| 00:03.27 | scooby2 | do I need duplicate settings for everything in zapata.conf if I have two T1's? |
| 00:04.43 | ZX81 | no, settings fall through |
| 00:04.59 | ZX81 | unless you've changed them in between |
| 00:06.56 | l2trace99 | anyone know how I can bind sip in asterisk to 2 out of 3 ips ? |
| 00:07.08 | scooby2 | ZX81: thx |
| 00:07.19 | ZX81 | l2trace99 you can't |
| 00:07.25 | ZX81 | scooby2 no probs |
| 00:07.49 | l2trace99 | thats what I though |
| 00:15.19 | *** join/#asterisk mihinomenest (n=argh@66.255.220.17) |
| 00:20.46 | aliver | I'd be eternally grateful if someone could look at http://pastebin.com/d3ce87279 and tell me all the ways I'm fucked up. |
| 00:21.03 | aliver | I'm having trouble figuring out where to use +1 and not in my dial plan I think. |
| 00:21.09 | aliver | I've got my sip.conf and extensions.conf |
| 00:21.42 | irieKen | This polycom problem is going to make me rip my hair out! |
| 00:21.50 | aliver | When I call the DID, I get this message from *: http://pastebin.com/d6834d730 |
| 00:22.53 | aliver | It basically complains that it can't find the SIP phone (404 not found) because it isn't in extensions.conf. Well, yah, but it looks like it should find it to me according to what I've read. However, I obviously not doing this shit right. |
| 00:25.50 | _ShrikE | aliver: Looking for +19707442251 in internalphones |
| 00:26.01 | aliver | yep. |
| 00:26.21 | aliver | but shouldn't that stuff in extensions.conf tack on the +1 ? |
| 00:26.23 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 00:28.07 | _ShrikE | You still dont have an +19707442251 extension in internalphones |
| 00:30.01 | aliver | but do I have to? Shouldn't I need 9707442251 instead since that line 98 and line 99 of that first pastebin (my extensions.conf) should add it on? |
| 00:30.09 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 00:30.14 | aliver | add on the "+1" I mean |
| 00:30.31 | [TK]D-Fender | aliver: WRONG CONTEXT |
| 00:30.45 | aliver | BTW, you are totally right, because when I change the phone to register with +19707442251 it works fine. |
| 00:31.07 | _ShrikE | the call is coming to internalphones, not frombandwidth |
| 00:31.41 | aliver | fender, I can see that you are right. I just wonder why the bandwidth.com docs say to have that.. |
| 00:31.49 | aliver | I'm probably just making this way to complicated. |
| 00:32.26 | aliver | I guess the bottom line is screw them. Go with what works. |
| 00:32.35 | [TK]D-Fender | aliver: You are. You probably don't need the 2-peer setup you have there. |
| 00:32.35 | aliver | Their docs suck anyhow. |
| 00:33.13 | aliver | I hope I don't, it's confusing the living shit outta me not to mention being broken. |
| 00:34.37 | aliver | They have this crap in the docs, and I'm wondering exactly what it's meant to do: |
| 00:34.40 | aliver | [internalphones] |
| 00:34.41 | aliver | exten => _1NXXNXXXXXX,1,Dial(SIP/+${EXTEN}@bandwidth.com_outbound,60) |
| 00:35.21 | lunaphyte | i'm trying to troubleshoot connection problems sending calls to a provider - i get these messages with a verbosity of 14: http://rafb.net/p/PLTGVh83.html |
| 00:36.14 | aliver | is that saying "anybody who dials a phone number, use the sip.conf context called "bandwidth.com_outbound" and our Extension as the originating DID and dial it" ??? |
| 00:36.39 | lunaphyte | what in my config could cause a provider to return circuit-busy? |
| 00:39.48 | *** join/#asterisk ryoohki (n=ryoohki@208.96.15.252) |
| 00:41.13 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 00:41.17 | *** join/#asterisk bkw_ (n=bkw_@freeswitch/developer/bkw) |
| 00:41.26 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:41.26 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:44.59 | _ShrikE | aliver: It means that within the [internalphone] context, any number matching the pattern 1NXXNXXXXXX will have a + added and will be sent to the peer bandwidth.com_inbound as configured in your sip.conf and will ring for 60 seconds. |
| 00:45.37 | aliver | ah thanks for the translation! |
| 00:45.53 | aliver | What is the difference between the "N" and the "X" chars in the pattern? |
| 00:47.04 | _ShrikE | N= 2-9 X= any number |
| 00:47.35 | aliver | wierd |
| 00:47.38 | aliver | but okay. |
| 00:47.47 | [TK]D-Fender | aliver: its a standard. |
| 00:47.50 | _ShrikE | pretty standard telecom stuff |
| 00:48.01 | aliver | I believe. I'm your standard igmo, so don't mind the looks of astonishment. |
| 00:48.27 | mosty | is there a good way to do multi tenant parking with asterisk? |
| 00:50.48 | aliver | Okay, another brilliant question. I have the phone ringing on the DID. However, I have half-duplex audio. I can't hear the outside caller's audio, but they can hear my SIP phone's audio. I know for sure that my firewall is letting the UDP SIP traffic come into my asterisk box from the outside because I nmap'd it from the outside and saw that UDP port was open. I'm using full 1-1 NAT for this server (pf using "binat" rule). I'm binding the a |
| 00:51.33 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:51.38 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-5a50d335daded51e) |
| 00:52.48 | _ShrikE | ~sipnat |
| 00:52.49 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 00:53.39 | JT | aliver: don't write so long sentences, they get chopped off :) |
| 00:53.56 | JT | the sip port does not carry any audio traffic. |
| 00:54.05 | [TK]D-Fender | mosty: app_supervaletparking |
| 00:54.20 | aliver | ack, sorry. |
| 00:54.27 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:55.40 | aliver | I'm getting confused with the nomenclature. I've got a asterisk box with a bunch of sip phones on the same subnet, I NAT outside to the bandwidth.com guys who have our DIDs and forward traffic to my outside (real-IP non-nat) IP which get's natted to me on my internal address. |
| 00:55.45 | [TK]D-Fender | aliver: pastebin a failed call with SIP debug. |
| 00:55.50 | aliver | Now, who is the proxy, who is the server, and who are the clients? |
| 00:56.01 | aliver | Will do. |
| 00:56.17 | mosty | [TK]D-Fender, do you know if that is for asterisk 1.2 or 1.4? |
| 00:56.32 | [TK]D-Fender | mosty: Releases for both |
| 00:56.49 | *** join/#asterisk hi365_m (n=hi365@213.151.56.78) |
| 00:57.15 | mosty | [TK]D-Fender, ok i must be looking at the wrong place, the link from the wiki only has one file |
| 00:58.31 | mosty | bkw_, where can i find your supervaletparking code? |
| 00:59.15 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 01:00.23 | *** join/#asterisk l2cache (n=l2cache@155.132.204.68.cfl.res.rr.com) |
| 01:02.23 | *** join/#asterisk korihor (n=korihor@190.78.32.60) |
| 01:06.53 | bkw_ | mosty: I didn't write it... anthm did.. I just cheer lead for it :P |
| 01:08.37 | mosty | oh ok |
| 01:09.22 | *** join/#asterisk salmander (n=omi@CPE0002b323839e-CM001225024984.cpe.net.cable.rogers.com) |
| 01:09.51 | salmander | hello, I have an asterisk server that sends 00000000 as caller ID. Where can I change the setting to send a string that I want? |
| 01:12.39 | [TK]D-Fender | salmander: In your dialplan |
| 01:13.38 | jaytee | salamander, use Set(CALLERID(num)="somenumber" |
| 01:13.46 | salmander | is that in extensions? |
| 01:13.53 | [TK]D-Fender | salmander: Yes. |
| 01:13.59 | jaytee | yep and I left off a ) at the end |
| 01:14.08 | salmander | http://www.voip-info.org/wiki/view/Setting+Callerid |
| 01:14.10 | salmander | like so? |
| 01:14.10 | lunaphyte | i'm trying to troubleshoot connection problems sending calls to a provider - i get these messages with a verbosity of 14: http://rafb.net/p/PLTGVh83.html |
| 01:14.14 | salmander | it just didn't say where to put it. |
| 01:14.31 | [TK]D-Fender | salmander: anywhere BEFORE the dial is placed would be nice. |
| 01:14.46 | salmander | I figure that much :) |
| 01:16.11 | salmander | I have 2 phones |
| 01:16.19 | salmander | and I have 2 DIDs that my server receives |
| 01:16.32 | mosty | bkw_, i can see app_valetparking on freeswitch.org, but where can i find app_supervaletparking ? |
| 01:16.45 | bkw_ | someone else did supervaletparking |
| 01:16.46 | salmander | how can I map outgoing calls from one extension to have one callerID number and the second phone to have another number? |
| 01:17.01 | russellb | and in Asterisk 1.6, the parking stuff that is included should do what you want |
| 01:17.04 | *** join/#asterisk ez` (n=ez@c66.203.221-242.clta.globetrotter.net) |
| 01:17.13 | lunaphyte | how can i further troubleshot why i'm getting a circuit-busy after the call is accepted? |
| 01:17.17 | mosty | bkw_, ah ok, i might change the wiki then, it says it's from you |
| 01:17.20 | lunaphyte | *troubleshoot |
| 01:17.32 | bkw_ | weird |
| 01:17.38 | salmander | In this page http://www.voip-info.org/wiki/view/Setting+Callerid is /2100 the number of the extension doing the dialing? |
| 01:17.39 | jaytee | salmander, you can set your preferred callerid for each SIP phone in SIP.CONF |
| 01:18.03 | jaytee | for each SIP account |
| 01:18.16 | salmander | I do |
| 01:18.31 | salmander | whenever I call an outgoing phone number they get 000000 |
| 01:18.37 | mosty | russellb, i am trying to fix an ancient asterisk 1.2 svn box that has an out of tree call parking patch which dies randomly, i'm looking for a more minimal change than 1.6 unfortunately |
| 01:19.11 | jaytee | add callerid="thenumberyouwant" to each SIP account in SIP.CONF and do a sip reload then test |
| 01:19.27 | russellb | mosty: fair enough :) |
| 01:19.28 | salmander | I tried restarting the server |
| 01:19.43 | russellb | just throwing it out there while on the topic, i guess |
| 01:19.56 | salmander | when I call from one extension to the other the caller id shows up |
| 01:20.04 | salmander | but when I call outgoing numbers, they get zeros |
| 01:20.25 | mosty | i think my best bet is a replacement for the standard call parking for 1.2 |
| 01:20.37 | russellb | nods |
| 01:21.05 | jaytee | salmander, how are your calls going out? POTS? PRI? or SIP to an ITSP? |
| 01:23.31 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 01:23.50 | salmander | IAX |
| 01:26.39 | russellb | You should be able to do an IAX debug of an outgoing call to see what callerid is being sent |
| 01:27.14 | russellb | It will be in the first packet sent to initiate the call (NEW) |
| 01:27.38 | salmander | russellb I should capture packets? |
| 01:27.41 | russellb | you don't have to |
| 01:27.44 | salmander | or there's an asterisk tool for that? |
| 01:27.47 | russellb | "Iax2 set debug on" or something |
| 01:27.51 | russellb | from the Asterisk CLI, yeah |
| 01:28.33 | *** join/#asterisk MaliutaLap (n=nikolai@kiev.lusan.id.au) |
| 01:28.41 | salmander | caller ID is set |
| 01:28.45 | russellb | being sent properly? |
| 01:28.50 | salmander | but should it be set as a number or can it be sent as a string? |
| 01:29.02 | salmander | Fname Lname <number> |
| 01:29.07 | lunaphyte | i'm getting a circuit-busy on calls passed to my provider. what else can i do on my side to determine what might be happening before i contact my provider? |
| 01:29.14 | russellb | It's fine like that |
| 01:29.22 | russellb | So then you need to talk to your provider ... |
| 01:29.40 | salmander | I'll try to switch my server to sip |
| 01:29.51 | russellb | well, it's not a protocol issue |
| 01:30.07 | salmander | when I talked to their tech guy, he said that I should switch to SIP from IAX |
| 01:30.17 | russellb | that doesn't make any sense, but ok ... |
| 01:30.23 | salmander | I thought so too. |
| 01:30.36 | russellb | tell him "how about you fix your system" |
| 01:30.45 | russellb | or switch to SIP, whatever works for you |
| 01:30.46 | russellb | heh |
| 01:30.52 | russellb | is just partial to IAX |
| 01:31.51 | jaytee | is partial to IAX2 |
| 01:32.04 | jaytee | :-) |
| 01:33.31 | russellb | heh, yes, I meant IAX2 |
| 01:33.41 | russellb | but IAX == IAX2 at this point ... |
| 01:35.34 | jaytee | russellb, first off you should know that I'm a chronic wiseass and not to take anything I say too seriously. secondly, I'm still quite the "noob" imho and would never argue or contradict an asterisk developer. |
| 01:35.44 | Nugget | IAX is passé. chan_skype is where it's at. |
| 01:35.49 | jaytee | LOL |
| 01:36.11 | russellb | jaytee: heh, you made a correct point, though :) |
| 01:36.43 | jaytee | whatever happened to that chan_flash2Skype2Sip some guy claimed to have invented in here one weekend but would never show us the code. |
| 01:36.55 | russellb | lol |
| 01:37.20 | jaytee | drmessano and I were getting alot of laughs out of messing with the guy |
| 01:37.20 | russellb | He invented it in his mind |
| 01:40.10 | jaytee | speaking of minds, has anyone seen mine around? I seem to have lost it. :-) |
| 01:40.56 | *** join/#asterisk drdrain (n=drdrain@cpe-066-057-105-080.nc.res.rr.com) |
| 01:42.45 | drdrain | Is there a way to make AMI refuse non ssl encrypted requests? |
| 01:42.56 | pcrane | jaytee over there? ---> |
| 01:43.21 | jaytee | ooooh!! |
| 01:43.43 | russellb | drdrain: Interesting question. I suppose you could do it by setting the regular bindaddress to localhost |
| 01:43.55 | drdrain | Well thats the thing |
| 01:43.58 | russellb | And only set the public bind address for the SSL connection |
| 01:44.23 | drdrain | Hummmm |
| 01:44.49 | russellb | set bindaddr=127.0.0.1, and sslbindaddr=0.0.0.0 |
| 01:45.29 | drdrain | What I want to do is run a click2call php script on my webserver that controls a call back on my * box |
| 01:45.56 | russellb | In that case, you may want to use the Manager over HTTP interface |
| 01:46.07 | drdrain | Of course I don't want to expose the AMI port in the clear |
| 01:46.24 | JT | salmander: makes sense to me, providers hate IAX |
| 01:47.04 | jaytee | buncha stinking haters! |
| 01:47.17 | drdrain | russellb: Any good docs on AMI over HTTP? |
| 01:48.17 | russellb | drdrain: Hm. There is a little bit in the O'Reilly book, in the section about the Asterisk-GUI |
| 01:48.30 | russellb | drdrain: The Asterisk GUI is entirely built on that interface, using javascript. |
| 01:48.47 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 01:49.01 | jaytee | I'm getting so sick of these ads with pictures of kids with severe hairlips |
| 01:49.05 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 01:50.29 | drdrain | russellb: Thx for the help. I'll go off and see what I can figure out. |
| 01:50.36 | aliver | which is better g726 or ilbc? |
| 01:50.57 | russellb | drdrain: You're welcome. I wish I had more info on that interface for you. I'm thinking about writing a tutorial ... but of course, it doesn't exist yet |
| 01:51.40 | russellb | but yeah, check out the book ... In the Asterisk-GUI chapter, there is a section "Developing for the Asterisk GUI" which should provide some insight |
| 01:52.08 | russellb | And also, see the appendix at the end of the book which covers all of the available manager actions. All of those are available over HTTP via the methods described in the GUI chapter .. |
| 01:52.15 | jeev | mr russel! |
| 01:52.27 | jeev | pump, pump the jam, pump it up |
| 01:52.31 | jaytee | oh ffs |
| 01:52.48 | *** join/#asterisk alephcom (n=Darren@host234117.airenet.com) |
| 01:52.49 | russellb | plumps it up |
| 01:52.52 | russellb | wait, that's not the same |
| 01:53.11 | jeev | lol |
| 01:54.42 | russellb | drdrain: last thing. For your application, you will be specifically interested in the "originate" manager action. |
| 01:55.39 | drdrain | russellb: Actually I have a working script that just uses the clear text interface |
| 01:55.47 | russellb | Cool, that works, too :) |
| 01:56.06 | drdrain | The thing is is is designed to run on the PBX box (local). |
| 01:56.24 | russellb | Ah. Well it should work just as well remotely |
| 01:56.31 | drdrain | I want to put it out on my webserver |
| 01:56.51 | drdrain | Yeah bit I'd have to expose the AMI to the 'net |
| 01:56.59 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 01:57.20 | drdrain | That can't be too secure right? |
| 01:57.28 | russellb | Yeah, you don't want to expose the raw interface ... |
| 01:57.29 | aliver | what does the underscore "_" mean as a wildcard in a number in extensions.conf? "any number of numbers" ? |
| 01:57.33 | russellb | either expose the SSL interface only |
| 01:57.35 | Sargun | Asterisk is secure. |
| 01:57.44 | russellb | or use astmanproxy to expose an SSL interface |
| 01:57.56 | russellb | aliver: _ just means you're specifying a pattern |
| 01:57.57 | mosty | aliver, it means "treat the following as a pattern, not a literal extension" |
| 01:57.58 | *** join/#asterisk jameswf-home (n=james@66.18.113.20) |
| 01:58.13 | aliver | Ahh, gotcha. |
| 01:58.15 | aliver | thanks. |
| 01:58.34 | drdrain | russellb: Right. Thats what I had in mind |
| 01:58.40 | russellb | cool. |
| 01:58.49 | drdrain | Just not clear how to make it happen |
| 01:59.03 | drdrain | Config-wise |
| 01:59.50 | russellb | what Asterisk version are you using? |
| 01:59.59 | drdrain | 1.4 |
| 02:00.23 | jaytee | aliver, page 137 and 138 of the book |
| 02:00.25 | russellb | Ok, the Manager SSL interface is only in 1.6. You'll have to use astmanproxy |
| 02:00.27 | jaytee | ~book |
| 02:00.28 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 02:00.43 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 02:00.53 | aliver | *nod* okay, duly noted and I'm trying to grab the book now. |
| 02:00.55 | *** join/#asterisk fury_ (i=fury@ool-44c2f380.dyn.optonline.net) |
| 02:01.00 | drdrain | Really, I thought the ssl stuff was added to the 1.4 tree |
| 02:01.01 | *** join/#asterisk `Sean (i=Un1x@CPE001d451b875f-CM00111ade88b6.cpe.net.cable.rogers.com) |
| 02:01.03 | aliver | ahh.. html |
| 02:01.13 | jaytee | you can download the pdf from the other link |
| 02:01.17 | jeev | russellb |
| 02:01.26 | russellb | drdrain: nope, sorry .. |
| 02:01.26 | jeev | as i told lmadsen, he didn't give me credit for writing the book with him |
| 02:01.28 | russellb | jeev: |
| 02:01.32 | jameswf-home | ~wrfiab |
| 02:01.32 | russellb | shakes his head |
| 02:01.33 | jeev | would you like to sign my petition t oget my credit ? |
| 02:01.36 | jameswf-home | ~wtfiab |
| 02:02.02 | jaytee | petition? |
| 02:02.12 | russellb | jeev: sorry, but you've got nothing on lmadsen :-p |
| 02:02.22 | jeev | lol |
| 02:02.42 | jameswf-home | Georga is at war go Hoyas |
| 02:02.49 | [TK]D-Fender | jeev: Sorry, your sub-prime contribution has gone into default. |
| 02:02.54 | fury_ | I have extensions _NXXNXXXXXX, 0, and _.! in my dial plan. I want (a) the extension 0 to be triggered immediately on pressing 0 if no other digits have been pressed, (b) _NXXNXXXXXX to be triggered upon dialing exactly ten digits that match that expression, (c) and _.! to be triggered within two seconds of no input from the user. Right now this all works, except for that (a) and (c) take about 5 seconds of idle time from the user on the pho |
| 02:03.25 | Strom_M | fury_: take out your _.! |
| 02:03.41 | fury_ | what will happen if there is an error? |
| 02:03.42 | Strom_M | you want a timeout extension instead |
| 02:03.44 | drdrain | russellb: Ok. Thx again for the advice. L8R |
| 02:03.49 | russellb | drdrain: sure, cya |
| 02:03.49 | Strom_M | look up the asterisk special extensions |
| 02:03.59 | fury_ | ya, I read somewhere that it's not good to use _.! |
| 02:04.17 | jameswf-home | I like exten => 911,1,congestion |
| 02:04.26 | fury_ | right now I have Set(TIMEOUT(digit)=2) before WaitExten |
| 02:04.28 | fury_ | is that correct? |
| 02:04.44 | russellb | jameswf-home: terrible advice :-p |
| 02:05.00 | Strom_M | fury_: no |
| 02:05.04 | Strom_M | look up the asterisk special extensions |
| 02:05.13 | jameswf-home | russellb: would playback(tt-allbusy) be better? |
| 02:06.14 | jeev | sub prime contribution |
| 02:06.19 | jeev | what the hell do i look like? countrywide? |
| 02:06.33 | Strom_M | jeev: yes |
| 02:06.44 | jeev | mr. strom! |
| 02:06.49 | jameswf-home | jeev: no you dont have that mutch $$ |
| 02:07.00 | jeev | eh, not as much as a bank. |
| 02:07.26 | russellb | mutch? what is that, a breed of dog? :-p |
| 02:07.29 | jameswf-home | I saw an ad tht said make CEO income.... I was thinking yes but ceo of what company... |
| 02:07.34 | jaytee | countrywide? no, I thought you looked more like an ad for the dangers of huffing paint |
| 02:09.04 | jameswf-home | I saw an intervention where the lady inhaled computer duster.... the computer duster we use has this crap in it that taste horrible... we spray it on peoples cups and keyboards pretty much last the whole day no matter what you do.. |
| 02:09.05 | jeev | lol jaytee |
| 02:09.22 | jeev | jameswf-home, you must hate your job! |
| 02:09.25 | *** part/#asterisk Paige (n=Paige@2001:470:1f05:531:21f:c6ff:fe48:8988) |
| 02:09.41 | jameswf-home | love my job..... horseplay makes the world a better place |
| 02:09.54 | pcrane | needs some of that stuff jameswf-home |
| 02:10.30 | jeev | can of air is a rip off, i bought a compressor for my office so the bitches stop wasting my money |
| 02:10.53 | jameswf-home | I get paid to work on linux, realize i use to do it free... my job rocks |
| 02:11.14 | jeev | eh, i run my own business.. the only thing that sucks is that i'm lazy. |
| 02:11.23 | jaytee | one of my co-workers switched a bunch of keycaps on my keyboard. It was probably 3 months before I even noticed it. |
| 02:11.43 | jeev | lol |
| 02:12.08 | scooby2 | trying to move from a very old Sangoma with no echo cancellation to a new Digium but when I call the new digium the voice recording is chopped off. By the time i hear anything it is 5-8 seconds into the greeting message. |
| 02:12.46 | jameswf-home | lay off the scobby snacks |
| 02:12.49 | scooby2 | lol |
| 02:13.30 | scooby2 | adding Wait()'s doesnt seem to affect anything |
| 02:14.09 | jeev | haha this gay guy called my black friend a scooby snack |
| 02:14.14 | jameswf-home | I hear sangona "just works" |
| 02:14.38 | jameswf-home | s/sangona/sangoma/ |
| 02:14.46 | scooby2 | i want to kill our "asterisk consultant" for having us buy this Digium |
| 02:14.53 | jameswf-home | but their beer sucks |
| 02:14.59 | russellb | scooby2: Please contact Digium tech support. |
| 02:15.12 | russellb | scooby2: However, this doesn't sound like it has anything to do with hardware ... |
| 02:15.35 | scooby2 | if I put the sangoma in the same machine it works perfectly |
| 02:15.37 | russellb | scooby2: Do you have an Answer() in the dialplan before the Wait() or before going into voicemail? |
| 02:15.44 | jameswf-home | scooby setup a test context that answers and playsback... will probably work |
| 02:15.51 | [TK]D-Fender | scooby2: pastebin a failed call at verbose 10, include your zapata.conf and zaptel.conf |
| 02:15.53 | [TK]D-Fender | ~pb |
| 02:15.53 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 02:15.55 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 02:15.55 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 02:16.17 | [TK]D-Fender | scooby2: And tell us what model the card is. |
| 02:16.19 | jaytee | I think his echo cancellation settings are set wrong |
| 02:16.24 | jaytee | for the new card |
| 02:16.34 | scooby2 | [TK]D-Fender: one sec |
| 02:16.41 | russellb | jaytee: good call |
| 02:16.45 | jameswf-home | oh that could be... set echotraining=0 |
| 02:16.46 | jaytee | set echotraining=no |
| 02:17.01 | jameswf-home | jinx |
| 02:17.02 | [TK]D-Fender | trust NOTHING until the pastebin arrives and assume the worse till it does. |
| 02:17.06 | jameswf-home | sorta |
| 02:17.12 | jaytee | [TK]D-Fender, good point |
| 02:17.19 | scooby2 | jaytee: set that in console? |
| 02:17.29 | [TK]D-Fender | scooby2: "set verbose 10" |
| 02:17.39 | jaytee | scooby2, no it's in the zapata.conf |
| 02:17.39 | [TK]D-Fender | scooby2: just move along with the pastebin. |
| 02:17.46 | russellb | scooby2: Make sure you have an Answer() in the dialplan, and check your echotraining option ... |
| 02:17.49 | [TK]D-Fender | scooby2: don't mind the chatter here |
| 02:17.49 | jeev | there's a recall on katana's. |
| 02:17.55 | jaytee | scooby2, do what [TK]D-Fender said first |
| 02:17.59 | russellb | smacks [TK]D-Fender |
| 02:18.02 | russellb | let us speculate!! |
| 02:18.25 | [TK]D-Fender | russellb : you can speculate all you want, just don't slow him down. |
| 02:18.47 | scooby2 | http://pastebin.com/d791f8a51 |
| 02:19.27 | [TK]D-Fender | scooby2: First, you have no switchtype selected. BAD |
| 02:19.56 | [TK]D-Fender | scooby2: Next you set BOTH ports as the primary timing source |
| 02:20.09 | [TK]D-Fender | scooby2: span=1,1,0,esf,b8zs -- span=2,1,0,esf,b8zs |
| 02:20.21 | [TK]D-Fender | scooby2: typically your 2nd should look like : span=2,2,0,esf,b8zs |
| 02:20.40 | jameswf-home | [TK]D-Fender: always gotta be right |
| 02:20.45 | jameswf-home | jerk |
| 02:20.59 | [TK]D-Fender | jameswf-home: its a rough life. |
| 02:21.04 | pcrane | hook flash timeout, how do I find out what it is? |
| 02:21.13 | scooby2 | ok, those things fixed |
| 02:21.25 | [TK]D-Fender | scooby2: stop *. ztcfg -vvvv, then restart * |
| 02:22.02 | [TK]D-Fender | scooby2: then test. And when you do this test please include the pastebin of the failed call (like I asked the first time) with PRI debug enabled |
| 02:22.33 | [TK]D-Fender | pcrane: Hook flash on what exactly? |
| 02:22.45 | pcrane | analouge phone |
| 02:22.47 | jameswf-home | ~wikis |
| 02:22.47 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 02:22.56 | [TK]D-Fender | pcrane: ATTACHED TO............. |
| 02:23.09 | pcrane | asterisk |
| 02:23.14 | jameswf-home | :)) |
| 02:23.18 | jaytee | groans |
| 02:23.22 | [TK]D-Fender | pcrane: Asterisk is a piece of SOFTWARE |
| 02:23.27 | jaytee | it's gonna be a long nite |
| 02:23.28 | pcrane | heh |
| 02:23.29 | pcrane | sorry |
| 02:23.34 | [TK]D-Fender | pcrane: You can't plug a wire into SOFTWARE |
| 02:23.36 | pcrane | is miles away |
| 02:23.42 | [TK]D-Fender | ~cluebat pcrane |
| 02:23.42 | jbot | ACTION pulls out a ClueBat (tm) and thwaps pcrane. |
| 02:23.47 | pcrane | ouch |
| 02:23.54 | jaytee | hahahaa |
| 02:23.59 | jaytee | never saw that one before |
| 02:24.07 | pcrane | i didn't see it coming either |
| 02:24.10 | jaytee | awww, russellb quit |
| 02:24.16 | pcrane | nvm... it's ok |
| 02:24.27 | jameswf-home | SELECT * FROM #asterisk WHERE clue > 0; |
| 02:24.37 | pcrane | zero rows returned |
| 02:24.50 | scooby2 | [TK]D-Fender: http://pastebin.com/m5f730026 |
| 02:25.22 | scooby2 | its not a failed call persay it is just skipping part of the zz_closed_msg |
| 02:25.56 | Strom_M | ENGLISH LESSON TIME |
| 02:25.58 | Strom_M | "per se" |
| 02:25.59 | Strom_M | THANK YOU |
| 02:26.07 | scooby2 | sorry |
| 02:26.09 | Strom_M | goes back to doing other things |
| 02:26.12 | jameswf-home | scooby2: you fixed echotraining |
| 02:26.14 | [TK]D-Fender | scooby2: New configs please. Also please make the first line of your dilaplan an "Answer" |
| 02:26.43 | [TK]D-Fender | Strom_M: .... that's LATIN :p |
| 02:26.49 | [TK]D-Fender | pwned |
| 02:27.07 | jaytee | Strom reminds me of my 7th grade English teacher that made me stay after school and write all the linking verbs on the blackboard a 100 times. |
| 02:27.14 | Strom_M | GARDEZ-MOI DANS LE REFRIGERATEUR |
| 02:27.16 | [TK]D-Fender | jameswf-home: See? A constrant struggle! |
| 02:27.26 | jameswf-home | http://en.wikipedia.org/wiki/Per_se |
| 02:27.52 | aliver | I'm having trouble understanding when STUN is going to be useful. |
| 02:27.53 | jaytee | appear, become, feel, grow, look, remain, seem, smell, sound, taste and all forms of the verb "to be" |
| 02:28.06 | aliver | Is that when I have NAT between the SIP device (phone) and asterisk? |
| 02:28.21 | [TK]D-Fender | Strom_M: Reste froind mon ostie :p |
| 02:28.22 | jaytee | and that was 41 years ago |
| 02:28.26 | [TK]D-Fender | froid* |
| 02:28.43 | [TK]D-Fender | aliver: Considering * doesn't support STUN.... probably not. |
| 02:29.23 | scooby2 | well my consultant is an idiot. Answer() fixes the problem completely |
| 02:29.24 | [TK]D-Fender | is still trying to find out how to set his laser printers at work on "kill". |
| 02:29.32 | [TK]D-Fender | scooby2: You're welcome. |
| 02:29.41 | scooby2 | thank you [TK]D-Fender |
| 02:29.46 | scooby2 | and everyone else |
| 02:30.08 | [TK]D-Fender | scooby2: For future reference, my rates are very accessible ;) |
| 02:30.23 | scooby2 | hook me up with an email or a webpage. :) |
| 02:30.33 | *** join/#asterisk macros73 (n=cs@c-67-163-224-69.hsd1.pa.comcast.net) |
| 02:30.34 | aliver | [TK]D-Fender, hmm. okay. I'm just wondering what category I fit into on the voip-info's NAT problem solving page. |
| 02:30.37 | jameswf-home | Pouvoir j'a s'il vous plaît le sexe avec votre femme. Je ne soigne pas qu'elle ait l'air d'une chèvre |
| 02:30.46 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 02:30.47 | aliver | I can't get their nomenclature "proxy", "server", and "client". |
| 02:31.09 | [TK]D-Fender | aliver: * got a WAN IP interface? |
| 02:31.10 | scooby2 | cause I'm not sure what we are paying this guy $180 a month for if he cannot fix a simple problem such as this. |
| 02:31.21 | [TK]D-Fender | scooby2: PM'd |
| 02:31.26 | scooby2 | thx |
| 02:31.27 | aliver | I have an * box which is 1:1 Nat'd with a legal IP and everything is allowed through. I have clients talking directly to the * box on it's real (non-NAT) internal IP. |
| 02:31.46 | jameswf-home | scooby2: you will be at astricon buying everyone beer with the money saved, correct? |
| 02:31.49 | *** join/#asterisk l2cache (n=l2cache@155.132.204.68.cfl.res.rr.com) |
| 02:31.50 | [TK]D-Fender | aliver: And hours ago I recall asking you to pastebin a failed call with SIP debug.... |
| 02:31.51 | aliver | [TK]D-Fender, yeah, it does, in effect, because there is not any PNAT going on. |
| 02:32.21 | aliver | [TK]D-Fender, yeah sorry, my CEO came and sat down and started asking me stupid questions about unrelated crap. I'll do that now. |
| 02:32.28 | aliver | It's going to be large, though. |
| 02:32.32 | jaytee | eev pauvre, il couldn' ; trouvaille de t sien derrière avec des mains et un chien de chasse. |
| 02:33.23 | [TK]D-Fender | jaytee: "pouvait" |
| 02:33.35 | jaytee | damn, I forgot to type a j at the beginning |
| 02:33.57 | jameswf-home | El queso es viejo y mohoso. Dónde está el cuarto de baño |
| 02:34.15 | [TK]D-Fender | jaytee: :and the translation butchered the rest |
| 02:34.35 | jaytee | yep, I suck so bad at French |
| 02:34.48 | jameswf-home | jaytee: must me canadien |
| 02:34.54 | jameswf-home | s/me/be/ |
| 02:35.08 | jaytee | nope, just a dumb damn yankee |
| 02:35.16 | [TK]D-Fender | jameswf-home: Canadian <- |
| 02:35.27 | jameswf-home | feminised :) |
| 02:35.28 | jaytee | Canada won't let me emigrate |
| 02:35.44 | [TK]D-Fender | We shall therefor denigrate! |
| 02:36.19 | jaytee | I'd love to live in Vancouver |
| 02:36.21 | aliver | http://pastebin.com/d4d22ed3f |
| 02:36.26 | [TK]D-Fender | jameswf-home: Nope, that'd be french & masculine. Canadienne would be female |
| 02:36.36 | jaytee | or maybe Toronto, got friends there |
| 02:36.38 | aliver | [TK]D-Fender there's the output. |
| 02:36.42 | aliver | all 90 pages of it. |
| 02:36.46 | aliver | or whatever. |
| 02:36.47 | [TK]D-Fender | jaytee: Not terribly different that being in the US anyways |
| 02:36.55 | aliver | I really don't see any errors in it. |
| 02:37.02 | aliver | I see a lot of stuff I don't understand, though. |
| 02:37.04 | jaytee | I'm tired of excessively humid summers. I grew up with cold winters so I can deal with that. |
| 02:37.20 | jameswf-home | The U.S. Shall invade canada make everyone including the Q province folks speak bad engrish and call ham ham and not bacon |
| 02:37.31 | jaytee | [TK]D-Fender, yeah but you don't have a moron in your equivalent of the White House |
| 02:38.00 | [TK]D-Fender | jaytee: True |
| 02:38.06 | aliver | I've tried changing nat=yes to nat=no and fiddling with the bind address. |
| 02:38.09 | *** join/#asterisk brodiem (n=brodiem@e2.72.1243.static.theplanet.com) |
| 02:38.24 | aliver | Still can't get audio in from outside callers that go through the SIP provider. |
| 02:39.11 | mosty | aliver: is your asterisk server behind nat? if so have you tried setting externip in sip.conf? |
| 02:39.11 | jaytee | aliver, what about your firewall ports? |
| 02:39.20 | jameswf-home | the problem is not the moron in the white house the problem is all the idiots at large who have failed to grasp basic civics and learn that moron has about 2% of the power to do what he is accused of or hated for |
| 02:39.39 | *** join/#asterisk hoegaatit (n=laa@c-67-160-226-244.hsd1.ca.comcast.net) |
| 02:39.42 | jaytee | jameswf-home, that's unfortunately true |
| 02:40.00 | aliver | mosty yes/sortof (1:1 NAT) and yes. |
| 02:40.31 | aliver | jaytee firewall is wide open to that IP. Tested that by scanning externall with nmap in (nmap -sU <myip>) |
| 02:40.33 | hoegaatit | if I change a conf file from AMI--is that a 'live edit' that goes into effect immediately or do I need to do more? |
| 02:40.36 | jameswf-home | bush is like bill gates.... not all that great but can market like a son of a bi** and make folks do his bidding... if you fall for the sales pitch of an idiot what does that make you |
| 02:40.38 | aliver | jaytee 5060 incoming is definitely open. |
| 02:40.54 | jaytee | what about ports 10000-20000 for RTP? |
| 02:41.18 | aliver | jaytee yep, they are all open (UDP) but only 5060 is open for TCP |
| 02:41.18 | mosty | aliver, run a packet trace on your firewall and check that the RTP packets are going to the correct place |
| 02:41.22 | aliver | Does RTP use TCP? |
| 02:41.25 | [TK]D-Fender | aliver: pastebin "rounte -an", your sip.conf masking only passwords, "iptables --list |
| 02:41.27 | jameswf-home | jaytee: you dont need all that simply do whats in rtp.conf... probably can bring that down |
| 02:41.38 | [TK]D-Fender | aliver: No |
| 02:41.39 | jaytee | no, and SIP uses UDP |
| 02:41.42 | [TK]D-Fender | aliver: RTP = UDP |
| 02:41.48 | aliver | okay. |
| 02:41.49 | jameswf-home | ~ports |
| 02:41.50 | jbot | well, ports is http://www.debian.org/ports/, or http://www.isi.edu/in-notes/iana/assignments/port-numbers, or the FreeBSD ports system etc etc, or http://www.portforward.com/routers.htm |
| 02:41.51 | aliver | and okay one sec. |
| 02:41.57 | jameswf-home | argh |
| 02:42.09 | jaytee | * 1.6 will do SIP TCP |
| 02:42.43 | [TK]D-Fender | aliver: And your PB should not have been skipping lines like that. |
| 02:44.24 | jaytee | jameswf-home, ok but could you confirm something for me? RTP uses 2 ports for each call? so 100 concurrent calls would need 200 RTP ports? |
| 02:44.51 | jameswf-home | Thats my understanding.. I dont really use sip |
| 02:45.12 | jaytee | really? what do you use? skinny? |
| 02:45.14 | jameswf-home | has an unlimited supply of zap hardware |
| 02:45.22 | aliver | http://pastebin.com/d715530b8 |
| 02:45.31 | aliver | I've adapted the info you asked for for BSD. |
| 02:45.37 | aliver | route -an == netstat -rnf inet |
| 02:45.48 | aliver | and iptables --list == look at your pf.conf |
| 02:46.33 | jameswf-home | well meetings over I am off |
| 02:47.01 | brodiem | Has anyone tried out any of the Aastra 5xi's? Comments? |
| 02:47.52 | aliver | If you notice that in my pf.conf I don't specify to let UDP through on port 5060, but I've tested it from the outside with nmap and it does, in fact, get through. |
| 02:48.01 | *** join/#asterisk bkw_ (n=bkw_@freeswitch/developer/bkw) |
| 02:48.04 | aliver | Which means all UDP is getting through. |
| 02:48.21 | [TK]D-Fender | aliver: All of your peers : "nat=no", "canreinvite=no". under [general] do "nat=yes", "canreinvite=no". |
| 02:48.51 | [TK]D-Fender | aliver: and don't name your phone entirels crazy things like [+19707442251] |
| 02:49.13 | [TK]D-Fender | brodiem: Meh |
| 02:49.28 | aliver | okay, but but the SIP phones are on the local internal subnet with the * box. |
| 02:50.04 | [TK]D-Fender | brodiem: Rubbery shit buttons, pooer LCD viewing angle, handset has NO weight, rubber feen on phone are weak and it slides. Speaker is a little rattly (treble) |
| 02:50.10 | [TK]D-Fender | aliver: Thats fine. |
| 02:50.20 | [TK]D-Fender | aliver: it is not NAT'd to * |
| 02:50.39 | [TK]D-Fender | aliver: it comes in on a local interface so the source & dest are local |
| 02:50.40 | aliver | [TK]D-Fender okay tried that. My phone rings when I call it (using a cell phone) and I can hear audio FROM the SIP phone but not from the Cell phone (external) |
| 02:51.11 | [TK]D-Fender | aliver: check your OUTSIDE firewall and your INSIDE firewall. |
| 02:51.14 | aliver | [TK]D-Fender right the SIP phones are all local and on the same subnet as *'s interface. |
| 02:51.20 | [TK]D-Fender | aliver: And PB your new config |
| 02:51.38 | aliver | I only have one firewall? |
| 02:51.50 | aliver | Not sure what you mean OUTSIDE and INSIDE? |
| 02:52.16 | brodiem | [TK]D-Fender: hmmm, I was looking at a 57i or Polycom 550 for personal use. I currently use a 480i quite often and like its build relatively well. Have you used the 480i also? |
| 02:52.17 | aliver | <PROTECTED> |
| 02:52.44 | jaytee | Polycom 550 is a schweeet phone |
| 02:52.55 | [TK]D-Fender | brodiem: 480 is better than 5i. What do you need in a phone? all seem overkill personally. |
| 02:53.14 | [TK]D-Fender | aliver: firewall on the * box itself <- |
| 02:53.23 | [TK]D-Fender | aliver: as well as the gateway one |
| 02:53.27 | aliver | [TK]D-Fender new http://pastebin.com/m28331f17 |
| 02:53.33 | brodiem | my thing is I like to tinker with XML scripts and the large LCD was attractive on the 57i, plus there are a lot of neat scripts already after converting them for vanilla asterisk use |
| 02:53.42 | [TK]D-Fender | aliver: Also make sure PF isn't doing any SIP transform, etc |
| 02:53.44 | jaytee | the only thing I don't like about Polycoms is the snap on base and the back make it hard to connect the handset cord, network cable and power if not using PoE with the base attached. |
| 02:53.44 | aliver | [TK]D-Fender there isn't a firewall on the * box, it's wide open with no rules. |
| 02:53.56 | jaytee | the angle is too steep |
| 02:54.05 | aliver | pf (the bsd firewall) isn't even enabled on my * box. |
| 02:54.07 | Strom_M | jaytee: that's why you connect the cables first and /then/ attach the base |
| 02:54.21 | [TK]D-Fender | AlAnd the outside.... |
| 02:54.27 | Strom_M | COMPLEX NONSOLUTIONS TO SIMPLE NONPROBLEMS |
| 02:54.38 | jaytee | Strom, that's exactly what I do |
| 02:54.50 | brodiem | lo |
| 02:54.51 | brodiem | lol |
| 02:54.56 | [TK]D-Fender | aliver: new = BAD |
| 02:55.06 | [TK]D-Fender | aliver: you don't seem to have done any of what I wrote. |
| 02:55.21 | brodiem | my thing is I like to tinker with XML scripts and the large LCD was attractive on the 57i, plus there are a lot of neat scripts already after converting them for vanilla asterisk use |
| 02:55.25 | brodiem | whoops |
| 02:56.30 | aliver | [TK]D-Fender I changed all the nat=no to nat=yes, that's what you wanted right? Oh shit the reinvite stuff. hold on. |
| 02:56.41 | aliver | I'll retry with that. |
| 02:56.54 | [TK]D-Fender | [22:48]<[TK]D-Fender>aliver: All of your peers : "nat=no", "canreinvite=no". under [general] do "nat=yes", "canreinvite=no". |
| 02:57.37 | *** join/#asterisk alephcom (n=Darren@host234117.airenet.com) |
| 02:57.50 | aliver | [TK]D-Fender I am a moron and you rule. Changing canreinvite=yes just fixed it! |
| 02:58.03 | aliver | I guess I need to know WTF that is! |
| 02:58.20 | [TK]D-Fender | aliver: there you go |
| 02:58.31 | aliver | does a little dance. |
| 02:58.43 | aliver | Something between the funky chicken and the superior dance. |
| 02:59.08 | aliver | [TK]D-Fender thanks big for your insight/help, man. |
| 02:59.43 | [TK]D-Fender | aliver: You're welcome. |
| 03:00.59 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 03:03.26 | brodiem | [TK]D-Fender: do you recall if the firmware between 480i/5xi is much different as far as configuration options? |
| 03:05.01 | [TK]D-Fender | brodiem: From what I recall looks just about the same |
| 03:05.28 | [TK]D-Fender | brothats another gripe : they moved to a full-pixel display and their retard firmware treats it like char-matrix |
| 03:06.24 | brodiem | errr I just noticed 57i: 144x128, P550: 320x160 |
| 03:06.56 | [TK]D-Fender | brodiem: its the USAGE that's stupid |
| 03:07.22 | brodiem | yeah I understand what you mean |
| 03:08.04 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 03:08.58 | brodiem | Well since the display doesn't seem to be an advantage on 5xi, no other real reason to consider that one.. |
| 03:11.27 | jaytee | mmmmm, Leinenkugels Honey Weiss. cool, crisp, refreshing |
| 03:12.06 | aliver | Say someone from the outside dials into a DID but the SIP phones on your * server are registered with just the last three digits. Would this line Allow an incoming call with the full DID to ring a phone configured that way? |
| 03:12.08 | aliver | exten => _7XX,1,Goto(internal,${EXTEN},1) |
| 03:12.32 | aliver | I'm trying to get rid of the dumbass full phone number SIP ID I've been using. |
| 03:12.59 | aliver | and just shorten it down to only be as long as it has to be to define the unique DID. Ie.. all the numbers in our DID block are the same but the last three digits. |
| 03:14.01 | [TK]D-Fender | aliver: you mas as well "#include internal" |
| 03:16.27 | aliver | [TK]D-Fender Well, I'm just asking in general 'cause I copied that line off someone else's * config. |
| 03:16.38 | aliver | I guess you can't answer without knowing what "internal" does. |
| 03:16.42 | aliver | that must be a context. |
| 03:16.42 | [TK]D-Fender | aliver: yuck |
| 03:17.06 | aliver | Just trying to understand in general. |
| 03:17.18 | aliver | I guess I'll read the * book on this for a bit until it congeals a little more. |
| 03:17.26 | [TK]D-Fender | aliver: Tell you what... I'll have you auto parts from misc vehicles and you install them on your car and see how it runs. ok? :) |
| 03:17.32 | [TK]D-Fender | hand* |
| 03:18.03 | [TK]D-Fender | aliver: watch out for frost-bite |
| 03:18.28 | aliver | Hehe, no I don't plan on using other peoples crappy configs. I just wanted to ask so I understood. It's clear I need to read about incoming calls in the book some at least before asking anymore stupid questions. |
| 03:19.28 | aliver | I'm just in a crappy situation because I'm a new Unix admin here at my new job and I know Unix/Networking stuff pretty well, but I've never touched * and now I have to learn it and redeploy it as they move to a new building. |
| 03:19.45 | aliver | The stuff they have now is using a PRI but they decided to use SIP trunking in the new place. |
| 03:19.59 | aliver | and now they want me to pick all this shit up in 4 days and make it work for them. |
| 03:20.26 | aliver | Not that * is impossible to understand but couple that with 50 other things to do and a bunch of crappy grandstream phones and it's somewhat of a challenge. |
| 03:20.29 | jaytee | sounds like they dumped you in the deep end of the pool to see how well you swim |
| 03:20.39 | [TK]D-Fender | aliver: My rates are very accessible :) |
| 03:20.40 | aliver | jaytee definitely. |
| 03:20.44 | aliver | hehe |
| 03:20.54 | *** join/#asterisk sucituanbo (n=blah@c-24-21-121-148.hsd1.wa.comcast.net) |
| 03:21.52 | jaytee | my boss shoved Grandstreams on me because they were cheap when we first started with Asterisk. When it came time to buy more phones I gave him an ultimatum, Get Polycoms or get someone else. |
| 03:22.16 | *** join/#asterisk Mike8861 (n=Mike8861@059148244254.ctinets.com) |
| 03:22.25 | TJNII | snorts |
| 03:22.29 | aliver | jaytee so far, these grandstreams have worked okay, but they are clearly cheap crap. I upgraded the firmware and now I'm just praying. |
| 03:22.32 | Mike8861 | good morning everybody!! |
| 03:22.35 | TJNII | My bos would say "Okay, GTFO" |
| 03:22.55 | aliver | Most bosses would. |
| 03:22.56 | jaytee | TJNII, yeah I was waiting to hear that myself but he caved |
| 03:23.12 | aliver | They typically don't tolerate ultimatums very well. |
| 03:23.18 | Mike8861 | aliver: which model do u own |
| 03:23.26 | aliver | I have gpx2000's |
| 03:23.43 | aliver | and some pingtels and a couple of polycoms. |
| 03:23.51 | aliver | a lovely mix |
| 03:24.10 | Mike8861 | gpx2000 is the midrange model of grandstream eh ? |
| 03:24.28 | aliver | if by midrange you mean something between Tonka and Fisher Price, yes. |
| 03:24.32 | *** join/#asterisk irisht (n=irisht@cpe-70-122-11-142.austin.res.rr.com) |
| 03:24.32 | jaytee | the gpx2000 is the Cadillac of Yugos |
| 03:24.39 | [TK]D-Fender | aliver: Matchbox FTW!!! |
| 03:24.46 | aliver | hehe |
| 03:24.52 | Mike8861 | [TK]D-Fender: Hello long time no see |
| 03:25.03 | [TK]D-Fender | jaytee: A king amongst fools... |
| 03:25.49 | jaytee | hehe |
| 03:27.10 | codefreeze-lap | Now, this is ***clever***: http://online.wsj.com/article/SB121857740590934637.html |
| 03:28.01 | jaytee | Apple had a fire at it's headquarters. Wonder if it was set by one of the people that bought the "I Am Rich" iPhone app. |
| 03:28.48 | ZX81 | lol |
| 03:29.23 | JT | they were probably testing the lithium batteries |
| 03:29.28 | JT | sony promised they were really good |
| 03:30.10 | Mike8861 | they are good at explosion and overheat |
| 03:30.51 | jaytee | if I had the money I'd buy a Moko or maybe an Instinct |
| 03:39.02 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 03:40.40 | [TK]D-Fender | is off shopping for Minolta lenses |
| 03:43.18 | JT | throws a Nikkor at [TK]D-Fender |
| 03:43.40 | [TK]D-Fender | JT : EW |
| 03:43.52 | JT | ew? |
| 03:44.04 | JT | a joke i'm guessing |
| 03:44.33 | [TK]D-Fender | JT : No. I'm not a Canonikon drone :p |
| 03:45.01 | JT | either way, nikkor have made some of the best lenses ever made |
| 03:45.10 | JT | for an SLR |
| 03:45.11 | [TK]D-Fender | JT : I went A-Mount and intend to benifit from IS + AF on all of my lenses without the added premium those ones require |
| 03:45.37 | [TK]D-Fender | JT : Same goes for every company. Minolta has tons, Sigma, Tamron, etc. |
| 03:46.05 | JT | lol as if sigma and tamron make lenses that are classed as "awesome" |
| 03:46.39 | [TK]D-Fender | JT : They've got quite a few actually... |
| 03:49.02 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 03:50.08 | jaytee | I just wanna build a rocket that'll reach the moon so I can recover all the custom made Hasselblad 500CM medium format cameras the Apollo missions left behind. Carl Zeiss lenses. |
| 03:50.57 | JT | haha |
| 03:51.38 | JT | stanley kubrich used one of the Carl Zeiss lenses made for the moon landings in Barry Lyndon |
| 03:51.43 | JT | kubrick |
| 03:51.53 | JT | f/0.4 or something insane |
| 03:52.04 | jaytee | damn! I just found out Jeff Healy died back in March |
| 03:52.15 | JT | he shot a candle lit dinner table scene with NO studio lights |
| 03:52.18 | JT | just candles |
| 03:52.54 | jaytee | Stanley Kubrick? haha. "Anyone up for a night of ultraviolence?" |
| 03:53.12 | JT | awesome movie |
| 03:53.14 | [TK]D-Fender | JT : Who needs an perture anyways ;) |
| 03:53.22 | [TK]D-Fender | JT : Just let it all in! |
| 03:55.07 | jaytee | one of my friends almost left his wife because she used Windex to clean his f2.8 300mm APO telephoto |
| 03:55.17 | [TK]D-Fender | OMFG |
| 03:55.24 | [TK]D-Fender | $$$ |
| 03:55.27 | jaytee | that was about my reaction |
| 03:56.38 | aliver | what is ${EXTEN:2} going to represent in a dial-plan? |
| 03:56.45 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-16-61.revip2.asianet.co.th) |
| 03:56.45 | jeev | boo, it's fender. |
| 03:57.00 | jaytee | aliver, your extension minus the first 2 digits |
| 03:57.06 | aliver | ah ! |
| 03:57.07 | aliver | thanks. |
| 03:57.25 | jeev | jaytee |
| 03:57.30 | jaytee | jeev |
| 03:57.32 | jeev | dood i'm a retard when it comes to context |
| 03:57.36 | jaytee | I know |
| 03:57.46 | JT | jaytee: the question is why wasn't a UV filter protecting the lens surface? |
| 03:57.52 | JT | or was it the other end of the lens |
| 03:58.03 | jeev | there is no i know's |
| 03:58.25 | jeev | you're supposed to say, no jeev.. you're the best at it |
| 03:58.25 | jeev | and NOT be sarcastic. |
| 03:58.42 | jaytee | JT, I think he's taken the filter off an put it in it's case. |
| 03:59.03 | JT | :o i always leave the filters on |
| 03:59.48 | [TK]D-Fender | JT : I bought a 50mm F1.7 prime I need a cap & filter for... thats part of my shopping ;) |
| 03:59.51 | jaytee | jeev, you are a friggin genius. Compared to you, Einstein looks like Forrest Gump. You wrote the book on Asterisk!!! |
| 04:00.04 | jeev | more lkike it |
| 04:00.06 | jeev | more like it |
| 04:01.10 | jaytee | [TK]D-Fender, I went Nikon when I started doing professional photography. My favorite lens is a 55mm Micro-Nikkor that focuses down to 6" and comes with an adapter that gives you 1:1 ratio on the negative. |
| 04:01.34 | jeev | jaytee, i'm good at photography too |
| 04:01.53 | jaytee | excellent macro lens and also worked great as a normal lens too |
| 04:02.04 | jaytee | I miss my darkroom |
| 04:02.11 | TJNII | Okay, I'm curious and google is getting me nowhere. I know almost nothing about optics except for the basices from physics II. What does Windex do to the lens? |
| 04:02.30 | jaytee | the ammonia destroys the apochromatic coating |
| 04:02.47 | [TK]D-Fender | jaytee: I'm trying to go for "variety" now and am close to placing my order for an 18-150mm Tamron (rates over 4.5/5 on IQ, etc) for my Sony A200 |
| 04:02.51 | jaytee | Windex will also ruin a good LCD panel so never use it on that either |
| 04:03.02 | TJNII | nods |
| 04:03.34 | [TK]D-Fender | jaytee: Focus to 17", but I'm buying a 1/2/4 Hoya diopter set for it as well as a UV filter & SMC polarizer |
| 04:03.37 | jaytee | I like Tamron's lenses. I think they're great quality without the brand name price markup you get for Nikon and Canon lenses. |
| 04:03.51 | jameswf-home | My big fat greek wedding says windex fixes everything |
| 04:03.56 | [TK]D-Fender | jaytee: Kinda a 1-stop shop |
| 04:06.02 | jaytee | [TK]D-Fender, I used to prefer fixed focal length but the newer lenses over the past 15 years have gotten much better. most of them used to suck at one end or the other. |
| 04:06.47 | aliver | Does $EXTEN always represent the extension of the person doing the dialing? |
| 04:06.56 | jaytee | hopefully at 18mm it will still be rectilinear |
| 04:07.00 | oilinki3 | good morning |
| 04:07.06 | jaytee | aliver, no |
| 04:07.11 | *** part/#asterisk baliktad (i=baliktad@c-24-16-27-4.hsd1.wa.comcast.net) |
| 04:07.16 | jaytee | it's the number your dialing |
| 04:07.38 | [TK]D-Fender | aliver: no, it represents where you ARE |
| 04:07.52 | [TK]D-Fender | jaytee: http://www.aocomputing.net/moo.jpg |
| 04:08.09 | [TK]D-Fender | jaytee: Last night's dinner :) |
| 04:08.11 | *** join/#asterisk hi365_m (n=hi365@213.151.56.78) |
| 04:08.15 | aliver | hmm. okay |
| 04:08.34 | [TK]D-Fender | jaytee: 50mm F2, and bokeh to spare |
| 04:08.57 | jaytee | oh, man!!!! ya had to do that, didn't ya? |
| 04:09.01 | [TK]D-Fender | :D |
| 04:09.14 | brodiem | god that looks tasty |
| 04:09.23 | jaytee | now I wanna go to Ruth's Chris |
| 04:09.24 | [TK]D-Fender | jaytee: Not to fear... I took Bessy's bell off first ;) |
| 04:09.57 | jaytee | I love my steak and roasts rare like that |
| 04:10.18 | aliver | I want to name my phones with just the unique part of the incoming phone number. Since the SIP trunk send me the numbers ala +19995551212 or some such thing I'd name the phone "212" if that was the unique part. How can I create a variable in my extensions.conf to just represent the _first_ part so I can use the Dial() application to get there? |
| 04:10.26 | aliver | Or does that just make no sense at all? |
| 04:10.59 | jaytee | when I was growing up we'd usually have a nice marbled roast beef cooked rare. We'd even spoon up the blood from platter. felt like a cow vampire :-) |
| 04:11.48 | [TK]D-Fender | :F |
| 04:12.31 | aliver | For example this works but looks like crap: |
| 04:12.33 | aliver | exten => +19707442251,1,Dial(SIP/2251,20) |
| 04:12.36 | jaytee | [TK]D-Fender, would you agree with me that using A-1 on meat like that is sacrilege? |
| 04:12.50 | [TK]D-Fender | jaytee: Actually the tiny bit of fat you see on the inside edge is actually bacon-wrap. |
| 04:12.56 | aliver | I want that Dial() part to use a nice variable so I don't have to write a rule for 200 extensions. |
| 04:13.06 | jaytee | drowns in his own drool |
| 04:13.25 | [TK]D-Fender | jaytee: Yes. If the meat was really good, it'd have its OWN flavour |
| 04:13.27 | aliver | I'm drooling enough for both of us. Cut that out. |
| 04:13.43 | aliver | (and I'm not even hungry, it's just natural) |
| 04:13.50 | [TK]D-Fender | crops the image |
| 04:13.54 | jaytee | yep, really good meat doesn't need anything.......well, maybe just a light dash of salt. |
| 04:14.27 | drmessano | I'm a vaginatarian.. so I am really getting a kick out of these replies. |
| 04:14.47 | jaytee | lol |
| 04:14.53 | [TK]D-Fender | jaytee: Yup, thats a good trick for PREP as well. use heavy rock salt and COAT it pure. Not just regular grain, the COURSE stuff. that'll leach the excess water out. Scrap it all off then broil away |
| 04:14.54 | jaytee | how was bowling? |
| 04:15.00 | jeev | i just farted like crazy at my girlfriends |
| 04:15.10 | [TK]D-Fender | drmessano: I can eat a peach for hours.... |
| 04:15.10 | drmessano | Pretty good.. I am not gonna be able to walk tomorrow |
| 04:15.15 | jaytee | I use coarse sea salt |
| 04:15.20 | drmessano | I bowled all night with a 15 lb ball |
| 04:16.15 | jeev | there is a place here in LA |
| 04:16.17 | jeev | i forgot the name |
| 04:16.21 | jaytee | ouch! that's exactly what happened to me when I went from a 12.5 lb ball to a 16lb ball. I started breaking 200 consistently but by the end of league play I was pulling muscles in my lower back and getting shooting pains down my legs. |
| 04:16.25 | jeev | it's brazillian, all you can eat with amazing quality meat |
| 04:16.30 | drmessano | LOL |
| 04:16.56 | drmessano | Heavy balls kick ass.. You just have to be able to back them up with some ass.. which, with my IT guy physique, is hard to do. |
| 04:17.44 | jaytee | yeah, I wanted the extra mass for the inertia cuz I was hitting the pocket just right but there wasn't enough transfer of energy to really scatter the pins with the 12.5 |
| 04:17.52 | jaytee | the 16lb ball kicked butt |
| 04:17.53 | [TK]D-Fender | jeev: Got 2 places like that here. Awesome. |
| 04:18.09 | jaytee | ever try Kobe beef? |
| 04:18.19 | jeev | fogo de caio or somehting |
| 04:18.24 | [TK]D-Fender | jaytee: Nope, not yet, nor tartar.... thats ASAP |
| 04:18.30 | jeev | we got a bunch of places here for that, but i heard this one was the best |
| 04:18.44 | jeev | never had Kobe, is it seriously that good ? |
| 04:18.49 | jaytee | I've had steak tartar but never tried Kobe yet |
| 04:18.55 | jeev | i dunno what steak tartar is |
| 04:19.00 | jaytee | jeev, they actually give the cows massages |
| 04:19.11 | jeev | ahh |
| 04:19.12 | jeev | cool. |
| 04:19.17 | jeev | i'm on a damn diet. |
| 04:19.17 | jaytee | jeev, it's ground raw steak |
| 04:19.20 | jaytee | with spices |
| 04:19.26 | jeev | oh, nasty |
| 04:19.38 | jeev | we have that in our food too |
| 04:19.46 | [TK]D-Fender | jaytee: Have you had? |
| 04:19.52 | jeev | my mom loves it |
| 04:19.53 | jaytee | had? |
| 04:19.56 | mosty | raw onion, and a raw egg yolk, i thought |
| 04:19.58 | jaytee | steak tartar? |
| 04:20.04 | jaytee | mosty, yep |
| 04:20.08 | mosty | with the beef obviously |
| 04:20.18 | jeev | nasty thought |
| 04:20.59 | jaytee | and they usually use a hint of garlic. mine came with chopped chives on top too with the egg |
| 04:21.41 | jaytee | [TK]D-Fender, were you asking me about the tartar or the brazillian place? |
| 04:21.43 | jeev | i want to eat like crazy now |
| 04:22.11 | [TK]D-Fender | jaytee: tartar. This one girl I went out with briefly would have gone with me. |
| 04:22.32 | [TK]D-Fender | jaytee: And yup, I've heard of the prep you just described |
| 04:23.06 | jeev | i hate my life, i'm at my girlfriends and she is stuffing her face with everything.. and her stomach is flatter than paper, with muscles :D ex-gymnist |
| 04:23.18 | jaytee | yeah, I did have it but most of the women I've met wouldn't touch it. If I met one that would I'd marry her on the spot. :-) ....... well, after having sex of course. |
| 04:23.20 | jeev | and i'm sitting here hating my life caue i can't pound ice cream |
| 04:24.26 | jaytee | I was in better shape until I moved to Indiana. I used to rollerblade 12 miles a day and worked out with free weights and a Nordic Track. I moved here and it went all downhill. |
| 04:24.47 | jeev | rollerblade? LOL |
| 04:24.57 | jaytee | I was trying for a full six pack of abs but all I could manage was four cans and a baby cheese loaf. |
| 04:25.14 | jaytee | what's so funny about rollerblading? |
| 04:25.18 | jeev | i never try for a full six pack, i eat too much. i used to dunk until i rolled my ankle during practice |
| 04:25.27 | jeev | i dunno, it's like.. you know. |
| 04:25.27 | *** join/#asterisk admin0 (n=admin@bb121-7-192-166.singnet.com.sg) |
| 04:26.09 | jaytee | yeah, I know what you're insinuating but that's just for California because all the guys your age in California are either gay or bi |
| 04:26.22 | jeev | ahhh |
| 04:26.24 | [TK]D-Fender | jaytee: http://www.break.com/index/the-hardest-part-of-rollerblading.html |
| 04:26.53 | jeev | hahahahahahahaha |
| 04:27.12 | jaytee | I like rollerblading because I used to ice skate in New England all the time when I was a kid and it's good aerobic/cardio and most important low impact. |
| 04:28.05 | jaytee | and if you've ever almost lost your right leg from the knee down because you accidentally jammed it in between two sections of boat dock you'd appreciate low impact sports |
| 04:28.47 | [TK]D-Fender | jaytee: ouch |
| 04:28.48 | jaytee | [TK]D-Fender, yeah I've seen that one |
| 04:29.43 | jeev | jaytee... so did you tell them ? |
| 04:29.57 | [TK]D-Fender | \o/ --- http://www.huffingtonpost.com/2008/08/13/mccain-in-the-21st-centur_n_118759.html |
| 04:30.05 | jaytee | [TK]D-Fender, my best friend is crazy drunk of Irish ancestry. One night we went out boating to watch the fireworks in Boston harbor. when we came back he decided to play Kung fu on the boat dock. I backed up and put my foot right where two section of dock met and went down fast. |
| 04:31.11 | [TK]D-Fender | jaytee: I took out my shoulder on a volleyball smash in lat Nov. It still hasn't recovered... |
| 04:31.53 | [TK]D-Fender | jaytee: it pops occasionally, and if I use it in a game I risk a nasty stabbing pain |
| 04:32.06 | jaytee | jeev, tell who? my parents? my dad died when I was 8. My mom just turned 90 and most importantly I'm not gay. I wouldn't mind bending a young moron like yourself over and listening to you whimper though if I'd gone through a long dry spell. |
| 04:32.27 | jeev | heh sorry |
| 04:33.22 | jeev | ah i didn't even see the moron comment.. so i'm not sorry then |
| 04:33.33 | jaytee | [TK]D-Fender, the older you get the longer it takes to heal. I had a problem with my right shoulder, torn ligaments in the rotator cuff. took almost 5 months before I could raise my arm over my head or move it behind my back. |
| 04:34.43 | jaytee | jeev, it's cool buddy. I'm just fuckin wit ya! |
| 04:35.15 | jeev | i can't tell, people in here are pretty damn nasty to me. |
| 04:36.01 | [TK]D-Fender | jeev: Careful when he says he's "fukin wit ya".... could be that dry spell coming on! |
| 04:36.06 | jaytee | you need to grow a thicker hide to play with the big boys |
| 04:36.11 | jaytee | heehehe |
| 04:37.07 | jeev | heh |
| 04:37.24 | jaytee | [TK]D-Fender, so are you still not planning on Astricon? |
| 04:38.37 | jeev | i should go to astricon and kick everyone's ass |
| 04:38.39 | [TK]D-Fender | jaytee: No passport, its far, and it'd be on my own bill. And I'm not a developer |
| 04:38.48 | jaytee | ah |
| 04:39.05 | jeev | "this is for what you said to me at 9:33 PM in #asterisk on aug 13th, man" |
| 04:39.14 | jeev | passport? damn, where you from |
| 04:39.42 | jaytee | I'm not really a developer either but I'd still like to hoist a few frostys with russell and a few others. |
| 04:40.30 | [TK]D-Fender | jaytee: I do that while Leif , Josh, etc come to town |
| 04:40.52 | jeev | fender seems to only see 1 out of every 6 of my messages |
| 04:40.57 | jeev | or questions |
| 04:41.22 | jeev | jaytee: i asked russel if there'd be any hot chix and he didn't say 1/0 |
| 04:41.32 | jaytee | only thing I'm "developing" is a speech rec IVR to replace our Locus Liaison system with LumenVox and after a 6 hour stretch of working on it I start thinking of running amok with an AK-47 |
| 04:42.36 | jaytee | jeev, he sees all and knows almost all. maybe it's cuz he only responds to the ones that make sense :-) |
| 04:43.41 | [TK]D-Fender | jaytee: http://www.demotivateus.com/tag/mag/ |
| 04:43.44 | jeev | ahh, yea, why would hot chix be at astricon! |
| 04:44.22 | jaytee | hahahahaahaha, EXCELLENT!!! I'm saving that one! |
| 04:44.23 | mchou | cause they'd be "working" the geeks? |
| 04:45.34 | jeev | i dont think any astricon attendees have had sex |
| 04:46.01 | mchou | jeev: that's known as a target rich environment |
| 04:46.08 | mchou | lol |
| 04:46.24 | [intra]lanman | jeev: a few cluecon attendees have ;-) |
| 04:47.13 | jaytee | [TK]D-Fender, http://www.demotivateus.com/page/2/ scroll down to the Retards one! |
| 04:48.02 | *** join/#asterisk astrutt (i=astrutt@pinky.ratman.org) |
| 04:48.10 | oilinki3 | btw. which one do you fellows prefere? Crying before or after the asterisk-realtime configuration? |
| 04:48.23 | astrutt | and during. |
| 04:48.27 | [intra]lanman | i like to cry during |
| 04:48.35 | astrutt | http://slashdot.org/firehose.pl?op=view&id=817763 Hey, if you have a slashdot account please vote this up =) and comment, hackers are people too =) great documentary =) premiered @ defcon on 08.08.08 |
| 04:50.24 | jeev | http://i17.photobucket.com/albums/b86/cocek/file000.jpg |
| 04:50.44 | oilinki3 | I don't understand this. I defined rtcachefriends=yes on both sip.conf and extensions.conf [global]. I can load peer with realtime load sippeers name voip.foo.com .. but sip show peers returns me nothing |
| 04:51.03 | oilinki3 | how can I know that the realtime is working? |
| 04:53.07 | jaytee | [TK]D-Fender, OMG!!! ya gotta see this. http://www.demotivateus.com/peta-just-because-i-masturbate-to-your-slutty-ads-doesnt-mean-im-going-to-stop-eating-delicious-animal-meat-demotivational-poster/ |
| 04:53.27 | [TK]D-Fender | jaytee: 3 pages ahead of you already :) |
| 04:53.39 | jaytee | and we were just talking about that earlier today |
| 04:53.39 | [TK]D-Fender | oilinki3: USE IT |
| 04:53.49 | oilinki3 | [TK]D-Fender: how? |
| 04:54.04 | jeev | hey for IAX, if i transfer calls, do i need something like Tt at the end or not ? |
| 04:54.19 | [TK]D-Fender | oilinki3: Do something the would use it |
| 04:54.41 | [TK]D-Fender | jeev: no. |
| 04:54.55 | oilinki3 | as a call to the number? |
| 04:56.08 | [TK]D-Fender | oilinki3: as in ANYTHING that will require * to access the resource via realtime |
| 04:57.41 | oilinki3 | I thought that it would be used as the peer supposed to make registration to external server |
| 04:59.03 | jaytee | [TK]D-Fender, how about the Winter one? ROFL |
| 04:59.22 | [TK]D-Fender | jaytee: Yup.... saving that.... |
| 04:59.32 | *** join/#asterisk heedly (n=heedly@purplehaze.lamedomain.net) |
| 04:59.39 | jaytee | these are great!! |
| 04:59.52 | heedly | hi |
| 05:00.21 | heedly | I'm getting stuff like file.c:1204 waitstream_core: Unexpected control subclass |
| 05:00.26 | heedly | when playing back a file. |
| 05:00.45 | heedly | and that file seems to be being played back not at the begining. |
| 05:01.07 | [TK]D-Fender | jaytee: EMO... omg |
| 05:01.24 | jaytee | hahahahaa |
| 05:01.35 | jaytee | Cybersex |
| 05:02.41 | astrutt | http://slashdot.org/firehose.pl?op=view&id=817763 If you have a slashdot account please vote this up =) and comment, hackers are people too =) great documentary =) premiered @ defcon on 08.08.08 |
| 05:03.12 | astrutt | oh and something more relevant, telephreak.org asterisk project. =) |
| 05:04.26 | jeev | what causes a failure because of congestion.. cause it was congested? lol |
| 05:07.38 | jaytee | [TK]D-Fender, MISSPELLING? |
| 05:07.46 | [TK]D-Fender | jeev: Encephalitis |
| 05:08.09 | [TK]D-Fender | jaytee: Clearly suffers from Ass-burgers syndrome ;) |
| 05:08.37 | jeev | jesus christ, the people in this country are idiots. so is hte stupid government.. they're threatening russia and saying to sotp bullying other countries. look at yourself you dumb shit moron |
| 05:08.41 | jaytee | omg, Agent Orange!!!! |
| 05:11.58 | [TK]D-Fender | jaytee: "Asians & Deer"!!!!!!! |
| 05:12.29 | jaytee | what page #? |
| 05:12.40 | [TK]D-Fender | 19 |
| 05:13.01 | [TK]D-Fender | Holy shit I can't stop laughing on that one |
| 05:13.47 | jeev | Fender, is class in session? i hvae questions |
| 05:14.41 | *** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) |
| 05:14.41 | drmessano | [TK]D-Fender: I have a question |
| 05:14.56 | [TK]D-Fender | 42 |
| 05:15.38 | drmessano | [TK]D-Fender: I read that asterisk is an open source telephony engine. Can you tell me what an "Open source" is, what "telephony" is, and why is it an "engine"? Also, what now? |
| 05:16.02 | drmessano | I am having a problem with my dialplan |
| 05:16.07 | jaytee | ROFL |
| 05:16.07 | drmessano | Let me show you what I have |
| 05:16.43 | drmessano | http://pastebin.com/m2423fe93 |
| 05:16.55 | jaytee | hahahaaha |
| 05:16.57 | drmessano | Why can't I send or receive calls to my SIP trunk? |
| 05:17.06 | drmessano | Why won't my phones register? |
| 05:17.11 | drmessano | and why does my IVR not work? |
| 05:17.45 | drmessano | Can you copy what I have there and add the relevant parts to make my callcenter fully operational for me? |
| 05:17.48 | drmessano | You rock! |
| 05:18.03 | jeev | ding ding the bell rang |
| 05:19.09 | drmessano | Jaytee: I am behind a firewall, I have taken no time to read the book or posts sent to me, I am lazy, have idea what I am doing, and I like to add shit at random and hope that it works. Can you tell my why I have one way audio? |
| 05:19.13 | drmessano | me* |
| 05:20.38 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 05:20.41 | jaytee | hey! I wrote the book *:TFOT.....um, what does ${EXTEN} mean? |
| 05:21.46 | jeev | jaytee |
| 05:21.59 | jeev | har har. |
| 05:22.29 | jaytee | [TK]D-Fender, MUFFIN TOP page 21 |
| 05:23.12 | [TK]D-Fender | jaytee: on the topic of 300mm F2.8 : http://forums.dpreview.com/forums/readflat.asp?forum=1037&thread=28950316 |
| 05:23.37 | jeev | takes out his katana |
| 05:24.35 | jaytee | [TK]D-Fender, awesome shot. a lens like that is worth every penny |
| 05:25.12 | jeev | hahaha what a pointless shot, where's the rest of the bird! |
| 05:25.19 | jeev | wait, is that a bird or bird shit! |
| 05:26.54 | jaytee | [TK]D-Fender, what's that lens cost avg retail? |
| 05:29.04 | [TK]D-Fender | jaytee: Old : http://www.dyxum.com/lenses/resultsAdv.asp |
| 05:29.13 | [TK]D-Fender | darn, can't link it |
| 05:29.48 | [TK]D-Fender | jaytee: Somewhere between 3-5K$ it seems |
| 05:30.06 | jaytee | sounds about right |
| 05:30.49 | jaytee | the Nikkor 300mm f2.8 goes for about the same |
| 05:30.55 | [TK]D-Fender | jaytee: size of a Raw-Cat Lawn-Chair for sure! |
| 05:31.00 | jaytee | I think it's like 3K or sow |
| 05:31.23 | [TK]D-Fender | alrighty... checkout time. Later all |
| 05:31.27 | jaytee | nite |
| 05:33.34 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 05:41.18 | heedly | I'm curious what prices people pay for local T1s? |
| 05:41.29 | heedly | for outbound dialing. |
| 05:44.01 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:48.56 | SwK | heedly, that varies from market to market and carrier to carrier.... |
| 05:49.26 | mosty | and the amount of calls you make |
| 05:49.58 | SwK | not really for local T1s per say.. but it does matter on LD T1s |
| 05:51.55 | jeev | mosty, you professional with iax and stuffs? |
| 05:52.21 | mosty | maybe. what's the problem? |
| 05:53.01 | jeev | no prob, i've gotten pretty succesful. i manage to get the internal asterisk to iax to the external, external makes the call.. all is well. incoming works well, it goes back |
| 05:53.31 | jeev | my issue is, what if i wnated to plug into the external, using an extension from the internal system.. i know i can set that up but how would i dial in? |
| 05:53.34 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 05:53.59 | mosty | where is the call going now, and where do you want it to go? |
| 05:54.20 | jeev | ok, right now, IAX between server internal and server external = fine. i can dial out of the network and make a succesful phone call |
| 05:54.34 | *** join/#asterisk CrashSys (i=Kumba@azrael.crashsys.com) |
| 05:54.53 | CrashSys | Anyone know of any good US-Based ITSP's that offer IAX trunking? |
| 05:55.09 | jeev | incoming works oto, if i call the #, it forwards it to the asterisk box then distributed it to the polycoms |
| 05:56.57 | jeev | but if i connect to the external asterisk with a softphone to test.., i want to be able to connect to the other asterisk box at the office.. so do i do IAX and set up a new context? |
| 05:57.01 | jeev | i'm not explaining it right, am i ? |
| 05:58.21 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-f95e2d9ee6e57a18) |
| 05:58.49 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-f73ed1110aae34ed) |
| 05:59.56 | SwK | CrashSys, they all gave up on it and switched to sip |
| 06:01.13 | CrashSys | blah |
| 06:06.20 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:06.47 | *** join/#asterisk foexle (n=heiko@router.moltomedia.de) |
| 06:07.22 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 06:13.06 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 06:14.44 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 06:19.07 | *** join/#asterisk hoegaatit (n=laa@c-67-160-226-244.hsd1.ca.comcast.net) |
| 06:21.33 | mosty | jeev: the simplest thing to do is setup extension ranges on both asterisk servers, and make sure that each asterisk server knows which extensions are on the other server |
| 06:22.29 | mosty | when delivering a call, all the server has to do is check "is this a local extension?" if so, Dial(SIP/whatever), if not, Dial(IAX2/theotherserver/whatever) |
| 06:22.46 | jeev | hmm yea i gotta figure it out |
| 06:22.52 | *** part/#asterisk jeev (n=email@unaffiliated/jeev) |
| 06:23.00 | *** join/#asterisk jeev (n=email@unaffiliated/jeev) |
| 06:23.01 | jeev | wow |
| 06:23.03 | jeev | how'd that happen |
| 06:23.27 | jeev | yea i know but i dont understand ${EXTEN} that much |
| 06:25.48 | *** join/#asterisk MaliutaLap (n=nikolai@kiev.lusan.id.au) |
| 06:26.29 | hoegaatit | has anyone noticed 1.6* crashing when doing things in AMI? |
| 06:28.33 | *** join/#asterisk pukkita (n=pukkita@137.Red-80-59-10.staticIP.rima-tde.net) |
| 06:30.35 | pukkita | hi |
| 06:36.52 | *** join/#asterisk oilinki (n=oil@ppp-124-120-18-45.revip2.asianet.co.th) |
| 06:42.25 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 06:43.34 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:43.51 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-6b5d530598e5d3a1) |
| 06:45.29 | obnauticus | anyone here know what this means: |
| 06:45.30 | obnauticus | [2008-08-13 23:45:11] NOTICE[6714]: chan_sip.c:14262 handle_request_invite: Unable to create/find SIP channel for this INVITE |
| 06:45.49 | obnauticus | nevermind got it |
| 06:47.18 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:49.18 | obnauticus | Can anyone here help with this probme: http://pastebin.ca/1171104 |
| 06:49.20 | obnauticus | problem* |
| 06:50.06 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 06:52.30 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 06:53.13 | *** part/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 06:56.16 | obnauticus | :( |
| 07:03.25 | *** join/#asterisk Sniper_VOIP (n=michofr@mail.splendor.net) |
| 07:16.14 | *** join/#asterisk Muc_Animus (n=Muc_Anim@62.214.130.132) |
| 07:16.25 | Muc_Animus | good morning to all ... |
| 07:16.35 | Muc_Animus | is this a eng or ger room? |
| 07:18.24 | obnauticus | This is a party room, Muc_Animus. |
| 07:18.35 | florz | suaheli, ausschliesslich |
| 07:19.45 | Muc_Animus | hört sich fair an :) |
| 07:19.58 | Muc_Animus | gut, dann schau ma mal ob schon jemand wach ist .... |
| 07:20.43 | Muc_Animus | ich scheitere daran den Asterisk mit dem Exchange Server Unified Messaging zum laufen zu bringen ... hat das jemand (mit der 1.6er ohne sipx) schon am laufen? *neugierigschau* |
| 07:21.02 | florz | seriously, this channel is basically English-only ;-) |
| 07:21.15 | obnauticus | ROFL |
| 07:21.46 | Muc_Animus | *grml* |
| 07:22.05 | Muc_Animus | that isn't fair as i speak english not very well :) |
| 07:22.13 | SwK | grml? is that like german for rofl? |
| 07:22.30 | SwK | Muc_Animus, dont worry most americans dont speak english well either |
| 07:22.49 | Muc_Animus | okay, lets see if anybody can understand it :) |
| 07:23.06 | obnauticus | Mein hose schlang ist sehr gross |
| 07:23.21 | *** join/#asterisk PepOSX (n=angeldav@190.72.129.75) |
| 07:23.23 | Muc_Animus | i need someone who has implemented a running configuration for asterisk 1.6 which interacts directly with MS Exchange UM 2007 ... |
| 07:23.49 | florz | obnauticus: what was that? babelfish? |
| 07:24.07 | Muc_Animus | nope, that's my really great english :) |
| 07:24.36 | florz | I was referring to his much worse ... erm ... German, it was meant to be, I suppose =:-) |
| 07:25.09 | obnauticus | :\ |
| 07:25.14 | obnauticus | I took german for like 2 years |
| 07:25.26 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:26.13 | Muc_Animus | okay, when we have finished the lang. quest. can anybody please shed some light into my issue? :) |
| 07:26.42 | SwK | what seems to be the issue... afaik... UM uses SIP... |
| 07:27.12 | Muc_Animus | uses sip / TCP .... to fix this i use asterisk 1.6 which also support sip / TCP ... |
| 07:28.29 | Muc_Animus | the problem seems to be that the call is created on port 5060 ... um move it to 5065/66/67 so, the answer from UM side didn't go to the origin call |
| 07:29.14 | obnauticus | I found this conversation really entertaining: http://forum.skype.com/lofiversion/index.php/t90911.html |
| 07:29.58 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 07:31.21 | Muc_Animus | hm... i didn't see my issue in the above post ... |
| 07:33.41 | obnauticus | Anyone: http://pastebin.ca/1171104 ? |
| 07:34.11 | SwK | something doesnt seem right in that... and unfortunately for me i havent set up UM (or 1.6 for that matter) ... being ahtts its like 0230 here in huntsville and most of the people that would probably have a good answer are sleeping or coming home from the pub hammered, I would have to suggest an email to the mailing list (sorry) |
| 07:35.00 | SwK | hah |
| 07:35.20 | SwK | I just love it when people think actually doing a real static linked binary is such a bad thing... |
| 07:35.33 | Muc_Animus | np ... thanks for your assistance, i will stay here till the guys are up later :) |
| 07:35.37 | SwK | look at 3/4s of the stuff out there for linux thats a large package... they static everything |
| 07:36.59 | SwK | sleep& |
| 07:37.11 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 07:41.54 | obnauticus | SwK: i know |
| 07:42.29 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.137) |
| 07:49.52 | *** join/#asterisk benneton (n=DELL@89.111.209.74) |
| 07:50.05 | benneton | hi guys! |
| 07:50.42 | mkillebrew | HAI!!!11one |
| 07:51.27 | obnauticus | HAI2U |
| 07:52.04 | benneton | Whats up? |
| 07:52.59 | benneton | Is it possible to disable agent ringing sound while in queue? I want moh only. Tnx |
| 07:53.56 | benneton | :) (is there something impossible with asterisk??) |
| 07:54.16 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 07:55.58 | benneton | AddQueueMember option, maybe? |
| 07:56.28 | *** join/#asterisk wizzy_ (n=andyr@musselcracker.aims.ac.za) |
| 07:56.55 | benneton | i can see "j" option for jumping to +101 |
| 07:58.34 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 07:59.20 | wizzy_ | I have a SIP phone (snom 300), asterisk 1.4 from svn (I backed out of 1.6.0 - too hectic), and a B410P ISDN card. When I make a call out using the snom phone, I can hear the other side, but quality is terrible for them. Where should I start looking ? |
| 08:01.29 | wizzy_ | Is it more likely to be the SIP end, the ISDN end, or some codec thing ? |
| 08:02.47 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 08:04.40 | *** part/#asterisk benneton (n=DELL@89.111.209.74) |
| 08:05.01 | *** join/#asterisk The_LightSide (n=root@dsl-241-107-54.telkomadsl.co.za) |
| 08:07.35 | *** join/#asterisk benneton (n=DELL@89.111.209.74) |
| 08:08.38 | metastable | tzafrir_laptop: do you know the variable to change to get rid of the kernel CFLAGS error in compiling b410p ? |
| 08:08.46 | benneton | in app_queue.c there is no other option than 'j' for AddQueueMeme |
| 08:08.48 | *** join/#asterisk speekac (n=alwin@60.51.217.61) |
| 08:09.00 | speekac | hi all |
| 08:09.10 | tzafrir_laptop | metastable, I'm not familiar with building misdn, sorry |
| 08:09.12 | speekac | does anyone know if asterisk support utf-8 in its variables |
| 08:09.21 | speekac | eg. via Set() |
| 08:09.25 | metastable | tzafrir_laptop: ok, np |
| 08:09.34 | metastable | tzafrir_laptop: qwell thaught you knew |
| 08:09.39 | EmleyMoor | Is there a way to limit a SIP peer to one registration at a time? |
| 08:09.46 | tzafrir_laptop | metastable, can you post a log? |
| 08:09.53 | metastable | sure |
| 08:09.56 | tzafrir_laptop | What kernel version? |
| 08:10.09 | metastable | 2.6.24-19-generic on an ubuntu server |
| 08:10.48 | metastable | tzafrir_laptop: http://pastebin.com/mdb562 |
| 08:10.55 | metastable | zaptel 1.4.11 compiles fine |
| 08:11.16 | speekac | EmleyMoor: check sip.conf options |
| 08:11.27 | tzafrir_laptop | first off, misdn is not related to zaptel |
| 08:11.32 | wizzy_ | metastable: there is newer code |
| 08:11.47 | metastable | beta ? |
| 08:11.48 | tzafrir_laptop | There's also misdn 1.1.8 now. Any chance it was fixed there? |
| 08:12.06 | wizzy_ | tzafrir_laptop: I think that is what I used |
| 08:12.27 | metastable | the zaptel make process chooses that misdn version |
| 08:12.47 | metastable | should i start off with a beta version of zaptel, or download misdn separately ? |
| 08:13.04 | tzafrir_laptop | download it separately |
| 08:13.05 | wizzy_ | metastable: I used dahdi |
| 08:13.18 | metastable | ok, i'll give it a try |
| 08:13.19 | tzafrir_laptop | dahdi removed the b410p target altogether |
| 08:13.49 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 08:14.10 | wizzy_ | metastable: I have built asterisk 1.4 / B410P on 8.04 - don't use 1.6 |
| 08:14.39 | metastable | i have 1.4.9 |
| 08:16.34 | metastable | http://pastebin.com/m5575b10a |
| 08:16.39 | metastable | An error |
| 08:16.42 | wizzy_ | Asterisk SVN-branch-1.4-r137405M |
| 08:16.58 | metastable | I had the same error when trying to compile 1.4.11 from a separate download |
| 08:17.14 | metastable | 1.1.7* :) |
| 08:17.56 | tzafrir_laptop | metastable, there are a host of other bad warnings there. For instance: |
| 08:18.00 | tzafrir_laptop | /root/mISDN-1_1_8/drivers/isdn/hardware/mISDN/avm_fritz.c:889: warning: passing argument 2 of ârequest_irqâ from incompatible pointer type |
| 08:18.08 | wizzy_ | hmm - I have capiutil.h - /usr/src/linux-headers-2.6.24-19/include/linux/isdn/capiutil.h |
| 08:18.11 | tzafrir_laptop | This probably means that the module would fail to load |
| 08:19.39 | EmleyMoor | It looks like there isn't |
| 08:19.51 | metastable | ok, it's compiling now |
| 08:20.26 | metastable | i removed some files relating to isdn from kernel sources earlier |
| 08:20.31 | tzafrir_laptop | Those warnings there scare me |
| 08:20.35 | wizzy_ | I have sound quality issues from snom phone -> asterisk 1.4 -> B410P |
| 08:20.46 | wizzy_ | how do I debug this ? |
| 08:20.49 | metastable | because some installer told me |
| 08:21.04 | metastable | tzafrir_laptop: it has compiled :) |
| 08:21.30 | *** join/#asterisk xenonex (n=xenonex@89.218.237.133) |
| 08:21.37 | wizzy_ | I get coredumps from rmmod - but it seems OK otherwise :) |
| 08:22.20 | *** join/#asterisk steliosk (n=Stelios@athedsl-309630.home.otenet.gr) |
| 08:24.08 | metastable | ok guys, the card is now detected |
| 08:24.12 | metastable | thanks for your advice |
| 08:24.24 | metastable | all your telephony are belong to us |
| 08:26.39 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-6874908d61527a72) |
| 08:27.48 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 08:30.26 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:30.53 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 08:31.44 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:35.52 | *** join/#asterisk zepmantra (n=asd@124.107.177.41) |
| 08:36.12 | zepmantra | ~centos52bug |
| 08:36.13 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 08:39.11 | qp | can anyone help me setting the outbound caller ID on the fly? I use freepbx, so would prefer a solution that is happy with freepbx :) |
| 08:39.39 | denon | you might be in the wrong channel then |
| 08:39.50 | denon | ~freepbx |
| 08:39.51 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 08:39.59 | qp | I know, its all a bit dead in there |
| 08:40.38 | denon | they'll probably wake up eventually |
| 08:40.43 | qp | I can drop stuff in the extensions override file tho |
| 08:41.12 | qp | no worries, just getting a bit desperate |
| 08:46.07 | EmleyMoor | What exactly are you trying to achieve? |
| 08:47.17 | qp | I use a dial prefix to dial out as a specific "company", so an operator dials *001*123456 and its logged against company "001" but I wuold also like the called number to see company 001's number |
| 08:47.44 | qp | the numbers are all owned by us, and its a voip only platform |
| 08:47.51 | *** join/#asterisk admin0 (n=admin@bb121-7-192-166.singnet.com.sg) |
| 08:48.03 | EmleyMoor | Oh right - well, in Asterisk, it's easy enough... |
| 08:48.31 | EmleyMoor | Assuming dialing 123456 would put you through regardless, that is... |
| 08:48.37 | qp | yeah |
| 08:48.46 | qp | on default outbound cid |
| 08:49.06 | EmleyMoor | How many such caller IDs do you have? |
| 08:49.24 | qp | potentially over a hundred, starting with 10 |
| 08:49.39 | qp | we handle calls for many one or 2 man companies |
| 08:49.57 | EmleyMoor | Well, you could put them in a database - use something like /callerid/001 = whatever... |
| 08:50.29 | qp | yeah I saw this in the freepbx forums, see the last post: |
| 08:50.29 | qp | http://freepbx.org/forum/freepbx/users/how-do-i-modify-outbound-caller-id-without-using-freepbx |
| 08:51.00 | *** join/#asterisk aksyn (n=aksyn@cp1242056-a.schoo1.lb.home.nl) |
| 08:51.35 | ghenry | Hi |
| 08:51.37 | ghenry | at http://nopaste.com/p/aUlLeUvub |
| 08:52.03 | ghenry | a sip endpoint is in [default] |
| 08:52.06 | *** join/#asterisk aksyn (n=aksyn@cp1242056-a.schoo1.lb.home.nl) |
| 08:52.51 | ghenry | should that sip endpoint be able to dial a exten in [numberplan-custom-1] without the include => numberplan-custom-1 in [default] |
| 08:52.58 | EmleyMoor | And match _*XXX*.,1,Set(CALLERID(num)={DB(callerid/${EXTEN:1,4})}) then Goto(${EXTEN:5},1) - how successful it might be without true asterisk I don't know. |
| 08:53.10 | EmleyMoor | 1:4 even |
| 08:54.05 | EmleyMoor | qp: You will of course have to allow for any other sequences with *X carefully... |
| 08:54.34 | qp | yeah |
| 08:54.51 | EmleyMoor | will have to rewrite his extensions.conf soon... some 07 numbers are changing to 06 |
| 08:55.27 | qp | cheers emley, will look into this |
| 09:02.22 | *** part/#asterisk speekac (n=alwin@60.51.217.61) |
| 09:11.48 | *** join/#asterisk micheluntu (n=michele@dhcp197-193.cu.mi.it) |
| 09:12.49 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-877fc2426218d91d) |
| 09:13.09 | *** join/#asterisk zydoon (n=zydoon@213.150.170.26) |
| 09:15.21 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 09:16.35 | micheluntu | is no way to see where the program attempts audio files? I get a generic Unable to open ..... No such file or directory |
| 09:18.13 | NoxIn- | micheluntu: the path should be defined in /etc/asterisk/asterisk.conf like this : astdatadir => /var/lib/asterisk |
| 09:18.20 | NoxIn- | for instance |
| 09:18.44 | *** part/#asterisk zydoon (n=zydoon@213.150.170.26) |
| 09:20.01 | micheluntu | NoxIn-: this entry is missing in asterisk.conf... is there a way from CLI? |
| 09:21.24 | NoxIn- | none that I know |
| 09:22.41 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-34c6bee24928d0f1) |
| 09:23.12 | micheluntu | ok, i'll try to set the parameter. astdatadir => /var/lib/asterisk/sounds should be right? |
| 09:23.29 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 09:23.42 | NoxIn- | on my debian install it's /var/lib/asterisk |
| 09:24.07 | NoxIn- | it's not the sounds directory but the datas directory, asterisk will look for a sounds/ folder inside it |
| 09:25.23 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 09:58.26 | *** join/#asterisk sysadmin-lb22 (n=asdf@mail.splendor.net) |
| 10:03.21 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.141) |
| 10:17.23 | *** join/#asterisk eDead|Work (n=woot@rsm3724-7258-216-235.cnt.nerim.net) |
| 10:20.10 | *** join/#asterisk jarod115 (n=jarod@rsm3724-7258-216-235.cnt.nerim.net) |
| 10:20.31 | jarod115 | hi |
| 10:20.37 | jarod115 | all people |
| 10:20.54 | jarod115 | first sorry for my english cause i'm french |
| 10:20.55 | jarod115 | ;-) |
| 10:22.43 | jarod115 | i use asterisKnow |
| 10:22.49 | jarod115 | last version |
| 10:23.05 | jarod115 | with freephonie SIP |
| 10:23.19 | jarod115 | and it's ok to call |
| 10:23.22 | jarod115 | and receive |
| 10:23.40 | jarod115 | but when i try to call a client |
| 10:24.11 | jarod115 | he ave a IVR ( Serveur vocal) |
| 10:25.25 | jarod115 | and i need to press number 1 on my thomson 2030 phone |
| 10:25.45 | jarod115 | but the IVR don't reconize the number |
| 10:26.20 | jarod115 | i d'ont know why he do this?????? |
| 10:26.25 | jarod115 | thanks for ur help |
| 10:32.32 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 10:39.57 | yang | ~asterisknow |
| 10:39.58 | jbot | somebody said asterisknow was based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 10:46.01 | *** join/#asterisk UD (n=steve@unaffiliated/underdawg) |
| 10:51.22 | *** join/#asterisk bark115 (n=jarod@car75-9-88-178-249-191.fbx.proxad.net) |
| 10:57.51 | *** join/#asterisk The_LightSide (n=root@dsl-241-107-54.telkomadsl.co.za) |
| 11:00.14 | qp | whats the best way of changing outbound called ID in asterisk? Set() what? |
| 11:00.21 | qp | *caller ID |
| 11:07.13 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-068-200.pools.arcor-ip.net) |
| 11:07.16 | *** join/#asterisk gnorbert (n=gnorbert@228-177.static.ew.hu) |
| 11:08.40 | Blackvel | hi guys. what's the BEST Asterisk CTI/Windows Outlook Integration tool out there (can be commercial)? snapanumber? outCall? snom flexor manager? phonesuite? any others? has to support SIP as well as ZAP (bristuff BRI) |
| 11:10.42 | *** join/#asterisk mort_gib (n=mjensen@dsl-p4-177.gibconnect.com) |
| 11:11.20 | mort_gib | HI, quick question |
| 11:12.41 | mort_gib | I don't need Zaptel fro my HW, but I need to use a IAX provider as a trunk, and * complains that this can't be done unless Zaptel can be used as timing source. So I would need to load ztdummy -Right?? |
| 11:12.52 | heedly | ya |
| 11:19.50 | pukkita | hiya mort_gib |
| 11:20.10 | mort_gib | Hi pukkita |
| 11:20.35 | qp | sorry to ask again, but whats the best way of changing outbound caller ID in asterisk? Set() what? |
| 11:20.58 | Blackvel | mort_gib: can remember that I used it with meetme and it worked (that was 2005). now I have HW card |
| 11:21.38 | mort_gib | Yeah, but I use Sangoma/BRI with this client so nogo :-) |
| 11:21.50 | *** join/#asterisk elmo2k3 (n=bjoern@i59F4EDB2.versanet.de) |
| 11:21.57 | mort_gib | I need to load ztdummy (I think) So how do I do that?? |
| 11:21.59 | Blackvel | qp: Set(CALLERID(number)= or Set(CALLERID(name)= |
| 11:22.29 | *** join/#asterisk pputman (n=centrex@conference/asterisk/x-c9935b2c47208c29) |
| 11:22.31 | qp | ok thanks, tried that but its not working, think freepbx is screwing something up, *sigh* |
| 11:22.35 | gnorbert | Hi all. Does somebody have an idea, how can I kick somebody from a meetme conference from outside the CLI? |
| 11:22.55 | Blackvel | tell him ;) |
| 11:24.00 | Blackvel | mort_gib: maybe something described on wiki? Probably I would change to zaptel directory, check Makefile for ztdummy module and it make, make install |
| 11:24.07 | mort_gib | Walk up to him slowly, savoring the moment, and kick him, using your hip |
| 11:24.20 | gnorbert | Blackvel: Thank you, I'll try, it'll be handled with programs. :) Maybe they will understand, how important it is for me. :) |
| 11:24.23 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:24.36 | pputman | I'm having a problem with chan_zap and 1.4.21.2. When I restart asterisk, zap show channels command doesn't work, but I can see the b-channels restarting and incoming calls on zaptel channels on the cli. It also shows chan_zap.so as loaded. But like I said, none of the zap cli commands work. If I issue a "module load chan_zap.so" it says it loaded it successfully, and I get all the zap commands back. Has anyone seen anythi |
| 11:24.36 | pputman | ng like this before? Thanks. |
| 11:24.43 | Blackvel | after that you have a ztdummy.ko which you can insmod /etc/sysconfig/zaptel and /etc/modprobe.conf for ztdummy options |
| 11:25.53 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-18-45.revip2.asianet.co.th) |
| 11:26.17 | gnorbert | Can't I use something like meetme kick in dialplan, or handle the CLI with a program, or script? |
| 11:26.19 | Blackvel | gnorbert: ah found it. there is cmd meetme kick? can't you integrate it into your dailplan so you can pass an argument? |
| 11:27.02 | Blackvel | questions is what user/channel it is on meetme room :) |
| 11:27.27 | *** part/#asterisk pukkita (n=pukkita@137.Red-80-59-10.staticIP.rima-tde.net) |
| 11:27.30 | gnorbert | Blackvel: If I can do it from dialplan, it's already good. :) |
| 11:27.32 | Blackvel | I am not an meetme expert, but weren't there programs to handle room management including a display? |
| 11:27.43 | *** join/#asterisk pukkita (n=pukkita@137.Red-80-59-10.staticIP.rima-tde.net) |
| 11:27.48 | gnorbert | I must kick everybody in a conference, if a given person quits. :) |
| 11:28.15 | pukkita | has the callerid setting in 1.14 changed? Set(CALLERID) doesn't seem to work?? |
| 11:28.16 | gnorbert | So if he quits, I can call the extension, that I need and kick all. :) |
| 11:28.47 | Blackvel | gnorbert: looks like you can use php and meetme list lists all users |
| 11:29.01 | Blackvel | probably there are solutions which you can use out of the box |
| 11:29.49 | gnorbert | Blackvel: Thanks, but is it sure, I can put meetme kick in dialplan? |
| 11:29.57 | mort_gib | Blackvel: None of that made much sense |
| 11:30.01 | mort_gib | I'm using Debian |
| 11:30.14 | Blackvel | pukkita: 1.4? Its Set(CALLERID(number)=123) |
| 11:30.22 | mort_gib | I have a ztdummy.ko file, but it will not load... |
| 11:30.30 | Blackvel | gnorbert: not so sure |
| 11:30.56 | Blackvel | mort_gib: any errors? |
| 11:31.27 | mort_gib | Yeah, insmod: error inserting '/lib/modules/2.6.18-6-686/misc/ztdummy.ko': -1 Unknown symbol in module |
| 11:31.44 | Blackvel | does it depend on libs? |
| 11:31.44 | pukkita | Blackvel ok, so to set the Callerid to the extension the call is being originated from? |
| 11:32.07 | pukkita | ${CALLERIDNUM} doesn't seem to work |
| 11:33.18 | tzafrir_laptop | mort_gib, modprobe ztdummy |
| 11:33.28 | tzafrir_laptop | this will insmod zaptel first |
| 11:33.50 | pukkita | I'm using the same dialplan from a working 1.12 on a 1.14 setup and looks like the callerid not being sent |
| 11:33.51 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 11:34.14 | Blackvel | pukkita: 1.1, 1.14? you don'T mean 1.4? |
| 11:34.20 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-6dcf43b50fbaa409) |
| 11:35.12 | Blackvel | pukkita: what does ${CALLERIDNUM} display with NoOp? maybe you use zaptel and something is wrong? |
| 11:35.19 | mort_gib | modprobe ztdummy -Gives nothing, insmod ztdummy (can't find file)) |
| 11:35.41 | Blackvel | I would go for /etc/init.d/zaptel start |
| 11:35.50 | Blackvel | it automatically starts the modules |
| 11:36.08 | Blackvel | not sure if I used 2005 ztcfg |
| 11:36.08 | pukkita | sorry 1.4 yep |
| 11:36.30 | pukkita | Blackvel empty, CALLERIDNUM is Empty |
| 11:36.43 | pukkita | I'm using zaptel, yep |
| 11:36.50 | pukkita | sangoma PRI card |
| 11:37.03 | mort_gib | <PROTECTED> |
| 11:37.03 | mort_gib | ztdummy 5160 0 |
| 11:37.04 | mort_gib | zaptel 186180 1 ztdummy |
| 11:37.04 | mort_gib | rtc 12372 1 ztdummy |
| 11:37.13 | *** part/#asterisk benneton (n=DELL@89.111.209.74) |
| 11:37.21 | Blackvel | pukkita: CALLERIDNUM is deprecated (not sure it works even on 1.4. I think it was even deprecated on 1.2) |
| 11:37.32 | Blackvel | I try too much the past days :) |
| 11:39.00 | Blackvel | found someting: ${CALLERID(num)}: The current Caller ID number - ${CALLERIDNUM} was used in versions of Asterisk prior to 1.2.0, it was DEPRECATED in 1.2.0 and removed in 1.4. |
| 11:39.53 | mort_gib | Thing is, now * won't start because I have an issue with my Zaptel configuration! |
| 11:40.57 | mort_gib | If I rmmod ztdummy * starts fine again |
| 11:41.03 | Blackvel | hey tzafrir_laptop :) do you use any CTI / Outlook integration with Asterisk like callOUT, snapanumber or phonesuite? |
| 11:41.28 | tzafrir_laptop | Blackvel, no |
| 11:42.02 | tzafrir_laptop | mort_gib, you modprobed ztdummy and you have a zaptel card (a digital one)? |
| 11:42.13 | gnorbert | It looks like I can't kick somebody from dialplan, or anyway I can't do it with meetme command. |
| 11:42.18 | Blackvel | http://www.voip-info.org/wiki-Asterisk+timer+ztdummy |
| 11:42.20 | tzafrir_laptop | if ztdummy is the first span, it changed span numbers for others |
| 11:42.56 | mort_gib | Blackvel: I use http://www.voip.com.sg/asterisk/voip_asterisk_outlook_dialer.html |
| 11:43.21 | pukkita | thanks Blackvel found out |
| 11:43.27 | mort_gib | tzafrir_laptop: I don't have any zaptel hardware, I'm using a Sangoma A500 card only |
| 11:43.39 | pukkita | looks like the Telco rejects the callerid if bigger than 4 digits |
| 11:43.48 | Blackvel | yesterday I saw an online video about tribox gui and ivr menu. I was pretty impressed |
| 11:44.22 | Blackvel | pukkita: ah for the telco side? outgoing telco call? do you use clip no screening? |
| 11:44.28 | mort_gib | But I get: WARNING[19924] chan_iax2.c: Unable to support trunking on peer 'voipon' without zaptel timing |
| 11:45.33 | Blackvel | mort_gib: BRI Sangoma A500 drivers are using zaptel or not? I thought they bypass zaptel (and soft echo MG2) at all? |
| 11:45.41 | Blackvel | so shouldn't be a matter of zaptel configuration |
| 11:46.03 | Blackvel | probably /etc/sysconfig/zaptel loads all the digium modules, which it shouldn't? |
| 11:46.29 | *** join/#asterisk macros73 (n=cs@c-67-163-224-69.hsd1.pa.comcast.net) |
| 11:46.40 | Blackvel | you have to uncomment all the modules really |
| 11:46.48 | mort_gib | Well apparently I need zaptel for timing, not for anything else |
| 11:46.52 | JT | wtf |
| 11:46.58 | mort_gib | I don't have a /etc/sysconfig |
| 11:46.59 | JT | why would you load all the modules |
| 11:47.08 | JT | people who load all zaptel modules are retarded |
| 11:47.16 | JT | only load what you use. |
| 11:47.36 | Blackvel | mort_gib: maybe /etc/default/zaptel? what linux distro? |
| 11:47.52 | mort_gib | Debian |
| 11:48.51 | pputman | JT, not to mention there is a great thing called make menuselect =). |
| 11:48.59 | JT | heh |
| 11:49.16 | mort_gib | -Found it |
| 11:49.28 | mort_gib | Commented anything but ztdummy out |
| 11:49.49 | tzafrir_laptop | mort_gib, /etc/defaut/zaptel ? What do you want to set there? |
| 11:50.19 | mort_gib | I need IAX to work in trunk mode! |
| 11:50.35 | gnorbert | So does anybody have any idea, how can I kick somebody out from a conference, if not from the CLI? |
| 11:50.44 | Blackvel | tzafrir_laptop: on centos its /etc/sysconfig/zaptel and there it loads all the modules when you call /etc/init.d/zaptel start |
| 11:50.55 | mort_gib | But it can't if I don't have a zatpel timing |
| 11:51.04 | gnorbert | If I can kick only the whole room, that's also good. |
| 11:51.26 | pukkita | anyone knows if the Read() timeout parameters are seconds? |
| 11:51.43 | Blackvel | tzafrir_laptop: I had to disable all digium modules (no matter of make menuselect) |
| 11:52.08 | Blackvel | ....lunch time ....away for some minutes.... |
| 11:52.09 | pukkita | sorry, already saw on voip.,info they're seconds |
| 11:52.26 | pukkita | my TFOT copy doesn't specify :) |
| 11:52.27 | tzafrir_laptop | mort_gib, why did you have to disable them? |
| 11:52.48 | mort_gib | I don't have any Zaptel hardware, or hardware that needs Zaptel |
| 11:52.58 | mort_gib | But IAX is complaining |
| 11:53.26 | pukkita | mort_gib: looks like you have to recompile zaptel |
| 11:53.34 | mort_gib | Been there, done that... |
| 11:53.46 | pukkita | unknown symbols is ussually a sign of build mismatch |
| 11:54.12 | pukkita | I recall also ztdummy timing uses USB for timing source |
| 11:54.20 | Blackvel | mort_gib: can't you show us the exact error of asterisk start on pastebin? |
| 11:54.31 | pukkita | but can't remember if it was FreeBSD or Linux specific |
| 11:54.32 | mort_gib | Yes, hang on |
| 11:55.02 | Blackvel | pukkita: I believe there are two timing options and depending on use it must be included in kernel |
| 11:55.13 | pukkita | guess you already tried depmod -a? (though I don't know if that's still used in linux nowadays) |
| 11:55.15 | Blackvel | somewhere that MUST be written down :) |
| 11:58.12 | *** join/#asterisk ez` (n=ez@c66.203.221-242.clta.globetrotter.net) |
| 11:58.30 | mort_gib | http://pastebin.com/d1d58756c |
| 11:59.23 | tzafrir_laptop | Blackvel, try cat /proc/zaptel/1 |
| 11:59.23 | tzafrir_laptop | (or whatever the span number is for ztdummy) |
| 11:59.23 | tzafrir_laptop | It will tell you the timing source being used |
| 11:59.23 | pputman | mort_gib, what happens when you insmod zaptel.ko ? |
| 11:59.36 | tzafrir_laptop | strings ztdummy.ko will tell you that as well |
| 11:59.40 | tzafrir_laptop | see the README |
| 12:00.13 | mort_gib | Asterisk wont start when I have zaptel running... |
| 12:00.41 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 12:00.58 | pputman | mort_gib, I missed half the conversation, did you already pastebin your zaptel.conf/zapata.conf ? |
| 12:02.16 | mort_gib | No, I don't have any hardware that requires zaptel, but IAX is complaining that trunk mode is not avail until timing (zaptel) is loaded |
| 12:02.57 | mort_gib | * is complaining that my zaptel.conf/zapata.conf are empty really |
| 12:03.13 | mort_gib | as in I have everything commented out |
| 12:03.23 | tzafrir_laptop | mort_gib, can you try the etch-and-a-half kernel? It is 2.6.24, and hence will be able to use hi-res timers |
| 12:03.33 | tzafrir_laptop | If you have problems with RTC |
| 12:04.12 | pputman | mort_gib, that's pretty odd, might want to file a bug |
| 12:04.23 | mort_gib | Yeah?? |
| 12:04.55 | mort_gib | I'm on :Linux asterisk 2.6.18-6-686 |
| 12:05.57 | *** join/#asterisk salzh (n=chatzill@116.233.207.96) |
| 12:06.32 | mort_gib | It is a quite long winded to go to 2.6.24.... |
| 12:08.24 | mort_gib | Thing is as far as I know meetme and similar things wont work unless I'm using timing of some sorts |
| 12:08.39 | mort_gib | But timing is supposed to be included in Wanrouter from Sangoma |
| 12:08.52 | qp | anyone done a direct update to MySQL from within a conf? |
| 12:12.08 | tzafrir_laptop | mort_gib, what's the output of zttest -v ? |
| 12:12.21 | tzafrir_laptop | anything close to 100% or just hangs? |
| 12:13.18 | mort_gib | Looks like it just hangs |
| 12:13.36 | mort_gib | --- Results after 0 passes --- |
| 12:13.36 | mort_gib | Best: 0.000 -- Worst: 100.000 -- Average: 100.000000, Difference: 100.000000 |
| 12:13.46 | mort_gib | ?? |
| 12:14.34 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:14.34 | tzafrir_laptop | mort_gib, do you have that sangoma card on your system? |
| 12:15.05 | mort_gib | Yes, ------------------------------- |
| 12:15.05 | mort_gib | | Wanpipe Hardware Probe Info | |
| 12:15.05 | mort_gib | ------------------------------- |
| 12:15.05 | mort_gib | 1 . AFT-A500-SH : SLOT=2 : BUS=2 : IRQ=82 : PORT=1 : HWEC=64 : V=32 |
| 12:15.05 | mort_gib | 2 . AFT-A500-SH : SLOT=2 : BUS=2 : IRQ=82 : PORT=2 : HWEC=64 : V=32 |
| 12:15.06 | mort_gib | 3 . AFT-A500-SH : SLOT=2 : BUS=2 : IRQ=82 : PORT=3 : HWEC=64 : V=32 |
| 12:15.08 | mort_gib | 4 . AFT-A500-SH : SLOT=2 : BUS=2 : IRQ=82 : PORT=4 : HWEC=64 : V=32 |
| 12:15.10 | mort_gib | Card Cnt: A500=1 |
| 12:15.25 | [TK]D-Fender | mort_gib: PASTEBIN |
| 12:15.31 | [TK]D-Fender | mort_gib: Please don't spam |
| 12:15.47 | Muc_Animus | anybody with asterisk 1.6 and UM expirience here? *hope* |
| 12:16.35 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 12:23.03 | gnorbert | Does somebody have any idea about how can I kick a user from a meetme conference, if not from the CLI? (Or how can I use CLI with a program or scrpit) |
| 12:23.57 | pukkita | [TK]D-Fender: thanks for your yesterday's tip about the pseudo IVR to address a PBX dialling the null number then DTMF'ing |
| 12:24.19 | pukkita | [TK]D-Fender: worked like a charm, I guess a slight delay is unavoidable, but it works |
| 12:24.22 | [TK]D-Fender | pukkita: so that was indeed what it was doing? |
| 12:24.31 | [TK]D-Fender | pukkita: heh... unusual, but not unheard of. |
| 12:24.44 | pukkita | [TK]D-Fender yep, and wasn't settable on the PBX, an old Bosch Digital PBX |
| 12:24.48 | [TK]D-Fender | pukkita: this is how analog DID works in the cases I've run into before. |
| 12:24.57 | [TK]D-Fender | pukkita: Sometimes wink-start indicated |
| 12:25.36 | pukkita | [TK]D-Fender I just put up a small IVR reading the outgoinf number DTMF codes then had to put a little conditional logic to decide which outbound trunk to use |
| 12:25.38 | [TK]D-Fender | gnorbert: "help meetme" |
| 12:26.19 | *** join/#asterisk AndyML (n=alauppe@pool-71-185-74-86.phlapa.fios.verizon.net) |
| 12:26.38 | mort_gib | So the conclusion is that IAX can only work in trunk mode if I have Zaptel hardware??? |
| 12:26.43 | gnorbert | [TK]D-Fender: But that gives only the CLI commands and I should solve it not from the CLI. |
| 12:27.27 | [TK]D-Fender | gnorbert: Go read the 2 chapters on AGI in the BOOK then. |
| 12:27.38 | pukkita | mort_gib I use it on a FBSD box with i4b, no zaptel hardware, on two other boxes with Sangoma cards w/o problems |
| 12:27.40 | gnorbert | [TK]D-Fender: Ok, thanks. |
| 12:27.49 | gnorbert | ~book |
| 12:27.51 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 12:27.58 | AndyML | mort_gib - I could be wrong, but I thought pseudo-zap used the usb controller for trunk timing. anyone? |
| 12:27.59 | gnorbert | Also found it. :) |
| 12:28.01 | [TK]D-Fender | mort_gib: No, only if you have a Zaptel timing source. |
| 12:28.49 | mort_gib | So how do I load a Zaptel timing device, when I have no Zapel hardware |
| 12:28.59 | *** join/#asterisk Gat0rvean (n=gredish@64.191.128.145) |
| 12:29.22 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-b08d8ae093b419f9) |
| 12:30.07 | [TK]D-Fender | mort_gib: ZTDUMMY <--- |
| 12:30.43 | mort_gib | Yeah, I tried, but when I load ztdummy * says that I have a "problem with my Zaptel config" |
| 12:31.05 | [TK]D-Fender | mort_gib: Guess you'd better look more closely at that then. |
| 12:31.18 | mort_gib | Which is strange, because I have everything commented out in zapata.conf/zaptel.conf |
| 12:31.53 | mort_gib | And if I ONLY load ztdummy, then those files are not loaded are they?? |
| 12:32.20 | mort_gib | ztdummy is just the timing "device" -Right?? |
| 12:32.25 | [TK]D-Fender | mort_gib: well let us know when you actually want to try something with it. |
| 12:32.30 | [TK]D-Fender | mort_gib: correct |
| 12:32.58 | *** join/#asterisk JenniferAkemi (n=akemi@76-10-152-16.dsl.teksavvy.com) |
| 12:33.31 | JT | iax trunking, what a waste of time :P |
| 12:34.25 | [TK]D-Fender | JT : trunking is jsut about the only reason to use it period. |
| 12:34.47 | JT | yeah, well s/trunking// at will :P |
| 12:37.00 | [TK]D-Fender | JT : well IAX had to have a real point somewhere, right? :) |
| 12:37.01 | Muc_Animus | has anybody experience with sipx? |
| 12:40.38 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 12:41.16 | *** join/#asterisk nikolaev (n=nikolaev@78.83.145.98) |
| 12:41.29 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 12:41.55 | qp | anyone done a direct update to MySQL from within a conf? |
| 12:42.13 | mort_gib | http://pastebin.com/m275e9ab9 |
| 12:42.43 | mort_gib | So what is wrong with IAX trunking?? |
| 12:43.49 | mort_gib | If I did cd /usr/src. |
| 12:43.54 | mort_gib | asterisk |
| 12:44.26 | mort_gib | make samples |
| 12:45.21 | [TK]D-Fender | mort_gib: broken description. Just do "modprobe ztdummy" , "ztcfg -vvvv", and then start *. Then "zap show status" from CLI |
| 12:46.33 | mort_gib | * is not starting |
| 12:47.36 | mort_gib | http://pastebin.com/d2657360f |
| 12:47.40 | *** part/#asterisk nikolaev (n=nikolaev@78.83.145.98) |
| 12:48.25 | [TK]D-Fender | mort_gib: pastebin "ztcfg -vvvv" and then your attempt to "asterisk -gvvvvvvvvc" |
| 12:48.59 | mort_gib | I have to load ztdummy again though, but hang on |
| 12:51.15 | mort_gib | http://pastebin.com/d79d76eed |
| 12:51.43 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 12:51.44 | [TK]D-Fender | mort_gib: ALL of * starting please... |
| 12:51.55 | *** join/#asterisk ManxPower (n=manxpowe@45.sub-75-248-217.myvzw.com) |
| 12:51.59 | mort_gib | That's it, * is NOT starting.... |
| 12:52.13 | *** join/#asterisk blogbasti (n=blogbast@calypso.planet-ic.de) |
| 12:52.13 | mort_gib | I might add that this is my problem ;-) |
| 12:52.22 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 12:52.28 | [TK]D-Fender | mort_gib: I want to SEE "asterisk -gvvvvvvvvc" from the line where you ISSUE IT |
| 12:53.19 | *** join/#asterisk coppice (n=chatzill@175.202.17.210.dyn.pacific.net.hk) |
| 12:53.32 | mort_gib | Ok, hang on |
| 12:54.30 | mort_gib | http://pastebin.com/d36f5d391 |
| 13:00.05 | tzafrir_laptop | mort_gib, it is exactly because you have a zaptel span (ztdummy) but it doesn't really produce timing |
| 13:00.15 | [TK]D-Fender | mort_gib: modprobe ztdummy. Then go to your * source folder, trash it and recompile from scratch |
| 13:01.02 | mort_gib | like make clean in /usr/src/asterisk (where my source happens to be) |
| 13:01.06 | tzafrir_laptop | [TK]D-Fender, that won't help |
| 13:01.18 | [TK]D-Fender | mort_gib: TRASH |
| 13:01.31 | tzafrir_laptop | zttest shows that there's no timing provided by zaptel |
| 13:02.11 | tzafrir_laptop | The test of asterisk (as of 1.4.21?) is: |
| 13:02.52 | tzafrir_laptop | if there is actually a zaptel (kernel) that can open the timing devices, |
| 13:03.14 | tzafrir_laptop | but zaptel does not really provide timing, we know that something is bad |
| 13:03.32 | tzafrir_laptop | And in that case Asterisk will have strange issues with e.g. playback |
| 13:03.46 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 13:07.40 | Wayhigh | so.. if I created a system to look through cdr records and only direct incoming calls that we haven't ever dialed or answered before to an IVR, would anyone but me find that useful? |
| 13:08.48 | Wayhigh | I'm thinking that it would make it so that if it recognized the number then it would direct it to the right extension or group.. if not.. it can pick up and help me get rid of telemarketers.. magazine subscription peeps.. etc. |
| 13:09.46 | NoxIn- | Wayhigh: telemarketers usually hide their number |
| 13:10.09 | NoxIn- | (at least they do here) |
| 13:10.30 | *** join/#asterisk washburnello (n=johnsy@h23.238.30.71.dynamic.ip.windstream.net) |
| 13:10.33 | Wayhigh | yeah so all unknowns get answered by ivr |
| 13:12.37 | *** part/#asterisk washburnello (n=johnsy@h23.238.30.71.dynamic.ip.windstream.net) |
| 13:15.00 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbplc.com) |
| 13:15.02 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:15.09 | Wayhigh | hmm.. someone give me an unlisted number to try looking up |
| 13:20.51 | n3hxs | OK "Unknown" |
| 13:21.00 | errr | Wayhigh: 8675309 |
| 13:21.01 | n3hxs | they called last night. |
| 13:21.49 | Muc_Animus | can anybody explain what is meant with "415: Unsupported Media Type" when dialing UM via SipX ? |
| 13:24.40 | wizzy_ | should I be able to plug a POTS phone straight into an Astribank ? |
| 13:24.58 | Wayhigh | errr: That's Jenny.. |
| 13:25.15 | Wayhigh | has 8675309 on fwd |
| 13:25.45 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 13:25.49 | *** join/#asterisk Sniper_VOIP (n=michofr@mail.splendor.net) |
| 13:26.04 | Wayhigh | anyways.. I may have found something else to convert to an awesome CNAM lookup tool |
| 13:26.09 | Wayhigh | one that actually gets unlisted numbers too and is free |
| 13:27.18 | *** join/#asterisk moy (n=moy@nat/ibm/x-b872ca95e7124cea) |
| 13:28.01 | rickross | I'm trying to track down how an unauthorized user is gaining access to my * system |
| 13:29.12 | rickross | is there any way that someone could make outbound calls (or even just REGISTER) without knowing the correct password? |
| 13:29.22 | *** join/#asterisk E-bola (n=bola@smtp.techbiz.dk) |
| 13:29.28 | rickross | eg - a password bypass or exploit? |
| 13:29.39 | E-bola | Hi all, im having a hard time getting my new polycom soundstation 6000 configured with asterisk |
| 13:29.49 | E-bola | the polycom seems extremely weird compared to snom/linksys phones |
| 13:29.52 | Qwell | rickross: see SECURITY.txt |
| 13:29.55 | E-bola | Are anybody experienced with polycoms? |
| 13:29.55 | Wayhigh | depending on the systemyou are using yes, it's possible.. it's possible they just got your password too |
| 13:30.25 | _ShrikE | There are also known exploits that you should be sure to patch against |
| 13:30.45 | [TK]D-Fender | E-bola: Plenty of us. And the WIKI has a decent guide |
| 13:31.00 | *** join/#asterisk Dalvion (n=Somethin@171.209.dsl.syd.iprimus.net.au) |
| 13:31.14 | E-bola | [TK]D-Fender: All the docs/guides i find deal with deploying in big nubmers, via tftp etc. I dont want to do that, i simply need to setup 1 phone |
| 13:31.23 | E-bola | But the web interface is really poor |
| 13:31.31 | [TK]D-Fender | rickross: You aren't showing us anything so we can't advise much |
| 13:31.32 | E-bola | the username field is called address apparently? |
| 13:31.36 | E-bola | wtf is up with that... |
| 13:32.02 | [TK]D-Fender | E-bola: Polycom should remover the web interface outright. |
| 13:32.04 | rickross | thx, Qwll - I will look |
| 13:32.14 | [TK]D-Fender | E-bola: Go provision it like it was intended |
| 13:32.52 | E-bola | [TK]D-Fender: are you sayingf its necesary to setup tftp servers boottp servers and what not just to get the phone to register with asterisk? |
| 13:32.55 | E-bola | thats horrible, if true |
| 13:33.10 | rickross | [TK]D - I will grab some log data and place it in a pastebin |
| 13:33.29 | [TK]D-Fender | E-bola: Necessary? probably not. Advisable? Yes. And I only use FTP personally. |
| 13:34.00 | E-bola | Its just plain dumb, there is no benifit at all just 5x more time spend on itm, compared to setting a few variables in a web interface :( |
| 13:35.32 | [TK]D-Fender | E-bola: for YOU maybe not, but Polycom phones are amongst the most configurable out there |
| 13:35.51 | [TK]D-Fender | E-bola: And plenty of other people out there are not whining about it. |
| 13:36.12 | E-bola | I know, i would expect it to be great. And the sound and build quality seems superb |
| 13:36.38 | E-bola | But the web interface looks terrible, i guess since most ppl provision them in greater numbers |
| 13:36.42 | ManxPower | E-bola: you can easily configure the phone to work with Asterisk. However, that's about all you can do with the polycom web interface |
| 13:37.28 | E-bola | ManxPower: Ive set both the address field and the username field to a sip account ont he asterisk server. the polycom apears to send the address field value to asterisk as username, but its not working |
| 13:37.44 | E-bola | Will sip debug show me precisely what the phone is sending (including user/password) ? |
| 13:38.05 | ManxPower | E-bola: I didn't say it was working for you. I said you can configure it to work with Asterisk by using only the GUI. |
| 13:38.57 | E-bola | Have you done so? |
| 13:38.59 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 13:39.06 | ManxPower | E-bola: Yes. Multiple times. |
| 13:39.25 | E-bola | Where do i configure the user and password and asterisk server ip? |
| 13:39.33 | E-bola | It sounds simple but i find the GUI very confusing |
| 13:39.43 | [TK]D-Fender | E-bola: Free tip : the field labelled "Address" is NOT THE IP ADDRESS. This should be your sip USERNAME |
| 13:39.49 | ManxPower | The vast majority of our 150 or so Polycom phones are provisioned using FTP, but there are a few remote phones that were configured manually |
| 13:40.18 | E-bola | [TK]D-Fender: yep i figured that much, If thats the sip username, what is the polycom auth user field for? |
| 13:40.34 | [TK]D-Fender | E-bola: CID VS authuser. Standard SIP stuff |
| 13:41.29 | E-bola | tries another account |
| 13:42.08 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:42.42 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:42.56 | E-bola | Nope, same error. It keeps giving me Username/auth name mismatch |
| 13:43.00 | jaytee | morning folks |
| 13:43.08 | E-bola | the username is correct though |
| 13:43.26 | E-bola | and ive filled out Auth Password with the sip account's password |
| 13:43.32 | E-bola | any clues to what I mgiht be doing wrong? |
| 13:43.43 | [TK]D-Fender | E-bola: SHOW US |
| 13:43.53 | [TK]D-Fender | E-bola: We are not going to play this "psychic" game today |
| 13:44.00 | E-bola | What you wanna see? |
| 13:44.14 | [TK]D-Fender | E-bola: the config on the phone & your server obviously |
| 13:44.24 | E-bola | alright |
| 13:44.40 | bradleyprice86 | E-bola: I'm not an expert, but I did have trouble with one polycom that I setup. It would never authenticate. I finally had to reset to default settings and reconfigur to get it to work correctly. |
| 13:44.58 | *** join/#asterisk mmartinn (n=martins@n128-227-41-215.xlate.ufl.edu) |
| 13:45.53 | E-bola | wonders if he can pull the phones config in a text format somehow... |
| 13:47.47 | E-bola | transport=naptr... wonder what that is |
| 13:47.54 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:48.17 | ManxPower | Ebola: leave it alone |
| 13:48.24 | ManxPower | you only need to set like 3 or 4 options |
| 13:48.31 | ManxPower | user/auth info and passwords |
| 13:48.43 | ManxPower | and server address, of course |
| 13:49.05 | E-bola | I tried setting the options on the phone it self now, to see if that works |
| 13:49.22 | E-bola | i have a feeling its somehow mixing up the interface user password with the sip user password |
| 13:50.10 | E-bola | hmm nope still Username/auth name mismatch |
| 13:51.01 | *** join/#asterisk scampbell (n=scampbel@mail.contplastics.com) |
| 13:51.03 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-068-200.pools.arcor-ip.net) |
| 13:51.06 | wizzy_ | the only zaptel (actually dahdi) hardware I have is an astribank. Is that a good enough timer source ? I have been loading dahdi_dummy for that, after the xpp stuff, but now it is not cooperating, and now dahdi_dummy is appearing as /proc/dahdi/1 - and all the scripts break |
| 13:51.33 | E-bola | [TK]D-Fender / ManxPower: Anyway i can dump the phone config into a textfile somehow? |
| 13:51.40 | Qwell | wizzy_: You have to use Asterisk from svn right now, if you use dahdi |
| 13:51.54 | Blackvel | hi. do the asterisk cti outlook integration tools (click to dial like snapanumber) work with ZAP channels besides SIP VoIP phones? |
| 13:51.54 | Qwell | expect rc releases that support dahdi in the immediate future |
| 13:51.59 | [TK]D-Fender | E-bola: there is no "export" What you see is what you've got |
| 13:52.26 | Qwell | wizzy_: and I believe xpp modules do provide timing, though you'd probably have to ask tzafrir_laptop |
| 13:53.03 | *** join/#asterisk Meaulnes (n=Meaulnes@nat.finelight.com) |
| 13:53.15 | wizzy_ | Qwell: I am using asterisk 1.4 from svn |
| 13:53.29 | Qwell | how recent? |
| 13:53.35 | Qwell | I think that just got fixed like...yesterday |
| 13:53.35 | wizzy_ | yesterday |
| 13:53.37 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 13:53.39 | Qwell | :D |
| 13:53.43 | wizzy_ | oh |
| 13:53.50 | Qwell | try updating today.. if not, please report a bug |
| 13:54.01 | wizzy_ | the timing thing, or the dummy loads last thing ? |
| 13:54.18 | Qwell | the timing thing.. some things were trying to get timing from zap still, even with dahdi installed |
| 13:54.20 | Meaulnes | hello. Is this the appropriate channel for asking questions about MOH? |
| 13:54.27 | wizzy_ | Qwell: will do |
| 13:54.32 | *** join/#asterisk l2trace99 (n=jr@75.112.133.235) |
| 13:54.39 | Qwell | I think it was meetme, specifically |
| 13:54.45 | wizzy_ | do I need to re-pull dahdi, or asterisk ? |
| 13:54.48 | Qwell | though I may be mistaken |
| 13:54.58 | Qwell | wizzy_: best to do both, but Asterisk at least |
| 13:55.04 | wizzy_ | k |
| 13:55.19 | Qwell | you should be okay if you're running the dahdi rc |
| 13:55.53 | *** join/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at) |
| 13:56.14 | *** part/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at) |
| 13:56.18 | *** mode/#asterisk [+b *!*@mail.splendor.net] by [TK]D-Fender |
| 13:56.24 | wizzy_ | I had it working by loading dummy last, but now I am fixing my astribank config (making them all fxs_ks) the /proc/dahdi order changed |
| 13:56.38 | Meaulnes | I am running Asterisk 1.4.17 and I have 2 contexts defined in musiconhold.conf, both are using mode=files. The default context is crazy loud, and I was wondering if there was a way that I might be able to adjust the volume? |
| 13:57.10 | E-bola | [TK]D-Fender / ManxPower: There are 4 server fields... 2 under SIP and 2 under Line |
| 13:57.10 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:57.16 | E-bola | Which should i fill out? |
| 13:57.48 | mmartinn | Is anyone here well versed in Local channels? |
| 13:58.13 | [TK]D-Fender | mmartinn: Fairly. Ask your question... |
| 13:58.50 | [TK]D-Fender | Meaulnes: Either your sound files are not "normalized", or your endpoints are jacked up |
| 13:59.06 | Meaulnes | ok |
| 13:59.17 | Meaulnes | thanks ... good place to start looking |
| 13:59.34 | [TK]D-Fender | Meaulnes: I advise you to test that using *'s sample MoH |
| 13:59.42 | mmartinn | [TK]D-Fender: I'm helping someone out who's using the Manager interface and local channels (I work on the Asterisk-Java project, and this got reported as a bug). Am I supposed to infer that any time I see the two parts of a local channel (blah,1 and blah,2) that they are linked? It doesn't generate any link events for the manager interface, and if it did, it would help a *whole* lot with not having to write special cases all over for loc |
| 14:00.53 | Meaulnes | Well it was working when a vendor set it up using mpg123, but with 2 contexts, mpg123 was easting everything up. So I used sox to comvert the mp3s and switched to file mode. It is probably something amiss in the conversion of the sound files |
| 14:00.57 | *** join/#asterisk lowtek (n=anonymou@mail.heavylogic.com) |
| 14:01.12 | [TK]D-Fender | mmartinn: No, they are not linked. These are unique channels. the ",1" is so that the overall name is unique |
| 14:02.44 | *** join/#asterisk hi365_m (n=hi365@213.151.57.118) |
| 14:03.29 | mmartinn | [TK]D-Fender: I think I'm confused about how they work, then. I was under the impression the local channel did a redirection between two endpoints, and each endpoint appeared to be linked to localchannel with a ,1 or a ,2 |
| 14:03.56 | Wayhigh | anyone mind advertisements for cheap hardware on here? |
| 14:04.06 | Wayhigh | I'd rather these tdm400's I have go to people here :) |
| 14:04.16 | [TK]D-Fender | mmartinn: No, a Local channel is just dialplan executing by itself without a "user" on the other end. |
| 14:04.29 | [TK]D-Fender | mmartinn: This has nothing to do with any other channel |
| 14:04.40 | E-bola | ManxPower: Do you have any polycom ip 6000? |
| 14:04.49 | ManxPower | E-bola: no |
| 14:04.53 | mmartinn | [TK]D-Fender: So when this guy does an originate across one, how am I supposed to tie the local channel back to the previous part of the call? |
| 14:05.00 | [TK]D-Fender | E-bola: SHOW US the configs already. |
| 14:05.04 | *** join/#asterisk riksta (n=rick@92.63.131.41) |
| 14:05.04 | ManxPower | Fortunatly, all the polycoms use the same config interface. |
| 14:05.27 | [TK]D-Fender | mmartinn: "Originate"... there is no previous part of a call. this is a NEW channel |
| 14:05.40 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 14:05.53 | E-bola | [TK]D-Fender: How do you want me to show you, if i cant export it in any way? |
| 14:06.09 | [TK]D-Fender | E-bola: If you can see it we should be seeing it. |
| 14:06.11 | E-bola | http://pastebin.com/m1f432c64 |
| 14:06.13 | E-bola | is the sip.conf entry |
| 14:07.11 | ManxPower | Ebola: Have you tried ATRL-A, CTRL-C, CTRL-V ? |
| 14:08.03 | E-bola | hmm not sure that will take content of the form fields,but let me try |
| 14:08.18 | ManxPower | E-bola: I'm not sure either, but it's stupid to not try |
| 14:08.50 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:08.51 | E-bola | nope it doesnt :/ |
| 14:09.02 | mmartinn | [TK]D-Fender: I think you've answered my question. We shouldn't be treating a local channel as two channels. I kept seeing the unique names and thinking, "Well geez do I need to assume they are connected?" |
| 14:09.27 | [TK]D-Fender | mmartinn: They aren't. |
| 14:09.38 | [TK]D-Fender | E-bola: If you can see it we should be seeing it. <---- |
| 14:09.59 | E-bola | let me upload a screenshot somewhere |
| 14:10.18 | mmartinn | [TK]D-Fender: Is there any reason to pay attention to the ,1 and ,2, then? From an outside point of view? Or is that just to help with internal channel juggling? |
| 14:11.12 | mmartinn | [TK]D-Fender: In other words, can you think of why I might care about the ,1 and ,2 from an application developer's standpoint? |
| 14:11.15 | [TK]D-Fender | mmartinn: the # itself won't tell you anything unique at a glance. Only matters when you need to tie CDR's, taget with hangups, or something else.. |
| 14:11.33 | [TK]D-Fender | mmartinn: Depends if you have an app that needs to target one specifically. |
| 14:12.19 | mmartinn | [TK]D-Fender: The goal in this particular use case is a separate CDR for each leg of a call, and sending through a local channel was making it hard to follow the CDR |
| 14:12.41 | [TK]D-Fender | mmartinn: Yup. |
| 14:12.56 | [TK]D-Fender | mmartinn: Nesting calls like that is messy |
| 14:13.05 | Great_Anta_Baka | does asterisk store the cdr records in the mysql database immediately? |
| 14:13.09 | [TK]D-Fender | mmartinn: Might try to see about using Goto's instead |
| 14:13.18 | Ebola | stabs ManxPower in the tab key with the feared triskelion of the ninjas of the isle of man. |
| 14:13.21 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:13.21 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:13.23 | [TK]D-Fender | Great_Anta_Baka: Should be only when the call is finished |
| 14:13.27 | Ebola | returns to his book. |
| 14:13.43 | mmartinn | [TK]D-Fender: Agreed. I've been suggesting alternative possibilities for the person doing this, but it's now the principal of the thing, that I want it to work :) |
| 14:13.51 | Great_Anta_Baka | what could result in me have 18 days of missing cdr records in my mysql database [TK]D-Fender? |
| 14:14.20 | hi365 | a server crash; locked tables |
| 14:14.21 | [TK]D-Fender | Great_Anta_Baka: MySQL being down. Auth problems. etc |
| 14:14.48 | mmartinn | [TK]D-Fender: Thanks for the brain pickage :) |
| 14:14.53 | Great_Anta_Baka | mmmm will check that out but as far as i know the server was running pretty smoothly and nothing has been restarted |
| 14:15.07 | Great_Anta_Baka | after the 22 of last month i have records |
| 14:15.23 | Great_Anta_Baka | but not between 4th and the 22nd of last month |
| 14:15.33 | ManxPower | I'm so glad I don't bill for calls. |
| 14:15.51 | Great_Anta_Baka | i think i am screwed for last months billing |
| 14:16.07 | ManxPower | You are unless you also stored into in the csv (default) |
| 14:16.42 | Great_Anta_Baka | indeed i am screwed |
| 14:17.58 | E-bola | [TK]D-Fender / ManxPower: http://profile.imageshack.us/user/e_bola/ |
| 14:18.04 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 14:18.21 | E-bola | [TK]D-Fender / ManxPower: Thats how my web itnerface looks (sory for using such a lame service, but it seemed like the fastest way to get an image online) |
| 14:18.57 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 14:20.27 | brodiem | Great_Anta_Baka: rebuild the table just in case it's the result of a crashed table, although unlikely (dump the table before running repair) |
| 14:20.44 | *** part/#asterisk mmartinn (n=martins@n128-227-41-215.xlate.ufl.edu) |
| 14:21.55 | wizzy_ | Qwell: just recompiling asterisk is not enough - it still complains about the lack of a timer |
| 14:22.07 | wizzy_ | tries dahdi as well |
| 14:22.26 | Great_Anta_Baka | looking through the mysql logs and see OMG brokebn table broken table... thats on the 4th |
| 14:23.41 | [TK]D-Fender | E-bola: Why is "Address" blank on your Line1 page? |
| 14:23.43 | brodiem | Great_Anta_Baka: try a myisamchk -o on the table in case it actually accepted the inserts |
| 14:24.19 | Great_Anta_Baka | ok brodiem will try that out |
| 14:24.24 | brodiem | backup first! |
| 14:24.32 | Great_Anta_Baka | eish |
| 14:24.33 | Great_Anta_Baka | tah |
| 14:25.04 | Great_Anta_Baka | do i just copy the db file or do i do a dump? |
| 14:25.24 | brodiem | I always do both |
| 14:28.13 | brodiem | Great_Anta_Baka: also do a 'repair table' from the mysql console after myisamchk is finished |
| 14:28.32 | Great_Anta_Baka | ok |
| 14:30.33 | *** part/#asterisk micheluntu (n=michele@dhcp197-193.cu.mi.it) |
| 14:31.07 | *** join/#asterisk nate187 (n=nate@gw.bigrivertel.net) |
| 14:31.59 | E-bola | [TK]D-Fenderr: because i asumed it wasnt necesary to fill out. Anyway ive tried having the same value in auth username and address |
| 14:32.09 | E-bola | it still complains its bad username/auth missmatch |
| 14:33.09 | [TK]D-Fender | E-bola: SIP debug please.. |
| 14:34.49 | E-bola | Can you pipe sip debug to a file? |
| 14:34.57 | rickross | how do you use the CLI to force Hangup on a channel that seems to be stuck? |
| 14:35.07 | brodiem | rickross: soft hangup |
| 14:35.32 | rickross | thx, brodiem - is that all I need to type? What is the exact command? |
| 14:35.37 | wizzy_ | Qwell: redoing the dahdi kernel modules fixed it, thanks |
| 14:35.48 | rickross | ahh, got it |
| 14:35.52 | brodiem | rickross: help soft hangup |
| 14:37.36 | wizzy_ | I have "signalling=fxs_ks" in my asterisk/dahdi-channels.conf (same as asterisk/zapata.conf ??) but I still get no dialtone from my Astribank. What else should I look at ? |
| 14:38.16 | E-bola | [TK]D-Fender: http://pastebin.com/m9343737 |
| 14:38.22 | E-bola | crosses fingers [TK]D-Fender can make sence of it |
| 14:39.05 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:39.14 | E-bola | hmm username apers to be empty in one of the sip packets if im interpreting it correctly |
| 14:39.57 | ManxPower | Ebola: he's not going to help you until you provide the requested information. There is nothing we hate more than users that ignore requests for specific information. |
| 14:40.08 | ManxPower | He REQUESTED the info on the polycom config. |
| 14:40.37 | ManxPower | You realize this would have been fixed 20 mins ago if you had provided the requested info. |
| 14:40.41 | E-bola | ManxPower: umm yes and he got it? I pasted the url to 2 screenshots. Then he asked for sip debug output, which i just pasted |
| 14:41.23 | ManxPower | sorry, I missed that paste |
| 14:41.37 | ManxPower | I still can't actually see the paste |
| 14:41.55 | E-bola | The sip debug one, or the screenshots of the web interface? |
| 14:42.00 | ManxPower | screenshot |
| 14:42.24 | Great_Anta_Baka | brodiem what file do i do the myisam check on |
| 14:42.26 | ManxPower | oh, the site is slow as molasses in january. |
| 14:42.46 | wizzy_ | in the northern hemisphere |
| 14:42.54 | ManxPower | So slow as to be useless. I'll let [TK]D-Fender deal with the pain of 5 min loading times for images on that PoS site. |
| 14:42.54 | E-bola | ManxPower: ya as I said im afraid its a quite horrible site, but it seemed like the easiest one to get signed up to , to be able to put the images online |
| 14:43.09 | brodiem | Great_Anta_Baka: the .MYI |
| 14:43.17 | Great_Anta_Baka | kk |
| 14:43.42 | ManxPower | E-bola: easiest for you at least |
| 14:44.04 | Great_Anta_Baka | brodiem: is says cdr.MYI is usable but should be fixed |
| 14:45.08 | brodiem | Great_Anta_Baka: ok do a myisamchk -r on it |
| 14:45.08 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 14:45.15 | brodiem | Great_Anta_Baka: followed by a repair table from console |
| 14:45.29 | E-bola | ManxPower: http://kampagne.techbiz.dk/Jonas/untitled.jpg http://kampagne.techbiz.dk/Jonas/untitled2.jpg |
| 14:45.36 | E-bola | ManxPower: Those should load fast.... |
| 14:46.22 | ManxPower | E-bola: make the address: the same as Auth ID: |
| 14:46.43 | E-bola | ManxPower: I have tried it. Same error |
| 14:46.47 | lowtek | Has anyone ever seen an issue where a Polycom 550 won't initiate MOH in Asterisk 1.4? All my other polycom's do just fine, it's just the 550's. This is SIP 3.0.0.0258 |
| 14:46.51 | Ebola | stabs ManxPower in the eye with the feared triskelion of the ninjas of the isle of man. |
| 14:46.56 | Ebola | returns to his book yet again. |
| 14:47.10 | ManxPower | E-bola: then do it again. |
| 14:47.13 | E-bola | ManxPower: Display Name, Address, Auth User ID are all set to Jonas2 |
| 14:47.18 | *** part/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 14:47.21 | Great_Anta_Baka | brodiem: it still says cdr.MYI is usable but should be fixed |
| 14:47.25 | ManxPower | E-bola: not according to your pastebin! |
| 14:48.00 | ManxPower | E-bola: Now we run into a problem. Your pastebins are showing one thing and you are telling me something different. Resolve the problem before we continue. |
| 14:48.05 | brodiem | Great_Anta_Baka: even after 'repair table'? |
| 14:48.19 | E-bola | ManxPower: Will do |
| 14:48.28 | Great_Anta_Baka | yep |
| 14:48.43 | ManxPower | Good. Just let me know when your pastebins match what you are telling me you set. |
| 14:48.52 | Great_Anta_Baka | it says at the top myisamchk:warning: 1 client is using or hasnt close the table properly |
| 14:49.20 | *** join/#asterisk ElSonico (n=tav@dyn35-116.vpn.utu.fi) |
| 14:49.33 | Great_Anta_Baka | ok got rid of that |
| 14:49.44 | brodiem | Great_Anta_Baka: ahh, yeah you don't really have a choice but to stop mysqld to unlock the table |
| 14:50.39 | E-bola | ManxPower: http://kampagne.techbiz.dk/Jonas/new2.jpg and http://kampagne.techbiz.dk/Jonas/new.jpg are the current setting. Still giving me a Registration from '<sip:Jonas2@213.150.48.236>' failed for '213.150.48.226' - Username/auth name mismatch |
| 14:50.39 | brodiem | Great_Anta_Baka: you could include the force option with myisamchk but much recommended to stop mysqld |
| 14:51.10 | E-bola | ManxPower: Am I correct in understanding that the problem is that Asterisk is not receiving a username from the polycom? |
| 14:51.11 | ManxPower | E-bola: you do not have a [Jonas2] in sip.conf |
| 14:51.27 | ManxPower | it's getting the username just fine, Jonas2 |
| 14:51.29 | Great_Anta_Baka | sigh i think i lost that data brodium :/ |
| 14:51.50 | Great_Anta_Baka | my balls are going to in a vice @ next weeks it meeting |
| 14:52.02 | ManxPower | E-bola: stop thinking and start doing. |
| 14:52.09 | E-bola | <E-bola> http://pastebin.com/m1f432c64 |
| 14:52.22 | brodiem | Great_Anta_Baka: more than likely, but you'll have to fix that table at some point |
| 14:52.22 | E-bola | As you can see from my earlier paste, i certainly do have a [Jonas2] |
| 14:52.32 | Great_Anta_Baka | well its fixed now |
| 14:52.34 | Great_Anta_Baka | thanks :P |
| 14:52.52 | brodiem | Great_Anta_Baka: well at least in your meeting you tell them you setup a secondary CDR source to avoid this in the future :) |
| 14:53.01 | brodiem | i.e. CSV |
| 14:53.13 | Great_Anta_Baka | yup.. |
| 14:53.14 | ManxPower | E-bola: I don't see an auth password in that screen shot |
| 14:53.48 | E-bola | ManxPower: everytime u save the settings the field is cleared. But i asume it saves it. |
| 14:54.22 | E-bola | I guess its so the password isnt send in cleartext over http |
| 14:54.22 | ManxPower | E-bola: Do you have ANY working sip devices connected to this asterisk? |
| 14:54.32 | mmlj4 | ManxPower: have you met the new luser yet? |
| 14:54.33 | E-bola | ManxPower: yes lots. Both snom's and linksys's |
| 14:54.40 | ManxPower | mmlj4: no |
| 14:54.51 | *** join/#asterisk spokra (n=spokra@host093-179-187.sea0.speakeasy.net) |
| 14:54.56 | mmlj4 | can you spell "clueless"? |
| 14:55.12 | ManxPower | mmlj4: I can spell "I have not met him yet" |
| 14:55.30 | mmlj4 | John says he was starting at a prompt and didn't know how to su to root |
| 14:55.47 | ManxPower | and really, I don't care if the guy has the tech abilities of a turnip. As long as he can get more money for IT I'm happy. |
| 14:55.48 | mmlj4 | <PROTECTED> |
| 14:55.56 | mmlj4 | yeah, roger that |
| 14:56.07 | ManxPower | and THAT is John's major weakness |
| 14:56.12 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 14:56.18 | mmlj4 | i was introduced to him on Monday, he didn't impress me |
| 14:56.34 | ManxPower | I'll be meeting him mext week |
| 14:56.36 | lowtek | Has anyone ever seen an issue where a Polycom 550 won't initiate MOH in Asterisk 1.4? All my other polycom's do just fine, it's just the 550's. This is SIP 3.0.0.0258 |
| 14:57.24 | E-bola | ManxPower: Isnt this the problem? Authorization: Digest username="", realm="asterisk", nonce="66226ba2", uri="sip:213.150.48.236:5060", response="7caa545586283ba7acb506d563c6417e", algorithm=MD5 |
| 14:57.45 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 14:58.07 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 15:00.01 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:00.44 | ManxPower | E-bola: that is SIP debug. I don't read SIP debug for free. |
| 15:00.50 | ManxPower | [TK]D-Fender: does, however. |
| 15:01.13 | rwaite | meow. |
| 15:01.18 | E-bola | Hmmm k... |
| 15:01.46 | ManxPower | E-bola: Reading SIP debug is *hard work* |
| 15:02.00 | rwaite | as is iax debug. at least for me |
| 15:02.10 | E-bola | Sure, atleast its harder than reading normal conf's |
| 15:02.48 | E-bola | I just wouldnt think that the digest username should be empty, i dont know the SIP protocol well enough to be sure though |
| 15:02.49 | oilinki3 | ah. with realtime should I define another sip server (incoming, outgoing) as a 'friend' and not as an peer? |
| 15:03.05 | rwaite | read the rfc |
| 15:03.56 | wizzy_ | AStribank manual: ON light is no, but no H/W OK light. "Check if Astribank is synchronised with the Asterisk server or with the ATU". Translation, please ? |
| 15:04.27 | lowtek | Anyone know how to turn off silence surpression in a Polycom 550? |
| 15:04.31 | ManxPower | wizzy_: it's not plugged in |
| 15:04.43 | ManxPower | lowtek: it's off by default |
| 15:04.57 | wizzy_ | ManxPower: the power light is on, and I get the right numbers from lsusb |
| 15:05.16 | *** join/#asterisk PepOSX (n=angeldav@190.72.129.75) |
| 15:05.16 | ManxPower | wizzy_: and the kernel modules are loaded and Asterisk is started |
| 15:05.18 | ManxPower | ? |
| 15:05.21 | wizzy_ | yes |
| 15:05.31 | lowtek | ManxPower: Thanks. Do you know the xml config setting? I think ours is being enabled (an assumption based on the lack of background noise when talking to someone on the phone as compared to a 501 or 330) |
| 15:05.31 | ManxPower | wizzy_: I guess it's time to call tech support |
| 15:05.58 | ManxPower | lowtek: I don't know where it is because I have never ever needed to set it. |
| 15:06.10 | ManxPower | lowtek: why is this a problem? |
| 15:06.38 | lowtek | ManxPower: You know, it should't be, but end-users are picky. It sounds different and therefor it's a problem. Silly, really. I found the setting, it's off: voice.vadEnable="0" |
| 15:07.04 | wizzy_ | Hmm - dahdi command not found in CLI. |
| 15:07.31 | lowtek | Just a random comment -- dundi rocks. |
| 15:07.33 | ManxPower | wizzy_: sounds like you have a version mismatch then |
| 15:07.42 | wizzy_ | has more work to do |
| 15:08.32 | E-bola | w00t |
| 15:08.35 | E-bola | breakthrough |
| 15:08.39 | *** join/#asterisk ariel_ (n=ariel_@74.8.35.6) |
| 15:09.07 | tzafrir_laptop | wizzy_, if the ATU is mentioned , this must be an old version of the manual |
| 15:09.10 | E-bola | you apparently failed to mention the line key stuff |
| 15:09.39 | E-bola | ManxPower / [TK]D-Fender: Once i configured the line key stuff it registered correctly |
| 15:09.42 | Qwell | tzafrir_laptop: he had a question about whether xpp provided timing or not.. I didn't want to say yes and be wrong |
| 15:09.50 | ManxPower | E-bola: that would be the Line 1 section, which you have filled out. |
| 15:10.27 | oilinki3 | does anyone else use/try to use realtime? |
| 15:10.37 | oilinki3 | or is the static files still way to go? |
| 15:10.44 | E-bola | Well i entered 1 and 1, in the calls per line field or whatever its called |
| 15:11.20 | E-bola | Num Line Keys and Calls Per Line |
| 15:11.23 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:11.23 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:11.37 | E-bola | not sure if this is because its a new ip soundstation 6000, but you guys can note that this needs to be filled out |
| 15:11.56 | wizzy_ | tzafrir_laptop: the main problem is that my only dahdi hardware is the astribank. Seems asterisk is missing a timer. Yesterday, I loaded dahdi_dummy to use as the timer, which appeared as /proc/dahdi/5 |
| 15:11.59 | ManxPower | E-bola: never needed to be filled out on any of my phones |
| 15:12.28 | wizzy_ | Today, dahdi_dummy appears as /proc/dahdi/1 - which stuffs up all the scripts |
| 15:12.29 | E-bola | ManxPower: Maybe you never used a IP 6000 with newest firmware, i dont know |
| 15:12.33 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:12.36 | E-bola | atleast this is what needed to be filled out for it to work |
| 15:12.45 | tzafrir_laptop | wizzy_, for starters: http://docs.tzafrir.org.il/README.Astribank.html#_troubleshooting |
| 15:12.57 | tzafrir_laptop | do you have anything in /proc/zaptel ? |
| 15:12.59 | ManxPower | E-bola: now take it out and see what happens.... |
| 15:13.39 | *** join/#asterisk iamhrh (n=iamhrh@74.7.128.162) |
| 15:13.46 | wizzy_ | tzafrir_laptop: no /proc/zaptel. I *do* have /proc/dahdi/[1-5] |
| 15:13.57 | tzafrir_laptop | ok |
| 15:13.57 | *** part/#asterisk iamhrh (n=iamhrh@74.7.128.162) |
| 15:14.05 | wizzy_ | 1 is dummy, 2-5 are astribank |
| 15:14.07 | tzafrir_laptop | so you have spans . |
| 15:14.24 | wizzy_ | If I do not load dummy, asterisk refuses to start |
| 15:14.36 | tzafrir_laptop | hmm... maybe ztdummy managed to register as the master |
| 15:14.40 | tzafrir_laptop | rmmod ztdummy |
| 15:14.51 | tzafrir_laptop | are those all analog (FXS/FXO)? |
| 15:14.51 | Qwell | zt or dahdi? |
| 15:14.52 | ManxPower | tzafrir_laptop: he has no dhadi commands in the CLI |
| 15:15.01 | tzafrir_laptop | right, rmmod dahdi_dummy |
| 15:15.07 | Qwell | ManxPower: if it isn't configured right, chan_dahdi wouldn't load |
| 15:16.33 | wizzy_ | tzafrir_laptop: all analog, I want to use them for existing pots phones |
| 15:16.37 | lowtek | lol |
| 15:16.44 | Qwell | lowtek: ?.. |
| 15:16.49 | tzafrir_laptop | wizzy_, good. So the span number doesn't matter |
| 15:16.56 | wizzy_ | I rmmod, and modprobed, and now it appears as 5, which I like |
| 15:17.06 | tzafrir_laptop | don't mosprobe it back |
| 15:17.13 | wizzy_ | ok |
| 15:17.24 | ManxPower | wizzy_: if you keep doing stuff you are not told to do you'll never get this fixed. |
| 15:17.37 | Qwell | tzafrir_laptop: for future reference, do the Astribanks provide timing? |
| 15:17.44 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 15:17.49 | tzafrir_laptop | Qwell, yes |
| 15:17.54 | Qwell | k |
| 15:18.06 | wizzy_ | tzafrir_laptop: asteris will not start without dummy loaded |
| 15:18.08 | lowtek | Qwell: wrong window ... oops ... Hey, I'm a big dundi fan right now ... we just deployed it service-wide, 34 servers, very nice |
| 15:18.17 | Qwell | lowtek: nice |
| 15:18.28 | Qwell | how much volume? 34 servers is...quite a few |
| 15:18.33 | ManxPower | wizzy_: you are wrong. Some in YOUR setup is causing this. |
| 15:18.37 | wizzy_ | asterisk.c: Asterisk has detected a problem with your DAHDI configuration and will shutdown for your protection. You have 2 options: |
| 15:18.46 | wizzy_ | ok |
| 15:19.19 | ManxPower | wizzy_: Dude, tzafrir_laptop WROTE the drivers and DESIGNED the hardware for the Xorcom. If you don't listen to him just who do you think is better qualified to help you? |
| 15:20.13 | ManxPower | wizzy_: do you get any errors from "ztcfg -vvv" |
| 15:20.20 | lowtek | Qwell: 20k/minutes day per server .. We've had that volume, but the dundi deployment has greatly simplified our setup and we were able to retire our border/gateway call processors making for a much more federated setup ... |
| 15:20.22 | tzafrir_laptop | notes Oron actually wrote most of the drivers and the hardware was designed by someone else altogether. But I know them well enough :-) |
| 15:20.52 | Qwell | lowtek: that's awesome |
| 15:21.39 | Qwell | tzafrir_laptop: too modest.. :p |
| 15:21.42 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:22.19 | wizzy_ | ManxPower: dahdi_cfg -vvv == http://pastebin.com/m1b47a65c |
| 15:22.21 | lowtek | Qwell: truth be told we fought and fought how to handle getting calls to and from their destination servers, trying everything, always dismissing dundi. Looking back, we could have saved a tremendous amount of effort just deploying dundi to begin with. It looks challenging at first glance, but once you set it up it's quite elegant and easy. Kudos (again) |
| 15:22.28 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 15:23.12 | ManxPower | wizzy_: That would be a "no". |
| 15:24.20 | *** join/#asterisk aksyn (n=aksyn@cp1242056-a.schoo1.lb.home.nl) |
| 15:24.46 | ManxPower | wizzy_: ls -l /usr/lib/asterisk/modules/chan_dahdi.so What do you get? |
| 15:25.12 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:25.31 | wizzy_ | ManxPower: -rwxr-xr-x 1 root root 656995 2008-08-14 15:58 /usr/lib/asterisk/modules/chan_dahdi.so |
| 15:25.55 | ManxPower | wizzy_: the only thing left is /etc/asterisk/zapata.conf |
| 15:26.31 | ManxPower | wizzy_: in /etc/asterisk/modules.conf put noload => chan_dahdi.so then start asterisk as "asterisk -cvvv", then do a load chan_dhadi.so in the CLI. |
| 15:26.32 | Qwell | zapata.conf isn't used with dahdi |
| 15:26.50 | ManxPower | Qwell: ok then whatever the hell they call the new asterisk pstn interface config file. |
| 15:27.05 | Qwell | Zaptel-to-DAHDI.txt |
| 15:27.35 | wizzy_ | ManxPower: the "dahdi_cfg -vvv" above would not verify /etc/asterisk/chan_dahdi.conf ? |
| 15:27.36 | ManxPower | Qwell: I'll leave it up the the DAHDI user to translate zap commands into the correct commands/options/names. |
| 15:27.56 | ManxPower | wizzy_: no, it does not. It verifies /etc/dahdi.conf or /etc/zaptel.conf |
| 15:28.15 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 15:28.17 | ManxPower | wizzy_: this is standard stuff. |
| 15:28.30 | wizzy_ | ok |
| 15:28.38 | blebleble | is there a way to monitor a sip trunk that is using ip based authentication to the provider? (no user/pass so doesnt shop in sip show registry) |
| 15:28.40 | ManxPower | I'm waiting, BTW. |
| 15:29.02 | ManxPower | blebleble: "sip show registry" shows remote servers Asterisk is registered TO |
| 15:29.29 | ManxPower | Nevermind -- you used the word "trunk" and I don't help people like that. best of luck |
| 15:29.43 | oilinki3 | ManxPower: but it does not show the realtime registrations? |
| 15:29.49 | ManxPower | wizzy_: I'm still waiting |
| 15:29.58 | ManxPower | oilinki3: I know nothing of Realtime |
| 15:30.10 | blebleble | Correct it doesnt, thats my piont i know sip show registry shows but not ip based auth |
| 15:30.12 | wizzy_ | ManxPower: getting there - with tzafrir_laptop on priv |
| 15:30.13 | oilinki3 | ManxPower: ok. |
| 15:30.25 | ManxPower | wizzy_: then I will leave you to him. |
| 15:30.42 | ManxPower | oilinki3: but oddly, blebleble did not say he was using Realtime |
| 15:30.57 | oilinki3 | I just wonder if enyone is using realtime. the documentation seems to stop at some point and it's difficult to find information of it. |
| 15:31.21 | pukkita | see ya guys! |
| 15:31.42 | lowtek | Is there an official way to say DAHDI yet? |
| 15:31.49 | wizzy_ | ManxPower: thanks |
| 15:31.53 | lowtek | dah-dee? |
| 15:31.56 | lowtek | daddyy? |
| 15:31.58 | [TK]D-Fender | ~dahdi |
| 15:31.59 | jbot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 15:32.01 | [TK]D-Fender | ^^^^ |
| 15:32.07 | lowtek | Cool, thanks, TK ;p |
| 15:32.25 | ManxPower | I prefer to call it "name change without community input" 8=| |
| 15:32.37 | *** join/#asterisk wonderworld (n=ww@ip-62-143-216-14.hsi.ish.de) |
| 15:32.55 | Qwell | lowtek: russellb put something up on his blog about it...I think. |
| 15:32.56 | ManxPower | But I guess we are stuck with it now. |
| 15:33.08 | Qwell | russellb: if not, you should :D I know you made that recording... |
| 15:33.17 | russellb | what did I do? |
| 15:33.21 | russellb | about how to say it? |
| 15:33.22 | Qwell | dahdi.wav |
| 15:33.23 | Qwell | yeah |
| 15:33.26 | russellb | yeah .. |
| 15:33.30 | Qwell | or maybe you could link it :p |
| 15:33.31 | russellb | i linked it on my blog, yeah |
| 15:33.38 | Qwell | lowtek: ^^^ |
| 15:33.41 | jaytee | russell has a blog? where? |
| 15:33.45 | russellb | russellbryant.net |
| 15:33.49 | jaytee | cool, thanks |
| 15:33.52 | russellb | but it's just the link you saw from jbot |
| 15:33.57 | Qwell | oh, duh |
| 15:33.57 | lowtek | Got it, thanks!!! |
| 15:34.00 | Qwell | I totally just missed that |
| 15:34.22 | jaytee | I've wanted to make a blog but then I rarely have anything interesting to say except wiseass comments |
| 15:34.38 | russellb | feel free to put your wiseass comments on mine :-p |
| 15:34.48 | mvanbaak | jaytee: gheh, my blog has 3 posts |
| 15:34.52 | mvanbaak | that's it |
| 15:35.21 | *** part/#asterisk blebleble (i=godie@caesar.godie.net) |
| 15:38.00 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 15:38.15 | *** join/#asterisk wonderworld (n=ww@ip-62-143-216-14.hsi.ish.de) |
| 15:38.44 | bijit | russellb: which db is best for asterisk? |
| 15:38.58 | russellb | that's a loaded question ... |
| 15:39.06 | russellb | depends on the application |
| 15:40.15 | *** join/#asterisk cnile (n=can@wsip-70-183-197-158.br.br.cox.net) |
| 15:40.16 | *** join/#asterisk aliver (n=aliver@ip-216-17-160-99.rev.frii.com) |
| 15:40.18 | bijit | I don't like GUI |
| 15:40.57 | [TK]D-Fender | bijit: No-one is forcing you to use it. |
| 15:41.15 | cnile | how can I disable the 0 and # feature when leaving a voicemail? |
| 15:41.19 | aliver | I'd like to make a macro or something in my dialplan that will allow people to call their own extension from the outside and dial *99 to access their voicemail (prompted for p/w after the *99) any idea what * Application to use for this? |
| 15:41.49 | bijit | [TK]D-Fender: boss. |
| 15:42.15 | [TK]D-Fender | bijit: Ok, ONE person is. Sucks to be you. |
| 15:42.49 | jaytee | bijit, join the club |
| 15:43.11 | bijit | :( |
| 15:43.20 | jaytee | I'm in the same boat |
| 15:43.33 | russellb | throws a tomato at [TK]D-Fender and runs |
| 15:43.46 | bijit | russellb: lol |
| 15:43.54 | aliver | I take it that once someone dials an extension and gets connected to their voicemail, I can't use "exten =>" to do things in the dial plan, right? |
| 15:43.58 | jaytee | I've got to somehow glom a gui onto my present asterisk system without breaking it. I don't want to, I have to. |
| 15:44.19 | aliver | jaytee trixbox? |
| 15:44.20 | bijit | jaytee: sucks huh |
| 15:44.21 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 15:44.22 | jaytee | just so some dumbass admin can add phones and extensions with some dumbed down web interafce |
| 15:44.38 | russellb | jaytee: that's going to be near impossible on something you have already done .. |
| 15:44.39 | jaytee | aliver, don't ever use that curse word around me again |
| 15:44.46 | jaytee | russellb, I know |
| 15:44.54 | jasonwoot | I always join at the best time |
| 15:44.58 | bijit | editing files is much more simple and fun...than using cursor for tabs. |
| 15:44.58 | jaytee | but I think I've found a custom way to do it. |
| 15:45.04 | russellb | jaytee: Here is what you do. Install the Asterisk-GUI, but remove all of the tabs except for the config file editor tab. |
| 15:45.14 | russellb | jaytee: so you edit the same files through a web GUI. :) |
| 15:45.16 | jaytee | russellb, lol |
| 15:45.21 | aliver | jaytee Well what else is there? You want to write your own in Visual Basic, PHP, Haskell or something? |
| 15:45.32 | [TK]D-Fender | pulls out a straw |
| 15:45.44 | [TK]D-Fender | Anyone want a jelly-filled doughnut without the jelly? |
| 15:45.59 | jaytee | awwww, man! no jelly? no way! |
| 15:46.09 | bijit | russellb: :) |
| 15:46.29 | russellb | [TK]D-Fender: I'll take it |
| 15:46.58 | bijit | donuts make me a fatty. |
| 15:46.59 | jaytee | Everytime I go into the supermarket and go by the boxes of rasberry filled glazed Krispy Kremes I have to reach down deep and draw on the last of my reserves of will power and soldier on. |
| 15:47.09 | jaytee | donuts make everyone fatty |
| 15:47.32 | bijit | so russellb you drink alot of red-bull. |
| 15:47.40 | russellb | I would if I could afford it |
| 15:47.43 | aliver | Fat boy pills, so they are called. |
| 15:48.08 | ManxPower | Krispy Kreme are disgusting. |
| 15:48.21 | jaytee | ManxPower, yes they are but they taste soooooo good |
| 15:48.49 | ManxPower | jaytee: not even as good as the dolly madison doughnuts make 3 weeks ago 2,000 miles away. |
| 15:49.04 | *** join/#asterisk wonderworld (n=ww@ip-62-143-216-14.hsi.ish.de) |
| 15:49.07 | jaytee | ManxPower, can we agree to disagree on that? |
| 15:49.22 | ManxPower | jaytee: I agree that you are wrong. *grin* |
| 15:49.24 | lmadsen | I don't agree to that |
| 15:49.33 | jaytee | lol |
| 15:49.36 | jasonwoot | manxpower, that's just unamerican |
| 15:49.37 | jaytee | hi Leif |
| 15:49.42 | lmadsen | howdy :) |
| 15:50.43 | bijit | does asterisk need special parameter for blf to work? |
| 15:50.50 | ManxPower | jasonwoot: Um, it's VERY american |
| 15:51.01 | ManxPower | bijit: asterisk, the phones, your brain. |
| 15:51.40 | bijit | ok. |
| 15:51.41 | [TK]D-Fender | bijit: Go read the WIKI page on "presence" |
| 15:51.52 | bijit | [TK]D-Fender: ty. |
| 15:52.45 | jasonwoot | ManxPower: need your mailing address please |
| 15:52.50 | jasonwoot | https://shop.krispykreme.com/mugs5.html |
| 15:52.52 | ManxPower | is trying to call someone with dialup, one phone line, no voicemail, no call waiting. 8-) |
| 15:53.03 | ManxPower | jasonwoot: "~manxpower" |
| 15:53.40 | ManxPower | ~manxpower |
| 15:53.41 | jbot | methinks manxpower is NOT an employee of Digium. He is looking for a training/teaching job in networking and/or Asterisk. Contact: eric@fnords.org |
| 15:54.34 | *** part/#asterisk E-bola (n=bola@smtp.techbiz.dk) |
| 15:55.03 | bijit | ManxPower: you can come work here comany asterisco spanish for asterisk. |
| 15:55.58 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 15:56.01 | jaytee | I'm not looking forward to working on the Spanish part of this LumenVox IVR I'm building |
| 15:56.15 | ManxPower | no habala espanol |
| 15:56.31 | jaytee | hablo un poco |
| 15:56.42 | jaytee | pero no no fluido |
| 15:56.47 | bijit | Its Costa Rica. Spanish required. |
| 15:56.52 | ManxPower | I said that quite often when I was in Madrid |
| 15:57.10 | bijit | oh wow madrid. |
| 15:57.13 | bijit | nice. |
| 15:57.22 | jaytee | there's always the mad dash into freeway traffic as an alternative......for me, not for anyone else. |
| 15:57.49 | jaytee | or finding another job but who would hire a cranky old fart in his 50's in this economy |
| 15:58.10 | bijit | in costa rica..age doesn't matter |
| 15:58.14 | bijit | and women are fine. |
| 15:58.41 | ManxPower | I'm not much interested in women. Are the men fine? |
| 15:58.43 | jaytee | in america age matters (regardless of any laws passed) and beauty, sex and horror are the new gods |
| 15:58.47 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:59.30 | bijit | hmm you got to look for them ManxPower |
| 15:59.42 | coppice | in america age matters, unless you are running for president |
| 16:00.05 | jaytee | coppice, and then all you need is a fresh pair of Depends and a good firm handshake |
| 16:00.16 | jasonwoot | what are your crimes jaytee? |
| 16:00.46 | jaytee | jasonwoot, pardon me? I don't understand the [question] |
| 16:00.51 | coppice | littering.... and disturbing the peace |
| 16:01.06 | ManxPower | Crimes against bacteria -- the list goes on and on. |
| 16:01.22 | cnile | jaytee, I am very new, but I thought you could put FreePBX on top of an existing Asterisk install |
| 16:01.37 | jaytee | cnile, not without fubaring the dialplan logic I have at present |
| 16:01.52 | jasonwoot | what, no scientologists here? |
| 16:02.26 | tzafrir_laptop | ok. The issue wizzy_ had was an issue with xpp in dahdi . It basically didn't install the required files to /usr/share/dahdi . Working on it now |
| 16:02.36 | cnile | ah. would be sweet if there was gui you could install on a Windows box that would interface an existing Asterisk system |
| 16:03.03 | cnile | or just a extension adding gui to put on top of a windows box |
| 16:03.04 | jaytee | I have a strange kludged setup. PSTN -> PRI-> Nortel Option 11C -> outbound PRI < - > Asterisk -> PSTN |
| 16:03.19 | mags2 | jaytee got similar here |
| 16:03.22 | jaytee | with 4 digit dial between the two pbx's |
| 16:03.28 | mags2 | cant wait to drop the nortel in the ocean |
| 16:03.39 | jaytee | mags2, amen to that |
| 16:03.53 | jaytee | and kick my AT&T support people to the curb |
| 16:04.05 | lowtek | cnile: roll your own in LAMP |
| 16:04.05 | tzafrir_laptop | replace it with a sipX pbx? |
| 16:04.06 | mags2 | zackly |
| 16:04.18 | lowtek | cnile: use static_realtiem for sip.conf |
| 16:04.25 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 16:04.29 | lowtek | s/realtiem/realtime |
| 16:05.07 | jaytee | I'm using sipX on a VM to do UDP to TCP proxy to Exchange 2007 Unified Messaging |
| 16:05.09 | mags2 | cnile I bet you could install the * gui on windows webserver and point it at the remote * AMI |
| 16:05.25 | jaytee | once 1.6 is pretty solid I'll dump sipX |
| 16:05.57 | mags2 | (actually that makes no sense at all, forget it lol) |
| 16:07.04 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 16:09.11 | *** join/#asterisk tkbeat (n=tk@f051196148.adsl.alicedsl.de) |
| 16:09.19 | jaytee | a short time back, maybe a month I swore I saw a new book advertised from PACKT Publishing by Flavio Goncalves that focused on integrating Asterisk with openSER but now I can't find the book anywhere. |
| 16:10.11 | coppice | .... or whatever openser is called this week :-) |
| 16:10.14 | jaytee | maybe it wasn't in print yet and they've pulled the ads since openSER is now Kamalio |
| 16:10.35 | jaytee | I saw Kamailio or something like that and also openSIPS |
| 16:11.12 | coppice | next they'll call it YAF - yet another fork |
| 16:11.27 | Qwell | coppice: forkingSER |
| 16:11.28 | coppice | or forks'r'us |
| 16:12.15 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:12.46 | coppice | its a kinda ser-eal development |
| 16:13.00 | ManxPower | or a ser-eal killer |
| 16:13.38 | coppice | The most effective killer app I worked on was missile guidance |
| 16:14.09 | jaytee | AEGIS? |
| 16:15.35 | coppice | you mean the system for shooting down airbuses? |
| 16:16.30 | jaytee | when did the U.S. Navy shoot down an airbus? I usually avoid the "news" nowadays as it's all "fair and balanced horseshit" |
| 16:16.57 | coppice | some years ago they very famously shot one down in the gulf |
| 16:17.21 | jaytee | gulf of mexico? |
| 16:17.42 | coppice | no. the one with the arabs around it |
| 16:18.00 | jaytee | Persian Gulf? |
| 16:18.53 | coppice | shouldn't it be the Iranian Gulf or something similar these days? |
| 16:19.32 | *** join/#asterisk Deeewayne (n=Deeewayn@conference/asterisk/x-314a5b5196d6c078) |
| 16:19.32 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:20.25 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
| 16:20.31 | tzanger | hmm I could have sworn that a number with a decimal place for gains in zapata.conf meant decibels |
| 16:20.47 | tzanger | i.e. rxgain=3.0 meant a gain of 3.0dB |
| 16:21.32 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 16:22.20 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 16:23.24 | *** join/#asterisk LND (n=Lee@89.192.172.133) |
| 16:23.44 | ManxPower | tzanger: it all means DB |
| 16:23.58 | ManxPower | dB even |
| 16:24.31 | tzanger | ManxPower: odd, when I ask for rxgain of 3.0 I do not get audio twice as loud as before |
| 16:24.40 | tzanger | with rxgain of 11.0 I am getting close to doube |
| 16:24.42 | tzanger | er double |
| 16:24.55 | ManxPower | tzanger: is the volume change the same when using 3.0 .vs 3 or 11.0 .vs. 11 |
| 16:25.28 | brodiem | tzanger: 3db does not mean twice as loud |
| 16:25.30 | ManxPower | tzafrir_laptop: I never thought the gain numbers ended up being set as dB, even if they are supposed to. |
| 16:25.31 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 16:25.37 | brodiem | tzanger: 3db means twice the transmitting power |
| 16:25.57 | tzanger | brodiem: on rxgain it should be a 2x amplification |
| 16:26.04 | brodiem | tzanger: 6db is technically twice the _volume_, and for the human ear to "perceive" 2x volume, it's more like 10db |
| 16:29.47 | *** join/#asterisk chackal_sjc (n=pet@200-168-20-217.dsl.telesp.net.br) |
| 16:30.08 | mags2 | ugh it is time to start going thru the changelogs of opensips and kama'ilio and decide which one to stick with |
| 16:30.35 | *** join/#asterisk mascool (n=george@c-68-84-164-71.hsd1.mi.comcast.net) |
| 16:32.03 | coppice | tzanger: you mean you have the model that goes all the way to 11, and it still isn't loud enough? |
| 16:32.12 | mags2 | hehe |
| 16:32.39 | tzanger | coppice: precisely |
| 16:33.03 | tzanger | oh coppice: I synthesized and loaded my first ever opencores PCI master core last night |
| 16:33.15 | tzanger | I didn't do a proper timing analysis but it does load and work reasonably well |
| 16:33.20 | coppice | someone actually makes a guitar amp that goes to 11 now |
| 16:34.31 | coppice | such a pity PCI is already obsolete :-) |
| 16:34.43 | Qwell | PCI guitar amp? |
| 16:34.52 | Qwell | that conversation just took a serious shift |
| 16:35.54 | tzanger | yeah well I didn't want to pay $500 for the pci express version, and there's no free PCI express core :-) |
| 16:35.58 | tzanger | this is a good start for me |
| 16:37.21 | coppice | actually, I think PCI will be around for a while. loads of embedded stuff uses it, and changes slowly. also, it seems to be coming back to PC motherboards |
| 16:37.31 | [TK]D-Fender | coppice: I've seen amps marked like that much more than a decade ago |
| 16:37.41 | cnile | I disable "review" in voicemail.conf, but when I caller presses # while leaving a message it still says some shit, anyway to keep that bitch from saying anything when your press # while leaving a voicemail? |
| 16:39.00 | [TK]D-Fender | cnile: pastebin the call & your configs. I also hope you restarted * since applying your "change" |
| 16:39.03 | [TK]D-Fender | ~pb |
| 16:39.04 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:39.13 | cnile | ok |
| 16:41.53 | coppice | [TK]D-Fender: they probably have to pay licence fees to Spinal Tap for a patent |
| 16:43.58 | cnile | if I could just get Asterisk to ignore all DTMF I could accomplish what I am trying to do |
| 16:44.09 | cnile | is there a simple way to do that? |
| 16:46.27 | *** join/#asterisk ar3dam (n=ar3dam@189.156.217.200) |
| 16:46.31 | [TK]D-Fender | cnile: No. Please provide the pastebin. |
| 16:46.44 | jameswf-home | thinks alison could whoop your arsefor callin her a b**ch |
| 16:46.54 | jameswf-home | :)) |
| 16:47.28 | ar3dam | hi there!... how i can do, if some extension is ringing, how i can take with other extension? |
| 16:47.45 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
| 16:48.12 | jameswf-home | what? |
| 16:48.20 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 16:48.21 | [TK]D-Fender | ar3dam: Show applications like pickup" |
| 16:48.49 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 16:48.53 | jameswf-home | request a copy of [TK]D-Fender wtf to english dictionary |
| 16:49.38 | [TK]D-Fender | is fluent in gibberish |
| 16:53.38 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.static.dsl.tele.dk) |
| 16:54.43 | jameswf-home | mad tv had a "how to speak dumbass" cant fing it |
| 16:54.47 | jameswf-home | *find |
| 16:55.30 | ar3dam | [TK]D-Fender, yes how i can make the pickup? |
| 16:56.04 | [TK]D-Fender | ar3dam: look at the applications, read the isntructions. |
| 16:56.58 | jameswf-home | wait something like http://www.youtube.com/watch?v=9fDpy2qZP7I |
| 16:58.39 | wizzy_ | tzafrir_laptop: after re-generating my dahdi-channels.conf file I can make calls from a POTS phone out through the B410P to the great beyond |
| 16:59.05 | wizzy_ | tzafrir_laptop: I appreciate your help |
| 16:59.24 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 17:00.48 | *** join/#asterisk kfife (n=mIRC@home.chicagoventure.com) |
| 17:01.58 | kfife | <PROTECTED> |
| 17:02.30 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:02.42 | kfife | Anybody know if there would be any issue with putting 151000+ npa/nxx lookups in an AstDB table that I woulld create for Least call routing rate lookups? Obviously it will work, but the question is whether it is optimal. What I do NOT want is a 150k member linked list, and I do NOT want b-tree that is 150k entries tall because the values were inserted in order etc. |
| 17:03.25 | kfife | The keyspace for an LCR table (the primary key npa/nxx) is UNIQUE and EVENLY DISTRIBUTED so it's perfect for a hash (index) function, or even a balanced tree. but I don't know whether the AstDB is sophisticated enough to do efficient lookups when the table size becomes large. |
| 17:05.33 | kfife | I must not be connected properly--I can't see any chatter. |
| 17:05.35 | kfife | I'll be bakc |
| 17:05.37 | kfife | ck |
| 17:05.39 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 17:06.29 | cnile | [TK]D-Fender |
| 17:07.09 | cnile | regarding disabling "review" and disabling the caller from being allowed to hit "#" while recording a voicemail: |
| 17:07.16 | cnile | http://rafb.net/p/POA2mM68.html |
| 17:08.15 | cnile | I set review=no in voicemail.conf, and restarted asterisk, but when the caller presses # it plays the "Im sorry an error has occurred" message |
| 17:08.23 | *** join/#asterisk kfife (n=Miranda@home.chicagoventure.com) |
| 17:08.31 | [TK]D-Fender | cnile: Ok, it isn't asking to review. Whats the problem? |
| 17:08.35 | ManxPower | kfife: or you were not patient enough. |
| 17:08.36 | brodiem | cnile: -- Recording was 4 seconds long but needs to be at least 15 - abandoning |
| 17:08.54 | kfife | OK I'm back |
| 17:09.04 | [TK]D-Fender | cnile: and that recording you see is YOUR FAULT. its in your DIALPLAN. |
| 17:09.05 | ManxPower | kfife: AstDB is really a Berkley DB (d think db1, but I'm not sure), you can look up performance info on that. |
| 17:09.10 | [TK]D-Fender | cnile: -- Executing [exit-FAILED@macro-vm:1] Playback("Zap/8-1", "im-sorry&an-error-has-occured") in new stack |
| 17:09.15 | kfife | My IRC client was only showing me logins and logoffs |
| 17:09.18 | jasonwoot | you guys are wearing your IRC handle on your shirt at astricon, right? |
| 17:09.21 | kfife | now I can see chatter |
| 17:09.33 | [TK]D-Fender | kfife: Because nobody was TALKING |
| 17:09.36 | brodiem | kfife: it was working the first time but nobody said anything |
| 17:09.57 | cnile | [TK]D-Fender, thanks for your input, I'll take a look at it |
| 17:10.12 | [TK]D-Fender | kfife: And I would advise you to use a real DB for your storage. |
| 17:10.35 | *** join/#asterisk Yourname` (i=Yourname@unaffiliated/yourname/x-837320) |
| 17:10.50 | kfife | A real DB has so much overhead. |
| 17:11.11 | [TK]D-Fender | kfife: try SQLite |
| 17:11.35 | [TK]D-Fender | kfife: IIRC there is even a res file for it |
| 17:12.57 | kfife | The Berkley DB will be good enough as long as it uses either a hash function or some kind of balanced tree |
| 17:13.16 | kfife | TK: IF not I'll definitely take your suggestion. |
| 17:13.34 | kfife | How do most of you do LCR rate lookups? |
| 17:14.54 | ManxPower | kfife: I imagine most of us don't. |
| 17:15.21 | kfife | Really? |
| 17:15.46 | ManxPower | kfife: unless you are a large volume account or a carrier, you are not going to save much money. |
| 17:16.03 | ManxPower | Get a 3/cent/min account and not worry about it. |
| 17:16.48 | jeev | hmmmmmmm |
| 17:17.27 | kfife | But a SMB can easily spend $500 mo on 3¢ LD. Cutting that down to .5-1.5¢/min can save several hundreds per month into perpetuity. |
| 17:17.37 | kfife | Seems like effort well spent |
| 17:17.43 | brodiem | kfife: FWIW I'd probably write a script using php/mysql to accept the called number and do its pattern matching for the destination route, passing it back to the dial plan (AGI) |
| 17:17.59 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:18.30 | kfife | brodiem: Thanks. |
| 17:19.02 | ManxPower | kfife: exactly how much are you paying currently for LD? |
| 17:19.41 | mvanbaak | I pay 50$ for every shot of LSD |
| 17:19.55 | ManxPower | mvanbaak: you are paying way too much. |
| 17:20.07 | mvanbaak | nah, this is real good stuff |
| 17:20.12 | tzafrir_laptop | What is channel type Tor in chan_zap? (the post in asterisk-users) |
| 17:20.28 | mvanbaak | torisa ? |
| 17:20.28 | ManxPower | tzafrir_laptop: old ISA Zaptel card |
| 17:20.51 | tzafrir_laptop | ManxPower, but did it ever have a different Asterisk channel type? |
| 17:20.56 | ManxPower | IIRC it was removed in 1.6 |
| 17:21.03 | kfife | Between .5 and 3.9 depending on the route, technology and carrier. |
| 17:21.04 | coppice | I have one of those. An E1 version :-) |
| 17:21.15 | tzafrir_laptop | I can't find it in 1.4.0 |
| 17:21.16 | Qwell | tzafrir_laptop: nope |
| 17:21.21 | ManxPower | kfife: and how many mins do you do per month? |
| 17:21.47 | tzafrir_laptop | It's also The Onion Router |
| 17:22.10 | tzafrir_laptop | TDMoTOR? |
| 17:22.18 | Qwell | tzafrir_laptop: patent that |
| 17:23.07 | coppice | although I do still have a tormenta 1 card, modded to do E1, I don't have an ISA slot to plug it into any more :-) |
| 17:23.27 | jeev | cusses DTMF |
| 17:23.52 | ManxPower | coppice: donate it to the Digium Museum (formerly The Smithsonian) in 20 years. |
| 17:25.11 | coppice | the schematic for that card is very close to one we built in about 1986 or 87, when the Mitel chips it contains first came out |
| 17:25.18 | Blackvel | wow. I am testing www.phonesuite.de (outlook cti client with asterisk and ZAP integration). it seems to work very cool (for the incoming calls spoken). |
| 17:25.30 | kfife | ManxPower: we use about 15-20000 minutes depending on the time of year |
| 17:26.07 | coppice | wow == the following announcement is a sales pitch :-) |
| 17:26.36 | kfife | Buy the abdominizer!! |
| 17:26.45 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:26.46 | mvanbaak | wow = World of Warcraft |
| 17:26.46 | Blackvel | nah, i am not the programmer...i am a user |
| 17:26.48 | jeev | haha coppice |
| 17:26.48 | tzafrir_laptop | In Asterisk 1.0.0 I see: static char *typecompat = "Tor"; /* Retain compatibility with chan_tor */ |
| 17:26.50 | ManxPower | kfife: then in your case it may make sense to do LCR. |
| 17:26.57 | Qwell | tzafrir_laptop: oh dear.. |
| 17:27.05 | jeev | Qwell, where is russel? |
| 17:27.06 | tzafrir_laptop | But not in 1.2.0 |
| 17:27.12 | Qwell | jeev: hiding from you |
| 17:27.16 | jeev | crap. |
| 17:27.19 | mvanbaak | 1.0.0 ? |
| 17:27.26 | coppice | was there ever a chan_tor? I don't remember that |
| 17:27.40 | Qwell | coppice: would've been pre 1.0.. |
| 17:27.40 | mvanbaak | coppice: maybe before cvs ? |
| 17:27.44 | Blackvel | but i am searching for ages. most dailers only seem to support SIP/IAX e.g callOUT, not ZAP (BRI) channels. others only support click to dail, not incoming caller id display with outlook full contact details |
| 17:28.23 | kfife | ManxPower: Correct. I have 2 carrier rate tables that I want to load into a lighweight, fast DB that doesn't add much complexity or additional failure modes. AstDB is perfect if only it is up to the task |
| 17:28.43 | ManxPower | kfife: you mean if BerkleyDBv1 is up to the task, right? |
| 17:29.30 | kfife | ManxPower: bingo |
| 17:29.46 | Qwell | how is it not up to the task? |
| 17:29.51 | jeev | one day i'm gonna buy Digium and make everyone there sit beside me and explain DTMF. like alcoholics anonymous |
| 17:29.54 | jeev | i'll let fender watch |
| 17:30.32 | cnile | [TK]D-Fender, throw me another bone...I am looking all through the dialplan and dont see how to disable # from being processed |
| 17:30.36 | coppice | what is there to explain? 16 variations on "beep" give 16 codes |
| 17:30.44 | *** join/#asterisk mgroman (n=miles@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 17:31.01 | [TK]D-Fender | cnile: You can't. The user is always permitted to hit # to terminate the VM. |
| 17:31.03 | jeev | coppice, the point is i'm going to do that while farting.. continually.. watching them melt and hurt for all the pain and yelling they've put me through. |
| 17:31.29 | cnile | mmmm...no way to disable then eh? |
| 17:31.34 | Strom_C | jeev: so wait...your failure to understand a concept is now someone else's fault? |
| 17:31.48 | [TK]D-Fender | cnile: That would the "can't" part. |
| 17:31.57 | jeev | haha. nobody's fault but fenders |
| 17:31.59 | jeev | i mean mine. |
| 17:32.12 | [TK]D-Fender | cnile: Aside from of course going in and recoding app_voicemail.so yourself |
| 17:32.12 | cnile | ok, can I disable rx dtmf? |
| 17:32.25 | [TK]D-Fender | cnile: Not on demand. |
| 17:32.44 | cnile | I dont want asterisk to "hear" any dtmf at all |
| 17:32.59 | [TK]D-Fender | cnile: Are your calls being answered by a receptionist at all times? |
| 17:32.59 | Qwell | so...don't hit keys? |
| 17:33.11 | Qwell | cnile: There is a reason you want that - how about you explain why that is? |
| 17:33.17 | tzafrir_laptop | chan_tor.c was added in 0.1.7 |
| 17:33.19 | [TK]D-Fender | cnile: What is the problem with hearing "thank you - goodbye"? |
| 17:33.27 | Qwell | tzafrir_laptop: and then swiftly removed in 0.1.8? :D |
| 17:34.03 | tzafrir_laptop | Qwell, almost: 0.1.9 |
| 17:34.25 | tzafrir_laptop | The compat string survived all the wat through 1.0.x |
| 17:34.36 | tzafrir_laptop | loves git grep |
| 17:35.05 | cnile | I have an Avaya system and we need to record calls. I setup a button on the Avaya system that basically 1-step conferences in another extension. That extension is on the Asterisk box...so the Avaya calls extension 3000 and 3000 automatically answers, goes straight to voicemail and starts recording. I dont want the person recording the call to accidentally hit the # key while recording the call. |
| 17:35.36 | ManxPower | cnile: then don't use Voicemail to do your recording. Use something else like....Record() |
| 17:35.55 | kfife | Anybody know if channel application works for boosting volume in a sip channel. I know it was created for ZAP, but the doc says it works in 'supported' channels. Anyone ever try this? |
| 17:35.55 | [TK]D-Fender | cnile: Wait...you're having someone use VM to record a CONFERENCE? |
| 17:36.45 | [TK]D-Fender | cnile: indeed VP is the wrong tool for this. |
| 17:37.13 | cnile | its easy, they hit a button while on a call, it dials an extension on Asterisk and starts recording. Using VM gives them a web interface to search for their recordings |
| 17:37.17 | [TK]D-Fender | VM* |
| 17:37.50 | cnile | all I need is an fxo port to pick up as soon as it rings and start recording regardless of how it happens |
| 17:39.01 | ManxPower | cnile: use Record with a post hangup script that locks the mailbox, creates the msgxxx.txt file and moves the sound file into the mailbox and name it correctly. You'll have to figure out how voicemail does its stuff, but it should work just fine. |
| 17:39.23 | ManxPower | Last I saw, it used .dot locking |
| 17:39.37 | cnile | ManxPower, I am not understanding why I need to reinvent the wheel |
| 17:39.40 | Blackvel | can you tell me, how the telco PSTN is using callerID name and number? it looks like only numbers can be sent and not a callerid name like "asterisk specialist", isn't it? :) |
| 17:39.41 | cnile | whats the difference? |
| 17:41.01 | DarienWork | CID includes the calling number and the billing name |
| 17:41.48 | DarienWork | but the receiving telco is responsible for looking the number up |
| 17:42.20 | Blackvel | so sending a real name only works with Voip phones and SIP/IAX? |
| 17:42.32 | brodiem | Blackvel: an LIDB lookup is done on the calling number sent |
| 17:42.59 | brodiem | Blackvel: depends if the receiving carrier accepts the caller name it was sent or if it requires an LIDB entry |
| 17:43.34 | DarienWork | the originator sends a number, the recipient telco looks up the name... that's how I understand it, anyway |
| 17:43.38 | Blackvel | my customer where I do java consulting uses now cisco call manager and cisco voip phones. I would love to cal and have Java specialist in display :) |
| 17:44.03 | Blackvel | I see |
| 17:44.19 | Blackvel | maybe I should go and ask them how I can issue an direct ip call to them |
| 17:44.24 | brodiem | Blackvel: you can if you buy a DID from a vendor who provisions to LIDB |
| 17:44.51 | [TK]D-Fender | cnile: VM is not a "call recording and web review tool for seamless call monitoring" |
| 17:44.57 | Blackvel | cisco call manager probably is just connected to S2m |
| 17:45.08 | Blackvel | not directly for SIP/IAX to internet |
| 17:45.10 | DarienWork | does anyone know any good SIP/IAX trunking in/around Vancouver, BC? |
| 17:45.28 | DarienWork | I've found a few, but dunno how good any of them are |
| 17:45.42 | Blackvel | brodiem: I'll check LIDB with Germany Telco |
| 17:45.51 | kfife | DarienWork: Try les.net I believe they're close |
| 17:46.02 | *** join/#asterisk foexle (n=heiko@88-134-242-24-dynip.superkabel.de) |
| 17:46.10 | DarienWork | taking a look |
| 17:46.19 | DarienWork | I found BroadConnect, which seems to be based in Calgary and Vancouver |
| 17:46.55 | DarienWork | hmm, les.net is in Winnipeg |
| 17:47.07 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 17:47.30 | kfife | DarienWork: but distributed media gateways. |
| 17:47.42 | cnile | heh, most people have trouble getting Asterisk to recognize dtmf and all I want to do is the opposite! |
| 17:47.59 | kfife | Les Bester used to be a customer of Telasip.com, but started his own ITSP |
| 17:48.14 | kfife | We use Telasip. They're great. |
| 17:48.16 | DarienWork | their website is horrific |
| 17:48.21 | kfife | indeed |
| 17:48.24 | ar3dam | hi there, i try to enable the pickup in features.conf, after save i reload, but whent i dial *8 showme this in cli "Nothing to pick up for" any ideas? |
| 17:48.29 | DarienWork | and I can't find any of the information I want |
| 17:48.33 | kfife | But they've got it where it counts |
| 17:48.36 | kfife | What do you want? |
| 17:49.48 | brodiem | ar3dam: you know the other channel has to be ringing for this to work right? |
| 17:50.31 | kfife | DarienWork: I may be able to point you to the right thing |
| 17:50.37 | DarienWork | hmm |
| 17:51.07 | DarienWork | basically, a SIP trunk and a DID |
| 17:52.10 | kfife | INBOUND Flat Rate DID |
| 17:52.10 | kfife | <PROTECTED> |
| 17:52.10 | kfife | <PROTECTED> |
| 17:52.10 | kfife | <PROTECTED> |
| 17:52.10 | kfife | <PROTECTED> |
| 17:52.11 | kfife | <PROTECTED> |
| 17:52.13 | kfife | <PROTECTED> |
| 17:52.15 | kfife | INBOUND Per Minute DID |
| 17:52.17 | kfife | <PROTECTED> |
| 17:52.19 | kfife | <PROTECTED> |
| 17:52.21 | kfife | <PROTECTED> |
| 17:52.23 | kfife | OUTBOUND TERMINATION |
| 17:52.23 | jeev | STOP SCrolling |
| 17:52.25 | kfife | <PROTECTED> |
| 17:52.56 | kfife | Those are my notes on Les.net |
| 17:53.08 | ManxPower | kfife: never post anything more than 3 lines to the channel. |
| 17:53.12 | jeev | haven't you heard of pastebin ? |
| 17:53.14 | ManxPower | use pastebin |
| 17:53.16 | ManxPower | ~pb |
| 17:53.16 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:54.20 | kfife | Sorry to flood the channel. I'm new to IRC |
| 17:54.57 | *** join/#asterisk arturo_era (n=chatzill@189.171.156.125) |
| 17:56.24 | kfife | THANKS EVERYONE! Les.net rate breakdown ala KFIFE: http://rafb.net/p/Ospgi696.html |
| 17:56.56 | Yourname` | There's no lookin' back anymo, kfife. IRC is where you find an answer to everything, including an itchy scrotum. |
| 17:57.12 | Yourname` | So welcome to the world of IRC! |
| 17:57.14 | n3hxs | I use les.net and am happy with it. |
| 17:57.28 | jeev | are you two salesmen ? |
| 17:57.32 | n3hxs | nope, |
| 17:57.34 | Qwell | Yourname`: what channel would one go to to ask that? |
| 17:57.46 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 17:57.48 | Qwell | I mean, besides just randomly asking in channels you frequent |
| 17:57.54 | kfife | Ixnay on the alesmansay |
| 17:58.05 | kfife | Les.net |
| 17:58.18 | n3hxs | I used to be one, but couldn't lie enough to sell Nissans |
| 17:58.30 | Yourname` | Qwell: There's a whole network out there to dish out medical answers! |
| 17:58.52 | n3hxs | good and bad answers. |
| 17:58.57 | kfife | les.net's blended rate is not bad, but you can find better termination, but they're great for setting up a one-off did for a short period of time |
| 17:59.21 | Yourname` | jeev: No, I love sales for the revenue it generates. However, other peoples do it. |
| 17:59.35 | kfife | I always use them for CRAIGSLIST ads and then dump the number so I dont' have to publish my REAL number |
| 17:59.52 | Yourname` | kfife: You use craigsnumber.com for that. |
| 17:59.55 | *** join/#asterisk spokra (n=spokra@host093-179-187.sea0.speakeasy.net) |
| 18:00.02 | [TK]D-Fender | cnile: Just go modify app_voicemail.so |
| 18:00.33 | kfife | Yourname': can you get a LOCAL number from craigsnumber.com? |
| 18:00.35 | brodiem | I just don't publish numbers to CL.. |
| 18:00.47 | Yourname` | kfife: I don't know whats local to you. |
| 18:01.33 | kfife | I'm in Chicago, and we have many NPA's. I like to have a 773 number to indicate that I'm conveniently located even though all of my contact numbers are 773's |
| 18:01.33 | kfife | I mean 312's |
| 18:01.42 | kfife | my contact numbers are 312's |
| 18:02.23 | *** join/#asterisk murdock_ut (n=chatzill@70.99.184.194) |
| 18:02.43 | kfife | I can get free DID's from IPKall, but they're going to make non-telephiles suspicious when you numbe is in Washington State |
| 18:03.35 | Yourname` | GrandCentral it up then! |
| 18:05.26 | *** join/#asterisk jtodd (i=hnkhiooo@ns.fox-den.com) |
| 18:08.18 | kfife | Yourname': about that itchy scrotum... |
| 18:08.27 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552046.dsl.bell.ca) |
| 18:12.22 | *** join/#asterisk zoid_99 (n=chris@router.asteriasgi.com) |
| 18:12.53 | *** join/#asterisk The-Bat (n=karthik@203.199.114.33) |
| 18:13.03 | zoid_99 | so.. how can I get a CONNECT before an ENTERQUEUE in queue_log |
| 18:13.22 | zoid_99 | app_queue is psychic |
| 18:14.01 | jeev | [TK]D-Fender, my IAX - IAX connection is great... is there something wrong on my end is DTMF is working to an outbound number and not to another? through IAX? SIP - IAX - INTERNOT - IAX - SIP(ITSP) - OUTBOUND |
| 18:14.32 | [TK]D-Fender | jeev: You're going to have break it down yourself. |
| 18:14.43 | lowtek | Did the HINT functionality change between 1.2 and 1.4? Suddenly, call status isn't being relayed to Polycom phones ... Presense is enabled in the config ... Any ideas? |
| 18:15.00 | jeev | ok, i'll look at sip debug again.. cause that's pretty much where its left, iax to iax is doing it properly. |
| 18:15.05 | [TK]D-Fender | lowtek: In 1.4 you need to be "type=peer" , "call-limit=99" |
| 18:15.12 | lowtek | Ahh! Thanks, TK! |
| 18:15.21 | brodiem | zoid_99: the logic in my mind tells me that you can "connect" to a queue by Queue() being launched in the dial plan, but depending on the queue's configuration you won't necessarily be a queued caller (enterqueue) |
| 18:17.47 | [TK]D-Fender | zoid_99: The logic in my mind is you haven't shown us the backup |
| 18:18.04 | zoid_99 | ok |
| 18:18.19 | zoid_99 | i'll pastebin it |
| 18:19.30 | ar3dam | hi kfife... i make the question on this room about the pickup, r u there? |
| 18:19.38 | brodiem | err yeah I realized what I said makes no sense, heh |
| 18:19.58 | zoid_99 | http://pastebin.com/m2a136554 |
| 18:20.14 | zoid_99 | let me find the actual file it was pulled from |
| 18:21.46 | [TK]D-Fender | zoid_99: Interesting. |
| 18:22.00 | brodiem | zoid_99: check the unix timestamps in the flatfile log? |
| 18:22.05 | [TK]D-Fender | ar3dam: And I pointed you to the list of pickup apps. GO READ |
| 18:22.19 | zoid_99 | that's what I'm looking for |
| 18:22.52 | lowtek | Hey TK, that was an easy fix, thanks again -- Does Asterisk 1.4 + HINT priorities + Polycom support anyting beyond just showing that a caller is on the phone? i.e., should DND and setting the status work? |
| 18:24.16 | zoid_99 | TK: looks like php is wrong here... Timestamps in the file are in order, the php conversion screwed up somehow |
| 18:24.42 | mvanbaak | yay for PHP |
| 18:25.25 | [TK]D-Fender | lowtek: Nope. |
| 18:25.54 | lowtek | TK: Thanks again for your help, it's much appreciated. |
| 18:26.18 | [TK]D-Fender | lowtek: np |
| 18:26.34 | *** join/#asterisk bl4q (n=Bl@dslb-088-066-228-150.pools.arcor-ip.net) |
| 18:26.42 | *** join/#asterisk fedya (n=fedya@rrcs-71-43-222-2.se.biz.rr.com) |
| 18:29.20 | lmadsen | Yourname`: you're even cooler in person than on IRC, lol |
| 18:29.34 | Qwell | lmadsen: are you saying he's a nub on IRC? |
| 18:30.02 | lmadsen | something like that |
| 18:30.03 | jeev | i'm too cool for you |
| 18:30.27 | jeev | leif, i said it once, i'll say it for you.. i'm gonna buy digium and make all the digium employees sit in a circle like AA and ask questions. |
| 18:30.38 | lmadsen | I'm too sexy for my shirt |
| 18:30.39 | jeev | i'm gonna ask questions |
| 18:30.58 | russellb | jeev: I sell consulting services at $100/minute |
| 18:31.08 | Qwell | 8 hour minimum? |
| 18:31.18 | russellb | nah, 1 minute minimum |
| 18:31.28 | jameswf-home | russellb: your to cheap you make the rest of us look greedy |
| 18:31.35 | russellb | jameswf-home: :) |
| 18:31.36 | *** part/#asterisk DarienWork (n=dan@209.17.173.13) |
| 18:31.48 | jeev | russellb, i got monopoly money for you |
| 18:32.06 | jameswf-home | I want internet money |
| 18:32.12 | Qwell | beanz? |
| 18:32.17 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-190-rrdg-esr-2.dynamic.isadsl.co.za) |
| 18:33.54 | jeev | has sent $1000 dollars to russellb. |
| 18:34.42 | russellb | doesn't believe you |
| 18:34.44 | Strom_C | Qwell: beenz.com buys digium for world-record ten billion beenz |
| 18:34.47 | jameswf-home | heh http://www.youtube.com/watch?v=YW6Lm9boMYk |
| 18:35.02 | Qwell | Strom_C: that's probably only like $3.50 these days |
| 18:36.12 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 18:36.48 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:38.33 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 18:41.25 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 18:42.51 | jameswf-home | chocolate rain.... |
| 18:43.35 | [TK]D-Fender | \o/ I just my FWD final payment notice..... (TRASHED) |
| 18:43.41 | [TK]D-Fender | got* |
| 18:44.00 | *** part/#asterisk fedya (n=fedya@rrcs-71-43-222-2.se.biz.rr.com) |
| 18:45.44 | *** join/#asterisk maszlo (n=reckenro@65.223.240.146) |
| 18:45.58 | mgroman | beenz is old skool |
| 18:46.04 | jaytee | Question about the Background application. As I understand it if I use an existing context in my dialplan as one of the optional parms and say the user presses 1 it will look in that context for the 1 extension and not the current context if I leave the option out. Am I right or wrong? |
| 18:46.53 | maszlo | im having a little brain fart I can't seem to find where to change the value for the time on unavailable calls before they get transferred.. can someone point me in the correct direction |
| 18:47.09 | maszlo | call fwd that is |
| 18:47.27 | *** join/#asterisk tobias (n=tobias@user-0c2hj2e.cable.mindspring.com) |
| 18:47.58 | Kobaz | is there a way to cause a channel to go on hold through the AMI (or something similar) |
| 18:48.30 | *** join/#asterisk eric_hill (i=EricHill@204.94.175.11) |
| 18:49.29 | jeev | FOR WHAT fENDER? |
| 18:49.39 | jeev | woops sorry |
| 18:49.50 | jeev | whats the final payment notice.. i just remembered i had to pay my parking ticket.. been 22 days but they didn't double it |
| 18:49.54 | jeev | i get 5 parking tickets a month at my OWN OFFICE |
| 18:50.24 | jaytee | man, you've got more sob stories than a soap opera |
| 18:51.01 | *** join/#asterisk Greek-Boy (n=email@41.222.89.77) |
| 18:51.42 | eric_hill | Can anyone tell me how the "T"ransfer option inside a Queue works? |
| 18:52.40 | russellb | magic! |
| 18:52.48 | jeev | heh |
| 18:52.51 | jeev | RUSSEL, serious question. |
| 18:52.58 | jeev | i used promo code "FREE" so i dont owe you anything |
| 18:53.11 | russellb | My serious question queue is shut down right now |
| 18:53.19 | jeev | oh, i just started it. |
| 18:53.58 | jeev | *what* am i looking for while debugging DTMF? |
| 18:54.16 | russellb | reasons why it's not working? |
| 18:54.35 | jaytee | tiny little men carrying little musical notation symbols? |
| 18:54.35 | jeev | yes sir.. like what.. tips and tricks? |
| 18:54.51 | russellb | it being there or not being there is a good start |
| 18:54.59 | jeev | jaytee, if i had tiny little men, i'd send them up your rear end |
| 18:55.00 | jaytee | jeev, it's a series of tubes! buy some Draino |
| 18:55.17 | eric_hill | Is it possible to have an escape from the queue with a single digit to get to one of 9 predetermined people? |
| 18:55.32 | jaytee | jeev, let's step outside and settle this mano y whinermoron |
| 18:55.51 | jeev | lol |
| 18:56.11 | [TK]D-Fender | eric_hill: Yes. Go read the sample queues.conf |
| 18:56.24 | [TK]D-Fender | maszlo: "core show application dial" |
| 18:56.26 | jeev | heh |
| 18:57.06 | *** join/#asterisk jmillikan (n=sa@70.228.181.123) |
| 18:57.24 | eric_hill | [TK]D-Fender: queues.conf doesn't even have the word Transfer in it anywhere... |
| 18:57.38 | [TK]D-Fender | eric_hill: BECAUSE YOU AREN'T transferring ANYTHING. |
| 18:57.45 | [TK]D-Fender | And my cpas is NEVER inverted |
| 18:57.50 | [TK]D-Fender | caps* |
| 18:58.01 | [TK]D-Fender | eric_hill: Go read the sample config. its stated clear as day |
| 18:58.34 | jeev | i'm gonna write my own book on asterisk |
| 18:58.42 | jaytee | is the pipe symbol and commas equivalent as seperators in an application? |
| 18:58.45 | Qwell | jeev: call Packt Publishing. They'll publish it |
| 18:59.06 | [TK]D-Fender | jaytee: pipes are defunct as of 1.6 |
| 18:59.16 | jeev | i'm gonna name it Asterisk 2, what it really should do. then as an example dial plan.. "get the call from 213xxxxxxx, call this extension" |
| 18:59.16 | [TK]D-Fender | jaytee: "," is a safer bet all around |
| 18:59.18 | Kobaz | is there a way to cause a channel to go on hold through the AMI (or something similar)... i can redirect the channel to a context that goes to musiconhold, but then i lose the other leg |
| 18:59.20 | jeev | everything written in plain english! |
| 18:59.36 | *** join/#asterisk aksyn (n=aksyn@cp1242056-a.schoo1.lb.home.nl) |
| 19:00.01 | [TK]D-Fender | Kobaz: No, thats an unfortunate side effect of AMI hijacking. |
| 19:00.21 | jaytee | [TK]D-Fender, yeah, that's what russell said earlier, core show application background displays pipes in the syntax example and the book uses commas. Glad I've used commas so far for my stuff |
| 19:00.28 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 19:00.43 | Kobaz | [TK]D-Fender: would it be possible to move both channels at once? |
| 19:00.56 | Kobaz | [TK]D-Fender: say, put both channels in a meetme as muted participants |
| 19:01.34 | *** join/#asterisk Greek-Boy (n=email@41.221.58.7) |
| 19:02.33 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 19:02.40 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:02.59 | [TK]D-Fender | Kobaz: Every option I've seen kills one end |
| 19:03.48 | Kobaz | mm |
| 19:08.00 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 19:08.01 | ReDNeQ | hello [TK]D-Fender |
| 19:08.05 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 19:08.07 | ReDNeQ | how are you today?? |
| 19:09.14 | [TK]D-Fender | ReDNeQ: Still breathing... |
| 19:09.38 | ReDNeQ | yeah i hear yah.. Can you spare a minute helping me figure out some time conditions.. |
| 19:10.54 | jeev | waits till fender stops breathing then will throw a party |
| 19:13.21 | [TK]D-Fender | jeev: There's a Dale Carnegie book or two you seem desperately in need of reading... |
| 19:14.42 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 19:14.56 | jeev | i can't believe after 3 years, my cage in one wilshire just had routing issues.. for 3 minutes. |
| 19:15.34 | jeev | wow i'm mesmerized |
| 19:17.45 | jeev | Fender, is it possible that i can kill a call? i got a call that is stuck there from when the route died. |
| 19:18.07 | [TK]D-Fender | jeev: "soft hangup [channelwithoutbraces]" |
| 19:18.23 | jeev | thank you |
| 19:18.46 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:19.12 | jeev | how do i know which channel it is, Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message |
| 19:19.46 | jeev | call id ? |
| 19:20.03 | jeev | nop |
| 19:29.43 | ManxPower | "show channels" |
| 19:29.47 | ManxPower | that info |
| 19:31.44 | *** part/#asterisk maszlo (n=reckenro@65.223.240.146) |
| 19:32.12 | *** join/#asterisk hydrashock (n=Ricardo@200.71.48.212) |
| 19:34.39 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 19:35.24 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 19:35.34 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 19:35.45 | exothermc | if a carrier is over capacity, what sip code should they return? |
| 19:36.52 | [TK]D-Fender | exothermc: Congestion |
| 19:37.02 | exothermc | 503? |
| 19:37.16 | [TK]D-Fender | exothermc: think thats it... |
| 19:37.29 | exothermc | but that is server specific, not carrier specific |
| 19:37.43 | [TK]D-Fender | exothermc: ... HUH? |
| 19:38.24 | exothermc | [TK]D-Fender: Say a carrier gives me 5 endpoints to reach them. if I get a 50X I believe that is only for that specific endpoint not the entire carrier. |
| 19:38.40 | aliver | Is faxing (even 9600 fax) going to work over a ulaw codec going out to an internet SIP trunker like FWD or Bandwidth.com ? |
| 19:38.40 | exothermc | so then hammer each of the 5 trying to get the call to go through. |
| 19:38.51 | ManxPower | exothermc: you know nothing about the carrier, only the end point. |
| 19:38.59 | russellb | aliver: unlikely |
| 19:39.38 | aliver | russellb Currently we make it work but we only use SIP internally up to a PRI. Is it the latency out to the net that kills the fax? |
| 19:39.39 | russellb | Fax over IP using ulaw is possible ... Fax over public internet is another story |
| 19:39.40 | exothermc | ManxPower: Yes, of course but what *should* a carrier return if none of their network can complete the call because of congestion. aka try someone else. |
| 19:39.59 | aliver | russellb that's what I suspected. |
| 19:40.06 | *** join/#asterisk Nivex (n=kjotte@user-0c8hvoj.cable.mindspring.com) |
| 19:40.10 | ManxPower | exothermc: I guess you'll have to read the SIP RFC and try to figure it out. |
| 19:40.17 | aliver | russellb Is there a heuristic for latency where you know it's going to fail, say > 30 ms? |
| 19:40.23 | ManxPower | I doubt such a situation is defined in SIP. |
| 19:40.34 | russellb | aliver: it's more than just latency though .. |
| 19:40.34 | exothermc | ManxPower: Ya I have been. |
| 19:40.46 | exothermc | I thought maybe a 600 code |
| 19:41.13 | aliver | russellb what other factors come to mind? |
| 19:41.37 | russellb | jitter, packet loss ... things done to compat those (jitter buffers with packet loss concealment, for example) |
| 19:42.02 | aliver | ahhh, that makes total sense... UDP/RTP. |
| 19:43.11 | russellb | aliver: any audio compression done along the path will also kill it |
| 19:43.33 | aliver | Anyone know what the length restrictions are for POTS lines? Can I run a POTS line over CAT5 for 435 feet? |
| 19:43.45 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:43.48 | aliver | russellb yeah, I figured the artifacts generated by compression would butcher fax. |
| 19:44.16 | *** join/#asterisk ShaunWing (n=chatzill@dsl-244-214-169.telkomadsl.co.za) |
| 19:45.42 | ShaunWing | VEry puzeled get ERROR[924] Zaptel driver not loaded issue every now and again when I try and restart asterisk and then seems to go away... any ideas? |
| 19:47.07 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 19:47.20 | *** join/#asterisk bl4q (n=Bl@dslb-088-067-047-201.pools.arcor-ip.net) |
| 19:48.13 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 19:48.28 | ShaunWing | Perhaps can be solved easily by getting Asterisk to run as non root properly |
| 19:48.52 | blebleble | i just restarted my asterisk box and now am getting spammed with "== Connect attempt from '127.0.0.1' unable to authenticate" everything seems to be working right and isnt giving much more detail of what is failling anyone have suggestions on where to start to troubleshoot,its actually its happening every 5seconds exactly if that helps (and amportal.conf and manager.conf do have matching passwords) |
| 19:49.12 | ShaunWing | I've followed instructions of the book "AsteriskFutureOf TelephonySecEdit" on page 295 onwards and get an error when running service asterisk start. The error is: cat: /var/run/asterisk.pid: No such file or directory . Please help |
| 19:49.25 | [TK]D-Fender | blebleble: you've got some process checking AMI and set up wrong. |
| 19:49.27 | ManxPower | ShaunWing: that means "asterisk not running" |
| 19:49.45 | ManxPower | [TK]D-Fender: Shall I refer GUI users to you from now on? |
| 19:50.10 | [TK]D-Fender | ManxPower: Nope, I take whatever cases I feel like, and I'm not putting much into this one. |
| 19:50.58 | ShaunWing | However service asterisk start should get it running surely? |
| 19:50.59 | blebleble | TK: thanks, think i just found it (hudlite) |
| 19:51.07 | tzanger | wtf |
| 19:51.21 | ManxPower | ShaunWing: try "asterisk -cvvv" to see if it starts or not |
| 19:51.32 | ShaunWing | It does start |
| 19:51.45 | ManxPower | then you can do a "stop now" and use service asterisk start |
| 19:52.04 | ShaunWing | However occasionally get the problem above Error 924 regarding Zaptel driver |
| 19:52.16 | ManxPower | I've never seen that error. |
| 19:52.35 | ShaunWing | I know its an issue with running as non root |
| 19:52.40 | tzanger | have you guys run across asterisk behind nat (5060 and 10000-20000 forwarded), polycom registers, asterisk accepts registration, but then spends the rest of its life retransmitting OPTIONS and NOTIFY? |
| 19:52.57 | ShaunWing | that I'm getting the cat: /var/run/asterisk.pid: No such file or directory |
| 19:53.23 | ShaunWing | I've followed all instruction in the book but no joy |
| 19:53.41 | [TK]D-Fender | tzanger: "qualify=yes" <--- |
| 19:53.56 | rob0 | Yeah, you need to know some basic Unix. Your non-root * cannot write files where it has no write permissions. |
| 19:53.57 | ManxPower | tzafrir_laptop: not really. Asterisk sends lots of OPTIONS (qualify=) and NOTIFY (mailbox=) anyway. |
| 19:54.15 | [TK]D-Fender | tzanger: Notify could also be presence. |
| 19:54.22 | tzanger | [TK]D-Fender: yes, qualify is set, but it's retransmitting (NAT=yes) #1, #2, #3... bullshit |
| 19:54.27 | tzanger | it doesn't seem to get a response after it registers |
| 19:54.35 | [TK]D-Fender | tzanger: PB it up |
| 19:54.49 | rob0 | mmmm peanut butter |
| 19:54.58 | tzanger | and if I place a call, I get no audio (and also no responses for qualify, so it becomes unreachable |
| 19:55.00 | *** join/#asterisk tobias (n=tobias@user-0c2hj2e.cable.mindspring.com) |
| 19:56.08 | [hC] | tzanger: you may just need to set externip= and localnet= in your sip.conf |
| 19:56.31 | rob0 | externip, what does that default to? |
| 19:56.41 | [hC] | no default. |
| 19:57.09 | [hC] | other than the ip address of the interface, i suppose. |
| 19:57.27 | rob0 | yeah, just wondering which one on a multi-homed box |
| 19:57.46 | tzanger | [hC]: you were right on the money |
| 19:57.50 | tzanger | they had externhost but it wasn't working |
| 19:58.04 | [hC] | tzanger: right on. |
| 19:58.13 | [TK]D-Fender | tzanger: Did they have the matching externrefresh? |
| 19:58.26 | tzanger | [TK]D-Fender: yes, 30 |
| 19:58.43 | jasonwoot | non-rhetorical question: should I even think about doing SIP via VPN for remote staff? |
| 19:58.45 | tzanger | however, I don't think I have RTP streaming just yet... |
| 19:58.47 | [TK]D-Fender | tzanger: At least it was a sane failure :) |
| 19:58.52 | tzanger | it seems to ring anyway |
| 19:58.56 | rob0 | I've noticed in my own sip debug that externip seems to choose the internal IP, not the one of the default gateway interface. |
| 19:58.57 | tzanger | nobody there to verify |
| 19:59.40 | tzanger | but it's the * box itself not sending RTP to the other side... might be because there is no call actually "up" yet |
| 20:00.03 | [hC] | tzanger: correct, they have to answer the phone before RTP moves |
| 20:00.20 | tzanger | [hC]: ok so basically hurry up and wait stuff :-) |
| 20:00.33 | rob0 | hmmm, maybe if I uncommented the localnet settings this would be fixed? |
| 20:00.53 | rob0 | I could set externip, but the IP is dynamic. |
| 20:00.56 | [hC] | if you set externip and not localnet, any phones on the local interface wont work |
| 20:01.07 | Qwell | rob0: hence the existence of externhost |
| 20:01.14 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
| 20:01.16 | rob0 | yeah I see that too |
| 20:03.03 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 20:03.13 | [TK]D-Fender | Wow for a second there I read "externhost" as "exorcist" |
| 20:03.15 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:03.28 | blebleble | should i beable to reboot a polycom 501 with "sip notify polycom-reboot $extension" ? it says notifying phone etc however phone does not reboot am i missing something |
| 20:03.29 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-78-183.vif.net) |
| 20:03.31 | tzanger | [TK]D-Fender: pretty much, eys |
| 20:03.39 | tzanger | I thought polycom phones supported the gsm codec? |
| 20:03.45 | tzanger | man I am slippiung |
| 20:03.50 | tzanger | slipping even |
| 20:04.01 | *** join/#asterisk Gershwin (n=fake@63.250.233.162) |
| 20:04.09 | rob0 | ISTM that * isn't DTRT, though. If I'm going to talk SIP with sip.example.com, why aren't we using the IP address that we would use to reach sip.example.com? |
| 20:04.15 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 20:04.15 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 20:04.47 | [TK]D-Fender | blebleble: Yes, you are. There is a guide on the WIKI for this. Go read it |
| 20:05.42 | [TK]D-Fender | tzanger: Nope, Polycom does G.711u/a, G.729, G.723, and in those rare cases G.722 |
| 20:06.42 | tzanger | hmm okay |
| 20:06.53 | ShaunWing | Say any idea how to kill ztdummy |
| 20:07.11 | ShaunWing | modprobe ztdummy |
| 20:07.20 | ShaunWing | starts it but how do I stop it |
| 20:07.22 | blebleble | [TK]D-Fender: are you speaking of the voip-info wiki? |
| 20:08.09 | ShaunWing | ps auxwww |grep asterisk |
| 20:08.11 | ShaunWing | shaunw 5282 0.0 0.0 61160 716 pts/0 S+ 22:07 0:00 grep asterisk |
| 20:08.24 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 20:08.29 | ShaunWing | Does this mean asterisk is running? |
| 20:08.48 | blebleble | ShuanWing: no |
| 20:08.57 | blebleble | its just finding your grep |
| 20:09.17 | ShaunWing | ok |
| 20:09.22 | ShaunWing | [shaunw@messaging asterisk-bin]$ ps auxwww |grep ztdummy |
| 20:09.24 | ShaunWing | shaunw 5284 0.0 0.0 61160 716 pts/0 S+ 22:08 0:00 grep ztdummy |
| 20:09.31 | ShaunWing | And ztdummy |
| 20:09.37 | [TK]D-Fender | blebleble: Yes |
| 20:09.48 | russellb | ztdummy is not a process ... |
| 20:10.03 | russellb | modprobe -r ztdummy |
| 20:10.31 | ShaunWing | ps auxwww |grep asterisk |
| 20:10.32 | ShaunWing | root 5299 0.1 0.1 447592 9480 pts/1 Sl+ 22:09 0:00 ./sbin/asterisk -cvvv |
| 20:10.34 | ShaunWing | shaunw 5337 0.0 0.0 61164 720 pts/0 S+ 22:09 0:00 grep asterisk |
| 20:10.36 | blebleble | my sip_notify is correct, thats where i originally got my info yet still doesnt seem to do anything possbly because my phones are behind a pix |
| 20:10.44 | ShaunWing | How do you see that is running now? |
| 20:10.58 | Qwell | random: my favorite command args are `ps awwfux` |
| 20:11.25 | [TK]D-Fender | blebleble: No. If the phone works at all then you have simply not followed all of the instructions. |
| 20:11.37 | blebleble | ok |
| 20:12.11 | rob0 | man woman ; cat dog ; less more ; which way |
| 20:12.12 | Wayhigh | anyone here remember where that article on unmasking unlisted numbers was? |
| 20:12.21 | ShaunWing | modprobe ztdummy where do I put it to get ztdummy to load automatically when asterisk starts? |
| 20:12.41 | *** join/#asterisk javawizard2539 (n=javawiza@c-76-23-28-244.hsd1.ut.comcast.net) |
| 20:12.54 | rob0 | Shaun would want to learn how his OS works. |
| 20:14.35 | *** part/#asterisk javawizard2539 (n=javawiza@c-76-23-28-244.hsd1.ut.comcast.net) |
| 20:16.26 | tzanger | Qwell: hahahahahaha |
| 20:16.30 | tzanger | I will have to remember that |
| 20:19.59 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 20:20.34 | Qwell | tzanger: 2nd place is netstat -plant |
| 20:20.42 | Qwell | (I like making words out of args..) |
| 20:20.44 | *** join/#asterisk MarkJenks (n=MarkJenk@yosemite.cellcom.com) |
| 20:20.57 | tzanger | Qwell: me too, but you're clearly ahead of me here |
| 20:21.01 | Qwell | :p |
| 20:21.04 | tzanger | used to use diff -Nurd |
| 20:21.08 | tzanger | or patch can't remember |
| 20:21.19 | Qwell | diff |
| 20:21.51 | rob0 | Okay, since my * isn't using NAT at all (internal connection to local SIP clients, external to remote SIP services), maybe I want a "localnet=0.0.0.0/0" setting? Or does "nat=no" do the same thing? |
| 20:22.23 | blebleble | [TK]: ok after like my 8th article i found on the subject is the answer to my question that it must be setup for FTP and not TFTP |
| 20:22.27 | *** join/#asterisk rabelais (n=blank@unaffiliated/rabelais) |
| 20:22.27 | MarkJenks | Does anyone out there have experience getting SS7 to work over AMI using the Digium TE122? |
| 20:22.29 | [hC] | rob0: if you arent using nat at all to get out to the internet, you dont need externip or externhost or localnet |
| 20:23.01 | Qwell | MarkJenks: ss7 over AMI? You may need to explain |
| 20:24.24 | rob0 | [hC], that's what I thought and how it is configured (and it works), but in my console I've seen it using "...@192.168.7.96" (which is the internal IP.) |
| 20:24.29 | MarkJenks | We are trying to get dahdi to work with out DMS100 switch, which can only do AMI,D4. All I get is "HDLC Abort!" |
| 20:24.38 | rob0 | (in SIP messages and headers) |
| 20:24.40 | MarkJenks | r/out/our |
| 20:25.03 | MarkJenks | AMI=56k |
| 20:25.48 | Blackvel | now my money bag is empty ... :) |
| 20:25.51 | Qwell | oh, THAT AMI |
| 20:25.54 | Qwell | ~ami |
| 20:25.54 | jbot | methinks ami is the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
| 20:26.05 | Qwell | I thought you meant that... I was confused.. |
| 20:26.17 | MarkJenks | Sorry about that. B8zs vs AMI |
| 20:27.04 | *** join/#asterisk Hardal (n=freesty@88.255.77.200) |
| 20:27.08 | Hardal | Hello |
| 20:27.19 | eric_hill | herro |
| 20:27.41 | mkillebrew | harro |
| 20:27.49 | Hardal | I am really stucked on somewhere.. anyone can help me?.. |
| 20:28.01 | eric_hill | I hear goo-gone is nice. |
| 20:28.21 | Hardal | i've tried but i should describe my problem i think. |
| 20:28.31 | rob0 | I am stuck on band-aid, 'cuz band-aid's stuck on me! |
| 20:28.41 | seanbright | in soviet russia, road forks you! |
| 20:28.44 | MarkJenks | Is there a better group for dahdi issues? |
| 20:28.44 | eric_hill | rob0: I thought that same thing! |
| 20:29.00 | Hardal | i am using Dial() in PHPAGI. i am trying to do |
| 20:29.06 | rob0 | Nonono! Don't describe the problem. That takes away all the fun of guessing!! |
| 20:29.23 | Hardal | if DIALSTATUS == Answer, go to xyx context |
| 20:29.36 | Hardal | but when i get ANSWER status, it's quiting with -1 |
| 20:29.46 | *** join/#asterisk hoegaatit (n=laa@c-67-160-226-244.hsd1.ca.comcast.net) |
| 20:30.07 | Hardal | exec_dial("IAX2/company","555125",60,"rg"); |
| 20:30.09 | Hardal | i am using that |
| 20:30.47 | Hardal | then getting status with DIALSTATUS |
| 20:30.53 | Hardal | it's "ANSWER" |
| 20:31.06 | [TK]D-Fender | ok, checkout time, heading home... |
| 20:31.10 | Hardal | i am trying to do, if status ANSWER go to context xyx |
| 20:31.16 | MarkJenks | I am running the latest trunks for libss7, dahdi and asterisk. |
| 20:31.17 | Hardal | but it's quitting |
| 20:31.26 | Hardal | any ideas? |
| 20:31.56 | MarkJenks | my /etc/dahdi/system.conf is span=1,1,0,d4,ami. bchan=1-23 mtp2=24 |
| 20:32.00 | Juggie | you need to tell dial not to go to the h context |
| 20:32.02 | Juggie | its an option |
| 20:32.10 | Juggie | dial terminates the call if it fails unless you tell it not to |
| 20:32.57 | MarkJenks | when I run ss7linktest I get NOTALIGNED -> IDLE, and IDLE -> NOTALIGNED. then it runs off to HDLC Abort! |
| 20:32.58 | Hardal | what is that flag Juggie? |
| 20:33.23 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 20:34.15 | MarkJenks | when I throw a loop on the T1, It does see my pointcode come back to me. Is it possible that the card I am using is having problems with 56k? |
| 20:34.20 | eric_hill | Hardal: core show application dial |
| 20:34.46 | MarkJenks | our DMS100 can only do 56k. What else can I look at for options? |
| 20:36.27 | Hardal | eric_hill and Juggie |
| 20:36.31 | Hardal | g: When the called party hangs up, exit to execute more commands in the current context. |
| 20:36.36 | Hardal | i am already using g |
| 20:36.42 | Hardal | exec_dial("IAX2/company","555125",60,"rg"); |
| 20:36.48 | Wayhigh | hmm |
| 20:37.20 | eric_hill | Hardal: what's the purpose of placing the call after the caller hangs up? Dialback? |
| 20:37.25 | Wayhigh | anyone know if vitelity passes P-Asserted-Identity? |
| 20:37.27 | *** part/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 20:37.50 | Hardal | but i am scripting an IVR, i should take back to user my top menu |
| 20:37.55 | Hardal | i mean i can't use dialback i think. |
| 20:38.22 | eric_hill | If a caller hangs up, shouldn't that be the end of the call? Or are you making a predictive dialer? |
| 20:39.08 | Hardal | when caller hangs up i am trying to stream a file: "you are disconnected" and try to keep user on my IVR |
| 20:39.46 | Hardal | an user calling somewhere from my IVR, with exec_dial() when call ends (other party hangs up) user should stay in my AGI script. |
| 20:40.22 | MarkJenks | Has anyone here used SS7 with Dahdi? |
| 20:41.11 | eric_hill | In the call context, just put an h,1,Goto(s|1) to put the caller back at the top of the context. Won't that work? |
| 20:41.12 | *** join/#asterisk Paige_ (n=Paige@208.89.241.31) |
| 20:41.22 | eric_hill | i.e. re-execute the AGI script again. |
| 20:41.39 | Paige_ | hi. anyone know the command line for sox to decreare the volume of my music on hold files? |
| 20:41.51 | Hardal | eric my context is like |
| 20:41.55 | Hardal | [chats] |
| 20:43.10 | bkruse | davevg-btwtech: you there? |
| 20:43.17 | davevg-btwtech | yes sir |
| 20:43.29 | bkruse | see priv msg |
| 20:46.57 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 20:47.51 | *** part/#asterisk Paige_ (n=Paige@208.89.241.31) |
| 20:48.02 | *** join/#asterisk nighty^ (n=nighty@217.37.109.17) |
| 20:51.09 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:51.37 | *** join/#asterisk ShaunWing (n=chatzill@dsl-244-214-169.telkomadsl.co.za) |
| 20:54.21 | jeev | FENDER!!! you just missed some crazy stuff |
| 20:54.29 | *** join/#asterisk nix8n82 (n=nate@63.162.28.92) |
| 20:55.46 | *** join/#asterisk andrewct (n=andrewct@204.213.178.234) |
| 20:57.32 | exothermc | [TK]D-Fender: Does asterisk correctly implement SRV records? Or multiple A records for a host? |
| 20:57.40 | nix8n82 | what is the best language to write asterisk agi in? |
| 20:57.53 | brodiem | nix8n82: whatever you prefer to write in |
| 20:58.03 | nix8n82 | as far a execution speed is concerned |
| 20:58.29 | brodiem | nix8n82: that depends on what you're doing :) |
| 20:58.31 | [TK]D-Fender | nix8n82: Whichever one executes at a speed you like |
| 20:58.56 | [TK]D-Fender | exothermc: No idea. |
| 21:00.18 | nix8n82 | how would you catch hang ups in C++? so it doesn't just kill the agi where it's at and would let it finish with some statiscal data |
| 21:00.23 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 21:00.50 | [TK]D-Fender | nix8n82: the same way you would in any other language |
| 21:01.03 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 21:01.11 | russellb | register a handler for SIGHUP ... |
| 21:01.19 | jeev | . |
| 21:02.04 | rickross | is there a command I can type in the CLI to kick off a particular sip registration? |
| 21:02.24 | nix8n82 | thanks russelib..would that still give me access to local variables? I have never registered a handler hor SIGHUP |
| 21:02.55 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:03.14 | mgroman | nix8n82: It might be of worth for you to look at existing AGI wrappers to see how they do it |
| 21:05.00 | jeev | russellb, one of the neighbors at the office found out i['m setting up voip here and he aske my friend if he could get it to. do you want to email the guy and tell him who he's dealing with ? |
| 21:05.28 | Qwell | jeev: hey, you want a cool flag in #asterisk? |
| 21:05.33 | jeev | nope. |
| 21:05.34 | russellb | jeev: i have no idea what the hell you are talking about |
| 21:05.35 | Qwell | +q. ask and you can have it |
| 21:05.51 | nix8n82 | mgroman: yes one would think so but I haven't found one yet that answers my question |
| 21:05.58 | jeev | i dunno what +q is on freenode, i'll have to ask around |
| 21:06.15 | mgroman | nix8n82: how to catch a hangup? |
| 21:06.24 | bkruse | russellb: I just read that and was like "maybe I am missing something from a previous conversation, because that makes no since" |
| 21:06.35 | russellb | jeev: every time you randomly bug me, the asterisk project makes less progress, just FYI :-p |
| 21:06.40 | jeev | haha |
| 21:06.44 | jeev | sorry, i stop i stop. |
| 21:06.53 | nix8n82 | and app failure |
| 21:08.37 | brodiem | anyone know if the queue-state (or whatever it was called, that allowed a second "monitoring" channel to be added to a queue member) get introduced into 1.4 anywhere? Last I remember there was a patch for 1.4.15 or something like that |
| 21:08.39 | Gershwin | test |
| 21:09.05 | Juggie | jeev, the trick is bribery |
| 21:09.13 | Juggie | like sending a pallet of red bull or dr pepper |
| 21:09.25 | brodiem | found a ref on mantis but a dead link in it |
| 21:09.26 | russellb | Juggie: +42 ! |
| 21:09.38 | jeev | Juggie: i offered it to him, he didn't accept. |
| 21:09.58 | mgroman | nix8n82: pyst has a instance variable called "_got_sighup" initiated to false. then it has a function "_handle_sighup()" that sets _got_sighup to true. and finally there is a test for hangup called "test_hangup()" and this will raise an exception if _got_sighup is in fact true |
| 21:10.31 | mgroman | runs away |
| 21:10.54 | jeev | wtf, a pallet of redbull is 2600 drinks. |
| 21:11.03 | jeev | he'd die.. and it'd cost me at least $1 each |
| 21:11.36 | [TK]D-Fender | jeev: Thats far cheaper than calling a hit... |
| 21:11.55 | [TK]D-Fender | fuggedabaoutit! |
| 21:13.16 | jeev | hmm, at least there's a bright side |
| 21:17.52 | TJNII | Well, there's my FWD final notice |
| 21:18.02 | TJNII | Time to delete that sip account.... |
| 21:18.13 | Nivex | TJNII: I got one of those too. Am I to infer that my account is going bye-bye ? |
| 21:18.28 | Qwell | Nivex: all unpaid accounts are gone |
| 21:18.29 | jeev | TJNII, customer not paying ? |
| 21:18.29 | keith4 | kinda take the F out of FWD, doesn't it? |
| 21:18.32 | russellb | I'm not sure why anyone would want to keep it ... |
| 21:18.43 | Qwell | keith4: just changes the F |
| 21:18.43 | Nivex | Free as in speech not as in beer? |
| 21:18.45 | russellb | what is 1 good reason to pay for that service? |
| 21:18.54 | TJNII | Nivex: I believe so |
| 21:18.54 | rob0 | yeah, who are they fooling? |
| 21:18.56 | Qwell | russellb: to keep it free! forever! |
| 21:18.58 | Qwell | :P |
| 21:19.02 | TJNII | jeev: FWD is going from free to pay |
| 21:19.03 | russellb | apparently .. |
| 21:19.18 | rob0 | ahhhh there's a new meaning for their "F" ... Fools |
| 21:19.29 | jeev | i've been bit by mosquitos like 50 places on my legs, i'm not exaggerating |
| 21:19.38 | Nivex | ok, I'll start carving it out of my configs with a rusty spork tomorrow |
| 21:20.10 | Blackvel | whats a good us/canada sip/iax account besides fwd? |
| 21:20.24 | Nivex | thank goodness sipphone.com (now gizmo project) is still around |
| 21:20.27 | Blackvel | what service is used by masses? |
| 21:20.27 | Nivex | would be nice if they did IAX |
| 21:20.54 | rob0 | Isn't gizmo also stanaphone? If so, they've been awful lately. |
| 21:20.58 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 21:21.07 | Blackvel | when by * is back and runs (hopefully linux doesn't crash anymore) i would like to be somehow connect to uk/us ;) |
| 21:21.54 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 21:22.22 | Hardal | in PHPAGI, after exec_dial it returns -1 |
| 21:22.25 | jeev | i set up dtmf logging on the main server and iax shows the dtmf key's for iax but lists nothing for SIP, how could i get the sip part of DTMF? sip debug isnt doin it.. |
| 21:22.27 | Hardal | and my script cannot continue |
| 21:22.29 | Hardal | any ideas? |
| 21:24.31 | rickross | we have someone who has connected their asterisk as a sip client to ours (not authorized) |
| 21:24.43 | rickross | we have no idea how they did it, and no idea how to kick them off |
| 21:24.48 | Qwell | rickross: Did you read the SECURITY.txt that I said to read? |
| 21:24.53 | rickross | Qwll, I did |
| 21:25.18 | TJNII | 5660 miles between oil changes. Oops. |
| 21:25.35 | rickross | I did not get any clear next steps from that |
| 21:26.13 | jeev | tjnii, i changed my oil every 8k lol |
| 21:26.15 | jeev | i dont care |
| 21:26.16 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 21:27.05 | TJNII | doesn't want to need any major engine work anytime soon. |
| 21:28.34 | Hardal | $agi->exec_dial("IAX2/company","555125",60,"rg"); |
| 21:28.34 | Hardal | $call_status=vget("DIALSTATUS"); |
| 21:28.34 | Hardal | $agi->verbose("Our call status" . $arama_durumu); |
| 21:28.34 | Hardal | if ($call_status == "ANSWER"){ |
| 21:28.34 | Hardal | $agi->stream_file('anonslar/'.$path.'/Operator_Busy_4_B',''); |
| 21:28.35 | Hardal | } |
| 21:28.37 | jeev | hardal |
| 21:28.38 | jeev | pastebin.com |
| 21:28.43 | jeev | ~pb |
| 21:28.44 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 21:28.44 | Hardal | OOPS |
| 21:28.46 | Hardal | sorry |
| 21:28.58 | rickross | when I do "sip show peers" I see an entry that hsouldn't be there, and I'd like to know if there is any convenient way to "deregister" just that one entry? |
| 21:28.59 | jeev | doesn't care, is under warranty |
| 21:29.37 | Hardal | http://rafb.net/p/10261L50.html |
| 21:29.44 | Hardal | anyone check my script please? |
| 21:29.53 | Hardal | i am getting -1 and script wont continue |
| 21:30.53 | Hardal | i am really stucked and need help.. |
| 21:32.59 | lesouvage | I have to configure autoprisioning for 60 polycom phones tomorow and I read the whole story on the wiki and it doesn't seems to be that complicated. Are there any pitfalls I should be aware of? |
| 21:33.33 | jeev | lesouvage, provisioning polycoms are so easy |
| 21:35.42 | lesouvage | jeev: thanks. |
| 21:36.06 | jeev | what ones are they ? |
| 21:36.35 | *** join/#asterisk _nn (n=nn@unaffiliated/nn) |
| 21:37.20 | lesouvage | jeev: soundpoint@ip |
| 21:37.50 | jeev | if it's the same method as a polycom 330.. damn man, that was easy, i added the option to my dhcpd, bound the ips to a special subnet and viola. |
| 21:38.01 | jeev | well i had to make a single config file and then write a script to deverything.. but it's so easy |
| 21:43.41 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 21:44.48 | Hardal | http://rafb.net/p/10261L50.html |
| 21:44.53 | Hardal | i am getting -1, any ideas? |
| 21:45.08 | Hardal | i can't control call after exec_dial |
| 21:45.19 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 21:47.45 | jeev | any explanations for dtmf working with the same ITSP on 1 number and not the other ? |
| 21:49.18 | Hardal | please someone help me |
| 21:49.26 | Hardal | i am really stucked |
| 21:49.34 | Hardal | its buggy or something |
| 21:49.37 | Hardal | i cannot understand |
| 21:52.19 | seanbright | where are you "getting -1"? $call_status? |
| 21:52.36 | tzafrir_laptop | Hardal, exec_dial is like having a Dial() in the dialplan, right? |
| 21:52.49 | Hardal | my console outputs: |
| 21:52.54 | Hardal | <PROTECTED> |
| 21:52.54 | Hardal | <PROTECTED> |
| 21:53.00 | Hardal | but it's not playing anything |
| 21:53.06 | Hardal | returning -1 |
| 21:53.09 | Hardal | and hangs up |
| 21:53.36 | Hardal | i can't control call if exec_dial() status ANSWER |
| 21:53.37 | seanbright | Hardal: turn on agi debugging in the CLI (run 'agi debug') |
| 21:53.49 | Hardal | oh |
| 21:53.53 | seanbright | Hardal: then place another test call and pastebin the CLI output |
| 21:53.59 | seanbright | Hardal: so that i can see it |
| 21:54.04 | Hardal | ok 1 sec |
| 21:54.26 | Hardal | i hope i can catch from console |
| 21:54.30 | Hardal | too many users rite now |
| 21:54.38 | seanbright | 'core set verbose 0' |
| 21:54.44 | seanbright | 'agi debug' |
| 21:54.49 | Hardal | ah okay |
| 21:54.55 | Hardal | verbosity was 10 :D |
| 21:56.00 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 21:56.12 | *** join/#asterisk jeffspeff (i=jeff@c-68-53-54-229.hsd1.ky.comcast.net) |
| 21:56.45 | Hardal | sorry but i cant really catch |
| 21:56.50 | Hardal | 30 lines per seconds.. |
| 21:56.56 | Hardal | too many users here rite now |
| 21:56.57 | Hardal | any ideas? |
| 21:57.00 | lesouvage | jeef: if you use a compressed codec one number falls out of the Hertz range after decompression and the other nuber doesn't. (I'm not completely sure that this is correct, but this is how i understand it) |
| 21:57.03 | seanbright | core set verbose 0 |
| 21:57.08 | Hardal | i already did |
| 21:57.13 | Hardal | when i turn on agi debug |
| 21:57.21 | Hardal | its really goes crazy |
| 21:57.27 | seanbright | lots of AGIs? |
| 21:57.53 | Hardal | no lots of users calling. i am using E1 / SS7 |
| 21:57.58 | Hardal | it a chat service IVR. |
| 21:58.11 | Hardal | also lots of AGI |
| 21:58.17 | seanbright | heh. |
| 21:58.21 | Hardal | :) |
| 21:58.43 | Hardal | sean |
| 21:59.08 | Hardal | nevermind AGI. may i use extension for checking dialstatus? |
| 21:59.21 | seanbright | yes |
| 21:59.23 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:59.23 | Hardal | like exten => s-ANSWER,... |
| 21:59.27 | seanbright | yes |
| 21:59.29 | Hardal | how can i do that? |
| 21:59.33 | Hardal | may i pm? |
| 21:59.36 | seanbright | no |
| 21:59.42 | Hardal | oh wait 1 sec |
| 21:59.44 | Hardal | going to pastebin |
| 21:59.49 | Hardal | i am pasting my context |
| 21:59.54 | seanbright | http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS |
| 22:00.03 | seanbright | that shows how to use DIALSTATUS |
| 22:00.27 | seanbright | the first example on the page shows how to jump to another extension based on DIALSTATUS |
| 22:00.36 | Hardal | but my context is little complicated.. |
| 22:00.41 | rickross | http://pastebin.ca/1172013 <- someone apparently trying (succeeding?) to register on my system over and over? |
| 22:00.42 | seanbright | Hardal: that's fine |
| 22:01.02 | seanbright | Hardal: but you have all the information you need at the link i just sent |
| 22:01.13 | seanbright | Hardal: so no need to paste your context. |
| 22:01.26 | Hardal | i know but i am messed up in my context. it's complicated |
| 22:01.34 | seanbright | Hardal: ok... well then |
| 22:01.36 | seanbright | ~book |
| 22:01.36 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 22:01.39 | seanbright | ^^^ |
| 22:03.37 | Qwell | rickross: so, why did you give them the password to 203, if you don't want them registering? |
| 22:04.00 | rickross | Qwell - we never gave it to anyone |
| 22:04.15 | seanbright | *97 for voicemail? i smell freepbx. |
| 22:04.21 | Hardal | http://rafb.net/p/nwuPiM57.html |
| 22:04.32 | Qwell | rickross: obviously somebody did... |
| 22:04.42 | Hardal | sean, how can i apply ANSWER for the groupchat?.. |
| 22:04.51 | Hardal | should i sepearete groupchat to another context? |
| 22:05.09 | seanbright | Hardal: you need to read the book |
| 22:05.52 | Hardal | sean |
| 22:05.58 | seanbright | hard |
| 22:06.00 | Hardal | if you can help me now it would be great. |
| 22:06.12 | seanbright | i'm sure it would be great. |
| 22:06.14 | Qwell | ~help |
| 22:06.21 | Qwell | damn bot |
| 22:06.26 | seanbright | he's already asked his question |
| 22:06.54 | Hardal | sean let me describe my problem |
| 22:07.00 | seanbright | Hardal: no thank you |
| 22:07.06 | seanbright | Hardal: i've grown tired of this. |
| 22:07.06 | Hardal | i want to run DEADAGI for groupchat too |
| 22:07.14 | seanbright | i am going to go get some food and head home instead. |
| 22:07.24 | lesouvage | rickross: it seems that phone 203 is trying to register and succeed. Does the used IP number ring a bell (74.52.192.52) |
| 22:07.29 | seanbright | leaves |
| 22:07.35 | Hardal | sean |
| 22:08.34 | Hardal | Qwell can you help me please? |
| 22:08.42 | Hardal | i dont understand it |
| 22:08.53 | rickross | Qwell, we just changed the device pw, restarted asterisk, and the user is immediately showing up in "sip show peers" again |
| 22:09.10 | rickross | this person does NOT have the new pw |
| 22:09.59 | rickross | I think "sip show peers" is showing the user in the table, even though they have not successfully completed the full register with authentication |
| 22:10.11 | rickross | hmm, maybe not |
| 22:10.20 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 22:10.45 | Qwell | rickross: what is dzone.com? |
| 22:11.15 | rickross | Qwell: we run a network of online community sites for tech developers |
| 22:11.38 | Qwell | so, you registering to yourself isn't normal? |
| 22:12.09 | Hardal | Qwell just 2 mins please. |
| 22:12.10 | jeev | rickross the rapper ? |
| 22:12.45 | rob0 | Rickroll with a twist. |
| 22:12.52 | jeev | push it to the limit |
| 22:13.05 | TJNII | He's never gonna give you up. |
| 22:13.15 | bkw_ | I have a voip rickroll setup |
| 22:13.24 | TJNII | Oh god. |
| 22:13.31 | rob0 | extension is 7425? |
| 22:14.46 | rickross | Qwell: no, we would not be registering to ourselves |
| 22:15.01 | rickross | where does it appear that we are? |
| 22:16.22 | rickross | this seems to be a remote asterisk system trying to access us as a sip client on 203 as s@63.214.136.151 |
| 22:16.47 | rickross | and it DOES seem that they had compromised the passwd (oh, joy) |
| 22:17.11 | Qwell | well, when you have freepbx running on...THE PUBLIC INTERNET |
| 22:17.12 | Qwell | *cough* |
| 22:17.27 | Qwell | an rc nonetheless |
| 22:17.45 | Qwell | you fix that, then we'll pretend we never had this discussion. k? |
| 22:18.23 | rickross | ever friendly :) |
| 22:18.52 | Qwell | did you at least bother to switch from the default password? |
| 22:19.05 | rickross | lemme check...yes - duh |
| 22:20.44 | bminish | Anyone got any thoughts on the dCAP ? got a potential employer offering to pay for me to do it |
| 22:21.07 | Qwell | rickross: secure your network |
| 22:21.15 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 22:23.09 | rickross | Qwell, thanks to you for that good help :) |
| 22:23.37 | russellb | bminish: it totally rocks (disclaimer: I work for digium :-p ) |
| 22:25.44 | bminish | heh, you offer it in london (I am in Ireland) is dublin on the cards at any stage |
| 22:27.01 | *** part/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 22:27.39 | Hardal | this is a wrong extension? anyone can help? |
| 22:27.40 | Hardal | exten => groupchat-ANSWER,1,AGI(Ortak/rastgele.php); |
| 22:27.53 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 22:28.09 | TJNII | Hardal: Have you read the book yet? |
| 22:28.18 | Hardal | i did |
| 22:28.28 | Hardal | but my context has extension with names |
| 22:28.30 | Hardal | it's confusing me |
| 22:28.37 | Hardal | should i seperate groupchat? |
| 22:28.47 | Hardal | or i can use that way? |
| 22:28.50 | Hardal | i am asking that.. |
| 22:28.55 | TJNII | AFAIK that ia legal. |
| 22:28.59 | TJNII | s/ia/is/ |
| 22:29.21 | Hardal | groupchat-BUSY, groupchat-ANSWER, groupchat-NOANSWER |
| 22:29.22 | Hardal | right? |
| 22:29.46 | Hardal | for controlling DIALSTATUS |
| 22:30.01 | Hardal | if ANSWER go .. if BUSY go xyz etc... |
| 22:30.17 | lesouvage | Hardal: there are more dialstatusses then this 3 |
| 22:30.34 | Hardal | i know but my dialstatus is always ANSWER |
| 22:30.44 | Hardal | i am printing it with AGI. its answer. |
| 22:30.46 | Hardal | i am trying to do |
| 22:30.55 | Hardal | if dialstatus is answer, go to xyz context |
| 22:31.07 | Hardal | but its not possible with after exec_dial |
| 22:31.19 | Hardal | its buggy or something? i am trying to understand it |
| 22:31.25 | Hardal | i am using PHPAGI |
| 22:32.35 | Hardal | it's my context: http://rafb.net/p/nwuPiM57.html |
| 22:32.43 | Hardal | and it's my AGI script: http://rafb.net/p/10261L50.html |
| 22:33.37 | *** join/#asterisk jpcansa (n=jpbenavi@190.10.2.87) |
| 22:33.57 | Hardal | its executing DEADAGI after exec_dial ends |
| 22:34.10 | Hardal | executing stream_file too, but its not streaming anything |
| 22:34.20 | Hardal | console: AGI Script Ortak/group.php completed, returning -1 |
| 22:34.25 | Hardal | and then goes to DEADAGI |
| 22:35.05 | Hardal | i want to send to user xyz context if DIALSTATUS == Answer |
| 22:39.49 | Hardal | any ideas? its really makes me crazy. i spent 3 hours about it |
| 22:40.13 | *** join/#asterisk macros73 (n=cs@c-67-163-224-69.hsd1.pa.comcast.net) |
| 22:42.00 | Hardal | jbot? |
| 22:45.26 | *** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net) |
| 22:47.42 | bijit | fbox ftw! |
| 22:48.08 | bijit | sorry always get wrong window :) |
| 22:48.18 | mchou | ok, some pls explain this to me in plain English. This is part of the help blurb associated with the * Dial command. |
| 22:48.18 | mchou | <PROTECTED> |
| 22:48.18 | mchou | <PROTECTED> |
| 22:48.18 | mchou | <PROTECTED> |
| 22:48.19 | mchou | <PROTECTED> |
| 22:48.21 | mchou | <PROTECTED> |
| 22:48.43 | russellb | Hm, that looks like English to me |
| 22:48.44 | russellb | :) |
| 22:49.03 | mchou | What would be an example of a bridged call? |
| 22:49.07 | bijit | lol |
| 22:49.26 | russellb | a bridged call is ... Inbound channel <-- bridge --> Outbound channel |
| 22:51.11 | mchou | russellb: so what exactly does this option accomplish? I mean what's all the babble about caller/callee? |
| 22:51.34 | russellb | I don't know how else to say it |
| 22:51.39 | russellb | The callee is the channel that made the call |
| 22:51.45 | russellb | that's backwards ... |
| 22:51.52 | russellb | The caller is the channel that made the call |
| 22:51.56 | mchou | lol |
| 22:52.02 | russellb | The callee is the channel that is receiving the call (outbound channel) |
| 22:52.08 | mchou | you sure you understand English :) |
| 22:52.18 | russellb | Yes. |
| 22:52.25 | russellb | I'm also pretty confident in my Asterisk understanding. |
| 22:52.38 | bijit | mchou: do you understand english? |
| 22:52.49 | *** join/#asterisk chkngumbo (n=warble@adsl-69-232-228-26.dsl.pltn13.pacbell.net) |
| 22:52.51 | chkngumbo | hi |
| 22:53.29 | mchou | yeah, but what's the purpose of sending DTMF if the inbound is "not connected" (i.e. bridged) to the outbound channel? |
| 22:53.45 | mchou | Is this some out of band means to send DTMF? |
| 22:54.01 | bijit | russellb: I am confident in you asterisk skills ;-) |
| 22:54.10 | chkngumbo | i'm a pre-newbie to asterisk and i'm wondering what hardware i might want to get for the pbx system i'm thinking of building |
| 22:54.30 | *** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 22:54.40 | russellb | It's just telling you the point in processing that the DTMF gets sent. Essentially, it is immediately after the called party answers, but before the two parties are connected together and start talking to each other. |
| 22:54.52 | Qwell | chkngumbo: we might need a little more detail than that :) |
| 22:54.59 | russellb | chkngumbo: you might want to look at the first (maybe 2nd?) chapter of the book that covers hardware |
| 22:55.01 | russellb | ~book |
| 22:55.01 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 22:55.10 | Ritzerisk | Hey i have a quicky question what port number do i need to allow for sip to work is it just 5060 udp ?? |
| 22:55.12 | Ritzerisk | thanks |
| 22:55.34 | russellb | ~ports |
| 22:55.35 | jbot | ports is, like, http://www.debian.org/ports/, or http://www.isi.edu/in-notes/iana/assignments/port-numbers, or the FreeBSD ports system etc etc, or http://www.portforward.com/routers.htm |
| 22:55.39 | russellb | damn, that's not it. |
| 22:55.55 | russellb | Ritzerisk: and whatever RTP port range you configure in rtp.conf |
| 22:56.43 | russellb | [ 10000-20000 ) by default, i think ... |
| 22:56.56 | russellb | [ == inclusive, ) == non-inclusive .. |
| 22:57.13 | Ritzerisk | hmmm so is it just the udp 5060 because im getting no audio but i see im connected ( WM6 sip client .. trying to find a good one too |
| 22:57.28 | russellb | I just said, 5060, and the ports for RTP (which is your media) |
| 22:57.45 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:58.50 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:59.03 | Ritzerisk | hmm rtp ... ill try seeing if thats in the cisco to add it.. |
| 22:59.15 | chkngumbo | i've got 2 voice lines, 4 phones, and i'd like to make it so that all the phones can be single-line and ring all phones which aren't currently busy |
| 22:59.19 | TJNII | s/ia/is/ |
| 22:59.30 | TJNII | Oops. Wrong window |
| 22:59.46 | Qwell | chkngumbo: TDM800P with 2 FXO and 4 FXS modules |
| 22:59.59 | Qwell | umm |
| 23:00.13 | Qwell | yeah, let me try that again. |
| 23:00.31 | Qwell | chkngumbo: TDM800P with 1 quad FXO module and 1 quad FXS module |
| 23:01.18 | Qwell | (you can get 2 single FXO modules instead of the quad module - it's up to you) |
| 23:01.59 | chkngumbo | i'm thinking of phasing out the second voice line, so 2 singles might make sense |
| 23:02.51 | Qwell | the way that card works, is it has 2 "slots" for modules...up to 8 ports. each "slot" can take either 1 quad module, or 2 single modules |
| 23:03.04 | Ritzerisk | darn thats alot of ports to allow though sucks i have only 1 ip right now eeek |
| 23:03.16 | Qwell | (up to 2 single modules, I should really say. having 1 single module is just fine) |
| 23:03.18 | *** join/#asterisk dlynes (n=dlynes@S01060016b68219f1.vs.shawcable.net) |
| 23:03.21 | Qwell | chkngumbo: make sense? |
| 23:03.27 | *** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net) |
| 23:03.32 | gaetronik | hi there |
| 23:03.59 | gaetronik | i have to set up a queue with a special property |
| 23:04.48 | gaetronik | if all agents are on line the call must be redirect to other num |
| 23:04.51 | gaetronik | is this possible |
| 23:04.55 | chkngumbo | do the modules change the functionality of the rj11 ports on the card? |
| 23:05.55 | Qwell | chkngumbo: yes |
| 23:06.02 | *** join/#asterisk Gringo_ (n=raf@213.219.166.60.adsl.dyn.edpnet.net) |
| 23:06.17 | Qwell | FXS modules allow you to connect phones, FXO modules allow you to connect lines |
| 23:07.01 | Gringo_ | I'm looking for a way to do attended transfers with the manager. Can this be done? Blind xfer works, but that's not always useful... |
| 23:07.35 | chkngumbo | makes sense |
| 23:07.35 | gaetronik | Gringo_, there is a patch on bugs.digium.com |
| 23:07.42 | chkngumbo | thank you, qwell |
| 23:07.59 | gaetronik | you have to change it to adapt to new versions |
| 23:08.17 | gaetronik | since there is a new option in a structure but this is quite nothing |
| 23:08.19 | obnauticus | Anyone here know of a good way to get visual voicemail on * ? |
| 23:08.53 | jaytee | visual voicemail? |
| 23:09.17 | obnauticus | Well managment of voicemail over a GUI interface |
| 23:09.25 | obnauticus | i.e. via the web |
| 23:09.33 | obnauticus | And possibly voice -> text |
| 23:09.43 | Gringo_ | gaetronik: is this it? http://bugs.digium.com/view.php?id=0010585 |
| 23:10.06 | Gringo_ | tnx, i'll look into it |
| 23:11.40 | gaetronik | Gringo_, yes |
| 23:11.57 | Gringo_ | hey, is this patch merged into 1.6? |
| 23:12.06 | gaetronik | Gringo_, no idea |
| 23:12.24 | Gringo_ | the last of the comments seems to indicate so...? |
| 23:12.44 | gaetronik | seems |
| 23:12.53 | gaetronik | i only use 1.4 |
| 23:13.14 | Gringo_ | that would really be an incentive for me to upgrade :) patches are nice, but when you need to upgrade the software to get rid of some bug, they can be a huge pain |
| 23:14.35 | Gringo_ | gaetronik: tnx for the tip, i'm off to bed |
| 23:14.40 | *** part/#asterisk Gringo_ (n=raf@213.219.166.60.adsl.dyn.edpnet.net) |
| 23:27.32 | chkngumbo | if i use a phone to ethernet gateway will i be able to use just about any voip phone i pick up at the store? |
| 23:27.46 | Qwell | chkngumbo: huh? |
| 23:28.00 | mchou | chkngumbo: you wont need a voip phone :) |
| 23:28.26 | chkngumbo | http://www.voip-info.org/wiki/view/Analog+Telephone+Adapters |
| 23:28.34 | chkngumbo | i'm looking under "Analogue Phone to Ethernet gateways" |
| 23:28.35 | Qwell | those are so you can connect an analog phone |
| 23:28.36 | mchou | chkngumbo: that's what presumably the phone to ethernet gateway is for |
| 23:29.18 | chkngumbo | oh, am i interpreting it backwards? it converts VoIP service into regular phone lines, not the other way around? |
| 23:30.22 | Qwell | chkngumbo: most ATAs have an FXS port. |
| 23:30.29 | Qwell | so it would be analog phone > ATA > Asterisk |
| 23:31.53 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 23:32.04 | Qwell | bbl |
| 23:35.51 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 23:38.37 | chkngumbo | i'm wondering if this setup makes sense: 2 Telco Lines >> Asterisk >> Ethernet/WiFi >> VoIP phones |
| 23:39.05 | chkngumbo | can that be done? |
| 23:39.38 | jeev | yea why not |
| 23:41.01 | chkngumbo | my phones can use an update and VoIP phones seem to be close to regular phones in price for cordless phones |
| 23:41.39 | chkngumbo | would that 2 Telco Lines >> Asterisk bit be done with an ATA? |
| 23:41.41 | jeev | voip phones suck walls. |
| 23:41.51 | jeev | no... |
| 23:42.04 | jeev | you'd need some sort of device from digium.com, i've never used hardware before so i wouldn't know. |
| 23:42.17 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 23:42.46 | chkngumbo | are they not as reliable as regular phones? |
| 23:42.48 | JT | err you could use ATAs to terminate those telco lines |
| 23:43.28 | jeev | oh, dunno |
| 23:43.42 | jeev | how would you use an ATA? |
| 23:43.47 | jeev | wouldn't that be using it reverse? |
| 23:44.17 | JT | they can have FXO or FXS ports |
| 23:44.27 | JT | the linksys SPA-3102 has 1 FXO port and 1 FXS port |
| 23:44.51 | jeev | why do you want to use telco and not itsp ? just curious |
| 23:45.24 | chkngumbo | no reason. actually, i was considering it |
| 23:46.38 | JT | telco is more reliable |
| 23:46.50 | chkngumbo | how much so? |
| 23:47.12 | jeev | well, obviously. |
| 23:52.14 | chkngumbo | so, is the most popular route with asterisk users to stick with POTS and regular phone connections, not dealing with conversion to ethernet/wifi and voip phones? |
| 23:52.46 | jeev | oh, well.. my suggestion and it's a weak one if you ask the op's in here and fender.. and jaytee and everyone else who hats me.. |
| 23:52.53 | jeev | get the POTS, get the digium or whteve hardware |
| 23:52.57 | jeev | and get some nice polycom |
| 23:53.00 | jeev | stay away from wifi phones |
| 23:53.20 | jeev | it's not "cool" and it makes your penis smaller. i learned it the hard ware. |
| 23:53.21 | jeev | way |
| 23:54.02 | chkngumbo | connection issues or security issues? |
| 23:55.02 | jeev | connection. it's stupid anyway |
| 23:55.15 | jeev | i had the WIP330 and had to fight and threaten linksys and actually file something to get them to help me. |
| 23:56.14 | *** join/#asterisk Dalvion (n=Somethin@139.232.dsl.syd.iprimus.net.au) |
| 23:58.13 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
