| 00:00.51 | drmessano | Skype becomes the de facto method for exchanging calls across the public internet and we've gone from locked into AT&T to locked into Skype |
| 00:01.03 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 00:01.03 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:01.22 | mchou | bah. there is no need to load _chan_skype :) |
| 00:01.27 | jaytee | I need to burn everything to DVD for archiving most soon. |
| 00:01.42 | jaytee | and get ready to go underground |
| 00:01.59 | LiNeTuX | drmessano: On a plus note, I found out AT&T supports Asterisk for their native SIP stuff. |
| 00:02.34 | jaytee | SkypeNazis and UberMenschenOpenSIPS stormtroopers busting down my door in the middle of the night. |
| 00:02.48 | mvanbaak_ | hey russellb |
| 00:03.58 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 00:05.17 | *** part/#asterisk friedrich| (i=friedric@trem-servers.com) |
| 00:06.46 | jeev | :> |
| 00:06.54 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 00:08.26 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 00:09.02 | drmessano | Back, I think |
| 00:10.33 | drmessano | wonders if he mentions "Skype" again if his modem will die |
| 00:10.40 | drmessano | oh no |
| 00:11.43 | drmessano | amihere? |
| 00:11.58 | RB2 | haha |
| 00:11.59 | drmessano | %ping% |
| 00:13.43 | RB2 | has never made a voice call using Skype... |
| 00:14.06 | RB2 | ok, I lied... I called the Skype Test Call twice... |
| 00:14.55 | drmessano | If you're gonna have to pay for Skype + Asterisk, I think it should come with friends |
| 00:15.16 | drmessano | Perhaps one friend per channel |
| 00:16.25 | RB2 | I can see where skype is useful for integrating customer service on the web. Click a link and someone is connected via skype to your call center, etc.... |
| 00:16.32 | RB2 | But, beyond that, I have no use for it. |
| 00:16.51 | jaytee | after the merger they'll move to acquire MySpace and it will become MySkypeAss@risk |
| 00:17.13 | RB2 | Merger? |
| 00:17.19 | RB2 | Did I miss something? |
| 00:17.25 | *** join/#asterisk sucituanbo (n=john@c-24-21-121-148.hsd1.wa.comcast.net) |
| 00:17.43 | jaytee | it starts with just a card and some demos and from there who knows? |
| 00:17.58 | *** join/#asterisk hardwire (n=hardwire@197-198-137-216.mtaonline.net) |
| 00:18.01 | hardwire | blah |
| 00:18.20 | jaytee | [TK]D-Fender, how you doin tonight? |
| 00:19.07 | drmessano | I would have been much happier seeing a partnership with AOL, since they at least have persued SIP |
| 00:19.31 | *** join/#asterisk Segnale007 (n=Segnale0@host112-254-dynamic.32-79-r.retail.telecomitalia.it) |
| 00:19.51 | drmessano | and XMPP actually |
| 00:20.07 | *** join/#asterisk Dystemper (i=phaze2@babble-on.org) |
| 00:22.00 | hardwire | shoots Dystemper |
| 00:22.46 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 00:22.49 | jaytee | had to put him down, he wuz old and had the distemper |
| 00:23.01 | RB2 | wb drmessano^ |
| 00:23.03 | hardwire | Dystemper.. required in 7 states |
| 00:24.07 | Dystemper | lol |
| 00:24.18 | drmessano^ | Internet keeps going down |
| 00:24.42 | hardwire | drmessano: in this day and age, the porno age, that's bound to happen. |
| 00:24.46 | v4mp | aol sucks no matter what they even suck in the uk... and aol no longer own it they got bought out by a mobile phone shop |
| 00:24.51 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 00:25.04 | hardwire | ok.. that was a stretch.. sorry guys. |
| 00:25.04 | tvirus | Any reason why, when someone calls into a Zap line, that it doesn't detect DTMF. This happens randomly and I need to do 'service zaptel restart' to get it to work. |
| 00:25.24 | tvirus | vpmdtmfsupport option for the module doesn't affect it. |
| 00:25.34 | drmessano^ | AOL has an extremely large userbase, and supports SIP for their voice offerings |
| 00:25.43 | drmessano^ | They also have plans to move to XMPP for their IM |
| 00:25.49 | drmessano^ | To me, thats sizeable |
| 00:26.27 | v4mp | dont think they do SIP uk side |
| 00:26.44 | drmessano^ | I could care less about the UK |
| 00:26.51 | v4mp | pfft |
| 00:27.02 | drmessano^ | UK != here |
| 00:27.10 | v4mp | lol |
| 00:27.18 | v4mp | aye true |
| 00:29.54 | RB2 | That's two people I've seen that say they know of a java applet browser-based sip client, but nobody has provided links. Lies, all lies! |
| 00:33.30 | RB2 | oh, there's one on java.net based on jain |
| 00:35.01 | mchou | drmessano: lemme get this straight. Say I load chan_skype. There's a chance my * box will become a skype supernode? |
| 00:42.18 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 00:44.07 | jaytee | having connection issues? |
| 00:45.15 | v4mp | oh dr |
| 00:45.36 | v4mp | uptime[13w 4d 2h 38m 28s] |
| 00:45.45 | jaytee | (Uptime) 14 days 2 hours 56 seconds [Record: 14d 2h 0m 56s] | Users: 1 | Load: 0.02 0.19 0.17 |
| 00:46.05 | v4mp | linux ? |
| 00:46.12 | jaytee | yep |
| 00:46.16 | v4mp | xchat ? |
| 00:46.20 | jaytee | yep |
| 00:46.30 | v4mp | what script u use to get record too ? |
| 00:47.05 | v4mp | or does that just read current uptime rather than saving longest uptime ? |
| 00:47.07 | jaytee | it's a python script |
| 00:47.32 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 00:48.01 | jaytee | I've got another one that dumps all kinds of info but I won't flood the channel |
| 00:48.22 | jaytee | I should really reboot this pig |
| 00:49.06 | RB2 | I had a linux box that was up almost a year and wouldn't you know it, on day 364, I had to shut it down because of maintenance on one of the power mains in the building. :-/ |
| 00:49.24 | jaytee | aw man! that totally sucks! |
| 00:49.29 | v4mp | nice :| |
| 00:50.14 | RB2 | I was considering keeping it on backup power until it rolled over. LOL |
| 00:50.28 | jblack | It seems to me that bragging about uptime is a way to state "I have a machine that is vulnerable to every security hole since X/Y/2007" |
| 00:50.29 | jaytee | like Frogger on Seinfeld? |
| 00:50.41 | RB2 | lol |
| 00:50.50 | jaytee | jblack, good point |
| 00:51.00 | RB2 | Not bragging, just thought it was humorous. ;) |
| 00:52.43 | v4mp | evening jblack |
| 00:53.04 | jblack | hi |
| 00:53.07 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 00:53.25 | jaytee | wb |
| 00:53.43 | v4mp | wanna try get other bits working on * but cant until can receive incoming calls to test |
| 00:54.24 | jblack | do you have a did? |
| 00:54.30 | RB2 | v4mp, 7777 won't do the trick for what you're testing? |
| 00:54.31 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 00:54.37 | drmessano^ | nice |
| 00:54.54 | drmessano^ | Comcast tells me my downstream power is 11db |
| 00:55.03 | drmessano^ | Which is way too high |
| 00:55.07 | drmessano^ | Needs to be 2 <> 7db |
| 00:55.16 | drmessano^ | So I found a radio shack cheap ass 4 port splitter |
| 00:55.21 | drmessano^ | Shoved it inline |
| 00:55.34 | drmessano^ | 3.7db will hold me over until tuesday when they come onsite |
| 00:56.22 | mchou | drmessano^: lemme get this straight. Say I load chan_skype. There's a chance my * box will become a skype supernode? |
| 00:56.49 | drmessano^ | If I understand correctly, no |
| 00:57.26 | drmessano^ | I believe the channel module gives you some path into the Skype cloud.. |
| 00:57.38 | v4mp | RB2, 7777 ? |
| 00:57.39 | drmessano^ | I don't believe it's a bridge to a client |
| 00:58.20 | mchou | wow |
| 00:59.20 | v4mp | jblack, i do weren't it you helping me for hours last nite and ended up being that the fault seems to be at providers end ? |
| 00:59.52 | RB2 | v4mp, I believe it simulates an incoming call. |
| 01:01.36 | jblack | v4mp: Ok. msg me the number, and I'll call it real quick. |
| 01:03.19 | v4mp | jblack, but its a uk number and its still not be sorted as only just had chance to contact them a few hours ago so wont hear nothing til maybe 2moz |
| 01:04.12 | jblack | suit yourself. |
| 01:04.40 | v4mp | u can if you want but woulda thought u wouldn't want to because its a uk national or local rate number |
| 01:09.34 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 01:09.34 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:14.52 | jblack | v4mp: You understand that calls from the US to the uk are all of 2 cents a minute? |
| 01:15.27 | v4mp | but that would be to a landline though ? |
| 01:16.20 | jblack | Double check the number you gave me |
| 01:17.07 | jblack | The number as you gave it to me lists as inmarsat. |
| 01:17.14 | Qwell | left |
| 01:17.19 | Qwell | erm |
| 01:17.26 | jeev | Qwell, did you pump up the jam today |
| 01:17.37 | v4mp | inmarsat whats that ? |
| 01:17.50 | jblack | Satellite phone, I believe. |
| 01:18.10 | Qwell | jeev: I pumped it up, yes |
| 01:18.11 | v4mp | well thats the correct number |
| 01:18.31 | jblack | What area? |
| 01:19.02 | v4mp | area ? o_O u wont use an area code for 1 of those numbers |
| 01:19.22 | v4mp | its like the 1-800 numbers but u pay to call it |
| 01:19.40 | jblack | What country. |
| 01:19.52 | v4mp | 44 |
| 01:20.25 | [TK]D-Fender | v4mp: Here that'd be "976" numbers |
| 01:20.41 | v4mp | correct :) |
| 01:20.52 | [TK]D-Fender | v4mp: Several of which lead to drmessano^ no doubt ;) |
| 01:21.47 | v4mp | 0970/0871/0844/0845/07077/0808 |
| 01:21.55 | v4mp | *0870 |
| 01:22.24 | v4mp | jblack, did you just call it ? |
| 01:23.19 | jblack | Yes, I did. |
| 01:23.24 | v4mp | 14413 handle_request_invite: Call from '84455707' to extension '84455707' rejected because extension not found. <<< from chan_sip.c |
| 01:24.11 | [TK]D-Fender | v4mp: Blatantly obvious enough.... |
| 01:24.33 | [TK]D-Fender | v4mp: Clearly you don't have an exten to match that # in the context it's being looked for in. |
| 01:25.00 | v4mp | its not meant to use that as an extension :/ |
| 01:25.43 | jblack | You should set up an incoming context, which you put all incoming calls to, then, when something comes in on 84455707, you can Goto() it to the rightp lace. |
| 01:26.50 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 01:27.00 | v4mp | went right over my head then lol |
| 01:28.49 | jblack | gets beer |
| 01:35.46 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-211-254-206.dsl.pltn13.sbcglobal.net) |
| 01:36.08 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:36.14 | v4mp | wtf jblack how come when u call the number it reaches * but when i call it doesn't :S |
| 01:38.29 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 01:45.57 | jblack | double checks UK dialing rates with diamondcard |
| 01:47.16 | jblack | Ouch! |
| 01:47.27 | Qwell | they seem so shady to me... ;/ |
| 01:47.58 | Qwell | parent company is "an MLM company" |
| 01:48.31 | jblack | 44-1 is .021, about like I'd expect. for 44-7, it shoots up to .22/min |
| 01:48.53 | Qwell | is that cellular or something? |
| 01:49.35 | jblack | From their call rates page, it's mobile. Split between o2, orange, tmobile, etc. |
| 01:49.45 | Qwell | makes sense |
| 01:50.49 | jblack | They dont' list 44-78 at all, which shouldn't be mobile, as it's a did for v4mp |
| 01:51.35 | timholum1 | HELLO |
| 01:51.51 | timholum1 | sorry for the caps |
| 01:51.57 | Qwell | EHLO |
| 01:52.02 | jblack | I'm just burning off diamoncard minutes anyways, as they're not very nimble. |
| 01:52.06 | timholum1 | oups caps lock was on :) |
| 01:53.23 | jblack | I have four providers right now. That's 2 too many. |
| 01:55.03 | timholum1 | i have exten => 604,n,VERBOSE("CALLERIDNUM = ${CALLERIDNUM} AND ${EXTEN}") it outputs "CALLERID = AND 604, in my sip.conf i have callerid="CouleTechLink" <200> ? |
| 01:55.22 | timholum1 | shouldint it output CALLERID = 200 AND 604 |
| 01:56.03 | jblack | why I have 4 is a long story involving a combination of ex-wives and impatience on my part. |
| 01:56.17 | Qwell | timholum1: ${CALLERID(num)} |
| 01:56.21 | jblack | Neither is good. |
| 01:56.29 | timholum1 | ok ill try that thanks :) |
| 01:56.43 | Qwell | timholum1: also, don't put quotes in your callerid |
| 01:58.29 | timholum1 | Qwell: it was just for verbose, i was trying to debug why it was not working i was trying VoiceMailMain($CALLERIDNUM@ctl) and it wasnt working so but now i know i need VoiceMailMain($CALLERID(num)@ctl) Thankyou |
| 01:59.09 | Qwell | no, both are wrong |
| 01:59.14 | [TK]D-Fender | timholum1: You you also need to re-read the chapter on variable & function referencing that tells you that get the value out of them they need to be put in ${} |
| 01:59.18 | Qwell | ${}, not just $ |
| 01:59.25 | [TK]D-Fender | timholum1: You don't just shove a $ in front. |
| 01:59.48 | [TK]D-Fender | timholum1: Go read channelvariables.txt in your doc folder |
| 01:59.50 | Qwell | and I mean when you set it in sip.conf |
| 01:59.52 | timholum1 | ok, thanks, i had the {} in my conf, just not when i read |
| 01:59.59 | timholum1 | read* typed |
| 02:00.02 | jblack | * needs an assisting daemon that can determine least cost routing by doing an xml-rpc request. |
| 02:00.33 | jblack | backed by some publicly managed database for the providers Out There. |
| 02:00.53 | *** join/#asterisk jtodd (i=hnmewse0@ns2.loligo.com) |
| 02:01.39 | timholum1 | ok i will do that [TK]D-Fender |
| 02:02.25 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 02:06.42 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 02:06.43 | drmessano^ | fascinating |
| 02:07.24 | *** join/#asterisk newmember (n=chatzill@S010600036d1139bb.cg.shawcable.net) |
| 02:08.04 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 02:09.40 | mosty | how can i dial two sip clients, with each of those two channels having different TRANSFER_CONTEXT values? i tried setvar in the peer's entry in sip.conf but it did not seem to work |
| 02:13.17 | jblack | Later all |
| 02:32.40 | *** join/#asterisk mateo_au (n=chatzill@12.144.159.231) |
| 03:16.33 | *** join/#asterisk Segnale007 (n=Segnale0@host112-254-dynamic.32-79-r.retail.telecomitalia.it) |
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| 03:28.36 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 03:59.01 | *** join/#asterisk hardwire (n=hardwire@rdbk-5166.mtaonline.net) |
| 03:59.05 | hardwire | oh hai |
| 04:00.36 | *** join/#asterisk jaybinks (n=jay@ppp118-208-84-213.lns2.bne4.internode.on.net) |
| 04:01.19 | jaybinks | hey ... Im integrating Asterisk ( using the AMI ) into our companies CRM software... how do I know if a call is inbound or outbound ... ? ( from the events on the AMI ) |
| 04:01.45 | hardwire | jaybinks: there is no difference |
| 04:02.03 | jaybinks | so there is no easy way to know then ? |
| 04:02.05 | hardwire | you simply have to know the dialplan, or set channel variables when calls come in or out from the dialplan |
| 04:02.24 | jaybinks | hmmm ok .. thanks |
| 04:02.28 | hardwire | YOU BETCHA |
| 04:02.46 | jaybinks | not a bad idea though... I can set a variable on my incoming sip trunk |
| 04:02.54 | jaybinks | and will that then be available in the event messages ?? |
| 04:02.59 | jaybinks | or do I need to "look" for it somehow |
| 04:03.00 | hardwire | if your company uses CRM software then it should associate phones/queues to specific CRM accounts at some level |
| 04:03.02 | hardwire | so you could use that info |
| 04:03.14 | tzafrir_laptop | jaybinks, make incoming calls go to a different context |
| 04:03.22 | hardwire | oh that's a good one too. |
| 04:03.24 | tzafrir_laptop | Set up a variable in that context |
| 04:03.33 | jaybinks | yea... the incoming ones do go "through" a different context |
| 04:03.35 | tzafrir_laptop | or any similar method |
| 04:03.37 | hardwire | or use that context "as" the variable |
| 04:03.38 | jaybinks | then they get put into the general one I Think |
| 04:03.50 | hardwire | interesting |
| 04:03.52 | tzafrir_laptop | You can also set a variable directly in the channel configuration |
| 04:03.59 | jaybinks | ( using Elastix .. so not 100% sure of how that works ) |
| 04:04.10 | jaybinks | how do I set the variable like you say ? |
| 04:04.12 | tzafrir_laptop | Elastix uses freepbx |
| 04:04.19 | jaybinks | yea exactly |
| 04:04.26 | tzafrir_laptop | freepbx has a rather complex dialplan |
| 04:04.34 | jaybinks | yea |
| 04:04.45 | hardwire | I've never used it |
| 04:04.45 | jaybinks | I only have incoming calls come in 1 provider |
| 04:04.52 | tzafrir_laptop | But you should see that incoming and outgoing calls have different CONTEXT |
| 04:04.53 | jaybinks | so I dont mind setting that up in the sip trunk though |
| 04:05.32 | jaybinks | ... Im not sure that the "Events" have the context in them though |
| 04:05.34 | jaybinks | ( or can I turn that on ) |
| 04:05.46 | hardwire | jaybinks: you can follow a calls unique id |
| 04:05.48 | tzafrir_laptop | it depends what events |
| 04:05.59 | hardwire | and correlate transfers with them, etc.. |
| 04:06.03 | jaybinks | oh ok |
| 04:06.09 | jaybinks | what event has the context then ? |
| 04:06.24 | jaybinks | I just grepped my manager log ( that Im generating in my software ) |
| 04:06.28 | jaybinks | and I dont see any context anywhere |
| 04:06.28 | hardwire | muck with it and find out :) |
| 04:06.33 | jaybinks | so maybe Im not getting that event |
| 04:06.54 | jaybinks | oh there one is .. oops |
| 04:06.59 | jaybinks | NewExtn : sets it |
| 04:07.14 | hardwire | does the uniqueid match later events? |
| 04:07.22 | jaybinks | yea fairly sure it will |
| 04:07.24 | hardwire | that's the important part :) |
| 04:07.27 | jaybinks | im already matching up unique id's |
| 04:08.09 | hardwire | on to other important business.. anybody know how long I should wait until using the teeth on a cement trowel on the surface? |
| 04:09.26 | hardwire | going once? |
| 04:11.06 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 04:19.52 | *** part/#asterisk gcarrillog (n=gcarrill@201.151.84.240) |
| 04:21.07 | *** join/#asterisk drbrown (n=drbrown@rrcs-24-172-153-82.central.biz.rr.com) |
| 04:25.12 | drbrown | has anyone tried octasic octware echo canceler? |
| 04:25.57 | hardwire | that sounds too fantastic to be real |
| 04:26.04 | hardwire | kinda like everlasting gobstoppers |
| 04:27.18 | drbrown | I have an interesting echo problem. |
| 04:27.35 | drbrown | one that a hardware echo canceler will be unable to solve |
| 04:28.48 | drbrown | I have echo on a voip line delivered by time warner cable via an arris cable modem |
| 04:29.28 | hardwire | hmm |
| 04:29.37 | hardwire | call time warner and tell them not to screw it up |
| 04:29.38 | hardwire | seriously |
| 04:30.02 | drbrown | if I use a hardware echo canceler it cancels the echo twice and causes the conversation to cut out. |
| 04:30.04 | hardwire | theres a 90% chance it's not your fault |
| 04:30.09 | drbrown | really |
| 04:30.27 | hardwire | echo is most prevalent in PSTN gateways. |
| 04:30.29 | drbrown | when I hook up a butt tester the line sounds fine |
| 04:30.33 | hardwire | wait |
| 04:30.36 | hardwire | butt tester? |
| 04:30.39 | drbrown | but |
| 04:30.42 | hardwire | so you have PSTN? |
| 04:31.00 | hardwire | you're using an ATA? |
| 04:31.00 | drbrown | they will only deliver it via fxs ports |
| 04:31.03 | hardwire | ah |
| 04:31.04 | hardwire | yeh |
| 04:31.06 | drbrown | essentially |
| 04:31.12 | hardwire | so they own the ATA? |
| 04:31.16 | drbrown | yes |
| 04:31.25 | hardwire | and there is no echo.. gotcha |
| 04:31.28 | hardwire | well..that stinks. |
| 04:31.38 | hardwire | used fxotune? |
| 04:31.50 | drbrown | if I cancel the echo at 128 taps it's too far out. |
| 04:32.00 | drbrown | it actually cuts the conversation |
| 04:32.37 | drbrown | No I haven't used fxotune |
| 04:32.40 | hardwire | you should see if they can kill echocan on their ATA |
| 04:32.55 | drbrown | I thought about doing that as well |
| 04:33.32 | drbrown | I am also having problems with call hangup detection as well, thought they might be able to help out with that as well |
| 04:33.42 | hardwire | it's a shame they are so limited. |
| 04:33.51 | hardwire | MTA's are a pain |
| 04:33.55 | drbrown | yes |
| 04:33.58 | hardwire | is that via the motorola? |
| 04:34.04 | drbrown | arris |
| 04:34.34 | hardwire | that's really the only way they can garantee certain things and not pay out the nostril for e911 |
| 04:34.56 | drbrown | I know |
| 04:35.24 | drbrown | I have managed to eliminate echo now, but have 0 sidetone |
| 04:35.53 | hardwire | so you used fxotune? |
| 04:36.37 | drbrown | not yet, getting ready too |
| 04:36.39 | drbrown | to |
| 04:37.23 | hardwire | WhiteWolf: I fancy your nick.. where are you located? |
| 04:37.45 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 04:41.10 | drbrown | all zeros |
| 04:41.25 | hardwire | hmm |
| 04:41.30 | hardwire | well I'm off to bed |
| 04:41.43 | hardwire | call, complain, subscribe them to dirty magazines |
| 04:41.50 | hardwire | do whatever you have to do.. |
| 04:41.54 | drbrown | thanks for the help hardwire |
| 04:42.04 | hardwire | word.. sorry I'm not on top of this atm :) |
| 04:42.23 | jeev | WOW |
| 04:42.24 | jeev | you liED |
| 04:42.57 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 04:43.10 | *** join/#asterisk keith4__ (n=keith@216-164-144-176.c3-0.eas-ubr9.atw-eas.pa.cable.rcn.com) |
| 04:45.27 | drbrown | can you use digium's hpec with sangoma equipment? |
| 04:45.59 | russellb | that might create a black hole |
| 04:46.06 | russellb | but in theory, yeah |
| 04:46.21 | hardwire | russellb: I forgot to bring my hpec/g729 stuff to digiums booth |
| 04:46.22 | hardwire | like |
| 04:46.28 | hardwire | I didn't even look in the bag till I got home |
| 04:46.33 | russellb | hardwire: fail! |
| 04:46.34 | hardwire | I thought it was full of confetti and shirts. |
| 04:46.36 | hardwire | tutally |
| 04:46.38 | hardwire | totally |
| 04:46.42 | hardwire | I'll live |
| 04:46.48 | hardwire | I buy echocan cards anyways |
| 04:48.07 | drbrown | you can't reduce the card from 128 to 64 though can you???? |
| 04:48.46 | drbrown | it cuts out too much of the conversation and actually cuts off the calling party |
| 04:48.49 | hardwire | no clue.. haven't had to tweak them yet. |
| 04:49.08 | drbrown | because the echo is being canceled twice |
| 04:49.20 | hardwire | like some sort of black hole. |
| 04:49.27 | hardwire | russellb is on to something. |
| 04:49.39 | drbrown | black hole????? |
| 04:50.13 | keith4__ | if you cancel the echo twice, do you end up canceling the original sound? |
| 04:50.19 | hardwire | you're going to give helicopters cancer. |
| 04:50.46 | drbrown | not sure what you're sayin but yeah that's what's going on |
| 04:51.02 | keith4__ | helicopters... cancer? |
| 04:51.12 | drbrown | maybe????? |
| 04:51.48 | drbrown | you cancel the other side of the conversation completely at times. |
| 04:54.09 | hardwire | keith4__: http://xkcd.com/401/ |
| 04:54.32 | keith4__ | nice |
| 04:56.25 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 04:56.29 | drbrown | nice |
| 04:57.01 | hardwire | -> bed for reals |
| 04:58.54 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 05:02.13 | *** join/#asterisk mateo_au (n=chatzill@12.144.159.231) |
| 05:06.03 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 05:12.05 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 05:13.25 | drmessano | ah joy |
| 05:13.32 | keith4__ | the output of netstat -an on my asterisk box only shows it listening on 5060 via udp? is this correct? |
| 05:13.42 | keith4__ | (as opposed to also listening on tcp:5060) |
| 05:14.08 | drmessano | Are you on 1.2 or 1.4? |
| 05:14.19 | keith4__ | uh... this one is still 1.2 |
| 05:14.28 | drmessano | No TCP until 1.6 |
| 05:14.36 | keith4__ | oh, ok |
| 05:16.17 | StephenF | where can I find a list of new features in 1.6? |
| 05:16.17 | StephenF | is there a wiki or changelist somewhere? |
| 05:18.28 | keith4__ | yah |
| 05:18.33 | keith4__ | ~wiki |
| 05:18.52 | keith4__ | hmm |
| 05:18.59 | drbrown | latter guys |
| 05:19.01 | keith4__ | shakes his fist at jbot |
| 05:19.51 | keith4__ | http://www.voip-info.org/wiki-Asterisk |
| 05:20.58 | StephenF | keith4__ ahh thanks |
| 05:26.22 | tzafrir_laptop | ~wikis |
| 05:26.30 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 05:26.35 | keith4__ | sighs |
| 05:26.36 | tzafrir_laptop | ~wiki wikipedia |
| 05:32.08 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 05:34.31 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 05:41.44 | Defraz | I installed Trixbox, and I know this is the trixbox channel but I have narrowed my issue down to asterisk and rxfax. my asterisk server seems to crash when a fax comes in. |
| 05:41.57 | Defraz | rxfax is called and blam seg fault. |
| 05:42.40 | Defraz | I have read everywhere, and on the forms people say it works but it isn't. Does anyone have a how to. |
| 05:52.00 | keith4__ | ~trixbox |
| 05:52.20 | jbot | i guess trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
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| 06:00.48 | jeev | heh |
| 06:06.50 | Defraz | haha |
| 06:07.14 | Defraz | figured that would be the answer. But Trixbox aside I am having trouble with asterisk and rxfax module |
| 06:07.26 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 06:07.30 | Defraz | and spandsp might be the issue or libtiff |
| 06:07.31 | drmessano^ | You're also dealing with an asterisk that wasn't compiled on your system |
| 06:07.47 | Defraz | I have tried recompiling it on my system actually |
| 06:07.49 | drmessano^ | @ whatever spec they decided to compile it at |
| 06:08.04 | Defraz | it won't compile right, with the proper Rxfax |
| 06:08.13 | drmessano^ | Probably not |
| 06:08.21 | Defraz | I don't think the patches that the forms reffer to is correct. |
| 06:08.28 | drmessano^ | Not really an asterisk problem |
| 06:08.32 | Defraz | I downloaded the source actually from asterisk.org |
| 06:08.36 | drmessano^ | Moreso the enviornment |
| 06:08.45 | Defraz | well I am not even using the trixbox config files now |
| 06:08.48 | Defraz | just asterisk. |
| 06:08.58 | drmessano^ | From the trixbox you downloaded |
| 06:09.15 | Defraz | No no, I just renamed the /etc/asterisk directory |
| 06:09.21 | Defraz | created a new one and when I did an asterisk install |
| 06:09.27 | Defraz | it created all the default config files |
| 06:09.31 | drmessano^ | Why dont you do a fresh OS load and try it |
| 06:09.36 | Defraz | and I renamed the modules |
| 06:09.39 | tzafrir_laptop | Defraz, rxfax is just a wrapper for spandsp |
| 06:09.45 | Defraz | it is like a vanilla asterisk install. |
| 06:09.52 | drmessano^ | heh, no it's not |
| 06:09.59 | tzafrir_laptop | First off, get a decent version of spandsp |
| 06:10.10 | Defraz | What version is the best |
| 06:10.23 | tzafrir_laptop | asterisk + asteirsk-devel should be good enough for building asterisk modules |
| 06:10.36 | Defraz | I have tried spandsp spandsp-0.0.4-22.pre15 and a 3.x version |
| 06:10.49 | Defraz | that is what I am using Tzafrir |
| 06:11.42 | Defraz | Drmessano for some reason is being difficult. I know how to extract the trixbox mess. And I did actually start with a fresh copy on a vm machine. the box is remote so I couldn't really reload the real box. |
| 06:11.58 | Defraz | can't test the pri on the vmware though |
| 06:12.00 | tzafrir_laptop | I'm using 0.0.5pre4 internally |
| 06:12.10 | Defraz | okay that might be the difference |
| 06:12.18 | Defraz | if I uninstalled the spandsp I am using |
| 06:12.18 | drmessano^ | I'm not being difficult, but you're using a shit unknown environment and insisting "oh no man, it's the same as a fresh load" |
| 06:12.21 | Defraz | and tyr that one. |
| 06:12.24 | drmessano^ | Don't insult my intelligence |
| 06:12.35 | Defraz | I am not, just not keeping an open mind. |
| 06:12.50 | Defraz | I know exactly what environment it is using. |
| 06:14.38 | Defraz | Thanks for the help tzafrir, that is what I will try next. Might even try a fresh install of CentOS just to rule out my "Unknown Environment" |
| 06:15.04 | Defraz | what version of libtiff are you using tzalfrir? |
| 06:20.42 | Defraz | off to try Tzafrir idea. |
| 06:20.57 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
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| 06:29.15 | drmessano | wonders how questioning ones use of a known flaky environment is considered "being difficult" |
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| 06:35.01 | *** part/#asterisk knobo (n=bohmersp@148.122.202.168) |
| 06:40.19 | jblack | drmessano: I wonder why questioning one's used of a limited editor is considered "elitism". =P |
| 06:40.48 | jblack | looks away and whistles an innocent tune |
| 06:42.55 | *** join/#asterisk drbrown (n=chatzill@rrcs-24-123-224-200.central.biz.rr.com) |
| 06:43.03 | jeev | there needs to be a cheaper method of KVM over IP in the market, single server use |
| 06:43.09 | jeev | around a hundred bux |
| 06:43.34 | drbrown | does the hpec echo canceler usually have distortion? |
| 06:43.40 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 06:44.00 | drbrown | or is it just my install? |
| 06:44.59 | jblack | jeev: ssh! |
| 06:45.30 | jblack | jeev: vnc! xvmcc! |
| 06:46.33 | jblack | no. xdmrpc.. xdmp |
| 06:46.35 | creativx | aten kh1516 |
| 06:46.36 | creativx | =) |
| 06:46.40 | jblack | something like that. |
| 06:48.29 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 06:49.41 | tzafrir_laptop | KVM? Xen? |
| 06:49.45 | jeev | jblack |
| 06:49.47 | jeev | KVM over IP dood |
| 06:49.50 | jeev | how is ssh kvm |
| 06:49.54 | jeev | that' not an oob solution |
| 06:50.14 | tzafrir_laptop | Get a system that supports LinuxBios |
| 06:51.19 | tzafrir_laptop | err.. renamed to coreboot |
| 06:54.17 | jeev | heh |
| 06:54.22 | jeev | or i'll get a real kvm |
| 06:54.28 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 07:04.45 | jblack | Linuxbios isn't linuxbios any more? |
| 07:05.10 | mosty | jeev, what kind of servers do you use? |
| 07:07.01 | jblack | ahhh, I see. |
| 07:13.13 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 07:20.21 | jeev | mosty, yougonna suggest IPMI ? |
| 07:20.48 | mosty | drac/elom/whatever |
| 07:21.14 | *** part/#asterisk ZefK (n=ZefK@wsc-fo.b.astral.ro) |
| 07:22.03 | jeev | i think it has IPMI support, i've notified the manufacturer asking. i messed up on a networking issue today that has got me thinking that i have to get this thing stable before i leave for london |
| 07:22.15 | jeev | the box i broke wasn't being used.. but it's 2 boxes at a single datacenter and i need perfection |
| 07:22.23 | jeev | i would've been happy with minicom access today. |
| 07:29.41 | *** join/#asterisk hi365_m (n=hi365@213.151.44.101) |
| 07:30.49 | jeev | sleep time |
| 07:30.49 | jeev | night |
| 07:31.06 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
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| 08:37.32 | marc7 | if I'm running something like: exten => _X.,1,Dial(IAX2/sonora_out/${EXTEN}) |
| 08:37.56 | marc7 | (so that the dial string is basically IAX2/sonora_out/2025551234 |
| 08:38.13 | marc7 | is there any way to then pass an argument or variable along for the opposite end to pick up? |
| 08:38.54 | marc7 | eg: dial "2025551234" with var/arg "John Smith" |
| 08:39.28 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
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| 09:12.52 | *** join/#asterisk brainopia (n=brainopi@d25k221.dialup.corbina.ru) |
| 09:14.07 | brainopia | i installed asterisk, but /etc/asterisk folder is empty, wtf? |
| 09:15.15 | ToTo | brainopia, make samples |
| 09:15.26 | brainopia | ToTo: thx |
| 09:16.42 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
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| 09:33.17 | gpkz8113 | Greetings guys, long time no visit. |
| 09:33.50 | gpkz8113 | Is there a simple way to hook up a dead end extension? (IE: I want to divert an incomming number to an extension that rings like normal but has no phone or endpoint on it |
| 09:34.43 | justdave | Freman: |
| 09:34.54 | justdave | exten => 1234,1,Wait(20) |
| 09:35.01 | justdave | exten => 1234,n,Hangup() |
| 09:35.22 | Freman | will that play the ring tone? |
| 09:35.40 | justdave | although I haven't actually tried that, I'm guessing because it didn't start with Answer() that it'll keep ringing |
| 09:35.45 | justdave | would have to try it to make sure though |
| 09:35.58 | Freman | nah, it's already made it through a privacy system so it's already answered |
| 09:36.10 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:36.12 | justdave | ah. |
| 09:36.23 | justdave | seems like there's an app that plays the ring tones |
| 09:36.40 | Freman | it's just one of those nusence callers who constantly rings for things he wants fixed and doesn't ever have money so... |
| 09:36.52 | Freman | I'd rather just know he called and call him back when I feel like it :) |
| 09:37.10 | Freman | hey, Playtones(dial) might do it |
| 09:37.21 | justdave | Ringing() |
| 09:37.32 | justdave | core show application Ringing |
| 09:41.36 | Freman | ah I have that good |
| 09:41.37 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:41.38 | Freman | thanks dave |
| 09:43.02 | *** part/#asterisk brainopia (n=brainopi@d25k221.dialup.corbina.ru) |
| 09:44.47 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:46.01 | jblack | My kid deserves a refund on her buffet. : http://gallery.linuxguru.net/cool-stuff/fortoon |
| 09:53.21 | Freman | ok, this has worked out REALLY REALLY cool |
| 09:54.05 | Freman | my privacy script handles a redirect flag, so I just redirect to Local/104@internal/n which is wait(100), it dials that as per normal and presto it just works (tm) |
| 09:54.17 | Freman | <PROTECTED> |
| 09:54.18 | Freman | hehe |
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| 10:27.06 | *** join/#asterisk gewuerzwiesel (n=gewuerzw@unaffiliated/gewuerzwiesel) |
| 10:27.39 | gewuerzwiesel | hello world :) |
| 10:31.16 | gewuerzwiesel | I need some help configuring capi with asterisk. I fixed all the kernelbugs and capi works (incoming calls are listed in the syslog) |
| 10:31.45 | gewuerzwiesel | so, if I "module load chan_capi.so" I get that warning: CAPI not installed, CAPI disabled! |
| 10:32.15 | *** join/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 10:35.08 | gsiener | Hi all. I installed asterisk-gui not long ago, and I started running into an issue where logging in results in an infinite loop of parsing config files. The only solution I've found via searching is to reinstall. Surely there's another option? |
| 10:38.05 | Freman | tthanks again dave |
| 10:41.11 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 10:42.57 | gewuerzwiesel | ok, now I get the incoming calls inside the console :) |
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| 10:58.49 | IsUp | hello |
| 11:00.01 | IsUp | i've just upgraded to Asterisk 1.4.21.2 on ubuntu 8.04 server |
| 11:00.14 | IsUp | i am getting too many <defunct> AGI processes |
| 11:01.22 | IsUp | i've tried perl and php. i've tried agi-test.agi after channel dies, process goes to <defunct> and restarting asterisk is the one solution. |
| 11:02.15 | IsUp | also i have a choppy playback on GSM files. my gcc version 4.2.3... my system was fine with older version of Asterisk. any ideas? |
| 11:02.19 | *** join/#asterisk angryuser (n=Miranda@128.237.146.195.dynamic.adsl.abo.nordnet.fr) |
| 11:06.34 | Blackvel | what is the worst thing to do with IVR programming? |
| 11:08.18 | Blackvel | what is the best way to do that asterisk 1.2.x recognices that the caller hangup already? It appears that the IVR is endlessly playing backgrounds on non-active channels (hangup gets recognized too late) |
| 11:08.58 | Blackvel | it may be the cause of the front patton isdn bri-sip gateway, but I am not sure |
| 11:09.35 | v4mp | O_O |
| 11:15.14 | *** join/#asterisk zap0 (n=moofy@123-243-103-30.static.tpgi.com.au) |
| 11:16.55 | zap0 | hello, im interested in building a system to interpret tone dialing for use in an application.. from what i've seen of asterisk, it can do it, but it'll take me quite some time to wade thru all the off-topic info. |
| 11:17.04 | zap0 | can anyone suggest a simpler solution? |
| 11:19.04 | *** join/#asterisk XnOSX (n=XnOSX@212.145.173.80) |
| 11:19.15 | Blackvel | zap0: what do you mean in detail? |
| 11:19.26 | Blackvel | tone dailing for use in an application? what does that mean? |
| 11:20.05 | zap0 | i just want to capture a phones' 1-9 keys being pressed.. i dont need PBX features, or even voice response. |
| 11:20.18 | zap0 | just want the keypresses.. passed to another applcaition. |
| 11:20.45 | Blackvel | depeinding on 1-9 key presses you want to launch different applications? |
| 11:20.47 | Blackvel | what application? |
| 11:21.32 | zap0 | my application. |
| 11:23.04 | zap0 | i just want the key values passed to my application. |
| 11:23.42 | *** join/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 11:24.03 | Blackvel | yeah but what? php? perl? |
| 11:24.30 | Blackvel | should astersk spawn a new process with your php/perl/script application on key press 1? |
| 11:24.42 | zap0 | how is that important? just about any interface i can deal with. |
| 11:25.12 | zap0 | it should pass the value '1' to my application via whatever method is easiest/works. |
| 11:25.27 | zap0 | is output to a php script possible. |
| 11:25.35 | zap0 | ? that would be ideal. |
| 11:25.40 | zap0 | i can write php. |
| 11:25.47 | IsUp | then try using phpagi, lol |
| 11:26.12 | Blackvel | if its an external application you have to use AGI afaik |
| 11:26.58 | Blackvel | different languages are supported by different interfaces, etc FastAGI Java |
| 11:27.35 | Blackvel | but if there is an simpler solution dunno |
| 11:27.36 | zap0 | im looking for answers that dont involve * also... |
| 11:27.49 | zap0 | perhaps * is overkill for what i need |
| 11:27.54 | Blackvel | could be |
| 11:28.35 | Blackvel | maybe the US guys join once they are awake/starting to work :) |
| 11:28.57 | UnixDawg | well untill you say what you need/want non of us can help you |
| 11:28.57 | zap0 | what time is that typically? (hrs from now) ? |
| 11:29.24 | UnixDawg | 7am on the east now |
| 11:29.25 | zap0 | UnixDawg, 3rd time explaining: i want to capture just the keypresses of a phone |
| 11:29.40 | IsUp | zap0, 3rd time saying |
| 11:29.42 | IsUp | use AGI |
| 11:29.44 | zap0 | i dont care for voice, or PBX functions. |
| 11:29.47 | IsUp | and dont be a jerk |
| 11:29.51 | UnixDawg | thast easy |
| 11:29.55 | UnixDawg | dood |
| 11:30.20 | UnixDawg | you trying to record numbers dialed |
| 11:30.20 | Blackvel | you are europe? 13:30pm (cet / gmt+1)? US is behind some hours |
| 11:30.57 | Blackvel | http://www.voip-info.org/wiki-Asterisk+AGI |
| 11:31.05 | zap0 | UnixDawg, i want the numbers passed to my application real-time (within a second~) |
| 11:31.45 | IsUp | zap0, this is last time i am saying |
| 11:31.49 | IsUp | use AGI |
| 11:31.51 | IsUp | ok? |
| 11:32.45 | zap0 | IsUp, you seem rather irratic... chill out. |
| 11:32.46 | Blackvel | you could check if all the ALL-IN asterisk solutions/ web gui solutions support AGI as well as |
| 11:32.49 | Blackvel | I doubt that |
| 11:33.25 | Blackvel | otherwise...if you have a business its easy to find the right asterisk consultant here in this channel |
| 11:33.30 | zap0 | im in au GMT+10. its 9:30pm |
| 11:33.32 | Blackvel | he can setup a solution which just works |
| 11:33.59 | UnixDawg | zap you looking to use a phone to control something like mr house |
| 11:35.07 | UnixDawg | zap stop listen |
| 11:35.25 | UnixDawg | and go into detail at what you want the dtmf to do . |
| 11:35.36 | UnixDawg | what does the application control |
| 11:35.50 | UnixDawg | how long will it need to be offhook to work it |
| 11:36.04 | UnixDawg | there is alot of info your not telling us to help you |
| 11:36.24 | UnixDawg | and I bet you would have to write a agi to control it |
| 11:36.26 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 11:36.54 | Blackvel | he cant "just pass" dtmf digits/control flow to any application afaik |
| 11:37.11 | Blackvel | the application has to follow the AGI/FastAGI interface |
| 11:37.13 | UnixDawg | you can with system calls |
| 11:37.25 | UnixDawg | but a agi might be best |
| 11:37.37 | Blackvel | UnixDawg: right, but that spawns a separate application process |
| 11:37.45 | UnixDawg | but eunless he elaborates and evplains better it does no help |
| 11:37.49 | Blackvel | but he has an existing running external application |
| 11:38.00 | UnixDawg | ok |
| 11:38.12 | Blackvel | he's just afraid of asterisk intro/setup |
| 11:38.33 | Blackvel | because of overkill (which might be true) |
| 11:39.03 | UnixDawg | you can dumb down asterisk to a minimum |
| 11:39.07 | Blackvel | there are so many consultants out to get easily / in time started with a running * solution |
| 11:39.15 | Blackvel | and just to focus on the agi thing |
| 11:39.23 | UnixDawg | yeah |
| 11:39.34 | Blackvel | noone has to go the 5-10 days setup learing process by himself |
| 11:42.57 | *** join/#asterisk Squeeb (i=squeeb@eggwee.co.uk) |
| 11:43.18 | zap0 | ok, i've found this wait for digit <timeout> thats a 'command' is that part of the * scripting language? |
| 11:44.35 | IsUp | zap0 |
| 11:44.40 | IsUp | go and read book |
| 11:46.13 | Blackvel | zap0: yes |
| 11:46.26 | Blackvel | zap0: are you private or do you have a business? |
| 11:46.37 | IsUp | hes got a business |
| 11:46.43 | zap0 | private at the moment.. |
| 11:46.52 | IsUp | and his boss want something from him |
| 11:46.54 | zap0 | im thinking of using it in an art project. |
| 11:46.57 | IsUp | and he is trying to resolve |
| 11:47.01 | IsUp | lol |
| 11:47.10 | zap0 | IsUp, what is your problem? You have nothing better to do but troll? |
| 11:47.24 | IsUp | ah well, you are just funny |
| 11:47.31 | UnixDawg | ok kids |
| 11:47.33 | zap0 | you are just pathetic. |
| 11:47.40 | UnixDawg | zap you need to go read |
| 11:47.44 | IsUp | you are asking a question and i am trying to help |
| 11:47.58 | IsUp | use AGI, read book, go to voip-info.org |
| 11:48.05 | UnixDawg | <PROTECTED> |
| 11:48.37 | UnixDawg | you need to go read about asterisk and look on voip-info.org |
| 11:48.45 | UnixDawg | it will help for the most part |
| 11:49.29 | Squeeb | Hello, I'm having some odd problems with incoming sip calls. I'm using the digium web manager and I've added an incoming rule for _X. but when I ring the sip number it says "extension 's' can not be found" |
| 11:49.35 | Squeeb | I thought _X. would match |
| 11:50.20 | IsUp | can you paste your extension please? http://pastebin.ca/ |
| 11:50.58 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 11:50.58 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 11:51.11 | zap0 | im more familiar with C, of the list of C interfaces on http://www.voip-info.org/wiki/view/Asterisk+AGI which is mature? |
| 11:51.13 | IsUp | zap0 |
| 11:51.16 | IsUp | take a look here: http://pastebin.ca/1213532 |
| 11:51.22 | IsUp | i made an example for you |
| 11:51.38 | IsUp | if caller press "5" it'll execute php script |
| 11:51.40 | *** part/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 11:51.43 | Squeeb | IsUp: http://pastebin.ca/1213533 |
| 11:52.23 | Squeeb | that DID_8676419 thing is weird |
| 11:52.31 | Squeeb | Shouldn't it be included in the DLPN_DialPlan1 ? |
| 11:52.52 | IsUp | what's your sip users context? |
| 11:53.41 | zap0 | do have to duplicate the entire script for each digit or modify this single script? |
| 11:53.57 | Squeeb | My sip users context is simply "sipgate" |
| 11:54.28 | IsUp | zap0 take a look here: http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf |
| 11:54.33 | IsUp | you can write an if condition |
| 11:55.31 | IsUp | well Squeeb, i cant see your "sipgate" context is defined |
| 11:55.44 | Squeeb | hmm |
| 11:55.45 | IsUp | did you paste just your extensions? or your extensions.conf? |
| 11:55.51 | Squeeb | extensions.conf |
| 11:55.56 | Blackvel | do you guys think its O.K for IVR programming to route back to the current background menu once the timeout/waitexten timeout finished? |
| 11:56.02 | Squeeb | did you want dialplan show output? |
| 11:56.16 | Blackvel | like user didn't choose and option, timeout 5 secs, play again the menu |
| 11:56.32 | Blackvel | of course this will endless loop if the user is not going to choose any menu option |
| 11:56.45 | IsUp | i think, it should be limited to 3 loops |
| 11:56.47 | IsUp | Squeeb |
| 11:56.50 | Blackvel | this might be the case with my endless background loop even the user already hangup |
| 11:56.52 | IsUp | simply do that |
| 11:57.03 | IsUp | edit your sip user |
| 11:57.06 | IsUp | context = DID_8676419_default |
| 11:57.17 | *** join/#asterisk zydoon (n=zydoon@41.225.157.209) |
| 11:57.18 | Squeeb | aah |
| 11:57.20 | Blackvel | so at the end of the 3 loops it should go to Hangup instead of background again |
| 11:57.31 | *** part/#asterisk zydoon (n=zydoon@41.225.157.209) |
| 11:57.31 | IsUp | and test your extension |
| 11:57.41 | Squeeb | Oh |
| 11:57.44 | IsUp | yeah it should hangup |
| 11:57.45 | Squeeb | turns out it exists |
| 11:57.49 | Squeeb | 8676419] |
| 11:57.50 | Squeeb | context = DID_8676419 |
| 11:57.50 | IsUp | it's my opinion |
| 11:57.51 | Squeeb | :( |
| 11:57.55 | Blackvel | i like that idea (stops endless playing problem on dead channels) |
| 11:57.56 | *** join/#asterisk propellerhead (n=yogurt2u@host113.190-136-116.telecom.net.ar) |
| 11:58.03 | Blackvel | how do you code this loop 3 times thing? |
| 11:58.19 | IsUp | well, i prefer using PHP on my AGI scripts |
| 11:58.25 | IsUp | for example write a function with |
| 11:58.43 | Blackvel | oh... will this system isn't AGI :( |
| 11:58.56 | Blackvel | just extension.conf with background, waitexten, etc. |
| 11:59.03 | IsUp | playMenu($count), do $count = 0 first, if user not going to choose anything then count +1. if count = 3, hangup |
| 11:59.13 | IsUp | ah, it's not good for ivr programming i think. |
| 11:59.33 | IsUp | Squeeb |
| 11:59.36 | Blackvel | was there any do/for/while loop in asterisk extensions programming? |
| 11:59.49 | IsUp | can you paste your user? mask your passwords. and which number you are trying to dial? |
| 11:59.58 | IsUp | i have no idea. |
| 12:00.06 | IsUp | but you can make something like |
| 12:00.21 | IsUp | t,1,goto(xxx) |
| 12:00.24 | IsUp | or something like that |
| 12:00.38 | IsUp | there was a "t" priority in extension if i am not wrong |
| 12:01.09 | Blackvel | IsUp: yes there is 't' |
| 12:01.14 | IsUp | and you can make set(count=1) |
| 12:01.30 | IsUp | when users reach to "t" again, test if users count equals to 3 |
| 12:01.42 | IsUp | if its not, then count + 1 |
| 12:01.46 | Blackvel | well yes, probably have to introduce set/gotoif count loop construct |
| 12:01.55 | Squeeb | IsUp: http://pastebin.ca/1213536 |
| 12:02.59 | IsUp | well Squeeb |
| 12:03.08 | IsUp | you are sending user to (default|6000|1) |
| 12:03.20 | IsUp | but theres no "6000" |
| 12:03.27 | Squeeb | oh sorry, there's also a 6000 |
| 12:03.31 | v4mp | hm wehere have i seen Squeeb before |
| 12:03.44 | Squeeb | I thought you meant just the context for the trunk sip |
| 12:04.15 | *** join/#asterisk [gnubie] (n=gnubie@cm141.omega112.maxonline.com.sg) |
| 12:04.15 | IsUp | well, Squee. let's try that. under your DID_8676419_default |
| 12:04.19 | [gnubie] | waves |
| 12:04.29 | IsUp | _X.,1,MusicOnHold() |
| 12:04.59 | IsUp | and 'dialplan reload' see what you get |
| 12:05.21 | [gnubie] | i'm having a problem with an outbound call to a pots line from a sip phone.. the callee cannot hear me at all but the caller can hear the callee's voice |
| 12:05.29 | Squeeb | ok reloaded ok, calling now |
| 12:05.38 | [gnubie] | i'm running asterisk 1.4.21.2 here.. |
| 12:05.47 | Squeeb | Sep 29 13:05:40] NOTICE[46487]: chan_sip.c:14035 handle_request_invite: Call from '8676419' to extension 's' rejected because extension not found. |
| 12:05.55 | [gnubie] | that's the only scenario that has a problem with me |
| 12:06.14 | IsUp | well try: |
| 12:06.18 | IsUp | _.,1,MusicOnHold() |
| 12:06.19 | jblack | squeeb: You sent the call into a context that doesn't have extension s. |
| 12:06.34 | [gnubie] | anyone has a similar problem with what i'm experiencing now? |
| 12:07.03 | jblack | squeeb: Look at the sip debug to see what context it's trying to go into, and make an s extension. |
| 12:07.10 | jblack | [gnubie]: sounds like typical firewalling issues to me. |
| 12:07.12 | jblack | ~sipnat |
| 12:07.13 | 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 |
| 12:07.21 | v4mp | jblack, my provider asked if i was behind a nat for server after i had already said there server and mine connect fine of which they wouldn't behind a nat |
| 12:07.34 | jblack | v4mp: Leave me a lone. |
| 12:07.35 | Squeeb | Hmm that's weird |
| 12:07.39 | Squeeb | that _. just rings |
| 12:07.42 | Squeeb | but no onhold music |
| 12:07.47 | v4mp | jblack, im not asking for ur help lol |
| 12:07.58 | Squeeb | jblack: I'll have a look now |
| 12:07.58 | [gnubie] | jblack: i don't think firewall is involve on this scenario |
| 12:08.08 | jblack | v4mp: Did I say you were? I just want you to leave me alone. |
| 12:08.17 | v4mp | ahh ok |
| 12:08.35 | jblack | Squeeb: btw, dont' use exten => _. It has nasty side effects, most of which I'm not aware of. I just know it gets you sent to asterisk hell. |
| 12:08.43 | Squeeb | yea |
| 12:08.47 | Squeeb | it warned me when I reloaded |
| 12:08.51 | Squeeb | worth it for diagnostics |
| 12:08.52 | [gnubie] | [sip_phone] ==sip==> [asterisk] ==pots==> [analog_phone] |
| 12:09.04 | Squeeb | what abut |
| 12:09.07 | Squeeb | exten = s,1,MusicOnHold() |
| 12:09.26 | jblack | That would be perfect, if you put it in the context that maches the account in sip.conf. :) |
| 12:09.48 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 12:09.59 | jblack | in sip.conf, in the account, you should have "context=something". exten => s,1,MusicOnHold() should be in something. |
| 12:10.10 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.8) |
| 12:10.12 | Squeeb | yea |
| 12:10.24 | Squeeb | well in the [something |
| 12:10.27 | Squeeb | pft can't type |
| 12:10.37 | Squeeb | well in the [something] context there's another include to ANOTHER context |
| 12:10.43 | Squeeb | not sure what the web manager was trying to do there |
| 12:12.48 | Squeeb | I'm still unsure of the s and _X. things |
| 12:13.11 | Squeeb | but I don't understand why I get the error "call from 8676419' to extension 's' rejected" .. I thought that _X. would match any number |
| 12:13.18 | Squeeb | as does s.. as far as I understand |
| 12:13.49 | Squeeb | Looking for s in DID_8676419 (domain 192.168.168.6) |
| 12:13.55 | Squeeb | from sip debug |
| 12:15.29 | Squeeb | [ Context 'DID_8676419' created by 'pbx_config' ] |
| 12:15.29 | Squeeb | <PROTECTED> |
| 12:15.29 | Squeeb | <PROTECTED> |
| 12:15.46 | Squeeb | So anything destined for "s" in DID_8676419 should Dial my extension right?? |
| 12:15.48 | *** join/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 12:15.51 | Squeeb | whoops.. random ? there. |
| 12:16.09 | Squeeb | woo |
| 12:16.10 | Squeeb | that worked |
| 12:16.16 | Squeeb | how random |
| 12:16.26 | jblack | Thanks for following basic instructions. |
| 12:16.42 | jblack | You're 1 up on 1/2 the people that come in here. =) |
| 12:16.44 | UnixDawg | core show brain |
| 12:16.55 | UnixDawg | core show websites |
| 12:16.58 | jblack | module not loaded. |
| 12:17.15 | [gnubie] | anyone here ever encountered my problem? |
| 12:17.20 | Squeeb | although now I'm getting one way audio .. *sigh* |
| 12:17.21 | UnixDawg | ? thebook |
| 12:17.26 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:17.28 | jblack | Squeeb: Firewall. |
| 12:17.31 | jblack | ~sipnat |
| 12:17.31 | 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 |
| 12:17.33 | Squeeb | checked |
| 12:17.41 | Squeeb | will check again |
| 12:17.46 | gsiener | hi all -- has anyone seen any asterisk tutorials that walk through using the gui? |
| 12:17.51 | [gnubie] | jblack: again, i don't think nat is the problem.. |
| 12:17.53 | [gnubie] | [sip_phone] ==sip==> [asterisk] ==pots==> [analog_phone] |
| 12:18.02 | UnixDawg | the gui is 2.0 and not yet documented |
| 12:18.03 | [gnubie] | that is the call flow |
| 12:18.06 | *** join/#asterisk lvl- (n=lvl@145.52.248.101) |
| 12:18.13 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:18.35 | jblack | [TK]D-Fender: Good morning. Glad you're here. [gnubie] is due for a larting |
| 12:19.02 | [gnubie] | the sip phone is connected inside the lan |
| 12:19.15 | [TK]D-Fender | jblack: "If it is weak, kill i or ignore it. Anything else honors it" - Volrath |
| 12:19.21 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 12:19.30 | [TK]D-Fender | it* |
| 12:19.39 | jblack | [gnubie]: And.... did you turn off redirect? If not, again, firewall. |
| 12:19.40 | Squeeb | weird issue |
| 12:19.52 | Squeeb | if I make a call, I can hear audio both ways, if I call in, I can only hear incoming audio |
| 12:20.01 | Squeeb | *reads suggested document* .. |
| 12:20.05 | [gnubie] | redirect? |
| 12:20.15 | jblack | ~book |
| 12:20.15 | jbot | hmm... book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 12:20.40 | jblack | is ready to go postal |
| 12:20.47 | [gnubie] | jblack: [sip_phone] ==sip_over_the_lan==> [asterisk] ==pots_via_tdm_fxo_card==> [analog_phone] |
| 12:20.52 | *** join/#asterisk bbryant (n=Brett_Br@adsl-159-32-97.flo.bellsouth.net) |
| 12:21.14 | jblack | [gnubie]: stop humping my leg. |
| 12:22.01 | *** join/#asterisk bbryant (n=Brett_Br@adsl-159-32-97.flo.bellsouth.net) |
| 12:22.40 | *** join/#asterisk bbryant (n=Brett_Br@adsl-159-32-97.flo.bellsouth.net) |
| 12:23.22 | gewuerzwiesel | which codec do I need to get the PSTN to SIP connection? I get these Warning atm: ...Unable to find a codec translation path from unknown to gsm... |
| 12:23.47 | [TK]D-Fender | grabs jblack, stamps him and tosses him into the back of a non-descript brown van. |
| 12:23.48 | *** join/#asterisk write_erase (n=Olivier@goodgw.m2m-fleet.com) |
| 12:24.01 | Squeeb | aha |
| 12:24.09 | Squeeb | RTP ports are too wide a range |
| 12:24.21 | jblack | Yeah, there are pretty obnoxious. |
| 12:24.29 | Squeeb | although that's weird as I have them set to 10000:20000 AND on the router |
| 12:24.34 | Squeeb | yet it's using 65something |
| 12:24.57 | jblack | I'd knock them down to about 100... 2.2*max calls you want to support. |
| 12:25.09 | *** join/#asterisk dwagner (n=dwagner@195.202.166.182) |
| 12:25.10 | Squeeb | yea |
| 12:25.12 | write_erase | Hi... I have an exten that connect to a SIP provider (exten => _0.,1,Dial(SIP/${EXTEN:1}@SIP_OUT1,,tT) . How can I add a second SIP provider if the 1st one is unavailable, or already busy ? |
| 12:25.37 | IsUp | s-BUSY |
| 12:25.43 | jblack | write_erase: exten => _0.,n,Dial(SIP/${EXTEN:1}@SIP_OUT2,,tT) |
| 12:25.46 | [gnubie] | jblack: kindly take a look at this => http://paste.debian.net/18178/ |
| 12:26.10 | write_erase | jblack, I just a second n(next) exten ? |
| 12:26.25 | jblack | [gnubie]: And? |
| 12:26.26 | [TK]D-Fender | write_erase: Any failure will send it continuing on in the dialplan |
| 12:26.44 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:26.45 | write_erase | [TK]D-Fender, oh OK! as simpls as that ! |
| 12:26.55 | [TK]D-Fender | write_erase: "core show application dial" <- read its instructions and the vars you can check if you care why it didn't go through |
| 12:27.04 | jblack | [gnubie]: Let me help you out in a different languege. "El Firewallo es tu problemo" |
| 12:27.27 | [TK]D-Fender | IsUp: tell me that wasn't a response to write_erase's question.... |
| 12:27.39 | jblack | [gnubie]: Your answer is in ~sipnat. really. |
| 12:27.54 | [TK]D-Fender | jblack: He's only been pointed to it time and again. |
| 12:28.08 | jblack | [TK]D-Fender: I've told him something like 4 times this morning myself. |
| 12:28.14 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 12:28.29 | [gnubie] | jblack: http://paste.debian.net/18178/ is the output of my firewall |
| 12:28.35 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 12:28.58 | dwagner | is there a chan_sip.c programmer? |
| 12:28.58 | Squeeb | hmmmmm |
| 12:29.06 | Squeeb | it seems rtp.conf is ignoring me |
| 12:29.09 | Squeeb | anybody any ideas why? |
| 12:29.14 | jblack | [gnubie]: Which I looked at, which told me that you're not following all of the instructions in ~sipnat. Now leave me alone. |
| 12:29.34 | jblack | Squeeb: restart *. Or see if you can find the module to reload for rtp. |
| 12:29.35 | angryuser | jblack : who cares about port range ? its the same daemon |
| 12:29.40 | IsUp | [TK]D-Fender: actually it was :) i thought if BUSY or CHANUNAVAIL returns |
| 12:30.12 | jblack | angryuser: The other daemons on the system care, as does any port forwarding router with limited memory or lots of client machines. |
| 12:30.13 | Blackvel | how can I speed up asterisk hangup detection with a sip patton gw? e.g the snom phones ring, I hangup the pstn call, it keeps ringing far too long |
| 12:31.02 | Blackvel | is this hangup detection only on asterisk side or must it be on the patton isdn bri gw side? what command must patton send to asterisk so it hangs up the channel? |
| 12:31.06 | Squeeb | Is there any way from the asterisk console to show what RTP port range is used? |
| 12:31.09 | Blackvel | what sip command |
| 12:31.12 | [TK]D-Fender | IsUp: No, you are showing a cut&paste grade answer based on some piece of dialplan only you feel necessarily applies to him. Never assume any macro sample you've ever just dropped into your dialplan out of a book or WIKI page is being used by anyone else in any manner. |
| 12:31.13 | angryuser | jblack : a router not capable all ports to any directions is a piece o junk |
| 12:31.29 | jblack | angryuser: attempting to forward 10K ports can crash certain natting comcast modems. I wouldn't be surprised if you could give a linksys an aneurism as well. Not to mention that the default range is in the same as dynamically assigned ports on many operating systems. |
| 12:31.37 | angryuser | to route* |
| 12:31.46 | IsUp | okay Fender, sorry =) |
| 12:31.46 | jblack | angryuser: Weren't you asking me a question? |
| 12:32.06 | jblack | If you didn't want the answer, why did you ask? |
| 12:32.15 | [TK]D-Fender | jblack: I really can't imagine why forwarding that many ports would be a problem. passing traffic on all of them sure I guess, but sitting idle & waiting? |
| 12:32.24 | angryuser | jblack : no |
| 12:32.56 | jblack | So, you didn't say "jblack : who cares about port range ?" ? |
| 12:33.04 | jblack | Aww, screw it. plonk |
| 12:33.15 | dwagner | is there a problem in the 1.4.21.2 ? can it be used in a production environment? |
| 12:33.38 | jblack | [TK]D-Fender: Because some of them have a naive implementation, where they try to make 10K individual forwarding rules. |
| 12:33.38 | angryuser | it was rhetorical but whatever |
| 12:34.11 | [TK]D-Fender | jblack: Ok, completely stupid idea, but comprehensible. |
| 12:34.31 | *** part/#asterisk dwagner (n=dwagner@195.202.166.182) |
| 12:34.36 | IsUp | well, i have a really big trouble with 1.4.21.2. after i've upgraded my servers to 1.4.21.2 i am getting hundreds of AGI defunct proccesses. |
| 12:34.38 | jblack | In fact, I think netfilter has the same problem too. One can't forward a range to a range, so you end up with a mongo pile of rules. :( |
| 12:34.40 | *** join/#asterisk dwagner (n=dwagner@195.202.166.182) |
| 12:34.50 | [TK]D-Fender | Yup... he waited long.. |
| 12:34.52 | jblack | probably you can, but I'm not smart enough. |
| 12:35.07 | IsUp | ive tried Perl or PHP but AGI is going to zombie process after channel closes |
| 12:35.14 | [gnubie] | jblack: i checked this one => http://www.aocomputing.net/?p=3 and i have a similar setup already |
| 12:35.16 | [TK]D-Fender | dwagner: IsUp>well, i have a really big trouble with 1.4.21.2. after i've upgraded my servers to 1.4.21.2 i am getting hundreds of AGI defunct proccesses. |
| 12:35.22 | jblack | [gnubie]: plonk |
| 12:35.23 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:35.48 | write_erase | [TK]D-Fender, do you have a working configuration for mp3 moh ? My moh starts playing for a couple of seconds only, then silent .... |
| 12:35.58 | IsUp | also i have troubles with GSM playback. now i am just using ulaw format. |
| 12:36.10 | jblack | write_erase: Calls work otherwise? |
| 12:36.11 | IsUp | i've read bugtracker, its about with GCC version or something. i am not sure. |
| 12:36.12 | [TK]D-Fender | IsUp: thats usually ... |
| 12:36.16 | [TK]D-Fender | ~centos52bug |
| 12:36.17 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889, or trouble compiling x86_64 packages? Report a bug to CentOS! ".i386 packages should not satisfy dependencies for .x86_64 packages." |
| 12:36.18 | [TK]D-Fender | ^^^ |
| 12:36.23 | [TK]D-Fender | oops... |
| 12:36.23 | write_erase | jblack, yeah... everything fine... |
| 12:36.25 | [TK]D-Fender | wrong one. |
| 12:36.33 | jblack | write_erase: literally mp3s? |
| 12:36.35 | Blackvel | is there anything I can do about HANGUP detection on ANSWERED asterisk channels? |
| 12:36.37 | [TK]D-Fender | ~gsmbug |
| 12:36.37 | jbot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 |
| 12:36.40 | [TK]D-Fender | ^^^ |
| 12:36.52 | IsUp | by the way, i am using Ubuntu 8.04 server with kernel 2.6.24-16-server |
| 12:37.04 | [TK]D-Fender | write_erase: Does it really start playing them? |
| 12:37.16 | IsUp | hmm thank you Fender. what about AGI proccesses? any idea? |
| 12:37.23 | jblack | write_erase: If so, try using the external player example. The built in mp3 codec is a bit... well, football helmet wearing, short bus riding, twit. |
| 12:37.23 | IsUp | they are staying as "zombie" |
| 12:37.25 | [TK]D-Fender | IsUp: Nope. |
| 12:37.29 | write_erase | YEs starts playing, then silence... |
| 12:37.36 | IsUp | i've tried 'agi-test.agi' too to see if its about my script or not |
| 12:37.38 | IsUp | but its failed too |
| 12:37.50 | [TK]D-Fender | write_erase: make sure they are non VBR (highly suspect), and have no ID3 tags |
| 12:37.52 | angryuser | write_erase : your mp3 file got fixed bitrate ? |
| 12:38.20 | jblack | write_erase: For the least, it can get tripped up by tags. I think it's limited on bit rates. |
| 12:38.20 | jblack | So, use the mpg123 example. |
| 12:38.20 | jblack | Or convert them to gsm. =) |
| 12:38.28 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 12:38.42 | IsUp | Fender |
| 12:38.49 | write_erase | jblack, is it working for you ? |
| 12:38.55 | jblack | Though most any music converted to gsm ends up sounding much like my ex-wife. Perhaps .wav, if you can afford the space. |
| 12:39.03 | IsUp | what's the difference between SVN 1.4 branche and tarball in Asterisk downloads? |
| 12:39.05 | write_erase | anyone has a working custom mp3 configuration ? |
| 12:39.12 | IsUp | should i use svn checkout to get latest 1.4? |
| 12:39.29 | jblack | using an external player worked for me, but I use gsm now. There's an example in the musiconhold.conf file. |
| 12:39.36 | yang | jblack: lol |
| 12:39.43 | [TK]D-Fender | IsUp: SVN branch is "at a point in time" which could be at any time. Release is just a specific point in SVN |
| 12:40.07 | IsUp | hmm. using SVN branch better? |
| 12:40.17 | jblack | yang: Yeah. Incomprehensible, scratchy, noise. Eeeeeaaaheeaaeee eeeaaa chchccccchhhhhrrhhheeee |
| 12:40.23 | [TK]D-Fender | IsUp: depends. |
| 12:40.29 | write_erase | jblack, what do you mean by 'I use gsm' ? is that a codec ? |
| 12:40.49 | jblack | write_erase: Yes. Look for "sox convert mp3 to gsm" |
| 12:41.07 | IsUp | well, i have just problems with GSM and AGI stuff. it's really bad. i am carrying high volume calls, about 15.000/day |
| 12:41.12 | IsUp | all is going to AGI |
| 12:41.32 | yang | write_erase: after sox converts the files for you to gsm it works well (yes its a codec) |
| 12:41.38 | [TK]D-Fender | LOL! Someone tried a trixbox hack against my webserver!!!! |
| 12:41.48 | write_erase | jblack, ok... then astrisk can read gms files natively ? or I need an external player again ? |
| 12:41.58 | jblack | ~book |
| 12:41.59 | jbot | extra, extra, read all about it, book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 12:42.02 | yang | [TK]D-Fender: yipeee, free VOIP long distance calls ! |
| 12:42.07 | jblack | Read the book. There's a section on music on hold. |
| 12:42.27 | jblack | starts loading his shotgun |
| 12:42.52 | jblack | [TK]D-Fender: What's the farthest north airport that you know of? |
| 12:43.55 | [TK]D-Fender | jblack: Civilian or military? |
| 12:43.59 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 12:44.00 | jblack | civilian |
| 12:44.25 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 12:44.27 | jblack | I'm gonna hop in a flight simulator and fly up to the north pole, and see if there's penguins I can chop up with the rotor blades. |
| 12:44.35 | [TK]D-Fender | jblack: Probably somewhere in Nunavut |
| 12:44.41 | jblack | thanks much |
| 12:44.47 | dwagner | i need help with chan_sip.c, is there anybody out there? |
| 12:44.59 | IsUp | Fender, last question. how can i apply this patch? and which patch should i use? |
| 12:45.03 | jblack | I bet "igloolik" is pretty far north. :) |
| 12:45.07 | [TK]D-Fender | jblack: Go over to Anchorage. You'll be able to see Putin.... And SANTA! |
| 12:45.18 | IsUp | 1.2-gsm-gcc4.2.patch, 1.4-gsm-gcc4.2.patch, 11243-maybe-asm.diff |
| 12:45.19 | Squeeb | Argh .. this is weird |
| 12:45.26 | [TK]D-Fender | dwagner: Ask a specific question, get a specific answer. |
| 12:45.35 | Squeeb | sometimes when I call into my asterisk machine, the caller can't hear anything I say until I put them on hold and release them again |
| 12:45.44 | jblack | Heh. I can buz Palin's hunting grounds. :) |
| 12:46.38 | dwagner | in the latest svn, i've the problem, that if i answer and transfer a call, the call will be dropped. if i transfer the call directly the transfer works. |
| 12:46.58 | yang | write_erase: beside the book you can get help from http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Musiconhold |
| 12:47.31 | [TK]D-Fender | dwagner: Please provide CLI output for both attempts including full SIP debug from beginning to end. Truncate nothing. |
| 12:47.43 | dwagner | ok |
| 12:47.59 | IsUp | Fender: how can i apply GSM patch? |
| 12:49.26 | *** join/#asterisk Linuturk (n=Linuturk@fluxbuntu/developer/Linuturk) |
| 12:52.53 | *** join/#asterisk datacompboy (n=datacomp@l64-89-222.cn.ru) |
| 12:53.03 | datacompboy | Hi all! :) Anybody there? |
| 12:53.30 | slima | no, no one. |
| 12:53.31 | datacompboy | I'm still with problem -- unable to dial to numbers with letters in it (A-D) |
| 12:53.34 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 12:53.58 | datacompboy | :( one more day without any ideas... |
| 12:54.35 | [gnubie] | i didn't changed anything with my /etc/asterisk/*.conf in upgrading from 1.4.17 to 1.4.21.2 and now i'm havign this one way audio in placing an outbound call |
| 12:55.17 | IsUp | did you read UPGRADE.txt? |
| 12:55.21 | [gnubie] | my asterisk server is facing the internet and using a public ip |
| 12:55.39 | IsUp | there can be some configuration changes or dynamics |
| 12:55.55 | write_erase | yang, thx |
| 12:56.06 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 12:56.11 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:57.20 | [gnubie] | IsUp: honestly, i didn't.. i got my asterisk source from the debian unstable repository and built it as a .deb package |
| 12:58.25 | IsUp | ah, i dont recommend to do that. if you are just testing some stuff, you can try to use tarball. it would be better i think. |
| 12:58.28 | Blackvel | found the problem with asterisk hangup problem. patton isdn bri gw has to use "early disconnect". otherwise is does not hangup message to asterisk |
| 12:59.10 | IsUp | what you mean Blackvel? |
| 12:59.31 | Blackvel | [gnubie]: weird. I even run the nearly identical configuration with asterisk 1.2 and 1.4 and have no audio problems. |
| 12:59.47 | [gnubie] | IsUp: the asterisk source from the debian unstable is from the project upstream with some patches for debian |
| 13:00.05 | Blackvel | <Blackvel> is there anything I can do about HANGUP detection on ANSWERED asterisk channels? |
| 13:00.05 | Blackvel | thats the answer to my question |
| 13:00.11 | [TK]D-Fender | dwagner: www.pastebin.com |
| 13:00.42 | [TK]D-Fender | Blackvel: taht made no sense |
| 13:00.49 | [gnubie] | Blackvel: i am even surprised that this only happens now when a sip phone inside a lan places an outbound call to an analog phone via my tdm fxo card and the pots network |
| 13:00.55 | IsUp | gnubie, i can recommend to you read upgrade.txt for changes |
| 13:01.46 | IsUp | Fender, i don't know how to apply patch. Should i apply to tarball release or svn branch? i think branch already has updated code. and how can i apply? |
| 13:01.48 | dwagner | http://pastebin.com/pastebin.php?dl=m6e6a25a5 |
| 13:01.55 | [gnubie] | because, when an analog phone that is connected to my fxs port and calls an analog phone on the pots network via the tdm fxo port, i don't have any problem |
| 13:02.28 | dwagner | call 12 -> 13, 13 answer, transfer to 14, hangup |
| 13:02.34 | *** part/#asterisk datacompboy (n=datacomp@l64-89-222.cn.ru) |
| 13:02.47 | [gnubie] | IsUp: ok.. |
| 13:03.12 | Blackvel | [TK]D-Fender: that made no sense? |
| 13:03.34 | Blackvel | its the solution to my problem which noone answered?! |
| 13:04.08 | [TK]D-Fender | dwagner: Bad link. Try pastebin.ca |
| 13:05.42 | [gnubie] | other than this one way audio problem, i also have a problem with an inter-asterisk trunking via iax2 with encryption=yes that after sometime during a call session, both parties cannot hear each other already.. when i checked the cli, it says something like a problem with the decryption or something |
| 13:05.45 | Squeeb | I've found a bug in the digium web admin |
| 13:05.46 | dwagner | http://www.pastebin.ca/1213580 |
| 13:05.59 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:06.07 | Squeeb | when you add an incoming call rule it puts exten = BLAH and not exten => BLAH |
| 13:07.07 | *** join/#asterisk wiscados (n=mint@81.25.184.155) |
| 13:07.41 | *** join/#asterisk feeds (n=feeds@85-135-232-6.adsl.slovanet.sk) |
| 13:07.43 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 13:08.17 | *** join/#asterisk c4t3l (n=root@74.95.210.124) |
| 13:08.25 | c4t3l | hello world |
| 13:11.46 | dwagner | here is the trace: http://www.pastebin.ca/1213580 |
| 13:12.46 | *** join/#asterisk _Roman (n=roman@87.254.78.150) |
| 13:13.39 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 13:13.50 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 13:13.53 | IsUp | [TK]D-Fender: can you help me about patch? |
| 13:14.12 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 13:15.03 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 13:15.50 | [TK]D-Fender | IsUp: Nope. |
| 13:17.35 | IsUp | whys that? |
| 13:17.57 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 13:19.37 | Blackvel | weird. DB SET does NOT survice asterisk stop now |
| 13:19.57 | Blackvel | survive |
| 13:20.17 | [TK]D-Fender | IsUp: Because I've only ever applied one patch ever. "man patch" <- I'm pretty sure this is a very basic thing. |
| 13:20.33 | Blackvel | when I start Asterisk again with asterisk -vvc -C /etc/asterisk_dev/asterisk.conf the DB SET variable is not there |
| 13:20.46 | IsUp | ah, patch diff stuff |
| 13:20.48 | IsUp | understood =) |
| 13:20.58 | [TK]D-Fender | Blackvel: Could it be that your other instance si not lokoing at the same AstDB file? |
| 13:21.03 | IsUp | i am just not sure about applying patch to tarball release or svn branch |
| 13:21.26 | _Roman | <PROTECTED> |
| 13:21.31 | Blackvel | thought I changed that (for dev instance) |
| 13:21.40 | [TK]D-Fender | dwagner: You have masked things where I have expressly told you not to. I cannot help you. |
| 13:21.43 | Blackvel | but when my snom calls 401 which does Set(DB...) |
| 13:21.45 | Blackvel | its working |
| 13:22.29 | Blackvel | so once its set again (after Asterisk startup), its working fine... the GotoIf just works with DBGet |
| 13:23.08 | Blackvel | must be config problem then. you all told me that DB variables overlive asterisk shutdown (as of file system db) |
| 13:23.30 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 13:23.33 | Katty | good morning!!! |
| 13:23.42 | IsUp | morning |
| 13:23.43 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 13:23.50 | [TK]D-Fender | Blackvel: they are stored in astdb (actual file) and if you have multiple configs set up you might not be pointing to the same one on some other instance. |
| 13:23.55 | [TK]D-Fender | Katty: Mew. |
| 13:23.59 | Katty | [TK]D-Fender: woof :> |
| 13:24.02 | Katty | [TK]D-Fender: also |
| 13:24.10 | [TK]D-Fender | Katty: Pick up over the weekend? |
| 13:24.24 | Katty | [TK]D-Fender: you don't by chance know how to either make these polycom phones louder by default, or asterisk louder by default, do you? |
| 13:24.28 | Katty | [TK]D-Fender: yes, last night around 6pm |
| 13:25.05 | [TK]D-Fender | Katty: Typically your PSTN interface should be normalized. Everything else already is. |
| 13:25.38 | Katty | [TK]D-Fender: well they want it louder and i'm trying to appease them. |
| 13:25.50 | Blackvel | [TK]D-Fender: thank you so much for your tip! :) |
| 13:26.02 | [TK]D-Fender | Katty: They have volume controls on the phone... |
| 13:26.13 | Katty | [TK]D-Fender: that's not good enough |
| 13:26.20 | [TK]D-Fender | Katty: Even at max? |
| 13:26.29 | [gnubie] | please enlighten me.. this will be my last to ask about this question.. do i need to treat my sip phones inside my lan as nat=yes when the one way audio problem scenario is: internal_sip_phone ==lan==> asterisk ==fxo_tdm_pots==> analog_telephone |
| 13:26.34 | Katty | [TK]D-Fender: well apparently they don't want to have to crank it up each time. |
| 13:26.47 | Katty | [TK]D-Fender: but i do believe that cellphone to cellphone calls routed through the server at max are unacceptable |
| 13:26.52 | [TK]D-Fender | Katty: Then you should set the persistence in your provisioning. |
| 13:26.53 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:27.01 | [TK]D-Fender | Katty: sip.cfg |
| 13:27.15 | [TK]D-Fender | Katty: I have given you all you should need to correct this. happy hunting :) |
| 13:27.44 | Katty | [TK]D-Fender: i'm sure i will have more questions, but thanks for pointing me in the right direction (= |
| 13:28.14 | *** join/#asterisk stencil (n=stencil@76-10-151-37.dsl.teksavvy.com) |
| 13:28.23 | [TK]D-Fender | [gnubie]: Nat should not matter, check your firewall |
| 13:28.25 | [gnubie] | i am not talking about connection to the internet here |
| 13:29.06 | Katty | [TK]D-Fender: riddick is a bundle of paws and ears :> |
| 13:29.09 | Katty | [TK]D-Fender: he's so adorable |
| 13:29.27 | [gnubie] | [TK]D-Fender: what do i need to check? udp ports 5060 and 10000-10100 are accepted already |
| 13:29.28 | [TK]D-Fender | [gnubie]: meaningless. |
| 13:29.44 | [TK]D-Fender | [gnubie]: REMOVE it completely for testing |
| 13:30.49 | [gnubie] | [TK]D-Fender: ok.. |
| 13:31.02 | *** join/#asterisk af_ (n=getsmart@88-149-241-240.dynamic.ngi.it) |
| 13:31.04 | Katty | [TK]D-Fender: ah, so the persistant keeps the changes of previous? (= |
| 13:31.07 | Katty | [TK]D-Fender: this is very neat. |
| 13:31.24 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:31.28 | stencil | Katty: I would have thought you would have called Asterisk or maybe Obelix |
| 13:32.10 | stencil | your pet dog |
| 13:32.41 | Katty | stencil: Kaiser Riddick der Kleine Hobbit mit Waggytail |
| 13:33.29 | Katty | [TK]D-Fender: so what's all this voice.gain.rx.digital.handset="-15" stuff i'm seeing |
| 13:33.37 | *** join/#asterisk [netman] (n=netman@199.Red-83-38-222.dynamicIP.rima-tde.net) |
| 13:33.48 | [TK]D-Fender | Katty: Stuff not to touch unless absolutely necessary |
| 13:34.13 | Katty | [TK]D-Fender: so the goal is to try to get the settings on the phone to stay at max, and if that doesn't work to tweak the rx volume? |
| 13:34.37 | [TK]D-Fender | Katty: I've answered you twice on this. It isn't changing... |
| 13:35.12 | Katty | [TK]D-Fender: you do not parse. |
| 13:35.26 | Katty | [TK]D-Fender: if persistant does not tell the phone to retain it's volume settings |
| 13:35.29 | Katty | [TK]D-Fender: then what is it doing? |
| 13:35.35 | [TK]D-Fender | Katty: I told you which parameter to use, and what not to touch. What part of that doesn't parse? |
| 13:35.44 | Katty | [TK]D-Fender: the part where you tell me what it's doing (= |
| 13:35.51 | [TK]D-Fender | Katty: Do X, don't do Y. Simple as that |
| 13:35.57 | Katty | [TK]D-Fender: why why why |
| 13:35.59 | Katty | [TK]D-Fender: consider me 2! |
| 13:36.25 | Katty | [TK]D-Fender: i guess i'll try it and see! |
| 13:36.28 | [TK]D-Fender | Katty: Changing base gains can lead to sidetone issues, distortion, AEC probelms, etc |
| 13:36.40 | Katty | [TK]D-Fender: we're not talking about base gains |
| 13:36.50 | Katty | [TK]D-Fender: i'm talking about what voice.volume.persistant.handset="1" does. |
| 13:37.10 | [TK]D-Fender | Katty>[TK]D-Fender: so what's all this voice.gain.rx.digital.handset="-15" stuff i'm seeing <- you were |
| 13:37.21 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 13:37.27 | Katty | [TK]D-Fender: that was a side question :P |
| 13:37.43 | [TK]D-Fender | Katty: and voice.volume.persistant.handset="1" is as obvious as it sounds. the phone will remember the HANDSET voluem from call to call. |
| 13:37.53 | Katty | [TK]D-Fender: oh jolly good. |
| 13:37.59 | [TK]D-Fender | Katty: Your train of thought is permanently derailed... |
| 13:38.02 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 13:38.05 | Katty | [TK]D-Fender: yes, yes it is. |
| 13:38.10 | Squeeb | Hmm .. I'm reading the manual but I'm not understanding some of the terminology.. |
| 13:38.20 | Katty | Squeeb: welcome to my life. |
| 13:38.37 | Squeeb | How can I have some background music or simply "ringing" while asterisk hunts for an available agent |
| 13:38.48 | Katty | Squeeb: riddick went *whinywhinywhiny* *barkbark* *whinewhinewhine* all night. |
| 13:38.50 | Squeeb | at the moment it's just silent to the caller while the hunt is in progress |
| 13:39.04 | Katty | presumes lonely puppy. |
| 13:39.08 | jblack | Try queues. |
| 13:39.13 | jblack | They have musiconhold |
| 13:39.13 | Katty | if only these things came with readmes |
| 13:39.25 | Squeeb | it only seems to happen when you put them back on hld |
| 13:39.28 | [TK]D-Fender | queues are the only reason to through to term "agent" around |
| 13:39.30 | [gnubie] | [TK]D-Fender: i already flushed the firewall and placed an outbound call again via tdm/pots but still, the callee cannot hear me.. |
| 13:39.31 | Squeeb | hold * |
| 13:39.41 | [TK]D-Fender | [gnubie]: Show us something useful. |
| 13:39.42 | IsUp | Fender, thank you :) i've applied patch. well i am gonna try it later on my production server. |
| 13:39.55 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-84bad83f9543db3b) |
| 13:39.55 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:40.28 | [gnubie] | [TK]D-Fender: wait, i'll run tcpdump here.. |
| 13:46.33 | c4t3l | Katty: you can download the SIP admin guide from Polycom's website |
| 13:48.42 | dwagner | hello, can the asterisk version 1.4.21.2 be used in a production environment? |
| 13:50.08 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 13:50.34 | [TK]D-Fender | dwagner: Sure |
| 13:50.35 | *** join/#asterisk SteveTotaro (n=Administ@pool-141-157-95-245.balt.east.verizon.net) |
| 13:51.21 | dwagner | i've heard that there is a problem with the sip channel, that sometimes they hang and block the isdn channels. |
| 13:52.15 | [TK]D-Fender | dwagner: Go try. If you have a problem, change versions. |
| 13:52.46 | *** join/#asterisk happytiger (n=happytig@2607ds3-fb.0.fullrate.dk) |
| 13:54.25 | dwagner | haha |
| 13:56.36 | *** join/#asterisk fasttrack (n=joeljose@218.248.68.63) |
| 13:56.50 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 13:57.14 | *** join/#asterisk klictel (n=klictel@207.61.107.242) |
| 13:57.23 | fasttrack | hy all, i have seen the asterisk site. I was planning for a replacement/alternative for cisco's call agent(cucm).. have i come to the right place? |
| 13:57.34 | *** join/#asterisk Jax (n=Jax@pdpc/supporter/active/Jax) |
| 13:57.36 | Jax | hello! |
| 13:58.08 | fasttrack | Mostly, just for call conferences(loads of them!!).. over internet |
| 13:58.22 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 13:58.31 | happytiger | Hey all i just fiddled around and made my sip.conf and extension.conf live in a postgressql database, and now when i try to write a command in the asterisk conolse like sip show peers or any sip * command seems to have dissapeared any advice?? |
| 13:58.55 | Jax | i'm pretty new to VoIP and just installed asterisk. if i only use IP Phones, and order VoIP from my provider, do i need any additional hardware? or will the PBX Server with a NIC be enough? |
| 13:59.16 | happytiger | The only files that has ben thouched is ext.config |
| 13:59.30 | *** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com) |
| 13:59.51 | happytiger | sorry the file extconfig.conf |
| 14:00.40 | happytiger | If I revert my extconfig.conf to use default cdr instead my commands reaper?? |
| 14:01.13 | happytiger | Jax: no other recuired |
| 14:01.39 | Jax | is there something special i need to watch out for when registering at my ISP? i.e so they don't do the PBX stuff or something? |
| 14:01.42 | Jax | sorry, i'm very new ;) |
| 14:02.06 | *** join/#asterisk festr__ (n=festr@ns.hiro.cz) |
| 14:02.54 | happytiger | jax I am new to but I anyt sip provider will do problem is quality of lines get from them some voip providers drop lines to get exztra conenctioncharges etc |
| 14:02.59 | Katty | c4t3l: somehow, i don't think the SIP admin guide's going to have any info on my new puppy |
| 14:03.06 | Katty | c4t3l: i want a readme for my PUPPEH |
| 14:03.15 | Katty | c4t3l: specifically for my puppy. |
| 14:03.24 | Katty | c4t3l: i've ready tons of stuff off google already (= |
| 14:03.26 | c4t3l | oh... hehe |
| 14:03.32 | c4t3l | cool |
| 14:03.38 | Jax | happytiger try using . and , :D |
| 14:03.40 | Katty | Angela's Guide to Riddick.pdf |
| 14:03.49 | [TK]D-Fender | Katty: Sorry, your puppeh is sold as-is with no documentation or warranty. Puppeh is only sold in OEM |
| 14:03.55 | c4t3l | nice! |
| 14:04.02 | Katty | [TK]D-Fender: acutally puppeh does have a warranty |
| 14:04.15 | Katty | [TK]D-Fender: puppeh comes with 4 month, parts only |
| 14:05.38 | Katty | Qwell: i think i'm going to ditch wow. |
| 14:05.54 | Katty | Qwell: it's the same thing, over and over. |
| 14:07.03 | happytiger | jax lol yup though a line like this seems more fun in the long run despite the general connotations you know what i mean? |
| 14:12.16 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 14:17.44 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:17.53 | *** join/#asterisk jets (n=brian@pdpc/supporter/active/jets) |
| 14:19.07 | *** join/#asterisk phpboy (n=shane@196.211.1.45) |
| 14:19.52 | phpboy | Hey all, how would I go about sending out a specific callerid regardless of which extension is dialing out trough a trunk? |
| 14:20.29 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 14:21.53 | Katty | phpboy: Set(callerid(num=0000000000)) |
| 14:22.28 | Katty | phpboy: exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=5733344439) <- sample |
| 14:22.57 | adr3nalin3 | I am attempting to connect two asterisk servers via iax2. I am getting registration errors and it says that my peers are not online. I have setup user and peer on both ends. I am using static ips on both do I need to register? Anything I am missing? |
| 14:23.01 | Katty | phpboy: and then your dial a command thingy |
| 14:23.15 | *** join/#asterisk ManxPower (n=manxpowe@38.sub-75-201-15.myvzw.com) |
| 14:24.09 | *** join/#asterisk mog (n=mog@nat/digium/x-13b8b30b05869c44) |
| 14:24.09 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:26.24 | [TK]D-Fender | adr3nalin3: You do not need to register |
| 14:27.27 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 14:27.27 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:29.08 | c4t3l | adr3nalin3: use debug |
| 14:29.17 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:30.23 | adr3nalin3 | If I do not register, when I do a iax2 show peers should the host show online? |
| 14:30.48 | write_erase | What's the separator when declaring multiple allow= on the same line ? |
| 14:30.51 | russellb | adr3nalin3: depends. If you configured the peer as host=dynamic, then no |
| 14:30.57 | russellb | write_erase: comma |
| 14:31.01 | write_erase | thx |
| 14:31.11 | Katty | hai russell! |
| 14:31.13 | Katty | hai anthm! |
| 14:31.15 | Katty | hugs anthm |
| 14:31.20 | Katty | hugs mog |
| 14:31.22 | Katty | hugs ManxPower |
| 14:31.28 | mog | hey Katty |
| 14:31.40 | Katty | mog: got my puppy last night :> |
| 14:31.45 | klictel | adr3nalin3: a peer must register either when you want to know their IP address is it is dynamic, or if it is static sometimes your itsp might still want you to register so they don't go through the pain of maintining a table of ips |
| 14:32.06 | adr3nalin3 | thanks everybody |
| 14:32.21 | anthm | hi |
| 14:33.15 | IsUp | anyone knows whats really "safe_asterisk"? |
| 14:33.31 | IsUp | should i start with asterisk safe_asterisk or asterisk start? |
| 14:33.37 | c4t3l | def safe! |
| 14:33.44 | Squeeb | safe asterisk runs in a loop |
| 14:33.47 | Squeeb | so if it crashes |
| 14:33.49 | Squeeb | it relaunches |
| 14:33.49 | Squeeb | etc.. |
| 14:33.50 | klictel | asterisk start will run safe asterisk |
| 14:34.00 | IsUp | by the way, i've resolved AGI defunct process with starting: 'asterisk start > /dev/null 2>&1' |
| 14:34.18 | Katty | can you fix puppeh defunct for me? |
| 14:34.29 | IsUp | now AGI scripts are closing properly =) |
| 14:34.44 | ManxPower | safe_asterisk is a SHELL SCRIPT. |
| 14:35.03 | IsUp | yeah i know, for core dumping. anything else than? |
| 14:35.08 | ManxPower | IsUp: you can remove the "start" |
| 14:35.08 | IsUp | setting ulimit to maximum |
| 14:35.33 | ManxPower | I always user "chkconfig asterisk on" and "service asterisk start" |
| 14:35.36 | IsUp | so i can use 'asterisk > /dev/null 2>&1' |
| 14:35.47 | IsUp | and i dont have color on my CLI |
| 14:35.49 | IsUp | any ideas? |
| 14:35.57 | IsUp | everything works fine but no colorize |
| 14:36.11 | ManxPower | IsUp: officially it would be "asterisk -c", but "asterisk" should work. |
| 14:36.20 | ManxPower | IsUp: Yes, that is correct. |
| 14:36.50 | ManxPower | The -r does not support color. If you want color, run asterisk as "asterisk -c" and put it in a screen process. |
| 14:37.06 | IsUp | i want to start asterisk in background mode. because i am on SSH env. so i should start my server and close my SSH connection. |
| 14:37.16 | IsUp | but in my other servers i have color as well |
| 14:37.24 | IsUp | when i am connected with 'asterisk -r' |
| 14:37.29 | ManxPower | IsUp: I cannot help you further. |
| 14:37.45 | IsUp | '/usr/sbin/asterisk -f -vvvg -c' |
| 14:37.54 | IsUp | it's giving me color but giving defunct AGIs too =) |
| 14:38.09 | phpboy | Katty: you are too kind, thanks :D |
| 14:38.49 | phpboy | Last question for today (hopefully), how would I go about watching a specific call on console as opposed to everything |
| 14:38.50 | ManxPower | IsUp: -c does NOT start Asterisk in the background, which is one of your requirements |
| 14:38.51 | phpboy | ? |
| 14:39.06 | ManxPower | phpboy: you can't. |
| 14:39.11 | phpboy | :( |
| 14:39.11 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-ca60e7d0a746d145) |
| 14:39.11 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:39.19 | phpboy | That is kinda sucky :/ |
| 14:39.21 | Katty | phpboy: you might try Flash Operator Panel or Isymphony |
| 14:39.32 | ManxPower | IIRC, 1.6 was rumored to support this, but I've not seen anything further about it. |
| 14:39.35 | Katty | phpboy: it's addon software...connected via manager |
| 14:39.39 | phpboy | I'll Goggle it |
| 14:39.44 | IsUp | safe_asterisk is starting asterisk in background with flags, "-f -vvvg -c" |
| 14:39.55 | Katty | phpboy: it's really considered recipetionist software--drag and drop calls around |
| 14:40.03 | Katty | phpboy: i prefer isymphony... |
| 14:40.06 | ManxPower | IsUp: "man asterisk" |
| 14:40.09 | fasttrack | frnds.... to have a decent qos, for atleast 10 teleconferences(simultaneous).. with 10 participants each... a celeron D 2.53GHz with 2gb ram and wan of 100mbps burst rate.. is enough? |
| 14:40.30 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 14:40.32 | ManxPower | fasttrack: Yes. No. |
| 14:40.34 | Katty | hai CunningPike! |
| 14:40.42 | Katty | hugs CunningPike |
| 14:40.49 | fasttrack | Yes.. coz? |
| 14:41.01 | ManxPower | 10 G729 transcodes might not work. ulaw might work |
| 14:41.09 | [TK]D-Fender | fasttrack: Codec is crucial in this calculation. |
| 14:41.17 | CunningPike | Hey, Katty! |
| 14:41.18 | [TK]D-Fender | fasttrack: Individual jitter, etc. |
| 14:41.24 | ManxPower | I would not expect 10 iLBC transcodes to work, but 10 alaw calls should. |
| 14:41.25 | Katty | CunningPike: i got my new puppeh last night!!!! |
| 14:41.33 | fasttrack | Dell PowerEdge 1435 Dual AMD DUal Core opteron, and 2gb ram ? |
| 14:41.33 | CunningPike | brb - vpn switcherooo |
| 14:41.34 | Katty | CunningPike: he's all chin fluff and paws! |
| 14:41.55 | fasttrack | where is the bottle neck?? cpu or ram? |
| 14:41.56 | [TK]D-Fender | Katty: You've killed him with peer pressure! |
| 14:42.04 | [TK]D-Fender | fasttrack: CPU |
| 14:42.18 | [TK]D-Fender | fasttrack: So, what CODEC? |
| 14:42.21 | fasttrack | using g711 should help a bit right? |
| 14:42.28 | Katty | [TK]D-Fender: i killed him by oozing excitement |
| 14:42.28 | [TK]D-Fender | fasttrack: Yes, a lot |
| 14:42.36 | *** join/#asterisk seanmh (i=HydraIRC@216.31.101.83) |
| 14:42.37 | [TK]D-Fender | fasttrack: Might do for your needs |
| 14:42.41 | fasttrack | since i have enought speed... |
| 14:42.44 | fasttrack | in wan.. |
| 14:42.46 | fasttrack | ;) |
| 14:43.04 | jblack | hrmm. the stock market is sad today. |
| 14:43.05 | *** join/#asterisk campari3 (n=campari@72-48-3-116.dyn.grandenetworks.net) |
| 14:43.14 | *** join/#asterisk MrGabu (n=gbdurant@200-170-192-194.static.spo.ifx.net.br) |
| 14:43.24 | fasttrack | so the dual core amd opteron?? over celeron d??... u sure celron will not suffice..?? |
| 14:43.26 | [TK]D-Fender | fasttrack: I'd highly recommend running a hardware zaptel timing source like a TDM410P. |
| 14:43.38 | campari3 | hello all |
| 14:43.41 | fasttrack | k... thanks for insights... |
| 14:43.46 | [TK]D-Fender | fasttrack: for the diff in cost, don't go asking for trouble |
| 14:43.49 | jblack | From 11,143 down to 10,838, so far. |
| 14:43.51 | fasttrack | k.. will..look at that... |
| 14:43.52 | campari3 | has anyone seen an exploit for asterisk that allows a remote user to take over a registration and make calls? |
| 14:44.03 | fasttrack | hardware? |
| 14:44.05 | campari3 | using a REGISTER with the contact header set to sip:s@<ip> ? |
| 14:44.18 | fasttrack | i am using cisco gateway and gatekeeper... |
| 14:44.26 | fasttrack | will that give timming???... |
| 14:44.28 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 14:44.35 | fasttrack | i haven looked at that yet,.. |
| 14:44.43 | [TK]D-Fender | campari3: That isn't any inherent takeover. |
| 14:44.56 | ManxPower | campari3: Not in recent versions of Asterisk. HOWEVER, a poorly designed dialplan and sip.conf could easily let people route phone calls thru your system. |
| 14:45.05 | fasttrack | i know there is option for "getting time form upstream gateway" and generating time.. |
| 14:45.06 | [TK]D-Fender | fasttrack: No, you need a ZAPTEL timing source to keep MeetMe in synch. |
| 14:45.17 | IsUp | ManxPower: i am unable to colorize =) |
| 14:45.20 | fasttrack | k.. an asterisk dependency? |
| 14:45.22 | ManxPower | fasttrack: not TIME, SYNC. |
| 14:45.31 | fasttrack | kkk... getting.. it.. |
| 14:45.32 | [TK]D-Fender | fasttrack: For which there is no way I'd trust ztdmmy for more than a tenth of your stated qty |
| 14:45.43 | campari3 | manx, dialplan and sip.conf are not poorly designed.. this is an active exploit |
| 14:46.06 | ManxPower | campari3: then it's unknown and you should report it on bugs.digium.com |
| 14:46.12 | [TK]D-Fender | campari3: you're offering no real details. And this is something you should PM to an admin in #asterisk-dev |
| 14:46.19 | campari3 | okay |
| 14:46.48 | campari3 | i am using v 1.2, upgrading to latest 1.2 now |
| 14:47.00 | campari3 | had a customer have the same issue and asking them what ver they are using |
| 14:47.22 | fasttrack | one more quest... |
| 14:47.24 | ManxPower | In my experience putting context=INVALID and put the proper context= lines for each of the friend/user/peer seems to keep things secure -- barring actual bugs |
| 14:47.38 | campari3 | yes, we do that |
| 14:47.43 | fasttrack | will a cisco gwgk.. improve performance??.. or is it just an unwanted overhead? |
| 14:47.44 | campari3 | there is a special REGISTER the attacker sends |
| 14:47.50 | campari3 | that causes asterisk to allow them access |
| 14:47.53 | ManxPower | campari3: we are assuming you are running the lastest version. |
| 14:48.00 | ManxPower | campari3: CITE. YOUR. COURCE> |
| 14:48.03 | ManxPower | and SOURCE too |
| 14:48.07 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 14:48.10 | campari3 | i am looking at the sip debug :-p |
| 14:48.11 | fasttrack | i am looking at colocating the servers.. |
| 14:48.12 | campari3 | that's my source |
| 14:49.12 | AndyMillar | hmm, what's the most reliable analogue card? |
| 14:49.18 | fasttrack | in my opinion... gwgk , i am planning to use gwgk for qos... |
| 14:49.21 | ManxPower | campari3: so go report the bug then |
| 14:49.42 | fasttrack | does asterisk support qos?? i mean can it police and shape and queue and classify? |
| 14:49.57 | fasttrack | i thought only hardware routers can do that |
| 14:50.03 | ManxPower | fasttrack: no. that is not the job of a server, that is the job of your network |
| 14:50.29 | ManxPower | [TK]D-Fender: how skeptical of campari3 are you? |
| 14:50.29 | fasttrack | k.... thanks |
| 14:50.46 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 14:51.18 | CunningPike | Katty: A new puppy, eh? Congrats |
| 14:54.07 | *** join/#asterisk Hertzy3 (n=ahertz@ahertz.atlantic.net) |
| 14:56.03 | Hertzy3 | anyone know how to boot up a polycom 501 and have it not look for a server? Just so I can pull it up to reset the local configs |
| 14:57.55 | CunningPike | Hertzy3: There's a key sequence to reset to the factory defaults |
| 14:58.15 | Hertzy3 | but do you need to be booted all the way up? |
| 14:58.21 | Hertzy3 | or can you do it any time? |
| 14:58.23 | ManxPower | Hertzy3: The polycoms REALLY REALLY REALLY like trying to find a server |
| 14:59.09 | Hertzy3 | yeah, well im having a lot of difficulty with that. I have brought up plenty of these phones onto my server with no problem, but i cant seem to get this one working at all |
| 15:00.08 | Hertzy3 | doesnt like dhcp, doesnt like static. never connects. cant get boot parameters. error downloading macaddr.cfg. no matter what I do I cant get this thing to even boot without finding a server. |
| 15:00.30 | *** join/#asterisk edwin_quijada (n=macaruch@190.80.159.93) |
| 15:00.30 | Hertzy3 | i saw somewhere it was like *468 or soemthing to reset the configs. but i assume u have to be fully booted for that to work |
| 15:00.31 | ManxPower | Hertzy3: My suggestion is to go into the bootrom (pre-boot) and change the DHCP option number to something you are not using, that way the phone should timout quickly |
| 15:00.52 | IsUp | well ManxPower... i am still in trouble with defunts... |
| 15:00.55 | ManxPower | yes, you have to be booted to do factory reset. Factory reset does not apply to boot options, IIRC. |
| 15:01.07 | CunningPike | Hertzy3: No - you can enter that key sequence during the pre-boot countdown |
| 15:01.23 | ManxPower | CunningPike: really? Interesting. |
| 15:01.57 | CunningPike | Press and hold the keys until you're prompted for the admin password (usually '456') |
| 15:02.12 | edwin_quijada | Hi! I am trying to use a zap channel but when incoming a call I get Zap/1-1 status is UNKNOWN |
| 15:02.51 | IsUp | I am using Ubuntu, i dont have init.d script. is that ok? |
| 15:03.22 | edwin_quijada | it recieves the call but doesnt asnwer |
| 15:03.24 | Hertzy3 | CunningPike: thank you very much, it appears to be working |
| 15:03.33 | CunningPike | Hertzy3: No problem |
| 15:03.42 | edwin_quijada | i can see that * process the call |
| 15:04.25 | ManxPower | edwin_quijada: run Answer() ? |
| 15:04.57 | edwin_quijada | ManxPower: where ? |
| 15:05.06 | ManxPower | edwin_quijada: the dialplan |
| 15:05.23 | edwin_quijada | yes |
| 15:07.10 | edwin_quijada | when i try to use Playback I get status UNKNOWN |
| 15:07.42 | *** part/#asterisk fasttrack (n=joeljose@218.248.68.63) |
| 15:07.56 | edwin_quijada | exten => s,1,Answer |
| 15:08.09 | edwin_quijada | exten => s,2,wait,2 |
| 15:08.24 | *** join/#asterisk Sinist3r (n=IamLegio@209.160.40.98) |
| 15:08.36 | ManxPower | edwin_quijada: Try Wait(2) |
| 15:08.41 | Sinist3r | Anyone know of a good opensource predictive dialer? |
| 15:09.00 | edwin_quijada | exten => s,Playback(vm-goodby) |
| 15:09.17 | edwin_quijada | Sinist3r: www.gnudialer.org |
| 15:09.35 | Squeeb | What's the difference between exten = and exten => |
| 15:09.35 | ManxPower | try exten => s,1,Playback(vm-goodbye) |
| 15:09.36 | Sinist3r | saw that one, couldn't tell if it was that good |
| 15:09.48 | ManxPower | ..er s,3 |
| 15:11.34 | edwin_quijada | ManxPower: sorry, ihave s,3,Pla... |
| 15:11.50 | Sinist3r | do I just install gnudialer as a module? |
| 15:12.16 | edwin_quijada | Sinist3r:no |
| 15:12.25 | edwin_quijada | ManxPower: i get the same |
| 15:12.29 | Sinist3r | So how does this work? |
| 15:12.40 | edwin_quijada | i cant hear anytging |
| 15:12.45 | edwin_quijada | anything |
| 15:12.59 | *** join/#asterisk hardwire (n=hardwire@rdbk-11713.mtaonline.net) |
| 15:13.20 | ManxPower | Sinist3r: It works like this: You come here and ask a question about software. Your question is answered. Then you go download the suggested software and read the docs. |
| 15:13.51 | edwin_quijada | Auto falltrough, channel 'Zap/1-1' status 'UNKNOWN' |
| 15:13.56 | [TK]D-Fender | ManxPower: Lets see... no real details, mention of a generic normal looking contact.... |
| 15:14.14 | edwin_quijada | after hangup |
| 15:14.22 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-211-254-206.dsl.pltn13.sbcglobal.net) |
| 15:14.32 | [TK]D-Fender | edwin_quijada>exten => s,Playback(vm-goodby) <- and somehow you feel a priority isn't needed for this line? |
| 15:14.50 | [TK]D-Fender | edwin_quijada: pastebin your actual call and your actual dialplan |
| 15:15.06 | edwin_quijada | ok |
| 15:15.13 | Blackvel | is there any configuration file for the sounds directory (e.g Record)? asterisk.conf astdatadir or astvarlibdir is the right place? or is there like moh another config file for absolute path mapping? |
| 15:15.48 | [TK]D-Fender | Blackvel: "core show application record" |
| 15:16.18 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 15:17.13 | Blackvel | it uses the standard directory...I know... |
| 15:17.39 | Sinist3r | is gnudialer the same as vicidialer? |
| 15:17.58 | [TK]D-Fender | Sinist3r: No. If it were, it'd be called vicidialer |
| 15:18.09 | Blackvel | its unfortunately not a specific Record question as it also counts for Playback |
| 15:18.19 | Sinist3r | A lot of sites speak of both as if they were one in the same. |
| 15:18.20 | [TK]D-Fender | Blackvel: "core show application playback" |
| 15:18.29 | [TK]D-Fender | Sinist3r: Well they aren't |
| 15:18.51 | Sinist3r | which one is betteR? |
| 15:19.35 | Blackvel | asterisk 1.2.x does not explain anything about sounds directory .... |
| 15:20.13 | Blackvel | I could try to rephrase the question to: is changing asterisk.conf astdatadir or astvarlibdir to specifiy the "sounds" directory? |
| 15:20.22 | Squeeb | To make asterisk dial out to a landline over a sip account, I use SIP/context/NUMBER right? |
| 15:20.23 | Blackvel | is changing...enough |
| 15:20.32 | Blackvel | SIP/number@context |
| 15:20.35 | Squeeb | aah yes |
| 15:20.54 | IsUp | how should i start Asterisk over SSH? i am using Ubuntu. my AGIs are going to <defunct> processes. |
| 15:21.07 | Blackvel | start? asterisk -vvvc? |
| 15:21.18 | Blackvel | or use init.d script |
| 15:21.54 | edwin_quijada | [TK]D-Fender: http://pastebin.com/m6f0b9f8b |
| 15:21.58 | IsUp | i can't use console cli, because i am on SSH. so when i am exit, |
| 15:22.04 | IsUp | asterisk will close |
| 15:23.12 | Blackvel | right |
| 15:23.17 | *** join/#asterisk ToTo (n=ToTo@207.176.6.132) |
| 15:23.17 | [TK]D-Fender | edwin_quijada: "dialplan show from-pstn" |
| 15:23.30 | Blackvel | isup: /etc/init.d/asterisk start? |
| 15:23.37 | jaytee | if you start asterisk as a service and then use asterisk -r it will reconnect |
| 15:23.45 | *** join/#asterisk markgreene (n=markgree@130.160.45.34) |
| 15:24.02 | Blackvel | if neither works try temporary starting with asterisk -vvc & |
| 15:24.06 | Blackvel | or run screen |
| 15:24.08 | markgreene | Hello everyone. Has anyone here used, or is familiar with, the sagnoma a108 card? |
| 15:24.11 | Blackvel | (only temp solution) |
| 15:24.15 | edwin_quijada | [TK]D-Fender: i dont understand you? |
| 15:24.22 | IsUp | i dont have init.d scripts |
| 15:24.25 | IsUp | yes markgreene |
| 15:24.29 | IsUp | i am using Sangoma |
| 15:24.36 | [TK]D-Fender | edwin_quijada: I just gave you a CLI command to PB. DO IT |
| 15:24.40 | Blackvel | isup: there is a sample (you can use it 1:1) in the src directory |
| 15:24.51 | IsUp | contrib/init.d/rc.debian... ? |
| 15:25.00 | markgreene | IsUp, I am looking at getting one but I am little confused on how it hooks up 8 T1s using 4 ports. Could you help me understand how this works? |
| 15:25.06 | Blackvel | isup: should be yes |
| 15:25.11 | Blackvel | just copy it to /etc/init.d/asterisk |
| 15:25.20 | IsUp | okay Blackvel, i am gonna try |
| 15:25.32 | markgreene | IsUp, are you using the A108? |
| 15:25.36 | Blackvel | if you use chkconfig you can enable the service "asterisk" for the run levels |
| 15:25.40 | edwin_quijada | ok |
| 15:25.52 | IsUp | i have A102, A104 and A108. i have all Sangoma cards i think. |
| 15:26.27 | IsUp | markgreene, ports are seperated. they use special cables. 1 RJ 45 splits two cable and provides 2 ports |
| 15:26.43 | IsUp | Sangoma great on stability, installation and support i think. |
| 15:27.00 | markgreene | IsUp, do these special cables come with the card? |
| 15:27.32 | IsUp | of course |
| 15:27.52 | IsUp | 3 stickers, 1 cd, a manual, card and cables =) |
| 15:28.38 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 15:30.28 | IsUp | i am using Sangoma A108 on my production servers. about 7 x HP server |
| 15:30.30 | IsUp | with ss7 |
| 15:30.45 | edwin_quijada | [TK]D-Fender:http://pastebin.com/m25971f11 |
| 15:30.54 | markgreene | IsUp, OK that makes more sense. Thanks for the help |
| 15:31.04 | IsUp | you are wellcome |
| 15:35.54 | [gnubie] | waves to all.. |
| 15:35.59 | [gnubie] | gtg now.. |
| 15:36.00 | [gnubie] | thanks.. |
| 15:37.35 | [TK]D-Fender | edwin_quijada: 4. Playback(vm-goodby) [pbx_config] <-- |
| 15:37.49 | Qwell | where'd the e go? |
| 15:38.10 | [TK]D-Fender | Qwell: I'd prefer to know where priority **3** went. |
| 15:38.30 | Qwell | [TK]D-Fender: $20 says our concerns are very much related. |
| 15:38.36 | [TK]D-Fender | edwin_quijada: Either you're not working in the right file or you need to learn to apply your changes. |
| 15:38.48 | *** part/#asterisk wasabi (n=wasabi@ubuntu/member/wasabi) |
| 15:38.50 | [TK]D-Fender | Qwell: Yes... they are both dialplan... its ends there ;) |
| 15:39.06 | Qwell | wait, wha? |
| 15:39.30 | [TK]D-Fender | Qwell: Doesn't matter if he picked a bad file... it'll never play anyways, and thats his first worry. |
| 15:39.39 | Qwell | yeah, didn't notice the 4 at the bottom |
| 15:39.44 | [TK]D-Fender | Qwell: Let him fix his mistakes in the right order ;) |
| 15:40.02 | [TK]D-Fender | Qwell: Trust NOTHING. Thats why I had him dump it from CLI as well |
| 15:40.11 | Qwell | indeed |
| 15:40.11 | edwin_quijada | [TK]D-Fender:that was the solution i had 4 instead of 3 |
| 15:40.26 | MrGabu | hello all |
| 15:40.27 | edwin_quijada | i solve thks a lot! |
| 15:40.48 | [TK]D-Fender | edwin_quijada: And you might want to examin the spelling for the sound file you want to play. We suspect you might have made an error there as well. |
| 15:40.54 | edwin_quijada | i thougth it didnt matter |
| 15:41.31 | [TK]D-Fender | edwin_quijada: NO COMMENT |
| 15:41.34 | edwin_quijada | [TK]D-Fender: thks a lot !!:) |
| 15:41.51 | MrGabu | what is the concept of "dynamic spans" (eth) ? |
| 15:42.30 | *** join/#asterisk grEvenX (n=even@c9A7F8BC3.dhcp.bluecom.no) |
| 15:42.35 | *** part/#asterisk edwin_quijada (n=macaruch@190.80.159.93) |
| 15:42.43 | *** join/#asterisk GreenCult (n=greencul@200.48.85.18) |
| 15:44.24 | *** join/#asterisk matsk (n=Mats@host-90-235-55-123.mobileonline.telia.com) |
| 15:44.35 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:45.53 | *** join/#asterisk moy (n=moy@65-122-15-169.dia.static.qwest.net) |
| 15:46.36 | *** join/#asterisk hardwire (n=hardwire@rdbk-857.mtaonline.net) |
| 15:46.53 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 15:47.56 | keith4 | [TK]D-Fender: i just choked on my coffee, i was laughing so hard at that last exchange |
| 15:48.17 | jeev | anyone dealt with Avocent DSR before ? |
| 15:51.47 | *** join/#asterisk bijit (n=benji@200.122.158.243) |
| 15:52.24 | *** join/#asterisk CarlaWinchester (n=chatzill@unaffiliated/quirkycarla) |
| 15:53.34 | *** join/#asterisk whymarkwh (n=dsfsdfsd@196.211.34.2) |
| 15:53.39 | bijit | I am having problems when internet fails..my asterisk goes crazy. I have to remove gateway and restart asterisk to start working normal. Anone has any ideas why? |
| 15:54.18 | *** part/#asterisk CarlaWinchester (n=chatzill@unaffiliated/quirkycarla) |
| 15:54.31 | whymarkwh | hi there i am connection a jurrhans 4 port bri card to my telko with asterisk can some please tell me if i need to set it to nt ot te mode, this confuses me |
| 16:01.22 | ManxPower | bijit: Asterisk expects to be able to look up the hostname associated with each IP address on the server. It normally uses DNS for this. If your Asterisk server does not need to use DNS, then edit /etc/nsswitch.conf and remove dns from that file. Also you can edit /etc/hosts and make sure ALL IP addresses of the system are listed |
| 16:01.48 | hardwire | you can use fake hostnames |
| 16:01.55 | hardwire | it just needs to resolve front and reverse |
| 16:01.59 | ManxPower | hardwire: correct. |
| 16:02.17 | hardwire | I use bind and voip.hq.soandso.com |
| 16:02.18 | ManxPower | hardwire: amazing how people don't know the basics of networking |
| 16:02.37 | hardwire | with iax and sip SRV records |
| 16:02.42 | hardwire | it makes everything so happy |
| 16:02.59 | Blackvel | whymarkwh: to telco pstn? TE |
| 16:03.41 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 16:03.46 | Blackvel | whymarkwh: and let me know if you encounter any echo problems in 2-3 weeks |
| 16:04.36 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:06.29 | whymarkwh | thx |
| 16:10.22 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 16:11.36 | *** join/#asterisk Xaviertoor (i=Fagner@189-015-117-066.xd-dynamic.ctbcnetsuper.com.br) |
| 16:13.55 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.static.dsl.tele.dk) |
| 16:13.56 | *** join/#asterisk propellerhead (n=yogurt2u@host113.190-136-116.telecom.net.ar) |
| 16:14.02 | *** part/#asterisk MrGabu (n=gbdurant@200-170-192-194.static.spo.ifx.net.br) |
| 16:15.55 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:17.46 | bijit | ManxPower: when you refer to the ip address, are you reffering to the ip of the phones? or from the asterisk box? |
| 16:20.39 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-215-215.phlapa.east.verizon.net) |
| 16:21.15 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:26.32 | gr0mit | Loadrunner, did you get your bristuff working? |
| 16:29.00 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:29.13 | ManxPower | bijit: of all IP addresses on the Asterisk box |
| 16:30.41 | ManxPower | hardwire: I used to use SRV records to let my ATA roam seamlessly between local LAN, remote non-NAT lan, and remove NAT lan. |
| 16:37.25 | *** join/#asterisk grEvenX (n=even@c9A7F8BC3.dhcp.bluecom.no) |
| 16:37.49 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 16:45.26 | *** join/#asterisk roe_ (n=roe___@216-164-160-45.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 16:48.04 | roe_ | anyone know if the aastra 480i support dns for the various server values? It looks like the web page has prefilled in 0.0.0.0 but that isn't necessarily deterministic |
| 16:49.48 | C4colo | roe_: yes |
| 16:50.09 | C4colo | you can put in a FQDN if you want |
| 16:50.33 | C4colo | just make sure that you either provide a valid DNS server value under network, or you run a DNS forwarder on the asterisk server |
| 16:51.25 | C4colo | running a dns forwarder on the asterisk server is the best way I have found as you can override values to resolve certian FQDNs to internal or external addresses depending on if the request came from internal or external |
| 17:00.56 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 17:01.15 | Carlos_PHX | I am trying to set up a T.38 fax receiver. I have installed 1.6rc6 and made sure that app_fax was selected in menuselect, however, show applications doesn't show a fax app. I can't find any docs on using the fax app. Any pointers? |
| 17:01.34 | Qwell | the answer is, of course, that there is no app named "Fax" |
| 17:01.40 | Qwell | It's SendFax and ReceiveFax |
| 17:02.17 | Carlos_PHX | Oddly, I don't see those in the app list either. |
| 17:02.20 | ManxPower | Carlos_PHX: "core show applications like fax" |
| 17:02.57 | Carlos_PHX | faxserver*CLI> core show applications like fax |
| 17:02.57 | Carlos_PHX | <PROTECTED> |
| 17:02.58 | Carlos_PHX | <PROTECTED> |
| 17:02.58 | Carlos_PHX | faxserver*CLI> |
| 17:03.08 | ManxPower | Carlos_PHX: then it's not installed. 8-) |
| 17:03.23 | ManxPower | is it listed in /usr/lib/asterisk/modules ? |
| 17:03.24 | Carlos_PHX | Heh, right. However I've recompiled twice, making sure it was selected. |
| 17:03.41 | Qwell | is app_fax.so loaded? |
| 17:05.48 | ManxPower | I just ordered a wire wrap tool. I feel so old school now. |
| 17:06.03 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 17:06.08 | coppice | I still have one of those somewhere |
| 17:07.11 | Carlos_PHX | Hmm |
| 17:07.12 | Carlos_PHX | faxserver*CLI> module load app_fax.so |
| 17:07.12 | Carlos_PHX | Unable to load module app_fax.so |
| 17:07.12 | Carlos_PHX | Command 'module load app_fax.so' failed. |
| 17:07.13 | Carlos_PHX | [Sep 29 10:06:55] WARNING[4184]: loader.c:371 load_dynamic_module: Error loading module 'app_fax.so': libspandsp.so.1: cannot open shared object file: No such file or directory |
| 17:07.13 | Carlos_PHX | [Sep 29 10:06:55] WARNING[4184]: loader.c:662 load_resource: Module 'app_fax.so' could not be loaded. |
| 17:07.32 | Carlos_PHX | Looks like my spandsp installation is not working. |
| 17:07.36 | ajohnson | interesting |
| 17:10.11 | ManxPower | Carlos_PHX: There you go. |
| 17:10.51 | ManxPower | I assume spandsp is not included in 1.6. (unless Digium and coppice managed to work out licensing issues) |
| 17:11.04 | Katty | CunningPike: thanks (= |
| 17:11.38 | CunningPike | Katty: You ha(d|ve) a cat, right? |
| 17:11.47 | ManxPower | coppice: I'm going to try my hand at selling used, preconfigured, prebuilt tellabs echo cancel systems in e-bay. All the chassis are wirewrap |
| 17:12.02 | Katty | CunningPike: i grew up with cats. |
| 17:12.06 | *** join/#asterisk MrGabu (n=gbdurant@200-170-192-194.static.spo.ifx.net.br) |
| 17:12.10 | Katty | CunningPike: always at least 2 in the house. |
| 17:12.11 | *** part/#asterisk MrGabu (n=gbdurant@200-170-192-194.static.spo.ifx.net.br) |
| 17:12.14 | Katty | CunningPike: http://www.new.facebook.com/photo.php?pid=34075286&l=ac49c&id=37617946 <- Riddick |
| 17:12.23 | coppice | ManxPower: the Smithsonian might make a good offer |
| 17:12.31 | ajohnson | ManxPower: I had heard recently that the licensing for spandsp had changed to allow it to be merged into 1.6 |
| 17:12.38 | ajohnson | but maybe it was going to be merged into 1.6.2 |
| 17:12.43 | Katty | CunningPike: he's such a sweety, tho currently whining a lot because of missing littermates |
| 17:12.48 | ManxPower | coppice: *bap* These things are rock solid and Just Work |
| 17:13.08 | ManxPower | I have 6 or 8 of them in production at clients |
| 17:13.21 | ManxPower | ajohnson: coppiece would know. |
| 17:13.53 | CunningPike | Katty: Cute |
| 17:14.03 | coppice | tellabs can be quirky. they are pretty old, and from before a lot of the modern tricks for rock solid performance |
| 17:14.28 | coppice | I made spandsp LGPL, so it can be used by Freeswitch. |
| 17:14.31 | ManxPower | coppice: we've never had problems with them. |
| 17:14.40 | ManxPower | coppice: Thank you for changing the license. |
| 17:14.49 | coppice | sheltered life, maybe :-) |
| 17:14.51 | Katty | CunningPike: very (= |
| 17:15.01 | Katty | CunningPike: house breaking is going well... |
| 17:15.03 | *** join/#asterisk jazzmann (n=chatzill@cpc1-lutn9-0-0-cust163.lutn.cable.ntl.com) |
| 17:15.11 | Katty | CunningPike: so far, only one accident in the house.... and it was on the way out the door |
| 17:15.15 | coppice | the licence change sucks, but I think working with Freeswitch is probably the greater ood |
| 17:15.16 | ManxPower | coppice: I bet they work better than the Digium EC cards. |
| 17:15.24 | CunningPike | Katty: Ah yes - rarely a problem with cats |
| 17:15.54 | Carlos_PHX | spandsp is still not included. I obtained it from the usual source and compiled, but apparently it's not working. Doing it again. |
| 17:16.00 | Katty | CunningPike: cats still need to be litter trained (= |
| 17:16.08 | Katty | CunningPike: and i've done my fair share of it |
| 17:16.20 | jazzmann | hi can anyone help me set my gizmo5 voip account with asterisk both are on same machine.Using centos/I tried the setting from this url http://support.gizmoproject.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=325#10 but could not succed |
| 17:16.23 | ManxPower | coppice: you COULD have offered to license spandsp to Digium for a reasonable fee |
| 17:16.32 | CunningPike | Katty: Guess we were lucky - our cat just used the tray from day one |
| 17:16.34 | *** join/#asterisk steliosk_laptop (n=stelios@emile.ath.forthnet.gr) |
| 17:16.36 | coppice | well they certainly shouldn't do, for several reasons. the are too old to incorporate the modern tricks, and anything sitting in the E1 or T1 path does additional alaw/ulaw conversions which reduces voice quality |
| 17:16.54 | Katty | CunningPike: some kittens do |
| 17:17.04 | Katty | CunningPike: some kittens i've bottle fed from 2 weeks of age. |
| 17:17.13 | Katty | CunningPike: just depends on the kitty |
| 17:17.18 | CunningPike | Katty: ya - we've had ours since she was a little kitty |
| 17:17.27 | coppice | ManxPower: Digium pay for something? are you new around here? |
| 17:17.33 | ManxPower | coppice: what specific modern tricks? |
| 17:17.46 | ManxPower | coppice: Bitter, table for one! |
| 17:18.16 | ManxPower | coppice: They paid for G729 license and a GPL H323 channel driver. |
| 17:19.17 | coppice | A lot of work was done in the late 90s on the quirks of cancellation in the real world. much of that resulted in tests in G.168, thought G.168 is far from comprehensive. |
| 17:19.55 | coppice | The make money from G.729. I don't know what happened with the H.323 code, but it never went anywhere, so I assume someone got pissed off |
| 17:20.46 | *** join/#asterisk EI5GTB (n=Paul@78.16.228.160) |
| 17:21.04 | ManxPower | these cards claim to be fully compliant with G.168 (as well as G.164 and G.165, but anything should support those) |
| 17:21.05 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 17:21.08 | coppice | a lot of modern ecs really suck, still. the Linksys ATAs are terrible |
| 17:21.14 | EI5GTB | afternoon gys. anyone familiar with the pap2 from linksys? i can't seem to get a dialtone |
| 17:21.21 | EI5GTB | hahaha |
| 17:21.24 | EI5GTB | about sums it up |
| 17:21.27 | Qwell | EI5GTB: nice timing |
| 17:21.35 | EI5GTB | that has to be quoted somewhere! |
| 17:21.37 | EI5GTB | lol |
| 17:21.42 | coppice | yeah, but G.168 has evolved a lot over the years, their ages means they relate to an old version |
| 17:22.14 | ManxPower | coppice: delightful how they change things and don't change the spec number 8-) |
| 17:22.29 | [TK]D-Fender | EI5GTB: It won't give dialtone until it has successfully registered |
| 17:22.41 | Qwell | ManxPower: I imagine it's versioned. |
| 17:22.50 | EI5GTB | it has sucesfully registered tho. according to the *cli anyway.. |
| 17:23.00 | ManxPower | These cards also claim to support all sorts for "enhancements" called CLEARCALL. |
| 17:23.12 | EI5GTB | Registration State:Online |
| 17:23.25 | coppice | ManxPower: a good one to try with a lot of older ECs is to use an IP phone that has response down to pretty low frequencies, and call out through an E1 or T1 port, through your EC, and to a distant analogue phone. Quite a few go really funky |
| 17:23.28 | ManxPower | the manual is dated 1999 |
| 17:23.28 | jazzmann | guys is there anyway I can use my gizmo5 account in a way that asterisk picks ups the calls for anyone calling it |
| 17:23.29 | jazzmann | thanks |
| 17:23.52 | Qwell | jazzmann: sure, it's just like any other SIP account, right? |
| 17:23.58 | *** join/#asterisk shazaum (n=shazaum@unaffiliated/shazaum) |
| 17:24.02 | coppice | and they were still wirewrapping then? I thought those ones were older |
| 17:24.23 | EI5GTB | [TK]D-Fender, accoring to the info page the phone is not going off hook |
| 17:24.33 | EI5GTB | im wondering is it to do with line impedence? |
| 17:24.37 | ManxPower | coppice: I have new-in-box chassis, and the only cheap chassis are the wire wrap versions |
| 17:24.44 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 17:25.02 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 17:25.09 | ManxPower | every once in a while you seen an AMP type connection on them, but they are hard to find cheap. |
| 17:25.12 | [TK]D-Fender | EI5GTB: Check your phone, then try another. Wiriing is equally suspect |
| 17:25.21 | coppice | I expect they were a pretty old design in 1999 |
| 17:25.44 | ManxPower | coppice: I assume they produce them for COs and telcos that use wire wrap |
| 17:25.57 | jazzmann | I tried the setting on their website but the different is I have to login in with my gizmo username and password and no setting up of ports like 5060 |
| 17:26.16 | ManxPower | coppice: I'll put on RJ-48/RJ-45 jacks on them |
| 17:26.19 | coppice | telcos mostly stopped using wire wrap in the early 80s :-) |
| 17:26.45 | jazzmann | http://www.zimbio.com/Asterisk/articles/50/AstGizmo+Setup+Gizmo5+Asterisk+Work+Together |
| 17:27.01 | jazzmann | http://www.zimbio.com/Asterisk/articles/50/AstGizmo+Setup+Gizmo5+Asterisk+Work+Together |
| 17:27.04 | ManxPower | coppice: *shrug* It's a new in box chassis with shrink wrap docs |
| 17:27.38 | ManxPower | at $25 each for a 16 slot chassis, I figure is not too bad. |
| 17:28.29 | ManxPower | coppice: you may be the most experience old school teleco guy here so I'm interested in your opinions and info |
| 17:28.34 | coppice | The G.168 doc does change. If you really want to know what people are talking about you look for things like G.168-2002 |
| 17:29.41 | ManxPower | coppice: what ms tail do you recommend, 32ms, 64,ms or 128ms? |
| 17:30.09 | coppice | 32ms spread over 128ms :-) |
| 17:31.44 | ManxPower | 8-) |
| 17:32.36 | ManxPower | in the real world whats the max endpath you would typically need |
| 17:33.43 | coppice | well, most cancellers beyond 32ms are actually running in a sparse mode cancelling 4 x 8ms chunks. Unless you disable the network cancellers, then from a CPE you seldom see more than 16ms of echo, and very rarely more than 32ms. The longer echos are taken care of my in network cancellers. |
| 17:34.27 | ManxPower | coppice: awesome |
| 17:34.36 | *** part/#asterisk matsk (n=Mats@host-90-235-55-123.mobileonline.telia.com) |
| 17:35.37 | ManxPower | The most commonly available used cards are the 32ms versions |
| 17:36.23 | tzafrir_laptop | coppice, what if "network" is "voip"? |
| 17:36.45 | ManxPower | tzafrir_laptop: Then you're screwed. |
| 17:36.54 | tzafrir_laptop | due to a bad terminal that didn't cancel the echo at its end? |
| 17:37.07 | coppice | don't let echos into VoIP paths. they won't come out :-) |
| 17:37.37 | tzafrir_laptop | I don't control all the voip path |
| 17:38.03 | tzafrir_laptop | Do I trust my voip provider to cancel the horror echos of the internet? |
| 17:38.14 | ManxPower | tzafrir_laptop: As I understand it you CANNOT cancel echo at the latencies required on VoIP paths. |
| 17:38.20 | coppice | that's irrelevant. one echo gets into a VoIP segment you are normally screwed. |
| 17:39.06 | *** join/#asterisk EI5GTB (n=Paul@78.16.171.55) |
| 17:39.53 | EI5GTB | dumb internet |
| 17:39.54 | coppice | it nothing to do with the latencies, although they don't help. the loop needs to be of a stable length for cancellers to adapt. Most VoIP paths are endlessly changing. If the path uses a low bit rate codec, even a stable length one won't cancel |
| 17:40.20 | ManxPower | coppice: so it is less the echo latency and more the echo jitter? |
| 17:41.23 | coppice | that comfort noise stuff that echo cancellers do is needed because of the lossiness of alaw and ulaw codecs. anything more lossy than those, and cancellation falls apart. |
| 17:42.54 | *** join/#asterisk simNIX (n=simNIX@82-204-21-111.dsl.bbeyond.nl) |
| 17:43.15 | ManxPower | coppice: what do you think about G.722 (Polycom calls it "HD Voice") |
| 17:43.26 | EI5GTB | ok, so my pap2 is registerd. Has the line light turned on. When i go off hook i can hear my voice, i.e there is power to the line, and if i press **** i get a prompt from the device |
| 17:43.29 | EI5GTB | but i get no dial tonwe |
| 17:43.47 | [TK]D-Fender | EI5GTB: Forget the light. Look at the info page |
| 17:44.06 | coppice | wideband voice is a huge improvement over traditional telephony. 24kb AMR-WB or G.722.1 makes 64k alaw sound dreadful |
| 17:44.07 | [TK]D-Fender | EI5GTB: None of the rest of what you have indicated confirms anything of value |
| 17:44.51 | ManxPower | EI5GTB: IIRC, SIPura (aka Linksys) ATAs default to not providing dialtone if the device is not registered. |
| 17:44.53 | EI5GTB | according to the info page the hook state is on. call 1 state: idle. Reg. stat.: online |
| 17:45.22 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:45.48 | ManxPower | do a "sip show peers" and see if Asterisk thinks the device is registered (with the correct IP address, port does not really matter). I assume you've checked the ATA's web interface status page? |
| 17:45.51 | EI5GTB | accoring to "sip show peers" it is online and unmonitord |
| 17:46.13 | jblack | The bailout failed the house. |
| 17:46.17 | ManxPower | EI5GTB: put qualify=yes on there for testing. I don't like to leave it on by default. |
| 17:46.28 | ManxPower | jblack: good. |
| 17:46.32 | coppice | we're all doomed |
| 17:46.37 | [TK]D-Fender | jblack: Safe at first! |
| 17:46.40 | jblack | The dow is down 618 and falling. |
| 17:46.47 | freakazoid0223 | nice |
| 17:47.04 | *** join/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 17:47.11 | jameswf | anyone got a snow board |
| 17:47.20 | freakazoid0223 | gold is up 3% |
| 17:47.22 | ManxPower | jblack: I would prefer a delay while more issues are addressed then pass a panic induced bill that really is not effective (aka Patriot Act) |
| 17:47.38 | jblack | I think there's bettery solutions than that thing. |
| 17:47.44 | EI5GTB | ManxPower, nothing |
| 17:47.53 | ManxPower | EI5GTB: what is nothing? |
| 17:48.06 | EI5GTB | adding qualify=yes makes no changes |
| 17:48.33 | ManxPower | EI5GTB: It should no longer be "Unmonitored" |
| 17:48.41 | jblack | predicts that bush will be on TV shortly, saying "The house of repressents are terrorists" |
| 17:49.00 | EI5GTB | yes, its monitored, but no dial tone |
| 17:49.16 | ManxPower | jblack: I'll watch my favorite left liberal news show and get the summary 8-) |
| 17:49.27 | EI5GTB | when i call it, as far as * sees it is ringing |
| 17:49.28 | ManxPower | EI5GTB: what is the latency listed? |
| 17:49.34 | EI5GTB | 13ms |
| 17:50.38 | ManxPower | EI5GTB: My next suggestion can be complicated. Configure syslog to accept remote logging messages, configure the ATA to log to syslog of the syslog server, turn up the debugging on the ATA and watch the logs. |
| 17:51.15 | EI5GTB | ok, ill have to see if this piece of.... can do that |
| 17:51.44 | ManxPower | EI5GTB: SIPura/Linksys normally work very well. |
| 17:51.49 | jameswf | be cool to adapt the line racer game to the stock market graph |
| 17:52.17 | jameswf | jblack: those are terrorist no one would appose eliminating |
| 17:52.31 | ManxPower | coppice: thanks for the information. |
| 17:52.33 | jameswf | My prediction: Marshal law prior to NOV 11th |
| 17:52.59 | jameswf | then bush's 3rd term :) |
| 17:53.15 | jameswf | viva la revolution |
| 17:53.32 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:53.33 | nny_1 | so I have a customer complaining about dropped calls on a T1 and I have no idea where to start with it. Every day a new freaking issue. First it's echo, now calls get dropped randomly. I am so f*king close to ordering a Sangoma card, ripping out this Digium tdm card and pissing on it. |
| 17:53.51 | jameswf | nny what kernel |
| 17:54.22 | EI5GTB | ManxPower, yea, it can do that. ill sort that put after dinner and homewrok |
| 17:54.22 | ManxPower | nny_1: Welcome to "I thought I tested everything and now that I put it in production its not working" land. |
| 17:54.26 | EI5GTB | thats for your help |
| 17:54.32 | jameswf | nny_1: this may sound silly but have you called support |
| 17:54.35 | nny_1 | 2.6.18-8.1.14.3. |
| 17:54.39 | nny_1 | jameswf: yeah |
| 17:54.42 | ManxPower | nny_1: do you get HDLC errors? |
| 17:54.56 | nny_1 | jameswf: I get "Hey you should try this." although never stating why |
| 17:55.02 | nny_1 | ManxPower: only when it is down |
| 17:55.15 | ManxPower | The only time I've personally seen a Sangoma card work where a Digium card did not was with HDLC errors and then only with older Digium cards. |
| 17:55.38 | jameswf | I had a customer who was in crc error hdlc abbort land, swapped cards cables everything a week later the phone company says our bad we had a bad box |
| 17:56.25 | ManxPower | nny_1: I recommend you make the system fail over to calling out via an ITSP, get the incoming calls forwarded to a VoIP DID if your T-1 is down. |
| 17:56.46 | nny_1 | jameswf: ManxPower I have been logging verbosely the messages verbosely |
| 17:56.49 | nny_1 | god |
| 17:56.51 | jameswf | nny_1: if you have hdlc abborts and it is a hardware problem then the errors should remain with a loopback in place |
| 17:56.52 | ManxPower | then you can actually diagnose and fix rather than run around rushing because people are screaming at you. |
| 17:56.57 | nny_1 | one sec phone |
| 17:57.34 | *** join/#asterisk jan1607 (n=niewerth@dslb-084-061-207-082.pools.arcor-ip.net) |
| 17:57.37 | jameswf | I would set the t1 to clear with a loopback and run patlooptest |
| 17:57.39 | jan1607 | hi all |
| 17:58.00 | nny_1 | jameswf: I have |
| 17:58.12 | jameswf | does it fix it |
| 17:58.17 | nny_1 | ManxPower: yeah I am trying to get the provider to forward to an ITSP line |
| 17:58.36 | nny_1 | ManxPower: jameswf only time I have hdlc errors is the 2 times the T1 has gone down |
| 17:58.36 | jameswf | if a loopback + patloop test fails it is hardware/driver |
| 17:58.49 | Blackvel | good luck nny_1 |
| 17:58.58 | Blackvel | have a good evening. bye all |
| 17:59.09 | nny_1 | jameswf: patlooptest ran clear |
| 17:59.16 | nny_1 | jameswf: did full testing locally |
| 17:59.19 | jameswf | then it is cabling/telco |
| 17:59.37 | nny_1 | jameswf yeah I agree |
| 17:59.45 | nny_1 | but they just come out and blame the system |
| 18:00.00 | nny_1 | and test the line in some generic fashion |
| 18:00.12 | jan1607 | anybody experienced asterisk segfaults in conjuction with sangoma a104-cards and E1-lines (2 in cpe, 2 in net-mode)?? |
| 18:00.25 | nny_1 | yeah zttool reports 1 irg miss |
| 18:00.27 | jameswf | nny_1: this is how it is dealing with a telco... a sangoma card will not fix the telco and their games |
| 18:00.46 | jan1607 | (Asterisk 1.4.21.2, libpri 1.4.7, zaptel 1.4.12.1) |
| 18:01.08 | *** join/#asterisk sacitec (n=tobi@201.144.211.82) |
| 18:01.15 | jameswf | put a loopback in the smart jack |
| 18:01.28 | nny_1 | jameswf: yeah i hope you're right.. I just have no other resources to expend.. they are trying to say the system is inferior |
| 18:01.37 | jameswf | yay open source |
| 18:02.57 | jameswf | nny_1: get to an engineering level @ digium and conference in the telco. allow a digium engineer to spank the telco techs a bit and they will likely swap their equipment and all will be well |
| 18:03.05 | ManxPower | nny_1: put it in a system with a different motherboard |
| 18:03.12 | nny_1 | jameswf: yeah basically we had no issues (other than echo) until last week when the telco started mucking about |
| 18:03.23 | nny_1 | ManxPower: not sure if I have that option but I can try |
| 18:03.37 | nny_1 | jameswf: yeah that's what I am working on |
| 18:03.50 | jameswf | ManxPower: it cleared a patlooptest |
| 18:05.28 | *** join/#asterisk feeds (n=feeds@85-135-232-6.adsl.slovanet.sk) |
| 18:09.17 | nny_1 | ManxPower: jameswf thanks for the advice. I feel pretty strongly this issue is with the telco, but they are such dicks about anything it is frustrating. |
| 18:09.24 | nny_1 | sorryto rant in channel, gonna go try to kick some ass |
| 18:10.06 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 18:10.33 | ManxPower | nny_1: get the failover working ASAP |
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| 18:14.21 | *** part/#asterisk Guest267890 (n=feeds@85-135-232-6.adsl.slovanet.sk) |
| 18:17.39 | nny_1 | ManxPower: working on that now |
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| 18:32.29 | Qwell | *sigh*.. I hate being a computer enthusiast sometimes. |
| 18:32.36 | Qwell | $450 to get my desktop and laptop working properly again |
| 18:33.35 | *** join/#asterisk sacitec (n=tobi@201.144.211.82) |
| 18:34.01 | Qwell | need a new PSU? Sorry, you'll need a new MB too. Oh, and your CPU doesn't work in your new MB, so you need a new CPU. And guess what - you need new RAM too |
| 18:35.54 | leif[astricon] | Qwell: why a new MB for a PSU? |
| 18:36.13 | Qwell | leif[astricon]: unless I get an old PSU |
| 18:36.14 | leif[astricon] | you should have been able to just get a PSU, unless the MB was toasted |
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| 18:38.01 | *** join/#asterisk envisean (n=envisean@ip70-181-105-234.oc.oc.cox.net) |
| 18:41.24 | *** join/#asterisk edibrac (n=edibrac4@206.173.193.34.ptr.us.xo.net) |
| 18:43.10 | edibrac | i guess this might sound silly, but is there a website out there that has recordings of commonly known phone sounds? i mean, for example, towards the end of a conference call, I observed a buzzing sound which people have said comes and goes |
| 18:43.53 | *** join/#asterisk gaetronik (n=gaetan@200.111.138.170) |
| 18:43.57 | Simon-- | electrical buzz? somebody's phone is probably too close to a lamp |
| 18:43.58 | edibrac | i guess it would be hard to match the "bad sound" to a certain cause.. since it might be hardware dependent |
| 18:44.56 | gaetronik | Hi there |
| 18:45.14 | gaetronik | i've an issue with agents logoff |
| 18:45.54 | gaetronik | when an agent is logged, in pause and this computer shutdown it remains conected for asterisk |
| 18:46.25 | gaetronik | is there any way to have a qualify like option for agents? |
| 18:47.00 | *** join/#asterisk ctooley (n=ctooley@209.33.108.195) |
| 18:48.48 | bpgoldsb | Anyone know how to get the functionality of 1.2's 'Macro' command in AEL? |
| 18:49.48 | envisean | edibrac: there are a bunch, we just put up a new soundpack for free that's open sourced http://www.flowroute.com/services/voice/ with Pat Fleet (aka Ma Bell and the original AT&T voice) |
| 18:50.25 | envisean | edibrac: ahh i'm sorry i misread that, i thought u wanted sound packs for prompts |
| 18:50.26 | *** join/#asterisk hardwire (n=hardwire@rdbk-4562.mtaonline.net) |
| 18:50.37 | edibrac | envisean: yeah it was badly worded |
| 18:50.38 | envisean | edibrac: are you looking for sound effects? |
| 18:51.44 | edibrac | sound effects? not really but it sounds like a fun idea for prank calls |
| 18:51.48 | envisean | edibrac: I don't believe that sound you are talking about is intentional, I may be wrong though |
| 18:52.11 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:52.35 | edibrac | no i was hoping somehow that certain odd sounds during a call were telltale signs of where the problem lied |
| 18:52.35 | *** join/#asterisk leif[astricon] (n=Leif@65-122-15-169.dia.static.qwest.net) |
| 18:52.45 | edibrac | lies. |
| 18:53.53 | *** join/#asterisk sacitec (n=tobi@201.144.211.82) |
| 18:54.37 | envisean | edibrac: like debugging sounds? heh |
| 18:55.19 | edibrac | yeah i know, it's not like there's something equivalent to motherboard BIOS beeps |
| 18:55.37 | edibrac | i wish there were. |
| 18:56.08 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 18:58.25 | edibrac | oh well that intermittent problem isn't totally crucial |
| 18:59.03 | edibrac | can i do IF/THEN logic in zapata.conf? |
| 18:59.37 | edibrac | we have a new block of DIDs that i want totally separate and not to go to the default context that we currently use |
| 19:00.46 | envisean | edibrac: yeah, you can route specific DIDs somewhere else definitely |
| 19:01.08 | edibrac | i'm not sure if it matters if i "branch off" from zapata.conf or from extensions.conf |
| 19:01.51 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:01.57 | edibrac | seems i can technicallly accomplish it either way. starting it off in zapata.conf seems the most logical way |
| 19:02.03 | *** join/#asterisk Segnale007 (n=Segnale0@host112-254-dynamic.32-79-r.retail.telecomitalia.it) |
| 19:03.59 | envisean | edibrac: yeah, shouldn't matter too much, i would probably do it in the extensions.conf though just to keep extensions & DIDs in the same file |
| 19:04.16 | envisean | you want a DID to be associated directly w/ an extension? or to follow a completely separate dial-in system? |
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| 19:06.05 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 19:06.48 | rwaite | hi all. sometimes one of the tdm400p zap channels will get stuck in an endless loop in an ivr |
| 19:06.55 | rwaite | i have no idea how to diagnose this |
| 19:07.12 | hardwire | you should set an absolutetimeout while diagnosing this |
| 19:07.17 | hardwire | to keep you from having some fancy bills |
| 19:07.19 | hardwire | or causing some |
| 19:07.24 | rwaite | lol |
| 19:07.34 | hardwire | I'm serious |
| 19:07.38 | edibrac | holy crap. stocks are getting killed |
| 19:07.39 | hardwire | a bug like that almost cost my company $10k |
| 19:07.44 | rwaite | is it the zap card though? |
| 19:07.51 | rwaite | failing to hangup the channel? |
| 19:07.58 | hardwire | rwaite: what POTS type? |
| 19:08.14 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 19:08.22 | rwaite | a regular pots line from our local provider |
| 19:08.25 | rwaite | in the US |
| 19:08.29 | hardwire | kewlstart? |
| 19:08.34 | rwaite | yes |
| 19:08.55 | hardwire | anyhoot.. you said endless loop in the IVR |
| 19:09.11 | hardwire | like it keeps moving around in the dialplan? |
| 19:09.32 | hardwire | or does the audio stop and start repeating itself like a broken cd? |
| 19:09.49 | rwaite | the dialplan is an endless loop |
| 19:10.01 | hardwire | so not really a pots issue? |
| 19:10.02 | rwaite | im watching with asterisk -r and it just keeps going thru |
| 19:10.19 | edibrac | is there an equivalent to denyhosts, a python script that blacklists your ip if your ssh login fails too many times |
| 19:10.22 | rwaite | well, that's my question. what would cause a line to get "lost" i mean someone isnt actually on the line |
| 19:10.30 | hardwire | edibrac: fail2ban |
| 19:10.38 | hardwire | it's python.. and it's apt-gettable |
| 19:10.44 | rwaite | i assume they've hung up and the ivr keeps going as if they are still connected |
| 19:10.52 | edibrac | hardwire: i can use it for conference lines right? |
| 19:11.47 | edibrac | er meetme.conf lines |
| 19:11.47 | hardwire | edibrac: uh.. sure.. fail2ban can take regular expressions from log files and do whatever |
| 19:11.53 | edibrac | ah it's like a more generic denyhosts? |
| 19:11.53 | hardwire | rwaite: how often does this happen? |
| 19:12.18 | hardwire | edibrac: sure |
| 19:12.23 | hardwire | check it out |
| 19:12.51 | hardwire | rwaite: if it happens often enough you should take a peek at the phone bill you get from your telco |
| 19:13.20 | hardwire | see if the call time differs from when the remote phone actually hangs up |
| 19:14.04 | hardwire | rwaite: does asterisk show the hangup event? |
| 19:14.40 | rwaite | these are incoming calls so they are not chargable |
| 19:14.50 | rwaite | this happens like, 3 or 4 times a week |
| 19:14.52 | hardwire | rwaite: our telco shows incoming calls for some reason, on the bills |
| 19:15.09 | hardwire | rwaite: hmm.. what did your telco say? |
| 19:15.28 | rwaite | i recently have been just "zap restart"ing it |
| 19:15.53 | rwaite | but i cant be sure if its a bug with my dialplan or a bug in asterisk |
| 19:16.06 | hardwire | rwaite: with ivr you should always have it timeout somehow |
| 19:16.12 | hardwire | just in case bugs like this happen |
| 19:17.15 | hardwire | like.. you don't really expect somebody to listen to "press 1 for english, 2 for igpay atinlay, ..." for an hour straight |
| 19:17.17 | rwaite | another race condition it looks like |
| 19:17.35 | rwaite | because if i do a "core show channels" it shows output, but if i do it again, no output |
| 19:18.00 | rwaite | im like this --->>><<<<--- close to installing 22rc5 |
| 19:20.00 | *** join/#asterisk grEvenX (n=even@c9A7F8BC3.dhcp.bluecom.no) |
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| 19:21.03 | rwaite | i sometimes wonder if i should move back to 1.2 to escape the bugs |
| 19:21.13 | rwaite | but then again, there are probably bugs there too |
| 19:23.10 | C4colo | can someone take a look at this and tell me if they see anything wrong with it? |
| 19:23.11 | C4colo | http://pastebin.ca/1213936 |
| 19:24.06 | citywok | connecting 2 digium 4 port T1 cards i need to use a T1 crossover, not a normal Cat5 patch cable, correct? |
| 19:24.35 | jan1607 | citywok: correct |
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| 19:25.28 | *** part/#asterisk lasko (n=chatzill@70.102.15.210) |
| 19:28.57 | C4colo | what is called an ethernet crossover will work as well, just needs to be TIA/EIA 568-A on one end and TIA/EIA 568-B on the other end |
| 19:30.38 | citywok | gar, i've been having problems talking to my pBX over E&M (asterisk -> old inter-tel pbx), i can dial into asterisk from the pbx, but not into the pbx from asterisk -- i decided i would connect to asterisk boxes via t1 and see if i could make them talk. i tried using em, em_w, and now i'm trying to use fxoks/fxsks -- still no joy -- am i retarded or am i missing something important? |
| 19:31.00 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:31.41 | citywok | good to know C4colo -- i was wondering if the t1 crossover was the same, but hadnt put enough effort into it to look it up :-) |
| 19:31.53 | citywok | i just know swap pins 1&2 with pins 4&5 |
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| 19:48.18 | ManxPower | Qwell: That is why I hate upgrading anything. |
| 19:48.24 | *** join/#asterisk errr_ (n=errr@fedora/errr) |
| 19:48.35 | Katty | errr_: you haz a tail! |
| 19:48.59 | Qwell | huh? |
| 19:49.46 | Linuturk | "The internet is a telephone system that's gotten uppity." -Clifford Stoll |
| 19:49.47 | Linuturk | lol |
| 19:49.59 | Qwell | ManxPower: what did I miss? |
| 19:50.12 | Qwell | oh |
| 19:50.21 | Qwell | ManxPower: yeah - want to fund the purchases? :p |
| 19:51.00 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:51.35 | lesouvage | #astricon |
| 19:51.40 | Qwell | is over |
| 19:51.44 | leif[astricon] | totally |
| 19:51.47 | leif[astricon] | bit late |
| 19:51.54 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc6, 1.4.22-rc5 (2008/09/09), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc4+2.0.0-rc2 (2008/09/08), Zaptel 1.4.12.1 (2008/09/09), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 19:52.10 | *** join/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 19:52.22 | lesouvage | you are all right :-( |
| 19:52.37 | *** join/#asterisk devhen (n=devhen@66.236.68.230.ptr.us.xo.net) |
| 19:56.36 | *** join/#asterisk smth (n=chatzill@CPE00014e01df2e-CM00195eda2522.cpe.net.cable.rogers.com) |
| 19:58.40 | citywok | why can i dial into asterisk from my pbx, but not dial out from asterisk into my pbx? -- why can i not dial from asterisk to asterisk? I have followed the basic configurations online trying PRI, E&M, and FXO/FXS, and no matter what i do, i can never dial into my pbx, or from asterisk server to asterisk server, over quad port T1 cards |
| 19:59.42 | citywok | according to my telco, currently sending calls into an inbound E&M T1, they dont send ANI or anything, just 4digit DNIS, so i'm trying to replicate that, but to no avail, my pbx never receives the digits (i can see in debug mode that it sends them) |
| 20:01.19 | [TK]D-Fender | citywok: Is that the only issue? that you aren't getting the digits? |
| 20:02.08 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 20:02.38 | citywok | [TK]D-Fender: i'm not certain if thats the _only_ issue, but it's the issue i'm currently facing. there could probably be a million reasons its happening :'-( |
| 20:02.57 | citywok | for instance, when i dial into my pbx, i get chan_zap.c:5142 zt_write: Dropping frame since I'm still dialing on Zap/25-1.. |
| 20:03.08 | [TK]D-Fender | citywok: pastebin an incoming call's CLI output at verbose 10, and your dialplan. |
| 20:03.29 | citywok | incoming into my asterisk box, from my PBX? -- the calsl that work? |
| 20:04.46 | citywok | here is a call from my pbx -> asterisk -> relayed over SIP to my cellphone |
| 20:04.47 | citywok | http://pastebin.com/d2ffc8826 |
| 20:05.40 | citywok | the instant i open the trunk, i get messaging in the CLI output, as i push the digits, i see it tell me each DTMF digit in the cli, and then it works |
| 20:06.34 | citywok | here is my dialplan for that: http://pastebin.com/d5a535c13 |
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| 20:07.54 | [TK]D-Fender | citywok: Your description looks like its going in circles. What is that Zap call coming in from? |
| 20:08.20 | citywok | coming in from my PBX to asterisk, and then forwarding the call out through SIP (acting as a SIP bridge for a non-sip pbx) |
| 20:08.49 | [TK]D-Fender | citywok: I see, so it LOOKS like that's what it is doing... |
| 20:08.57 | citywok | yea, and it works that direction |
| 20:09.05 | citywok | but i can not call back the other direction |
| 20:09.05 | [TK]D-Fender | citywok: show me the other |
| 20:09.23 | [TK]D-Fender | citywok: I asked you to show me the PROBLEM. |
| 20:10.48 | citywok | i dial 2 in my xlite sip phone, it follwos the dialplan: http://pastebin.com/dbc7949a |
| 20:11.04 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 20:11.17 | citywok | you can see i was tryig a few combinations to see what might work, if anything, but to no avail |
| 20:12.43 | citywok | is it this? zt_handle_event: Sent deferred digit string: T1593w |
| 20:13.00 | citywok | is that just not what it should be sending? i'm not sure what the T means in DTMF land |
| 20:13.19 | outtolunc | tone vice pulse |
| 20:14.23 | jblack | whoah. the market closed down 750? |
| 20:14.48 | citywok | holy shit, 770 points down |
| 20:14.48 | leif[astricon] | yep |
| 20:15.06 | citywok | [TK]D-Fender, this is what it looks like with immediate=no http://pastebin.com/d286e51cf -- the earlier one was immediate=yes |
| 20:15.07 | leif[astricon] | that's what happens when you tell the market you're going to bail them out, then not |
| 20:16.15 | *** part/#asterisk GreenCult (n=greencul@200.48.85.18) |
| 20:22.12 | [TK]D-Fender | citywok: Ok, not sure what to advise here... |
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| 20:23.15 | citywok | can you not see anything wrong? It all seems fine to me, but alas i get into my pbx and it says "the extension you dialed does not exist", and my PBX doesnt show that it received anything |
| 20:23.58 | codefreeze-lap | bpgoldsb: did you get an anwer? |
| 20:24.34 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 20:26.04 | codefreeze-lap | bpgoldsb: look at http://voip-info.org/wiki/view/Asterisk+AEL2#Macros |
| 20:27.48 | [TK]D-Fender | ok, checkout time. Heading home. Later all |
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| 20:31.05 | logicwrath | im trying to register a softphone over a cisco VPN client and it is failing. I cannot even see it attempting in the asterisk console. 2 other people are also using a VPN client, is there something I can test? |
| 20:31.18 | logicwrath | the other two people connect fine |
| 20:33.59 | nny_1 | having dtmf issues on a T1 |
| 20:34.15 | nny_1 | client says it's not getting passed. Any safe setting adjustments I can make? |
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| 20:36.58 | exothermc | are there some recent rpms out there for asterisk? |
| 20:37.08 | smth | <PROTECTED> |
| 20:37.28 | UnixDawg | 1.4.21 |
| 20:37.48 | UnixDawg | i know that there should be a new iso coming out |
| 20:38.15 | bpgoldsb | codefreeze-lap: I didn't get one, but I figured that out after enough time. Appreciate the effort though |
| 20:39.11 | jaytee | anyone here ever experienced flickering on Polycom displays after upgrading SIP firmware? |
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| 20:39.34 | codefreeze-lap | bpgoldsb: np. I usually find what I need about 10 seconds after I ask.... :) |
| 20:39.44 | smth | particularly, only one inband digit be recognized at ' background' . it was not collected at 'Waitexten' or ' Read' .. any idea? |
| 20:42.33 | *** join/#asterisk Alan_Hicks (n=alan@cardinal.lizella.net) |
| 20:44.07 | Alan_Hicks | Howdy folks. I was wondering if fxotune helps with echo cancellation on Digium cards with hardware echo cancellation. I checked the man page but couldn't determine this answer from it. I have a brief period of echo when a user dials in. The echo vanishes by the time the caller begins to speak. |
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| 20:45.23 | jaytee | Alan_Hicks, do you have echotraining=yes in zapata.conf? |
| 20:45.38 | Alan_Hicks | jaytee: Yes. Dumb-ass me forgot to pastebin that file. |
| 20:45.57 | jaytee | comment it out and restart zaptel |
| 20:46.12 | Alan_Hicks | But according to the sample zapata.conf that shouldn't have an effect on hardware echo cancellation cards. |
| 20:46.26 | jaytee | or better yet, set it to =no and restart |
| 20:46.27 | Alan_Hicks | Seriously? |
| 20:46.39 | jaytee | it shouldn't......but |
| 20:47.29 | Alan_Hicks | I was intending to do that, but since the business is currently open, I didn't want to shut-down asterisk for even a brief period of time. |
| 20:47.39 | jaytee | are these Digium cards with HWEC for analog or digital? |
| 20:47.42 | Alan_Hicks | I'll give that a try after COB though. |
| 20:47.47 | Alan_Hicks | analog. |
| 20:48.00 | Alan_Hicks | wc24xxp IIRC |
| 20:49.47 | *** join/#asterisk anarcat (n=anarcat@pdpc/supporter/active/anarcat) |
| 20:49.55 | jaytee | that's probably normal behavior for the card then, analog circuits tend to be more prone to echo. If it's only happening when the line first goes active and drops out by the time someone starts talking then it's working, if it continues or happens in the middle of a call then it's a problem |
| 20:50.01 | anarcat | so our termination provider crapped on me |
| 20:50.10 | anarcat | and we're wanting to add a second one |
| 20:50.31 | anarcat | so i'm wondering what's the best way to have two termination providers in a dial plan? |
| 20:50.53 | anarcat | i take it that Dial(foo&bar) will not work in this case because it will ring the two providers at the same time |
| 20:51.01 | Alan_Hicks | jaytee: Ick. The customer doesn't like the echo at all. Wants it all gone. |
| 20:51.47 | Alan_Hicks | Granted, I haven't yet run fxotune on the modules. Would that potentially help at all? |
| 20:52.11 | jaytee | it might, I've never run fxotune on a card with HWEC |
| 20:52.18 | Alan_Hicks | TY |
| 20:52.27 | jaytee | just on old crap x100p cards about twice a day. |
| 20:54.37 | jaytee | I have a couple TE212P cards and I've never had to do any tweaking for echo on my T1's. |
| 20:58.13 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
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| 21:03.23 | *** part/#asterisk anarcat (n=anarcat@pdpc/supporter/active/anarcat) |
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| 21:09.23 | *** join/#asterisk eric_hill (i=EricHill@204.94.175.11) |
| 21:10.07 | eric_hill | Anyone know what bit to twiddle in the Polycom XML to get the Fwd softkey to work with Asterisk? |
| 21:10.59 | [TK]D-Fender | eric_hill: it does already |
| 21:11.14 | [TK]D-Fender | eric_hill: nothing to play with |
| 21:13.40 | eric_hill | I think my XML config got 'borked and it stopped working at some point. I think it has to do with the <reg/> section.... |
| 21:15.12 | [TK]D-Fender | eric_hill: Nope. |
| 21:15.30 | eric_hill | Yep, sure enough. |
| 21:15.31 | [TK]D-Fender | eric_hill: go look at actual SIP debug & CLI to see whats happening. |
| 21:15.45 | jplank | anyone ever play with Asterisk BE? Is it just a paid for version of AsteriskNOW? |
| 21:16.08 | [TK]D-Fender | jplank: No, ABE is just a paid ver of * with a few proprietary add-ins. |
| 21:16.19 | jplank | anything worth its cost? |
| 21:16.22 | Qwell | and support... |
| 21:16.30 | [TK]D-Fender | jplank: And one for which more support may be paid for |
| 21:16.44 | [TK]D-Fender | Qwell: What support does it come with by default? |
| 21:16.46 | jplank | my boss keeps grilling me why most vendors sell it instead of just building it themselves |
| 21:16.53 | Qwell | [TK]D-Fender: no idea |
| 21:17.03 | [TK]D-Fender | Qwell: Everything I recall says "zip" |
| 21:17.10 | Qwell | let's see |
| 21:17.30 | [TK]D-Fender | jplank: What vendors? Selling BE? |
| 21:17.32 | jplank | I was thinking about paying the $300 for the 10 call license to see if its worth anything |
| 21:17.32 | Qwell | yeah, I think you need at least a silver subscription to buy it |
| 21:17.37 | *** join/#asterisk angryuser (n=Miranda@128.237.146.195.dynamic.adsl.abo.nordnet.fr) |
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| 21:17.40 | jplank | shelton|johns |
| 21:17.45 | jplank | EUS Communications |
| 21:17.59 | jplank | pretty much every asterisk vendor we've come up against lately |
| 21:18.39 | [TK]D-Fender | jplank: Maybe covering their butts with an air of "officiality" and "credibilty" in not using the "OSS" release |
| 21:18.46 | jplank | ahhhh |
| 21:18.56 | jplank | well, thats pretty much what I've come up with |
| 21:19.03 | dlynes | Is there a known issue with dtmf problems where an asterisk box receives the dtmf in on sip just fine (1.4.18), but when it transmits it over to another box (1.4.19.1), some of the dtmf tones get duplicated, and/or others get corrupted into some other digit? |
| 21:19.03 | jplank | but I was wondering if it was more |
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| 21:19.22 | jplank | it seems silly that its based off of 1.2, so I thought there had to be something |
| 21:19.31 | [TK]D-Fender | jplank: Which is a complete illusion and you are forced to run notably older releases. This would likely have an incremental effect with the 1.6 release changes |
| 21:19.48 | jplank | thats what I was thinking |
| 21:19.49 | ctooley | [TK]D-Fender, Compatibility testing of modules included, inclusion of modules that can't be open sourced and "no question" support if you choose to invoke it. |
| 21:20.13 | jplank | thats why I was wondering what the draw was (besides the support) |
| 21:20.23 | [TK]D-Fender | ctooley: For the right price per hour I'm sure they wouldn't need to CARE what the problem is :) |
| 21:20.44 | jplank | the way I look at it though, if you can't support the system, you probably shouldn't be selling it |
| 21:20.55 | jplank | isn't that were all the money is anyway |
| 21:20.59 | jplank | support contracts |
| 21:21.07 | [TK]D-Fender | jplank: Its great being able to blame someone else, no? |
| 21:21.11 | ctooley | [TK]D-Fender, The price per hour is definitely not unreasonable, I charge more per hour to support my customers. And they do care, anything that's been modified will not be supported |
| 21:21.22 | *** join/#asterisk leif[astricon] (n=Leif@65-122-15-169.dia.static.qwest.net) |
| 21:21.30 | jplank | [TK]D-Fender: thats def true |
| 21:21.38 | citywok | [TK]D-Fender, does this look possible? http://bugs.digium.com/view.php?id=10360 |
| 21:21.43 | [TK]D-Fender | ctooley: In the grand scheme of things I rank BE as a "whatever". |
| 21:22.02 | citywok | <PROTECTED> |
| 21:22.21 | [TK]D-Fender | citywok: Looks nearly identical no? |
| 21:22.42 | citywok | yea, exactly. seems like it's an old enough post htough that it should be bug-fixed (they say 1.4.8, i am 1.4.21) |
| 21:25.21 | [TK]D-Fender | citywok: Well its something to go on. Maybe you should attempt to re-open that case |
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| 21:34.52 | citywok | [TK]D-Fender, i've found if i set my PBX side trunks to OPX instead of E&M, that i am able to dial into my pbx from asterisk like i should be able to (OPX is for the internal PBX communications, so when i dial in i am an internal station instead of a remote dial in), and leave the asterisk side alone, it works |
| 21:36.12 | citywok | \ |
| 21:37.00 | *** join/#asterisk devhen|Work (n=devhen@216.194.118.110) |
| 21:37.51 | citywok | that tells me its probably something in my pbx then, or the method of dialing into it (*ANI*DNIS*) or something like that could be wrong? no? |
| 21:38.09 | *** part/#asterisk Alan_Hicks (n=alan@cardinal.lizella.net) |
| 21:39.05 | *** part/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 21:42.25 | angryuser | [TK]D-Fender : you pointed on a file called asterisk-skype 2-3 days ago, it was g729 promts packed with that strange name |
| 21:42.49 | Qwell | asterisk-sounds? |
| 21:43.05 | Qwell | got skype on the brain? :p |
| 21:43.18 | [TK]D-Fender | angryuser: And no, I didn't |
| 21:43.23 | angryuser | oh just a anonymous pack, uploaded by someone |
| 21:49.33 | *** join/#asterisk propellerhead (n=yogurt2u@host130.190-226-46.telecom.net.ar) |
| 21:51.15 | eric_hill | We have inbound caller number on our dedicated ISDN 800 number, but not name. Is that something that the carrier could provide, or is 800 number service numeric-only for inbound calls? |
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| 22:26.20 | lanthica | hey all |
| 22:26.57 | lanthica | I just kind of stumbled across asterisk, and was just wondering a little more about it |
| 22:27.16 | *** join/#asterisk simprix (n=simprix@c-24-11-27-93.hsd1.mi.comcast.net) |
| 22:27.39 | lanthica | so if I wanted to setup a menu, and interact with the server, would I do that via C, or a scripting language or is there some library I import in whatever language I'm using? |
| 22:31.56 | [TK]D-Fender | lanthica: What kind of "interaction"? |
| 22:32.22 | [TK]D-Fender | lanthica: *'s call processing is handled by the dialplan through which you can call extrenal scripts, etc as well. |
| 22:32.28 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-ca60e7d0a746d145) |
| 22:32.35 | [TK]D-Fender | lanthica: Go download the book, and get started. |
| 22:32.36 | [TK]D-Fender | ~book |
| 22:32.37 | jbot | hmm... book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 22:32.38 | [TK]D-Fender | ^^^^^^^^^ |
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| 23:08.11 | jaytee | wow, what a day |
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