| 00:00.44 | flenders | ~pb |
| 00:00.55 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 00:01.05 | *** join/#asterisk rene- (n=rene@200.34.66.137) |
| 00:01.39 | JT | J4k3: the plesiochronous timing isn't working for you? ;) |
| 00:02.17 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 00:02.32 | J4k3 | JT: something isn't. I'm personally laying my money on the far-end router right now |
| 00:02.44 | ruied_ | [TK]D-Fender, yesterday you helped me making a call redirection qith the inbound number authentication (CallerIDString). Is there any way so I can have a kind of phone numbers Array so several inbound numbers could make a call redirection from the pstn to voip? |
| 00:02.51 | J4k3 | JT: weird stuff... the latency/jitter was starting every 63 seconds last night, today its every 59 seconds. |
| 00:03.03 | J4k3 | has no relation to traffic, at least not my traffic. |
| 00:03.28 | [TK]D-Fender | ruied_, there are dozens of ways to do this. |
| 00:04.15 | [TK]D-Fender | ruied_, If we want to stick to the simplest built-in options go read up on AstDB on the wiki and "show function DB" |
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| 00:06.07 | ruied_ | ok, going to check that, I was thinking of reading a text file with the phone numbers, but did'nt know if it is possible... |
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| 00:08.20 | DrukenHME | has someone done up an automated 411 system yet? |
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| 00:08.52 | [TK]D-Fender | ruied_, If you learn AGI, and want to dcode it, sure. |
| 00:10.03 | stridernzl | does anyone use DynDNS with eyebeam and sucessfully have a remote extension working ? I'm more specifically interesting in any troubles you had? |
| 00:15.23 | rene- | hey i got mitel to do paging with asterisk! |
| 00:15.42 | rene- | what i dont know, and would like to learn is how to do hint-subscriptions |
| 00:16.22 | rene- | i want BLF with it, but well the progamabble keys only allow for speed dial and for other dumb things like seeing the phones call logs and stuff |
| 00:16.51 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 00:17.21 | rene- | the only interesting thing was with sip advanced mode, where you can define a line as a sylantro or broadsoft line appearance |
| 00:18.12 | rene- | with the sylantro line appearance, it says in the phone line error seizure, and i see a suscribe flying to asterisk, with a weird Event: call-info, asterisk prompty returns a 489 Error |
| 00:18.48 | rene- | what event should i see |
| 00:18.56 | rene- | ? when a phone is trying to subscribe for BLF ? |
| 00:19.29 | rene- | and has anyone seen a way to do this with a mitel? |
| 00:20.39 | [TK]D-Fender | rene-, You are describing SIP-B SLA. * does not support this feature |
| 00:20.44 | rene- | what about SIP Notify, can i push the events to the phone from asterisk? i found that the Sip headers for doing page were very similar to those than snom and polycom, mitel is far less documented since is lees used in asterisk |
| 00:21.19 | rene- | [TK]D-Fender: is it possible to use hints? with a mitel? what should the mitel need to send to asterisk? |
| 00:21.33 | [TK]D-Fender | rene-, No. |
| 00:21.43 | [TK]D-Fender | rene-, Oops... strike that. |
| 00:21.48 | rene- | heh |
| 00:22.11 | [TK]D-Fender | rene-, Hints yes.. thats basic PRESENCE. Which I don't know for certain, I'd still bet they support that. |
| 00:22.25 | [TK]D-Fender | rene-, But that IS only for speed-dial & in-use lighting |
| 00:22.36 | rene- | in-use lighting would be very cool to have |
| 00:23.18 | rene- | i have speed dialing but i see no way to use basic presence with the mitel |
| 00:23.57 | rene- | i mean a speed dial key would send an INVITE right? |
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| 00:27.41 | blitzrage | rene-: 5220? (just curious) |
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| 00:32.49 | rene- | blitz |
| 00:33.09 | rene- | i have 5212,5215,5220, and 5224, i would rather have some snoms |
| 00:33.44 | blitzrage | heh :) |
| 00:33.52 | blitzrage | the 5220 is decent... the handset is way too small though |
| 00:34.28 | rene- | yeah but mitel is very closed about them |
| 00:34.35 | blitzrage | no kidding! |
| 00:34.47 | rene- | and there are not popular enough to attract some enterprising hackers |
| 00:35.00 | blitzrage | I have about 12 of them sitting in my basement that have a bad firmware on them... |
| 00:35.03 | blitzrage | any idea how I can reflash them? |
| 00:35.30 | blitzrage | apparently they auto-upgraded themselves, and Mitel wants $150 to reflash each of them... but there is nothing wrong with them |
| 00:35.50 | blitzrage | free phone to anyone who can help me get them reimaged |
| 00:36.01 | JT | lame, reminds me of motorola flashport |
| 00:36.07 | blitzrage | no kidding |
| 00:36.11 | rene- | well the guy before me did mess up some phones but those were sent back to mitel and they fixed them up for free |
| 00:36.22 | rene- | yeah i get nervous when i upgrade the things |
| 00:36.38 | JT | they charge for everything software related with motorola flashport |
| 00:36.43 | rene- | you know they still ship phones with v 4 firmware when they released v 6 like more than a year ago? |
| 00:37.42 | blitzrage | I think these had v4 on them, and the phone had the tftp server at Mitel setup in them when they got them, which upgraded to v6 on the first boot (with no warning), which basically toasted them, and Mitel wouldn't fix them |
| 00:38.13 | JT | so why are you guys using these piece of junk phones? :) |
| 00:38.44 | blitzrage | I got 12 of them for free :) |
| 00:38.57 | blitzrage | but they are basically bricks... been hoping I could find someone who knew how to reflash them |
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| 00:58.57 | tectoris | Is the latest stable version of Asterisk meant to run on a kernel 2.4 setup? |
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| 00:59.44 | mog | asterisk is a userland program |
| 00:59.49 | mog | kernel version does not matter |
| 01:00.00 | mog | zapltel however is now pretty much 2.6 only for the newer cards |
| 01:00.04 | BSD_Tech[laptop] | well it does for zaptel |
| 01:00.19 | BSD_Tech[laptop] | some versions of zaptel dont build on older kernels |
| 01:00.21 | mog | BSD_Tech, see above |
| 01:00.33 | BSD_Tech[laptop] | ok\ |
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| 01:07.22 | tectoris | Actually, I'm playing around with old x100ps and asterisknow in one partition. Op Sys Centos 5 (2.6). Now I want to install asterisk in the "main" partition. Should that work? I heard something about asterisk preferring [2.4]. Should I just go ahead and give it a stab, or is it going to be issuematic? |
| 01:07.47 | Qwell | tectoris: no, asterisk doesn't "prefer" any kernel... |
| 01:07.59 | Qwell | I'm pretty sure 99% of people would recommend using 2.6 though |
| 01:08.25 | tectoris | Qwell - Thanks. Thanks All. |
| 01:08.26 | anzen | is there anything i have to open on a PIX to allow outbound calls? |
| 01:09.26 | mog | x100p will work with 2.4 though |
| 01:10.59 | tectoris | mog - thanks |
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| 01:45.06 | mocker | Anyone know why incoming calls would only give the 4 digits they are dialing and not the full number that was dialed? |
| 01:45.29 | mocker | I'm guessing that's what the PRI is giving and there's not much I can do. |
| 01:46.55 | JT | check what comes in on pri intense debug |
| 01:46.55 | JT | but probably is a telco setting |
| 01:48.11 | mocker | JT: Yup, pri debug confirms. |
| 01:54.29 | JT | right, then that's what you're getting from the telco |
| 01:55.00 | [hC] | just call them and explain and they should be able to help you fix it |
| 01:55.02 | [hC] | or fix it on their site |
| 01:55.04 | [hC] | side |
| 01:59.35 | *** part/#asterisk Ryzer (n=registra@83.101.1.70) |
| 02:00.16 | infinity1 | hey . in ael, how can i assign a variable with a variable? |
| 02:01.47 | infinity1 | e.g. PHONES2=${EXT203}&${EXT205}&${EXT204}; |
| 02:01.50 | infinity1 | doesnt work |
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| 02:20.22 | mrdigital-work | anyone use sendmail? |
| 02:27.06 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 02:27.06 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 02:34.57 | infinity1 | mrdigital-work: rofl |
| 02:35.06 | infinity1 | sendmail eh? you want pain, don't you. |
| 02:35.43 | shido6 | yes |
| 02:35.43 | infinity1 | i haven't touched sendmail.cf in hmmm ..at least 5 years. i've uninstalled it a lot though. |
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| 02:36.14 | shido6 | LOL |
| 02:36.19 | shido6 | whats wrong with your sendmail? |
| 02:36.28 | shido6 | apt-get install post-fix and call it a day |
| 02:36.59 | infinity1 | shido6: here yee! |
| 02:37.57 | JT | shido6: "postfix" even :) |
| 02:39.41 | *** join/#asterisk ixx (i=foobar@cpe-70-112-123-132.austin.res.rr.com) |
| 02:40.40 | ixx | who do you contact if your number has been ported away w/o authorization from your provider? |
| 02:42.47 | *** part/#asterisk dracosilv (n=draco@CPE-65-29-47-173.wi.res.rr.com) |
| 02:42.49 | mosty | the provider of the number |
| 02:43.01 | ixx | hmmm |
| 02:43.07 | ixx | they are just telling me to pick a new number! |
| 02:43.20 | ixx | err my provider |
| 02:43.27 | ixx | I do not know who has the number now |
| 02:43.36 | mosty | call it and see, heh |
| 02:43.36 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:43.50 | ixx | doing that now :) |
| 02:44.06 | ixx | did not work from my mobile... |
| 02:44.08 | ixx | hmm |
| 02:44.47 | ixx | not allowed to place call from my line... |
| 02:44.59 | ixx | maybe pushed for international? |
| 02:45.32 | mosty | anyway, it's only something that the provider of the number can fix, if they refuse you can threaten legal action. at least you can do that here in australia |
| 02:46.24 | ixx | is there someway to do the equivalent of a whois on a number? |
| 02:46.39 | ixx | I have no idea who ported it away |
| 02:46.58 | ixx | and my provider has not answered that question though I have asked repeatedly |
| 02:47.23 | mosty | i don't know, here there is a national body that handles that stuff. don't know what you do in other countries |
| 02:47.50 | ixx | hmmm yeh trying to find info on that |
| 02:48.06 | JT | nanpa |
| 02:50.12 | shido6 | :) |
| 02:52.15 | ixx | hmm looks promising... thanks |
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| 03:04.20 | rrolfe | is there a place on the net for updated documentation on asterisk 1.4? Most of the syntax I have found has been for 1.2 |
| 03:06.35 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
| 03:06.41 | DrukenHME | voip-info.org |
| 03:08.01 | rrolfe | I think I have been there before, but I will check it out again. Thanks :) |
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| 03:08.45 | `Sean | what are some good |
| 03:08.50 | `Sean | Pay as you go providers |
| 03:09.03 | tzanger | sevard: where are you calling to |
| 03:09.08 | tzanger | er `Sean |
| 03:09.31 | `Sean | USA only |
| 03:09.35 | `Sean | maybe canada once in a while |
| 03:11.18 | tzanger | `Sean: nufone, asterlink... I've had experiences with both of them and they work well |
| 03:11.27 | tzanger | both support asterisk development as well |
| 03:11.30 | `Sean | asterlink is whom i use currently |
| 03:11.39 | `Sean | but the problem is hes cut support for toll free calls |
| 03:11.46 | `Sean | so basicly you cant call a toll free number |
| 03:11.51 | `Sean | using there service wich is stupid |
| 03:11.58 | `Sean | seeing how people paid for using there service |
| 03:12.09 | JT | s/there/their/ |
| 03:12.30 | `Sean | s/there/their |
| 03:12.36 | `Sean | heh too late wont work |
| 03:12.38 | `Sean | where is jbot? |
| 03:12.43 | `Sean | ah hes there |
| 03:12.53 | JT | it's lagging, you missed a trailing slash anyway :P |
| 03:13.00 | `Sean | s/there/their/ |
| 03:13.04 | `Sean | heh |
| 03:18.36 | `Sean | JT, any service providers? |
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| 03:39.04 | codefreeze | infinity1: Set(var=${val}&${var2}); |
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| 04:16.02 | TUplink_ | i jost the ability to make a call...... the client registers and i see it on the CLI but i dont even get an error about not being able to find the extension' |
| 04:16.08 | TUplink_ | verbose is 99 |
| 04:16.26 | TUplink_ | is there a way to reload the extensions? |
| 04:16.30 | bkw__ | reload |
| 04:16.44 | TUplink_ | bi was using one like extensions reload |
| 04:16.52 | TUplink_ | but* |
| 04:17.05 | TUplink_ | when i do a reload i dont see anything about extensions |
| 04:18.55 | TUplink_ | i think i lost the extensions part of asteirsk |
| 04:19.08 | TUplink_ | i have 2 server... both 1.4 |
| 04:19.20 | TUplink_ | extensions reload works on the old one |
| 04:19.31 | TUplink_ | but Akita*CLI> extensions reload |
| 04:19.31 | TUplink_ | No such command 'extensions reload' (type 'help' for help) |
| 04:20.46 | russellb | dialplan reload |
| 04:21.23 | TUplink_ | Akita*CLI> dialplan reload |
| 04:21.23 | TUplink_ | No such command 'dialplan reload' (type 'help' for help) |
| 04:21.48 | TUplink_ | have dialplan show |
| 04:22.14 | TUplink_ | errr.... |
| 04:22.16 | russellb | core show modules like pbx_config |
| 04:23.09 | TUplink_ | core show works but not with modules |
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| 04:24.26 | TUplink_ | pbx_config.so Text Extension Configuration 0 |
| 04:24.26 | TUplink_ | 1 modules loaded |
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| 04:26.10 | TUplink_ | any other ideas? |
| 04:26.44 | bkw__ | reload reloads extensions |
| 04:26.52 | TUplink_ | got it |
| 04:26.52 | TUplink_ | [Jun 7 00:26:26] WARNING[710]: config.c:599 process_text_line: parse error: no closing ']', line 262 of /usr/local/etc/asterisk/extensions.conf |
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| 04:27.00 | TUplink_ | module load pbx_config.so |
| 04:27.08 | bkw__ | you can do reload which does it also |
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| 04:28.02 | TUplink_ | damn..... i pres the ctl key in ssh and it put a funky char in my dialplan |
| 04:29.14 | *** part/#asterisk TUplink_ (n=TUplink@c-24-126-39-22.hsd1.wv.comcast.net) |
| 04:30.52 | hansin321 | At my work we have recently moved to a 'hosted' VoIP solution that uses MGCP. We are currently still get ?x (not sure how many) T1 lines comming into the old PBX system solely for the conference bridge. Yeah, I know create an IP bridge; hopefully in the long-term. But for now my question is, do you think I could use * to act as a gateway from to MGCP to T1 (via installed card) that I could then connect to the PBX conference bridg |
| 04:32.36 | JT | easier to make it gateway the T1 with no MGCP involved |
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| 04:37.06 | hansin321 | JT: I think I get what you mean. But the idea would be to get rid of the T1's dedicated to the conference bridge. They/we are spending money on these and the idea would be to get rid of them. The company went with a hosted VoIP sollution from Qwest that utilizes MGCP. That also run on another set of T1's, along with data. I thought instead of leasing the T1's just for the PBX bridge, we could gateway the MGCP to T1 and run that |
| 04:39.15 | Aquavette | I need to route voicemail to exchange 2007 UM. I have SIPX as an go-between and it works just fine for everything except voicemail. Basically I can't figure out how to send voicemail to Exchange 2007. Any suggestions? |
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| 04:43.07 | JT | hansin321: yeah, does sounds like excessive T1s right now |
| 04:44.30 | Corydon76-home | Aquavette: sounds like you need help in configuring Exchange, not help on configuring Asterisk. |
| 04:44.41 | Aquavette | the voicemail in Exchange works fine |
| 04:44.56 | Aquavette | I just can't get Asterisk to forward its voicemail to exchange |
| 04:45.14 | kiscokid | forward as in email? |
| 04:45.16 | Corydon76-home | Do you have an email address defined for the user? |
| 04:45.35 | Aquavette | not using email, I mean physically routing the voicemail audio to the Exchange 2007 Unified Message Voicemail Service |
| 04:45.48 | Aquavette | So when no one answers that extension |
| 04:45.56 | Aquavette | it gets routed to Exchange 2007 Auto attendant |
| 04:46.06 | Corydon76-home | Uh, you need to pay Microsoft about $2.4 million to get them to open the APIs |
| 04:46.20 | Corydon76-home | We can't write to closed APIs |
| 04:46.47 | Aquavette | So basically your tellin me that there is no way to forward a call to exchanges voicemail system? |
| 04:46.53 | Corydon76-home | Correct |
| 04:48.33 | Corydon76-home | Or you could pay someone else to write the compatibility layer |
| 04:48.47 | Corydon76-home | Still, it's unlikely to be cheap |
| 04:48.48 | Aquavette | see I was thinking |
| 04:49.16 | Aquavette | is i could inject a header |
| 04:49.23 | Aquavette | onto the transfer |
| 04:49.44 | Corydon76-home | Asterisk sends emails. It is compatible with Exchange insofar as Exchange supports email. |
| 04:49.57 | Aquavette | but not with the unified messaging aspect of it |
| 04:50.35 | Corydon76-home | Nope, it's a nice wishlist, but I doubt anybody is going to write that layer for you for free, at least in the short term |
| 04:51.20 | Corydon76-home | 1.4 supports IMAP unified messaging, though. |
| 04:51.56 | Corydon76-home | but IMAP is an open standard, not a proprietary solution |
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| 04:52.12 | Aquavette | fair enough |
| 04:52.16 | Aquavette | thanks for the information |
| 04:58.21 | kiscokid | Aquavette: have you looked at this: http://technet.microsoft.com/en-us/library/2516dac1-dfdc-47eb-8e6f-18b1537a57b2.aspx ? |
| 04:58.26 | Aquavette | yea |
| 04:59.32 | Aquavette | it lists Cisco Call manager |
| 04:59.39 | Aquavette | which I have, but my client wants to move away from |
| 05:00.33 | Aces1Up | if i have a sip uri: of sip:freeman:abel9939@kingdom.dnsdojo.com how do i set this up on my asterisk box in sip.conf? |
| 05:02.46 | Corydon76-home | Aces1Up: that's username, secret, and host parameters, respectively |
| 05:05.39 | Aces1Up | i'm confuses, if i use a DID forwarder and they want a sip URI to my asterisk box what do i use? |
| 05:07.34 | kiscokid | Aquavette: seems like cisco call mgr connects to exchange UM via SIP |
| 05:07.55 | Aquavette | right, but my client wants to toss call manager |
| 05:08.01 | Aquavette | and use Asterisk |
| 05:08.07 | Aquavette | because of the cost |
| 05:08.45 | kiscokid | so, can you make * do the same thing Call Mgr does with respect to sip? |
| 05:09.14 | Aquavette | I can make * gothrough a program called sipX and then go to exchange |
| 05:09.25 | Aquavette | Outlook Voice Access and everything works |
| 05:10.01 | Aquavette | but then I run into the problem that any phone calls that don't get answered, I can't send the unanswered calls to Outlook Voice Access |
| 05:12.21 | kiscokid | you can't do this when the call goes unanswered? dial(SIP/Exchange/xxx) |
| 05:12.31 | Aquavette | yea |
| 05:12.39 | Aquavette | I can't force it |
| 05:13.32 | kiscokid | not sure what you mean by force it |
| 05:14.26 | Aquavette | I can't make it route the call from the users handsiet to outlook voice access |
| 05:16.09 | kiscokid | I'm confused, if the user doesn't answer the phone in say 20 seconds doesn't the dialplan jump to the next priority which could be another dial command? |
| 05:16.49 | Aquavette | I guess thats what I don't understand |
| 05:16.52 | Aquavette | can i make it do that? |
| 05:16.57 | kiscokid | yeah |
| 05:17.49 | kiscokid | if the primary dial command is dial(SIP/1234,20) and the call is not answered, the call will jump to the next priority |
| 05:17.49 | Aquavette | where do I do that at extensions.conf? |
| 05:17.54 | kiscokid | yeah |
| 05:18.53 | Aquavette | what would the command look like? |
| 05:18.59 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 05:19.06 | Putzz | hello |
| 05:19.15 | Putzz | anyone here using sms with asterisk? |
| 05:19.19 | kiscokid | hard to say, you might have extension macros defined |
| 05:19.36 | kiscokid | let me see if I can find a simple example |
| 05:21.13 | kiscokid | exten => 1000,1,Playback(hello-world) |
| 05:21.13 | kiscokid | exten => 1000,2,Dial(SIP/1000,10) |
| 05:21.13 | kiscokid | exten => 1000,3,Voicemail(1000@default,u) |
| 05:21.41 | kiscokid | exten => 1000,1,Playback(hello-world) |
| 05:21.41 | kiscokid | exten => 1000,2,Dial(SIP/1000,10) |
| 05:21.41 | kiscokid | exten => 1000,3,Voicemail(1000@default,u) |
| 05:21.54 | kiscokid | sorry for spamming |
| 05:22.06 | kiscokid | I sent it twice by mistake |
| 05:22.41 | kiscokid | these three lines from extensions.conf show a simple example. |
| 05:23.32 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 05:23.34 | kiscokid | When someone calls extension 1000 * plays "Hello World", dials the extension and if there's no answer in 10 secs, connects the caller to voicemail |
| 05:24.15 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:24.46 | kiscokid | you could replace priority 3 with exten => 1000,3,Dial(SIP/Exchange/1000) |
| 05:25.26 | kiscokid | Aquavette are you still here? |
| 05:25.31 | Aquavette | reading |
| 05:25.35 | Aquavette | and comprehending |
| 05:25.44 | Aquavette | and comparing with the file that I got |
| 05:27.39 | Aquavette | thanks for hte info |
| 05:27.46 | Aquavette | going to look at what I have in detail |
| 05:27.54 | Aquavette | you've given me some useful informationm |
| 05:27.59 | kiscokid | ok np |
| 05:28.07 | kiscokid | glad to help |
| 05:29.36 | *** join/#asterisk jacq (n=jal@203.187.143.130) |
| 05:29.55 | adorah | well kiscokid..do u happen to know how I apply Set(TIMEOUT(absolute)value) command at once to all the extensions and/or trunks? |
| 05:31.35 | kiscokid | adorah: offhand no, sorry |
| 05:31.54 | Aquavette | kiscokid, the people that setup * (which i did not do) actually used trixbox |
| 05:32.05 | Aquavette | so it has all kinds of freepbx rules in place for call routing |
| 05:32.16 | kiscokid | Aquavette should be the same concept |
| 05:32.40 | adorah | Aquavette: still u can use in many cases palin dial plan rules |
| 05:32.49 | adorah | =plain.. |
| 05:33.03 | Aquavette | just remove the rules that freepbx has sat |
| 05:33.08 | Aquavette | and add what I want instead? |
| 05:33.35 | adorah | in y'r case I guess u just need to add them to the priorities |
| 05:34.24 | kiscokid | haven't looked at FreePBX rules but there should be some way to direct a call to a local extension and then if it is not answered direct it to a voicemail system |
| 05:34.59 | Aquavette | see i'm manually in the configs for Asteriks right now |
| 05:35.12 | Aquavette | since that places uses Trixbox |
| 05:35.31 | Aquavette | it has those extra additions on top of it |
| 05:35.33 | Aces1Up | if i have no context configured for an incoming sip channel, should i still get something when debugging in the cli of some sort of call coming into the box? |
| 05:36.55 | kiscokid | Aces1UP never tried that but I doubt it |
| 05:37.11 | kiscokid | you might try setting verbosity to 10 |
| 05:38.34 | Aces1Up | i'm using ipkall, and when i call the number they gave me it rings, but nothing shows up on my asterisk box. |
| 05:38.42 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 05:39.38 | Aces1Up | sip set debug off |
| 05:39.45 | Aces1Up | sip set debug off |
| 05:39.51 | Aces1Up | lll |
| 05:39.56 | kiscokid | Aces1Up did you define the proper SIP context? |
| 05:40.41 | Aces1Up | i can't pick my own? i just used context=from-ipkall |
| 05:41.46 | Aces1Up | or do i have to use something specific? |
| 05:41.53 | kiscokid | that would work, probably, you also need to define some commands in the dialplan in the context pointed to by the SIP context |
| 05:42.05 | *** join/#asterisk Keizer (n=keizer@c-69-138-121-223.hsd1.fl.comcast.net) |
| 05:42.07 | Keizer | Hey guys |
| 05:42.10 | Aces1Up | yes i have it dialing my softphone. |
| 05:42.19 | Keizer | If I am just going to use a SIP trunk do I need to build zaptel? |
| 05:42.38 | Aces1Up | Keizer i didn't and i can make sip trunk calls. |
| 05:42.57 | Keizer | Aces1Up: Did you build libpri? |
| 05:43.01 | Aces1Up | nope |
| 05:43.05 | Keizer | Sweet |
| 05:43.06 | Aces1Up | just the core module. |
| 05:43.50 | kiscokid | Keizer but if you ever want to use MeetMe conferenecing you will have to build zaptel |
| 05:44.11 | Aces1Up | kiscko is that just for the timing thing? |
| 05:44.14 | Aces1Up | to do meetme? |
| 05:44.41 | kiscokid | yeah, but MeetMe won't build without zaptel actually installed and running |
| 05:45.00 | Aces1Up | i see. |
| 05:45.46 | Aces1Up | kiscko, if my ipkall is ringing, how can i tell if it is coming to my box at all? |
| 05:46.00 | Aces1Up | need to figure out where to start troubleshooting. |
| 05:46.46 | kiscokid | Aces: in sip.conf for IPKALL what do you have in the context= line |
| 05:47.56 | Aces1Up | context=from-ipkall |
| 05:48.16 | *** join/#asterisk DrCron (n=rszasz@c-67-174-231-152.hsd1.ca.comcast.net) |
| 05:48.28 | kiscokid | aces: do you have a [from-ipkall] context in extensions.conf? |
| 05:48.32 | Aces1Up | yes |
| 05:49.15 | kiscokid | I assume username and secret are correct in sip.conf |
| 05:49.29 | kiscokid | correct as in for your account |
| 05:50.21 | Aces1Up | i didn't include those as after reading others experiences with ipkall and asterisk they did not include those in their configs. |
| 05:50.28 | *** join/#asterisk [o^o] (n=k3pc@c-71-60-125-243.hsd1.pa.comcast.net) |
| 05:50.42 | [o^o] | hello, can someone point me in right dir on this? |
| 05:50.44 | [o^o] | [DB Error: extension not found] ** mysql://asteriskuser:eLaStIx.asteriskuser.2oo7@localhost/asterisk |
| 05:51.04 | *** join/#asterisk [hC] (n=hardcore@190.10.13.145) |
| 05:51.09 | kiscokid | Aces, let me look at my voicepulse config |
| 05:51.29 | Aces1Up | here is the configs i'm using basically http://www.voip-info.org/wiki/view/IPKall |
| 05:52.58 | Aces1Up | kisco becuase its ringing, does that mean it hitting my box? |
| 05:53.26 | kiscokid | Aces not sure |
| 05:55.22 | Aces1Up | is there a way to see if data is coming from that ip when calling the number like sip debug <ip address> of ipkall? |
| 05:55.43 | kiscokid | is IPKALL a free service? |
| 05:55.46 | Aces1Up | yes |
| 05:56.15 | kiscokid | well, just so you know, I never got any calls to work until I tried a paid service |
| 05:56.28 | Aces1Up | ok doke. |
| 05:56.32 | kiscokid | I tried Voipjet and FWD and they never worked |
| 05:57.09 | Aces1Up | alright, well i'm going to call it a night, thanks for your help kisco |
| 05:57.15 | kiscokid | Voipjet has 25 cents worth of free calling, supposedly but no calls ever went through |
| 05:57.27 | kiscokid | ok, aces, good luck |
| 05:57.28 | Aces1Up | kisco you have any experience with didx exchange? |
| 05:57.38 | kiscokid | didx? |
| 05:57.46 | Aces1Up | didx is a DID provider |
| 05:58.08 | kiscokid | I set up a DID with Voicepulse and its working |
| 05:58.13 | [o^o] | I _think_ my fwd works |
| 05:58.19 | [o^o] | never got or made a call |
| 05:58.27 | Putzz | I've used didx |
| 05:58.38 | Aces1Up | putzz, how do you like them? |
| 05:59.06 | Putzz | they r ok |
| 05:59.24 | Putzz | they dont provide the dids tho |
| 05:59.33 | Putzz | they just link you to it basicly |
| 05:59.39 | Putzz | providers sign up and sell their dids |
| 05:59.42 | Putzz | and u buy |
| 05:59.48 | Putzz | some providers suck |
| 05:59.54 | Putzz | so u have to look at their rating |
| 05:59.55 | kiscokid | how much per did? |
| 06:00.01 | Putzz | depends where |
| 06:00.16 | kiscokid | US, CA, Palo Alto |
| 06:00.20 | Putzz | cheap |
| 06:00.25 | Putzz | like 2.5 |
| 06:00.33 | Putzz | some r 5 |
| 06:00.35 | Putzz | but unlimited incoming |
| 06:00.44 | Putzz | or alot of minutes like 5000 - 10000 incoming |
| 06:00.47 | Aces1Up | putzz what level of rating od the DID did you see till you received good quality, i saw most of them are at like 5. |
| 06:01.09 | Putzz | 3-5 |
| 06:01.11 | Putzz | dont go lower |
| 06:01.15 | Putzz | or u might loose number |
| 06:01.19 | Putzz | or bad quality |
| 06:01.20 | Aces1Up | lol yeh, hrmm i thought 5 was low. |
| 06:01.23 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
| 06:01.26 | Aces1Up | didn't want to buy lower than 5 |
| 06:01.31 | Putzz | right not their highest is 5 |
| 06:01.36 | Aces1Up | as i will be using these numbers as a service.. |
| 06:01.40 | Putzz | us dids there r good |
| 06:01.43 | Putzz | straight from XO |
| 06:01.50 | Aces1Up | ohhh i thought the highest was 10 |
| 06:01.57 | Putzz | the rating goes to 10 |
| 06:01.59 | Aces1Up | guess i din't read something right. |
| 06:02.02 | Aces1Up | ohh ok. |
| 06:02.05 | Putzz | but right now all providers go no hgher then 5 |
| 06:02.11 | Putzz | 5 is very good |
| 06:02.26 | Aces1Up | putzz, but you have to have like 10 in your account to use them right? |
| 06:02.44 | *** join/#asterisk groogs (n=gregmac@d38-54-164.commercial1.cgocable.net) |
| 06:03.23 | Aces1Up | putzz i know you can search for it, but i couldn't find it, where do i go to search if the DID is unlimited incoming or can be used for calling card service, couldn't find it with the basic search. |
| 06:05.10 | *** join/#asterisk waptaxi (n=cahe@45.151-224-87.telenet.ru) |
| 06:06.47 | Putzz | sorry back |
| 06:07.10 | Putzz | u need to have 20 dids or they charge u $20 service fee |
| 06:07.19 | Putzz | soon it will be 50 or $50 |
| 06:08.15 | Aces1Up | thats not bad though. |
| 06:08.40 | Aces1Up | do you know where i search for calling card, or unlimited minutes on didx? couldn't find the search for that. |
| 06:08.41 | [o^o] | for usa there is always freedigits |
| 06:08.43 | *** join/#asterisk hfb (n=hfb@75.80.37.175) |
| 06:09.05 | Putzz | u have to look for the number |
| 06:09.10 | Putzz | then click into it |
| 06:09.11 | Putzz | and see |
| 06:09.16 | Aces1Up | man. |
| 06:09.17 | Aces1Up | that sucks. |
| 06:09.18 | Putzz | it will say calling card: yes |
| 06:09.20 | Putzz | yes |
| 06:09.26 | Putzz | most of them will say calling card: no |
| 06:09.27 | Aces1Up | there are hundreds of frickin numbers. |
| 06:09.30 | Putzz | tf will say yeah |
| 06:09.39 | Putzz | well search for the area code u want |
| 06:09.46 | Putzz | and they will be all same provider |
| 06:09.51 | Putzz | if one says YES |
| 06:09.55 | Putzz | they will all say YES |
| 06:09.57 | *** part/#asterisk [o^o] (n=k3pc@c-71-60-125-243.hsd1.pa.comcast.net) |
| 06:10.08 | Aces1Up | hrmm when i look it doesn't have a calling card section. |
| 06:10.50 | Putzz | u search for number |
| 06:10.53 | Putzz | click on the number |
| 06:10.58 | Putzz | and u will see the details for did |
| 06:11.26 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 06:11.35 | Putzz | their site sucks tho |
| 06:11.43 | Putzz | should have more options and be faster |
| 06:11.45 | Putzz | its so slow |
| 06:12.08 | Putzz | the good thing is the did is directly connected from carrier to you unlike most dids u buy from voip providers |
| 06:12.12 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 06:12.42 | Putzz | I got few international dids from them and they r great |
| 06:12.54 | Aces1Up | putzz yeh i want to use them to provide calling to mexico. |
| 06:13.09 | Putzz | to mexico? |
| 06:13.13 | Putzz | or from mexico? |
| 06:13.45 | Aces1Up | from mexico. |
| 06:13.51 | Aces1Up | sorry. |
| 06:14.02 | Putzz | I have that |
| 06:14.03 | Putzz | ;-) |
| 06:14.21 | Aces1Up | alot of those off 5000min per month, but i wonder what happens when i use way more than that, i don't want them to shut me down. |
| 06:14.22 | Putzz | LD in mexico is expensive |
| 06:14.26 | Putzz | a did definatly helps |
| 06:14.33 | Putzz | they charge per min |
| 06:14.35 | Putzz | but its cheap |
| 06:14.38 | Putzz | its like 1.5c per min |
| 06:14.48 | Putzz | it tells u on there |
| 06:15.33 | *** join/#asterisk redax (n=redax@mail.caracom.hu) |
| 06:15.35 | redax | hi, |
| 06:15.35 | Putzz | I've done my job. now anyone here use SMS and asterisk? |
| 06:16.24 | redax | time to time my asterisk stop working and kern.log full with "mISDN_rdata: rport queue overflow" |
| 06:16.51 | redax | I've tried the jitterbuffer settings as the chan_misdn wiki told |
| 06:16.54 | Aces1Up | putzz no, but i would like to. |
| 06:16.56 | redax | doesn't helps. |
| 06:17.53 | kiscokid | how do you mean SMS and Asterisk? |
| 06:18.03 | Putzz | using sms with asterisk |
| 06:18.06 | Putzz | what else ;-) |
| 06:18.23 | kiscokid | SMS is text messaging right? |
| 06:18.47 | Putzz | yes sir |
| 06:18.53 | Putzz | short message service |
| 06:19.41 | kiscokid | so how does SMS interface with none cell phones? |
| 06:20.09 | Putzz | well I know I can send sms with asterisk using a provider |
| 06:20.12 | Putzz | and I can receive it too |
| 06:20.20 | Putzz | but Im studying it still ;-) |
| 06:20.26 | *** join/#asterisk xtr (i=94752345@216.19.191.191.novuscom.net) |
| 06:21.10 | kiscokid | so, you want to receive text messages and display them on phones? |
| 06:21.43 | Putzz | receive / send |
| 06:34.28 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.f5.ngi.it) |
| 06:34.54 | *** join/#asterisk [o^o] (n=k3pc@c-71-60-125-243.hsd1.pa.comcast.net) |
| 06:35.04 | [o^o] | tzafrir ping |
| 06:35.12 | tzafrir | pong |
| 06:35.18 | [o^o] | got a sec? |
| 06:35.21 | tzafrir | yes |
| 06:35.32 | [o^o] | trying to get elastix running |
| 06:35.43 | [o^o] | it all works but when trying to get into freepbx I get |
| 06:35.43 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:35.44 | [o^o] | mysql://asteriskuser:eLaStIx.asteriskuser.2oo7 |
| 06:35.46 | [o^o] | ooops |
| 06:36.04 | [o^o] | [DB Error: extension not found] ** mysql://asteriskuser:eLaStIx.asteriskuser.2oo7@localhost/asterisk |
| 06:36.35 | tzafrir | well, I'm not the greatest elastix expert. I only saw that at first glance it looks more properly done than TrixBox |
| 06:36.46 | tzafrir | never actually tried to use it |
| 06:36.49 | [o^o] | it looks terriffic |
| 06:36.54 | [o^o] | but |
| 06:37.02 | [o^o] | if can't access the pbx config................. |
| 06:37.15 | [o^o] | looks like sql login error |
| 06:37.27 | [o^o] | but is issue in sql or in the amportal ? |
| 06:37.47 | tzafrir | that message seems to tell you that it fails to connect to the mysql server |
| 06:37.51 | tzafrir | Is it running? |
| 06:38.09 | [o^o] | yes as far as I can tell (can log into it via shell) |
| 06:38.33 | tzafrir | [DB Error: extension not found] ** |
| 06:38.40 | tzafrir | is that really a login error? |
| 06:39.22 | [o^o] | I dunno, don't know squat abt mysql |
| 06:39.28 | [o^o] | or sql in general |
| 06:39.29 | [o^o] | lol |
| 06:39.32 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 06:40.11 | Putzz | tzafrir: have u used SMS with asterisk? |
| 06:41.13 | tzafrir | Putzz, there are a number of variants of "SMS". The one I used is the one of the SMS app |
| 06:41.37 | tzafrir | to send an SMS through a PSTN. This tends to work in european telcos |
| 06:41.53 | Putzz | can I send and receive SMS using asterisk? |
| 06:45.38 | *** join/#asterisk fbffff (n=fbffff@c-67-167-98-42.hsd1.il.comcast.net) |
| 06:49.43 | tzafrir | There are also a bunch of other ways. E.g: if you happen to have an good adapter to a CDMA / GSM network, you may get SMS-s as well. |
| 06:49.53 | tzafrir | But there are also independent SMS services |
| 06:50.08 | tzafrir | those require no hardware at all |
| 06:50.18 | tzafrir | I have no experince with those |
| 06:50.30 | tzafrir | So the answer to your question is basically "yes" |
| 06:50.39 | Putzz | k cool |
| 06:50.44 | Putzz | not too much info out there |
| 06:50.51 | Putzz | I've read everything there is |
| 06:51.00 | Putzz | but still not enough info on sms with asterisk |
| 06:51.12 | JT | yes it basically has next to no inbuilt ability to sms |
| 06:51.58 | tzafrir | http://www.google.com/search?q=sms+site%3Avoip-info.org |
| 06:52.53 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 06:53.43 | Putzz | would I be able to lets say build a sms carrier with asterisk for office sms exchange using a did? |
| 06:54.11 | Putzz | cellphone <->sms<->asterisk<->sms<->cellphone |
| 06:54.33 | JT | sms carrier? |
| 06:54.41 | JT | are you crazy |
| 06:54.47 | Putzz | yes |
| 06:54.48 | Putzz | ;-) |
| 06:54.51 | JT | there's almost no open source carrier grade software |
| 06:54.53 | JT | for anything |
| 06:55.29 | JT | that said, there's also very little open source sms software |
| 06:56.05 | Putzz | damn |
| 06:56.08 | Putzz | hehehe |
| 06:56.28 | Putzz | so what is app sms for? msn sms? |
| 06:56.44 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:57.34 | waptaxi | open source wap & sms gateway www.kannel.org |
| 06:58.32 | Putzz | ah nice |
| 06:58.39 | mitcheloc | someone sms me! |
| 06:59.07 | JT | waptaxi: ever used it? |
| 06:59.37 | JT | Putzz: there's a difference between sending a couple of smses, and being a gateway/service provider |
| 07:00.29 | Putzz | well or send / receive |
| 07:00.35 | waptaxi | yeah, we use it as wap gateway and for sending sms via SMPP and GSM phones |
| 07:00.39 | Putzz | I dont need to be a gateway or service provider |
| 07:00.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:01.35 | JT | waptaxi: any good? |
| 07:01.41 | *** join/#asterisk yassaccan (n=yassacca@admin155.hgo.se) |
| 07:02.01 | JT | waptaxi: so you only interfaced with gsm phones, no proper gateway interface? |
| 07:03.14 | waptaxi | works nice mostly, as I said also with SMPP protocol |
| 07:05.07 | JT | smpp, how does that connect? |
| 07:06.24 | waptaxi | sorry, maybe i didn't understand your question about "proper gateway interface" |
| 07:06.55 | JT | waptaxi: refresh me, what is smpp again? |
| 07:07.31 | waptaxi | Short Message Peer-to-Peer (protocol) |
| 07:08.22 | waptaxi | proto to communicate with provider's SMS Center |
| 07:08.31 | JT | dial dialup? |
| 07:08.33 | JT | via |
| 07:08.40 | JT | or gprs |
| 07:08.44 | JT | or ss7? |
| 07:08.48 | waptaxi | internet |
| 07:09.13 | JT | does that mean the provider needs to provide a gateway service? |
| 07:10.02 | waptaxi | correct |
| 07:11.45 | *** join/#asterisk awk (n=awk@65.111.177.74) |
| 07:13.28 | *** join/#asterisk FlatFoot (n=simon@80.88.192.83) |
| 07:16.59 | *** join/#asterisk [o^o] (n=k3pc@c-71-60-125-243.hsd1.pa.comcast.net) |
| 07:17.05 | [o^o] | hey tzafrir |
| 07:17.14 | tzafrir | hi |
| 07:17.21 | [o^o] | u got that line to install freepbx on rapid? |
| 07:17.29 | [o^o] | redoing my rapid 1.1 |
| 07:17.42 | tzafrir | in 1.1 there's no freepbx |
| 07:17.46 | tzafrir | only amp |
| 07:17.53 | tzafrir | and it should be installed by default |
| 07:17.56 | [o^o] | ok |
| 07:18.03 | [o^o] | thought there was some line u gave me |
| 07:18.32 | tzafrir | that was for the 1.2 version |
| 07:18.45 | tzafrir | what version of asterisk do you have there? |
| 07:19.08 | tzafrir | so have you resolved that problem with elastix? |
| 07:20.05 | tzafrir | what did you need to do? |
| 07:21.11 | [o^o] | looking for the web management on rapid |
| 07:21.18 | [o^o] | could not resolve elastix |
| 07:21.37 | [o^o] | so did the ultimate solution,,, install over it with rapid |
| 07:21.38 | [o^o] | lol |
| 07:21.45 | tzafrir | it should come with one. |
| 07:21.59 | [o^o] | waiting for someone to fix the issue on elastix |
| 07:22.14 | tzafrir | hmm... please look in the mene System Information => Package Versions |
| 07:22.19 | tzafrir | Please pastebin that |
| 07:22.40 | *** join/#asterisk oej (n=olle@guest-rocq-135234.inria.fr) |
| 07:23.34 | [o^o] | huh? |
| 07:23.38 | [o^o] | disk wipe is done |
| 07:23.40 | drako | Hello, I'm trying to set up a RDSI line but i get this from asterisk |
| 07:23.48 | drako | asterisk1*CLI> misdn show channels |
| 07:23.48 | drako | Chan List: (nil) |
| 07:23.59 | *** join/#asterisk zamba (i=marius@flage.org) |
| 07:24.23 | [o^o] | it was centos 5 and asterisk 1.4.4 |
| 07:24.40 | zamba | i need help getting callerid from openser to asterisk.. so far i've been able to append the remote-party-id header into the sip headers, but this doesn't get applied to the callerid-setting |
| 07:27.41 | *** join/#asterisk oej_ (n=olle@dhcp-rocq-152.inria.fr) |
| 07:29.11 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 07:33.48 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 07:38.29 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-169-41.dsl.irvnca.pacbell.net) |
| 07:39.49 | *** join/#asterisk drrt (n=junior@ip242-64.baltnet.ru) |
| 07:46.46 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:50.57 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 07:51.03 | Zeeek | coffee |
| 07:51.08 | creativx | beer |
| 07:52.15 | JT | creativx: were you from .au or was i thinking of someone else? |
| 07:52.26 | *** join/#asterisk BSD_Tech[laptop] (n=BSDTech@adsl-69-230-169-41.dsl.irvnca.pacbell.net) |
| 07:53.47 | Zeeek | so far centos5 seems like it's an ok distro |
| 07:53.59 | JT | maybe :/ |
| 07:54.00 | Zeeek | I have only one thing against it |
| 07:54.00 | Aces1Up | anyon here use didx? |
| 07:54.16 | JT | .rpm |
| 07:54.32 | Zeeek | didx helps a lot with the development of my spam filters |
| 07:54.57 | Zeeek | so the difficulty is that zaptel won't compile |
| 07:56.05 | creativx | JT: im from norway :> |
| 07:56.08 | *** join/#asterisk matsk (n=mk@194.68.102.172) |
| 07:56.09 | Zeeek | can someone confirm that zaptel 1.4.2.1 works with the latest 1.4 |
| 07:56.10 | creativx | hence the beer.. at 10 am |
| 07:56.20 | JT | creativx: hmm okay |
| 07:56.22 | Zeeek | ls .. |
| 07:56.36 | Zeeek | asterisk 1.4.4 |
| 07:56.46 | *** join/#asterisk Curus (n=Curus@10.8.185.213.dk-amb.res.sta.perspektivbredband.net) |
| 07:58.30 | creativx | nene |
| 07:58.42 | creativx | i wonder how exiting it would be to set up realtime against a winbox sqlserver |
| 07:58.42 | zamba | i'm able to successfully set remote-party-id in the sip headers, but asterisk doesn't include this in its callerid settings |
| 07:59.00 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:05.05 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.144) |
| 08:05.56 | JT | winbox sql? |
| 08:07.15 | *** join/#asterisk friedrich| (n=friedric@e177244185.adsl.alicedsl.de) |
| 08:07.59 | *** join/#asterisk af_ (n=getsmart@81-174-9-192.f5.ngi.it) |
| 08:08.05 | *** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 08:08.39 | creativx | JT: windows server with sql server :) |
| 08:08.47 | JT | right, ms sql |
| 08:08.52 | creativx | indeedo |
| 08:09.04 | JT | of you can connect to it, should be fine |
| 08:09.10 | JT | one of microsoft's best products |
| 08:09.14 | creativx | yeah i guess using res_odbc it should work |
| 08:09.25 | creativx | im gonna have to investigate it when im back from vacation this week |
| 08:09.27 | JT | came from the sybase source tree originally |
| 08:09.46 | creativx | i guess its not much better than the dbadmin who sets it up |
| 08:10.00 | creativx | but it seems to be running atleast some mission critical apps around the globe |
| 08:10.09 | JT | it has real rdbms capabilities |
| 08:10.12 | JT | unlike mysql |
| 08:11.47 | creativx | well my idea was to move the sip user reg to a sql express db |
| 08:11.52 | creativx | and sync it against our crm |
| 08:12.23 | CBU[^_^]M`` | weee |
| 08:12.39 | CBU[^_^]M`` | is SPA 3102 compatible with asterisk? |
| 08:15.47 | JT | yes |
| 08:19.26 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 08:19.39 | creativx | wtf |
| 08:19.53 | creativx | after the recent update it seems microsuck live search has taken over my google settings |
| 08:19.55 | creativx | damn you to hell |
| 08:24.26 | Keizer | Hey guys |
| 08:25.41 | *** join/#asterisk jacq (n=jal@203.187.143.130) |
| 08:25.59 | Keizer | When I test out my Asterisk I can hear the the asterisk server talk but if I call one of my extensions I can't here anything. If I leave a voicemail for a extension it is recorded though |
| 08:26.16 | Keizer | I'm wondering if this has anything to do with my channeltypes options |
| 08:26.52 | *** join/#asterisk saftsack (n=oliver@p54A7D00B.dip.t-dialin.net) |
| 08:27.55 | *** join/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 08:29.17 | JT | in sip.conf, canreinvite=no |
| 08:33.48 | *** join/#asterisk euthanasie (n=kvirc@80.146.187.238) |
| 08:33.57 | euthanasie | Hi |
| 08:35.35 | *** join/#asterisk porche (n=porche@88.239.132.50) |
| 08:35.42 | porche | hi all |
| 08:36.10 | euthanasie | hi porche |
| 08:36.42 | porche | i have got a problem with busy detection, after some search, i find out that I need to configure enum busy_detect |
| 08:36.53 | porche | hi euthanasie |
| 08:37.03 | JT | Keizer: you there? |
| 08:37.30 | porche | is there someone here, can point me the right way, I do know the busy pattern, |
| 08:37.40 | porche | hi JT, this this porche, your headache |
| 08:38.01 | JT | okay.... |
| 08:38.01 | euthanasie | i have a problem with AgentCallBackLogin, always getting the error Extension `101@agenten` is not valid fpr automatic login of agent `101` but the extension exists and it's a simple call function .... |
| 08:38.02 | Keizer | JT: Yes |
| 08:38.10 | JT | Keizer: tried my suggestion? |
| 08:38.28 | Keizer | JT: I'll try it real quick |
| 08:41.32 | Keizer | JT: Not so far |
| 08:41.57 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:42.21 | JT | Keizer: how are you calling the asterisk server? |
| 08:43.00 | Keizer | Calling extension 101 |
| 08:43.08 | Keizer | Or 103 |
| 08:43.13 | Keizer | Using a software phone |
| 08:43.16 | JT | sorry that's useless to me |
| 08:43.18 | JT | ah s |
| 08:43.18 | Keizer | One is on Linux it's called Ekiga |
| 08:43.21 | JT | so sip sotphone |
| 08:43.24 | JT | softphone |
| 08:43.28 | Keizer | Yes |
| 08:43.48 | JT | how many extensions do you have? |
| 08:43.55 | Keizer | 3 |
| 08:44.04 | JT | all sip on lan? |
| 08:46.16 | Keizer | Yep |
| 08:46.20 | Keizer | On a CentOS box |
| 08:46.27 | Keizer | Compiled Asterisk 1.4.4 |
| 08:46.34 | JT | pastebin sip.conf after masking the passwords |
| 08:46.45 | Keizer | I didn't compile Zaptel or libpri |
| 08:48.01 | *** join/#asterisk saftsack (n=oliver@p54A7F7FA.dip.t-dialin.net) |
| 08:49.45 | euthanasie | can anyone help me plz? I don't no whould is should do, tried everything, but with the same result Extension `101@agenten` is not valid for automatic login of agent `101` |
| 08:50.25 | s0ck | any of you chaps use vmware |
| 08:51.13 | JT | ~pb |
| 08:51.24 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 08:51.25 | Keizer | JT: http://rafb.net/p/ZKfmdN16.html |
| 08:52.00 | JT | ..wtf |
| 08:52.05 | JT | Keizer: where's the rest of the file? |
| 08:52.12 | JT | that is not enough |
| 08:52.16 | JT | i need the whole file |
| 08:52.26 | Keizer | The rest looked commented out |
| 08:52.32 | Keizer | With ; in front of everything |
| 08:52.58 | JT | did it include any other files? |
| 08:54.59 | porche | euthanasie, what's the context of |
| 08:55.01 | Keizer | JT: Dunno |
| 08:55.02 | porche | agent 101 |
| 08:55.23 | euthanasie | agenten @ porche |
| 08:56.08 | JT | Keizer: you're in serious trouble if that's your whole sip.conf file |
| 08:56.16 | JT | phones shouldn't work at all |
| 08:57.22 | porche | is there a definition on agents.conf |
| 08:57.27 | porche | is 101 an extension |
| 08:57.29 | porche | or agent? |
| 08:57.47 | euthanasie | both |
| 08:58.00 | Keizer | They're working though |
| 08:58.04 | Keizer | I left many voice mails |
| 08:58.11 | porche | ok |
| 08:58.22 | porche | is the a group defitinition for agents? |
| 08:59.25 | Keizer | JT: [Jun 7 04:58:45] WARNING[7302] chan_zap.c: Ignoring switchtype |
| 08:59.47 | euthanasie | no @ porche |
| 09:00.02 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 09:00.12 | porche | what's the exact command euthanasie |
| 09:00.31 | euthanasie | wait a moment |
| 09:00.33 | JT | Keizer: dude i suggest you read the book |
| 09:00.35 | JT | ~thebookl |
| 09:00.37 | JT | ~thebook |
| 09:00.39 | jbot | well, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:00.46 | JT | zap has nothing to do with sip |
| 09:01.00 | porche | JT, may I ask a question? |
| 09:02.04 | *** join/#asterisk Kigh (n=kai@ciphron.de) |
| 09:02.13 | JT | i'm going to be away from irc for a bit |
| 09:02.23 | euthanasie | http://www.pastebin.ca/547156 here u have ^^ |
| 09:02.23 | porche | ok |
| 09:08.28 | Keizer | Dang |
| 09:10.18 | *** join/#asterisk saftsack (n=oliver@p54A7D007.dip.t-dialin.net) |
| 09:14.51 | zdrulio | i want to load module cpp.radius, but "Error loading module 'app_radius.so': librad.so: cannot open shared object file: No such file or directory". I have installed librad. and file librad.so exist. any ideas ? |
| 09:15.10 | porche | zdrulio |
| 09:15.11 | porche | run |
| 09:15.25 | porche | find / -name librad.so |
| 09:15.37 | zdrulio | yes |
| 09:15.40 | porche | it's most probably under |
| 09:15.42 | zdrulio | it exist |
| 09:15.49 | porche | ? |
| 09:15.54 | porche | is it under |
| 09:16.01 | porche | <PROTECTED> |
| 09:16.01 | porche | ? |
| 09:16.12 | zdrulio | /usr/local/lib/librad.so |
| 09:16.16 | porche | yes |
| 09:16.17 | zdrulio | yes |
| 09:16.18 | porche | run |
| 09:16.29 | porche | ln -s /usr/local/lib/librad.so /usr/lib |
| 09:16.30 | zdrulio | in asterisk CLI i write |
| 09:16.53 | porche | no run it at command line, not cli |
| 09:17.03 | zdrulio | what ? |
| 09:17.20 | porche | quit cli |
| 09:17.25 | porche | then run |
| 09:17.36 | porche | ln -s /usr/local/lib/librad.so /usr/lib |
| 09:17.40 | zdrulio | rdy |
| 09:17.43 | porche | then re-run asterisk |
| 09:17.48 | zdrulio | aha |
| 09:18.12 | porche | euthanasie, i think all ok |
| 09:18.15 | Zeeek | anyone have a TDM400P? |
| 09:18.19 | porche | may be you need to change the |
| 09:18.26 | *** join/#asterisk TimothyP (n=timothy@116.252-243-81.adsl-static.isp.belgacom.be) |
| 09:18.30 | porche | agent id + extension |
| 09:18.36 | porche | it may be needed to be different |
| 09:18.48 | porche | zeeek have got a tdm2400 |
| 09:19.03 | Zeeek | what modules do you have in it? |
| 09:19.08 | TimothyP | Hi, how can I change the files that are played when the systems says "the person at extension bla bla bla is not available right now" , because we have our own messages and a series of options |
| 09:19.09 | euthanasie | no porche, they may be identic |
| 09:19.24 | Keizer | [Jun 7 05:15:47] WARNING[17679]: app_dial.c:1099 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 09:19.27 | porche | the command seems to be right, i have no clue |
| 09:19.31 | creativx | TimothyP: /var/lib/asterisk/sounds |
| 09:19.37 | porche | standard, what's the problem zeeek? |
| 09:19.46 | TimothyP | creativx, yes I know I have new files in there |
| 09:19.51 | TimothyP | but I don't want to overwrite the old ones |
| 09:19.57 | Zeeek | I have 2FXO and one FXS |
| 09:20.01 | zdrulio | porche: thanks but i want to load app_radius.so and cdr_radius.so. |
| 09:20.02 | TimothyP | I want to change which ones are played, because I need a series of them to be played |
| 09:20.16 | porche | best to |
| 09:20.17 | porche | stop it |
| 09:20.22 | porche | and restart |
| 09:20.23 | Zeeek | none of the channels want to be fxoks |
| 09:20.27 | porche | or you can reload |
| 09:20.42 | zdrulio | when i write in CLi module load ... Module 'app_radius.so' did not register itself during load |
| 09:20.57 | TimothyP | our system needs to say "the person you are trying to dail is not available right now, press 1 to leave a voice message, press 2 to reach the person on his mobile" , I have those text in our native language , but need to change voicemail settings then |
| 09:21.04 | porche | any error message, zdrulio |
| 09:21.15 | zdrulio | this ? |
| 09:21.21 | zdrulio | ot this is not a error ? |
| 09:21.35 | porche | i mean, any clue, it may be again another library or current library |
| 09:21.48 | porche | zeek have you tried |
| 09:21.51 | porche | one sec |
| 09:22.04 | porche | genzaptelconf |
| 09:22.21 | porche | it automatically checks the cards, and produces zaptel.conf |
| 09:22.27 | zdrulio | porche: http://pastebin.ca/547197 |
| 09:22.54 | porche | zdrulio |
| 09:23.04 | porche | is this a different message ? |
| 09:23.12 | porche | i mean b4 linking (ln -s |
| 09:23.18 | zdrulio | yes |
| 09:23.24 | zdrulio | befor linking |
| 09:23.49 | porche | can you stop cli and re-run asterisk -vvvvvc |
| 09:24.27 | Zeeek | is it normal that with only 3 modules lugged in, all channels can be configured? |
| 09:24.30 | extr3m | i call your -v's and raise you -vvvvvvvv |
| 09:24.50 | TimothyP | I could use Playback to play back the messages, but then the user ends up in voicemail and here's the voicemail stuff all over again |
| 09:24.55 | zdrulio | porche: http://pastebin.ca/547207 |
| 09:26.06 | porche | zdrulio, are you sure, you have configured the radius for asterisk and also is radius server up and running? |
| 09:26.18 | porche | zeeek yes |
| 09:26.23 | porche | you can see the config |
| 09:26.24 | porche | with |
| 09:26.34 | porche | ztcfg |
| 09:26.41 | porche | ztcfg -vv |
| 09:27.04 | Zeeek | I've been doing that. There are three modules and only two leds lit |
| 09:27.20 | porche | what does it say? |
| 09:27.25 | zdrulio | porche: yes. in the moment i have asterisk with runing and wirking radius, but i want to make a copy of this server. |
| 09:27.29 | porche | only two modules? |
| 09:27.43 | Zeeek | 2 FXO one FXS |
| 09:27.49 | Zeeek | only one LED lit |
| 09:27.58 | TimothyP | aha would it help if I changed the zonemessages in voicemail.conf |
| 09:28.04 | Zeeek | that aleready does't look right |
| 09:28.34 | porche | well if zaptel ztcfg sees 3 modules |
| 09:28.37 | Zeeek | ah, maybe they still need a power to card? |
| 09:28.41 | *** join/#asterisk snuffy22 (n=na@61.29.30.137) |
| 09:29.10 | Zeeek | what an idiot! |
| 09:29.12 | porche | dont knwo that part zeek, in tdm2400 its just a pci card, all power from pci |
| 09:29.31 | Zeeek | no IIRC you still need power for the ring |
| 09:29.43 | Zeeek | so that must be it. I totally forgot about that |
| 09:30.40 | porche | ok good |
| 09:31.12 | *** join/#asterisk andyd (n=andyd@77.75.24.10) |
| 09:31.19 | zdrulio | porche: the working asterisk is 1.2 and i want to up 1.4 |
| 09:31.23 | zdrulio | may be this is a problem |
| 09:32.48 | porche | may be zdrulio |
| 09:36.34 | Zeeek | there must be a lwa about making the power cable inside the PC case being 1 inch too short |
| 09:36.48 | Zeeek | s/lwa/law/ |
| 09:40.01 | Zeeek | ~oddtdm Did you connect power molex? |
| 09:42.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:49.21 | Zeeek | .. |
| 09:52.36 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 10:03.28 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 10:04.42 | *** join/#asterisk Avochelm (n=damien@CPE-121-209-54-60.sa.bigpond.net.au) |
| 10:06.46 | Uatec | Zeeek, you're right |
| 10:06.53 | Uatec | athough not always as much as 1 inch |
| 10:07.06 | Zeeek | I had to move a drive to get it to work |
| 10:07.11 | Uatec | sometimes i have to bend a card to strech a cable round :\ |
| 10:07.32 | Uatec | lol, a lwa |
| 10:07.51 | Zeeek | one nice thing about Dell is the way the cable them |
| 10:08.19 | Uatec | i have a compaq here |
| 10:08.24 | *** join/#asterisk oddd (n=lund@203-206-92-144.dyn.iinet.net.au) |
| 10:08.25 | Uatec | it's quite nice inside |
| 10:08.33 | Uatec | although a server i'm building, that's based on a compaq |
| 10:08.39 | Uatec | not as good as my workstation |
| 10:08.40 | Uatec | so cluttered |
| 10:08.49 | oddd | hello |
| 10:08.56 | *** part/#asterisk porche (n=porche@88.239.132.50) |
| 10:08.58 | Zeeek | my homemade PC have improved now that IDE cables are better and the SATA which are even better |
| 10:09.00 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 10:09.02 | zeeesh | hi |
| 10:09.07 | Zeeek | who |
| 10:09.08 | oddd | I was wondering if anyone around was willing to help out with a small problem I'm having |
| 10:09.17 | oddd | I cannot get any sound out of playback |
| 10:09.27 | oddd | it doesn't seem to have any error associated with it. |
| 10:09.29 | Zeeek | make install |
| 10:09.36 | oddd | perhaps it is a codec problem |
| 10:09.39 | oddd | ? |
| 10:10.07 | oddd | err... |
| 10:10.14 | oddd | yes.. that is how I installed it |
| 10:10.23 | Zeeek | no that wans'"t for you |
| 10:10.39 | oddd | oh. sorry |
| 10:11.28 | Zeeek | I don't have ZAP listed in channeltypes |
| 10:13.12 | zeeesh | hi |
| 10:13.34 | CBU[^_^]M`` | when i install asterisk... does it automatically have h232 and SIP codec? |
| 10:13.56 | Zeeek | those are protocols not codecs |
| 10:14.14 | *** join/#asterisk snuffy22 (n=na@61.29.30.137) |
| 10:14.14 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 10:14.15 | *** join/#asterisk ptiggerdine (n=ptiggerd@203-219-14-182.static.tpgi.com.au) [NETSPLIT VICTIM] |
| 10:14.15 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) [NETSPLIT VICTIM] |
| 10:14.16 | CBU[^_^]M`` | hehhe |
| 10:14.20 | oddd | modulo that, SIP is, but I think h232 need a library |
| 10:14.56 | CBU[^_^]M`` | hmmm ... whats the port of h323 and sip? |
| 10:15.06 | oddd | listening port? |
| 10:15.10 | oddd | 5060 for SIP |
| 10:15.13 | oddd | not sure about h232 |
| 10:15.55 | CBU[^_^]M`` | oks |
| 10:16.27 | CBU[^_^]M`` | im still waiting for my sipura 3102 ... im planning to set up my asterisk when it arrives |
| 10:16.42 | oddd | just hope that Playback works |
| 10:16.51 | oddd | it really is a show stopper when it doesn't |
| 10:17.01 | *** join/#asterisk mattfletcher (n=matt@62-249-226-101.no-dns-yet.enta.net) |
| 10:17.03 | CBU[^_^]M`` | does linksys have any GMS gateway products? |
| 10:17.32 | mattfletcher | i'm having problems getting my head round a Pickup() configuration, can anyone help? |
| 10:17.54 | CBU[^_^]M`` | <oddd> just hope that Playback works <= what do you mean? |
| 10:18.44 | oddd | I'm trying to get it to work right now |
| 10:18.51 | oddd | I get nothing out of it. |
| 10:18.53 | oddd | from the console |
| 10:19.04 | oddd | but other things like Milliwatt and Echo work |
| 10:19.23 | oddd | I'm assuming that it cannot seem to convert GSM to whatever it wants |
| 10:19.36 | oddd | but I get no error messages with 'asterisk -rvvvvvvvvvvv' |
| 10:20.23 | CBU[^_^]M`` | heheh ... i saw some GSM voip gateways ... but it usually costs 300 USD |
| 10:20.30 | CBU[^_^]M`` | tooo expensive for homeuse |
| 10:21.06 | oddd | I think I did something wrong in my compile |
| 10:21.18 | oddd | but I would have thought I'd get warned |
| 10:22.50 | mattfletcher | I have posted my Pickup() problem to pastebin, can anyone with experience of using Pickup() help? http://pastebin.ca/547276 |
| 10:31.20 | oddd | hurm... Playtones doesn't seem to work either |
| 10:33.10 | *** join/#asterisk gardo (n=gardo@124.107.38.214) |
| 10:34.08 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:34.27 | puzzled | hi |
| 10:39.37 | oddd | hurm.. might be a config problem |
| 10:46.14 | zdrulio | hm. i`m searching for radius modul for asterisk 1.4 |
| 10:46.17 | zdrulio | any ideas ? |
| 10:46.22 | *** join/#asterisk RedBack (n=lukeblac@82.152.56.113) |
| 10:47.06 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 10:48.15 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:51.56 | *** join/#asterisk kkeil (n=kkeil@p549790f0.dip0.t-ipconnect.de) |
| 10:53.47 | *** join/#asterisk Corydon76-home (i=yellow@pdpc/supporter/sustaining/Corydon76-home) |
| 10:53.47 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 10:56.33 | *** join/#asterisk andyd (n=andyd@77.75.24.10) |
| 10:57.35 | *** join/#asterisk nullvariable (n=nullvari@ip70-173-244-97.lv.lv.cox.net) |
| 11:08.49 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-118-241.ph.ph.cox.net) |
| 11:12.21 | *** join/#asterisk Fieldy (i=9iHJR5KD@gentoo/contributor/Fieldy) |
| 11:17.17 | *** join/#asterisk waptaxi (n=cahe@45.151-224-87.telenet.ru) |
| 11:17.21 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 11:18.41 | jeremy_g | How many concurrent registered users and concurrent calls I can have on a 1.8 Ghz AMD Opteron 2210 HE with 1GB RAM and 1MB L2 Cache |
| 11:21.17 | nullvariable | jeremy_g: what kind of bandwidth will you have? |
| 11:22.10 | redax | hi, |
| 11:23.04 | redax | is there a way to know the name of the trunk channel. the ${CHANNEL} contains the SIP/ext-xxxx, and I'd like to know which mISDN channel was used for the call |
| 11:26.02 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 11:28.19 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 11:32.09 | zdrulio | i have problem again |
| 11:32.37 | *** join/#asterisk saftsack (n=oliver@p54a7dbd1.dip.t-dialin.net) |
| 11:34.07 | zdrulio | if i call from 1 asterisk to another aster i have "exten => _14.,1,Dial(IAX2/USER:password@HOST:PORT/${EXTEN:2},30,r)" but another asterisk`s users viw only USER ID. i want to another asterisk users view original numer who was dial him. can i do this ? |
| 11:43.36 | redax | seems like there's no o->chan->name set as variable |
| 11:44.58 | *** join/#asterisk RedBack (n=lukeblac@82.152.56.113) |
| 11:45.41 | redax | ah. ${CDR(dstchannel)} |
| 11:45.43 | redax | thanks ;-) |
| 11:46.36 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:47.11 | oddd | well |
| 11:47.17 | oddd | it isn't a decoder issue |
| 11:47.28 | oddd | the same problem with .wav files |
| 11:47.31 | oddd | as with .gsm files |
| 11:47.39 | oddd | so it is in the output somehow |
| 11:47.50 | oddd | but it doesn't matter if I use the console or an SIP client |
| 11:49.00 | oddd | is there any easy way to debug app_playback.c? |
| 11:51.32 | *** part/#asterisk mattfletcher (n=matt@62-249-226-101.no-dns-yet.enta.net) |
| 11:54.54 | jeremy_g | tzanger:what is your experience in terms of performance limits on * |
| 11:54.57 | jeremy_g | tzanger:what is your experience in terms of performance limits on * |
| 11:55.03 | jeremy_g | sorry |
| 11:55.15 | tzanger | performance limits? based entirely on what you're trying to do |
| 11:55.21 | tzanger | I have no high-volume systems |
| 11:55.43 | tzanger | my biggest one spends most of its time bridging two PRIs, which is NOT very resource-intensive |
| 11:56.09 | tzanger | if you're doing any audio processing (echo can, transcoding) you're going to be CPU-bound |
| 11:56.26 | tzanger | recording isn't very intensive unless it's lots of concurrent channels, at which point you start becoming I/O bound |
| 11:57.09 | *** join/#asterisk saftsack (n=oliver@p54A7FF98.dip.t-dialin.net) |
| 11:57.30 | jeremy_g | tzanger:no transcoding |
| 12:03.11 | tzanger | jeremy_g: sip, tdm, what |
| 12:03.33 | tzanger | lots of small calls or lots of longer calls? |
| 12:05.57 | *** join/#asterisk denke (n=denke@mehess.adsl.datanet.hu) |
| 12:07.04 | denke | Hello Everyone! |
| 12:07.34 | denke | Can anybody help me with asterisk - xlite? |
| 12:07.41 | denke | in asterisk I get:Internal RTCP NTP clock skew detected: lsr=1925527781, now=1925721092, dlsr=193331 (2:949ms), diff=20 |
| 12:08.04 | denke | and in xlite: [07-06-07]14:00:14.299 | Warning | Audio | "Not enough data in latency reducer 0" | sua::CAudioManager::CAudioOutgoingStream::MixOneFrame |
| 12:08.35 | denke | and the audio keeps the speed, but there are losses in both ways... |
| 12:09.25 | *** join/#asterisk lwh (n=lwh192@rdsl-0469.tor.pathcom.com) |
| 12:11.10 | *** join/#asterisk Strom_M (n=strom@208.47.199.4) |
| 12:11.42 | denke | noone? ... please |
| 12:11.44 | jeremy_g | tzanger:only sip |
| 12:12.19 | jeremy_g | tzanger:23% long calls, 70% small calls |
| 12:12.43 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 12:14.09 | tzanger | lots of small calls involve more setup/teardown |
| 12:14.19 | tzanger | will the asterisk server be able to step out of the audio path? |
| 12:18.31 | *** join/#asterisk saftsack (n=oliver@p54A7F562.dip.t-dialin.net) |
| 12:18.56 | *** join/#asterisk mirco (n=mirco@p54B264E7.dip.t-dialin.net) |
| 12:19.24 | jeremy_g | tzanger:never |
| 12:19.28 | jeremy_g | unfortunately |
| 12:22.33 | *** join/#asterisk boch (n=fran@190.48.201.232) |
| 12:22.37 | tzanger | jeremy_g: how many concurrnet calls (approx) |
| 12:29.37 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:30.45 | zdrulio | <PROTECTED> |
| 12:31.47 | tzanger | zdrulio: take a REALLY GOOD LOOK at your dial command |
| 12:31.57 | tzanger | what *extension* are you dialing? |
| 12:32.00 | tzanger | print it here |
| 12:33.19 | *** join/#asterisk Dantix (n=Dante@200.68.70.100) |
| 12:35.32 | Dantix | hi all!! are there a way to force a collective ring from a program or from an external event like a cron job? |
| 12:35.55 | tzanger | Dantix: use the local channel |
| 12:36.57 | Dantix | tzanger: please give me some link to start learning about |
| 12:37.07 | tzanger | is voip-info still down? |
| 12:37.49 | Dantix | tzanger: I don't know, I'll try with that. thanks |
| 12:38.27 | zdrulio | tzanger: i`m with user 1000 registred in A1(asterisk1) and i call to A2(Asterisk2). User in A2 view call from 5000 |
| 12:38.34 | zdrulio | i want to view 1000 |
| 12:38.48 | [TK]D-Fender | Dantix: lookup ".call files" and "AMI Originate" on the WIKI when it comes back up. |
| 12:38.56 | *** join/#asterisk mightnare (n=mike@p1015-ipad02motosinmat.mie.ocn.ne.jp) |
| 12:39.17 | tzanger | zdrulio: what's your caller id look like on A2 and A2 |
| 12:39.20 | tzanger | er A1 and A2 |
| 12:39.39 | [TK]D-Fender | zdrulio: First set up a proper peer/user between each site and stop putting your auth in your dial. |
| 12:39.40 | zdrulio | er ? |
| 12:39.49 | tzanger | morning [TK]D-Fender |
| 12:39.55 | [TK]D-Fender | *yawn* |
| 12:39.59 | tzanger | yeah same here |
| 12:40.01 | [TK]D-Fender | tzanger: Mornin' |
| 12:40.02 | tzanger | missed my timmie's on the way in |
| 12:40.11 | tzanger | drinking the free house coffee but it ain't hte same |
| 12:40.21 | [TK]D-Fender | tzanger: After my first 3 cups I'm the best so-and-so around ;) |
| 12:40.23 | zdrulio | [TK]D-Fender: why ? explain me plz |
| 12:40.36 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 12:40.36 | tzanger | *sigh* we need an #asterisk-newbies |
| 12:40.52 | tzanger | of course that's impossible, though... |
| 12:40.53 | [TK]D-Fender | zdrulio: Dialplan is much more exposed. JUST DO IT. |
| 12:40.55 | Zeeek | tzanger this *is* asterisk-newbies |
| 12:40.58 | tzanger | I wonder what about an asterisk best practices document? |
| 12:41.08 | [TK]D-Fender | tzanger: But think of the breathing room we'd have! |
| 12:41.25 | tzanger | [TK]D-Fender: ha |
| 12:41.46 | Zeeek | does anyone thing AEL should totally replace extensions.conf? |
| 12:41.49 | [TK]D-Fender | tzanger: Think of it like the Kyoto Accord. Sometimes you jsut need to know what to do so you do the exact opposite ANWAYS ;) |
| 12:42.04 | Zeeek | [TK]D-Fender on in the US |
| 12:42.06 | tzanger | *coughs* |
| 12:42.12 | tzanger | dammit I nearly spit my coffee on the screen |
| 12:42.13 | tzanger | hahaha |
| 12:42.17 | [TK]D-Fender | Zeeek: All AEL does is PARSE BACK to extensions.conf. |
| 12:42.22 | coppice | I've been to Kyoto. It was full of people driving Accords |
| 12:42.36 | rob0 | I want an Accord. |
| 12:42.43 | Zeeek | so I'm setting up a brand new asterisk. SHould I mess with AEL or not? |
| 12:43.35 | [TK]D-Fender | Zeeek: How many docs use AEL1/2 (OMG it changed!??!). How many people here use it and can reall help out. If your dialplan so complex as to deserve it? |
| 12:43.54 | [TK]D-Fender | Zeeek: This for me adds up to "collossal waste of time" |
| 12:44.04 | Zeeek | Well, at that level one doesn't usually need help |
| 12:44.22 | Zeeek | I'm more helpneedy in the linux and make paryt |
| 12:44.44 | Zeeek | for example I got it all installed on centos 5 but I have no idea how |
| 12:44.47 | *** part/#asterisk Dantix (n=Dante@200.68.70.100) |
| 12:45.08 | Zeeek | A lot of weird shit went on durning make and I just removed stuff I know I don't want until it worked |
| 12:45.13 | *** join/#asterisk floppp (n=flop@nat-staff.b3g-telecom.com) |
| 12:45.48 | [TK]D-Fender | Zeeek: I chalk it up to AEL just being back compiled and adding another point of failure to my system. |
| 12:45.55 | Zeeek | ok, noted |
| 12:46.31 | Zeeek | so I just need to be sure not to fall in the same traps as my current 22,000 line extensions.conf file |
| 12:47.14 | *** join/#asterisk saftsack (n=oliver@p54A7FA12.dip.t-dialin.net) |
| 12:47.14 | Corydon76-home | Ye gods... 22,000 lines? |
| 12:47.17 | *** join/#asterisk The_Lightside (n=dialt@mtngprs2.mtn.co.za) |
| 12:47.20 | [TK]D-Fender | Zeeek: When AEL changes the complete way apps, dialplan patterns, evaluations and priorities really work, then maybe. But by then it would be a CORE tech. think of the apps that would have to change. |
| 12:47.21 | rob0 | And they're all busy. |
| 12:47.52 | The_Lightside | hi guys, sorry to bring this up again... |
| 12:47.53 | Zeeek | It's an odd feeling to be using a linux box that has a P4 2.4ghz CPU after doing it for years on a P-III 800 |
| 12:48.01 | [TK]D-Fender | Zeeek: Yes, thats PSYCHO. I had about 400 lines of astdb super-dialplan in my worst setup.... |
| 12:48.14 | [TK]D-Fender | Zeeek: I wanna see your dialplan :) |
| 12:48.15 | Zeeek | my current setup is always my worst setup |
| 12:48.20 | The_Lightside | user A phones user B, is there any way to get the caller ID of user B to display on USer A's screen? |
| 12:48.33 | [TK]D-Fender | Zeeek: Then stop now before your system explodes ;) |
| 12:48.36 | Corydon76-home | Sounds like you need more pattern matching and/or integration with a database |
| 12:48.48 | Zeeek | I exaggerated, wc says 2830 lines |
| 12:49.11 | Corydon76-home | That's better, but still rather large |
| 12:49.15 | [TK]D-Fender | The_Lightside: Sure. Before you do the dial, place a system call to some sort of app (that you may wel have to INVENT) that will do the job. Then dial. |
| 12:49.18 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:49.24 | Zeeek | every marco I ever tried is in there |
| 12:49.27 | [TK]D-Fender | 2830 = bad. |
| 12:49.37 | Zeeek | but I'll look for lines that don't start with ';' |
| 12:49.39 | [TK]D-Fender | Zeeek: Trim the fat... |
| 12:49.45 | Corydon76-home | Oh, so it's mostly unused... |
| 12:50.01 | rob0 | 5 lines without the comments. |
| 12:50.12 | The_Lightside | [TK]D-Fender, sure... was hoping someone had some ideas :) |
| 12:50.20 | Corydon76-home | That's better, but having lots of unused contexts slows down the search everytime you advance a priority |
| 12:50.40 | Zeeek | Corydon76-home the known to be unused are commented out |
| 12:50.48 | Corydon76-home | Ah |
| 12:51.17 | Corydon76-home | grep -cE '^[^;]' /etc/asterisk/extensions.conf |
| 12:51.39 | Zeeek | so about 2424 lines |
| 12:51.51 | redax | if I have a SIP/124-xxx-yyy how can I cut down the `-yyy' ? |
| 12:52.07 | Corydon76-home | redax: use CUT() |
| 12:52.21 | Zeeek | 1637 lines begin with exten |
| 12:52.26 | Zeeek | ca va |
| 12:52.42 | Zeeek | we've come a long way from 22,000 |
| 12:53.01 | Corydon76-home | redax: ${CUT(CHANNEL,-,1-2)} |
| 12:53.23 | Zeeek | sorry, 1463 lines begin with exten |
| 12:53.40 | Corydon76-home | Okay, that's more reasonable |
| 12:54.03 | Zeeek | I think the new box won't need a lot of the soho stuff |
| 12:54.08 | redax | Corydon76-home: the problem is sometimes I don't have -xxx |
| 12:54.10 | Zeeek | it's just for texting 1.4 |
| 12:54.25 | Corydon76-home | redax: then it gets a little more complex. |
| 12:54.40 | [TK]D-Fender | Zeeek: Gimme! I'm sure I could cop off plenty for you REALLY fast :) |
| 12:54.44 | [TK]D-Fender | chop* |
| 12:55.00 | redax | is there a way to specify the fields like 1-(n-1) ? :-) |
| 12:55.13 | Corydon76-home | redax: ${CUT(CHANNEL,-,1-$[${FIELDQTY(CHANNEL,-)} - 1])} |
| 12:55.19 | redax | whoa. |
| 12:55.40 | Corydon76-home | It gets simpler in 1.4, though |
| 12:56.15 | redax | actually it's ast 1.2 |
| 12:56.16 | Zeeek | [TK]D-Fender 113 NoOps |
| 12:56.45 | Corydon76-home | redax: that's what you need to do, then |
| 12:56.48 | Zeeek | so 145O lines are actually doing something |
| 12:56.54 | [TK]D-Fender | Zeeek: C'mon.... pastebin! |
| 12:56.56 | Zeeek | so 135O lines are actually doing something |
| 12:56.59 | Corydon76-home | redax: that's 1 - n-1 |
| 12:57.31 | redax | great, thank you very much corydon |
| 12:57.55 | Corydon76-home | Hmmm, I guess 1.4 doesn't get simpler |
| 12:58.08 | Corydon76-home | Oh well |
| 12:58.57 | Zeeek | I could get rid of about 20 lines removing [incoming-iaxtel] |
| 13:01.16 | *** join/#asterisk bapril (n=bapril@pool-70-109-158-237.cncdnh.east.verizon.net) |
| 13:01.37 | Corydon76-home | I guess there's no better syntax to be had |
| 13:01.55 | Zeeek | Corydon76-home than what? |
| 13:02.11 | Zeeek | oh, CUT? |
| 13:02.22 | Corydon76-home | Correct |
| 13:04.45 | Zeeek | I can't figure this out. zaptel: make clean ( no erros) make "autoconfig file not found" then if I go to menuselect and make, then cd.. ; make ; it works |
| 13:05.12 | blitzrage | Corydon76-home: that's not too bad |
| 13:05.20 | Corydon76-home | Did you type: ./configure ? |
| 13:05.24 | Zeeek | ya |
| 13:05.32 | Corydon76-home | 'make menuselect' |
| 13:05.44 | blitzrage | hopefully by July I can start working on the cookbook |
| 13:05.47 | Zeeek | same autoconfigure; file not found" err |
| 13:06.00 | Zeeek | in the selectmenu dir |
| 13:06.00 | blitzrage | and I just realized I got a copyedit of TFoT2 in the mail, and I need to go pick it up |
| 13:06.06 | blitzrage | ugh... so much work to do |
| 13:06.09 | Corydon76-home | blitzrage: woot |
| 13:06.19 | *** join/#asterisk Strom_M (n=strom@192.41.247.50) |
| 13:07.10 | Corydon76-home | blitzrage: so we're about a week from the print run? ;-) |
| 13:07.21 | blitzrage | Corydon76-home: probably 2-3..... |
| 13:07.45 | Corydon76-home | Heheheh |
| 13:08.40 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:09.05 | Zeeek | blitzrage concentrating on 1.2 or 1.4 for Book ? |
| 13:09.14 | blitzrage | 1.4 |
| 13:09.17 | blitzrage | of course |
| 13:09.20 | *** join/#asterisk mattfletcher (n=matt@62-249-226-101.no-dns-yet.enta.net) |
| 13:09.30 | Katty | so is there a way to answer an incoming call, ring a blast group, after timeout answer the call and play them an audio file, then Page(sip/100&sip/101) play an audio file, hangup page, then ring the blast group again? |
| 13:09.35 | Zeeek | the vast majority of prod sys are 1.2 .... of course |
| 13:09.52 | mattfletcher | how can i make my asterisk server answer a zap call more quickly. it takes two rings before asterisk picks it up every time |
| 13:10.10 | Zeeek | mattfletcher make sure fax detect is off |
| 13:10.17 | Zeeek | and disable cid |
| 13:10.37 | mattfletcher | zeeek: where are they hiding? |
| 13:10.49 | JT | mattfletcher: get digital lines |
| 13:10.52 | Zeeek | zapata.conf |
| 13:11.13 | Zeeek | grep fax zapata.conf |
| 13:11.16 | blitzrage | Zeeek: for now... by end of summer 1.2 will go into maintenance mode |
| 13:11.22 | Corydon76-home | blitzrage: what's the page count? |
| 13:11.27 | blitzrage | Corydon76-home: no idea :( |
| 13:11.30 | Zeeek | I'm talking installed base, not The Wiz |
| 13:11.33 | blitzrage | I'm curious too! |
| 13:11.36 | mattfletcher | faxdetect is commented out, what do i need to do to cid? |
| 13:11.37 | [TK]D-Fender | Katty: Probably. But you should really eliminate that term "blast group" from your vocabulary. It does not parse.... |
| 13:11.38 | JT | summer, what an international season :P |
| 13:11.44 | [TK]D-Fender | Katty: Mew. (belated) |
| 13:11.55 | blitzrage | JT: talking NA summer of course since it's the only one that matters :) |
| 13:12.07 | Zeeek | Katty is sendmail still working? |
| 13:12.08 | JT | not sure when that is |
| 13:12.26 | blitzrage | May-September |
| 13:12.43 | blitzrage | or something like that |
| 13:13.09 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:13.09 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:13.11 | Corydon76-home | Officially, June - September |
| 13:13.35 | Zeeek | that's the Apple version :) |
| 13:13.45 | jeremy_g | does anyone have any idea of dimensioning an asterisk box in terms of erlang b model (we know number of lines, busy hour traffic and have to find the call blocking rate) |
| 13:13.48 | JT | 4 month season, does not compute |
| 13:14.00 | oddd | ok |
| 13:14.06 | oddd | I have narrowed down my problem |
| 13:14.10 | oddd | but I have no idea what it means |
| 13:14.16 | JT | seasons should be 3 months long :/ |
| 13:14.37 | oddd | playback does a ast_streamfile() |
| 13:14.39 | Corydon76-home | JT: it is... June 23rd - September 22nd |
| 13:14.45 | oddd | and then ast_waitstream() |
| 13:14.46 | Corydon76-home | or thereabouts |
| 13:14.48 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:14.49 | JT | strange country |
| 13:14.53 | oddd | but the waitstream never returns |
| 13:15.09 | Corydon76-home | JT: summer solstice until the autumn equinox |
| 13:15.20 | oddd | so for some reason the streamfile() function is putting data in, but it never being got out |
| 13:15.30 | JT | summer here is 01/12 to 28/02 |
| 13:15.57 | Corydon76-home | JT: we use celestial seasons |
| 13:15.58 | Katty | Zeeek: i think so (= |
| 13:16.22 | JT | hmm |
| 13:16.42 | Zeeek | Katty then you should have thrown a friendly hug my way |
| 13:17.01 | Zeeek | after making sure no rice carob milk was on it |
| 13:17.27 | [TK]D-Fender | JT : I'd guess jsut about all of them... |
| 13:17.28 | coppice | JT: if they don't, they suck |
| 13:17.48 | JT | well they seem to never mention it in their spec sheets |
| 13:17.59 | [TK]D-Fender | JT : Sure they do.... |
| 13:18.06 | JT | which one? :) |
| 13:18.42 | coppice | don't look at the stuff from joe's honest voip emporium |
| 13:19.29 | JT | heh |
| 13:19.33 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:19.33 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:19.34 | [TK]D-Fender | JT : http://www.voipsupply.com/product_info.php?products_id=459 |
| 13:19.41 | *** join/#asterisk sav_mcfly (n=R00T@pergamo.zonaz.net) |
| 13:19.44 | JT | also considering gear that wasn't originally intended for sip |
| 13:19.45 | [TK]D-Fender | JT : > Echo cancellation (G.168 up to 128ms) |
| 13:19.49 | JT | like lucent gear |
| 13:19.49 | mattfletcher | i'm using the switch statement to link two servers together over a vpn, but it has caused a problem. if i try and park a call, by transferring it to 700, it doesn't see that as a valid local extension and tries it across the switch bridge. can this be prevented? |
| 13:20.04 | jeremy_g | tzanger:no transcoding, 1,8 Ghz opteron (dual core) with 1 GB, all my web surfing concludes 100 concurrent calls is the most realistic figure |
| 13:20.14 | JT | 128ms, that's the same as the digium and sangoma cards with hwec isn't it? |
| 13:20.17 | [TK]D-Fender | JT : http://www.voipsupply.com/product_info.php?products_id=2847 |
| 13:20.19 | tzanger | jeremy_g: so try it |
| 13:20.22 | jeremy_g | all sip |
| 13:20.25 | [TK]D-Fender | JT : G.168 echo cancellation |
| 13:20.36 | tzanger | g168 is just a test specification is it not? |
| 13:21.04 | tzanger | i.e. you can see if the echo can passes g168 (how many varieties/annexes?) or not, but g168 does not in itself define an echo canceller |
| 13:21.04 | [TK]D-Fender | JT : I could go on and on... but if I found that never having specced each one out myself in so little time... you clearly aren't trying! |
| 13:21.20 | zdrulio | [TK]D-Fender: hum if i want to use "exten => _14.,1,Dial(IAX2/5000:password@HOST:PORT/${EXTEN:2},30,r)" can i view real number it nother side ? it is posible ? |
| 13:21.23 | coppice | tzanger: tests define most things |
| 13:21.48 | tzanger | coppice: they just determine whether the implementation passes the test |
| 13:21.48 | jeremy_g | tzanger:i am comfortable only with sipp for testing and i have no idea how the voice quality is being degraded when the call number is rising, it certainly crosses 100 and there are still active calls, i gotta have human users who can feedback the voice quality they get |
| 13:21.52 | tzanger | it doesn't define the implementation |
| 13:22.07 | [TK]D-Fender | zdrulio: Stop being a broken record and start doing the changes that were suggested to you. tzanger made some good points earlier as well and YOU are in need of some echo cancellation... |
| 13:22.11 | jeremy_g | need some itsp guy |
| 13:22.13 | tzanger | jeremy_g: so when it starts to degrade, throw more hardware at it |
| 13:22.25 | tzanger | there are some EXCELLENT posts on -users this past two weeks by Matt Roth I think which go into this in detail |
| 13:22.34 | coppice | nobody would want to define the implementation of an EC. that would be stupid. you want to define its required behavoiur. the G.168 spec does that |
| 13:22.41 | jeremy_g | tzanger:how can i check the voice degradation with sipp agents calling |
| 13:22.48 | tzanger | coppice: that's my point |
| 13:23.03 | jeremy_g | is there someone running an asterisk powered itsp |
| 13:23.04 | mattfletcher | has anyone got any experience of leadtek videophones on an asterisk system? |
| 13:23.05 | tzanger | EC#1 and EC#2 could acheive their results in very different ways, but they both pass g168 |
| 13:23.12 | jeremy_g | give me the concurrent call number |
| 13:23.26 | tzanger | they're not hte same thing, but they give the same result, or at least to the level of what g168 conformance gives |
| 13:23.28 | coppice | tzanger: ITU specs rarely define how. they define what |
| 13:23.41 | tzanger | coppice: I'm pretty picky on this, as I'm in the middle of ODVA certification right now |
| 13:24.01 | tzanger | the spec is tight, but even if you pass it there are corner cases that one implementation can work well where another doesn't |
| 13:24.06 | tzanger | even though they both pass spec |
| 13:24.07 | [TK]D-Fender | tzanger: the "how" is patentable, you can't have everyone using the same method under their own names now can we? ;) |
| 13:24.21 | coppice | tzanger: passing most specs is only the beginning |
| 13:24.25 | tzanger | now whether the conformance tests are sufficient or not... that's another point :-) |
| 13:24.49 | mattfletcher | can anyone help me with a switch statement problem. i've linked two servers together over a vpn, but if i try and park a call by transferring it to 700, it doesn't see that as a valid local extension and tries it across the switch bridge. can this be prevented? |
| 13:24.51 | [TK]D-Fender | tzanger: Standards we're meant to be lowered ;) Welcome to congress ..... |
| 13:25.09 | [TK]D-Fender | were* |
| 13:25.11 | [TK]D-Fender | asldklasj;dkjsfdsfdg |
| 13:25.15 | tzanger | [TK]D-Fender: heh |
| 13:25.19 | tzanger | you're still fat fingering too? |
| 13:25.32 | [TK]D-Fender | tzanger: No, that was deliberate (brain fart) |
| 13:25.42 | [TK]D-Fender | tzanger: I need a ^%$@#ing vacation |
| 13:26.27 | JT | dodgy ITU site, won't let you download G.168 for free for some reason |
| 13:26.59 | *** part/#asterisk sav_mcfly (n=R00T@pergamo.zonaz.net) |
| 13:27.02 | Katty | Zeeek: oh? |
| 13:27.05 | Katty | Zeeek: and why is that? |
| 13:27.10 | Katty | Zeeek: did you send one to me that i missed? |
| 13:30.36 | JT | oh, you can download the G.168 standards except not the new pre-published 2007 revision |
| 13:30.43 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:31.50 | coppice | new ITU trick. you can download anything released for free. all the interesting stuff is suddenly pre-release |
| 13:32.18 | JT | heh |
| 13:32.33 | JT | need to use the 3 free standards per email trick for that then |
| 13:32.46 | JT | but yeah, G.168 does appear to be a bunch of tests |
| 13:32.49 | coppice | the 3 downloads thing has gone |
| 13:33.29 | JT | weird it still talks about it on the web site |
| 13:33.30 | coppice | you can download the previous revisions for free |
| 13:34.00 | coppice | does it? maybe I missed that. i thought the 3 downloads had gone, and there was just the free offer |
| 13:34.29 | JT | i haven't tried to use it recently... but all the stuff refering to it is there |
| 13:34.49 | coppice | URL? |
| 13:34.57 | oddd | well |
| 13:35.00 | oddd | I solved it |
| 13:35.03 | oddd | no help to you guys |
| 13:35.06 | oddd | :P |
| 13:35.24 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:36.59 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:37.00 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:37.08 | JT | http://ecs.itu.ch/cgi-bin/ebookshop |
| 13:38.57 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:39.10 | *** join/#asterisk plantseeker (n=chatzill@83.167.161.28) |
| 13:39.36 | zdrulio | hm i have another question. about radius modul for asterisk. i have working asterisk 1.2 with wirking radius modul but now when i want to setup 1.4 aster, radius modul don`t work. any ideas ? |
| 13:41.44 | [TK]D-Fender | zdrulio: I'd bet that 1.4 is a major change and that module is in now way compatible with it. Go find a 1.4 monitoring module. |
| 13:42.35 | zdrulio | i`m searching but .. with no success |
| 13:42.56 | Corydon76-home | Then ask the author (nicely) if he will port it to 1.4 |
| 13:44.02 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 13:44.02 | *** mode/#asterisk [+o russellb] by ChanServ |
| 13:45.00 | Corydon76-home | If the author has the necessary licensing paperwork on file with Digium, contributing it back to the Asterisk project would be an option... and the author would no longer need to keep up with API changes |
| 13:45.27 | mihinomenest | I still can't get asterisk to respond to dtmf. |
| 13:45.56 | Corydon76-home | mihinomenest: on what channel? |
| 13:46.12 | mihinomenest | sip. |
| 13:46.28 | Corydon76-home | Is the DTMF mode lined up between client and server? |
| 13:46.44 | mattfletcher | can anyone help me with a switch statement problem. i've linked two servers together over a vpn, but if i try and park a call by transferring it to 700, it doesn't see that as a valid local extension and tries it across the switch bridge. can this be prevented? |
| 13:47.02 | mihinomenest | Corydon76-home: theoretically. |
| 13:47.19 | Corydon76-home | mihinomenest: which are you using? |
| 13:48.00 | mihinomenest | I've tried in-band, rtp, and sip on a grandstream that's registered directly to the provider's SIP server. |
| 13:48.24 | Corydon76-home | Oh, so it's your provider that is the problem |
| 13:48.25 | [TK]D-Fender | mattfletcher: Clearly you should ahve pastebin'd your dialplan aready so we could point out what was wrong instead of having to ask of guess blindly :) |
| 13:48.39 | mihinomenest | Corydon76-home: story of my life. |
| 13:49.02 | *** join/#asterisk galeras (n=root@201.244.240.115) |
| 13:49.18 | [TK]D-Fender | mihinomenest: What does your GS being connected directly to an ITSP have to do with *? |
| 13:49.31 | mihinomenest | that's how I'm calling *. |
| 13:49.36 | mihinomenest | it shouldn't. |
| 13:49.49 | mihinomenest | I should be able to call from a landline as well, but that doesn't work either. |
| 13:49.54 | Corydon76-home | Then it's not about the GS, it's about your provider and Asterisk |
| 13:50.02 | [TK]D-Fender | mihinomenest: Eliminate * from the equation and dial some other outside IVR and test your DTMF. |
| 13:50.18 | [TK]D-Fender | mihinomenest: See if it is your provider and not * on calling IN. |
| 13:50.26 | [TK]D-Fender | mihinomenest: Too many variables your way. |
| 13:50.57 | Corydon76-home | Or call your provider over a land line and connect to Asterisk that way |
| 13:51.01 | mihinomenest | it definitely works calling someone else's IVR. |
| 13:51.27 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 13:52.50 | mihinomenest | I'm going to guess that my provider's Sip server is sending DTMF in-band and asterisk can't read it. |
| 13:52.58 | mihinomenest | I suppose I need to do something about that. |
| 13:53.13 | *** join/#asterisk kissand (i=kissandd@pc1.ucnet.uoc.gr) |
| 13:53.21 | kissand | hello/help |
| 13:53.34 | kissand | i have purchased two tdm400p |
| 13:54.01 | kissand | i have installed the first on a pc it works fine (fax,ivr, etc etc) |
| 13:54.39 | kissand | i installed the second on HP proliant ML110 PCI-X slot without connecting the POTS line (yet) |
| 13:54.58 | kissand | when i give the command zap show status i get ok |
| 13:55.08 | kissand | rob0 i suspend something with irq too |
| 13:55.37 | kissand | the problem is that 1) the irq on tdm is 58? 2) in /proc/interrupts i dont see any conflicts |
| 13:55.48 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 13:55.57 | tzafrir_laptop | kissand, what is the actual problem? |
| 13:56.06 | syzygyBSD | how can I hangup a channel? |
| 13:56.11 | syzygyBSD | a local channel |
| 13:56.20 | tzafrir_laptop | soft hangup? |
| 13:56.24 | tzafrir_laptop | from the CLI? |
| 13:56.27 | syzygyBSD | :( nope |
| 13:56.45 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 13:56.47 | kissand | Wildcard TDM400P REV I Board 1 OK 0 0 0 |
| 13:56.47 | syzygyBSD | I think there was a problem, it did hangup, but was kept in the queue... |
| 13:56.56 | kissand | i didnt connect any line on tdm yet |
| 13:57.04 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:57.04 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:57.10 | kissand | it should be ALARM not OK |
| 13:57.16 | tzafrir_laptop | syzygyBSD, so soft hangup the other channel? |
| 13:57.28 | tzafrir_laptop | kissand, why should it be alarm? |
| 13:57.50 | kissand | and when i try to call through zap channel it shows that it is answered |
| 13:58.02 | kissand | tzafrir because i didnt connect any pstn line on tdm |
| 13:58.08 | syzygyBSD | there is no other channel... :( at least that I can find, 'show queues' shows that it is ringing, but I can't find it in show channels, |
| 13:58.50 | syzygyBSD | ahh, both ends are in the queue, |
| 13:59.20 | *** join/#asterisk bobbytux (n=bobbytux@LNeuilly-152-21-159-81.w193-253.abo.wanadoo.fr) |
| 13:59.26 | bobbytux | hello |
| 13:59.33 | tzafrir_laptop | kissand, this is a generally a known issue of the TDM400P card |
| 13:59.43 | syzygyBSD | how can I restart only 1 queue |
| 13:59.51 | tzafrir_laptop | This happns to work on X100P for the wrong reasons |
| 14:00.09 | tzafrir_laptop | And it works on our Astribank |
| 14:00.12 | coppice | JT: hey, waddaya know. the 3 downloads still works :-) |
| 14:00.18 | JT | coppice: :) |
| 14:00.50 | kissand | tzafrir the irq doesnt look too good |
| 14:00.55 | tzafrir_laptop | But even on the Astribank you won't see an alarm there, as this is a channel alarm and not a span alarm: the card itself is OK. Just some of its channels are disconnected |
| 14:01.01 | jkiff | Greetings, ya'll. I'm reading that fax over IP is... not so hot. This is mainly due to fax's low tolerance of jitter, packet loss, etc, so trying to fax via IAX, SIP, etc over the Internet would be pretty crappy. However, if I have a stable enough LAN, could I use SIP to get a fax as far as my Asterisk box where I then dump it onto the T1 just like any other outbound call? |
| 14:01.06 | kissand | SERVER:~ # more /proc/interrupts |
| 14:01.06 | kissand | <PROTECTED> |
| 14:01.06 | kissand | <PROTECTED> |
| 14:01.06 | kissand | <PROTECTED> |
| 14:01.06 | kissand | <PROTECTED> |
| 14:01.07 | kissand | <PROTECTED> |
| 14:01.09 | kissand | <PROTECTED> |
| 14:01.11 | kissand | <PROTECTED> |
| 14:01.12 | tzafrir_laptop | ~pb |
| 14:01.24 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 14:01.24 | kissand | 169: 1148368 0 IO-APIC-level eth0 |
| 14:01.24 | kissand | 217: 0 0 IO-APIC-level libata |
| 14:01.24 | kissand | 225: 364246 0 IO-APIC-level ioc0 |
| 14:01.24 | kissand | (sorry for the flood) |
| 14:01.25 | Strom_M | kissand: idiot, don't do thqt |
| 14:01.31 | Strom_M | use pastebin |
| 14:01.45 | tzafrir_laptop | looks like wctdm is sending tons of interupts |
| 14:01.57 | tzanger | what's wrong with that IRQ? |
| 14:02.00 | tzafrir_laptop | watch -n1 -d cat /proc/interrupts |
| 14:02.13 | JT | kissand: ALARMs are for digital lines |
| 14:02.16 | tzafrir_laptop | then you'll see clearly how this changes every second |
| 14:02.28 | tzafrir_laptop | You should have there some 1000 interrupts per second |
| 14:02.31 | kissand | 1000 interrupts per second |
| 14:02.40 | JT | kissand: you are coming into this with a lot of assumptions as to how things "should" run |
| 14:02.55 | bobbytux | does any of you have a W6692 working with chan_misdn.so ? |
| 14:02.58 | JT | alarms are generally not a feature relevant to analogue lines |
| 14:02.58 | bobbytux | thank you |
| 14:03.29 | *** part/#asterisk mattfletcher (n=matt@62-249-226-101.no-dns-yet.enta.net) |
| 14:03.34 | kissand | JT i saw alarm in the firts tdm400 when the pstn line was not connected |
| 14:04.04 | mihinomenest | if I put "dtmfmode=inband" in sip.conf, is that telling * to look for dtmf in-band, or is that telling asterisk to send dtmf in-band? |
| 14:04.12 | JT | beginner's luck ;) |
| 14:04.18 | JT | mihinomenest: both |
| 14:04.23 | JT | avoid inband, generally |
| 14:04.34 | kissand | the only differenct between the two installation is that the firts (working) is connected directly to public network while the other on a small classic telephne center |
| 14:04.39 | mihinomenest | indeed. |
| 14:05.00 | mihinomenest | unfortunately, to get the password from my provider's support department is sometimes asking too much. |
| 14:05.32 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 14:05.40 | kissand | i will connect a pstn line on the tdm and i will be back if it still wont work |
| 14:05.46 | kissand | lets go to the second issue |
| 14:06.04 | kissand | i have installed a 2-isdn port beronet card, which works fine |
| 14:06.32 | kissand | but when receiving fax i get |
| 14:06.35 | kissand | [Jun 6 19:51:57] NOTICE[20802]: channel.c:2353 __ast_read: Dropping incompatible voice frame on mISDN/1-u36 of format slin since our native format has changed to alaw |
| 14:06.36 | JT | kissand: wouldn't it be more important to focus on if the line works with asterisk instead of if asterisk says "alarm" or not? |
| 14:06.41 | mihinomenest | that didn't help. |
| 14:06.49 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:06.52 | JT | hah |
| 14:07.02 | JT | yet another misdn bug i bet |
| 14:07.04 | JT | misdn is junk |
| 14:07.27 | kissand | JT the installation that had "alarm" became "OK" when i connect the pstn line |
| 14:07.40 | JT | so? |
| 14:07.46 | JT | who cares, try and get it working |
| 14:07.54 | JT | alarms aren't important for analogue |
| 14:08.06 | mihinomenest | well, "Jun 7 06:06:45 WARNING[8647]: dsp.c:1426 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833" |
| 14:08.17 | tzafrir_laptop | kissand, you can see the same alarms in zttool, and in cat /proc/zaptel/* |
| 14:08.22 | syzygyBSD | how can I force the hangup of this local channel? http://pastebin.ca/547736 |
| 14:08.27 | JT | mihinomenest: that's obvious |
| 14:08.38 | kissand | JT hmm is this an misdn bug? :> i am trying to get fax with app_rxfax, iaxmodem and ATA and the only message i get is "Dropping... |
| 14:08.42 | mihinomenest | JT: unless you've never don it before. |
| 14:08.49 | kissand | should i use cvs for misdn? |
| 14:08.49 | *** join/#asterisk eeos (n=eeos@86.53.50.16) |
| 14:08.53 | JT | mihinomenest: inband means send in audio stream |
| 14:08.58 | eeos | hi everybody |
| 14:08.59 | syzygyBSD | soft hangup Local/12004@direct-600e ------------------ Local/12004@direct-600e is not a known channel |
| 14:09.00 | mihinomenest | I know. |
| 14:09.02 | JT | dtmf only works on uncompressed audio |
| 14:09.08 | Strom_M | mihinomenest: it should also be obvious if you know how the g729 codec works |
| 14:09.09 | JT | g.729 is highly compressed |
| 14:09.16 | tzafrir_laptop | "RED", "NOT_OPEN" or whateer shows as an extra text in the first line. "OK" means that there are no alarms, and hence no extra messages |
| 14:09.34 | mihinomenest | I don't really know how it works, just that it's retardedly compressed and you have to license it. |
| 14:09.51 | mihinomenest | and that the only real reason my provider wants to use it is so they don't have to pay for bandwidth. |
| 14:09.53 | JT | mihinomenest: that should tip you off that tones may not work through it :) |
| 14:10.12 | mihinomenest | well, it did. |
| 14:10.23 | mihinomenest | the problem is, I still need to make it work. |
| 14:10.33 | JT | there are 2 other dtmf modes |
| 14:10.39 | mihinomenest | yes, I know. |
| 14:10.47 | JT | if your provider doesn't work with either of them, junk your provider |
| 14:10.47 | mihinomenest | RFC 2833 and SIP. |
| 14:11.14 | kissand | mihinomenest dtmf=rfc2833 |
| 14:11.18 | JT | SIP INFO to be accurate |
| 14:11.19 | mihinomenest | I'm about to. |
| 14:11.22 | kissand | and in your sip client the same |
| 14:11.48 | kissand | JT is the "Dropping blahblah" an misdn bug? |
| 14:11.55 | JT | kissand: i think so |
| 14:12.04 | JT | avoid misdn if possible |
| 14:12.50 | kissand | is a replacement for beronet cards? |
| 14:12.50 | JT | not sure about the 2 port |
| 14:12.50 | JT | but bristuff works with other ones |
| 14:12.50 | JT | for sure |
| 14:13.25 | kissand | should i try *capi* something? |
| 14:13.49 | JT | only if bristuff doesn't have a driver for your card |
| 14:14.10 | JT | bristuff allows you to access the channels in a zaptel like fashion |
| 14:14.36 | kissand | yeap so i read, i will try ing |
| 14:14.46 | kissand | does it work with asterisk 1.4 versions? |
| 14:15.09 | kissand | i will find that out myself :> |
| 14:15.19 | kissand | JT thank you very much for yor help |
| 14:15.32 | JT | do to the download directory at junghanns, and download the latest version, it has a script inside that downloads the correct version of asterisk, libpri and zaptel for you and patches and compiles it alll |
| 14:15.37 | JT | no, 1.2 |
| 14:17.44 | kissand | it seems that there is a -test1 version for asterisk 1.4 |
| 14:18.34 | JT | i wouldnt use it |
| 14:19.13 | kissand | oh well -test1 and the first version is not very promising indeed :> |
| 14:19.26 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:20.01 | kissand | JT thank you for your help |
| 14:20.04 | kissand | cu |
| 14:21.02 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 14:25.06 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:28.51 | De_Mon | heh. |
| 14:29.16 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 14:39.23 | Bladerunner05 | Hi all I have tdm400p, with 4 fxo. in zaptel.conf I have to write fxsks=1-4 loadzone=xx defaultzone=xx ? |
| 14:41.33 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-169-41.dsl.irvnca.pacbell.net) |
| 14:42.09 | [TK]D-Fender | Bladerunner05: Looks about right |
| 14:42.52 | *** join/#asterisk btsteve (n=btsteve@204.10.20.30) |
| 14:44.14 | Strom_M | but all on separate lines, of course |
| 14:45.25 | Uatec | damn |
| 14:45.36 | Uatec | how to test credit card processing in a live environment without becoming poor |
| 14:45.53 | Strom_M | you call the credit card companies and get test card numbers |
| 14:45.53 | Strom_M | duh |
| 14:46.03 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:46.12 | Uatec | they don't do test card numbers on the live system |
| 14:46.20 | Uatec | that's why they call it live |
| 14:47.14 | Strom_M | charge your own card then refund your money |
| 14:48.05 | Uatec | oh |
| 14:48.08 | Uatec | maybe i should have said |
| 14:48.11 | Uatec | i'm a lazy git |
| 14:48.16 | Mercestes | .....damn, I meant to join #asterisk....how did I mistype that and get #credit-cards |
| 14:48.22 | *** join/#asterisk ToyMan (n=Stuart@74-32-22-252.dsl1.mdl.ny.frontiernet.net) |
| 14:48.29 | Uatec | i know Mercestes, i did the same |
| 14:48.30 | [TK]D-Fender | Uatec: Oh don't worry, thats well established here already ;) |
| 14:49.00 | Mercestes | Uatec, Did you try using your own credit cards test #? |
| 14:49.47 | Mercestes | There is a credit card hash you can run your real number against to generate a test number that you can use. |
| 14:49.54 | Mercestes | google credit card hash program |
| 14:50.16 | *** join/#asterisk eeos (n=eeos@86.53.50.16) |
| 14:50.23 | eeos | hi there! |
| 14:50.31 | Mercestes | Hi eeos. |
| 14:50.33 | key2 | Mercestes: url ? |
| 14:50.44 | Mercestes | key2: http://www.google.com duh |
| 14:51.35 | eeos | [TK]D-Fender: Zeeek after many attempts, finally our asterisk box connects properly to oe of our external voip providers! any user on the newtrok can open calls, and everybody is happy |
| 14:51.39 | eeos | :D |
| 14:51.39 | key2 | Mercestes: ahh u talking about the algorithm that multiply every other digit by 2 ? |
| 14:51.49 | eeos | for outbound calls .... |
| 14:51.57 | eeos | hi Mercestes |
| 14:52.04 | Mercestes | Key2: well, no it doesn't just do that it's a full hash, but yea, something very similar. |
| 14:52.23 | Mercestes | eeos: Congratz. |
| 14:52.25 | *** join/#asterisk waptaxi (n=cahe@45.151-224-87.telenet.ru) |
| 14:53.01 | eeos | Mercestes: more or less :) we cannot receive inbound calls (yet) and cannot use the other provider (we have two) |
| 14:53.04 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 14:53.41 | *** join/#asterisk ToyMan (n=Stuart@74-32-22-252.dsl1.mdl.ny.frontiernet.net) |
| 14:53.42 | Bladerunner05 | Don't know why.. using tdm400p (with 4 fxo) ztcfg -vv show me all channels correctly (fxs) I configure zaptel and zapata following the guide asterisk (1.4.4 with latest zaptel) don't show any errors but don't answer a call |
| 14:54.02 | Strom_M | have you told it to answer a call? |
| 14:54.37 | Bladerunner05 | in extension.conf exten => s,1,Answer() exten => s,2,Echo() |
| 14:54.42 | [[blah]asfd | does chanisavail work differently in 1.4 than in 1.2? The reason I ask is because I am using chanisavail to check to see if a queuemember is already on a call, if they are then the queue should get a busy response and move on to the next agent. However last night i upgraded the server to 1.4 and now they get presented with a call from the queue, even if they are already on a call. |
| 14:55.15 | syzygyBSD | are local channels ever created if not explicitly defined? |
| 14:55.51 | [[blah]asfd | here is how i do chanisavail http://pastebin.ca/547843 |
| 14:56.04 | [[blah]asfd | what I am thinking is that priority jumping is not supported any more. |
| 14:56.06 | [[blah]asfd | is that correct? |
| 14:57.04 | Bladerunner05 | <Strom_M> here is my config http://www.pastebin.ca/547847 |
| 14:57.25 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:57.35 | syzygyBSD | [[blah]asfd: well, your n+101 isn't right... |
| 14:58.16 | blitzrage | n+101? ewwwww :) |
| 14:58.25 | blitzrage | someone needs to learn about priority labels! |
| 14:58.34 | syzygyBSD | lol |
| 14:58.41 | syzygyBSD | I'll look that up sometime... |
| 14:59.58 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:59.58 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:01.13 | *** join/#asterisk Dorphalsig (n=root@200.71.58.39) |
| 15:01.17 | *** join/#asterisk AdamB0122 (n=adam@207.200.28.175) |
| 15:01.17 | Dorphalsig | Hello |
| 15:01.36 | AdamB0122 | Has anyone been able to properly use a PIX firewall for a Asterisk Box? |
| 15:02.33 | Dorphalsig | I have * 1.2.16 |
| 15:02.41 | [TK]D-Fender | AdamB0122: PIX is amongst the very WORSE firewalls to get * involved with. |
| 15:02.42 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 15:02.48 | [TK]D-Fender | AdamB0122: Go cehck the WIKI |
| 15:02.52 | [TK]D-Fender | check* |
| 15:02.54 | Dorphalsig | and Imḿ trying to compile |
| 15:02.54 | AdamB0122 | [TK]D-Fender Awesome |
| 15:02.55 | AdamB0122 | thanks |
| 15:02.57 | Dorphalsig | spandsp |
| 15:02.59 | Zeeek | check's in da mail |
| 15:03.10 | Dorphalsig | actually its app_rxfax and app_txfax |
| 15:03.18 | Dorphalsig | however I am getting compilations errors |
| 15:03.19 | Dorphalsig | such as |
| 15:03.28 | Bladerunner05 | <Strom_M>: any ideas ? |
| 15:03.56 | Dorphalsig | ./include/asterisk/plc.h:150: error: conflicting types for 'plc_fillin' |
| 15:04.14 | Dorphalsig | Kin anybody plz help me out? |
| 15:05.03 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:06.42 | *** join/#asterisk bbryant (i=brett@nat/digium/x-61985a85f3ee1db0) |
| 15:06.50 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 15:07.00 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 15:07.28 | IOscanner | Is there a way to monitor sip debug traffic via the manager port (5038) with a program? |
| 15:07.45 | Dorphalsig | hello? |
| 15:07.46 | IOscanner | Or even a perl or php script? |
| 15:09.06 | *** join/#asterisk b11d|bbl (n=no@234-200-29-134.hcc.mnscu.edu) |
| 15:09.20 | b11d|bbl | hi all |
| 15:09.32 | Zeeek | GET A ROOM |
| 15:09.37 | Dorphalsig | I dunno |
| 15:09.45 | Dorphalsig | he/she/it started it |
| 15:09.51 | b11d | can anyone tell me if it |
| 15:09.52 | b11d | doh |
| 15:10.00 | Dorphalsig | I ust wanna know if its a she |
| 15:10.06 | b11d | if it's possible to harvest uptime info from polycom phones via snmp or something like that? |
| 15:10.09 | Dorphalsig | Uatec: are you a she¿ |
| 15:10.41 | blitzrage | you're in #asterisk... chances it's a straight girl are slim :) |
| 15:10.49 | Dorphalsig | LOL |
| 15:10.54 | Zeeek | blitzrage very observant |
| 15:11.05 | Dorphalsig | Hrmmm |
| 15:11.14 | Dorphalsig | Ok, so Iĺl just back a bit |
| 15:11.20 | Dorphalsig | and try to get an answer for my question |
| 15:11.30 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 15:11.33 | Dorphalsig | Kin anybody help me get spandsp working with apprxfax? |
| 15:11.47 | Zeeek | doe spandsp work with 1.4 ? |
| 15:11.57 | Dorphalsig | Theoretically yes |
| 15:12.02 | Dorphalsig | I never managed to make it work |
| 15:12.04 | *** join/#asterisk ecoleman (n=eric@24.75.47.130) |
| 15:12.06 | Dorphalsig | then again |
| 15:12.06 | ecoleman | howdy folks |
| 15:12.10 | Zeeek | man is good, theoretically |
| 15:12.22 | Zeeek | salut écoleman |
| 15:12.28 | Dorphalsig | right now im having a hard time making it work with 1.2 |
| 15:13.01 | *** join/#asterisk saftsack (n=oliver@p54A7F024.dip.t-dialin.net) |
| 15:13.21 | ecoleman | since soxmix is gone in sox v13.0.0, is it safe to just create a shell script that just has: /usr/local/bin/sox -m "$@" ? |
| 15:13.56 | [[blah]asfd | syzygyBSD: k made a change... I am not sure if i got this correct or not. http://pastebin.ca/547890 |
| 15:14.03 | Dorphalsig | Come on.... sombody must know spandsp |
| 15:14.07 | Dorphalsig | and how to make it work |
| 15:14.09 | Dorphalsig | damnit |
| 15:14.11 | [[blah]asfd | am i understanding how to do that correctly? |
| 15:14.17 | Dorphalsig | Im getting compilation issues |
| 15:14.49 | [[blah]asfd | Dorphalsig: issues compiling, you may want to check a linux room. what disrto are you using? |
| 15:19.08 | IOscanner | Is there a way to dump sip debug messages to a file only? |
| 15:19.20 | IOscanner | What kind of impact will this have on asterisk? |
| 15:19.43 | s0ck | well, it's not going to improve performance :P |
| 15:20.24 | Bladerunner05 | Don't know why.. using tdm400p (with 4 fxo) ztcfg -vv show me all channels correctly (fxs) I configure zaptel and zapata following the guide asterisk (1.4.4 with latest zaptel) don't show any errors but don't answer a call |
| 15:20.34 | *** join/#asterisk Waverly360 (n=Waverly3@209.12.249.243) |
| 15:21.10 | Bladerunner05 | here is my config http://www.pastebin.ca/547847 |
| 15:21.20 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 15:21.35 | IOscanner | I know that, but how much |
| 15:21.57 | brodiem | hey guys, how can I quickly bridge a zap channel to another channel (i.e. a Zap channel sitting in a queue) |
| 15:22.36 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:23.08 | Zeeek | Bladerunner05 I missed your question, what was it? |
| 15:23.44 | *** join/#asterisk murdmath (n=mmurdock@mail.kimballequipment.com) |
| 15:23.57 | Waverly360 | I'm still having some callerid issues. How can the callerid(name) variable be empty if I know for a fact that the callerid is being passed into asterisk from a pri? |
| 15:24.01 | Bladerunner05 | Don't know why.. using tdm400p (with 4 fxo) ztcfg -vv show me all channels correctly (fxs) I configure zaptel and zapata following the guide asterisk (1.4.4 with latest zaptel) don't show any errors but don't answer a call |
| 15:24.25 | Zeeek | did you try calling out? |
| 15:24.57 | Bladerunner05 | no I don't |
| 15:25.03 | Bladerunner05 | do it |
| 15:25.23 | Zeeek | Dial(ZAP/1) and see what happens |
| 15:26.20 | Zeeek | and no indication of call on CLI when it is coming in? |
| 15:26.28 | Uatec | Hi, i'm trying to send my caller ID when I make a call using Set(CALLERID(number)=XXXXX) but all i get is Unknown |
| 15:26.37 | Uatec | i'm using asterisk business edition 1.3 |
| 15:26.41 | Uatec | is that not the way to do it? |
| 15:27.09 | blitzrage | Uatec: where are you sending it to? |
| 15:27.18 | Uatec | my isdn card |
| 15:27.20 | *** join/#asterisk bkw__ (n=brian@adsl-70-143-39-83.dsl.tul2ok.sbcglobal.net) |
| 15:27.31 | Uatec | and eventually my mobile phone, which definately does support it |
| 15:27.35 | blitzrage | whoever is providing the ISDN connection probably isn't allowing you to set CID |
| 15:27.37 | Uatec | becuase i've received it from this line before... |
| 15:27.46 | blitzrage | received -- not send |
| 15:27.55 | eeos | Zeeek: fiished the exercise AND the for levels above ....HA! |
| 15:28.03 | Uatec | i have sent if from the isdn to the mobile before |
| 15:29.49 | [TK]D-Fender | Bladerunner05: Stop repeating the same complaint over and over again every 5 minutes. We heard you the first 10 times. |
| 15:29.55 | Zeeek | eeos now you're ready for certification |
| 15:30.08 | [TK]D-Fender | Bladerunner05: Pastebin your zaptel.conf, zapata.conf, and extensions.conf. |
| 15:30.10 | [TK]D-Fender | ~pb |
| 15:30.26 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 15:30.26 | eeos | Zeeek: :P |
| 15:30.38 | eeos | Zeeek: well, outbound calls work, but inbound calls not yet |
| 15:30.46 | Zeeek | so, no cigar |
| 15:31.00 | Waverly360 | [TK]D-Fender: so I shouldn't keep asking about callerid then? :P |
| 15:31.02 | *** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net) |
| 15:31.15 | eeos | Zeeek: :( |
| 15:31.18 | AndrewGearhart | morning [TK]D-Fender |
| 15:31.25 | Zeeek | there's a special on callerid on aisle 4 today only |
| 15:32.19 | [TK]D-Fender | Waverly360: "You have the right to remain silent... so STFU!" |
| 15:32.19 | Zeeek | [TK]D-Fender heh |
| 15:32.19 | eeos | Zeeek: oe thing at a time, man :) |
| 15:32.19 | [TK]D-Fender | > unload chan_bile.so |
| 15:32.19 | Zeeek | the word finished means finished |
| 15:32.26 | AndrewGearhart | hey folks, I'm seeing "SIP Trunking" how is that different from, say, an ITSP? |
| 15:32.38 | *** join/#asterisk AvoidingDeadlock (n=brian@adsl-70-143-39-83.dsl.tul2ok.sbcglobal.net) |
| 15:32.54 | Zeeek | there should be a way to have everything you type in the CLI say $what_you_typed is now deprecated |
| 15:32.57 | *** join/#asterisk floppp (n=flop@nat-staff.b3g-telecom.com) |
| 15:33.31 | *** mode/#asterisk [+b *!*n=brian@*.dsl.tul2ok.sbcglobal.net] by russellb |
| 15:33.31 | *** kick/#asterisk [AvoidingDeadlock!i=russellb@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb) |
| 15:33.39 | *** join/#asterisk ^majik^ (n=kvirc@68-187-20-73.static.uncty.tn.ken-tennwireless.com) |
| 15:33.39 | *** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
| 15:33.45 | eeos | Zeeek: well, I finished with redirecting outbund calls on an external provider, and I am pretty happy with the result :P |
| 15:34.05 | Zeeek | eeos after about 10 days on that, you can be proud |
| 15:34.06 | Dorphalsig | [[blah]asfd: I have FC4 |
| 15:34.07 | [TK]D-Fender | AndrewGearhart: Probably synonymous in intent, and inappropriately worded. |
| 15:34.51 | eeos | Zeeek: ah, ah, ah , ah (actually 4 days 2 hour a day) |
| 15:34.54 | *** mode/#asterisk [-b *!*n=brian@*.dsl.tul2ok.sbcglobal.net] by russellb |
| 15:35.03 | Waverly360 | [TK]D-Fender: Unfortunately, I don't know you well enough at this point to tell if you're joking, or just in a bad mood :P. |
| 15:35.05 | Zeeek | for three lines, it's still a lot :) |
| 15:35.20 | Dorphalsig | [[blah]asfd: The error is: ../include/asterisk/plc.h:156: error: conflicting types for 'plc_init' |
| 15:35.28 | *** join/#asterisk Ironhand (n=arjen@meek.xs4all.nl) |
| 15:35.35 | ^majik^ | does anyone know where I can find some good info on asterisk 1.4's users.conf file? |
| 15:35.42 | Dorphalsig | wiki |
| 15:35.49 | eeos | Zeeek: not everybody has esoteric capability of interpreting cryptic documentation |
| 15:36.05 | [TK]D-Fender | Waverly360: If I quote like that its likely in jest, but there might be a grain to it in the "don't push your luck" category :) |
| 15:36.21 | AndrewGearhart | I'm trying to decipher http://www.simplesignal.com/siptrunking.html it seems to be what I'd like to do running VoIP on the outside. |
| 15:36.32 | Waverly360 | [TK]D-Fender: Understood. I'll try to be wary :). |
| 15:36.33 | Zeeek | eeos BS! The documentation at that level is perfectly clear |
| 15:36.55 | ^majik^ | Dorphalsig: not much of anything about users.conf on voip-info.org. is there another wiki? |
| 15:36.59 | Bladerunner05 | Zeek: in my extension.conf I put exten => _0.,1,Dial(Zap/1) but not found |
| 15:37.04 | [TK]D-Fender | AndrewGearhart: Ok, forget that and just tell me straight what you want to do exactly. |
| 15:37.39 | [TK]D-Fender | AndrewGearhart: These guys seem to be a poorly worded ITSP |
| 15:37.41 | Zeeek | Bladerunner05 what is not found? The extension? |
| 15:37.59 | AndrewGearhart | [TK]D-Fender: k. typing and trying to keep it concise so it's short, sweet and clear. be just a moment. |
| 15:38.09 | Bladerunner05 | Zeek: when I make a call xlite answer not found |
| 15:38.12 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 15:38.14 | eeos | Zeeek: I will have to seriously disagree with you |
| 15:38.25 | [TK]D-Fender | Bladerunner05: I asked you for your configs to see where things may have gone wrong. Sop asking for help until you provide them. |
| 15:38.25 | mocker | Anyone have any tips for centralizing voicemail from multiple servers to one? |
| 15:38.34 | Zeeek | eeosgive it a rest and write a better documentation |
| 15:38.42 | jkiff | Is there a way to redirect what shows up in the CLI to a file? `asterisk -rvvvvvvvvvv > asterisk.log` doesn't seem to work. |
| 15:38.44 | mocker | I tried NFS, but the lag between the servers causes a huge pause when checking for new mail. |
| 15:38.55 | Defraz | Is there some place that explains the different Dtmfmodes and how to use them and so on. |
| 15:39.14 | Zeeek | http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode |
| 15:39.44 | Dorphalsig | jkiff: just go to logger.conf |
| 15:39.46 | mihinomenest | Turns out my provider wanted more money to turn on "extension dialing" |
| 15:39.59 | Dorphalsig | jkiff: and set full debugging options |
| 15:40.49 | Bladerunner05 | Zeek: this is all my configuration: http://www.pastebin.ca/547950 |
| 15:40.51 | jkiff | Dorphalsig: Ah, I see! Thanks. :) |
| 15:42.16 | Zeeek | Bladerunner05 what is the output of ZAP SHOW CHANNELS ? |
| 15:43.42 | Bladerunner05 | http://www.pastebin.ca/547957 |
| 15:45.05 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 15:45.07 | [TK]D-Fender | Bladerunner05: X-Lite tells you "not found"? |
| 15:45.07 | [TK]D-Fender | Bladerunner05: What are you dialing on it? |
| 15:45.07 | Bladerunner05 | <[TK]D-Fender>: yes when i try to call out ? |
| 15:45.07 | [TK]D-Fender | Bladerunner05: What # are you dialing? |
| 15:45.19 | Bladerunner05 | <[TK]D-Fender>: an external number like 099xxxxx |
| 15:45.49 | [TK]D-Fender | Bladerunner05: turn on SIP debug, and try your call again. pastebin ALL of the output. |
| 15:46.00 | eeos | Zeeek: I have written documentation for fsos projects, so .... |
| 15:47.01 | Zeeek | I think the new ps solved the hum on the Polycom ip500 with headset |
| 15:47.29 | Zeeek | spent entire morning installing 1.4 on new box |
| 15:47.45 | Zeeek | didn't reach the pint of port forwarding so no ssh |
| 15:47.49 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 15:48.10 | Zeeek | basically, the success would indicate I should now quit until Monday |
| 15:48.22 | Zeeek | or risk screwing up something badly tomorrow |
| 15:48.43 | AndrewGearhart | GOAL: Lower our phone/internet bills & maintain quality phone & data service. |
| 15:49.04 | AndrewGearhart | We need to have the ability for 5 "lines" to 10 extensions. I've decided it will be VoIP inside the office from desks to asterisk. |
| 15:49.12 | AndrewGearhart | I now need to decide how asterisk will be connecting to POTS... 1) via Sangoma A200d to analog lines or 2) to an ITSP |
| 15:49.13 | [TK]D-Fender | AndrewGearhart: How much are you paying for LD now? What kind of lines? What cost/line/channel? |
| 15:49.34 | [TK]D-Fender | AndrewGearhart: many would advise against using VoIP as your primary business telco link. |
| 15:49.45 | *** join/#asterisk saftsack (n=oliver@p54A7EE38.dip.t-dialin.net) |
| 15:49.47 | Bladerunner05 | this is the output when I try to call out http://www.pastebin.ca/547970 |
| 15:49.58 | AndrewGearhart | LD is $0.06/m s-to-s $0.065 in-state |
| 15:50.10 | Bladerunner05 | <[TK]D-Fender> this is the output when I try to call out http://www.pastebin.ca/547970 |
| 15:50.16 | ManxPower | AndrewGearhart: Do you like working where you work? |
| 15:50.17 | Zeeek | horribly expensive if that's the US |
| 15:50.20 | AndrewGearhart | each line is currently $45-$48 each |
| 15:50.48 | AndrewGearhart | ManxPower: yes. this is a side project. why? |
| 15:51.10 | ManxPower | AndrewGearhart: If you hate your job then go with the ITSP. If you like your job then go with the PSTN lines. |
| 15:51.11 | [TK]D-Fender | Bladerunner05: Pastebin "show dialplan" |
| 15:51.33 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 15:51.34 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 15:51.46 | [TK]D-Fender | AndrewGearhart: Your LD rate is pure BS. I get .03$/min Cdn easy. |
| 15:52.00 | ManxPower | AndrewGearhart: You could go with a hybrid approach. Have the number of PSTN lines to handle most of your call volume, then use an ITSP for overflow |
| 15:52.13 | [TK]D-Fender | AndrewGearhart: Threaten your telco rep with departure and negociate your rate down. |
| 15:52.15 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-fe74fe70049e094f) |
| 15:52.25 | Bladerunner05 | ] <[TK]D-Fender> : the result are many lines .... |
| 15:52.33 | [TK]D-Fender | Bladerunner05: ALL OF IT. |
| 15:52.37 | Zeeek | dialplan show interni |
| 15:52.43 | [TK]D-Fender | Bladerunner05: Let me sift through the crap. |
| 15:52.57 | [TK]D-Fender | Zeeek: I'm sure he didn't apply his changes.... |
| 15:53.04 | Zeeek | yeah no ext rel |
| 15:53.05 | [TK]D-Fender | Zeeek: I'm jsut waiting to SEE it. |
| 15:53.16 | Zeeek | and then leap mercilessly |
| 15:53.19 | [TK]D-Fender | Bladerunner05: Oh and turn off SIP debug now. |
| 15:53.23 | [TK]D-Fender | "sip no debug" |
| 15:53.35 | Bladerunner05 | dialpan show interni return: There is no existence of 'interni' context |
| 15:53.54 | Zeeek | exten <tab> rel <tab> |
| 15:53.54 | [TK]D-Fender | Bladerunner05: Looks like you updated extensions.conf but didn't put the change in EFFECT |
| 15:54.03 | [TK]D-Fender | Bladerunner05: "reload" ! |
| 15:54.08 | [TK]D-Fender | Bladerunner05: And try again |
| 15:54.25 | Bladerunner05 | if I do show dialplan the latest line is: -= 68 extensions (134 priorities) in 40 contexts. =- |
| 15:54.32 | Zeeek | of course this wouldn't have happened with polycoms. |
| 15:54.39 | AndrewGearhart | [TK]D-Fender: I've found a CLEC that will do unlimited for the regular lines... at about the same rate / mo for the regular line charge we currently pay |
| 15:55.00 | Bladerunner05 | I reload but the problem is the same |
| 15:55.02 | [TK]D-Fender | AndrewGearhart: Sounds good.... wheres the problem? |
| 15:55.03 | ManxPower | AndrewGearhart: both analog ports and voip each have their drawbacks |
| 15:55.25 | [TK]D-Fender | Bladerunner05: pastebin "ls -l /etc/asterisk" I have a strong suspicion. |
| 15:55.25 | ManxPower | AndrewGearhart: a PRI would be BEST, of course. |
| 15:55.44 | Zeeek | ManxPower we even left one analog phone on the main line just in case asterisk itslef is down and we haven't notied |
| 15:55.47 | Uatec | hey, does anybody know how to persuade an aastra 9133i to connect to line 2, rather than line 1? |
| 15:55.50 | AndrewGearhart | [TK]D-Fender: I still have to pay $.045 for the toll-free lines on inbound. |
| 15:55.52 | Bladerunner05 | <[TK]D-Fender>: here is the result of make samples after compiling * |
| 15:56.06 | [TK]D-Fender | AndrewGearhart: Again thats BS... I'm gett .03c/m |
| 15:56.09 | Bladerunner05 | <[TK]D-Fender>: this is your suspect? |
| 15:56.19 | AndrewGearhart | ManxPower: the only provider of PRI in our area is the same JAs that we're using for the analog lines and wants an absolute fortune. |
| 15:56.21 | ManxPower | Bladerunner05: The sample config files are NOT a working configuration |
| 15:56.25 | [TK]D-Fender | Bladerunner05: If you make samples you are KILLING YOUR CONFIGS |
| 15:56.34 | Zeeek | which could be a problem |
| 15:56.45 | AndrewGearhart | [TK]D-Fender: any idea if your provider is available in Saint Marys, PA, USA? |
| 15:56.46 | Bladerunner05 | OK what file I have to delete ? |
| 15:56.56 | [TK]D-Fender | Bladerunner05: Provide me what I last requested of you |
| 15:57.06 | Bladerunner05 | The other strange problem is that id didn't respond when I call it |
| 15:57.14 | [TK]D-Fender | Bladerunner05: No, "MAKE SAMPLES" is KILLING YOUR CONFIGS. Stop doing !. you ahve nothing to delete! |
| 15:57.37 | Bladerunner05 | <[TK]D-Fender>: I do make samples before write my configuration |
| 15:57.42 | [TK]D-Fender | Bladerunner05: Stop everything else and provide what I jsut requested. and "cat" out your dialplan from Linux CLI |
| 15:58.02 | [TK]D-Fender | Bladerunner05: And make sure I can see it ALL (include the catul commands used) |
| 15:58.22 | Bladerunner05 | <[TK]D-Fender>: how do that ? |
| 15:58.36 | [TK]D-Fender | cat [your extensions file] |
| 15:58.41 | Zeeek | [TK]D-Fender what do you know about selectmenu ? |
| 15:59.04 | ManxPower | Bladerunner05: "do" is current tense, "did" is past tense. |
| 15:59.04 | [TK]D-Fender | Bladerunner05: And I asked you to do Bladerunner05: pastebin "ls -l /etc/asterisk" I have a strong suspicion. |
| 15:59.04 | Bladerunner05 | <[TK]D-Fender> You need to see all config files ? |
| 15:59.16 | [TK]D-Fender | Bladerunner05: I told you EXACTLY what I wanted... |
| 15:59.16 | Bladerunner05 | ok |
| 15:59.56 | [TK]D-Fender | Bladerunner05: Along with the "cat" of your dialplan showing me your linux CLI commands being called as well as their output. |
| 16:00.27 | Bladerunner05 | http://www.pastebin.ca/547999 |
| 16:01.40 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-6e191363ebbbffa6) |
| 16:01.40 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 16:02.16 | Zeeek | ummmm what day is there, Bladerunner05 ? |
| 16:02.39 | [TK]D-Fender | Bladerunner05: Please "cat" out your dialplan... |
| 16:02.42 | Zeeek | I'd say |
| 16:02.48 | [TK]D-Fender | Bladerunner05: the last thing I asked you for |
| 16:02.52 | Zeeek | extension.conf from today is a good place to begin |
| 16:03.01 | [TK]D-Fender | ManxPower: Do YOU see whats about to happen? |
| 16:03.05 | Zeeek | 2007-06-07 18:40 extension.conf |
| 16:03.09 | [TK]D-Fender | Zeeek: SHUSH! |
| 16:03.20 | [TK]D-Fender | Zeeek: I'm giving him rope! |
| 16:03.22 | ManxPower | [TK]D-Fender: you are going to tell him he is an idiot and is editing the wrong files. |
| 16:03.24 | Zeeek | you guys are worse than cats playing with mice! |
| 16:03.33 | Bladerunner05 | This is the result of my dialplan: http://www.pastebin.ca/548006 |
| 16:03.33 | Zeeek | no stop |
| 16:03.42 | *** join/#asterisk hegars (n=hegars@202-154-103-68.people.net.au) |
| 16:03.43 | ManxPower | oh, and be sure to mention that you will never get that hour back. |
| 16:03.49 | hegars | hello |
| 16:03.57 | [TK]D-Fender | Bladerunner05: I said CAT it from linux CLI |
| 16:04.05 | ManxPower | Zeeek: But it is FUN to see them twitch. |
| 16:04.09 | Zeeek | stop, I can't bear to watch this |
| 16:04.18 | Bladerunner05 | I do asterisk -r -x "show dialplan" is correct ? |
| 16:04.30 | [TK]D-Fender | Bladerunner05: "CAT" <------- |
| 16:04.35 | Zeeek | [TK]D-Fender YOU DIDN'T ANSWER MY QUESTION |
| 16:04.35 | hegars | does anyone have the beat copies of the firemwares for the Grandstream GXP2000 that they can send me |
| 16:04.39 | [TK]D-Fender | Bladerunner05: From Linux CLI. No, NOT Asterisk -anything |
| 16:04.53 | ManxPower | "Love to eat 'dem mousies. Mousies is what I love to eat! Bite their little heads off, nibble on their tiny feet!" |
| 16:05.22 | [TK]D-Fender | Zeeek: Not a question-mark anywhere in the last 20 mins from you directed at me :) PUNCTUATE! |
| 16:05.24 | Zeeek | a sense of community |
| 16:05.44 | Zeeek | [17:58] <Zeeek> [TK]D-Fender what do you know about selectmenu ? |
| 16:05.53 | [TK]D-Fender | Zeeek: We are a community... you can't have a good lynching without "community"! ;) |
| 16:06.13 | Bladerunner05 | try this http://www.pastebin.ca/548013 |
| 16:06.19 | [TK]D-Fender | Zeeek: Term doesn't ring a bell.... |
| 16:06.31 | Zeeek | term.... <bell> muhahaha |
| 16:07.06 | [TK]D-Fender | Bladerunner05: You are in ASTERICK CLI there. |
| 16:07.11 | [TK]D-Fender | ASTERISK* |
| 16:07.18 | [TK]D-Fender | Bladerunner05: I said CAT it from LINUX. |
| 16:07.37 | Zeeek | this is no baptism by fire. You guys don't know what is was like with JerJer "put the crack pipe down" |
| 16:07.57 | ManxPower | I like JerJer |
| 16:07.59 | Zeeek | Bladerunner05 ils se moquent de toi mon pote! |
| 16:08.04 | Zeeek | I miss JerJer |
| 16:08.05 | [TK]D-Fender | Zeeek: I've been here for years.... he's far from the first to use it... |
| 16:08.09 | Uatec | http://asymptotia.com/wp-images/2006/12/copy_cat_copies.jpg <-- Bladerunner05??? |
| 16:08.26 | Bladerunner05 | this is cutted from shell command cat extension.conf http://www.pastebin.ca/548024 |
| 16:08.33 | Bladerunner05 | I hope I understand |
| 16:08.39 | [TK]D-Fender | THERE!!! |
| 16:08.44 | [TK]D-Fender | Finally the EVIDENCE! |
| 16:08.51 | [TK]D-Fender | anapbx:/etc/asterisk# cat extension.conf <- this is NOT EXTENSIONS.CONF! |
| 16:09.03 | cpm | Zeeek, lol! |
| 16:09.05 | [TK]D-Fender | Bladerunner05: You may as well have named it FRED.TXT |
| 16:09.18 | Zeeek | c'est trop penible les mecs |
| 16:09.21 | Bladerunner05 | AZZZZZZZZZZZZZZZZZ |
| 16:09.21 | *** part/#asterisk galeras (n=root@201.244.240.115) |
| 16:09.28 | [TK]D-Fender | Bladerunner05: WTF are you doing editing a file named "extension.conf" and thinking it MATTERS? |
| 16:09.33 | murdmath | Uatec: Did you figure out your caller id thing? |
| 16:09.35 | Bladerunner05 | Sure................ |
| 16:09.44 | Bladerunner05 | Now I try to rename it |
| 16:09.46 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:09.56 | Zeeek | well, linux is so hot, it should be able to guess. |
| 16:10.18 | Zeeek | didn't VAX-VMS have a thing like "did you mean extensions.conf?" |
| 16:10.25 | [TK]D-Fender | Zeeek: No, that'd be "Asterisk" being "so smart".... to linux its jsut files, but * should be able to guess your configs! |
| 16:10.53 | cy303 | heh |
| 16:10.54 | Zeeek | ah but the underlying FS should see the sys call and see the similarity |
| 16:10.54 | [TK]D-Fender | My work here is done.... |
| 16:10.57 | [TK]D-Fender | *sigh* |
| 16:10.58 | cpm | Zeeek, penible ? |
| 16:11.07 | Zeeek | gesus H triste |
| 16:11.13 | Uatec | what is the opposite to a blind transfer? |
| 16:11.31 | Uatec | murdmath, nope |
| 16:11.35 | [TK]D-Fender | Uatec: Hanging up on the caller |
| 16:11.46 | hegars | lol |
| 16:12.00 | Zeeek | C&C |
| 16:12.34 | Bladerunner05 | <[TK]D-Fender> : I do that but the problem is the same when I make a call outside.... not found |
| 16:12.44 | murdmath | Uatec: What switchtype is your PRI? |
| 16:12.58 | Zeeek | Bladerunner05 restart asterisk |
| 16:12.59 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 16:13.01 | Bladerunner05 | this is the result of show dialplan from cli> http://www.pastebin.ca/548038 |
| 16:13.04 | [TK]D-Fender | Bladerunner05: Show me that the file is in the right state and IN EFFECT. |
| 16:13.05 | Bladerunner05 | Zeek: done |
| 16:13.12 | Bladerunner05 | ok do that |
| 16:13.38 | [TK]D-Fender | Bladerunner05: Ok, that looks better. Now re-enable sip debug and pastebin the new call attempt |
| 16:13.51 | Bladerunner05 | this is the content of the file http://www.pastebin.ca/548044 |
| 16:13.54 | Zeeek | and add an X to the '_0X.' => |
| 16:13.58 | Zeeek | just for fun |
| 16:15.02 | Daejeo1 | I want to handle 1000-1300 concurrent calls. what kind of server(rack/tower) should I buy? |
| 16:15.09 | Bladerunner05 | Now works the call start..... |
| 16:15.13 | sunsmasher | we should have a cpan contest.. who can try to install and get the most required dependencies |
| 16:15.14 | Bladerunner05 | thanks guys |
| 16:15.27 | Bladerunner05 | now I try to see if * answer an incoming call |
| 16:15.27 | Zeeek | why does it work now? |
| 16:15.39 | *** join/#asterisk tdonahue-laptop (n=tdonahue@static-acs-24-154-94-8.zoominternet.net) |
| 16:15.51 | Sweeper | Daejeo1: it should be a tower, and for the calls to sound really good, it should probably be blue |
| 16:16.05 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:16.13 | tdonahue-laptop | anyone here successfully running zaptel 1.4.2 on debian etch? |
| 16:16.30 | Daejeo1 | Sweeper: brand name? |
| 16:16.37 | [TK]D-Fender | Bladerunner05: Yay! |
| 16:16.39 | *** join/#asterisk flot (n=flot@user183.hovrino.net) |
| 16:16.42 | Uatec | murdmath, BRI mate... |
| 16:16.44 | Sweeper | Daejeo1: Initech |
| 16:16.54 | hegars | yep |
| 16:16.56 | tdonahue-laptop | i'm getting kernel panics on my amd64 version of etch any time i attempt to start a meetme conference |
| 16:17.03 | cy303 | tdonahue-laptop: on ubuntu.. |
| 16:17.20 | tdonahue-laptop | cy303, i386 or amd64? |
| 16:17.27 | cy303 | i386 |
| 16:17.41 | Daejeo1 | Sweeper: going to use E1 connections |
| 16:17.48 | hegars | tdonahue-laptop: i run it on debian etch i386 too |
| 16:18.09 | tdonahue-laptop | hmm... wonders if he has a i386 box around to test on... |
| 16:18.09 | *** join/#asterisk n0n4m3 (n=NoName@noname.rula.net) |
| 16:18.09 | *** part/#asterisk n0n4m3 (n=NoName@noname.rula.net) |
| 16:18.09 | *** join/#asterisk n0n4m3 (n=NoName@noname.rula.net) |
| 16:18.09 | Sweeper | Daejeo1: then you should get the Initech RS-E |
| 16:18.09 | murdmath | Uatec: It think the options are National, ni1, dms, 4ess, 5ess, euroisdn, and qsig I think. |
| 16:18.16 | tdonahue-laptop | s/wonders/i wonder/ |
| 16:18.19 | Sweeper | it has e1 connectors on the front panel |
| 16:18.20 | flot | hello. what is asnparser ? |
| 16:18.40 | tdonahue-laptop | oooo.. thats cool |
| 16:19.02 | rene- | hello |
| 16:19.08 | rene- | any mitel users out there? |
| 16:19.13 | *** join/#asterisk n0n4m3 (n=NoName@noname.rula.net) |
| 16:19.28 | tdonahue-laptop | no voip on our mitel system though |
| 16:19.29 | Bladerunner05 | Really don't works this is the result: http://www.pastebin.ca/548052 |
| 16:20.17 | *** join/#asterisk QbY (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 16:20.30 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 16:20.44 | Daejeo1 | Sweeper: can you point me to url? |
| 16:21.05 | QbY | If I have a user dialing a 11 digit number that is actually in my dialplan, how is the best way to check for its existance before sending it off to the PSTN? |
| 16:21.06 | *** join/#asterisk b4ka (n=jh@200.69.198.45) |
| 16:21.11 | [TK]D-Fender | Bladerunner05: Ok this one I'll jsut GIVE you... |
| 16:21.28 | [TK]D-Fender | Bladerunner05: You are doing : exten => _0.,1,Dial(ZAP/1) |
| 16:21.48 | [TK]D-Fender | Bladerunner05: this mean it will take ANY number that starts with a "0" and has at least 1 more digit. |
| 16:22.04 | b4ka | hey, anyone knows why im getting this message on some calls after a couple minutes? DEBUG[25626] channel.c: Didn't get a frame from channel: SIP/2617-083157e0 |
| 16:22.16 | b4ka | and then the call drops |
| 16:22.19 | [TK]D-Fender | Bladerunner05: And then it will take your 1st line and just give you DIALTONE without passing on the whole number or any aprt of what you dialed. |
| 16:22.21 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 16:22.29 | Uatec | how can i do different things if my call is left unanswered rather than if the phone is busy |
| 16:22.37 | Uatec | i.e. i have a sip device and i Dial() it |
| 16:22.58 | Uatec | i want to go to answerphone if the phone is not answered |
| 16:23.03 | Sweeper | Daejeo1: sorry, not |
| 16:23.08 | Sweeper | you could call them tho |
| 16:23.10 | [TK]D-Fender | Bladerunner05: If you wanted to dial the ENTIRE # the way you dialed it from your SIP soft-phone you would do : exten => _0.,1,Dial(ZAP/1/${EXTEN]) |
| 16:23.12 | Uatec | and i want to dial a different sip device if the phone is busy, i.e. in use |
| 16:23.18 | [TK]D-Fender | Bladerunner05: If you wanted to dial the ENTIRE # the way you dialed it from your SIP soft-phone you would do : exten => _0.,1,Dial(ZAP/1/${EXTEN}) |
| 16:23.22 | [TK]D-Fender | (fixed for typo) |
| 16:23.43 | [TK]D-Fender | Uatec: Look at the DIALSTATUS vriable |
| 16:23.47 | Bladerunner05 | that's right |
| 16:23.49 | [TK]D-Fender | Uatec: "show application dial" |
| 16:23.56 | [TK]D-Fender | Uatec: ..... you lazy ass! ;) |
| 16:24.17 | Sweeper | Daejeo1: http://www.initech.com/english/about/info.jsp <-- number at the bottom. remember to reference model RS-E, with the front panel connectors |
| 16:24.21 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:24.28 | [TK]D-Fender | Bladerunner05: ok, REALLY heading to lunch now... |
| 16:24.37 | Bladerunner05 | now I try |
| 16:25.16 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 16:25.58 | Bladerunner05 | <[TK]D-Fender> Ok now works, I do a 15 min pause.... |
| 16:26.20 | Bladerunner05 | <[TK]D-Fender> So now I have to understand why * don't answer the incoming calls |
| 16:26.49 | murdmath | Uatec: It is my understanding that your pri needs to be configured for national by your telco for you to be able to modify the caller id. |
| 16:27.34 | Daejeo1 | Sweeper: also I want to run few machines as soft-switches. which one do you recommend? |
| 16:28.14 | b4ka | anyone? |
| 16:28.31 | Uatec | murdmath, it is |
| 16:28.34 | b4ka | ive seen some messages asking the same o n the list, no response |
| 16:29.19 | Uatec | [TK]D-Fender, exten => s,n,GotoIf(${DIALEDSTATUS} = ANSWERED) ????!?!?!? |
| 16:29.26 | Uatec | that's an awful way of doing it |
| 16:32.43 | *** part/#asterisk eeos (n=eeos@86.53.50.16) |
| 16:33.11 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 16:33.24 | QbY | is there a way to have Goto return to the dialplan if the context and extension don't exist in the dialplan? |
| 16:33.35 | QbY | intsead of just dumping the call |
| 16:37.05 | Uatec | QbY, us the s extension... that's "anything else" |
| 16:37.58 | Uatec | well it's Start, but it works as anything else... |
| 16:37.59 | Uatec | i think |
| 16:39.15 | QbY | Uatec.. Here's my problem. We have DIDs that ring extensions.. When an extension calls out his full DID shows in the Caller ID. So if he calls another person in the office his full number shows up. If they go through their recent calls and see someone called and selects dial it will create a loop in the proxy because it sends it right back to Asterisk. What I woudl like to do, is to check the dialplan for the existence of that DID before s |
| 16:39.54 | QbY | I tried Goto(context_with_all_dids,${exten},1) but, if the DID doesn't exist it bombs, never making it to the Dial(SIP/${EXTEN}@PSTNGATEWAY) |
| 16:41.28 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 16:41.33 | [TK]D-Fender | Uatec: If its answered you won't even GET to the next priority... |
| 16:43.00 | [TK]D-Fender | QbY: ChanIsAvail(Local/123@context,j) |
| 16:43.12 | [TK]D-Fender | QbY: Chanisavail will tell you if its legit. |
| 16:43.33 | Zeeek | [TK]D-Fender how was lunch? |
| 16:43.53 | Uatec | [TK]D-Fender, what about when the receiver of the Dial command hangs up? |
| 16:43.54 | [TK]D-Fender | Zeeek: Easy to swallow.. unlike most of the stuff in here ;) |
| 16:44.41 | [TK]D-Fender | Uatec: Same deal. If you want * to deal with a hangup, thats what "exten => h," and Dial's "g" option are for. |
| 16:45.07 | Uatec | ahh..... |
| 16:45.23 | Uatec | what i actually want to do is ring Phone A, and if they don't answer, ring phone B |
| 16:45.27 | Uatec | wait |
| 16:45.27 | Uatec | shit |
| 16:45.34 | Uatec | what i actually want to do is ring Phone A, and if they don't answer go to voicemail |
| 16:45.41 | Uatec | but if phone A is busy, then ring Phone B |
| 16:45.52 | Uatec | and if Phone B is busy, or they don't answer, then go to the voicemail |
| 16:46.03 | [TK]D-Fender | Uatec: then you don't need to do ANYTHING |
| 16:46.17 | [TK]D-Fender | Uatec: Just dial them back-to-back, and 3rd priority to VM. |
| 16:46.19 | n0n4m3 | o_O |
| 16:46.24 | Aquavette | thats what I was gonna say... |
| 16:46.26 | [TK]D-Fender | Uatec: 3 puny lines of dialplan for the whole mess |
| 16:46.33 | Uatec | 3rd priority to VM? |
| 16:46.33 | n0n4m3 | Dial(sip/a,30); |
| 16:46.34 | Uatec | what? |
| 16:46.35 | n0n4m3 | Dial(sip/b,30); |
| 16:46.41 | n0n4m3 | Voice(...) |
| 16:46.50 | [TK]D-Fender | Voicemail(123@somevmcontext) |
| 16:46.57 | Uatec | n0n4m3, that will dial sip/a, then if they don't answer dial sip/b |
| 16:46.58 | n0n4m3 | yeah.. or something like that |
| 16:47.00 | Uatec | that's not what i want |
| 16:47.09 | [TK]D-Fender | Uatec: Thats what you just ASKED |
| 16:47.28 | [TK]D-Fender | Uatec: Ah, missed something |
| 16:47.31 | n0n4m3 | Uatec: what i actually want to do is ring Phone A, and if they don't answer go to voicemail |
| 16:47.32 | n0n4m3 | :D |
| 16:47.46 | [TK]D-Fender | Uatec: So you really don't want to ring A & B simultaneously? |
| 16:48.08 | Uatec | no, i don't |
| 16:48.09 | n0n4m3 | so ring phone b only if a is busy |
| 16:48.15 | Uatec | correct, n0n4m3 |
| 16:49.21 | *** join/#asterisk Innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
| 16:49.59 | [TK]D-Fender | Uatec: Ok, so dial the first, check the fail reason and jump to either VM or dialing #2 |
| 16:50.01 | [TK]D-Fender | (b) |
| 16:51.15 | Uatec | that sounds like lines and lines of horrible exten => s |
| 16:51.42 | [TK]D-Fender | Uatec: I never said what exten you were using, and its 2 extra lines. |
| 16:51.59 | AndrewGearhart | [TK]D-Fender: what about: keep two analog lines (1 fax, 1 voice both switched to the unlimited plan at 49 and 39/mo), run the additional lines (five) over VoIP to an ITSP. Think that's enough fault tolerance? |
| 16:52.13 | *** join/#asterisk hugohagogo (n=cleber@189.23.20.2) |
| 16:52.21 | [TK]D-Fender | AndrewGearhart: If your lines are unlimited how is paying for outside service (again billed per minute) CHEAPER or BETTER |
| 16:52.25 | Uatec | it's 2 extra lines per dialed phone |
| 16:52.39 | [TK]D-Fender | Uatec: Time to make macros |
| 16:53.01 | [TK]D-Fender | Uatec: And should be *0* extra after 3 lines of macro overhead. |
| 16:53.03 | AndrewGearhart | [TK]D-Fender: each additional line is $39/mo we usually have several folks on the phone at the same time... |
| 16:53.26 | Innatech | I'm trying to remember a recommendation for a favorite phone from a couple people from a few weeks back. Polycom or Aastra, I think, for heavy executive use. Sound familiar to anyone? It came up during a round of Snom-bashing. |
| 16:53.35 | *** join/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 16:53.43 | [TK]D-Fender | AndrewGearhart: Calc out your termination costs then based on the # of minutes required. |
| 16:53.53 | [TK]D-Fender | Innatech: Polycom > all |
| 16:53.57 | [TK]D-Fender | ~phones |
| 16:54.31 | jbot | phones is probably http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 16:54.34 | [TK]D-Fender | ~gs |
| 16:54.47 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 16:54.47 | ramindia | can some one tell me. how can i add meetme using mysql, realtime ? |
| 16:54.47 | Innatech | OK, that jibes with my memory. Thnx. |
| 16:54.48 | Uatec | macro? |
| 16:54.48 | Uatec | i have no idea how you would write a macro to do that |
| 16:54.48 | Uatec | unless |
| 16:54.59 | Innatech | I need to tattoo that URL backwards on my forehead. |
| 16:55.00 | [TK]D-Fender | Uatec: You should hire me to rebuild your entire setup for you. If you're going to be lazy, be EFFICIENT about it ;) |
| 16:55.04 | ramindia | any one done meetme with mysql realtime, i dont see any document for the same |
| 16:55.15 | [TK]D-Fender | Innatech: that URL is OLD and I should make my own at this point. |
| 16:55.29 | [TK]D-Fender | Innatech: Same as for documenting other things like SIP/NAT settings, etc. |
| 16:55.40 | Innatech | heh. I've been seeing it forever, you'd think I'd remember it by now w/o the reminder. |
| 16:55.57 | Corydon76-home | I wouldn't recommend Cisco phones. Ever. But I would recommend Grandstream phones for people who need a cheap phone. |
| 16:56.10 | Aquavette | I like my Cisco Phones |
| 16:56.40 | Innatech | Yeah, I've had decent luck with recent revision Snom's with revised firmware too. But, still, I wanted to refresh my memory so far as the conventional wisdom goes. |
| 16:56.49 | [TK]D-Fender | Corydon76-Linksys costs a tine amount more, and c'mon.... Polycom IP 320 = $95 USD. Thats only a little more than a GXP.... |
| 16:56.56 | Corydon76-home | They're expensive pieces of shit that should be confined to the dustbin until they a) come down in price, and b) Cisco gets their head out of their ass with respect to licensing the software image |
| 16:57.15 | Uatec | [TK]D-Fender, this is my job |
| 16:57.18 | Uatec | but i'm in the learning phase |
| 16:57.42 | [TK]D-Fender | Corydon76-home: Licensing needs to change, and the QUALITY of their SIP app too.... no presence, crappier call handling, etc... but then it might compete with CCM! ;) |
| 16:58.16 | Aquavette | see, I'm a CCNA, so i get access to the images for free. and I have the 7940's I own working great with Asteriks, no issues at all. |
| 16:58.21 | [TK]D-Fender | Uatec: Even better... Company money. My rates are very accessable and can save them (through you) lost of time to do a 1-shot clean-up and training. |
| 16:58.32 | Aquavette | Anyone that wants the image, I'll supply it too them |
| 16:58.34 | Corydon76-home | [TK]D-Fender: more grief from a vendor is not what I need. We deploy Polycom and SNOM almost exclusively. |
| 16:59.16 | Innatech | While we're beating the Grandstream horse, I take it their cheap FXO gateway is also to be avioded? |
| 16:59.23 | [TK]D-Fender | Corydon76-home: I agree, just saying what Cisco could do to bring themselves towards parity with Polycom. Polycom of course still KILLS everything else on price/quality ratio. |
| 16:59.32 | Innatech | perhaps in favor of audiocodes? |
| 16:59.49 | Corydon76-home | Innatech: why aren't you using Asterisk for FXO ports? |
| 16:59.54 | [TK]D-Fender | Innatech: Before talking FXO, how many ports, what kind of sue & location reltive to *? |
| 16:59.56 | Innatech | Rackmount server. |
| 17:00.08 | [TK]D-Fender | use* |
| 17:00.09 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 17:00.15 | Corydon76-home | If it deserves rackmount, it deserves a T1 PRI interface |
| 17:00.15 | Innatech | tk-d: just two, so we can make use of the lines carrying the DSLs. |
| 17:00.22 | *** part/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 17:00.45 | Innatech | Corydon76: Perhaps in an ideal world ;P |
| 17:00.45 | [TK]D-Fender | Innatech: What kind of call volume? |
| 17:00.50 | Corydon76-home | Using POTS with a rackmount interface is like towing a boat with a Yugo. |
| 17:00.58 | Uatec | [TK]D-Fender, it's my job to learn how and then we sell our services... |
| 17:01.02 | Uatec | i.e. my asterisk skills |
| 17:01.13 | [TK]D-Fender | Uatec: ... where.. WHERE?!?! ;) |
| 17:01.36 | Uatec | where? |
| 17:01.45 | Uatec | inglend |
| 17:01.46 | [TK]D-Fender | Uatec: If you're beating yourself up over a little piece of dialplan like this not even know the tools at your disposal you're a long way away... |
| 17:01.50 | Uatec | lol |
| 17:01.54 | Uatec | it's called learning |
| 17:02.00 | [TK]D-Fender | Uatec: I was talking about your SKILLS! |
| 17:02.09 | [TK]D-Fender | ]:D |
| 17:02.19 | Innatech | [TK]D : I suppose that would depend on the rules I put into place. It's a small office, but they tend to use their phones fairly heavily. They're going to have 6 channels over VOIP, and would also like to use the POTS lines they can't get rid of for local outbound calling. |
| 17:03.45 | Innatech | [TK]D: A couple extra ports wouldn't hurt, as they'd like to be able to sublet an office or two, but the main idea is not to waste the physical lines neccesitated by the DSL. |
| 17:04.14 | [TK]D-Fender | Innatech: Well on the cheap you might consider 2 x SPA -3102's or a TDM400P / A200. To save a few bucks you might try living with software EC as well. |
| 17:07.38 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 17:08.23 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 17:08.30 | Bladerunner05 | <[TK]D-Fender>: I'm back so... why my asterisk don't answer incoming call ? |
| 17:08.31 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-7-112.w81-251.abo.wanadoo.fr) |
| 17:08.51 | Innatech | [TK]D: the extra couple hundred dollars wouldn't be a deal breaker if I were to recommend something like the Audiocodes MP114 . Is there a significant benefit to be had? I've not had much experience in doing VOIP alongside traditional lines. |
| 17:08.53 | Zeeek | Bladerunner05 - long time no see! |
| 17:08.58 | [TK]D-Fender | Bladerunner05: Don't know yet. * shows NOTHING on the CLI for thie incoming ring? |
| 17:09.23 | Bladerunner05 | <[TK]D-Fender>: absolutly nothing |
| 17:09.27 | Innatech | [TK]D: I've dealt with Zaptel cards before, but I mostly work with plain VOIP. |
| 17:09.44 | [TK]D-Fender | Innatech: I prefer PCI usually for FXO. If you want quality, then A200d |
| 17:10.43 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-118-241.ph.ph.cox.net) |
| 17:10.48 | Zeeek | Bladerunner05 are the 4 LEDs lit on the card? |
| 17:11.17 | Innatech | [TK] D: Hmm. I suppose I could slap them into their own * box and tie it into the main one. |
| 17:11.23 | Bladerunner05 | Zeek: all on |
| 17:11.48 | Zeeek | zap show channel 2 |
| 17:12.25 | Bladerunner05 | http://www.pastebin.ca/548158 |
| 17:13.36 | Zeeek | same as mine except for File descriptor |
| 17:13.37 | Aquavette | Anyone tell me how to structure this, correctly. I want to ring an extension, IF NO ANSWER, I want to add something to the sipheader, and then have it call another extension. What I need to add to the SIP header is this: exten => s-NOANSWER,1,SIPAddHeader(Diversion: <tel:${ARG1}>\;reason=user-busy\;screen=no\;privacy=off) |
| 17:14.22 | Zeeek | Bladerunner05 maybe italy has funky states on telco lines? |
| 17:14.56 | Bladerunner05 | Zeek: don't think |
| 17:15.34 | [TK]D-Fender | Aquavette: Looks fine |
| 17:16.02 | Aquavette | see, whats that I thought, but I can't get the order working right |
| 17:16.13 | Aquavette | It will ring, but it will never add the header, and dial hte new extension |
| 17:16.37 | n0n4m3 | bruteforcing DOES pay up! |
| 17:18.54 | Zeeek | Bladerunner05 look at this: http://www.asteriskguru.com/archives/asterisk-users-tdm400-hungup-problem-vt91346.html |
| 17:19.28 | Zeeek | see what answeronpolarityswitch=yes is set in your zaptel |
| 17:19.33 | ^majik^ | anyone know if I can get the x64 version of the g729 codec through digium? |
| 17:19.41 | Qwell[] | ^majik^: for linux, yes |
| 17:20.00 | ^majik^ | Qwell[]: ok, thanks |
| 17:20.50 | [TK]D-Fender | Aquavette: Trun up sip debug, verbose 10 and pastebin the attempt from beginning to end |
| 17:21.04 | Aquavette | gotcha, thanks |
| 17:22.14 | *** join/#asterisk EduHard (n=edward2a@201.254.104.117) |
| 17:22.21 | EduHard | hello everyone |
| 17:22.27 | EduHard | Need little help urgent |
| 17:23.31 | EduHard | extensions cannot communicate |
| 17:23.36 | EduHard | any idea? thx |
| 17:23.43 | jkiff | So I'm reading that fax over IP is... not so hot. I see that this is mainly due to fax's low tolerance of jitter, packet loss, etc, so trying to fax via IAX, SIP, etc over the Internet would be pretty crappy. However, if I have a stable enough LAN and it's using g711, could I use SIP to get a fax as far as my Asterisk box where I then dump it onto the T1 just like any other outbound call? |
| 17:23.53 | jkiff | EduHard: You'll have to be more specific than that. |
| 17:24.19 | Bladerunner05 | Zeek I set answeronpolarityswitch=yes on zaptel.conf reload but nothig.... |
| 17:24.34 | Zeeek | it should be no |
| 17:24.56 | Zeeek | by the way, you may have to reload zaptel for changes like that |
| 17:25.06 | [TK]D-Fender | EduHard: You gave no details whatsoever. No, we have NO IDEA. |
| 17:25.09 | *** join/#asterisk AdamB0122 (n=adam@207.200.28.175) |
| 17:25.18 | AdamB0122 | Quick Question reguarding Asterisk |
| 17:25.24 | AdamB0122 | and its more of confirmation for me |
| 17:25.28 | Zeeek | [TK]D-Fender don't you have to reload the zaptel drivers when you change zaptel.conf ? |
| 17:25.29 | EduHard | fxotune -i 4 |
| 17:25.35 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 17:25.35 | EduHard | wait for fxotune to finish |
| 17:25.37 | [TK]D-Fender | Zeeek: Yup |
| 17:25.39 | EduHard | fxotune -s |
| 17:25.43 | AdamB0122 | If I have this PBX configured to use a T1 card |
| 17:25.45 | EduHard | asterisk start |
| 17:26.00 | AdamB0122 | I dont need to worry about it being behind a pix |
| 17:26.06 | AdamB0122 | because all the lines are analog |
| 17:26.07 | Bladerunner05 | •zeedo• do that but nothing.... |
| 17:26.14 | EduHard | then phones can't communucate with each other |
| 17:26.36 | [TK]D-Fender | EduHard: That makes NO sense. The facts have nothing to do with each other. |
| 17:26.44 | AdamB0122 | EduHard - Network to Network transactions are all done behind the PIx, so i dont need to worry about that either |
| 17:26.52 | [TK]D-Fender | EduHard: A hundred things could be wrong and we don't know what kind of phones you are using. |
| 17:27.09 | AdamB0122 | inside to inside the PIX doesn't montior anything, actually nothing even talks to the pix, it just goes directly to that device. |
| 17:27.22 | Bladerunner05 | Zeek: there is a gui that help to configure correctly all? |
| 17:27.24 | EduHard | I know, that's why I'm turning crazy.... maybe the phones... |
| 17:27.29 | Zeeek | [TK]D-Fender what is needed as a compliment to pastebin is a forum to fill out with the answers to all these questions |
| 17:28.11 | [TK]D-Fender | Zeeek: Not a bad idea. A form, not a forum. |
| 17:28.41 | Zeeek | Bladerunner05 yes: http://www.voip-info.org/wiki/view/Asterisk+consultants |
| 17:28.49 | Zeeek | s/forum/form/ |
| 17:28.53 | Zeeek | indeed |
| 17:29.08 | Zeeek | one like pastebin that issues an id number |
| 17:29.50 | Zeeek | someone could prolly modify a pb to show an initial document that had the questions |
| 17:30.00 | Aquavette | exten => s-NOANSWER,1,SIPAddHeader(Diversion: <tel:${ARG1}>\;reason=user-busy\;screen=no\;privacy=off) |
| 17:30.00 | Aquavette | exten => s-NOANSWER,2,Dial(SIP/222) |
| 17:30.02 | [TK]D-Fender | Zeeek: basically a live bug-tracker. |
| 17:30.03 | Aquavette | oops, sorry, |
| 17:30.16 | Zeeek | [TK]D-Fender more directed that that even |
| 17:30.18 | [TK]D-Fender | Aquavette: Pastebint he dialplan, its exectution and the SIP debug along-with |
| 17:30.31 | Aquavette | that what I was doing |
| 17:30.34 | Aquavette | just hit hte wrong window... |
| 17:30.35 | Aquavette | hehehe |
| 17:31.10 | Bladerunner05 | Zeek: I intend graphic interface |
| 17:31.15 | Zeeek | asterisk: version, nat?, phone 1:nat?,SIP/IAX/ZAP etc tec |
| 17:31.33 | Zeeek | Bladerunner05 not that I know of |
| 17:31.53 | Zeeek | but you should now call Digium because you own a Digium card |
| 17:32.01 | Zeeek | they will help you |
| 17:32.10 | [TK]D-Fender | Bladerunner05: Go read THE BOOK. Go use the WIKI and google up some guides. This is * 101 stuff... |
| 17:32.13 | [TK]D-Fender | ~book |
| 17:32.26 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:32.26 | [TK]D-Fender | ~wikis |
| 17:32.28 | jbot | extra, extra, read all about it, wikis is http://www.voip-info.org |
| 17:32.28 | [TK]D-Fender | ~docs |
| 17:32.30 | jbot | rumour has it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 17:32.31 | Zeeek | I think he has a not obvious problem |
| 17:32.46 | [TK]D-Fender | Zeeek: That doesn't forgive the first 10 that WERE ;) |
| 17:32.52 | Zeeek | hmmmmm |
| 17:33.02 | Zeeek | indeed, it is a test of moral fiber |
| 17:33.16 | [TK]D-Fender | Zeeek: I'm and not in the "posthumous forgiveness" business here ;) |
| 17:33.41 | Zeeek | you don't seem to be in any business at all with the time wasted here :) |
| 17:34.01 | Zeeek | must be waiting for the polycom to boot... |
| 17:34.15 | [TK]D-Fender | Zeeek: I get the occasional contract from here, and my clients all happy with my efforst. |
| 17:34.50 | Zeeek | contract? Tueur à gages? |
| 17:35.14 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 17:35.37 | Zeeek | FON wants to know why I haven't been sharing my internet connection |
| 17:36.10 | Aquavette | Fon router? |
| 17:36.24 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 17:36.26 | Zeeek | After their router brought down our DSL a few times with no one even near it and what with it being illegal, I decided to to put it back on the shelf |
| 17:36.41 | Zeeek | Yeah, they were shipped out free |
| 17:36.46 | Zeeek | at a conference I attended |
| 17:37.02 | Zeeek | Katty? |
| 17:37.08 | Aquavette | I went in and reloaded dd-wrt when it did that too mine |
| 17:37.13 | *** join/#asterisk bkunyiha (i=Billk@66-113-79-5.rev.ibsinc.com) |
| 17:37.32 | *** join/#asterisk catpants (n=catling@12-214-191-244.client.mchsi.com) |
| 17:37.39 | Zeeek | I do,n't have time to play with FON. IN fact someone less than a block away has one |
| 17:37.47 | Zeeek | so I'm superfluous |
| 17:37.55 | `pariah | i have twi X100P's in my asterisk server. i have both of these hooked up to POTS lines. when i ring either of the numbers, the asterisk console shows nothing. any help? |
| 17:37.59 | `pariah | *two |
| 17:37.59 | Aquavette | see, it made me mad, I erased the installed firmware and install something different |
| 17:38.07 | Aquavette | its not a glorifed access point |
| 17:38.19 | [TK]D-Fender | `pariah: Without seeing all realted configs, don't expect any... |
| 17:38.21 | Zeeek | a lot of ppeaopl have hacked the fos I hear |
| 17:38.25 | [TK]D-Fender | related* |
| 17:38.29 | Aquavette | its easy |
| 17:38.41 | Zeeek | anyone want one? |
| 17:38.41 | `pariah | [TK]D-Fender: ill show configs which ones zaptel.conf and zapata.conf? |
| 17:38.47 | Katty | pastebin.ca/548227 |
| 17:38.52 | Katty | i must be missing something little |
| 17:38.54 | [TK]D-Fender | `pariah: And related dialplan |
| 17:38.56 | *** join/#asterisk waptaxi (n=cahe@45.151-224-87.telenet.ru) |
| 17:39.07 | Zeeek | what about seeing if the cards are even recognized? |
| 17:39.22 | *** join/#asterisk casimir (n=casimir@rrcs-71-43-154-55.se.biz.rr.com) |
| 17:39.23 | `pariah | both cards are shown with zap show status |
| 17:39.28 | Zeeek | BIG ZAPTEL DAY today! Even I actually installed one |
| 17:39.29 | Innatech | So I just got back from looking at a clients new space on another floor in my building. They're inheriting oldish CAT5 wiring. Is it a terrible idea to try and run their phones and data over VLANS on that cabling? The servers routers switches modems etc will be tied together on gigabit, but it's the old cabling out to the offices and sec'y stalls. Should we bid bidding CAT6 installs now to avoid pain down the road? I'm thinking yes |
| 17:39.29 | Innatech | , am I overbuilding? |
| 17:39.40 | [TK]D-Fender | Katty: #INCLUDE => /etc/asterisk/zap-theritegroup <- no "#" in front |
| 17:39.52 | `pariah | [TK]D-Fender: i haven't gotten to a dialplan, im just trying to see if they will ring the console, then chose what to do when it rings |
| 17:39.54 | [TK]D-Fender | Katty: umm... oops |
| 17:40.07 | Katty | [TK]D-Fender: i don't believe you (= |
| 17:40.18 | [TK]D-Fender | Katty: that should be : include "fullpath" |
| 17:40.32 | [TK]D-Fender | Katty: You hybridized your syntax... |
| 17:40.33 | Aquavette | Cat6 is expensive, and the benefits it gives are not that much |
| 17:40.38 | Aquavette | unless your doing all gigabit |
| 17:40.52 | Katty | [TK]D-Fender: but all my other ones like #INCLUDE /etc/asterisk/sipupstairs works |
| 17:41.06 | [TK]D-Fender | Katty: Was unsure what you were trying to do, although I'm afraid to ask why you are including an entire FILE in that piece of IVR. This screams "not good" |
| 17:41.15 | Katty | [TK]D-Fender: don't ask. |
| 17:41.19 | Katty | [TK]D-Fender: it's horribly complicated. |
| 17:41.21 | [TK]D-Fender | Katty: no "=>" |
| 17:41.29 | Katty | [TK]D-Fender: and i don't want to explain i for 50th billion time. |
| 17:41.32 | [TK]D-Fender | Katty: You had me at "horribly" ;) |
| 17:42.03 | Katty | [TK]D-Fender: there are several different companies, with several different lines. |
| 17:42.15 | Katty | [TK]D-Fender: each [zap-context] needs to run through a different set of stuff. |
| 17:42.56 | casimir | Innatech, I tend to agree w/ Aquavette the benefits of cat6 to all the endpoints may not be worth the hassles |
| 17:42.56 | Katty | [TK]D-Fender: i don't want extensions conf to be huge and confusing, so i'm breaking it down into smaller bits. |
| 17:42.56 | `pariah | here are zaptel.conf and zapata.conf |
| 17:42.56 | `pariah | http://www.pastebin.ca/548240 |
| 17:42.56 | [TK]D-Fender | Katty: I'm clearly not seeing the dialplan being pumped out by that call.. its coming from somewhere else... |
| 17:42.58 | casimir | and cat6 won't be cat6 anymore after it goes around a corner on its way to a desk |
| 17:43.03 | Aquavette | yup |
| 17:43.04 | Zeeek | paria what does ztconfig tell you? |
| 17:43.11 | Katty | any other advice on my pastebin anyone? |
| 17:43.14 | [TK]D-Fender | `pariah: yOUR SIGNALLING IS REVERSED BETWEENT HE TWO... |
| 17:43.31 | [TK]D-Fender | `pariah: and you have not defined any CHANNELS in zapata.conf |
| 17:43.55 | [TK]D-Fender | Katty: I jsut told you that the CLI output doesn't match your dialplan in your pastebin... it won't do anyone any good :) |
| 17:43.59 | Trevor_b | :q |
| 17:44.02 | Trevor_b | doh |
| 17:44.07 | Katty | [TK]D-Fender: yes, yes you did. |
| 17:44.08 | [TK]D-Fender | Katty: You're gonna have to spill the beans to get out of this one! |
| 17:44.18 | Katty | [TK]D-Fender: but that doesn't help me any. |
| 17:44.24 | Katty | [TK]D-Fender: i already know that something is wrong, but i don't know what. |
| 17:44.36 | [TK]D-Fender | Katty: you aren't showing use the configs being affected.... |
| 17:45.09 | *** join/#asterisk frocos11292 (n=ask@firewall.vipvoz.com) |
| 17:45.14 | Uatec | [TK]D-Fender, ytou know what we were talking about?? |
| 17:45.14 | Uatec | exten => s,1,Dial(${ARG1},3,t) |
| 17:45.14 | Uatec | exten => s,2,GotoIf(${DIALSTATUS} = NOANSWER?3:5) |
| 17:45.15 | Uatec | exten => s,3,VoiceMail(u${ARG2}@default) |
| 17:45.15 | Uatec | exten => s,4,Hangup() |
| 17:45.15 | Uatec | exten => s,5,Noop() |
| 17:45.18 | Uatec | does that look right? |
| 17:45.25 | Zeeek | no, it should be in pb |
| 17:45.27 | Uatec | if ther eis no answer then go to the voicemail |
| 17:45.32 | Uatec | otherwise, don't do anything |
| 17:45.37 | [TK]D-Fender | Uatec: PASTEBIN !!!!!!!!!!!!!!!! |
| 17:45.39 | Uatec | then it would go on to the next macro for the next phone |
| 17:45.41 | Uatec | sorry :( |
| 17:45.53 | Katty | [TK]D-Fender: you sure are snippy today. |
| 17:46.05 | Katty | [TK]D-Fender: in fact, you have been for the last 3 or 4 days :P |
| 17:46.07 | [TK]D-Fender | Katty: The 1 D 10 T errors run WILD today. |
| 17:46.15 | flot | Where to take the program asnparser ? |
| 17:46.36 | Bladerunner05 | <[TK]D-Fender> : how can I resolve the problem for incoming calls? |
| 17:46.46 | [TK]D-Fender | Uatec: I'll make this simple : GO TRY IT AND SEE. |
| 17:46.50 | Zeeek | Bladerunner05 call Digium, they will help you |
| 17:46.56 | [TK]D-Fender | Bladerunner05: indeed. |
| 17:47.22 | Bladerunner05 | :-( |
| 17:47.26 | Zeeek | Bladerunner05 there are people there who can ssh into your machine and fix it |
| 17:47.54 | Uatec | [TK]D-Fender, I HAVE FUCKING TRIED IT |
| 17:48.05 | Uatec | DO YOU THINK I JUST SPEND 20 MINUTES LOOKING AT IT?!??!?! |
| 17:48.08 | Zeeek | o,, take it outside |
| 17:48.17 | [TK]D-Fender | Uatec: wel... what happened? |
| 17:48.23 | Uatec | nothing |
| 17:48.30 | Uatec | it just hung up |
| 17:48.36 | Uatec | apparently there was no voicemail '' |
| 17:48.40 | [TK]D-Fender | Uatec: If nothing happened then you ren't even executing your dialplan. Go fix it. |
| 17:48.41 | Uatec | but i said go to ${ARG2} |
| 17:49.02 | Uatec | it did the Voicemail route, then hungup |
| 17:49.38 | [TK]D-Fender | Uatec: Pastebin <- |
| 17:50.41 | Zeeek | ~oddtdm |
| 17:50.53 | Zeeek | ~oddzaptel |
| 17:50.57 | Zeeek | ~oddtdm400 |
| 17:51.58 | Katty | Uatec: you're doing great, btw (= |
| 17:52.00 | Zeeek | hey! |
| 17:52.33 | Bladerunner05 | I compile zaptel but don't find zttool..... |
| 17:52.37 | Katty | Zeeek: you're doing great too, keep it up. |
| 17:52.46 | Zeeek | Incidentally, folks.... |
| 17:53.13 | Zeeek | If you're tired of answering questions, tell me what's wrong with this headline?: http://www.wsoctv.com/mlb/13222064/detail.html |
| 17:53.38 | Aquavette | bad play on words |
| 17:53.54 | Zeeek | seriously wrong headline |
| 17:54.20 | Uatec | it doesn't appear to be differnatiating between NOANSWER and BUSY |
| 17:54.23 | Uatec | i know that the phone is busy |
| 17:54.29 | Uatec | it's in a call |
| 17:54.37 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 17:54.40 | Uatec | but it's ${DIALSTATUS} is still NOANSWER |
| 17:54.51 | Zeeek | Bladerunner05 http://www.voip-info.org/wiki/view/Asterisk+zttool |
| 17:55.39 | Qwell[] | Uatec: With SIP, just because a phone is in use, doesn't mean that it's busy |
| 17:55.42 | [TK]D-Fender | Uatec: It rang on phone A, didn't it? |
| 17:55.55 | Qwell[] | unless you set the busy limit to 1 or something |
| 17:56.02 | [TK]D-Fender | Uatec: If a phone does not REJECT a call while on the phone, then it is accepted |
| 17:56.12 | [TK]D-Fender | Uatec: That means call-waiting, etc. |
| 17:56.13 | Qwell[] | well, it can still REJECT and be NOANSWER :) |
| 17:56.21 | Zeeek | in all fairness to aspiring asterisk gurus everywhere, sometimes looking at the same "misteak" for hours doesn't help |
| 17:56.25 | [TK]D-Fender | Uatec: To take this into accout : "show application chanisavail" |
| 17:56.41 | Zeeek | ah, positive reinforcement |
| 17:57.07 | [TK]D-Fender | Qwell[]: Can it? Didn't think any response short of "ringing/trying" could do that... |
| 17:57.19 | *** join/#asterisk chodorenko (n=chodoren@etm005.nl.ded.neolocation.net) |
| 17:57.25 | *** join/#asterisk ikey (i=ikey@220.226.13.56) |
| 17:57.29 | [TK]D-Fender | Zeeek: Steak heads right where it should.... MY GRILL! |
| 17:57.30 | Qwell[] | well, my polycom has a button I can hit to ignore a call.. that sends a reject, doesn't it? |
| 17:57.37 | Qwell[] | it stops ringing when I hit it :D |
| 17:57.49 | Zeeek | In March of 2004, I asked the following question on the FWD fourm: "what exactly is a dialplan?" |
| 17:57.58 | [TK]D-Fender | Qwell[]: Applicable inter-office skills! |
| 17:58.03 | Zeeek | and this was regarding a SIP phone! |
| 17:59.22 | [TK]D-Fender | Qwell[]: And Polycom's "reject" won't trigger a "NOANSWER". |
| 17:59.32 | [TK]D-Fender | Qwell[]: Only a Dial timeout should do that. |
| 18:00.02 | [TK]D-Fender | Qwell[]: You can specify a response to translate to "congestion" and 1 other thing IIRC |
| 18:00.54 | *** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
| 18:00.55 | Qwell[] | oh |
| 18:00.59 | Zeeek | I never notivced the reject button on the poly |
| 18:01.03 | Qwell[] | Well, what do I know? I'm just a user :P |
| 18:01.11 | Zeeek | so anxious to talk to someone |
| 18:01.25 | Zeeek | DND, yes |
| 18:01.29 | Qwell[] | Zeeek: I've only ever used the reject button with one certain person calling |
| 18:01.46 | Qwell[] | well, and our receptionist - but it was that same person calling through her to reach me |
| 18:01.52 | Zeeek | heh |
| 18:02.11 | Qwell[] | I've never had a call with him shorter than 30 minutes... |
| 18:02.16 | Qwell[] | umm, him/her |
| 18:02.26 | Zeeek | o.....k..... |
| 18:02.29 | Qwell[] | :p |
| 18:02.42 | *** join/#asterisk EduHard (n=edward2a@201.254.104.117) |
| 18:02.42 | EduHard | hello again |
| 18:02.56 | EduHard | Now I have a more detailed problem |
| 18:03.10 | Zeeek | which merits a detauiled question |
| 18:04.13 | EduHard | call from an extension (4003) to another (4000) or any other in the network, actually 4, and it gives me no answer and the destination extension does'n even rings |
| 18:04.44 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 18:04.45 | EduHard | sorry for typing errors, this keyboard is almost for recycle |
| 18:04.48 | *** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
| 18:05.50 | flot | Where to take the asnparser ? |
| 18:06.25 | `pariah | now one more question. when i dial the zap cards, it thows an error after a couple rings. this error ca/548296 |
| 18:06.28 | `pariah | ca/548296 |
| 18:06.47 | `pariah | shit http://www.pastebin.ca/548296 |
| 18:08.01 | *** join/#asterisk ikey1 (i=ikey@220.226.13.56) |
| 18:08.51 | Uatec | dammit |
| 18:08.57 | jkiff | Just a heads up, the link in jbot's entry for t38 is 404. |
| 18:09.01 | Uatec | stupid callwaiting was messing me about |
| 18:09.35 | n0n4m3 | crappy |
| 18:09.46 | n0n4m3 | i have problems with belco bcip-300 |
| 18:09.55 | n0n4m3 | it just doesn't want to register to asterisk 1.4.4 |
| 18:09.57 | coppice | ~t38 |
| 18:09.59 | jbot | methinks t38 is see http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf for a decent overview of how it all works, no, it's not ready yet, we'll let you know. a really lousy spec. a lightweight fighter, also known as the Talon |
| 18:10.29 | [TK]D-Fender | `pariah: You didn't specify what context to send incoming calls to. |
| 18:10.43 | Innatech | casimir, aqauvatte: thanks for the input (I was on the phone). |
| 18:10.45 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 18:10.50 | coppice | brooktrout is no more. it has ceased to be. |
| 18:11.08 | [TK]D-Fender | EduHard: pastebin your CLI output |
| 18:11.16 | *** join/#asterisk karlhaines (n=karl@unaffiliated/karlhaines) |
| 18:11.25 | karlhaines | how can i make music on hold louder ? |
| 18:11.33 | karlhaines | it's really quiet |
| 18:11.35 | [TK]D-Fender | karlhaines: you shouldn't ahve to. |
| 18:11.52 | [TK]D-Fender | karlhaines: if these are your own recordings then use SOX to normalize them |
| 18:12.54 | karlhaines | its an mp3 |
| 18:12.57 | karlhaines | maybe thats why |
| 18:13.25 | [TK]D-Fender | karlhaines: no. |
| 18:13.57 | `pariah | [TK]D-Fender: is that part of zapata.conf? |
| 18:13.59 | [TK]D-Fender | karlhaines: Most recordings made for MoH are pre-normalized to some sort of standard |
| 18:14.04 | [TK]D-Fender | `pariah: Yes |
| 18:14.57 | IOscanner | Anyone use the presence feature from Eyebeam Soft phone asterisk? |
| 18:15.36 | [TK]D-Fender | IOscanner: I have. Seems fine |
| 18:16.31 | `pariah | [TK]D-Fender: say if i wanted whoever calls zap-1 to be forwarded to SIP 101@default how would i go about doing that? you know of any examples? |
| 18:16.40 | *** join/#asterisk Ironhand (i=arjen@mjolnir.xyx.nl) |
| 18:17.11 | [TK]D-Fender | `pariah: this is all DIALPLAN. You have to specify the dialplan context to use in your zapata.conf and then tell it what to do in there. |
| 18:17.21 | [TK]D-Fender | `pariah: ..... |
| 18:17.22 | [TK]D-Fender | ~book |
| 18:17.24 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:17.59 | [TK]D-Fender | `pariah: there is no such thing as "SIP 101@default" |
| 18:18.22 | `pariah | [TK]D-Fender: i have which context i want specified in zapata.conf |
| 18:18.30 | [TK]D-Fender | `pariah: SIP devices don't have a 2nd context like heirarchy |
| 18:18.46 | [TK]D-Fender | `pariah: Your CLI output begs to differ. |
| 18:19.19 | [TK]D-Fender | <PROTECTED> |
| 18:19.21 | [TK]D-Fender | <PROTECTED> |
| 18:19.22 | [TK]D-Fender | [Jun 7 06:02:21] WARNING[7937]: pbx.c:2450 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler |
| 18:19.39 | [TK]D-Fender | `pariah: No. You. Don't. |
| 18:19.50 | [TK]D-Fender | `pariah: this should be a hint. |
| 18:20.17 | `pariah | http://www.pastebin.ca/548346 zapata.conf |
| 18:20.57 | [TK]D-Fender | `pariah: the error above clearly tells you what you are missing. It is NOT lying. |
| 18:21.17 | `pariah | i thought the context=default would specify the context defualt.... |
| 18:21.32 | `pariah | which is in my dialplan |
| 18:21.48 | [TK]D-Fender | `pariah: pastebin it then. |
| 18:22.06 | [TK]D-Fender | `pariah: and PAY ATTENTION TO : == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's' |
| 18:22.07 | `pariah | which file? |
| 18:22.18 | [TK]D-Fender | `pariah: extensions.conf obviously. its complainging about your dialplan |
| 18:23.19 | ikey1 | hi |
| 18:23.26 | `pariah | i have a really simple extensions.conf with only 1 sip extension for testing purposes |
| 18:23.57 | `pariah | http://www.pastebin.ca/548362 |
| 18:26.31 | Uatec | how can i set my pound key linux distro to use a time server? |
| 18:27.28 | EduHard | call from an extension (4003) to another (4000) or any other in the network, actually 4, and it gives me no answer and the destination extension doesn't even rings |
| 18:27.31 | EduHard | any idea? |
| 18:27.36 | [TK]D-Fender | `pariah: You sure as hell don't have an "s" exten in there just like it said.... |
| 18:27.41 | [TK]D-Fender | ~book |
| 18:27.47 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:27.47 | cpm | install ntpd |
| 18:27.48 | [TK]D-Fender | ^^^^^^^^ |
| 18:28.25 | [TK]D-Fender | EduHard: that is NO information at all. I asked you for a pasebin of the CLI output already.... |
| 18:29.16 | `pariah | [TK]D-Fender: I dont want an s extension, i want the zap cards to call a SIP extension after they have been dialed. |
| 18:29.44 | [TK]D-Fender | `pariah: Incoming calls land on "s". PERIOD. you make it do what you want FROM THERE. |
| 18:31.45 | IOscanner | Fender: What did you do to set it up with asterisk? |
| 18:32.37 | flot | Where to take the "asnparser" ? |
| 18:32.44 | [TK]D-Fender | IOscanner: You enter in 3 fields. user, pass & IP. |
| 18:32.47 | [TK]D-Fender | IOscanner: Thats it. |
| 18:32.55 | [TK]D-Fender | IOscanner: hints "just work". |
| 18:33.04 | karlhaines | exit |
| 18:33.07 | EduHard | sorry, what's a pasebin? |
| 18:33.10 | [TK]D-Fender | ~pb |
| 18:33.14 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 18:33.16 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 18:34.31 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 18:35.01 | n0n4m3 | anyone of you guys uses belco bcip-300 sip phone? seems like mine doesn't like asterisk :( |
| 18:35.55 | n0n4m3 | it tries to connect |
| 18:35.56 | EduHard | oh sorry, gimme a minute... i feel dumb ;) |
| 18:36.05 | n0n4m3 | but asterisk keeps sending 401 :/ |
| 18:37.25 | IOscanner | Fender: the precense feature so I can see who is busy and available. |
| 18:37.41 | Katty | [TK]D-Fender: so about that problem i was having. |
| 18:37.54 | Katty | [TK]D-Fender: with the include it doesn't work, but if i move the contents of the include to the main file, it works. |
| 18:38.01 | Katty | [TK]D-Fender: yet, all the other includes work fine. |
| 18:38.07 | [TK]D-Fender | Katty: yOU INCLUDE FORMAT WAS WRONG. |
| 18:38.15 | Katty | [TK]D-Fender: it's the same as the rest of them. |
| 18:38.15 | [TK]D-Fender | darn caps |
| 18:38.33 | [TK]D-Fender | IOscanner: nothing to configure in eyebeam. |
| 18:39.24 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 18:39.43 | IOscanner | Okay then I am going to write a service to handle it. I also have to handle status from cisco phones too. |
| 18:39.44 | `pariah | [TK]D-Fender: i am confused how the extension s will know which zap channel is calling, do you know of any examples i can look at to maybe figure it out? |
| 18:40.02 | IOscanner | Is there a way to write sip debug messages to a different file? |
| 18:40.24 | Mercestes | `pariah, Why would your extensions care which zap channel is calling? |
| 18:40.40 | IOscanner | Unless I can open a socket to Asterisk and be able to monitor sip debug messages in real-time. |
| 18:40.51 | n0n4m3 | any ideas? |
| 18:40.51 | n0n4m3 | http://rula.net/32 |
| 18:41.13 | *** join/#asterisk n3glv (n=n3glv@c-71-60-125-243.hsd1.pa.comcast.net) |
| 18:41.13 | IOscanner | I don't need the logs, I just need to montior the status of a few things. |
| 18:41.17 | n3glv | hi guys |
| 18:41.27 | n3glv | anyone have any insight |
| 18:41.29 | n3glv | [DB Error: extension not found] ** mysql://asteriskuser:eLaStIx.asteriskuser.2oo7@localhost/asterisk |
| 18:41.58 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 18:42.05 | `pariah | Mercestes: i want say, zap chanel 1 when called to dial sip ext. 101 and when zap chan 2 is dialed to call sip ext. 102 |
| 18:42.24 | n0n4m3 | argh |
| 18:42.25 | Strom_M | `pariah: so put the channels in separate contexts |
| 18:42.27 | n0n4m3 | forgot to reload |
| 18:42.27 | n0n4m3 | :D |
| 18:42.54 | IOscanner | Is there a way to open a CLI from a program that will pass the data into a program? |
| 18:43.26 | n3glv | can you set the extension for DID=zap channel? I never tried |
| 18:43.49 | n3glv | IOscanner,I suspect that's asterisk api stuff |
| 18:44.23 | IOscanner | can the API do sip debug streams? |
| 18:44.31 | n3glv | dunno |
| 18:44.52 | n3glv | but I think they end up in /var/log/asterisk/full |
| 18:45.00 | n3glv | perhaps u could parse that |
| 18:45.17 | IOscanner | yes, but if you reboot someone has to login and enable sip debug again |
| 18:45.21 | *** part/#asterisk EduHard (n=edward2a@201.254.104.117) |
| 18:45.57 | n3glv | I'd like to get elastix runnin |
| 18:45.58 | n3glv | g |
| 18:46.04 | n3glv | get a sql error |
| 18:46.22 | `pariah | Strom_M: ok, that sounds like a good idea but how do i specify each context in zapata.conf? i only see it being specified for both zap channels? |
| 18:46.30 | Katty | [TK]D-Fender: so if my include syntax is wrong, what should it be? |
| 18:46.47 | *** part/#asterisk n3glv (n=n3glv@c-71-60-125-243.hsd1.pa.comcast.net) |
| 18:46.50 | Strom_M | `pariah: you specify it for each channel separately |
| 18:47.12 | Mercestes | `pariah, do you have a good reason for wanting it that way? |
| 18:47.21 | Mercestes | `pariah, Any reason why all of your lines can't simply go to all of the phones? |
| 18:47.56 | Aquavette | is there any plan in the future for Asteriks to do SIP over TCP |
| 18:47.56 | Aquavette | ? |
| 18:48.13 | Mercestes | Aquavette, as opposed to what? |
| 18:48.23 | Aquavette | sip over udp? |
| 18:48.29 | Mercestes | Sip doesn't go over udp |
| 18:48.33 | Mercestes | RTP goes over udp |
| 18:49.16 | Aquavette | but RTP can go over TCP |
| 18:49.22 | Aquavette | if the server supports it |
| 18:49.37 | Mercestes | Not that I'm aware of |
| 18:50.03 | Aquavette | Like Cisco Call Manager can do SIP over TCP and RTP over TCP |
| 18:50.17 | Aquavette | Asterisk does Sip over TCP, and then RTP over UDP |
| 18:50.25 | Mercestes | Ok. |
| 18:50.40 | `pariah | Mercestes: yes. what we are doing in is seting up SIP phones at each of our houses. so if the secretarty at work gets a call she can transfer to ext. 202 which is a port on our KSU at work that is hooked up to an X100P in the box. so if she transfers the call we got on POTS to 202 on our KSU it will go to the X100P and then to my SIP phone at my house |
| 18:50.47 | Mercestes | Aquavette, you are correct. Asterisk does. |
| 18:50.58 | Mercestes | Aquavette, So what want to know is, will we ever move our RTP stream to TCP? |
| 18:51.23 | Mercestes | `pariah, Define KSU |
| 18:51.27 | Aquavette | yrs |
| 18:51.29 | Aquavette | *yes |
| 18:51.34 | Aquavette | to increase interoperability |
| 18:51.39 | Mercestes | Aquavette, That's an #asterisk-dev question. |
| 18:51.44 | Mercestes | Or a feature request. |
| 18:52.09 | Aquavette | fair enough |
| 18:52.09 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:53.14 | `pariah | Mercestes: Panasonic Key Station unit. not really a PBX but more of a thing that decodes DTMF tones for a business telephony environment. |
| 18:53.19 | Mercestes | `pariah, So basically you are trying to implement the "lines" implementation the KSU is expecting? |
| 18:53.37 | [[blah]asfd | i am trying to understand how to use chanisavail. Can anyone check my work and see if I have done this correctly? It is not working. is my n+101, right? http://pastebin.ca/548453 |
| 18:54.11 | Mercestes | `pariah, I am pretty sure you can specify a channel, like Dial(zap/1-${EXTEN}/${EXTEN}) if you want to statically map channels to endpoints.. |
| 18:54.32 | Mercestes | `pariah, or like Strom set, set up specific contexts in zapata.conf and define each of your channels individually |
| 18:54.38 | Mercestes | `pariah, or both. |
| 18:55.10 | `Sean | Mercestes know of a provider that has instant setup |
| 18:55.14 | `Sean | tollfreegateway isn't working for me |
| 18:55.22 | `Sean | and well asterlink doesn't support tollfree |
| 18:55.25 | Mercestes | `Sean, Try teliax.com |
| 18:55.34 | `pariah | http://www.pastebin.ca/548458 zapata.conf |
| 18:55.58 | Mercestes | `pariah, yea, like that. |
| 18:56.11 | Mercestes | `pariah, I'm pretty sur eyou can also dial Zap/1-1 and 1-2 and 1-3 etc. too. |
| 18:56.19 | `pariah | Mercestes: but each of those are going to the context defualt |
| 18:56.21 | Mercestes | Not supposed to but I believe you can |
| 18:56.36 | Mercestes | `pariah, so add a context line, context=zap1 context=zap2 etc. |
| 18:56.38 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 18:57.19 | *** join/#asterisk TUplink_ (n=TUplink@c-24-126-39-22.hsd1.wv.comcast.net) |
| 18:57.44 | TUplink_ | is MailboxExists() suposed to jump to the next priorty if the vmbox does not exist? |
| 18:58.04 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 18:58.19 | _VoiceMeUp_COM | anyone know the max length of a dial ( ARG$ ? |
| 18:58.26 | _VoiceMeUp_COM | the arg can be up to xxx ? |
| 18:58.32 | _VoiceMeUp_COM | to know how many peers i can sim ring |
| 18:59.00 | Mercestes | Wild Guess: 65535 characters. |
| 18:59.20 | *** join/#asterisk komradebob (n=komradeb@164.55.254.106) |
| 18:59.21 | *** join/#asterisk ^majik^ (n=kvirc@68-187-20-73.static.uncty.tn.ken-tennwireless.com) |
| 18:59.28 | Mercestes | whatever limit it's int, int16, int32, float, double or string varaible type allows for. |
| 18:59.44 | ^majik^ | Qwell[], you work for digium, right? |
| 18:59.58 | TUplink_ | <PROTECTED> |
| 18:59.59 | _VoiceMeUp_COM | no idea hey ;) |
| 19:02.12 | `pariah | Mercestes: so say i put context=default1 for the first device |
| 19:02.37 | _VoiceMeUp_COM | mercetes yeah but where can i find that limit ? |
| 19:02.47 | `pariah | Mercestes: when i call that card i get == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's' |
| 19:02.50 | Corydon76-home | TUplink_: yes. It sets VMBOXEXISTSSTATUS with the result |
| 19:02.52 | Mercestes | works |
| 19:03.11 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:03.20 | TUplink_ | exten => _20XXX,106,MailboxExists(${EXTEN},j) |
| 19:03.20 | TUplink_ | exetn => _20XXX,107,Background(extension) but it never goes to background |
| 19:03.26 | Mercestes | `pariah, should work. I'm pretty sure you can match the DNID that's coming at you too if the Telco sends it. |
| 19:03.32 | `pariah | why would it still be looking in context defaukt? |
| 19:03.42 | Mercestes | `pariah, Did you reload your zaptel? |
| 19:03.53 | `pariah | i restarted asterisk completely |
| 19:03.58 | TUplink_ | <PROTECTED> |
| 19:03.58 | TUplink_ | <PROTECTED> |
| 19:04.18 | [TK]D-Fender | `pariah: Go set the context for each of your channels in zapata.conf and make sure you make that context in extensions.conf and create an "s" exte for the calls to land on. |
| 19:04.57 | _VoiceMeUp_COM | would tha code be in chan_sip.c ? |
| 19:05.06 | _VoiceMeUp_COM | the max length of the dial peers i can ring simultaneously |
| 19:05.42 | TUplink_ | you could just use a queue |
| 19:06.14 | _VoiceMeUp_COM | cant |
| 19:06.14 | _VoiceMeUp_COM | lol |
| 19:06.14 | TUplink_ | why not? |
| 19:06.14 | _VoiceMeUp_COM | caus |
| 19:06.14 | _VoiceMeUp_COM | its from our frontend.. to let clients choose multiple peer to ring for an exten |
| 19:06.20 | TUplink_ | well how else can you make a group? |
| 19:06.21 | *** join/#asterisk bakermd (n=bakermd@204.10.20.30) |
| 19:06.21 | _VoiceMeUp_COM | so we want to knwo a safe value for max # of peers |
| 19:06.33 | bakermd | Hey all, Caller ID is showing in the CDR, but ${CALLERIDNUM} is not working - any ideas? (returns null) |
| 19:06.35 | _VoiceMeUp_COM | DIAL(SIP/1&sip/2&etc.. |
| 19:06.42 | `pariah | would this be correct for the incoming call to dial a sip ext. after it rings and gets picked up? exten => s,3,Dial(SIP/101,20) |
| 19:06.59 | Strom_M | bakermd: CALLERIDNUM was deprecated in 1.2 and doesnt exist in 1.4 |
| 19:07.10 | Strom_M | use CALLERID() and read upgrade.txt |
| 19:07.10 | bakermd | Strom_M: Aah - whats the replacement? |
| 19:07.14 | bakermd | THANKS!!! |
| 19:07.15 | [TK]D-Fender | `pariah: You need priority 1 & 2 as well, but sure, that would cause it to ring a sip device jsut fine. |
| 19:08.07 | _VoiceMeUp_COM | or waht is the function called ? |
| 19:08.34 | TUplink_ | SET(CALLERID(number)= |
| 19:08.55 | TUplink_ | orSET(CALLERID(name)= |
| 19:09.03 | *** part/#asterisk QbY (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 19:09.08 | [TK]D-Fender | _VoiceMeUp_COM: I'm not aware of a "safe number" bun you WILL hit dialplan STRING LENGTH limits long before that becomes a worry. |
| 19:09.30 | _VoiceMeUp_COM | yeah ok |
| 19:09.31 | komradebob | is there some reason a stock-out-of-the-box install would not be recording cdrs in CSV? |
| 19:09.42 | _VoiceMeUp_COM | ok so 10 -15 ? |
| 19:10.17 | `pariah | http://www.pastebin.ca/548496 extensions.conf |
| 19:10.44 | [TK]D-Fender | `pariah: Looks fine |
| 19:11.19 | `pariah | http://www.pastebin.ca/548503 zapata.conf |
| 19:11.23 | [TK]D-Fender | `pariah: But you don't need exten => 101 under [default1] at all, and after your dial you should do a "hangup" or something else. |
| 19:11.28 | bakermd | Strom_M: I see that you can use set(callerid(ani)) - but I am having trouble finding how to pull the callerid |
| 19:12.03 | _VoiceMeUp_COM | ast_get_extension_app_data |
| 19:12.07 | TUplink_ | ${CALLERID(name)} |
| 19:12.11 | *** join/#asterisk [hC] (n=hardcore@190.10.13.145) |
| 19:12.13 | `pariah | [TK]D-Fender: that is the only place i have the sip extensions defined |
| 19:12.14 | TUplink_ | or number |
| 19:12.17 | [TK]D-Fender | `pariah: but i would rename those contexts tos omething like [from-zap-line1] and so on... |
| 19:12.49 | `pariah | [TK]D-Fender: i'm just trying to get it functional ATM, i will be renaming them after i can get it to sort of work |
| 19:12.54 | [TK]D-Fender | `pariah: You are mixing things up. those contexts are do your Zaptel channels to send contexts to and your callers don't GET aA MENU. |
| 19:13.10 | [TK]D-Fender | `pariah: thy don't "dial 100" with thier phone... * jsut dials it automatically. |
| 19:13.35 | Innatech | [TK} D : Do you have a preferred vendor for Polycom? Like one who might give a bit of a break on the purchase of ~10 IP-501s and a few of their cheaper siblings (like the 320)? |
| 19:13.42 | [TK]D-Fender | `pariah: those numbered extens are so your SIP phones are able to call EACH OTHER. These should all be in a DIFFERNT context. |
| 19:14.10 | [TK]D-Fender | Innatech: in order : www.telephonydepot.com , www.atacomm.com , www.voipsupply.com |
| 19:14.14 | Innatech | thanks. |
| 19:15.19 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 19:15.26 | *** part/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 19:15.37 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 19:17.05 | [TK]D-Fender | Innatech: np |
| 19:17.08 | Innatech | for anyone who's worked with the polycomms much: 601 vs. 501 -- worth the $100 / per ? |
| 19:17.33 | [TK]D-Fender | Innatech: 501 is a rare case to justify. 6012 is for receptionists. |
| 19:17.43 | [TK]D-Fender | Innatech: Got PoE? |
| 19:18.05 | _VoiceMeUp_COM | 6012? |
| 19:18.12 | [TK]D-Fender | 601 |
| 19:18.28 | Innatech | [TK]D : Hopefully. If it isn't present at install, we'll wind up putting in a PoE switch within a few months. People hate the wall warts, and they'll want the phones on a central UPS. |
| 19:18.53 | [TK]D-Fender | Innatech: IP430's for everybody else then. |
| 19:19.15 | Innatech | [TK] D : BRB, gonna eyeball the 430's specs. |
| 19:20.23 | [TK]D-Fender | Innatech: 2-line, 5 calls/line-key, PoE + Comes with Brick. 4 soft-keys. Speakerphone. |
| 19:20.26 | _VoiceMeUp_COM | ah |
| 19:20.36 | bakermd | In asterisk 1.4 can you still set the first entry in a dialplan to be 's' ? |
| 19:20.57 | [TK]D-Fender | Innatech: Edge on the 320/330 for having the brick included and the extra soft-key. |
| 19:21.00 | *** join/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
| 19:21.11 | [TK]D-Fender | Innatech: if you don't need passthrough, ip 320 may be a consideration. |
| 19:21.25 | [TK]D-Fender | Innatech: slightly tougher call. |
| 19:21.28 | Innatech | [TK] D: so its included brick & extra soft key on the 430 vs. an extra line appearance on the 501 |
| 19:21.49 | Innatech | Or are there other features the 501 has the 430 lacks? (or vice-versa) |
| 19:21.58 | [TK]D-Fender | Innatech: IP501 is not a standard "work" phone. Its more of a "I want a cool phone for my home desk and it'll be my primary w/o PoE" |
| 19:22.47 | [TK]D-Fender | Innatech: 501 costs $170 and needs a special cable for PoE upping the cost further. Bulky too. 501 has a bigger screen (good for microbrowser), but seriously more than astandard office user needs. |
| 19:22.47 | Innatech | [TK] D : The look-and-feel issue is actually significant. These users are giving up Nortel/Meridian desk sets. They want similar functionality and solid feel. |
| 19:22.56 | ^majik^ | anyone from digium not afk/busy? :) |
| 19:23.05 | [TK]D-Fender | Innatech: All of the polycom's are good quality & feel. |
| 19:23.48 | [TK]D-Fender | Innatech: I migrated my company from a Norster 8x24 2 years ago |
| 19:23.51 | Innatech | [TK] D : OK, good stuff to know. |
| 19:24.01 | *** part/#asterisk TUplink_ (n=TUplink@c-24-126-39-22.hsd1.wv.comcast.net) |
| 19:24.03 | Innatech | [TK] D : Yeah, there's a part of me that will be sad to see the beast go. |
| 19:24.14 | [TK]D-Fender | Innatech: Its also what I use at home and suggest to my customers. |
| 19:24.38 | Innatech | [TK] D : Those meridian handsets feel like a natural appendage after all these years ;) |
| 19:24.45 | [TK]D-Fender | Innatech: No..go CELEBRATE. Norhell can eat a bag of @^%# in my books... |
| 19:24.46 | Katty | [TK]D-Fender: i'd be willing to bet that include didn't like the dash. |
| 19:24.55 | [TK]D-Fender | Innatech: I used mine for 10 years... |
| 19:25.12 | [TK]D-Fender | Katty: Extraneous chars = bad |
| 19:25.20 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 19:25.34 | Innatech | TK: This one has been up and running since the early 90s. |
| 19:25.45 | Katty | [TK]D-Fender: yep, it was the dash. |
| 19:25.54 | [TK]D-Fender | Innatech: Same. 1994-2005 RIP <- |
| 19:26.19 | [TK]D-Fender | or was taht 92... hrm |
| 19:26.29 | Innatech | TK: Ours was '92, I think. |
| 19:26.33 | *** join/#asterisk umay (n=chris@71-208-191-53.hlrn.qwest.net) |
| 19:28.04 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
| 19:28.13 | [TK]D-Fender | Innatech: I was on DR5 |
| 19:28.14 | bakermd | Can anyone help me with this pls: Channel 'SIP/5080-0a0fd158' sent into invalid extension 's' in context 'bridge', but no invalid handler |
| 19:28.39 | [TK]D-Fender | Innatech: With a NAM2 VM unit and a shitty 4 seat ACD |
| 19:28.43 | Innatech | TK: In all fairness, and even if its a nightmare to support/maintain/upgrade, it did a good job. If I hadn't been given the chance to tinker with it growing up, I wouldn't have gotten into telecomms quite as much as early. |
| 19:28.57 | [TK]D-Fender | bakermd: It tells you EXACTLY whats missing in your dialplan. Do the math |
| 19:29.14 | [TK]D-Fender | Innatech: Same here... but i was ready to replace it YEARS prior. |
| 19:29.19 | Innatech | TK: The system here is a mishmash of meridian and nortel modules, I don't even know whats in the cabinet these days. |
| 19:29.34 | bakermd | [TK]D-Fender: This worked on 1.2 |
| 19:30.06 | [TK]D-Fender | bakermd: Whatever happened, it doesn't now. Go fix up your dialplan. |
| 19:30.29 | bakermd | [TK]D-Fender: s exists as an extension in the dialplan context bridge - this is correct, isn't it? |
| 19:30.32 | *** join/#asterisk sav_mcfly (n=R00T@pergamo.zonaz.net) |
| 19:30.43 | Innatech | TK: Heh, yeah. I'm ready to replace it, but I imagine I'll still feel a little misty when we take it off the wall and it becomes just another pile of legacy junk. |
| 19:30.49 | [TK]D-Fender | bakermd: it is NOT lying. |
| 19:31.10 | [TK]D-Fender | Innatech: its trash, and congrats on not going down the BCM route. |
| 19:31.33 | Innatech | TK D : yeah, eff that. |
| 19:31.37 | [TK]D-Fender | Innatech: the hadsets can be made reusable with * but not at a great cost ratio. |
| 19:31.52 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 19:32.00 | Innatech | TK D : I suppose I might keep one or two just for me to play with. Could be fun. |
| 19:32.03 | [TK]D-Fender | Innatech: Citel had a SICK deal on Atacom a while back, but thats gone now. |
| 19:32.25 | bakermd | [TK]D-Fender: I'm not trying to be argumentative, just trying to find out why the config worked on 1.2 and not on 1.4 |
| 19:32.25 | [TK]D-Fender | Innatech: 24 port Nortsar>SIP converter for $1K |
| 19:32.53 | Innatech | TK D : Yeah, the subtenants whose portion of the former PBX I'm dealing with don't actually own the handsets so they don't care. |
| 19:32.57 | [TK]D-Fender | bakermd: Its telling yuo exactly whats missing in your dilaplan. Thats all there is to say. |
| 19:33.58 | Innatech | TK D : The main tenant the owns the handsets might end up going that way but I'm not dealing directly with them (at least, not yet. We'll see ;) ) |
| 19:34.10 | bakermd | [TK]D-Fender: Does invalid extension mean that the extension is missing, or that I cannot use s as an extension name? |
| 19:34.25 | [TK]D-Fender | bakermd: Its not there as expected. |
| 19:34.26 | *** join/#asterisk karlhaines (n=karl@unaffiliated/karlhaines) |
| 19:34.41 | [TK]D-Fender | bakermd: Pastebin the whole mess if you'd like to try to prove otherwise |
| 19:34.54 | bakermd | [TK]D-Fender: Cool - will do |
| 19:35.13 | Innatech | [TK] D : That *IS* an interesting piece of kit, tho. Have you set one up? Is it nightmarish? |
| 19:35.41 | `pariah | ok now i have some of my issues resovled, but there is one strange issue remaining. here is my zapata.conf and extensions.conf and the error. http://www.pastebin.ca/548564 |
| 19:36.37 | `pariah | i know the second error is because i dont have the extension setup on a device, so the second one would actually work if a valid ext. were there, but the first one confuses me. where is default coming from? i haven't defined it anywhere :( |
| 19:37.34 | bakermd | [TK]D-Fender: http://rafb.net/p/P8516x81.html |
| 19:37.41 | [TK]D-Fender | Innatech: Set what up? Not sure exactly what you're referring to. |
| 19:37.42 | bakermd | [TK]D-Fender: Let me know what else you need to see |
| 19:38.25 | [TK]D-Fender | `pariah: You have to completely restart * for zaptel changes to take effect (or do "reload chan_zap.so") |
| 19:38.43 | [TK]D-Fender | bakermd: extensions.conf please..... |
| 19:38.50 | `pariah | [TK]D-Fender: its been restarted many times since the changes |
| 19:39.17 | bakermd | [TK]D-Fender: I use RealTime ODBC - that was the snippet of the DB for this app |
| 19:39.21 | [TK]D-Fender | `pariah: == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's' <--- look at this... you tell ME... |
| 19:39.33 | [TK]D-Fender | bakermd: Don't make me ask again.... |
| 19:39.37 | Innatech | [TK] D: that Nortel/SIP gateway....I imagine it needs some amount of configuration/massage/animal sacrifice to make it go. |
| 19:40.01 | [TK]D-Fender | Innatech: Actually it doesn't looklike and I've spoken to someone who's used it. |
| 19:40.09 | [TK]D-Fender | Innatech: Its not "glorious", but it works. |
| 19:40.18 | [TK]D-Fender | Innatech: Earlier revision have been buggy... |
| 19:40.29 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 19:41.19 | Innatech | [TK] D: Interesting. I'll have to talk to the people who owned all the handsets about what they're moving to. If it's a viable piece of equipment to rely on, it might be perfect for them. |
| 19:41.26 | `pariah | [TK]D-Fender: that is where i am confued notice DEFAULT! |
| 19:41.44 | `pariah | [TK]D-Fender: I dont have default anywhere or have zap setup to look in DEFAULT |
| 19:42.06 | bakermd | [TK]D-Fender: http://rafb.net/p/gErFVK10.html |
| 19:42.07 | `pariah | notice my zapata.conf |
| 19:42.08 | [TK]D-Fender | Innatech: I'd still suggest you ditch it ALL if I were you... its a nominal level of functioning, and at $3k for 24 ports now.... costs about as much as new polycoms... so "screw that!" |
| 19:42.43 | [TK]D-Fender | Innatech: Only good if they have almost exactly 24 ext's and can't change their wiring. |
| 19:42.52 | [TK]D-Fender | bakermd: I asked for EXTENSION.CONF. What are you not getting? |
| 19:43.07 | [TK]D-Fender | `pariah: then your changes did not take effect. |
| 19:43.07 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 19:43.14 | [TK]D-Fender | `pariah: Kill * and start it up again. |
| 19:43.27 | `pariah | [TK]D-Fender: ok will will kill the process |
| 19:43.35 | bakermd | [TK]D-Fender: ODBC... There is no extensions.conf - data is stored in DB for realtime changes |
| 19:43.36 | [TK]D-Fender | `pariah: "stop now" |
| 19:43.38 | rene- | hey [TK]D-Fender |
| 19:44.00 | rene- | i have found out that my cisco catalyst express 500 is rebooting for apparently no reason |
| 19:45.24 | rene- | and since i am using it for voice and data vlans, the users arent finding it funny at all |
| 19:45.25 | [TK]D-Fender | bakermd: then you have not "read the manual". You need to do "switch => realtime" in [bridge] for it to know to USE realtime. |
| 19:45.25 | `pariah | ok [TK]D-Fender did that now i restarted the server |
| 19:45.25 | [TK]D-Fender | rene-: Ah the house of cards begins to fall! |
| 19:45.25 | Innatech | [TK] D: You're quite right! I think that particular situation is not too far off of where those folks are, but I don't really know. As I said, they're not my direct client at this point, I'm working for one of their subtenants currently hanging off of the EOL'd Norstar. |
| 19:45.25 | bakermd | [TK]D-Fender: Thanks man!! |
| 19:45.27 | [TK]D-Fender | bakermd: * isn't too bright about their implementation and you can't jsut invent contexts in a database. |
| 19:45.29 | rene- | [TK]D-Fender; yes but i thought the weakest link in the chain was my linux vlan to eth bridge |
| 19:45.35 | rene- | not the cisco gear |
| 19:45.39 | [TK]D-Fender | bakermd: While I don't like it, you need to learn to LIVE with it,. |
| 19:45.56 | `pariah | [TK]D-Fender: == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's' |
| 19:46.11 | [TK]D-Fender | `pariah: pastebin your configs again |
| 19:46.18 | `pariah | same thing after restart! the zapata.conf is correct though and the context is set! |
| 19:46.26 | `pariah | ok 1 sec |
| 19:46.40 | rene- | POE power consumption is under 65% so i wonder |
| 19:46.50 | [TK]D-Fender | `pariah: and your channel 1 you set the context AFTER "channel=". this is BAD!!!! and the problem. |
| 19:46.53 | mihinomenest | is there a document somewhere that lists all of the * applications and their arguments? |
| 19:47.02 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 19:47.03 | [TK]D-Fender | mihinomenest: "show applications" |
| 19:47.08 | [TK]D-Fender | mihinomenest: "show application [appname]" |
| 19:47.25 | *** join/#asterisk deb_user (n=deb_user@albuquerque.agroinnovations.com) |
| 19:47.34 | mihinomenest | I don't suppose there's a pdf, html doc, or man page as well? |
| 19:47.50 | deb_user | anyhbody know how to get zapateller to work on * 1.4.0 with the (nocallerid) option? |
| 19:48.02 | bakermd | [TK]D-Fender: Turns out that sip.conf has the config for Realtime/bridge@sipextensions - adding it to extensions.conf didnt help |
| 19:48.13 | deb_user | i can get it to work, but when i do zapateller(nocallerid), it doesn't do anything, no tone or anything |
| 19:48.55 | [TK]D-Fender | mihinomenest: I gave you "the way it is". You can check the WIKI, but it can be deprecated at any time. |
| 19:49.31 | [TK]D-Fender | `pariah: Also means your zap/2 is bad... |
| 19:49.44 | rene- | [TK]D-fender: have you seen the dlink gear rebooting randomly like that? |
| 19:49.54 | [TK]D-Fender | rene-: nope. |
| 19:50.04 | [TK]D-Fender | rene-: All 100% solid here. |
| 19:50.11 | rene- | and it is like < 1000 USD right? |
| 19:50.17 | [TK]D-Fender | rene-: DES-1526 PoE Switches. |
| 19:50.24 | [TK]D-Fender | rene-: $400 +/- |
| 19:50.30 | [TK]D-Fender | rene-: there is a newer model now. |
| 19:50.36 | rene- | i paid 1500 for each of those cisco switches |
| 19:50.38 | rene- | damn |
| 19:50.53 | bakermd | [TK]D-Fender: The config for contexts bridge, inbound, and forward are set up with switch realtime. |
| 19:50.55 | [TK]D-Fender | rene-: K-Y probably came included with the bundle ;) |
| 19:51.28 | *** join/#asterisk Dovid (n=Dovid@bzq-82-81-102-119.red.bezeqint.net) |
| 19:51.38 | bakermd | [TK]D-Fender: context inbound is apparently functioning correctly |
| 19:51.50 | Dovid | hi guys. how do i change the amount of frames asterisk sends over g729 |
| 19:52.02 | [TK]D-Fender | rene-: http://www.antonline.com/p_D-Link-Systems--Inc.--DES-1228P--Web-Smart-24-PORT-10-100-Poe-switch--4-Gigabit--2-Combo-Sfp-_275788.htm |
| 19:52.39 | Dovid | my ITSP said I am seding 60/240 and I need to be sending 20/20 |
| 19:53.28 | `pariah | [TK]D-Fender: hey sorry cant post now, have to take lunch before its too late, but when i get back can i page you or msg you? |
| 19:53.44 | [TK]D-Fender | `pariah: I' might be around. Feel free to try |
| 19:53.52 | IOscanner | Anyone know if there is a patch to read SIP 'PUBLISH' options? |
| 19:53.53 | Innatech | [TK] D & Rene: how do you like those DES switches? I'm thinking about buying a couple for this project. |
| 19:54.04 | `pariah | [TK]D-Fender: ok great. thanks for all the help |
| 19:54.23 | Innatech | [TK] D & Rene: (as opposed to something a little heavier feature wise but more expensive.) |
| 19:54.26 | [TK]D-Fender | Innatech: I use their predecessor's and they work jsut great |
| 19:54.57 | Innatech | [TK] D : No issues with the smallish MAC table? |
| 19:55.13 | [TK]D-Fender | Innatech: They do VLAN, PoE on all ports, are managed, do SNMP & have dual dual-format GBIT uplinks. What MORE do you want? ;) |
| 19:55.25 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 19:55.48 | rene- | D-Fender: thanks man |
| 19:55.53 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 19:56.04 | rene- | i want have bought some of those instead :) |
| 19:56.52 | Innatech | TK D : Well, their equivs in the Netgear product line used to hold 8x more MAC addresses, and I'm not familiar enough with their management features vs competition to know what the price differentials might account for. |
| 19:57.39 | Innatech | TK D : But, if you say they rock thats more or less good enough for me. |
| 19:58.03 | [TK]D-Fender | Innatech: http://www.dlink.com/products/?sec=0&pid=540 |
| 19:58.17 | bakermd | OK - so making progress here - Problem now is No such application 'SetVar' |
| 19:58.18 | [TK]D-Fender | Innatech: Go read and see if you like the specs |
| 19:58.28 | [TK]D-Fender | bakermd: Yuo are SO 1.0.x.... |
| 19:58.32 | Innatech | Will do, 'preciate it. |
| 19:58.36 | [TK]D-Fender | bakermd: welcome to 2007! |
| 19:59.41 | bakermd | [TK]D-Fender: I know, right? This box ran like a champ forever, but its starting to hiccup, so I am moving to the current release - looks like you use set instead of setvar now |
| 20:00.10 | [TK]D-Fender | Innatech: OMG, link looks bad |
| 20:00.31 | Waverly360 | [TK]D-Fender: So I figured out my callerid issue. |
| 20:00.45 | [TK]D-Fender | Innatech: http://www.dlink.com/products/?sec=2&pid=541 |
| 20:01.00 | Innatech | :) |
| 20:01.02 | Waverly360 | [TK]D-Fender: I'm not waiting long enough for the telco to send me the information. As soon as the line rings, I pick it up and shoot it to my agi script. |
| 20:01.21 | [TK]D-Fender | Waverly360: BRILLIANT :) |
| 20:02.22 | Waverly360 | [TK]D-Fender: I didn't really think about it..I thought all that information made it across the moment the line starts ringing..but I guess callerids take at least a couple of rings to get that info. |
| 20:04.50 | *** join/#asterisk kimosabe (n=kimosabe@189.175.37.61) |
| 20:05.09 | kimosabe | can some one helpme find a good voice overip provider that has no signup fee ? |
| 20:05.30 | rob0 | FWD :) |
| 20:05.47 | rob0 | iaxtel |
| 20:06.05 | [TK]D-Fender | rob0++ |
| 20:06.10 | rob0 | :) |
| 20:07.15 | kimosabe | is aiaxtell unlimeted usa canada |
| 20:08.44 | rob0 | Iaxtel doesn't connect to the PSTN. |
| 20:10.04 | kimosabe | ok thanks |
| 20:13.26 | [hC] | So, if my telco provisioned my PRI as National ISDN1, but i use National ISDN2 in zapata.conf, what does that mean? |
| 20:13.28 | [hC] | It works.. |
| 20:14.34 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 20:16.38 | mihinomenest | ParkAndAnnounce(announce:template|timeout|dial|[return_context]) - is that "return context" the sip.conf context or the extensions.conf context? |
| 20:18.21 | [TK]D-Fender | [hC]: Backwards compatability. |
| 20:18.39 | [TK]D-Fender | mihinomenest: extensions.conf clearly. |
| 20:18.52 | [hC] | [TK]D-Fender: would it be better to force ni1? sometimes i get circuit busy messages on their circuit and dont know why |
| 20:19.04 | [TK]D-Fender | [hC]: #34? |
| 20:19.20 | [hC] | [TK]D-Fender: sounds familiar, but not sure, id have to check. |
| 20:19.40 | [TK]D-Fender | [hC]: Dont' forget that some ISDN codes come back LOOKING like you didn't get a channel, but its actually the switch telling your the person you are cALLING is busy |
| 20:19.47 | [TK]D-Fender | [hC]: I get that here on NI1. |
| 20:19.54 | Katty | if i do a System() command, how do i get multiple commands onto the same line? is it || or ; or something? |
| 20:20.10 | [TK]D-Fender | Katty: call multiple or make a script. |
| 20:20.48 | [hC] | [TK]D-Fender: yeah, thats not the case. it immedialtey calls back and it goes thru. it happens a lot. |
| 20:21.17 | Katty | [TK]D-Fender: call multiple? |
| 20:21.27 | Katty | [TK]D-Fender: you do not parse. |
| 20:21.32 | [hC] | [TK]D-Fender: just wanted to make sure that moving from ni2 to ni1 is not going to cause a problem. |
| 20:22.34 | *** join/#asterisk doolittlework (i=doolittl@196.211.34.2) |
| 20:23.11 | doolittlework | thank you Mark Spencer |
| 20:23.23 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 20:23.38 | doolittlework | hi dennis |
| 20:23.40 | [TK]D-Fender | Katty: exten => s,2,System(blah...) |
| 20:23.43 | [TK]D-Fender | Katty: exten => s,3,System(blah...) |
| 20:23.45 | [TK]D-Fender | Katty: exten => s,4,System(blah...) |
| 20:25.29 | De_Mon | add extension 123123,1,NoOp(${REGEX("[abc]" his)}) into local replace |
| 20:25.46 | De_Mon | see anything wrong with this line? the CLI just gives me add extension usage when I try it |
| 20:27.45 | doolittlework | DeMon: What are you trying to do? |
| 20:30.28 | De_Mon | doolittlework test a REGEX patern |
| 20:30.59 | De_Mon | I put it in extensions.conf and reloaded and all worked as expected, CLI didn't like me adding it through add extension tho.. very weird |
| 20:31.36 | De_Mon | just sucks having to extensions reload till I got the pattern correct |
| 20:32.51 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 20:35.10 | doolittlework | sorry your ask is over my head way over De_Mon |
| 20:35.16 | *** join/#asterisk duckz (n=duckz@141.85.3.18) |
| 20:36.40 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 20:38.58 | De_Mon | np |
| 20:40.15 | doolittlework | what does the REGEX DO? |
| 20:41.55 | De_Mon | its just a function |
| 20:42.02 | De_Mon | show function REGEX |
| 20:42.20 | redax | doing gex again :) |
| 20:42.20 | doolittlework | k let me see |
| 20:42.21 | De_Mon | the issue is that I can add the extension to extensions.conf but not using 'add extension' from the CLI |
| 20:42.48 | doolittlework | ok that makes sence |
| 20:45.55 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:48.03 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 20:48.47 | *** join/#asterisk astguy (n=astguy@c-24-8-95-194.hsd1.co.comcast.net) |
| 20:53.15 | De_Mon | ahhhh each extension should be in its own context so you can handle invalid/timeout uniquely |
| 20:54.21 | neverblue2 | can I listen in a current call, with two headsets ? |
| 20:55.03 | neverblue2 | can I get maybe a usb to "audio" plug, then use a dual "splitter" to split the "audio" into to headsets? |
| 20:55.23 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:55.26 | rene- | i want to say thanks to all the nerds that help other nerds |
| 20:55.36 | neverblue2 | hey Fender |
| 20:55.41 | neverblue2 | hard question for you :) |
| 20:55.52 | neverblue2 | can I listen in a current call, with two headsets ? |
| 20:55.54 | [TK]D-Fender | neverblue2 : VIAGRA |
| 20:55.58 | neverblue2 | lmao |
| 20:56.04 | rene- | hehehe |
| 20:56.12 | rene- | viagra is a hard answer |
| 20:56.15 | doolittlework | :} |
| 20:56.15 | Katty | does asterisk have a built in file that says please wait while your call is being transfered. |
| 20:56.28 | [TK]D-Fender | neverblue2 : Sure, Plantronics sells Polaris Y adaptors |
| 20:56.33 | neverblue2 | they do |
| 20:56.35 | neverblue2 | k, thanks |
| 20:56.51 | rene- | Katty: ls /var/lib/asterisk/sounds/* | grep wait |
| 20:56.56 | [TK]D-Fender | neverblue2 : Good for training CSR's |
| 20:57.04 | neverblue2 | yeah |
| 20:57.06 | neverblue2 | thats what its for |
| 20:57.07 | Innatech | (So is a sharp stick.) |
| 20:57.07 | Katty | rene-: thank you |
| 20:57.18 | neverblue2 | you work for a VOIP company? |
| 20:57.37 | rene- | :) |
| 20:57.38 | [TK]D-Fender | neverblue2 : Who me? |
| 20:57.56 | neverblue2 | yes, you |
| 20:58.00 | [TK]D-Fender | neverblue2, Nope, but I do minor consulting in *. |
| 20:58.08 | [TK]D-Fender | privately |
| 20:58.09 | neverblue2 | minor? |
| 20:58.10 | neverblue2 | lol |
| 20:58.19 | [TK]D-Fender | neverblue2, Well no SER, etc... |
| 20:58.25 | neverblue2 | im looking for a new VOIP provider |
| 20:58.33 | neverblue2 | can you recommend one or two? |
| 20:58.35 | [TK]D-Fender | ~itsp |
| 20:58.36 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others." (tm) (c) 2007 ManxPower |
| 20:58.38 | [TK]D-Fender | ^^^^^ |
| 20:58.46 | neverblue2 | no, you personnally |
| 20:59.02 | [TK]D-Fender | neverblue2 : Teliax is considered "less sucky" than most. |
| 20:59.19 | [TK]D-Fender | neverblue2, Well.... I don't really use them.... |
| 20:59.50 | neverblue2 | this isnt a residential plan I am looking for |
| 21:00.57 | [TK]D-Fender | neverblue2, I DEFINATELY don't suggest them as a primary link for business |
| 21:01.07 | [TK]D-Fender | (itsp's in general) |
| 21:01.24 | neverblue2 | is there anyone you would recommend for business VOIP service? |
| 21:01.35 | neverblue2 | or anyone in the channel for that matter... |
| 21:01.42 | [TK]D-Fender | neverblue2, Everything depends on usage in channels, volume, location, etc. |
| 21:01.50 | De_Mon | neverblue2 you can also use the ChanSpy application |
| 21:01.54 | [TK]D-Fender | channels, so on |
| 21:02.08 | Trevor_b | neverblue2: I use teliax, and have used broadvoice (still do a little). I wouldnt suggest broadvoice even with there prices, they just have various issues. Teliax has been a good service, and their phone support is really good. Email less so, but call from a cell if you need something fast and have no outside hardlines. |
| 21:02.22 | De_Mon | add extension 123123,1,NoOp(${REGEX("[abc]" his)}) into local replace |
| 21:02.34 | redax | wheee.. what a swivel-eyed reporter on the tv... |
| 21:02.35 | De_Mon | [TK]D-Fender you see any reason why this command shouldnt work? |
| 21:02.40 | neverblue2 | ChanSpy? |
| 21:02.44 | neverblue2 | ill look into that |
| 21:02.45 | [TK]D-Fender | De_Mon, Sorry.... don't do regex... |
| 21:02.53 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 21:03.11 | De_Mon | [TK]D-Fender its not a regex question I promise |
| 21:03.16 | neverblue2 | Trevor_b, youre referring to personal service, correct? |
| 21:03.24 | Trevor_b | neverblue: I use hard lines for main inbound and forward to teliax corporate. |
| 21:03.27 | Trevor_b | No business |
| 21:03.29 | De_Mon | [TK]D-Fender exten => 123123,1,NoOp(${REGEX("[abc]" his)}) |
| 21:03.42 | De_Mon | works fine from extensions.conf, but it doesn't let me add it on CLI |
| 21:04.02 | [TK]D-Fender | De_Mon, ? |
| 21:04.22 | neverblue2 | ah De_Mon was looking for something to handle the hardware issue with listening in on a call :) |
| 21:04.41 | De_Mon | I can create an extension thru extensions.conf that I can't create on the CLI |
| 21:04.49 | Trevor_b | Hardlines for my main on TDM (dont need PRI for my needs) then i use their corporate account on Teliax. |
| 21:05.01 | De_Mon | neverblue2 ah |
| 21:05.10 | Trevor_b | so if my hardlines roll busy up the line, then i forward to non hardline. |
| 21:05.12 | neverblue2 | Fender, so what can I tell you so you could recommend a good business VOIP provider? |
| 21:06.14 | De_Mon | [TK]D-Fender it looks like I can't create add a line containing a function on the CLI... |
| 21:06.39 | astguy | Anyone recommend an ITSP that offers the best rate on multiple channels? |
| 21:07.06 | astguy | I'm looking for 10-20 channels, but don't really want to spend $20/month/channel |
| 21:07.10 | De_Mon | oops nm 2nd test was broken |
| 21:07.33 | Trevor_b | neverblue2: The only one I can highly suggest is Teliax. Been using it for business for over a year. fairly minor issues compared to service during the same time for broadvoice. |
| 21:07.34 | *** join/#asterisk sandorp (n=sandor@firewall2.wsi.net) |
| 21:08.11 | Trevor_b | astguy: teliax pay as you go, depends on how many minutes you expect to run, but its like 10 a month before minutes and 2 cents a minute, 20 channels max |
| 21:09.01 | astguy | Trevor_b: thx. Do they charge for inbound calls? |
| 21:09.02 | Katty | how do you... |
| 21:09.08 | sandorp | I seem to have a problem in my dialplan that I can't figure out ... if I redial the same exact long distance number, 1 out of 3 times it calls a local number by stripping the area code; I have checked my dialplan to make sure that local numbers are exactly 7 digits and long distance starts with a 1 + 10 digits |
| 21:09.08 | Katty | how do you put someone on hold? |
| 21:09.19 | Trevor_b | pay as you go i think is both directions 2 cents, but check the website. |
| 21:09.55 | astguy | Trevor_b: thanks |
| 21:10.18 | Katty | s,1,Dial(5 phones) ~ s,2,Answer ~ s,3,playback(pleasehold) ~ s,4,somehowputthispersononhold ~ s,5,Dial(thesephonesagain) ~ s,6,Goto(afterhoursthingy,s,1) |
| 21:10.27 | sandorp | anyone know why the pattern matching for the same dialed number would resolve to 2 different entries in the dialplan? |
| 21:11.03 | Trevor_b | Katty: Can play music while you dial |
| 21:11.17 | Katty | Trevor_b: that's what i'm trying to do |
| 21:11.27 | Katty | Trevor_b: but i dunno how |
| 21:11.28 | [TK]D-Fender | Katty, "m" in Dial |
| 21:11.44 | Katty | Dial(m)? |
| 21:12.43 | [TK]D-Fender | Katty, Dial(SIP/1&SIP/2&SIP/3,20,m) |
| 21:12.44 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 21:12.56 | Katty | hmm |
| 21:13.00 | [TK]D-Fender | Katty, "show application dial" |
| 21:13.06 | [TK]D-Fender | Katty, Read up! |
| 21:13.30 | Katty | i don't think that's going to do what i want it to do |
| 21:13.32 | bakermd | What is the best way to take digit input from a user? |
| 21:13.42 | Katty | but maybe |
| 21:13.45 | Katty | we'll see |
| 21:13.45 | [TK]D-Fender | Katty, What do yuo want to do then? |
| 21:13.58 | bakermd | i.e. I want them to dial a 10 or so digit string that I store in a variable |
| 21:14.03 | [TK]D-Fender | bakermd, Depends how many digits, what you have to do with them, etc. |
| 21:14.06 | De_Mon | sorry, its: add extension 123123,1,NoOp(${REGEX("[abc]" foo)}) into local replace |
| 21:14.12 | [TK]D-Fender | bakermd, "show application read" |
| 21:14.18 | bakermd | [TK]D-Fender: Variable - 10 to 20 |
| 21:14.21 | bakermd | [TK]D-Fender: Thanks! |
| 21:14.43 | Katty | [TK]D-Fender: i'll pastebin i a minute. |
| 21:14.57 | Katty | [TK]D-Fender: yeah, that's too quick |
| 21:15.03 | Katty | [TK]D-Fender: sec |
| 21:15.25 | De_Mon | which is better, Read() or putting the function into its own context and using WaitExten() and a dialplan pattern... |
| 21:16.06 | Katty | [TK]D-Fender: pastebin.ca/548786 |
| 21:16.26 | Katty | [TK]D-Fender: look at line 9-11 |
| 21:16.34 | Katty | [TK]D-Fender: i need it to pause just long enough for the intercom to pick up |
| 21:16.43 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 21:16.47 | Katty | [TK]D-Fender: say its thing, and then start ringing again |
| 21:16.50 | De_Mon | waitexten lets you use background() and they can interupt at any time, but other than that |
| 21:17.14 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 21:19.13 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:19.19 | [TK]D-Fender | Katty, Why page all phones and then ring then right away after? Isn't ringing incentive enough to answer? |
| 21:19.28 | Katty | nope |
| 21:19.42 | Katty | the 2 second wait works wonderfully |
| 21:19.45 | [TK]D-Fender | Katty, They are going to geta TON of extr calls then. |
| 21:19.50 | Katty | [TK]D-Fender: yep |
| 21:19.56 | [TK]D-Fender | FUGLY |
| 21:19.58 | Katty | [TK]D-Fender: one girl is supposed to answer the phone. |
| 21:20.02 | Katty | [TK]D-Fender: but sometimes she's too busy |
| 21:20.08 | [TK]D-Fender | Katty, What psycho wants it this way? |
| 21:20.25 | Katty | [TK]D-Fender: and the other people in the group are WAY on the other side of the building, and have no way of knowing if she's gotten the call. |
| 21:20.39 | Katty | [TK]D-Fender: so, the bossman wants a way to alert the rest of the people that she can't get to the call. |
| 21:20.50 | Katty | [TK]D-Fender: when, in fact, every single friggen polycom phone downstairs will pick that up |
| 21:21.06 | Katty | [TK]D-Fender: just to Make Sure(tm) one of the other people who answers the phone will still know, even if they're in my office. |
| 21:21.12 | Katty | [TK]D-Fender: you're right. it's stupid. |
| 21:21.15 | Katty | [TK]D-Fender: but it's what he wants. |
| 21:21.35 | De_Mon | Katty is bossman a he or a she? |
| 21:21.40 | Katty | he |
| 21:21.46 | Katty | besides, it's kinda neat. |
| 21:21.58 | Katty | i never knew you could create a call just by dumping a file into /outgoing |
| 21:22.07 | De_Mon | eh |
| 21:22.55 | De_Mon | how do I announce a caller to someone that is xfered from a queue? |
| 21:23.16 | [TK]D-Fender | De_Mon, depends how they were called. |
| 21:23.57 | De_Mon | xfered isn't the right term, they were handed to the queue member |
| 21:26.59 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 21:27.31 | [TK]D-Fender | De_Mon, handed HOW.... |
| 21:30.32 | doolittlework | atty how do you do this----i never knew you could create a call just by dumping a file into /outgoing |
| 21:30.40 | doolittlework | katty how do you do this----i never knew you could create a call just by dumping a file into /outgoing |
| 21:36.40 | FuriousGeorge | i have no idea how my users mnanage to do this or how to prevent ithttp://pastebin.ca/548829 |
| 21:36.50 | FuriousGeorge | and that kinda scares me :) |
| 21:36.55 | Innatech | doolittlework: check out Katty's pastebin. Look for the lines with the cp and mv commands. |
| 21:37.21 | FuriousGeorge | hey Katty lonf time no see |
| 21:37.24 | FuriousGeorge | *long |
| 21:38.22 | De_Mon | [TK]D-Fender I didn't know there was more than one way for a queue member to be given someone in the queue |
| 21:39.01 | [TK]D-Fender | De_Mon, indeed there is. Consider how it is that that they are MEMBERS of the queue and login |
| 21:39.09 | *** part/#asterisk komradebob (n=komradeb@164.55.254.106) |
| 21:39.48 | De_Mon | they are added thru addqueuemember? |
| 21:40.13 | De_Mon | the phone rings queuemember picks up, and there's the person from the queue |
| 21:41.07 | [TK]D-Fender | De_Mon, Ask yourself "what makes them RING" <- |
| 21:42.12 | De_Mon | I really have no clue what you're getting at... Addqueuemember adds SIP/somephone the QUEUE rings sip/somephone |
| 21:43.36 | [TK]D-Fender | De_Mon, Think what ELSE you do than just dialing a SIP device DIRECTLY that would allow you to prepend the call for announce... |
| 21:44.10 | yannj_fr | is there someone interested in the new version of the book Asterisk , TFOT? |
| 21:44.14 | De_Mon | i have no IDEA or i wouldn't ASK |
| 21:45.59 | [TK]D-Fender | De_Mon, *sigh* |
| 21:45.59 | [TK]D-Fender | De_Mon, Use a local channel and Dial them yourself. I'm sure you'll know which options to use. |
| 21:46.11 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:46.37 | [TK]D-Fender | yannj_fr, No, the old one will do just fine through * 3.0.1 |
| 21:46.57 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:47.19 | *** join/#asterisk grandy (n=chatzill@c-69-181-153-165.hsd1.ca.comcast.net) |
| 21:47.30 | yannj_fr | [TK] D-Fender, reallly? |
| 21:48.28 | [TK]D-Fender | </sarcasm> |
| 21:49.04 | yannj_fr | My company bought it on oreilly , it is the second release |
| 21:49.14 | yannj_fr | compliant to Asterisk 1.4 |
| 21:49.34 | yannj_fr | and I think the PDF shhould be under Creative Commons |
| 21:51.57 | [TK]D-Fender | De_Mon, "show application dial" |
| 21:53.50 | sandorp | I have a problem dialing via an analog line ... I call the same long distance number 3 times in a row and at least 1 time it strips the area code and calls a local number; any idea why asterisk might do that? my local pattern is _9NXXXXXX and the long distance one is _91NXXXXXXXXX |
| 21:54.34 | sandorp | I can't see how the local pattern could ever match a long distance number |
| 21:54.47 | De_Mon | im trying to figure out how that would work |
| 21:55.03 | sandorp | btw, I strip the 9 |
| 21:55.13 | Dovid | how do i see what channels are using what codecs ? |
| 21:55.19 | De_Mon | instead of sending the caller directly to the queue, dial(local/thequeue)? |
| 21:56.01 | [TK]D-Fender | De_Mon, him... its when you log them IN. SIP/ <- this is what should be changing ;) |
| 21:56.01 | grandy | Hello... does anyone feel like helping me debug my configuration? I set up DNS SRV records for sip at the instruction of my origination provider, but I'm not getting any DTMF recognition and sometimes the calls do not appear to even be reaching my asterisk box.... I'd very much appreciate some advice... My provider's asterisk points to the location of my server(s) via the SRV records... |
| 21:56.03 | De_Mon | with options for when the call is answered... that just doesn't sound workable |
| 21:56.37 | [TK]D-Fender | De_Mon, SIP/[device] doesnt' let you do anything. Local DOES. in there you can do the dial YOURSELF. |
| 21:57.40 | De_Mon | addqueuemember(local/some@phones) or something along those lines? |
| 21:57.57 | *** join/#asterisk [[blah]asfd (n=ckwall@54.sub-70-193-70.myvzw.com) |
| 21:57.59 | De_Mon | talk about added layer of complexity |
| 21:59.17 | [TK]D-Fender | De_Mon, very small actually... |
| 21:59.19 | De_Mon | seems like using local/ would remove them from the queue to the new dialplan |
| 21:59.22 | [TK]D-Fender | De_Mon, run with it :) |
| 21:59.41 | *** join/#asterisk lwh (n=lwh192@rdsl-0469.tor.pathcom.com) |
| 21:59.44 | *** join/#asterisk monstertruck (n=monstert@c-75-74-251-82.hsd1.fl.comcast.net) |
| 21:59.53 | [TK]D-Fender | De_Mon, nope.... Queue is in effect and its purpose it to BRIDGE channels. Doesn't matter what either side is. |
| 22:01.28 | *** join/#asterisk mike38533 (n=omar@c-66-176-110-20.hsd1.fl.comcast.net) |
| 22:02.20 | [[blah]asfd | when using chanisavail i never get any status back except for 0. even when the phone is on a call already. this changed after upgrading from 1.2.14 to 1.4.4. anyone have any ideas why that could be? here is what I am trying to do. http://pastebin.ca/548886 like i said earlier today... I was able to do this before the upgrade. (using prio jumping. now trying to do prio lable) |
| 22:06.27 | FuriousGeorge | somehow, sometimes, when my users attempt to transfer a call this happens http://pastebin.ca/548829 |
| 22:06.47 | FuriousGeorge | [[blah]asfd: use dialstatus var instead of canisavail |
| 22:06.57 | FuriousGeorge | chanisavail* is so asterisk 1.0 |
| 22:07.24 | FuriousGeorge | it might even be deprecated, and the n+101 jumping behavior is disabled by default in 1.4.4 |
| 22:07.44 | FuriousGeorge | i mean in 1.4.X |
| 22:08.11 | *** join/#asterisk kimosabe (n=kimosabe@189.175.37.61) |
| 22:08.12 | [[blah]asfd | FuriousGeorge: so do you mean to do: exten => s,1,dialstatus(SIP/${ARG1}) |
| 22:08.13 | rene- | FuriousGeorge n+101 is alive and kicking in 1.4.4 |
| 22:08.18 | *** join/#asterisk woolbeo (n=woolbeo@exchange.services.daqe.com) |
| 22:08.27 | FuriousGeorge | rene-: i said it was disabled by default |
| 22:08.37 | FuriousGeorge | which is only what ive read here and there, i dont use that |
| 22:08.39 | [[blah]asfd | i moved away from n+101 |
| 22:08.47 | [[blah]asfd | but did i do the change right? |
| 22:09.01 | FuriousGeorge | [[blah]asfd: no, you do a dial, and the next line you do a goto(s-{$DIALSTATUS} |
| 22:09.21 | woolbeo | How do I remove a busy status of an extension with a hint? the exten is not busy, but asterisk thinks it is. |
| 22:09.28 | kimosabe | does anyone know a good voice/ip to pstn provider with no signup fee ? |
| 22:09.29 | FuriousGeorge | then you have extensions to catch the possible dialstatus responses, such as s-UNAVAIL |
| 22:09.44 | FuriousGeorge | s/response/return |
| 22:09.51 | FuriousGeorge | s |
| 22:10.06 | rene- | FuriousGeorge: chanisavail was not good for me for agents coming from a queue context what is working for me is set group with enforcing call-limit+1 |
| 22:10.11 | FuriousGeorge | you can read all about it, it behaves very intuitively |
| 22:10.33 | FuriousGeorge | rene-: [[blah]asfd is the one using chanisavail, not me |
| 22:10.43 | [[blah]asfd | FuriousGeorge: I need it to generate a busy signal if the phone that is in use. |
| 22:11.01 | rene- | FuriousGeorge: however it is being useful to me now for implementing the Page function, if a phone is in use at all then dont send them a call with auto-answer header indications |
| 22:11.03 | [[blah]asfd | i am using chanisavail to generate that |
| 22:11.05 | FuriousGeorge | so s-BUSY(playback(congestion)) or something |
| 22:11.34 | rene- | blah: as FuriousGeorge said the dial application will set a dialstatus |
| 22:11.38 | [[blah]asfd | but i have to create the reason for this phone to be busy... cuz i can send 8 calls to the phone before it is busy. |
| 22:11.52 | [[blah]asfd | i use chanisavail to say that if it is in use at all, then it is busy. |
| 22:12.00 | FuriousGeorge | rene-: my sip channels just dont page when they are in use by default, so if i try, even with header, you are not interrupted |
| 22:12.04 | rene- | that is true for things like Queue() and Voicemail() |
| 22:12.10 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 22:12.12 | FuriousGeorge | [[blah]asfd: set call-limit in sip.conf |
| 22:12.23 | rene- | FuriousGeorge: my sip channels take the other call and then lockup and reboot |
| 22:12.31 | FuriousGeorge | lol |
| 22:12.38 | FuriousGeorge | i know the feeling |
| 22:12.56 | [[blah]asfd | i did incominglimit=1 then i could no longer transfer |
| 22:12.59 | FuriousGeorge | speaking of sip channels, any idea what is causing this http://pastebin.ca/548829 |
| 22:13.15 | rene- | blah: it is tricky, but you cant set it right |
| 22:13.23 | rene- | i meant you can |
| 22:13.51 | rene- | if on 1.4 try using the limitonpeer=yes option in the general section of sip |
| 22:14.19 | FuriousGeorge | [[blah]asfd: so you want to be able to transfer, but you dont want call waiting on the phone by default then |
| 22:14.34 | [[blah]asfd | something like that, right |
| 22:14.52 | FuriousGeorge | so set call-limit to 2 and disable callwaiting in the client, it will say its busy if someone tries to call it |
| 22:15.04 | sandorp | is the SayDigits application built into asterisk 1.4? I'm trying to figure out why the number that I dial on x-lite is not the number dialed by asterisk |
| 22:15.27 | [[blah]asfd | how do i disable the client for callwaiting? |
| 22:15.30 | [[blah]asfd | in sio.conf? |
| 22:15.34 | FuriousGeorge | rene-: nah, im still using 1.2 on this server... forget how to fix it, i'd like to know how they do it to begin with |
| 22:15.47 | FuriousGeorge | [[blah]asfd: what is the client? |
| 22:15.54 | FuriousGeorge | (the phone) |
| 22:15.56 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 22:16.00 | [[blah]asfd | linksys spa942 |
| 22:16.15 | FuriousGeorge | log into the things firmware and look around, ill bet you its in there |
| 22:18.52 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 22:19.59 | *** join/#asterisk karlhaines (n=karl@unaffiliated/karlhaines) |
| 22:22.29 | *** join/#asterisk MrWup (i=root@80-46-38-124.static.dsl.as9105.com) |
| 22:22.30 | MrWup | hey |
| 22:22.31 | MrWup | exten =>_#1XX,1,Goto(voicemail,s,1) |
| 22:22.33 | MrWup | this wont work |
| 22:22.39 | MrWup | just says call failed |
| 22:22.41 | MrWup | any idea why? |
| 22:22.47 | rene- | ahh so the star book is updated to 1.4? |
| 22:22.50 | MrWup | the aim is to let the user type in say #100 |
| 22:22.55 | rene- | gimme |
| 22:22.59 | MrWup | and get to the management for voicemail of extension 100 |
| 22:23.22 | MrWup | i got it working with exten =>_*1XX,1,Goto(voicemail,s,1) |
| 22:23.58 | FuriousGeorge | MrWup: dont use # |
| 22:24.06 | MrWup | why? |
| 22:24.20 | FuriousGeorge | # means send on some clients, or something |
| 22:24.24 | FuriousGeorge | use * |
| 22:24.25 | MrWup | oh |
| 22:24.26 | MrWup | ok |
| 22:26.07 | MrWup | when i do record... can i specify the path to a samba share? |
| 22:26.18 | MrWup | eg.. i have a share mounted which puts files on my windows 2003 fileserver |
| 22:26.48 | MrWup | could i do record(/mnt/telecomstore/voicemail/greeting:alaw) |
| 22:26.49 | MrWup | ? |
| 22:27.06 | *** join/#asterisk Know1 (i=know1@creep.bur.st) |
| 22:28.29 | MrWup | ah |
| 22:28.30 | MrWup | i can |
| 22:28.33 | MrWup | yeehaw |
| 22:28.57 | [[blah]asfd | yeah, there is nothing in the phone to stop call waiting. |
| 22:29.43 | [[blah]asfd | i looked up limitonpeer on the wiki and did not see anything there. |
| 22:29.53 | FuriousGeorge | lookup call-limit |
| 22:30.25 | *** part/#asterisk mike38533 (n=omar@c-66-176-110-20.hsd1.fl.comcast.net) |
| 22:31.51 | [[blah]asfd | but without call waiting control, call-limit=2 would still ring the phone if a second call came in wouldnt it? and if i set it to 1 then they could not transfer. |
| 22:33.23 | [[blah]asfd | chanisavail worked perfectly for this. but now no matter what, in 1.4 chanisavail always shows the phone avail no matter what its physical status is. |
| 22:33.44 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 22:34.51 | FuriousGeorge | im sure there is a way |
| 22:36.55 | FuriousGeorge | how is it that incominglimit preventing you from transfering |
| 22:38.53 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 22:39.03 | *** join/#asterisk Strom_M (n=strom@208.47.199.4) |
| 22:39.11 | yannj_fr | [TK] D-Fender ... m |
| 22:39.14 | yannj_fr | pm* |
| 22:40.12 | [[blah]asfd | not sure... i insert incomminglimit=1 and i cant transfer i take it out, i can |
| 22:40.17 | [[blah]asfd | dunno |
| 22:42.39 | [[blah]asfd | it seems like the core of the issue is that the chanisavail is not correctly reporting the status of the phone. |
| 22:42.50 | [[blah]asfd | it always reports as 0 rather than the accurate status. |
| 22:43.11 | [[blah]asfd | sigh |
| 22:50.18 | tdonahue-laptop | i'm seeing a message like "zaptel.c:771 (pid 3836: asterisk) got signal 80000000" scrolling on my console occasionally.... |
| 22:50.27 | tdonahue-laptop | is this something that i should be worried about? |
| 22:50.32 | *** join/#asterisk CoffeeIV (i=rgr@rrcs-71-42-183-82.sw.biz.rr.com) |
| 22:50.56 | tdonahue-laptop | i saw that the message was removed in TRUNK, so i'm thinking probably not, but I just want to be sure |
| 22:51.47 | CoffeeIV | If I am installing asterisk on a server with a voicetronix card and no digium hardware on it, do I need to install the zaptel stuff ? That stuff is modules and utilities only needed for digium's hardware, right ? |
| 22:51.47 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 22:52.56 | _VoiceMeUp_COM | Addr->IP : 69.70.72.134 Port 0 |
| 22:53.00 | _VoiceMeUp_COM | why is it using this ? |
| 22:53.03 | _VoiceMeUp_COM | its static ip |
| 22:53.05 | _VoiceMeUp_COM | from realtime |
| 22:53.18 | tdonahue-laptop | CoffeeIV, zaptel is also used as a timing device for some of the modules |
| 22:53.41 | tdonahue-laptop | specifcally (since i'm configuring it right now) MeetMe comes to mind |
| 22:55.22 | CoffeeIV | tdonahue-laptop, thanks, I will install it anyway, can't hurt I suppose ( although it is such an old linux distro that I may have to use an older zaptel and asterisk 1.2) |
| 22:56.18 | tdonahue-laptop | CoffeeIV, even alot of the newer ones are still packaging 1.2 |
| 22:57.46 | CoffeeIV | I'm trying to put it on RH 7.1, and I will probably have to upgrade the kernel and a lot of stuff by hand |
| 22:58.16 | *** join/#asterisk frocos11292 (n=ask@firewall.vipvoz.com) |
| 22:59.37 | aptura | came across a industrial cordless/push to talk wireless phone that can cover up to 3,000 acres. |
| 23:00.10 | CoffeeIV | aptura: I want it, how much $ ? |
| 23:00.33 | _VoiceMeUp_COM | i still get default expiry 130 |
| 23:00.38 | _VoiceMeUp_COM | i tried to make it 604800 |
| 23:00.45 | _VoiceMeUp_COM | not sure why |
| 23:00.46 | aptura | covers 12 floors in a building or 250,000 sq feet in a warehouse. |
| 23:00.51 | _VoiceMeUp_COM | trying to simulate 1 week expiry |
| 23:00.59 | _VoiceMeUp_COM | cause hes static and ARA sip is broken |
| 23:01.06 | tdonahue-laptop | CoffeeIV, Redhat 7.1... you are a glutton for punishment |
| 23:01.32 | aptura | http://www.engeniuscanada.com/engmain.htm I got a quote from greybar and the starting price is around 700 dollars and up. |
| 23:01.42 | aptura | Alot of these phones can work together. |
| 23:02.16 | aptura | It sounds like a nextel setup without the service cost for a fixed area. This sounds like the perfect wifi phone replacment. |
| 23:02.32 | *** join/#asterisk dlynes_laptop (n=dlynes@d207-216-161-56.bchsia.telus.net) |
| 23:02.44 | *** join/#asterisk Rusty1 (n=Rusty1@cpe-72-226-96-74.nycap.res.rr.com) |
| 23:05.47 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 23:05.55 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 23:06.49 | tzafrir_laptop | CoffeeIV, RH7.1??? |
| 23:06.51 | tzafrir_laptop | wow |
| 23:06.59 | CoffeeIV | yeah |
| 23:07.07 | CoffeeIV | kernel 2.4.2 |
| 23:07.12 | CoffeeIV | those were the days |
| 23:07.14 | tzafrir_laptop | why not upgrade to 7.3 at least? something that is half-supported |
| 23:08.33 | tzafrir_laptop | I recall seeing somewhere in the zaptel code requirement for kernel >= 2.4.8 |
| 23:08.35 | aptura | tzafrir you know of software that can hunt for a open wifi spot and it connects me with a audible beep? |
| 23:08.47 | tzafrir_laptop | aptura, no |
| 23:08.50 | aptura | k |
| 23:08.56 | CoffeeIV | I looked in the code and zaptel wants 2.4.5 or greater |
| 23:09.57 | tzafrir_laptop | CoffeeIV, another path of least resistance: centos 2. It's supposed to be equivalent of RHEL2.1, which is a bit close to RH7.2 |
| 23:10.51 | CoffeeIV | well, if I have to re-install, I will go straight to CentOS 5.0 or something that modern -- there is software on this old box that I want to interact with Asterisk, and if I do a re-install of something modern, then I have to try to get that stuff working |
| 23:11.27 | CoffeeIV | upgrading might be what I need to do . . . I'll put a couple of hours into getting asterisk working on RH 7.1 first, though |
| 23:12.14 | tzafrir_laptop | if you upgrade the kernel, make and whatever, how much "RH7.1" will it be? |
| 23:12.32 | kFuQ | rh7.1 is ancient lol |
| 23:14.07 | CoffeeIV | well, if after upgrading the kernel, make, binutils (objcopy needs to be newer), and whatever else, I don't care how 7.1 it is so long as the ancient custom app on there still works for asterisk to talk to |
| 23:14.51 | _VoiceMeUp_COM | i had a client on wiundwows 3.1 |
| 23:14.57 | _VoiceMeUp_COM | he wanted to run xten |
| 23:15.52 | neverblue2 | ok back again |
| 23:16.12 | neverblue2 | Fender, you still here? |
| 23:16.23 | *** join/#asterisk CrashSys (n=kumba@158-211.187-72.tampabay.res.rr.com) |
| 23:16.38 | *** part/#asterisk woolbeo (n=woolbeo@exchange.services.daqe.com) |
| 23:16.58 | CrashSys | Anyone know a good source for 24-pair telco cable? |
| 23:17.29 | JT | electrical/cable wholesaler? |
| 23:17.30 | lee_is_me | where can I tell asterisk to wait x seconds before picking up a ringing FXO on an analog card? |
| 23:17.52 | lee_is_me | is that just a matter of the Wait() command? |
| 23:18.07 | CrashSys | Blah... they're not open... off to teh google... |
| 23:18.13 | _VoiceMeUp_COM | wait(5) |
| 23:18.28 | lee_is_me | before calling Answer()?? |
| 23:18.42 | neverblue2 | ~thebook |
| 23:18.52 | jbot | methinks thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:18.52 | neverblue2 | ~book |
| 23:18.54 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:19.00 | neverblue2 | laggy bot |
| 23:19.15 | yannj_fr | jbot try it : www.intelunix.fr/tfot |
| 23:19.23 | neverblue2 | anyone a VOIP provider? |
| 23:19.27 | _VoiceMeUp_COM | yes |
| 23:19.30 | _VoiceMeUp_COM | how can i help ? |
| 23:19.38 | neverblue2 | business provider? |
| 23:19.43 | _VoiceMeUp_COM | and if commerical nature please pm me to keep out of this channel |
| 23:19.45 | _VoiceMeUp_COM | yes |
| 23:19.48 | neverblue2 | k, pm |
| 23:21.01 | CrashSys | Hmmm... $21 + $1.25/ft for pre-fab amp cables... |
| 23:21.28 | JT | sounds about right |
| 23:21.49 | *** join/#asterisk angom_h (n=Angel@189.178.3.55) |
| 23:21.52 | CrashSys | in plenum... |
| 23:23.21 | mitcheloc | try it in black plenum for fun |
| 23:23.33 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 23:24.03 | CrashSys | Yeah... let them think it's a CATV/Satelite wire |
| 23:27.00 | J4k3 | when deploying coax-based networking, buy duplex coax. Apply 5-10kV to one wire, and run the data down the other |
| 23:27.11 | J4k3 | so you'll know wherever the wire gets cut by the smell of burning flsh. |
| 23:27.12 | J4k3 | er flesh |
| 23:28.01 | CrashSys | Maybe i'll market that as "Armed" cable... |
| 23:28.07 | CrashSys | I'm sure someone would pay for that... |
| 23:28.20 | aptura | Crash as a kid I was messing with one of those once. I became the fence. |
| 23:28.43 | aptura | In my youth it was all about electronics ;) |
| 23:28.52 | J4k3 | well, the ~300VAC that runs the T1 repeaters coming out here definetly lets you know you're tapping a hi-cap. |
| 23:28.57 | CrashSys | I remember taking a whiz on one when I visited my grandparents... I learned the hard-way... |
| 23:29.29 | *** join/#asterisk remmo (n=junk@203.62.147.6) |
| 23:29.31 | CrashSys | Course, I didn't know it was electrified till about 2-ms into it |
| 23:29.52 | J4k3 | "until my genitals were on fire" |
| 23:29.54 | aptura | J4 man that is a bit of power. :) Crash.. I accidently touches both terminals and my legs shot out from under mee and went flying backwards 5 feet hitting the door ;) |
| 23:30.19 | CrashSys | I jumped backwards about 5 feet... |
| 23:30.37 | aptura | yea I did build a 200kv telsa coil before :) |
| 23:30.46 | lee_is_me | phantom calling problem on sangoma anlog: tried busydetect=yes, busycount=6, answeronpolarityswitch=no don't seem to fix the problem. Write using WaitForRing but only seemed to freeze the line for many seconds |
| 23:30.54 | aptura | plays absolute havic with phone lines. |
| 23:31.17 | lee_is_me | that is why i was asking about waiting before asterisk/zaptel picks up a ringing fxo line |
| 23:31.19 | CrashSys | I'm sure... tesla coils scare me |
| 23:31.41 | aptura | Some day out of fun will build one that stands 8 feet tall. |
| 23:31.49 | J4k3 | a man that spent most of his life |
| 23:31.53 | J4k3 | getting screwed |
| 23:31.56 | J4k3 | in the wlalet |
| 23:31.58 | J4k3 | er wallet |
| 23:32.00 | aptura | Down the street from me is one that stands 2 stories tall. |
| 23:32.31 | aptura | J4k3 I know pretty sad story. westinghouse put him in the poor house all right. |
| 23:32.31 | CrashSys | Why build a tesla coil when you can just take one of those old magentic VHS tape erasers and hold it up to a telco line to screw it up? |
| 23:32.39 | CrashSys | Tons safer |
| 23:33.08 | remmo | but there is no shock effect exposure |
| 23:33.13 | CrashSys | True |
| 23:33.25 | remmo | you know you really gotta put your tongue on the line to actually test if it works. |
| 23:33.26 | CrashSys | That does deminish the rush |
| 23:33.34 | remmo | otherwise its hearsay |
| 23:34.03 | aptura | Did you know any thermal nuclear device detonated probebly 10 miles in the atmosphere will kill any electronic device in most states in a geographic region? |
| 23:34.04 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 23:34.11 | CrashSys | 1000' Cat5e Plenum $716... *googles another website* |
| 23:34.32 | JT | aptura: i thought everyone knew that... |
| 23:34.52 | CrashSys | I plan to use the 1950's approach to nuclear bombs... and hide under my desk... |
| 23:34.57 | bapril | we don't need no stinkin' T-bird... "Yup that tastes like ESF" :-) |
| 23:35.04 | aptura | While it is extreemly remote that a emp attach would occour it could be done. |
| 23:35.34 | CrashSys | Did you also mention how the fall-out will spread like a rain storm as it rains down? |
| 23:35.55 | CrashSys | And that while we wont have our cell-phones our the illustrious i-pod, we probably will grow a third eye or testicle... |
| 23:36.10 | *** join/#asterisk bkw__ (n=brian@adsl-70-143-39-83.dsl.tul2ok.sbcglobal.net) |
| 23:36.53 | aptura | JT I was stationed at one of two EMP test facilities while in the airforce and the warning signs near the test facility are pretty clearly what would happen if you approached during the test phase..especially with a pace maker :) One of our HH-60s was flying by the facility and did not know it was in test mode and the aircraft was experaincing violent yaw kicks. The tail controls are both electrical and hydralic powered. |
| 23:37.38 | *** join/#asterisk benofsky (n=benofsky@86.43.88.82) |
| 23:37.48 | benofsky | what ports need to be forwarded for iax and sip? |
| 23:37.57 | aptura | Now the question is how deep can you bury telcom or any electrical gear so its not exposed to the electrostatic charges. |
| 23:38.17 | CrashSys | Aptura: Encase it in grounded metal conduit? |
| 23:38.25 | aptura | true |
| 23:38.36 | CrashSys | Sucks for subterranean tho... |
| 23:38.51 | CrashSys | or however you spell it |
| 23:39.00 | aptura | but even that may not work. Lighting once hit the ground and went up the ground wire frying some radio transmitters once. |
| 23:39.09 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 23:39.09 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:42.04 | JT | CrashSys: that doesn't work. |
| 23:42.04 | lee_is_me | I don't think a phantom ring can be avoided in my a customer's case because of the way their system is setup... |
| 23:42.04 | CrashSys | jt: Metal will not block EMP? |
| 23:42.04 | aptura | it will |
| 23:42.16 | lee_is_me | they are having a group of devices ring as soon as a call comes in from the telco |
| 23:42.16 | JT | CrashSys: only if done right |
| 23:42.19 | aptura | electrical shilding will |
| 23:42.35 | aptura | second emp test facility is at white sands missle range. |
| 23:42.45 | lee_is_me | is that just more or is there no way to avoid a "phantom" ring if a calls starts to call, but then hangs up? |
| 23:42.48 | JT | CrashSys: problem is wires etc going in and out of metal case |
| 23:43.01 | lee_is_me | lol, that should have been "is that just me..." |
| 23:43.13 | aptura | a very small one that is used for small objects. Ours is 50% larger then the largest cargo transport aircraft. |
| 23:43.19 | CrashSys | Well yeah... but I was thinking more like how you would run secure lines... |
| 23:43.20 | JT | eg in lightning, there are 3 forms of discharge transfer, direct hit, capacitive coupling, and inductive coupling |
| 23:43.35 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 23:43.46 | aptura | heat lightning |
| 23:43.52 | aptura | ever hear of that one? |
| 23:44.05 | CrashSys | lee: The way I handled phantom calls it to start a channel-timeout at start of call until call leaves the automated attendant... |
| 23:44.28 | JT | wires are usually caught by capacitive coupling if they're underground |
| 23:44.38 | aptura | no clouds at all in this case. Two layers of air that have a potential difference in ionic charge minus the visable cloud vapor. |
| 23:44.40 | CrashSys | that way i'm only dealing with a phantom call for 2-mins tops... |
| 23:45.04 | lee_is_me | CrashSys: that's the problem. This customer has the phones ring directly to devices when the call comes in. No IVR <--- in between ---> |
| 23:45.07 | aptura | Crash thats happened to me before. |
| 23:45.26 | lee_is_me | CrashSys: only IVR if they don't pick up on the attempt group ring. |
| 23:45.32 | aptura | phantom LCD light up on pstn phone. |
| 23:45.42 | lee_is_me | i don't think it can be avoided personally. |
| 23:45.45 | aptura | would not ring. |
| 23:46.00 | CrashSys | Hmmmm |
| 23:46.14 | lee_is_me | did I make sense? |
| 23:46.19 | CrashSys | Isn't there silence detection in asterisk? |
| 23:46.20 | aptura | I ignore it. the phone would light up for 2 seconds then go out. |
| 23:47.21 | tdonahue-laptop | CrashSys, there was work toward it, i don't know that it is completed at this point |
| 23:53.11 | CrashSys | With my old digium te205p using RBS I would have a problem where someone would call and hang-up right as asterisk answered and the hang-up wouldn't be detected, and the IVR would go on forever and the line was technically idle... |
| 23:53.11 | lee_is_me | CrashSys: That appears to be the problem with this site |
| 23:53.11 | lee_is_me | except that they are having a group of phone ring immediately so its annoying. |
| 23:53.11 | lee_is_me | lol |
| 23:53.11 | CrashSys | I handled it by adding a time-out to the incoming call and removing it when they left the IVR... |
| 23:53.11 | *** join/#asterisk HeMan (n=jimmy@1-1-7-40a.far.sth.bostream.se) |
| 23:53.11 | CrashSys | That way the line could only be dead for 2-mins tops... |
| 23:53.11 | CrashSys | What are they doing? Faxes? |
| 23:53.11 | lee_is_me | no just small company that want the phone to ring directly before IVR |
| 23:53.12 | lee_is_me | go to IVR if noone picks up on the intial group ring. |
| 23:53.12 | lee_is_me | 1. Ring Group 2. Ether someone answers or go to IVR |
| 23:53.12 | JT | CrashSys: silence detection in what sense? |
| 23:53.12 | CrashSys | jt: to detect a hangup/phantom channel... |
| 23:53.12 | CrashSys | like if you have 60-seconds of silence, hangup... |
| 23:53.12 | *** part/#asterisk putnopvut (i=putnopvu@nat/digium/x-fe74fe70049e094f) |
| 23:53.12 | CrashSys | Lee: If they pick up the line that has the phantom call, hear nothing, and hang-up... does asterisk not hang-up the channel? |
| 23:53.12 | JT | well that's a bad way to detect hangups |
| 23:53.12 | lee_is_me | no |
| 23:53.12 | lee_is_me | sometimes it rings back again |
| 23:53.12 | JT | a perfectly legitimate call could be silent for 60 seconds |
| 23:53.12 | lee_is_me | most times its just once though |
| 23:53.13 | *** join/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 23:53.13 | CrashSys | jt: Well silence detection would be good for answering machine's... |
| 23:53.13 | *** part/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 23:53.22 | CrashSys | just in general it has it's applicable uses... |
| 23:53.22 | JT | asterisk isn't an answering machine though? |
| 23:53.37 | CrashSys | But if you were doing outbound dialing it would be useful... |
| 23:53.41 | lee_is_me | I tried WaitForRing but that just froze the channel for a while (way longer than the 2 second param I supplied) |
| 23:54.06 | JT | what would be more useful is disconnect supervision or digital signalling, imho |
| 23:54.24 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 23:54.35 | *** join/#asterisk saftsack (n=saftsack@pD9E0440E.dip0.t-ipconnect.de) |
| 23:54.42 | CrashSys | jt: Silence detection would be nice if i'm using an auto-dialer at a vetinary hospital to call people and let them know that fluffy is due for his shots... |
| 23:55.55 | CrashSys | It would be useful... having a "waitforsilence(<seconds>)" command in the dial-plan... |
| 23:56.26 | CrashSys | Probably over-complicated tho |
| 23:57.03 | CrashSys | There's a couple AMD things floating around already tho... |
| 23:57.21 | lee_is_me | Hmmm. I just tried using WaitForRing() BEFORE any other commands such as Answer() and that seemed to work |
| 23:57.28 | lee_is_me | (dialed into customer site now) |
| 23:58.31 | lee_is_me | I did kick out this error: Jun 7 20:06:00 WARNING[25526]: chan_zap.c:6309 ss_thread: CallerID returned with error on channel 'Zap/3-1' |
| 23:58.32 | lee_is_me | <PROTECTED> |
| 23:58.51 | CrashSys | Either you dont have caller ID or have the wrong settings... |
| 23:59.46 | lee_is_me | Well, I've been changing things around a bit trying to track this down. I'll put the original files back in place and see if that fixes it as I was not having that error before I started "fixing" it. |
| 23:59.48 | *** join/#asterisk john-eman0n (n=eman0n@cpe-76-181-123-158.columbus.res.rr.com) |
