00:00.46 | *** join/#asterisk LetsTalkPBX (n=LetsTalk@pool-71-182-164-82.pitbpa.east.verizon.net) |
00:04.59 | *** join/#asterisk khronos (n=khronos@c-76-110-134-230.hsd1.fl.comcast.net) |
00:12.19 | danicholson | Any-one have the *.conf files that they are using with a Cisco AS5400 that they are willing to share? |
00:18.31 | khronos | Anybody ever connected an Asterisk system to an Altigen? |
00:18.31 | aydiosmio | khronos: clever. |
00:18.31 | aydiosmio | let me know how that works out |
00:18.55 | khronos | What I want to do is link my exten on the work pbx to my phone at home since I work from home a lot. |
00:19.43 | khronos | Right now I have ti breaking out a line and call my home phone, but I'd like to take that out of the loop. |
00:21.19 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
00:33.15 | khronos | Ah, looking around and talking to my boss I see I've got to use h323 to talk to the Altigen. |
00:33.46 | khronos | Now just have to figure out how to format the h323 config file. |
00:34.14 | khronos | A question I have though, should I use the h323 driver that comes with Asterisk or openh323. |
00:36.52 | bkruse_home | ewwwwww |
00:36.55 | bkruse_home | file: <3 |
00:42.49 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
00:42.57 | *** join/#asterisk ctaloi (n=ctaloi@pool-72-90-82-84.syrcny.fios.verizon.net) |
00:43.21 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
00:46.18 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
00:47.36 | paavum | can I integrate any fax 2 mail solution (spanDSP/HylaFax/Asterfax) with Asterisknow B4? |
00:47.41 | paavum | b4 = beta 4 |
00:51.00 | jovannotti | something here knows what version I should use for pwlib and openh323 in fedora core 6 ? |
00:52.51 | ManxPower | jovannotti: only the version listed in the readme or install file will work |
00:56.12 | jovannotti | then I need to download and install Open H.323 version v1.17.1, PWLib v1.9.0 ? |
00:56.25 | jovannotti | I tried to install these, but it looks so older to fedora core 6 |
00:56.32 | jovannotti | and I have problems compiling it |
00:56.57 | *** join/#asterisk hmm-home (n=hmm-home@24-117-131-41.cpe.cableone.net) |
00:57.08 | hmm-home | I forgot how colorful gaim is |
01:04.22 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
01:05.52 | *** join/#asterisk JT_ (n=jon@unaffiliated/jt) |
01:11.46 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
01:16.39 | *** join/#asterisk jcaceres (n=josexato@190.40.70.66) |
01:16.54 | jcaceres | hello does anybody used RAGI? |
01:18.56 | danicholson | Hello, any-one here with a Cisco AS5400 connected to *1.4? |
01:23.37 | ManxPower | jcaceres: RAGI lets your app LISTEN for audio, it does not allow your app to SEND audio. |
01:31.33 | ManxPower | jcaceres: Sorry, I was thinking of EAGI. What I said does not apply to RAGI |
01:32.32 | jcaceres | oka ManxPower, |
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01:44.16 | danicholson | Hello, any-one here with a Cisco AS5400 connected to *1.4? |
01:47.34 | blitzrage | danicholson: there wasn't 30 mins ago, and there still isn't |
01:53.21 | *** join/#asterisk bkw_ (i=brian@ppp-70-128-123-137.dsl.tulsok.swbell.net) |
01:57.34 | *** join/#asterisk digiterata (n=digitera@bas1-montreal02-1096716595.dsl.bell.ca) |
01:59.03 | digiterata | hey folks |
01:59.29 | digiterata | very frustrated, looking for some advice. |
02:00.11 | jovannotti | idem |
02:00.29 | digiterata | I'm not a linux pro but I'm trying to get Asterisk 1.4 up and running on a virtual machine. |
02:01.02 | *** join/#asterisk remmo (n=junk@smack.isp.net.au) |
02:01.47 | khronos | I've actually thought about running Asterisk in a vm, but I think the hardware modules for interface cards you have will need to be compiled in the the host machine's kernel as well as the ztdummy module if you don't have any cards. |
02:02.21 | digiterata | I've played with AsteriskNow but I'm looking to get a clean installation up and running and I'm not sure where to start. Trying to find a good simple distro that works nicely with Asterisk |
02:02.39 | *** join/#asterisk bkw_ (i=brian@adsl-70-142-43-193.dsl.tul2ok.sbcglobal.net) |
02:03.12 | digiterata | I've played around with AsteriskNow and it's pretty nice actually. just that it's a bit locked down and I'd like to be able to talk to Asterisk through the AGI |
02:03.45 | digiterata | I think for anything real, I'd still rather have asterisk on bare metal, but at the moment I'm purely in development. |
02:04.11 | digiterata | what distro do you use to base your Asterisk machines on? |
02:04.43 | *** join/#asterisk phix (i=threat@60-240-43-214.static.tpgi.com.au) |
02:04.56 | Nugget | I run asterisk in Slackware, FreeBSD, Mac OS X. |
02:05.13 | phix | <PROTECTED> |
02:05.32 | digiterata | I've heard good things about Slackware. Is it a good distro to learn on? |
02:05.51 | phix | Ihave two asterisk servers, A and B, I am trying to call a user on A from B |
02:06.06 | phix | the user on A has an extension of 3, and is using SIP |
02:06.11 | digiterata | Slackware, it's known for stability, yes? |
02:06.19 | Nugget | Linux is Linux. |
02:06.22 | phix | hello |
02:07.10 | digiterata | ok, so it doesn't really matter. except it does. they all seem to use different package managers. |
02:07.14 | Nugget | Slackware is not a particularly good distro to learn on, though. |
02:07.18 | Nugget | I use it because I hate it the least. |
02:07.27 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
02:07.27 | *** mode/#asterisk [+o mog] by ChanServ |
02:07.30 | Nugget | It's the least Linuxy Linux. |
02:07.44 | phix | can any one give me a hand? |
02:07.57 | bluelinq | is there a way to increase the volume using some configuration ina 5960 on sip? |
02:08.06 | digiterata | I have a VMware image running Ubuntu 6.10 server. |
02:08.50 | digiterata | It seems pretty good except for the fact that I can't seem to wget asterisk from digium. |
02:08.58 | phix | grrrrrrrrrrrr |
02:09.22 | bluelinq | weird part is that under the sccp firmware the volume is a lot louder... |
02:09.28 | digiterata | hey phix, I'm sorry. wish I knew. |
02:10.24 | phix | digiterata: ok |
02:10.30 | phix | digiterata: you know anyting about iax? |
02:11.57 | bluelinq | sip volume anyone? |
02:16.56 | bluelinq | phix: here you go...http://www.pastebin.ca/453539 |
02:19.30 | digiterata | yeah I know about iax (sorry was on the phone) |
02:19.44 | digiterata | enough to be dangerous anyway |
02:22.12 | phix | grrrr |
02:22.16 | phix | why am I getting rejected for? |
02:23.58 | *** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net) |
02:25.24 | nDuff | Anyone know how to modify the dialplan on a GXP-2000? |
02:26.57 | nDuff | I'm looking at the web interface, and I don't see anything at all about the dialplan (except for a field for a fixed prefix to be added to outgoing calls) |
02:26.58 | phix | ok so why am I getting rejected for? |
02:27.13 | phix | ...... |
02:28.15 | nDuff | phix: That's typically an authentication-related message, which implies some obvious initial places to look. |
02:28.34 | nDuff | oh, wait, you posted more above. |
02:30.50 | nDuff | phix: How is the host defined in your iax.conf? A user, a peer, or a friend? |
02:32.01 | *** join/#asterisk Fieldy (i=6nD2Oy7n@gentoo/contributor/Fieldy) |
02:32.35 | phix | friend |
02:32.45 | phix | nDuff: on some servers A and B |
02:33.01 | phix | as I would like either end to make or receive calls |
02:33.12 | phix | some = both |
02:33.18 | phix | typo :P |
02:34.09 | nDuff | ...and your secrets match, and your host entries match the actual IP the remote traffic is coming from, and the IAX module has been reloaded since you did all this? |
02:34.14 | phix | yes |
02:34.37 | phix | I have /etc/init.d/asterisk restart on both servers |
02:34.53 | phix | (both are debian system running asterisks 1.2) |
02:34.54 | nDuff | Huh. Dunno. Last time I did that, it Just Worked. |
02:35.00 | phix | hmmm |
02:35.38 | phix | nDuff: I want to dial server A extension 3 from server B |
02:36.24 | phix | I have added in exten => 501,1,Dial(IAX2/name/3) on server B |
02:36.57 | phix | do I need to add anything on Server A to give server B permission to ring that extension? |
02:37.49 | phix | do I need a hostfrom directive? |
02:38.17 | nDuff | IIRC, I ended up specifying everything (hostname, password, etc) in my outgoing Dial path. (Okay, it didn't Just Work -- there was a little trial/error/adjustment on the outgoing side) |
02:38.37 | phix | do I need to put it in an outgoing dial plan? |
02:38.49 | nDuff | not to say it should be necessary to do that -- I was just in a hurry, and that's what worked for me. |
02:38.59 | phix | I have setup a SIP user which is in context sip, the extension I want to call is in context sip too |
02:39.03 | phix | is that all I need to do ? |
02:39.09 | phix | hmm |
02:39.49 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
02:40.26 | digiterata | hey guys, I've been trying all weekend to get up and running on Asterisk 1.4 My latest attempt is from a VM running Astlinux. Could anyone tell me if it's possible to upgrade the Asterisk 1.2 on this distro to 1.4? |
02:40.57 | nDuff | phix, the whole thing is more like IAX2/username:password@host:port/number?context |
02:41.14 | phix | hmmm |
02:41.27 | phix | do I need to specify username and password? |
02:41.33 | nDuff | digiterata: Of course it's possible, as long as you've got a compiler; it's just a matter of how *hard* it is. |
02:41.35 | phix | I have guest enabled on both servers |
02:42.03 | nDuff | phix: I had to. Shouldn't be necessary, then or now, but I was in a hurry and it worked for me. |
02:42.41 | phix | ok |
02:42.58 | nDuff | phix: If that doesn't work, I'm not sure what to tell you -- I'd probably turn on debug logs or (if they're inadequate) start instrumenting the relevant source to figure out what's going on, but that's just me. |
02:43.05 | digiterata | right, about that *hard* part. |
02:43.47 | nDuff | digiterata: I don't know the first thing about Astlinux, so I can't tell you. |
02:43.58 | digiterata | i'm a complete linux n00b but I'm good with google - haven't found the answer yet though |
02:44.50 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9ui.cable.mindspring.com) |
02:47.31 | VJFROMGT | knock knock |
02:56.05 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
02:58.12 | *** join/#asterisk jovannotti (n=jovannot@190.84.99.36) |
02:58.26 | jovannotti | something has tested TC400B card from digium > |
02:58.27 | jovannotti | ? |
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03:16.05 | piper69 | is LINKSYS PAP2 locked to vonage? |
03:16.26 | jql | typically |
03:16.33 | jql | PAP2-NA is the unlocked variety |
03:18.46 | piper69 | where can i get that , all what is in the market is for vonage |
03:19.07 | piper69 | aren't vonage suppouse to go out of busniess |
03:20.37 | khronos | Possibly, but we don't know yet. |
03:21.09 | khronos | From what it looks like we'll have to wait til the courts decide the pattent issues that are being faught over. |
03:21.46 | mog | vonage wont go under |
03:21.54 | mog | verizon might buy em for their customers |
03:21.59 | mog | after they crush them of course |
03:22.11 | mog | or someone who has indemnity from verizon on said patents |
03:22.18 | jql | vonage has liabilities, though. bad purchasing decision |
03:22.57 | jql | all that cash in the bank still doesn't make them worth it... |
03:22.58 | piper69 | http://shop4.outpost.com/%7BVq1PeD5uecMRKyYm34Ehcw**.node2%7D/product/4175693;jsessionid=Vq1PeD5uecMRKyYm34Ehcw**.node2?site=sr:SEARCH:MAIN_RSLT_PG |
03:23.04 | file | you silly goose |
03:23.09 | mog | yeah i know |
03:23.13 | mog | i am retarded |
03:23.52 | danicholson | http://www.voipnow.org/vonage/index.html |
03:25.00 | khronos | In the next couple months I will be setting up a voip over satellite system for a couple of locations. |
03:25.14 | khronos | What are my options for doing fax over ip? |
03:25.31 | Corydon76-home | LOL... oh, wait, were you serious? |
03:25.54 | khronos | I was thinking of doing some sort of trunking transcoding server at the satellite company that will take the calls and transcode them to gsm off the pri. |
03:26.43 | khronos | If I set the asterisk servers at the remote locations to use gsm how would I be able to have the clients pass faxes over their atas at the different locations and still have them work? |
03:27.04 | nDuff | khronos: T.37 and T.38. |
03:27.10 | CunningPike | khronos: What he said |
03:27.14 | file | you were thinking of sending faxes over that? |
03:27.19 | VJFROMGT | how can i tell what codec is been used on a particular iax2 call? |
03:27.19 | CunningPike | I can't type fast enough, obviously |
03:28.06 | khronos | Would I have to set certain extensions up as fax lines or will I be able to have the atas at the homes pick which protocol to use? |
03:28.12 | CunningPike | khronos: T.38 is your only chance, and even then, you're getting involved in a whole heap of hurt, my friend |
03:28.48 | nDuff | khronos: See http://www.soft-switch.org/foip.html |
03:28.52 | *** join/#asterisk mekong (n=josh@cpe-69-203-218-147.nyc.res.rr.com) |
03:35.27 | mekong | s |
03:37.32 | nDuff | khronos: given you enough to think about? |
03:38.36 | *** join/#asterisk darkladywolf (n=root@wolf.tpgi.com.au) |
03:38.54 | nDuff | khronos: If I were in your shoes, I'd rig up something T.37-style (maybe even real T.37), using iaxmodem or something like it at each POTS connection and then sending the content as email between them. |
03:40.55 | *** join/#asterisk thoughtpolice (n=austin@c75-111-137-154.plaicmtc01.tx.dh.suddenlink.net) |
03:41.04 | nDuff | khronos: that said, I'm very happy to not be in your shoes right now. |
03:42.40 | darkladywolf | Hi folks. Anyone here a DUNDi expert? Or at least willing to help out someoene who knows they've got it wrong, but can't work out where? |
03:44.19 | khronos | Problem is I'm basically being a service provider to homes. |
03:44.26 | khronos | on islands. |
03:44.39 | khronos | The only way they have to get to the world is over satellite. |
03:48.04 | khronos | Basically here's the situation. |
03:48.39 | khronos | We've got a bunch or rich people who come from the us and they have homes on this island and they want to have the same services there as they do when their in the US. |
03:49.13 | CunningPike | ~wglwat |
03:49.18 | jbot | well, wglwat is well, good luck with all that |
03:49.22 | CunningPike | :) |
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03:53.57 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
03:54.59 | jql | it's good to be rich |
03:57.15 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
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04:03.07 | *** join/#asterisk JerJer[mobile] (n=jj@199.45.11.90) |
04:05.10 | JerJer[mobile] | anyone seen a compile error in menuselect/mxml ? |
04:05.34 | JerJer[mobile] | make[3]: Entering directory `/usr/src/asterisk/menuselect/mxml' |
04:05.34 | JerJer[mobile] | autoconf |
04:05.35 | JerJer[mobile] | autoconf: no input file |
04:07.04 | mosty | how can i get asterisk to transcode my prompts to g729 and save the files? |
04:07.40 | nDuff | khronos: would they be actively unhappy about getting and sending their faxes via computer? |
04:07.53 | nDuff | khronos: because if you can give them a TCP connection to a hylafax server, that's much much less trouble. |
04:08.19 | nDuff | khronos: if that wouldn't fly -- then it's store-and-forward. |
04:11.40 | [TK]D-Fender | mosty, you mean to transcode prompts you already have? |
04:12.22 | mosty | [TK]D-Fender, yes |
04:13.50 | mosty | i have lots of files, i don't want to use that online util |
04:14.08 | [TK]D-Fender | mosty, don't know a really "convenient" way, sorry |
04:15.25 | *** join/#asterisk ChkDigit (n=mrw@static24-72-71-175.regina.accesscomm.ca) |
04:16.14 | ChkDigit | Hey guys, I had Asterisk<->Polycom presence working when I let a client. I walked out the door, and it stopped. |
04:16.34 | mosty | there are links to a dead website which had a "convert" cli command, this is as close as i have found to a good solution |
04:16.54 | ChkDigit | What is the reason for Asterisk to report that a set is Idle, despite the fact the CLI says it is rining? |
04:17.02 | ManxPower | convert is a 1.4+ feature, I believe |
04:17.35 | ManxPower | ChkDigit: the hint and exten do not match |
04:17.49 | mosty | manxpower: do you have 1.4 installed? could you see if it includes res_conv.so ? |
04:17.58 | ManxPower | ChkDigit: What 1version of Asterisk? |
04:18.11 | ChkDigit | They match. However, I just did /etc/init.d/asterisk restart and things look okay... |
04:18.19 | ManxPower | mosty: Gads no, I currently have no timetable to update my machines to 1.4 |
04:18.32 | ChkDigit | It is Asterisk Bus.Ed. 1.3.0.b1 |
04:19.17 | ManxPower | ChkDigit: you'll have to talk to Digium for support of that |
04:20.47 | *** part/#asterisk darkladywolf (n=root@wolf.tpgi.com.au) |
04:22.41 | ChkDigit | Believe me. I did., |
04:23.09 | ChkDigit | Spent 2 hours on tier 1, and 3 on 2 with angler. |
04:23.20 | ChkDigit | Solved the problem, then it crept back. |
04:25.43 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
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04:31.33 | *** part/#asterisk danicholson (n=danichol@203.89.191.222) |
04:33.32 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
04:41.09 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
04:45.19 | *** join/#asterisk threat (n=phix@60-240-43-214.static.tpgi.com.au) |
04:45.41 | threat | well that was fun |
04:47.33 | threat | iax refused to work |
04:49.45 | threat | is there a iax work directive for asterisk? |
04:56.16 | *** join/#asterisk squish102 (n=squish10@cpe-024-074-100-250.carolina.res.rr.com) |
04:59.20 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
05:01.39 | ChkDigit | In VoiceMail() what would cause it to not respond to the caller pressing 0 or *? |
05:04.10 | *** join/#asterisk Strom_C (n=strom@135.196.213.180) |
05:06.15 | *** part/#asterisk squish102 (n=squish10@cpe-024-074-100-250.carolina.res.rr.com) |
05:06.42 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
05:07.15 | Keltus | is there an actual hardware limitation to regular computer modems that make them not work with asterisk? or is it purely software |
05:08.41 | mitcheloc | i imagine it's software |
05:16.23 | nDuff | I've seen a discussion of why a voice modem is inadequate for telephony, but that was long ago and far away, and I certainly couldn't come up with a link right now. |
05:16.46 | nDuff | ...for that matter, it might have been in something published by way of dead tree. |
05:17.01 | Keltus | hah |
05:17.13 | Keltus | I'm trying to avoid buying new hardware for my voip stuff |
05:17.23 | Keltus | so I want my phone line to hit my computer |
05:17.32 | nDuff | I mean, there *is* chan_modem |
05:19.36 | JT | yes |
05:19.36 | nDuff | ahh. *was* chan modem, but it was removed because the hardware generally couldn't do what chan_modem asked it to. |
05:19.36 | JT | cheap modems are cheap shit |
05:19.37 | JT | with poor components like hybrids |
05:19.37 | JT | and you need a driver for the device |
05:19.37 | JT | and it must be a soft modem for it to be even possible to make a driver |
05:19.37 | Strom_C | ~ygwypf |
05:19.50 | jbot | somebody said ygwypf was You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
05:19.50 | JT | just spend a couple of dollars and get something proper |
05:19.50 | *** join/#asterisk Bazy (n=bazy@exodus.upctm.ro) |
05:29.20 | *** join/#asterisk JoelSolanki (i=Joel@202.160.161.94) |
05:29.33 | threat | JT, hi |
05:29.33 | JT | hi |
05:29.33 | threat | JT, I require your assistance |
05:29.33 | threat | I am trying to setup IAX between two asterisk servers, however I keep getting authorisation denied messages |
05:29.33 | threat | is there some type of ACL settings I need to configure? |
05:29.34 | threat | would you like a pastebin dump of both of the servers iax.conf and extensions.conf file? |
05:29.34 | Keltus | JT: what do you guys recommend? |
05:29.34 | JT | Keltus: what is the scenario? |
05:29.34 | nDuff | threat: picking out a single individual and demanding assistance is not exactly good ettiquette. Actually, neither of those actions is. |
05:29.35 | Keltus | JT: I want to run a customer service center |
05:29.35 | nDuff | Keltus: How many lines? |
05:29.35 | Keltus | we have about 10 CSRs rotating, and we get about 200 calls a day |
05:29.35 | Keltus | we have 1 toll free number |
05:29.35 | Keltus | anything else is up in the air |
05:29.35 | nDuff | Keltus: I'd be getting a proper PRI and a T1 card (or E1, if appropriate for your region) at that point. |
05:29.36 | Keltus | can you explain what those things are and what they ? |
05:29.36 | Keltus | they do* |
05:29.36 | nDuff | Keltus: a PRI is where your phone company gives you a single drop with 24 voice channels (or maybe fewer voice channels and runs data over the rest) |
05:29.36 | nDuff | Keltus: a T1 card goes in the Asterisk box; you plug the PRI into it. |
05:29.36 | threat | nDuff, ok |
05:29.36 | threat | nDuff, how do you suggest I go about this then? |
05:29.37 | threat | I singled JT out since he helped me earlier today |
05:29.37 | JT | nDuff: ask the channel |
05:29.37 | JT | err |
05:29.37 | JT | threat: |
05:29.39 | JT | people don't like being singled out |
05:29.39 | JT | or dobbed in |
05:29.59 | Keltus | hmmm okay, what would the data channel be for? |
05:30.05 | Keltus | we have standard internet access |
05:30.13 | Keltus | I think T1 lines already, over ethernet |
05:30.19 | Keltus | can I just use those? |
05:30.28 | *** join/#asterisk dhakatel (n=ashrar@58.65.224.5) |
05:30.32 | nDuff | threat: about what -- asking for help? There's plenty of documentation on that; although it's not quite IRC-centric, http://www.catb.org/~esr/faqs/smart-questions.html is a good place to start. |
05:30.54 | nDuff | Keltus: who are you getting your T1 lines from? Are they also a telco? |
05:31.17 | nDuff | Keltus: They might be able to split it up and put some voice channels in, but it very much depends on who they are; I'd expect they'd want to sell you something slightly different. |
05:31.48 | nDuff | Keltus: If the full T1's bandwidth is more than you need, you might talk to them about splitting it into 12 voice lines and the other half data. |
05:32.10 | threat | nDuff, ok great |
05:33.09 | Keltus | our T1 line is from XO communications |
05:33.22 | Keltus | okay, I didn't know about the voice + data package |
05:33.23 | nDuff | I don't know them. |
05:33.26 | Keltus | I'll check that out |
05:33.57 | jql | it's usually listed under "fractional" service |
05:34.41 | Keltus | gotcha |
05:34.47 | threat | ok here we go, general questions 1) how do I setup IAX asterisk - asterisk 2) How do I stop POTS incoming calls from ringing analog phone when the caller has hanged up |
05:38.26 | nDuff | threat: (1) - that's a pretty general question. Have you read http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers? |
05:39.13 | nDuff | threat: (2) - err... they *do* stop. For me, anyhow. Presumably something's broken on your site if they don't, and that needs to be localized before it can be debugged very effectively. |
05:42.04 | Keltus | what card would you guys recommend for just some testing? ie. something I can do at home and configure asterisk until I want to make the big step? |
05:52.45 | threat | nDuff, hmmm |
05:53.15 | threat | nDuff, well if I ring the asterisks server on the POTS line then hang up the asterisk server keeps ringing the internal analog lines |
05:53.18 | JT | Keltus: you don't need any cards to do testing |
05:53.23 | JT | Keltus: you can just use VoIP |
05:53.24 | threat | what setting would control this? |
05:53.27 | threat | timeout? |
05:54.50 | nDuff | Keltus: the only thing you might buy hardware for in test-phase is if you want to decide on what kind of phones to use. |
05:56.22 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
05:57.08 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
05:57.22 | JoelSolanki | Hi all. was reading on asterisk 1.4 and found that it doesnt required re-invites. it tell like this. ' if we know at call setup that we can release media, we will do that directly ' |
05:57.34 | JoelSolanki | is this early media disable stuff ? |
05:57.44 | THX2000 | Is there a way to get asterisk to store voicemail wav's and info w/ r/w access rights for users? |
05:58.10 | mosty | THX2000, what are you trying to do? |
05:58.51 | THX2000 | get php to parse the info stored in msg0000.txt |
06:00.11 | Keltus | well I want to be able to test calling my toll free number --> ringing out yahoo messenger over gizmo |
06:00.35 | mosty | THX2000, running php from apache, cgi, cron, what? |
06:00.40 | Keltus | how would I do it without any hardware? the toll free number needs to redirect to an actual phone # |
06:00.45 | THX2000 | thttpd |
06:01.02 | THX2000 | php-cli |
06:01.11 | JT | Keltus: you can get DIDs over voip |
06:01.24 | JT | not as reliable as a PSTN one, but good enough a lot of the time |
06:01.26 | mosty | THX2000, you could make the voicemail dirs setgid |
06:01.54 | mosty | THX2000, and put them in group that the thttpd user is a member of |
06:02.28 | nDuff | Keltus: does it need to be the same toll-free number you'll be using in production? |
06:03.02 | Keltus | yea, I want to make sure it works end-to-end |
06:03.08 | THX2000 | that kinda makes sense. Im in a bit over my head here, but i guess thats how ya learn :P |
06:03.24 | Keltus | basically, be able to handle 5 calls per day to our customer service rep, as a beta test |
06:04.02 | Keltus | what kind of hardware would I need to just have a very small number of calls? I have an old linux computer I can set up asterisk on |
06:04.13 | *** join/#asterisk SwordManX (n=sword@ip70-161-179-101.hr.hr.cox.net) |
06:04.20 | Keltus | and then I'll be using gizmo -> yahoo messenger for the VoIP for a free test call |
06:05.09 | nDuff | Keltus: ahh. For just one line, an IAXy might be your cheapest bet. |
06:05.59 | nDuff | ...actually, wait, that's the wrong end. |
06:06.18 | nDuff | ...that'll talk to a phone, not to an outside line. (I never keep "FXO" vs "FXS" straight, but that's the distinction) |
06:06.21 | JT | IAXy isn't that cheap anyway |
06:06.39 | nDuff | true. |
06:06.49 | nDuff | Something from Sipura, then. |
06:07.03 | nDuff | I think they've got an ATA that does both FXO and FXS. |
06:07.24 | Keltus | is a IAXy a FXS device? |
06:07.26 | JT | SPA-3102 has 1 FXS and 1 FXO port |
06:07.30 | JT | Keltus: yes |
06:07.34 | Keltus | I just need a FXO don't I |
06:07.42 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
06:07.46 | nDuff | Keltus: depends, what kind of phone(s) will you be using? |
06:07.50 | Keltus | since for outbound, I will use gizmo and yahoo messenger for testing |
06:07.53 | nDuff | ahh. |
06:07.55 | Keltus | I'll use speakers and a mic |
06:07.56 | Keltus | for now |
06:08.36 | nDuff | I don't exactly advise that. Something built for telephony will give you proper echo cancellation, and otherwise sound a whole lot better to folks on both ends. |
06:08.52 | JT | nDuff: he's just testing |
06:09.34 | nDuff | well, yeah, but if your support staff hate it during the test... |
06:09.43 | Keltus | should I get them a headset? |
06:10.01 | nDuff | ...but then, I guess whether it's a sell-it-to-the-staff test or a get-asterisk-working test. |
06:10.09 | nDuff | Keltus: I'd recommend it, yes. |
06:10.21 | Keltus | gotcha |
06:10.30 | Keltus | it's more get-asterisk-working but it would be nice if everyone was excited about it |
06:10.52 | JT | Keltus: yeah you must use headset, pc speakers and mic pretty much won't work |
06:11.25 | Keltus | and then you recommend SPA-3102 for the incoming call part? |
06:11.40 | JT | for business a card is probably better |
06:11.44 | JT | like a TDM400P |
06:12.11 | JT | but as someone else suggested, if you have quite a few agents, you'll want PRI ISDN T1/E1 |
06:12.23 | Keltus | yeah I want to just test a few calls on one line first |
06:12.34 | Keltus | to make sure this is what we want |
06:13.08 | JT | the problem is you'll need a totally different card, but i guess an spa-3102 might be cheap enough to test |
06:13.18 | JT | otherwise you can get phone service over the Internet |
06:14.05 | Keltus | the different between the two cards is the spa-3102 does incoming and outgoing right? so wouldn't the headset be good enough |
06:14.13 | Keltus | the CSR will have computers, and plug their headsets in them |
06:15.08 | JT | not really |
06:15.20 | JT | the audio quality from a SIP hardphone like a Polycom is much better |
06:15.45 | Keltus | ok |
06:15.57 | Keltus | I'll put in orders for those tomorrow. any other equipment we'll need? I read about the X100P card for asterisk calls. |
06:15.59 | flenders | Keltus: I have an SPA-3000 at home, and sound quality is not that good when using the FXO channel |
06:16.20 | JT | Keltus: headsets to suit the phone as well i guess |
06:16.38 | JT | Keltus: you should probably do a bit more research and testing before buying a whole bunch |
06:16.42 | Keltus | right. so SPA-3102 and a headset |
06:16.57 | Keltus | well it prices to about $100 right now so it's not much |
06:17.01 | JT | yeah |
06:18.21 | Keltus | great |
06:18.42 | Keltus | thanks for all the advise |
06:39.09 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
06:43.20 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
06:43.29 | mosty | i have a strange problem with one-touch recording. it works with some calls but not others. sip->sip calls work, but sip->iax (->sip) calls don't work from the originating sip end |
06:43.45 | mosty | all my dial commands have wW in the options |
06:43.52 | mosty | what could be wrong? |
06:52.19 | *** join/#asterisk Ast001 (n=uros@77-105-44-230.adsl-2.sezampro.yu) |
06:52.32 | Ast001 | hello |
06:52.43 | Ast001 | I need your advice |
06:53.23 | Ast001 | is it good to set priority to -19 to asterisk |
06:53.30 | Ast001 | with renice -19 asterisk pid |
06:53.35 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
06:54.00 | mosty | are you having trouble with running asterisk at the default nice level? |
06:54.07 | Ast001 | well I have |
06:54.30 | Ast001 | My operators who works from hove and have adsl 512/128 can not hear properly musiconhold |
06:54.48 | Ast001 | they can hear but it stops for 1-2 sec etc... |
06:55.00 | JT | you must have rtp silence supression or a highly compressed codec like g.729 |
06:55.17 | Ast001 | no i dont use g729 |
06:55.19 | mosty | i doubt that's load-related |
06:55.32 | Ast001 | i use ulaw alaw gsm ilbs |
06:55.33 | Ast001 | ilbc |
06:55.43 | Ast001 | operators have xlite |
06:55.56 | mosty | turn off silence sippression in x-lite |
06:56.07 | JT | mosty: load related? |
06:56.13 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.voop.net) |
06:56.20 | *** join/#asterisk hellop (n=hellop@udp112969uds.hawaiiantel.net) |
06:56.23 | JT | yes but silence supression in x-lite only controls its stream to asterisk |
06:56.35 | Ast001 | well ok |
06:56.43 | Ast001 | i will do it |
06:56.55 | mosty | jt: i doubt the issue would have anything to do with the nice level |
06:57.12 | JT | mosty: ah yeah |
06:57.17 | mosty | Ast001, does your pbx have a zaptel timer? |
06:57.17 | hermuli | does anyone have an idea why app_mysql would leave connections open (sometimes) even when i call the closing thing from dialplan? |
06:57.19 | Ast001 | is adsl 512/128 is enough for ulaw alaw |
06:57.30 | JT | yes unless it's over a bad dsl network |
06:57.40 | JT | you need 85kbit/s |
06:57.45 | mosty | Ast001, try with gsm |
06:57.46 | Ast001 | i have pri isdn |
06:57.57 | JT | mosty: he's already tried though |
06:57.58 | Ast001 | i dont know about zaptel timer |
06:58.05 | Ast001 | I tryed gsm |
06:58.16 | Ast001 | They can hear gsm but moh stops and continue |
06:58.37 | JT | moh does the exact same thing? |
06:58.37 | mosty | how long do the pauses last? |
06:58.48 | Ast001 | 1-2 sec |
06:58.51 | Ast001 | to 4-5 sec |
06:59.27 | Ast001 | silence suppresion |
06:59.30 | mosty | is it just moh that does that? does it happen when you're speaking? |
06:59.32 | Ast001 | can not find that |
06:59.35 | Ast001 | no |
06:59.40 | Ast001 | they can speak well |
07:00.29 | Ast001 | is that excacll named SILENCE SUPRESION ? |
07:00.56 | JT | might be something similar |
07:01.07 | Ast001 | use concealment ? |
07:01.09 | mosty | i don't know what x-lite calls it. don't worry, it probably wont help |
07:01.15 | Ast001 | transmit silence ? |
07:01.24 | JT | yes |
07:01.36 | Ast001 | transmit silence=yes |
07:01.44 | Ast001 | at the moment |
07:02.22 | Ast001 | now its now |
07:02.24 | Ast001 | now |
07:02.37 | Ast001 | transmit silence=no |
07:02.57 | *** join/#asterisk af_ (n=getsmart@81-174-47-36.f5.ngi.it) |
07:03.13 | Ast001 | it did not help |
07:03.30 | Ast001 | pauses on moh continues on ilbc and gsm |
07:03.43 | JT | Ast001: so there is a PRI on the same machine that provides MoH? |
07:03.50 | mosty | what version of asterisk? |
07:03.55 | Ast001 | 1.2.17 |
07:04.01 | Ast001 | yes everything on the same machine |
07:04.06 | JT | hrm |
07:04.11 | JT | weird |
07:04.18 | JT | many channels? |
07:04.35 | Ast001 | well about 10 channels work at the same time |
07:05.09 | Ast001 | is this SIP related problem ? |
07:05.16 | Ast001 | would it be better with mozphone and iax2 ? |
07:05.22 | JT | maybe, who knows |
07:05.28 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
07:05.33 | JT | run rtp debug on a peer |
07:05.42 | Ast001 | i tryed that |
07:05.46 | JT | and have a call on that peer, then put it on MoH |
07:05.55 | JT | does the flow or RTP packets slow during MoH? |
07:06.07 | Ast001 | didnt noticed that |
07:06.15 | Ast001 | enormous number of lines of type |
07:06.29 | Ast001 | send rtp packet recieved rtp packet ... |
07:06.42 | JT | yes it sends a line for every packet |
07:07.15 | Ast001 | rtp debug ip xx.xx.xx.xx ? |
07:07.21 | JT | yes |
07:07.43 | JT | see if it sends less rtp packets during MoH silence |
07:08.50 | Ast001 | only this line Sent RTP packet to 77.105.44.230:50419 (type 98, seq 44253, ts 2830616, len 50) |
07:09.37 | JT | there should be more than one line, just take note of if the flow of packets is reduced during the problem silence |
07:10.03 | *** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.3.121.revip2.asianet.co.th) |
07:10.06 | Ast001 | this line continue to repeat indefinitly without stop |
07:10.23 | Ast001 | until i do no debug |
07:10.43 | JT | hrm ok, so the flow is probably unchanged by silence |
07:10.52 | Ast001 | yes |
07:11.00 | JT | try a different SIP client or phone |
07:11.00 | BugKhaM | hi, what's the best package used to recieve/send faxes? spandsp? |
07:11.12 | JT | BugKhaM: explain the problem a little more |
07:12.07 | BugKhaM | JT: I wanna receive faxes and send it out automatically by email, just wondering what to use |
07:12.30 | BugKhaM | JT: http://www.voip-info.org/wiki-Asterisk+Fax+to+email is what I found |
07:12.39 | JT | i guess either spandsp or hylafax should work |
07:13.18 | BugKhaM | JT: application spandsp used to be ported with *, but not anymore? |
07:13.51 | JT | yeah only guaranteed to work with certain versions |
07:13.56 | Ast001 | does it have something with irq priorities |
07:14.07 | Ast001 | because eth0 is sharing resource |
07:14.17 | JT | Ast001: with what? |
07:15.49 | Ast001 | no |
07:15.54 | Ast001 | its at 23: 66489101 IO-APIC-fasteoi eth0 |
07:16.16 | JT | err so what's it sharing with? |
07:16.16 | Ast001 | but its on 23 do i need to move it ti 2 or 3 ? |
07:16.28 | JT | no why would you |
07:16.28 | Ast001 | no eth1 is sharing it is for LAN |
07:16.32 | Ast001 | sorry |
07:16.37 | Ast001 | ok |
07:16.47 | JT | i see no evidence of irq sharing from what you've pasted |
07:17.01 | JT | now try running zttest whilst doing problem MoH calls |
07:17.02 | Ast001 | yes eth0 is not sharing |
07:17.12 | JT | see if the scores drop below 99.97% |
07:17.16 | Ast001 | i made error confused eth0 with eth1 |
07:18.20 | Ast001 | -bash: zttest: command not found |
07:18.32 | JT | you should install it then |
07:19.06 | Ast001 | where can i found it ? |
07:19.14 | Ast001 | is it part of zaptel ? |
07:19.19 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
07:19.19 | JT | yes |
07:19.41 | Ast001 | well I did ./configure make make install during compiling zaptel |
07:20.04 | Ast001 | why did it install that zttest ? |
07:20.09 | Ast001 | didn't ? |
07:20.16 | JT | it might not by default |
07:20.23 | JT | just check the documentation on how to do it |
07:21.27 | *** join/#asterisk mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
07:21.43 | mattwj2005 | anyone of the 265 people in this room own a solar panel |
07:21.45 | mattwj2005 | ? |
07:22.04 | mattwj2005 | :) |
07:25.17 | *** join/#asterisk squall (n=squall@ns2.squallnetwork.net) |
07:26.39 | Keltus | what's the difference between a x100 and a x100p? |
07:27.03 | JT | one has a p missing |
07:27.06 | mattwj2005 | digium cards? |
07:27.28 | Keltus | yea |
07:27.50 | JT | Keltus: one has a letter p missing |
07:27.51 | mattwj2005 | no idea off hand |
07:28.01 | Keltus | funny JT |
07:28.02 | mattwj2005 | have you googled it? |
07:28.04 | Keltus | yeah |
07:28.05 | mattwj2005 | :) |
07:28.06 | Keltus | nothing |
07:28.15 | Keltus | I'm thinking x100 is just shorthand for x100p |
07:28.17 | mattwj2005 | www.voip-info.org ? |
07:28.18 | JT | Keltus: and it's the CORRECT ANSWER :) |
07:28.25 | Keltus | ah |
07:28.27 | JT | there is no such thing as an X100 |
07:28.44 | mattwj2005 | the p guy was off somewhere |
07:28.45 | mattwj2005 | :) |
07:28.47 | mattwj2005 | lol |
07:28.52 | JT | the X100P is a crappy discontinued FXO card based off an old discontinued Intel winmodem chipset |
07:29.18 | mattwj2005 | I think he had to use the bathroom |
07:29.51 | Keltus | yea. the one I purchased for testing is called TDM01B |
07:30.04 | mattwj2005 | sorry 2:30 am humor |
07:30.11 | JT | which is a form of the TDM400P |
07:30.18 | Keltus | yup. I know |
07:30.28 | Keltus | it was $150 |
07:35.14 | Ast001 | rtp debug ip does not give me anything when I use iax2 and mozphone |
07:35.22 | Ast001 | and with mozphone its much better |
07:35.31 | JT | well of course not, iax2 doesn't use rtp |
07:35.45 | Ast001 | so maybe its rtp thing ? |
07:36.03 | JT | rtp or sip maybe or xlite |
07:36.11 | Ast001 | i see |
07:36.18 | Ast001 | then I will migrate to mozphone |
07:36.18 | JT | that's a minimum off all the variables that have changed :P |
07:36.49 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
07:37.06 | mattwj2005 | I am thinking about using the natural Minneasota sunlight in the Summer |
07:37.28 | mattwj2005 | thinking about buying a solar panel....assuming my apartment doesn't mind |
07:37.32 | santoshr | i am using asterisk 1.2.9 .. is thr a way to tell a call to goto a context,s,1 after the "S" time in Dial |
07:37.47 | JT | mattwj2005: for what purposes? |
07:38.19 | santoshr | so when the call disconnects it goes to a specific playback or something |
07:38.29 | mattwj2005 | charge batteries |
07:38.47 | mattwj2005 | be kind to the earth |
07:38.50 | mattwj2005 | that kind of thing |
07:39.00 | JT | mattwj2005: batteries for anything in specific? |
07:39.17 | mattwj2005 | well I own a 80 GB iPod |
07:39.45 | mattwj2005 | it would be cool if I had battery power for it this summer |
07:39.57 | mattwj2005 | I have a few other usb devices |
07:40.10 | mattwj2005 | I have a nice radio that runs on AA |
07:40.51 | mattwj2005 | I also have a gameboy that takes AA |
07:41.18 | mattwj2005 | energy cost in the US are always so expensive during the summer |
07:41.45 | mattwj2005 | and I am a hobbist at heart |
07:42.26 | flenders | haha, a solar panel to recharge your ipod |
07:42.29 | flenders | that was a good one |
07:42.38 | mattwj2005 | lol |
07:44.30 | JT | the solar panel would probably cost more, but it could be a fun project :) |
07:44.32 | Keltus | another newbie question - if I have 2 FXO ports, do I need 2 wall jacks that each have a different line on it? or is there some sort of splitter |
07:45.06 | flenders | Keltus: you need 2 wall jacks |
07:45.12 | mattwj2005 | ~$100 get some good voltage |
07:45.18 | mattwj2005 | that is usd |
07:45.27 | santoshr | . is thr a way to tell a call to goto a context,s,1 after the "S" time in Dial |
07:45.30 | JT | mattwj2005: wattage you mean? |
07:45.39 | Keltus | flenders: do telcos usually give you two different numbers for each jack? or one number for the two jacks |
07:45.51 | mattwj2005 | 15 W |
07:45.54 | flenders | you want to split one single line? |
07:45.59 | JT | Keltus: that would depend if they're different lines |
07:46.00 | flenders | or you want 2 lines? |
07:46.02 | mattwj2005 | http://www.amazon.com/Sunforce-50032-Solar-Battery-Charger/dp/B0006JO0X8/ref=pd_bbs_sr_4/002-9570840-7785656?ie=UTF8&s=automotive&qid=1177312165&sr=8-4 |
07:46.05 | Ast001 | its much better on iax2 |
07:46.06 | *** join/#asterisk Bananaskin (n=Banana@81-86-102-88.dsl.pipex.com) |
07:46.18 | Keltus | I have a 1 800 number |
07:46.29 | JT | Ast001: must be a bug in something, somewhere..... |
07:46.33 | Keltus | if I have one line and split it, will it alternate the jacks? |
07:46.43 | Keltus | I mean, I want to be able to take 2 calls simultaneously |
07:47.09 | Ast001 | well it was default isntalation of Asterisk 1-2-17 and zaptel for 1-2 and libpri |
07:47.27 | mattwj2005 | 41 inches might be too big though |
07:47.40 | Ast001 | it is strange it worked perfecly before I moved to optical cable |
07:47.41 | mattwj2005 | ~ 1 meter |
07:48.10 | JT | optical cable where? |
07:48.30 | Ast001 | at headquarters where server is |
07:48.51 | flenders | Keltus: you need 2 lines |
07:49.17 | flenders | Keltus: and 2 FXO channels |
07:49.53 | JT | Ast001: yeah, what does the fibre do? |
07:49.55 | flenders | mattwj2005: massive panel |
07:50.06 | Keltus | alright, so I guess I need to call the telco to set this up. it's not something I can do in hardware, right? |
07:50.08 | JT | that's like saying "I connected a piece of copper to my server" |
07:50.37 | mattwj2005 | maximum output at 12 V....1.5 A |
07:50.42 | Ast001 | I had cable link 2mbit/sec |
07:50.48 | Ast001 | cable internet |
07:50.56 | JT | oh, INTERNET, ok |
07:50.59 | Ast001 | but link was down to meny times |
07:51.14 | Ast001 | and ISP said optical cable is much better |
07:51.18 | JT | maybe your new isp does something funky to rtp, who knows |
07:51.18 | mattwj2005 | using ohms law |
07:51.20 | JT | same isp? |
07:51.21 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
07:51.21 | JT | hrm |
07:51.21 | Ast001 | stable |
07:51.36 | Ast001 | no isp is the same |
07:51.43 | Ast001 | new is just optical cable |
07:51.53 | JT | Ast001: is it optical ethernet? |
07:52.02 | Ast001 | they say no barrieras no firewall nothing |
07:52.18 | Ast001 | some box is connected to server with ethernet |
07:52.25 | JT | but you still can't tell me where along the line the fibre comes into it |
07:52.48 | mattwj2005 | I am guessing that'll be too big |
07:53.04 | Ast001 | other wire is tv cable signal and goes to tv cable box |
07:53.19 | mattwj2005 | here is a little more conversative solution |
07:53.21 | mattwj2005 | http://www.amazon.com/Sunforce-50022-Battery-Trickle-Charger/dp/B0006JO0TC/ref=pd_bbs_12/002-9570840-7785656?ie=UTF8&s=automotive&qid=1177312165&sr=8-12 |
07:53.33 | Ast001 | don't know really |
07:53.45 | Ast001 | but server can go to internet ok and ping is ok too |
07:53.49 | *** join/#asterisk Uatec_ (n=uatecuk@adsl.ntsols.com) |
07:53.52 | Uatec_ | Greetings |
07:54.11 | Ast001 | no firewalls |
07:54.18 | Ast001 | connection speed is ok too |
07:54.24 | Ast001 | said speakeasy.net/speedtest |
07:54.40 | JT | Ast001: have you even seen the fibre cable? |
07:55.11 | mattwj2005 | 416 mA |
07:55.26 | mattwj2005 | @ 12 V |
07:55.31 | *** join/#asterisk af_ (n=getsmart@81-174-46-10.f5.ngi.it) |
07:57.03 | flenders | Ast001: so, using cable it worked, with "fiber" it doesn't? |
07:57.26 | Ast001 | with cable it worked great |
07:57.39 | Ast001 | now it is working with problems |
07:57.40 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
07:58.05 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
07:59.49 | JT | Ast001: is the fibre the ethernet connection between the modem and the PC or what? |
08:00.04 | mattwj2005 | Ast001 do you have any network certs? |
08:00.44 | Ast001 | between pc and that box is ordinary ethernet cable |
08:00.57 | Ast001 | used before |
08:00.57 | JT | so you have no idea where the fibre is? awesome |
08:01.03 | Ast001 | no |
08:01.09 | mattwj2005 | there is a short list of things that kill voip quality |
08:01.26 | nemski | latentcy |
08:01.32 | JT | add magic optical cable to it |
08:01.32 | nemski | cpu load |
08:01.33 | Ast001 | well isp sets all of that |
08:01.51 | mattwj2005 | lack of bandwidth......people that set up bad qos or traffic shaping......greedy telephone companies.......and bent cables |
08:01.59 | Ast001 | I just put ethernet cable in that box and configured it on server |
08:02.22 | JT | so you have no clue about how you connect to the Internet? great |
08:02.33 | mattwj2005 | you can also add poor cables...loose cables....cables without clips etc |
08:03.05 | Ast001 | throug tv signal I guess |
08:03.10 | JT | i saw a cable in a datacentre that was a flat 8 core cable between switch and server |
08:03.17 | JT | all i thought was "what an idiot" |
08:03.40 | mattwj2005 | I make 46.1k usd doing networking |
08:03.47 | flenders | Ast001: do you have a cable modem? |
08:03.49 | mattwj2005 | I love to help for free if I can |
08:03.52 | JT | Ast001: so the coaxial cable tv network |
08:03.59 | JT | that's not fibre :) |
08:04.28 | flenders | mattwj2005: is that your salary? |
08:04.54 | mattwj2005 | indeed |
08:05.15 | flenders | is that a lot in the US? |
08:05.20 | Ast001 | i have some strange box wich is bigger that cable modem i had for previous connection |
08:05.29 | mattwj2005 | I am not rich |
08:05.32 | mattwj2005 | I am not poor |
08:05.42 | flenders | and you're not starving I guess |
08:06.04 | mattwj2005 | nope |
08:06.25 | mattwj2005 | every time I go to the grocery store I try to give to the hungry |
08:07.21 | mattwj2005 | no man on this planet should go hungry if he has neighbors and a city or town |
08:07.43 | flenders | true! |
08:07.48 | flenders | what state are you in? |
08:08.04 | mattwj2005 | Minneasota....we touch Canada! |
08:08.37 | JT | does canada enjoy being touched? |
08:08.54 | flenders | hahahha |
08:09.09 | mattwj2005 | I am not Canada |
08:09.11 | *** join/#asterisk izaak (n=izaak@modemcable097.151-202-24.mc.videotron.ca) |
08:09.20 | mattwj2005 | you'll have to ask them |
08:09.24 | JT | heh |
08:09.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
08:11.11 | mattwj2005 | we are pretty safe...we touch just the top part |
08:11.24 | mattwj2005 | *with the top part |
08:12.04 | mattwj2005 | http://en.wikipedia.org/wiki/Minnesota |
08:12.14 | mattwj2005 | http://upload.wikimedia.org/wikipedia/commons/thumb/5/54/Map_of_USA_MN.svg/286px-Map_of_USA_MN.svg.png |
08:13.26 | *** join/#asterisk mkl1525 (n=mkl1525@pD953083D.dip0.t-ipconnect.de) |
08:16.38 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
08:18.30 | mkl1525 | Hi, I've got agents and I'd like to distinguish if a direct call is made or the queue application is calling so that I can set different Dial() parameters. But haven't found any variable that I could use for this, setting my own variable didn't help either - any hints? |
08:20.59 | dhakatel | yap |
08:21.42 | dhakatel | i used it |
08:22.37 | dhakatel | what type of reduncy u need |
08:22.55 | JT | dhakatel: wrong channel? |
08:22.59 | dhakatel | just fail over and take over |
08:23.06 | dhakatel | or data replecation |
08:23.16 | JT | dhakatel: WRONG CHANNEL |
08:24.06 | mattwj2005 | anyone have any good suggestions for AA to USB adapters? |
08:24.33 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
08:24.51 | *** part/#asterisk Ast001 (n=uros@77-105-44-230.adsl-2.sezampro.yu) |
08:26.47 | santoshr | can the call flow be controlled after the call disconnects after S(time) in Dial |
08:28.22 | *** join/#asterisk jm|laptop (n=jm@sentry.flags.co.uk) |
08:28.33 | flenders | mattwj2005: wanna charge your ipod using AA? |
08:28.33 | *** join/#asterisk saftsack (n=oliver@p54a72e07.dip0.t-ipconnect.de) |
08:28.45 | mattwj2005 | yup |
08:29.02 | flenders | I think apple has something |
08:29.22 | flenders | I think it plugs into the ipod connector |
08:29.45 | mattwj2005 | see here is my idea |
08:29.57 | mattwj2005 | solar panel charges AA batteries |
08:29.59 | tzafrir | I saw some in ebaw. But I really don't understand how is this related to * |
08:30.07 | mattwj2005 | when done I charge my iPod |
08:30.12 | tzafrir | Unless * is considered to be a wildcard |
08:30.35 | flenders | :D |
08:30.43 | mattwj2005 | I am sorry I am off topic |
08:30.44 | mattwj2005 | :( |
08:31.22 | JT | not a bad off topic though |
08:31.51 | mattwj2005 | I am saving money on energy though |
08:31.51 | mattwj2005 | probably shutdown some computers too |
08:31.51 | mattwj2005 | :( |
08:32.27 | mattwj2005 | I will be running a form of an asterisk server...even if it is virtual |
08:33.25 | mattwj2005 | speaking of Asterisk... |
08:33.35 | mattwj2005 | I can get one up and running without using a make samples |
08:33.38 | mattwj2005 | :) |
08:35.23 | *** join/#asterisk Nickle (i=Loots@c-71-204-146-212.hsd1.ca.comcast.net) |
08:35.58 | Nickle | Can anyone recommend a low cost yet good quality outbound service? |
08:36.23 | Nickle | US maybe Canada/unlimited calling. |
08:36.23 | mattwj2005 | www.teliax.com is my favorite service provider |
08:36.33 | Nickle | Ahh, good price? |
08:36.43 | mattwj2005 | ~$25 usd |
08:36.47 | Nickle | Not bad |
08:37.12 | mattwj2005 | good quality...they are looking to hire people too |
08:37.17 | mattwj2005 | ATTN ROOM |
08:37.21 | Nickle | Heh |
08:37.26 | Nickle | You work for them? |
08:37.29 | Nickle | ;P |
08:37.47 | jql | shill! |
08:38.08 | mattwj2005 | nope |
08:38.43 | mattwj2005 | just say if your between Diet Mt and a desecent Internet connection.... |
08:38.48 | Nickle | that $24 plan is unlimited outboud and incomming eh |
08:39.10 | mattwj2005 | yeah...but read the fine print |
08:39.17 | Nickle | my AIX server is in LA |
08:39.34 | Nickle | IAX |
08:39.35 | mattwj2005 | they give you so many mins and then they charge over that amount |
08:39.36 | Nickle | rather |
08:40.23 | mattwj2005 | they have been pretty reliable and good quality |
08:40.25 | Nickle | is there a free service which will allow me to call toll free numbers? |
08:40.36 | Nickle | actually, here's the scoop |
08:40.43 | mattwj2005 | not sure |
08:40.54 | Nickle | i recently was promoted, and i work with a global team. i just need to dial ATT teleconference |
08:40.57 | Nickle | but i have a CELL phone |
08:41.07 | Nickle | and i work out of my house 4 days a week |
08:41.13 | Nickle | haha |
08:41.31 | Nickle | but i have my asterisk server, so i figure, what the heck |
08:43.48 | mattwj2005 | ISBN 0-596-00962-3 |
08:43.54 | mattwj2005 | read that |
08:44.35 | mattwj2005 | it has a lot of good info on what asterisk can and cannot do |
08:44.48 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
08:45.10 | mattwj2005 | I have a copy...haven't read it yet |
08:46.04 | Nickle | so there's no toll free forwarding service |
08:46.06 | Nickle | :\ |
08:46.12 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
08:46.59 | mattwj2005 | http://www.tollfreegateway.com/ |
08:47.12 | Nickle | Nice |
08:47.14 | Nickle | Thanks! |
08:47.23 | mattwj2005 | your welcome |
08:48.51 | mkl1525 | Is there any way to proceed with a queue call after the agent hangs up? tried the h extension -> works but can't use the channel with an ivr anymore, Dial with g parameter executes the parameters immediately after the caller gets in the queue and not when he leaves the queue - any further suggestions? |
08:57.36 | *** join/#asterisk af_ (n=getsmart@81-174-46-10.f5.ngi.it) |
08:59.52 | *** join/#asterisk FreezeS (n=bla@82.208.157.125) |
09:06.07 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
09:14.08 | *** part/#asterisk mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
09:15.21 | FreezeS | I've got a problem with a TE110P. I'm getting: ZT_CHANCONFIG failed on channel 25: No such device or address (6) |
09:15.38 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
09:23.51 | *** join/#asterisk uski (n=uski@stargate.esiee.fr) |
09:24.07 | tzafrir | FreezeS, is it T1 or E1? |
09:24.35 | uski | hi, i have a very stupid question for you; when using authenticate in a dialplan, how the hell do i enter the code from my phone ? it says "enter the code followed by square" but there is no "square" key ! and it's not # or * so what is it ? |
09:24.36 | tzafrir | sounds like you have T1 and trying to ocnfigure it as E1 |
09:24.53 | FreezeS | tzafrir: the jumper is set for E1 |
09:25.26 | tzafrir | FreezeS, and is that reflected in /proc/zaptel/1 ? |
09:25.46 | *** part/#asterisk nettie (n=nettie@ns.coolgadgets.it) |
09:25.48 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
09:26.21 | uski | noone uses Authenticate ??? |
09:27.18 | tzafrir | uski, "#"? |
09:27.28 | uski | didn't work, but i can try again... |
09:27.42 | tzafrir | I don't use Authenticate, BTW |
09:27.55 | hermuli | i did, for a moment, and it was # |
09:28.53 | FreezeS | tzafrir: can I paste it to you on private ? |
09:29.00 | tzafrir | yes |
09:29.20 | tzafrir | FreezeS, youy can also pasebin it |
09:29.55 | uski | hmm ok, that's strange, it seems that no audio is received from my SIP channel in fact, so that'd explain why it doesn't work, i think it's just timeouting |
09:29.56 | *** join/#asterisk nettie (n=nettie@ns.coolgadgets.it) |
09:29.58 | uski | thanks ;) |
09:30.20 | FreezeS | tzafrir: is there a way to check from software the jumper setting ? |
09:30.21 | nettie | hey tzafrir at the end I wasnt able to fix it at the end :( |
09:30.36 | nettie | still getting IRQ #16 disabled and crash |
09:32.08 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
09:32.10 | zeeesh | hi |
09:35.33 | *** join/#asterisk kore (i=kore@mindwipe.org) |
09:35.50 | uski | any idea of the way i can check if my asterisk server received something from the sip server |
09:38.45 | uski | is there something special that i should use to enable DTMF recognition ? |
09:42.34 | uski | fixed: i needed to use dtmfmode=inband |
09:42.52 | Uatec_ | uski, thankyou for sharing your solutions with us. |
09:44.06 | hermuli | I'm still wondering the mysql connections don't time out :P |
09:44.14 | hermuli | why* |
09:45.37 | hermuli | i have no idea what to try next... |
09:47.17 | uski | anyone knows why I can't place an outgoing call with DISA ? if I use Channel: SIP/provider/number in a call file it works so the setup is good, but I can't place a call with DISA. I have the tone, but as soon as I enter a number the tone changes to a "busy" tone or sth like that. Any example extensions.conf file for this ? |
09:47.47 | uski | maybe i need to prefix the number with 9 or so ? |
09:51.33 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
09:56.06 | FreezeS | tzafrir: that was it, the jumper was set in the wrong way |
09:57.04 | *** join/#asterisk sebadel (n=seb@162.36-246-81.adsl-static.isp.belgacom.be) |
09:57.42 | sebadel | hello, is this the right room foor AADK issues ? |
10:11.38 | *** join/#asterisk neoalex (n=neoalex@user-0ccengj.cable.mindspring.com) |
10:12.16 | creativx | woah is this a room |
10:12.29 | sebadel | hello |
10:12.36 | sebadel | any experience with AADK ? |
10:12.55 | neoalex | does anyone know of any service allowing unlimited calling to the US for the same price as skypeout (29.95/year) |
10:17.44 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
10:22.17 | sebadel | does anyone know why save_config could hang on a brand new AADK |
10:22.57 | *** join/#asterisk toot (n=toot@84.19.250.3) |
10:36.08 | uski | any idea of how I can place an external call using a SIP channel with DISA ? |
10:38.26 | *** join/#asterisk ccesario (n=ccesario@201-0-124-188.dsl.telesp.net.br) |
10:39.54 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
10:52.02 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
10:56.21 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
10:58.52 | *** join/#asterisk Aurs (n=Aurs@ti500720a080-4354.bb.online.no) |
10:58.59 | mosty | anyone good at debugging one touch recording? it works on some calls for me, but on internal -> external calls only the external end can turn it on/off. all my dial commands have wW though |
10:59.10 | *** join/#asterisk Fibres (i=Fibres@cpc2-leic3-0-0-cust157.lei3.cable.ntl.com) |
10:59.13 | Fibres | Hi all |
11:03.11 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
11:14.48 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
11:16.06 | Fibres | Im wondering can anyine help me. I need to provide a gateway to allow a PRI ISDN based pbx to connect to voip. |
11:19.27 | *** join/#asterisk SoftIce (n=bongo@dsl-242-115-118.telkomadsl.co.za) |
11:19.28 | SoftIce | hi |
11:19.49 | SoftIce | listen, if I have asterisk realtime, and I just want sip.conf in the db, do I also need extensions.conf |
11:19.54 | SoftIce | or can extensions.conf be read locally? |
11:20.17 | Uatec_ | AADK? |
11:25.02 | DrukenHME | SoftIce: nothing has to be in realtime... |
11:25.15 | DrukenHME | it's whatever you want... realtime is just a choice... an option |
11:25.37 | SoftIce | DrukenHME: yes but what im asking is can just sip sit in realtime |
11:25.40 | nemski | it's all relative |
11:25.43 | SoftIce | or if i wanted sip would i need the extentions too ? |
11:25.55 | SoftIce | all i want is sip.conf |
11:26.02 | SoftIce | can that be done by itself, yes/no ? |
11:30.28 | DrukenHME | yes/no |
11:32.50 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-208647.home.otenet.gr) |
11:42.22 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
11:47.21 | *** join/#asterisk grEvenX (n=even@ti500720a080-4354.bb.online.no) |
11:51.08 | *** join/#asterisk darkmug (n=dennis@143.106.7.170) |
11:56.46 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
12:02.08 | uski | assuming that my SIP provider supports emitting more than one call simultaneously, is one context enough in SIP.conf, or should i specify one context per outgoing slot ? |
12:03.17 | *** join/#asterisk sebadel_ (n=seb@162.36-246-81.adsl-static.isp.belgacom.be) |
12:04.31 | *** part/#asterisk sebadel_ (n=seb@162.36-246-81.adsl-static.isp.belgacom.be) |
12:04.39 | *** join/#asterisk sebadel_ (n=seb@162.36-246-81.adsl-static.isp.belgacom.be) |
12:05.34 | *** join/#asterisk frigidzephyr (i=frigidze@nat/digium/x-9e33bd3ecc6c6953) |
12:05.53 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
12:07.41 | DrukenHME | uski: one context is fine... |
12:08.29 | sebadel_ | hello, I have a brand new AADK but the webinterface is very unstable |
12:08.48 | sebadel_ | I'd like to know how to save my config via the SSH console |
12:08.48 | DrukenHME | wtf is aadk ? |
12:08.59 | sebadel_ | Asterisk appliance Develompment Kit |
12:09.11 | sebadel_ | save_config is hanginf |
12:09.14 | sebadel_ | hanging |
12:09.37 | sebadel_ | is there any better web interface for it ? |
12:09.42 | uski | so... "SIP/xxx is circuit-busy" means that my operator refuses to send another call, right ? |
12:09.44 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
12:09.51 | DrukenHME | oh... the nice sexy machine that's way over priced.... |
12:09.57 | sebadel_ | indeed |
12:10.32 | DrukenHME | sebadel_: i know nothing about it, except that tried to get me to buy one... and i wasen't havin no part of it |
12:10.46 | DrukenHME | uski: or the line your calling is busy... |
12:11.11 | sebadel_ | I had t buy one for a customer |
12:11.22 | sebadel_ | and it doesn't work as expected |
12:11.31 | sebadel_ | it's supposed to be out-of-the-box |
12:11.35 | DrukenHME | does anything really work as expected? |
12:11.37 | uski | no it isn't, this happens immediatly after a Dial :( so they don't want me to send another call, sh** (thx DrukenHME ;)) |
12:11.56 | sebadel_ | not really |
12:12.12 | Geert | I've got a call (in show channels) |
12:12.17 | sebadel_ | I get the web interface, the SSH prompt but I can't save anything |
12:12.24 | Geert | which is stuck, how do I stop/kill it? |
12:12.33 | DrukenHME | geert, restart |
12:12.35 | sebadel_ | and the webinterface crashes every time I change anything |
12:12.44 | Geert | not an option, there are 23 current calls :p |
12:12.54 | DrukenHME | restart when convient?? :) |
12:13.16 | Geert | oh, that'll work :p |
12:13.18 | *** join/#asterisk jovannotti (n=jovannot@190.84.99.36) |
12:13.49 | jovannotti | someone has tested TC400B with G723 succesfully ? |
12:14.19 | Geert | DrukenHME: doesn't work, the call hangs so asterisk won't restart :) |
12:14.29 | DrukenHME | sebadel_: you'd probably be better to go into the ast-dev or call asterisk tech over that thing?? |
12:14.33 | *** join/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
12:14.44 | DrukenHME | oh yes... too true... |
12:16.17 | DrukenHME | Geert: is it a zap interface? |
12:16.31 | Geert | yes |
12:17.42 | [[blah]asfd | got a new error message today: |
12:17.42 | [[blah]asfd | Connected to Asterisk <Version Unknown> currently running on No more connections allowed |
12:17.42 | [[blah]asfd | <PROTECTED> |
12:17.42 | [[blah]asfd | No more connections allowed |
12:17.42 | [[blah]asfd | *CLI> |
12:17.48 | jovannotti | I have tested TC400B, it only works for me in G729, in G723 only in one direction, could someone help me please |
12:17.56 | [[blah]asfd | anyone know what happened |
12:18.00 | [[blah]asfd | ? |
12:18.08 | DrukenHME | Geert: zap destroy channel work ? |
12:18.16 | *** join/#asterisk ctaloi (n=ctaloi@pool-72-90-82-84.syrcny.fios.verizon.net) |
12:19.37 | [TK]D-Fender | jovannotti: Virtually noone here has a TC400. Go call Digium support for it. |
12:19.44 | Geert | DrukenHME: I just did "restart now" |
12:20.00 | DrukenHME | that'll work :) |
12:20.44 | jovannotti | thanks Fender, I am trying to tall with them |
12:25.54 | JT | talk? |
12:26.43 | jovannotti | talk, sorry, but since yesterday I am trygin to call but anyone answer the phone |
12:27.01 | JT | keep in mind US business hours :) |
12:27.05 | JT | it was the weekend |
12:28.04 | frigidzephyr | they are open 7:00AM to 7:00PM , CST |
12:28.19 | frigidzephyr | or CDT right now i think |
12:29.52 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-208647.home.otenet.gr) |
12:30.01 | jovannotti | let me try right now , thanks |
12:30.11 | *** join/#asterisk ecze (n=ecze@eczema.ecze.com) |
12:34.17 | jovannotti | this is the address |
12:34.17 | jovannotti | Digium, Inc. |
12:34.17 | jovannotti | 150 West Park Loop, Suite 100 |
12:34.17 | jovannotti | Huntsville, AL 35806 |
12:34.17 | jovannotti | United States |
12:34.47 | jovannotti | what rime do you think is ther right now ? because I am in the line , waiting for assistance ,,, some minugtes ago |
12:35.34 | nasls_lsa | do you know any good vo-ip provider in Greece ? |
12:35.49 | nasls_lsa | on in Europe .. |
12:36.25 | florz | well, in .de, there are quite a few "acceptable" ones ... |
12:37.03 | florz | have a look at the voip-info.org wiki |
12:38.31 | frigidzephyr | i am in huntsville where digium is, its 7:37 AM |
12:38.32 | nasls_lsa | thanks |
12:38.32 | ecze | re |
12:38.32 | nasls_lsa | nai ? |
12:42.32 | jovannotti | ok, nobody pickup the phone, I am waiting for aprox 10 minutes on line :( |
12:42.43 | ecze | Yesterday I was talking about a bug in the old deprecated chan_modem channel. Finaly I have posted the bug in the bug systems from digium but right now I need to apply the best solution for testing purpose... |
12:51.15 | Uatec_ | jovannotti, it's 7.50 abouts i think |
12:51.20 | Uatec_ | maybe earlier |
12:52.06 | Uatec_ | i checked |
12:52.15 | Uatec_ | it's 0751 hours there |
12:53.16 | JT | maybe he doesn't understand |
12:53.16 | JT | there's probably 1 person there |
12:53.18 | JT | :P |
12:53.22 | Uatec_ | and they start at 7am |
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12:57.18 | jovannotti | ok ok I'll try later, I already posted in digium support he problem too |
12:57.49 | mosty | stay on the phone |
12:58.09 | mosty | i've emailed digium's support a number of times, and never received any response |
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13:05.13 | uski | anyone using the "failann" macro to say the reason of a failed call ? it seems that my version of asterisk (1.2) doesn't have this macro builtin |
13:05.22 | uski | as a lot of people are using it i assume it's builtin |
13:05.34 | blitzrage | uski: nothing is "built in" |
13:05.39 | blitzrage | you have to create your dialplan |
13:06.20 | uski | yea ok, thanks |
13:06.29 | blitzrage | you may be able to find a copy of that macro somewhere on the Internet |
13:07.18 | uski | i tried, everyone uses it, and no one seems to have it, thus my question ;) |
13:07.27 | blitzrage | who is everyone? |
13:07.29 | blitzrage | I've never heard of it |
13:07.29 | uski | i bet that they all copied the line from somewhere and they didn't test |
13:07.34 | uski | well at least 20 people |
13:07.42 | blitzrage | from where? |
13:07.57 | blitzrage | sounds like something that probably came from trixbox or something |
13:08.04 | uski | ah yea that's possible |
13:08.23 | [TK]D-Fender | ~trixbox |
13:08.29 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums at http://www.trixbox.org/modules/newbb/ |
13:08.29 | blitzrage | its the most probable |
13:09.10 | uski | ok thanks, i'll try to see if i can find that macro, else i'll create it myself with several Goto and Playback |
13:09.26 | uski | BTW, the link provided by jbot is out of date (404) |
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13:19.23 | [TK]D-Fender | ~trixbox |
13:19.25 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums at http://www.trixbox.org/modules/newbb/ |
13:19.36 | [TK]D-Fender | hrm |
13:19.57 | [TK]D-Fender | ~trixbox |
13:19.59 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
13:20.01 | [TK]D-Fender | There |
13:22.02 | JT | [TK]D-Fender: you had to allows the unclean presence of trixbox.org grace your pc? :P |
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13:23.00 | [TK]D-Fender | JT : No, I jsut fix the bot messages on where to go for stuff so we can more efficiently dispatch trolls. |
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13:23.45 | JT | heh fair enough |
13:24.21 | *** join/#asterisk nettie (n=nettie@ns.coolgadgets.it) |
13:25.04 | [TK]D-Fender | JT : And I have considered getting a spare PC to test things like Trixbox / * GUI, etc on that is not my home server for the purpose of perhaps being able to support them (business is business). |
13:25.38 | JT | wglwat ;_ |
13:25.41 | JT | ;) |
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13:26.37 | nettie | hi guys, what's the most common sip users / extensions dialplan naming strategy please? how are you naming sip phones, users and extensions? you just user numerical values or there's a known/suggested "sanity checked" :) strategy please? |
13:26.39 | [TK]D-Fender | JT : Yeah, you know you won't see ME around here asking for help with it :) |
13:26.54 | JT | heh |
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13:28.12 | blitzrage | nettie: In my vPBX environment I just name it [username#vpbx] |
13:28.39 | blitzrage | nettie: don't use extension numbers though -- you could use MAC addresses, or some unique username, or a combination of things |
13:28.58 | nettie | so, phones mac address as username? |
13:29.02 | nettie | in sip.conf |
13:29.13 | blitzrage | it really depends on what you're doing and what makes sense... there isn't really any official way of naming your extensions |
13:29.24 | nettie | then variables with name of the users in extensions.conf and related mac address |
13:29.38 | nettie | and then real extension pointing to the variables? |
13:29.38 | blitzrage | nettie: sure, you could... I personally don't because I find a mac address conveys very little information when looking at it |
13:29.47 | blitzrage | nettie: sure, you got the idea |
13:30.13 | blitzrage | as long as it's a unique value, that's pretty much all that matters |
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13:32.22 | defsdoor | anyone use aastra phones know how to stop it recording missed calls for group rang calls ? |
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13:33.40 | DrukenHME | i use aastra phones, but don't record anything... |
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13:34.38 | defsdoor | it displayed n missed calls on the lcd display |
13:34.47 | DrukenHME | or... you mean the caller id... |
13:34.54 | defsdoor | I'd rather it didn't if the calls was a group calls |
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13:35.19 | Muhadib | I need help with echo-cancellation issues... Is there anyone that could help me? please... |
13:35.35 | [TK]D-Fender | defsdoor: You can't Period. It doesn't know anything about calls it receives. SIP phones are smart & don't assume anything about the servers sending them calls. |
13:35.36 | mosty | defsdoor, the phone probably can't tell the difference between "group" calls and singlular calls |
13:35.42 | DrukenHME | defsdoor: i don't think that can be accomplished |
13:35.53 | defsdoor | I thought so :( |
13:36.23 | defsdoor | I'll take it off the display altogether if I can |
13:36.33 | defsdoor | everyone has voicemail so important missed calls will be there |
13:37.09 | DrukenHME | ewhh... no caller id.... i don't like how you think... |
13:37.30 | *** part/#asterisk LouieDog (n=louiedog@pool-71-164-37-158.chrlwv.east.verizon.net) |
13:37.40 | defsdoor | DrukenHME: not cancel cli |
13:37.49 | defsdoor | DrukenHME: just dont display missed calls on the lcd |
13:38.08 | DrukenHME | oh, well that's a phone option.... |
13:38.26 | DrukenHME | you can probably fix that in the cfg files |
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13:41.36 | blitzrage | SwK: oh no you di-ant! |
13:41.48 | SwK | ? |
13:42.09 | SwK | i di-ant whut? |
13:42.38 | blitzrage | oh you know |
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13:46.21 | elriah | Hi all. Anyone using (successfully) the Cisco 7941/7961's behind NAT going to a public Asterisk box? |
13:47.17 | blitzrage | elriah: yep -- all I did was set the NAT stuff on the phone, and it worked |
13:47.28 | elriah | Did you have to make any firewall changes? |
13:47.31 | blitzrage | nope |
13:47.41 | elriah | That's on the 7941's, right? Not the 7940? |
13:47.49 | blitzrage | 7960 actually |
13:47.55 | nasls_lsa | I am little bit confused about mISDN and dial plan ... how to configure msns , and which line rings to a phone .. |
13:47.55 | elriah | Yea, different firmware. |
13:48.03 | blitzrage | oh yah -- uses the java stuff right? |
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13:48.12 | elriah | The 79x1's use different firmware than the 79x0's. (sigh) thanks anyway. |
13:48.17 | blitzrage | I have a 7970 here, but couldn't get it to register anywhere but locally |
13:48.33 | mosty | nasls_lsa, msns are configured in the misdn.conf from memory |
13:48.46 | blitzrage | but I gave up on it because it was taking too much of my time trying to get it to run |
13:48.47 | elriah | Our 7940's work great. But we bought 25 7941's and have to use them via a VPN. |
13:48.51 | nasls_lsa | and how do I get them to my dialplan ? |
13:48.52 | blitzrage | I should really just sell it... |
13:49.19 | elriah | blitzrage: Your probably like us, the phone is so cool you don't want to get rid of it. Plus the quality is top-notch. |
13:49.19 | blitzrage | anyone want a 7970? :) |
13:49.41 | blitzrage | elriah: indeed -- and I love my 7960, except it stopped taking incoming calls for some reason... |
13:50.00 | blitzrage | all the other phones on my network are fine though |
13:52.14 | nasls_lsa | [studio] ports=3 context=studio msns=210515 with that will I take in my dialplan a call doing studio,1,Dial(SIP/110) ? |
13:52.30 | elriah | Anyone using 1.4.2 w/asterisk realtime in production? If so, what's the results? Any major issues? |
13:53.21 | mosty | nasls_lsa, no. do you know what a context is in extensions.conf? |
13:53.28 | nasls_lsa | elriah: I am setting it up , works very good at the moment , and saw my isdn card and worked fine .. I have some problems with the configurations ( dial plan - misdn ) but that is my problem :) |
13:53.35 | blitzrage | elriah: I am -- works great |
13:54.10 | blitzrage | elriah: I load tested a 2x quad-core Xeon box and got at least 500 calls with media... with like... < 20% CPU usage |
13:54.26 | elriah | Cool. Looks like it's time to consider upgrading from 1.2.16 ... |
13:54.29 | blitzrage | using realtime, func_odbc, lots of stuff |
13:54.40 | blitzrage | I found a lot of bugs and had them fixed in my testing |
13:54.57 | blitzrage | so you're welcome :D |
13:55.13 | elriah | Hey, thanks! |
13:55.14 | elriah | ;) |
13:55.30 | blitzrage | make sure you setup a separate development platform and test first |
13:55.36 | elriah | Always... |
13:55.41 | blitzrage | don't just upgrade your production server, or you're gonna cry :) |
13:55.48 | nasls_lsa | mosty: yes .. |
13:56.01 | elriah | lol, we process about 18k calls a month, I wouldn't dare... |
13:56.31 | mosty | nasls_lsa, well context=foo in misdn.conf refers to a context in extensions.conf, not an extension |
13:57.55 | nasls_lsa | aah ! mISDN/studio ?!?!?!? |
13:58.18 | GreyFoxx | elriah: hehe we're paranoid too. We're hovering around 78k a month so upgrades are very strictly planned :) |
13:59.17 | nasls_lsa | mosty: I quit ... out of ideas |
13:59.31 | GreyFoxx | they want me to setup Openser infront of the asterisk boxes now, partly so I can redirect traffic away from a box getting updates/maintenance |
14:03.00 | blitzrage | GreyFoxx: that's what I just setup -- OpenSER is the registration point, and it can just randomly drop calls to any Asterisk box in the cluster, and its all good |
14:03.13 | blitzrage | practically have a self-healing network now :) |
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14:09.21 | GreyFoxx | blitzrage: Yeah, that's pretty much what they want me to setu. Authenticating to our existing SQL database as much as possible. I just need the time to sit down and really look over the openser documentation |
14:10.51 | blitzrage | GreyFoxx: yah... its a pain in the butt :) |
14:11.10 | blitzrage | I setup views in our pgsql table to do the authentication and subscriber tables, etc... |
14:11.17 | blitzrage | so Asterisk and SER both know about the same users |
14:11.38 | blitzrage | then the GUI can configure it, change something, and both SER and Asterisk know about it at the same time, with no reload scripts or anything |
14:19.50 | GreyFoxx | That's basically what I planned to do using views. |
14:20.35 | GreyFoxx | Basically a view for the voicemail and sip users |
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14:22.57 | blitzrage | GreyFoxx: yeppers, it works very well |
14:27.08 | nettie | uhmm blitzrage I keep getting extension XXXXXXXX in context 'bri' does not exist. Rejecting call on channel 0/1, span 2 |
14:27.13 | nettie | where the extension is there |
14:27.17 | nettie | I quadchecked eheh |
14:27.33 | nettie | any idea what coul dbe the problem please? |
14:27.49 | blitzrage | sounds like you don't have a pattern match that matches the number coming in |
14:27.52 | [TK]D-Fender | nettie: Your dialplan does NOT match. It does not lie about this. Go pastebint he whole mess including CLI output so we can find out what you di wrong. |
14:27.54 | blitzrage | or the bri context doesn't exist |
14:28.06 | [TK]D-Fender | ~pb |
14:28.10 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
14:28.15 | nettie | it's pretty basic |
14:28.21 | nettie | will pastebin |
14:28.21 | blitzrage | still have a bug though |
14:28.27 | [TK]D-Fender | nettie: PASTEBIN |
14:28.35 | nettie | eheh sure |
14:29.13 | [TK]D-Fender | nettie: its not that we don't trust you..... actually... yeah that really does sum it up well.... |
14:29.16 | [TK]D-Fender | PASTEBIN! ;) |
14:29.22 | nettie | http://pastebin.ca/454283 |
14:29.23 | nettie | :) |
14:29.33 | nettie | I'm sure I' missing something |
14:29.44 | blitzrage | I'm sure you are too :) |
14:29.47 | blitzrage | Asterisk is positive of it |
14:31.09 | [TK]D-Fender | [zero_local} |
14:31.42 | [TK]D-Fender | fix your braces, and do a reload to make sure what you're showing us is actually in effect |
14:32.37 | DrukenHME | [TK]D-Fender: are you being your typical self today ? |
14:33.01 | [TK]D-Fender | DrukenHME: Think so... |
14:33.10 | *** join/#asterisk denzs (n=denzs@carbon.gonicus.de) |
14:33.25 | [TK]D-Fender | DrukenHME: Yup.. chan_bile.so is indeed loaded ;) |
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14:34.01 | Muhadib | Despite having configured in zapata.conf file "echocancel = yes" and "echotraining = yes", when I write in * CLI "zap show channel 4 it displays "echo cancellation currently off"!!! Anyone that passed the same problem???? |
14:36.01 | denzs | hi, iam trying to limit the call time and let asterisk play a warning after 10seconds with Dial(IAX/xxx/${EXTE:1}|30|ttL(20000:10000)), but after 10secs the call gets hanged up with the following message in the CLI: Apr 23 16:33:49 DEBUG[32746]: channel.c:3361 ast_generic_bridge: Nobody there, continuing... |
14:36.01 | denzs | Apr 23 16:33:49 DEBUG[32746]: channel.c:3361 ast_generic_bridge: Nobody there, continuing... |
14:36.32 | denzs | is there something to mention when using the L option? |
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14:37.21 | nettie | It's a whiargh |
14:37.24 | nettie | thanx TK !! |
14:37.26 | nettie | doh |
14:37.28 | nettie | lemme retry |
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14:40.01 | nettie | works.. |
14:40.02 | nettie | damn |
14:40.06 | nettie | eheh |
14:40.08 | nettie | typo |
14:40.41 | nettie | evil typo :) .. I was sure it was a stupid thing.. I configured a quite complex IVR a cuple of months ago with macros and stuff |
14:40.42 | nettie | eeh |
14:41.10 | [TK]D-Fender | nettie: I'm sure it was stunningly complicated :) |
14:46.36 | nettie | TK :)))))))))))0 |
14:46.58 | nettie | phear me ehehe |
14:47.41 | Muhadib | Asterisk sucks... It's echo cancellation doesn't work by far... Anyone that configured echo-cancellation properly? |
14:48.11 | GreyFoxx | We just use hardware that does echo cancellation well |
14:48.21 | GreyFoxx | You aren't using a X100P/clone are you ? |
14:48.23 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
14:48.41 | Muhadib | nops...I have a digium TDM400P |
14:49.27 | Muhadib | and I'm trying to use zaptel echo canceller |
14:49.50 | GreyFoxx | We didn't like the ones on the TE405. We found it wasn't all that great, so we have an army of Telabs 81-2572 cards |
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14:50.07 | mosty | bah, i hate when google thinks my queries look like virus/worm queries |
14:50.20 | GreyFoxx | the 2572's work very well, they onboard support 64ms, but with a daughter card will do 128ms |
14:51.14 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.91) |
14:51.16 | Muhadib | TDM400P is an analogue card with 4 fxo/fxo ports |
14:51.49 | Muhadib | I only have echo issues with analogue interfaces |
14:52.14 | GreyFoxx | We've never had any luck with the zaptel echo cancelling here with the zaptel stuff |
14:52.48 | [[blah]asfd | could anyone recommend a way to sync my linux laptop to my works MS exchange calendar? I have been using thunderbird for my email, that works just fine. |
14:53.34 | [TK]D-Fender | [[blah]asfd: Evolution |
14:53.47 | Muhadib | so which software echo-canceler do you recomend using with zaptel stuff? |
14:54.04 | [[blah]asfd | i installed and am trying that, is there something more i need to do to configure that? i dont see where to set up any server information. |
14:54.41 | [TK]D-Fender | [[blah]asfd: Sorry, that answer is about as much as you should expect here. I think you might be "lost" :) |
14:54.44 | ChkDigit | Isn't evolution-connector required to hook into exchange? |
14:54.54 | *** join/#asterisk heison (n=heison@gw-yyz1.somanetworks.com) |
14:55.50 | ChkDigit | Anyway, I have an Asterisk question: What would prevent VoiceMail() from receiving 0 or * when the user presses them? |
14:55.59 | GreyFoxx | Muhadib: I don't recommend any, and I've never seen any t6hat worked all that well myself. ymmv |
14:56.05 | GreyFoxx | ooh lunch time |
14:57.27 | mosty | ChkDigit, are both directions of audio actually working? |
14:57.50 | Muhadib | thanks Greyfoxx |
14:58.14 | ChkDigit | mosty: I'd assume so. Directory, VoiceMailMain, and other apps work. |
14:58.21 | [[blah]asfd | [TK]D-Fender: dammit... habit, i meant to post that in #linux ;-) |
14:59.10 | [[blah]asfd | join /#linux |
14:59.12 | [[blah]asfd | errrrrrr |
14:59.18 | [[blah]asfd | leaving now, sorry all. |
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15:03.38 | [TK]D-Fender | ChkDigit: you have to configure your mailbox's to accept "0", and need to have extens to support "*" and "0" |
15:03.49 | paavum | hello, I am using asterisk GUI, however when I try to log in I get a "404 Not found" error |
15:04.11 | paavum | I've tried to look in the logs but I cant see any error there |
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15:04.12 | *** mode/#asterisk [+o mog] by ChanServ |
15:04.12 | paavum | :s |
15:04.37 | tsurko | paavum, check the topic:) |
15:04.38 | paavum | can anybody gimme a hand? |
15:05.08 | paavum | They are Fun Channels, not support channels :P |
15:05.10 | paavum | ;) |
15:05.13 | *** join/#asterisk karlhaines (n=karl@209.12.254.71) |
15:05.56 | [TK]D-Fender | paavum: GUI's are not supported here. |
15:06.01 | [TK]D-Fender | paavum: ANY of them. |
15:06.58 | paavum | one final q... can I get spanfax to wrk with * 1.4? |
15:07.06 | *** join/#asterisk astawerksdotcom (n=astawerk@cpe-75-179-164-7.woh.res.rr.com) |
15:07.21 | [TK]D-Fender | paavum: Never heard of spanfax.... |
15:07.28 | *** join/#asterisk akitogo (n=chatzill@213.221.85.2) |
15:07.34 | paavum | its spandsp's fax thingy |
15:07.46 | [TK]D-Fender | paavum: get your names right :) |
15:08.06 | [TK]D-Fender | paavum: And not sure... I've seen some talk about real problems with 1.4 |
15:08.13 | tsurko | does anybody have experience with softphones on thin clients? |
15:08.24 | paavum | tsurko... I'm trying to get that working |
15:08.25 | [TK]D-Fender | paavum: I think IAXModem + Hylafax is working properly un 1.4 though |
15:08.42 | astawerksdotcom | i got xlite to work on a wyse box before |
15:08.48 | tsurko | paavum, what terminals are you using? |
15:09.02 | paavum | tsurko... old pentium II/III |
15:09.16 | tsurko | i mean the software part:) |
15:09.24 | paavum | ltsp |
15:10.05 | tsurko | paavum, me too:) |
15:10.09 | akitogo | Hi, anybody here who succeeded to installed a digium B410p with asteriskNow? |
15:10.16 | tsurko | and how it's going? |
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15:11.06 | paavum | tsurko... I worked on it untill last week, this week I'm trying to get asterisk now in virtual machines |
15:11.18 | paavum | tsurko... there are a couple of howtos around |
15:11.23 | tsurko | paavum, I'm trying to do something similar |
15:11.45 | tsurko | but I have some porblems with the softphones - the sound is a little mettalic |
15:12.08 | tsurko | Why you're installing * on virtual machines? |
15:12.27 | paavum | cuz I need 4 independent installations for 4 different companies |
15:12.42 | paavum | and I dont wanna use 4 servers :P |
15:12.53 | tsurko | I see |
15:13.18 | tsurko | it's just for testing purposes I suppose? |
15:13.21 | paavum | [TK]D-Fender --> cant you help me with the GUI thing? seems like nobody's around in the other channels |
15:13.34 | [TK]D-Fender | paavum: No, I can't |
15:13.38 | paavum | :( |
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15:14.29 | putzz | GUI = for Noobs that dont want to learn. LOL |
15:14.49 | paavum | putzz... GUI makes things quicker |
15:15.05 | [TK]D-Fender | paavum: I can mount entire sysems in the time you've asked about that ;) |
15:15.35 | paavum | and what makes it hurt is that its true |
15:16.06 | paavum | one more thing... asterisknow does not support mysql cdr? |
15:16.47 | putzz | I dont think u really understand that GUI is not supported here |
15:17.01 | paavum | I'm not speaking of guis now ... |
15:17.24 | paavum | I just wanna know if I have to compile the damn thing myself |
15:17.36 | putzz | well asterisknow is what? |
15:17.49 | paavum | a distro |
15:17.54 | paavum | asterisgui is a gui |
15:18.12 | paavum | ^^ |
15:18.57 | [TK]D-Fender | paavum: Again, this is not the pace to ask about either. |
15:19.37 | paavum | sigh |
15:21.06 | *** join/#asterisk naitram (n=ttech@216.77.58.40) |
15:21.53 | naitram | how do you force asterisk to unregister a sip client |
15:22.51 | [TK]D-Fender | naitram: typcailly if you comment out the register & reload your sip config I believe * will send the unregister |
15:23.48 | naitram | ok, thkns |
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15:25.40 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
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15:28.54 | uwe | hello, i have asterisk and sip clients connected to it, and when i do sip show peers the status shows time ~150 ms , this is a lot i think ... but when i ping the clients i get very good response time (<2 ms), i think this can be something in the network filtering non icmp traffic or something, so i was wondering if there is a way to test that time at any given point ? an application like ping but for sip :) |
15:30.34 | *** join/#asterisk agile (n=mike@63.98.55.146) |
15:30.52 | JTgiri | good morning, I hope somebody could help me with this...i want to use as my pstn gateway..i can call out from asterisk without any problem..but if i call from another sip server i get following error message |
15:30.56 | JTgiri | <PROTECTED> |
15:30.56 | JTgiri | <PROTECTED> |
15:30.56 | JTgiri | <PROTECTED> |
15:30.56 | JTgiri | <PROTECTED> |
15:30.56 | JTgiri | <PROTECTED> |
15:30.58 | JTgiri | <PROTECTED> |
15:30.59 | mosty | uwe: is it a problem when you call? |
15:31.00 | JTgiri | <PROTECTED> |
15:31.11 | putzz | omg |
15:31.15 | putzz | ~pb |
15:31.25 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
15:31.48 | JTgiri | ok sorry |
15:32.54 | uwe | mosty, i have not very good sound qulity and sometimes the sound skips a little |
15:35.55 | [TK]D-Fender | JTgiri: Looks like the call is coming in just fine. Its your dialplan, go make it do what you want. |
15:36.06 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
15:36.12 | mosty | uwe: you will need to be more descriptive, what do you mean by "not very good sound quality" ? |
15:38.44 | uwe | i mean the sound is slightly inturrupted |
15:43.25 | nettie | <PROTECTED> |
15:44.15 | joe--f | hey, what do you guys think of voxbone for voip forwarding? |
15:44.57 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
15:48.45 | *** join/#asterisk Mad|Cow (n=madcow@000-202-109.area3.spcsdns.net) |
15:49.32 | Mad|Cow | Does anyone have a an examples on how to match international numbers? I'm in the US and having issues dialing the UK and other countries..... |
15:49.56 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
15:50.48 | mosty | what do you want to match these numbers for? |
15:51.17 | Kigh | Mad|Cow: exten => _00XXXX.,s,NoOp(${EXTEN:2:4} is the country prefix) |
15:51.27 | Mad|Cow | mosty: in my extensions.conf file... so I can dial international |
15:51.33 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-9bcafbb44472e732) |
15:52.05 | Kigh | you need to dial 00XXYYYYYYYYY .. where XX is the country prefix an YYYY the parties number, without leading zero |
15:52.11 | Mad|Cow | Kigh: I dont need 011? |
15:52.18 | Kigh | whats 011 |
15:52.41 | Mercestes | Prefix for an international call in the US |
15:52.42 | Mad|Cow | Kigh: I thought 011 was how you dialed international |
15:52.51 | Kigh | im from germany .. i.E. a german number in internation format: 00491511 where 1511 is the city |
15:53.09 | Kigh | Mad|Cow: urm dunno what it is in your country, in europe that "00" |
15:53.17 | Kigh | *thats |
15:53.32 | putzz | us to UK is: 01144 |
15:53.50 | putzz | *U.S to UK is: 01144 |
15:53.55 | Kigh | then you need to match exten => _011XXX.,s,NoOp() |
15:54.38 | *** join/#asterisk Exstatica (i=exstatic@redline.mednor.net) |
15:54.39 | Kigh | didnt know its 011 in U.S., i've never been there sorry |
15:54.55 | Mad|Cow | Kigh: would I still need the ${EXTEN:2:4} in your example above? |
15:55.33 | putzz | no you wont need it |
15:55.59 | Kigh | Mad|Cow: no you will just dial(${EXTEN}) on a normal line. |
15:56.21 | jm|laptop | ~enum |
15:56.31 | jbot | somebody said enum was http://www.voip-info.org/wiki-Enum |
15:56.40 | mvanbaak | enum rox ! |
15:56.47 | Kigh | ack |
15:56.57 | putzz | ew |
15:56.59 | mosty | Mercestes, it depends what your upstream provider requires |
15:57.11 | jm|laptop | mvanbaak: am I right in thinking that I register my phone number in reverse or something and I get calls routed via IP where possible? |
15:57.15 | Mad|Cow | got it... thanks guys... I have to run but I'll try it in a bit |
15:57.17 | Mad|Cow | Thanks again! |
15:57.40 | mosty | er, stupid nick-completion |
15:57.45 | mvanbaak | jm|laptop: something like that yeah. You put your phone number in reverse and you can add different records to it |
15:57.51 | jm|laptop | mvanbaak: weird. What stops me putting our competitors numbers in the directory so that anyone trying to contact them will get patched to MY voip?! |
15:58.04 | mvanbaak | like: preference 10: call IAX2/jm@my.voice.box |
15:58.16 | mvanbaak | pref20: call SIP/jm@my.voice.box |
15:58.30 | mvanbaak | jm|laptop: because of validation calls and stuff |
15:58.35 | jm|laptop | oh |
15:58.37 | jm|laptop | by whom? |
15:58.44 | jm|laptop | paid administrators? |
15:58.49 | mvanbaak | the enum registry |
15:58.50 | mvanbaak | yeah |
15:58.56 | jm|laptop | still sounds a little dodgy |
15:59.00 | jm|laptop | I might try it; all the same |
15:59.03 | mvanbaak | every country has a delegation |
15:59.04 | jm|laptop | ~dundi |
15:59.06 | jbot | extra, extra, read all about it, dundi is http://www.dundi.com |
15:59.21 | mvanbaak | the delegation is based on country code |
15:59.22 | jm|laptop | oh dundi is something different |
15:59.22 | blitzrage | dundi is pretty sweet |
15:59.32 | mvanbaak | dundy is very sweet :) |
15:59.40 | jm|laptop | least cost routing, right? |
15:59.51 | blitzrage | you can use it for that, ya |
15:59.52 | mvanbaak | jm|laptop: here in .nl you have to give your landline nr for enum |
15:59.55 | jm|laptop | often free when the local calls are for that local code |
16:00.00 | blitzrage | dundi is how you pass information between two Asterisk boxes |
16:00.04 | mvanbaak | that number gets a call and reads a pin |
16:00.10 | mvanbaak | you have to provide that pin |
16:00.23 | mvanbaak | that will give you access to the enum records of your phone nr |
16:00.29 | mvanbaak | that way you can attach info to it |
16:00.37 | jm|laptop | hmm |
16:00.40 | jm|laptop | I'll read up later |
16:00.48 | jm|laptop | you understand my concern, though? |
16:00.52 | jm|laptop | re: lying |
16:00.55 | mvanbaak | if the enum nr is already taken (think, I try to get my neighbours nr) they gonna verify it manually |
16:00.58 | mvanbaak | uhhuh |
16:01.02 | mvanbaak | I do |
16:01.09 | mvanbaak | but the risk is the same as with domain names |
16:01.21 | jm|laptop | I suppose |
16:01.26 | mvanbaak | brb, going to visit neighbour |
16:01.28 | mvanbaak | latero |
16:01.32 | paavum | tsurko ... |
16:01.34 | paavum | u there? |
16:01.56 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
16:03.12 | *** part/#asterisk frigidzephyr (i=frigidze@nat/digium/x-9e33bd3ecc6c6953) |
16:04.10 | sivana | does anyone know of a way to check to see if a channel is in use (ie: SIP/chan1234) |
16:04.20 | sivana | using 1.4 |
16:04.43 | sivana | IsChanAvail is broke it seems |
16:04.47 | *** part/#asterisk deoptima (n=deoptima@c-71-228-222-87.hsd1.tn.comcast.net) |
16:04.51 | *** join/#asterisk ToyMan (n=Stuart@ool-45784fde.dyn.optonline.net) |
16:05.14 | [TK]D-Fender | sivana: Works for me... show how you're using it. |
16:06.12 | sivana | http://www.pastebin.ca/454394 |
16:08.01 | brad_mssw | i just upgraded from asterisk 1.2 to 1.4.2 ... I take it using jbforce=yes in sip.conf is not recommended? on incoming calls, it appears to cut the channel off in about 3 seconds with a critical jb error ... though I can get some calls through, randomly it seems. Also, setting jbimpl=adaptive appears to not work at all .. i get no sound from either end |
16:08.59 | brad_mssw | just wondering if having jb enabled is worth it at all |
16:09.59 | brad_mssw | ... that said, I should have kept the log from when this occurred, i had to disable it real fast because it was on my office's production phone system, so i can't give the exact errors until after business hours |
16:11.11 | mosty | brad_mssw, 1.4 isn't really ready for production use yet, in my experience |
16:12.26 | brad_mssw | hmm, nice |
16:12.51 | brad_mssw | spent a couple hours trying to port my extensions.ael over ... thought that'd be all there was to it |
16:14.24 | mosty | if you stick with 1.4 now, expect a few things to be broken :/ |
16:15.26 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
16:15.58 | brad_mssw | i was mainly upgrading to get access to the jitterbuffer as we have some quality issues from time to time |
16:17.25 | *** join/#asterisk StealthHe (n=StealthH@151.204.60.4) |
16:17.47 | *** join/#asterisk rg1_ (n=rg1@www.airlinksystems.com) |
16:17.51 | sivana | lots of core dumps in 1.4 |
16:18.43 | rg1_ | question: does anyone know if there is a way when asterisk receives a call, if the caller-id that is presented is "spoofed" from someone? |
16:18.57 | brad_mssw | i'm not running a high-load system at all, max is usually 3-4 active calls ... hopefully i won't have too many issues |
16:20.46 | brad_mssw | actually, found something on bugs.digium.com that looks to be the exact error message I was getting: [Apr 12 09:26:32] WARNING[26247]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission a11ce-10262-461e3382-1cff-216.109.205.115 for seqno 7771 (Critical Response) |
16:21.02 | brad_mssw | with jbforce=yes ... |
16:21.13 | brad_mssw | it would hang up after that |
16:21.37 | mosty | well that warning message appears in other situations too |
16:22.44 | StealthHe | Is anyone aware of the existence of any helpful documentation for incorporating Asterisk voicemail into an existing Cisco CM architecture? |
16:22.50 | rg1_ | anyone know anything about caller-id's out there? Specificially, about the caller-id of incoming calls |
16:23.23 | mosty | rg1_, you can set the callerid that local extensions see |
16:23.43 | brad_mssw | rg1_: as far as I know, there's no way to tell if the callerid was spoofed for not ... |
16:23.51 | rg1_ | incoming calls that hit the PRI trunk line |
16:24.14 | mosty | rg1_, what are you trying to do exactly? |
16:24.19 | rg1_ | brad_mssw-thats what i thought too, but when i spoofed an out-going call to my bank to have them think i was calling from my house, they could tell |
16:24.45 | rg1_ | yet, when i call a "normal" phone, it looks like the # I am spoofing |
16:25.09 | rg1_ | so I'm wondering if there is another variable that the switch is getting (NI2) that might be available on asterisk to check |
16:25.22 | brad_mssw | rg1_: was the number you were spoofing a legitimate number? |
16:25.33 | mosty | rg1_, no this is most likely done by your service provider |
16:25.48 | rg1_ | yes |
16:25.52 | rg1_ | legitimate |
16:26.07 | rg1_ | mosty - don't understand your comment |
16:26.15 | [TK]D-Fender | sivana: You are calling it wrong. |
16:26.36 | rg1_ | XO is my service provider and I can make calls with the callier-id/name I set in asterisk |
16:26.56 | mosty | rg1_, your service provider can pass on the callerid you provide them, but it's their option wether to trust it or not |
16:27.05 | rg1_ | same provider sends both calls (XO) - 1) to normal phone customer looks like my home# |
16:27.17 | rg1_ | mosty, i understand that, but they could tell it was NOT my home phone |
16:27.38 | rg1_ | so they must be able to see the actual phone# (at least that's what i'm thinking) |
16:27.51 | mosty | plus, big businesses and government agencies get more callerid data than regular telephone service |
16:27.53 | rg1_ | maybe a more sophisticated card/switch gets both numbers? |
16:28.00 | brad_mssw | rg1_: guarantee your provider doesn't allow you to set the name ... at least not in the US |
16:28.03 | rg1_ | yeah, thats what I was thinking |
16:28.11 | brad_mssw | rg1_: it may populate the name calling to another phone on their own network, but not across networks |
16:28.15 | rg1_ | brad_mssw - true |
16:28.40 | rg1_ | I think mosty's latest comment must be correct |
16:28.58 | *** join/#asterisk FreezeS (n=bla@82.208.157.125) |
16:29.11 | rg1_ | I'm wondering what kind of switch/card/service can tell that, because for another application I'm doing I really need to KNOW that the caller-id is 100% from that phone# |
16:29.53 | mosty | rg1_, talk to your service provider. you can probably only get what you want by forking out a lot more money for your service |
16:30.03 | rg1_ | right |
16:30.23 | mosty | as an alternative, you can setup a callback service |
16:30.39 | rg1_ | right, thats what we're doing - but outgoing costs us call charges |
16:30.45 | rg1_ | so we're trying to avoid that if possible |
16:30.54 | rg1_ | and we don't want to have to ask for passwords |
16:30.57 | mosty | depends how secure you want to be |
16:31.00 | rg1_ | right |
16:31.42 | StealthHe | Is there a hardware compatibility list for Asterisk? More accurately, is it compatible with a Cisco 1760 using VIC2-4FXO cards? |
16:32.14 | Qwell[] | StealthHe: wouldn't the cisco just be speaking SIP? |
16:32.28 | StealthHe | I believe so. |
16:32.45 | Qwell[] | then there isn't much for the hardware to be compatible with... |
16:33.00 | Qwell[] | asterisk doesn't know/care what hardware the cisco has |
16:33.11 | aydiosmio | I have all sorts of junk talking SIp to our AS5300 (IOS 12.2) |
16:33.24 | aydiosmio | no big deal |
16:33.26 | StealthHe | My apologies, I've pretty much been thrown into this by my company. I am just starting to research all this information, so if I seem a little less knowledgable, it's because I am at the moment. |
16:33.35 | Qwell[] | !book |
16:33.36 | Qwell[] | erm |
16:33.38 | Qwell[] | ~book |
16:33.41 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
16:33.41 | Qwell[] | ~wikis |
16:33.42 | jbot | well, wikis is http://www.voip-info.org |
16:33.46 | aydiosmio | ~consultant |
16:33.48 | jbot | Hire a consultant. |
16:33.49 | Qwell[] | StealthHe: start there |
16:33.52 | aydiosmio | hahaha |
16:33.54 | Qwell[] | ...or there |
16:33.57 | aydiosmio | nice one jbot |
16:34.36 | *** join/#asterisk jeffik (n=Valued@h-64-105-236-252.chcgilgm.covad.net) |
16:34.59 | StealthHe | I believe I have that in pdf format. Hopefully, I'll be given the time to actually get through it. |
16:35.16 | Qwell[] | option 3: quit |
16:35.20 | StealthHe | Indeed! |
16:35.45 | aydiosmio | good way to learn though |
16:35.56 | aydiosmio | thight deadline on something you know nothing about |
16:36.09 | aydiosmio | it may not be right, but damned it it doesn't work |
16:36.40 | *** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
16:37.00 | aydiosmio | just don't bother us with things that can be answered with a little bit of research |
16:38.23 | aydiosmio | and by "us" I mean these fine men and women with large repositories of asterisk knowlege |
16:38.36 | Sweeper | holy fuck is openser a nasty nasty thing to configure |
16:40.44 | StealthHe | The deadline is really the only reason I bothered. I'll refrain from asking further questions until I've had the chance to read through the book. |
16:41.09 | *** join/#asterisk jovannotti (n=saravia@190.144.48.1) |
16:41.35 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
16:42.33 | joshaidan | I have a simple question... What does PVT stand for? Such as in SIP_PVT. |
16:44.39 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
16:47.10 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
16:47.37 | puzzled | hi |
16:48.47 | jm|laptop | does enum work in 1.2 ? |
16:49.00 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
16:49.51 | jm|laptop | and anyway - surely p2p > enum ? |
16:50.18 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
16:52.24 | jovannotti | helpppppp, I need to use the TC400B card with G723. I wrote to support@digium.com and call to the phone numbers, but nobody answer to me !! |
16:54.25 | aydiosmio | damnit |
16:54.29 | *** join/#asterisk riksta (n=rick@rhamnett.plus.com) |
16:54.33 | aydiosmio | I told you to stop whining yesterday |
16:54.39 | aydiosmio | same applies today |
16:54.57 | riksta | Hi can someone tell me how i can find out about sending SMS over my E1 channel? I can't seem to find anything useful via google |
16:55.16 | ManxPower | jovannotti: when did you contact them? |
16:55.42 | jovannotti | I wrote an e-mail this morning. 5 hours ago aprox |
16:55.50 | ManxPower | riksta: did you try voip-info.org or the mailing list archives? |
16:56.01 | riksta | ManxPower, yeah, went there first! |
16:56.11 | ManxPower | jovannotti: Digium's offices opened 3 hours ago |
16:56.25 | ManxPower | Support usually takes 24 - 48 hours to respond. |
16:56.34 | ManxPower | If you want faster support buy a support contract. |
16:57.14 | riksta | ManxPower, i can only really find people wanting to sell me a sms gateway :) I want to find out how to do it myself |
16:57.22 | riksta | (if possible) |
16:59.13 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
17:01.45 | aydiosmio | riksta: what do you mean sms gateway? |
17:01.48 | aydiosmio | (you can build one on linux) |
17:02.13 | aydiosmio | heck I built one on windows |
17:02.15 | riksta | aydiosmio, i want to know how to send sms data down my E1 |
17:02.26 | riksta | i don't want to use an sms gateway from someone |
17:02.30 | aydiosmio | all you need is a cell phone with an AT interface and some software to send/recieve |
17:02.45 | riksta | no, i dont want this |
17:02.55 | aydiosmio | ...E1? |
17:02.55 | *** join/#asterisk qdk (n=qdk@193.164.155.27) |
17:02.55 | riksta | read what I am saying ;) |
17:03.12 | riksta | aydiosmio, if you don't know what an E1 is then I'm pretty sure you can't help me. |
17:03.13 | aydiosmio | I don't think you understand how SMS works |
17:03.31 | aydiosmio | I know what an E1 is, I 'm wondering why the heck you'd want to send SMS with it |
17:03.38 | paavum | riksta,... I thnk you need some kind of interface with the cellular network |
17:03.46 | paavum | you cant just send sms over e1s |
17:03.53 | ManxPower | paavum: that would be called the SMSC |
17:03.57 | aydiosmio | You can only send SMS over the internet in the form of a TCP conneciton to a SMS gateway provider |
17:04.03 | ManxPower | paavum: are you not in Europe? |
17:04.18 | ManxPower | aydiosmio: you are not in Europe either, are you? |
17:04.23 | riksta | hmmm, ok conflicting info :) please can someone clarify |
17:04.37 | riksta | I just heard it's possible to push your own SMS over an E |
17:04.38 | riksta | 1 |
17:04.45 | aydiosmio | no I'm not |
17:05.04 | ManxPower | riksta: In the USA you can only send SMS over the internet. In Europe you should be able to send an SMS from Asterisk to the SMSC of the carrier of the destination. |
17:05.04 | aydiosmio | I'm not from the land where the PSTN allows SMS sending to home phones |
17:05.12 | ManxPower | The SMSC will have a telephone number. |
17:05.29 | riksta | ok, and I imagine you need to send it to the correct network's SMSC? |
17:05.39 | ManxPower | aydiosmio: SMS in the USA and SMS in the rest of the world work totally differently |
17:05.51 | riksta | ManxPower, also, i'm looking for some documentation, on actually HOW to send this using asterisk |
17:06.12 | ManxPower | riksta: I don't believe they exist |
17:06.12 | ManxPower | there has been discusstions on the mailinglists, however. |
17:06.13 | riksta | ok , question resolved :) |
17:06.20 | riksta | yeah, i saw that one thread |
17:06.40 | aydiosmio | http://lists.digium.com/pipermail/asterisk-dev/2003-December/002425.html |
17:08.23 | riksta | yeah i read that one |
17:08.36 | riksta | confused me ;) |
17:09.01 | Qwell[] | ManxPower: discusstions? |
17:09.12 | Qwell[] | ManxPower: surely you didn't think we'd let that slide? :) |
17:09.16 | riksta | hehe |
17:09.48 | ManxPower | Us people in Alabama ain't good speelers |
17:09.55 | Qwell[] | peepol? |
17:10.50 | jm|laptop | what is the point of free164 numbers? |
17:10.55 | [TK]D-Fender | disgustion : (n) an unlgy conversation you'd rather not be having. |
17:10.59 | [TK]D-Fender | ugly* |
17:11.05 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
17:13.31 | *** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com) |
17:15.20 | mvanbaak | back |
17:17.52 | *** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
17:18.14 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
17:20.25 | sivana | [TK]D-Fender: I changed it.. added sj as the options |
17:20.38 | sivana | not sure why the syntax was different from 1.2 when it worked.. but we'll see |
17:21.10 | [TK]D-Fender | sivana: Priority jumping is even MORE of a dead item in 1.4 |
17:21.24 | [TK]D-Fender | sivana: You should not be using that means where at all avoidable. |
17:21.59 | sivana | that's the only call where I use it |
17:22.22 | sivana | but I'll take some time later and incorporate the status variable |
17:23.11 | [TK]D-Fender | sivana: Yeah, for a down & dirty 1-shot dea, what the heck.. |
17:23.14 | [TK]D-Fender | deal* |
17:23.54 | jm|laptop | hmm |
17:24.07 | jm|laptop | I should be able to receive calls in the format IAX2/foo@bar.com right? |
17:24.19 | jm|laptop | rather than IAX2/guest@bar.com/foo ? |
17:25.39 | ManxPower | jm|laptop: IAX2/foo@bar.com will send the call to the "s" extension on the destination server |
17:26.00 | jm|laptop | that's what I thought |
17:28.07 | russellb | ooh, extensions.com |
17:28.07 | jm|laptop | ah |
17:28.08 | jm|laptop | .conf lol |
17:28.08 | ManxPower | the context of "s" is whatever is the context= line from the [foo] section of iax.conf |
17:28.08 | ManxPower | The reason it goes to the "s" extension, of course, is because there is no destination extension on the Dial line. |
17:28.09 | jm|laptop | sure |
17:28.09 | ManxPower | It is a little known fact that "s" is in fact French for "no destination extension" |
17:28.09 | jm|laptop | but I can grab the foo@ ? |
17:28.19 | ManxPower | "grab"? |
17:28.30 | jm|laptop | Apr 23 18:28:14 NOTICE[9373]: chan_iax2.c:6925 socket_read: Rejected connect attempt from 204.55.81.11, who was trying to reach 's@' |
17:28.43 | jm|laptop | oh wait |
17:28.49 | ManxPower | s@ means "s" at "no context found" |
17:29.13 | jm|laptop | where did aix.conf go?! :O |
17:29.25 | Qwell[] | jm|laptop: it's still on your IBM |
17:29.26 | ManxPower | there never was an aix.conf |
17:29.32 | jm|laptop | mail:/etc/asterisk# find . | grep aix.conf |
17:29.33 | jm|laptop | mail:/etc/asterisk# |
17:29.34 | jm|laptop | :( |
17:29.55 | ManxPower | and there never will be an aix.conf in Asterisk. |
17:29.59 | Qwell[] | there might |
17:29.59 | jm|laptop | ok I should have used -name |
17:30.11 | ManxPower | perhaps you are thinking of iax.conf and for some reason lost half your brain? |
17:30.20 | Qwell[] | ManxPower: if somebody ports it to AIX, I imagine they'll have AIX specific customizations |
17:30.27 | jm|laptop | ManxPower: it's been a long day :( |
17:30.29 | ManxPower | Look under the sofa. That's where I always find mine. |
17:30.38 | ManxPower | jm|laptop: then go to bed and stop wasting our time. |
17:30.46 | jm|laptop | :O |
17:31.24 | ManxPower | I'm not joking. If you are so tired as to make such simple mistakes you will not actually accomplish anything no matter how much you try. |
17:31.52 | jm|laptop | my default context goes in [general] ? |
17:36.34 | *** join/#asterisk jmacz (n=jmacz@201.244.170.241) |
17:37.37 | jm|laptop | " ... No application 'Dail' for extension ..." |
17:37.47 | jm|laptop | ManxPower: I'm beginning to agree with you |
17:37.48 | jm|laptop | :| |
17:38.44 | ManxPower | I have been wrong before, but that was years ago. |
17:39.36 | [TK]D-Fender | (as measured in may-fly years, not human) |
17:39.38 | [TK]D-Fender | ;) |
17:40.54 | [TK]D-Fender | ~book |
17:40.56 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
17:41.40 | *** join/#asterisk ucfMethod (n=ucfmetho@office.eyestreet.com) |
17:43.38 | anonymouz666 | vars inside ${CURL must be passed as ${var} or just as var |
17:43.48 | jm|laptop | so |
17:43.55 | jm|laptop | I appear to have e164.org sort of set up |
17:44.33 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
17:44.57 | jm|laptop | although I'm still not entirely convinced |
17:46.17 | mvanbaak | lol |
17:46.20 | mvanbaak | get some sleep |
17:46.22 | mvanbaak | and try again |
17:46.39 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
17:53.39 | *** join/#asterisk candyman50 (n=mdecandi@pool69-59-255-25.kewr1.s.vonagenetworks.net) |
17:54.28 | mvanbaak | works for me |
17:55.18 | candyman50 | Question: Does anyone know how to mainupulate the Sip TO: Header in *? |
17:55.44 | candyman50 | I know I can manipulate the FROM with CALLERID(name/num) |
17:56.10 | *** join/#asterisk Dandan (i=dandan@wsip-70-167-100-158.ri.ri.cox.net) |
17:56.15 | Dandan | hey all |
17:56.17 | Dandan | :) |
17:56.21 | Dandan | just jumped to 1.4 |
17:56.28 | Dandan | is there any reason why Echo app would be REAAAALY jerky? |
17:56.49 | Dandan | as in: the sound is probably 10 seconds long and it takes the app over 1 minute to play it? |
18:04.37 | *** join/#asterisk bkw_ (i=brian@adsl-70-142-43-193.dsl.tul2ok.sbcglobal.net) |
18:04.38 | anonymouz666 | anyone know how can I debug the function ${CURL()} |
18:04.48 | anonymouz666 | it's not sending anything to the server |
18:06.59 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
18:11.51 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
18:11.54 | hrmphh | [Apr 23 10:04:28] NOTICE[9062] chan_zap.c: Avoiding deadlock... |
18:11.57 | hrmphh | over and over in the logs |
18:12.02 | hrmphh | how can i fix that? |
18:12.30 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
18:12.36 | anonymouz666 | just a quick question |
18:12.46 | anonymouz666 | does ${curl()} support https? |
18:16.05 | *** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net) |
18:16.19 | *** join/#asterisk elriah (i=elriah@adsl-074-164-217-189.sip.bhm.bellsouth.net) |
18:16.22 | ManxPower | hrmphh: stop using chan_spy or Monitor |
18:16.35 | elriah | Hi all. Does anyone know if the T.38 patch is in asterisk 1.2.16? |
18:16.55 | ManxPower | elriah: since it is a NEW FEATURE the patch would not be in 1.2.x |
18:16.56 | elriah | Would it be in show codecs if it were? |
18:17.22 | elriah | Ahh. Didn't know when it was introduced. Found a mantis thread that was for 1.2... |
18:17.35 | elriah | says committed into trunk 33890. |
18:17.39 | AndrewGearhart | anybody here familiar with the Polycom 320? |
18:18.01 | elriah | AndrewGearhart: what's the question? |
18:18.03 | hrmphh | manx; i wasnt aware that i was? |
18:18.06 | hrmphh | manx; how to disable? |
18:18.21 | *** join/#asterisk trevarthan (n=trevarth@c-71-59-54-137.hsd1.ga.comcast.net) |
18:18.22 | ManxPower | elriah: trunk is whatever the next version will be at the time of the comit |
18:18.30 | ManxPower | so if 1.2 was out at the time, then it would be in 1.4 |
18:18.34 | elriah | Got ya. |
18:18.37 | hmm-home | i'm having some string problems SayDigits(${foo_${bar}}) does not work. ${bar} = 0 and the value for ${foo_0} is 12345 |
18:18.41 | elriah | Is there a backport that you know of? |
18:18.42 | hrmphh | also i was told that the person making the outbound call heard nothing, but when he later spoke to the person on the receiving end, they said they heard hold music |
18:18.47 | AndrewGearhart | elriah: it says that it is a two-line phone... what exactly does that mean? that it can only handle two lines period? or that it only has buttons for two lines... |
18:19.00 | ManxPower | hrmphh: you don't disable. You would have to be running chanspy or monitor to record or listen to your calls |
18:19.03 | elriah | AndrewGearhart: No, that it can display two line labels. |
18:19.37 | ManxPower | AndrewGearhart: A "line" on a polycom can handle more than 1 call, but we always disable that feature because it confuses users |
18:19.39 | elriah | AndrewGearhart: Most people just provision a single line lable, i.e., extension 805, but that doesn't limit the number of inbound/outbound calls. I believe that number is 64. |
18:19.49 | ManxPower | if you want more than 2 "lines/calls" get a different model |
18:20.05 | hrmphh | manx; youre saying i would have had to interactively run it? |
18:20.30 | hmm-home | yeah nevermind i'm retarded |
18:20.30 | trevarthan | Hello. I have an IVR running on an asterisk box connected to a linksys spa3102. After a day or so of up-time, the IVR starts exhibiting an annoying "ringing" noise, sort of like a feedback loop, immediately after DTMF presses. As far as I can tell, this problem is only manifest on the spa3102 <-> SIP/asterisk link. Other SIP phones connecting to the asterisk box sound fine. Any ideas? |
18:20.41 | ManxPower | hrmphh: NO! You would have to put it in extensions.conf |
18:21.18 | AndrewGearhart | elriah & ManxPower : thanks |
18:21.34 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
18:21.44 | ManxPower | I'm starting to think that nobody actually manages to convert one video format to another. All docs about this using any tool do not work or are just plain wrong. |
18:22.00 | ManxPower | or are Mac specific |
18:22.03 | Qwell[] | ManxPower: that sounds like it's just asking for trouble, really |
18:22.12 | elriah | Ahh, the trials and tribulations of bridging fax calls on Asterisk 1.2... |
18:22.14 | Qwell[] | though, I have thought about using transcode/mencoder to do it |
18:22.21 | elriah | Anyone have some chicken bones? |
18:22.30 | Qwell[] | but, I don't *ever* violate patents... |
18:22.35 | trevarthan | ManxPower: yeah, transcode works. I use it with MythTV all the time. |
18:22.40 | syzygyBSD | :) Qwell |
18:22.56 | file | Qwell[]: you are violating a Telcomjoshvoxmart patent by saying you aren't! |
18:23.00 | syzygyBSD | I never knowingly violate patents... |
18:23.04 | Qwell[] | file: I have a license, nub |
18:23.10 | ManxPower | Qwell: I'm trying to remux .ty (tivo) into a format Quicktime Pro can read (.vob, .mpg) to convert it to mpeg4 |
18:23.15 | hrmphh | Manx; theres nothing ine xtensions relating to chanspy or monitor |
18:23.20 | Qwell[] | ManxPower: oh, not with asterisk? heh |
18:23.21 | hrmphh | something else must be causing the deadlock |
18:23.21 | ManxPower | trevarthan: every ffmpeg based transcode looks terrible to me |
18:23.26 | Qwell[] | transcode works great |
18:23.29 | elriah | At this point, just by getting up in the morning, I'm pretty sure everyone violates some technology patent. |
18:23.34 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
18:23.35 | ManxPower | hrmphh: then you must be having some other problem |
18:23.51 | AndrewGearhart | our current phone system shows 12 buttons for the extensions and 6 buttons for outside lines... and I'm trying to figure out what the necessary minimum price would be to carry that kind of feature set over to a new phone system |
18:24.00 | ManxPower | Qwell: does transcode support 1.5Mbps - 8.0Mbps bitrates? |
18:24.08 | AndrewGearhart | that's the root of my questions. ;-) |
18:24.15 | Qwell[] | ManxPower: it supports whatever you tell it... |
18:24.21 | syzygyBSD | AndrewGearhart: what is your question? |
18:24.31 | ManxPower | Qwell: ffmpeg is supposed to but, but that never works |
18:24.59 | elriah | AndrewGearhart: If you want a good speakerphone, consider at least a Polycom 501. |
18:25.08 | syzygyBSD | ManxPower: do you have the codec for .ty installed? |
18:25.15 | AndrewGearhart | syzygyBSD: was just explaining more about what I had previously asked in case anybody was interested. |
18:25.24 | syzygyBSD | oh, I just joined |
18:25.30 | ManxPower | Qwell: well it "works" with lots of buffer underruns and if I tell it to use a MASSIVE buffer then the underruns go away but the file cannot be played in quicktime for windoews |
18:25.30 | syzygyBSD | or rejoined... |
18:25.40 | ManxPower | syzygyBSD: yes. |
18:25.41 | Qwell[] | ManxPower: That's a feature |
18:25.46 | AndrewGearhart | elriah: that would be one of the things that would be part of it... at least one of the phones needs to have a nice speakerphone on it |
18:25.46 | ManxPower | it is called vsplit-linux |
18:25.53 | syzygyBSD | ya, why would you want to use windows? |
18:26.02 | syzygyBSD | j/k |
18:26.04 | Qwell[] | ManxPower: You just have to give it a good codec.. something windows supports |
18:26.14 | Qwell[] | (which, by default, is very little) |
18:26.18 | anonymouz666 | I think app curl does not support HTTPS :( |
18:26.20 | syzygyBSD | will vlc on windows play it? |
18:26.23 | anonymouz666 | Corydon-w ? |
18:26.27 | anonymouz666 | am I right ? |
18:26.28 | ManxPower | syzygyBSD: because I believe that a commercial mpeg4 encoder is better than a free one. |
18:26.42 | Qwell[] | ManxPower: There's your first mistake ;) |
18:26.49 | Corydon-w | anonymouz666: it depends |
18:26.50 | ManxPower | Qwell: Uh, by the time the file gets to windows is it a "standard" mpeg2 file. |
18:27.06 | Qwell[] | ManxPower: mpeg2 is just a wrapper, isn't it? |
18:27.14 | syzygyBSD | lol... |
18:27.17 | ManxPower | I can PLAY the files all day long in VLC or WMP |
18:27.27 | ManxPower | Qwell: both a wrapper and a codec |
18:27.28 | anonymouz666 | Corydon-w: on what? |
18:27.30 | Qwell[] | oh |
18:27.33 | syzygyBSD | well that sounds like a quicktime problem |
18:27.46 | Corydon-w | anonymouz666: depends on the installation of libcurl |
18:28.03 | ManxPower | syzygyBSD: My point still stands. I cannot get from point A to point B |
18:28.11 | syzygyBSD | if you can use vlc, why would you want to use quicktime? |
18:28.19 | syzygyBSD | oh.. you can... just no reason to... |
18:28.23 | ManxPower | syzygyBSD: because I do not want to make 500 people install VLC |
18:28.28 | syzygyBSD | hence, the lack of tutorials |
18:28.43 | syzygyBSD | oh, so this isn't for personal.... |
18:28.53 | anonymouz666 | Corydon76-home: thanks again! |
18:29.00 | syzygyBSD | wait.. you want to make 500 people install itunes, but not vlc? |
18:29.03 | ManxPower | And since WMP will only play streams that are in the MS version of mpeg4 in an .ASF container if I do that then I lock out all the mac users |
18:29.08 | anonymouz666 | I will rebuild the lib |
18:29.26 | Dandan | Hey Guys? Anyone might know why Echo app (in Ast 1.4.2) is slow ancd choppy? |
18:29.48 | Dandan | I have ztdummy, uhci, everything done properly... |
18:29.49 | ManxPower | syzygyBSD: I figure quicktime will be more useful than vlc for the majority of "uh, where is the any key" users. |
18:30.18 | ManxPower | and honestly VLC's seeking features suck |
18:30.21 | syzygyBSD | I stopped supporting those users a long time ago |
18:30.29 | syzygyBSD | my answer is go ask your 5 year old |
18:30.44 | ManxPower | syzygyBSD: no one under 21 is allowed on the property |
18:31.01 | syzygyBSD | oh.. so it is a business... changes even more... |
18:31.10 | syzygyBSD | you work at a brewery? |
18:31.11 | syzygyBSD | mmmmm |
18:31.18 | syzygyBSD | can i get a job? |
18:31.30 | ManxPower | syzygyBSD: gay campground. |
18:31.41 | ManxPower | The interview process is somewhat non-tradisional |
18:31.43 | Dandan | lol, even better prolly :> |
18:31.45 | *** join/#asterisk Strom_M (n=strom@135.196.213.180) |
18:32.05 | elriah | ManxPower: Eh? What's a gay campground? |
18:32.15 | elriah | ManxPower: Only gay couples allowed? |
18:32.16 | illsci | word of life |
18:32.35 | ManxPower | elriah: A campground where everyone is happy. |
18:32.46 | syzygyBSD | very happy... |
18:33.02 | trevarthan | I have an IVR running on an asterisk box connected to a linksys spa3102. After a day or so of up-time, the IVR starts exhibiting an annoying "ringing" noise, sort of like a feedback loop, immediately after DTMF presses. As far as I can tell, this problem is only manifest on the spa3102 <-> SIP/asterisk link. Other SIP phones connecting to the asterisk box sound fine. Any ideas? |
18:33.08 | elriah | Who would have thunk it... A quick google search yieled: http://www.campgayusa.com |
18:33.09 | syzygyBSD | except for the IT support... |
18:33.15 | elriah | It's official, there's a website for everything. |
18:34.15 | syzygyBSD | trevarthan: bad zap connection or echo canceler issues |
18:34.43 | trevarthan | syzygyBSD: not a zap connection. It's a terminal adapter. a linksys 3102. |
18:35.03 | trevarthan | echo canceler on the asterisk end? Or the spa3102 end? |
18:35.04 | anonymouz666 | Corydon-w: I need to use something like that in app_curl.c: curl_easy_setopt(curl, CURLOPT_SSL_VERIFYPEER, FALSE); |
18:35.10 | ManxPower | trevarthan: change the packet side on the SPA from .3 to .2 |
18:35.26 | anonymouz666 | Corydon-w: you authorize me to do that in app_curl.c ? |
18:35.27 | ManxPower | I don't recall what the exact option is called. |
18:35.34 | ManxPower | maybe ms per packet or something like that |
18:35.47 | trevarthan | ManxPower: ok. I'll give that a shot. Thanks. |
18:35.58 | hrmphh | what ist he setting that determines when it will start dialing? |
18:36.05 | hrmphh | i.e. no more DTMF |
18:36.05 | hrmphh | tones |
18:36.07 | syzygyBSD | trevarthan: either, maybe both (the way they cancel out the echo the other already canceled....) the analog connection to your phone |
18:36.49 | elriah | hrmphh: Dial Plan, usually on the phone. |
18:36.49 | syzygyBSD | hrmphh: what do you mean |
18:37.00 | hrmphh | its an analog phone |
18:37.02 | hrmphh | connected to fxs port |
18:37.26 | syzygyBSD | digittimeout? |
18:37.32 | elriah | The # key will do it. Dunno how you would control a dial plan for a phone, digittimeout maybe. |
18:38.21 | *** join/#asterisk wundaboy (n=look@adsl-68-122-118-112.dsl.pltn13.pacbell.net) |
18:38.27 | ManxPower | hrmphh: your dialplan in extensions.conf if you are on an FXS port. the dialplan on the ATA if you are using an ATA |
18:38.35 | ManxPower | digittimeout only works for IVR, not dialing |
18:38.51 | ManxPower | and # will NOT do it if you are on an fxs card |
18:39.04 | naitram | i have a linksys ATA, anyone know what to set so that i get a notification of an off hook when the phone is lifted off the hook? |
18:39.11 | *** join/#asterisk Meaty` (n=meaty3@office.abi.ca) |
18:39.11 | elriah | Well, then I stand corrected. |
18:39.32 | hrmphh | hmm ok |
18:39.34 | hrmphh | so im sol then |
18:39.36 | hrmphh | no way to change |
18:39.38 | wundaboy | I am using a hosted pbx that requires an 11digit dialstring, is there anyway i can setup some kind of proxy to modify the dialstring? |
18:39.40 | hrmphh | seems like it takes a while to start ringing |
18:39.42 | hrmphh | after it dials |
18:40.49 | ManxPower | hrmphh: that will only happen if you have a badly designed dialplan |
18:41.18 | ManxPower | i.e. you are dialing 5551212 but you have another exten line that can match 2125551212 |
18:41.28 | ManxPower | how does asterisk know that 5551212 is the full number? |
18:42.27 | *** join/#asterisk Strom_C (n=strom@135.196.213.180) |
18:43.24 | ManxPower | Qwell: According to a FAQ from apple.com Quicktime Pro with the MPEG2 support can transcode MPEG2 video to other formats, but not MPEG2 audio. |
18:43.43 | Qwell[] | ManxPower: lame |
18:44.01 | *** join/#asterisk DoDaT69 (n=dodat69@internal.digitalson.com) |
18:44.05 | ManxPower | hrmphh: my users never have to wait for more than 1/10th of a second for the call to be processed after they dial |
18:44.09 | ManxPower | Qwell: I agree. |
18:44.21 | Qwell[] | ManxPower: can it do different audio in an mpeg2 wrapper? |
18:45.54 | ManxPower | Qwell: Should be able to, working on that now |
18:46.10 | *** join/#asterisk docelmo (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net) |
18:47.22 | hrmphh | manx; i dont know if the call is waiting to be processed |
18:47.25 | hrmphh | or just processing takes a while |
18:47.27 | hrmphh | http://pastebin.ca/454623 |
18:47.30 | hrmphh | theres the dial plan |
18:48.23 | tzafrir_laptop | flash works nicely on an FXS card |
18:48.41 | hrmphh | flash? |
18:48.44 | hrmphh | to transfer? |
18:48.46 | hrmphh | or conf? |
18:50.42 | tzafrir_laptop | transfer |
18:51.35 | anonymouz666 | Corydon-w! I broke the app :) niiice |
18:51.40 | hrmphh | ok heres a really strange problem description: user makes outbound call (e.g. dials 91800WHATEVER), hears nothing but silence on the phone after dialing, hangs up after a while. later gets in touch with person he thought he was calling and person said their phone rang, they answered, and they heard the music on hold! |
18:52.23 | ManxPower | hrmphh: put the contents of extensions.conf not the output of "show dialplan" |
18:52.32 | hrmphh | its extensions.ael |
18:52.35 | hrmphh | thats what i put |
18:52.41 | ManxPower | hrmphh: can't help you with that |
18:52.50 | *** join/#asterisk Ambrose (n=ambrose@we-dont.gotdns.org) |
18:52.50 | hrmphh | basically the same thing |
18:52.53 | hrmphh | just diff syntax |
18:53.17 | Ambrose | Anyone know how to change the limit of the Dial command? I have a really long dial command with a bunch of w's and it seems like it's getting cut off and not sending the whole ting |
18:53.59 | Ambrose | In the cli it show "Executing Dial <....>" but the next line says "Called <....>" and that's where it's cut off |
18:54.10 | ManxPower | hrmphh: correct. I don't really feel like learning an entire new syntax for extensions today. |
18:54.18 | ManxPower | hrmphh: what FXS card are you using? |
18:54.29 | hrmphh | TDM400P |
18:54.34 | hrmphh | its the 1 fxs, 3 fxo |
18:54.48 | joe--f | hey, what do you guys think of voxbone for voip forwarding? |
18:57.10 | ManxPower | Ambrose: you would have to modify the asterisk source for get more than: channel.h:#define AST_MAX_EXTENSION 80 |
18:57.28 | Ambrose | ManxPower : Thanks. |
18:58.08 | trevarthan | ManxPower: so.... change 'RTP Packet Size:' under the SIP tab from .03 to 0.2? |
18:59.07 | trevarthan | any idea why that is necessary? |
19:00.30 | ManxPower | trevarthan: because The SPAs default to 30ms packets and Asterisk is hardcoded and designed for 20ms packets. So if you leave the defaults the last 10ms of each packet is dropped. |
19:02.04 | trevarthan | oh. ok. yeah, that's good to know, eh? |
19:02.42 | naitram | anyone know how to enable hook flash on linksys voip gateway |
19:02.45 | ManxPower | trevarthan: of it is .03 in the SPA then change it to .02 if it is .3 in the ATA then change it to .2 |
19:03.33 | *** join/#asterisk Goodjoke (n=Goodjoke@74.202.86.23) |
19:04.12 | Ambrose | ManxPower : When I uncomment that and changed it, now I get an error when I try to run make. Any ideas? |
19:05.50 | *** join/#asterisk sysreq (n=sysreq@193.245-ppp.3menatwork.com) |
19:07.50 | Hmmhesays | is a variable set anywhere that tells you what context your gosub came from? |
19:10.08 | ManxPower | Hmmhesays: See README.variables |
19:10.28 | ManxPower | Ambrose: # is not a comment |
19:10.33 | ManxPower | put the # back in |
19:11.03 | *** join/#asterisk JerJer[mobile] (n=jj@199.45.11.90) |
19:11.17 | Ambrose | ManxPower : Oh, ok |
19:16.48 | *** part/#asterisk trevarthan (n=trevarth@c-71-59-54-137.hsd1.ga.comcast.net) |
19:18.23 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
19:22.22 | AndrewGearhart | is there a feature in asterisk that, if the extension that you're calling is busy, you can set it to ring both of you when they become free? |
19:26.11 | *** part/#asterisk Ambrose (n=ambrose@we-dont.gotdns.org) |
19:29.59 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-136017.home.otenet.gr) |
19:31.15 | JerJer[mobile] | AndrewGearhart: In the telco world this is called CampON |
19:31.22 | JerJer[mobile] | and i have not seen that feature |
19:36.14 | AndrewGearhart | JerJer[mobile]: thx. |
19:36.31 | AndrewGearhart | how does one become a contributor to asterisk? |
19:38.21 | Dandan | ~choppy |
19:38.23 | Dandan | ~slow |
19:38.25 | jbot | no I'm not |
19:39.17 | Dandan | I will answer myself: you you EVER get a SLOW and CHOPPY sound, make sure, you add "noapic" to the kernel boot. |
19:39.33 | Dandan | jbot: choppy:you EVER get a SLOW and CHOPPY sound, make sure, you add "noapic" to the kernel boot. |
19:39.38 | Dandan | ~jbot |
19:40.12 | Dandan | hm, how do you train him? |
19:41.16 | redax | hi, |
19:41.49 | Corydon-w | jbot: choppy is <reply> If you ever get a slow and choppy sound, make sure, you add "noapic" to the kernel boot parameters. |
19:41.51 | jbot | Corydon-w: okay |
19:41.56 | redax | is the bristuffed zaptel give the DID number ? |
19:42.03 | Corydon-w | ~choppy |
19:42.05 | jbot | If you ever get a slow and choppy sound, make sure, you add "noapic" to the kernel boot parameters. |
19:42.07 | redax | seems like I miss the DID now |
19:42.22 | Corydon-w | jbot: no, choppy is <reply> If you ever get a slow and choppy sound, make sure you add "noapic" to the kernel boot parameters. |
19:42.24 | jbot | Corydon-w: okay |
19:42.30 | joe--f | has anyone used VoxBone with astericks? |
19:43.04 | Corydon-w | Yes, but it's Asterisk, not astericks |
19:43.15 | joe--f | ha |
19:43.16 | joe--f | sry |
19:44.09 | Corydon-w | voxbone works fine |
19:44.09 | joe--f | Corydon-w: Would you recommend going with VoxBone for VOIP origination? |
19:44.09 | Corydon-w | I don't recommend any particular service in my role here. |
19:44.38 | joe--f | I'm planning on having asterick manage a bunch of conference calls, and was wondering what would be best.. |
19:45.33 | joe--f | Each conference call will have a unique PIN, that will only be available for 'X' hours.. and i'm going to parse the Call Records to verify that the two people made the conference call based on the pin# they were assigned.. |
19:46.05 | joe--f | just wondering if voxbone is a good choice to start this idea with.. and potentially move over to a large hosting provider.. |
19:46.18 | _VoiceMeUp_COM | hmm |
19:46.44 | _VoiceMeUp_COM | joe depends on where your clients from |
19:46.53 | _VoiceMeUp_COM | USA/CAN or EUROPE |
19:46.55 | joe--f | they'll be primarily within the US |
19:46.58 | joe--f | for now |
19:47.04 | _VoiceMeUp_COM | and concurent cannels |
19:47.26 | _VoiceMeUp_COM | we have a few Datelines clients and call center clients that scaled up to 300 calls on a toll free number without isues |
19:47.29 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
19:47.31 | joe--f | yeah, isn't it cheaper to just get a bunch of DID's instead of a bunch of channels for one DID? |
19:47.34 | _VoiceMeUp_COM | how many people per conf and how many confs ? |
19:47.36 | Defraz | shouldn't this work exten => _12083467090,1,Dial(IAX2/12083467090@wc-pbx.fuzecore.com) |
19:47.42 | _VoiceMeUp_COM | no |
19:47.48 | _VoiceMeUp_COM | cheaper per channel |
19:47.58 | _VoiceMeUp_COM | since you can resuse channel banks for active calls |
19:48.09 | _VoiceMeUp_COM | and nothave lets say the 4 channels per number sitting and idling |
19:48.11 | joe--f | _VoiceMeUp_COM: we're just going to be having... 2. (the reasoning is that the people aren't allowed to call directly to one another, since we need to log everything) |
19:48.16 | _VoiceMeUp_COM | per channel is usually best on volume |
19:48.32 | joe--f | _VoiceMeUp_COM: hmm ok, that sounds good |
19:48.39 | CunningPike | Would there be any detrimental effects of storing voicemail on a remote NFS export with nolock? |
19:48.40 | _VoiceMeUp_COM | so 2 channels toll free USA ? |
19:49.20 | joe--f | _VoiceMeUp_COM: yes, i mean we're potentially hoping to have a couple dozen conferences.. |
19:49.22 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-72.ph.ph.cox.net) |
19:49.33 | _VoiceMeUp_COM | ok at same time ? |
19:49.36 | joe--f | yes |
19:49.42 | joe--f | but just 2 people per conference |
19:49.45 | Dandan | Corydon-w: thanks :) |
19:49.49 | _VoiceMeUp_COM | your best bet is hosting these on different weekdays to scatter the volume |
19:49.53 | cr4z3d | is 5060 the only port asterisk binds too when starting up? |
19:49.55 | redax | I don't have incoming DID with Zaptel (bristuff) what's did I missconfigured? |
19:50.25 | _VoiceMeUp_COM | telse youll have 24 channels idling on 99% of the time.. ( bette ryet .. 6/7 of the time ) if ocnf are weekly |
19:50.35 | _VoiceMeUp_COM | cr4z3d no |
19:50.37 | Dandan | anyone can recommend voipdiscount? anyone using them? |
19:50.55 | _VoiceMeUp_COM | cr4z3d these lots |
19:50.57 | cr4z3d | _VoiceMeUp_COM, what else does it bind too? cuz i keep getting a bind error.. never happened until i actually had to restart my computer |
19:51.08 | _VoiceMeUp_COM | dundi/iax/sip/manager/etc |
19:51.15 | _VoiceMeUp_COM | well it shoudl tell you the service in the error line |
19:51.18 | cr4z3d | _VoiceMeUp_COM, seems to be while starting manager.c |
19:51.22 | _VoiceMeUp_COM | like..DUndi cant bind to XXXXX |
19:51.26 | _VoiceMeUp_COM | then its manager |
19:51.32 | cr4z3d | it didn't say what por though |
19:51.36 | _VoiceMeUp_COM | nano /etc/manager.conf |
19:51.48 | Dandan | use VI, Luke! |
19:51.50 | _VoiceMeUp_COM | port = 5038 |
19:51.56 | _VoiceMeUp_COM | you must have another process runing |
19:52.01 | _VoiceMeUp_COM | Or another app |
19:52.13 | _VoiceMeUp_COM | if you need to run multipleinstances increment that port by 2 |
19:52.15 | Dandan | or another instance of asterisk (<= done that) |
19:52.16 | _VoiceMeUp_COM | to be sure |
19:52.42 | _VoiceMeUp_COM | i meant /etc/asterisk/manager.conf |
19:52.43 | _VoiceMeUp_COM | sorry |
19:52.53 | joe--f | _VoiceMeUp_COM: what do you mean by hosting these on different weekdays.. like use a different DID for each day of the week?.. |
19:52.56 | cr4z3d | yeah _VoiceMeUp_COM 5038 but netstat -p shows nothing on that port |
19:54.00 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
19:54.37 | JerJer[mobile] | Dandan: perhaps take a look at discountvoipoutlet.com |
19:54.53 | JerJer[mobile] | unless you need something in .EU |
19:54.54 | Dandan | JerJer[mobile]: thanks, I need something with 0 to Poland |
19:54.59 | Dandan | I do :/ |
19:55.03 | JerJer[mobile] | ah |
19:55.08 | _VoiceMeUp_COM | nah |
19:55.09 | JerJer[mobile] | then i can't help |
19:55.09 | Dandan | Well, I need termination in poland for free |
19:55.14 | _VoiceMeUp_COM | i meant host conferences on each day.. |
19:55.20 | _VoiceMeUp_COM | conf 1 on monday..conf2 on tuesday etc |
19:55.21 | Dandan | but I will originate from the US |
19:55.36 | JerJer[mobile] | nothing is free |
19:55.46 | Dandan | well, flat fee |
19:55.47 | _VoiceMeUp_COM | cr4z3d telnet localhost 5038 |
19:55.47 | Nugget | telnet is eeeeeeevil! |
19:56.03 | cr4z3d | yes yes it is |
19:56.05 | cr4z3d | but ok |
19:56.09 | _VoiceMeUp_COM | telnet |
19:56.11 | _VoiceMeUp_COM | lol |
19:56.20 | *** join/#asterisk joshaidan (n=brianj@thunderbay-voip-4.vianet.ca) |
19:56.21 | _VoiceMeUp_COM | ok hmm tought nugget ws a chicken mcnugget bot |
19:56.21 | joe--f | _VoiceMeUp_COM: well, so, i have no control over when the conferences take place.. they'll be fairly random, since caller1 and caller2 will both select a time that's good for them.. |
19:56.29 | _VoiceMeUp_COM | oh |
19:56.30 | _VoiceMeUp_COM | i get it |
19:56.33 | _VoiceMeUp_COM | then you are ok |
19:56.42 | Defraz | exten => _12345564343,1,Dial(SIP/12345564343@blah.blahblah.com) |
19:56.43 | _VoiceMeUp_COM | you can sassume its scattered around the biz week |
19:56.46 | joe--f | so potentially one day there could be a ton of conferences, and the next day none. |
19:56.50 | joe--f | ya |
19:57.24 | Defraz | Okay that line work with SIP but when I replace the SIP with IAX2 it doesn't' work. |
19:57.30 | Defraz | What am I missing? |
19:58.24 | Defraz | from what I have found I can't see any difference between sip and iax on the dial command. |
19:59.08 | cr4z3d | _VoiceMeUp_COM, hmm interesting i actually was able to get in |
19:59.26 | cr4z3d | now how do i kill that |
19:59.33 | cr4z3d | and why didn't it show up in netstat |
19:59.59 | JerJer[mobile] | Defraz: matching without any N or X ? |
20:00.49 | Defraz | When a call comes into my [from-pstn-custom] context it will forward the call to the other asterisk box (blah.blahblah.com) |
20:01.03 | Defraz | and it works with SIP but when I replace sip with iax2 it doesn't work. |
20:01.13 | Defraz | I have incoming trunks setup for both sip and iax2 |
20:06.17 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
20:07.03 | _VoiceMeUp_COM | netstat -anl |
20:07.16 | _VoiceMeUp_COM | netstat -an |grep 5038 |
20:13.01 | cr4z3d | _VoiceMeUp_COM, found it using ps -ef.. wonder why i couldn't asterisk -r |
20:14.59 | *** join/#asterisk ReD-MaN (n=redman@CPE0002b38bce8b-CM0018c0b357cc.cpe.net.cable.rogers.com) |
20:15.46 | *** join/#asterisk Bobocop (n=Bobocop@83.168.90.41) |
20:16.32 | _VoiceMeUp_COM | ah |
20:16.34 | _VoiceMeUp_COM | hmm |
20:16.39 | _VoiceMeUp_COM | killall -9 asterisk |
20:16.44 | _VoiceMeUp_COM | then restart it |
20:17.00 | cr4z3d | i just did a pkill asterisk and started it up and worked fine |
20:17.01 | *** join/#asterisk DrukenHME (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
20:17.13 | _VoiceMeUp_COM | perfect |
20:17.21 | Bobocop | what's wrong with RxFAX and detecting end of incoming fax transmission? Why don't RxFAX hang up? |
20:17.30 | *** join/#asterisk Buglouse (n=Buglouse@adsl-69-215-134-89.dsl.milwwi.ameritech.net) |
20:17.53 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-4c440cc43b449661) |
20:18.04 | syzygyBSD | that isn't rxfaxes job? |
20:18.27 | syzygyBSD | it should return after finished, just not hangup |
20:18.56 | Mercestes | <PROTECTED> |
20:19.18 | *** join/#asterisk thoughtpolice (n=austin@c75-111-146-82.plaicmtc01.tx.dh.suddenlink.net) |
20:19.32 | jm|laptop | someone with a FWD account want to help me? |
20:19.44 | jm|laptop | with some test calls |
20:21.53 | Bobocop | you're right :) but right now RxFAX stays on the line, waiting for something, even after remote fax machine disconnected... So line stays busy forever. Is the AbsoluteTimeout only solution? |
20:21.56 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
20:22.02 | Bobocop | Mercestes: thx :) |
20:24.19 | Bobocop | I'm wondering if this is normal behaviour of RxFAX, or it's just my broken setup.... |
20:25.45 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
20:26.22 | _VoiceMeUp_COM | Bobocop add ==== exten => h,1,Hangup() |
20:26.42 | *** join/#asterisk malcolmd (i=malcolmd@pdpc/sponsor/digium/malcolmd) |
20:31.18 | *** join/#asterisk X-Rob_ (n=rob-x@CPE-58-167-128-40.qld.bigpond.net.au) |
20:34.29 | Bobocop | _VoiceMeUp_COM: no change :( |
20:36.24 | *** join/#asterisk CyberPony (n=CyberPon@66-194-25-11.static.twtelecom.net) |
20:36.27 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
20:37.11 | Bobocop | I'm assuming, that * gives total call handling responsibility to RxFAX, so it seems that RxFAX is the one who should release line... or at least exit after successfull transmission, and return control to * |
20:37.26 | Bobocop | or am I totally wrong? ;) |
20:39.48 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
20:44.01 | _VoiceMeUp_COM | hmm |
20:44.07 | _VoiceMeUp_COM | sip debug it |
20:44.15 | _VoiceMeUp_COM | make it verbose and ebug 999 |
20:44.39 | _VoiceMeUp_COM | no idea , its really easier when you see the logical aspects |
20:44.52 | Bobocop | sip debug?.... this is handled by FXO on local machine |
20:45.54 | _VoiceMeUp_COM | hmm didnt know |
20:45.55 | _VoiceMeUp_COM | ok |
20:46.43 | Bobocop | yup, I wish I knew more about its logic too :) |
20:46.45 | *** join/#asterisk hallship (n=hallship@70.103.238.2) |
20:47.36 | Bobocop | the worst thing is, that I have to in fact limit maximum fax size because of setting resonable timeout value |
20:48.42 | Bobocop | so when I set timeout for lets say 2 minutes, the maximum fax length will be no more than 2 pages. or even 1. |
20:48.51 | hallship | I am stumped. I am trying to setup a sip call relay through my asterisk server. Call comes in on a sip trunk and is then rerouted right back out to a different number. The issue is with callerid. I can't seem to get the originating number to show on the final destination. |
20:49.07 | hallship | It seems be sending the sip account info for the second leg of the call. |
20:49.38 | hallship | Does anybody know how I can edith the sip From: section on an outbound sip call? |
20:51.03 | *** join/#asterisk BruXo (n=celio@c91192a6.static.bhz.virtua.com.br) |
20:51.24 | _VoiceMeUp_COM | fromsuer |
20:51.30 | Bobocop | anyway, thx for help _VoiceMeUp_COM :) cu |
20:51.31 | _VoiceMeUp_COM | fromuser = blah |
20:51.39 | _VoiceMeUp_COM | no prob |
20:51.40 | *** part/#asterisk Bobocop (n=Bobocop@83.168.90.41) |
20:52.02 | hallship | and blah can be whatever? |
20:52.18 | _VoiceMeUp_COM | its gonna be username |
20:52.22 | _VoiceMeUp_COM | the from:BLAH |
20:52.25 | _VoiceMeUp_COM | is the callerid |
20:52.35 | _VoiceMeUp_COM | you culd set it on the phone |
20:52.39 | _VoiceMeUp_COM | or use SER to rewrite |
20:52.50 | _VoiceMeUp_COM | one way or another youll use trustrpid |
20:52.52 | _VoiceMeUp_COM | =yes |
20:54.21 | _VoiceMeUp_COM | so from:"BLAH "<554544333> or from;user@server or whatever you need to pass it .. but best bet is the SER soluition , you could add headers and or modifiy the full line to whatever you need it to be |
20:54.53 | _VoiceMeUp_COM | example : SER is user based.. bob@domain.com , asterisk is extension based.. NXXNXXXXXX@domain.com |
20:55.25 | _VoiceMeUp_COM | thats when you ccan use SER to actually poll a db to figure the did to show for bob and replace all this by from:"BOB"<5554443333> |
20:57.45 | hallship | Hmm, Okay that's interesting. Let me give it a try. Thanks! |
20:59.08 | _VoiceMeUp_COM | k |
21:02.01 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-87cbb2a170f6d9ea) |
21:03.55 | Hmmhesays | hmm can you use # with read? |
21:04.18 | Hmmhesays | i mean can cmd read read the # key and return it |
21:04.41 | _VoiceMeUp_COM | eads a #-terminated string of digits a certain number of times from the |
21:04.42 | _VoiceMeUp_COM | user in to the given variable. |
21:04.48 | _VoiceMeUp_COM | oh |
21:04.50 | Hmmhesays | apparently not |
21:04.51 | _VoiceMeUp_COM | #.. hmm |
21:05.05 | _VoiceMeUp_COM | <PROTECTED> |
21:05.05 | _VoiceMeUp_COM | <PROTECTED> |
21:05.11 | _VoiceMeUp_COM | no since its a end of line terminator |
21:07.28 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
21:07.29 | hrmphh | [Apr 23 14:03:07] DEBUG[9751] chan_zap.c: Got event Wink/Flash(3) on channel 1 (index 0) |
21:07.33 | hrmphh | [Apr 23 14:03:07] DEBUG[9751] chan_zap.c: Winkflash, index: 0, normal: 20, callwait: -1, thirdcall: -1 |
21:07.35 | hrmphh | [Apr 23 14:03:07] DEBUG[9751] chan_zap.c: Already have a dsp on Zap/1-2? |
21:07.40 | hrmphh | what does that mean exactly? |
21:10.27 | *** join/#asterisk pvanstam (n=Pim@dsl-083-247-093-018.solcon.nl) |
21:10.35 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195) |
21:11.08 | khronos | Hmm, having a small problem stripping off a number in my dial string. |
21:11.31 | khronos | What I want to do is hit 9 as a prefix and everything after that excluding the 9 get sent to the other system. |
21:11.34 | khronos | I have |
21:12.15 | khronos | exten => _9.,1,Dial(IAX2/me@my_friend/{$EXTEN:1}) |
21:12.22 | *** join/#asterisk |BLiX| (i=asa@12.192.197.15) |
21:12.45 | khronos | On the other system it seems to not be stripping off the 9 at the beginning of the dial string. |
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21:17.20 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
21:22.40 | *** part/#asterisk danicholson (n=danichol@203.89.191.222) |
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21:30.07 | |BLiX| | anyone know where to find documentation on users.conf? |
21:38.41 | *** join/#asterisk SomethingISODD (n=dan@xplr-ts-v10-208-114-188-192.barrettxplore.com) |
21:38.49 | Vec | Do iax2 friends authenticate eachother in both directions i.e. incoming and obviously outgoing calls ? |
21:39.11 | SomethingISODD | hello all question using G729 how many concurrent calls can be on a 10MG line |
21:39.12 | Vec | |BLiX| : the documentation within users.conf is pretty good, its basically iax.conf and sip.conf in one |
21:39.14 | SomethingISODD | MG=MEG |
21:39.45 | jm|laptop | I have updated to 1.4 and now I don't see call progress despite core set verbose 999 and core set debug 999 |
21:39.47 | jm|laptop | why is that? |
21:40.06 | Vec | SomethingISODD : I would say 1000 max |
21:40.20 | SomethingISODD | ok Vec thanks |
21:40.31 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
21:40.46 | Vec | jm|laptop : no idea, works fine for me |
21:40.54 | jm|laptop | oh |
21:40.57 | Vec | jm|laptop : try core set verbose 10 |
21:40.59 | jm|laptop | ok thanks :) |
21:41.09 | Vec | and take off core set debug |
21:41.15 | Vec | I think its core set no debug |
21:41.22 | Vec | or core set debug 0 |
21:41.34 | jm|laptop | Core debug is now OFF |
21:41.43 | Vec | set core verbose 10 |
21:41.47 | Vec | then see what happens |
21:41.49 | jm|laptop | did that |
21:41.52 | jm|laptop | still nothing :S |
21:41.52 | Vec | 999 seems like a little much |
21:42.03 | jm|laptop | it's Debian, though - so this package might still be 'unstable' |
21:42.38 | Vec | jm|laptop : u can try compiling it from source, also try run it asterisk -cvvv |
21:46.04 | jm|laptop | Vec: my bad; phone was still connecting to OLD asterisk box - sorry. |
21:46.32 | jm|laptop | blimey - lots changed 1.2 --> 1.4 :S |
21:49.22 | Uatec_ | does anybody use 1.3? |
21:50.48 | *** join/#asterisk danicholson (n=danichol@203.89.191.222) |
21:54.14 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
21:55.01 | elriah | Does SayDigits not take a variable? |
21:56.56 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
22:02.38 | anonymouz666 | it takes |
22:02.53 | anonymouz666 | saydigits(${blah}) |
22:02.54 | elriah | You guys see anything wrong with this? exten => s,n,Set(FILENAME=${${TIMESTAMP}:-11:6}) |
22:03.04 | elriah | It keeps evaluating null. |
22:03.33 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
22:03.33 | *** mode/#asterisk [+o denon] by ChanServ |
22:04.21 | elriah | i.e., in the consule it says Executing Set("SIP/whatever-b71c7228", "FILENAME=") |
22:04.25 | *** join/#asterisk allankardec (n=root@189-19-59-138.dsl.telesp.net.br) |
22:04.29 | allankardec | hello |
22:04.54 | elriah | And it should set FILENAME to the timestamp seconds. |
22:05.15 | Qwell[] | elriah: You're trying to execute a number |
22:06.32 | elriah | Qwell[]: I'm looking but I don't see it. Throw me a bone (please) |
22:06.44 | Qwell[] | You don't need the outer ${} |
22:06.54 | Qwell[] | move the :-11:6 inside the ${TIMESTAMP} |
22:07.07 | elriah | hrm.. Ok, thanks.. Going to try it now.. |
22:08.27 | elriah | Thanks, Qwell. |
22:09.06 | *** part/#asterisk pvanstam (n=Pim@dsl-083-247-093-018.solcon.nl) |
22:09.38 | allankardec | Somebody could help me with the instalation of g729 of intel? |
22:09.46 | Qwell[] | allankardec: no |
22:09.58 | allankardec | thanks |
22:10.30 | anonymouz666 | lol=yes |
22:12.48 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
22:12.52 | FuriousGeorge | hey all |
22:13.34 | jovannotti | did you pay for the license allankardec ? |
22:13.57 | allankardec | is one license free |
22:14.12 | allankardec | sorry, I'm Brazillian |
22:14.48 | jovannotti | g729 is not free for installation allankardec, at least do you have bought TC400B card |
22:16.20 | jovannotti | or do you pay 10USD for each licence |
22:16.29 | FuriousGeorge | does anyone use gentoo here? ive always used the svn builds, but i notice gentoo installs all these distro specific patches, and while i have no idea what they are for exactly, im wondering if they may be a good thing |
22:17.06 | allankardec | the intel fornush |
22:17.21 | jovannotti | do you mean IPP ? |
22:17.31 | allankardec | yes, i do |
22:18.02 | allankardec | I get non commerce, |
22:18.10 | allankardec | I got non commerce |
22:18.14 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
22:18.14 | FuriousGeorge | another thing. i didnt mess with the -march setting when i built it (i have a socket 939 opteron 165) and i noticed "k8" was automatically selected |
22:18.26 | FuriousGeorge | but gentoo recommends using "opteron" for my march setting |
22:18.27 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
22:18.49 | jovannotti | ok, I tried but I have not deep knowledge in C to create the drivers, I saw a page for a guy from Latvia or England that developed some years ago, but the versions are obsolete now |
22:19.01 | FuriousGeorge | the reason im asking about all this is b/c i have two identical systems. one deadlocks once a week one doesnt. |
22:19.26 | *** join/#asterisk pvanstam (n=Pim@dsl-083-247-093-018.solcon.nl) |
22:19.54 | FuriousGeorge | i built the kernels separately, so maybe there is some difference there. i could try taking the kernels from the good system. the other difference is that the bad system has two tdm400p cards |
22:20.50 | *** join/#asterisk BrianR___ (i=brianr@static-72-70-36-11.bstnma.fios.verizon.net) |
22:21.02 | BrianR___ | Anyone here use asterisk with vitelity? |
22:21.22 | *** join/#asterisk dniel (n=ary@host224.190-30-210.telecom.net.ar) |
22:21.39 | dniel | chupenme bien un huevo |
22:21.51 | dniel | aguante la linea con tono |
22:22.03 | dniel | aguante avaya y nortel |
22:22.19 | dniel | file: tocame bien la punta de la pija con tus dientes |
22:22.23 | *** join/#asterisk ardor (n=Miranda@las-static-66.18.135.148.mpowercom.net) |
22:22.33 | ardor | Hi everyone |
22:22.38 | file | dniel: ...hi? |
22:22.40 | FuriousGeorge | heh |
22:23.04 | dniel | file: agarra tu boquita y pegate una flor de lamida de pija. |
22:23.12 | jovannotti | its an argentinian guy that needs to be banned |
22:23.26 | file | oh, banning |
22:23.27 | file | I can do that! |
22:23.30 | FuriousGeorge | :) |
22:23.37 | *** part/#asterisk dniel (n=ary@host224.190-30-210.telecom.net.ar) |
22:24.38 | FuriousGeorge | so, any gentoo users in here that dont use the svn packages? im thinking of doing that myself as I'm having issues with deadlocks |
22:25.06 | *** join/#asterisk dniel (n=ary@host224.190-30-210.telecom.net.ar) |
22:25.09 | dniel | :p |
22:25.16 | dniel | file: :p |
22:25.18 | *** part/#asterisk dniel (n=ary@host224.190-30-210.telecom.net.ar) |
22:25.41 | file | well! |
22:25.56 | *** mode/#asterisk [+b #asterisk!*@*] by file |
22:26.00 | file | oh crap |
22:26.10 | Qwell[] | eh? |
22:26.13 | FuriousGeorge | i bet this kid is in here under another alias and wants nothing more than some extended attention |
22:26.23 | file | manually typing out ban statements sucks |
22:26.28 | file | when you haven't done it in years |
22:26.39 | anonymouz666 | they think that we can't understand the damn spanish |
22:26.47 | Qwell[] | /mode +b hostmask channel |
22:26.48 | Qwell[] | :D |
22:26.56 | FuriousGeorge | yeah! damn spanish speakers de la verga |
22:27.06 | Qwell[] | file: or just /kickban <user> |
22:27.29 | Qwell[] | /quote PONG |
22:27.37 | anonymouz666 | telnet sucks. BitchX was nice |
22:27.48 | Qwell[] | erm, I guess you wouldn't /quote that on telnet, heh |
22:28.17 | *** join/#asterisk alexhopper (n=a27386@mctnnbsa24w-142167039254.pppoe-dynamic.nb.aliant.net) |
22:28.35 | file | it's my room mate! |
22:29.02 | ardor | I'm trying to get some Music inside of my app_conference, Is this possable? |
22:31.08 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
22:32.37 | allankardec | jovannotti, I installed the g729 codec, after the concluid download, if i run it, i do receive the info: permission denied |
22:33.22 | Qwell[] | allankardec: I was serious earlier. No, you won't get help with that here. |
22:33.26 | allankardec | jovanotti: sorry i am slow in tzping in english |
22:34.25 | *** join/#asterisk ctaloi (n=ctaloi@pool-72-90-82-84.syrcny.fios.verizon.net) |
22:34.46 | *** join/#asterisk dlynes_laptop (n=dlynes@d207-216-161-56.bchsia.telus.net) |
22:36.29 | digiterata | Hi all, I'm doing some Asterisk testing with Jabber/Jingle clients and was wondering if anyone can recommend a decent voice client that runs on linux. At the moment my test subject is running GoogleTalk in a Virtual Machine. Hoping theirs a native solution. |
22:36.51 | allankardec | quell, sorry i missed your info, can zou repead that again pl |
22:37.29 | anonymouz666 | allankardec: ele não vai te ajudar pq esse codec é patenteado e vc precisa comprá-lo. |
22:37.43 | Qwell[] | anonymouz666: something like that |
22:38.10 | allankardec | anonymouz666, ehhehehehehehe, valeu pelo português, agora conseguir ler rapido |
22:38.28 | Qwell[] | no, no rapido |
22:39.05 | jm|laptop | holy crap: I have really broken IAX :s |
22:42.55 | allankardec | speaking serious, do you know this site www.readytechnology.co.uk/open/ipp-codecs-g729.1 ? |
22:45.08 | bulle | allankardec: the situation is a bit complicated when it comes to g729 |
22:45.21 | bulle | allankardec: as the codec has parts of it being patented |
22:46.04 | bulle | allankardec: now, in some countries, like the u$a, there are software patents, and laws against aiding people to commit crimes, eg, helping people to use patented source without the correct licence |
22:46.18 | *** join/#asterisk mholman (n=mholman@203-206-107-167.dyn.iinet.net.au) |
22:46.23 | jovannotti | actually you can do it, but downloading IPP from Intel Page, and then trying to compile allakardec, I tried but I prefer pay 10 USD for each license |
22:46.25 | bulle | allankardec: so, depending on where you are, you might or might not be required by law, to have a licence, in order to run g729 |
22:46.59 | jovannotti | I was looking for codecs g729 and g723, I tried paying to digium 10 USD for each licence. and it works fine |
22:47.01 | bulle | allankardec: i would suggest you buy a licence, if you live in a country where its required, but if your country doesnt require it, dont pay for the licence |
22:47.44 | bulle | allankardec: anyway, as many people here are american citizens, they are not going to discuss any unlicenced software with you |
22:48.45 | JT | err what the hell |
22:48.55 | JT | there is nothing wrong with discussing patented software |
22:49.34 | bulle | JT: im of the same opinion, but i got to take lots of flack, for helping out with non licenced stuff |
22:49.38 | allankardec | brazil don't require to buy the licences |
22:49.52 | bulle | allankardec: ye, brazil is a sane country in that respect |
22:50.18 | bulle | allankardec: you guys also got wanderlei silva =) |
22:50.46 | anonymouz666 | rodrigo minotauro is a better fighter |
22:50.48 | JT | bulle: there's a differennce between helping someone out with stolen code and discussing code that a patent is required due to something it provides, in a business environment :) |
22:51.24 | bulle | JT: well, this chap seems to want to get help on how to use this code without having a valid licence |
22:51.26 | anonymouz666 | allankardec: how do you know that? |
22:51.31 | bulle | JT: as i said, i dont disagree |
22:51.38 | bulle | anonymouz666: brazil doesnt have software patents ........... |
22:51.51 | JT | bulle: so it'd depend if the code was subject to a copyright violation or not |
22:52.08 | JT | whether he gets a patent for a function it provides is something else |
22:52.10 | anonymouz666 | bulle: where did you read that? |
22:52.15 | bulle | JT: well, as i said, last time i tried to help, i got shitloads of rants about how i was a thief and criminal etc |
22:52.33 | JT | bulle: i think the intel code might be stolen |
22:53.15 | jm|laptop | I have a silly question :/ |
22:53.30 | bulle | anonymouz666: nowhere, i have been talking to some brazilian friends of mine |
22:53.36 | jm|laptop | in 1.2 my verbose and debug were colourful - in 1.4 monochrome. What can I do? |
22:53.36 | allankardec | My asterisk is logined the of vonage, the out call is perfect, but, coming call is problem with g729 |
22:54.02 | anonymouz666 | bulle: ok. |
22:54.26 | bulle | anonymouz666: software patents are realy the exception, not the rule, out in the world |
22:55.31 | anonymouz666 | there are a lots of companies in here that uses a paid version of g729, must have a reason for that. |
22:56.15 | DoDaT69 | how can I add enum checking to outbound route? |
22:57.58 | [TK]D-Fender | DoDaT69, Its all just dialplan, go look at the apps related to enum |
22:58.38 | DoDaT69 | okay |
22:59.02 | bulle | JT: well, if its all software, then its per definition a software patent |
22:59.26 | bulle | JT: atleast here, you cant patent algorithms and software |
23:00.29 | *** join/#asterisk evilbuny (n=flycasua@kirk.ozwide.net.au) |
23:01.33 | bulle | hmm, time to slepp |
23:01.35 | JT | bulle: i believe the g.729 patent covers the algorithm |
23:01.55 | allankardec | hello, I have friends that actived the g729 of intel, but I dont speak with him |
23:03.16 | jm|laptop | ~skype |
23:03.26 | jbot | methinks skype is evil. see gizmoproject.com, or offering free landlines: http://share.skype.com/sites/en/2006/05/free_calls_to_all_landlines_an.html |
23:03.57 | Qwell[] | jbot: no, skype is evil. see gizmoproject.com |
23:04.00 | jbot | Qwell[]: okay |
23:04.22 | Qwell[] | jbot: no, skype is evil. see gizmoproject.com |
23:04.26 | jbot | Qwell[]: i already had it that way |
23:04.32 | Qwell[] | No you didn't |
23:04.39 | allankardec | hello all, I tired my decision is going to buy. |
23:04.45 | jm|laptop | Qwell: way to argue with a bot :S |
23:04.49 | JT | ~skype |
23:04.51 | jbot | somebody said skype was evil. see gizmoproject.com |
23:04.52 | jm|laptop | but skype is evil, right? |
23:04.55 | Qwell[] | way to argue with an op arguing with a bot |
23:04.56 | allankardec | to licences from digium |
23:05.06 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
23:05.09 | JT | looks like the bot did listen the first time |
23:05.09 | jm|laptop | Qwell: touche |
23:05.25 | allankardec | thanks all |
23:05.40 | Qwell[] | jbot: a smart bot wouldn't have stripped the whitespace automatically |
23:05.59 | anonymouz666 | chan_msn.c would be possible? |
23:06.03 | Qwell[] | anonymouz666: sure |
23:06.11 | Qwell[] | it's just software |
23:06.37 | bulle | msn nowadays just uses sip and rtp or ? |
23:06.43 | jm|laptop | h.323? |
23:06.56 | jm|laptop | some I'm guessing I can't just add a skype 'trunk' still? |
23:07.12 | jm|laptop | (for free) |
23:07.14 | jm|laptop | s/some/so/ |
23:07.15 | allankardec | but, I have losing time |
23:07.22 | Qwell[] | sure you can, if you write a module for it |
23:07.34 | jm|laptop | (: |
23:08.15 | allankardec | qwell, jonajona, anonymou666 and bulle thank for help me |
23:08.17 | jm|laptop | I don't think I like Skype users anyway |
23:08.38 | jm|laptop | with enum/p2p sip uri they will soon lose their foothold? |
23:08.40 | anonymouz666 | I don't like, but my customers do. |
23:08.51 | Qwell[] | customers are nubs |
23:09.34 | jm|laptop | seems $600 is too little bounty |
23:09.41 | jm|laptop | not surprising with such a weak dollar atm ;) |
23:09.52 | JT | the old windows client that did voip was h.323 |
23:09.59 | JT | msn messenger is supposedly sip |
23:10.34 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
23:11.02 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
23:11.45 | jm|laptop | so why *hasn't* someone reverse engineered Skype? Licensing? or is it just a pita? |
23:12.14 | digiterata | Sorry to jump in. Did I hear someone say I can add MSN via a standard SIP channel? |
23:12.31 | JT | jm|laptop: some people have, but no-one cares about it, it's rubbish |
23:12.41 | jm|laptop | JT: agreed |
23:12.50 | jm|laptop | wait: I didn't post that line?! |
23:13.05 | digiterata | or is this still theoretical. rubbish to nerds like us, but 40 million people use MSN - not rubbish. |
23:13.17 | Qwell[] | digiterata: skype is rubbish |
23:13.20 | jm|laptop | oh yes I did; [00:08] <jm|laptop> I don't think I like Skype users anyway |
23:13.21 | Qwell[] | MSN just slightly less so |
23:13.47 | jm|laptop | JT: isn't it one of the biggest worldwide 'private' PBXes though? |
23:14.06 | digiterata | yeah agreed, Skype - closed, proprietary, crappy p2p architecture, too inconsistent to be useful; too closed to be inter-operable |
23:14.08 | paavum | Hi |
23:14.08 | JT | jm|laptop: possibly |
23:14.17 | paavum | How can I disable zttranscode from beeing loaded? |
23:14.32 | Qwell[] | paavum: don't load codec_zap |
23:14.46 | *** join/#asterisk hallship (n=hallship@70.103.238.2) |
23:15.00 | digiterata | I'm actually working actively on getting gTalk working with Asterisk - but I'm not very experienced and the documentation is a bit sparse at the moment. |
23:16.10 | anonymouz666 | but it works. |
23:16.28 | anonymouz666 | I already configure that |
23:16.55 | digiterata | it works, just not for me - yet. |
23:16.56 | hallship | Hello, I dropped in earlier with a callid issue. I got some advice but that doesn't seem to be the answer to my delima. Here is the issue in a nutshell. |
23:17.10 | Qwell[] | hallship: I'm allergic to nuts. |
23:17.13 | hallship | I am trying to setup a sip call relay through my asterisk server. Call comes in on a sip trunk and is then rerouted right back out to a different number. The issue is with callerid. I can't seem to get the originating number to show on the final destination. |
23:17.31 | hallship | :) |
23:17.44 | hallship | <PROTECTED> |
23:18.07 | digiterata | anonymouz666: you know how to configure Jabber/Jingle? Could you show me how? I've got a simple AsteriskNow virtual machine running. not sure how to configure. |
23:18.18 | JT | hallship: what does it show instead? |
23:18.25 | hallship | _VoiceMeUp_COM_ suggested setting the fromuser to the callerid I wanted. |
23:18.46 | hallship | It shows the info that is configured for the sip account that the sip trunks are registerd with. |
23:19.13 | paavum | Hi.. Im having severe problems with * 1.4 |
23:19.17 | anonymouz666 | digiterata: there is an example in voip-info that can help you |
23:19.20 | JT | hallship: why dob people in like that? |
23:19.23 | hallship | The same scenario works fine using IAX or ZAP channels, btw. |
23:19.25 | paavum | I have a queue with analog extns |
23:19.46 | paavum | but every time I get into the queue I get a "Cant ring Zap/xx" msg on the cli |
23:19.57 | paavum | and then I get "tt-allbusy" |
23:20.06 | paavum | played to me |
23:20.15 | paavum | as if it couldnt get into the queue |
23:20.32 | paavum | can anybody give me a hand? |
23:20.33 | digiterata | I'll go look again. might have questions for you though. Hope you don't mind. It would make my night if I could figure this out tonight. |
23:20.35 | hallship | JT: Sorry I didn't get your question..? |
23:20.48 | JT | <PROTECTED> |
23:21.00 | JT | people will be weary to help you when you say things like that |
23:21.16 | hallship | Oh, ya, I'm sure he meant well. |
23:21.22 | JT | there is no need to bring up who told you something previously |
23:22.03 | [TK]D-Fender | paavum, pastebin the clie output of the failed call from beginning to end, and then your applicable config files |
23:22.16 | hallship | Oh I see. Well I wasn't trying to bad mouth him. Mostly just trying to spark some recollection if anyone was still here from the previous discussion. |
23:22.52 | JT | hallship: so what callerid are you getting? |
23:23.06 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
23:23.11 | [TK]D-Fender | hallship, Do you have something in writing saying that the provider you are calling out through even permits you to set your callerid? |
23:23.27 | hallship | I get the callerid of the SIP account that the trunks ring into and out on. |
23:23.46 | JT | please don't use the word trunks, they're not trunks in that context :) |
23:24.23 | [TK]D-Fender | hallship, who's callerid are you seeing, the INCOMING calls, or the callerid of the account you are using to dial OUT from? |
23:24.29 | hallship | [TK]D-Fender: no, althought their support says it should be possible. |
23:24.41 | IOscanner | Anyone have good recommendation of good outbound terminication for US. |
23:25.12 | IOscanner | Looking for a carrier that has good rates with good uptime and support. |
23:25.16 | Qwell[] | terminication? |
23:25.23 | hallship | JT: The call is ringing in on what I've come to know as SIP trunks. What shall I call them instead? |
23:25.33 | *** join/#asterisk uski (n=uski@ALagny-153-1-18-27.w86-198.abo.wanadoo.fr) |
23:25.36 | [TK]D-Fender | hallship, I did NOT ask you a yes/no question. |
23:25.36 | JT | hallship: connections, sessions |
23:25.54 | IOscanner | sorry Termination |
23:25.57 | *** join/#asterisk kopeah (n=kopeah@cpe-70-115-242-122.satx.res.rr.com) |
23:26.00 | [TK]D-Fender | hallship, Get your head on straight and answer the very direct question I asked. |
23:26.18 | IOscanner | wow missed that |
23:26.19 | hallship | I am seeing the account I am dialing out with. I want to see the callid of the account I am dialing in with. |
23:26.24 | uski | hi, i have asterisk 1.2.13 and it segfaults... where do i send the core dump ? ;) |
23:26.38 | *** join/#asterisk bawb2 (n=bawb2@ip50210.estcmp.ku.edu) |
23:27.00 | [TK]D-Fender | hallship, before you dial out change the callerID inyour dialplan and see if you can see the change. if you can't, you're up a creak as they say... |
23:28.19 | hallship | You mean change the CallerID variable to the whatever I want to to see? If I noOp the ${CALLERID} variable before I dial out it's correct. But they are keying off of the From: in the sip header. |
23:28.44 | *** join/#asterisk anthm][ (n=anthm@m015f36d0.tmodns.net) |
23:29.58 | *** join/#asterisk AndyCap_ (n=aoy@pdpc/supporter/sustaining/AndyCap) |
23:30.17 | hallship | I did find a forum talking about setting some sort of custom variable and getting your provider to key off of that for callerid. Anyone here had any experience with that? |
23:30.50 | JT | the ${CALLERID(num)} function |
23:31.05 | JT | it's not ${CALLERID} anymore |
23:31.28 | *** join/#asterisk threat (n=phix@60-240-43-214.static.tpgi.com.au) |
23:32.01 | hallship | Okay, that's the varaible I have been playing with. And no, setting it doesn't do the trick. |
23:32.06 | [TK]D-Fender | hallship, are you setting "fromuser" in your SIP peer? |
23:33.41 | hallship | Well I'm fairly sure that will work but that doesn't address this issue, in my understanding. becuase I need to send the originating callers callid info. |
23:34.32 | hallship | I can't set the fromuser on the fly right? I haven't found anything showing how that could be done anyway. |
23:35.10 | [TK]D-Fender | hallship, I just asked if you had a very specific value set in your sip.conf peer entry. can you just answer the question...... |
23:36.27 | hallship | no |
23:37.27 | hallship | I haven't changed it in the sip peer. |
23:37.40 | hallship | why do you ask? |
23:39.40 | hallship | [TK]D-Fender and JT thanks, I'll check back after some more digging. I get the feeling this isn't the best time :) |
23:40.06 | [TK]D-Fender | hallship, I asked if you SET it to something |
23:40.25 | [TK]D-Fender | hallship, you are getting all your answers mismatched to the question asked. |
23:41.05 | hallship | What do you mean? I ran this in my dialplan. ;exten => 1235,1,set(fromuser=${CALLERID} |
23:41.27 | hallship | - the ; of course :) |
23:41.31 | [TK]D-Fender | hallship, again WRONG ANSWER. sip.conf. I've said it like 3 times now. |
23:41.54 | [TK]D-Fender | hallship, "fromuser=[somethinghere]" in SIP.CONF. |
23:41.54 | hallship | I did answer that. I said no I haven't set that in my sip peer. |
23:42.22 | JT | hallship: it will never be a good time if you don't answer peoples' questions when they attempt to help you for free :) |
23:42.29 | hallship | Because I don't see how that would help. This needs to be a dynamic/automatic change that happens on each call. |
23:42.38 | [TK]D-Fender | hallship, No, you said "I haven't changed it in the sip peer." changed from WHAT is what I have to ask myself... does this mean it has a value and you haven't been messing with it *lately*? |
23:42.57 | [TK]D-Fender | hallship, I'm trying to pin down all the things you can do that will screw up your attempts |
23:43.28 | [TK]D-Fender | hallship, ambiguity does not help. and that "set" you jsut showed me is not valid. |
23:44.05 | [TK]D-Fender | hallship, That is not a variable of any significance in the dialplan |
23:45.09 | hallship | [TK]D-Fender: that's what I found. I was coming back to try and clarify why that was suggested. That's all. I don't know why we are so far down this road. I am quite sure that the fromuser isn't the correct solution at this point. I merely mentioned it to start a dialogue. |
23:45.40 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
23:45.43 | hallship | JT: I'm doing my best to answer what you guys are asking but you seem to have a bone to pick with me. |
23:46.07 | hallship | I am not trying to be ambibous at all. I am just trying to answer what you are asking. |
23:46.18 | [TK]D-Fender | hallship, No, it could have been involder if you were setting that in your peer entry... wasn't a bad thing to consider, but the wording used was garaunteed to confuse all, and your clairifcations didn't help :) |
23:46.27 | bkruse_home | JT: keeping the nubs under control? |
23:46.40 | [TK]D-Fender | Anyways, I've got to be heading out for a bit. |
23:47.32 | mholman | Hi Guys, I have a small problem with asterisk 1.4 + TDM400P... |
23:47.43 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
23:47.46 | bkruse_home | mholman: let me guess...........it doesnt work? |
23:47.53 | mholman | My ZAP channels seem to be disconnected exactly 30 seconds after they are put on hold. |
23:48.03 | mholman | Does this sound familiar to anyone? |
23:48.21 | JerJer[mobile] | mholman: any SIP channels involve d? |
23:48.28 | hallship | This is a new setup. callerid has never worked correctly. I have removed the fromuser setting out of the sip.conf table. didn't help. I agree that is what is being sent out. I am looking for some sort of work around that will allow me to send the originating callerid instead. |
23:49.18 | mholman | the call is coming through as follows: ZAP -> Asterisk -> SIP -> eyeBeam clients |
23:49.46 | hallship | Okay, thanks for your time. JT and TK. |
23:50.15 | mholman | Ive also tested it using an IAX connection instead of the ZAP one, and on hold works fine. |
23:50.51 | flenders | mholman: a similar thing happened to me, and it went away when I got rid of echo on the TDM400 |
23:51.54 | mholman | flenders: thx, what setting did you change/did you start getting echo as a result? |
23:52.16 | flenders | mholman: also try disabling the busydetect on zapata.conf |
23:52.33 | flenders | mholman: you have to play around with tx/rx gains |
23:52.42 | flenders | mholman: and also use fxotune |
23:53.29 | mholman | flenders: wow great, I'll go and have a play with those settings. thx |
23:53.43 | mholman | flenders: haven't run fxotune at all yet. |
23:54.32 | flenders | mholman: hope that helps |
23:54.53 | bkruse_home | flenders: good advice |
23:55.47 | *** join/#asterisk elriah (i=elriah@adsl-074-164-217-189.sip.bhm.bellsouth.net) |
23:55.54 | elriah | Woohoo! My first wiki contribution: http://www.voip-info.org/wiki/view/Prompt+Recording+with+Unique+IDs |
23:56.11 | bkruse_home | yay |
23:56.19 | mholman | thanks a lot :-) |
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23:57.23 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
23:58.11 | FuriousGeorge | any gentoo users here? |
23:58.25 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
23:58.48 | flenders | asterisk+gentoo? or just gentoo? |
23:58.59 | FuriousGeorge | asterisk plus gentoo of course :) |
23:59.26 | bkruse_home | FuriousGeorge: i HAVE, whats up; |
23:59.33 | FuriousGeorge | im tying to isolate a deadlock problem, and i'm wondering if i should be using the gentoo ebuilds vs svn |
23:59.53 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
23:59.55 | FuriousGeorge | basically i dont know if its software or hadware or what |