| 00:00.09 | Daejeo | Katty: meow :) |
| 00:00.10 | [TK]D-Fender | jasonwoot: You have not exorcised everything then |
| 00:00.14 | jasonwoot | but something must be looking at it then, no? with the 'remote unix...' |
| 00:00.27 | drmessano | jasonwoot: I gave you the answer.. its the web interface |
| 00:00.30 | [TK]D-Fender | jasonwoot: What would be "yes" |
| 00:00.36 | drmessano | Its checking for status |
| 00:01.37 | drmessano | Someone needs to write an AMI 1.0 to 1.1 proxy |
| 00:02.01 | drmessano | Effing app devs not updating plugins that use AMI |
| 00:02.49 | drmessano | Better yet legacyamiport= |
| 00:02.50 | drmessano | done |
| 00:03.32 | drmessano | bindlegacyami= ftw |
| 00:04.01 | [TK]D-Fender | justfingguesswhatimdoing=yes |
| 00:04.16 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 00:08.52 | drmessano | The Asterisk integration plugin for Openfire hasnt been updated for 1.6 so it's horribly broken with the AMI changes |
| 00:09.06 | drmessano | I blame the dev completely, but the users end up suffering |
| 00:10.31 | Akiyuki | ~seen CARLOS_PHB |
| 00:10.35 | jbot | i haven't seen 'carlos_phb', Akiyuki |
| 00:11.13 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 00:11.34 | Akiyuki | ~seen CARLOS_PHX |
| 00:11.35 | jbot | carlos_phx <n=Carlos@ip68-3-162-244.ph.ph.cox.net> was last seen on IRC in channel #asterisk, 9h 25m 19s ago, saying: 'Brrr...had to fire up the heater for the first time this year. Got up and it was down to 72 in the house. The humanity.'. |
| 00:14.02 | Daejeo | ~seen [TK]D-Fender |
| 00:14.03 | jbot | [tk]d-fender is currently on #asterisk (2h 17m 16s). Has said a total of 60 messages. Is idling for 10m 2s, last said: 'justfingguesswhatimdoing=yes'. |
| 00:14.31 | Akiyuki | :D |
| 00:14.36 | Daejeo | ~seen Katty |
| 00:14.37 | jbot | katty is currently on #asterisk, last said: 'tzanger: in a perfect world, that would work.'. |
| 00:14.44 | Akiyuki | Is it possible to tell an IP phone to ring, without passing it through asterisk? |
| 00:14.50 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 00:15.20 | [TK]D-Fender | Akiyuki: Yes. Call it from something else |
| 00:15.24 | Akiyuki | :P |
| 00:15.48 | Akiyuki | I mean, say I have a telephone that connects to a local asterisk server. Can another device on the network call that phone directly without hitting the * aserver? |
| 00:16.03 | [TK]D-Fender | Akiyuki: thats what I just said |
| 00:16.31 | Akiyuki | Like what? |
| 00:16.41 | Akiyuki | Can I just telnet or ssh to the phone and tell it to ring? |
| 00:16.42 | [TK]D-Fender | Akiyuki: like anything |
| 00:16.55 | [TK]D-Fender | Akiyuki: Take aonther SIP device and call your target directly |
| 00:17.31 | Akiyuki | Would the other SIP device have to routh through a PBX? or could it go SIP->SIP or SIP->MGCP without a PBX? |
| 00:17.40 | *** join/#asterisk JayTee52 (n=jforde05@unaffiliated/jaytee) |
| 00:17.46 | [TK]D-Fender | Akiyuki: ... EVERYTHING is SIP |
| 00:17.52 | [TK]D-Fender | Akiyuki: just call it DIRECTLY |
| 00:18.03 | Daejeo | [TK]D-Fender: what do you eat in the morning? |
| 00:18.05 | Akiyuki | foo@192.168.1.x? |
| 00:18.14 | [TK]D-Fender | Daejeo: KITTENS |
| 00:18.20 | [TK]D-Fender | Akiyuki: Yes. |
| 00:18.32 | Akiyuki | What about an MGCP telephone? Same thing/ |
| 00:19.04 | [TK]D-Fender | Akiyuki: MGCP is a very dumb protocol and intended for client-server. SIP has evrything as a client |
| 00:19.10 | Daejeo | you are never tired answering anything . does mark Spencer cook for you ? |
| 00:19.25 | [TK]D-Fender | Daejeo: Oh, I tire alright... |
| 00:19.41 | x86 | why did the deaf blonde sit on the newspaper? |
| 00:19.42 | Akiyuki | These phones only do MGCP |
| 00:20.11 | [TK]D-Fender | Akiyuki: Then why did you start, asking about SIP? |
| 00:20.39 | x86 | so she could lip read ;) |
| 00:20.45 | Akiyuki | rimshots |
| 00:20.46 | [TK]D-Fender | Akiyuki: Know anything about cooking? GOOD. So... how do I fix an automatic transmission on my car?!?!?! |
| 00:20.55 | Akiyuki | [TK]D-Fender: :> |
| 00:21.24 | Akiyuki | Is making a call to an MGCP phone the same ? |
| 00:21.50 | [TK]D-Fender | Akiyuki: Not to my knowledge. as I said, MGCP is client/server. SIP is P2P |
| 00:22.00 | [TK]D-Fender | (by comparison) |
| 00:22.19 | Daejeo | Akiyuki: Japanese? |
| 00:22.29 | Akiyuki | Daejeo: No :( |
| 00:22.35 | *** join/#asterisk fiXXXerMet (n=kyle@cmu-24-35-53-185.mivlmd.cablespeed.com) |
| 00:22.35 | Akiyuki | Daejeo: Spitalian |
| 00:22.43 | [TK]D-Fender | Akiyuki: So why are you stuck with MGCP exactly? |
| 00:23.04 | Akiyuki | We bought ~100+ MGCP phones for our current provider and are moving. |
| 00:23.07 | Akiyuki | moving providers |
| 00:24.01 | [TK]D-Fender | Akiyuki: Those were Polycoms, weren't they? |
| 00:24.09 | Akiyuki | Nortel 6812 |
| 00:24.14 | [TK]D-Fender | Oh yeah... WORSE |
| 00:24.20 | [TK]D-Fender | Akiyuki: BRILLIANT choice.. |
| 00:24.33 | Akiyuki | It wasn't my choice. It was a ~cheap issue by the boss :) |
| 00:25.09 | [TK]D-Fender | Akiyuki: How much each? |
| 00:25.26 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 00:25.26 | *** mode/#asterisk [+o denon] by ChanServ |
| 00:25.41 | Akiyuki | ~100 |
| 00:26.20 | [TK]D-Fender | Akiyuki: You got ass-raped... and no KY even. |
| 00:26.23 | *** part/#asterisk lukeb (n=outkast@office.telifon.com) |
| 00:26.57 | [TK]D-Fender | Akiyuki: So much for "cheap". a GOOD phone would have cost you less |
| 00:27.03 | Akiyuki | Like I said, it wasn't my call. I wasnt even involved in the decision. |
| 00:27.29 | Akiyuki | We have unlimited calls and unlimited long distnace for $24.00 through this voip provider, per user. |
| 00:28.33 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 00:28.50 | [TK]D-Fender | Akiyuki: How much could any 1 user cost you in the first place? |
| 00:29.03 | [TK]D-Fender | Akiyuki: thats actually pretty horrific |
| 00:29.29 | [TK]D-Fender | Akiyuki: Because thats what it would cost residentially for the same, except you wouldn't have to pay per user. |
| 00:29.45 | [TK]D-Fender | Akiyuki: What % of uyour users are on the phone at any given time? |
| 00:30.01 | [TK]D-Fender | Akiyuki: because you'll see the math really doesn't add up in your favor |
| 00:30.25 | *** join/#asterisk talntid (n=eric@66.208.251.170) |
| 00:30.51 | [TK]D-Fender | Akiyuki: How many minutes a month? Concurrent calls? |
| 00:30.58 | [TK]D-Fender | Akiyuki: etc... |
| 00:31.37 | Akiyuki | a half million minutes per month |
| 00:31.47 | Akiyuki | about 70% of the users or more are on the phone at any one time. |
| 00:31.51 | [TK]D-Fender | Akiyuki: Across how many users? |
| 00:31.55 | Akiyuki | 100 |
| 00:32.54 | Akiyuki | Was looking for an unlimited calls, unlimited simultaneous calls that I could use to build an auto dial feature/predictive dialer, for a couple hundred bucks a month or less, and i could have that call our current system back |
| 00:33.22 | [TK]D-Fender | Akiyuki: thats a really high concurrency rate working HUGE shifts ao the phone all the time... |
| 00:33.47 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 00:34.01 | [TK]D-Fender | Akiyuki: to rack up that much |
| 00:34.24 | Akiyuki | I know :/ |
| 00:34.30 | [TK]D-Fender | Akiyuki: hrm |
| 00:34.34 | Akiyuki | It's almost 24 hrs |
| 00:34.57 | [TK]D-Fender | Akiyuki: Well, maybe this IS a good deall... the most psycho usage I've seen for the to let it go like that |
| 00:35.19 | Akiyuki | I dont want to replace our current system. Just get a SIP trunk that I can build an auto dialer on, and then it connects back to our local #s |
| 00:35.59 | [TK]D-Fender | Akiyuki: Well you claim to average out .005$/min... I guess you really can't complain at all with that, now can you? |
| 00:36.12 | Akiyuki | well, no |
| 00:36.27 | Akiyuki | But they dont allow auto dialers, or predictive dialers, or SIP accounts w/ the plan we have |
| 00:36.31 | [TK]D-Fender | Akiyuki: You win. Freak-case of the year. |
| 00:36.37 | [TK]D-Fender | :) |
| 00:37.06 | Akiyuki | hehe |
| 00:37.38 | Akiyuki | So , since we are stuck w/ mgcp phones, I was just going to get 1 sip account some where that can allow us to have 100 simultaneous calls and unlimited calling per month |
| 00:37.46 | Akiyuki | What do you think that will run? |
| 00:42.18 | *** join/#asterisk neurosys (n=neurosys@adsl-153-223-41.mia.bellsouth.net) |
| 00:44.14 | beek | [TK]D-Fender: If I may bug you for another question... I double-checked the /etc/dahdi/system.conf that was generated by the sangoma utility. No wonder I was getting the occasional PRI issue! I had no master source. I've corrected that. I have my channel bank using the T1 for a clock reference -- will the A104D put that clock on all ports not marked as a master? |
| 00:45.11 | [TK]D-Fender | Akiyuki: I don't even want to think about it... |
| 00:45.58 | [TK]D-Fender | beek: No, "Master" means acting as a timing source. "Normal" means TAKING timing |
| 00:46.26 | beek | [TK]D-Fender: So I need to allow the Adit 600 channel bank to use it's own internal clock? |
| 00:47.43 | beek | [TK]D-Fender: http://www.pastebin.ca/1266265 . Span 1 is to the telco, Span 2 is to the PBX and Span 3 is to the Adit |
| 00:49.52 | *** join/#asterisk lucasb (n=outkast@s154-5-252-231.bc.hsia.telus.net) |
| 00:50.54 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 00:51.04 | *** join/#asterisk rhombus (n=rhombus@dsl-vlan435-66-18-218-36.nucleus.com) |
| 00:51.40 | Akiyuki | grr |
| 00:51.43 | Akiyuki | just called vonage |
| 00:52.23 | [TK]D-Fender | beek: your ADIT should be ":MASTER |
| 00:53.48 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 00:53.56 | beek | [TK]D-Fender: So I'll turn its internal clock back on. Would span=3,2,0,est,b8zs be appropriate? |
| 00:54.50 | [TK]D-Fender | beek: 3,0,0 |
| 00:56.10 | beek | [TK]D-Fender: Done. I hadn't made that change yet. I hadn't looked at this file once I configured it with the sangoma tool and noted the "do not hand edit" at the top. ;-) |
| 00:56.28 | [TK]D-Fender | bekkBAH |
| 00:56.39 | rhombus | How can I get hold music to play continuously, instead of starting a new track for every call? |
| 00:57.34 | Akiyuki | Vonage offers $50 unlimited simultaneous outbound calls, unlimited calls. |
| 00:57.35 | beek | [TK]D-Fender: Thanks again for your help. This project is coming along nicely. |
| 00:57.50 | beek | (and I'm learning a metric shitload in the process) |
| 00:58.02 | [TK]D-Fender | beek: Metric... good |
| 00:58.40 | *** join/#asterisk SQLDarkly (n=dakendri@192.147.57.6) |
| 01:00.13 | *** join/#asterisk cvnet (n=dahitler@74.210.103.241) |
| 01:00.26 | cvnet | back |
| 01:00.40 | cvnet | how do you setup callerID name (not the number) in asterisk ? |
| 01:03.45 | SQLDarkly | "name" <number> |
| 01:05.12 | [TK]D-Fender | callerid="joe" <> |
| 01:05.59 | *** join/#asterisk Docfxit (n=Docfxit@netblock-68-183-215-195.dslextreme.com) |
| 01:07.10 | Docfxit | Does anyone know if the GUI works with Dahdi? |
| 01:09.37 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-9dd6dc0939fb80aa) |
| 01:10.29 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 01:19.26 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
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| 01:28.25 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:29.41 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:31.04 | *** join/#asterisk DrAkOSX (n=ljd@nelug/coreteam/luisjose) |
| 01:32.46 | UnixDawg | ok what is the properline for sox for converting wav files for asterisk ? |
| 01:32.57 | FruitBasket | Hey, I got a line in my log that says, "Spawn extension (macro-dialexten, dial, 3) exited non-zero on 'SIP/provider-nv-1dc85270'". That suggests that the call was disconnected on the provider's end, right? I e-mailed them and asked, they said they received a BYE from my asterisk box... help? |
| 01:33.08 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 01:34.45 | [TK]D-Fender | FruitBasket: means nothing. Go look at SIP debug |
| 01:35.06 | StephenF | sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql |
| 01:35.12 | FruitBasket | fender: I can't. I don't have SIP debug on all my calls.. it mostly makes it hard to read the dialplan. |
| 01:35.26 | StephenF | UnixDawg ^^^ |
| 01:35.31 | FruitBasket | I turned it on Thurs,Fri,Sat and.. there were _no_ problems. I come in today and they've been having tons :-/ |
| 01:35.37 | StephenF | thats what I use |
| 01:35.48 | FruitBasket | is sip debug generally good to just have on, to trace the random call? |
| 01:36.35 | FruitBasket | has found WavePad to do a far better job of wave -> ulaw/gsm/other than sox |
| 01:37.55 | [TK]D-Fender | FruitBasket: if they said you hung up, I'd take it at that for now and look at what happened on your end |
| 01:38.14 | *** part/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 01:38.16 | FruitBasket | fender: already did. We lost two calls at the same time from two providers. |
| 01:38.35 | FruitBasket | The only thing I can say is that it said exited non-zero in SIP/remoteprovider-id |
| 01:38.50 | FruitBasket | I have no idea why, but now I know it wasn't the provider.. |
| 01:39.24 | FruitBasket | losses like this are pretty hit and miss, so I'd have to log all sip to get any info. |
| 01:40.00 | FruitBasket | yet again... I'm _really_ stumped. |
| 01:41.58 | Akiyuki | What costs more? Inbound or outbound calls? |
| 01:42.06 | FruitBasket | depends on provider. |
| 01:44.38 | *** join/#asterisk plasmid (n=noway@c-71-225-10-10.hsd1.pa.comcast.net) |
| 01:47.26 | UnixDawg | thanks |
| 01:47.35 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 01:47.51 | Akiyuki | [TK]D-Fender: Correction, I just looked at the billing, we are doing 2,520,000 minutes per month |
| 01:48.29 | [TK]D-Fender | Akiyuki: more than hard to beat at $25/channel 8 100 = $2500/mo |
| 01:49.12 | Akiyuki | I called packet 8, and the guy laughed me off the phone when I told him what I wanted to do :) |
| 01:49.46 | [TK]D-Fender | Akiyuki: You seem to have a killer deal.. I don't think I'd want to fuck that up if I were you... |
| 01:50.09 | [TK]D-Fender | Akiyuki: As I said, "you win"... your situation is extremely hard to imagine being more cost-effective on this basis |
| 01:50.17 | [TK]D-Fender | Akiyuki: Unless its a staffing question... |
| 01:50.20 | UnixDawg | i am not getting any audio |
| 01:50.22 | [TK]D-Fender | Akiyuki: then again.... |
| 01:50.51 | Akiyuki | Well, im not trying to replace them.. just find a supplement so i can build an auto dialer |
| 01:50.56 | *** join/#asterisk chendy (n=chatzill@59.40.223.115) |
| 01:51.59 | [TK]D-Fender | Akiyuki: They seem good as a provider. It sounds like you need a SIP> MGCP gateway in between |
| 01:52.09 | [TK]D-Fender | Akiyuki: that sounds quite doable |
| 01:52.10 | UnixDawg | I converted the file but no audio when I play it |
| 01:52.29 | Akiyuki | [TK]D-Fender: Do you know of a SIP -> MGCP gateway? |
| 01:52.39 | Akiyuki | Because we can just purchase another "seat" and fake it :) |
| 01:52.45 | [TK]D-Fender | Akiyuki: Not off-hand |
| 01:53.35 | Akiyuki | Everytime I google for it, I come up empty handed |
| 01:55.07 | [TK]D-Fender | Akiyuki: Check out FreeSWITCH. Maybe they support acting as a phone. |
| 01:55.19 | Akiyuki | Is that a channel? |
| 01:58.08 | *** join/#asterisk andresmujica (n=andresmu@190.25.101.78) |
| 02:00.13 | *** join/#asterisk ipguy (n=dkavadas@129.94.190.121) |
| 02:01.43 | *** part/#asterisk ipguy (n=dkavadas@129.94.190.121) |
| 02:05.44 | *** part/#asterisk fiXXXerMet (n=kyle@cmu-24-35-53-185.mivlmd.cablespeed.com) |
| 02:18.00 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 02:19.27 | *** join/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 02:20.22 | Titanous | Has anyone seen "Peer audio RTP is at port 0.0.0.0:3001" (no ip) with remote phones? |
| 02:20.37 | Titanous | I get it when the remote phone tries to initiate the call |
| 02:20.51 | Titanous | no RTP audio is passed in either direction |
| 02:21.07 | [TK]D-Fender | Titanous: remote phones should almost never be allowed to try to reinvite |
| 02:22.27 | Titanous | [TK]D-Fender: the phone has canreinvite: no set |
| 02:22.59 | Titanous | is there anything else I can do to fix it? |
| 02:23.07 | Titanous | it works when the asterisk side initates the call |
| 02:25.27 | *** join/#asterisk sasargen_ (n=chatzill@68-244-175-239.pools.spcsdns.net) |
| 02:26.00 | [TK]D-Fender | Titanous: You can start by showing us the SIP debug for the call and your configs. pastebin is your friend |
| 02:26.02 | [TK]D-Fender | ~pb |
| 02:26.02 | jbot | [~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/ |
| 02:30.20 | *** join/#asterisk SQLDarkly (n=dakendri@192.147.57.6) |
| 02:31.02 | SQLDarkly | What does chan_sip.c:4512 sip_alloc: Unable to create RTP audio session: Too many open files |
| 02:31.14 | SQLDarkly | I am load testing right now |
| 02:31.31 | SQLDarkly | I have 200 users in 1 conference |
| 02:31.37 | SQLDarkly | all streaming moh |
| 02:31.50 | SQLDarkly | no more calls are allowed in so I am assuming this to be my max |
| 02:32.15 | Titanous | [TK]D-Fender: http://pastebin.ca/1266339 |
| 02:32.20 | SQLDarkly | Would streaming MoH from another server expand the number of participants in a given conference |
| 02:33.12 | [TK]D-Fender | Titanous: Ok, I'm not working on masked IP's, when IP's are the problem. |
| 02:34.49 | Titanous | [TK]D-Fender: k, http://pastebin.ca/1266341 |
| 02:34.57 | Titanous | oops |
| 02:35.05 | Titanous | wrong paste, one sec |
| 02:35.56 | *** join/#asterisk nikko (n=nikko@173-17-214-107.client.mchsi.com) |
| 02:36.21 | Titanous | [TK]D-Fender: sorry about that, http://pastebin.ca/1266342 |
| 02:37.52 | [TK]D-Fender | Titanous: You've still masked all the read sources, and not provided your configs either |
| 02:38.00 | [TK]D-Fender | (removed outright) |
| 02:42.06 | Titanous | [TK]D-Fender: http://pastebin.ca/1266347 http://pastebin.ca/1266346 |
| 02:42.24 | Titanous | need any other configs? |
| 02:45.22 | [TK]D-Fender | Titanous: Waht about this other phone, for which we don't get to see the aCTUAL DIAL... |
| 02:46.13 | Titanous | [TK]D-Fender: the dial is handled by Adhearsion, is that releveant? |
| 02:47.11 | [TK]D-Fender | Titanous: Well so far the only thing I've heard is "failure". Any leg can be responsible. I'd seriously PROVE which is working independant of the other |
| 02:48.04 | [TK]D-Fender | Titanous: And is * itself behind NAT? |
| 02:48.13 | Titanous | [TK]D-Fender: same results when the remote phone calls any extension/trunk on the system (including VM, etc) no audio in either direction |
| 02:48.25 | [TK]D-Fender | Titanous: What router are they behind? |
| 02:48.37 | Titanous | [TK]D-Fender: yes, with all configured ports forwarded, Tomato on WRT54G |
| 02:48.42 | [TK]D-Fender | Titanous: And have any special settings been taken into account? |
| 02:48.57 | Titanous | [TK]D-Fender: no issues with incoming calls from trunks, etc |
| 02:49.05 | Titanous | [TK]D-Fender: special settings? |
| 02:49.25 | [TK]D-Fender | Titanous: Clarify, A ) * behind NAT? B ) What router is the phone heind. C ) Any forwarding on remote side? |
| 02:49.41 | [TK]D-Fender | behind* |
| 02:50.51 | Titanous | [TK]D-Fender: A) yes, B) Buffalo router with DD-WRT, C) yes, all traffic from the asterisk IP on all prots/protocols |
| 02:51.00 | Titanous | *ports |
| 02:51.10 | *** join/#asterisk jc_yyz2bkk (n=jc@ppp-58-8-64-183.revip2.asianet.co.th) |
| 02:51.26 | jc_yyz2bkk | anyone using bash for the agi scripts? |
| 02:51.32 | [TK]D-Fender | Titanous: C ) = the PHONE side. *'s router's ports SHOULD be forwarded.. on the REMOTE side (the phone's), they SHOULDN'T |
| 02:51.54 | [TK]D-Fender | jc_yyz2bkk: Probably someone... |
| 02:53.17 | jc_yyz2bkk | i just started with it, im using Sunny Woos stdout reader... which it looks like most people are, even if they dont know it, and i was just wondering how i get the results from the WAIT FOR DIGIT... |
| 02:53.28 | Titanous | [TK]D-Fender: all traffic FROM the Asterisk IP to the remote phone is forwarded directly for troubleshooting purposes, all ports (RTP, SIP, IAX2) are forwarded on the asterisk side |
| 02:53.58 | [TK]D-Fender | Titanous: Never forward on the remote side and I have heard of DD-WRT being an issue for things like this... attempt a swap |
| 03:10.11 | jc_yyz2bkk | when agi scripting, and reading the stdout... if i use WAIT FOR DIGIT, it should give me more than just a '200' |
| 03:30.45 | troy- | what command should i use to play music while an extension is ringing? |
| 03:31.28 | Docfxit | Any ideas when the GUI will work with Dahdi? |
| 03:32.53 | [TK]D-Fender | troy-: "core show application dial" |
| 03:33.22 | troy- | [TK]D-Fender, already got it, much thanks |
| 03:33.51 | *** join/#asterisk jameswf (n=james@ip68-3-60-66.ph.ph.cox.net) |
| 03:47.46 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
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| 03:51.10 | Daejeo | troy: are you the guy from a movie "TROY" |
| 03:51.14 | Daejeo | ? |
| 03:52.48 | drmessano | I love "Tron".. that was the best documentary ever |
| 03:54.39 | goobsoft | Well, I figured out my problem. The firewall in front of asterisk was configured improperly. I thought ports 10000-60000 was open, but it was not. I needed to specify the port range with a colon not a dash. The web-interface, didn't report that issue. Just thought I'd report back. I'm going to suggest changes to OpenWRT. Thanks the help. |
| 03:57.19 | *** join/#asterisk robba (n=robert@mail.ampwest.com.au) |
| 03:57.22 | drmessano | wow |
| 03:57.26 | drmessano | 10000-60000? |
| 03:57.58 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 03:58.05 | robba | does anyone know what causes -- Got SIP response 489 "Bad event" |
| 03:58.09 | goobsoft | Yes, Link2Voip requires 10000:60000 <- see, I can learn :) |
| 03:58.37 | drmessano | Thats 20000 concurrent calls worth |
| 03:59.08 | goobsoft | I guess they are doing well... |
| 03:59.14 | drmessano | Um |
| 03:59.31 | drmessano | Nom like, what YOU would need for 20000 simulaneous calls |
| 03:59.36 | drmessano | That makes no sense |
| 03:59.51 | drmessano | The RTP ports are defined for YOUR asterisk box anyway |
| 04:00.00 | drmessano | So you only need what is defined in RTP.conf |
| 04:02.05 | goobsoft | Hmm, you might be right. I'll look into that. |
| 04:02.07 | drmessano | If anything, they may require allowing OUTBOUND traffic from your router in the 20000-60000 range, as some routers/admins would do so |
| 04:02.20 | drmessano | For most this would not be a problem |
| 04:02.52 | drmessano | But as far as open ports on your end, you can use as little as say 10000-10500 so long as its defined in rtp.conf |
| 04:03.37 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 04:04.18 | goobsoft | I see. I just need to make changes when I have a lot of time to test. |
| 04:04.56 | goobsoft | I spend all day just finding the first problem. |
| 04:05.20 | drmessano | What do you have set in rtp.conf? |
| 04:05.22 | goobsoft | But I do appreciate your suggestion and what your saying sounds right to me. |
| 04:05.35 | goobsoft | the default 10000 to 20000 |
| 04:06.03 | drmessano | So right now you have 20000 extra ports open, possibly exposing other daemons/services to the net for no reason |
| 04:06.48 | drmessano | sorry, my math was off.. 40000 extra ports |
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| 04:07.23 | goobsoft | true |
| 04:07.26 | pyite | clear |
| 04:07.41 | drmessano | Yes, so you need to seriously consider fixing that ASAP |
| 04:08.31 | Daejeo | ~seen [TK]D-Fender |
| 04:08.33 | jbot | [tk]d-fender is currently on #asterisk (6h 11m 46s). Has said a total of 131 messages. Is idling for 35m 40s, last said: 'troy-: "core show application dial"'. |
| 04:09.06 | goobsoft | well it's openwrt, it's not running anything except asterisk. When I run netstat -nl, it shows that nothing else is listening. Wouldn't any packet to those other ports just be dropped unless I installed some other software? |
| 04:12.41 | drmessano | Any apps you run behind it that try to map high numbered ports through the NAT stand a good chance of failing miserably, and if you're sure theres nothing else running, then go ahead and leave those 40000 additional ports open |
| 04:12.46 | drmessano | Its your router |
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| 04:15.29 | [TK]D-Fender | As long as RTP.conf matches, what the hell... |
| 04:16.31 | goobsoft | You make good points. Thanks again for all of the help. Good night. |
| 04:16.35 | *** part/#asterisk goobsoft (n=chad@cpe-66-25-84-137.satx.res.rr.com) |
| 04:16.44 | Juggie | rtp.conf port config would be the ports you receive rtp packets on |
| 04:16.48 | Juggie | not send |
| 04:17.11 | [TK]D-Fender | Juggie: Yes, and we're talking about inbound port vulnerability possibilities |
| 04:17.20 | Juggie | i dont see why link2voip or anyone else would care what ports your local end was configured to use |
| 04:17.34 | Juggie | so long as their firewall allows them to send out to those ports. |
| 04:17.40 | Juggie | it should not matter the range |
| 04:17.57 | Juggie | from their perspective the dst port is irrelevant |
| 04:18.02 | Juggie | its the source port that matters |
| 04:18.20 | Juggie | as thats where you have to send your RTP back to |
| 04:18.51 | Juggie | as for opening UDP ports |
| 04:18.55 | Juggie | thats not a big security hole |
| 04:19.05 | drmessano | They probably mentioned to allow outbound connections to 10000:60000 for users like n00b softphone users with personal firewalls that bitch about every outbound.. (Vista, etc) |
| 04:19.09 | Juggie | but you can do a couple of things to solve that, one of which would be to do port triggering |
| 04:19.28 | Juggie | eg, when a sucessful connection is setup on 5060 then allow rtp ports. |
| 04:19.35 | drmessano | Which became "I need to forward 10000-60000 to asterisk" |
| 04:19.40 | drmessano | Which is stupid |
| 04:19.48 | Juggie | of course, even if your box replies login incorrect |
| 04:19.54 | Juggie | that will be viewed as a good connection |
| 04:20.16 | Juggie | so that wont be much help to someone who knows to get access to the udp ports, you need to send a sip packet |
| 04:20.19 | Juggie | but it would help |
| 04:20.47 | Juggie | you could also take it a step further and setup a trigger w/ asterisk some way based on the registration |
| 04:20.55 | Juggie | or using openser, etc |
| 04:20.58 | Juggie | lots of solutions :) |
| 04:21.31 | drmessano | Thats a lot of work for that 30% that cant even open whats defined in a config file |
| 04:22.14 | drmessano | 1000-2000, 10000-11000, or 10000-60000, but never 10000-20000 |
| 04:22.20 | drmessano | Why oh why is that so hard |
| 04:23.10 | drmessano | I left off 0-0, or better known as "I am using SIP, not RTP, so I didnt need those open" |
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| 04:51.57 | drmessano | Anyone here using CarrieXchange? |
| 04:54.22 | *** part/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
| 04:56.59 | *** part/#asterisk Docfxit (n=Docfxit@netblock-68-183-215-195.dslextreme.com) |
| 04:58.03 | [TK]D-Fender | drmessano: Nah... got burned by them a while back... |
| 04:58.25 | drmessano | How so? |
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| 05:05.17 | [TK]D-Fender | drmessano: .... |
| 05:05.23 | [TK]D-Fender | drmessano: Work with here... |
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| 05:21.18 | JayTee52 | mmmmm, dirty pillows |
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| 05:44.48 | luke-jr | why can't one go from Line In/Out to diodes to a RJ11? |
| 05:44.56 | joelsolanki | WARNING[6334]: chan_sip.c:3036 sip_call: No audio format found to offer |
| 05:45.07 | joelsolanki | i am using asterisk 1.4.22 |
| 05:45.13 | joelsolanki | and getting error at cli WARNING[6334]: chan_sip.c:3036 sip_call: No audio format found to offer |
| 05:45.18 | joelsolanki | is this codec issue ? |
| 05:45.30 | jql | smells like a codec issue |
| 05:45.38 | joelsolanki | hmm. |
| 05:45.59 | joelsolanki | i have 2 asterisk |
| 05:46.11 | joelsolanki | eyebeam --> box1 --> box2 |
| 05:46.47 | joelsolanki | eyebeam uses g711 --> box1 has g711 --> box2 is also asterisk accepting g729/g711 so it should do transcoding |
| 05:47.03 | joelsolanki | can this work ? |
| 05:47.06 | jql | with sufficient debug levels, I suspect you'd see messages complaining about the failure to match up codecs |
| 05:47.21 | joelsolanki | or will also need g729 licenses installed at box1 ? |
| 05:47.35 | joelsolanki | i want to have transcoding to be done at box2 level |
| 05:48.14 | joelsolanki | jql: possible ? |
| 05:48.50 | luke-jr | let me rephrase: I have a cable with a RJ11 on one end, and two 3.5mm audio plugs on the other; what good is it? |
| 05:50.43 | jql | much is possible |
| 05:50.59 | jql | g729 can work in passthru mode |
| 05:51.07 | jql | just have both box 1 and box 2 accept g729 |
| 05:51.30 | jql | box 1 will happily pass through the g729 without attempting to dsp it |
| 05:52.16 | joelsolanki | ok |
| 05:52.22 | joelsolanki | but what i want is like this |
| 05:52.53 | joelsolanki | eyebeam sends box1 g711 and box1 passes box2 g711 only but box2 transcodes to g729 and send to voip provider |
| 05:52.56 | joelsolanki | this is what i need |
| 05:53.04 | jql | that's fine, too |
| 05:53.12 | joelsolanki | so how can i do that ? |
| 05:53.18 | joelsolanki | let me give you my current config |
| 05:53.26 | jql | pastebin it, yeah |
| 06:00.01 | joelsolanki | jql: 1 min |
| 06:00.45 | *** join/#asterisk itiliti (n=itiliti@75.150.198.1) |
| 06:01.00 | joelsolanki | http://www.pastebin.ca/1266439 |
| 06:01.04 | joelsolanki | plz check |
| 06:02.10 | itiliti | can anyone help me with a CDR issue? |
| 06:03.38 | joelsolanki | jql: what do you think ? |
| 06:03.57 | jql | did I miss something? |
| 06:04.20 | joelsolanki | means ? |
| 06:04.25 | joelsolanki | did you check the pastebin ? |
| 06:04.26 | joelsolanki | http://www.pastebin.ca/1266439 |
| 06:04.56 | jql | on box1, disallow g729 |
| 06:05.12 | joelsolanki | oklet me do it |
| 06:05.41 | jql | phone ----[ulaw]---> box1 ----[ulaw]---> box2 ---[g729]---> pstn right? |
| 06:05.58 | jql | if so, take my suggestion |
| 06:06.14 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 06:06.18 | joelsolanki | yes correct |
| 06:06.20 | joelsolanki | tell me plz |
| 06:07.16 | jql | g729 should only be 'allow'ed on the [peer] for your provider |
| 06:07.40 | jql | if you intend to transcode on box2 |
| 06:07.52 | joelsolanki | hmm l |
| 06:07.55 | joelsolanki | let me try |
| 06:08.15 | joelsolanki | disabled g729 on box2 |
| 06:08.30 | joelsolanki | and only g711 is allowed on box1 is that ok ? |
| 06:08.46 | joelsolanki | same results |
| 06:09.02 | jql | which box gives the error? 1 or 2? |
| 06:09.51 | joelsolanki | box1 gives out error |
| 06:09.59 | joelsolanki | [Nov 25 00:02:53] WARNING[6391]: chan_sip.c:3036 sip_call: No audio format found to offer |
| 06:10.20 | jql | hrm. what's the peer config for your phone? it also needs to be ulaw |
| 06:10.36 | joelsolanki | ok let me try |
| 06:10.39 | jql | box1 has two peers -- one for the phone, and one for box2. both need to be ulaw |
| 06:11.02 | joelsolanki | i had allow all in phone config |
| 06:11.14 | joelsolanki | yes |
| 06:11.18 | joelsolanki | 1 sec |
| 06:11.33 | jql | while that technically should work ... |
| 06:11.55 | jql | codec negotiation is always a bitch |
| 06:12.30 | jql | also, it's not allow=g711, it's allow=ulaw |
| 06:12.35 | jql | I only just noticed that |
| 06:12.39 | jql | might be hurting you |
| 06:12.55 | joelsolanki | oh k |
| 06:12.58 | joelsolanki | let me put that |
| 06:13.01 | jql | change that everywhere |
| 06:15.39 | joelsolanki | cool. that worked :) |
| 06:15.54 | joelsolanki | thanks JQL :):) |
| 06:16.03 | jql | enjoy |
| 06:16.07 | joelsolanki | i wasnt aware of g711 and ulaw issue :) |
| 06:16.35 | [TK]D-Fender | Issue? |
| 06:16.35 | jql | it's that way for hysterical raisins |
| 06:16.41 | [TK]D-Fender | sees crazy people |
| 06:16.51 | joelsolanki | not issue i mean |
| 06:17.12 | joelsolanki | i kept g711 which didnt worked. then as per jql i kept ulaw and it worked. |
| 06:17.30 | joelsolanki | i thought g711 should work in same manner but i will remember this :) |
| 06:18.35 | joelsolanki | now working on extensions. having some problem. trying to figure. let me checkout |
| 06:20.19 | itiliti | I am trying to figure out the best way to write the called DID to the CDRDB when a call comes in, or ends. |
| 06:20.30 | itiliti | ANy ideas as to the best way to do it? |
| 06:20.39 | itiliti | I was thinkingof using the userfield |
| 06:21.13 | jql | umm... why is the called DID different that what's written by default into your CDR? |
| 06:22.47 | *** join/#asterisk Chris-NB (n=chris@85-126-61-10.work.xdsl-line.inode.at) |
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| 06:25.15 | itiliti | the called DID is not written to the CDRDB. |
| 06:25.29 | itiliti | At least not on SIP trunking |
| 06:25.34 | jql | what is written instead, in its place? |
| 06:26.11 | itiliti | s, or the destination that matches the inbound context, and the first destination. |
| 06:26.26 | drmessano | hysterical raisins? |
| 06:26.28 | drmessano | I love that |
| 06:26.33 | drmessano | I will be using that tomorrow |
| 06:26.54 | jql | heh |
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| 06:28.25 | itiliti | http://pastebin.com/m36c226e9 |
| 06:29.00 | itiliti | it never adds the DID that was called. I am trying to setup a trcking report that a client can show which DID's they are using for Marketing were called and whioch one werent. |
| 06:29.33 | jql | there's two phone-numbers, and neither of them is right? |
| 06:29.38 | jql | frowns |
| 06:29.43 | itiliti | So far I have this when the call is hung up: exten => s,1,Set(CDR(userfield)=${SIP_HEADER(To):16:10}) |
| 06:30.14 | itiliti | nope those numbers are the outbound CID. The DID is never wrtten to the CDRDB. I have been trying to get this figured out for about a month... |
| 06:30.58 | itiliti | The issue with my cide is that it counts in a number of characters from the sip header. it doesn work on all numbers. Especially outgoing ones, just incoming. It sort of works, but isnt very clean. |
| 06:31.16 | itiliti | is there a variable for the DID in Asterisk? |
| 06:31.49 | jql | theoretically, DNID |
| 06:32.47 | jql | in the pri world, I'm used to the DNIS |
| 06:33.34 | itiliti | I have looked into those too, just to see if there is a variable or something. The weird part is that Asterisk knows what it is, but I just cant seem to figure out how to write it to the damn CDR DB.. |
| 06:47.21 | Chris-NB | hi |
| 06:47.41 | Chris-NB | anyone using a snom phone with firmware v7 and additinal languages? |
| 06:48.11 | Chris-NB | I try to load German to my phone (via provisioning), but the phone tries to download my xml files and hangs |
| 06:48.59 | Chris-NB | display says snom 320 Checking Configuration.. |
| 06:49.09 | Chris-NB | but does nothing further ... |
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| 07:09.43 | BeeBuu | anyone tell me why the callee can't active the appliction by setting "testfeature => 88,peer/callee,Macro,calltest"? |
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| 07:27.07 | orkid | dtmf issues? |
| 07:28.24 | BeeBuu | orkid: i don't think so. cause "testfeature => 88,peer/caller,Macro,calltes" work! |
| 07:35.05 | cvnet | is there any customizable softphone out there? |
| 07:35.17 | cvnet | where you can label it with your own? |
| 07:35.43 | drmessano | Find one thats open source and get someone to hack at the code |
| 07:36.18 | cvnet | hum |
| 07:36.19 | cvnet | thanks |
| 07:36.34 | cvnet | im ok with coding if i get a open source i think i can do it myself, do you know any btw?> |
| 07:38.10 | drmessano | http://www.voip-info.org/wiki-Open+Source+VOIP+Software |
| 07:39.28 | cvnet | thanks |
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| 08:24.28 | SwK | anyone know a good IPv6 soft phone? |
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| 08:49.29 | HaMYaI | which is the proper sip's dtmf mode for asterisk 1.2 dtfmode,dtmf or dtmfmode? |
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| 09:12.09 | itiliti | dtmfmode=auto/inband/rfc2833 |
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| 09:31.24 | Nunners | hi all - could someone take a look at my log file for last night, and explain what they think might be happening. The result of the main bit (at 2am) is that we are unable to make/receive calls using Zap.... http://pastebin.com/m76088a23 |
| 09:35.00 | Nunners | anyone in? |
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| 09:41.48 | gr0mit | Nunners, yes |
| 09:42.36 | gr0mit | eew- its analoguey stuff |
| 09:42.45 | Nunners | unfortunately... |
| 09:43.08 | *** join/#asterisk h-idrisi (n=h-idrisi@212.100.196.195) |
| 09:43.26 | Nunners | would I be correct by saying it would seem the analogue line is bringing up an alarm/failure/polarity change, thus causing a problem? |
| 09:43.34 | gr0mit | has not touched analogue for years |
| 09:43.53 | gr0mit | this could be |
| 09:43.57 | *** part/#asterisk Strom_C (n=strom@netblock-208-127-61-171.dslextreme.com) |
| 09:44.05 | gr0mit | maybe BT running line tests at 2am? |
| 09:44.25 | Neo|Laptop | anything's possible with BT |
| 09:47.36 | Nunners | Would a zap module reset run every hour overnight possibly fix the problem? |
| 09:48.08 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 09:52.13 | Nunners | If so, is it possible to setup as a cronjob? |
| 09:57.05 | angryuser | who is using aastra 55i ? is there any way to see the call's received history ? ;) |
| 09:58.17 | angryuser | Nunners: to reset all modules you need to stop asterisk and unload zaptel as also all analog/digital driver's |
| 09:59.15 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 09:59.17 | angryuser | Nunners: i had the same problem, i had to roll back to 1.4.19/all and all become normal |
| 09:59.28 | itiliti | on the 55i, you just need to0 bind a button to the callers type. then you can see the last 200 calls you have gotten, you can also use the redial to see the last 100 you have called... |
| 10:03.40 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
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| 10:12.49 | Nunners | angryuser: Do you know what caused it? |
| 10:13.03 | angryuser | Nunners: nope |
| 10:13.41 | Superbartt | Is anyone familiar with Call Deflection? I currently am running Asterisk 1.4.21.2-BRIstuffed-0.4.0-RC3c but the call just gets dropped eventually using ZapCD() |
| 10:13.49 | angryuser | Nunners: all my analog zap's stop working after some time, so i rolled back |
| 10:13.53 | Nunners | someone has suggested on the forum that it used to be called SALT - the problem that is... an automatic line test being run |
| 10:14.48 | angryuser | Nunners: in my case 1.4.21 zaptew > zap offline 1.4.19 > normal , so why search a solution tell me ? |
| 10:18.50 | Chris-NB | anyone using language xml file for a snom-3X0 phone |
| 10:19.09 | Nunners | angryuser: Because I'm worried going back will not fix the problem if it's being caused by the line, and not asterisk. |
| 10:19.25 | Chris-NB | I try to change the language from the webserver, but the phone does not boot if the xml file is sent during provisioning |
| 10:19.26 | angryuser | Nunners: all you zap channel go offline ? |
| 10:19.34 | Chris-NB | anyone got this working? |
| 10:19.55 | Nunners | angryuser: I'm also not aware of any reported bugs with 1.4.21 to do with this, having checked the bugs lists! |
| 10:20.40 | Nunners | angryuser: It's probably the one thing I haven't checked yet, as don't get time when everyone starts coming in and wants to use their phone.... I have just been resetting it up to now... I plan to check it in the early hour of tomorrow morning to see what's happened.... |
| 10:20.57 | angryuser | Nunners: answer my question please |
| 10:21.16 | angryuser | Nunners: all you zap channels (analog) stop working ? |
| 10:23.47 | Nunners | angryuser: yes they all stop working... |
| 10:24.32 | Nunners | both fxs and fxo |
| 10:28.05 | angryuser | Nunners: do you want to pass tour time searching a problem or start usuing your system ? |
| 10:28.12 | angryuser | using* ? |
| 10:31.33 | Nunners | angryuser: I want to get it working reliably, but I haven't got an hour spare, during downtime to spend re-installing an older version... also are there not security consequences for going backwards rather than staying with the most up-to-date? |
| 10:33.59 | angryuser | Nunners: i you have no the time to get it work reliably, ask someone how does, as for the security you got the reason, but there are a lot of solution to limit the risk |
| 10:34.13 | angryuser | who* |
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| 11:01.57 | *** join/#asterisk farah (n=fauf@11-71-72-193.adsl.switzerland.net) |
| 11:02.24 | farah | hi all |
| 11:03.01 | farah | anyone confortable with the CLI command "iax2 show netstats"? |
| 11:04.09 | farah | anyone can help me plz? |
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| 11:10.17 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 11:14.39 | stmaher | Hello everyone. Im trying to use Asterisk AMI to generate an outbound call which connects to a meeting.. (for paging multiple phones) can you please take a look at this and tell me whats wrong? http://www.pastebin.ca/1266555 |
| 11:14.41 | stmaher | thanks |
| 11:16.00 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 11:17.06 | angryuser | stmaher: check your permissions in manager.conf |
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| 11:21.50 | stmaher | angryuser thanks.. Permissions are fine.. 10.0.0.0/255.0.0.0 |
| 11:22.29 | angryuser | stmaher: paste all your file please |
| 11:23.00 | angryuser | you need the Call permission enabled |
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| 11:24.53 | stmaher | angryuser http://www.pastebin.ca/1266556 |
| 11:25.38 | stmaher | angryuser thanks |
| 11:25.45 | stmaher | angryuser its pretty much a default asterisk 1.6 install |
| 11:26.27 | angryuser | stmaher: you are using [admin] ? |
| 11:26.36 | stmaher | angryuser yep |
| 11:26.57 | stmaher | angryuser http://www.pastebin.ca/1266555 |
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| 11:29.03 | angryuser | stmaher: strange, normally you got all the right's , are you sure about the originate syntax ? if peer/1234 exist as all-page context ? |
| 11:30.04 | stmaher | angryuser yes.. It doest.. I think Im missing originnate in the permissions of commands |
| 11:31.45 | angryuser | stmaher: originate - Permission to originate new calls. Write-only. yes |
| 11:33.19 | angryuser | that was my first guess but i admit i forgot that originate is a spare option ;) |
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| 11:51.22 | stmaher | angryuser thanks for your help.. the script isnt working itself but I now know the Manager is .. THanks again for your help! |
| 11:57.57 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 12:00.01 | nfi|ermes | when someone call me from outside (zap channel), my sip extensions can t see caller id |
| 12:00.09 | nfi|ermes | my zapata.conf : http://pastebin.com/m2b7ae607 |
| 12:00.52 | nfi|ermes | in my extensions.conf i try exten => s,5,NoOp(${CALLERID(num)}) |
| 12:01.25 | nfi|ermes | and the result is: -- Executing [s@from-pstn:5] NoOp("Zap/1-1", "") in new stack |
| 12:08.29 | *** join/#asterisk Nunners (n=james@mail.nadn.co.uk) |
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| 12:12.16 | Nunners | angryuser: I've solved the problem. My telecoms provider have enabled CPC signals and stopped the auto testing overnight. Speaking with someone outside of this group, they looked at the debug, and said that was definitely the problem. They couldn't understand why reverting to an older version would work..... thanks for you help anyway... |
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| 12:29.53 | Superbartt | Does anyone has any idea how call forwarding on an isdn line actually takes place? Is it something set in the telco, or something configured in the local equipment? |
| 12:33.14 | florz | Superbartt: either is possible |
| 12:34.03 | Superbartt | hmmfg ok, well my telco has a *21*XXX# number to call to do the forwarding but that doesn't seem to be working, So i'm looking at a solution to do it locally |
| 12:34.29 | Superbartt | in my asterisk (with bristuff) i have the zapCD function, but that eventually just disconnect the caller that should be forwarded |
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| 12:43.51 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 13:01.56 | *** join/#asterisk feeds (n=feeds@stip-static-181.213-81-187.telecom.sk) |
| 13:04.52 | nfi|ermes | hi florz |
| 13:09.43 | florz | hi |
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| 13:13.11 | E-bola | In manager.conf how can i specify to bind to 2 different ip addresses? |
| 13:13.22 | E-bola | the wiki on voip-info.org doesnt say |
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| 13:20.57 | feeds|School | Could someone help me please? http://asterisk.pastebin.com/f5299e332 |
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| 13:25.35 | donnib | i have two phones on a network (one subnet), a asterisk server (another subnet). when one phone calls the other phone on the same network and on the same subnet they cannot hear eachother. what is the reason ? |
| 13:25.47 | donnib | i am not running NAT |
| 13:26.33 | donnib | i am not sure how SIP calls work but they are not going thru the Asterisk server right ? they only use the server as the negotiator ? |
| 13:27.38 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:29.06 | donnib | anyone ? |
| 13:30.04 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 13:30.32 | feeds|School | donnib: Could you pastebin your extensions.conf please? |
| 13:30.43 | WimpMan | donnib: Depends. Mostly on sip.conf / canreinvite= |
| 13:31.45 | feeds|School | If youre callin exten@asteriskIP then you're calling thru asterisk, but when dialling exten@phone2IP then only through the local subnet... I think ;) |
| 13:31.55 | *** part/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 13:32.02 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:32.09 | feeds|School | hi [TK]D-Fender |
| 13:32.18 | feeds|School | Could someone help me please? http://asterisk.pastebin.com/f5299e332 |
| 13:34.47 | *** join/#asterisk unixdawg_ (n=unixdawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 13:35.03 | feeds|School | anyone? |
| 13:35.08 | unixdawg_ | morning |
| 13:35.26 | beek | unixdawg_: morning. |
| 13:35.29 | unixdawg_ | I am working on a asterisknow 1.5 box |
| 13:35.35 | unixdawg_ | but having a issue |
| 13:35.43 | unixdawg_ | sox main-greating-en-es.wav -r 8000 -c 1 -s -w new-main-greating-en-es.wav resample -ql |
| 13:35.51 | unixdawg_ | I do this |
| 13:36.13 | unixdawg_ | but then I upload the file threw the gui and then pick it in the ivr |
| 13:36.34 | unixdawg_ | but when you dial the ivr it says its playing it but no audio |
| 13:37.11 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 13:38.32 | feeds|School | unixdawg_: you mean no audio in sip client? |
| 13:38.43 | unixdawg_ | correct |
| 13:38.54 | unixdawg_ | I pick up my polycom dial the ivr |
| 13:38.57 | [TK]D-Fender | feeds|School: you do NOT use the same priority label repetitively in the same exten. it is a SPECIFIC point which let it know where to jump to. When you have 5 tags the same how will it know which one to go to? |
| 13:38.58 | unixdawg_ | no audio |
| 13:39.32 | feeds|School | [TK]D-Fender: No idea, wait a few minutes, have to go bye |
| 13:40.10 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.195.82) |
| 13:42.41 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 13:43.23 | [TK]D-Fender | And Dear God is that exten theory wrong... |
| 13:47.26 | [TK]D-Fender | unixdawg_: if others work in its place then your file format is bad or your source was bad |
| 13:48.14 | farah | anyone could help me please: when i run the CLI command "iax2 show netstats", on the remote side, the value are all 0..why? |
| 13:48.43 | [TK]D-Fender | farah: waht "remote side"? |
| 13:49.39 | mort_gib | TK: I found "some" posts indicating that other Users of the same card is having the same problems I have.... |
| 13:49.48 | mort_gib | Trying with another card now... |
| 13:50.08 | [TK]D-Fender | mort_gib: namely? |
| 13:50.20 | farah | [TK]D-Fender: the output of the command "iax2 show netstats" gives statistics for the local side and the remote side |
| 13:50.31 | mort_gib | ?? Card or problem?? |
| 13:50.41 | [TK]D-Fender | mort_gib: YES :p |
| 13:50.50 | mort_gib | LOL |
| 13:50.58 | mort_gib | I'll be trying a Digium card |
| 13:51.11 | [TK]D-Fender | farah: Because maybe * can't tell what the other side didn't get? |
| 13:51.21 | mort_gib | Others had intermitted dropped calls that looked like "normal hangup" |
| 13:52.25 | farah | [TK]D-Fender:for the local side, i get value for the packet lostm the jitter, and the number of packets, but for the remote side, i get values just for the jitter, and all the other values are 0 |
| 13:52.32 | [TK]D-Fender | mort_gib: Ah yes... well... remember that clearing code doesn't come from thin air... would make no sense for a card to throw that out for nothing |
| 13:52.52 | [TK]D-Fender | farmaybe its a placeholder and is simply not possible to know ATM |
| 13:52.59 | mort_gib | TK: but if the driver is dodgy... |
| 13:53.41 | [TK]D-Fender | mort_gib: Dodgy enough to invent a properly formatted HANGUP request? :) |
| 13:53.51 | mort_gib | I use the same setup for ALL my clients, but ONLY my Gibraltar clients, using the A500 card has these issues |
| 13:53.56 | [TK]D-Fender | mort_gib: this is grasping at straws... you know this, right? |
| 13:54.06 | *** join/#asterisk feeds (n=feeds@stip-static-181.213-81-187.telecom.sk) |
| 13:54.10 | [TK]D-Fender | mort_gib: But if thats all you've got, hey, go for it |
| 13:54.58 | mort_gib | Well... on the posts I found they DID state that this seemed to happen when another user would hang up. Marc from Sangoma seemed to think it was a driver issue |
| 13:55.15 | mort_gib | -And Yes, I AM grasping for straws :-) |
| 13:55.42 | mort_gib | I WOULD also get some experience with Digiums cards.... |
| 13:55.43 | [TK]D-Fender | feeds: Your use of labels was wrong, that exten check no "status" that we see the origin of (where's that var coming from?", there is no MoH in there, just a hangup, and your trailing hangup will never be called an is thus worthless |
| 13:55.52 | mort_gib | Nice to know, first hand, what is out there... |
| 13:56.02 | feeds|School | [TK]D-Fender: so what did you say? So I have to do something like : 123,n(busy),Goto(xyz,1) and create another exten with that what I put in the busys |
| 13:56.07 | feeds|School | ? |
| 13:56.24 | [TK]D-Fender | mort_gib: Sure thing... we don't have anything solid to go on either, so hey, if it doesn't hit the pocketbook much, gwhy not? |
| 13:56.43 | [TK]D-Fender | feeds|School: http://asterisk.pastebin.com/f5299e332 |
| 13:56.50 | [TK]D-Fender | feeds|School: Answer is above |
| 13:56.52 | mort_gib | In this case doing nothing will hit my pocket book much harder... |
| 13:57.16 | [TK]D-Fender | mort_gib: Reason enough... |
| 13:57.26 | [TK]D-Fender | mort_gib: Friggen BRI :p |
| 13:57.49 | mort_gib | Yes, as I mentioned I have two really large (for me) installations coming up, that won't be using BRI :-) |
| 13:58.00 | mort_gib | But those clients speak together so.... |
| 13:58.34 | coppice | how does a really large installation use BRI? :-\ |
| 13:58.50 | feeds|School | [TK]D-Fender: so something like this ? |
| 13:58.55 | feeds|School | wait a sec.. |
| 13:59.00 | mort_gib | coppice: They shouldn't |
| 13:59.14 | *** join/#asterisk JayTee52 (n=jforde@unaffiliated/jaytee) |
| 13:59.31 | mort_gib | coppice: Unless they are different department etc etc |
| 14:00.06 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:00.32 | mort_gib | coppice: and take "really large" installations with a pinch of salt, I'm a one man band... :-) |
| 14:01.11 | feeds|School | [TK]D-Fender: http://asterisk.pastebin.com/f7869dc3f |
| 14:03.32 | [TK]D-Fender | feeds|School: http://asterisk.pastebin.com/mc40f003 |
| 14:04.18 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 14:06.01 | feeds|School | http://asterisk.pastebin.com/f6620348b |
| 14:08.14 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:10.11 | [TK]D-Fender | feeds|School: http://asterisk.pastebin.com/m33cf65b3 |
| 14:11.13 | mort_gib | TK: Come on, he read the book :-) |
| 14:12.12 | *** join/#asterisk nikko (n=nikko@69.57.49.100) |
| 14:12.14 | [TK]D-Fender | feeds|School: What page can I see that on? |
| 14:12.32 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-aba30e4396cd3203) |
| 14:13.25 | feeds|School | [TK]D-Fender: I know you're right, but I just tried, and well... failed :( |
| 14:13.33 | feeds|School | wait I'll find it ;) |
| 14:14.24 | feeds|School | oops, that was DIALSTATUS not CALLSTATUS |
| 14:14.33 | feeds|School | nevermind, page 159 |
| 14:14.39 | [TK]D-Fender | feeds|School: and that doesn't check if they are on a call or not. |
| 14:15.00 | feeds|School | and what does it check? |
| 14:15.10 | ajohnson | The status of an attempted dial |
| 14:15.13 | ajohnson | you have to dial first |
| 14:16.34 | feeds|School | ajohnson: I know, but what are the statuses of Dial? | Please dont bet me up, I'll try core show application Dial , before you can write it ;) |
| 14:17.15 | ajohnson | Then I won't bother writing it :-D |
| 14:18.10 | feeds|School | :D |
| 14:18.43 | ajohnson | feeds|School: I assume you're trying to find out if someone is already on a call before you dial? |
| 14:18.51 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 14:18.54 | ajohnson | If so, you may want to check out group counting |
| 14:20.06 | feeds|School | ajohnson: Yes; WHat's group counting? | Don't say it, I know its in the wiki ;) |
| 14:20.19 | ajohnson | http://www.voip-info.org/wiki/view/Asterisk+cmd+SetGroup |
| 14:20.27 | ajohnson | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+GetGroupCount |
| 14:20.40 | feeds|School | [TK]D-Fender: is this better? http://asterisk.pastebin.com/fcef7392 |
| 14:20.43 | ajohnson | mmmm coffee time |
| 14:21.01 | *** join/#asterisk pyite (n=pyite@c-76-21-105-195.hsd1.ca.comcast.net) |
| 14:21.32 | ajohnson | feeds|School: You are not specifying a dialing timeout, which means if they are not there, the phone will continue to ring forever |
| 14:22.27 | mort_gib | ajohnson: Which means that the caller will die of old age waiting for the phone to be answered :-) |
| 14:22.56 | ajohnson | Yes, and we don't have a DIED_OLD_AGE call result |
| 14:23.00 | feeds|School | ajohnson: http://asterisk.pastebin.com/f48e1f568 |
| 14:23.23 | mort_gib | ajohnson: Damn, * is completely useless! |
| 14:23.32 | [TK]D-Fender | feeds|School: http://asterisk.pastebin.com/m2a68055 |
| 14:23.36 | ajohnson | feeds|School: Logically wrong, but it would work |
| 14:23.52 | ajohnson | well I mean... it wouldn't work the way you want it to work :) |
| 14:24.43 | ajohnson | you will have to switch to using group counting to do what you want to do |
| 14:25.25 | feeds|School | http://asterisk.pastebin.com/f5f314f89 |
| 14:25.35 | feeds|School | Ok gonna have a look on group counting |
| 14:25.50 | ajohnson | feeds|School: You can't check DIALSTATUS when you HAVEN'T DIALED |
| 14:26.21 | telnettech | anybody on here that can decipher SIP messages |
| 14:26.27 | feeds|School | omg, how then?, nvm having a look on grp counting |
| 14:27.34 | [TK]D-Fender | feeds|School: No, you're going to have READ THE INSTRUCTIONS I GAVE YOU |
| 14:28.31 | *** join/#asterisk IPkaf (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 14:28.36 | *** part/#asterisk IPkaf (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 14:29.02 | Katty | hi |
| 14:29.06 | *** join/#asterisk riotzy (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 14:29.18 | *** part/#asterisk feeds|School (n=feeds@stip-static-181.213-81-187.telecom.sk) |
| 14:29.27 | riotzy | hi everybody |
| 14:30.06 | riotzy | i work in cyber cafe |
| 14:30.20 | mort_gib | riotzy: Lucky for you :-) |
| 14:30.24 | SQLDarkly | Weights in DUNDi are supported in 1.4.22 correct? |
| 14:30.29 | Katty | sighs |
| 14:30.42 | Katty | there is a samsung dealer here this morning, to peddle his el cheapo phone system. |
| 14:30.45 | riotzy | right now i use a soft ifcash to control all my phone |
| 14:30.49 | anonymouz666 | mort_gib: I wouldn't say lucky :-) |
| 14:30.58 | mort_gib | Me neither.... |
| 14:30.59 | Katty | i'm going to have to sit through 3 hours of sales and marketing propoganda |
| 14:31.08 | Katty | why me )= |
| 14:31.10 | riotzy | i plan to change all my computer to linux |
| 14:31.20 | riotzy | and ifcash not working |
| 14:31.36 | mort_gib | Katty: Maybe he has stickers :-O |
| 14:31.50 | Katty | mort_gib: yeah. and maybe he can stick them where the sun don't...erm. |
| 14:31.52 | *** join/#asterisk panolex (n=olex@194.44.160.178) |
| 14:31.55 | Katty | breathes. |
| 14:32.11 | Katty | yes. maybe he has stickers. |
| 14:32.12 | mort_gib | Katty: lol, yeah, we all get those meetings... |
| 14:32.17 | riotzy | as my pabx is an asterisk is there any software for Linux OS |
| 14:32.24 | Katty | i don't want to install crap phone systems. |
| 14:32.30 | riotzy | to control all my phone call time ? |
| 14:32.31 | Katty | and have people bitching at me when they realize they're crap |
| 14:32.33 | *** join/#asterisk jeff (i=jeff@unaffiliated/jeff) |
| 14:32.53 | Katty | riotzy: that does not parse. please try agian. |
| 14:32.57 | Katty | s/agian/again/ |
| 14:33.28 | mort_gib | Katty: That's the beauty of it, see you make sure your "financial controller" endorses the system, and he is in for the "reward" when they realize it won't work |
| 14:33.45 | Katty | mort_gib: yeah. |
| 14:33.46 | riotzy | what ? |
| 14:33.59 | Katty | mort_gib: we're already a samsung dealer for copiers. so i guess they thought they wanted to do their phone systems too. |
| 14:34.10 | Katty | mort_gib: sadly, they've already made quotes to TWO places on this samsung system. |
| 14:34.19 | mort_gib | -Mind you. I predicted that a SAP implementation worth £1m would fail, which it did and the clown is still there |
| 14:34.21 | Katty | mort_gib: i have no idea if it's worth ANYTHING, much less how to mkae it work. |
| 14:34.46 | Katty | i really hate when they try to sell stuff we don't know how to do. it's going to bite them in the tail someday. |
| 14:34.51 | mort_gib | Katty: Recommend * to them! |
| 14:34.53 | Katty | and it's going to be my tail. not theirs. |
| 14:35.06 | Katty | mort_gib: the sales rep doesn't want to. she claims * is too expensive |
| 14:35.14 | Katty | mort_gib: and we're talking el cheapo server and sip trunks ONLY |
| 14:35.23 | Katty | mort_gib: they want cheaper than cheap. |
| 14:35.29 | Katty | mort_gib: i guess they want tin cans and string. |
| 14:35.45 | mort_gib | Which is exactly what they get with the Samsung system |
| 14:35.47 | telnettech | anybody no how to read SIP messages |
| 14:35.53 | Katty | mort_gib: we sell plenty of * boxes tho. |
| 14:36.03 | Katty | mort_gib: the company just wants something even cheaper :/ |
| 14:36.21 | Katty | did i mention i didn't want to take the heat for a crappy phone system? |
| 14:36.32 | mort_gib | Katty: So they don't even want to use a product they sell?? |
| 14:36.37 | [TK]D-Fender | riotzy: * IS that program, and its your job to configure it |
| 14:36.39 | kaldemar | telnettech: "sip set debug on" in CLI |
| 14:36.42 | Katty | mort_gib: not for this particular bid, no |
| 14:36.42 | mort_gib | Katty: In passing |
| 14:37.07 | Katty | mort_gib: asterisk is too expensive for this particular company |
| 14:37.15 | Katty | mort_gib: so our company quoted them something cheaper. |
| 14:37.18 | telnettech | no i got the debug on but i need to be able to decipher the messages to find out why I dont get a normal US ringing on internal calls |
| 14:37.24 | [TK]D-Fender | Katty: SAP... its more than a logisitcs package... its a FEELING |
| 14:37.27 | mort_gib | Katty: What size install?? |
| 14:37.27 | Katty | mort_gib: which i've no idea how to setup, or if it's even worth getting. |
| 14:37.34 | Katty | mort_gib: 4 phones |
| 14:37.38 | Katty | mort_gib: i think. |
| 14:37.49 | mort_gib | Katty: FXS?? |
| 14:37.49 | telnettech | i am very new to this asterisk stuff( less than 3 months) |
| 14:38.02 | Katty | mort_gib: i'm praying it's analog lines and IP phones |
| 14:38.07 | Katty | mort_gib: with my luck...tho... |
| 14:38.11 | Katty | sighs |
| 14:38.14 | Katty | let's not even go there. |
| 14:38.22 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 14:38.25 | Katty | [TK]D-Fender: well i have a VERY bad feeling about this. |
| 14:38.38 | Katty | [TK]D-Fender: worse than seeing your drunkle over thanksgiving feeling. |
| 14:38.40 | [TK]D-Fender | Katty: Sure thing han :) |
| 14:38.41 | mort_gib | Katty: Samsung would be some kind of digital phones.... |
| 14:38.49 | Katty | mort_gib: digital doesn't mean IP |
| 14:38.51 | *** part/#asterisk nikko (n=nikko@69.57.49.100) |
| 14:38.59 | Katty | mort_gib: it just means digital display on an analog phone :/ |
| 14:39.01 | kaldemar | telnettech: there's nothing to decipher, the messages are in text format. you need to learn SIP. |
| 14:39.03 | mort_gib | Katty: I know |
| 14:39.03 | [TK]D-Fender | Katty: Thats what he's saying |
| 14:39.05 | Katty | mort_gib: but i hope you're right. |
| 14:39.27 | [TK]D-Fender | Katty: Some stupid key system |
| 14:39.38 | [TK]D-Fender | Katty: a dead-end choice |
| 14:40.14 | mort_gib | Well, it's like the Avaya systems, where the phones are "provisioned" via an analogue line.... Ehm interresting idea |
| 14:40.43 | telnettech | I understand that but in the mean time, I need someone that does know SIP to assist in this issue. I am an old Telecom guy who has not had to learn this until I was thrown into it 3 months back. I am trying to get up to speed but need a little help |
| 14:41.05 | Katty | wibbles |
| 14:41.25 | Katty | [TK]D-Fender: i guess i could always threaten to quit if i installed one and people started bitching at me. |
| 14:41.35 | telnettech | here is the pastebin location: http://pastebin.com/d1d35001f |
| 14:41.53 | Katty | [TK]D-Fender: i actually had a dream our company went out of business, and the boss man was going to open a resturant. |
| 14:41.58 | Katty | [TK]D-Fender: the boss wanted me to be a waiter. :/ |
| 14:42.04 | mort_gib | Katty: Just refer them to Samsung... If Samsung provides it they can provide support too |
| 14:42.19 | Katty | mort_gib: our company doesn't work like that :< |
| 14:42.39 | Katty | mort_gib: the boss doesn't want them going to samsung. he wants them to have a contract with us for support. |
| 14:42.51 | Katty | mort_gib: which means yours truly has to deal with it. |
| 14:43.10 | mort_gib | Katty: Then you MIGHT want change, restaurant sounds nice :-P |
| 14:43.20 | Katty | screw that. |
| 14:43.22 | mort_gib | Katty: That sounds very familiar |
| 14:43.34 | *** join/#asterisk n3hxs (n=HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 14:43.35 | mort_gib | Uhm, not the screw part |
| 14:43.38 | Katty | i'm going to go work at a zoo. |
| 14:43.46 | Katty | maybe jaytee will get me hired. |
| 14:44.03 | Katty | i can work with a bunch of monkeys and asses |
| 14:44.10 | mort_gib | Katty: Yeah, I'm sure they don't use phones or computers for that matter... |
| 14:44.26 | mort_gib | Katty: By the sound of it you already do :-) |
| 14:44.28 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 14:44.32 | Katty | maybe the usa will have a civil war, and i can just help farm. |
| 14:44.38 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 14:44.58 | Katty | we'll be the new iceland. |
| 14:45.44 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 14:48.12 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:48.18 | coppice | Katty: iceland? you're thinking of the Day After Tomorrow, not civil war |
| 14:49.12 | Katty | coppice: well, granted, it's not civilw ar... but the people are definately protesting. |
| 14:49.56 | Katty | coppice: their krona lost half it's value, their government has bought its three bigest banks... |
| 14:50.19 | Katty | coppice: and they've also taken a nearly 5 billion dollar loan |
| 14:50.19 | coppice | for loose change, too :-) |
| 14:50.19 | [TK]D-Fender | remembers that Iceland is mostly green, and Greenland is mostly ice... |
| 14:50.32 | etfonhomey | Katty, how much does this Samsung thing sell for? |
| 14:50.39 | Katty | of course we've given way more than 5 billion here to bail out and loans and such. |
| 14:50.49 | Katty | etfonhomey: my guess is peanuts and cheeseburgers. |
| 14:51.27 | etfonhomey | Katty, For 4 phones you can get an * box in there for under $4K easily. |
| 14:51.33 | rwaite | i only want a million |
| 14:51.35 | Katty | etfonhomey: yeah. too expensive. |
| 14:51.49 | Katty | etfonhomey: think tin cans and string, and you've got a mental picture of my company |
| 14:51.58 | Katty | etfonhomey: i've seen a coworker run cat5 with a coat hanger. |
| 14:52.09 | Katty | etfonhomey: down a wall. with ELECTRICAL WIRING |
| 14:52.19 | etfonhomey | Katty, oh dear |
| 14:52.24 | riotzy | and my today's question is how to configure it ? |
| 14:52.37 | riotzy | how to do ? |
| 14:52.40 | mort_gib | Katty: You need to change your job asap |
| 14:52.40 | Katty | how to configure a coat hanger? |
| 14:52.43 | riotzy | that 's m y question ? |
| 14:52.49 | Katty | mort_gib: indeed. |
| 14:52.57 | rwaite | how to do? |
| 14:53.03 | riotzy | tk d-fender ? |
| 14:53.09 | etfonhomey | Katty, when I run into a client like you, I do not renew my contract. |
| 14:53.09 | Katty | to configure a coathanger, you bend it |
| 14:53.12 | Katty | to do [TK]D-Fender well... |
| 14:53.18 | Katty | uhh. |
| 14:53.20 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:53.21 | rwaite | steak? |
| 14:53.22 | Katty | i'm not going to answer that one. |
| 14:53.24 | Katty | hugs fskrotzki |
| 14:53.27 | riotzy | lol |
| 14:53.33 | riotzy | i ask hi m ? |
| 14:53.40 | riotzy | how to configure it ? |
| 14:53.42 | Katty | etfonhomey: we're not an end user. |
| 14:53.44 | fskrotzki | say's morning darling.. |
| 14:53.54 | Katty | etfonhomey: we use asterisk here, and sell asterisk. |
| 14:53.58 | [TK]D-Fender | looks around nervously... |
| 14:54.03 | Katty | etfonhomey: but the sale rep has gotten REALLY cheap. |
| 14:54.08 | riotzy | <PROTECTED> |
| 14:54.13 | Katty | etfonhomey: so she's looking for something even cheaper to sell. |
| 14:55.00 | etfonhomey | Katty, seems like your sales rep would like to keep clients. So, why sell s**t? |
| 14:55.07 | mort_gib | Katty: Well, technically you can just split one analogue line in 4 outlets... |
| 14:55.08 | Katty | etfonhomey: i'm not sure why she's doing it |
| 14:55.13 | Katty | etfonhomey: she's giving me heart burn |
| 14:55.15 | *** join/#asterisk lirakis (n=lirakis@65.200.189.231) |
| 14:55.19 | lirakis | hey all |
| 14:55.28 | Katty | etfonhomey: not to mention making me grumpy on a tuesday morning |
| 14:55.30 | all | hey |
| 14:55.36 | Katty | etfonhomey: and now I HAVE TO GO SIT THROUGH A SAMSUNG SALES MEETING |
| 14:55.42 | lirakis | i am having trouble with a background() command. It keeps handing off only a single digit from the extensions that the user enters |
| 14:55.42 | Katty | etfonhomey: did i mention i was grumpy? |
| 14:55.52 | *** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 14:56.27 | corr | * Cannot join #gentoo (You are banned). |
| 14:56.29 | corr | f-ers |
| 14:56.45 | Katty | is done doing her female bitch rant. |
| 14:57.02 | Katty | [TK]D-Fender: you can come out now. |
| 14:57.04 | etfonhomey | Katty, ha! |
| 14:58.22 | Guest14354 | grr |
| 14:58.44 | Katty | they really must not like you. |
| 14:58.56 | *** join/#asterisk jeffik (n=jeffik@69-196-165-181.dsl.teksavvy.com) |
| 14:59.30 | *** join/#asterisk nirs (n=nirs@212.235.43.194) |
| 14:59.54 | nirs | hi guys |
| 15:00.05 | nirs | I'm having the weirdest IAX2 problem ever |
| 15:00.09 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:00.16 | nirs | I've got 2 servers, configured identically |
| 15:00.26 | nirs | while one accepts IAX2 calls without a problem |
| 15:00.30 | nirs | the other does not |
| 15:00.33 | nirs | both are on the same LAN |
| 15:00.37 | nirs | same config files |
| 15:00.40 | nirs | same everything |
| 15:00.56 | lirakis | damn |
| 15:01.02 | lirakis | i had the stupid m flag set |
| 15:01.05 | lirakis | that would do it |
| 15:05.53 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 15:06.01 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 15:06.03 | unixdawg_ | sox main-greating-en-es.wav -r 8000 -c 1 -s -w new-main-greating-en-es.wav resample -ql |
| 15:06.43 | jaytee | I just got back to the puter and my Xchat icon was all bouncy. |
| 15:06.48 | jaytee | Hi Katty! |
| 15:06.58 | Katty | hihi |
| 15:07.13 | jaytee | your sleekgeek.org site's still down :-( |
| 15:09.02 | rwaite | so i finally figured out why xchat is being so slow |
| 15:09.08 | rwaite | my damn logs |
| 15:09.45 | stintel | :) |
| 15:09.45 | stintel | 25|16:10:49 autolog_path = ~/irclogs/%Y/%m/$tag/$0.log |
| 15:09.53 | stintel | but donno if that works for xchat |
| 15:10.10 | Katty | jaytee: yeah probably |
| 15:11.18 | etfonhomey | Totally off topic, but just got a support call that an end-user's computer started running terribly slow. I logged into it and the user was sending an email with a 530MB attachment. This company refuses to set email attachment and mailbox size limites. Sorry, I just had to tell someone. |
| 15:12.15 | stintel | people never understood that there is FTP to transfer files :) |
| 15:12.54 | etfonhomey | They HAVE a working FTP server setup with the folder shared on their desktop to copy the files on to and instructions they can send to their clients. |
| 15:12.59 | mort_gib | etfonhomey: Don't worry, I think we all understand. I have to deal with Artichitects |
| 15:14.19 | etfonhomey | They also have an account at a place like Web Cargo that they could use as well. Exchange 2003 SP2 has a size limit of 75GB. This company has one user with a 10GB mailbox. OK. Done venting. |
| 15:15.23 | etfonhomey | mort_gib, I was reading above. How big is your new install? |
| 15:15.37 | unixdawg_ | ok this bite |
| 15:15.37 | Katty | happy again. |
| 15:15.42 | mort_gib | One 110+ and one 90+ |
| 15:15.45 | unixdawg_ | Iam convertin 3 sound files |
| 15:15.50 | Katty | nothing that a cute video of a baby polar can't fix. |
| 15:16.05 | unixdawg_ | and no audion from asterisk when Idial the ivr |
| 15:16.13 | mort_gib | Prospects now, but pretty sure they will sign up.. |
| 15:16.46 | mort_gib | Not that big, I know, but I remain a one man band, despite my efforts to try an hire more hands |
| 15:16.53 | *** join/#asterisk funxion (n=funxion@63.214.236.169) |
| 15:17.13 | etfonhomey | mort_gib, what phones are you using? |
| 15:17.30 | mort_gib | etfonhomey: Snoms mostly |
| 15:17.46 | mort_gib | Nice handsets, users like the webinterface a lot |
| 15:18.05 | etfonhomey | mort_gib, where are you located? |
| 15:18.09 | mort_gib | And I can "lock" the setting I don't want them to fiddle with |
| 15:18.16 | mort_gib | Gibraltar/Spain |
| 15:18.20 | unixdawg_ | likes polycoms |
| 15:18.59 | etfonhomey | mort_gib, Guess I can't help you out until Ryan Air starts their 12 euro transatlantic flights. |
| 15:19.19 | mort_gib | unixdawg: Yes, they are nice but Snoms allows the users to "play" with their phones and that makes them more interested in "their new system" |
| 15:19.55 | mort_gib | etfonhomey: Well... We don't always need physical presence do we... |
| 15:21.04 | etfonhomey | mort_gib, not always except when physically installing all those phones. |
| 15:22.51 | *** join/#asterisk ZaVoid (n=zavoid@75.147.121.177) |
| 15:24.03 | telnettech | is there a better channel than this one for beginners? |
| 15:25.20 | [TK]D-Fender | telnettech: Nope. What is your actual question? |
| 15:26.17 | anonymouz666 | 22 active channels |
| 15:26.17 | anonymouz666 | 8 active calls |
| 15:26.26 | anonymouz666 | this is what ast_masquerade can do for you |
| 15:26.34 | anonymouz666 | 22 active channels for just ONLY 8 calls |
| 15:26.37 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 15:27.14 | [TK]D-Fender | anonymouz666: You I can have 100 channels & NO calls... its called "sitting in IVR Hell" |
| 15:27.38 | anonymouz666 | there's no IVR in this machine ;) |
| 15:28.09 | [TK]D-Fender | anonymouz666: Parking, Queues, plenty of other ways to keep channels from being bridged .... also 3-way calls lowering that count. |
| 15:28.51 | anonymouz666 | I would say atxfer has its part on it |
| 15:29.03 | telnettech | TKD: it just seems that this channel is for people that have been working on Asterisk for awhile and when someone that is a beginner comes on to ask questions, people feel the need to tell them what they need to read and not assist in the actual problem at that moment. I know I have to learn how to manuever around, how to read SIP, how to enter commands in Linux and all the other networking stuff that this is built on. I am just looking for some |
| 15:29.03 | telnettech | answers to the questions I hav now so that I can learn as I go. I am a telecom guy that worked for a RBOC, went into working on PBX systems and dont have the computer knowledge that some of the other people on here have. |
| 15:29.45 | jameswf | is a billionaire weeeeeeeeeeeeeee |
| 15:30.02 | [TK]D-Fender | telnettech: Don't get discouraged... we get all kinds here and some questions don't get answered very fast. there are varying levels of interest by all parties here all the time. |
| 15:30.11 | [TK]D-Fender | telnettech: So what can we help you with specifically? |
| 15:30.35 | telnettech | I know I have a steep learning curve to travel on. My company was bought by another larger company and that scared the guys that worked on these few systems we have out there. I have been thrown into the pit with a lion and am expected to be an expert but it is going to take time. |
| 15:30.57 | telnettech | TKD: http://pastebin.com/d1d35001f |
| 15:30.58 | etfonhomey | telnettech, the key to getting help in here is to ask specific questions. |
| 15:31.35 | telnettech | this is SIP messages between 2 devices that I need to be able to figure out where the message tells it what type of ringing to initiate |
| 15:31.42 | jameswf | telnettech: you should not take people in this room to seriously... most are jaded by the "hey I just installed asterisk now what" newbs, if you show your making a valid effort to learn on your own and are not overly annoying in the process most will warm up to you... |
| 15:31.46 | anonymouz666 | [TK]D-Fender: one atxfer makes 4 active channels for just one active call. that's the math. |
| 15:31.58 | [TK]D-Fender | anonymouz666: :) |
| 15:32.19 | [TK]D-Fender | telnettech: What do you hear currently? |
| 15:32.23 | etfonhomey | telnettech, jameswf is exactly right. |
| 15:32.57 | jameswf | ~nowwhat |
| 15:32.58 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 15:32.59 | jameswf | heh |
| 15:33.01 | telnettech | TKD: The ringing for internal calls is not US it is like a European ring (Double Ring) |
| 15:33.26 | jameswf | telnettech: what is your tonezone and loadzone... |
| 15:33.42 | telnettech | jameswf: where do i find that out |
| 15:33.49 | telnettech | sip.conf? |
| 15:39.21 | jeff | grumbles at callfiles |
| 15:39.41 | jeff | okay, if anyone's up for a challenge... |
| 15:39.56 | jeff | i'm setting up something very much based off of the example here: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out+deliver+message |
| 15:40.41 | jeff | the problem is that when i'm dropping callfiles into /var/spool/asterisk/outgoing, asterisk (as documented) tries to place the calls immediately. |
| 15:40.57 | jeff | (or based on the timestamp of the file, either way...) |
| 15:41.32 | jeff | my problem is that i'm using SIP trunking to make the outgoing calls, and i have call-limit=1 on the sip trunk in sip.conf |
| 15:41.54 | jeff | so i drop four callfiles in the outgoing dir, asterisk starts to dial the first and then fails the other three. |
| 15:41.57 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:42.50 | jeff | the fix seems to be only have one callfile in the outgoing dir at a time, since i only have the ability to make one call at a time. |
| 15:43.02 | jeff | is this the only way, or am i missing something simpler? |
| 15:43.51 | jeff | and if i need to keep one file and only one file in that dir, is there an existing script that can do that for me? watch /var/spool/asterisk/outgoing and dump one callfile in at a time if the dir is empty? |
| 15:44.25 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 15:49.58 | etfonhomey | telnettech, did you find where those were? |
| 15:50.59 | neurosys | wow. telnet, that was the old tymnet rival back in the dialup days. no? |
| 15:51.10 | neurosys | Or that was Telenet. |
| 15:51.19 | telnettech | ET: no i didnt. It appears to be only in a zaptel file but we are not using zaptel as these are copper truks thru a mediatrix 1204 which talks to the Asterisk with SIP |
| 15:51.20 | neurosys | :) |
| 15:51.34 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 15:53.09 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 15:54.08 | telnettech | neuro: no you are correct, it is telnet |
| 15:54.28 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 15:54.40 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-a0e7c8ab0a7ed58c) |
| 15:54.40 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:55.07 | neurosys | telnettech: Actually it was Telenet. Sprint private WAN. |
| 15:55.38 | neurosys | telnettech: But this is going back to pre 91' |
| 15:55.41 | telnettech | correct, sprint had a product named Telenet for the private WAN service |
| 15:55.51 | jaytee | I could never hear the pin drop |
| 15:55.55 | harry_v | heheh |
| 15:56.02 | neurosys | heheh |
| 15:56.05 | harry_v | I remember that commerical |
| 15:56.39 | telnettech | anyways ET, where else would i find the tonezone and loadzone info |
| 15:56.41 | etfonhomey | telnettech, what kind of phones are you using? |
| 15:56.57 | jaytee | I remember cigarette commercials for Lucky Strike and Tareytons |
| 15:57.02 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 15:57.07 | telnettech | ET: Grandstream GXP 2000 |
| 15:57.09 | etfonhomey | telnettech, in polycoms (and probably most others), you can change the ring sound. |
| 15:57.30 | telnettech | yeah I checked that and they are set to system ring tone |
| 15:57.42 | telnettech | that is under the account settings |
| 15:57.50 | harry_v | I was wondering if anyone may have experainced * getting stuck half when it was loading? |
| 15:57.58 | etfonhomey | telnettech, maybe try changing them to a different ring just to see if it has an affect. |
| 15:58.08 | telnettech | it reads " F1=440, F2=480, C=200/400" |
| 15:58.08 | harry_v | half way when loading. I really suspect its just old hardware. |
| 15:58.34 | [TK]D-Fender | telnettech: User-Agent: MxSipApp/5.0.23.153 MxSF/v3.2.2.4 <-- I'm wondering what the native indication zone is for this UA. |
| 15:59.06 | [TK]D-Fender | etfonhomey: We are not talking about the station ring-tone, we are talking call progress signalling |
| 15:59.10 | [TK]D-Fender | etfonhomey: to the CALLER |
| 15:59.23 | telnettech | TK: correct |
| 15:59.42 | etfonhomey | etfonhomey, oh, so the caller is hearing a different ring tone after they dial the number? |
| 15:59.57 | telnettech | the caller hears the double ring but the phone rings normally |
| 16:00.30 | telnettech | TK: how would I find this out, the native indication zone |
| 16:00.52 | etfonhomey | telnttech, gotcha. Could it be some config on your gateway? |
| 16:01.46 | harry_v | TK, ever test the RN device for SIP redirection? |
| 16:02.09 | [TK]D-Fender | telnettech: When the caller gets it you need o loko at the UA generating ring. Yuo ccan have INBAND ringing (generated byt he bridging device, usually *), or OOB at which point the near-side gateway is responsible. I'd look at that MxSIP device if I were you |
| 16:02.22 | [TK]D-Fender | harry_v: RN? |
| 16:03.09 | harry_v | RN Appliance. bit dated article. http://searchunifiedcommunications.techtarget.com/news/article/0,289142,sid186_gci1187287,00.html |
| 16:03.37 | harry_v | seems it will switch currently running sip connections from a failed server to a backup |
| 16:04.06 | [TK]D-Fender | harry_v: Never touched HA or redundancy myself |
| 16:04.44 | harry_v | Intereting article never less. |
| 16:05.12 | [TK]D-Fender | harry_v: I'm sure it is. |
| 16:06.17 | harry_v | interesting, ranch networks is the company that owns the product but thay are out of business. |
| 16:06.51 | SuPrSluG | if i call a wrong internal number. my blf stays on after hangup on the call. calling a correct number works perfectly. |
| 16:06.56 | harry_v | Funding pulled and let everyone go. |
| 16:07.15 | harry_v | Sounds as if thay were in the red. To bad. |
| 16:07.17 | SuPrSluG | any ideas? |
| 16:07.27 | telnettech | TK: thanks for your help |
| 16:07.41 | [TK]D-Fender | telnettech: Have you found something in there? |
| 16:07.50 | telnettech | im looking now |
| 16:07.57 | [TK]D-Fender | SuPrSluG: Some details would be nice... |
| 16:09.13 | etfonhomey | telnettech, what did you say the settings on your Mediatrix GW were? |
| 16:09.32 | harry_v | Anyway, have appointment and need to go. Before I do, TK, have you seen a case where asterisk would hang stall, and the continue some time after a set period of time? went to use the phone and no DT. Checked console and pushed a key on he keyboard then rest of dialplan ect loaded. |
| 16:10.03 | harry_v | Ive seen this in 1.2 but now this is 1.4. |
| 16:10.14 | SuPrSluG | sorry. polycom 670 using asterisk and opensips. |
| 16:10.17 | harry_v | I suspect some how it could be hardware related. |
| 16:10.58 | [TK]D-Fender | SuPrSluG: When its ALL nice & neat in a pastebin I'll take a look at it... |
| 16:11.12 | unixdawg_ | ok bbiab have to figure out why sox is not converting files correctly |
| 16:11.28 | SuPrSluG | k |
| 16:14.19 | unixdawg_ | TK need input |
| 16:14.47 | unixdawg_ | I am converting 3 audio files |
| 16:15.09 | unixdawg_ | like this |
| 16:15.13 | unixdawg_ | sox main-greating-en-es.wav -r 8000 -c 1 -s -w new-main-greating-en-es.wav resample -ql |
| 16:15.24 | unixdawg_ | and then uploading them to a asterisk server |
| 16:15.35 | unixdawg_ | I have a ivr that points to them |
| 16:15.57 | unixdawg_ | but when I idal the ivr after I do a restart of aaterisk I get no audio |
| 16:17.34 | *** join/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 16:17.45 | nfi|ermes | [13:00] <nfi|ermes> when someone call me from outside (zap channel), my sip extensions can t see caller id |
| 16:17.46 | nfi|ermes | [13:00] <nfi|ermes> my zapata.conf : http://pastebin.com/m2b7ae607 |
| 16:17.46 | nfi|ermes | [13:00] <nfi|ermes> in my extensions.conf i try exten => s,5,NoOp(${CALLERID(num)}) |
| 16:17.46 | nfi|ermes | [13:01] <nfi|ermes> and the result is: -- Executing [s@from-pstn:5] NoOp("Zap/1-1", "") in new stack |
| 16:18.07 | *** join/#asterisk nikko (n=nikko@69.57.49.100) |
| 16:18.09 | mikealeonetti | on a Cisco phone, if I don't enable the NAT, even though "nat" is enabled for the client on the server, will it still work? |
| 16:18.20 | etfonhomey | telnettech, I'm sure you've already done this, but have you checked the "Custom Tone" section on your Medatrix? |
| 16:18.22 | telnettech | ET: Under the voiceIfDtmfTransportTable of the mediatrix SNMP settings, I have OOB Using RTP, Payload type is 101 and the enforce default is enabled |
| 16:18.24 | mikealeonetti | my Cisco phone won't connect out of the office and I'm trying to troubleshoot why |
| 16:18.51 | mikealeonetti | it gets an IP, correct DNS, and the proxy is correct |
| 16:19.08 | [TK]D-Fender | unixdawg_: Go prove that its the file that is bad |
| 16:19.17 | telnettech | ET: this is per our vendor tech assistance to get DTMF to pass over to the Telco for IVR menus |
| 16:19.32 | [TK]D-Fender | unixdawg_: And you should already know I ahve no faith in the initial good condition of the files you're trying to convert I hope.... |
| 16:19.43 | unixdawg_ | well I have other audio files that I converted that 2 months ago that work fine |
| 16:20.06 | [TK]D-Fender | unixdawg_: Changes nothing... |
| 16:20.37 | unixdawg_ | I can play them fine on the laptop with xmms |
| 16:20.45 | unixdawg_ | and the sound correct |
| 16:20.48 | telnettech | ET: yeah the tones are set for NOrth America 1 |
| 16:21.02 | [TK]D-Fender | nfi|ermes: You're missing "callerid=asreceived" <- |
| 16:21.30 | etfonhomey | telnettech, and under the Custom Tone section, you're not overriding anything? |
| 16:22.05 | telnettech | no all options on my country customization table are disabled |
| 16:22.05 | unixdawg_ | ok no audio |
| 16:22.57 | mikealeonetti | maybe it's a router setting |
| 16:22.58 | mikealeonetti | hopefully |
| 16:23.03 | etfonhomey | telnettech, have you tried North American 2 just in case? |
| 16:23.35 | telnettech | no i havent changed this at all |
| 16:23.36 | telnettech | this is the default setting |
| 16:24.41 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 16:27.39 | etfonhomey | etfonhomey, is it possible for you to just set it to bypass * and ring directly to a phone when a call on a certain FXO port is made? |
| 16:27.50 | etfonhomey | oops |
| 16:27.57 | *** part/#asterisk farah (n=fauf@11-71-72-193.adsl.switzerland.net) |
| 16:28.34 | etfonhomey | telnettech, ^^^^^ and I mean "when a call is received" |
| 16:29.03 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 16:33.10 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.149) |
| 16:38.10 | *** join/#asterisk Johnsie (n=jdlewis@c-24-23-118-1.hsd1.pa.comcast.net) |
| 16:38.30 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 16:39.12 | telnettech | ET: changing to northamerica2 did nothing to change the ringing |
| 16:44.05 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:47.24 | *** join/#asterisk Johnsie (n=jdlewis@c-24-23-118-1.hsd1.pa.comcast.net) |
| 16:47.53 | *** part/#asterisk Johnsie (n=jdlewis@c-24-23-118-1.hsd1.pa.comcast.net) |
| 16:50.08 | mikealeonetti | is there anything in Linksys routers that would prevent a phoen from regiwtering? |
| 16:51.51 | [TK]D-Fender | ~sipnat |
| 16:51.52 | jbot | [~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 |
| 16:51.54 | [TK]D-Fender | mikealeonetti: ^^^ |
| 16:52.09 | mikealeonetti | [TK]D-Fender: thanks |
| 16:56.49 | mikealeonetti | [TK]D-Fender: all those are set, and I do have other phones working outside with the same configuration. This phone TRIES to register but fails |
| 16:57.04 | [TK]D-Fender | mikealeonetti: PASTEBIN is your friend |
| 16:57.31 | mikealeonetti | sure |
| 16:57.35 | mikealeonetti | I can paste the Cisco logs |
| 16:57.43 | [TK]D-Fender | mikealeonetti: No./ |
| 16:57.51 | mikealeonetti | heh |
| 16:57.56 | [TK]D-Fender | mikealeonetti: Don't care what Cisco thinks... I care what ASTERISK thinks |
| 16:58.04 | mikealeonetti | [TK]D-Fender: Asterisk doesn't see it |
| 16:58.27 | [TK]D-Fender | mikealeonetti: if traffic never gets to *, then check your firewalls |
| 16:58.48 | mikealeonetti | [TK]D-Fender: firewall allows everything UDP on 5060 and from 1024:64000 |
| 16:59.02 | [TK]D-Fender | mikealeonetti: then check your cisco side. |
| 16:59.09 | [TK]D-Fender | mikealeonetti: Maybe its never getting out. |
| 16:59.22 | [TK]D-Fender | mikealeonetti: And test with a soft-phone on that side |
| 16:59.33 | mikealeonetti | [TK]D-Fender: what's a good windows soft phone? |
| 17:00.32 | mikealeonetti | quick and easy install |
| 17:02.33 | [TK]D-Fender | mikealeonetti: Any |
| 17:03.20 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 17:04.45 | mikealeonetti | I'll get Ekiga |
| 17:05.02 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 17:07.11 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:08.39 | mikealeonetti | [TK]D-Fender: well, it registers |
| 17:10.07 | *** join/#asterisk bijit (n=benji@200.122.158.243) |
| 17:10.59 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 17:11.03 | *** join/#asterisk hi365_m (n=hi365@213.151.43.64) |
| 17:11.47 | [TK]D-Fender | mikealeonetti: then fix your Cisco |
| 17:12.12 | unixdawg_ | ok back |
| 17:12.17 | unixdawg_ | this day is killing me |
| 17:12.35 | mikealeonetti | [TK]D-Fender: too bad it's only configurable by TFTP. Maybe I can set her up with a soft phone if I find one that dials out. |
| 17:15.50 | mikealeonetti | let's try this |
| 17:15.58 | mikealeonetti | I do hate screwing up |
| 17:16.10 | *** join/#asterisk |Torg| (n=mdm@ppp-70-251-227-151.dsl.rcsntx.swbell.net) |
| 17:16.38 | unixdawg_ | ciscos do ftp |
| 17:16.50 | unixdawg_ | and http as far as I know |
| 17:16.56 | mikealeonetti | unixdawg_: how can I reconfig through it? |
| 17:17.08 | unixdawg_ | threw the phone lcd screen |
| 17:17.23 | mikealeonetti | it's locked |
| 17:17.31 | unixdawg_ | you should have a configuration or menu button that allows you into the setup |
| 17:17.36 | unixdawg_ | thats a issue |
| 17:17.50 | unixdawg_ | you then have to hard wipe the phone and unlock it |
| 17:18.08 | unixdawg_ | never done it but I know it can be done |
| 17:18.13 | unixdawg_ | why is it locked |
| 17:18.24 | mikealeonetti | 'cause it was last configured through TFTP |
| 17:18.32 | mikealeonetti | it's probably something stupid like I forgot to turn on the NAT option |
| 17:18.33 | mikealeonetti | great |
| 17:19.01 | *** part/#asterisk unixdawg_ (n=unixdawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 17:19.02 | |Torg| | can someone give me some help with a x100p fxo that is always offhook? |
| 17:20.17 | *** join/#asterisk unixdawg_ (n=unixdawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 17:20.29 | unixdawg_ | man I hate the net today |
| 17:20.57 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 17:22.48 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:24.28 | *** join/#asterisk xieliwei (i=dcff041d@gateway/web/ajax/mibbit.com/x-f608f9f5c7436149) |
| 17:24.53 | xieliwei | Still having problems with chan_mobile, although a big step closer |
| 17:25.19 | xieliwei | I can get a few mobiles to pair, but it seems that sms does not work for all of them |
| 17:25.53 | xieliwei | I have tried a Nokia E51 which I think should be fully supported, but mobile show devices still reports it as No |
| 17:26.51 | Qwell | xieliwei: the Nokia bluetooth stuff is very poor |
| 17:26.58 | Qwell | very very non-conformant |
| 17:27.15 | xieliwei | hmm, how about a motorola V3? |
| 17:27.27 | xieliwei | also tried a HTC 3600 with winmo 6.1 |
| 17:27.40 | Qwell | xieliwei: My moto works pretty well. Similar to the V3 |
| 17:27.45 | Qwell | (Razr, right?) |
| 17:27.45 | xieliwei | all shows up as phone but sms is disabled |
| 17:27.48 | xieliwei | yeah |
| 17:27.57 | Qwell | v195 - sameish firmware. works fairly well |
| 17:28.00 | xieliwei | i have a problem pairing the razr actually |
| 17:28.10 | xieliwei | it keeps telling me invalid pin |
| 17:28.12 | Qwell | it was tricky pairing it iirc. umm |
| 17:28.21 | xieliwei | how did you do it? |
| 17:28.29 | Qwell | I think I had to unpair it from the phone, then let Asterisk connect to it |
| 17:28.43 | Qwell | lemme see |
| 17:28.46 | xieliwei | yeah, it keeps asking me when asterisk tries to connect |
| 17:29.06 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:29.42 | Qwell | Settings>Connection>Bluetooth Link>Device History>highlight your server>Delete |
| 17:30.01 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 17:30.10 | xieliwei | hmm, not even bonded in the first place, the list is empty |
| 17:30.11 | Qwell | then I think the next time it tried to connect, it asked for pin and worked. |
| 17:30.14 | Qwell | O.o |
| 17:30.40 | Qwell | under Device History? |
| 17:30.42 | xieliwei | hmm, yeah, it tried to connect |
| 17:30.42 | Qwell | not Handsfree |
| 17:30.43 | xieliwei | yeah |
| 17:30.47 | xieliwei | yup |
| 17:30.57 | xieliwei | i have a bad feeling its my dongle again |
| 17:30.58 | Qwell | does a scan from Handsfree see the server? |
| 17:31.02 | xieliwei | nope |
| 17:31.05 | Qwell | funky |
| 17:31.07 | SkramX | I keep getting -- Remote UNIX connection\n -- Remote UNIX connection disconnected in my Asterisk console |
| 17:31.15 | SkramX | any ideas? |
| 17:31.19 | Qwell | SkramX: Something is connecting to the console. |
| 17:31.25 | xieliwei | SkramX, are you using freepbx? |
| 17:31.28 | SkramX | right... |
| 17:31.30 | SkramX | and no |
| 17:31.37 | SkramX | Asterisk-GUI is installed but I'm not using it |
| 17:31.43 | xieliwei | hmm |
| 17:32.16 | SkramX | yeah |
| 17:32.30 | xieliwei | Qwell: you know of any way for my moto to successfully bond to my server? |
| 17:33.16 | mikealeonetti | man |
| 17:33.20 | mikealeonetti | I wish I sdidn't screw that up |
| 17:33.23 | mikealeonetti | I take things to personally |
| 17:34.06 | Qwell | xieliwei: iirc, those were the steps I took.. it was a pain the first couple times |
| 17:34.15 | Qwell | I'd have to set everything up and try it again |
| 17:34.25 | xieliwei | hmm, but the moto should work properly right? with sms? |
| 17:34.36 | Qwell | mine does, and it's very similar to yours |
| 17:34.46 | xieliwei | okay, so there's a chance there |
| 17:34.59 | Qwell | I'm pretty sure the V3 was specifically tested |
| 17:34.59 | xieliwei | i guess i'll go get another dongle tomorrow, this would be the fourth |
| 17:35.10 | |Torg| | can someone give me some help with a x100p fxo that is always offhook? |
| 17:35.21 | Qwell | xieliwei: My jabra has worked pretty well. |
| 17:35.29 | xieliwei | weird thing with the dongle is that my config in hcid.conf does not seem to work |
| 17:35.44 | Qwell | Jabra A320s |
| 17:35.53 | xieliwei | hmm, I'll see if i can get that off ebay |
| 17:36.03 | Qwell | I picked mine up in like Staples or something, heh |
| 17:36.11 | Qwell | little more expensive, but...it's served me well |
| 17:36.34 | Qwell | the range is nice too. With this and the V3 firmware (assuming it has the same bluetooth chip as mine) can go about 100 yards at max |
| 17:36.44 | xieliwei | ohh, that's pretty far |
| 17:36.50 | Qwell | yards? feet? I forget. it's good though |
| 17:37.03 | xieliwei | mine becomes undiscoverable in the adjacent room |
| 17:37.04 | Qwell | I can get to my mailbox down my long driveway, and keep a conversation going |
| 17:38.19 | xieliwei | great, no local sellers |
| 17:38.25 | [TK]D-Fender | SkramX: You've clearly install some monitoring sofware that is still doing lookups. |
| 17:39.47 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 17:40.09 | SkramX | other than tcpdump, how do I find out who the culprit i? |
| 17:40.30 | Qwell | the connection is local. tcpdump won't help |
| 17:40.34 | SkramX | okay |
| 17:40.40 | *** join/#asterisk sp00k3y (n=sp00k3y@wsip-98-190-136-194.ph.ph.cox.net) |
| 17:40.51 | SkramX | well. |
| 17:41.22 | xieliwei | would a SMC-BT10 work well? anyone? |
| 17:42.50 | *** join/#asterisk xieliwei (i=dcff041d@gateway/web/ajax/mibbit.com/x-7ea88a2726c805ce) |
| 17:42.56 | xieliwei | whoops |
| 17:43.03 | xieliwei | would a SMC-BT10 work well? anyone? |
| 17:45.00 | *** join/#asterisk ACK-NAK (n=Miranda@home.chicagoventure.com) |
| 17:45.20 | [TK]D-Fender | SkramX: Some leftover GUI crap is responsible. You shold already know which. |
| 17:48.04 | ACK-NAK | T1 question. Can I have one of the ports on a Digium T1 card show up to our legacy switch just like the Telco? In other words, if Asterisk were to fail, we could get by just by moving the T1 cable from Asterisk to the legacy switch (with no change in provisioning)? |
| 17:48.42 | Daviey | ACK-NAK: surely you'd need two phones on each desk? |
| 17:50.03 | ACK-NAK | Daviey: When a call comes in on the T1, it would be routed to the legacy switch just as before unless it's a number that Asterisk needs to handle |
| 17:51.17 | WimpMan | ACK-NAK: Yes |
| 17:52.04 | ACK-NAK | WimpMan: so it could be set up so that my switch wouldn't even know that asterisk is functioning as a 'bump in the wire' |
| 17:52.25 | ACK-NAK | WimpMan: the wire being the T1 circuit provisioned by the telco att/xo etc |
| 17:52.55 | WimpMan | More or less. |
| 17:53.07 | ACK-NAK | WimpMan: what's the less part? |
| 17:53.17 | *** join/#asterisk feeds (n=feeds@85-135-245-223.adsl.slovanet.sk) |
| 17:53.33 | WimpMan | For ordinary calls, yes. If you use supplementary services, Asterisk is likely to fail. |
| 17:54.02 | ACK-NAK | supplementary services such as facilities messages? |
| 17:54.12 | WimpMan | Yes |
| 17:54.32 | ACK-NAK | What other supplementary services would present a problem? |
| 17:54.58 | ACK-NAK | WimpMan: I appreciate your help by the way. |
| 17:55.09 | WimpMan | I'm not entirely sure, but I guess most of them. |
| 17:55.14 | WimpMan | HOLD works. |
| 17:55.18 | feeds | feeds |
| 17:55.43 | WimpMan | Not sure how likely you are to use that on a PRI, however. |
| 17:56.06 | hi365_m | anyone using a snom m3? |
| 17:56.22 | ACK-NAK | I can't imagine why we'd want the far-end to hold a call. |
| 17:56.39 | ACK-NAK | WimpMan: ...but that's what you're referring to, correct? |
| 17:56.58 | *** join/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com) |
| 17:57.00 | WimpMan | No, it's about what you do. |
| 17:57.33 | ACK-NAK | by far end I mean the DMS100 or whatever switch I'm connected to. |
| 17:57.48 | WimpMan | But you will indeed probably lose most information sent from the telco such as when you're being held or sent into a conference. |
| 17:58.53 | WimpMan | Charging information isn't forwarded either, as far as I remember. |
| 17:59.02 | *** join/#asterisk luar (n=weechat@cyberplant-1-pt.tunnel.tserv13.ash1.ipv6.he.net) |
| 17:59.08 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:59.22 | ACK-NAK | But if we're doing simple DID stuff like receiving DNIS digits for inbound calls, I could simply move the cable (or use a SPF device) to failover to the legacy switch, without having to change any provisioning on the legacy switch? |
| 17:59.23 | *** join/#asterisk s0lid (n=s0lid@acl1-34bts.gw.smartbro.net) |
| 17:59.41 | WimpMan | yes |
| 18:00.35 | ACK-NAK | WimpMan: Thanks. |
| 18:01.27 | ACK-NAK | I'm getting the feeling that TDMoE is dying. Agree? Disagree? |
| 18:01.47 | ACK-NAK | ...in favor of TDM hardware on PCI/PCX |
| 18:02.04 | mark_csi | hi all, I in here last night discussing missed incoming calls on a digium card. I've since found that the issue looks to be callerid related. |
| 18:02.12 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 18:03.01 | mark_csi | I've posted logs at http://www.pastebin.ca/1266798 |
| 18:03.21 | mark_csi | anyone any ideas? thx |
| 18:03.22 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:05.37 | SkramX | there aren't any Cepstral employees chilling in here.. are there? |
| 18:06.39 | ACK-NAK | T1 Echo Cancellation question. Why is echo cancellation necessary on a T1 card? Wouldn't that only be necessary if you were muxing out to a 2 wire system via channel bank? Isn't it usually the channel bank/ATA that does the echo cancellation? |
| 18:07.07 | [TK]D-Fender | ACK-NAK: Only 1 guy cared about TDMoE.... and we've disposed of his body last week :) |
| 18:07.17 | SkramX | hahaha |
| 18:07.38 | stintel | 1 more body coming up :] |
| 18:07.55 | ACK-NAK | [TK]D-Fender: :-) Funny. TDMoE seems great in theory. Why is it dying? |
| 18:07.58 | [TK]D-Fender | ACK-NAK: Or the second you bridge a call out to the PSTN and relaize the far side is analog |
| 18:08.15 | [TK]D-Fender | ACK-NAK: No, whats funny is that you should it was ever "lively". |
| 18:08.24 | [TK]D-Fender | ever though* |
| 18:08.31 | [TK]D-Fender | ever thought* |
| 18:08.51 | ACK-NAK | [TK]D-Fender: so the terminating carrier doesn't handle the echo? |
| 18:09.02 | [TK]D-Fender | ACK-NAK: don't bet on it. |
| 18:09.08 | [TK]D-Fender | ACK-NAK: EC is a reality for a reason |
| 18:09.45 | ACK-NAK | so the ten-to-fifteen-year-old legacy PRI cards in the nortel switches of the world most likely have hardware echo can built in? |
| 18:10.10 | [TK]D-Fender | ACK-NAK: yup |
| 18:10.27 | ACK-NAK | [TK]D-Fender: Now I understand. Thanks. |
| 18:11.06 | ACK-NAK | [TK]D-Fender: So TDMoE sounds like a great idea in theory. What makes it suck in practice? |
| 18:11.14 | jaytee | Katty: ping |
| 18:11.20 | [TK]D-Fender | ACK-NAK: What uses it beyond *? |
| 18:11.53 | [TK]D-Fender | (And that stupid RedFone crap) |
| 18:12.11 | jaytee | ugh, RedFone (jaytee shudders) |
| 18:14.37 | ACK-NAK | [TK]D-Fender: Even if it's only used by *, it seems like ethernet would be a nice way to abstract the physical transport layer. If it sucks in practice it's because it runs into some limitation that makes it lame. What is that limitation? What makes it go lame? |
| 18:16.00 | [TK]D-Fender | ACK-NAK: Yes, but its only useful between 2 boxes. It cannot be made redundant, adds pretty much nothing as far as stability goes really. Its a DEAD END. |
| 18:16.45 | *** join/#asterisk spokra (n=spokra@blockhead.sea0.speakeasy.net) |
| 18:18.16 | *** join/#asterisk wiscados (n=mint@81.25.184.155) |
| 18:19.04 | ACK-NAK | [TK]D-Fender: So I think you're saying that adding another bump in the wire can only decrease reliability, and what you achieve with the abstraction can be just as effectively replicated using physical interface cards? (...Such as the earlier discussion about failover to a legacy system? Easier to just use a 2-port card) |
| 18:19.17 | ACK-NAK | [TK]D-Fender: Is that a valid summary or am I missing another point? |
| 18:21.13 | [TK]D-Fender | ACK-NAK: the point is its only usable directly between 2 *'s. Requires special setp. Offers WHAT by comparison to any other VoIP transport, etc? If you want a dumb dedicated NIC, go right ahead. You can use a private subnet'd NIC but then at least you can route it, and debug it, do other stuf.... |
| 18:21.40 | jaytee | be back later |
| 18:21.41 | [TK]D-Fender | ACK-NAK: Just to FAKE some other protocol? retarded. It'd make more sense if it were interoperable with anything else but it isn't. |
| 18:21.55 | [TK]D-Fender | ACK-NAK: and FORGET that Redfone crap :p |
| 18:22.06 | [TK]D-Fender | ACK-NAK: So basically... worhtless |
| 18:22.28 | ACK-NAK | [TK]D-Fender: I was thinking of TDMoE with a redfone or an AddTran box. |
| 18:23.05 | ACK-NAK | Failover to *1 or *2 or legacy, handled by the box. No need for multiple physical interfaces. |
| 18:23.51 | [TK]D-Fender | ACK-NAK: Except when you realize WHAT is doing the failover. * is a shit failover engine |
| 18:24.07 | [TK]D-Fender | ACK-NAK: thats what proxies, RR DNS and all sort of other fun things are good at... |
| 18:24.36 | [TK]D-Fender | ACK-NAK: Put * at the center of the universe and the next segfault will send your entire slution right to hell |
| 18:26.02 | ACK-NAK | so a failover solution like the rhino SPF will move my T1 interface to a backup Asterisk server? What's the best way to deal with seg faults in a TDM world? |
| 18:26.31 | [TK]D-Fender | ACK-NAK: When * bombs, your external T1 failover will do its job :) |
| 18:26.49 | mikealeonetti | word to your momz |
| 18:26.53 | mikealeonetti | I came to drop bombz |
| 18:27.03 | mikealeonetti | know what I'm sayin'? |
| 18:27.12 | [TK]D-Fender | ACK-NAK: Now on the concept of SIP staying up, etc... a more hardened solution like SRE comes to mind. |
| 18:27.33 | ACK-NAK | SRE or SER? |
| 18:27.34 | [TK]D-Fender | ACK-NAK: at which point * tends to become a terminator / application server |
| 18:27.38 | [TK]D-Fender | SER* |
| 18:27.51 | ACK-NAK | I'd never heard of SRE. |
| 18:28.03 | [TK]D-Fender | ACK-NAK: This is the mehtodd used by a lot of larger deployments. |
| 18:28.13 | ACK-NAK | Is OpenSER the one to use? |
| 18:28.15 | [TK]D-Fender | ACK-NAK: Solar realms Elite... great BBS game! |
| 18:28.23 | [TK]D-Fender | ACK-NAK: But I'm talking SER... |
| 18:28.26 | [TK]D-Fender | ~ser |
| 18:28.27 | jbot | ser is probably [~ser] Sip Express Router - see http://www.iptel.org/ser/, or at #ser |
| 18:28.28 | ACK-NAK | :) |
| 18:29.02 | *** join/#asterisk wardenxvx (n=stepheng@pool-98-110-5-204.cmdnnj.east.verizon.net) |
| 18:29.03 | ACK-NAK | Can you point me to some failover devices you like? |
| 18:29.12 | ACK-NAK | [TK]D-Fender: This is very helpful. Thank you. |
| 18:29.42 | wardenxvx | curious.. what version of linux is best to run w/ asterick.. im going ot also be using my box as a file/print server also |
| 18:29.56 | [TK]D-Fender | ACK-NAK: I've never personally depolyed a HA setup myself... your T1 failover is a good step.. the rest requires you to look at your entire arch |
| 18:30.06 | [TK]D-Fender | ACK-NAK: I'm really not the best to advise you beyond these basics |
| 18:30.13 | [TK]D-Fender | ACK-NAK: Just glad to share some "insight" |
| 18:30.59 | ACK-NAK | [TK]D-Fender: I appreciate it. So the only good use for the cheaper non-echo-can digium cards are for things like point-to-point all digital stuff. |
| 18:31.00 | ACK-NAK | right? |
| 18:31.24 | *** join/#asterisk etfonhomey (n=chatzill@www2.askpri.org) |
| 18:31.25 | [TK]D-Fender | ACK-NAK: To me, *'s greatest strength should be its REPLACABILITY. I use Rhino CB (T1), a T1-PRI, and Polycom SIP phones. EVERYTHING is transportable to another base solution if need be and NO piece of my solution owns my ass. |
| 18:32.09 | [TK]D-Fender | ACK-NAK: Lets jsut say that I would never trust SWEC in its current state where I care about call quality |
| 18:32.19 | *** join/#asterisk khussein78 (n=khussein@a22-209.adsl.paltel.net) |
| 18:32.25 | etfonhomey | [TK]D-Fender, did telnettech ever find the problem? |
| 18:32.25 | Nugget | telnet is eeeeeeevil! |
| 18:32.26 | khussein78 | hi |
| 18:32.38 | khussein78 | hi |
| 18:32.53 | [TK]D-Fender | etfonhomey: not that I noticed... I was off the case at that point |
| 18:33.14 | etfonhomey | I've been away for a couple of hours. Just curious. |
| 18:33.17 | khussein78 | i see on my system that php command eat the cpu with agi scripts |
| 18:33.32 | khussein78 | and the CPU usage around 90% |
| 18:33.34 | ACK-NAK | [TK]D-Fender: Got it. I'd sort of ruled out, and thereby forgotten about T1 with SWEC. |
| 18:33.42 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-87-202.w86-215.abo.wanadoo.fr) |
| 18:33.57 | khussein78 | after i run killall -HUP php |
| 18:34.05 | Ericounet | hello all |
| 18:34.08 | ACK-NAK | [TK]D-Fender: Thanks for your input. |
| 18:34.09 | khussein78 | it back to normal, why this issue happened |
| 18:34.15 | [TK]D-Fender | ACK-NAK: You're welcome |
| 18:34.17 | khussein78 | and how can i debug this |
| 18:35.06 | *** join/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com) |
| 18:35.09 | *** join/#asterisk pawsmacker (n=dingo@206.124.12.162) |
| 18:36.20 | mark_csi | hi all, how do you capture the asterisk console data to a file? |
| 18:36.33 | kaldemar | use "script" for example. |
| 18:38.11 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 18:41.32 | etfonhomey | [TK]D-Fender, on your production systems did you go through and make sure that * doesn't load any modules you don't use, such as chan_mgcp? |
| 18:42.45 | [TK]D-Fender | etfonhomey: Certain secific ones like that, DAHDI, etc. Only the ones that run listening interfaces, etc that I don't need. Cust on security risks, etc. All apps, etc I leave up |
| 18:46.04 | telnettech | ET: sorry it was lunch time |
| 18:46.20 | telnettech | ET: the problem remains even after changing to NorthAmerica2 |
| 18:46.34 | jaytee | I hate the corporate world. I really miss kindergarten. You could take naps after lunch back then. |
| 18:47.04 | telnettech | I dont think you want to take a nap there in the elephant pen jaytee |
| 18:49.38 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 18:49.55 | etfonhomey | telnettech, did you see my suggestion about taking your * box out of the loop? Can you setup that Mediatrix to ring directly to a phone rather than having * between the phone and the gateway? That might tell you if the issue is some config within * or not. |
| 18:50.21 | etfonhomey | Not sure why some of that is in bold. |
| 18:50.26 | telnettech | I didnt see it but I cant take it out |
| 18:51.00 | telnettech | the mediatrix is the analog gatewau for the phones and they are not capable of working by themselves |
| 18:51.41 | telnettech | the wierd thing is it is just this 1 site |
| 18:51.53 | beek | DAMMIT! chan_dahdi.c:4290 handle_alarms: Detected alarm on channel 1: Red Alarm I thought I'd have had that fixed with the timing. |
| 18:51.59 | telnettech | we have the same setup at about 4 other customer's and they are all the same |
| 18:52.15 | etfonhomey | telnettech, the ringing issue exists at all the customers? |
| 18:52.29 | telnettech | the only difference is that this is analog CO trunks and not a T-1 or PRI |
| 18:52.37 | telnettech | no it is just this 1 customer |
| 18:53.03 | etfonhomey | telnettech, OK. How many analog lines do you have? |
| 18:53.29 | telnettech | the difference between the different customers is that this customer has CO lines thru the Meditrix gateway and the others have T-1 or PRI thru a red-fone device |
| 18:54.13 | telnettech | this customer is not programmed for zaptel because of the gateway "acting" as a SIP device to communicate to the Asterisk box |
| 18:54.53 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 18:55.24 | telnettech | there are 10 analog lines spread among 3 mediatrix FXO gateways |
| 18:55.27 | MrTelephone | is there a way to stop asterisk from denying authentication where a URI username does not equal the auth username? |
| 18:55.28 | etfonhomey | telnettech, I'm talking about how many of the FXO ports on the Mediatrix are you uing. |
| 18:55.35 | etfonhomey | using* |
| 18:55.49 | telnettech | 4 ports on the 1st and 2nd device |
| 18:57.04 | etfonhomey | telnettech, this is a ridiculous test, but might yield new information. What if you hook up analog phones to each of the 10 ports at the same time and dial in? |
| 18:58.43 | telnettech | When the Telco line hits the mediatri, the ring is fine, you will hear an auto attd and when you select off the menu, this is when you get the "funky" ringing |
| 18:58.48 | *** join/#asterisk lucasb (n=lbussey@office.telifon.com) |
| 18:59.06 | telnettech | even if you have mulitple calls the ringing is fine until after the Auto Attd |
| 18:59.07 | etfonhomey | telnettech, OK. Then scratch my ridiculous test. |
| 18:59.33 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 18:59.35 | etfonhomey | telnettech, is the auto attd. setup on your * box? |
| 18:59.41 | telnettech | yes |
| 19:00.42 | *** part/#asterisk feeds (n=feeds@85-135-245-223.adsl.slovanet.sk) |
| 19:00.49 | *** join/#asterisk wardenxvx (n=stepheng@pool-98-110-5-204.cmdnnj.east.verizon.net) |
| 19:00.51 | *** part/#asterisk wardenxvx (n=stepheng@pool-98-110-5-204.cmdnnj.east.verizon.net) |
| 19:01.31 | *** join/#asterisk wolfelectronic (n=wolfelec@91.112.227.150) |
| 19:02.52 | *** join/#asterisk wardenxvx (n=stepheng@pool-98-110-5-204.cmdnnj.east.verizon.net) |
| 19:03.42 | wardenxvx | for running asterisk along w/ file/printserver which distribution of linux do you reccomend |
| 19:04.46 | [TK]D-Fender | wardenxvx: Whichever youa re best capable of administering |
| 19:04.56 | wardenxvx | ok thanks |
| 19:05.01 | jameswf | thinks he will make the new guns n roses album his MOH |
| 19:05.06 | etfonhomey | telnettech, ok. Then I definitely think it's an * issue. |
| 19:05.16 | etfonhomey | jameswf, didn't they ban that in China? |
| 19:05.33 | jameswf | etfonhomey: they should have in america :) |
| 19:05.43 | etfonhomey | jameswf, lol! |
| 19:05.54 | telnettech | did you see the sip messaging that I posted? ET |
| 19:06.40 | etfonhomey | opening it back up. |
| 19:06.54 | jameswf | wardenxvx: I run asterisk in conjunction with a print server on ubuntu but it's workload is pretty light |
| 19:07.04 | telnettech | TKD: to answer your question about the MxSipApp, that is the mediatrix device |
| 19:08.27 | MrTelephone | 403 Authentication user name does not match account name <-- what is the purpose of this filter? |
| 19:11.37 | MrTelephone | does anyone know if the newer asterisk versions support differential from and auth username fields? |
| 19:12.08 | wardenxvx | yea.. i was debating of ubuntu and fedora.. i havent used linux since redhat 6 so its been a while |
| 19:12.28 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 19:12.47 | wardenxvx | jamesswf: have you tried the gui ver. of asterisk on it? |
| 19:13.11 | etfonhomey | telnettech, pastebin your sip.conf |
| 19:13.15 | [TK]D-Fender | GUI version.... LOL |
| 19:13.16 | *** join/#asterisk synchris (n=synchris@athedsl-05425.home.otenet.gr) |
| 19:13.59 | Katty | jaytee: pong. |
| 19:15.06 | Katty | jaytee: taking a break from samsung meeting :< |
| 19:16.55 | telnettech | ET: here is the sip.conf pastebin http://pastebin.com/d69222373 |
| 19:16.58 | etfonhomey | Katty, figure out the pricepoint of that thing? |
| 19:17.11 | Katty | etfonhomey: no |
| 19:17.25 | Katty | etfonhomey: they're not pitching price. they're pitching what it can do. |
| 19:17.33 | Katty | etfonhomey: i've learned i need to take a pillow with me when i go back. |
| 19:17.53 | etfonhomey | Katty, so you don't know what your at-cost price is? |
| 19:18.05 | [TK]D-Fender | Katty: Why don't they just pitch the Samsung... RIGHT OUT THE F-ING WINDOW! |
| 19:18.07 | Katty | etfonhomey: pricing isn't my problem. |
| 19:18.07 | [TK]D-Fender | aing! |
| 19:18.11 | [TK]D-Fender | zing! |
| 19:18.14 | Katty | [TK]D-Fender: hehe |
| 19:18.29 | Katty | etfonhomey: wether or not it's a piece of crap, is my problem. |
| 19:18.31 | [TK]D-Fender | ................................................ *poof* |
| 19:22.10 | Katty | etfonhomey: if you are so concerned about the price, why don't you call up samsung |
| 19:22.37 | etfonhomey | Katty, I don't want to waste my time, like you have to. :) |
| 19:22.57 | Katty | good idea ;) |
| 19:24.25 | etfonhomey | telnettech, I notice you're not doing "canreinvite=no" in your sip.conf. |
| 19:26.38 | *** part/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 19:28.45 | *** join/#asterisk aksyn (n=aksyn@gw.na.nu) |
| 19:30.39 | etfonhomey | telnettech, that forces the 2 endpoints to keep * in the middle of the call. |
| 19:32.22 | *** join/#asterisk aksyn (n=aksyn@gw.na.nu) |
| 19:33.39 | *** join/#asterisk saftsack (n=oliver@e176224156.adsl.alicedsl.de) |
| 19:37.49 | *** join/#asterisk Hadi- (n=AlanS@cmr-208-124-195-107.cr.net.cable.rogers.com) |
| 19:37.52 | Hadi- | hello everyone |
| 19:38.07 | Hadi- | im having a problem sending calls from asterisk to our new AS5400 |
| 19:38.14 | Hadi- | call connections but no audio |
| 19:38.21 | Hadi- | call connects |
| 19:38.33 | *** part/#asterisk Hadi- (n=AlanS@cmr-208-124-195-107.cr.net.cable.rogers.com) |
| 19:38.36 | *** join/#asterisk Hadi- (n=AlanS@cmr-208-124-195-107.cr.net.cable.rogers.com) |
| 19:39.36 | jameswf | sports people: orlando or boston |
| 19:47.08 | *** part/#asterisk pawsmacker (n=dingo@206.124.12.162) |
| 19:47.56 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 19:49.19 | mikealeonetti | is there such a thing as aliases in the voicemail.conf? |
| 19:52.43 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 19:54.14 | *** join/#asterisk kerx (n=prepro@adsl-68-125-33-177.dsl.irvnca.pacbell.net) |
| 19:54.31 | *** join/#asterisk Micc (n=dotirc@97-113-1-157.tukw.qwest.net) |
| 19:55.15 | Micc | I just setup a new aastra 480i and I set it up just like the one before I thought but it can't keep a call for more than a few seconds. It hangs up automatically. |
| 19:55.30 | Micc | It worked fine before with a different 480i. |
| 19:56.03 | *** join/#asterisk riddlebox (i=441ed6ab@gateway/web/ajax/mibbit.com/x-81e92c6e808be491) |
| 19:56.13 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 19:57.14 | [TK]D-Fender | mikealeonetti: Fake it on the file system |
| 19:57.57 | mikealeonetti | [TK]D-Fender: as in what? |
| 19:58.12 | etfonhomey | Hadi, that sounds like a NAT issue. Where is the AS5400 located in your network topology? |
| 20:00.28 | [TK]D-Fender | mikealeonetti: as in symlink the folder |
| 20:00.55 | mikealeonetti | I mean, can I have both Mike and Michael for the directory? is what I'm asking |
| 20:02.05 | *** join/#asterisk styelz (n=yoohoo@2001:5c0:8adb:0:0:0:0:1) |
| 20:04.09 | [TK]D-Fender | mikealeonetti: Sure, Just control the context they dump into |
| 20:04.24 | *** join/#asterisk aksyn (n=aksyn@gw.na.nu) |
| 20:05.09 | mikealeonetti | [TK]D-Fender: so for example, if I dial Mik or Mic the directory both sends it to my extension |
| 20:05.31 | [TK]D-Fender | mikealeonetti: Directory throws them into the DIALPLAN, its YOUR job |
| 20:06.02 | mikealeonetti | hrm |
| 20:09.21 | *** join/#asterisk bijit (n=benji@200.122.158.243) |
| 20:16.17 | telnettech | ET: sorry had to walk away to a meeting.....you are correct, we are not doing the Reinvite. This allows the users to do transfers which requires the * box |
| 20:16.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:17.07 | Micc | Ok, so why do I keep getting these chan_sip.c:4029 set_destination: can't find address for host '712' |
| 20:17.12 | Micc | 712 is the extension of the phone. |
| 20:22.14 | Micc | any ideas? |
| 20:23.23 | Micc | why is it that some extensions/phones work fine but others give that error then hangup. |
| 20:23.38 | [TK]D-Fender | Micc: pastebin is your friend.... |
| 20:24.30 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 20:26.24 | mikealeonetti | are blonde women still stupid if they dye their hair? |
| 20:28.16 | telnettech | TKD: any idea where you put what extension number people should reach when they "zero" out of voicemail? |
| 20:28.36 | telnettech | I have 0 as an extension in my extensions.conf |
| 20:32.29 | telnettech | TKD: never mind......i found it!!!! our developers suck |
| 20:33.37 | Micc | TKD, http://pastebin.com/m525c76de would you like my sip.conf or dialplan too? |
| 20:34.08 | etfonhomey | telnettech, is this a separate problem or related to the ringing issue? |
| 20:34.17 | telnettech | seperate issue |
| 20:34.35 | telnettech | new install last week.....customer already calling to get this turned on |
| 20:34.55 | telnettech | did you see the sip.conf pastebin ET? |
| 20:36.00 | etfonhomey | telnettech, yeah, I'm wondering if things would change if you set all your sip hosts to "canreinvite=no". |
| 20:36.17 | telnettech | i can try |
| 20:36.31 | etfonhomey | telnettech, I was looking through your SIP debug and it seemed like that should be more information. |
| 20:38.17 | telnettech | but from what i understand, if the reinvite is done, and the 2 devices reinvite to each other, then the people cant do a transfer. This is kinda important since we are calling a hotel front desk and may need transferred to either an admin or guest |
| 20:40.11 | telnettech | ET: sorry i need to learn to read better.....i can set the reinvite to "no" |
| 20:40.54 | telnettech | ET: the SIP messages that I posted had to do with the selection of the menu option off the auto attd and what the system did afterwards |
| 20:41.02 | etfonhomey | telnettech, I think that's the preferred way for most SIP endpoints. |
| 20:41.19 | etfonhomey | telnettech, Yeah, I thought I would see the reinvite messages, though. |
| 20:41.19 | *** join/#asterisk nny_1 (n=scott@64.203.244.146) |
| 20:42.39 | mikealeonetti | I get that if I added another line in the voicemail.conf and put "Mike Leonetti" to suppliment "Michael Leonetti" Directory would recognize that, but it has to have a different extension. I am very unclear how I can make both Mike and Michael go to my extension |
| 20:42.44 | mikealeonetti | and be the same vmail |
| 20:43.46 | nny_1 | anyone using teliax have an issue with a 480 client error/ 503 server error? First time trying them out. pastebin goodness with SIP debug etc here http://pastebin.com/mc44960f |
| 20:44.04 | nny_1 | I can use the account by connecting directly with my SPA962 |
| 20:44.22 | nny_1 | just making sure I didn't fudge something up on my end |
| 20:50.14 | *** part/#asterisk wolfelectronic (n=wolfelec@91.112.227.150) |
| 20:51.08 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) |
| 20:51.30 | Linuturk | what would happen if I restarted asterisk in the middle of the day? would any current calls drop? |
| 20:52.03 | etfonhomey | Linuturk, yes, you could do a "restart gracefully" and it will restart when there are no current calls. |
| 20:52.34 | Linuturk | etfonhomey: I've just set the holiday dates for thanksgiving, so I need to restart asterisk for that to take effect. what's the graceful restart? |
| 20:52.50 | etfonhomey | "restart gracefully" |
| 20:52.57 | Linuturk | lol |
| 20:52.57 | etfonhomey | From the CLI |
| 20:52.59 | Linuturk | that simple |
| 20:53.05 | Linuturk | thanks :) |
| 20:53.11 | etfonhomey | Sometimes developers make things easy |
| 20:54.46 | Linuturk | what will I see when it actually restarts? ie, how do I confirm? |
| 20:57.26 | etfonhomey | Linuturk, if you do it at the CLI, it will kick you out of the CLI (because the CLI is killed when asterisk restarts). Then you can get back to the CLI by typing asterisk -r |
| 20:57.40 | etfonhomey | If asterisk doesn't restart asterisk -r will fail. |
| 20:59.10 | Linuturk | got it :) |
| 20:59.13 | Linuturk | thanks :) |
| 20:59.53 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 21:01.01 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 21:02.41 | *** join/#asterisk aksyn (n=aksyn@gw.na.nu) |
| 21:06.59 | *** join/#asterisk aksyn (n=aksyn@gw.na.nu) |
| 21:07.37 | *** join/#asterisk Akiyuki (n=root@ip-208-109-154-197.ip.secureserver.net) |
| 21:07.54 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:07.54 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 21:08.01 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:08.12 | Micc | TKD, I have a sip debug, do you want to take a look at it? |
| 21:08.51 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 21:09.39 | Micc | In the SIP header its sending its internal ip address, not the external nat address, could that be the problem? |
| 21:12.23 | [TK]D-Fender | Micc: Sounds pretty clear to me... |
| 21:12.59 | Micc | So how do I tell the phone to use its nat address, I've put the external IP in the NAT IP field in the config but it still sends its internal IP. |
| 21:14.33 | Micc | NAT port is set to 0, should that be something else? |
| 21:14.41 | Micc | all the other phones are configured the exact same way. |
| 21:16.09 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:17.12 | Micc | The phones that work are also using their internal IP, so that is not the problem. |
| 21:20.40 | Micc | This makes no sense. |
| 21:22.34 | *** join/#asterisk etfonhomey_ (n=chatzill@www2.askpri.org) |
| 21:23.53 | [TK]D-Fender | ~sipnat |
| 21:23.54 | jbot | [~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 |
| 21:23.56 | [TK]D-Fender | read up |
| 21:24.05 | [TK]D-Fender | checkout time, back in a bit |
| 21:24.11 | mikealeonetti | ~sipchicks |
| 21:24.16 | mikealeonetti | ~hotgirls |
| 21:24.30 | mikealeonetti | jbot hates me |
| 21:24.30 | jbot | yes. |
| 21:24.39 | mikealeonetti | lol |
| 21:24.40 | jaytee | hehe |
| 21:24.46 | jaytee | jbot loves me |
| 21:24.47 | jbot | Yes, I do love you! |
| 21:24.50 | jaytee | see! |
| 21:24.57 | jaytee | I love jbot |
| 21:25.09 | jaytee | hmmm |
| 21:25.11 | mikealeonetti | jbot loves me |
| 21:25.12 | jbot | Yes, I do love you! |
| 21:25.32 | mikealeonetti | interesting |
| 21:25.34 | jaytee | must be one of those love/hate relationships |
| 21:25.43 | mikealeonetti | jbot is a hot girl (bot) |
| 21:25.55 | jaytee | jbot, you rock! |
| 21:25.55 | jbot | jaytee: aw, gee |
| 21:26.01 | mikealeonetti | lol |
| 21:26.47 | Micc | could this be a router issue? maybe switching out the phones has caused the router to get confused? |
| 21:27.09 | jaytee | the bot in Ubuntuforums used to have Chuck Norris'ims, Mr. T'isms and other internet meme crap in it. |
| 21:27.25 | mikealeonetti | heh |
| 21:27.32 | mikealeonetti | what's wrong with the bot we got? |
| 21:27.38 | mikealeonetti | he talk pretty good. |
| 21:27.43 | Micc | what is "got 400 out of order" ? |
| 21:27.49 | jaytee | ~chuck |
| 21:27.50 | jbot | somebody said chuck was the freebsd daemon |
| 21:28.07 | jaytee | ~Mr T. |
| 21:28.08 | jbot | i guess mr t is the man who pities fools and throws helluva far |
| 21:28.08 | mikealeonetti | Micc: that's when the machine won't put out any more soda, right? |
| 21:29.46 | mikealeonetti | how about |
| 21:29.50 | mikealeonetti | ~donaldtrump |
| 21:30.12 | Micc | nice. |
| 21:30.53 | *** join/#asterisk intralanman (n=lanman@va-67-76-163-209.sta.embarqhsd.net) |
| 21:31.14 | ReDNeQ | ?nat |
| 21:31.23 | mikealeonetti | ~nat |
| 21:31.24 | jbot | rumour has it, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 21:31.37 | ReDNeQ | ~ports |
| 21:31.37 | jbot | from memory, ports is http://www.debian.org/ports/, or http://www.isi.edu/in-notes/iana/assignments/port-numbers, or the FreeBSD ports system etc etc, or http://www.portforward.com/routers.htm |
| 21:31.44 | ReDNeQ | ~nat ports |
| 21:31.46 | mikealeonetti | ~thundercats! |
| 21:31.49 | ReDNeQ | doH |
| 21:31.54 | mikealeonetti | ~natsip |
| 21:32.01 | mikealeonetti | ~sipnat |
| 21:32.02 | jbot | [~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 |
| 21:35.13 | Micc | This is driving me nuts! This should be so simple! WTF!! |
| 21:35.22 | mikealeonetti | whjat's the issue? |
| 21:35.41 | SQLDarkly | Indeed. :) I almost expected another ! |
| 21:38.12 | Micc | outgoing calls work fine, incoming calls hangup after a couple seconds. |
| 21:38.38 | mikealeonetti | no matter what phone |
| 21:38.40 | Micc | and I get this chan_sip set_destination: can't find address for host error |
| 21:38.40 | mikealeonetti | even softphones? |
| 21:38.56 | Micc | only on a couple of phones. |
| 21:39.10 | mikealeonetti | what's different about the phones? |
| 21:39.23 | Micc | same model, same config. |
| 21:39.30 | Micc | I can't find a different. |
| 21:39.46 | mikealeonetti | they don't have conflicting IPs or anything like that, do they? |
| 21:45.31 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 21:45.39 | Micc | The only thing I can think of is the router has saved the wrong information in the nat table. |
| 21:46.02 | mikealeonetti | is it behind the network |
| 21:46.06 | mikealeonetti | another network rather |
| 21:47.23 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:57.01 | mikealeonetti | there goes my other identity |
| 21:59.27 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:59.39 | mikealeonetti | look who it is |
| 22:01.15 | nny_1 | gonna re post the question one more time to see if anyone has an idea of where I can dig further |
| 22:01.16 | nny_1 | anyone using teliax have an issue with a 480 client error/ 503 server error? First time trying them out. pastebin goodness with SIP debug etc here http://pastebin.com/mc44960f |
| 22:04.29 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 22:05.38 | edoceo | Is there a way in the Asterisk CLI to see the definitoins of the CallGroup and PickupGroup ? |
| 22:06.17 | [TK]D-Fender | edoceo: this is held in the channel definition |
| 22:08.18 | edoceo | sip show channels doesn't have it and I can't figure out what to specifiy for channel with sip show channel |
| 22:09.11 | edoceo | Ah Ha! |
| 22:09.18 | edoceo | sip show peer 65002 |
| 22:11.12 | SkramX | anyone ever used a Cisco phone through VPN? is that possible? |
| 22:12.46 | [TK]D-Fender | SkramX: Any reason the packets carried over your VPN are somehow "magical"? |
| 22:13.02 | orkid | (they're encrypted) |
| 22:13.31 | [TK]D-Fender | orkid: are they encryped when the REACH their destination? |
| 22:14.33 | orkid | i dunno, ask skramx |
| 22:14.35 | orkid | :) |
| 22:14.45 | nny_1 | nm on my question, looks like it is server side and teliax is looking at it |
| 22:16.01 | SkramX | [TK]D-Fender: im working on a project for work at home and asterisk server can't get an external IP... |
| 22:17.00 | [TK]D-Fender | SkramX: And why wouldn't * be able to get an external IP? |
| 22:17.07 | SkramX | Company policy |
| 22:17.19 | [TK]D-Fender | SkramX: that makes no sense... |
| 22:17.25 | SkramX | forget it. |
| 22:17.39 | [TK]D-Fender | SkramX: Extern IP isn't a policy, its a fact. |
| 22:17.45 | [TK]D-Fender | SkramX: who DOESN'T have one? |
| 22:18.06 | [TK]D-Fender | senses a linguistic/terminology failure |
| 22:21.00 | [TK]D-Fender | <PROTECTED> |
| 22:21.12 | [TK]D-Fender | darn caps... takes the edge off a great joke.. |
| 22:22.20 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:22.20 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 22:22.32 | *** join/#asterisk CrazyTux (n=brandon@rrcs-67-52-124-226.west.biz.rr.com) |
| 22:22.44 | lmadsen | hey all, looking for someone in the US I can get 40 Polycom ip330 phones from. Anyone have recommendations of people they have used? |
| 22:23.17 | *** join/#asterisk stevie123 (n=stevie@e177146026.adsl.alicedsl.de) |
| 22:23.35 | SkramX | [TK]D-Fender: haha |
| 22:24.20 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 22:24.48 | *** part/#asterisk nny_1 (n=scott@64.203.244.146) |
| 22:25.01 | stevie123 | hello, i'm using asterisk 1.6.0.1 with freepbx 2.5, i set up my trunk with host and port, transport=tcp and context=from-internal, but incomming calls aren recognized "unknown peer" |
| 22:25.27 | stevie123 | is this a tcp problem, that my host isnt matched correctly? |
| 22:26.43 | [TK]D-Fender | lmadsen: www.telephonydepot.com |
| 22:28.18 | lmadsen | US based? (I don't want to ship across the border) |
| 22:31.50 | [TK]D-Fender | lmadsen: Yes |
| 22:32.08 | [TK]D-Fender | lmadsen: This isn't the sam request as Yourname`s is it? |
| 22:32.13 | lmadsen | it is |
| 22:32.18 | lmadsen | I didn't know he has already asked |
| 22:32.20 | lmadsen | had* |
| 22:32.26 | [TK]D-Fender | lmadsen: PM'd |
| 22:32.26 | lmadsen | goes back to work |
| 22:32.30 | lmadsen | [TK]D-Fender: thx |
| 22:33.06 | [TK]D-Fender | lmadsen: Just about the best price around, and I and my clients on both sides of the border have bought from them and were very happy with the service |
| 22:34.29 | stevie123 | no idea? |
| 22:37.12 | stevie123 | how can i log the matching process for host=ip for incoming calls? |
| 22:42.15 | stevie123 | debug doenst help me here |
| 22:43.16 | [TK]D-Fender | stevie123: What debug? We don't see any... |
| 22:44.56 | stevie123 | core set debug 90 |
| 22:45.24 | [TK]D-Fender | stevie123: that is NOT "sip debug" |
| 22:45.39 | stevie123 | i use wireshark for sip debug |
| 22:45.45 | stevie123 | i dont have a SIP problem |
| 22:45.46 | [TK]D-Fender | stevie123: Don't |
| 22:46.20 | *** join/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net) |
| 22:46.24 | stevie123 | i setup a trunk but asterisk dont uses this for incoming calls |
| 22:46.43 | stevie123 | it says "Received incoming SIP connection from unknown peer to 63"" |
| 22:46.51 | [TK]D-Fender | stevie123: steForget about your "trunk", forget about your peers, and enable GLOBAL SIP debug. |
| 22:47.03 | *** join/#asterisk s0lid (n=s0lid@122.53.110.85) |
| 22:47.04 | [TK]D-Fender | stevie123: Open your eyes and step back |
| 22:47.34 | stevie123 | is this a CLI command? |
| 22:47.49 | [TK]D-Fender | "sip set debug on" |
| 22:48.00 | [TK]D-Fender | stevie123: Things you should really know by now... |
| 22:48.17 | pta200 | For Asterisk's Google Talk module, does there need to an entry in the gtalk.conf file for each Google user you want to call? |
| 22:48.36 | stevie123 | thx |
| 22:48.49 | stevie123 | i didnt know there was a difference |
| 22:49.34 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
| 22:51.22 | *** join/#asterisk aep (n=aep@thor.asgaartech.com) |
| 22:51.40 | aep | hey, i'd like to do phone conferencing with SIP or something similar. is asterisk what i want? |
| 22:51.56 | [TK]D-Fender | aep: big group? |
| 22:52.06 | aep | no. up to 5 |
| 22:52.26 | stevie123 | <PROTECTED> |
| 22:52.29 | [TK]D-Fender | aep: * can accomodate fairly large groups with MeetMe (dialplan application) |
| 22:52.50 | [TK]D-Fender | stevie123: just because the host matches dosn't mean the AUTH does |
| 22:53.05 | aep | sounds cool |
| 22:53.09 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
| 22:53.12 | aep | so asterisk is what i want? |
| 22:53.34 | [TK]D-Fender | aep: It's certainly do the job |
| 22:53.47 | aep | sounds like there is something better suited? |
| 22:53.56 | [TK]D-Fender | aep: nothing I've heard of. |
| 22:54.09 | aep | ok thanks, where do i start? |
| 22:54.11 | [TK]D-Fender | aep: * is of course capable of being so much more, but I've seen people use it for less... |
| 22:54.13 | [TK]D-Fender | ~book |
| 22:54.14 | jbot | i heard book is probably 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://tfot.leifmadsen.com or see ~buybook |
| 22:54.18 | [TK]D-Fender | ^^^^^ |
| 22:54.22 | aep | thanks |
| 22:54.27 | [TK]D-Fender | aep: and go download and install it. |
| 22:54.32 | aep | obviously :D |
| 22:54.35 | stevie123 | is there a difference between asterisk 1.4 and 1.6 with this auth process? |
| 22:54.54 | aep | is it fine to start testing on a real server right away or are there some security issues i should better read about before? |
| 22:54.56 | [TK]D-Fender | aep: Pay special attention that you'll be needing to install 2 major packages : Asterisk itself, and DAHDI (needed for timing for your conferencing) |
| 22:55.14 | [TK]D-Fender | stevie123: Shouldn''t be much |
| 22:55.56 | stevie123 | because my setup worked in 1.4 |
| 22:56.51 | [TK]D-Fender | stevie123: And you keep making empty statements and showing us nothing. |
| 22:57.57 | stevie123 | k |
| 22:59.31 | aep | uh wow asterisk looks fairly professional |
| 22:59.42 | aep | i'm sure it is the right choice for my simple needs |
| 22:59.44 | pta200 | anybody play with the gtalk channel? |
| 23:00.11 | *** join/#asterisk der_soenke (n=soenke@dslb-088-065-008-156.pools.arcor-ip.net) |
| 23:00.26 | [TK]D-Fender | aep: what you actually need is probably fairly simple. |
| 23:00.35 | [TK]D-Fender | aep: as for as configuration is concerned |
| 23:00.42 | aep | ie also simple to configure? |
| 23:00.43 | aep | ok cool |
| 23:00.53 | aep | i imagine the astersik configuration is huge |
| 23:01.06 | [TK]D-Fender | aep: in a matter of scale, but th pieces are spare and the learning curve steep on your own. |
| 23:02.04 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:02.04 | [TK]D-Fender | aep: not so much really... just a particular scaling depending... there is no perfect guide to home you in on what you want... its all about setting up your SIP peers for auth, and a minimal dialplan which basically only really consists of dumping most callers into a ocnference room |
| 23:02.08 | stevie123 | Here is my sip debug |
| 23:02.09 | stevie123 | http://pastebin.ca/1267031 |
| 23:02.17 | aep | oof the manual steps right into confiruation without explaining how stuff actually works |
| 23:02.47 | aep | i never touched anything with telephony |
| 23:03.06 | aep | any book i should read previous to that? |
| 23:08.32 | stevie123 | i found the problem, my caller uses a different port as i specified |
| 23:08.49 | stevie123 | thx D-Fender for the debug command |
| 23:09.42 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:09.43 | aep | haha the ascii art from configure is neat |
| 23:11.01 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 23:11.28 | [TK]D-Fender | aep: the book has a good primer on general telephony tech |
| 23:11.37 | [TK]D-Fender | ok, I'm off for a bit. |
| 23:11.39 | [TK]D-Fender | BBL |
| 23:11.46 | lesouvage | I just read "elastix without tears" ( see: http://www.elastixconnection.com/downloads/elastix_without_tears.pdf ) and this is certainly a good book without the "we have a gui and you don't need to know anything further" |
| 23:13.40 | lesouvage | All kind of explenation about the dialplan, configuring trunks, integration with openfire and sugercrm, setting up fax, router configuration, etc. etc. |
| 23:13.57 | *** join/#asterisk Segnale007 (n=Pietro@host86-135-19-235.range86-135.btcentralplus.com) |
| 23:14.28 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 23:14.39 | *** join/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 23:15.09 | UnixDawg | digium needs to rename the asterisk-now 1.5 beta something else and leave the asterisk-now project to the digium gui |
| 23:16.10 | *** join/#asterisk DeVilSoulBlacK (n=aandaluz@srv.ec-gye.internet.geainternacional.com) |
| 23:16.31 | DeVilSoulBlacK | Hi all !! |
| 23:19.15 | Deeewayne | O.O |
| 23:19.24 | pta200 | aloha |
| 23:19.38 | lesouvage | ?-) |
| 23:20.43 | Qwell | UnixDawg: suggestions@digium.com |
| 23:21.47 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 23:21.54 | pcrane | weirdness here: http://pastebin.com/m69848c1e |
| 23:22.13 | pcrane | asterisk 1.4 using dahdi... all config files refer to the old zap stuff |
| 23:22.33 | jdnWEST | So i'm setting up an * box, and I have 2 spare PRI's in my local market that are doing nothing, is there a website or somewhere that I could sell this excess capacity to? |
| 23:22.45 | tompaw | jdnWEST: arbinet? |
| 23:23.11 | tompaw | jdnWEST: www.arbinet.com |
| 23:24.35 | jdnWEST | interesting, i'll look into it, i'm just not sure if 2 pri's is enough for a large company like that to care |
| 23:24.48 | tompaw | they're traffic exchange |
| 23:24.58 | tompaw | you can set up an offer, if you find a buyer - why not? |
| 23:26.46 | jdnWEST | any idea on what the standard rates are based on, minutes, channels, number of jellybeans in a jar? |
| 23:28.26 | *** part/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net) |
| 23:28.37 | tompaw | jdnWEST: current market situation ;) |
| 23:30.08 | aep | what do i need to do so asterisk picks up my sip.conf and starts listening? |
| 23:30.22 | pcrane | cli> sip reload |
| 23:30.22 | tompaw | reload |
| 23:30.35 | aep | i did a restart yes |
| 23:30.44 | aep | root@sophia:/etc/asterisk$sip reload |
| 23:30.45 | aep | -bash: sip: command not found |
| 23:30.45 | aep | :P |
| 23:30.55 | tompaw | erm |
| 23:30.59 | tompaw | asterisk -r |
| 23:31.01 | tompaw | [ENTER] |
| 23:31.03 | tompaw | reload |
| 23:31.05 | tompaw | [ENTER] |
| 23:31.08 | aep | ew |
| 23:31.09 | aep | thanks |
| 23:31.13 | pcrane | asterisk -rx 'sip relod' |
| 23:31.15 | pcrane | reload* |
| 23:31.16 | pcrane | :p |
| 23:31.18 | etfonhomey | tompaw, good thing [TK]D-Fender has left for the day. |
| 23:31.22 | pcrane | mmm |
| 23:31.23 | tompaw | you're welcome[ENTER] |
| 23:31.24 | tompaw | ;-) |
| 23:31.43 | tompaw | etfonhomey: what harm did I cause? |
| 23:31.44 | aep | well still it didnt open a port |
| 23:32.12 | tompaw | you didn't mention that before, you said Asterisk doesn't listen to your commands from sip.conf |
| 23:32.12 | etfonhomey | tompaw, I'm just saying that he might have exploaded over aep's questions. |
| 23:32.15 | aep | my sip.conf is very basic. just a copy of that book |
| 23:32.33 | aep | listen as in open a port, yes |
| 23:32.48 | aep | the manual says "modify that file and then youäre good to go" |
| 23:33.00 | aep | it doesnt say how to actually make asterisk use sip |
| 23:33.16 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:33.18 | aep | some comandline option? |
| 23:33.29 | seanbright | you need to keep reading the book i think |
| 23:33.50 | tompaw | or you can pastebin your config for us |
| 23:34.00 | tompaw | come on, who reads books nowadays. |
| 23:34.04 | aep | sure, which file? i only modified sip.conf |
| 23:34.06 | seanbright | oh man... |
| 23:34.11 | etfonhomey | aep, * uses sip right out of the box |
| 23:34.17 | seanbright | i thought we had reached out minimum troll requirement for this channel |
| 23:34.24 | seanbright | apparently i was wrong. |
| 23:34.31 | joat | what? not enough? |
| 23:34.43 | aep | http://rafb.net/p/b8yK0S55.html |
| 23:35.17 | aep | netstat doesnt show any open ports from asterisk |
| 23:35.24 | aep | so i wont even try to connect |
| 23:35.33 | tompaw | aep: before they crucify you - how do you know it's not listening? |
| 23:35.57 | aep | http://rafb.net/p/8DwZbz49.html ;) |
| 23:36.40 | tompaw | -tulpen almost sounds like a word in... I don't know... Dutch? |
| 23:36.50 | tompaw | but does it also show udp? |
| 23:36.53 | aep | yeah its how i keep that in memory |
| 23:36.56 | aep | yes -u is udp |
| 23:37.03 | pcrane | yep just did that on an asterisk machine |
| 23:37.05 | pcrane | and it works |
| 23:37.12 | pcrane | (i.e. it should show asterisk stuff) |
| 23:37.34 | pcrane | e.g. udp 0 0 0.0.0.0:5060 0.0.0.0:* 0 35857 24195/asterisk |
| 23:37.38 | aep | asterisk is running however |
| 23:37.39 | joat | is asterisk actually running? |
| 23:37.40 | aep | 15661 ? Ssl 0:00 /usr/sbin/asterisk -G asterisk -U asterisk |
| 23:37.41 | tompaw | aep: is your asterisk running? |
| 23:37.50 | etfonhomey | aep, how bout a ps -aux | grep asterisk for us. |
| 23:38.14 | joat | what do the log files say? |
| 23:38.21 | aep | ps aux? ;) |
| 23:38.22 | aep | asterisk 15661 0.0 1.0 19272 1372 ? Ssl 16:31 0:00 /usr/sbin/asterisk -G asterisk -U asterisk |
| 23:38.26 | aep | log files! good idea |
| 23:38.44 | *** join/#asterisk SibRphrek (i=SibrPhre@cpe-67-243-43-136.nyc.res.rr.com) |
| 23:38.47 | aep | 1227659832|NONE|NONE|NONE|CONFIGRELOAD| thats everything i find |
| 23:38.50 | tompaw | that's the smartest thing that's been said here in the past 15 minutes. |
| 23:39.00 | joat | also, instead of running it as a service, try running it from the command line |
| 23:39.07 | aep | okay |
| 23:39.11 | joat | that's not the log you should be looking at |
| 23:39.19 | seanbright | /var/log/asterisk/full |
| 23:39.54 | tompaw | seanbright: how did you guess his distro? |
| 23:40.01 | aep | seanbright: no such file |
| 23:40.06 | aep | console says: |
| 23:40.06 | tompaw | ah, you didn't. |
| 23:40.08 | aep | [Nov 25 16:45:32] WARNING[15774]: loader.c:786 load_modules: No 'modules.conf' found, no modules will be loaded. |
| 23:40.14 | aep | maybe thats relevant |
| 23:40.18 | seanbright | tompaw: i didn't |
| 23:40.26 | *** join/#asterisk sasargen (n=chatzill@174-146-168-108.pools.spcsdns.net) |
| 23:41.04 | aep | ew that was it |
| 23:41.11 | aep | thanks! |
| 23:41.12 | seanbright | aep: did you install from source? |
| 23:41.37 | aep | nope. almost. throught a pkgbuiild, but thats close to from source |
| 23:41.49 | aep | ie hardly modifications |
| 23:41.58 | aep | the book assumes you're on debian |
| 23:42.10 | seanbright | should have sample confs in there somewhere |
| 23:42.10 | aep | so it didnt mention that |
| 23:42.13 | aep | but console provides good debug info :) |
| 23:42.15 | seanbright | with the source tarball you can do 'make samples' |
| 23:42.29 | aep | yep i have them. they're good |
| 23:42.30 | seanbright | not sure if there is an equivalent with pkgbuild |
| 23:42.32 | aep | just missed that one |
| 23:42.36 | seanbright | gotcha |
| 23:42.44 | seanbright | logger.conf is another good one :) |
| 23:42.52 | aep | hehe yeah |
| 23:49.09 | jaytee | <PROTECTED> |
| 23:49.54 | drmessano | I hear Vonage is better |
| 23:50.40 | jaytee | I want a voip account I can use with an Asterisk box |
| 23:51.28 | *** join/#asterisk DarkRift (n=dark@65.92.249.153) |
| 23:51.59 | jdnWEST | jaytee: I use vitelity there are better, and there are cheaper, but it works |
| 23:53.30 | jaytee | how much for a single DID #? |
| 23:53.38 | etfonhomey | jaytee, I use Vitelity as well. I've been using them for almost a year now and the only problem I have had is when my Internet goes down and I can't get to them, which is of course not their fault. |
| 23:54.17 | *** join/#asterisk propellerhead (n=yogurt2u@190.245.220.103) |
| 23:54.28 | etfonhomey | jaytee, $1.49 / month for a DID. |
| 23:54.39 | jaytee | I have Comcast as an ISP so I'm not sure how well some other companies VOIP will work. I'm suspicious of a major player like Comcast playing dirty. |
| 23:54.50 | jaytee | 1.49 a month for a DID and then what? |
| 23:54.54 | etfonhomey | jaytee, 1.39 cents / minute |
| 23:55.57 | etfonhomey | Minimum $30 to open it up, but then minimum of $15 refills. pay as you go |
| 23:56.02 | pcrane | I'm having problems with Unable to request channel Zap |
| 23:56.08 | pcrane | it's on a T1 line |
| 23:56.12 | jaytee | so for a 10 minute local call with Vitelity I'd be paying about a third of my monthly service with AT&T |
| 23:56.13 | pcrane | they can't receive calls on it |
| 23:56.23 | pcrane | so I'm using call files to check to make sure it works |
| 23:56.42 | *** join/#asterisk inv_arp (n=junya@b07s03mr.corenetworks.net) |
| 23:56.46 | etfonhomey | 10 cents? You pay 30 cents / month for AT&T? |
| 23:56.56 | etfonhomey | Excuse me 14 cents. |
| 23:57.08 | jaytee | I pay about 44 bucks a month for AT&T local and LD total |
| 23:58.03 | etfonhomey | jaytee, if my math is right, that comes to 0.3 % of your monthly AT&T bill for a 10 minute call via Vitelity. |
| 23:58.35 | etfonhomey | jaytee, I forgot to add in the DID cost. |
| 23:59.06 | etfonhomey | jaytee, with the DID cost added in, that's 3.7% of your monthly AT&T bill. |
| 23:59.11 | jaytee | no, I'm talking the minute rate charge of 1.39. do the math, a ten minute call for 1.39 a minute is 13.90 cents |
| 23:59.21 | *** join/#asterisk lucasb (n=lbussey@office.telifon.com) |
| 23:59.42 | etfonhomey | $0.139 != $44.00 / 3 |