ | sometimes it's easyer to put a pri card into the norstar then plug in a te110p board to it from the asterisk box. |
| 02:29.35 | *** join/#asterisk pengyong (n=lala@222.188.138.165) |
| 02:29.40 | AsteriskNewbie | Oh ... I don't mean to put the the asterisk box at the remote location anyway .. |
| 02:29.57 | AsteriskNewbie | I want it to be at the old location and have everyone at the new location used a sip-type phone .. |
| 02:30.17 | _Sam-- | that'll work |
| 02:30.19 | SibRphrek | yay got it |
| 02:30.29 | *** join/#asterisk DrukenHME (n=druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 02:30.54 | AsteriskNewbie | In this case then .. I don't really need the Norstar anymore ... do I? |
| 02:31.04 | Qwell | AsteriskNewbie: not so much, no |
| 02:31.07 | SibRphrek | AsteriskNewbie: what are you trying to do - get a new T1? |
| 02:31.08 | _Sam-- | if you want it to talk to the norstar phones |
| 02:31.25 | AsteriskNewbie | Ah yes! the goddamn Norstar phones .. |
| 02:31.29 | AsteriskNewbie | I was forgetting that .. |
| 02:31.53 | DrukenHME | might as well offload them on ebay and buy some ip phones :) |
| 02:32.01 | AsteriskNewbie | SibRphrek: No .. am trying to hook up the new office to the old one .. via asterisk ..am hoping .. |
| 02:32.09 | SibRphrek | oh |
| 02:32.16 | SibRphrek | you want asterisk to talk to pots? |
| 02:32.27 | AsteriskNewbie | Yes, SibRphrek .. |
| 02:32.34 | SibRphrek | ah |
| 02:32.37 | SibRphrek | i have to do that now |
| 02:32.43 | SibRphrek | so my pots customers have voicemail |
| 02:32.48 | AsteriskNewbie | But the pots are at the old location and the new location should be all voip .. |
| 02:32.49 | _Sam-- | AsteriskNewbie : you are very much on the right track. ditch the norstar, get all SIP phones, and connect the remote office with SIP phones |
| 02:32.52 | SibRphrek | it's all about call fowarding w/verizon |
| 02:33.09 | AsteriskNewbie | Hmm .. |
| 02:33.22 | _Sam-- | how many phones are you talking about on the norstar system? |
| 02:33.27 | _Sam-- | 20? |
| 02:33.46 | AsteriskNewbie | Sam: Sound good. using sip phones for new location is easy. convinging management to ditch the old nortel phones is difficult .. |
| 02:34.14 | _Sam-- | you need to figure out how much that nortel is costing in service contracts |
| 02:34.18 | AsteriskNewbie | No .. about 15 phones .. |
| 02:34.20 | _Sam-- | and then show them how they could save money |
| 02:34.28 | _Sam-- | 15 sip phones is a drop in the bucket compared the nortel system |
| 02:34.30 | Qwell | pfft, 15 phones is like one service visit |
| 02:34.33 | DrukenHME | i bet management dumped 15+ grand into the old shitty norstar pbx :) |
| 02:34.43 | AsteriskNewbie | hehehe! ya bet! |
| 02:34.46 | SibRphrek | AsteriskNewbie: you need phones? |
| 02:34.51 | SibRphrek | we got phones |
| 02:35.01 | _Sam-- | maybe setup a cheap asterisk box and convince them why its better |
| 02:35.10 | _Sam-- | buy a single nice phone, a cheap linux box |
| 02:35.11 | _Sam-- | and make your case |
| 02:35.12 | AsteriskNewbie | SibRphrek .. ip phones? |
| 02:35.19 | SibRphrek | yeah |
| 02:35.32 | *** join/#asterisk iq (n=iq@71-38-79-126.omah.qwest.net) |
| 02:35.34 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:35.35 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:35.37 | iq | Hi All |
| 02:35.38 | iq | g |
| 02:35.42 | SibRphrek | hi iq |
| 02:35.44 | _Sam-- | i work with alot of businesses about your size... |
| 02:35.55 | DrukenHME | i'd like to get my hands on some cisco's but i'm a cheap fuck and refuse to pay the price... |
| 02:35.56 | AsteriskNewbie | yes .. sam? |
| 02:36.02 | _Sam-- | a good strong selling point it that you can connect to the asterisk server from anywhere, like for the execs... |
| 02:36.07 | _Sam-- | so they could be part of the system from home |
| 02:36.12 | _Sam-- | get their voicemail there, etc... |
| 02:36.16 | _Sam-- | small biz likes that |
| 02:36.31 | _Sam-- | that and email to voicemail is a good point to sell them on |
| 02:36.37 | *** join/#asterisk welles (n=welles@61.150.43.114) |
| 02:36.42 | Qwell | heh |
| 02:36.45 | Qwell | I can't do vm emails |
| 02:36.46 | _Sam-- | if you have outside sales force that needs to access your phone system , that is a good one too |
| 02:36.49 | Qwell | bastards filter .wav files |
| 02:37.04 | kratzers | it's be nice to have them in .mp3 |
| 02:37.08 | _Sam-- | how many lines does each of your nortel phones support? |
| 02:37.23 | kratzers | as .wav are fairly large and many common players don't support gsm by default |
| 02:37.24 | _Sam-- | it should be an easy sale, really, if you can make a good presentation |
| 02:37.25 | AsteriskNewbie | one each .. pretty much .. |
| 02:37.27 | *** part/#asterisk dja_ (n=alden@cpe-24-95-62-160.columbus.res.rr.com) |
| 02:37.32 | _Sam-- | the sip phones all do 3+ lines |
| 02:37.36 | _Sam-- | for the most part, that is. |
| 02:37.43 | _Sam-- | you have caller ID with name? |
| 02:37.51 | _Sam-- | how about click to dial from web pages? |
| 02:38.12 | _Sam-- | about web access to their voicemail? |
| 02:38.27 | AsteriskNewbie | Hmm ... |
| 02:38.30 | _Sam-- | how about "follow me" type features, where if someone calls their desk phone, it can then forward to their cell if they dont answer |
| 02:38.31 | DrukenHME | i think sam is in heat |
| 02:38.41 | _Sam-- | nah just working on my sales ideas :) |
| 02:38.48 | AsteriskNewbie | hehehe |
| 02:38.50 | _Sam-- | i will have to pitch the same ideas sooner or later myself |
| 02:39.14 | DrukenHME | :) |
| 02:39.34 | *** join/#asterisk dseeb_ (n=dcb@CPE-144-131-35-116.vic.bigpond.net.au) |
| 02:39.46 | shido6 | or all of thte above at the same time and whichever one picks up thats the one that gets the call |
| 02:40.13 | Druken | bout fucken time |
| 02:40.20 | Druken | only took like 15 mins... |
| 02:40.22 | _Sam-- | what about incoming call menus? call queues? |
| 02:40.43 | Druken | most pbx's will do ivr's and queues |
| 02:41.08 | _Sam-- | probably not the pbx's that small and medium businesses can afford |
| 02:41.09 | Druken | voicemail is usually a killer on boxed pbx's... hehe |
| 02:41.19 | Qwell | okay |
| 02:41.20 | _Sam-- | an office with 15 phones...not sure they are getting IVR |
| 02:41.27 | Qwell | have any of you EVER seen an office that DIDN'T have voicemail? |
| 02:41.41 | AsteriskNewbie | _Sam: the real problem is .. |
| 02:41.42 | Druken | hehe my office, has 3 phones, and we have an ivr... |
| 02:41.44 | Druken | :) |
| 02:41.53 | _Sam-- | we honestly didnt have voicemail at my office until * |
| 02:41.56 | Druken | Qwell: i've seen offices without voicemail on the pbx.... |
| 02:41.57 | _Sam-- | we didnt need it though either |
| 02:42.07 | AsteriskNewbie | most of thse smal businesses have already invested in a phone system .. probably a norstar type sytem .. |
| 02:42.07 | xtrvd | Qwell: My retail office never had voicemail until last year... (23 years in business at the time) |
| 02:42.32 | AsteriskNewbie | You need to sell a solution that will integrate into the existing system .. |
| 02:42.38 | _Sam-- | the nortel type system is called a "key" type system? |
| 02:43.24 | AsteriskNewbie | ?? |
| 02:43.26 | AsteriskNewbie | Whatcha mean? |
| 02:43.33 | Druken | AsteriskNewbie: why integrate? destroy and conqurer... REPLACE! :) |
| 02:43.35 | _Sam-- | AsteriskNewbie : will be a hard pill for lots to swallow, but with * its cheaper to just dump the old crap and start again. the old days of expensive proprietary hardware are gone...... |
| 02:43.56 | _Sam-- | AsteriskNewbie : im not sure what i meant, but i dont know what are considered "key" type pbx's |
| 02:44.04 | _Sam-- | i dont understand what that means, thats why i was asking. |
| 02:44.26 | _Sam-- | but it seems a common term..."key type pbx" or something like that, just trying to learn |
| 02:44.28 | kratzers | key systems |
| 02:44.33 | _Sam-- | yeah whats that mean |
| 02:45.53 | _Sam-- | nevermind...figured it out |
| 02:46.00 | AsteriskNewbie | hehehe .. |
| 02:46.02 | _Sam-- | i used to have a key system |
| 02:46.11 | _Sam-- | AsteriskNewbie : key system info: http://www.allworx.com/XQ/ASP/p.2302/QX/default.htm |
| 02:46.20 | _Sam-- | yours is probably a key system |
| 02:46.29 | AsteriskNewbie | Hmm .. |
| 02:46.47 | kratzers | http://experts.about.com/q/Telecommunications-2419/pbx-vs-key-system.htm |
| 02:47.21 | _Sam-- | thanks kratz |
| 02:47.46 | shido6 | muhahahahahahaa |
| 02:47.46 | kratzers | yup, I did a report recently on PBX systems and heard "key systems" mentioned some but skimmed over |
| 02:47.56 | *** join/#asterisk wellng (n=welles@61.150.43.114) |
| 02:48.05 | kratzers | as they didn't really pertain too much |
| 02:48.13 | _Sam-- | asterisk, in reality, could be considered a 'softswitch'? |
| 02:48.26 | russellb | asterisk is whatever you want it to be |
| 02:48.31 | _Sam-- | i see |
| 02:48.33 | kratzers | yeah, soft PBX |
| 02:49.06 | _Sam-- | how would asterisk speak directly to ss7 equipment? |
| 02:49.14 | _Sam-- | what protocol(s) does that take? |
| 02:49.20 | Qwell | _Sam--: ss7 |
| 02:49.36 | _Sam-- | asterisk speaks ss7? |
| 02:49.44 | Qwell | not yet |
| 02:49.45 | kratzers | I wish I knew C... I'd hack on Asterisk some |
| 02:49.53 | Qwell | but, that's what you'd talk to ss7 equipment with |
| 02:50.22 | _Sam-- | so, my CLEC has a softswitch, that from what i can tell...is nothing (does nothing) more than what i can do with asterisk...they have it at the CO... |
| 02:50.37 | _Sam-- | that device connects to SS7 / some other stuff i dont know about...and gets calls to PSTN... |
| 02:50.44 | _Sam-- | i was just trying to figure out how that happens |
| 02:51.03 | kratzers | I'd like to have a [globals] context in more files |
| 02:51.16 | kratzers | and have support for the #include directive in more files |
| 02:51.56 | *** join/#asterisk peted20 (n=chatzill@71.39.93.58) |
| 02:52.36 | _Sam-- | justinu : how do calls get from the tekelec that i connect to to PSTN? |
| 02:52.46 | _Sam-- | im just trying to understand the path, and how stuff works |
| 02:52.54 | Qwell | howstuffworks.com/ss7 |
| 02:53.05 | _Sam-- | what does that tekelec speak to the next device? sip still? |
| 02:53.41 | AsteriskNewbie | Sam ... are you using tekelec? |
| 02:53.57 | _Sam-- | i connect to a tekelec, which then gets my calls to pstn |
| 02:54.07 | _Sam-- | SIP-->tekelec-->pstn |
| 02:54.16 | AsteriskNewbie | Hmm ... |
| 02:54.32 | kratzers | I have an odd problem with queues. I'm using joinempty = yes, and calls still land in queues with no available phones on the first call. Then Asterisk tries every phone and sees that it's not available and all subsequent calls aren't queued (which is good), but I don't want the first call to be queued either |
| 02:54.36 | AsteriskNewbie | Funny ... I know someone currently evaluating the tekelec solution for voip service .. |
| 02:54.46 | _Sam-- | i think more accurately, it probably looks something like this....SIP-->tekelec-->cisco something-->PSTN |
| 02:55.06 | kratzers | any ideas? |
| 02:55.25 | _Sam-- | kratzers : do you use agent logins? |
| 02:55.30 | kratzers | nope |
| 02:55.33 | _Sam-- | thats probably why |
| 02:55.43 | _Sam-- | * doesnt know the queues are empty (no agents) |
| 02:55.59 | kratzers | I'm in the process of using switching to agents, but I'd rather not if I don't have to |
| 02:56.14 | _Sam-- | i hear that, i started using logged in agents and switched out, its a pain actually. |
| 02:56.30 | *** join/#asterisk wellng (n=welles@61.150.43.114) |
| 02:56.31 | kratzers | people won't be happy having to dial an extension and logging in and out |
| 02:56.43 | _Sam-- | its not that much to it though, and maybe you could make a speed dial on their phones do it |
| 02:56.51 | _Sam-- | but i think that may be the only way to do what you need |
| 02:56.56 | _Sam-- | im not positive by any means. |
| 02:57.01 | kratzers | I was afraid of that |
| 02:57.08 | *** part/#asterisk GeneG (n=GeneG@toronto-HSE-ppp4164687.sympatico.ca) |
| 02:57.23 | kratzers | users don't like change. especially if it requires them to do something more with no visible benefit (as they won't roam) |
| 02:57.25 | _Sam-- | the agent login part should only take about 5 minutes to set up |
| 02:57.58 | kratzers | My shift was up before I got the configs done, but I'm assuming it will solve the problem |
| 02:58.19 | kratzers | how does an agent know if he or she is logged in? |
| 02:58.43 | kratzers | like if they walk away and someone calls and they're logged out since they didn't pick up. how do they know they've been logged out? |
| 02:59.14 | _Sam-- | you have to expressly log out |
| 02:59.23 | _Sam-- | usually, at least the way i had mine. |
| 02:59.33 | _Sam-- | i used this: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AgentCallbackLogin |
| 02:59.40 | kratzers | I think the default config has a logout if an agent doesn't answer in 15 seconds |
| 02:59.47 | kratzers | but of course that could be changed |
| 02:59.51 | peted20 | anyone out there have any experience with vegastream? I am looking at buying an FXO gateway from them, but am not sure if vegastream makes good stuff? I just want something that sounds good and has good echo cancellation |
| 03:00.38 | _Sam-- | kratzers : "Alternatively, you can configure the autologoff setting in Asterisk config agents.conf. This will allow your agents to be automatically removed from the queue after a certain number of seconds if they do not answer calls from the queue. This is very useful for agents who forget to log out. |
| 03:00.58 | _Sam-- | i dont know if thats on by default...but you are right |
| 03:01.33 | kratzers | I'll have to do some playing. |
| 03:02.06 | kratzers | thanks for the link |
| 03:02.36 | _Sam-- | sure, i know the agent logins are a pain, but if could make it simple for the users maybe the will see the beneifts |
| 03:02.44 | _Sam-- | what type of ring strategy for the call queue? |
| 03:02.55 | kratzers | possibly I can make a login/logout button the the phone |
| 03:03.04 | kratzers | leastrecent |
| 03:03.13 | _Sam-- | they will appreciate a logged in agent setup |
| 03:03.21 | _Sam-- | because calls wont be ringing to places where nobody is |
| 03:04.00 | _Sam-- | i use ringall in my queue now, and stopped the logged in agents |
| 03:04.48 | AsteriskNewbie | _sam: do you know how to enable long tones .. as opposed to dtmf? |
| 03:05.14 | _Sam-- | there is a special nortel note on that.... |
| 03:05.25 | AsteriskNewbie | yeah ? |
| 03:05.28 | AsteriskNewbie | Must have missed it .. |
| 03:05.43 | _Sam-- | http://www.voip-info.org/wiki/view/Asterisk+Nortel |
| 03:06.14 | AsteriskNewbie | Oh .. I read all of that. Not much help am afraid .. |
| 03:06.23 | _Sam-- | "You should know that most of the Nortel phones cannot send DTMF tones to the FXO interface in Asterisk. You need to implement "Long Tones" (feature 808) before Asterisk will act on the tones so you can dial o use something like Voicemail. |
| 03:06.25 | _Sam-- | " |
| 03:06.32 | [av]bani | driving kids to school in a hummer. lovely. |
| 03:06.38 | AsteriskNewbie | yeah .. that's what I'm wondering .. |
| 03:06.46 | AsteriskNewbie | Do you know how to implement feature 808? |
| 03:06.54 | [av]bani | if thats not a perfect example of conspicuous consumption i dont know what is. it's like driving around in a vehicle shaped like a giant middle finger. |
| 03:07.54 | _Sam-- | AsteriskNewbie : i was more under the impression it is something on the nortel phones. |
| 03:08.25 | kratzers | later all |
| 03:08.26 | AsteriskNewbie | Oh? Hmm ... never thought of it that way. I will have to look it up ... |
| 03:08.33 | UdontKnow | [av]bani: hah |
| 03:09.58 | |omni| | anyone get festival using something other than the default voice with Asterisk's tts_textasterisk command? |
| 03:11.12 | _Sam-- | how are people doing the voice changes in real time in asterisk? |
| 03:11.22 | _Sam-- | like make a male voice sound like a female voice |
| 03:11.29 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.5 released! (March 3, 2006) -=- Asterisk-addons 1.2.2 now available (March 6, 2006) |
| 03:11.48 | _Sam-- | nevermind |
| 03:11.57 | _Sam-- | "Asterisk Voice Changers" |
| 03:12.00 | _Sam-- | i should have known |
| 03:12.35 | octothorpe_ | Please elaborate sam, or are you looking for answers |
| 03:12.45 | _Sam-- | there is a voice changer |
| 03:12.47 | _Sam-- | for asterisk |
| 03:12.51 | _Sam-- | http://www.lobstertech.com/voicechanger/ |
| 03:13.32 | octothorpe_ | sam: thanks, I'll look at it |
| 03:13.45 | _Sam-- | sure thing, have fun, let me know how it works |
| 03:15.37 | _Sam-- | i cant imagine that it works great if all you do is change the pitch of a voice |
| 03:15.39 | _Sam-- | but it might |
| 03:16.11 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 03:16.16 | MikeJ[Laptop] | it does some cool stuff |
| 03:16.30 | MikeJ[Laptop] | but is basically a toy |
| 03:16.40 | octothorpe_ | toys are cool |
| 03:18.12 | _Sam-- | MikeJ[Laptop] : i am not experiencing any problems currently, but i notice that switch-07 is a problem |
| 03:18.26 | _Sam-- | my asterisk cant register there (again, im not having any problems, just a heads up) |
| 03:20.11 | MikeJ[Laptop] | ummm |
| 03:20.25 | MikeJ[Laptop] | well.. we did have somone overwhelming some of our switches. |
| 03:20.30 | MikeJ[Laptop] | it should be better now |
| 03:20.59 | *** join/#asterisk encode (n=encode@blah.i.hate.w1ndo.ws) |
| 03:21.03 | _Sam-- | nah not yet, no biggie |
| 03:21.15 | _Sam-- | going to bed anyway...good luck :) |
| 03:21.49 | octothorpe_ | MikeJ[Laptop]: what service do you run? |
| 03:38.43 | *** part/#asterisk alephcom (n=darren@host75.net14.mcsnet.ca) |
| 03:45.55 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au) |
| 03:49.30 | *** join/#asterisk trym (n=trym@194.63.254.6) |
| 03:51.49 | *** join/#asterisk bmg505 (n=leon@dsl-146-24-50.telkomadsl.co.za) |
| 03:57.11 | *** part/#asterisk octothorpe_ (n=octothor@c-67-186-207-234.hsd1.ut.comcast.net) |
| 03:58.05 | *** join/#asterisk f^bar (n=octothor@c-67-186-207-234.hsd1.ut.comcast.net) |
| 03:58.17 | f^bar | ~bullshit |
| 03:58.18 | jbot | i heard bullshit is If you want to speak bullshit, please go to #debian.bullshit. sdf dflkj Linux sucks sfg yo momma dfg #debian.bullshit |
| 04:01.01 | f^bar | ~lart f^bar |
| 04:01.20 | Qwell | ~bot abuse |
| 04:01.21 | jbot | Leave me alone.. I feel abused and molested. |
| 04:04.39 | xtrvd | =) |
| 04:07.52 | *** part/#asterisk reallost1 (n=reallost@12-215-208-184.client.mchsi.com) |
| 04:10.18 | *** part/#asterisk f^bar (n=octothor@c-67-186-207-234.hsd1.ut.comcast.net) |
| 04:11.45 | *** join/#asterisk JunK-Y (n=junky@67.71.157.108) |
| 04:15.29 | xtrvd | Could somebody point me in the direction of a resource to gain more information in understanding how I can connect 2 * boxes together and have them talk via IAX and pass through conversations between their connecting SIP phones? |
| 04:22.59 | brookshire | voip-info.org! |
| 04:23.00 | brookshire | :D |
| 04:23.18 | xtrvd | I'm looking brookshire, but I can't find such an article! |
| 04:23.29 | xtrvd | ... nevermind: http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers |
| 04:34.57 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 04:37.12 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 04:40.06 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 04:52.12 | jeebusroxors | is it possible to encode a wav with g729? |
| 04:53.07 | Qwell | it wouldn't really be a wav anymore, no |
| 04:54.14 | jeebusroxors | well |
| 04:54.25 | jeebusroxors | i need to take pcm and encode... |
| 04:54.38 | jeebusroxors | was wondering if there are any linux apps for that |
| 04:59.44 | brookshire | sox ? |
| 05:01.05 | justinu | sox doesn't know g729 |
| 05:01.46 | Winkie | anyone happen to know offhand if autofill is in 1.2.5 |
| 05:01.49 | jeebusroxors | hm |
| 05:04.28 | *** join/#asterisk angom_h (n=angom@red-corp-201.130.118.46.telnor.net) |
| 05:06.16 | *** join/#asterisk clwnkllr (i=eurolite@suicide.nothingkillsfaster.net) |
| 05:06.19 | clwnkllr | With a ANA to VOIP broadband convereter can you split it off and connect multiple phones from one converter? Like if theres only 1 RJ-45 port on the converter can you split it off with a RJ-45 splitter and connect multiple phones? |
| 05:06.48 | *** join/#asterisk lucasjb (n=lucas@mail.stabat.com) |
| 05:07.08 | clwnkllr | I want to tunnel my current phone system to VOIP |
| 05:07.50 | clwnkllr | I have 7 phones in my home |
| 05:08.39 | *** join/#asterisk devin (i=devin@203.141.139.231.user.ad.il24.net) |
| 05:09.01 | jeebusroxors | clwnkllr: what layer does an ANA work on? |
| 05:09.51 | clwnkllr | hell if i know |
| 05:09.56 | clwnkllr | i wouldent be here if i knew what i was talking about |
| 05:10.04 | clwnkllr | what i mean by ana is analog |
| 05:10.11 | *** join/#asterisk devin (i=devin@203.141.139.231.user.ad.il24.net) |
| 05:10.45 | clwnkllr | dont give me hellish memories of cisco class |
| 05:10.50 | jeebusroxors | clwnkllr: im gonna say no... |
| 05:10.52 | jeebusroxors | heh |
| 05:11.00 | clwnkllr | I don't see why it would not work? |
| 05:11.11 | jeebusroxors | your splitting 45 or 12? |
| 05:11.56 | astra^^ | hello room can anyone help me set up asterisk .. in my server.. ? |
| 05:12.04 | clwnkllr | the box that u can buy to connect to ethernet (your modem) that gives you a RJ-45 output so you can connect a normal household telephone |
| 05:12.20 | clwnkllr | err |
| 05:12.24 | clwnkllr | RJ-11 I mean |
| 05:12.30 | jeebusroxors | clwnkllr: thats 11 ;) |
| 05:12.34 | clwnkllr | yea i know |
| 05:12.42 | jeebusroxors | i suppose that wold work - it would ring all phones at once though |
| 05:12.48 | clwnkllr | whats wrong with that? |
| 05:12.53 | clwnkllr | isint that what they do right now? |
| 05:13.03 | clwnkllr | asterisk would be useless tho :( |
| 05:13.17 | clwnkllr | I wanted to play with it but I am not spending 200$x6 to get 6 new phones |
| 05:13.42 | jeebusroxors | clwnkllr: you can find phones for ~80 |
| 05:13.52 | clwnkllr | still, I own a 4 story house |
| 05:13.57 | clwnkllr | it would be a nightmare wiring it all |
| 05:14.02 | jeebusroxors | heh nah |
| 05:14.16 | jeebusroxors | you got all your exsisting phones for snakes ;) |
| 05:14.40 | clwnkllr | how would I run RJ-45 through RJ-11 |
| 05:14.54 | Qwell | 45 through 11? |
| 05:14.57 | Qwell | you...don't |
| 05:15.03 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 05:15.07 | Qwell | 4 pairs vs 2 pairs |
| 05:15.17 | jeebusroxors | you would snake using the cat2, pull the rj45 with it |
| 05:15.26 | jeebusroxors | er cat5* |
| 05:15.36 | clwnkllr | hmm |
| 05:15.54 | clwnkllr | would one of those wall plugin network devices |
| 05:16.06 | clwnkllr | can't remember what its called now 1 sec |
| 05:16.43 | jeebusroxors | they have punchdown blocks... |
| 05:16.56 | jeebusroxors | youd have a wall jack with female rj45 in it |
| 05:17.42 | clwnkllr | a powerline bridge |
| 05:17.54 | clwnkllr | does it provide enough bandwidth for voip |
| 05:18.00 | jeebusroxors | i wouldnt know |
| 05:18.22 | clwnkllr | hmm |
| 05:18.23 | clwnkllr | Connects 10/100 Ethernet Network to Powerline Network for Seamless Integration |
| 05:18.42 | MikeJ[Laptop] | voip does not require much bandwidth |
| 05:18.50 | clwnkllr | I assume video phones do |
| 05:18.54 | MikeJ[Laptop] | just low latentcy and jitter |
| 05:19.22 | MikeJ[Laptop] | certainly compared to 10 mbit eth |
| 05:19.43 | jeebusroxors | ugh...i need to encode pcm with g729 or 723 |
| 05:19.50 | MikeJ[Laptop] | need? |
| 05:20.03 | clwnkllr | The PowerLine Bridge can also plug directly into a cable or DSL modem to allow |
| 05:20.03 | clwnkllr | Internet access and data transfer rates up to 14Mbps over |
| 05:20.03 | clwnkllr | home powerlines. |
| 05:20.11 | clwnkllr | woops, sorry for multiple lines. |
| 05:20.20 | jeebusroxors | MikeJ[Laptop]: yea... |
| 05:20.31 | clwnkllr | That's faster then a cable modem anyways |
| 05:20.39 | MikeJ[Laptop] | are we talkig about in a house still? |
| 05:20.46 | MikeJ[Laptop] | or somthing else? |
| 05:20.53 | clwnkllr | my house |
| 05:20.59 | clwnkllr | thinking of a cheap way to intigrate voip |
| 05:21.07 | MikeJ[Laptop] | pull wire |
| 05:21.09 | MikeJ[Laptop] | :D |
| 05:21.14 | MikeJ[Laptop] | it's not that hard |
| 05:21.20 | brookshire | jeebusroxors: http://www.asteriskguru.com/audio_conversion.php |
| 05:21.22 | MikeJ[Laptop] | that's what cold air returns are for |
| 05:21.28 | [av]bani | \o> |
| 05:22.09 | MikeJ[Laptop] | brookshire, sox is your friend? |
| 05:22.11 | jeebusroxors | brookshire: your a demigod |
| 05:22.15 | jeebusroxors | wav isnt working though heh |
| 05:22.23 | MikeJ[Laptop] | sox'll do it |
| 05:22.45 | clwnkllr | is there a vxml addon for asterisk? |
| 05:22.49 | clwnkllr | that would be nice :) |
| 05:22.51 | jeebusroxors | sox will? |
| 05:23.20 | brookshire | :( |
| 05:23.35 | jeebusroxors | damn sox will do it....someone said it didnt heh |
| 05:23.41 | brookshire | http://www.germanixsoft.de/index.php |
| 05:23.49 | brookshire | there is also that |
| 05:24.04 | justinu | sox can do g729 encoding? |
| 05:24.10 | MikeJ[Laptop] | oh... |
| 05:24.10 | jeebusroxors | justinu: yea |
| 05:24.13 | justinu | wow |
| 05:24.15 | MikeJ[Laptop] | ummm |
| 05:24.19 | justinu | does it actually work? |
| 05:24.19 | MikeJ[Laptop] | I dunno actually |
| 05:24.39 | MikeJ[Laptop] | I have a 729 lib I use... but not w/ sox |
| 05:24.43 | justinu | i can't believe an opensource app like sox can encode g729 |
| 05:24.49 | MikeJ[Laptop] | nope... |
| 05:24.56 | jeebusroxors | well it can play it at least |
| 05:24.58 | MikeJ[Laptop] | didn't know that is what we were talking aout |
| 05:25.07 | jeebusroxors | how bout 723? |
| 05:25.18 | clwnkllr | I don't understand this powerline bridge, ok so it converts 10/100 to your powerline now how do you convert it back so you can plug a cat5e cable into it? |
| 05:25.31 | MikeJ[Laptop] | it has a box on each side |
| 05:25.35 | MikeJ[Laptop] | jeebusroxors, dunno... |
| 05:25.45 | clwnkllr | linksys only sells the bridge |
| 05:25.50 | clwnkllr | no 'receivers' |
| 05:26.01 | MikeJ[Laptop] | clwnkllr, bridge is 2 pieces |
| 05:26.09 | MikeJ[Laptop] | trancievers |
| 05:26.37 | clwnkllr | http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1115416874725&pagename=Linksys%2FCommon%2FVisitorWrapper |
| 05:26.58 | jeebusroxors | brookshire: im on linux heh |
| 05:27.17 | clwnkllr | hell |
| 05:27.37 | clwnkllr | it better use token ring |
| 05:27.42 | clwnkllr | or it would be collsion central |
| 05:29.28 | clwnkllr | ahh, i see |
| 05:29.31 | clwnkllr | its homeplug compliant |
| 05:29.58 | lucasjb | Hiyas, I know how to setup an extension for AgentBarge, but how do I define who is a supervisor and what their password is? |
| 05:49.45 | clwnkllr | Anyways, thanks for the input I still have a lot of thinking to do ;d. |
| 05:49.45 | clwnkllr | cya. |
| 05:49.47 | *** part/#asterisk clwnkllr (i=eurolite@suicide.nothingkillsfaster.net) |
| 05:59.58 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 06:02.21 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:05.59 | *** join/#asterisk CpuID2 (n=nathan@dsl-202-173-176-82.qld.westnet.com.au) |
| 06:06.14 | CpuID2 | hey ppls, anyone here using l7-filter with asterisk? preferably with IAX[2]? |
| 06:24.02 | *** join/#asterisk Pegger (n=peg@pool-68-163-139-150.bos.east.verizon.net) |
| 06:26.52 | *** join/#asterisk tuxinator_linuxM (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 06:34.19 | *** join/#asterisk laichzeit (n=immortal@dsl-145-187-236.telkomadsl.co.za) |
| 06:35.41 | *** join/#asterisk welles (n=welles@219.144.202.56) |
| 06:36.52 | laichzeit | hi guys, I have a ring group (2 receptionists), when they're on lunch, people in the office want to be able to answer calls on their behalf, is it possible for someone in the office to transfer the call to themselves (they're using x-lite clients) ? |
| 06:37.43 | *** join/#asterisk Haris (n=Haris@fiber10-228.brain.net.pk) |
| 06:40.24 | *** join/#asterisk YoMama (n=rewt@d14-69-186-121.try.wideopenwest.com) |
| 06:40.58 | xtrvd | I believe there's a way that a user can enter an extension (onLunch switch) which will change your dialplan and thereby forward calls to a different set of extensions until your offLunch switch extension is activated reverting the dialplan back to its original state. |
| 06:41.16 | YoMama | xtrvd: yeah..it's not hard |
| 06:41.25 | YoMama | xtrvd: you just use the astdb to do it |
| 06:41.30 | xtrvd | [22:36] <laichzeit> hi guys, I have a ring group (2 receptionists), when they're on lunch, people in the office want to be able to answer calls on their behalf, is it possible for someone in the office to transfer the call to themselves (they're using x-lite clients) ? |
| 06:41.48 | MGSsancho | isn that a ramstein song? |
| 06:41.52 | MGSsancho | laichzeit? |
| 06:41.58 | xtrvd | YoMama: Thanks, I believe that's exactly what laichzeit needed. =) |
| 06:42.04 | YoMama | np |
| 06:42.57 | YoMama | yeah..a nifty way to do it is to create a context just for the receptionists |
| 06:43.02 | YoMama | call it [receptionist] let's say |
| 06:43.14 | YoMama | and have them include inside-lines |
| 06:43.20 | YoMama | so their extension acts normally |
| 06:43.32 | YoMama | and in their own context...createa dialcode like 999 or whatever |
| 06:43.55 | YoMama | exten => 999,1,uhh..i can't remember the command..but have it set a key/value in astdb |
| 06:44.03 | YoMama | then in your incoming... |
| 06:44.14 | YoMama | you do an if that value is set in the database...then transfer directly to the cover lines |
| 06:44.30 | YoMama | it's that easy..i could write it out..but it'd require i go look shit up..and u can do that..but that's conceptually how it works |
| 06:44.39 | YoMama | and then 998 can delete the db value |
| 06:44.55 | YoMama | or make it *998 |
| 06:44.57 | YoMama | whatever u want |
| 06:46.34 | YoMama | anyone who knows more about asterisk awake in here? |
| 06:46.38 | YoMama | i got a question :) |
| 06:47.51 | *** join/#asterisk Grizzy (i=Generic@ppp-69-238-229-72.dsl.pltn13.pacbell.net) |
| 06:49.59 | xtrvd | YoMama: What's your question. I may not be able to answer, but I'm still curious. =) |
| 06:50.14 | Winkie | anyone checked out asterisk's makefiles, i need some breif help |
| 06:50.52 | YoMama | xtrvd: if i'm trying to hook a legacy phone system that uses analog lines using asterisk as a proxy...what's the best strategy for extension transfer between asterisk and this phone system? |
| 06:52.41 | xtrvd | YoMama: I haven't the slightest idea... =| |
| 06:53.17 | laichzeit | err.. thanks for the advice on the call transfer stuff |
| 06:53.36 | FuriousGeorge | my trunk my trunk, my lovely asterisk trunk |
| 06:54.12 | xtrvd | No no Telco, you don't want no Telco |
| 06:54.21 | Qwell | FuriousGeorge: You're banned from music for a while |
| 06:54.30 | FuriousGeorge | lol |
| 06:55.32 | YoMama | Qwell: ha... |
| 06:55.37 | YoMama | i hate that song |
| 06:57.09 | FuriousGeorge | ima make make make make this call, make this call out on my trunk |
| 06:57.14 | YoMama | Qwell: hey man..maybe u can help me out with this |
| 06:57.35 | xtrvd | FuriousGeorge: your trunk? |
| 06:57.43 | FuriousGeorge | my lovely asterisk trunk |
| 06:57.47 | FuriousGeorge | (check it out) |
| 06:57.57 | Ukyo | Junk in the Trunk ? |
| 06:57.59 | YoMama | xtrvd: you're not from the States are you? |
| 06:58.15 | FuriousGeorge | dont have to set the nice-y, config the sip device-y |
| 06:58.23 | Ukyo | ausiland from the looks of it. :P |
| 06:58.24 | xtrvd | YoMama: From Canada, and yes, I do follow the parodical theme. |
| 06:58.32 | Ukyo | oh sorry |
| 06:58.34 | xtrvd | =) |
| 06:58.34 | jeebusroxors | *background* make this call make this call |
| 06:58.41 | Ukyo | thought he asked george :P |
| 06:58.45 | YoMama | FuriousGeorge: thank you |
| 06:59.12 | FuriousGeorge | spending all my money money, on tdm four hundy's |
| 06:59.15 | FuriousGeorge | I CANT HELP IT |
| 06:59.25 | xtrvd | ... |
| 06:59.27 | Ukyo | Do any of you buy dedicated servers to host * on? |
| 06:59.31 | jeebusroxors | so would a severe drop in bandwith (up speeds) result in no audio on one end? |
| 06:59.57 | xtrvd | Hopefully the wind is knocked out of him enough to get some silence in here. |
| 07:00.02 | tuxinator_linuxM | jeebusroxors: yep |
| 07:00.17 | jeebusroxors | tuxinator_linuxM: thats what i was afraid of :-\ thanks |
| 07:00.20 | FuriousGeorge | xtrvd: im sure if the black eyed peas were here they'd tell you not to hate |
| 07:00.38 | YoMama | Ukyo: u asking what machine to run it on? |
| 07:00.38 | xtrvd | Where is the love eh? |
| 07:00.49 | jeebusroxors | FuriousGeorge: watch out for those mpaa guys |
| 07:00.55 | FuriousGeorge | lol |
| 07:01.05 | xtrvd | They would tell me to stop the 'Monkey Business'? |
| 07:01.08 | Ukyo | nah, was just curious if anyone has ever put * on a DedicatedServer (Hosted server at a datacenter |
| 07:01.09 | Ukyo | ) |
| 07:01.12 | FuriousGeorge | jeebusroxors: dont worry, only the parody version will be on my MoH |
| 07:01.19 | Ukyo | lol |
| 07:01.27 | jeebusroxors | FuriousGeorge: theres an actual version? |
| 07:01.33 | Ukyo | -_- |
| 07:01.37 | xtrvd | FuriousGeorge is recording now... |
| 07:01.41 | jeebusroxors | i want it |
| 07:02.08 | YoMama | Ukyo: I know people who've done it |
| 07:02.09 | xtrvd | I can hear the "*background* make this call make this call" now. |
| 07:02.15 | FuriousGeorge | mine is the parody version. afaik, the BEPs have yet to write a song about PBXs |
| 07:02.36 | YoMama | Ukyo: to use it in a commercial environment, if you're gonna do calling card app or resell it...yeah |
| 07:02.49 | Ukyo | Yeah, that's what I was thinking |
| 07:03.38 | Ukyo | Finished getting my * server running and all, and stopped, to think about the market for it |
| 07:03.48 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:04.07 | YoMama | it's a good cheap way to get a fancy phone system |
| 07:04.08 | *** join/#asterisk af_ (n=af@ip-172-156.sn1.eutelia.it) |
| 07:04.13 | YoMama | the only problem is its lack of failover |
| 07:04.30 | Ukyo | (Trying to think of a nifty package I could offer people) |
| 07:04.40 | YoMama | i can think of tons |
| 07:04.47 | Ukyo | FIll me in. :) |
| 07:04.48 | YoMama | what infrastructure do u own? |
| 07:05.10 | Ukyo | I own a datacenter in the Dallas Infomart, we have GigE's with Level3, Savvis, etc |
| 07:05.29 | *** join/#asterisk X-Rob_ (n=Rob@dsl-220-235-230-122.vic.westnet.com.au) |
| 07:05.32 | Ukyo | Started the company doing colocation, but mostly sell dedicateds now |
| 07:05.37 | YoMama | do u supply connectivity to any customers? |
| 07:05.42 | *** join/#asterisk welles (n=welles@61.150.43.114) |
| 07:05.52 | Ukyo | That's what I do primarily. ;) |
| 07:05.58 | YoMama | connectivity? |
| 07:05.59 | Ukyo | Colocation, Dedicated servers, Cabinets. etc |
| 07:06.06 | YoMama | ok...so no connectivity |
| 07:06.14 | YoMama | just colos and shared servers |
| 07:06.16 | Ukyo | I can sell ds3's etc |
| 07:06.25 | Ukyo | In fact, I can beat most pricing on them in Dallas |
| 07:06.34 | Ukyo | But I don't bother pushing too much |
| 07:07.23 | YoMama | if u really wanna talk about this...we should take it offline |
| 07:07.23 | xtrvd | Ukyo: Side question, not intending to jack the conversation, but how much do T1/DS1's and T3/DS3's go for in Dallas? |
| 07:07.53 | Ukyo | xtrvd: its actually a pretty impressive range nowdays. With the zero-loop links you can get them dirt cheap |
| 07:07.57 | Ukyo | YoMama: sure :) |
| 07:08.05 | Ukyo | t1's can be had for $250/mo |
| 07:08.16 | Ukyo | full t3's for $2500 or less |
| 07:08.22 | Ukyo | including loop fees |
| 07:08.37 | Qwell | Dallas/Ft. Worth? |
| 07:08.40 | Ukyo | Yeah |
| 07:09.00 | YoMama | ukyo: i need to host my sun box without getting any bitching about how much bandwidth i'm using..haha |
| 07:09.14 | Ukyo | ultrasparc ? what? |
| 07:09.24 | YoMama | it's a 4 proc ultrasparc |
| 07:09.25 | tuxinator_linuxM | YoMama: porn site? |
| 07:09.27 | YoMama | no |
| 07:09.27 | Ukyo | ooold coffee table now :P |
| 07:09.30 | YoMama | just my own box |
| 07:09.47 | YoMama | i do a lot of testing on it...run a few blog sites..run an ecommerce site |
| 07:09.48 | xtrvd | For anybody to answer: T1's interface with Asterisk through a PRI card and you get 23 channels of audio to connect to the Telco, right? |
| 07:09.50 | YoMama | that's about it |
| 07:10.01 | Qwell | Ukyo: Is that $250 for PRI, or data T1? |
| 07:10.02 | tuxinator_linuxM | xtrvd: yep |
| 07:10.05 | Ukyo | assuming your t1 is going ot the telco |
| 07:10.14 | YoMama | xtrvd: 22 or 23 and 2 D channels |
| 07:10.23 | *** join/#asterisk dseeb_ (n=dcb@58.165.245.136) |
| 07:10.27 | tuxinator_linuxM | 2 D channels? |
| 07:10.30 | *** part/#asterisk dseeb_ (n=dcb@58.165.245.136) |
| 07:10.39 | Ukyo | Qwell: basically it would be a t1 loop between my datacenter, and wherever. I can channelize it if need be. |
| 07:10.39 | YoMama | yeah...D channels |
| 07:10.47 | Qwell | ahh |
| 07:11.04 | YoMama | Ukyo: you'd sell it for $250? |
| 07:11.05 | Ukyo | Break it down to say 13 channels, and 10channels combined for bandwidth, etc |
| 07:11.17 | tuxinator_linuxM | I though they only did redundant D's with multiple T1's |
| 07:11.27 | Qwell | NFAS |
| 07:11.30 | tuxinator_linuxM | err, PRI's |
| 07:11.33 | Ukyo | Depends on the config. t1 of bandwidth, Depending on location I belive $250 |
| 07:11.43 | xtrvd | ... So if I'm paying $500CDN a month on 10 standard analog telephone lines... it would be stupid not to move to a T1 where I can easily upgrade to 13 lines in a future upgrade (next month intended) |
| 07:11.55 | Qwell | xtrvd: yes |
| 07:11.56 | YoMama | Ukyo: does that $250 include local loop? |
| 07:12.04 | Ukyo | yes |
| 07:12.11 | xtrvd | Qwell: Sweet jebus, why don't people tell me these things!? |
| 07:12.14 | YoMama | xtrvd: $500 for 10 analog lines? holy crap |
| 07:12.18 | Qwell | xtrvd: You never asked? |
| 07:12.23 | Ukyo | lol |
| 07:12.27 | Ukyo | $500cdn |
| 07:12.30 | Ukyo | but yeah, thats still a ton |
| 07:12.33 | YoMama | xtrvd: i pay $400US for one PRI |
| 07:12.35 | Ukyo | I could buy 10 residentials for cheaper |
| 07:12.42 | Ukyo | YoMama: that expensive? |
| 07:12.47 | Ukyo | I used to work for an ISP here |
| 07:12.50 | YoMama | Ukyo: no..it's about average |
| 07:12.51 | xtrvd | YoMama: 44.50CAN / line, add applicable taxes, and whammy. |
| 07:12.53 | Ukyo | it was like $250/mo for a pri |
| 07:12.58 | Ukyo | all inbound tho |
| 07:13.05 | YoMama | Ukyo: depends on what state u live in |
| 07:13.10 | Ukyo | True |
| 07:13.21 | YoMama | Ukyo: this was a unverisal PRI that had DIDs, inbound and outbound service |
| 07:13.27 | Ukyo | PRI's from bel... *cough* at&T are pretty cheap here |
| 07:13.32 | Ukyo | ah |
| 07:13.37 | tuxinator_linuxM | I was quoted 550 - 650 USD here is SoCal |
| 07:13.41 | Ukyo | It was just an incoming dialup pri bank |
| 07:13.45 | xtrvd | What is a T1 interface card worth for connecting to an Asterisk Box? |
| 07:13.51 | Qwell | xtrvd: worth? |
| 07:13.52 | YoMama | xtrvd: what part of canada areu in? someone wastelling me that PRIs in canada aren't that expensive |
| 07:13.56 | xtrvd | Vancouver, BC. |
| 07:13.58 | *** join/#asterisk littlejohn (n=little@host215-5.pool8259.interbusiness.it) |
| 07:13.58 | Ukyo | xtrvd: ebay? :P |
| 07:14.01 | Qwell | for a single T1, about $600 |
| 07:14.02 | xtrvd | Price wise... Digium. |
| 07:14.09 | Qwell | or maybe it was $699...I don't recall |
| 07:14.12 | YoMama | xtrvd: www.voipsupply.com |
| 07:14.27 | YoMama | get the new card with the on-board echo cancellation |
| 07:14.29 | xtrvd | That would be a lot smarter than paying another $800 for a Panasonic card for the proprietary system that is in right now... |
| 07:14.38 | YoMama | echo canceling in * sucks my balls |
| 07:14.41 | Qwell | YoMama: I don't think single span T1 cards have echo can |
| 07:14.44 | Ukyo | hahhaa |
| 07:14.44 | YoMama | oh |
| 07:15.02 | YoMama | Qwell: still..i don't lie :) |
| 07:15.02 | Qwell | would have to ask one of the Digium boys |
| 07:15.04 | YoMama | it does suck balls |
| 07:15.14 | Qwell | YoMama: tried the latest echo cans? |
| 07:15.21 | Qwell | They work very well |
| 07:15.41 | Qwell | Cresl1n is the man |
| 07:15.49 | FuriousGeorge | <PROTECTED> |
| 07:15.52 | Ukyo | actually, I have been looking at doing like the crowd, and starting a hosted VOIP PBX |
| 07:15.55 | Qwell | no, don't think so |
| 07:15.57 | xtrvd | The diff between single span and dual span T1's is... *I'm trying to look this up at the same time as talking about it, but I can't keep up in the reading* |
| 07:16.03 | YoMama | Qwell: what do u mean the latest? |
| 07:16.17 | Ukyo | I have never worked with Bell on DID blocks before. Something I need to look into. (Costs, etc) |
| 07:16.20 | Qwell | umm...MG2? I don't know |
| 07:16.23 | YoMama | xtrvd: single span means it takes 1 T1..dual span means it takes 2 |
| 07:16.41 | YoMama | Ukyo: DIDs are easy to manage when they're coming over a PRI |
| 07:16.50 | YoMama | they suck the big one when they're coming over analog trunks |
| 07:16.53 | Ukyo | Yeah, but how do th ebells charge for them ? |
| 07:16.54 | xtrvd | So with dual, I could have the full 48 lines? |
| 07:17.02 | Qwell | Ukyo: DIDs are dirt cheap |
| 07:17.05 | Ukyo | like $X/mo for X did's ? |
| 07:17.11 | Ukyo | Really? |
| 07:17.16 | YoMama | Ukyo: typically u "rent" them for a block of 10, 25, 50, or 100 |
| 07:17.16 | Ukyo | What about toll free's ? |
| 07:17.21 | Qwell | not so cheap |
| 07:17.27 | Ukyo | hehe |
| 07:17.35 | Qwell | but still not all that much |
| 07:17.38 | YoMama | toll free and DIDs are totally different |
| 07:17.39 | Ukyo | I take it pricing discounts on the more blocks /dids you have |
| 07:17.53 | YoMama | they're pretty cheap |
| 07:17.59 | YoMama | i was renting 100 DIDs for $20/month |
| 07:18.05 | Ukyo | ... |
| 07:18.07 | Ukyo | Your kidding me |
| 07:18.10 | YoMama | no |
| 07:18.11 | Qwell | yeah, it's only several cents per DID |
| 07:18.29 | YoMama | it's just a phone # |
| 07:18.33 | YoMama | doesn't provide service |
| 07:18.37 | Ukyo | Time to call bell and start a side business. :P |
| 07:18.40 | YoMama | nothing except a map in the switch |
| 07:18.44 | YoMama | well |
| 07:18.52 | YoMama | lemme tell u what u can do :-P |
| 07:18.59 | Ukyo | I wanted to do hosted pbx. not voip |
| 07:19.09 | Ukyo | advertise locally to small businesses |
| 07:19.26 | Ukyo | $30/mo for a "phone system" basic menu. extensions fwd to (specified phone number) |
| 07:19.26 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 07:19.43 | Ukyo | and they pay 2c per minute. mebbe like 250 minutes included |
| 07:21.13 | FuriousGeorge | Ukyo: one problem your gonna have is when customer A parks a call customer B can answer it |
| 07:21.29 | Qwell | nah |
| 07:21.35 | Qwell | just setup entirely seperate contexts |
| 07:21.43 | xtrvd | No parking allowed. There, problem solved. |
| 07:22.00 | Ukyo | yeah |
| 07:22.10 | astra^^ | can anyone help me set up asterisk please.. |
| 07:22.13 | FuriousGeorge | Qwell: i thought that wasnt allowed?i thought that was a pretty accepted limitation? |
| 07:22.20 | Qwell | FuriousGeorge: dunno, maybe |
| 07:22.31 | Qwell | could be worked around, I'm sure |
| 07:22.35 | laichzeit | I was thinking about what you guys said earlier, creating a [reception] context with an extension say 999 that should set a value in astdb (I'm on lunch now).. how would you make it write a value in the db, like... an external script is executed or what? |
| 07:22.50 | Qwell | laichzeit: dbput |
| 07:22.55 | Qwell | dbget/dbput |
| 07:22.56 | FuriousGeorge | Qwell: i think people do it with meetmes |
| 07:22.59 | Qwell | show applications like db |
| 07:23.03 | astra^^ | can anyone help me set up asterisk please.. |
| 07:23.15 | Qwell | astra^^: ask again in 30 seconds |
| 07:23.26 | xtrvd | astra^^: What do you need help with, perhaps we could point you in the right direction. |
| 07:23.33 | Qwell | 60 seconds isn't long enough. people might miss it |
| 07:23.37 | Qwell | keep asking |
| 07:23.39 | laichzeit | Qwell, is that a macro? |
| 07:23.46 | Qwell | laichzeit: it's an application |
| 07:23.53 | laichzeit | ok |
| 07:24.00 | Qwell | laichzeit: show applications like db |
| 07:24.27 | astra^^ | xtrvd:.thanx dude.. am new to asterisk nw i want to set up astersik on my server |
| 07:24.36 | xtrvd | Question, is an individual line coming from a T1 able to handle fax just as easily as it can voice? |
| 07:24.39 | astra^^ | i worked on other server.. |
| 07:25.15 | laichzeit | Qwell, you have any idea what package I need to build for that? |
| 07:25.23 | Qwell | it's part of the core |
| 07:25.26 | xtrvd | astra^^: Have you downloaded the source and compiled it yet? |
| 07:25.28 | FuriousGeorge | astra^^: thats like going to#apache and saying you wanna be a web designer |
| 07:25.30 | FuriousGeorge | ~docs |
| 07:25.31 | jbot | docs is probably probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 07:25.50 | FuriousGeorge | i suggest voip-info |
| 07:26.02 | laichzeit | ah, thanks Qwell |
| 07:26.32 | FuriousGeorge | astra^^: grab a sip client, a free voip account (like one from sipphone.com) and get some people to log into your server |
| 07:26.49 | astra^^ | FuriousGeorge > am sorry but i am, nt so well in tis subject.. i am learning . and i tot u are the masters u could help me out sir |
| 07:27.05 | astra^^ | i have worked on n2p 6 months ago |
| 07:27.06 | xtrvd | astra^^: Have you downloaded the source and compiled it yet? If not, I recommend going to: http://www.asterisk.org/download and downloading asterisk first. |
| 07:27.08 | FuriousGeorge | lol, flattery will get you no where |
| 07:27.36 | FuriousGeorge | astra^^: there is a lot to learn, probably the first thing would be how to get a free sip client and a free sip account registered with your asterisk server |
| 07:28.01 | xtrvd | astra^^: Secondly, do what FuriousGeorge is recommending. |
| 07:28.23 | xtrvd | And feel free to come back for more redirection after you have completed those two simple steps. |
| 07:28.26 | Winkie | quick question before i go to bed, anyone know how to ensure a queue will not ring someone who is already on a call |
| 07:28.31 | Winkie | regardless of the number of queue members available |
| 07:28.44 | xtrvd | Set the phone to disable call waiting? |
| 07:28.52 | xtrvd | Oh! |
| 07:28.56 | xtrvd | Check the status of the call |
| 07:28.57 | astra^^ | thanx a lot sir.. i'll do it and cme bk.. thanx.. |
| 07:29.06 | Winkie | haha, i can't do that i'm afraid |
| 07:29.10 | Qwell | Winkie: There is a patch on the tracker I think |
| 07:29.26 | Winkie | Qwell: oh really? that'd be interesting, i've already patched the queue to include autofill |
| 07:29.37 | Winkie | any idea how i could check it out? |
| 07:29.43 | astra^^ | xtrvd: can u log on to my server .... :) |
| 07:29.56 | Qwell | Winkie: bugs.digium.com - you'll have to find it |
| 07:29.57 | xtrvd | Winkie: My recommendation would have been: Goto(XXX-${DIALSTATUS} |
| 07:30.27 | FuriousGeorge | Qwell: and whats this new echocan algorithm you mentioned before? is it something i uncomment in one of the source files? |
| 07:30.30 | Winkie | xtrvd: it's a queue? |
| 07:30.31 | xtrvd | astra^^: I'm afraid I don't have the time, as I am leaving for the evening. |
| 07:30.35 | Winkie | how the heck would i do that in a queue? |
| 07:30.43 | xtrvd | Winkie: I stand corrected.... Forgot the queue part. |
| 07:30.44 | xtrvd | Sorry |
| 07:31.01 | astra^^ | xtrvd: ohh... thanx anyway sir.. wen will u be back ... |
| 07:31.06 | *** join/#asterisk SibRphrek (i=SibrPhre@user-12lccke.cable.mindspring.com) |
| 07:31.25 | Qwell | FuriousGeorge: think so |
| 07:31.29 | xtrvd | astra^^: Probably in about 20 or so hours. |
| 07:31.54 | FuriousGeorge | astra^^: try to call us toll free numbers first |
| 07:32.01 | FuriousGeorge | that should keep you busy for 20 hours |
| 07:32.10 | xtrvd | =) |
| 07:32.18 | astra^^ | ohh.. ok .. sir.. :( |
| 07:33.58 | xtrvd | Alright, I have a quick physical configuration question before I get out of everyone's hair tonight. If I am to interface a T1 line into a T1 PCI card in an Asterisk box, and then use SIP phones at the office in question to interface with the asterisk card, how does one interface any fax lines at such a location? Just use FXS ports of a TDM400P (or a ATA)? |
| 07:34.00 | Winkie | Qwell: any idea what i'd search for with this? i'm having no luck, i found the autofill patch i made myself though :) |
| 07:34.09 | Winkie | xtrvd: or use app_rxfax |
| 07:34.12 | Winkie | and txfax |
| 07:34.26 | X-Rob_ | xtrvd, bad news. Fax-over-IP sucks. |
| 07:34.28 | laichzeit | thid DBput function, does it require me to go and create a table in the astdb first? |
| 07:34.37 | Qwell | laichzeit: don't think so |
| 07:34.41 | laichzeit | cool |
| 07:34.45 | xtrvd | X-Rob_: I would be using a T1 connecting directly to my telco. |
| 07:34.53 | X-Rob_ | xtrvd, fax-over-ip sucks. |
| 07:35.12 | Winkie | X-Rob_: there's no problem the way he's describing it |
| 07:35.24 | X-Rob_ | Yes there is. TDM's don't handle data well. |
| 07:35.28 | xtrvd | I'm using standard voice lines... not IP. |
| 07:35.45 | X-Rob_ | you need a channel bank to do fax properly |
| 07:35.48 | Winkie | he said or an ATA, he can buy what's known to work, or use the fax stuff if he needs to |
| 07:35.56 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 07:35.57 | Qwell | http://bugs.digium.com/view.php?id=6111 and http://bugs.digium.com/view.php?id=6315 |
| 07:36.00 | Qwell | Winkie: ^^ |
| 07:36.15 | Qwell | already committed |
| 07:37.13 | xtrvd | I'm just trying to figure out how to interface an asterisk setup which has a direct t1 to a telco to use faxes. |
| 07:37.58 | X-Rob_ | xtrvd, the way that works is to use a two-port PRI card, have your T1 coming in on one port, and the other port having analog lines connected to it with a channel bank. |
| 07:38.02 | YoMama | xtrvd: say that again? |
| 07:38.06 | Winkie | Qwell: yeah i just found it |
| 07:38.18 | Qwell | Winkie: so, just use the call limit stuff |
| 07:38.19 | Winkie | it looks like it was too late to make 1.2.5 |
| 07:38.27 | YoMama | xtrvd: is it a pri or a channelized T1? |
| 07:38.36 | Qwell | Winkie: backport it |
| 07:38.52 | xtrvd | YoMama: I haven't got it yet, so what ever I would be using to replace 13 analog telephone lines. |
| 07:39.02 | Winkie | Qwell: i will, i'm just saying the commit is of no use, and the call limit stuff is extra-queue? |
| 07:39.21 | YoMama | xtrvd: price will be the deciding factor |
| 07:39.34 | YoMama | xtrvd: channelized T1s are not very fun to set up though |
| 07:39.41 | Qwell | Winkie: call limit stuff is in sip.conf |
| 07:39.54 | xtrvd | What exactly is the difference? (Will accept explanation, or link to one) |
| 07:40.08 | YoMama | xtrvd: let's say you went PRI...u could just hang a fax machine off of a FXS port and pick one of the DIDs to be your fax line |
| 07:40.19 | Winkie | Qwell: yeah i thought so, they need to be able to have call waiting also |
| 07:40.28 | YoMama | or...even better..u could install the fax software that works with asterisk and have it send the faxes to your email |
| 07:40.34 | Winkie | (just in case we have important clients laffo) |
| 07:40.36 | xtrvd | YoMama: Alright, I understand that much. It sounds easy enough with the PRI. |
| 07:40.36 | X-Rob_ | YoMama, please don't encourage him to use a TDM4 for fax, nor an ATA. |
| 07:40.45 | YoMama | noo..not an ATA |
| 07:41.12 | YoMama | X-Rob: what's wrongwith the TDM4 FXS ports and a fax machine? |
| 07:41.41 | Ukyo | heck, I got a pap2, hooked it to my * server, with voice pulse. fax no problem |
| 07:41.46 | Ukyo | no special settings |
| 07:42.09 | X-Rob_ | YoMama, it doesn't work reliably, that's what's wrong with it 8) |
| 07:42.36 | YoMama | nothing is reliable about fax |
| 07:42.41 | X-Rob_ | (I should point out, that's _exactly_ how I do it, so I know it doesn't work reliably. It works 'ok', but not 'well') |
| 07:42.42 | YoMama | to begin with |
| 07:42.51 | Ukyo | I have faxed over 100 pages with it w/o problem :P |
| 07:43.17 | xtrvd | Ukyo: I've got one in my hand right now... I just opened the box last week. |
| 07:43.30 | YoMama | xtrvd: typically...if u want fax relability...X-Rob is right...u don't wanna pump it thru any digital system |
| 07:43.34 | Ukyo | takes all of a moment to unlock :) |
| 07:43.43 | Ukyo | had it configed for asterisk in 4 minutes |
| 07:44.00 | xtrvd | Okay, so with a Fax, the *best* way to do it is to isolate my Fax channels to be seperate of my incoming T1? |
| 07:44.16 | YoMama | xtrvd: how many fax lines do u have? |
| 07:44.20 | xtrvd | Just 2. |
| 07:44.25 | xtrvd | 2/13 |
| 07:44.28 | YoMama | ah |
| 07:44.57 | xtrvd | it's a retail business, so there isn't much in the way of a lot of desk jobs... so fax to email isn't much of an option. |
| 07:45.23 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 07:45.24 | YoMama | channelized T1s are not truly digital...they're analog lines that are assigned on various channels along a T1 line |
| 07:45.31 | wasim | ugh ... |
| 07:45.36 | wasim | i need aspirin |
| 07:45.37 | wasim | Mar 7 12:43:42 WARNING[26469]: app_dial.c:1020 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 07:45.47 | wasim | when i know i have a few channels free |
| 07:45.50 | YoMama | xtrvd: there a lot of outgoing faxes? |
| 07:45.58 | Ukyo | wasim: my new install of * keeps giving me dial exec full, but not the zap msg |
| 07:46.17 | YoMama | xtrvd: or is it mostly incoming? |
| 07:46.26 | Ukyo | if I pick up and hang up a few times with the 7960, it will do that |
| 07:46.26 | xtrvd | YoMama: Both ways, |
| 07:46.46 | tuxinator_linuxM | http://www.sellvoip.net looks like a good price for origination |
| 07:47.07 | wasim | Ukyo: how many channels do you have up... we're getting this at about 84 channels up out of 63 |
| 07:47.14 | wasim | err, 93 |
| 07:47.31 | YoMama | xtrvd: if u wanna avoid all issues..keep your fax lines |
| 07:47.42 | wasim | and it sucks, coz the switch thinks that there are still come channels available, so it keeps routing calls to this |
| 07:47.46 | YoMama | xtrvd: look at replacing the 11 lines with another solution |
| 07:47.59 | wasim | but 84 channels of g729 on mgcp ua isn't bad :) |
| 07:48.00 | xtrvd | YoMama: Thanks, that's a cut-through answer. I don't want to have problems in the long run. |
| 07:48.05 | wasim | 35k calls since this morning |
| 07:48.20 | xtrvd | YoMama: I should still be able to use 11 lines on a T1 for near equal cost. |
| 07:48.40 | Ukyo | wasim: as I said, mine is a different error tho |
| 07:48.44 | X-Rob_ | xtrvd, perfect. That's the best way to do it. |
| 07:48.50 | Ukyo | because I amn only using voip |
| 07:49.05 | YoMama | tuxinator: i don't see how the local did at that place is a good deal |
| 07:49.05 | *** join/#asterisk vgster (n=vg@host217-45-221-53.in-addr.btopenworld.com) |
| 07:49.20 | wasim | Ukyo: yeah, we're making a new channel driver |
| 07:49.25 | YoMama | xtrvd: maybe..i dunno...just get a quote and compare |
| 07:49.27 | Ukyo | yeah, esp for the price of did's |
| 07:49.31 | wasim | but the zap part is bugging |
| 07:49.35 | Ukyo | hm |
| 07:49.41 | xtrvd | What do you guys usually find is the number of standard analog lines which costs the same as a T1? |
| 07:49.49 | tuxinator_linuxM | YoMama: well, I have a few numbers that I need to have incoming only for a few months... and I don't have a PRI |
| 07:50.11 | YoMama | tuxinator: existing? |
| 07:50.23 | tuxinator_linuxM | yes, need to be ported |
| 07:50.26 | YoMama | tuxinator: do u need to port #'s or u just need inbound |
| 07:50.28 | YoMama | oh ok |
| 07:50.36 | tuxinator_linuxM | port and inbound |
| 07:51.20 | tuxinator_linuxM | for two numbers, port and both ways for 3 numbers... but those there can wait, i just need to hold them somewhere |
| 07:51.27 | wasim | we're approaching a BHCA of about 6k |
| 07:51.31 | YoMama | i just think it's goofy that u pay inbound fees |
| 07:51.46 | YoMama | unless.... |
| 07:51.46 | Winkie | anyone mind quickly telling me how the hell i get some debugging info up on the cli? |
| 07:51.56 | Ukyo | YoMama: alot of voip providers charge inbound |
| 07:51.58 | Ukyo | like asterlink |
| 07:52.15 | YoMama | I have an inbound "DID" that doesn't charge inbound |
| 07:52.29 | tuxinator_linuxM | YoMama: what's the monthy fee? |
| 07:52.31 | YoMama | they didn't offer to port #'s...but still |
| 07:52.56 | YoMama | tuxinator: $12 for 3 months |
| 07:53.15 | tuxinator_linuxM | not too bad of a price |
| 07:53.38 | YoMama | no |
| 07:54.37 | tuxinator_linuxM | but not porting.. too bad |
| 07:54.46 | Ukyo | yes, horrible price. everything should be free! -_- |
| 07:55.27 | [av]bani | http://www.faqs.org/qa/rfcc-1577.html |
| 07:56.23 | encode | if you were planning on setting up a remote voicemail server, and the local pbx sets outgoing callerids to be the same no matter the originating extension, how would you go about doing it? |
| 07:56.27 | Ukyo | av: lol |
| 07:57.12 | X-Rob_ | encode: I'd cry. |
| 07:57.14 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:57.36 | tuxinator_linuxM | [av]bani: is that guy for real? |
| 07:57.56 | xtrvd | Argh, one more question. If getting a T1 PRI from the local Telco, can I utilize the unused portion of the line for data transfer with minimal effort? |
| 07:58.35 | YoMama | xtrvd: minimal? no |
| 07:58.44 | xtrvd | It's a hassle? |
| 07:58.50 | YoMama | xtrvd: many phone companies offer voice/data over one "pipe" |
| 07:59.06 | encode | X-Rob_: hehe yeah, i can't think of any way to do it either |
| 07:59.12 | xtrvd | Would I be requiring seperate hardware... or?... |
| 07:59.14 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 08:00.38 | YoMama | xtrvd: the phone company would put in a channel bank |
| 08:00.46 | YoMama | certain channels would be desginated data |
| 08:00.52 | YoMama | the others would be voice |
| 08:01.12 | xtrvd | YoMama: Thank you. =) |
| 08:01.23 | xevo | xtrvd: you can get in dynamically allocated as well |
| 08:01.45 | YoMama | xevo: yes u can..but now we're talking some good fun |
| 08:01.51 | xtrvd | =| |
| 08:02.01 | xtrvd | You guys sure know how to make me feel insignificant. |
| 08:02.37 | xevo | ahh... it's easy... carriers have some good products out there right now |
| 08:02.43 | xtrvd | Thanks for everybody's help tonight, and goodnight. =) |
| 08:02.55 | dpryo | *yawns* |
| 08:02.57 | dpryo | Good morning :) |
| 08:02.58 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-118.sd.sd.cox.net) |
| 08:03.26 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 08:10.44 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-218.claranet.co.uk) |
| 08:14.10 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
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| 08:16.24 | wasim | arrrghhhh ..... |
| 08:17.19 | wasim | we're doing about 6000 calls per hour and this is not funny, to loose 1500 call attempts to stupid zap |
| 08:18.33 | wasim | Mar 7 13:16:52 WARNING[31669]: app_dial.c:1020 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 08:19.13 | trixter | I dont think that is just zap |
| 08:19.16 | trixter | I see that on sip a whole lot |
| 08:19.49 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:22.08 | *** join/#asterisk bweschke (n=bweschke@219.sub-70-193-112.myvzw.com) |
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| 08:30.15 | robin_z | bah ... setting nat=yes in general turned out to be a Bad Plan |
| 08:32.09 | encode | on the bright side...if you screw it up enough...there cant be any emergency phone calls cos no-one can use their phone |
| 08:33.15 | *** join/#asterisk pawal (n=pawal@212.247.14.36) |
| 08:38.00 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 08:41.18 | wasim | i really wish it wouldn't do this, so that the ss7 can route calls to other places |
| 08:41.33 | wasim | but since it won't, and shows 5 channels free, we keep getting calls here |
| 08:41.36 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 08:42.13 | *** join/#asterisk hardwire (n=spencers@209.112.194.39) |
| 08:44.01 | wasim | hahaha ... lol |
| 08:44.07 | *** join/#asterisk wellng (n=welles@61.150.43.114) |
| 08:44.15 | wasim | i renice * to -20 and it starts dropping calls |
| 08:44.24 | wasim | down to 77 from 84 |
| 08:44.33 | wasim | bring prio back to 0 and its ok |
| 08:44.40 | wasim | well, ok to 84, not up to 93 |
| 08:45.46 | wasim | 88! |
| 08:46.34 | *** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.225) |
| 08:46.36 | Kernel_Core | hi all |
| 08:50.54 | [av]bani | ... |
| 08:51.29 | [av]bani | wasim: you cant do that... use -p instead |
| 08:57.25 | robin_z | encode: sadly, one of them had a mobile ... |
| 08:59.40 | *** join/#asterisk jhnjwng (n=wj1918@pool-70-21-166-133.nwrk.east.verizon.net) |
| 09:02.22 | tzafrir | wasim, FWIW, I wonder if you can actually destroy and add channels on the fly |
| 09:02.38 | wasim | can i change a running * to -p |
| 09:02.46 | Kernel_Core | h323 can't forward the call .... |
| 09:02.46 | Kernel_Core | <PROTECTED> |
| 09:02.47 | Kernel_Core | <PROTECTED> |
| 09:02.47 | Kernel_Core | Mar 7 02:56:50 WARNING[7278]: pbx.c:2345 __ast_pbx_run: Channel 'H323/ip$217.218.80.225:1933/3188' sent into invalid extension 's' in context 'default', but no invalid handler |
| 09:02.54 | tzafrir | wasim, not that I know of |
| 09:03.13 | wasim | ok, time to undivert calls from these, wait for call hangup and then restart |
| 09:03.57 | tzafrir | wasim, that is: you can even today destroy channels. But it seems that creating them on the fly by re-reading zapata.conf is not such a big deal. Maybe. |
| 09:04.07 | tzafrir | Anyway, it is a matter of set_prio |
| 09:04.16 | wasim | tzafrir: but my zap channels are fine |
| 09:04.26 | wasim | tzafrir: it * that thinks it can't open it |
| 09:05.15 | tzafrir | wasim, hmmm, take a look at setpriority(2) |
| 09:07.02 | wasim | i hope none of these are long calls, last night we had to wait for 2 hours for a couple of lovers to get off the e1 so we could reset * :( |
| 09:07.12 | tzafrir | You can write a small C program to change the sched priority of a process |
| 09:07.30 | tzafrir | now where was that small command-line app to do that? |
| 09:07.39 | wasim | renice ? |
| 09:09.01 | tzafrir | wasim, also, a quick apt search gives me libbsd-resource-perl |
| 09:13.13 | [av]bani | tzafrir: the problem with that is all the threads. |
| 09:13.22 | [av]bani | tzafrir: better to use a graceful shutdown, then restart with -p |
| 09:15.22 | [av]bani | wasim: 'stop when convenient' will stop * when there is 0 calls |
| 09:15.41 | wasim | [av]bani: yep |
| 09:15.51 | jaike | stop gracefully? |
| 09:16.17 | jaike | whats the difference between the two |
| 09:16.32 | [av]bani | # stop gracefully: Gracefully shut down Asterisk |
| 09:16.32 | [av]bani | # stop now: Shut down Asterisk imediately |
| 09:16.33 | [av]bani | # stop when convenient: Shut down Asterisk at empty call volume |
| 09:16.37 | [av]bani | http://www.voip-info.org/wiki-Asterisk+CLI |
| 09:17.33 | tzafrir | wasim, no. setpriority is what -p inflicts |
| 09:17.58 | tzafrir | changes the "hard" scheduling priority ("real-time priority") |
| 09:18.22 | tzafrir | It means that if Asterisk has anything to do it will have priority over any other userspace program |
| 09:19.13 | tzafrir | As you can see from a simple look at 'top', there is a limit to the extra scheduling boot renicing gives |
| 09:20.29 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
| 09:22.00 | *** join/#asterisk nayyares (n=Nayyar@58.65.151.218) |
| 09:22.23 | nayyares | hi guys |
| 09:22.46 | nayyares | where i can find the PHP AGI of ASTCC? |
| 09:23.52 | tzafrir | BTW: anybody heard of a simple and nice way to test the effects of network latency using the netlink interface or whatever? |
| 09:24.44 | *** join/#asterisk bweschke (n=bweschke@229.sub-70-192-11.myvzw.com) |
| 09:25.09 | wasim | poor people they had 15k unanswered calls which they billed for |
| 09:25.38 | wasim | atleast another first for * |
| 09:25.47 | wasim | not the wrong billing, thats always been there |
| 09:26.00 | wasim | but mgcp user agent mode on a live network :) |
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| 09:30.58 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 09:32.13 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 09:34.09 | Gwynm | Hi guys. I've got Asterisk working nicely and running a test AGI python script. I'm now trying to get it to talk to RAGI. I get 'no application "DeadAgi" for extension'. Can anyone help? |
| 09:34.23 | Gwynm | This is a commercial project, so I'm willing to pay for support if necessary. |
| 09:34.50 | backblue | Gwynm: what its ragi? |
| 09:35.12 | Gwynm | It's an interface between Ruby On Rails and Asterisk. |
| 09:35.14 | tzafrir | ruby agi? |
| 09:35.22 | Gwynm | tzafrir: yes. |
| 09:35.25 | backblue | dont know what's that |
| 09:35.37 | Gwynm | Basically, I've got a server running on port 4573 that speaks AGI. |
| 09:36.12 | Gwynm | So I think I should be able to get to it with exten => 998,2,Agi(agi://127.0.0.1/hello/dialup) |
| 09:36.42 | tzafrir | Basically: Agi(/path/to/file) |
| 09:37.00 | tzafrir | or: Agi(relative/path/to/file) |
| 09:37.01 | Gwynm | tzafrir: But it's not a file... it's a tcp socket with a process at the far end that speaks agi. |
| 09:37.34 | backblue | agi can use agi://IP ? |
| 09:38.09 | Gwynm | backblue: Apparently. If you google for 'fastagi' you'll know as much as I do. |
| 09:38.47 | Gwynm | I'm using a tutorial at http://www.snapvine.com/code/ragi/ragi_tutorial_v1.pdf, which gives an extension line like exten => 998,2,deadagi(agi://127.0.0.1/hello/dialup) |
| 09:38.50 | tzafrir | Gwynm, "no such application" messages are probably unrelated to the ruby end. |
| 09:39.06 | tzafrir | Try: show application deadagi |
| 09:39.19 | tzafrir | in the asterisk CLI. Any chance it wasn't built? |
| 09:39.29 | Gwynm | tzafrir: Now this *is* a stupid question, but I've never actually used the asterisk CLI. Where is it? |
| 09:39.37 | tzafrir | asterisk -r |
| 09:39.45 | Gwynm | It's possible it wasn't built.. I'm using a macosx binary. |
| 09:40.00 | tzafrir | you need to be root, or otherwise with the ability to write to the asterisk socket. |
| 09:40.07 | wasim | and Mar 7 14:38:33 WARNING[5421]: app_dial.c:1020 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 09:40.13 | wasim | still ... even with -p |
| 09:40.15 | Gwynm | *CLI> show application deadagi |
| 09:40.19 | tzafrir | On Mac there should be a nice asterisk console interface, whose name escapes me |
| 09:40.26 | Gwynm | Your application(s) is (are) not registered |
| 09:40.37 | backblue | Gwynm: are you loading the module? |
| 09:40.52 | tzafrir | wasim, does such channel actually exist? What was the Dial command? |
| 09:40.56 | Gwynm | No. I didn't know I needed to - sounds like I missed a basic setup guide somewhere. |
| 09:41.04 | backblue | wasim: zap show channels? |
| 09:41.06 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 09:41.16 | tzafrir | Gwynm, sorry, I'm no osx guy |
| 09:41.16 | backblue | wasim: cat /proc/zaptel/* |
| 09:41.21 | Gwynm | backblue: Can you give me a URL to 'howto load module', or a line of code and the location of a config file to paste it? |
| 09:41.53 | Gwynm | tzafrir: Unfortunately, neither am I - trying to make the switch from Debian. |
| 09:42.05 | tzafrir | Gwynm, bad move |
| 09:42.15 | *** join/#asterisk steveaj (n=steve@82-71-15-37.dsl.in-addr.zen.co.uk) |
| 09:42.23 | backblue | Gwynm: ... cd /usr/lib/asterisk/modules |
| 09:42.53 | Gwynm | backblue: I've got app_agi.so .. is that it? |
| 09:42.57 | backblue | Gwynm: osx or any another unix it's ok, just make shor you know what you are doing |
| 09:43.17 | Gwynm | backblue: no 'app_deadagi' or anything like that. |
| 09:43.25 | backblue | Gwynm: i never used, let me take a look. |
| 09:43.55 | backblue | Gwynm: put this in your modules.conf |
| 09:43.56 | backblue | load => res_agi.so |
| 09:43.56 | backblue | res_agi.so=yes |
| 09:44.08 | backblue | and try it again. |
| 09:44.10 | viperdude | Gwynm: osx is based on FreeBSD if I remember correctly and FreeBSD is not that asterisk friendly, timer related issues, switch to linux if poss |
| 09:44.15 | tzafrir | I actually tend to aimply autoload them all |
| 09:44.34 | backblue | viperdude: asterisk runs fine in osx. |
| 09:44.37 | Gwynm | viperdude: This is just the development environment. I'm hosting on linux. |
| 09:44.41 | tzafrir | osx is not that close to freebsd |
| 09:45.10 | viperdude | ok well no experience of osx but in our call center FreeBSD chocked |
| 09:45.14 | Gwynm | I have autoload=yes in modules.conf |
| 09:45.21 | backblue | tzafrir: thats a bad idea, its pretty easy to know what you need, dont need to load them all, but its your machine, your cpu :P |
| 09:46.11 | backblue | why use osx, when you have linux, just for hosting asterisk? i think you should have the same developement eviroment than in production. |
| 09:46.28 | backblue | if you will host in linux, should develop in linux |
| 09:46.28 | Gwynm | backblue: I added those two lines. No change :/ |
| 09:46.29 | viperdude | backblue: i agree |
| 09:46.52 | backblue | Gwynm: you compiled asterisk or used some package? |
| 09:47.04 | Gwynm | backblue: Used 'stable' OSX package. |
| 09:47.07 | backblue | Gwynm: but you have res_agi.so? |
| 09:47.22 | backblue | stable? so which asterisk version do have? |
| 09:47.26 | Gwynm | No. I have app_agi.so . |
| 09:47.32 | Gwynm | But no res_agi.sp . |
| 09:47.44 | backblue | i dont have app_agi :P |
| 09:47.50 | backblue | and i use agi's |
| 09:47.55 | wasim | trying with 4e1 |
| 09:47.56 | Gwynm | CVS-10/28/03 |
| 09:47.57 | backblue | Gwynm: what version do you have? |
| 09:48.22 | wasim | top - 14:46:50 up 19:50, 10 users, load average: 19.90, 10.36, 5.46 |
| 09:48.26 | wasim | :S |
| 09:48.36 | Gwynm | Oh, shit. 'Asterisk cmd DeadAGI' in the wiki says "Added to CVS 2004-03-03". |
| 09:49.22 | backblue | CVS 10-28-03 its quite old dont you think? :o |
| 09:49.33 | backblue | Gwynm: compile asterisk. |
| 09:50.18 | Gwynm | -ibook-g4:~ lyn$ gcc |
| 09:50.18 | Gwynm | -bash: gcc: command not found |
| 09:50.29 | Gwynm | It's going to be a long haul :/. But I think I'll have to. |
| 09:50.49 | backblue | install apple developers kit |
| 09:50.58 | backblue | you have all you need i think |
| 09:51.25 | backblue | but your system will be all messed up, you should use linux, install linux on your mac -> www.gentoo.org |
| 09:51.36 | backblue | i use linux on my ibook |
| 09:52.31 | Gwynm | backblue: What's the compatibility like? Debian on my Toshiba was an uphill battle.. hence the switch. |
| 09:52.58 | dpryo | backblue: I understand airport extreme now works perfectly under linux? |
| 09:53.35 | Enth | Guys, what are the credentials I need to enter in sip.conf and extensions.conf if I wanted to add a SIP user outside NAT in another office? |
| 09:53.44 | backblue | dpryo: i dont have airport extreme. i have airport normal. dont know the actual stat of airport extreme driver, sorry. |
| 09:53.45 | Enth | and if I'm also using a dyndns |
| 09:53.57 | backblue | Gwynm: install gentoo, you will like a lot. |
| 09:54.07 | backblue | it's pretty nice, for developers. |
| 09:54.39 | UdontKnow | hmmm |
| 09:54.47 | UdontKnow | anyone used a zinwell zt-1000 ata? |
| 09:54.58 | UdontKnow | its not trying to register... |
| 09:55.01 | UdontKnow | weird |
| 09:55.54 | *** join/#asterisk hgaillac (n=Harry@186.14.119-80.rev.gaoland.net) |
| 09:57.00 | hgaillac | Hello, Is there a way to set an an outbound proxy in sip.conf |
| 09:57.08 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 10:00.17 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 10:01.12 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:01.46 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 10:01.47 | UdontKnow | yay |
| 10:01.49 | UdontKnow | connected |
| 10:04.48 | tzafrir | actually, Debian will work there just as well... |
| 10:05.30 | tzafrir | I have Debian work nicely on my laptop (actually third laptop I confiugred with Debian. no major problem with either of the three) |
| 10:08.00 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 10:10.32 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:10.45 | Gwynm | Alright, SOLVED, (for the benefit of the log files): If you're trying to use Asterisk, Mac OS X and Deadagi (eg for RAGI), you need a recent version of Asterisk. Compile your own or use http://www.sunrise-tel.com/asterisk-on-macosx.html . Don't use the "stable" version from 2003 or you'll waste hours :(. |
| 10:10.57 | Gwynm | Got it going with the 1.0.7 binary. |
| 10:11.09 | Gwynm | Now, of course, my actual AGI script is dying, but that's a ruby problem. |
| 10:11.28 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 10:12.35 | nayyares | is there any GPL softphone? |
| 10:12.44 | UdontKnow | of course |
| 10:12.47 | UdontKnow | several |
| 10:13.06 | UdontKnow | linphone and kphone are popular softphones |
| 10:13.13 | littlejohn | kphone, ekiga, gnomemeeting |
| 10:13.22 | nayyares | UdontKnow, which one is the best ! |
| 10:13.22 | UdontKnow | openwengo is published by wengo.fr folks |
| 10:13.35 | UdontKnow | nayyares: depends on your needs |
| 10:13.39 | littlejohn | i find ekiga a nice one |
| 10:13.55 | nayyares | hmm |
| 10:14.10 | *** join/#asterisk fulgas (n=fulgas@207.226.175.10) |
| 10:14.17 | UdontKnow | there are text-user-interface softphones too |
| 10:14.40 | nayyares | UdontKnow, wawoo |
| 10:15.38 | Gwynm | Woot, RAGI is working. This is awesome. Thanks for you help, guys. |
| 10:17.53 | shido6 | any c coders? |
| 10:18.27 | nayyares | shido6, #c :) |
| 10:23.15 | nayyares | littlejohn, ekiga is not ported to CentOS/RHEL yet :( |
| 10:25.51 | Enth | Guys, what are the credentials I need to enter in sip.conf and extensions.conf if my * is inside NAT and if I wanted to add a SIP user inside NAT in another office, and if i'm using dyndns ? |
| 10:29.13 | littlejohn | nayyares, download and compile ;) |
| 10:29.29 | *** part/#asterisk jaike (n=a@203.131.137.76) |
| 10:29.54 | littlejohn | enth, nap asterisk and nat sip client won't work |
| 10:30.08 | *** join/#asterisk _deg_ (n=deg@200.250.222.8) |
| 10:30.09 | Enth | oh |
| 10:30.13 | Enth | why not? |
| 10:30.31 | Enth | even if port forwarding is done? |
| 10:34.44 | Enth | so what's the solution for a scenario like that? |
| 10:36.03 | nayyares | i have single PC in my room, and want to play with all sort of configuration supported by Asterisk, i hope if i use multiple IPs on single machine, Asterisk w'nt mind it :) |
| 10:37.21 | *** join/#asterisk Modcuts (n=bob@proporta.gotadsl.co.uk) |
| 10:39.10 | wasim | ok, i need to tweak the kernel for optimum performance |
| 10:39.16 | wasim | i'll write this up for big-asterisk also |
| 10:39.40 | wasim | so, timing, should it be 100,250 or 1000 hz |
| 10:40.44 | wasim | what about PCI Access Mode .. direct? |
| 10:44.30 | tzafrir | wasim, IIRC this is for the case that the bios provides bogus data. But I'm not sure |
| 10:47.56 | wasim | ok, trying new one, and gcc optimizations while making kernel for a dual xeon |
| 10:48.17 | wasim | like should i get sse3 in the kernel make time as well, or not worry about it |
| 10:48.54 | tzafrir | nayyares, asterisk will. Unless you tell each asterisk copy to listen on a separate IP |
| 10:49.18 | littlejohn | Enth, on voip-info.org search for asterisk nat |
| 10:49.39 | Enth | that didnt help unfortunately :/ |
| 10:49.46 | Enth | I had a look. |
| 10:49.48 | Enth | :( |
| 10:51.07 | *** join/#asterisk mut (n=animenod@65.111.201.79) |
| 10:51.25 | mut | hey all |
| 10:51.37 | mut | anyone know what'de cause far end distortion of my voice? |
| 10:51.44 | mut | calling out a pri on a 405p |
| 10:52.22 | wasim | tzafrir: another concern is the cpu load on call setup |
| 10:52.50 | wasim | tzafrir: i'm sure the box will handle 120 calls once they are up, but handling 100 calls and setting up/tearning down another 20 every second isn't cheap |
| 10:53.41 | kippi | whats the name of the asterisk live boot cd? |
| 10:54.15 | tzafrir | kippi, I think that there are two by now... |
| 10:54.48 | kippi | tzafrir: do u have a name of them? or has someone used one? |
| 10:55.06 | tzafrir | kippi, I'm too lazy to check the wiki... |
| 10:55.13 | tzafrir | Haven't used any |
| 10:55.58 | kippi | ah |
| 10:56.37 | mut | these tellab echo cans suck |
| 10:56.38 | mut | ush |
| 10:56.40 | mut | ugh |
| 11:08.40 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 11:20.02 | tzafrir | When I first heard the term "echo can" I thought of a tin can that amplifies the echo |
| 11:33.26 | *** join/#asterisk Bambr (n=Bambr@213-35-233-174-dsl.end.estpak.ee) |
| 11:34.52 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 11:36.02 | Kumbang | hello guys, does anyone have problem with busydetect with tdm24xxp |
| 11:36.46 | Kumbang | busydetect in wctdm24xxp seems not working |
| 11:38.12 | kippi | has anyone used the avaya 4620 handsets with asterisk |
| 11:38.21 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 11:41.37 | sl16 | does anyone know what is the variable where the seconds are counting until getting to X : Dial(SIP/11,X) |
| 11:42.15 | sl16 | i want to detect on what second the human will pick up the phone |
| 11:48.40 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 11:48.50 | PoWeRKiLL | hi |
| 11:49.02 | PoWeRKiLL | Is there a uppercase to lowercase function in asterisk dialplan function ? |
| 11:49.53 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 11:53.00 | heison | has iaxtel.com been abandoned? |
| 11:54.52 | *** join/#asterisk gwynm (n=Gwyn@ppp158-45.lns3.adl2.internode.on.net) |
| 11:55.49 | gwynm | Hey guys. I've got a swift.agi that's basically just 'echo "stream file /tmp/f"', and /tmp/f.wav exists. When I dial in it doesn't play. There's nothing /var/log/asterisk/messages. Where should I start looking? |
| 11:56.35 | gwynm | (as before, I'm willing to pay for support) |
| 12:06.32 | fugitivo | gwynm: what do you need to do? |
| 12:12.00 | gwynm | fugitivo: Eventually I'm trying to get swift.agi to the point where I can do TTS. I've tracked it down a bit further: |
| 12:12.16 | gwynm | Mar 7 22:40:54 WARNING[314]: Unexpected header size 16 |
| 12:12.17 | gwynm | Mar 7 22:40:54 WARNING[314]: Unable to open fd on /tmp/f.WAV |
| 12:12.33 | gwynm | It looks like I'm getting bad wav files from swift. I generated that file with: |
| 12:12.46 | gwynm | sudo swift -o f.wav -p audio/channels=1,audio/sampling-rate=8000 "testing" |
| 12:13.14 | gwynm | (wav file up at dezyne.net/f.wav) |
| 12:14.15 | gwynm | According to 'file', it's: f.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 12:16.35 | fugitivo | if you play the file normally, can you hear sound? |
| 12:16.57 | gwynm | Yes. |
| 12:17.46 | *** join/#asterisk Jedirl (n=hhgds4@213.162.200.226) |
| 12:17.48 | Jedirl | Hello |
| 12:18.06 | Jedirl | I'm having very strange problems with the GotoIf command |
| 12:18.19 | Jedirl | it *always* goes true, even if the condition isn't |
| 12:18.33 | Jedirl | (I've double checked with NoOp's) |
| 12:19.55 | *** join/#asterisk fulgas (n=fulgas@207.226.175.10) |
| 12:20.17 | Druken | you probably have a fault in the syntax |
| 12:20.27 | Jedirl | http://pastebin.com/588743 <- here is it |
| 12:21.11 | Jedirl | it always goes 120, even if NoOp shows ${ESTADO} ok |
| 12:22.11 | Druken | perhaps the space is screwing it up ? |
| 12:22.15 | Druken | otherwise it looks good |
| 12:23.25 | Jedirl | I've finally found the problem... :/// it needs a $ just before the [ |
| 12:23.41 | Druken | oh shit.. yeah... hehehe |
| 12:23.45 | Jedirl | :)) |
| 12:23.55 | Druken | i totally missed that too... |
| 12:24.14 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 12:25.45 | *** join/#asterisk fjean (n=fjean@201009186209.user.veloxzone.com.br) |
| 12:26.11 | fjean | hello, anyone knows if the g729 license is per call or per channel ? :- ) |
| 12:26.25 | Druken | channel afaik |
| 12:26.26 | Jedirl | per concurrent use |
| 12:26.31 | Jedirl | oh |
| 12:26.34 | fjean | ok |
| 12:26.34 | Jedirl | maybe I'm wrong then |
| 12:26.50 | Druken | Jedirl: concurrent use would be channel :) |
| 12:27.00 | Jedirl | ops |
| 12:27.01 | Jedirl | :) |
| 12:27.04 | fjean | hehe |
| 12:27.18 | gwynm | Does anyone have a .wav file that they know Asterisk can play? |
| 12:27.32 | fjean | I have 25 of them and 'sip show channels' shows 24 channels, am I in trouble ? |
| 12:27.37 | vgster | cant you convert one of the gsm files? |
| 12:27.39 | fjean | :-) |
| 12:28.16 | Druken | fjean: i see 1 left... looks like your in good shape :) hehehehe |
| 12:28.31 | fjean | druken, well well |
| 12:28.33 | gwynm | vgster: The idea is to remove the probable broken part, which is me.. |
| 12:28.48 | vgster | ok |
| 12:28.54 | Druken | i would i could admit i'm broken..... |
| 12:29.07 | Druken | i'll just continue to blame the world for my imperfections |
| 12:29.12 | Druken | :) hehe |
| 12:29.45 | gwynm | Interestingly, if I add audio/encoding=pcm8 to the swift string, the error goes away.. but still nothing plays. |
| 12:29.56 | fjean | I thought show g729 would give the current number of licence in use...but... |
| 12:30.14 | vgster | it does doesnt it |
| 12:30.23 | fjean | not really |
| 12:30.33 | fjean | sgway*CLI> show g729 |
| 12:30.34 | fjean | 0/0 encoders/decoders of 25 licensed channels are currently in use |
| 12:30.45 | vgster | so 25 licenses channels with none using g729? |
| 12:31.05 | Druken | it would seem so |
| 12:31.21 | I-MOD | that just means you aren't using your licenses |
| 12:31.37 | I-MOD | they're only in use if you're on a call that uses them |
| 12:31.41 | Druken | how much is the g729 ? |
| 12:31.46 | vgster | $10 ap op |
| 12:32.21 | Druken | $10 for 25 channels? that's not t=so bad |
| 12:32.29 | vgster | per channel |
| 12:32.34 | Druken | ouch.... |
| 12:32.37 | vgster | indeed |
| 12:32.38 | Druken | rip me a new asshole |
| 12:32.41 | fjean | well if i do sip show channels i see 22 of them ! |
| 12:32.41 | vgster | ok |
| 12:32.54 | vgster | and your point is? |
| 12:33.02 | Druken | fjean: you only use the license when a call is being placed |
| 12:33.09 | Druken | the client doesn't hold the license |
| 12:33.42 | fjean | well you see I am tryning to see of I must buy new licenses, and from what I see on the system I am not sure |
| 12:33.46 | Druken | oh.. you have 22 CHANNELS open now? |
| 12:33.51 | fjean | yes |
| 12:33.59 | fjean | 10.1.1.111 968823 fea3bf24ac5 00101/09269 g729 |
| 12:34.09 | Enth | does anyone know which softphones support IAX2? |
| 12:34.11 | Druken | hmm.. |
| 12:34.12 | fjean | I have 23 of these now |
| 12:34.17 | Enth | Drunken! sup |
| 12:34.33 | Druken | hey Enth |
| 12:34.38 | Enth | :) |
| 12:35.09 | Enth | how's it goin? |
| 12:35.11 | Druken | dun ask me about iax2 softphones.. cause i dun use softphones, can't stand them :) |
| 12:35.18 | fjean | I will call digium and let you know, wait a sec |
| 12:35.25 | Enth | i know :/ |
| 12:35.25 | Druken | plus i don't reccomend iax2 on the outside... |
| 12:35.31 | Enth | hrmmm |
| 12:35.31 | fjean | enth: firefly |
| 12:35.32 | Enth | well |
| 12:35.36 | I-MOD | digium doesnt open for another 25 mins |
| 12:35.40 | RoyK | anyone here using spandsp with 1.2.5? |
| 12:35.45 | Druken | i only got 1 iax2 device, and it's sitting on the desk next to me |
| 12:35.47 | _Sam-- | idefisk = iax softphone |
| 12:35.50 | fjean | I-mod, ok |
| 12:35.53 | _Sam-- | www.asteriskguru.com |
| 12:36.07 | vgster | atcom do iax hardphones |
| 12:36.14 | Enth | ah yes |
| 12:36.41 | Druken | i treat voip like i treat sex... go hard or go home :) |
| 12:36.53 | vgster | dont you get drunk first? |
| 12:37.03 | Druken | don't drink... so no |
| 12:37.21 | iDunno | sounds careless |
| 12:37.25 | iDunno | you really should drink |
| 12:37.26 | *** join/#asterisk f7950qs0 (n=redhotte@61.17.213.99) |
| 12:37.27 | I-MOD | Enth: iaxcomm |
| 12:37.30 | iDunno | otherwise you'll dehydrate |
| 12:37.35 | f7950qs0 | hi everyone |
| 12:37.42 | vgster | yes 6 pints of wife beater please |
| 12:37.55 | Druken | iDunno: i don't drink intoxicating beverages |
| 12:37.59 | gwynm | Has anyone installed asterisk non-root, in a user account? |
| 12:38.10 | iDunno | no tea, coffee or hot chocolate either then? :) |
| 12:38.11 | gwynm | I have shared space on a server, so it'd have to be inside /home/gwyn .. |
| 12:38.26 | Druken | no tea or coffee, but i do enjoy hot chocolate |
| 12:38.32 | iDunno | caffeine! |
| 12:38.37 | iDunno | intoxicating! |
| 12:38.38 | Druken | coke is my drink of choice :) |
| 12:38.47 | iDunno | caffeine! intoxicating! ;) |
| 12:38.59 | f7950qs0 | I have an internet calling center and want to monitor the minutes used in my cabins that make calls through ATAs how do I do that |
| 12:39.06 | wasim | queuemetrics |
| 12:39.09 | Enth | Drunken: Well, I need yuor expert opnion here: Scenario is that I have a broadband at home with dynamic ip. Have NAT running. Got Asterisk running on a box internally. My colleague is external, is also behind NAT. I want both of us to be able to call eachother. Issues faced is that I am behind NAT and so is he, we have port forwarded SIP. |
| 12:39.35 | f7950qs0 | i just want a computer to track the minutes called out from an ATA |
| 12:39.57 | wasim | f7950qs0: setAccountCode and then do a query using that |
| 12:40.05 | Druken | Enth: you should know my stance on asterisk behind a nat.... |
| 12:40.10 | Druken | don't do it :) |
| 12:40.29 | Enth | true |
| 12:40.31 | Enth | hehe |
| 12:40.49 | f7950qs0 | wasim thanks, is there any way to do it without using asterisk? asterisk is too powerful for such a small cause |
| 12:41.00 | RoyK | f7950qs0: don't trust wasim |
| 12:41.18 | Enth | bah i'll just get a hosted server. |
| 12:41.19 | Enth | heh |
| 12:41.32 | *** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.140) |
| 12:41.34 | Kernel_Core | hi all |
| 12:41.43 | f7950qs0 | Enth a hosted server? |
| 12:41.49 | f7950qs0 | I use different ITSPs |
| 12:42.08 | RoyK | anyone here using spandsp with 1.2.5? |
| 12:42.26 | RoyK | i just get |
| 12:42.26 | RoyK | Mar 7 13:41:57 WARNING[3207]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: t30_completion_code_to_str |
| 12:42.31 | Enth | heh |
| 12:42.36 | f7950qs0 | I saw a callshop billing software but it is so complicated I couldn't get it to run even after 20 days of hard work |
| 12:44.02 | Druken | Enth: where are you btw? |
| 12:44.55 | f7950qs0 | how do you guys become so good at asterisk and linux and all that by the way just asking ! |
| 12:44.58 | Enth | Drunken: .uk |
| 12:45.12 | Druken | f7950qs0: practice |
| 12:45.14 | Druken | hehe |
| 12:45.19 | Enth | heh. |
| 12:45.34 | Druken | asterisk requires a certain..... learning curve |
| 12:45.49 | Enth | yep and did I mention that it also needs resources? |
| 12:45.52 | Druken | an idiocy if you may... |
| 12:46.14 | Druken | Enth: look at my nick carefully |
| 12:46.20 | Enth | like it's pointless running asterisk i fu have broadband/NAT and want to call someone externally who is on NAT |
| 12:46.46 | Druken | Enth: it can be done... i just don't like servers behind nat :) |
| 12:46.50 | Druken | just my preference |
| 12:46.57 | Enth | I agree. |
| 12:47.08 | vgster | i port forward |
| 12:47.20 | Druken | that works too |
| 12:47.26 | Enth | yup |
| 12:47.36 | Druken | personally, i'd just use the asterisk machine as a nat/router |
| 12:47.46 | Enth | but having a dynamic ip doesnt help |
| 12:47.47 | Enth | :) |
| 12:48.02 | Druken | nope.. it doesn't |
| 12:48.10 | Enth | exactly. |
| 12:48.34 | Enth | perhaps they should put a disclaimer - "sorry, not recommended if you have dynamic IP" |
| 12:48.36 | Druken | only way that works is if you both have dyn dns's and ya use those for lookups |
| 12:48.42 | Enth | yeah |
| 12:48.42 | f7950qs0 | how's astbill? |
| 12:48.49 | vgster | my ip at home hasnt changed for agessss |
| 12:48.51 | f7950qs0 | easy to configure? |
| 12:49.08 | Druken | Enth: why not use an outside source? toss it threw a server outside somewhere? |
| 12:49.17 | vgster | never used astbill sorry |
| 12:49.21 | Enth | Drunken: what do you mean? |
| 12:49.21 | *** join/#asterisk jijgeh (n=luken@static-66-182-95-76.bbsc.net) |
| 12:49.52 | Enth | you mean have asterisk running outside the nat on a static ip? |
| 12:49.55 | *** join/#asterisk Eitch (i=[U2FsdGV@unaffiliated/eitch) |
| 12:50.00 | vgster | phone your isp and beg them to static your internet ip |
| 12:52.04 | RoyK | ~seen coppice |
| 12:52.11 | jbot | coppice <n=chatzill@206.157.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 1d 23h 39m 53s ago, saying: 'depends on the phone, i think. I believe that is what theo was originally doing'. |
| 12:52.20 | Enth | Drunken ? |
| 12:52.36 | Druken | Enth: pm me |
| 12:53.01 | vgster | ~seen mywife |
| 12:53.03 | jbot | vgster: i haven't seen 'mywife' |
| 12:53.10 | vgster | crap hope she isnt out spending again |
| 12:53.22 | Druken | probably |
| 12:53.32 | gwynm | Hrm. Could someone please paste the output of 'show codecs' somewhere? I have this feeling that I'm missing some.. |
| 12:53.32 | vgster | its lunchtime too |
| 12:53.34 | Enth | most of them do. |
| 12:54.07 | f7950qs0 | is there any room for astbill? |
| 12:54.40 | vgster | http://pastebin.ca/44766 |
| 12:55.10 | Enth | gwynm: http://pastebin.ca/44767 |
| 12:55.22 | vgster | ooo beat you |
| 12:55.41 | Enth | damn you. |
| 12:55.44 | Enth | :) |
| 12:55.56 | vgster | why does pastebin always crash my IE |
| 12:56.04 | I-MOD | haha |
| 12:56.07 | I-MOD | good one vgster |
| 12:56.11 | Enth | that's IE for you. |
| 12:56.18 | Druken | which pastebin? |
| 12:56.19 | Enth | try using ff or opera |
| 12:56.21 | Druken | .ca or .com ? |
| 12:56.26 | tronix | gwynm: you're probably missing at least one: g729 ;) |
| 12:56.26 | vgster | .ca |
| 12:56.46 | Enth | Drunken: I pm'd you/ |
| 12:56.52 | Enth | -/ +. |
| 12:56.55 | Druken | shouldn't... it's all basic html, i don't think stephen has released the new version.... |
| 12:56.57 | vgster | ive got firefox ad its fine |
| 12:57.06 | vgster | but where do i tell mirc to use ff not ie |
| 12:57.14 | Druken | Enth: i seen that and i've typed shit in since, but i guess your not getting it? |
| 12:57.19 | Enth | damn, you use mirc? |
| 12:57.24 | Druken | yes |
| 12:57.26 | tronix | in windows somewhere, there's a setting for preferred browser, I think? vague recollection |
| 12:57.29 | gwynm | Thanks. Not missing any.. so that doesn't explain not being able to play sounds :/ |
| 12:57.32 | vgster | thats set to ff |
| 12:57.36 | tronix | hmm. |
| 12:57.48 | Enth | hrmmm |
| 12:57.56 | Enth | nope didnt get any pm |
| 12:58.02 | Enth | odd |
| 12:58.24 | Druken | stupid thing |
| 12:58.28 | Enth | well |
| 12:58.36 | Enth | I'm on multiple networks using irssi |
| 12:58.58 | Enth | works fine normally |
| 12:59.06 | Druken | :) |
| 12:59.09 | Enth | was chatting to a few others in pm. |
| 12:59.09 | Enth | hrmmm |
| 12:59.16 | Enth | blast. |
| 12:59.20 | Druken | just doesn't like me i guess |
| 12:59.24 | Enth | hah |
| 12:59.33 | Enth | got msn? |
| 12:59.45 | Druken | course |
| 13:00.08 | Enth | there. |
| 13:00.41 | f7950qs0 | what if I dont know any dial plan and I want all the numbers go through |
| 13:00.54 | f7950qs0 | is it o.k. if I just do XXX. |
| 13:01.59 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 13:02.21 | f7950qs0 | ??? |
| 13:03.44 | Enth | fn~f7950qs0: why dont u want a dial plan? |
| 13:04.07 | f7950qs0 | cause I dont know how to configure it |
| 13:04.38 | Enth | you have to have a dialplan |
| 13:04.52 | *** join/#asterisk _MartinCabrera_ (n=_MartinC@litigaractivos1.att.net.co) |
| 13:04.54 | Druken | use X. |
| 13:05.14 | f7950qs0 | in my asterisk I configured everything, got my ATA to register on an extension |
| 13:05.19 | f7950qs0 | and couldn't dial out because of a dial plan |
| 13:07.36 | f7950qs0 | the room's quiet all of a sudden |
| 13:07.51 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 13:08.36 | *** join/#asterisk but3k4 (n=but3k4@unaffiliated/but3k4) |
| 13:09.04 | gwynm | Hmm... what does register_verify: No registration for peer mean? |
| 13:11.51 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 13:11.54 | f7950qs0 | nobody's talkin or everybody's going to work |
| 13:12.27 | *** join/#asterisk guyee (n=izomtrik@nextra.nudli.equitas.hu) |
| 13:13.42 | vgster | im here |
| 13:13.57 | vgster | but im busy |
| 13:14.43 | guyee | Hi, I want to use SIP to connect to my provider, but when the provider answers with 484 Incomplete, * thinks that "Everyone is busy/congested at this time" |
| 13:15.02 | guyee | is it possible to get early dial work through asterisk? |
| 13:15.18 | RoyK | anyone here using spandsp with 1.2.5? |
| 13:15.55 | f7950qs0 | can I configure astbill from the gui? |
| 13:16.06 | f7950qs0 | I dont know how to configure dial plans and many other things |
| 13:16.12 | f7950qs0 | I just learned what a trunk is !! |
| 13:16.52 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.adnc.com) |
| 13:17.01 | FlyboySR22 | Good Morning Everyone |
| 13:18.17 | vgster | RoyK - i tried it but got problems |
| 13:18.28 | vgster | ive never use astbill |
| 13:18.28 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:18.38 | RoyK | i get symbols missing |
| 13:19.08 | vgster | is this the asterisk app bit? |
| 13:19.29 | RoyK | app_rxfax |
| 13:19.32 | RoyK | doesn't load |
| 13:19.42 | vgster | odd you should say that cos i think i had that too |
| 13:19.56 | vgster | let me play for a sec on my test box |
| 13:20.57 | x86 | can someone please try accessing my SIP URI: sip:bryce@shellshark.net |
| 13:21.00 | vgster | hmm makefile dont patch against 1.2.5 |
| 13:21.16 | Mike9 | anybody tried the new cisco 7940G/7960G SIP load 8.2? |
| 13:21.20 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
| 13:23.41 | PoWeRKiLL | which one to I have to sign to submit a patch http://www.digium.com/disclaimer.txt or http://www.digium.com/disclaim.changes ? |
| 13:24.40 | tronix | Mike9: nope.. interesting. 7.5 was latest on cisco's site when i looked 1-2 days ago. is 8.2 up there now? |
| 13:25.15 | *** join/#asterisk occam23 (n=seb@extgw.carmunity.de) |
| 13:25.25 | Mike9 | looks like they posted it last night.... strange thing tho, no 8.0 or 8.1... and it just says 7960G & 7940G |
| 13:25.53 | Mike9 | but there are release notes for 8.1 |
| 13:25.55 | vgster | RoyK i get no error using 1.2.4 on my box |
| 13:26.33 | RoyK | vgster: what versions of app_rxfax? |
| 13:26.41 | occam23 | whats the recommendation for a diva pri rev2, using the kernel driver or the eicon? for production |
| 13:27.08 | *** join/#asterisk jeffik (n=Jeff@CPE0050babf4cd6-CM014350000760.cpe.net.cable.rogers.com) |
| 13:27.23 | vgster | pre-25 |
| 13:27.47 | vgster | runs ok with 1.2.4 do you want me to try it with 1.2.5? actually i could do with getting it working myself |
| 13:28.50 | RoyK | strange |
| 13:29.04 | RoyK | spandsp 0.0.2? |
| 13:29.16 | vgster | yes, whatever is latest pre-25 |
| 13:29.25 | vgster | but i remember getting an error at some point |
| 13:29.29 | vgster | ive tried it before |
| 13:29.57 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
| 13:30.54 | PoWeRKiLL | http://bugs.digium.com/view.php?id=6668 |
| 13:33.13 | *** join/#asterisk lo2 (n=lo2@ti112210a080-10701.bb.online.no) |
| 13:33.13 | *** part/#asterisk fjean (n=fjean@201009186209.user.veloxzone.com.br) |
| 13:33.18 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 13:33.25 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 13:33.57 | *** join/#asterisk pengyong (n=lala@218.93.153.67) |
| 13:34.14 | guyee | NE1 knows how to get overlap dialing (from SIP to SIP) work via Asterisk? It handles 484 as "Invalid number format" instead of "Number incomplete" |
| 13:36.23 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 13:36.52 | vgster | RoyK - compiles fine with 1.2.5 |
| 13:37.47 | Nugget | 0.0.2 sure is a scary version number. |
| 13:37.58 | vgster | pre-25 |
| 13:38.28 | *** join/#asterisk guilherme-jorge (n=guilherm@200.155.185.1) |
| 13:38.40 | vgster | its a long version number thats for sure |
| 13:39.37 | guilherme-jorge | Hello all, I would like to know if there are way to discover the codec used in a call, through AGI script... Any idea? |
| 13:40.08 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 13:41.24 | [TK]D-Fender | guilherme-jorge : Did you try what I suggested yesterday? |
| 13:46.33 | guilherme-jorge | yes, I tried, but I didn't found any variable or command to get the codec. I got just the channel name. Help me, pls!! :) |
| 13:46.53 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 13:47.26 | guilherme-jorge | <[TK]D-Fender>: I got the channel name through get_variable function |
| 13:47.26 | [TK]D-Fender | guilherme-jorge : You haven't been paying attention. Once you have the channel name do a "sip show channels" and it will TELL you the codec for it! |
| 13:47.44 | [TK]D-Fender | Do it in CLI right now! |
| 13:48.11 | *** join/#asterisk fugitivo (n=ajf@201.255.179.22) |
| 13:48.33 | guilherme-jorge | But I would like to discover channel name through AGI script, because I need discover the codec before terminate the call... |
| 13:48.37 | guilherme-jorge | understand? |
| 13:51.26 | *** join/#asterisk SimplTon (n=bushah@user-108756k.cable.mindspring.com) |
| 13:52.09 | x86 | w00t! |
| 13:52.13 | x86 | DNS SRV++ |
| 13:52.27 | [TK]D-Fender | guilherme-jorge : I JUST TOLD YOU WHAT TO DO IN YOUR SCRIPT! JUST DO IT!. |
| 13:53.05 | x86 | hahaha |
| 13:53.07 | iDunno | don't shout! |
| 13:54.25 | x86 | [TK]D-Fender: it's weird that your CID info came across as just your name... no callback URI or anything... I normally use SIP to do outbound and inbound PSTN calling, so not really that familiar with SIP to SIP calling... Is that normal behaviour for CID to come across like that? |
| 13:55.10 | guilherme-jorge | I know the way to do this, but I don't know how to execute "sip show channel" (for example) in a AGI Script |
| 13:56.29 | *** join/#asterisk puzzled (n=yeahrigh@62.45.11.228) |
| 13:56.47 | guilherme-jorge | I tried to use the execute function, but it didn't recognized the command |
| 13:57.48 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:57.52 | *** join/#asterisk jsharp (n=jsharp@65.88.255.245) |
| 14:02.34 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 14:04.23 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 14:04.24 | guilherme-jorge | Hello all, Which function do I use to execute a "sip show channel" command (for example) through AGI script?? |
| 14:06.14 | *** join/#asterisk e3eli3h (n=not@82.102.94.82) |
| 14:10.50 | zamba | what text-to-speech-thingy is asterisk using? |
| 14:10.56 | jsharp | festival |
| 14:13.11 | zamba | jsharp: cool, thanks |
| 14:14.16 | *** join/#asterisk kpettit (n=keith@69.15.174.113) |
| 14:14.23 | *** join/#asterisk KentMentolado (n=KentMent@213.60.220.36) |
| 14:14.28 | KentMentolado | hello all |
| 14:14.48 | KentMentolado | I have strange problems using RealTime with IAX2 (and PostgreSQL). As application, I set 'Dial', and as appdata, 'IAX2/username'. Everything works with this configuration |
| 14:15.02 | KentMentolado | But if I set appdata to 'IAX2/username,timeout', I get the error 'unknow host username,timeout' |
| 14:15.31 | KentMentolado | what am I doing wrong? I read some documentation on www.voip-info.org, and I think my setup is correct. |
| 14:15.43 | russellb | replace the comma with a '|' |
| 14:15.49 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:16.44 | KentMentolado | thans russellb, it works :) |
| 14:16.52 | russellb | you're welcome |
| 14:19.14 | *** join/#asterisk miztic (n=gerard@rarcoa.com) |
| 14:20.19 | trelane` | is the asterisk ftp server down or just overloaded? |
| 14:20.22 | trelane` | I'm getting resets |
| 14:21.49 | trelane` | odd I got in on bsd ftp neither moz nor ie would grab it |
| 14:23.33 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 14:23.36 | [TK]D-Fender | blargh |
| 14:23.44 | *** join/#asterisk guilherme-jorge (n=guilherm@200.155.185.1) |
| 14:23.47 | Hmmhesays | never fear i have arrived |
| 14:24.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:24.30 | guilherme-jorge | Hello all, which function can be used to execute a "sip show channels" command (for example) through Agi script? Any idea? |
| 14:25.02 | Hmmhesays | ack |
| 14:25.03 | [TK]D-Fender | guilherme-jorge : use an * manager API call from within your script to do it. |
| 14:25.14 | Hmmhesays | i just bought these |
| 14:25.34 | [TK]D-Fender | guilherme-jorge : I do that on a script called by my Polycom phones to show live queue stats. |
| 14:25.37 | *** join/#asterisk aah_user (n=octothor@198.60.73.230) |
| 14:26.28 | viperdude | [TK]D-Fender: yeah I do that too... open up a telnet session to the manager with PHP |
| 14:26.48 | russellb | or you could just execute asterisk -rx "blah" ... |
| 14:26.53 | dgorski | [TK]D-Fender: any good links on crontrolling the polycom's display? |
| 14:27.00 | dgorski | I would be interested in that |
| 14:27.23 | viperdude | anybody using cisco 7960's with chan_sccp.so? |
| 14:27.59 | [TK]D-Fender | dgorski : there are just 2 settings for that in the provisioning. one for idle (plus the update interval), the other is through the "services" button on-demand |
| 14:28.03 | Enth | :( |
| 14:28.08 | [TK]D-Fender | Enth : Been done already |
| 14:28.13 | Enth | where? |
| 14:28.18 | Enth | dont tell me voip-info |
| 14:28.27 | [TK]D-Fender | dgorski : I do live queue stats on the |
| 14:28.40 | Enth | :) |
| 14:28.50 | [TK]D-Fender | "idle" browser page, and mass-presence, directories, etc on the "services one" |
| 14:29.06 | [TK]D-Fender | Enth : Actually, yes there, as well as a page file made. |
| 14:29.26 | Enth | [TK]D-Fender: where is it? |
| 14:29.40 | dgorski | [TK]D-Fender: I'm a bot new to the polycoms, where to start? I've got 301 and 501 to work with, I assume you are talking 501 at least... |
| 14:29.42 | Enth | voip-info doesnt have any info on dynamic ip+asterisk |
| 14:29.46 | dgorski | bot => bit |
| 14:30.14 | [TK]D-Fender | dgorski : only the 60x series has the MicroBrowser, so you're out of luck. |
| 14:30.40 | Druken | a browser on a phone.. hehehe |
| 14:30.45 | Druken | does it do WAP? :) |
| 14:30.47 | dgorski | [TK]D-Fender: thanks! I'll haev to get my hands on some 6xx ones then. |
| 14:31.49 | Enth | soooo.... |
| 14:32.00 | *** part/#asterisk aah_user (n=octothor@198.60.73.230) |
| 14:32.01 | Enth | where's this documentation then [TK]D-Fender |
| 14:32.51 | *** join/#asterisk svenna_ (n=svenna@p548D1ED4.dip0.t-ipconnect.de) |
| 14:33.16 | Enth | hrmmm |
| 14:34.33 | [TK]D-Fender | Enth : There are spots in the WIKI, some poorly linked. Externhost & externrefresh is it. |
| 14:34.53 | [TK]D-Fender | Enth : Its just not as "in your face" as some of the other info. |
| 14:35.22 | Enth | :) |
| 14:35.49 | Druken | [TK]D-Fender: generally we like things to be "in our face" :) |
| 14:36.11 | vgster | i agree |
| 14:36.14 | Enth | lol |
| 14:38.37 | Enth | what's the best way to check if your router has port forwarded port xxxx to whatrver machine? telnet to that ip/host: port? |
| 14:38.58 | dgorski | dynamic is no big deal, just make sure you register=> with something that is not dynamic |
| 14:39.01 | *** join/#asterisk jero (n=jero@savoirfairelinux.net) |
| 14:39.21 | Enth | fn~dgorski: define "something" |
| 14:39.44 | dgorski | something: peer |
| 14:39.55 | Druken | that's sip |
| 14:40.07 | dgorski | what's SIP? |
| 14:40.11 | Enth | heh |
| 14:40.17 | Enth | yeah |
| 14:40.22 | Druken | wrong damn window |
| 14:40.30 | *** join/#asterisk Tili (n=Tili@82-217-236-131.cable.quicknet.nl) |
| 14:40.30 | dgorski | ;) |
| 14:40.54 | dgorski | something: anything that is expected to find your extensions |
| 14:41.25 | [TK]D-Fender | dgorski : He mean where the SERVER is dynamic. |
| 14:41.38 | *** join/#asterisk gambolputty (n=root@64.74.225.131) |
| 14:41.40 | dgorski | why would you EVER do that? |
| 14:43.12 | gambolputty | Is it possible to automatically make outbound only calls from * that playback a recording? |
| 14:44.08 | dgorski | http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out |
| 14:45.14 | docelm0 | hay BSD users.. Whats a good book for reference? |
| 14:45.14 | gambolputty | I've been there, but * always wants to connect the call to a phone. |
| 14:45.23 | dgorski | read more |
| 14:45.34 | dgorski | "call files" |
| 14:45.49 | dgorski | "if the call answers, connect it here" |
| 14:46.34 | dgorski | Application: Playback |
| 14:46.38 | dgorski | Data: /a/recording |
| 14:46.47 | dgorski | or use your dialplan |
| 14:46.54 | gambolputty | I don't want to connect anywhere, except something like /dev/null |
| 14:47.11 | dgorski | so you want to call someone and have them hear the output of /dev/null? |
| 14:47.16 | dgorski | no, that's not what you want. |
| 14:47.19 | [TK]D-Fender | gambolputty : upon connect you send it to a macro or something |
| 14:48.03 | gambolputty | I want them to hear a recording |
| 14:49.23 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 14:49.50 | dgorski | Channel: Zap/1/some_poor_sucker_getting_spammed |
| 14:49.50 | dgorski | Callerid: Not Telemarketer <555-1212> |
| 14:49.51 | dgorski | MaxRetries: 100 |
| 14:49.51 | dgorski | Application: Playback |
| 14:49.51 | dgorski | Data: /some/spam/message/to/play/to/some_poor_sucker |
| 14:49.56 | cthompson | Johann Gambolputty de von Ausfern- schplenden- schlitter- crasscrenbon- fried- digger- dingle- dangle- dongle- dungle- burstein- von- knacker- thrasher- apple- banger- horowitz- ticolensic- grander- knotty- spelltinkle- grandlich- grumblemeyer- spelterwasser- kurstlich- himbleeisen- bahnwagen- gutenabend- bitte- ein- nürnburger- bratwustle- gerspurten- mitz- weimache- luber- hundsfut- gumberaber- shönedanker- kalbsfleisch- mittler- aucher von Hautkopft of Ul |
| 14:49.58 | *** join/#asterisk fjean (n=fjean@201009186209.user.veloxzone.com.br) |
| 14:50.13 | fjean | hi all ! I got the answer for the g729 license.. |
| 14:51.18 | *** join/#asterisk octothorpe (n=octothor@198.60.73.230) |
| 14:51.25 | fjean | it's not per channel/call its only really used when transcoding... |
| 14:51.39 | docelm0 | fjean, where have you been? |
| 14:51.46 | docelm0 | fjean, its always been like that |
| 14:51.46 | jsharp | Its per call that needs transcoding. |
| 14:51.56 | fjean | jsharp, right |
| 14:52.05 | jsharp | Its always been like that. |
| 14:52.11 | fjean | if its 729 right trough then you are not using a license... |
| 14:52.24 | jsharp | Right. |
| 14:52.34 | fjean | cool |
| 14:52.40 | docelm0 | fjean, if its pass thru or doesnt bridge in the server you dont need a license.. |
| 14:54.33 | Katty | yawn. |
| 14:54.51 | docelm0 | sigh.. MEW! |
| 14:54.58 | Katty | mew. |
| 14:55.20 | [TK]D-Fender | Katty: mew. |
| 14:55.21 | Hmmhesays | lovely i get to go do retard tech support now |
| 14:55.24 | docelm0 | hay Katty nice pic of you in your office loungeing in your chair.. |
| 14:55.27 | guyee | NE1 knows how to get overlap dialing (from SIP to SIP) work via Asterisk? It handles 484 as "Invalid number format" instead of "Number incomplete" |
| 14:55.46 | docelm0 | HAHAHA |
| 14:55.52 | *** join/#asterisk coppice (n=chatzill@210.22.134.146) |
| 14:56.27 | *** part/#asterisk fjean (n=fjean@201009186209.user.veloxzone.com.br) |
| 14:58.05 | Katty | docelm0: thanks (= |
| 14:58.17 | Katty | docelm0: co-worker caught me off guard. wanted to test his new camera. |
| 14:58.36 | Katty | docelm0: and that's not my office ;) |
| 14:58.41 | Hmmhesays | ahh another day fighting with chan_sip |
| 14:58.45 | Katty | hey iDunno (= |
| 15:01.21 | Hmmhesays | if anyone wants to help me out feel free |
| 15:01.22 | Hmmhesays | lol |
| 15:01.37 | *** join/#asterisk Nivex (i=kjotte@user-0ce2nsu.cable.mindspring.com) |
| 15:01.57 | *** join/#asterisk UlbabraB (n=salama@host-84-222-45-94.cust-adsl.tiscali.it) |
| 15:01.58 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 15:02.26 | Katty | Hmmhesays: i don't think i'm able. |
| 15:04.31 | [TK]D-Fender | Katty: I could use all the hugs I can get right now. I wok up this morning on a couch at a friends place on my birthday with my ex-gf having a complete nervous breakdown and in crisis since yesterday. |
| 15:04.58 | Katty | [TK]D-Fender: :<< |
| 15:05.06 | Katty | Skumling: that's because i didn't hand one out. |
| 15:05.18 | Skumling | Katty: heh ;) |
| 15:05.59 | [TK]D-Fender | Hmmhesays : Whats the problem> |
| 15:07.35 | *** join/#asterisk xtrvd (n=j@d209-121-36-44.bchsia.telus.net) |
| 15:09.11 | Druken | katty is a lil cutie |
| 15:09.56 | Skumling | Druken: I want to see the pic too... *whine whine* |
| 15:10.20 | Druken | i dun have it, nor would i give it out.... |
| 15:10.21 | iDunno | Katty is very cute :) |
| 15:10.28 | Druken | if she wanted you to see, she'd let you |
| 15:10.38 | Skumling | Druken: yeah yeah... I get it... |
| 15:10.44 | Druken | :) |
| 15:11.39 | Katty | you've all insaned. |
| 15:11.50 | Druken | course... i think it's coming on time for a new pic.. it's been a while... |
| 15:12.10 | iDunno | Katty: hmm - in my case, that's impossible, I was already insane :) |
| 15:12.23 | Katty | hee! ^_^ |
| 15:12.44 | Druken | i'd be insulted if you called me normal.... |
| 15:13.07 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 15:13.16 | jaiger | Skumling, http://www.puntarenas.com/carnavales/katty.html |
| 15:13.33 | jaiger | Skumling, her pic is easily found via google images |
| 15:13.43 | Katty | haha |
| 15:13.53 | Skumling | jaiger: damn, she needs to losse som weight |
| 15:13.55 | Skumling | loose |
| 15:13.57 | Skumling | ;-)))) |
| 15:13.57 | Katty | that'll be the day |
| 15:14.42 | jaiger | Skumling, never say that to a woman |
| 15:15.14 | *** part/#asterisk _deg_ (n=deg@200.250.222.8) |
| 15:15.37 | Druken | look at this ugly mofo http://www.efnetchatzone.net/Druken.html |
| 15:15.52 | `Sauron | grar |
| 15:16.05 | `Sauron | apparently everybody's all over the new lexar CF cards |
| 15:16.08 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 15:16.14 | Skumling | jaiger: heh a clothes salesman actually said that to one of my lady friends yesterday... or something like that... "oooh no, that jacket is just *way* too small for you"... damn she turned angry ;-) |
| 15:16.17 | Katty | cute kid. |
| 15:16.18 | `Sauron | cuz they have a manuf. rebate of $100 |
| 15:16.32 | jaiger | Druken, I dig the star graphic |
| 15:16.33 | tamp4x | anyone here use b2bua |
| 15:16.42 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
| 15:16.45 | Druken | Katty: old pic... he's 4 now... |
| 15:16.50 | Druken | not so cute anymore :) |
| 15:16.55 | Katty | Druken: probably still cute. |
| 15:17.10 | Druken | well, lemme see if i can find a recent pic... |
| 15:17.13 | iDunno | http://www.sommitrealweird.co.uk/photos/20060121-MeBeforeDuringAndAfter/imgp1270.jpg <-- something like what I look like currently :) |
| 15:17.37 | *** join/#asterisk kpettit (n=keith@69.15.174.113) |
| 15:17.42 | jaiger | Druken, he's a terror at 4 I'm sure |
| 15:18.00 | Katty | Druken: oooh, efnet. |
| 15:18.05 | Katty | Druken: should i go visit? |
| 15:18.52 | Druken | hmm... all my pics have disappeared |
| 15:19.16 | `Sauron | weird |
| 15:19.34 | Druken | me digital is mia too.... |
| 15:19.37 | Druken | god i hate moving |
| 15:20.00 | jaiger | Druken, you need to keep that pic up to date |
| 15:20.03 | `Sauron | http://www.adorama.com/ICADRXTB.html |
| 15:20.38 | Druken | why? |
| 15:20.53 | jaiger | Druken, joking. |
| 15:21.27 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfl2r.dialup.mindspring.com) |
| 15:22.04 | jaiger | don't think I've taken pics out of my digital in months |
| 15:22.11 | Druken | well, found the digital.... but good luck finding the usb cable for it :) |
| 15:22.43 | jaiger | `Sauron, that's what I use. I lost the usb cable years ago |
| 15:22.51 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:23.04 | `Sauron | last I bought pcmcia cards |
| 15:23.20 | `Sauron | I bought a stack of 'em, a 4-in-1 for nono-CF cards, and a CF card reader |
| 15:24.19 | `Sauron | s/nono/non |
| 15:24.26 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 15:24.33 | `Sauron | hum |
| 15:24.37 | h3x | linear flash is silly |
| 15:24.39 | `Sauron | should I do 3-5 business days |
| 15:24.44 | `Sauron | or 2nd day air |
| 15:25.32 | stoffell | anyone know why xten eyebeam video only works for the caller ? (not for called person) |
| 15:26.36 | [TK]D-Fender | stoffell : It works for both ends. |
| 15:26.54 | stoffell | [TK]D-Fender, i got the distinctive ringtone working for the polycom501 ;) |
| 15:27.07 | [TK]D-Fender | stoffell : you have to click on the "start video" button, AND have the proper codecs enabled in sip.conf |
| 15:27.10 | h3x | he means his local camera echo |
| 15:27.16 | zamba | how can i define what port range asterisk should use for udp? |
| 15:27.19 | [TK]D-Fender | stoffell : What way do you do it on yours? |
| 15:27.22 | zamba | source port |
| 15:27.37 | stoffell | [TK]D-Fender, it works, but only the persons that 'calls' can see both video sources |
| 15:28.08 | [TK]D-Fender | stoffell : Not in my experience. Each side has to start video and should work just fine. |
| 15:28.25 | stoffell | [TK]D-Fender, okay, will try other pc to make sure.. |
| 15:29.10 | octothorpe | [TK]D-Fender: What codecs should be enabled for video? |
| 15:29.22 | stoffell | [TK]D-Fender, polycom, it works by using this howto: http://www.voip-info.org/wiki/view/Polycom+auto-answer+config |
| 15:29.54 | stoffell | [TK]D-Fender, but for * 1.2 you must use : SIPAddHeader(Alert-Info: Ring Answer) |
| 15:32.19 | [TK]D-Fender | stoffell : So are you using that just for a distinctive ring, or for the auto-answer portion as well? |
| 15:32.50 | *** join/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 15:32.55 | stoffell | [TK]D-Fender, i only use it for distinctive ring, haven't tested auto answer (but that should also work I think) |
| 15:33.34 | [TK]D-Fender | stoffell : Yeah, I stalled at the change in how you add the header I think... I'll give it a try some day soon. |
| 15:34.52 | mroth_imm | anyone running large numbers of dynamic agents in a single queue (120+) and receiving these messages "Could not create persistent member string, out of space" |
| 15:36.01 | mroth_imm | it's from app_queue, and i see it's related to a constant, but we're using ABE, so all i can do is suggest things to Digium support...looking for input from someone who may have experienced the same problem |
| 15:36.13 | mroth_imm | (and possibly patched/solved it) |
| 15:36.48 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 15:36.58 | shmaltz | how do I fore a reload of resolv.conf? |
| 15:37.35 | mroth_imm | shmaltz: you can reload individual shared objects from the CLI |
| 15:37.45 | mroth_imm | type "reload <TAB>" to see what i mean |
| 15:37.48 | [TK]D-Fender | octothorpe : h261, h263, h263p |
| 15:37.53 | h3x | resolv.conf is the dns resolver |
| 15:38.00 | shmaltz | mroth_imm, not in Asteriks, in lynix |
| 15:38.07 | h3x | theres nothing to reload |
| 15:38.20 | octothorpe | thanks fender |
| 15:38.27 | h3x | your problem is probably the nsswitch cache |
| 15:39.14 | h3x | whatever its called in your distro |
| 15:39.14 | *** join/#asterisk _MartinCabrera_ (n=_MartinC@litigaractivos1.att.net.co) |
| 15:39.15 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfl2r.dialup.mindspring.com) |
| 15:40.07 | Hmmhesays | hey baby whatcha doing this evening can ya meet me down at the railroad tracks, I got tom petty playing in my silverado and I, iced down a 6 pack |
| 15:40.41 | *** join/#asterisk Fedoracore6 (n=deddd@60.50.141.168) |
| 15:42.10 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:42.11 | *** join/#asterisk sevard (i=sev@24-179-181-160.dhcp.dlth.mn.charter.com) |
| 15:42.19 | *** join/#asterisk mutilator (n=animenod@65.111.222.120) |
| 15:42.48 | sevard | I don't suppose there's a way to "auto sense" NAT or not? If you have a roaming SIP client it'd suck to have to reconfigure him everywhere he went. |
| 15:43.18 | mutilator | just nat=yes no matter what |
| 15:43.38 | [TK]D-Fender | sevard : what mutilator said.... |
| 15:44.13 | sevard | I thought that wouldn't work if the client wasn't NATd |
| 15:44.48 | asteriskmonkey | hey any any of the digium guys on? |
| 15:44.54 | *** join/#asterisk DrukenHME (n=druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 15:45.21 | mutilator | works fine |
| 15:45.22 | brookshire | what do you need? |
| 15:46.02 | asteriskmonkey | two questions 1)when i tune my pri using the test number and ztmonitor to get optimal of 14500 in -vv mode why does dtmf stop working and 2) why am i seeing spillover from rx=>tx and vise versa when running ztmonitor on a channel |
| 15:46.03 | brookshire | asteriskmonkey: :D |
| 15:46.21 | brookshire | oh.. i have no idea..i just make websites :D |
| 15:46.24 | sevard | Alright, another question.. what about having allow and disallow both in asterisk realtime static configs does one override the other? |
| 15:46.42 | sevard | or can you specify order, like host.conf |
| 15:46.45 | *** join/#asterisk rene- (n=rene-@dsl-200-95-25-160.prod-infinitum.com.mx) |
| 15:47.01 | brookshire | but i'm sure you will get a faster response from email |
| 15:47.22 | asteriskmonkey | darn |
| 15:47.23 | asteriskmonkey | ok if no one can answers those i have to call em :( |
| 15:47.26 | asteriskmonkey | ok |
| 15:47.30 | asteriskmonkey | i will email them then |
| 15:47.31 | asteriskmonkey | thanks |
| 15:47.36 | shmaltz | h3x, how do I do that in slackware? reload the nsswitch? |
| 15:47.42 | rene- | quick app_directory question, what happens when two persons have the same last name? does the system allow the caller to choose whom they are trying to reach? |
| 15:47.57 | shmaltz | rene-, try it and tell us |
| 15:48.15 | sevard | Disallow all first works fine in sip.conf, but if i put a disallow=all line in the database, it blocks all codecs, nevermind the fact that i have allow=speex in there too allow=speex in there too |
| 15:48.28 | brookshire | asteriskmonkey: i bet if you call right now, you will get to a free person instantly :) |
| 15:49.01 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:49.01 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:49.13 | Katty | anthm: (= |
| 15:49.23 | anthm | hello |
| 15:49.28 | sevard | Hey Katty, thanks for the b-day advice btw |
| 15:49.35 | Katty | sevard: how'd it go? |
| 15:49.43 | sevard | Katty: bad ass |
| 15:49.54 | Katty | :>> |
| 15:50.14 | h3x | actually i think its called nsd or something, name service daemon ? |
| 15:50.24 | sevard | Katty: my brother in a gorilla suit disrupted her class 5 minutes before it ended, i guess the professor took it well. From there the gorilla gave her a card which lead her on a hunt all around the city |
| 15:50.45 | rene- | shmaltz: the wiki has all teh answers: "If more than one matching last name is found, it will allow the caller to cycle through all the matches found." |
| 15:50.45 | octothorpe | rene: yes, it gives the first, and an option to get the second if the first isn't who you were looking for |
| 15:51.06 | Katty | sevard: goodness. |
| 15:51.08 | sevard | Katty: her last clue after her 2 hour hunt was "what the crap are you doing running around town? go home!" when she went home her she found me in her room, I filled it with over 450 balloons, to the brim. |
| 15:51.35 | *** join/#asterisk kratzers (n=kratzers@martha.pa.net) |
| 15:51.38 | Katty | sevard: aww! |
| 15:51.43 | kratzers | howdy |
| 15:51.58 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 15:52.05 | sevard | Katty: from there we went to a very nice dinner, i had halibut, delicious. After that we went to a show the highschool was putting on, they won state for show choir, it was really really good., |
| 15:52.17 | FlatFoot | ok i'm stummped anyone got config examples ( that work ) for the hint type thing on the 190 and 360 ? |
| 15:52.24 | Katty | sevard: sounds dreamy. |
| 15:52.35 | mutilator | heh |
| 15:52.36 | kratzers | probably common question, say I have an agent number 100 who always uses extension 100, how can I allow them to easily login? |
| 15:52.37 | mutilator | man |
| 15:52.44 | mutilator | i can hardly afford flowers and a card |
| 15:52.45 | sevard | Katty: After that we went home and split a bottle of 2001 merlot and watched a movie. |
| 15:52.54 | sevard | Katty: it effin ruled. |
| 15:53.09 | kratzers | currently have -> AgentCallbackLogin(${CALLERIDNUM}|${CALLERIDNUM}@some_context) |
| 15:53.23 | kratzers | but I'd like it to use the current extension rather than asking for a new location |
| 15:53.29 | DrukenHME | sevard: then she "thanked" you in her special little way |
| 15:53.49 | kratzers | any ideas? |
| 15:54.04 | sevard | DrukenHME: it's been 2 1/2 weeks and she won't stop thanking me. |
| 15:54.19 | DrukenHME | i notice your not complaining :) |
| 15:54.24 | sevard | :) |
| 15:54.42 | *** join/#asterisk notOnyx (n=email@208.19.245.194) |
| 15:55.22 | Katty | sevard: i'm glad you did something wonderful for her :> |
| 15:55.23 | *** join/#asterisk rAndal- (n=andy@216.158.99.146) |
| 15:55.59 | DrukenHME | i was going to take my girlfriend out for dinner and a movie.. but she didn't come home.. |
| 15:56.07 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 15:56.38 | sevard | DrukenHME: ..What? |
| 15:56.52 | DrukenHME | hehe |
| 15:57.06 | Katty | dinner and a movie is boring, guys |
| 15:57.08 | Katty | spice it up a little |
| 15:57.16 | DrukenHME | friday, i was going to take her out to dinner, and a movie. but she didn't come home till the next night |
| 15:57.18 | Katty | we're getting sick of the same old thing |
| 15:57.24 | jontow | FlatFoot: i've got a couple for the SNOM 320s.. should be the same( .oO{ ?? }) |
| 15:57.28 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 15:57.32 | sevard | Yeah, Take a hint from sevard. |
| 15:57.38 | DrukenHME | Katty: when you live together, dinner and a movie is spiced :) |
| 15:57.44 | FlatFoot | jontow: does it work ? |
| 15:57.49 | jontow | seemingly ;) |
| 15:57.56 | jontow | i can show appearances for the lines, can't yet pick them up though |
| 15:57.56 | Katty | DrukenHME: then you clearly need to work on that living together bit. |
| 15:58.16 | sevard | DrukenHME: Instead of dinner and a movie, what about splunking and robbing. |
| 15:58.29 | FlatFoot | how did you get it to work %****$£$$$ things won't show activity on other phones for me |
| 15:58.41 | jontow | lemme login to the machine and retrieve :) |
| 15:58.41 | jaiger | Katty, do you have kids? I don't remember the last time I had dinner & a movie w/ my wife. how can that be boring? |
| 15:58.48 | iDunno | Katty: what do you suggest, then? (other than finding someone to take out first ;) |
| 15:58.49 | FlatFoot | jontow: ta |
| 15:58.50 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 15:59.04 | Katty | jaiger: me? kids? |
| 15:59.10 | Katty | jaiger: you clearly have no clue how old i am. |
| 15:59.10 | DrukenHME | Katty: what is wrong with taking her to her favourite restaurant, and spending some quality time watching a movie of her choice? |
| 15:59.17 | sevard | DrukenHME: Dinner and a threesome? |
| 15:59.22 | *** join/#asterisk RoyK (n=roy@a217-118-45-74.bluecom.no) |
| 15:59.22 | Katty | DrukenHME: nothing, until it gets boring. |
| 15:59.26 | *** join/#asterisk TheoC (n=theochao@68-191-219-240.dhcp.dntn.tx.charter.com) |
| 15:59.28 | Katty | DrukenHME: and then it looses its charm. |
| 15:59.33 | DrukenHME | sevard: sounds good, send your girl over :) |
| 15:59.40 | *** part/#asterisk TheoC (n=theochao@68-191-219-240.dhcp.dntn.tx.charter.com) |
| 15:59.45 | Katty | have dinner on a cruise ship |
| 15:59.50 | jontow | FlatFoot: exten => 101,hint,SIP/101 |
| 15:59.51 | Katty | or rent a pontoon |
| 15:59.52 | jaiger | Katty, you're right, I do not. I have 2 young kids and my wife & I never get out |
| 15:59.55 | sevard | DrukenHMEL: Any other girl this one isn't for sharesies |
| 16:00.00 | jontow | and function key 12 on phone 100 is this: |
| 16:00.01 | *** join/#asterisk TheoC (n=theochao@68-191-219-240.dhcp.dntn.tx.charter.com) |
| 16:00.12 | `Sauron | Katty: so how old are you? |
| 16:00.20 | jontow | destination / <sip:101@pbx.ip.add.ress;user=phone> |
| 16:00.31 | jontow | i believe i just typed in '101' and it auto-completed |
| 16:00.33 | jaiger | `Sauron, I've been told you shouldn't ask a woman that either |
| 16:00.36 | DrukenHME | jaiger: between the two of us, we have 4... |
| 16:00.41 | asteriskmonkey | brookshire: sent em email :) |
| 16:00.42 | FlatFoot | jontow: got that but the light it no light up |
| 16:00.47 | `Sauron | jaiger: I don't follow "the rules" |
| 16:00.50 | jontow | seems to work for me.. |
| 16:00.52 | `Sauron | I like to live dangerously |
| 16:01.05 | jontow | Asterisk CVS HEAD built by root@romeo on a i686 running Linux on 2005-10-14 21:51:11 UTC |
| 16:01.08 | jontow | and its an oldass version of HEAD |
| 16:01.14 | jaiger | `Sauron, I also like to live dangerously |
| 16:01.18 | Katty | `Sauron: old enough to know that you really don't need to know. |
| 16:01.21 | heison | mr$evone |
| 16:01.22 | RoyK | zoa: ping |
| 16:01.25 | `Sauron | jontow: I think mine's older.. Hehn. |
| 16:01.27 | jontow | :D |
| 16:01.37 | TheoC | I'm having some problem with our dialplan.agi - when I try to dial a new extension I added I get "Returned from dialparties with no extensions to call" - but the original exts I set up work fine. What could the problem be? |
| 16:01.45 | jontow | well, i upgraded specifically to that branch and made sure it was stable.. for the appearances |
| 16:01.46 | FlatFoot | jontow: have you got the snom on DESTINATION ? |
| 16:02.07 | jontow | yeah, function key 12 is on destination.. |
| 16:02.08 | `Sauron | KAtty: You're no fun. Hehn. |
| 16:02.12 | sevard | Anyone know a bit about realtime asterisk to answer my question about allow lines? :) |
| 16:02.16 | RoyK | ~seen zoa |
| 16:02.20 | jbot | zoa is currently on #asterisk (3m 30s), last said: '1000 interrupts per second are ok'. |
| 16:02.23 | Katty | `Sauron: nope. not unless you know me. |
| 16:02.35 | jontow | sevard: i know .. literally 'a bit' .. been playing with it the last few days |
| 16:02.41 | FlatFoot | jontow; what * version are you on ? |
| 16:02.51 | rene- | ~seen jerjer |
| 16:02.52 | jbot | jerjer <n=jj@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 11d 19h 3m 59s ago, saying: 'Kovecses: ok there isn't a debian package for this? thanks'. |
| 16:03.02 | sevard | jontow: do you have allow and disallow in your database? |
| 16:03.10 | sevard | static configs |
| 16:03.12 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 16:03.36 | _Sam-- | `Sauron: did you used to be on #Php on efnet? |
| 16:03.42 | jaiger | `Sauron, I'll admit her comment did prompt the question tho |
| 16:03.46 | _Sam-- | trying to figure out where i remember that nick |
| 16:07.37 | *** join/#asterisk salviadud (n=ralfalfa@dsl-201-133-198-176.prod-infinitum.com.mx) |
| 16:08.02 | `Sauron | Sam: I still am. |
| 16:08.13 | _Sam-- | nice, what about derick? |
| 16:08.13 | `Sauron | But yes I'm the same guy. |
| 16:08.21 | `Sauron | Dunno, think he's still around |
| 16:08.26 | _Sam-- | that guy is a php genius |
| 16:08.39 | _Sam-- | what are you up with asterisk? |
| 16:08.52 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 16:12.12 | *** join/#asterisk octothorpe (n=octothor@198.60.73.230) |
| 16:12.34 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 16:13.52 | guyee | NE1 knows how to get overlap dialing (from SIP to SIP) work via Asterisk? It handles 484 as "Invalid number format" instead of "Number incomplete" |
| 16:15.55 | *** join/#asterisk CoolAcid (n=jason@216.99.98.39) |
| 16:16.22 | *** join/#asterisk bkw_ (n=bkw_@ppp-70-128-118-15.dsl.tulsok.swbell.net) |
| 16:17.43 | jontow | sevard: "permit" and "deny", and "allow" and "disallow" |
| 16:17.50 | jontow | disallow = "all" |
| 16:17.55 | jontow | allow = "gsm;ulaw;ilbc" |
| 16:18.05 | jontow | separate 'em (any key that has multiple values) with a ; |
| 16:18.34 | jontow | (sorry for the delay; was helping one of the admins fix one of the widebank's management ports :)) |
| 16:20.01 | FlatFoot | jontow; what * version are you on ? |
| 16:20.20 | *** join/#asterisk MatsK (n=mk@141.221.181.62.in-addr.dgcsystems.net) |
| 16:20.53 | TheoC | I'm having some problem with our dialplan.agi - when I try to dial a new extension I've added I get "Returned from dialparties with no extensions to call" - but the original extensions I set up work fine. What could the problem be? |
| 16:20.55 | *** join/#asterisk Seyr (n=Seyr_@pf01.grantgeo.com) |
| 16:22.53 | sevard | jontow: When I put disallow all first and the allow codecs after it disallows all codecs and doesn't allow any codecs to work. |
| 16:23.47 | `Sauron | Sam: for fun, playin around at home |
| 16:23.56 | `Sauron | and we might have a machine at work doing conferencing |
| 16:25.42 | jontow | weird.. |
| 16:25.54 | FlatFoot | jontow: sorry to be a pain but , done all that the hint light does not light up and if i prees the function button i get Not Found 199 ( 199 being the hint exten ) |
| 16:26.02 | jontow | <excuse the paste..> : |
| 16:26.03 | jontow | | disallow | varchar(25) | | | all | | |
| 16:26.06 | jontow | | allow | varchar(80) | | | gsm;ulaw;alau | | |
| 16:26.12 | jontow | thats the relevant bit of my table structure |
| 16:26.16 | jontow | defaults on the right :) |
| 16:26.19 | Beirdo | ~pastebin |
| 16:26.21 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste |
| 16:26.28 | Beirdo | easier to read too :) |
| 16:26.41 | *** part/#asterisk vgster (n=vg@host217-45-221-53.in-addr.btopenworld.com) |
| 16:27.10 | jontow | i'm completely aware of pastebin.. but its a bit overkill for 2 lines |
| 16:27.37 | luke-jr_ | OMG |
| 16:27.45 | luke-jr_ | PAP2-NA suck |
| 16:27.49 | luke-jr_ | (%)*#)% |
| 16:28.10 | jontow | (btw, i know alaw is spelled wrong above..) |
| 16:30.47 | sevard | jontow: That's exactly what I have.. and it doesn't work. |
| 16:31.14 | jontow | weird, i haven't had any problems with it.. did catch a spelling error in the resource i was using to set it up though.. thats in my sip_peers table, too.. |
| 16:31.57 | *** join/#asterisk HamYaI (i=HamYai@125.24.5.172) |
| 16:36.16 | *** join/#asterisk pointer (i=pointer@aj.catt.com) |
| 16:37.32 | *** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv) |
| 16:37.39 | pointer | I just wanted to hop in here and go on "public" record as saying that after trying the digium TDM400, TDM2400 w/EC, the sangoma a200 with and without EC, and even the digium single port T1 w/o EC...that the only card that didn't have echo was the sangoma a200 w/EC and that their support people are ridiculously helpful |
| 16:38.11 | rajiv|work | how do you know if you need the EC card add on for the sangoma a200 ? |
| 16:38.26 | *** join/#asterisk razu_ (n=razu@ip228.cab74.mus.starman.ee) |
| 16:38.50 | pointer | the sangoma single port T1 doesn't have echo either....we have a couple of the sangoma quad T1 w/EC on order...I expect good things out of them as well |
| 16:39.03 | pointer | rajiv|work: I have a card with and without EC |
| 16:39.12 | pointer | rajiv|work: they're separate cards completely |
| 16:39.22 | pointer | rajiv|work: the A200s w and w/o EC that is |
| 16:39.37 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 16:40.09 | pointer | anyone here using audiocodes gear? |
| 16:40.37 | rajiv|work | i thought that the EC card was a daughter card fro the a200 ? http://www.sangoma.com/datasheets/p_a200-specs seems to imply that |
| 16:40.41 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 16:40.58 | pointer | the card without EC doesn't even have the socket |
| 16:41.03 | rajiv|work | ah |
| 16:41.33 | pointer | really wierd...I expected it to just be a module that you could add to |
| 16:41.41 | pointer | s/to$/to/ |
| 16:41.43 | pointer | heh, too |
| 16:42.09 | pointer | but now I have to RMA this one without it |
| 16:42.27 | mutilator | man |
| 16:42.33 | mutilator | why do digium cards suck so much |
| 16:42.36 | mutilator | ugh |
| 16:43.02 | *** join/#asterisk redondos (n=redondos@190.48.45.160) |
| 16:43.08 | pointer | mutilator: I've spent a ton of money on them and came to the same conclusion. save yourself some time and go sangoma |
| 16:43.41 | mutilator | well i went out and got a tellab echo can |
| 16:43.44 | mutilator | just to get rid of it |
| 16:43.50 | mutilator | and putting that in sorta solved the problem |
| 16:44.03 | mutilator | far end gets super choppy audio |
| 16:44.15 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 16:44.15 | mutilator | so any faxing and dialup people did doesn't work either |
| 16:44.26 | pointer | mutilator: we've tested almost every card digium sells, audiocodes, vegastream, sangoma, and openswitch6(sp?) and the only thing that worked was the sangoma w/EC |
| 16:44.26 | backblue | does anyone have append when one incoming call is from zap and dialed & awnsered from a sip user, its ok, but if it is awnsered from asterisk (eg: ivr's), it dont work? |
| 16:44.47 | mutilator | i'll have to put my damn cisco box back in, wasted money on 2 of them 405p's |
| 16:44.48 | *** join/#asterisk RoyKa (n=roy@a217-118-45-74.bluecom.no) |
| 16:44.55 | rajiv|work | pointer: ok. but any ideas how to choose between EC and not? |
| 16:44.55 | backblue | sangoma its the top! :D |
| 16:45.05 | pointer | mutilator: get a sangoma! |
| 16:45.17 | mutilator | if i can get it approved |
| 16:45.21 | pointer | rajiv|work: there is no choice, get ECs |
| 16:45.21 | mutilator | i doubt that |
| 16:45.39 | pointer | mutilator: RMA the 405ps |
| 16:45.43 | *** join/#asterisk Darwin35 (n=Darwin@sta-208-139-193-162.rockynet.com) |
| 16:45.47 | mog_work | hey now.... |
| 16:45.51 | mog_work | whats wrong with the 405s |
| 16:45.52 | mog_work | ^_^ |
| 16:45.54 | msw | mog! |
| 16:45.59 | backblue | what its EC's? |
| 16:46.02 | mutilator | plenty |
| 16:46.06 | mog_work | oucha |
| 16:46.23 | mog_work | care to eleborate |
| 16:46.41 | rajiv|work | pointer: $660 vs $360 at voipsupply.com. ouch |
| 16:47.01 | pointer | mog_work: after $5-10k in PSTN term gear for asterisk, I think I can safely say that we've seen a lack of consistent quality out of the digium stuff...the 3 sangoma cards we've tested were flawless |
| 16:47.23 | pointer | mog_work: actually, more than $10k |
| 16:47.50 | pointer | mog_work: and 9 months of off and on testing with vendors (including digium/sangoma/vegastream/etc) |
| 16:47.54 | _Sam-- | what type of equipment is 10k to get to PSTN? |
| 16:48.00 | _Sam-- | 10 quad t1 cards? |
| 16:48.01 | mog_work | well |
| 16:48.03 | mutilator | well i'de need to rma both these 405's to afford a 104D |
| 16:48.36 | *** join/#asterisk kratzers (n=kratzers@martha.pa.net) |
| 16:48.40 | mog_work | but i have worked with probably hundreds to thousands of teXXX cards |
| 16:49.06 | mutilator | i have a 110 card |
| 16:49.12 | mutilator | and it works great, no echo or anything |
| 16:49.21 | backblue | sangoma it's the best, better then digium offcourse. |
| 16:49.21 | kratzers | is it possible to have agents on DND not receive calls? |
| 16:49.31 | mutilator | so i figured the 405 would be the same |
| 16:49.33 | pointer | _Sam--: 4-5 tmd400ps, 2-3 digium T1, 2-3 digium quad span t1, 2 sangoma T1, 2 sangoma quad t1, 2 sangoma a200s (1 w/EC), 3 vegastream 50s, 1 audiocodes mp104, 1 openswitch6, and some other stuff I'm forgetting |
| 16:49.40 | mutilator | both cards, 2 different uses do the same thing |
| 16:49.57 | *** part/#asterisk UlbabraB (n=salama@host-84-222-45-94.cust-adsl.tiscali.it) |
| 16:50.21 | pointer | _Sam--: I know I'm forgetting something in that list |
| 16:50.47 | rajiv|work | ok, so why a sangoma a200 4 fxo vs wellbridge or dlink fxo gateway? |
| 16:50.50 | kratzers | anyone? |
| 16:50.57 | pointer | _Sam--: the 4 port digium cards seem to work ok in the one production box we have them in (only 2 active PRIs there) |
| 16:51.24 | pointer | _Sam--: we haven't rolled the other one into production yet, but we went ahead and ordered a sangoma quad T1 card to replace the digium one |
| 16:54.33 | *** part/#asterisk Seyr (n=Seyr_@pf01.grantgeo.com) |
| 16:54.34 | Fedoracore6 | hai all |
| 16:54.34 | pointer | rajiv|work: the price diff for EC is worth it on the sangoma. we tried ironing out the echo issues with digium and voipsupply and decided to try the sangoma after hearing good things about them. |
| 16:54.46 | *** join/#asterisk ixos (n=ixos@mail.kneedraggers.com) |
| 16:54.50 | _Sam-- | hah |
| 16:54.54 | _Sam-- | good luck ixos :) |
| 16:55.01 | ixos | no luck in finding an answer? |
| 16:55.06 | _Sam-- | i didnt check |
| 16:55.08 | Fedoracore6 | where i can find script for check password must same in databasess |
| 16:55.49 | jbalcomb | Whats the best CDR package for asterisk? I need to bill clients based on calls we take and make for them charged by total call duration. |
| 16:56.14 | tzafrir | Fedoracore6, what do you mean? which passwords? which database? which access do you have to it? |
| 16:56.18 | _Sam-- | ixos : sauron might be a good one to ask, if he's here |
| 16:56.24 | _Sam-- | `Sauron : you there? |
| 16:56.38 | backblue | hi, what its the default password of asterisk@home? anyone knows? |
| 16:56.44 | mog_work | asterisk? |
| 16:56.45 | Fedoracore6 | i try this code but didnt work |
| 16:56.46 | mog_work | password |
| 16:56.47 | Fedoracore6 | http://pastebin.com/589130 |
| 16:56.49 | mog_work | imanubb |
| 16:56.56 | jbalcomb | backblue www.google.com? |
| 16:57.02 | Fedoracore6 | tzafrir |
| 16:57.07 | FlatFoot | exit |
| 16:57.13 | FlatFoot | opp's wrong screen |
| 16:57.18 | Fedoracore6 | http://pastebin.com/589130 |
| 16:57.37 | backblue | jbalcomb: i'm reading asterisk at home handbook, i found it, i was not finding it! :P |
| 16:57.54 | _Sam-- | ixos : if it dials out using the asterlink connection that is fine. |
| 16:58.01 | ixos | can the Call Manager API originate action be used to force the outgoing call to use a different destination SIP address? The originate action doesn't accomodate settings things up like 'SIP/teliax/[outgoing number]' in the Exten: field, only [phone_number] |
| 16:58.04 | ixos | that's what it's doing now |
| 16:58.15 | Fedoracore6 | my database name asterisk |
| 16:58.31 | Fedoracore6 | and have 2 table in there name student and cdr |
| 16:59.07 | _Sam-- | ixos: it seems like the originate command just uses which ever extension matches the pattern in extensions.conf |
| 16:59.17 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 16:59.21 | _Sam-- | and the only pattern that matches the string you are sending is the asterlink one... |
| 16:59.26 | _Sam-- | but zoa would know too |
| 17:00.12 | _Sam-- | ixos: i have an idea that may work |
| 17:00.29 | _Sam-- | we would send an extra 1 in the beginning of the extension |
| 17:00.36 | _Sam-- | then in extensions.conf i would setup another extension that matches |
| 17:00.50 | ixos | give it a try |
| 17:01.00 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 17:01.03 | _Sam-- | or, try having your script send 9,1302XXXXXXXX |
| 17:01.12 | ixos | that would probably be better |
| 17:01.18 | _Sam-- | working on my side now |
| 17:01.40 | _Sam-- | for now i am going to hard code it for your cell phone number for testing |
| 17:01.47 | ixos | k |
| 17:02.31 | _Sam-- | try having your side send 9 1 302+your cell number |
| 17:03.11 | _Sam-- | you didnt send it as one string |
| 17:03.16 | _Sam-- | send 91302562xxxx |
| 17:03.26 | ixos | bingo |
| 17:03.48 | _Sam-- | aight i will dick around with the extensions.conf now |
| 17:05.19 | _MartinCabrera_ | running |
| 17:05.19 | _MartinCabrera_ | "patch -p0 < asterisk-1.2.5-patch" |
| 17:05.19 | _MartinCabrera_ | What i'm doing wrong? |
| 17:05.38 | hardwire | -p1? |
| 17:05.38 | Fedoracore6 | did some budy have sample codeing to check password mus same in databases |
| 17:05.42 | _Sam-- | ixos : can you send 1areacodenumber |
| 17:05.44 | tzafrir | Fedoracore6, so what exactly is your question? You can try the same query from a different mysql client (e.g: the command-line mysql) |
| 17:05.44 | _Sam-- | er |
| 17:05.47 | *** join/#asterisk maruz (n=maumar@adsl-123-3.38-151.net24.it) |
| 17:05.48 | _Sam-- | 91areacodenumber |
| 17:05.53 | _Sam-- | and i think we are all done |
| 17:05.55 | _MartinCabrera_ | (i "get can't find file to patch at input line 5") |
| 17:06.41 | Fedoracore6 | sorry if my unswer u all didnt understand :) |
| 17:07.19 | jbalcomb | Whats the best CDR package for asterisk? I need to bill clients based on calls we take and make for them charged by total call duration. |
| 17:09.01 | *** part/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 17:09.49 | *** join/#asterisk jmanq (n=jquintus@pool-151-203-200-91.bos.east.verizon.net) |
| 17:12.59 | kippi | http://www.voip-info.org/wiki-Asterisk down? |
| 17:13.14 | *** join/#asterisk diego_br (n=brazilei@200.208.241.178) |
| 17:14.14 | SplasPood | kippi: seems that way |
| 17:14.18 | wasim | thats a MUST read, wiki-Asterisk-down |
| 17:14.40 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:14.43 | SplasPood | jbalcomb: from what I've found the OSS CDR packages have been crap in one way or another |
| 17:14.50 | SplasPood | jbalcomb: been working on my own.. |
| 17:15.13 | SplasPood | jbalcomb: they either just plain suck, or they try to takeover your total asterisk config/mgmt, etc.. |
| 17:15.16 | *** part/#asterisk ixos (n=ixos@mail.kneedraggers.com) |
| 17:15.26 | SplasPood | but thats just my opinion |
| 17:15.52 | mphill | my system picks up and say "the person at extension... then doesn't say anything" any ideas? |
| 17:20.22 | jbalcomb | SplasPood We are using Asterisk CDR Analyizer right now. It's decent but we had to do some sort of patch/update/recompile on Asterisk to get it to track calls across a transfer. |
| 17:20.46 | bkw_ | wasabi |
| 17:20.49 | jbalcomb | SplasPood From what I'm hearing, Asterisk doesn't do a very good job of providing information for CDR billing |
| 17:20.52 | PoWeRKiLL | hi bkw_ |
| 17:20.57 | Katty | DrukenHME: are you on efnet? |
| 17:21.12 | mphill | what is CDR? |
| 17:21.29 | Qwell | ~cdr |
| 17:21.30 | jbot | methinks cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw |
| 17:22.59 | *** join/#asterisk fjean (n=fjean@201009186209.user.veloxzone.com.br) |
| 17:23.44 | fjean | hello again, anyone knows if we can adjust call volume (transfer and receive) using ztdummy ?? |
| 17:23.52 | *** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
| 17:25.29 | fjean | would it be usefull to set rxgain and txgain from zapata.conf ? |
| 17:26.49 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 17:27.07 | [TK]D-Fender | fjean : You're supposed to do it in zapata in your channel defs |
| 17:28.46 | *** join/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 17:28.56 | TheoC | Some of our extensions are failing (with a message "The person on extension x is busy"). I've traced it back that the ExtState coming back from the dialplan.agi is 4 (The phone is ringing) even though that's not the case. What could be the cause? |
| 17:29.19 | fjean | sfs-fender : thanks I ll try that |
| 17:32.29 | *** join/#asterisk gunsch (n=linux@pD954160C.dip0.t-ipconnect.de) |
| 17:32.31 | *** join/#asterisk guilherme-jorge (n=guilherm@200.155.185.1) |
| 17:32.37 | *** join/#asterisk b0xii (i=b0xii@cpe-70-116-68-157.houston.res.rr.com) |
| 17:32.56 | x86 | hmm |
| 17:33.16 | *** part/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 17:33.27 | x86 | how can i make it so when someone calls my voicemail extension, it doesnt prompt for the mailbox number, it just uses whatever extension they are calling from as the mailbox number? |
| 17:33.33 | x86 | just asks them for a password |
| 17:33.36 | Hmmhesays | whats the easiest way to return part of a string before a specified character in C? |
| 17:35.18 | niZon | x86: try Voicemailmain(${CALLERIDNUM}) |
| 17:36.29 | I-MOD | Hmmhesays: strtok |
| 17:36.46 | I-MOD | or just look for that character and set it to null |
| 17:37.51 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 17:37.57 | Hmmhesays | hmm ok |
| 17:42.36 | *** join/#asterisk ToTo (n=ToTo@host58-138.pool872.interbusiness.it) |
| 17:42.41 | rajiv|work | pointer: any suggestion on where to buy sangoma hardware? that is any place besides voipsuplpy |
| 17:43.09 | *** join/#asterisk brettnem (n=brettnem@nemeroff.com) |
| 17:43.28 | fjean | How can I get one way voice with a Sipura and two way OK with a softfone, both SIP on same route ? |
| 17:43.57 | fjean | should I use a hammer ? :) |
| 17:44.45 | *** join/#asterisk trelane_ (n=trelane@mail.allthingsit.com) |
| 17:44.49 | *** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.144) |
| 17:44.53 | Kernel_Core | hi all |
| 17:45.05 | Kernel_Core | anybody here configured asterisk and cisco with H323 ? |
| 17:45.17 | *** part/#asterisk maruz (n=maumar@adsl-123-3.38-151.net24.it) |
| 17:45.33 | *** join/#asterisk Nel_ (n=er@207.237.156.254) |
| 17:45.37 | Nel_ | hello there |
| 17:46.25 | Hmmhesays | strtok makes pointer from integer without a cast |
| 17:46.27 | Nel_ | I need some help with an X-lite phone using NAT. The Phone authenticates with asterisk perfectly without NAT but when I try to use NAT I get 403 forbidden and wrong password on the asterisk log. If I just leave the password blank it works...any ideas? |
| 17:46.40 | Kernel_Core | anybody here familiar with h323 ?! |
| 17:47.19 | Hmmhesays | but i have char *new |
| 17:47.37 | Nel_ | anybody has experience with x-lite phones ? |
| 17:49.13 | fjean | kernel - i did install gnugk and oh323... |
| 17:49.36 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 17:49.42 | pointer | rajiv|work: that's where we've been buying them |
| 17:49.54 | *** part/#asterisk pointer (i=pointer@aj.catt.com) |
| 17:50.05 | Nel_ | anybody has experience with x-lite phones ? |
| 17:50.16 | Kernel_Core | fjean: are you familiar with the Nufone driver ? |
| 17:50.19 | *** join/#asterisk lrizzo (n=luigi@81-174-38-222.f5.ngi.it) |
| 17:50.26 | fjean | kernel - sorry no |
| 17:50.49 | fjean | nel_ - I have on my computer but I cant even find the options panel, hehe |
| 17:50.53 | Nel_ | I'm having problems authenticating x-lite phones with NAT |
| 17:50.58 | Nel_ | lol |
| 17:51.09 | Kernel_Core | fjean: is oh323 stable for you ? |
| 17:51.21 | fjean | it "was" |
| 17:51.35 | rajiv|work | pointer: any reason you didnt go with an external gateway like the wellbridge or dlink? |
| 17:51.37 | PakiPenguin | Kernel_Core, use inacess's h323 |
| 17:52.06 | nextime | Kernel_Core : i'm using oh323 in a production system, is enought stable in my opinion |
| 17:52.13 | *** join/#asterisk pawal (n=pawal@c-65fee253.203-1-64736c11.cust.bredbandsbolaget.se) |
| 17:52.13 | jontow | hmm.. weird.. realtime config for IAX2 on one end, flatfiles on the other.. the other can call the realtime box, but not the other way around.. type = friend |
| 17:52.16 | jontow | :/ |
| 17:52.26 | Kernel_Core | PakiPenguin: you mean ooh323 ? |
| 17:52.38 | *** part/#asterisk msw (n=msw@rdu-nat.rpath.com) |
| 17:52.44 | Nel_ | anybody has experience with x-lite phones ? |
| 17:53.20 | cthompson | I'm currently using it at home to test until I can afford a sipura |
| 17:53.26 | cthompson | but I'm a rank newb |
| 17:54.31 | *** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
| 17:54.48 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 17:55.07 | *** join/#asterisk pigpen2 (n=mark@m015f36d0.tmodns.net) |
| 17:55.29 | pigpen2 | Hey guys...I need some quick help....I need to do a test from the inside interface of the wrt through a vpn....so essentially " ping -i 192.168.1.1 172.16.10.10 " Where the 192.168.1.1 is the inside of the wrt...the 172.16.10.10 is a host across a vpn which the wrt is providing. |
| 17:55.42 | pigpen2 | But...the ver of ping on the wrt does not have this advanced function. |
| 17:55.43 | pigpen2 | Ideas? |
| 17:56.11 | pigpen2 | Why did I post this in this cannel..? |
| 17:56.14 | pigpen2 | sorry... |
| 17:57.13 | jontow | woo.. fixed |
| 17:57.13 | jontow | heh :) |
| 17:57.26 | *** part/#asterisk pigpen2 (n=mark@m015f36d0.tmodns.net) |
| 17:57.29 | jontow | man that was dumb.. port = NULL will definitely break things :) |
| 17:57.36 | fjean | anyone uses a Mera MVTS connected to an asterisk ? |
| 18:00.59 | [TK]D-Fender | Ok, I've got a big problem with UDP on an * server. How would I do a UDP port scan with nmap or similar tool to verify that nothing is being filtered? |
| 18:01.33 | [av]bani | ... |
| 18:02.22 | fugitivo | nmap -sU -O -P0 ip |
| 18:02.26 | x86 | how can i make it so when someone calls my voicemail extension, it doesnt prompt for the mailbox number, it just uses whatever extension they are calling from as the mailbox number? |
| 18:02.32 | fugitivo | [TK]D-Fender: ^^^^^ |
| 18:02.35 | *** join/#asterisk [Outcast] (n=outcast@222-152-110-218.jetstream.xtra.co.nz) |
| 18:04.00 | [TK]D-Fender | x86 : pastebin what you've got now |
| 18:05.34 | jbalcomb | x86 applications.conf:exten => _8999,3,VoicemailMain(s${CALLERIDNUM}) |
| 18:05.53 | *** join/#asterisk equanimity (n=alex@stu0254.keble.ox.ac.uk) |
| 18:06.54 | *** join/#asterisk N9URK (n=icechat5@rrcs-70-61-78-165.midsouth.biz.rr.com) |
| 18:07.18 | N9URK | hi guys, where might I find an article on using xlite with asterisk over the Internet? |
| 18:07.26 | jbalcomb | www.google.com |
| 18:07.40 | N9URK | that was funny |
| 18:07.44 | jbalcomb | indeed |
| 18:07.45 | [TK]D-Fender | N9URK : there is a good guid on the WIKI |
| 18:07.58 | N9URK | I can get it going over a lan but not over the internet |
| 18:08.16 | jbalcomb | N9URK http://www.voip-info.org |
| 18:08.29 | sundancer | Anyone going tomorrow to ISS Winter Summit 2006 to Czech / Liberec ? |
| 18:08.32 | N9URK | looking at it now |
| 18:08.35 | N9URK | thanks |
| 18:08.44 | N9URK | will see if that works |
| 18:08.45 | wasim | muahahah ! ... 106 active calls :) |
| 18:08.56 | wasim | we stripped app_dial compeltely ;) |
| 18:09.25 | jbalcomb | naked app_dial is hot |
| 18:09.27 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 18:10.50 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 18:12.12 | jmanq | I am at a loss for configuring a TE110P for a verizon T1 line |
| 18:12.23 | mog_work | anyone use chan_h323 in this channnel? |
| 18:12.32 | mroth_imm | anyone ever experience a flood of "too many open files" messages...got the open file ulimit set to about 1000000 but still getting it daily |
| 18:12.46 | jmanq | anyone have any suggestions for where to go / what to look at? |
| 18:13.00 | mroth_imm | lsof was showing around 2000 files system wide a few minutes prior |
| 18:13.09 | [TK]D-Fender | jmanq : Have you confirmed that the line is active and what settings you should be using with it>? |
| 18:13.13 | mroth_imm | hopefully i'm missing something obvious |
| 18:14.10 | jmanq | I know the line is active (long calls with verizon this morning), where should I look for the settings I should be using? |
| 18:14.30 | mroth_imm | maybe adjust something in /proc/sys/fs as well? |
| 18:15.07 | *** part/#asterisk redondos (n=redondos@190.48.45.160) |
| 18:15.30 | *** join/#asterisk justinu (n=justin@72.18.13.34) |
| 18:16.51 | N9URK | ON the xlite/* question, I followed the instructions at the wiki and now I am getting a "call not approved" msg after dialing a number. What does this indicate? |
| 18:16.57 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 18:17.04 | N9URK | does this indicate something is wrong in the setup? |
| 18:17.16 | Zodiacal | anyone know why the Flash Operator Panel unlocks it self after a refresh? |
| 18:17.18 | N9URK | It does say that I am registered to the * server |
| 18:17.51 | [TK]D-Fender | jmanq : Have you confirmed what signalling they are using on it? |
| 18:17.55 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 18:18.13 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 18:18.22 | jsharp | And are you using the correct kind of cable between your T1 card and the Verizon smartjack? |
| 18:19.28 | jmanq | jsharp: Yeah, we are using the cable the tech guys left for us |
| 18:20.31 | jmanq | [TK]D-Fender: How do I determine what signalling? I assumed this was a standard thing. Is this the type of thing I can call them and ask? |
| 18:20.41 | [TK]D-Fender | yes. |
| 18:20.43 | mroth_imm | wtf...i have no file-max under /proc/sys/fs/ |
| 18:20.46 | [TK]D-Fender | call them |
| 18:20.47 | *** join/#asterisk coolhp (n=crap@modemcable240.139-203-24.mc.videotron.ca) |
| 18:20.49 | mroth_imm | is it me, or is that unusual? |
| 18:21.07 | jsharp | jmanq: What kind of T1 did you get? A PRI or a voice/data circuit or what? |
| 18:21.08 | coolhp | Good day folks ! |
| 18:21.13 | mroth_imm | eh, nevermind |
| 18:21.18 | fjean | eh guys, can we create SIP accounts that don't use username/passowrd but just authentication by IP ? |
| 18:21.21 | coolhp | Anyone interested in trying out the Cisco SIP 8.2 Image for 7940/7960 ? |
| 18:21.32 | jmanq | jsharp: a voice/data circuit |
| 18:22.08 | *** join/#asterisk southtel (n=slester@c-67-191-211-148.hsd1.ga.comcast.net) |
| 18:22.41 | Zodiacal | any ideas? |
| 18:22.44 | jsharp | Can you stick your zaptel.conf and zapata.conf on pastebin.ca so we can look at em? |
| 18:22.48 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 18:23.08 | *** join/#asterisk NexGen (n=me@adsl-70-135-6-65.dsl.tulsok.sbcglobal.net) |
| 18:23.10 | [TK]D-Fender | fjean : I believe a username is required, though not a password... |
| 18:23.22 | NexGen | anyone use a gxp2000 in here? |
| 18:23.43 | fjean | fender - ok |
| 18:24.12 | [TK]D-Fender | fjean : it isn't an account if it doesn't have something to ID it by. The other option would be to treat all calls as "guests" and always do direct IP dialing in your extensions.conf. but thats all just UGLY.... |
| 18:24.27 | [TK]D-Fender | or worse... |
| 18:24.35 | fjean | got it |
| 18:24.40 | [TK]D-Fender | NexGen : Plenty of us have, whats your question? |
| 18:24.51 | jmanq | my configs are online already: http://www.ccs.neu.edu/home/jquintus/asterisk/current/etc/zaptel.conf and http://www.ccs.neu.edu/home/jquintus/asterisk/current/etc/asterisk/zapata.conf |
| 18:25.11 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 18:25.57 | NexGen | well i have one small problem, may not be fixable, but when i make calls using the speakerphone, and the remote party hangs up, is there anyway to get my phone to auto hangup? Right now it waits then gives me a busy signal till I hit speaker and manually hangup |
| 18:27.24 | jsharp | jmanq: Your configurations are for a PRI. |
| 18:27.59 | jsharp | And if you're not running a PRI, that would be your problem. |
| 18:28.00 | jmanq | jsharp: I was begining to think that... I couldn't find any othe examples / doco on the web |
| 18:28.38 | jsharp | You need to call verizon and ask them which T1 channels are for voice and which are for data. |
| 18:28.55 | jsharp | From there, we can configure your zaptel stuff accordingly. |
| 18:29.07 | jmanq | jsharp: ok, I am on the phone as we type |
| 18:31.56 | [TK]D-Fender | oh God... AMP..... |
| 18:33.10 | salviadud | <PROTECTED> |
| 18:35.20 | fugitivo | ~amp |
| 18:35.22 | jbot | amp is probably NOT supported here! people using it should join #amportal |
| 18:36.29 | jaiger | nioniopinasinpgosniepoinpsoainpoinsepoitnpoitmtmmghntnntn |
| 18:36.45 | jaiger | doh, sorry about that folks |
| 18:37.26 | jsharp | Bless you. |
| 18:37.52 | *** join/#asterisk RoyK (n=roy@g-001.osl255.netcom.no) |
| 18:38.01 | *** join/#asterisk southtel (n=slester@c-67-191-211-148.hsd1.ga.comcast.net) |
| 18:40.01 | [av]bani | yay gxp2000 |
| 18:40.47 | southtel | I'm trying to find out some technical details about hunt groups. |
| 18:41.18 | southtel | Is there any way to know what hunt group number was actually dialed? |
| 18:42.09 | southtel | That is to say, if I have a two number hunt group, NXX NXX 1000 and NXX NXX 1001... |
| 18:42.45 | _Sam-- | [av]bani : why yay? |
| 18:42.53 | [av]bani | :) |
| 18:43.05 | southtel | And I dial NXX NXX 1001, but I end up entering the asterisk system via the NXX NXX 1000 extension, is there any way for asterisk to know that I actually dialed NXX NXX 1001? |
| 18:43.19 | *** join/#asterisk htims (n=pd@Vcdff.v.pppool.de) |
| 18:43.40 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 18:43.43 | equanimity | hey all. has anyone managed to get a Netgear TA612V which is locked to Sipgate to connect to their own Asterisk box? |
| 18:44.11 | DrukenHME | eu: what works? |
| 18:44.26 | jmanq | jsharp: I just got off the phone with verizon. We have channels 1 - 5 for voice |
| 18:45.34 | _Sam-- | [av]bani : after all your testing of various phones, has your opinion changed either better or worse about the gxp? |
| 18:45.40 | [av]bani | no |
| 18:45.51 | [av]bani | i doubt it will, its the best $80 phone there is |
| 18:46.10 | _Sam-- | if you had 150 to spend per phone, what would you buy? |
| 18:46.15 | [av]bani | and if you buy a gxp2000 expecting cisco 7985 performance, tough shit |
| 18:46.24 | eu | Guys, what's the best way to eliminate echoing? |
| 18:46.34 | [av]bani | eu -> kill the person talking |
| 18:47.35 | [av]bani | _Sam--: $150? probably an spa-941 |
| 18:47.44 | [av]bani | not much in that price range |
| 18:48.01 | _Sam-- | thats interesting that you like the 941 over the gxp |
| 18:48.08 | _Sam-- | that was kind of what i was trying to see |
| 18:48.30 | *** join/#asterisk tmccrary (n=tmccrary@d47-69-35-227.try.wideopenwest.com) |
| 18:48.46 | [av]bani | _Sam--: for $150? |
| 18:48.59 | jsharp | jmanq: Okay. Lemme conjure up a zaptel & zapata.conf that should work for ya. |
| 18:49.02 | _Sam-- | yep...but for 150 you arent getting a 941 w/ 4 lines? |
| 18:49.06 | [av]bani | if grandstream made a $150 phone, no doubt i'd pick it over the 941. but... grandstream doesnt |
| 18:49.08 | tmccrary | Is Digium's hardware fairly good in general? |
| 18:49.20 | tmccrary | Grandstream = Horrible phones |
| 18:49.36 | [av]bani | _Sam--: now if you had said $180 or $200, the answer would have been very different |
| 18:49.37 | _Sam-- | tmccrary : you are entitled to an opinion, but the gxp really isnt "horrible" at all. |
| 18:50.10 | justinu | playing with the gxp2000 some more... |
| 18:50.13 | tmccrary | Well yes, I suppose I should rephrase that. In my experience, Grandstream phones are horrible flakey pieces of garbage. |
| 18:50.18 | justinu | i upgraded two phones to the latest 2.13 firmware |
| 18:50.24 | justinu | and had echo on SIP->SIP calls |
| 18:50.25 | [av]bani | tmccrary: "spa-841" |
| 18:50.26 | justinu | :( |
| 18:50.29 | tmccrary | I've had 3, one I've already destroyed in a rage |
| 18:50.42 | _Sam-- | i have 0 echo on my gxps |
| 18:50.43 | tmccrary | compared to something like a Snom phone, they're toys |
| 18:50.47 | tmccrary | at least they're cheap though |
| 18:50.51 | [av]bani | snom? HAHAHAHAHAHAHAHAHAHAHAHA |
| 18:50.54 | justinu | you running 2.13? |
| 18:51.00 | _Sam-- | nah im still on the .9 i think |
| 18:51.02 | *** join/#asterisk RoyK (n=roy@g-001.osl255.netcom.no) |
| 18:51.02 | jsharp | jmanq: Here's zapata.conf http://pastebin.ca/44797\ |
| 18:51.03 | [av]bani | you just lost all credibility with me |
| 18:51.04 | jsharp | jmanq: Here's zapata.conf http://pastebin.ca/44797 |
| 18:51.07 | justinu | 2.9? |
| 18:51.07 | _Sam-- | i havent done any updrades for a long time |
| 18:51.10 | justinu | or 1.9? |
| 18:51.11 | _Sam-- | yeah 1.2.9 |
| 18:51.18 | NexGen | i am running 2.13 on mine, its like a whole new phone really |
| 18:51.26 | justinu | i have one customer who's using 1.13 pretty happily |
| 18:51.46 | [av]bani | 2.13 is nice, it re-regs after asterisk restart, which polycoms don't |
| 18:51.49 | justinu | so, either a) 2.13 introduced handset echo again, or b) there are major QA issues with the phones |
| 18:52.04 | _Sam-- | i still have 3 or 4 phones at this office that are 1.13 |
| 18:52.10 | _Sam-- | and i havent had to upgrade |
| 18:52.11 | tmccrary | they're are, serious, SERIOUS QA issues with those phones. Say far, far away |
| 18:52.21 | [av]bani | tmccrary: yes, i agree. snom has serious qa issues |
| 18:52.31 | jsharp | jmanq: And zaptel.conf http://pastebin.ca/44798 |
| 18:52.31 | tmccrary | Are you a Grandstream shareholder or something? |
| 18:52.33 | [av]bani | the firmware is shit. stay far away |
| 18:52.43 | justinu | 2.13 looked good... but we weren't happy with the sound |
| 18:52.49 | [av]bani | tmccrary: no, i own cisco, grandstream, snom, polycom phones |
| 18:52.55 | [TK]D-Fender | SPA-941 isn't worth it vs the Polycom IP 501.... |
| 18:52.57 | tmccrary | wow, so do I |
| 18:53.04 | tmccrary | what a coincidence |
| 18:53.08 | _Sam-- | justinu : is the main complaint how you sound to other people when you are on the gxp? or how they sound to you? |
| 18:53.09 | [av]bani | tmccrary: and snom is a freaking waste of money. nice hardware, freaking shame about the firmware |
| 18:53.14 | [av]bani | tmccrary: the snom firmware is utter trash |
| 18:53.17 | [TK]D-Fender | If the 941 dropped a fair bit it might be worth something... |
| 18:53.17 | tmccrary | I guess if you can't follow directions |
| 18:53.23 | [av]bani | snom has NO IDEA how to design a UI |
| 18:53.27 | justinu | _Sam--: it's how I sound while I'm talking |
| 18:53.30 | tmccrary | Not that they're perfect, but they work well and consistant |
| 18:53.39 | [av]bani | tmccrary: the fucking snoms lock up all the goddamn time. |
| 18:53.50 | [av]bani | tmccrary: read the snom pages. lots of people have the fucking thing lock up. |
| 18:53.57 | NexGen | only thing with the gxp is I have not been able to get work, is the ability to be on one 1 (Account 1) and have asterisk ring my account 2 if another call comes in, dont know if its the phones fault tho |
| 18:54.11 | tmccrary | That's weird, I have had no problems like that with about 150+ phones (Snom 320's) |
| 18:54.12 | _Sam-- | justinu : on my gxps, the mic is so good that when i 'monitor' my sales guys, i hear all kinds of background converstations |
| 18:54.18 | [av]bani | and snom keeps introducing new undocumented "features" in new firmware which bork provisioning |
| 18:54.24 | [av]bani | tmccrary: snom 360 |
| 18:54.27 | _Sam-- | NexGen : enable call waiting |
| 18:54.34 | _Sam-- | login to the gxp and check the radio button |
| 18:54.41 | _Sam-- | that says "disable call waiting" "no" |
| 18:54.41 | justinu | _Sam--: my issue isn't with volume, or anything, it's with the echo... i have no idea why some do it, and some dont |
| 18:54.42 | tmccrary | I have never used a Snom 360, never had a good reason to justify the cost |
| 18:55.12 | tmccrary | With the GXP, is there a way to make the speaker work properly and not get crazy amounts of echo and awesome jimi hendrix style feedback? |
| 18:55.19 | tmccrary | I mean in speakerphone mode |
| 18:55.31 | [av]bani | tmccrary: $30 extra for a backlit display and xml support. pretty clear to me? |
| 18:55.36 | _Sam-- | tmccrary : what, you arent a hendrix fan ? :) |
| 18:55.39 | _Sam-- | you are right about that one. |
| 18:55.40 | tmccrary | Yeah, definately not worth it |
| 18:56.06 | [av]bani | tmccrary: snom good points: good jitter buffer. sips/srtp support. thats about it |
| 18:56.09 | tmccrary | Yes, I like hendrix alot (I generally like all aritistic music) |
| 18:56.28 | [av]bani | snom bad points: weedy speakerphone. buggy as hell firmware. |
| 18:56.37 | _Sam-- | i dont get echo on the speakerphone, but its not terribly loud. |
| 18:56.45 | [av]bani | oh yeah, and US indications are STILL FUCKING WRONG (hello snom? hello?) |
| 18:56.52 | _Sam-- | the speakerphone echo i think was fixed a ways back |
| 18:56.54 | justinu | we tried the speakerphone on gxp firmware 1.9 |
| 18:57.03 | justinu | and it was terrible, so that's what made me upgrade to 2.13 |
| 18:57.08 | *** join/#asterisk UdontKnow (i=udontkno@freenode/staff/udontknow) [NETSPLIT VICTIM] |
| 18:57.27 | _Sam-- | justinu : sip calls to sip devices on the same network even give echo? |
| 18:57.33 | *** join/#asterisk udk (i=udontkno@freenode/staff/udontknow) |
| 18:58.12 | justinu | _Sam--: yep |
| 18:58.17 | _Sam-- | thats insane |
| 18:58.20 | justinu | _Sam--: latency is pretty low... 2ms |
| 18:58.20 | [av]bani | tmccrary: worse yet, snom had the gall to claim to me that "nobody ever complained about US indications" which is bullshit because i have the email address of someone else who complained to snom of the exact same thing |
| 18:58.34 | FuriousGeorge | my trunk my trunk |
| 18:58.45 | PakiPenguin | justinu, got a test gxp? |
| 18:58.52 | _Sam-- | justinu : i noticed i get echo on the gxp when i call the guy who sits 8 feet away from me |
| 18:59.00 | _Sam-- | but thats cause his mic is picking up my call |
| 18:59.09 | [av]bani | heh.. no phone will ever EC that |
| 18:59.10 | backblue | anyone with asterisk clustering without ser? |
| 18:59.17 | [av]bani | not even ciscos |
| 18:59.33 | [av]bani | my cisco 7970 does that when i talk to a polycom in the same room (15 feet) |
| 18:59.44 | _Sam-- | yeah, that is what i was hoping may be justin's problem |
| 19:00.17 | [av]bani | justinu: 2ms ping, but rtp uses 20ms packets |
| 19:00.35 | [av]bani | justinu: most phones will only EC 4-8ms of handset |
| 19:00.41 | [av]bani | becuase thats all they have to EC |
| 19:00.47 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 19:00.57 | [av]bani | its not a phone's job to EC the remote side of the conversation :) |
| 19:03.04 | [TK]D-Fender | [av]bani : So what are your 2 favourite phones these days? |
| 19:04.04 | *** join/#asterisk ketanp (n=ketanp@67.132.43.2) |
| 19:04.24 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 19:04.38 | tmccrary | [av]bani: wow that's too bad. However, my Snom's have been flawless so far aside from one DOA phone that was promptly returned. |
| 19:04.56 | tehdely | weasels have eaten our phone system |
| 19:05.21 | [av]bani | tmccrary: for $200 i expect better. snom has been a complete downer. |
| 19:05.28 | ketanp | who has good intl rates for the major destinations: uk, india, germany, france, israel, etc? |
| 19:05.44 | [av]bani | tmccrary: if i had known these problems ahead of time, i would have bought a polycom instead of a $200 snom |
| 19:06.10 | [av]bani | [TK]D-Fender: favorite in what way? "best phone" or "best value" ? |
| 19:06.24 | [TK]D-Fender | [av]bani : Feel free to list both seperately. |
| 19:06.35 | ketanp | low volume |
| 19:06.41 | [TK]D-Fender | 2 favourites in each class you define |
| 19:06.44 | [av]bani | [TK]D-Fender: best phone -- cisco 7970 and polycom 601. both are FUCKING EXPENSIVE.. best value, gxp2000 |
| 19:07.11 | [av]bani | i think i'm going to sell my snom360. can't freaking stand the POS |
| 19:07.23 | [av]bani | total waste of $200 |
| 19:07.41 | [av]bani | what has happened to that fabled 'german engineering' ? i dont see it in snom at all |
| 19:07.49 | *** join/#asterisk skkip (n=Skipper@216.160.91.91) |
| 19:07.55 | tmccrary | I do. :) |
| 19:08.02 | jmanq | jsharp: Thanks for the config, but now asterisk won't startup |
| 19:08.25 | Zodiacal | anyone know why my outlook tapi dialer isn't adding a 1 in front of long distance numbers? |
| 19:08.39 | jmanq | jsarp: asterisk -vvvgc gets me the error message: Ouch ... error while writing audio data: : Broken pipe |
| 19:08.57 | jmanq | jsharp: the line before that is: "Parsing '/etc/asterisk/zapata.conf': Found" |
| 19:09.15 | [av]bani | [TK]D-Fender: i'm sure if i had a cisco 7985 that would be "best phone", but thats $2400... |
| 19:09.16 | Zodiacal | windows has my area code, so it knows when a number should be long distance and i told it to put a 1 for long distance.. but it doesn't. |
| 19:09.38 | tmccrary | yeah cisco phones are great, its too bad they're price themselves out of the market |
| 19:09.51 | tmccrary | because they're not THAT great |
| 19:10.02 | tmccrary | unless you have all cisco telephony hardware I guess |
| 19:10.04 | [av]bani | tmccrary: they totally blow snom away. in every possible way. |
| 19:10.22 | [av]bani | tmccrary: from a-z, cisco phones are superior to snom |
| 19:10.26 | tmccrary | wow, you definately have a thing for Snom. Jesus, you'd think they killed your father or something |
| 19:11.01 | [av]bani | tmccrary: they blow off my bug reports with bullshit stories like "nobody has ever reported that bug before" |
| 19:11.06 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 19:11.15 | tmccrary | [av]bani: Did it hurt? |
| 19:11.40 | tmccrary | You need to chill out, it's one phone and apparently lousy customer service. :) |
| 19:11.59 | tmccrary | I mean, I didn't even mention Snom when I mentioned Cisco and you started flaking |
| 19:12.03 | tmccrary | about Snom |
| 19:12.04 | zoa | i like snom |
| 19:12.09 | tmccrary | zoa: me too |
| 19:12.16 | tmccrary | But don't tell that to this guy |
| 19:12.19 | tmccrary | hehe :) |
| 19:12.24 | [av]bani | tmccrary: there are better phones for the money. |
| 19:12.26 | zoa | the st-302's are also cool, (but in a different price range) |
| 19:12.38 | tmccrary | is he a bot everyone or what? Like an anti-snom bot? |
| 19:12.39 | [av]bani | tmccrary: much, much, much better phones for the same amount of money. |
| 19:12.45 | tmccrary | I think he is, let me test: |
| 19:12.48 | tmccrary | SNOM SNOM SNOM SNOM |
| 19:12.49 | zoa | the snom 190's are now being sold as elmeg 190 |
| 19:12.57 | zoa | they are very good bang for the buck |
| 19:13.17 | [av]bani | oh, you are so funny. man. that is just so unspeakably awesome. i must bow to your incredible sense of humor. |
| 19:13.23 | [av]bani | please continue. |
| 19:13.29 | tmccrary | I imagined you saying that in a Stewie voice |
| 19:13.41 | [av]bani | good for you. |
| 19:14.13 | tmccrary | zoa: Who's elmeg? |
| 19:14.30 | [TK]D-Fender | [av]bani : The IP 601 @ $240 isn't so bad.... |
| 19:14.38 | jbalcomb | Subject: HELP |
| 19:14.39 | jbalcomb | "You are in a maze of twisty little passages, all alike." |
| 19:15.50 | [av]bani | [TK]D-Fender: one of these days polycom will discover the fabled 'backlight' |
| 19:16.01 | jbalcomb | [av]bani I dream of this day.. |
| 19:16.26 | jbalcomb | Can we go back to Cisco /pricing/ themselves out of the market? |
| 19:16.29 | [av]bani | jbalcomb it will be glorious, angels will descend from the heavens |
| 19:16.37 | [TK]D-Fender | [av]bani : True, but their # of failings is still seriously counter-balanced by their virtues, also for the price. |
| 19:17.09 | [av]bani | [TK]D-Fender: of course, polycom will do something stupid at the same time, like release an ip701 which is an ip601 with backlight, but at $350 |
| 19:17.42 | [av]bani | and it appears, polycom is finally fixing their buddy limit |
| 19:18.12 | [av]bani | i guess people kept getting pissed off that their expensive sidecars were useless with asterisk |
| 19:19.29 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:20.49 | [av]bani | and yay, 7970 now has sip firmware |
| 19:20.50 | jbalcomb | maybe you guys should all get together and make the most perfect phone, perfect PRI card, and perfect phone system? You could probably make bazillions of dollars. |
| 19:21.14 | [av]bani | jbalcomb: sangoma already makes the perfect pri card |
| 19:21.34 | [TK]D-Fender | [av]bani : what are the details of the buddy limit fix? |
| 19:21.35 | jbalcomb | 1 down, 2 to go. Quick, get on it before someone else does. |
| 19:21.39 | _Sam-- | if all the cisco phones support sip , what is the point of sccp? |
| 19:21.44 | bkw_ | [av]bani, it does more than PRI |
| 19:21.44 | [av]bani | [TK]D-Fender: they increased it to like 38 or something |
| 19:21.47 | _Sam-- | just trying to understand why its better |
| 19:21.49 | [av]bani | _Sam--: cisco is migrading to sip |
| 19:21.56 | [av]bani | migrating |
| 19:22.01 | jbalcomb | mitigating |
| 19:22.02 | [TK]D-Fender | [av]bani : in new SIP release? |
| 19:22.11 | [av]bani | [TK]D-Fender: thats what the rumor mill says |
| 19:22.24 | bkw_ | where is the 7970 firmware online? |
| 19:22.25 | bkw_ | in CCO? |
| 19:22.27 | bkw_ | I can't find it |
| 19:22.36 | [av]bani | _Sam--: sccp locks you in to cisco's call manager (or so cisco hopes...) |
| 19:22.37 | tmccrary | What do you guys think of Digium's PRI card? |
| 19:22.56 | [av]bani | bkw_: cmterm-7970_7971-sip.8-0-2-0.cop |
| 19:22.59 | jbalcomb | tmccrary I think they are awesome cause I own three of them. |
| 19:23.00 | [TK]D-Fender | [av]bani : Rumour mill tends to put out "white bread" material when what we really want is multi-grain and wholesome! |
| 19:23.13 | bkw_ | [av]bani, and thats all I need? |
| 19:23.13 | [av]bani | bkw_: it's in the release notes too. |
| 19:23.16 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 19:23.26 | [av]bani | bkw_: i guess, though nobody seems to have tested it yet. |
| 19:23.31 | bkw_ | does it work with asterisk yet? |
| 19:23.31 | bkw_ | I wonder |
| 19:23.44 | [av]bani | dunno, somoene said it doesnt support BLF yet |
| 19:23.45 | jbalcomb | tmccrary wait, what I mean is, I have three of them and I have no choice so they'd better be awesome. |
| 19:23.47 | [av]bani | which is a big downer |
| 19:23.58 | [av]bani | seems odd to me though |
| 19:24.51 | jalsot | hi |
| 19:26.47 | bkw_ | screw blf |
| 19:26.48 | jbalcomb | tmccrary I have had a bit of trouble with echo, jitter, and drops but the GXP-2000 seems more responsible than the PRI card. Of course, the config here is hodgepodge and I'm a nub so that may factor in as well. |
| 19:26.51 | bkw_ | I want a 7970 with sip |
| 19:26.52 | bkw_ | haha |
| 19:26.55 | bkw_ | that works |
| 19:27.46 | tmccrary | You mean your GXP-2000 is more reliable than Digium's PRI? |
| 19:28.11 | fugitivo | debian doesn't let you use 2 instances of apt-get at the same time? that sucks |
| 19:29.14 | tmccrary | what do you mean two instances? |
| 19:29.22 | tmccrary | oh, file locking |
| 19:29.26 | tmccrary | yeah, only one |
| 19:29.38 | fugitivo | yes that, sucks :) |
| 19:29.38 | tmccrary | You may be able to fetch with one, however, I've never tried |
| 19:30.18 | tmccrary | Why do you need two instances of apt going? you can just go like: apt-get install package1 package2 |
| 19:30.51 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 19:30.54 | fugitivo | waste of time |
| 19:31.51 | fugitivo | for example, downloading a package of 50mb when installing 5 small packages at the same time |
| 19:32.03 | fugitivo | if i download that package, i can't install anything else |
| 19:32.11 | fugitivo | (the big package) |
| 19:32.22 | tmccrary | you should try try using -d on that one |
| 19:32.29 | fugitivo | what does -d do? |
| 19:32.37 | tmccrary | -d, --download-only |
| 19:32.37 | tmccrary | <PROTECTED> |
| 19:32.37 | tmccrary | <PROTECTED> |
| 19:32.39 | mikefoo | just downloads |
| 19:32.40 | tmccrary | from man :) |
| 19:32.54 | tmccrary | honestly, it may still not work that way, I have never tried it |
| 19:33.18 | fugitivo | i'll just stick with gentoo and emerge |
| 19:33.43 | tmccrary | gentoo, eek |
| 19:34.03 | tmccrary | it works great if you're not using it in production or for anything serious. I found that one out the hardway |
| 19:34.06 | [av]bani | http://funroll-loops.org/ |
| 19:34.06 | Nel_ | anybody has experience with x-lite phones ? |
| 19:34.16 | mikefoo | gentoo the os that doesn't care about time.. |
| 19:34.18 | fugitivo | not true, i use a lot of gentoo boxes in production |
| 19:34.20 | mikefoo | 3 day installs |
| 19:34.38 | tmccrary | Believe me, you should SWITCH AWAY FROM GENTOO NOW BEFORE IT'S TOO LATE. |
| 19:34.59 | fugitivo | 2 1/2 years working with gentoo |
| 19:34.59 | dgorski | woa |
| 19:35.00 | tmccrary | Gentoo slowly degrades, it's like windows 98 |
| 19:35.00 | dgorski | wrong |
| 19:35.01 | fugitivo | no problems at all |
| 19:35.02 | dgorski | wrong |
| 19:35.10 | dgorski | what kind of nonsense is that? |
| 19:35.13 | dgorski | gentoo is awesome |
| 19:35.15 | Nel_ | I have authentication problems through NAT, with X-Lite. The phone registers fine with no password any ideas? |
| 19:35.28 | mikefoo | gentoo is horrible. |
| 19:35.34 | tmccrary | I'm not having thing argument here, but believe me, you are going to find out the hard way about this. |
| 19:35.51 | fugitivo | tmccrary: i didn't yet, so i don't think i'll find any problem with it |
| 19:35.54 | mikefoo | "oh lets just cource everything so we can be cool, and pretend everything works smoothly, because its source" |
| 19:35.59 | mikefoo | source* |
| 19:36.00 | fugitivo | tmccrary: i'm running big production boxes with gentoo |
| 19:36.15 | *** join/#asterisk Gamercjm (n=Gamercjm@pool-71-254-164-89.lsanca.fios.verizon.net) |
| 19:36.27 | fugitivo | mikefoo: when we started using linux, there was no packaging systems like today |
| 19:37.17 | tmccrary | its not bad because it's source-based, it's bad because it unravels itself after about 2 1/2 to 3 years time |
| 19:37.17 | mikefoo | ofcourse, not get with it.. |
| 19:37.17 | mikefoo | its 2006 not 1996 |
| 19:37.17 | *** join/#asterisk crich1999 (n=crich@port-212-202-198-154.dynamic.qsc.de) |
| 19:37.17 | fugitivo | mikefoo: i liked 1996 |
| 19:37.31 | mikefoo | tmccrary: I never said source is bad, I am saying they think source is the answer to all, no matter how long it takes. |
| 19:37.35 | dgorski | I use binary packages with gentoo all the time |
| 19:37.40 | [av]bani | fugitivo: even slackware had "packages". they were called tarballs ;) |
| 19:38.03 | fugitivo | [av]bani: yes, i still use that |
| 19:38.15 | fugitivo | oh hell! |
| 19:38.17 | fugitivo | look! |
| 19:38.21 | fugitivo | i'm compiling asterisk from source! |
| 19:38.25 | fugitivo | i'm a caveman! |
| 19:38.28 | tmccrary | hehe |
| 19:38.29 | mikefoo | even solaris has packages |
| 19:38.43 | fugitivo | oh hell, i'm applying patches!!! |
| 19:38.50 | fugitivo | i'm an old man! |
| 19:39.24 | *** join/#asterisk Ramzi-324 (n=Acme@fctnnbsc16w-156034225070.nb.aliant.net) |
| 19:39.39 | Gamercjm | Im having a problem trying to make a .call file, In the Channel, Do i just put the IAX2/user@voip/9510000000 ? |
| 19:39.54 | Gamercjm | the 9510000000 being the number to call? |
| 19:40.05 | *** join/#asterisk detatch (n=akwairc@209.107.188.100) |
| 19:40.57 | detatch | Can anyone tell me how I can divert calls into a different queue if there are no agents logged into a given queue? |
| 19:43.54 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 19:44.16 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 19:45.10 | wasim | detatch: joinempty=no |
| 19:45.20 | wasim | detatch: and next priority send them to a different queue |
| 19:49.17 | *** join/#asterisk trelane_ (n=trelane@208.64.32.51) |
| 19:49.23 | FuriousGeorge | can anyone comment on the sangoma a100 vs the tdm400? |
| 19:49.26 | Gamercjm | can anyone help getting the .call files to work? |
| 19:49.59 | *** join/#asterisk darby_t (i=darby_t@dlm81.neoplus.adsl.tpnet.pl) |
| 19:50.17 | [TK]D-Fender | FuriousGeorge : A200 can have on-board EC which is "godly", and expands cheaper than TDM's A200 is also IRQ friendly and PCI voltage agnostic. |
| 19:50.46 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 19:51.13 | [av]bani | [TK]D-Fender has an a200? |
| 19:51.49 | [av]bani | or are you just assuming based on the 104d :) |
| 19:52.09 | FuriousGeorge | [TK]D-Fender: yeah i heard about the hw echo can. im kinda ticked i got all these tdm everywhere :) |
| 19:52.11 | [TK]D-Fender | [av]bani : Enough confirmations and some assumptions, yes :) |
| 19:52.14 | FuriousGeorge | and they are about the same price :) |
| 19:52.31 | *** join/#asterisk AJay-MN (i=AJay@63.231.252.9) |
| 19:53.17 | AJay-MN | is there a reason why my SIP phones will register with Asterisk and have a reregistering set to every 3mins, but never does, then after 1 hour Asterisk unregisters the device??? |
| 19:53.49 | *** join/#asterisk backblue (n=moo@87-196-1-157.net.novis.pt) |
| 19:54.11 | detatch | wasim> next priority in the dialplan you mean? s,2,queue(whatever)? |
| 19:54.11 | sundancer | Hm my Swissvoice IP10S keeps sending BYE sip:2464@193.77.x.y message to asterisk.. any idea why? And when i call this particular phone it keeps telling me: SIP/2.0 486 Busy Here |
| 19:54.27 | wasim | detatch: oui |
| 19:54.35 | sundancer | But i can see it by `sip show peers` |
| 19:54.52 | detatch | cool |
| 19:54.53 | sundancer | And asterisk and phone are exchanging keepalive packets |
| 19:54.55 | detatch | thanks |
| 19:56.00 | *** join/#asterisk Ref^Smokey (i=biGhumaN@cpe-66-67-100-79.rochester.res.rr.com) |
| 19:58.43 | *** join/#asterisk Mw3 (n=mw3@national.t-error.hu) |
| 19:58.57 | *** join/#asterisk carb0n^ (n=carbo@137.101.17.34) |
| 20:00.06 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 20:00.12 | jorgito | somebody from czech republic here ? |
| 20:01.07 | carb0n^ | anyone from pakistan ? |
| 20:01.09 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 20:01.24 | jorgito | carb0n^, you are realy from pakistan? |
| 20:01.34 | carb0n^ | yes |
| 20:01.44 | [av]bani | anyone from north korea? |
| 20:01.53 | *** join/#asterisk mattwj2005 (n=Matt@dialup-4.158.216.91.Dial1.Chicago1.Level3.net) |
| 20:02.39 | jorgito | carb0n^, that is cool |
| 20:05.02 | *** join/#asterisk backblue (n=moo@87-196-32-230.net.novis.pt) |
| 20:05.07 | backblue | why use TDMoE in asterisk clustering and not use SIP or IAX trunks? |
| 20:05.40 | mattwj2005 | hey guys....any good ulimited incoming and outgoing voip service providers (for within the US)? |
| 20:06.18 | mattwj2005 | sorry I know I have asked this question quite a few times.....I thought I had one....but they their support service and web site suck |
| 20:06.30 | xachen | what? sixtel? :P |
| 20:07.29 | mattwj2005 | url? |
| 20:08.08 | xachen | I was asking if you were with iax.cc |
| 20:09.06 | mattwj2005 | I asked for more information from telasip.....they never got back to my e-mail and I tried calling a few times and no one was home |
| 20:09.07 | *** join/#asterisk Dr-Linux (n=Nothing@host202-147-168-130.lhr.dancom.net.pk) |
| 20:09.39 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 20:10.05 | mattwj2005 | I just want to set up an asterisk box for my home phone |
| 20:10.22 | mattwj2005 | looking to save a few bucks along the way :P |
| 20:13.10 | *** join/#asterisk carb0n^ (n=carbo@137.101.17.34) |
| 20:17.30 | *** join/#asterisk gaspiz (n=gaspiz@86.34.6.164) |
| 20:17.38 | mattwj2005 | anyone in the room have a favorite ulimited service? |
| 20:17.39 | mattwj2005 | :) |
| 20:17.51 | [TK]D-Fender | Most "unlimited" places cost more than they are used for... |
| 20:18.33 | AndyCap | didn't you just ask this btw? :-P |
| 20:18.52 | mattwj2005 | lol |
| 20:18.53 | mattwj2005 | true |
| 20:18.54 | gaspiz | hi, I have a problem with ser actually. in the responses it puts a Warning: 392 my_ip:port "Noisy feedback tells: pid=24601 |
| 20:19.00 | gaspiz | is this a problem? |
| 20:19.44 | mattwj2005 | I guess as far as service goes.....I could always use voipjet |
| 20:19.50 | mattwj2005 | they are only 1.3 cents a min |
| 20:22.08 | mattwj2005 | how about unlimited DID's.....anyone know of a place to get some in the 715-229-xxxx and 651-392-xxxx ranges? |
| 20:22.56 | mattwj2005 | sorry guys....I am kinda of a noob :P |
| 20:24.28 | DrukenHME | anyone setup a spa-3000 to work as a incoming fxo ? |
| 20:24.50 | mroth_imm | anybody ever encounter the "too many open files" error during socket allocation on a machine with a huge ulimit -n? |
| 20:24.55 | [TK]D-Fender | mattwj2005 : Stop spamming the same stupid question every 5 minutes! Check the WIKI for and mailing lists for recommendations, and try voxilla.com forums as well. |
| 20:25.03 | [TK]D-Fender | DrukenHME : I have one |
| 20:25.19 | DrukenHME | [TK]D-Fender: feel like giving me a hand? i just got one in today |
| 20:25.21 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
| 20:26.06 | [TK]D-Fender | DrukenHME Will see what I can do.... I don't have it handy. |
| 20:26.15 | jmanq | I have been having troubles making a call from a sip phone connected to my asterisk box to my cell phone |
| 20:26.34 | jmanq | my asterisk box is attached to the pstn with a T1 card |
| 20:26.47 | jsharp | Did you get your T1 to come up? |
| 20:27.01 | Dr-Linux | [TK]D-Fender: sir salaam |
| 20:27.09 | jmanq | jsharp: Maybe! |
| 20:27.15 | DrukenHME | [TK]D-Fender: can ya remember anything special about NOT getting the dialtone? |
| 20:27.46 | jmanq | jsharp: thanks for the help earlier btw. you left before I could thank you |
| 20:27.49 | [TK]D-Fender | DrukenHME : You working on the FXS port now, or the FXO port? |
| 20:27.56 | DrukenHME | the FXO |
| 20:28.01 | jsharp | No problem. Lunch was calling and my stomach won over. |
| 20:28.07 | [TK]D-Fender | DrukenHME : is it registering? |
| 20:28.13 | DrukenHME | yeah |
| 20:28.17 | DrukenHME | that was the easy part |
| 20:28.31 | [TK]D-Fender | show me what you use to try to dial out on it. |
| 20:28.42 | [TK]D-Fender | (extensions.conf) |
| 20:28.44 | DrukenHME | i don't want to dial out.... |
| 20:28.48 | DrukenHME | i want to dial in... |
| 20:29.04 | [TK]D-Fender | have you TRIED dialing out on it? a somewhat important test... |
| 20:29.28 | jmanq | jsharp: So it seems as if the call gets dropped as soon as I pick up my cell phone |
| 20:29.47 | DrukenHME | no... cause i want it as a dial-in only device :) |
| 20:29.48 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-145-52.red.bezeqint.net) |
| 20:30.06 | jsharp | What do you see on your asterisk console when the call goes out. |
| 20:30.28 | [TK]D-Fender | DrukenHME : this page covers it pretty well.... http://www.voip-info.org/wiki/index.php?page=Sipura+3000 |
| 20:30.32 | mattwj2005 | sorry [TK]D-Fender. Thanks for the website :) |
| 20:31.12 | DrukenHME | ya i guess i should have checked the wiki :) |
| 20:31.18 | DrukenHME | but that search annoys me :) |
| 20:31.40 | [TK]D-Fender | DrukenHME : pay attention to : Dial Plan 2: (S0<:15551234567>) in the web-config for the 3k. it will dial this exten in the FXO accounts sip.conf entry like a DID. |
| 20:31.49 | [TK]D-Fender | this is the key... |
| 20:31.57 | jmanq | jsharp: http://pastebin.ca/44816 |
| 20:32.07 | [TK]D-Fender | as well as the other little bits that tell it to route by IP. |
| 20:32.31 | jmanq | jsharp: I think i was getting an attempt at an incoming call at the same time |
| 20:33.04 | jmanq | those don't work yet either ;) |
| 20:33.28 | jsharp | Hmmm. You've got a pretty complicated dial plan to test with. |
| 20:35.02 | jmanq | how could i simplify it? |
| 20:35.43 | *** part/#asterisk gaspiz (n=gaspiz@86.34.6.164) |
| 20:35.57 | *** join/#asterisk delox99 (n=delox@modemcable246.108-203-24.mc.videotron.ca) |
| 20:36.07 | delox99 | Hi all |
| 20:36.17 | delox99 | i have a question on peers and codecs |
| 20:36.40 | [TK]D-Fender | delox99 : shoot |
| 20:36.43 | delox99 | im registering to a peer that accepts ulaw or g729 |
| 20:37.10 | delox99 | i would like my sip users to be able to connect to this peer using g729 |
| 20:37.15 | Katty | Hmmhesays: you around? |
| 20:37.22 | jsharp | jmanq: Post your extensions conf and I'll see about trimming it down to test. |
| 20:37.40 | delox99 | on the other, these users all have mailboxes that works like you know in gsm |
| 20:37.51 | delox99 | i want them to use the mailboxes using gsm |
| 20:37.56 | delox99 | is it possible? |
| 20:38.30 | *** join/#asterisk exonic (n=exonic@209.172.11.54) |
| 20:38.37 | [TK]D-Fender | delox99 : I believe if you set order of allowed codecs in sip.conf for those phones and peers that it will pick ULAW for VM locally... |
| 20:38.49 | delox99 | so if connection to trunk use g729 elseif connection to mailboxes use gsm |
| 20:38.50 | [TK]D-Fender | and do the rest as pass-through without requiring licenses. |
| 20:39.01 | exonic | Anyone familiar with asterisk manager API ? |
| 20:39.18 | delox99 | im able to do g729 as passthrough no problem |
| 20:40.01 | [TK]D-Fender | delox99 : if you allow a 2nd codec it should fallback to it for local stuff that isn't pass-through. |
| 20:40.02 | delox99 | if i set g729 ulaw in [general] in sip.conf, it always defaults to g729 even for mailboxes |
| 20:40.12 | *** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
| 20:40.20 | [TK]D-Fender | delox99 : pastebin your sip.conf |
| 20:40.24 | exonic | I'm writing my own manager interfcae (think a max tnt interface) and I don't know what to do when I get the 'Dial' event... |
| 20:40.25 | [TK]D-Fender | ~pb |
| 20:40.27 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 20:40.35 | delox99 | how do i use pastebin? |
| 20:40.50 | delox99 | sorry not an mirc guru |
| 20:40.52 | [TK]D-Fender | go to the site, copy&paste your sip.conf and change the passwords. |
| 20:41.00 | [TK]D-Fender | nothing to do with IRC |
| 20:41.21 | jmanq | jsharp: http://pastebin.ca/44818 As a warning, the file is very long |
| 20:41.53 | jmanq | jsharp: it is pretty much the same as the extensions.conf that I got working with my analog card several months ago |
| 20:44.03 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 20:44.15 | delox99 | if i put ulaw,g729 in general in sip.conf and specify allow g729 for the peer it still uses g729 |
| 20:44.19 | jsharp | Hrm. |
| 20:44.53 | [TK]D-Fender | delox99 : just do the pastebin and I'll see where its failing... |
| 20:45.04 | [av]bani | http://home.xnet.com/~raven/Sysadmin/VoiceMail.html |
| 20:45.37 | exonic | Anyone care to checkout a ncurses asterisk interface? |
| 20:45.43 | exonic | I'm looking for some feedback |
| 20:45.48 | *** join/#asterisk Lino` (n=Lino@i577BD070.versanet.de) |
| 20:45.56 | Juggie | exonic, screen shots? :) |
| 20:46.57 | backblue | anyone with experience in asterisk cluxtering? |
| 20:47.03 | *** join/#asterisk gaspiz (n=gaspiz@86.34.6.164) |
| 20:47.20 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 20:47.40 | gaspiz | hi, can anybody give me an example of a contact header in a 302 SIP message? |
| 20:50.19 | heison | has iaxtel been abandoned? |
| 20:50.34 | exonic | Juggie, I'll have to get some screenshots, but I don't like showing the personal information in 'em |
| 20:51.57 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 20:51.57 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:52.10 | DrukenHME | w00t! |
| 20:52.14 | DrukenHME | i think i got it working |
| 20:52.28 | delox99 | ok sorry for delay |
| 20:52.33 | delox99 | here it is |
| 20:52.34 | delox99 | http://pastebin.com/589606 |
| 20:53.01 | jbalcomb | [TK]D-Fender how do you feel about the Superdial Macro? http://www.voip-info.org/wiki/view/Superdial+macro |
| 20:53.05 | mikefoo | heison: you use iaxtel? |
| 20:53.57 | heison | mikefoo: it's been in my dialplan for a while, don't know if it's still alive... or should i just remove it altogether? |
| 20:54.24 | mikefoo | ::shrug:: |
| 20:54.38 | mikefoo | who is your main outbound dials? |
| 20:54.52 | heison | Bell and Nufone |
| 20:54.59 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 20:55.13 | mikefoo | url for bell? |
| 20:55.15 | [TK]D-Fender | delox99 : Where is your phone definition? I only see your carrier |
| 20:55.26 | heison | Bell Canada, CLEC |
| 20:55.41 | mikefoo | whoa.. nufone has got a new website |
| 20:55.44 | mikefoo | niice.. heh |
| 20:56.09 | [TK]D-Fender | jbalcomb : "Shit-on-a-stick". Mine is clearly superior (and 1.2 compliant ;)) |
| 20:56.17 | mikefoo | [TK]D-Fender: hey sup |
| 20:58.07 | [TK]D-Fender | jbalcomb : Mine allows for immediate VM access, selective operator for VM exit, and optional VM on busy/no-answer. |
| 20:58.14 | jbalcomb | [TK]D-Fender Superdial Macro has a 1.2 version. |
| 20:58.15 | [TK]D-Fender | mikefoo : blarg.... |
| 20:58.29 | [TK]D-Fender | jbalcomb : "show me the money".... |
| 20:59.05 | jbalcomb | [TK]D-Fender I'm on it. Still getting the new server ready. |
| 20:59.25 | nextime | notOnyx: anyone know a good termination for mexico destination? ( good = hight ASR and low rates for hight volume )? |
| 20:59.29 | nextime | ops |
| 20:59.37 | nextime | s/notonyx/OT |
| 20:59.40 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 21:00.36 | CunningPike | Does anyone know if the hint priority supports patterns e.g. exten => _401[0-2],hint,SIP/${EXTEN} |
| 21:00.41 | _Sam-- | pike = pi kappa alpha? |
| 21:01.31 | CunningPike | http://paul.merton.ox.ac.uk/ascii/cunning-pikes.html |
| 21:01.41 | zoa | hey ho sam |
| 21:01.46 | CunningPike | :D |
| 21:01.55 | zoa | were you looking for me some days ago ? |
| 21:01.59 | *** join/#asterisk aster1sK (n=bimbo@0x50c72269.adsl-fixed.tele.dk) |
| 21:02.06 | *** part/#asterisk aster1sK (n=bimbo@0x50c72269.adsl-fixed.tele.dk) |
| 21:02.24 | _Sam-- | zoa : probably...but whatever it was i forget now :) |
| 21:02.40 | _Sam-- | zoa: how long until i can test SIP on fisk? |
| 21:02.50 | delox99 | sorry for delay again i was on the phone |
| 21:02.51 | _Sam-- | i found with my setup that SIP sounds twice as nice as SIP, for ME. |
| 21:02.58 | _Sam-- | er twice as nice as IAX |
| 21:03.09 | _Sam-- | i switched everything over to sip, all connections...and its like POTs lines here |
| 21:03.19 | [TK]D-Fender | CunningPike : I doubt it highly, and for the sample you showed, just not worth it. |
| 21:03.27 | heison | is IAXtel really gone? |
| 21:04.01 | _Sam-- | whats going on Dr-Linseed |
| 21:04.06 | CunningPike | [TK]D-Fender: Thanks - agreed for the sample, but ideally I'd like it to be for all extensions |
| 21:04.59 | [TK]D-Fender | CunningPike : how many? |
| 21:05.03 | Dr-Linux | _Sam--: we hired 2 developers for C/Csharp , they are working with AGI stuff :) |
| 21:05.05 | DrukenHME | [TK]D-Fender: you got a stun server? |
| 21:05.10 | [TK]D-Fender | DrukenHME : nope |
| 21:05.17 | DrukenHME | k |
| 21:05.33 | CunningPike | [TK]D-Fender: 400 :D |
| 21:06.31 | CunningPike | What I'm looking at is providing a hint priority for all our extensions, allowing us to then configure BLF as needed, without adding 100's of lines to our extensions.conf |
| 21:06.54 | [av]bani | c# :| |
| 21:07.30 | CunningPike | Ideally what I would like to do is exten => _XXXX,hint,SIP/${EXTEN} |
| 21:07.32 | delox99 | sorry the damn phone again |
| 21:07.41 | delox99 | the sip users are in a database |
| 21:08.24 | *** part/#asterisk gunsch (n=linux@pD954160C.dip0.t-ipconnect.de) |
| 21:08.31 | backblue | no one with asterisk clustering experience? |
| 21:08.48 | delox99 | i dont have any codecs specified for the sip users so i think they should use the settings in the general section of sip.conf right? |
| 21:09.22 | [TK]D-Fender | CunningPike : Nows a good time to use "INCLUDE" in your dialplan :) |
| 21:09.27 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 21:09.47 | [TK]D-Fender | delox99 : I never trust what I don't set personally :) |
| 21:10.02 | delox99 | on the physical phones i can specify g729 first then ulaw but it still defaults to ulaw i fin sip.conf > general i have ulaw,g729 |
| 21:10.06 | jmanq | jsharp: any thoughts? |
| 21:10.07 | [TK]D-Fender | delox99 : and realtime codec prioritization I've heard funny things about... |
| 21:10.13 | *** join/#asterisk oatis (n=admin@ppp-71-142-33-177.dsl.scrm01.pacbell.net) |
| 21:10.18 | jsharp | No. I don't see why it wouldn't work. |
| 21:10.18 | delox99 | oh yeah? |
| 21:10.20 | delox99 | could be |
| 21:10.34 | [TK]D-Fender | delox99 : yup... you might want to consider buying a few licenses |
| 21:10.35 | CunningPike | [TK]D-Fender: Hmmm - the wiki seems to indicate that 1.2.2 onwards should allow global variables....... |
| 21:10.47 | oatis | Hi, I am getting an error on startup |
| 21:10.51 | oatis | This is what I am getting in the messages log... "loader.c: Loading module chan_modem.so failed!" |
| 21:11.23 | [TK]D-Fender | CunningPike : That wouldn't change anything... they wouls still be static and the hint is "registered", not "evaluated". so thats a no-go as far as practicality is concerned |
| 21:11.40 | delox99 | buying licenses, thats what ill end up doing i think |
| 21:11.44 | [TK]D-Fender | oatis : then do a "noload => chan_modem.so" in modules.conf |
| 21:13.09 | oatis | D-Fender, Cool, thanks |
| 21:16.16 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 21:16.21 | Hmmhesays | well i finally got mwi waiting indicators to work between asterisk and SER |
| 21:16.24 | Hmmhesays | pretty sweet |
| 21:16.39 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 21:16.40 | MikeJ[Laptop] | yes you are! |
| 21:16.44 | backblue | omfg, no one here, work under a voip isp? |
| 21:16.51 | MikeJ[Laptop] | ? |
| 21:16.55 | *** join/#asterisk e3eli3h (n=not@82.102.94.82) |
| 21:16.56 | MikeJ[Laptop] | backblue, huh? |
| 21:16.58 | Hmmhesays | haha |
| 21:18.08 | Dr-Linux | Katty: is the one |
| 21:18.12 | CunningPike | [TK]D-Fender: ${EXTEN} doesn't work, as you surmised. Oh well - massive included file coming right up..... |
| 21:19.08 | Katty | Dr-Linux: bwa? |
| 21:19.16 | Hmmhesays | ahh Dr-Linux |
| 21:19.18 | Hmmhesays | boobies? |
| 21:19.52 | oatis | I have 1 more questions. I changed servers and now with the new install my voice mail is not working although I am using the exact same config files.. it looks like /usr/share/asterisk/sounds/voicemail points to /var/spool/asterisk/voicemail and It doesn't record the message I leave for that particular extention |
| 21:20.10 | oatis | any extention |
| 21:20.12 | oatis | actually |
| 21:20.30 | Hmmhesays | did you compile one and use packages on another? |
| 21:20.46 | [TK]D-Fender | oatis L Could be a rights issue |
| 21:20.49 | DrukenHME | boobies? |
| 21:21.06 | *** join/#asterisk naturalblue (n=Administ@87.192.100.109) |
| 21:21.18 | oatis | D-Fender, looks like its owned by root, who and what permissions should be assigned to it? |
| 21:21.25 | *** join/#asterisk mattwj2006 (n=Matt@dialup-4.159.47.59.Dial1.Chicago1.Level3.net) |
| 21:21.50 | Dr-Linux | Hmmhesays: i'll cut your wife hair :@ |
| 21:22.07 | Hmmhesays | closest thing I have ot a wife is my engaged girlfriend |
| 21:22.14 | [TK]D-Fender | oatis : Depends what * runs under... |
| 21:22.16 | Hmmhesays | and she's not engaged to me so thats not very close |
| 21:22.23 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 21:22.31 | dgorski | <PROTECTED> |
| 21:22.38 | [TK]D-Fender | Hmmhesays : Engaged to YOU I hope or things could turn awkward :) |
| 21:22.51 | dgorski | one should not "point at" the other |
| 21:22.52 | Dr-Linux | Hmmhesays: lol |
| 21:23.17 | *** join/#asterisk AJay-MN (i=AJay@63.231.252.9) |
| 21:23.27 | Dr-Linux | Katty: bwa? :S |
| 21:23.33 | Katty | Dr-Linux: sign of confusion. |
| 21:23.34 | AJay-MN | Anyone having problems getting a Grandstream 101 to stay registered? |
| 21:23.54 | naturalblue | nyone here ever used ADM |
| 21:23.56 | Dr-Linux | [TK]D-Fender: sir what does bwa mean in your language? :S |
| 21:24.08 | Dr-Linux | Katty: ooo ic |
| 21:24.27 | Hmmhesays | [TK]D-Fender, negative on that one |
| 21:25.27 | MikeJ[Laptop] | Hmmhesays, BAD! |
| 21:25.28 | Darwin35 | TELIAX |
| 21:25.37 | MikeJ[Laptop] | hmm |
| 21:25.41 | MikeJ[Laptop] | it's Darwin35 |
| 21:25.57 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 21:26.00 | Darwin35 | whas going on |
| 21:26.17 | MikeJ[Laptop] | ummm |
| 21:26.20 | MikeJ[Laptop] | nothing too much |
| 21:26.28 | MikeJ[Laptop] | well.. quite a lot actually |
| 21:27.29 | Darwin35 | fill em in |
| 21:28.01 | oatis | Looks like /var/spool/asterisk/voicemail didn't get created when I installed the package |
| 21:28.14 | Qwell[] | oatis: When you get the first voicemail, it will be created |
| 21:28.20 | DrukenHME | hmm... figures |
| 21:28.24 | Qwell[] | or..should |
| 21:28.41 | [av]bani | \o> |
| 21:28.42 | Darwin35 | no it builds that dir the first time you run asterisk |
| 21:28.53 | oatis | Qwell, for some reason it wasn't creating it and my voicemails where getting lost |
| 21:29.04 | Qwell[] | oatis: Some ancient version of *? |
| 21:29.10 | Darwin35 | after you add users to voicemail.conf |
| 21:29.11 | Qwell[] | like 0.6.x? |
| 21:29.27 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.179) |
| 21:29.28 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 21:29.54 | oatis | nope, a new version |
| 21:30.32 | Darwin35 | 1.0.9 |
| 21:30.45 | Darwin35 | you should be on 1.2.4 by now |
| 21:30.50 | Qwell[] | 1.2.5 |
| 21:30.51 | jorgito | i have one question, is it possible to make call parking on my mobile phone , i forward one did to mobile phone, tried to pres # but doesnt work... |
| 21:31.14 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 21:31.31 | _Sam-- | uh...your mobile phone doesnt talk to * |
| 21:31.48 | jorgito | _Sam--, yes so no way ... |
| 21:31.48 | Katty | you can make asterisk talk to a mobile phone. |
| 21:31.55 | jorgito | Katty ??? |
| 21:31.58 | zoa | AJay-MN: yes |
| 21:31.58 | oatis | Qwell, 1.2.1 |
| 21:32.04 | _Sam-- | sure, if it answered the call, THEN called the cell phone and stayed in the middle |
| 21:32.05 | zoa | you need to reboot the phone every 30 minutes or so |
| 21:32.07 | Katty | jorgito: email notifications to sms. |
| 21:32.08 | zoa | then it works |
| 21:32.13 | Katty | jorgito: forwards, extensions |
| 21:32.23 | Katty | jorgito: i'd hardly say asterisk can't talk to a mobile. |
| 21:32.54 | jorgito | Katty well i need to park a call while * calls my mobile, i need to park the call on the mobile... |
| 21:33.09 | _Sam-- | you could do it, if you use the T in the dial command probably |
| 21:33.11 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 21:33.16 | _Sam-- | T when it calls your cell phone |
| 21:33.20 | Dr-Linux | Katty: maybe your mobile doen't have an ear :) |
| 21:33.36 | Katty | Dr-Linux: it's the k]at mobile |
| 21:33.47 | _Sam-- | and include => parkedcalls |
| 21:33.50 | *** join/#asterisk Tamarisk (n=adrian@user-6936.lns6-c10.dsl.pol.co.uk) |
| 21:33.57 | jorgito | Katty, t or T ? |
| 21:34.10 | Katty | jorgito: you should be talking to _Sam-- |
| 21:34.20 | Qwell[] | t |
| 21:34.36 | Dr-Linux | k[at , huh Katty your english is always difficult to me :S |
| 21:34.42 | _Sam-- | qwell is right like usual |
| 21:34.45 | _Sam-- | t: Allow the called user to transfer the call by hitting # |
| 21:34.45 | _Sam-- | T: Allow the calling user to transfer the call by hitting # |
| 21:34.51 | Qwell[] | no, not k[at...k]at |
| 21:35.13 | riksta | has anyone tried running the sample eagi-sphinx-test binary? |
| 21:35.18 | Katty | Dr-Linux: oh forget it. |
| 21:35.18 | *** join/#asterisk darby_t (i=darby_t@dld63.neoplus.adsl.tpnet.pl) |
| 21:35.45 | Katty | Dr-Linux: it's not important enough to explain. you either get it or you don't. savy? |
| 21:35.57 | Dr-Linux | riksta: that doesn't work for me |
| 21:36.10 | riksta | Dr-Linux: im looking for how to run it from the dialplan |
| 21:36.13 | Dr-Linux | savy? |
| 21:36.35 | Dr-Linux | Katty: sorry today i don't have dictionary |
| 21:36.44 | Qwell[] | .dict savvy |
| 21:36.47 | Qwell[] | erm |
| 21:36.50 | Qwell[] | ~dict savvy |
| 21:36.56 | *** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com) |
| 21:37.16 | *** join/#asterisk sener (n=sener@p213.54.181.171.tisdip.tiscali.de) |
| 21:37.24 | Dr-Linux | oooooooooooO:O |
| 21:37.29 | Dr-Linux | Katty: :@ |
| 21:38.09 | riksta | Dr-Linux: how did you try to run it |
| 21:38.59 | Dr-Linux | riksta: even i install all the sphinx package, for voice recognition functionality |
| 21:39.03 | Dr-Linux | but no luck |
| 21:39.45 | jorgito | Katty, well if i hit # on mobile phoen i hear transfer bla bla but when i am pressing 700 then Unable to find extension '' in context 'default |
| 21:40.34 | gammacoder | i'm starting to experiment with sip qualify for remote users behind NAT - anyone have any positive / negative experience you'd care to share |
| 21:40.52 | Katty | jorgito: k |
| 21:41.04 | jorgito | Katty, what ? |
| 21:41.13 | Katty | jorgito: k = ok. |
| 21:41.24 | Dr-Linux | gammacoder: one of my cisco phone user has this problem |
| 21:41.41 | Dr-Linux | his phone was behind NAT |
| 21:42.03 | Dr-Linux | i used qualify option for him, and that works fine now |
| 21:42.16 | jorgito | Katty could you please specify what is ok ? thanks Katty |
| 21:42.19 | riksta | Dr-Linux: what did you have in extensions.conf though |
| 21:42.54 | gammacoder | Dr-Linux: i can get the NAT traversal to work for some NAT firewalls / routers - others it seems like time-out their translation awfully quickly |
| 21:42.55 | Katty | jorgito: default reply. |
| 21:43.02 | Katty | jorgito: you talked to me. i said ok. |
| 21:43.35 | jorgito | Katty k |
| 21:43.38 | Qwell[] | ha |
| 21:44.26 | Katty | Qwell[]: it's amusing when people talk to me for no apparent reason. |
| 21:45.27 | Dr-Linux | riksta: thats problem for me, what should be in dialplan, even my script works fine |
| 21:45.33 | Dr-Linux | so i left it as it is |
| 21:45.44 | Qwell[] | Katty: indeed |
| 21:46.11 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 21:46.26 | Dr-Linux | riksta: someone from here recommended Sphinx, but even i never worked for him/her even |
| 21:46.40 | *** part/#asterisk jmanq (n=jquintus@pool-151-203-200-91.bos.east.verizon.net) |
| 21:46.58 | Dr-Linux | s/i/it |
| 21:49.14 | riksta | anyone know where i can get a free DID for incoming calls for california? |
| 21:49.22 | justinu | katty: i have no apparent reason for it, but hello |
| 21:49.24 | riksta | i just need it for a demonstration |
| 21:50.16 | Tamarisk | Please can someone explain a few pointers on codecs used in * and phones regards compatability? |
| 21:51.00 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-48-113.cybersurf.com) |
| 21:51.02 | Katty | justinu: hihi. |
| 21:51.18 | Tamarisk | I appreciate that G723.1 and G729 need licencing before they work in asterisk so are off as default is that correct? |
| 21:51.41 | justinu | basically |
| 21:52.00 | justinu | what pointers do you need? |
| 21:52.13 | jsharp | char * |
| 21:52.25 | Tamarisk | Just getting confused with reading all the numbers I will continue if you can help |
| 21:52.32 | justinu | i'll try |
| 21:53.01 | Tamarisk | G711 is also call alaw or ulaw depending where you are? |
| 21:53.02 | docelm0 | Hay Katty how do you like my reply to your nicknames bulletin? |
| 21:53.07 | jbalcomb | funny that i keep finding '101' in our dialplan |
| 21:53.33 | justinu | 101 is a freeway |
| 21:53.37 | justinu | or a depech mode album |
| 21:53.43 | jbalcomb | right on |
| 21:53.47 | justinu | or a room where they torture you |
| 21:54.10 | jbalcomb | so no amount of math would lead n+101 to equal 101 in the dialplan though right? |
| 21:54.29 | mattwj2006 | anyone use telasip? |
| 21:54.33 | justinu | i started using ast after that 101 stuff was deprecated |
| 21:54.49 | justinu | so i'm not sure, actually |
| 21:54.51 | jbalcomb | eh? 101 deprecated? |
| 21:55.06 | justinu | that priority jumping stuff got replaced with a DIALSTATUS variable |
| 21:55.09 | justinu | afaik |
| 21:55.25 | *** join/#asterisk thomasp (n=tomtom@p549284A0.dip0.t-ipconnect.de) |
| 21:55.31 | thomasp | hi |
| 21:55.47 | Tamarisk | Is GSM known as anything else such as say G728? |
| 21:55.51 | thomasp | is here anyone who can help me to plug a t sinus pbx to asterisk? tia |
| 21:55.54 | justinu | i do not believe so |
| 21:56.08 | justinu | thomasp: that kind of job costs money |
| 21:56.10 | justinu | fyi |
| 21:56.22 | justinu | i doubt anyone here will help for free |
| 21:56.29 | jbalcomb | justinu interesting. i guess i wont wate my time fixing the 101 entries. |
| 21:56.47 | Tamarisk | OK confused now, I will explain better, I thought I could sort it without being a complete dumbass |
| 21:56.48 | thomasp | justinu: ah. ic. i am a private person. so i have to pay?? |
| 21:56.54 | justinu | jbalcomb: read up on the wiki about the DIALSTATUS stuff... |
| 21:57.01 | justinu | thomasp: pbx integration is tricky work |
| 21:57.02 | jbalcomb | justinu am doing so now |
| 21:57.11 | Qwell[] | thomasp: We're happy to help with most things...but some things are a bit more...involved |
| 21:57.15 | Katty | docelm0: hmm? |
| 21:57.20 | AJay-MN | zoa: I never had an issue with it till i did an update of the firmware, and now i cant go back :( |
| 21:57.21 | justinu | thomasp: private or not, few people have the skill to do such things, and most who do want to be paid for their time |
| 21:57.37 | jbalcomb | thomasp plug a T-1 crossover cable into the PBX and into a port on the Asterisk's PRI card |
| 21:57.44 | justinu | if you wanna ask general questions about what's going on... we'll definitely help for free |
| 21:57.44 | thomasp | justinu: the docs are REALLY misleading. sth |
| 21:58.02 | Tamarisk | From what I read * uses gsm for music on hold, but the Grandstream ATA I use does not list GSM in its Vocoder list |
| 21:58.12 | thomasp | jbalcomb: that is done. debug 99 and verbose 99 doesnt show anything in asterisk when i phone a internal sip phone |
| 21:58.16 | Qwell[] | Tamarisk: Asterisk will transcode to whatever the call is |
| 21:58.19 | jbalcomb | thomasp http://www.voip-info.org/wiki/view/Asterisk+legacy+integration |
| 21:58.23 | Katty | docelm0: heh, sure. |
| 21:58.31 | thomasp | jbalcomb: i read that fucking doc. |
| 21:58.37 | Katty | docelm0: you don't strike me as the warrior type ;) |
| 21:58.48 | thomasp | jbalcomb: if you look for onther one, you will find another howto, another way. |
| 21:58.50 | jbalcomb | thomasp haha.. you sound nice. |
| 21:58.55 | justinu | heh |
| 21:59.21 | justinu | thomasp: you know anything about T1s, or PRI? |
| 21:59.39 | Tamarisk | OK so when I limit the grandstream to onluy use PCMA then the music on hold will be PCMA, |
| 21:59.46 | jbalcomb | thomasp well, I have a Telrad Key-BX hooked up to my Asterisk server and it works great. I figured it out all by myself so I guess you could manage the same. |
| 21:59.48 | thomasp | justinu: a little bit. |
| 21:59.49 | justinu | Tamarisk: correct. |
| 22:00.03 | jbalcomb | I am not special. |
| 22:00.12 | justinu | thomasp: if you're not seeing anything when you call, make sure the D channel and T1 carrier are up and working ok. |
| 22:00.23 | justinu | if indeed your PBX supports PRI\ |
| 22:00.37 | jbalcomb | justinu I'm sure he fucking tried that already. ;) |
| 22:00.39 | docelm0 | Katty, come to astericon and see me up close.. :) |
| 22:00.41 | thomasp | justinu: i get right settings with "ztcfg -vv" |
| 22:00.42 | docelm0 | GRRR!! |
| 22:00.43 | Tamarisk | OK checking on a few settings |
| 22:00.43 | docelm0 | :) |
| 22:00.48 | oatis | where do you define the RTP ports? which conf file is it? |
| 22:00.51 | justinu | ztcfg won't show you D channel status |
| 22:00.53 | Katty | docelm0: i'll be at cluecon. |
| 22:01.00 | justinu | jbalcomb: fuckin' a ;) |
| 22:01.06 | docelm0 | Katty, probably not gonna be able to do that.. |
| 22:01.08 | Darwin35 | cluelesscon |
| 22:01.11 | Katty | docelm0: then i shan't see you. |
| 22:01.14 | oatis | nm, lol rtp.conf :P |
| 22:01.15 | thomasp | justinu: what else? |
| 22:01.23 | jbalcomb | oatis is the the 'bindport' in sip.conf? |
| 22:01.23 | justinu | i'd like to go to cluecon |
| 22:01.25 | justinu | i think i will |
| 22:01.26 | docelm0 | astricon is where the PARTIES at.. |
| 22:01.33 | mog_work | amen |
| 22:01.38 | Katty | we partied pretty hard ourselves at cluecon |
| 22:01.41 | MikeJ[Laptop] | Darwin35, now that's not night |
| 22:01.43 | MikeJ[Laptop] | nice |
| 22:02.32 | jbalcomb | zap show status |
| 22:02.41 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 22:03.04 | Darwin35 | gawd cant I yank a few chains |
| 22:03.11 | justinu | pri show status too, i believe |
| 22:03.17 | Darwin35 | I have been around a long time now |
| 22:03.29 | Tamarisk | Can asterisk transcode between codec's when each end is using a different setting? |
| 22:03.34 | jbalcomb | pri show span <span> |
| 22:03.39 | justinu | ah yes |
| 22:03.46 | justinu | Tamarisk: yep |
| 22:03.52 | Katty | Darwin35: nope. |
| 22:04.02 | justinu | cats don't like chains, btw |
| 22:04.19 | jbalcomb | neither do bunnies |
| 22:04.23 | justinu | they tend to get pretty pissed if you yank on them |
| 22:04.24 | Tamarisk | Very clever, that must some processing power |
| 22:04.34 | justinu | Tamarisk: not so bad, but it depends on the codec. |
| 22:04.43 | justinu | Tamarisk: for example, speex is the one that takes the most CPU right now. |
| 22:04.47 | Darwin35 | not true my cat loves his plastic chain he plays with all the time |
| 22:05.06 | Darwin35 | drags the thing around the house |
| 22:05.08 | thomasp | i do not get it. i just did pri debug span 1,2,4,5 - nothing. the pbx is connected directly with a crossed cable, 36x45 |
| 22:05.13 | *** part/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 22:05.19 | Darwin35 | and will play tug of war with you and it |
| 22:05.25 | Tamarisk | Trying to keep bandwidth down between extensions on a home system,, basically playing to see what quality sounds like but have a |
| 22:05.58 | justinu | Tamarisk: imho, the best sounding low bandwidth codec is g729, but you need to buy a license per channel |
| 22:06.01 | jbalcomb | thomasp I think my $75/hr for contract work is pretty decent, might take two hours tops. |
| 22:06.04 | justinu | although at 10 bucks, it's not so expensive |
| 22:06.19 | justinu | jbalcomb charges less than I do, so his rate is decent. |
| 22:06.41 | Tamarisk | The book says it is good but no not want to purchase for home system. I guess geared towards commercial use? |
| 22:06.43 | Darwin35 | and ideas to see if they will work |
| 22:06.53 | thomasp | jbalcomb: not interested |
| 22:06.59 | Darwin35 | I also have to setup a dial a movie line |
| 22:07.04 | Darwin35 | and dial a yoke |
| 22:07.08 | Tamarisk | I have a funny when connecting between grandstream and Linphone |
| 22:07.11 | Darwin35 | dial a trick |
| 22:07.17 | Darwin35 | dial a tramp |
| 22:07.20 | Tamarisk | I though it may be down to codecs |
| 22:07.23 | jbalcomb | thomasp thats a shame. i wish you the best of luck. |
| 22:07.25 | Darwin35 | dial a homo |
| 22:07.32 | justinu | lol |
| 22:07.52 | Tamarisk | <PROTECTED> |
| 22:08.08 | justinu | Tamarisk: that's a new one |
| 22:08.15 | Tamarisk | then I hear a burst of DTMF then normal audio will follow |
| 22:08.25 | justinu | Tamarisk: make sure VAD is off on both phones |
| 22:08.28 | justinu | ~vad |
| 22:08.29 | jbot | vad is probably Voice Activity Detection or Silence Suppression. Asterisk does not currently support this, so please turn it off on the client |
| 22:08.32 | jbalcomb | Tamarisk sounds like a phone setting |
| 22:08.44 | *** join/#asterisk T`2 (i=id@pdpc/supporter/student/T) |
| 22:09.19 | jbalcomb | Darwin35 i can get you some 800 numbers so to that stuff up. ;) |
| 22:09.25 | Tamarisk | I was assuming differerent codecs................... The phone is a normal house phone plugged into the ATA but I have another to test |
| 22:09.39 | Darwin35 | I can get them free here at work |
| 22:09.40 | jbalcomb | s/so to that/to set that |
| 22:10.03 | justinu | Tamarisk: doesn't sound like a codec problem |
| 22:10.52 | Tamarisk | I will just try an even more basic dtmf phone in the ata to test |
| 22:11.18 | justinu | Tamarisk: have you verified VAD/silence supression is off on both of the phones? |
| 22:11.58 | jorgito | what europe based sip provider do you recommend me ? |
| 22:12.10 | Tamarisk | I will look at that also |
| 22:12.20 | *** join/#asterisk bigjb (n=bigjb@82-36-140-105.cable.ubr02.perr.blueyonder.co.uk) |
| 22:12.47 | bigjb | does anyone have experience with x100p card? |
| 22:13.17 | Deep6 | bigjb, x100p if you count the crap hole clone jobby I bought off ebay then yes |
| 22:13.39 | bigjb | NOTICE[3265]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'ZAP' (cause 0 - Unknown) << what about this error |
| 22:13.59 | justinu | no zaptel drivers loaded? |
| 22:14.00 | Deep6 | I've not got that far |
| 22:14.04 | bigjb | my card is showing as alarm status red from asterisk |
| 22:14.14 | justinu | no phone line plugged in? |
| 22:14.19 | justinu | incorrect signalling type? |
| 22:14.30 | jsharp | Might be your dishwashing detergent. |
| 22:14.30 | Deep6 | bigjb, where'd you get your card? |
| 22:14.35 | bigjb | ahh |
| 22:14.44 | bigjb | i cant remember |
| 22:14.46 | justinu | jsharp: i'm soaking in it |
| 22:14.50 | bigjb | let me see if i can find email |
| 22:15.04 | *** join/#asterisk trym (n=trym@194.63.254.6) |
| 22:15.51 | bigjb | ag |
| 22:15.57 | bigjb | possibly cable problem |
| 22:16.07 | justinu | it could be a bad card, but I used cheapo clone x100p cards that were fine |
| 22:16.12 | justinu | ymmv, i guess |
| 22:16.25 | Tamarisk | Should Silence Suppression: in the Grandstream be set to NO or YES |
| 22:16.27 | Deep6 | Justinu what distro? |
| 22:16.38 | justinu | Tamarisk: NO |
| 22:16.48 | bigjb | Deep6, what country you in? |
| 22:16.49 | justinu | Deep6: i prefer centos, but it works well on about any linux |
| 22:17.02 | Deep6 | I truthfully haven't worked on it very long so I've no idea if my cheapo card works |
| 22:17.09 | Tamarisk | It is presently set for No so that bit is correct trying to find it in Linphone |
| 22:17.09 | Deep6 | bigjb why may I ask? |
| 22:17.21 | bigjb | well the site i bought from appears to be uk only |
| 22:17.26 | Deep6 | Canada |
| 22:17.40 | Deep6 | bigjb, what does lspci tell you about yours? |
| 22:17.41 | justinu | X100p cards are designed for north america, so they may not work well in other places |
| 22:17.49 | bigjb | hang on |
| 22:18.20 | Tamarisk | I hope X100p cards work in the UK as I just bought a couple to try at home in the UK they just arrived |
| 22:18.41 | bigjb | 00:14.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 22:19.04 | Deep6 | 0000:00:0e.0 Communication controller: Motorola: Unknown device 5608 |
| 22:19.06 | Deep6 | is mine |
| 22:19.18 | Deep6 | does anyone know if this Motorola 5608 will work with asterisk? |
| 22:19.35 | bigjb | Tamarisk, i have found at least one site with config for telewest connection, i think that the issue with mine may be that i grabbed a modem cable from work instead of standard phone |
| 22:20.07 | justinu | tigerjet is what I've been using |
| 22:20.56 | Tamarisk | bigjb: I am not that for down the line yet with Asterisk still struggling with basic sip internally before I even try to connect to outside the house |
| 22:21.10 | bigjb | what sip phone you using? |
| 22:22.02 | Tamarisk | I have a grandstream 286 ATA with normal dtmf phones plugged in when it works the audio is fantastic. |
| 22:22.20 | bigjb | eeek |
| 22:22.53 | Tamarisk | I also use softphones such as Kphone and Linphone but not sure what is going wrong with them why eeeek |
| 22:23.08 | justinu | damn, if you think 286 ATA is fancy... |
| 22:23.12 | justinu | ;) |
| 22:23.21 | Tamarisk | £32 from ebay |
| 22:23.56 | *** join/#asterisk phatmonkey (n=phatmonk@65.98.2.81.in-addr.arpa) |
| 22:23.59 | Tamarisk | justinu: sorry for the delay I have gone through Linphone and unable to find any silence setting |
| 22:24.09 | Tamarisk | <PROTECTED> |
| 22:24.13 | justinu | hmm... i'm not sure what else to suggest |
| 22:24.17 | justinu | no, i haven't played with that one yet |
| 22:24.21 | justinu | only eyebeam and xlite |
| 22:24.34 | *** join/#asterisk TedC (n=ted@gray.impulse.net) |
| 22:24.52 | Tamarisk | Budget tone is up from the 286 ata |
| 22:25.04 | pv2b | oh, speaking of the budgetone 100, i may get rid of it in favour of some other phone, after discovering it generates heaps of EMI. |
| 22:25.06 | justinu | heh |
| 22:25.18 | Hmmhesays | haha a customer just told me the rdp session I was on was across a fiber connection... I just about asked him if it was 2 tin cans and a string |
| 22:25.25 | bigjb | shows how much i know |
| 22:25.30 | bigjb | =oP |
| 22:25.45 | justinu | one day you guys will get a chance to use a polycom IP phone |
| 22:25.57 | pv2b | anyone got any advice for any cheap SIP hardphones that don't leak much EMI, by the way? |
| 22:26.00 | justinu | and you'll probably keel over at how good it sounds |
| 22:26.25 | [av]bani | justinu: "bass boost" |
| 22:27.18 | Tamarisk | I started by looking at aastra 9112 as a basic started but then decided that before I got too far down the line to keep it cheep. That may be why I am having issues now |
| 22:29.32 | Tamarisk | Whan setting dtmf transmission is it best to set to use SIP info, or RFC 2833 which is the default in asterisk or is that also intelligent out of the box? |
| 22:29.38 | [av]bani | rfc2833 |
| 22:29.43 | [av]bani | if you can |
| 22:29.48 | justinu | correct |
| 22:30.22 | trixter | the difference largely is that rfc2833 goes in the rtp stream where sip-info goes on the signalling part.. so if you redirect media it causes problems to use sip-info |
| 22:30.32 | trixter | if you never redirect media it probably doesnt matter much |
| 22:30.48 | [av]bani | well you could use inband :P |
| 22:30.52 | [av]bani | but that suxxx |
| 22:31.00 | *** join/#asterisk bartpbx (n=bartpbx@p54B00451.dip0.t-ipconnect.de) |
| 22:31.03 | Tamarisk | OK changing over to 2833 on both the ata and Linphone |
| 22:31.08 | justinu | inband over g729 is da shiznit |
| 22:31.11 | [av]bani | ha |
| 22:31.15 | [av]bani | inband over lpc10 |
| 22:31.20 | bartpbx | hello, i need some help |
| 22:31.47 | bartpbx | today, two of my realtime peers are not registered any more |
| 22:31.54 | justinu | someone is hammering (wardialing) my DIDs in sacramento |
| 22:32.04 | bartpbx | i alywas recive Unable to find key '<userID>' in family 'SIP/Registry' |
| 22:32.25 | bartpbx | anyone has an idear what the reason could be for this? |
| 22:32.40 | bartpbx | all other peers are registering fine |
| 22:33.00 | Deep6 | anyone had any luck with working around this: |
| 22:33.02 | Deep6 | Failed to initailize DAA, giving up... |
| 22:33.03 | phatmonkey | I have heard all sorts about a web interface to access voicemail - what is this? |
| 22:33.42 | justinu | deep6: that sounds bad... like a bad card |
| 22:33.44 | justinu | or something |
| 22:34.07 | Deep6 | justinu, grrrr |
| 22:34.32 | justinu | i don't really know for sure, i'm just saying that I remember seeing something about DAA initializing, and mine didn't say failed |
| 22:34.55 | Tamarisk | The lack of audio until a dtmf key is pressed is still happening |
| 22:34.59 | Tamarisk | Asterisk reports |
| 22:35.43 | Tamarisk | Attempting native bridge of SIP/phone-b3ec and SIP/adrian-54f8 |
| 22:35.50 | justinu | that's normal |
| 22:36.16 | Tamarisk | when I hit a key audio is then patched and no other messages show |
| 22:36.23 | encode | http://find.walla.co.il/?w=/200&q=%E7%E9%E9%EC%E5%FA |
| 22:36.26 | encode | oops |
| 22:36.30 | encode | sorry people |
| 22:37.12 | justinu | encode: unforgivable |
| 22:38.07 | justinu | hehe |
| 22:39.03 | phatmonkey | voicemail web interface anyone? |
| 22:39.17 | phatmonkey | a google search brings up nothing, I swear I heard about it somewhere |
| 22:39.32 | Tamarisk | I have set both phones ATA and Linphone for PCMA codecs so both common |
| 22:39.42 | Tamarisk | <PROTECTED> |
| 22:39.48 | phatmonkey | aaah, found it |
| 22:39.48 | phatmonkey | http://www.voip-info.org/wiki/view/Asterisk+gui+vmail.cgi |
| 22:39.50 | Tamarisk | On hold music is good |
| 22:39.56 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 22:40.10 | Zodiacal | anyone know why the volume that pstn users hear is low... this is only for voicemail greetings and talking. the asterisk operators voice is at a fine volume level... maybe its my mic volume level some where? |
| 22:41.04 | Tamarisk | But I get a warning that flexibel rate not heavily tested, is that normal? |
| 22:41.34 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 22:42.14 | justinu | that has something to with mpg123, i believe... |
| 22:42.20 | justinu | i wouldn't worry about that |
| 22:42.46 | Tamarisk | OK just hoping some message i get may pint to why no initial audio patch |
| 22:43.01 | Tamarisk | may point to why no initial audio |
| 22:43.05 | Hmmhesays | why do sales people insist on selling garbage |
| 22:43.28 | Nivex | Hmmhesays: that's how they get paid. |
| 22:43.42 | justinu | Tamarisk: try turning on 'rtp debug' |
| 22:43.44 | justinu | when you make a call |
| 22:43.45 | Nivex | the more garbage they sel, the more they get paid |
| 22:43.52 | justinu | see if RTP packets begin to flow before or after you press the DTMF. |
| 22:44.01 | Tamarisk | is that in CLI ? |
| 22:44.05 | justinu | correct. |
| 22:44.08 | Hmmhesays | and the more of a pain in the ass it is for me |
| 22:44.41 | Tamarisk | aaaaahhhhhhhhhhhhhhhhhh how do I stop it |
| 22:44.53 | Katty | yawn. |
| 22:45.27 | Zodiacal | any ideas? |
| 22:45.29 | justinu | rtp no debug |
| 22:47.03 | Tamarisk | OK with rtp debug set. I ring ext 100 which is SIP/adrian. it rings, I answer but not rtp traffic |
| 22:47.24 | Tamarisk | when i hit the dtmf key on the grandstream off it goes |
| 22:47.50 | justinu | sip/adrian is linphone? |
| 22:48.07 | Tamarisk | yes it is |
| 22:48.41 | justinu | i think that linphone is at fault here... it probably behaves like asterisk does... |
| 22:48.48 | justinu | asterisk will not send RTP if it doesn't receive any RTP |
| 22:50.13 | justinu | for whatever reason your GS ata doesn't appear to be sending anything until you hit that DTMF key either, which is odd. |
| 22:50.13 | Tamarisk | I must have something wrong in config as it will not let me go the otherway from linphone to ata. Just tried and it will not find the phone |
| 22:50.15 | Tamarisk | But it recognises it when called? |
| 22:51.19 | bartpbx | hello, noone can help me with this " Unable to find key XYXX in family 'SIP/Registry" ? |
| 22:51.23 | *** join/#asterisk Yashy (n=yashy@mail.yashy.com) |
| 22:52.39 | jbalcomb | can I turn off modules or something to get rid of these error messages? http://pastebin.com/589890 |
| 22:53.48 | Tamarisk | I do appreciate all your assistance, would it be better for me to pastebin any config's to see if I have any glaring errors in SIP or extensions |
| 22:56.06 | Deep6 | actually guys switching a pci slot made my cheapo joe x100p work |
| 22:56.16 | bartpbx | @ jabalcomb you could turn of cdr_pgsql |
| 22:56.40 | bartpbx | and pbx_dundi |
| 22:56.50 | bigjb | Deep6, you using old machine? |
| 22:56.57 | encode | are x100p's worth it with echo issues? |
| 22:57.01 | *** join/#asterisk fjean (n=fjean@201009183198.user.veloxzone.com.br) |
| 22:57.24 | *** join/#asterisk mattwj2005 (n=Matt@dialup-4.159.47.59.Dial1.Chicago1.Level3.net) |
| 22:57.49 | fjean | hi all ! anyone understands the relationship between DIALSTATUS and QUALIFY=yes ? :) |
| 22:58.18 | mattwj2005 | iax.cc any good? |
| 22:58.21 | X-Rob_ | DIALSTATUS is set by Dial. qualify=yes is set in sip.conf. |
| 22:58.26 | X-Rob_ | Thats' the relationship (eg, none at all) |
| 22:59.03 | fjean | hi x-rob, hehe, well in the wiki there is a comment saying the contrary: http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+DIALSTATUS |
| 22:59.25 | fjean | that is why i am actually asking |
| 22:59.32 | Tamarisk | I have pasted! www.pastebin.com/589900 |
| 23:00.08 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 23:00.08 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:00.08 | Deep6 | bigjb, marginally p3500 |
| 23:00.12 | Deep6 | er p3 500 rather |
| 23:00.34 | bigjb | ahh |
| 23:00.36 | Tamarisk | The most part is the default sip.conf where I have added three entries [adrian], [carrie] and [phone] |
| 23:01.06 | Tamarisk | the last section is a short extensions conf where there is extensions 100, 101 and 102 |
| 23:01.19 | Tamarisk | <PROTECTED> |
| 23:01.23 | Tamarisk | also |
| 23:01.33 | bigjb | anyone know why festival seems to produce speech thats almost to quick to comprehend? |
| 23:01.48 | Deep6 | bigjb, the funniest thing is, I have no idea where to go after I have the zaptel.conf setup :).... all my efforts have been focused on ztcfg -vvv that now I have no idea what to do!:) |
| 23:02.06 | bigjb | hang on a mo |
| 23:02.32 | bigjb | http://users.pandora.be/Asterisk-PBX/InstallWildcard.htm |
| 23:02.34 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 23:03.06 | Mavvie | Mar 8 10:02:37 DEBUG[27850]: chan_iax2.c:6426 socket_read: For call=2, set last=98664 |
| 23:03.15 | Mavvie | brilliant flood when you have debug set to one. |
| 23:04.48 | *** part/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 23:07.39 | Yashy | http://www.devrandom.org/p/39 Trying an X100P card.. xttool shows the X100P as "OK", but when I call my home number, Asterisk doesn't appear to be doing anything with -cvvvvvvvvg |
| 23:07.57 | Darwin35 | TELIAX come over to TELIAX help us help you |
| 23:08.54 | *** join/#asterisk deam (i=fork@83.98.246.59) |
| 23:09.03 | deam | hi |
| 23:09.13 | deam | I have a small question before I even start looking at asterisk |
| 23:09.26 | fjean | x-rub : i searched the net but could not find something usefull... |
| 23:10.01 | deam | I have a config running atm.. it's a ADSL router which has POTS ports.. it supports SIP/ACS would it be possible to use asterisk as SIP gateway for this device ? |
| 23:10.08 | deam | and then use asterisk functionality |
| 23:11.22 | bigjb | is it a draytek? |
| 23:11.46 | deam | nope |
| 23:11.49 | deam | allieddata copperjet |
| 23:11.55 | deam | 1616-2P |
| 23:13.02 | Gamercjm | still trying to get help with the .call files for automatic outgoing calls |
| 23:14.15 | mattwj2005 | anyone in here use iax.cc? |
| 23:15.20 | simulated | mmm love them channel pings |
| 23:15.21 | bigjb | deam, i dont see why not |
| 23:15.35 | simulated | wow, 1.2.5 has been released... im so absorbed into svn i didnt even pay attention |
| 23:16.46 | mattwj2005 | I am looking for reliable service.....iax.cc seems real professional....anyone know where I can find good reviews of service providers? |
| 23:16.58 | deam | so bigjb: I basically can use asterisk as SIP gateway ? |
| 23:17.08 | deam | then let it connect to another external SIP gateway > |
| 23:17.17 | deam | I have voipbuster configured atm |
| 23:17.22 | deam | directly on my router |
| 23:17.30 | bigjb | well the looking at the manual the pot ports simply connect to a standard sip connection |
| 23:17.45 | Deep6 | after configuring zapata.conf shouldn't zap show channels show a configured channel? |
| 23:18.20 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 23:18.20 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:18.29 | simulated | deam, if your modem allows you to access the VoIP capabilities without manufacturer limitations, then you should have no problem using Asterisk as your SIP provider |
| 23:19.23 | simulated | in a worst case scenario, you can setup an an account on your asterisk that your router will use to authorize |
| 23:19.38 | *** join/#asterisk _deg_ (n=deg@201.22.12.20.adsl.gvt.net.br) |
| 23:20.40 | Tamarisk | I think deam has left the server |
| 23:20.49 | simulated | woops hehe |
| 23:21.04 | simulated | i scrolled up to read his question :\ |
| 23:21.36 | simulated | Tamarisk you have any experience w/ h323 on * ? |
| 23:22.12 | Tamarisk | No sorry I am definatly the wrong guy to ask I ask for much more help then many |
| 23:22.27 | simulated | hehe |
| 23:22.44 | simulated | I need to use a custom codec that all the asterisk implementations of h323 don't support |
| 23:22.57 | bartpbx | this is realy anoying.. anyone using realtime around here? |
| 23:23.30 | *** join/#asterisk Leland (n=leland@ws2.discpro.org) |
| 23:23.36 | Leland | evening all |
| 23:23.49 | Leland | quick question if I may ? |
| 23:24.09 | simulated | bartpbx shoot |
| 23:24.12 | simulated | bartpbx love that shit |
| 23:24.30 | *** join/#asterisk WAudette (n=WAudette@67.170.156.3) |
| 23:25.07 | SplasPood | Anyone know why asterisk would be generating caller id by taking the name of the stanza from sip.conf, removing all non-numeric characters, and using that? |
| 23:25.17 | Leland | have a problem with MoH on a G.729 channel... works fine over G.711 and GSM codecs, but if endpoint is g.729 the MoH is just a bunch of digital noise... anyone have any tips ? |
| 23:25.33 | bartpbx | simulated, normaly i like relatime.. but today it is rally anoying |
| 23:25.50 | bartpbx | but im not sure if it is a realtime problem |
| 23:25.58 | simulated | whats the message log showing |
| 23:26.07 | bartpbx | db.c: Unable to find key '108532' in family 'SIP/Registry' |
| 23:26.17 | simulated | is it in the database? |
| 23:26.25 | bartpbx | 108532 is the name of the peerure |
| 23:26.29 | bartpbx | sorry |
| 23:26.29 | simulated | you've tried restarting? |
| 23:26.29 | bartpbx | sure |
| 23:26.43 | simulated | no conflicts? |
| 23:26.51 | simulated | it's only happening for that one peer? |
| 23:27.04 | bartpbx | it happens for 4 peers of 150 |
| 23:27.08 | russellb | bartpbx: is that a DEBUG message? |
| 23:27.11 | *** join/#asterisk SGM (n=stoyan@home.marinov.us) |
| 23:27.25 | SGM | hi |
| 23:27.46 | bartpbx | russellb, yes, it is a debug message. |
| 23:28.01 | SGM | I have a question about SIP reinvites |
| 23:28.04 | bartpbx | and the peer is not registering any more |
| 23:28.22 | russellb | so this is expected and you can safely ignore it |
| 23:28.30 | bartpbx | this is the only message which is different on only these 4 peers |
| 23:28.52 | SGM | I read on voip-info.org that reinvites won't work if I'm using t, T, h, H, w, W or L |
| 23:29.22 | justinu | SGM: right, because those options require asterisk to stay in the media stream. |
| 23:29.31 | SplasPood | god, I've never seen this CID behavior |
| 23:29.34 | justinu | (to watch for the DTMF) |
| 23:29.35 | SplasPood | its freaking odd.. |
| 23:29.46 | bartpbx | is there anyway to get realtime to log the sql queries? |
| 23:29.48 | SGM | justinu: yes, but if I'm using sip info for dtmf |
| 23:29.50 | SGM | ? |
| 23:30.04 | justinu | SGM: theoretically, you're ok... however, in practice ??? |
| 23:30.09 | *** join/#asterisk ReD-MaN (i=daemon@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 23:30.52 | *** part/#asterisk ReD-MaN (i=daemon@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 23:31.07 | SGM | in practice.... haven't tried yet |
| 23:31.12 | *** join/#asterisk ReD-MaN (i=daemon@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 23:31.14 | justinu | good question tho |
| 23:31.19 | SGM | any idea where to touch? |
| 23:31.21 | SGM | :) |
| 23:31.50 | justinu | either read the code in chan_sip, or try it |
| 23:31.54 | justinu | the latter is probably easier ;) |
| 23:32.10 | litage | without having a FWD account, can Asterisk redirect calls to a FWD user? |
| 23:32.11 | SGM | I did tried it |
| 23:32.15 | SGM | it didn't work |
| 23:32.20 | SGM | that's why I'm asking |
| 23:32.20 | SGM | ;) |
| 23:33.04 | justinu | (3:34:31 PM) SGM: in practice.... haven't tried yet |
| 23:33.07 | justinu | ? |
| 23:33.30 | justinu | SGM: you'll probably have to get your coding hat on and patch chan_sip.c |
| 23:34.06 | DaPrivateer | hrm |
| 23:34.12 | bartpbx | what is the best way to debug an register try? |
| 23:34.21 | Zodiacal | if i change zapata.conf do i have to recompile? |
| 23:34.22 | SGM | justinu: I meant I haven't tried to modify the code |
| 23:34.26 | DaPrivateer | no one here would happen to have updated firmware for the Polycom Soundpoint IP's would they? :-p |
| 23:34.28 | SGM | to see if it's gonna work |
| 23:34.32 | justinu | ic |
| 23:34.48 | bartpbx | i see serveray SIP/2.0 401 Unauthorized sip messages and would like to now what the actual problem was |
| 23:34.53 | justinu | Zodiacal: no, you need to stop asterisk and restart it in many situations tho. |
| 23:35.01 | bartpbx | e.g. wrong password, invalid user, .. |
| 23:35.02 | Zodiacal | justinu okie thanks! |
| 23:35.15 | SplasPood | <PROTECTED> |
| 23:35.18 | SplasPood | thats freaking odd |
| 23:35.26 | SGM | reinvites should only pass the rtp directly, right? |
| 23:35.28 | SplasPood | why is it setting CALLERID(num) to the name of the stanza from sip.conf ? |
| 23:35.51 | justinu | SGM: reinvites will allow you to stop having * proxy the audio (aka, hairpinning) |
| 23:36.08 | justinu | but it still stays on the signalling plane, so you should continue to get your SIP info packets from the UAs |
| 23:36.35 | SGM | the thing is that I want to keep the asterisk server in the datacenter |
| 23:36.41 | SGM | while the phones are in the office |
| 23:37.00 | SGM | and I don't think it's a good idea to pass all the audio to the server and back |
| 23:37.06 | *** join/#asterisk UndiFineD (n=me@a82-93-111-205.adsl.xs4all.nl) |
| 23:37.08 | SGM | when the phones are in the same lan |
| 23:37.12 | justinu | you're on the right track, but i think you've run into a limit of asterisk's sip channel |
| 23:37.37 | SGM | ok, let's see what chan_sip.c looks like ;) |
| 23:38.12 | *** part/#asterisk fjean (n=fjean@201009183198.user.veloxzone.com.br) |
| 23:38.13 | justinu | you won't like it much ;) |
| 23:38.13 | justinu | kinda frustrating |
| 23:38.29 | Leland | *sniff* :( |
| 23:38.59 | *** join/#asterisk ManxPower (i=ewieling@192.sub-70-210-254.myvzw.com) |
| 23:39.17 | ManxPower | ~docs |
| 23:39.19 | jbot | rumour has it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 23:39.22 | ManxPower | ~mailinglist |
| 23:39.23 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 23:39.55 | SGM | ahhh |
| 23:39.55 | SGM | I think I found it! |
| 23:40.10 | SGM | I was playing with asterisk and T.38 |
| 23:40.11 | Leland | I'm sure I must be overlooking something obvious ... MoH over 711u/a or GSM works fine, but is totally corrupted over 729 ... I would have thought that it should be transcoded |
| 23:40.37 | SGM | and if T.38 support is enabled |
| 23:40.49 | SGM | it won't bridge the channels |
| 23:41.14 | justinu | Leland: yeah, that shouldn't be an issue. |
| 23:41.29 | Leland | justinu: hmm... seems to be an issue though |
| 23:41.40 | Leland | can't work out why though... source of the MoH is a normal mp3 |
| 23:41.42 | justinu | i play moh over g729 all the time. |
| 23:43.02 | Leland | did you modify the source files in any way, coding or bit rate ? |
| 23:43.11 | justinu | the source mp3s? no |
| 23:43.15 | Leland | hmm |
| 23:43.22 | justinu | however, i'm running asterisk 1.2.0 |
| 23:43.24 | justinu | not the latest stuff |
| 23:43.31 | justinu | no idea if that means anything or not |
| 23:43.57 | SplasPood | god this is the oddest asterisk issue i've ever had |
| 23:44.17 | Leland | I'm not running the latest stuff either.. running a commercial port of * (PBXWare) .. the underlying system is still * but all the call handling is done through AGI scripting and databases |
| 23:44.45 | *** join/#asterisk xtr (i=94752345@S0106000c41ed11e1.vf.shawcable.net) |
| 23:45.24 | *** join/#asterisk bigjb (n=bigjb@82-36-140-105.cable.ubr02.perr.blueyonder.co.uk) |
| 23:45.34 | bartpbx | can anyone outthere tell my why this register failes? http://pastebin.com/589971 |
| 23:46.03 | bartpbx | the peer XXXXXX is existing and yes, the password is correct |
| 23:47.17 | *** join/#asterisk ambriento (n=ambrient@www.cobranet.com.br) |
| 23:47.49 | justinu | leland: suspect your g729 encoder, perhaps |
| 23:48.07 | Leland | justinu: it's the digium licensed codec |
| 23:48.37 | mattwj2005 | anyone have any problems with iax.cc? |
| 23:48.49 | justinu | leland: one thing you could try doing, is converting your MoH into ulaw PCM files, then setting type=native in moh.conf |
| 23:48.57 | justinu | see maybe if that transcode works. |
| 23:49.15 | Leland | hmm.. |
| 23:50.09 | ManxPower | Every time I travel I hate it a little bit more. |
| 23:50.14 | Darwin35 | chat at home.... |
| 23:50.45 | ManxPower | I go to the New Orleans area about once a month for work. |
| 23:51.19 | justinu | i was just out in NYC... had a good time |
| 23:51.37 | SplasPood | NYC represent! |
| 23:51.44 | ManxPower | I've not been to the actual city of New Orleans since Katrian |
| 23:51.59 | Leland | justinu: off the top of your head you know what sox syntax could convert a .wav to raw ? |
| 23:52.04 | *** part/#asterisk Tamarisk (n=adrian@user-6936.lns6-c10.dsl.pol.co.uk) |
| 23:52.08 | SplasPood | I'll love anyone that can explain asterisk's behavior when no CID is set in sip.conf to me.. |
| 23:52.11 | ManxPower | I suppose I should during my next trip. |
| 23:52.19 | ManxPower | Leland, .wav is almost raw. |
| 23:52.35 | justinu | leland: i can't remember... it's pretty clear if you read the man page |
| 23:52.40 | justinu | -t raw or something |
| 23:52.52 | ManxPower | SplasPood, incoming? outgoing? device? service provider? |
| 23:52.57 | *** join/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
| 23:53.26 | SplasPood | Manx: basically I'm not setting callerid= in sip.conf for a particular user.. When calling out it appears to be setting the callerid from the [context] in sip.conf ?!? |
| 23:53.35 | *** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com) |
| 23:53.41 | SplasPood | ie if its [sip-user-here] CID ends up as sipuserhere |
| 23:53.50 | ManxPower | SplasPood, Ah. Weird. |
| 23:53.59 | SplasPood | I can duplicate this every time |
| 23:54.02 | SplasPood | with different sip clients |
| 23:54.10 | SplasPood | it's totally an asterisk issue... or erm.. feature |
| 23:54.17 | ManxPower | SplasPood, One would expect the CLID to be "asterisk" like in chan_zap. |
| 23:54.33 | SplasPood | yes, or I'd expect it to take the client's CID |
| 23:55.09 | ManxPower | SplasPood, it should take the info the client provides, assuming it's in a format asterisk sets, and that the client sends it. what verison of Asterisk? |
| 23:55.27 | SplasPood | 1.2.4 |
| 23:55.58 | bartpbx | hm. im still stuck with these 4 peers |
| 23:56.10 | ambriento | splaspood, which client are u using? |
| 23:56.24 | SplasPood | ambriento: I've tried two.. x-lite and a cisco ata 186 |
| 23:56.24 | bartpbx | I need to get them woking. anyone has an idear what i could look at |
| 23:56.33 | ManxPower | I don't allow my users to set their callerid. |
| 23:56.46 | SplasPood | Manx: nor do I, usually... |
| 23:56.52 | ManxPower | bartpbx, without more info about the problem..... |
| 23:56.57 | ambriento | manxpower, how do you block that? |
| 23:56.59 | SplasPood | this whole taking of the [context] and making it the CID sounds like a "feature" |
| 23:57.22 | ManxPower | ambriento, you put callerid=Robert Dobbs <6667> in the sip.conf [section-for-the-device] |
| 23:57.29 | *** join/#asterisk kratzers (n=kratzers@pool-151-205-208-110.cap.east.verizon.net) |
| 23:57.32 | bartpbx | I think the problem seams to be related to the router |
| 23:57.49 | bartpbx | we have an office with 4 peers behind a router |
| 23:58.02 | *** join/#asterisk niteowloz (n=niteowlo@203.185.195.84) |
| 23:58.06 | bartpbx | this afternoon the phones stopped working |
| 23:58.07 | SplasPood | Manx: any way I can force it to send 'UNAVAILABLE' as the CID? |
| 23:58.07 | ambriento | manxpower, ow, I see. :) |
| 23:58.29 | ManxPower | bartpbx, make sure you have qualify=yes in each sip.conf peer/friend/user |
| 23:58.38 | ManxPower | SplasPood, callerid=UNAVAILABLE |
| 23:58.49 | ManxPower | don't know what the number would come across as. |
| 23:58.51 | kratzers | anybody know if queued calls should land at phones on DND? |
| 23:58.58 | bartpbx | in the log i see them rejected with 401 (in sip debug) and the Unable to find key message |
| 23:59.15 | bartpbx | I've just recieved a message that they installed a new router on site |
| 23:59.17 | ManxPower | kratzers, that would depend on how the DND is done, and how the call is dialed. |
| 23:59.24 | bartpbx | I think i have to focus on that |
| 23:59.27 | ambriento | splaspood, if you don't set that callerid thing inside the [sip-user] x-lite will set it from its setup |
| 23:59.35 | Leland | hmm... |
| 23:59.39 | bartpbx | but how can a new router cause register rejects? |
| 23:59.46 | riddlebox | can you add a line in extensions.conf in your outgoing context that if someone dials 10 digits to add the one in front? |
