| 00:19.49 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 00:23.48 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 00:28.01 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 00:32.06 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:41.49 | *** join/#asterisk vegbox (n=kevinle@wireless-169-235-59-82.ucr.edu) |
| 00:46.27 | *** join/#asterisk ctp (n=ctp@brsg-d9bee9e1.pool.mediaWays.net) |
| 00:46.52 | WindowsUser | theres no dial command in 1.6? |
| 00:47.01 | *** join/#asterisk Gnewt (n=hackerle@because.iownyour.info) |
| 00:47.11 | Gnewt | Hey #asterisk :) |
| 00:47.40 | WindowsUser | I guess i could use originate :-/ |
| 00:47.40 | Gnewt | I'm looking into setting up a simple VoIP setup with two SIP outbound trunks and only a couple extensions, as well as a DISA |
| 00:47.49 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 00:47.51 | Gnewt | I'm on Ubuntu 9.04 |
| 00:48.02 | Gnewt | Wondering if there is a tutorial or something reccomended for a simple setup |
| 00:48.11 | Gnewt | Asterisk scares me :> |
| 00:49.08 | WindowsUser | ~book |
| 00:49.09 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 00:49.29 | sjobeck | hey, hi all, hope every one is well. I am wondering how I can show (hack) a SIP extension to be registered permanently. I have a few SIP extensions that are merely for forwarding. I want them to show as registered just so I can put calls to them (so they will then forward to where they need to). Unfortunately they need to SIP. |
| 00:49.46 | WindowsUser | Gnewt: you just need to read stuff on editing sip.conf and extensions.conf |
| 00:50.30 | WindowsUser | sjobeck: cant just Dial(sip/sjobeck@initech.com) ? |
| 00:51.11 | Gnewt | WindowsUser: Okay, and should apt-get install asterisk work fine? |
| 00:51.34 | WindowsUser | for sip stuff sure |
| 00:51.36 | WindowsUser | whats disa? |
| 00:51.51 | Gnewt | direct inward system access |
| 00:52.05 | Gnewt | basically a DID that passes through to a dialtone once I enter the passcode |
| 00:52.15 | sjobeck | windowsUser: i cant because my 3rd party software wont if it is not active |
| 00:52.17 | WindowsUser | ah |
| 00:52.38 | sjobeck | dumb thing. doingn sanity checks on me. i wont be sanity checked! |
| 00:52.38 | sjobeck | :) |
| 00:52.46 | WindowsUser | sjobeck: so they need to be registered to asterisk? |
| 00:52.54 | sjobeck | y |
| 00:53.08 | WindowsUser | you should look at my nick and realize that sanity checks dont exist for me :) |
| 00:53.24 | sjobeck | that is awesome |
| 00:53.31 | sjobeck | i with you, brother |
| 00:54.03 | sjobeck | i thought about settings expiry to 99999999999 (but reboots & etc) |
| 00:54.14 | sjobeck | i thought about IAX but it makes me use SIP |
| 00:54.25 | sjobeck | i thouht about Custom but it makes me use SIP |
| 00:54.38 | WindowsUser | oh so asterisk has to register with it, and not the other way around? |
| 00:55.34 | sjobeck | well, no, asterisk has to see that these handful of SIP extensions are active (ie: registered, online, reachable, etc, etc) before i can send calls to them |
| 00:55.46 | WindowsUser | pshaw |
| 00:56.01 | sjobeck | when it shows them offline, i wont let me send calls to them & i dead in the water |
| 00:56.09 | WindowsUser | I have a spa3102 and the handset gets registered, but the pstn isn't |
| 00:56.53 | sjobeck | i need some thing analogous to that. the ATA dosent pass presence between those 2 technologies inside of it. |
| 00:56.54 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-a68eceab632061d9) |
| 00:57.47 | WindowsUser | I have host=dynamic for the handset and host=192.168.0.6 and port=5061 for the pstn connection, the rest is the same |
| 00:58.23 | WindowsUser | afk gotta check my bios for serial port setting |
| 00:58.55 | sjobeck | what? bios? lost me. |
| 00:59.05 | sjobeck | thought we were talking SIP. |
| 00:59.43 | *** join/#asterisk BuSyAnToS (n=31749@93-44-80-124.ip96.fastwebnet.it) |
| 01:03.32 | carrar | SIP SAP I was taken a bath |
| 01:03.51 | *** join/#asterisk vegbox (n=kevinle@wireless-169-235-59-82.ucr.edu) |
| 01:04.09 | carrar | And then I saw the girl next door! |
| 01:13.52 | *** join/#asterisk palin (n=jcook@adsl-69-208-73-67.dsl.klmzmi.ameritech.net) |
| 01:14.18 | *** part/#asterisk palin (n=jcook@adsl-69-208-73-67.dsl.klmzmi.ameritech.net) |
| 01:14.29 | *** join/#asterisk blkry (n=blkry@68-115-163-250.static.ahvl.nc.charter.com) |
| 01:14.58 | blkry | I have the issue of all inbound calls dropping after 19 seconds. I have checked settings, switched out the router and the problem continues. The problem started when we switched ISP's to charter. I am starting to think that chater is blocking port 5060. searches don't suggest it though. |
| 01:16.06 | blkry | outbound calls are fine. |
| 01:20.05 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 01:20.45 | tengulre | hi,all |
| 01:40.47 | *** join/#asterisk zdennis (n=zdennis@24-236-150-85.dhcp.aldl.mi.charter.com) |
| 01:41.10 | *** join/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 01:41.10 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:41.23 | zdennis | I'm trying to a basic demo going with asterisk... each time I run "asterisk -r"... I continually get "Connect attempt from '127.0.0.1' unable to authenticate" every few seconds |
| 01:41.44 | zdennis | I can run asterisk commands just fine in the CLI console |
| 01:41.51 | zdennis | but I don't know how to find what is trying to connect |
| 01:42.11 | zdennis | I'm running the AsteriskNow image |
| 01:42.39 | zdennis | any pointers? |
| 01:44.01 | zdennis | nevermind, I think I found some pertinent info |
| 01:46.06 | *** join/#asterisk Kumbang (n=kellyswe@rusnas.paume.itb.ac.id) |
| 02:02.10 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 02:11.02 | *** join/#asterisk micols (n=mio@rlogin.dk) |
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| 02:46.43 | *** part/#asterisk lanning (n=lanning@173.8.187.197) |
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| 02:55.07 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:58.48 | *** join/#asterisk ctp (n=ctp@brsg-d9bef6be.pool.mediaWays.net) |
| 03:00.12 | WindowsUser | sjobeck: hows the SIP stuff? |
| 03:00.34 | sjobeck | thx for checking. no where. i afraid. |
| 03:00.43 | sjobeck | wondering how to hack the astdb now. |
| 03:00.51 | sjobeck | but ....... dont want to b0rk it |
| 03:01.18 | sjobeck | i could cron some sort of asterisk -rx "db set blah" to keep those extensions registered but yug ! |
| 03:01.21 | sjobeck | yuck |
| 03:01.22 | sjobeck | yug |
| 03:01.22 | sjobeck | guh |
| 03:01.23 | sjobeck | gah |
| 03:03.15 | WindowsUser | i dont see why you cant just dial(sip/user@device) |
| 03:03.48 | sjobeck | oh, sure, i with you, i sure i can, you can, script can, phones can, but ......... the stoooopid 3rd party app' cant. |
| 03:04.06 | sjobeck | it wont let you xfer or a call to, or dial a , ext that is offline |
| 03:08.41 | *** join/#asterisk tobias (n=tobias@user-0ce2hp1.cable.mindspring.com) |
| 03:18.54 | *** join/#asterisk OrNix (n=ornix@78.40.81.34) |
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| 03:32.49 | *** part/#asterisk yoblooc (n=yoblooc@210.83.214.163) |
| 03:34.42 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 03:53.06 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
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| 03:59.49 | *** join/#asterisk Borai (n=DYN@S0106001c109e98db.no.shawcable.net) |
| 04:00.09 | Borai | hi, XXX was not compiled with the same compile-time options as this version of Asterisk. what does that mean? |
| 04:04.19 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 04:13.58 | *** join/#asterisk Chinorro (n=Chino@54.225.117.91.dynamic.mundo-r.com) |
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| 04:29.47 | *** join/#asterisk techie (n=techie@60.sub-70-213-67.myvzw.com) |
| 04:32.48 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
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| 04:48.22 | Gnewt | Hey #asterisk |
| 04:48.31 | Gnewt | So I just installed asterisk on an Ubuntu 9.04 system |
| 04:48.41 | Gnewt | did some basic sip.conf and extensions.conf config |
| 04:48.49 | Gnewt | added all the right exceptions in my fw |
| 04:49.11 | Gnewt | But when I nmap the server I get nothing |
| 04:49.14 | Gnewt | as in no 5060 open |
| 04:49.20 | Gnewt | can't connect to it with a softphone either |
| 04:49.34 | carrar | aan you nmap from inside your firewall? |
| 04:51.42 | Gnewt | yeah one sec |
| 04:52.20 | Gnewt | No asterisk-related ports open |
| 04:52.34 | carrar | guess you need to go back and setup asterisk |
| 04:53.16 | carrar | bbl |
| 04:53.19 | [TK]D-Fender | And of course I'd blindly trust the testing methodology? |
| 04:53.28 | [TK]D-Fender | carrar: Wouldn't you? ;0 |
| 04:53.31 | carrar | hahha |
| 04:53.34 | carrar | oh yeah |
| 04:54.08 | Gnewt | Asterisk should listen on 5060 by default, right? |
| 04:54.39 | carrar | listen to whatever you have co nfigured |
| 04:54.44 | [TK]D-Fender | I love non-qualified numbers :) |
| 04:55.18 | [TK]D-Fender | Gnewt: Pastebin your tests and your firewall dump. |
| 04:56.01 | Gnewt | one sec |
| 04:57.59 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 04:58.20 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 04:59.22 | *** join/#asterisk errotan (n=errotan@5403E419.catv.pool.telekom.hu) |
| 04:59.41 | Gnewt | http://gnewt.pastebin.com/m410d6889 |
| 05:03.19 | Gnewt | [TK]D-Fender |
| 05:03.22 | [TK]D-Fender | Gnewt: And so far I'm seeing a TCP scan. SIP = UDP |
| 05:04.11 | Gnewt | smacks self |
| 05:04.13 | Gnewt | one sec |
| 05:04.38 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 05:05.26 | ectospasm | Why does MS do SIP over TCP? I guess it doesn't matter at that point... |
| 05:07.33 | [TK]D-Fender | ectospasm: SIP also supports TCP, and because of that allows for persistent connections. Better for NAT traversal of the signalling protoco |
| 05:07.35 | [TK]D-Fender | l |
| 05:07.49 | Gnewt | [TK]D-Fender: 5060/udp open|filtered sip |
| 05:09.33 | Gnewt | But no entry from asterisk -rvvvvvvv when I try to connect with X-Lite |
| 05:10.24 | [TK]D-Fender | Gnewt: And that doesn't tell me that you enabled SIP DEBUG. |
| 05:10.59 | [TK]D-Fender | Gnewt: Might help to get a better description of what is running where... |
| 05:12.50 | Gnewt | Ahh just enabled sip debug, realized what is most likely the problem: <--- Transmitting (no NAT) to 10.0.1.4:39553 ---> |
| 05:12.58 | Gnewt | 10.0.1.4 is my IP |
| 05:13.02 | Gnewt | in my NAT |
| 05:13.08 | Gnewt | my Asterisk server is out of NAT |
| 05:14.57 | Gnewt | in X-Lite "Discover global address" is on though |
| 05:15.49 | WindowsUser | well it didn't work then :) |
| 05:21.09 | [TK]D-Fender | And * shouldn't CARE what IP its claiming to be.... go read the guide |
| 05:21.10 | [TK]D-Fender | ~sipnat |
| 05:21.11 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 05:33.07 | [TK]D-Fender | Checkout time, later all |
| 05:33.33 | carrar | Gnewt, nice reverse |
| 05:33.34 | carrar | because.iownyour.info |
| 05:33.57 | Zuchmir2 | <[TK]D-Fender>: FYI the issue was ALG was enabled on the firewall, after disabling, the DTMF worked |
| 05:34.23 | Zuchmir2 | <[TK]D-Fender>: thanks for your help |
| 05:34.39 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 05:36.31 | Zuchmir2 | anyone with a clue why ALG would cause problems? |
| 05:38.18 | carrar | http://tinyurl.com/n8w3cj |
| 05:39.23 | *** join/#asterisk jsgoecke (n=Adium@c-71-202-25-141.hsd1.ca.comcast.net) |
| 05:42.10 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-67-247-172-185.rochester.res.rr.com) |
| 05:44.12 | Zuchmir2 | my question is, i would think ALG enabled would HELP the RTP traffic get through the firewall, instead when it was ENABLED the DTMF was NOT getting through |
| 05:45.15 | kaldemar | ALG's usually just screw up things in many ways when NAT and asterisk are involved. |
| 05:55.19 | Zuchmir2 | ok, thanks all |
| 05:56.47 | *** part/#asterisk jsgoecke (n=Adium@c-71-202-25-141.hsd1.ca.comcast.net) |
| 05:59.37 | frk2 | hey- anybody here used the chinese AT 620 phones that just came out? My past experience with Atcom has been exceptionally bad |
| 06:03.41 | *** part/#asterisk pfake (i=blank@slu.ms) |
| 06:08.11 | joobie | hey guys.. is it possible with the AMI interface, to give write permissions to only one command? Looks like you can only specify 'write = command', which lets the user send any command .. i want to restrict this to just 1 command...... |
| 06:22.45 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
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| 06:49.54 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
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| 07:09.20 | *** join/#asterisk blkry (n=blkry@96.37.27.72) |
| 07:11.28 | jql | good for you |
| 07:11.42 | jql | I am at home, but only drinking beer |
| 07:13.38 | nkohh | interesting |
| 07:14.22 | nkohh | i guess we know how you feel about drinking alone |
| 07:18.46 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
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| 07:54.26 | *** join/#asterisk Damir (n=da@89-201-235-215.dsl.optinet.hr) |
| 07:55.34 | Damir | hello, is there a version features compare link for asterisk. I am little confused and i don't know which version should i install |
| 07:56.11 | Damir | 1.6.1.1 / 1.6.0.10 / 1.4.2.51 |
| 07:56.13 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-c0e6738e68fd84ed) |
| 07:57.04 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:57.49 | *** join/#asterisk Chris-NB (n=chris@85-126-61-10.work.xdsl-line.inode.at) |
| 08:02.35 | mort_gib | Damir: How adventerous are you?? |
| 08:04.54 | Damir | much :) |
| 08:05.14 | mort_gib | Damir: then DO go bleeding edge |
| 08:06.06 | *** join/#asterisk juny4ng (i=juny4ng@119.234.37.237) |
| 08:06.16 | juny4ng | hi guys, anyone around who can help me with the asterisk manager inteface? |
| 08:06.59 | mort_gib | 1.4.XX is the conservative stable choice |
| 08:07.10 | mort_gib | 1.6.0.XX is becoming stable |
| 08:07.14 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:07.33 | mort_gib | 1.6.1.XX will allow you to get your hands dirty with some interesting things |
| 08:08.10 | mort_gib | devicestate and Bri support in Dahdi is only for 1.6.XX |
| 08:08.28 | mort_gib | Allthough point-to-multipoint is not yet supported |
| 08:08.37 | Damir | thanks for information |
| 08:08.52 | mort_gib | np |
| 08:09.13 | mort_gib | -If you are starting with Asterisk, then go 1.6.1.XX |
| 08:11.32 | *** join/#asterisk JackTheNipple (n=JackTheN@dialbs-213-023-036-202.static.arcor-ip.net) |
| 08:12.36 | juny4ng | Is it possible to originate a call to a specific extension automatically and then play out some voice (using TTS) based on certain parameters i can pass in? Afterwards, seek an acknowledgement (e.g. press 1 to acknowledge) and then hangup |
| 08:13.19 | jql | I could say "yes", but that would seem to obligate me towards a demonstration |
| 08:13.39 | juny4ng | alright thanks |
| 08:13.43 | juny4ng | nah, no problem |
| 08:13.54 | juny4ng | i was looking at originate action via the manager inteface |
| 08:14.06 | juny4ng | but that somehow requires the other channel to answer the call first |
| 08:14.13 | *** join/#asterisk jgoo (n=r3rman@athedsl-4535406.home.otenet.gr) |
| 08:14.31 | jql | yes, but when they do it jumps to the context you told it |
| 08:14.37 | jql | which can then do... whatever |
| 08:15.05 | jgoo | hey feeps, so I am setting up an SPA 2102 - and all looks right, I am connected on ethernet port - and it isn't registering to my pbx - does it force itself to connect via the internet port? |
| 08:15.23 | jgoo | I am googling but WTF, this should be the number 1 thing on the frikkin google, along with 110# 210# |
| 08:15.31 | juny4ng | jql: alright thanks man, i'll try it out |
| 08:16.14 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:17.57 | jgoo | does anyone know if the SPA2102 only registers to SIP on the internet port, not the ethernet port? |
| 08:18.31 | jgoo | fuckkkkkkkit gotta sleep macbook now |
| 08:22.20 | *** join/#asterisk jgoo (n=r3rman@athedsl-4535406.home.otenet.gr) |
| 08:22.31 | jgoo | so, right, wtf is the SPA2102's problem? |
| 08:23.44 | WindowsUser | it wont register on the lan port? good to know |
| 08:24.17 | jgoo | I don't know - I could try, but I don't know if it will fusk itself up if I make the ethernet and internet two ips on the same network... it shouldn't |
| 08:24.23 | jgoo | afterall, they are wire-agnorstic |
| 08:24.35 | jgoo | agnostic, damn deskis at wrong height |
| 08:24.52 | LinuxUser | WindowsUser, you suck |
| 08:24.55 | jgoo | anyway |
| 08:25.38 | jgoo | I am trying to use it as a PAP2, which all work absolutely fine, and it amazes me that some products actually work |
| 08:25.54 | jgoo | whereas almost every wireless device I own locks up after more than 60 seconds of clicking on the web interface |
| 08:25.56 | WindowsUser | if the ips are on the same network you'll have issues as to which one is used |
| 08:26.18 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 08:26.34 | jgoo | no... if they are different ips |
| 08:26.54 | jgoo | WindowsUser, as long as it doesn't become inaccesible, I was up late last night, and I usually write the default on a sticker and slap them on the bitch |
| 08:27.21 | jgoo | oh I did... that is how tired I was... so anyway, I'll try regardless, I was wanted to vent |
| 08:28.26 | jgoo | Oh, and FUCK my routers dhcp service, piece of shit just stops responding, and when you are switching between lots of repeaters and wired / wireless networks, suddenly not connecting because DHCP freezes is the most FUCKING TRIG thing to try and work out. |
| 08:28.28 | jgoo | heh. |
| 08:28.51 | *** join/#asterisk BeeBuu (n=beebuu@59.38.99.58) |
| 08:29.17 | WindowsUser | haha |
| 08:30.01 | BeeBuu | any one had tried conference manager from asterikast? |
| 08:30.26 | WindowsUser | I've had issues with every consumer router I've used |
| 08:36.11 | *** join/#asterisk ISO9001 (i=blank@slu.ms) |
| 08:41.50 | *** join/#asterisk LakeSolon (n=blake@173-21-208-150.client.mchsi.com) |
| 08:53.51 | BeeBuu | is there any one had tried conference manager from asterikast? |
| 08:56.04 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-93d3587e5a332984) |
| 08:58.17 | *** join/#asterisk stimpie (n=stimpie@84-104-5-142.cable.quicknet.nl) |
| 09:03.46 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
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| 09:34.21 | *** join/#asterisk Exstatica (i=Exstatic@freenode/staff/exstatica) |
| 09:35.23 | Exstatica | anyone ever had an issue with asterisk and festival? i show asterisks connecting to the festival server but nothing happens, it just terminates |
| 09:35.43 | Exstatica | festival log shows client connecting, and then disconnecting |
| 09:36.48 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-203.gibnet.gi) |
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| 10:25.56 | *** join/#asterisk joobie (n=joobie@210-84-27-242.dyn.iinet.net.au) |
| 10:26.03 | zagibu | what about capi? My boss just threw this at my head, but I have no idea what it is...i read it is an API for ISDN control? Is that it? |
| 10:26.14 | joobie | hey guys.. is it possible with the AMI interface, to give write permissions to only one command? Looks like you can only specify 'write = command', which lets the user send any command .. i want to restrict this to just 1 command...... |
| 10:27.09 | angryuser | zagibu, and what your boss want to do with capi ? |
| 10:27.48 | zagibu | that's what i'm trying to figure out...he's usually kinda vague and doesn't know what he wants himself most of the time... |
| 10:28.22 | joobie | tell him he is an ass |
| 10:28.35 | zagibu | nah, i like the money he pays |
| 10:28.49 | angryuser | zagibu, try to figure what he wants |
| 10:28.57 | hi365 | joobie: nope |
| 10:29.03 | joobie | telling him he is an ass is worth more than money |
| 10:29.03 | zagibu | yeah, I try asking him again |
| 10:29.13 | joobie | hi365, doh |
| 10:29.13 | *** join/#asterisk dexteruk (n=dexteruk@hst-8-67.cisbg.com) |
| 10:29.14 | *** part/#asterisk khronos (i=tom@sip02.kineticvoip.com) |
| 10:29.16 | joobie | thanks |
| 10:29.16 | zagibu | bah, I don't believe in karma |
| 10:29.30 | dexteruk | can anyone help me with a PRI |
| 10:29.50 | dexteruk | how do you set the MSN that the PRI will answer on |
| 10:30.11 | dexteruk | when calling the pri number i get busy |
| 10:30.25 | dexteruk | I have inbound routs for the DID number |
| 10:31.01 | angryuser | dexteruk, pastebin traces and a context where that call gets in |
| 10:31.48 | dexteruk | Im using asterisknow. I know there is a setting somewhere in zapata that you set it to * |
| 10:32.16 | dexteruk | to answer all inbound numbers |
| 10:32.21 | angryuser | dexteruk, are you sure that somewhere there is a setting ? |
| 10:32.44 | dexteruk | Im sure I did it a number of years ago, so things may have changes |
| 10:32.52 | angryuser | dexteruk, pastebin traces and a context where that call gets in |
| 10:32.57 | viraptor | does anyone know if rtpkeepalive depends on nat=yes, or not? |
| 10:33.40 | angryuser | anyway i need to go eat something xD bon appetit to all gmt + 2 |
| 10:34.36 | angryuser | viraptor, it does not depends on it.......normally |
| 10:35.11 | zagibu | just a last question about capi: is asterisk capable of this per default, or do you have to extend it? I'm asking because we use a "proprietary" asterisk appliance that doesn't like meddling... |
| 10:35.53 | viraptor | angryuser: what do you mean by "normally"? I wanted to know if I have to set nat=yes if I have a phone which is not behind nat, but I still want to have the rtpkeepalive sent |
| 10:36.06 | angryuser | zagibu, forget about capi... |
| 10:36.13 | zagibu | okay |
| 10:37.28 | angryuser | viraptor, qualify=yes does not care about nat |
| 10:38.40 | viraptor | angryuser: so rtpkeepalive will be used only if qualify=yes? |
| 10:39.15 | angryuser | zagibu, capi is old you dont want to mes around with bri and capi, if you want some social life of course |
| 10:39.22 | angryuser | mess* |
| 10:39.29 | angryuser | viraptor, yes |
| 10:40.47 | angryuser | viraptor, basicly it is used to know if the phone reachable or not, but as a side effect this option keeps ports open in nat table |
| 10:40.57 | angryuser | im out |
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| 10:57.34 | *** join/#asterisk jkroon (n=jkroon@dsl-240-167-45.telkomadsl.co.za) |
| 11:00.04 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 11:05.24 | jkroon | with isdn (capi specifically), what is the purpose of the nationalprefix and internationalprefix options? |
| 11:08.28 | Chainsaw | jkroon: The national prefix is 0 in most of the world, but 1 in the United States. It's the number you put in front of a full phone number, dialled nationally. |
| 11:08.46 | Chainsaw | jkroon: The international prefix is 00 in most of the world. It's the number you put in front of the country code. |
| 11:09.21 | Chainsaw | jkroon: (Generally abbreviated as +, it's whatever you replace that + with) |
| 11:09.32 | jkroon | that makes sense and is what I figured, why does asterisk need to know this? |
| 11:10.08 | Chainsaw | jkroon: Because it will be receiving & sending phone numbers over ISDN. |
| 11:10.27 | Chainsaw | jkroon: Getting the prefix wrong could result in a dropped or misrouted call. |
| 11:11.23 | jkroon | ok, working on a box in china, can i safely assume the prefixes are 0 and 00? If I then intend to Dial() a number locally, do I still dial it with the 0 prefix for local calls as with analog lines? |
| 11:11.51 | jkroon | and how do I set up CLI, because all I'm getting out of it currently is private number/number witheld. |
| 11:16.25 | mort_gib | Is there a sip client out for Nokias E71?? |
| 11:16.25 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 11:16.34 | Chainsaw | jkroon: I'm not familiar with ISDN/PSTN setup in China, sorry. |
| 11:16.51 | Chainsaw | jkroon: Perhaps others have those details for you, otherwise I'd ask google about it. |
| 11:17.17 | jkroon | i have asked google. |
| 11:18.17 | jkroon | ok, i had a similar issue in ZA, can get the clid off an incoming call just fine, as well as actually make calls out (nationally at least, by stripping off the leading zero when dialing), but I could not get outbound CLI working (this was with a quad-span Digium PRI card). |
| 11:18.24 | Pan3D | mort_gib: doesn't the Nokia have a built-in client? |
| 11:18.40 | Pan3D | http://www.geek.com/articles/mobile/feature-voip-with-nokia-e71-how-to-2008095/ |
| 11:18.53 | mort_gib | Pan3D: On the "tech spec" it says "VOIP Support"?? |
| 11:19.16 | Pan3D | right, and if you follow the link above, you will have teh details on how to configure it |
| 11:19.22 | Pan3D | it's just a couple menus |
| 11:19.46 | Chainsaw | jkroon: Outbound CLI tends to be filtered quite aggressively to avoid spoofing. |
| 11:19.56 | Chainsaw | jkroon: If you don't get it exactly right, your telco will likely strip it off. |
| 11:20.40 | jkroon | this is what i guessed, but how do i go about "getting it right", the telco (telkom, in ZA) at least is useless when attempting to obtain such information. |
| 11:20.59 | mort_gib | Pan3D: Hey that was VERY helpful!! Thanks! |
| 11:21.15 | Chainsaw | jkroon: If you have a number block assigned, try the very first number in the block. xxx00 |
| 11:21.19 | Pan3D | mort_gib: np |
| 11:21.42 | mort_gib | I'll buying that asap! |
| 11:21.50 | jkroon | I know what the numbers are I need to dial from the outside world to get to the PRI in ZA and that works, the number being presented as the destination extension is exactly the last 4 digits of the number dialed, with the CallerID(num) being the full number. |
| 11:22.50 | jkroon | i can dial out (minus the leading zero), but am not getting CLI, I've tried setting CallerID(num)=0123456789, =123456789 or 6789 and all of them give me private number. |
| 11:25.21 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 11:29.59 | Chainsaw | jkroon: The suggestion seems to be to bolt 0011 to the front of the number. Source: http://mybroadband.co.za/vb/showthread.php?t=96301 |
| 11:33.54 | *** join/#asterisk cusco (n=tralala@2001:0:53aa:64c:341e:7921:2ac0:762d) |
| 11:34.04 | cusco | hi |
| 11:37.36 | jkroon | Chainsaw, 0011 followed by 7 digits, so I'm guessing the 0011 refers to the jhb area code of 011, so in my case I should probably use 0012, thanks. |
| 11:37.42 | jkroon | now to figure out the china version. |
| 11:44.16 | joobie | guys anyone played with the sms feature of asterisk? |
| 11:44.41 | joobie | how the freak does it work? i mean does it require some sort of 3rd part provider to send an SMS? |
| 11:48.47 | viraptor | hi, let's say an asterisk (1.4) process suddenly starts slowing down, ignoring packets, it handles ~270 channels (same as every day), but it slowed down to a crawl only today - has anyone got good ideas on how to debug it? |
| 11:49.22 | *** join/#asterisk truz (n=truz@78.97.44.14) |
| 11:49.39 | truz | hello all |
| 11:49.44 | truz | can some1 help me with one problem ? |
| 11:49.55 | truz | i have a problem when i try to make an IVR menu |
| 11:49.55 | zagibu | what problem? |
| 11:50.07 | truz | WARNING[12269] pbx.c: No application 'DigitTimeout' for extension (mainmenu, s, 4) |
| 11:50.19 | truz | and my call hangup |
| 11:51.29 | wdoekes | perhaps there is no application DigitTimeout then? |
| 11:52.11 | wdoekes | pb the relevant parts of your extensions.conf |
| 11:53.27 | truz | wdoekes : i think there is no application for DigitTimeout but where can i get one ? |
| 11:53.38 | truz | can i paste my mainmenu from my extenstion.conf ? |
| 11:53.42 | wdoekes | ~pb |
| 11:53.43 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 11:54.34 | *** join/#asterisk AsteriskDom (n=amunoz@75.112.88.200.m.sta.codetel.net.do) |
| 11:55.40 | truz | http://pastebin.com/d70653d2 |
| 11:56.01 | truz | it's an example that i found it on web . |
| 11:56.20 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 11:57.16 | wdoekes | remove lines 4 and 5, and add after six: exten => s,7,WaitExten(10) (and renumber all) |
| 12:01.02 | wdoekes | (you can use the letter n instead of numbers, except for the first number (1). that way it will be easier to add/remove lines) |
| 12:01.23 | *** join/#asterisk ik_5 (n=ik@95.211.21.45) |
| 12:02.59 | truz | ty :) |
| 12:03.01 | truz | it works :P |
| 12:03.50 | *** join/#asterisk Jacke (i=jacke@valhalla.bofh.org) |
| 12:05.12 | *** join/#asterisk SuPrSluG (n=SuPrSluG@72.237.213.162) |
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| 12:07.01 | mechbangirc | hi how can I override my peer's caller ID in sip.conf |
| 12:08.55 | angryuser | Pan3D, noka's sip client is bad, use fring, he got sip profile and working nice |
| 12:11.35 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:12.12 | *** join/#asterisk zagibu (n=adsc@cust.static.213-180-179-82.cybernet.ch) |
| 12:13.36 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:17.27 | jkroon | Chainsaw, the 0011 (or 0012) prefix doesn't work either. |
| 12:17.52 | Chainsaw | jkroon: They pointed at a mailing list in that forum post. I hope they can help you there. |
| 12:18.17 | jkroon | so do I. thanks for the assistance, didn't think you were going to help further, just wanted to let you know. |
| 12:18.50 | Chainsaw | jkroon: I would if I was in South Africa. But I'm in the United Kingdom. Unless it's about BT's peculiar timings or connectors... my knowledge is limited. |
| 12:19.33 | jkroon | Chainsaw: lol. sounds like fun. rather you than me. then again ... i still need to go hunt an iax/2<->sip timing issue. |
| 12:22.40 | ik_5 | how do i do "not" inside an if of AEL ? |
| 12:34.32 | *** join/#asterisk zagibu (n=adsc@cust.static.213-180-179-65.cybernet.ch) |
| 12:39.43 | tzafrir_laptop | '!' ? |
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| 12:50.37 | zagibu | oookay, my boss gave another barely understandable speech about capi, and I think that maybe I know what he wants to do now...he thinks CAPI is some sort of magical interface between CRM/ERP applications and telephony |
| 12:51.28 | zagibu | he probably wants to install capi on a server somewhere, then point the CRM/ERP software and telephony software both to this server, so that they then magically cooperate |
| 12:53.11 | Chainsaw | zagibu: With faeries. And unicorns! |
| 12:55.51 | [TK]D-Fender | zagibu: that would be TAPI, not CAPI |
| 12:56.19 | [TK]D-Fender | zagibu: http://en.wikipedia.org/wiki/Telephony_Application_Programming_Interface |
| 12:57.09 | zagibu | i know tapi, he just came to me babbling about capi |
| 12:58.34 | [TK]D-Fender | zagibu: http://en.wikipedia.org/wiki/Common_ISDN_Application_Programming_Interface |
| 12:58.35 | zagibu | i think he confuses the expressions, but still, i have a document here from a company that produces a capi solution, and asterisk is in the compatibility list |
| 13:00.43 | zagibu | strangely enough, the solution requires Microsoft Windows as OS |
| 13:01.59 | zagibu | hmmm, seems it is talking to the asterisk server as a SIP client |
| 13:02.11 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 13:02.21 | mort_gib | zagibu: TK is right... you boss is talking about TAPI |
| 13:03.14 | zagibu | i know, what he WANTS to do is TAPI, but what he has in hand is indeed capi |
| 13:03.34 | zagibu | it's called Xcapi, a solution of a german company named te-systems |
| 13:04.05 | mort_gib | zagibu: Give him a Chineese fingertrap, save yourself time and sanity |
| 13:04.12 | zagibu | hehe, good idea |
| 13:04.43 | viraptor | I've enabled qualify on the peer, as angryuser suggested before, but still - rtpkeepalive doesn't seem to work - is there something I have to do to enable it? |
| 13:07.06 | angryuser | viraptor, why do you need it ? |
| 13:07.26 | angryuser | viraptor, you phone disconnect conversation ? |
| 13:07.45 | angryuser | viraptor, what version of asterisk do you use ? |
| 13:08.37 | viraptor | I have to interface with ms ocs gateway, it disconnects any call which is on hold for exactly 30 seconds - I saw some post saying that it can be solved with rtpkeepalive-s, but the comfort noise packets are simply not generated on my side |
| 13:08.38 | viraptor | 1.6.1.1 |
| 13:09.59 | viraptor | just to confuse me more, it does generate them in a test setup between my 2 snoms, but as soon as it goes to the real network, it stops sending keepalives |
| 13:10.23 | angryuser | viraptor, confrt noise is not complete in asterisk, what value do you have for rtpkeepalive ? |
| 13:10.55 | viraptor | 2 |
| 13:11.12 | *** join/#asterisk ruben23 (n=RPL@124.107.3.178) |
| 13:11.40 | Katty | mew. |
| 13:11.47 | angryuser | viraptor, can you put some music on hold during "on hold "? |
| 13:11.52 | mort_gib | Hi Katty |
| 13:12.01 | Katty | hi mort |
| 13:12.14 | *** join/#asterisk zdennis (n=zdennis@24-236-150-85.dhcp.aldl.mi.charter.com) |
| 13:13.09 | viraptor | angryuser: I mean a situation when I dial OCS gateway, it answers and puts me on hold - it sends the music, but asterisk doesn't send anything (as it should) |
| 13:13.38 | angryuser | viraptor, have you read this ? is it you post ? http://social.microsoft.com/Forums/en-US/communicationsservertelephony/thread/898c1b60-af05-4678-ae41-0550f282e286 |
| 13:13.40 | viraptor | I want to change it, so that asterisk sends comfort noise even when it's in recvonly mode |
| 13:13.54 | angryuser | your* |
| 13:14.21 | ruben23 | hi guys, when i run asterisk -rvvvvv i got this ------> Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) , tried running asterisk it worked, but when rebooted still the same i got the same error. |
| 13:14.25 | angryuser | viraptor, there is a solution at the end |
| 13:15.34 | angryuser | ruben23, try tu run asterisk with "asterisk -vvvvv" ans see where it crashes |
| 13:15.34 | viraptor | angryuser: yes - I'm trying to do exactly that... the problem I have is that it works when I test it with 2 snoms, but doesn't work on the real server (it just doesn't send those packets) |
| 13:15.47 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:15.49 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:16.00 | Katty | hi _ShrikE |
| 13:16.36 | viraptor | the only difference I can see is that real server uses 172.... network, everything else is the same really |
| 13:16.39 | angryuser | viraptor, you have added nat=yes in [general] section ? |
| 13:16.43 | ruben23 | angryuser:Asterisk already running on /var/run/asterisk.ctl. Use 'asterisk -r' to connect. |
| 13:16.44 | _ShrikE | Hugs Katty |
| 13:16.46 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:c452:e19f:d2b1:7757) |
| 13:16.52 | ruben23 | i got that messages |
| 13:17.35 | viraptor | tried with =yes and =no - neither of them works |
| 13:17.41 | angryuser | viraptor, have you entered localnet=172.x in sip.conf ? |
| 13:18.40 | viraptor | angryuser: do I need to? it's a multihomed host, 172.... is not a natted network, it's a vpn |
| 13:18.57 | ruben23 | <PROTECTED> |
| 13:19.02 | *** part/#asterisk ik_5 (n=ik@95.211.21.45) |
| 13:19.08 | viraptor | angryuser: also I have to support the public network, none of them are firewalled / natted |
| 13:19.20 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 13:19.28 | angryuser | viraptor, asterisk dont know what is local or what is not unless you told him |
| 13:19.48 | angryuser | viraptor, so you got all you peers with nat=yes ? |
| 13:20.13 | viraptor | no, only defined a global nat=yes |
| 13:20.37 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 13:21.24 | angryuser | viraptor, personal setting overwrites general, si if you set nat=no in your [peer] it will be off, you can check it with "sip show peers" NAT column |
| 13:21.51 | *** join/#asterisk tobias (n=tobias@216.27.28.176) |
| 13:22.05 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 13:22.05 | *** mode/#asterisk [+o denon] by ChanServ |
| 13:22.11 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
| 13:22.24 | angryuser | ruben23, many things, are you sure that you are root ? xD |
| 13:22.38 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 13:23.02 | [TK]D-Fender | [09:18]<ruben23> angryuser:what could be the problem asterisk wont automatically start on reboot <- because you didn't set up a startup script for it |
| 13:23.27 | angryuser | ruben23, try to see if asterisk is started "ps axjf | grep asterisk" |
| 13:24.10 | viraptor | angryuser: ok - changed it, user was overwriting the change... let me test that now... |
| 13:24.13 | brah | Also, 'ls /etc/init.d | grep asterisk' |
| 13:24.26 | SuPrSluG | anyone heard of dtmf issues with level 3 ? It doesn't consistently receive the input consistently. I've been using rcf2833 |
| 13:25.03 | ruben23 | angryuser: http://pastebin.com/mb7e0f5e |
| 13:25.04 | angryuser | Subdolus, you mean rfc2833 not rcf2833 ? |
| 13:25.41 | ruben23 | brah:i got no message with that command |
| 13:25.42 | SuPrSluG | oh forgot to mention I'm using 1.6.06 |
| 13:25.47 | angryuser | <SuPrSluG> ^^ |
| 13:26.23 | brah | Then, if you-re running FreeBSD, 'ls /etc/rc.d | grep asterisk' |
| 13:26.35 | ruben23 | im on centos |
| 13:26.40 | brah | Red hat? |
| 13:26.45 | ruben23 | centos |
| 13:26.48 | ruben23 | yeah |
| 13:26.50 | SuPrSluG | yes rfc2833 |
| 13:27.14 | brah | Then probably asterisk is not autostarting because it isn't set to. |
| 13:27.33 | ruben23 | [TK]D-Fender: it do start on reboot- but as i call suddenly it stop form doin it, until form my last reboot |
| 13:28.41 | brah | Do |
| 13:28.44 | ruben23 | brah: it was set during the compile, but suddenly it stop-how do i set it again a startup script for my asterisk |
| 13:28.56 | SuPrSluG | freeswitch also had this problem and documented it was an issue with the way they handled the rtp stream. wondering if asterisk had the same issue and if there was a fix in the newer versions |
| 13:28.57 | brah | ln -s `which asterisk` /etc/init.d/ |
| 13:29.26 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 13:29.30 | ruben23 | brah: nothing at all |
| 13:29.35 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:29.38 | brah | :) |
| 13:29.41 | ruben23 | it empty |
| 13:29.52 | brah | ls /etc/init.d | grep asterisk |
| 13:30.15 | ruben23 | i got asterisk |
| 13:30.57 | brah | Then that's all. |
| 13:31.00 | brah | Check if it works. |
| 13:32.16 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 13:33.55 | ruben23 | ill start rebooting it again now |
| 13:34.07 | [TK]D-Fender | ruben23: What did you actually DO? |
| 13:35.28 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 13:37.25 | ruben23 | i have multiple callers- suddenly it stop- so i try to reboot my asterisk server then- i try couple reboot asterisk wont start_ it will just start if a run asterisk command-but when rebooted still go this error--------->Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 13:37.42 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 13:38.07 | [TK]D-Fender | ruben23: What did you actually DO? <--- |
| 13:39.09 | Katty | sleepies. |
| 13:39.23 | *** join/#asterisk chantanito (n=santiago@200-71-150-172.static.telcel.net.ve) |
| 13:39.33 | chantanito | hi... |
| 13:39.48 | brah | ruben23, asterisk -cvvvvvvvv |
| 13:39.53 | brah | Is it starting gracefully? |
| 13:40.46 | ruben23 | yes |
| 13:42.02 | chantanito | i'm configuring zaptel in my server right now.. |
| 13:42.08 | brah | Check the permissions of /etc/init.d/asterisk |
| 13:42.08 | chantanito | i just have one doubt... |
| 13:42.15 | brah | ls -l /etc/init.d | grep asterisk |
| 13:42.20 | [TK]D-Fender | brah: looking at the wrong scale of things.... |
| 13:42.28 | chantanito | what does the bchan and dchan parameters mean? |
| 13:42.39 | [TK]D-Fender | ruben23: What did you DO to try to tell your system to start * on boot? |
| 13:42.42 | brah | I don't see why it should not be starting if doing it manually starts gracefully. |
| 13:42.44 | chantanito | does --> do |
| 13:42.52 | chantanito | please |
| 13:43.05 | brah | I told him to copy a soft symlink of `which asterisk` in /etc/init.d |
| 13:43.12 | [TK]D-Fender | brah: BEcause I don't see him configuring it to start. Just having a startup script doesn't say that it is USED |
| 13:43.12 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:43.36 | [TK]D-Fender | chantanito: they are for PRI |
| 13:44.13 | *** join/#asterisk SlipperyChicken (n=andrew@69.159.231.121) |
| 13:45.25 | brah | Also CentOS uses chkconfig |
| 13:45.36 | [TK]D-Fender | brFINALLY someone says it.. |
| 13:45.41 | [TK]D-Fender | brah: FINALLY someone says it.. |
| 13:46.10 | brah | Over 9000 hours in google. |
| 13:46.24 | [TK]D-Fender | ruben23: "chkconfig --list|grep asterisk |
| 13:49.02 | ruben23 | execute it done. |
| 13:51.06 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 13:51.28 | [TK]D-Fender | ruben23: PASTE the result |
| 13:51.52 | brah | There must not be any. |
| 13:52.44 | asaleem | can an internal iax extension call an internal sip extension? |
| 13:53.06 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:53.06 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:53.20 | [TK]D-Fender | brah: He didn't say that precisely and you might be so lucky as to learn to not trust what you don't see around here. |
| 13:53.35 | brah | :) |
| 13:53.38 | *** join/#asterisk SkramX (n=SkramX@rrcs-71-40-233-189.sw.biz.rr.com) |
| 13:53.46 | brah | Too much trust in the world perhaps. |
| 13:53.51 | SuPrSluG | asaleem:yes |
| 13:54.19 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 13:54.21 | [TK]D-Fender | asaleem: There is no such thing as "internal", and* processes calls. Calls in from whatever source your configured, and out to any other resource you've configured |
| 13:54.42 | [TK]D-Fender | asaleem: I use * as a JUKEBOX and to MAKE COFFEE. |
| 13:55.04 | asaleem | SuPrSluG, do I need to do something special to make that happen? |
| 13:55.39 | asaleem | SuPrSluG, or just create them in and create the dialpan |
| 13:55.48 | SuPrSluG | asaleem:yes, you need an iax and a sip extension working. then just dial |
| 13:56.10 | asaleem | [TK]D-Fender, wow, how do you do that? Any instructions/tutorial? |
| 13:56.12 | ruben23 | <PROTECTED> |
| 13:56.46 | asaleem | SuPrSluG, ok, great, thanks, I was unable to make call,I will give it another try |
| 13:56.53 | SuPrSluG | asaleem:that's it create an extension in iax2 and sip conf files and |
| 13:57.05 | SuPrSluG | add to dialplan |
| 13:57.36 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:58.19 | *** join/#asterisk Von_Lorenz (n=lorenzo@ip-89-162.sn1.eutelia.it) |
| 13:58.26 | Von_Lorenz | HI to all |
| 13:58.52 | [TK]D-Fender | ruben23: "chkconfig asterisk on |
| 13:59.10 | [TK]D-Fender | asaleem: You can DIAL whatever you want. Its YOUR dialplan |
| 13:59.52 | ruben23 | [TK]D-Fender:--->service asterisk does not support chkconfig |
| 14:00.13 | asaleem | [TK]D-Fender, can I DIAL my gf to turn her ON? |
| 14:00.28 | asaleem | [TK]D-Fender, :) |
| 14:00.29 | *** join/#asterisk moa_ (n=moa_@lab.vision.net) |
| 14:00.45 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:01.13 | jamesh1 | [TK]D-Fender: experience working with meetme conferences? |
| 14:01.16 | chantanito | thanks [TK]D-Fender |
| 14:01.25 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:02.31 | [TK]D-Fender | ruben23: previous advice for preparing the startup script was WRONG |
| 14:02.47 | [TK]D-Fender | ruben23: Go hunt downt he crap you copied over and delete them |
| 14:03.00 | [TK]D-Fender | ruben23: then go to your source folder and "make config" |
| 14:03.07 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 14:03.09 | [TK]D-Fender | ruben23: then redo the chkconfig |
| 14:03.29 | [TK]D-Fender | asaleem: Yes, thats what the "vibrate" option on her cell-phone is for |
| 14:03.42 | [TK]D-Fender | jamesh1: yes |
| 14:03.43 | ariel_ | Morning folks |
| 14:03.52 | *** join/#asterisk jkroon (n=jkroon@dsl-240-167-45.telkomadsl.co.za) |
| 14:05.32 | asaleem | [TK]D-Fender, we have to figure out what to do if VIBRATE option is off |
| 14:06.56 | ruben23 | <PROTECTED> |
| 14:09.27 | jamesh1 | I'm having difficulty nailing down a meetme conference recording problem. |
| 14:11.36 | [TK]D-Fender | jamesh1: How have have you tried looking at this problem? How much have you shown us since the start? |
| 14:11.58 | *** join/#asterisk zdennis (n=zdennis@75.119.89.19) |
| 14:12.48 | jamesh1 | I've tried making multiple conf#'s connecting in different ways, whether it be from the meetme misc-dest or from outside using DID or internal using SIP |
| 14:12.59 | ruben23 | TK]D-Fender: one more thing i ahve setup asterisk server running on 250 SATA hdd but its 90 percent utilize now- can i add up another drive on it an re route my var/spool/asterisk/monitorDONE storage on the new Disk..?------->how to do that..? |
| 14:13.23 | [TK]D-Fender | ruben23: Go learn Linux |
| 14:13.30 | jamesh1 | lol |
| 14:14.03 | ruben23 | :-( |
| 14:14.05 | [TK]D-Fender | ruben23: You don't understand mount points, the linux filesystem, startup scripts for your distro and so many other basic things. |
| 14:15.09 | ruben23 | ok, but lastly no need to setup asterisk on that new drive just mount the drive...just a hint |
| 14:15.22 | ruben23 | :-D |
| 14:15.34 | jamesh1 | you will just b using it as a hard drive |
| 14:15.37 | jamesh1 | as storage space. |
| 14:15.42 | jamesh1 | obviously you won't have to :P |
| 14:16.04 | *** part/#asterisk zagibu (n=adsc@cust.static.213-180-179-65.cybernet.ch) |
| 14:17.05 | ruben23 | ok thanks-----> ill go learn linux :-D |
| 14:17.34 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 14:20.07 | jamesh1 | okay if I don't specify a user pin the recording file gets made at first person logging into conference |
| 14:20.08 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:20.16 | jamesh1 | <PROTECTED> |
| 14:20.17 | jamesh1 | <PROTECTED> |
| 14:22.01 | *** join/#asterisk juanIMP (n=juan@200.71.41.254) |
| 14:22.11 | jamesh1 | whether internal or external the wav is not being created when a user pin is designated. |
| 14:24.15 | *** join/#asterisk [T]ank (n=ckwall@21.sub-75-216-89.myvzw.com) |
| 14:24.26 | [T]ank | how do i turn off privicy db? |
| 14:25.15 | *** join/#asterisk Corydon76-dig (i=indigo@pdpc/supporter/bronze/Corydon76-home) |
| 14:25.15 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 14:26.28 | *** join/#asterisk aethelrick (n=richard@217.113.164.66) |
| 14:28.05 | *** part/#asterisk [T]ank (n=ckwall@21.sub-75-216-89.myvzw.com) |
| 14:28.07 | jamesh1 | with a moderator pin and user pin the conference wav is not being recorded either. any ideas? |
| 14:28.17 | jamesh1 | they have in the past. |
| 14:29.16 | aethelrick | hi all, I'm using astmanproxy and AMI to automate our phone system from a web application, my problem is that I currently use AMI commands to get a list of channels (core show channels) in order to understand what calls are currently connected, this list has truncated channel names in it which can't be used in subsequent AMi commands like Redirect and Originate etc... what is the correct way to get a list of live channels from Asterisk, specificall |
| 14:29.16 | aethelrick | y from AMI?? any ideas? |
| 14:29.42 | [TK]D-Fender | aethelrick: "core show channels concise" |
| 14:29.47 | aethelrick | ooooo |
| 14:29.53 | aethelrick | that may be a HUGE help |
| 14:29.56 | aethelrick | brb |
| 14:30.35 | aethelrick | awesome... thanks so much, you saved me a lot of time :) |
| 14:30.56 | [TK]D-Fender | aethelrick: You're welcome |
| 14:33.40 | *** join/#asterisk scarecrowed (n=fabiano@200.172.91.227) |
| 14:36.54 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 14:37.14 | *** join/#asterisk ehsjoar (n=ehsjoar@24.9.91.203) |
| 14:37.50 | aethelrick | [TK]D-Fender: one other quick question... do you have a link to a document showing the names of the fields returned by "core show channels concise" |
| 14:38.34 | aethelrick | I think I could guess most... but I want to be certain and google-ing failed me :) |
| 14:38.56 | [TK]D-Fender | aethelrick: Don't recall offhand. Might be in DOC, might be on the WIKI or CLI help. The fields are mostly obvious |
| 14:39.07 | Corydon76-dig | aethelrick: it's the same as 'core show channels' |
| 14:39.20 | Corydon76-dig | aethelrick: only the fields aren't spaced out |
| 14:39.53 | aethelrick | I agree they are mostly obvious... it's just the few extra ones that were parts of other fields in "core show channels" they are now fields in their own right |
| 14:40.17 | *** join/#asterisk jthurman42 (n=jthurman@66.154.131.25) |
| 14:41.38 | Jumpie | when considering ATAs for connecting to fax machines, is the t.38 protocol only an issue at the providers and the asterisk software itself? or are ATAs supposed to be t.38 capable as well? |
| 14:41.45 | Jumpie | was lookin at the grandstream ht286 |
| 14:41.52 | Jumpie | which i've heard is pretty good, especially considering the price |
| 14:42.16 | Corydon76-dig | The ATA has to be T.38-capable, as well |
| 14:42.25 | Jumpie | hmm |
| 14:42.29 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 14:42.44 | Jumpie | well if t.38 is the baseline standard for this, why arent most of these ATAs im lookin at mentioning t.38 |
| 14:42.49 | *** join/#asterisk pagec (n=pagec@96.57.210.34) |
| 14:42.54 | Jumpie | Supports SIP 2.0, TCP/UDP/IP, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP protocols |
| 14:43.12 | Corydon76-dig | Because they're not designed to be used with fax machines? |
| 14:43.16 | Jumpie | good call :D |
| 14:43.23 | Jumpie | you have a recommended ata for use with faxing? |
| 14:43.35 | Jumpie | ah |
| 14:43.38 | Jumpie | the 486 |
| 14:43.41 | Katty | falls asleep on desk |
| 14:43.43 | *** join/#asterisk LeddyHM (n=NONE@75.63.105.141) |
| 14:43.49 | Corydon76-dig | The best ATA I've used for faxing is no longer produced |
| 14:43.54 | Jumpie | doh..really? |
| 14:44.21 | Corydon76-dig | but then again, I've been out of the reseller business for 2+ years |
| 14:44.25 | Jumpie | this is odd lookin at the ht486..it actually mentions fax over ip this time..the other did not, but still doesnt mention t.38 |
| 14:44.28 | rene- | steals Katty's lamp |
| 14:44.54 | pagec | hey anyone know what is going on http://pastebin.com/m4725448 ? |
| 14:48.55 | jamesh1 | I've been referring to web-meetme conference btw. |
| 14:50.10 | zdennis | each time you make a change to dialplan or to a configuration file do you have to restart asterisk or does it reload changes by itself? |
| 14:50.38 | jamesh1 | rasterisk / reload / ctrl+c |
| 14:50.54 | coppice | Corydon76-dig: which ATA did you find best? |
| 14:51.47 | zdennis | jamesh1: thanks |
| 14:52.21 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 14:52.30 | Corydon76-dig | coppice: let me go look for the name... slipped my mind |
| 14:52.30 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733615.dsl.bell.ca) |
| 14:53.34 | Katty | does anyone know if it's safe to get a pet a bath the day after giving them frontline? |
| 14:53.44 | [TK]D-Fender | Corydon76-dig: One of the mediatrix 2000 series? |
| 14:53.55 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:53.56 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:54.24 | Corydon76-dig | coppice: we suspected that the reason for its great performance were that the packets were all 10ms in length |
| 14:54.29 | coppice | the mediatrix T.38 has some awful bugs, but if the other end is tolerant of them it can work quite well |
| 14:54.49 | Corydon76-dig | [TK]D-Fender: it was not the mediatrix |
| 14:55.01 | Von_Lorenz | I have a problem when i try to compile the source of dahdi i recive this error for Firmware not found: ../XppConfig.pm USB_FW.hex FPGA_FXS.hex FPGA_1141.hex FPGA_1151.hex FPGA_1161.hex PIC_TYPE_1.hex PIC_TYPE_2.hex PIC_TYPE_3.hex PIC_TYPE_4.hex /usr/share/dahdi/ |
| 14:55.01 | ariel_ | Katty: 24 hours |
| 14:55.31 | coppice | 10ms packets shouldn't really improve FAXing |
| 14:56.12 | Katty | ariel_: k |
| 14:56.17 | Katty | ariel_: you have dogs? |
| 14:56.20 | brah | It's not like pets need baths anyway |
| 14:56.23 | brah | they can lick themselves |
| 14:56.26 | coppice | some ATAs do pretty weird things to the audio, which make them totally FAX proof :-) |
| 14:56.32 | Katty | brah: clearly you don't own pets. |
| 14:56.47 | tzafrir_laptop | Von_Lorenz, hmm.. where? |
| 14:56.48 | brah | I would pay good money to be able to lick my privates clean |
| 14:57.00 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 14:57.07 | markbest | Is anyone using the PAGE() command here? |
| 14:57.14 | Von_Lorenz | sorry in debian system |
| 14:57.20 | Katty | markbest: yes. |
| 14:57.36 | coppice | brah: I like to lick my wife, but only *after* she has bathed |
| 14:57.37 | markbest | Katty: When you use PAGE how many 'rings' do you get? |
| 14:57.44 | ariel_ | Katty: yes dog, small what I call a shits zoo.... just a white hair ball of a dog |
| 14:57.46 | tzafrir_laptop | Von_Lorenz, where do you get that message? on 'make'? |
| 14:57.54 | Katty | markbest: i set up mine to do be auto answer. |
| 14:57.56 | Von_Lorenz | ye of course |
| 14:57.59 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:58.07 | Katty | ariel_: what brand of flea shampoo do you use? |
| 14:58.08 | Von_Lorenz | make install |
| 14:58.43 | markbest | Katty: I don't think my phones activate with autoanswer they seem to only auto-pick-up after two rings. |
| 14:58.45 | tzafrir_laptop | Von_Lorenz, a standard source tarball? |
| 14:58.56 | Katty | markbest: what brand of phones are you using? |
| 14:58.58 | ariel_ | Katty: my wife does the cleaning and all, so I have no Idea. |
| 14:58.59 | Von_Lorenz | yes i take it from ftp of digium |
| 14:59.03 | Katty | ariel_: ah. |
| 14:59.07 | markbest | Katty: Ascom i75 WiFi phones. |
| 14:59.14 | Katty | markbest: ah. we use polycoms. |
| 14:59.23 | Von_Lorenz | wget http://downloads.asterisk.org/pub/telephony/dahdi-linux/releases/dahdi-linux-2.2.0.1.tar.gz |
| 14:59.35 | tzafrir_laptop | Von_Lorenz, what version? |
| 14:59.44 | markbest | Katty: Can you still provide a sample of you extentions.conf? Just so I can double check my work? |
| 14:59.51 | Von_Lorenz | 2.2.0.1 |
| 14:59.56 | tzafrir_laptop | can you pastebin all the output you got? |
| 14:59.57 | markbest | Katty: Or look over mine... |
| 15:00.20 | brah | >i take from ftp |
| 15:00.23 | brah | >wget http:// |
| 15:00.24 | brah | wat |
| 15:00.41 | Von_Lorenz | soory brah |
| 15:00.47 | brah | :P |
| 15:00.50 | Von_Lorenz | its to late for me |
| 15:00.55 | ariel_ | I have a meeting bbl folks |
| 15:00.57 | Katty | markbest: i surpose. sec |
| 15:00.59 | Von_Lorenz | and my mind |
| 15:01.06 | Von_Lorenz | this is the output |
| 15:01.10 | Von_Lorenz | install -m 644 ../XppConfig.pm USB_FW.hex FPGA_FXS.hex FPGA_1141.hex FPGA_1151.hex FPGA_1161.hex PIC_TYPE_1.hex PIC_TYPE_2.hex PIC_TYPE_3.hex PIC_TYPE_4.hex /usr/share/dahdi/ |
| 15:01.10 | Von_Lorenz | install: impossibile fare stat di `USB_FW.hex': No such file or directory |
| 15:01.10 | Von_Lorenz | install: impossibile fare stat di `FPGA_FXS.hex': No such file or directory |
| 15:01.10 | Von_Lorenz | install: impossibile fare stat di `FPGA_1141.hex': No such file or directory |
| 15:01.10 | Von_Lorenz | install: impossibile fare stat di `FPGA_1151.hex': No such file or directory |
| 15:01.11 | Von_Lorenz | install: impossibile fare stat di `FPGA_1161.hex': No such file or directory |
| 15:01.13 | Von_Lorenz | install: impossibile fare stat di `PIC_TYPE_1.hex': No such file or directory |
| 15:01.15 | Von_Lorenz | install: impossibile fare stat di `PIC_TYPE_2.hex': No such file or directory |
| 15:01.17 | Von_Lorenz | install: impossibile fare stat di `PIC_TYPE_3.hex': No such file or directory |
| 15:01.19 | Von_Lorenz | install: impossibile fare stat di `PIC_TYPE_4.hex': No such file or directory |
| 15:01.23 | Von_Lorenz | make[1]: *** [install] Error 1 |
| 15:01.25 | Von_Lorenz | make[1]: Leaving directory `/usr/src/modules/dahdi/drivers/dahdi/xpp/firmwares' |
| 15:01.26 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 15:01.27 | Von_Lorenz | make: *** [install-xpp-firm] Error 2 |
| 15:01.28 | brah | Pastebin, bro |
| 15:01.31 | *** kick/#asterisk [Von_Lorenz!n=joe@216.191.106.163] by [TK]D-Fender ([TK]D-Fender) |
| 15:01.56 | Katty | grins |
| 15:02.05 | [TK]D-Fender | :F |
| 15:02.15 | Katty | markbest: http://pastebin.ca/1486945 |
| 15:02.30 | jamesh1 | [TK]D-Fender: no input on web-meetme issue? lol |
| 15:02.45 | Katty | i have input. |
| 15:02.52 | Katty | call the people who wrote it. |
| 15:02.54 | [TK]D-Fender | jamesh1: I don't see cohesive pastebins from you showing the failure, a dump of the recording folder, etc |
| 15:03.18 | [TK]D-Fender | jamesh1: And if 3rd party junk is broken, it isn't our problem. |
| 15:03.26 | *** join/#asterisk Von_Lorenz (n=lorenzo@ip-89-162.sn1.eutelia.it) |
| 15:03.48 | jamesh1 | I didn't say it was your problem. I was just curious if you had dealt with anything remotely close to this issue. |
| 15:03.52 | Aiatek | ~ paste |
| 15:03.53 | infobot | paste is probably http://rafb.net/paste/, or see also pb, or http://bin.cakephp.org/ |
| 15:03.56 | Von_Lorenz | sorry guys i don't know who you mean with the word pastebin |
| 15:04.00 | Aiatek | ~ pastebin |
| 15:04.01 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:05.53 | jkroon | ok, does anybody have any constructive feedback regarding the hardware echo canceller from digium? |
| 15:06.11 | jkroon | as in, has anybody got a case where it actually works well? |
| 15:06.41 | jamesh1 | I use and have used digium cards w/ EC they work great. |
| 15:06.42 | Von_Lorenz | http://pastebin.com/d76184d3f |
| 15:06.49 | Von_Lorenz | is true this way? |
| 15:07.24 | jkroon | jamesh1, ok, could you maybe pastebin your dahdi/system.conf and chan_dahdi.conf files for me somewhere please? |
| 15:07.47 | jamesh1 | what's the issue you have? |
| 15:07.47 | aethelrick | hi all, is there a way to interrogate the dialplan from AMI to find out which extension(s) are associated with a channel |
| 15:08.01 | jkroon | i keep on getting what can only be described as call breakup, like the echo canceller is being overly aggressive. |
| 15:08.31 | jkroon | and changing the echocancel=yes to have some value doesn't work, i've tried values ranging from 32 to 1024 and nothing fixes it completely. |
| 15:08.46 | jkroon | basically the incoming voice seems to "cut out". |
| 15:09.21 | rene- | aethelrick: you dont need to interrogate the dialplan from AMI |
| 15:09.41 | jamesh1 | rxwink=300 echotraining=800 |
| 15:09.46 | [TK]D-Fender | jkroon: Also depends which EC its got, etc |
| 15:10.00 | rene- | you just want to issue the ami action status and that will give you every single device status, then just parse for your desired extension |
| 15:10.09 | [TK]D-Fender | 11:09]<jamesh1>rxwink=300 echotraining=800 <- neither is used with HWEC |
| 15:10.33 | jamesh1 | I know this. |
| 15:10.36 | jkroon | [TK]D-Fender, from dmesg: Booting VPMADT032 <-- in the one case. |
| 15:10.41 | aethelrick | ok... er... that sounds like it will work, but won't that be heavy on the server? |
| 15:11.26 | jkroon | would the firmware revision (107 currently) make a difference? |
| 15:11.28 | errr | hi, I have 2 locations. we are connected to each other via dedicated t-1. my 2 systems have an IAX trunk between them so we can call each other locally. If I am getting audio issues with 1 sided audio, or choppy audio, and from time to time echo in those calls, would it be asterisk, or my network? |
| 15:11.45 | Von_Lorenz | <tzafrir_laptop> had you see errors |
| 15:11.47 | jkroon | i've now just disabled the HW EC and am using oslec, default params and it's sounding a lot better. |
| 15:12.12 | jamesh1 | the hardware could always be faulty |
| 15:12.16 | jkroon | errr, using trunking or not? |
| 15:12.36 | errr | jkroon: Im not sure what trunking means |
| 15:12.36 | jkroon | jamesh1, i've now had three different sites where i had the exact same issue after switching to oslec it's gone. |
| 15:12.46 | jkroon | errr, do you have trunk=yes anywhere? |
| 15:12.50 | jkroon | in iax.conf |
| 15:14.17 | jkroon | i'm busy trying to track an issue with "choppy" sound when brdinging between iax/2 and sip that seems to be related to trunk=yes and trunktimestamps=yes with any value of trunkfreq (but worse when trunkfreq != codec sample length), which seems to be some kind of bug in asterisk itself. |
| 15:14.19 | errr | jkroon: no |
| 15:14.28 | Katty | eppigy: poke. |
| 15:14.35 | jkroon | errr, if you don't use trunk=yes I suggest switching to SIP. |
| 15:15.30 | jkroon | unless of course there is a specific reason (like auth related and multiple channels between the hosts) to use IAX/2. |
| 15:15.42 | errr | jkroon: ok, but lets pretend we are not trying to change my setup thats been working for ages. Im more just curious if the problem would more likely be my hardware running asterisk or software (asterisk), or if it is a miss configured network |
| 15:16.03 | jkroon | do you have jitter buffers enabled? |
| 15:16.09 | jkroon | if so, which one? |
| 15:16.11 | errr | I do not think so |
| 15:16.27 | jkroon | chances are you've got some slight variance in latency (jitter) that the phones can't handle. |
| 15:16.47 | *** join/#asterisk galeras (n=galeras@186.80.186.118) |
| 15:16.48 | errr | so network? |
| 15:17.19 | jkroon | generally I find that having jitterbuffer=yes, forcejitterbuffer=yes is a *good thing* in iax2.conf |
| 15:17.20 | errr | my boss went jacking with our network a couple weeks ago, then these problems started happening and he is forcing me to prove to him its not asterisk |
| 15:17.22 | jkroon | *iax.conf |
| 15:17.50 | tzafrir_laptop | Von_Lorenz, sounds odd. |
| 15:17.54 | jkroon | also, jbimpl=adaptive :) |
| 15:18.15 | errr | jkroon: do all these settings you are suggesting work in 1.2? |
| 15:18.30 | jkroon | errr, seriously doubt it. |
| 15:18.59 | jkroon | heck, i think trunking only came in at 1.4. the jb stuff has also (from what I've read) changed a LOT from 1.2 to 1.4 |
| 15:19.44 | errr | I dont think we really have the time right now to put into making the change to 1.4 |
| 15:20.19 | rene- | aethelrick, it shouldnt matter, it is just a bit of text every now and then, your other shot is to wait for newchannel events that match your desired extension |
| 15:20.39 | *** join/#asterisk mykhyggz (n=mykhyggz@evolone.org) |
| 15:21.09 | tzafrir_laptop | Von_Lorenz, because it sounds like you're using /usr/src/dahdi.tar.bz2 or something similar |
| 15:22.45 | Von_Lorenz | i have untar the source |
| 15:24.29 | jkroon | errr, i'd suggest going straight for 1.6.1.1 at this point. |
| 15:25.16 | Von_Lorenz | i need to go i come back in about 1 hr |
| 15:25.25 | errr | jkroon: even for business critical setups? |
| 15:25.38 | Von_Lorenz | see you later |
| 15:26.29 | Von_Lorenz | zan thanks for your help |
| 15:27.42 | Von_Lorenz | <tzafrir_laptop> thanks for your help |
| 15:28.22 | brah | Can asterisk 1.6 send and receive T.30? |
| 15:31.38 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 15:31.53 | rene- | brah: digium announced some t38 warez recently |
| 15:32.02 | rene- | oh sorry |
| 15:32.06 | rene- | you meant t30 |
| 15:32.17 | brah | :/ |
| 15:32.46 | rene- | coppice was around, he is the fax expert |
| 15:32.55 | mog | rene-, digium's asterisk fax stuff does t.30 as well i thought |
| 15:33.01 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:33.02 | mog | t.30 is regular analog fax right? |
| 15:33.05 | brah | Yeah |
| 15:33.11 | mog | ya it does that as well |
| 15:33.18 | mog | and you get one free channel to play with |
| 15:33.24 | brah | What do you mean by 'asterisk fax stuff'? |
| 15:33.45 | brah | Ah.. this is what they were talking about the other day |
| 15:33.55 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 15:33.57 | mog | http://www.digium.com/en/products/software/faxforasterisk.php |
| 15:33.58 | brah | Of how digium dropped software support for T.30 since they added it to their hardware |
| 15:34.08 | galeras | with "dahdi show status" span is OK, "pri show spans" show nothing. Any suggestion? (signalling=pri_cpe) |
| 15:34.15 | *** join/#asterisk tobias (n=tobias@216.27.28.176) |
| 15:34.18 | brah | right? |
| 15:34.26 | mog | galeras, do you have pri commands in the shell |
| 15:34.35 | mog | i hadnt heard they had dropped t.30 support |
| 15:34.47 | galeras | mog, at cli? yes |
| 15:35.47 | brah | Of how it isn't included in asterisk |
| 15:35.55 | brah | But T38 is |
| 15:36.11 | *** join/#asterisk ellert (n=ellert@80.101.124.67) |
| 15:36.42 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 15:37.11 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:37.20 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 15:38.27 | ellert | hello, i have a question that is probably easy to answer to the linux diehards around. I am trying to upgrade asterisk from 1.2 to 1.6, downloaded, configured and installed all the packages, but i get the following error all the time: |
| 15:38.28 | ellert | Running dahdi_cfg: DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (22) |
| 15:39.30 | brah | Wrong echo cancellation parameter for channel 1? |
| 15:39.32 | ellert | even though i let it install all the sample configs with the idea i could fix them again later to the situation i had before, something seems to be wrong here |
| 15:39.43 | galeras | any idea? with * 1.4 and zaptel my PRI was fine, but * 1.6 (current) and dahdi, "pri show spans" show nothing. |
| 15:39.44 | ellert | all default in the generated configs |
| 15:41.33 | ellert | span=1,1,0,ccs,hdb3,crc4 |
| 15:41.33 | ellert | # termtype: te |
| 15:41.33 | ellert | bchan=1-15,17-31 |
| 15:41.33 | ellert | dchan=16 |
| 15:41.34 | ellert | echocanceller=mg2,1-15,17-31 |
| 15:41.49 | viraptor | angryuser: I tried turning on nat, setting local nets, qualify, etc. - still no rtpkeepalive stuff visible in pcaps :/ |
| 15:42.24 | *** join/#asterisk sumasuma (n=root@64.92.218.196) |
| 15:42.27 | *** join/#asterisk icyValk77 (n=icyValk7@host81-153-93-26.range81-153.btcentralplus.com) |
| 15:42.40 | angryuser | viraptor, i cant help you anymore, i dont know rtpkeepalive stuff much, ask around |
| 15:43.06 | viraptor | yeah... mailing list then |
| 15:43.22 | *** join/#asterisk hesco (n=hesco@24.99.160.121) |
| 15:44.09 | ellert | so it can not find the echocanceler. can it be that it has a problem finding the libraries? |
| 15:44.34 | hesco | My office power just went out. I'm restoring my network, but don't yet seem to have phone service. * is running, I've set verbosity to 3, but I get no dial tone and my calls all end with a busy signal. |
| 15:44.57 | hesco | no information in the logs, though |
| 15:45.22 | hesco | sip show peers shows my desk handset |
| 15:45.31 | bmoraca | what would cause an ATA to not put the port on hook after the call is ended from the other side? |
| 15:45.33 | hesco | iax2 show peers shows my trunk |
| 15:46.03 | hesco | but even attempting to dial an internal extension does not seem to work. |
| 15:46.10 | hesco | what might I be missing here |
| 15:47.31 | jkroon | ok, what could all be wrong if (Diva eIcon Server board PRI card) I can make calls out, but I'm not seeing any activity when calling in to the box? |
| 15:47.31 | [TK]D-Fender | hesco: You're missing looking at SIP DEBUG to see if the call is even reaching * |
| 15:48.01 | Jumpie | anybody have any good pda type device implementation with asterisk? |
| 15:48.13 | jkroon | and is there any way to debug the issue? Say, some kind of console that allows me to look at what's going on at the ISDN level or something? |
| 15:48.23 | Jumpie | some friends and i are trying to nail down some devices, possibly one of the nokia linux based ones, using a sip client, provided i can get an ip and communicate with asterisk |
| 15:48.26 | Jumpie | anybody doin that? |
| 15:49.09 | *** join/#asterisk tzanger (n=tzanger@gromit.mixdown.ca) |
| 15:49.15 | tzanger | wow it's been a while since I've been here :-) |
| 15:49.28 | RoyCrowder | Jumpie: take a look at mobile sip phones, they are out there. |
| 15:49.47 | hesco | ok, with sip set debug on and sip set debug peer hugh_desk, nothing in the logs |
| 15:49.47 | RoyCrowder | Jumpie: you usually have to have some sort of base station though |
| 15:49.54 | tzanger | is the digium res_fax able to take my fax machine (connected via TDM400) and bridge it to a t.38 provider? |
| 15:49.57 | hesco | same busy signal |
| 15:49.57 | *** join/#asterisk LemensTS (n=customgt@70.238.180.76) |
| 15:50.05 | LemensTS | can you set a ring timeout on zap? i know u can on sip |
| 15:50.20 | tzanger | or would I have to tell the fax machine to fax it to an extension, and then have the system fax the fax out from there (store and forward) ? |
| 15:50.32 | Jumpie | RoyCrowder yea...i know they exist..just curious if ther were some particular favorites :D |
| 15:50.34 | [TK]D-Fender | LemensTS: Dial works the same regardless of tech |
| 15:50.57 | hesco | so does that indicate that the call is NOT reaching the asterisk server? |
| 15:51.15 | RoyCrowder | Jumpie: I worked on a job with my company where we set some up but for the life of me I don't remember who made them or what they were, it has been awhile :| |
| 15:51.15 | [TK]D-Fender | hesco: Busy signal != Looking at * CLI with SIP DEBUG |
| 15:52.12 | hesco | I did: `sip set debug on` and `sip set debug peer hugh_desk` and still see nothing in the *CLI when I test. |
| 15:52.56 | Katty | oh |
| 15:53.00 | hesco | CLI says: SIP Debugging Enabled for IP: 192.168.15.4:60003 |
| 15:53.01 | [TK]D-Fender | hesco: Then you either have a networking problem, or your device isn't configured right |
| 15:53.01 | Katty | i think i just might be waking up a bit. |
| 15:53.18 | [TK]D-Fender | hestAnd do NOT specify an IP or peer. use global debug only |
| 15:53.33 | *** join/#asterisk vegbox (n=kevinle@24-205-54-7.dhcp.gldl.ca.charter.com) |
| 15:55.39 | vegbox | the latest |
| 15:55.40 | vegbox | lol |
| 15:55.48 | vegbox | opps i left the chat haha |
| 15:56.11 | hesco | I have a ssh connection from this desktop machine to the * server that is routed through the same router as my handset |
| 15:56.36 | [TK]D-Fender | hesco: Already this is sounding like a NAT issue |
| 15:59.32 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 15:59.41 | ellert | hm interesting. regarding the DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (22) error: a reboot seems to have fixed the problem |
| 15:59.59 | ellert | a reboot on a unix machine..... yech :( |
| 16:03.12 | hesco | [TK]D-Fender: I am able to browse to the web interface of the HT-486 |
| 16:03.48 | [TK]D-Fender | hesco: MEANINGLESS |
| 16:04.01 | hesco | from inside the subnet on my desk, 192.168.0.0, and from my desk to ssh to the * server at 192.168.15.51 |
| 16:04.09 | hesco | can you say more about the NAT issue? |
| 16:04.38 | [TK]D-Fender | hesco: If you see nothing at * CLI with SIP debug enabled, then packets aren't reaching it. |
| 16:07.18 | *** join/#asterisk apeiron_ (i=apeiron@isuckatdomains.net) |
| 16:08.21 | LemensTS | TKD-Fender: is the dial timeout the same for dialing out an e1 pri? |
| 16:08.29 | LemensTS | group |
| 16:08.44 | hesco | I just dropped the firewall on the * server and see the same non-results |
| 16:09.14 | LemensTS | doing Dial(ZAP/g1/${EXTEN},20) but it wont dial out. Get a Requested transfer capability: 0x00 - SPEECH.... -- Channel 0/1, span 1 got hangup, cause 28 |
| 16:09.21 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 16:09.23 | LemensTS | If take the ,20 off, it dials |
| 16:10.18 | *** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 16:11.07 | hesco | !paste |
| 16:11.11 | hesco | ??paste |
| 16:12.57 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 16:13.11 | JackTheNipple | ~paste |
| 16:13.12 | infobot | well, paste is http://rafb.net/paste/, or see also pb, or http://bin.cakephp.org/ |
| 16:14.46 | hesco | OK, my ssh reset itself, when I went back in, suddenly I got this: http://pgsql.privatepaste.com/adlMKjBbOR and an attempt to call a local extension reached voicemail. But I'm still hearing no dialtone. |
| 16:15.44 | bmoraca | uhg |
| 16:15.52 | bmoraca | stupid linksys ATAs |
| 16:15.59 | hesco | well I now seem to make an outgoing call, even if I don't have a dialtone. |
| 16:16.32 | bmoraca | i can get the thing to reorder indefinitely or hookflash when the remote party hangs up...but i can't just get it to hang the fxs line up. |
| 16:18.07 | *** part/#asterisk ellert (n=ellert@80.101.124.67) |
| 16:18.16 | jameswf | Michael Jackson Funeral bigger than VUC! |
| 16:19.37 | *** join/#asterisk mbrevda (n=mbrevda@94.159.178.239) |
| 16:19.59 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:20.40 | bmoraca | hesco: you're almost assuredly having a NAT issue. |
| 16:21.24 | *** join/#asterisk shazaum (n=98u89weu@unaffiliated/shazaum) |
| 16:28.27 | SuPrSluG | LemensTS:use put other dial options after the 1st comma. like {BLAH}.ttrm,20 |
| 16:29.31 | SuPrSluG | LemensTS:if you don't want any -> {BLAH},,20 |
| 16:31.00 | [TK]D-Fender | SuPrSluG: Backwards... |
| 16:31.05 | [TK]D-Fender | LemensTS: PASTEBIN |
| 16:31.15 | SuPrSluG | is it? |
| 16:31.32 | [TK]D-Fender | hesco: And why are you showing the OK to a BYE? Show the entire damn call |
| 16:31.54 | SuPrSluG | doh |
| 16:32.00 | *** join/#asterisk ccesario_ (n=ccesario@189.88.2.216) |
| 16:32.13 | LemensTS | http://pastebin.com/m7b072ffb |
| 16:32.15 | *** join/#asterisk flashnet (n=fdfdsf@201-213-121-79.net.prima.net.ar) |
| 16:33.52 | *** join/#asterisk zingbats (n=zingbats@cpc2-sals2-0-0-cust799.sotn.cable.ntl.com) |
| 16:35.06 | [TK]D-Fender | LemensTS: ....... Show the actual &#!@ call......with CHANNEL DEBUG |
| 16:35.18 | zingbats | I have a really newbie question: I'm looking into asterisk just to fool around with, mainly. I am, however, completely naive and ignorant to the whole world of telecomms. I have an old box sitting around which I want to act as the server. Do I need any special hardware before I even start looking at the software? I doubt the 56K Modem is all I need. |
| 16:35.57 | [TK]D-Fender | zingbats: That modem is almost certainly worthless |
| 16:36.45 | [TK]D-Fender | zingbats: And no, you can learn * using jsut a softphone & * |
| 16:36.47 | [TK]D-Fender | ~softphone |
| 16:36.48 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 16:38.30 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:39.07 | zingbats | Right. What about in terms of using it as a replacement voicemail / answer phone in the landline side of things, rather than VoIP |
| 16:39.33 | *** join/#asterisk ChkDigit (n=mike@67-225-17-254.sktn.hsdb.sasknet.sk.ca) |
| 16:41.49 | [TK]D-Fender | zingbats: If you jsu want an answering machine, jsut go buy one. |
| 16:42.46 | zingbats | [TK]D-Fender, it's not that I need one, it's the process of learning. This kinda stuff interests me, and has nothing to do with any functionality especially. If achieving that goal is too difficult / expensive, then I will obviously find something else to occupy my time :-) |
| 16:43.14 | [TK]D-Fender | zingbats: Digium and Samgoma btoh make decent FXO interfaces, but ont he cheap I might recommend a Linksys SPA-3102 (~$70 USD) |
| 16:43.45 | [TK]D-Fender | zingbats: The SPA is a good idea for learning. It'll let you use your line with *, as well as 1 phone. |
| 16:45.30 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 16:45.41 | *** part/#asterisk tzanger (n=tzanger@gromit.mixdown.ca) |
| 16:46.55 | bmoraca | Sangoma makes a USB FXO device for about $130 |
| 16:47.48 | [TK]D-Fender | bmoraca: Looks interesting. I'd like to see som reviews of it first though. |
| 16:48.33 | zingbats | So the SPA translates the signal to packets, which I assume are then dealt with by * |
| 16:48.38 | bmoraca | yeah...i haven't tested it myself or heard anything about it...but it could be an interesting way to bring Asterisk PBX systems to mini- or picoITX systems |
| 16:48.52 | Qwell | zingbats: not exactly, but close enough |
| 16:49.23 | *** part/#asterisk viraptor (n=viraptor@80.175.164.6) |
| 16:49.57 | zingbats | I had an ADSL modem once that boasted VoIP technology... How would that benefit everyday consumers? Can Skype and similar user-friendly systems interface with the SPA devices? |
| 16:50.08 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 16:50.29 | Qwell | zingbats: Skype no, anything SIP - maybe |
| 16:50.31 | [TK]D-Fender | zingbats: No |
| 16:50.47 | [TK]D-Fender | zingbats: And DSL has nothing to do with VoIP specifically |
| 16:50.49 | zingbats | So how would users benefit from the VoIP feature? Essentially; they wouldn't |
| 16:50.50 | [TK]D-Fender | ADSL* |
| 16:50.51 | *** join/#asterisk propellerhead (n=yogurt2u@host172.190-30-200.telecom.net.ar) |
| 16:51.13 | zingbats | [TK]D-Fender, that was to illustrate it was included with a home environment. |
| 16:51.27 | [TK]D-Fender | zingbats: Well the pieces don't add up. |
| 16:51.45 | [TK]D-Fender | zingbats: Don't list a brick as an ingredient for making a milk-shake |
| 16:51.49 | Qwell | zingbats: the "VoIP" on modems is usually inaccessible to users. you just plug a phone in, and it does all the magic, out of your control |
| 16:52.02 | bmoraca | zingbats: typically, a router with "voip" features (specifically, the ability to trunk via SIP (or anything else) to provide service via an ATA) benefits the provider in being able to offer more options for telephony. |
| 16:52.12 | [TK]D-Fender | zingbats: Prehaps that modem had an ATA built-in |
| 16:52.16 | [TK]D-Fender | ~ATA |
| 16:52.16 | infobot | [ata] Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 16:52.27 | [TK]D-Fender | zingbats: the SPA is an ATA BTW. One with 1 FXS, 1 FXO. |
| 16:52.28 | zingbats | Oh I see, so you plug a special phone into it, and voila. |
| 16:52.46 | [TK]D-Fender | zingbats: ADSL modem/router's with ATA's built it are typically just FXS (for plugging in phones, not lines) |
| 16:52.53 | [TK]D-Fender | zingbats: No, not a special phone. |
| 16:52.58 | [TK]D-Fender | zingbats: regular analog |
| 16:53.42 | zingbats | So internet only calls can be received on a "normal" phone |
| 16:54.18 | [TK]D-Fender | zingbats: Pretty much |
| 16:54.45 | LemensTS | tkd-fender: ok got a full pri debug http://pastebin.com/d7b78aa26 |
| 16:55.36 | LemensTS | I imagine its line 28 of the pastebin |
| 16:55.39 | zingbats | I essentially would like a way to handle incoming calls through the PC. Whether or not I need * for this, I don't know. |
| 16:57.15 | [TK]D-Fender | LemensTS: You don't seem to get it.... I want to see the DIAL. The entire call. ALL OF IT. |
| 16:57.32 | [TK]D-Fender | LemensTS: Am I coming throught loud and clear? Stop cutting shit out of your pastebins! |
| 16:58.25 | [TK]D-Fender | zingbats: Well you need SOME piece of software. * may as well be it |
| 16:58.44 | brah | 'app_valetparking' |
| 16:58.47 | brah | any idea of what this is? |
| 16:58.59 | [TK]D-Fender | brah: 3rd party parking lot app |
| 16:59.17 | [TK]D-Fender | brah: Not ported to 1.6 AFAIK, but was awesome prior to that |
| 16:59.27 | brah | I had to comment out a few lines to get it to compile. I hope as long as I don't call it, everything will work |
| 17:00.54 | zingbats | [TK]D-Fender, So in order to get my PC to handle incoming calls, I'd need a SPA device, an * server or similar, and then some sort of PC interface. Whatever happened to plugging the line into a PC :) |
| 17:01.25 | *** part/#asterisk jamesh1 (n=jhenders@xob.neospire.net) |
| 17:01.26 | [TK]D-Fender | zingbats: No need for a PC interface. the SPA-3102 does it all and plugs onto your network switch and speaks SIP over it to * |
| 17:03.08 | *** join/#asterisk sjobeck (n=Adium@64.122.41.37) |
| 17:03.20 | *** join/#asterisk Selveste1 (n=Selveste@4103ds1-nks.0.fullrate.dk) |
| 17:03.20 | zingbats | [TK]D-Fender, I run windows on my PC, so was planning sticking * on my server... |
| 17:03.52 | [TK]D-Fender | zingbats: Sure. |
| 17:04.09 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 17:04.25 | zingbats | So by interface, I meant getting it from * to my PC. |
| 17:06.05 | bmoraca | zingbats: that's where the softphone comes in |
| 17:06.31 | [TK]D-Fender | zingbats: the SPA lets you sue a regular phone AND your home line, both independent of each other. If yuo want to run a soft-phone on your Windows PC as well, then by all means |
| 17:06.37 | *** join/#asterisk jamesh1 (n=jhenders@xob.neospire.net) |
| 17:08.14 | zingbats | Ok. I'm in the UK, and the postage on the SPA3102 is stupidly high... Any other recommendations? |
| 17:08.45 | bmoraca | order it from a UK provider? |
| 17:09.38 | *** join/#asterisk QaDeS (n=mklaus@dslb-084-056-237-091.pools.arcor-ip.net) |
| 17:09.48 | [TK]D-Fender | bmoraca: Now why go do something smart like that?! |
| 17:10.44 | zingbats | They're more expensive from UK suppliers. I guess I shouldn't complain, because they're still cheaper than the P&P + device |
| 17:11.58 | *** join/#asterisk blkry (n=blkry@64.147.222.130) |
| 17:13.03 | zingbats | 'tis the principle. |
| 17:13.40 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 17:15.49 | [TK]D-Fender | zingbats: the problem with principle is that interest comes bundled.... |
| 17:17.14 | zingbats | =] |
| 17:17.25 | zingbats | I assume the SPA3102 connects via ethernet to the server? |
| 17:17.42 | bmoraca | nope...bluetooth ;) |
| 17:17.55 | zingbats | Of course. |
| 17:18.00 | zingbats | The better of the two. |
| 17:18.15 | zingbats | I did think it would be better to look for a model with firewire... |
| 17:18.16 | bmoraca | i'm just yanking your chain |
| 17:18.27 | zingbats | Ditto |
| 17:20.28 | *** join/#asterisk juny4ng (n=junyangn@cm179.zeta200.maxonline.com.sg) |
| 17:21.56 | *** join/#asterisk scarecrowed (n=fabiano@200.172.91.227) |
| 17:25.02 | zingbats | So the SPA3102 gets an incoming call, and the * answers it... The analogue phone does nothing. Softphone then rings, and it's answered. That's a reasonable setup? But, what if my PC isn't on, and I want the analogue phone to ring... |
| 17:27.19 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:28.31 | [TK]D-Fender | zingbats: Sure |
| 17:30.34 | zingbats | Genius. Well I will put forward the case to my housemates, one of whom now wants a "If you wish to speak to James, press 1" system. I think I'll mess around with the softphones and * first :) Thanks for your help, guys. |
| 17:37.06 | Katty | dances. |
| 17:39.09 | Joel | zingbats get voip service, everyone gets their own number. |
| 17:41.05 | LemensTS | tdk-fender: sorry this is not an accessible external system so im having someone pastebin it for me...i got on it now http://pastebin.com/m59f7d8ac That is everything it says start to finish for the call, pri debug on, verbose at level 20 i think. |
| 17:42.39 | [TK]D-Fender | LemensTS: is this an ORIGINATE-d call? |
| 17:43.00 | [TK]D-Fender | LemensTS: Beacuse there are some indications that it is, and I still don't see the damn dial, or any "origin of call" stuff |
| 17:43.35 | [TK]D-Fender | -- Executing [failed@immediate-message:1] GotoIf("OutgoingSpoolFailed", "0?congestion|1") in new stack |
| 17:44.45 | LemensTS | its a dial strait from extensions.conf |
| 17:45.37 | [TK]D-Fender | LemensTS: then get the entire damn call |
| 17:46.11 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 17:49.58 | *** join/#asterisk hamush (n=hamush@24-155-177-34.dyn.grandenetworks.net) |
| 17:50.24 | *** join/#asterisk shazaum (n=98u89weu@unaffiliated/shazaum) |
| 17:53.17 | hamush | hi guys, i'm new with asterisk and had a working test setup, but i've managed to somehow break it. i keep getting the error: "chan_local.c:494 local_alloc: No such extension/context @phones creating local channel", even though the phones context is defined in my dialplan. can someone help me take a look at my problem? http://pastebin.com/m3a117adf |
| 17:54.23 | [TK]D-Fender | hamush: -- Now forwarding SIP/drennan-58af to 'Local/@phones' (thanks to SIP/1001-18f2) <-- there is no target EXTENSION. |
| 17:54.34 | [TK]D-Fender | NEXT!!@!!@ (c) BKW |
| 17:54.35 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 17:57.03 | hamush | [TK]D-Fender: does that probably mean there's something wrong with the definition of 1001 in my sip.conf? |
| 17:58.20 | [TK]D-Fender | hamush: <-- SIP read from 127.0.0.1:5060: SIP/2.0 482 Loop Detected <--- you are also doing something else insane here |
| 17:59.33 | [TK]D-Fender | hamush: Reliably Transmitting (no NAT) to 127.0.0.1:5060: <- * is calling ITSELF |
| 18:01.47 | *** join/#asterisk cusco (n=tralala@2001:0:53aa:64c:1821:6ebc:2ac0:762d) |
| 18:01.51 | cusco | hi |
| 18:01.58 | cusco | asterisk is taking 99% cpu |
| 18:02.12 | cusco | Im getting this mensage in debug: |
| 18:02.13 | cusco | DEBUG[25422] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=481 |
| 18:02.16 | cusco | over and over again |
| 18:02.38 | *** join/#asterisk JackTheNipple (n=JackTheN@static-87-79-237-194.netcologne.de) |
| 18:02.47 | Qwell | cusco: I want to say that's been fixed in a later version of 1.6 |
| 18:02.55 | Qwell | err, are you running 1.6? |
| 18:03.17 | cusco | 1.6.1.1 |
| 18:03.27 | Qwell | hrm |
| 18:03.48 | cusco | sorry 1.6.0 |
| 18:03.56 | Qwell | 1.6.0.1? |
| 18:04.03 | cusco | 1.6.1.0 will be the on that we will put into production tomorrow |
| 18:04.16 | cusco | not sure, how can I tell? |
| 18:04.20 | Qwell | core show version |
| 18:04.22 | bmoraca | sounds like you might be putting it in today ;) |
| 18:05.12 | cusco | Asterisk 1.6.0 built by root @ perfpbx on a i686 running Linux on 2008-10-08 10:12:52 UTC |
| 18:05.14 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 18:05.34 | cusco | bmoraca: tomorrow only, even if I have to restart asterisk |
| 18:05.40 | russellb | that version is about 9 months old :-) |
| 18:05.44 | cusco | I wouldn't want that tho, There are people talking... |
| 18:05.58 | russellb | at least update to the latest 1.6.0 release |
| 18:06.02 | russellb | 1.6.0.X |
| 18:06.34 | cusco | russellb: we run a 24h service, we can't afford to stop calls , we will change teh server, to update kernel etc on this one. that will happen tomorro 7am tho |
| 18:06.42 | eppigy | Katty:HI |
| 18:07.06 | cusco | how can I know what is making this 99% cpu usage? |
| 18:07.11 | cusco | and what is that error? |
| 18:07.26 | bmoraca | it's a bug...it's fixed in a newer version than you have |
| 18:07.40 | bmoraca | restart asterisk and it'll go away for the time being |
| 18:08.24 | Qwell | cusco: If you truly can't afford to stop calls, you need a backup server. One which should be used for exactly this purpose. |
| 18:08.26 | cusco | I soft hanged upa call that was really noisy... and it went back to normal |
| 18:08.46 | cusco | Qwell: thats what we will be worrying about tomorrow |
| 18:09.24 | cusco | we have 3 machines right now, and we are setting up the new server then update the current toa redundancy server |
| 18:09.48 | cusco | meanwhile killing that call, reduced the system load |
| 18:10.01 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-38c71be58d9bfaf8) |
| 18:12.48 | *** join/#asterisk malaiwah (n=mbelleau@modemcable209.80-20-96.mc.videotron.ca) |
| 18:12.58 | juny4ng | Hi guys |
| 18:13.12 | juny4ng | Anyone know if i can pass "parameters" from originateAction (Asterisk Manager) to a particular AGI script? |
| 18:13.46 | [TK]D-Fender | juny4ng: Yes, its in the docs |
| 18:14.26 | juny4ng | [TK]D-Fender: Do you know which part? I tried searching for it but had no answers.. Thanks! |
| 18:14.39 | [TK]D-Fender | juny4ng: Go read them again |
| 18:16.25 | juny4ng | yup.. do you mean the AGI(script.agi|arg1|arg2|...) portion? |
| 18:16.54 | [TK]D-Fender | juny4ng: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate |
| 18:17.46 | juny4ng | Okay, i see it now. Variable.. Thanks alot! |
| 18:27.53 | *** join/#asterisk zeroHalo (n=zeroHalo@75.150.77.161) |
| 18:29.05 | hamush | if i want a certain sip account to only accept connections from localhost, is host=localhost in the appropriate section in sip.conf the best way to do that? |
| 18:31.06 | hamush | i ask because i've tried and am getting auth issues :P (although host=dynamic seems to work just fine) |
| 18:31.25 | [TK]D-Fender | hamush: It works, but your dialplan is sending the call in CIRCLES |
| 18:33.36 | hamush | [TK]D-Fender: yeah, i think i resolved that issue... i've got a sip user agent running on the same machine as asterisk that is set to use a different port (5061), but asterisk was getting confused and trying to use an outdated successful registration on port 5060 because the registration was failing because of the host=localhost option |
| 18:34.19 | *** join/#asterisk AwayML (n=AndyML@pool-173-49-143-205.phlapa.fios.verizon.net) |
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| 18:43.15 | *** join/#asterisk pulpster (n=pulpster@p16.eregie.pub.ro) |
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| 19:05.44 | cusco | hi |
| 19:06.05 | cusco | queue show blah shows some callers |
| 19:06.17 | cusco | how can I relate that call with an extention or its unique id? |
| 19:06.25 | cusco | to take it out, and hang it up |
| 19:10.29 | cusco | or how to hang up a call that is on queue or evenif it has been answered? |
| 19:13.07 | *** join/#asterisk zdennis (n=zdennis@75.119.89.19) |
| 19:13.18 | zdennis | is it possible to hook into when a caller terminates from a call? |
| 19:17.12 | *** join/#asterisk errotan (n=errotan@5403E5C2.catv.pool.telekom.hu) |
| 19:19.39 | *** join/#asterisk apdg (n=grey@ash.stopavoiding.us) |
| 19:21.04 | zdennis | found it, 'h'' |
| 19:24.50 | mykhyggz | Does anyone know a replacement for "festival" application in 1.6.X? |
| 19:25.14 | jkroon | what does festival do again? |
| 19:25.27 | zeroHalo | text to speech |
| 19:25.32 | mykhyggz | Yeah. |
| 19:26.00 | jkroon | oss? |
| 19:26.12 | jkroon | or floss as people call it nowadays? |
| 19:26.22 | jkroon | cepstral is another option. |
| 19:27.11 | *** join/#asterisk Micc (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 19:27.20 | mykhyggz | Okay, I'll have a look. Thanks. |
| 19:30.36 | Katty | decides to do her nails. |
| 19:31.00 | *** join/#asterisk tobias (n=tobias@216.27.28.176) |
| 19:32.28 | pulpster | hello |
| 19:32.38 | mykhyggz | So... cepstral has a licence to purchase. That's out. FLOSS doesn't seem to relate to speech synthesis, unless I'm missing something. Is there a similar replacement to Festival? |
| 19:32.52 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 19:33.23 | pulpster | does anyone succesfully configured a cisco 7911 with asterisk using chap_sccp or chap_skinny. I need a working configuration |
| 19:33.56 | pulpster | please help - or take a look here: http://trixbox.org/forums/vendor-forums-non-certified/linksys/cisco/rtp-channel-problem-7911g-i-get-no-voice-payload-tr |
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| 19:34.33 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-105-164-34.nys.biz.rr.com) |
| 19:37.48 | pulpster | thanks all in advance - really any idea would be greatly apreciated |
| 19:37.50 | pulpster | I am stuck |
| 19:40.00 | Katty | blinks |
| 19:40.08 | Katty | so a coworker just walked into my office asking me to burn her a cd |
| 19:40.18 | Katty | normally i wouldn't be opposed |
| 19:40.23 | Katty | but it's for an american idol demo |
| 19:40.29 | Katty | wonders what on earth this has to do with her job |
| 19:40.55 | [TK]D-Fender | Katty: Give it to her.... then burn it... WITH FIRE!!!!! |
| 19:41.03 | *** join/#asterisk maagic (i=maagic@fsck.fi) |
| 19:44.32 | Katty | oh well |
| 19:44.38 | Katty | wasn't like i was doing anything but taking off nail polish anyway |
| 19:50.11 | *** part/#asterisk icyValk77 (n=icyValk7@host81-153-93-26.range81-153.btcentralplus.com) |
| 19:50.37 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 19:50.52 | eppigy | TRABAJO |
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| 19:53.19 | shazaum | eppigy, =/ |
| 19:56.05 | eppigy | 8[] |
| 20:03.05 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 20:03.39 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-246-78.lns10.mel6.internode.on.net) |
| 20:04.59 | *** join/#asterisk SkramX (n=SkramX@cpe-70-112-25-138.austin.res.rr.com) |
| 20:05.55 | Katty | HI EPPIGY |
| 20:05.59 | eppigy | hiya |
| 20:06.11 | [TK]D-Fender | WHAT ARE YOU DOING DAVE? |
| 20:08.02 | eppigy | I watched that for the first time a week or so ago |
| 20:08.10 | *** join/#asterisk lasko (n=lasko@70.102.15.210) |
| 20:08.50 | [TK]D-Fender | eppigy: Don't watch the ending when you're very tired.... its bad enough lucid, but it'll mess you up (more) |
| 20:08.52 | lasko | On asterisk1.6 trunk I'm trying to test the latest but when I try to compile I'm getting "chan_ooh323.h:53:26: error: asterisk/rtp.h: No such file or directory" -- Where is this file? |
| 20:09.48 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 20:12.34 | *** join/#asterisk Aiatek (n=amunoz@75.112.88.200.m.sta.codetel.net.do) |
| 20:15.25 | Winkie | hey guys, anyone got a budgettone, i need to know whether it will respond correctly to alert-info |
| 20:15.46 | Winkie | looking specifically at the BT-201s |
| 20:16.32 | Aiatek | Winkie you mean grandstream |
| 20:16.35 | Aiatek | ? |
| 20:16.50 | Winkie | Aiatek: yes |
| 20:16.59 | Aiatek | :{ |
| 20:17.35 | bmoraca | I have a pile of paper on my desk I use to pass notes to my coworkers...that's about as effective as a grandstream phone. |
| 20:17.56 | Aiatek | bmoraca you are right |
| 20:18.03 | Winkie | haha, well it's not my choice what phone to get |
| 20:18.10 | Winkie | only my job to research whether it supports the right features |
| 20:20.47 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:21.45 | Winkie | just hoping someone here has one and can pass it an Alert-info and see if it does support it |
| 20:22.28 | [TK]D-Fender | checkout time, later all |
| 20:26.09 | *** part/#asterisk zeroHalo (n=zeroHalo@75.150.77.161) |
| 20:31.59 | bmoraca | Winkie: shouldn't the feature research come BEFORE the phone purchase? |
| 20:32.18 | Winkie | bmoraca: it is |
| 20:32.28 | Winkie | they have picked that phone, and asked me to check it supports all the features |
| 20:32.32 | bmoraca | oh...well that makes it easy: buy polycoms. |
| 20:32.51 | bmoraca | Winkie: tell them no. it'll save you time, money, and headache in the future. |
| 20:33.13 | Winkie | bmoraca: yeah i'm fairly sure they won't go for that |
| 20:33.17 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 20:33.21 | Winkie | the only feature they care about is alert-info |
| 20:33.24 | Winkie | and there are alternative options |
| 20:33.39 | Winkie | so for the 100% difference in price i don't think they're going to buy it |
| 20:33.53 | Winkie | bmoraca: i don't have to support this client anyway :) |
| 20:34.03 | bmoraca | I guarantee you that two weeks after you buy them, you will be back here asking questions that would be a moot issue if you had invested in the higher quality phones |
| 20:34.21 | Winkie | haha, i know exactly those concerns |
| 20:34.53 | Winkie | if i could control it, i would probably get a bunch of linksys spa 921s |
| 20:35.03 | Winkie | but like i say, all i can tell them is that it doesn't support it |
| 20:35.13 | Winkie | and then they will look at the price of the next range up and buy the cheap ones anyway |
| 20:35.34 | Winkie | thanks for your concern though |
| 20:35.54 | Winkie | guessing you see plenty of "omg i bought this phone and it does not wor kwtf!!" in here :) |
| 20:36.02 | bmoraca | Winkie: as the system integrator, it's your job to make sure that the customer is getting the best possible solution. I don't offer my customers anything except Polycom and Cisco (not my decision, but my boss insists on offering Cisco) |
| 20:36.20 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 20:36.35 | Winkie | bmoraca: i am not really the system integrator, i am a third party contractor for a friend of theirs who they've asked to build them an ultimate budget system |
| 20:36.44 | Katty | hmm. i'm just not feelin this blue nail polish |
| 20:37.04 | Winkie | i agree with your concerns anyway, and if i had a choice i wouldn't be choosing these budgetones |
| 20:37.13 | bmoraca | Winkie: all I can say is to check the user manual to see whether or not they claim to support the feature you want. |
| 20:37.22 | *** join/#asterisk yerwish (n=geekangl@94.77.235.168) |
| 20:37.29 | Winkie | bmoraca: yeah it supports locally configured distinctive rings, so i suspect it might |
| 20:37.37 | Winkie | figured here was the best place to find people with a wide selection of phones |
| 20:37.51 | ariel_ | Cisco's are great expensive phones and routers. I love the polycom's and use them as primary phones. Both have there use. |
| 20:37.53 | bmoraca | that's not necessarily the same thing. distinctive ring doesn't have to come from alert-info |
| 20:38.06 | Katty | Winkie: you will find our knowledge doesn't cover cheaper things. |
| 20:38.29 | Winkie | bmoraca: that's true, their support is based off callerid, but it at least confirms that their chipset can sound different rings :) |
| 20:38.43 | Winkie | Katty: oh i'm sure plenty of people have taken the extreme budget route |
| 20:38.57 | Katty | oh i'm sure of that too |
| 20:39.01 | Katty | but i'm not sure you will find them here |
| 20:39.08 | Winkie | well no harm if not |
| 20:39.14 | Winkie | not going to cost me any money or reputation |
| 20:39.15 | Katty | of course |
| 20:39.16 | bmoraca | or if they are here, they'll be asking the questions, not answering them :P |
| 20:39.31 | Winkie | perhaps, i should hang around more and see if there's anything i can answer |
| 20:39.37 | Winkie | although i haven't installed 1.6 on anything yet |
| 20:40.15 | Katty | bmoraca: well said |
| 20:40.31 | Katty | eppigy: this blue is boring :< |
| 20:40.45 | Katty | eppigy: i'm thinking something more cherry red. |
| 20:40.52 | Katty | eppigy: thoughts? |
| 20:41.35 | anthm | it's #asterisk how bout orange =p |
| 20:41.49 | Katty | orange won't match a lot of my clothing. |
| 20:42.27 | Katty | all tho i do have anew orangey one i've been wanting to try |
| 20:42.29 | Katty | maybe i will. |
| 20:42.48 | eppigy | Katty: i like red but not so much on lips and nails |
| 20:43.09 | Katty | eppigy: i think i'm gonna try my new orangey color |
| 20:43.54 | *** join/#asterisk ManxPower (n=manxpowe@69.73.94.162) |
| 20:44.20 | Katty | hi manx |
| 20:44.25 | ManxPower | Hello Katty |
| 20:46.12 | bmoraca | Winkie: my statement wasn't based off of the intelligence of the user, but rather off the fact that cheaper equipment tends to have more issues than more expensive equipment...issues which wouldn't be there. |
| 20:46.53 | bmoraca | Winkie: for instance, when we installed our phone system, we decided to shave a little off the cost and put Cisco 7912 phones in certain locations instead of 7940s...the savings was about $40/phone... |
| 20:47.22 | ManxPower | bmoraca: At least you didn't use Grandstream to save money. |
| 20:47.24 | bmoraca | Winkie: the issue is that 7912 phones running SIP firmware are absolutely incapable of ringing for more than 19 seconds. that's a problem when you're putting htem in a call center. |
| 20:47.37 | Winkie | bmoraca: no i'm aware of these sorts of problems |
| 20:47.42 | Winkie | this is not my first asterisk system |
| 20:47.44 | Winkie | nor my first callcentre |
| 20:47.45 | *** join/#asterisk SaiSoma|AFK (n=SaiSoma@74.167.136.30) |
| 20:48.04 | Winkie | if i could make the choice, i would |
| 20:48.09 | Winkie | and in fact have |
| 20:48.10 | bmoraca | Winkie: I spend several days trying to figure it out...learned my lesson the hard way, and now I don't even bother to tell the customer that a phone cheaper than the Polycom 330 even exists :) |
| 20:48.29 | Winkie | bmoraca: well we have a few linksys spa 921 phones around the place |
| 20:48.37 | Winkie | and i've found them to be an excellent phone for general use |
| 20:48.40 | Winkie | so they are usually what i like |
| 20:48.56 | Winkie | but in this case it is literally a friend of a friend asking him to spec him up a $1500 asterisk system for 6 users on an E1 |
| 20:49.13 | Winkie | so i don't really have any control there, i'll mention that the phones aren't recommended etc |
| 20:49.17 | bmoraca | i'm not a big fan of linksys phones...played around with them a bit, but meh |
| 20:49.20 | Winkie | but they'll buy them anyway, i can basically guarantee it |
| 20:49.35 | bmoraca | 1500 on an e1? that's not gonna happen...nor is it gonna be nice...heh |
| 20:49.36 | Winkie | bmoraca: cisco own linksys now, and the 921 is one of the joint ones afaik |
| 20:49.56 | bmoraca | Winkie: actually, the linksys phones are sipura based, not cisco |
| 20:50.11 | ManxPower | Heck, 1/3 of that cost is just the E1 card. |
| 20:50.30 | Winkie | yep |
| 20:50.36 | Winkie | now you see why i have no choice in the matter |
| 20:50.39 | bmoraca | 1/3? a Sangoma A101 with echo cancel is $800 |
| 20:50.46 | *** join/#asterisk oglynn (n=me@173-9-102-81-NewEngland.hfc.comcastbusiness.net) |
| 20:51.00 | bmoraca | why do you need an e1 for 6 phones? |
| 20:51.01 | Winkie | yep, they're actually spending more on the card than on the server it's going into |
| 20:51.04 | ManxPower | Of course you have a choice. You can choose not to be involved in a project that sucks. |
| 20:51.06 | Winkie | bmoraca: it is cheaper than BRIs |
| 20:51.22 | Winkie | ManxPower: well it's a choice of me getting paid for sending 2 PDFs and writing an extensions.conf, or not getting paid |
| 20:51.25 | Winkie | so i prefer to get paid :) |
| 20:51.26 | oglynn | wondering if there is an easy way to light MWI from the dial plan or from a script? |
| 20:51.27 | bmoraca | oh, so you actually found a provider who will do a partial e1? |
| 20:51.38 | Winkie | bmoraca: they did :) |
| 20:51.46 | ManxPower | Winkie: that depends on how much you want the money |
| 20:51.50 | bmoraca | can't get partial T1s here in the states |
| 20:51.54 | Winkie | bmoraca: really? |
| 20:51.57 | ManxPower | bmoraca: of course you can. |
| 20:51.57 | Winkie | that's odd |
| 20:52.07 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 20:52.07 | Winkie | yeah i could swear i've worked with a fractional t1 before |
| 20:52.13 | bmoraca | well, you can...but most providers won't do it. none of the ones in central CA will do it. |
| 20:52.21 | Winkie | anyway ManxPower my responsibility for maintainance after installation: 0 |
| 20:52.21 | ManxPower | It's much harder with and RBOC, but I've personally installed many partial T-1 PRIs |
| 20:52.35 | Winkie | i will gladly charge them a large amount of cash to fix their problems |
| 20:52.59 | ManxPower | Winkie: exactly. You'll end up getting screwed by the maintainance as the customer won't want to pay for support for "that crappy phone system". 8-) |
| 20:53.15 | Winkie | ManxPower: haha, they'll just complain to their friend |
| 20:53.21 | Winkie | which is not me :D |
| 20:53.47 | Winkie | actually, considering this channel has more experience on expensive objects |
| 20:53.47 | ManxPower | bmoraca: Almost every T-1 PRI I've installed were less than 23 B-channels. Maybe it's much different in California |
| 20:53.58 | Winkie | i was discussing asterisk redundancy with a friend the other day |
| 20:54.11 | Winkie | but we were unable to find an acceptable e1 cross connect switch type arrangement for 4 PRIs |
| 20:54.14 | ManxPower | One of the oddities in USA telecom is that each state is different |
| 20:54.17 | Winkie | anyone got any preferences etc? |
| 20:54.48 | ManxPower | bmoraca: Find yourself a good CLEC. 8-) |
| 20:54.49 | *** join/#asterisk rwong (n=ricky@www.roflwaffle.com) |
| 20:54.52 | bmoraca | ManxPower: well, i understand where they're coming from...it costs just as much to maintain the copper if 3 or 23 channels...so they just don't bother if you have less. |
| 20:55.09 | Winkie | well in this situation they have fibre close to the building |
| 20:55.31 | Winkie | so it's a choice of splitting a single pri into another pri |
| 20:55.33 | Winkie | or splitting a single pri into some BRIs |
| 20:55.35 | bmoraca | ManxPower: we have one....but they're too far away from us to be of any use in that scenario...hence why I'm starting to do SIP-to-PRI media gateways for my customers |
| 20:55.41 | Winkie | less work and equipment for PRI i guess |
| 20:55.57 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 20:56.03 | oglynn | Is an easy way to light MWI from the dial plan or from a script? |
| 20:56.05 | Katty | wants a minpin/chihuahua puppy |
| 20:56.13 | ManxPower | bmoraca: All the Frac-PRIs I've installed were billed as the Local Loop + Per Channel Cost. At least in Louisiana Voice T-1s are much cheaper for the local loop. So we frequently put a couple of PRI B-channels on the local loop and use the rest for data. |
| 20:56.34 | Winkie | haha data over here is impossibly annoying |
| 20:56.38 | bmoraca | ManxPower: CA is a huge state...lots of mileage costs here. the per channel costs is such a small fraction that it just doesn't make any sense. |
| 20:56.40 | Winkie | we have two 10mbit site to site links |
| 20:56.43 | ManxPower | The local loop is always provided by the RBOC, of course. |
| 20:56.50 | Winkie | and they've pushed the cost up to like $15,000 a year for both |
| 20:56.57 | Winkie | we're replacing them with the same product from a competitor |
| 20:57.01 | ManxPower | bmoraca: Ah. Our local loops were usually about $150/month |
| 20:57.04 | Winkie | but because we have British Telecom, it's actually the same fibre |
| 20:57.08 | Winkie | going through the same ducts |
| 20:57.11 | Winkie | handled by the same engineers |
| 20:57.14 | Winkie | but it costs us 1/4 |
| 20:57.16 | bmoraca | ManxPower: yes, some of them do the integrated T1s where they split some channels off for PRI...but again, the mileage is the biggest cost |
| 20:57.18 | Winkie | even though we're now going through a middleman |
| 20:58.06 | ManxPower | I don't think we ever paid mileage on the voice T-1s |
| 20:58.15 | bmoraca | where are you located? |
| 20:59.04 | ManxPower | bmoraca: Former client, not current client. They had offices all over Louisiana and a few offices in Mississippi, they used I-55 as their CLEC |
| 20:59.24 | bmoraca | interesting |
| 21:00.04 | ManxPower | bmoraca: The client used to be the 2nd largest customer of the CLEC (the CLEC's ISP was the largest). Then they were bought by a big CLEC and customer service went all to hell. |
| 21:00.09 | bmoraca | I have Telepacific, AT&T, O1, and Pacwest available...and none of them will do a fractional PRI |
| 21:00.47 | bmoraca | well, Pacwest probably would...but only because I'd have to do the termination/origination myself |
| 21:00.48 | oglynn | anyone got any idea on sending an MWI command from script or dial plan? |
| 21:00.57 | ManxPower | bmoraca: Even BellSouth would do Frac if we pushed hard enough |
| 21:01.03 | Winkie | oglynn: iirc there's some info on voip-info.org that might help |
| 21:01.22 | ManxPower | Then they merged with AT&T and customer service went all to hell. |
| 21:01.27 | Winkie | bmoraca: how do you handle asterisk redundancy? specifically the PRI side? |
| 21:01.45 | ManxPower | bmoraca: much of the partial PRIs we used had data on the unused channels |
| 21:02.00 | bmoraca | they're all moving to SIP-based voice delivery...even for "integrated T1" customers. AT&T uses Cisco 1841s and Telepacific and O1 use Adtran |
| 21:02.50 | Winkie | we're seeing this with something called "21CN" over here |
| 21:02.57 | Winkie | a national initiative to modernise infrastructure |
| 21:03.03 | ManxPower | idly notes he is looking for employment, currently living in Huntsville |
| 21:03.05 | bmoraca | Winkie: I generally don't worry about it. if your T1 goes out, there are much larger issues at hand. if the provider is having upstream issues, survivability of your numbers is doomed, and if you just need outbound survivability, a couple analog lines or a SIP account is sufficient |
| 21:03.06 | Winkie | or conversely, a management initiative to make an assload in bonuses and waste money |
| 21:03.20 | Winkie | bmoraca: actually i mean more the connectivity to asterisk |
| 21:03.23 | Winkie | say for example you have 4 PRIs |
| 21:03.26 | Winkie | 4 asterisk servers |
| 21:03.30 | Winkie | and load balanced phone connectivity etc |
| 21:03.42 | Winkie | do you retain the ability to move a PRI termination if a server were to fail? |
| 21:03.57 | Winkie | or is it a case of going on site and patching manually? |
| 21:04.22 | Winkie | the only solutions we've found are intended to be used with 16+ PRIs |
| 21:04.26 | Winkie | and so are not really economical |
| 21:05.46 | bmoraca | Winkie: in a case like that, I would use an IP-based load balancing appliance for phone connectivity to multiple servers and have all 4 PRIs go into a single box, trunking via SIP to the SIP proxy servers |
| 21:06.11 | Winkie | bmoraca: aha that is one of the two options we came up with |
| 21:06.17 | Winkie | the other was to set up asterisk as a tree arrangement |
| 21:06.53 | bmoraca | Winkie: the issue with maintaining multiple POPs is that Asterisk wasn't really designed for it and it's a dialplan nightmare |
| 21:06.55 | Winkie | i guess until someone makes a small scale switch that's the best job |
| 21:07.08 | ManxPower | \ |
| 21:07.27 | Winkie | bmoraca: it is? we mocked up a simple query daemon to find the right termination which seemed ok |
| 21:07.34 | ISO9001 | DUNDi? |
| 21:07.37 | Winkie | using the manager interface |
| 21:07.57 | Winkie | ISO9001: nope, just some hacked on thing |
| 21:08.14 | ISO9001 | no, I'm suggesting that as a possible answer for multiple POPs. |
| 21:08.42 | Winkie | ISO9001: yeah, i'm just saying we didn't use it :) |
| 21:09.04 | ISO9001 | gotcha. |
| 21:09.32 | ISO9001 | pfft... I use it for a home setup, because I'm a glutton for punishment ;) |
| 21:09.41 | Winkie | o rly |
| 21:09.45 | Winkie | i might have to investigate |
| 21:09.58 | Winkie | one of our providers is exposing an API to move number termination between PRI groups |
| 21:10.08 | Winkie | so that would probably provide HA throughout |
| 21:10.44 | Winkie | i have like 4-5 1u machines downstairs |
| 21:10.44 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 21:11.02 | Winkie | ISO9001: o hey that looks quite simple |
| 21:11.09 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:11.16 | ISO9001 | it's not as bad as the various guides make it look. |
| 21:13.12 | *** join/#asterisk Zuchmir2 (n=Zuchmir-@ool-18bd3bfc.dyn.optonline.net) |
| 21:14.00 | Zuchmir2 | is there a prog similar to sox to convert g729 |
| 21:14.18 | russellb | asterisk with codec_g729, heh |
| 21:14.27 | russellb | asterisk -rx "convert foo.wav foo.g729" |
| 21:15.04 | Zuchmir2 | [TK]D-Fender: thanks for your help yesterday, the issue was ALG on the firewall was enabled and that caused the problems |
| 21:15.46 | *** join/#asterisk malaiwah (n=mbelleau@modemcable209.80-20-96.mc.videotron.ca) |
| 21:17.13 | Zuchmir2 | @russellb: is there something that's less "bulky", i have thousands of files that need conversion |
| 21:17.35 | bmoraca | a quick bash script? |
| 21:17.37 | russellb | not that I am aware of. |
| 21:17.56 | Zuchmir2 | ok, thx |
| 21:18.16 | Winkie | asterisk is not that 'bulky' in any sense i think |
| 21:18.39 | Winkie | the executable is about a megabyte on a random installation i picked, and will be cached in memory |
| 21:18.51 | Winkie | a bash script should work fine |
| 21:18.51 | [TK]D-Fender | Winkie: Thats what res_steriods.so is for |
| 21:19.12 | Winkie | [TK]D-Fender: i've heard that can lead to unexpected volume increases |
| 21:19.22 | Winkie | and problems with your 'termination' |
| 21:19.46 | [TK]D-Fender | Winkie: Can cause massive plugin deflation |
| 21:19.48 | hamush | is there a way to check if asterisk is getting any RTP packets from a sip client? |
| 21:19.58 | Winkie | hamush: tcpdump? |
| 21:20.06 | Zuchmir2 | bmoraca: if need be a can setup * on a secondary server, and run the conversions there... |
| 21:20.08 | Winkie | i always resort to tcpdump :( |
| 21:20.17 | Winkie | Zuchmir2: start it niced down? |
| 21:20.30 | hamush | Winkie: yeah, i might have to go down that road :-/ |
| 21:20.56 | Winkie | hamush: but tcpdump is nice? |
| 21:21.06 | *** join/#asterisk sjobeck (n=Adium@64.122.41.37) |
| 21:21.14 | Zuchmir2 | i have production servers, and dev servers, i can do the conv on non-production sever at full speed |
| 21:22.00 | Winkie | Zuchmir2: i've seen some talk about using ffmpeg in the past, but i can't really remember whether it was 729 or some other patent restricted format |
| 21:22.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:24.13 | cusco | im still trying to test sipp with asterisk. I can't understand why calls don't get to queue and they are hang up |
| 21:25.25 | sjobeck | hey, hi, all, i have a fun one for you to wrap your collective heads around. how can i have a SIP ext shown as registered all the time even though there is no phone attached? |
| 21:26.19 | Winkie | sjobeck: can i ask why you would want that? |
| 21:26.19 | bmoraca | sjobeck: better question is why would you want to? |
| 21:26.34 | sjobeck | hahahah |
| 21:27.17 | sjobeck | i understand your asking. doesn't seem sane does it, but i need these for call forwarding, need about 8 of them, but they MUST show to our software as online (AKA registered) |
| 21:28.06 | Winkie | sjobeck: i don't understand, does your software use 'sip show peers' or something to determine which extensions are active? |
| 21:28.19 | Winkie | (i forget the Manager command) |
| 21:28.58 | sjobeck | good question: not sure what it does to "know" but, yes, it does know, i guess i can try to find that out, but it wont let us call one of these 8 extension when theyre offline |
| 21:29.34 | Winkie | right, well how it figures out they're online is quite important |
| 21:33.56 | [TK]D-Fender | sjobeck: How is this piece of software even LOOKING at the state of this other device? |
| 21:34.27 | sjobeck | sorry, i just dont know yet. |
| 21:34.39 | [TK]D-Fender | sjobeck: that makes no sense. |
| 21:34.40 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 21:35.03 | [TK]D-Fender | sjobeck: And you can't expect to be handed a solution when you can't say how the device is even being monitored |
| 21:35.14 | sjobeck | :) |
| 21:35.15 | sjobeck | true |
| 21:35.19 | sjobeck | true |
| 21:35.24 | sjobeck | i dont know ........... yet |
| 21:35.29 | sjobeck | I'm finding out |
| 21:35.42 | [TK]D-Fender | sjobeck: Clues are in a bin to the left of the door... let us know when you've found one ;) |
| 21:35.50 | cusco | http://pastebin.com/m79996c19 |
| 21:35.55 | sjobeck | but at the same time, we dont know how (yet) how to force an ext to show as registered all the time ? |
| 21:36.17 | cusco | I can't see why it doesn't reach the extention, its not even put into a queue -- why does it hangs up?? |
| 21:36.52 | Winkie | sjobeck: well it's unlikely to be easy, but it may just be a case of altering the logic your application uses to detect whether an extension is 'registered' |
| 21:37.15 | *** join/#asterisk killfill (n=killfill@200.63.96.244) |
| 21:37.46 | [TK]D-Fender | sjobeck: Appearances depends on what is being looked at, and how <- |
| 21:37.56 | sjobeck | winkie: i am most likely not going to be able to hack this app', since its 3rd party. unfortunately. thx. |
| 21:38.03 | [TK]D-Fender | sjobeck: So tell your chicken to go eat the egg and come back with that clue. |
| 21:38.26 | Winkie | sjobeck: well even so, knowing how it detect availability is key to knowing how to fool it :) |
| 21:38.49 | killfill | hey.. when i call AgentLogoff on an agent that is currently ringing its phone, then the call get passed to the next agent, but the original agent gets not logged off the queue |
| 21:39.01 | Winkie | of course, the next stage is to contact your vendor |
| 21:39.03 | killfill | how would i 'force' to loggoff?.. |
| 21:39.21 | cusco | Im getting ----- Spawn extension (HF, s, 1) exited non-zero on 'SIP/5061-01e07a00' |
| 21:39.38 | cusco | is thsi the clinet hanging up? |
| 21:41.33 | [TK]D-Fender | cusco: Ending,yes. Why is harder to see |
| 21:41.34 | *** join/#asterisk jamesh1 (n=jhenders@xob.neospire.net) |
| 21:41.47 | killfill | any tips?.. :P |
| 21:43.25 | *** join/#asterisk ingenius (n=alektro@host117.190-138-52.telecom.net.ar) |
| 21:43.55 | *** join/#asterisk hardwire (n=hardwire@216-67-98-253.static.acsalaska.net) |
| 21:44.14 | hardwire | I need somewhere to terminate like. 40 simultaneous calls as a test.. anybody know of a good resource for that? |
| 21:44.20 | hardwire | pstn |
| 21:45.47 | [TK]D-Fender | hardwire: As a test of what? |
| 21:46.15 | hardwire | anything.. echo.. milliwatt.. looped recording |
| 21:46.25 | hardwire | something that won't make the feds knock on my door.. preferable. |
| 21:47.00 | [TK]D-Fender | hardwire: No, what is the point of this test? |
| 21:47.58 | hardwire | capacity testing for some channel recording |
| 21:48.13 | Winkie | hardwire: another asterisk server |
| 21:48.19 | hardwire | Winkie: $$$ |
| 21:48.21 | Winkie | is the way we tested our voice recording |
| 21:48.22 | hardwire | ok.. thanks anyways |
| 21:48.40 | hardwire | cant afford another PRI card (using VoIP doesn't tax the IRQs as much) |
| 21:48.44 | hardwire | bbl |
| 21:49.33 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 21:50.08 | cusco | [TK]D-Fender: ok.. Im just fiddling with sipp |
| 21:57.42 | hamush | can asterisk proxy rtp traffic between two sip user agents? |
| 22:01.47 | hamush | or does it just tell the user agents to point their rtp streams at each other? |
| 22:02.49 | Winkie | hamush: it can do both, that's what canreinvite controls |
| 22:02.59 | ManxPower | hamush: It can do either, assuming the two devices can communicate directly |
| 22:03.32 | hamush | cool, i will look into canreinvite, thanks! |
| 22:04.01 | ManxPower | reinvites are enabed by default |
| 22:04.20 | ManxPower | reinvites don't (generally) work with NAT, of course |
| 22:05.29 | [TK]D-Fender | Martial arts time, BBL |
| 22:18.28 | infinity1 | how can i remotely control my polycom phone DND? |
| 22:18.38 | infinity1 | so if i'm out of the office the busy message gets enabled? |
| 22:22.15 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:27.03 | ManxPower | infinity1: you can't |
| 22:27.15 | ManxPower | you'll need to do it on the Asterisk server. |
| 22:28.07 | jkroon | lisp ... why lisp?!? (festival interface stuff) |
| 22:30.59 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 22:35.51 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 22:36.05 | mmlj4 | 'lo, ManxPower |
| 22:36.44 | carrar | freenode support ssl yet? |
| 22:37.10 | ManxPower | Hello, mmlj4 |
| 22:37.28 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 22:38.29 | Qwell | carrar: #freenode? |
| 22:38.30 | *** join/#asterisk Viral_ (n=Viral@bas15-toronto63-1177890976.dsl.bell.ca) |
| 22:38.43 | carrar | the ircd |
| 22:38.47 | carrar | looks like they don't yet |
| 22:39.00 | Qwell | ask them, not us |
| 22:39.13 | infinity1 | ManxPower: can i run a script on the astersik server to do it? |
| 22:39.16 | carrar | too late |
| 22:39.46 | ManxPower | infinity1: The polycom does not support remote changing of DND |
| 22:40.49 | bmoraca | infinity1: any modification to busy states or which voicemail greeting is played has to be controlled through Asterisk's dialplan. so, yes, you can run a script on your asterisk server to do it. |
| 22:41.58 | Viral_ | Question: Does anyone have ideas on how to control the audio streaming inbound and outbound (I kinda want Flash -> Ruby RTMP Deamon -> Outbound dialing) |
| 22:42.49 | ManxPower | Viral_: As long as the audio is RTP and the signaling is SIP, I don't see a problem. |
| 22:45.09 | Viral_ | So how would I take control over the audio streams/channels? |
| 22:46.40 | Viral_ | [dial out] "you have a call from xxx" [audio stream] [hangup] |
| 22:47.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:47.04 | ManxPower | Viral_: for the most part you don't. |
| 22:47.07 | ManxPower | Not with Asterisk. |
| 22:47.13 | Viral_ | Then what? |
| 22:47.37 | ManxPower | Asterisk has some good call control stuff by using the AMI (Asterisk Manager Interface), but there really isn't anything to manage audio. |
| 22:47.46 | ManxPower | Viral_: I don't know what else might do what you want. |
| 22:48.21 | ManxPower | <PROTECTED> |
| 22:48.49 | ManxPower | you can do that with a .call file and the Asterisk dialplan. I think you are making this much more complicated than it is. |
| 22:48.59 | Viral_ | Well, asterisk to me and more so sip connections are really just like static http connections, do x, I need to share streaming audio objects to it |
| 22:49.03 | Viral_ | I very well might be. |
| 22:49.59 | ManxPower | Viral_: I wish you the best of luck. |
| 22:54.07 | Viral_ | @ManxPower np, the idea came from webphones, but I don't want to handle sip connections through the client-side via sip or iax, I rather just share a stream to an existing sip configuration, I'm really surprised nobodies touched on the topic |
| 22:55.32 | ManxPower | Asterisk is a PBX Toolkit. |
| 22:56.19 | ManxPower | I recommend you set up Asterisk as a PBX before you try to make it cure wold hunger, develop FTL drive, or turn it into an object oriented audio management system. |
| 22:59.06 | Viral_ | @ManxPower sure, thanks, clutter innovation with sarcasm |
| 23:01.17 | *** part/#asterisk Viral_ (n=Viral@bas15-toronto63-1177890976.dsl.bell.ca) |
| 23:03.22 | *** join/#asterisk SaiSoma|AFK (n=SaiSoma@74.167.136.30) |
| 23:06.05 | bmoraca | ManxPower: how did you understand anything that was said? runon sentences and what not...yikes! |
| 23:06.26 | *** join/#asterisk korcan (n=korcan@99.23.50.73) |
| 23:09.38 | *** join/#asterisk c4t3l (n=c4t3l@c-76-31-57-251.hsd1.tx.comcast.net) |
| 23:09.44 | c4t3l | greetings all! |
| 23:11.14 | c4t3l | i got a gang of polycom ip330s and I cant seem to dialout 4 digits to *. 10 digits & 911 work fine. Using defualt sip.cfg available from polycom. Using sip firmware version 3.0.4 |
| 23:13.15 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:13.18 | c4t3l | I've already modulated the flux capacitor |
| 23:13.56 | jql | c4t3l: I actually see that question on my support flow chart... Try dialing the number with the handset on the hook, and then lifting the handset to dial out. Does it work, then? |
| 23:14.19 | c4t3l | no bueno |
| 23:14.27 | jql | what error do you get? |
| 23:14.34 | c4t3l | on ast console? |
| 23:14.37 | c4t3l | nothing |
| 23:14.38 | jql | on the phone |
| 23:14.43 | c4t3l | one sec |
| 23:15.01 | c4t3l | connecting to:... enter more digits |
| 23:15.37 | lanning | check the dialplan on the phone |
| 23:15.55 | c4t3l | i double checked the digitmap in the sip.cfg and it looks like it should work |
| 23:16.00 | jql | hmm... at this point, I'd have to reset your dialplan to the default |
| 23:18.41 | c4t3l | very strange |
| 23:18.48 | c4t3l | I am using the default sip.cfg file |
| 23:19.06 | c4t3l | downloaded from polycom |
| 23:23.55 | bmoraca | c4t3l: always when I've had that issue, it's been a registration issue |
| 23:24.03 | c4t3l | hmm |
| 23:24.10 | bmoraca | or an asterisk dialplan issue |
| 23:24.24 | bmoraca | enable sip debug in asterisk and attempt...you should see a 404 |
| 23:24.31 | c4t3l | strange thing is that i can dial 10 digits and 911 and my special vm codes |
| 23:24.43 | c4t3l | four digits just dont get sent to asterisk |
| 23:24.44 | bmoraca | c4t3l: then it's a dialplan issue in asterisk |
| 23:25.00 | bmoraca | c4t3l: again, enable sip debug and try. |
| 23:25.15 | c4t3l | would I not see the failure on the ast console? |
| 23:25.31 | bmoraca | not if the location is not found, no |
| 23:26.07 | bmoraca | pastebinning your sip.conf and the relevant portions of extensions.conf might be helpful, too |
| 23:28.47 | Zuchmir2 | asterisk -rx "convert /voice/cvt/english/history/histo135.wav /voice/cvt/english/history/histo135.g729" comes back with: Failed to convert /voice/cvt/english/history/histo135.wav to /voice/cvt/english/history/histo135.g729! |
| 23:29.25 | bmoraca | do you have the g729 codec loaded into asterisk? |
| 23:31.28 | *** join/#asterisk Olobola (i=Olobola@104.sub-75-210-79.myvzw.com) |
| 23:32.27 | Zuchmir2 | 0/0 encoders/decoders of 1 licensed channels are currently in use |
| 23:32.48 | Zuchmir2 | some files work, others don't |
| 23:34.26 | Olobola | I can't get dtmf to pass for some reason. This only happens when calling out to a DID number. Phone mazes for bank of america and Verizon don |
| 23:34.36 | Olobola | 't work but voicemail does. |
| 23:35.08 | Olobola | I mean regular #'s |
| 23:37.15 | bmoraca | Olobola: what are you using for trunks? analog? if so, increase your gain. |
| 23:40.12 | Olobola | bmoraca: x-lite -> asterisk server -> voip provider |
| 23:41.26 | bmoraca | then i can't help you. softphones are notoriously unreliable, and SIP isn't exactly reliable itself. i'd wager it's probably a fluke of the setup, rather than a config issue. |
| 23:41.35 | c4t3l | bmoraca: thanks alot. I just needed to debug my sip to see a typo in my dialplan. I'd stepped away from asterisk for a couple of years and got rusty :P |
| 23:42.11 | bmoraca | c4t3l: a quick about the 330 is that it'll often give you that message, rather than play a reorder, when you attempt to dial an extension that doesn't exist |
| 23:42.19 | bmoraca | s/quick/quirk/ |