| 00:02.02 | v1s | what is a bad ms? lik 500 + ? |
| 00:03.26 | ChannelZ | Well, that's half a second |
| 00:05.10 | frigidzephyr | v1s: in what context? |
| 00:05.21 | frigidzephyr | v1s: latency on a voip trunk? |
| 00:05.26 | v1s | trunk |
| 00:05.32 | frigidzephyr | 500 is pretty bad |
| 00:05.37 | frigidzephyr | i really wouldn't want over 300 |
| 00:08.45 | v1s | so like 1-100 = good, 101-300 = ok, 300+ = bad |
| 00:10.20 | frigidzephyr | sure |
| 00:11.37 | citywok | we have 250 or so to our philippines agents and it can be annoying sometimes |
| 00:12.02 | frigidzephyr | yeah, ive heard a lot of people say anything over 150 is bad |
| 00:12.48 | WIMPy | There's a nice graph in G.114. |
| 00:13.07 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 00:13.27 | citywok | assuming there isn't much jitter you can get away with it, but people end up talking over each other a lot |
| 00:13.44 | WIMPy | That would make <=100 good, <=150 ok and <=200 acceptable. |
| 00:13.45 | citywok | for internal calls... if that's what you get it is what it is. but i wouldn't want to make sales calls with 500ms of latency |
| 00:14.09 | citywok | our 250ms has great call quality, it actually rivals our san francisco office, but the delay can be annoying. |
| 00:15.43 | v1s | and if I have sip prov - server 1 - server 2 - sip client. |
| 00:15.50 | v1s | i would add them all up |
| 00:16.03 | v1s | like sip prov to serv 1 = 30ms |
| 00:16.05 | *** join/#asterisk b2 (~ion@pdpc/supporter/active/beckb) |
| 00:16.14 | v1s | serv 1 to serv 2 = 75ms |
| 00:16.27 | v1s | serv 2 to sip client = 30ms |
| 00:16.27 | WIMPy | If bandwidth isn't a big isse, make smaller packets if you can. |
| 00:16.53 | citywok | yea fewer voice frames per packet |
| 00:17.16 | citywok | also, why do you need 75ms from serv1 to serv2? find a way to eliminate that if you can |
| 00:17.42 | citywok | or use canreinvite and let server2 talk to the media gateway directly rather than having to go through server1 |
| 00:17.49 | v1s | serv 1 in one country serv 2 in another country. how to eliminate. |
| 00:18.00 | WIMPy | That would be best. |
| 00:18.11 | citywok | canreinvite? |
| 00:18.50 | v1s | have set to: canreinvite=yes |
| 00:19.16 | citywok | that should tell asterisk to take itself out of the loop |
| 00:19.22 | citywok | if it can that is. |
| 00:20.00 | citywok | but you can't mixmonitor/monitor a call that you do that to, and i have to record everything, so i don't use the option or have much experience with it |
| 00:21.02 | v1s | what do I lookup or u have link for how to make smaller packets? |
| 00:21.36 | v1s | codecs.conf? |
| 00:21.57 | WIMPy | Yes, and the other ends as well. |
| 00:23.01 | citywok | alright, time to go to the gym. later |
| 00:23.19 | *** join/#asterisk JonnyD_work (~Jon@cpe-071-075-036-057.carolina.res.rr.com) |
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| 01:50.05 | *** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 01:51.24 | *** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 01:53.00 | Russ | I'm having a bad problem... |
| 01:53.21 | Russ | someone broke 7D dialing to my VoIP number which is in a 7D permissive area |
| 01:53.27 | Russ | http://pastebin.com/szFiMqHZ |
| 01:53.57 | Russ | Telcordia says that 7D dialing in my area code should work |
| 01:54.10 | Russ | 7D dialing in my area code still works for other numbers |
| 01:54.14 | Russ | but not mine.... |
| 01:55.34 | Russ | at best, a really bad mistake, or at worst, totally unfair practices |
| 01:56.41 | nix8n82 | Russ, can't you dial the 10 digit number and it still be considered a local call? |
| 01:56.52 | Russ | yes |
| 01:57.30 | Russ | but that's really annoying to tell anyone and everyone I give my number to, "I know you never have to dial 480 for any other number, but my number is special, you have to dial 480 first" |
| 02:00.29 | nix8n82 | Sorry I was thinking dialing out not inbound...yeah that is super retarded..I'm sure you would have to bitch to a lot of people before you can get that changed... |
| 02:03.51 | Russ | it broke sometime in the last week |
| 02:04.34 | nix8n82 | So it's something they can fix |
| 02:04.46 | Russ | I really hope so |
| 02:05.03 | nix8n82 | but working with the telco blows... as least where I am at. it's hard to get them to do anything |
| 02:05.54 | Russ | would filing a complaint with the FCC help? |
| 02:06.32 | nix8n82 | maybe...I would threaten that first with someone as high up as you can get to |
| 02:06.45 | Russ | perhaps a number portability complain, since it is a ported number |
| 02:07.06 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 02:07.41 | nix8n82 | right I think you would have to talk to a lawyer...to know for certain, but that would cost ya |
| 02:07.48 | Russ | I need to try to figure out from my provider who exactly to complain to |
| 02:08.08 | Russ | I think Level 3 fits in somewhere |
| 02:14.04 | *** join/#asterisk moos3 (~rgenthner@cpe-72-224-105-166.maine.res.rr.com) |
| 02:14.08 | moos3 | question |
| 02:14.19 | nix8n82 | answer |
| 02:15.13 | moos3 | lol sorry i didn't finish my thought |
| 02:15.31 | moos3 | I have mutliple sip trunks, how can i load balance accross them ? |
| 02:16.50 | nix8n82 | why load balance and not fail over dialing? |
| 02:16.52 | moos3 | ie siptrunk-a siptrunk-b, two calls go out first one goes to siptrunk-a and second call goes to siptrunk-b, etc |
| 02:17.04 | moos3 | nix8n82: failover dialing ? |
| 02:17.10 | moos3 | what do you mean please share |
| 02:17.29 | moos3 | failover as in the trunk isn't available ? |
| 02:17.36 | nix8n82 | if siptrunk-a fails to make the call the call tries siptrunk-b |
| 02:18.10 | moos3 | so how would my dialplan change for that currently i have 821 exten => _+1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@${GLOBAL(SIP_PROVIDER)},70) |
| 02:18.20 | moos3 | so how would i change that |
| 02:19.23 | moos3 | oh nvm |
| 02:19.29 | moos3 | i answered my own q |
| 02:19.32 | moos3 | doh |
| 02:21.11 | *** join/#asterisk JonnyD_work (~Jon@cpe-071-075-036-057.carolina.res.rr.com) |
| 02:22.35 | nix8n82 | ok |
| 02:27.14 | moos3 | nix8n82: can't i do something like this http://pastebin.com/n56X6gDG |
| 02:27.47 | moos3 | where is roundrobins between the two trunks to spilt the total of 20 channels between the two ? |
| 02:29.06 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 02:39.56 | ChannelZ | your syntax is a little off |
| 02:46.02 | WIMPy | Where does Asterisk go when the next priority of an extension should be executed, but doesn't exist? |
| 02:46.21 | ChannelZ | Nowhere |
| 02:47.16 | WIMPy | That's what I thought, but is obviousely incorrect. |
| 02:47.33 | ChannelZ | If autofallthrough is on, the dialplan is essentially over and the channel will torn down. If autofallthrough is off, it'll sit there and wait for a new extension to be dialed |
| 02:47.47 | WIMPy | It does something for 10 seconds and it is not in my dialplan. |
| 02:48.11 | WIMPy | The setting of autofallthrough makes no difference. |
| 02:49.11 | WIMPy | just feels like a complete noob ATM. |
| 02:49.45 | ChannelZ | better than a rube |
| 02:50.19 | WIMPy | Not sure there's a difference right now. |
| 02:52.14 | WIMPy | I did a Dial that failed, no further priorities in that extension and get a message from Asterisk saying that it suffers from overload and 10s later it clears the call. |
| 02:52.44 | WIMPy | If I however ass a Hangup() next, it clears the call immediately saying the number doesn't exist. |
| 02:53.13 | WIMPy | And I have no idea what that's about. |
| 02:54.36 | *** join/#asterisk ArchGT (~archgt@unaffiliated/archgt) |
| 02:55.18 | WIMPy | No pointers to enlighten me on some dialplan basics? |
| 02:56.27 | ChannelZ | I've no guesses without being able to see things |
| 02:56.56 | WIMPy | Well, the dialplan is pretty short, let me paste that. |
| 02:57.55 | WIMPy | http://wimpy.yeti.dk/pastebin |
| 02:58.44 | WIMPy | When I remove the Hangup at _5X! it goes nuts. |
| 02:59.55 | WIMPy | If I dial a number that isn't valid externally that is. |
| 03:02.50 | ChannelZ | well if you dial an 'invalid' number (which is what exactly?) my guess is it's being picked up by your _*. pattern which answers the channel and then does nothing |
| 03:03.07 | *** part/#asterisk Hydrant (~aj@unaffiliated/hydrant) |
| 03:03.10 | WIMPy | * is not a wildcard |
| 03:03.48 | ChannelZ | I know but you've said nothing about what you're dialing, shown no console output, so I can only make a guess as to maybe why you 'get nothing' based on what I can see |
| 03:04.09 | WIMPy | If I dial 51666 it will dial out 666. Which is not a valid number. |
| 03:04.29 | WIMPy | But it only tells the phone if the Hangup is present. |
| 03:05.03 | ChannelZ | ok so this is more to do with your external provider to whom you're sending 666. Is this POTS, PRI...? |
| 03:05.44 | WIMPy | Indeed. But I get the 'invalid number' immediately, but it does not get passed on to the phone unless that Hangup() is present. |
| 03:06.36 | WIMPy | here's a trace of the call without the Hangup(): https://issues.asterisk.org/file_download.php?file_id=28012&type=bug |
| 03:07.17 | ChannelZ | TMI |
| 03:07.59 | WIMPy | I just don't understand why it goes nuts without that Hangup(). |
| 03:08.22 | WIMPy | But I've been told that's (at least kind of) expected behaviour. |
| 03:09.34 | ChannelZ | And what are you dialing from? |
| 03:09.47 | WIMPy | Another dahdi channel. |
| 03:27.11 | WIMPy | http://wimpy.yeti.dk/pastebin shows the console output with and without Hangup(), but that doesn't tell me much, either. |
| 03:27.58 | *** join/#asterisk devmod (~devmod___@c-76-100-208-204.hsd1.md.comcast.net) |
| 03:38.56 | nix8n82 | moos3, I don't think your variable tr is being use properly..I would use a short agi script and the asterisk database to do what you want, but I'm sure you could work something out in just the dialplan |
| 03:42.01 | ChannelZ | I imagine it's something particular to your channels. |
| 03:43.47 | WIMPy | Yes, it seems to be dahdi only. But it is obviousely caused by the dialplan. |
| 03:44.01 | WIMPy | And that's the part I don't understand. |
| 03:46.29 | ChannelZ | Not really. at the point you see the autofallthrough, it indicates whatever status to the calling channel, and the pause might be caused by that channel type, playing an indication (a fast-busy or something) for 10 seconds. With more to do in the dialplan, it would start executing those steps immediately, and not giving back an indication to the calling channel (because maybe in your dialplan you choose to Playback an error message of your own, etc.) |
| 03:47.01 | WIMPy | Indeed. |
| 03:47.34 | ChannelZ | I think it's a specific behavior from not implicitly hanging up the channel immediately from the dialplan, so the channel is doing some indication of its own for a time before finishing. Which makes sense |
| 03:48.14 | WIMPy | But with the Hangup, the cause 1 is retained. Without it uses channel unavailable after those 10 seconds. |
| 03:48.42 | WIMPy | I have never heard any tones/announcements. |
| 03:49.42 | WIMPy | Wich is the part that makes it rather unsuitable for users, but that's a different story. |
| 03:50.21 | ChannelZ | Because your telco just rejects and hangs up THAT leg of the call, not providing an audible fast-busy or anything. * then indicates back to the calling channel 'CHANUNAVAIL' whose apparently deciding what to do next, but just not doing what you want. |
| 03:50.44 | WIMPy | They do, but Asterisk doesn't care. |
| 03:51.08 | ChannelZ | For instance if I make a similar test, calling from an analog phone through my ATA (so it appears as SIP to Asterisk), I get a busy tone.. but that's being generated by my ATA after getting the channel progress back from *. |
| 03:51.22 | ChannelZ | If I do the same thing from a softphone, the channel terminates immediately and the softphone tells me what happened. |
| 03:52.28 | ChannelZ | In your case DAHDI should be doing something but isn't. Why I don't know, that's a different story.. is the channel you're calling from an analog phone hooked up to a TDM card? |
| 03:53.15 | WIMPy | The fact that there is no concept of having audio after a call ends is another story alltogether. I'm only after the weird signalling for the moment. |
| 03:53.35 | WIMPy | No, both ISDN. |
| 03:53.59 | WIMPy | Can dahdi do such things at all? |
| 03:54.18 | ChannelZ | Put a Congestion(10) in after your dial and see what happens |
| 03:54.28 | ChannelZ | Do what such things? |
| 03:54.45 | WIMPy | *anything* after a call terminates. |
| 03:57.43 | ChannelZ | It shouldn't. You should be able to do anything after the dial. (in fact you are, Hangup, in your one version.) |
| 03:58.31 | WIMPy | Ok, placing a Congestion() does give an audible fast busy. But always gives the wrong indication, no matter if followed by Hangup or not. |
| 03:59.21 | ChannelZ | well Congestion is Congestion. |
| 03:59.26 | WIMPy | Actually the Hangup isn't executed. |
| 03:59.44 | ChannelZ | So what's this then: "-- Executing [52666@dahdi:2] Hangup("DAHDI/i7/23239-7", "") in new stack" |
| 03:59.51 | WIMPy | Yes, abd somehow wrong here. But at least it is something that produces audio. |
| 03:59.59 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 04:00.35 | WIMPy | That's the explicit Hangup() at priority 2 that makes it work. |
| 04:00.36 | ChannelZ | What I'm saying is that Congestion forefully indicates congestion, it has nothing to do with the result of the Dial command prior.. it has no idea |
| 04:01.05 | WIMPy | And it terminates the dialplan. |
| 04:01.05 | ChannelZ | I know. That's what I said. |
| 04:02.33 | WIMPy | Idealy I'd get the audio from the other channel, but I guess we're quite far away from such a thing. |
| 04:03.34 | ChannelZ | That's up to your telco - it's the one rejecting the call and terminating your connection. |
| 04:03.49 | WIMPy | That's the way the PSTN works. |
| 04:03.57 | ChannelZ | That's different. |
| 04:04.00 | WIMPy | But they still play an announcement. |
| 04:04.10 | ChannelZ | Have you tried dialing a legitimately-formatted but incorrect number? |
| 04:04.19 | WIMPy | Just that Asterisk won't take it. |
| 04:04.41 | WIMPy | We don't have a closed number plan. |
| 04:05.23 | WIMPy | Anything from 3 digits upwards can be valid. |
| 04:06.23 | ChannelZ | There are isms specific to ISDN I can't help with as I've no experience with it, perhaps there is some channel config parameters you can switch so that DAHDI doesn't take the rejection from the channel and keeps on it or something if indeed they do play an indication. |
| 04:07.17 | WIMPy | I'm not sure what it should do. As far as I know there is simply no concept of having audio after a call terminates. |
| 04:07.40 | WIMPy | In Asterisk that is. |
| 04:08.32 | ChannelZ | If you hook an ISDN phone up directly to the line and call that number, 666, you get audio from the telco saying 'the number you have dial sucks ass' or whatever? |
| 04:08.42 | WIMPy | yes |
| 04:08.48 | *** join/#asterisk dogonovmax (~Adium@78-106-142-109.broadband.corbina.ru) |
| 04:10.52 | ChannelZ | So it's the nature by which DAHDI is talking to your telco over the ISDN.. some other level of protocol (? dunno how ISDN works) |
| 04:11.19 | WIMPy | err, pardon? |
| 04:12.46 | ChannelZ | Again I don't have ISDN or know its innards so I can only speculate.. either there is some means by which DAHDI talks to the remote end and requests channel progress indications as opposed to letting the telco accept the call and play its own indications, or something similar |
| 04:13.28 | ChannelZ | (or I should say DAHDI talks to your card and asks of it to set the ISDN channel to behave a certain way) |
| 04:14.04 | WIMPy | You don't ask anything, you just get it. |
| 04:14.43 | WIMPy | Well, you ask for a connection to 666, and as that's not a valid number they will terminate the call, but indicate that an announcement will be played. |
| 04:15.10 | WIMPy | So that's kind of the reverse of early media. |
| 04:15.22 | ChannelZ | What I'm saying is that the remote end is requesting the hangup. Whether it then goes on to provide audio to the client and your ISDN phone ignores the hangup request and allows you to hear the message being played from the telco, I dunno |
| 04:15.47 | ChannelZ | There may or may not be a way to config DAHDI to behave the same way |
| 04:16.01 | WIMPy | It doesn't ignore it. It will indicate the reason on its display. |
| 04:16.56 | ChannelZ | ok so that answers my question |
| 04:17.04 | ChannelZ | or confirms my complete guess, rather |
| 04:17.11 | WIMPy | And if you're on speakerphone it might switch off either immediately or after some time, depending on the reason for call termination. |
| 04:17.32 | ChannelZ | How to fix is another matter, but I don't think it has anything to do with your dialplan. |
| 04:17.33 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 04:17.51 | WIMPy | Not the adio thing, no. |
| 04:19.05 | WIMPy | But the fact that I get a 10 second delay with the false indication that some network layer is busy has to do with the dialplan. |
| 04:19.31 | WIMPy | And that doesn't make sense to me. |
| 04:19.57 | ChannelZ | I don't know if 'priindication' and/or 'inbanddisconnect' in chan_dahdi has any relevance here |
| 04:20.39 | WIMPy | The later might be interesting for audio, yes. |
| 04:21.09 | WIMPy | priindication sounds lik it would be upstream only. |
| 04:22.03 | WIMPy | But that's the confusing part about dahdi. One configuration file for completely different technologies. A list of what option can be used with what might be a good idea. |
| 04:23.27 | ChannelZ | "Allows a caller to hear inband disconnect data that the remote |
| 04:23.27 | ChannelZ | end may play at disconnect, instead of releasing the B channel |
| 04:23.27 | ChannelZ | immediately" |
| 04:23.38 | ChannelZ | grrph damn embedded LFs |
| 04:23.50 | *** join/#asterisk r0d3nt (~astrutt@cheshire.telephreak.org) |
| 04:23.57 | ChannelZ | sounds like what you need |
| 04:24.13 | ChannelZ | or want anyway |
| 04:25.11 | *** join/#asterisk pinoyskull- (~pinoyskul@124.6.182.55) |
| 04:25.50 | WIMPy | Just tried. |
| 04:26.14 | WIMPy | It does keep the cahnnel up, but it does not forward audio. I hear a busy now. |
| 04:27.00 | ChannelZ | you restarted dahdi and everything |
| 04:27.15 | WIMPy | Seems to be a little random. Now I get a dialtone. |
| 04:27.44 | WIMPy | I module unload chan_dahdi and thn load. |
| 04:28.03 | WIMPy | kills it. |
| 04:28.20 | ChannelZ | hmm. well hopefully someone who actually knows ISDN will have more to say |
| 04:29.33 | WIMPy | Well it does not end the call now, but no (or random) audio. |
| 04:29.56 | WIMPy | Let's see what happens after the timeout. |
| 04:31.16 | ChannelZ | Do you still see "-- Channel 0/1, span 6 got hangup request, cause 1" immediately and while the random audio plays? |
| 04:32.05 | WIMPy | No. I do get it, off course, but it's not displayed on the console until the channel is released. |
| 04:43.02 | *** join/#asterisk amr922 (~chatzilla@41.235.138.184) |
| 04:43.25 | amr922 | Hello ! |
| 04:43.30 | amr922 | i need some help please |
| 04:43.56 | amr922 | [Dec 22 07:43:45] WARNING[28227] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 04:45.28 | WIMPy | Is chan_dahdi loaded? |
| 04:46.20 | amr922 | yes |
| 04:46.34 | amr922 | Elastix*CLI> dahdi show status |
| 04:46.35 | amr922 | Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO |
| 04:46.37 | amr922 | Wildcard AEX410 Board 1 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) |
| 04:51.08 | *** part/#asterisk tash (~Tommy@ks-76-7-1-196.sta.embarqhsd.net) |
| 04:52.44 | amr922 | hello :( |
| 04:53.30 | ChannelZ | then what are you Dial()ing ? |
| 04:54.51 | amr922 | exten => 5555,1,Dial(DAHDI/1/026279666) |
| 04:54.58 | ChannelZ | And is channel 1 configured? |
| 04:55.13 | ChannelZ | dahdi show channels |
| 04:55.42 | amr922 | Elastix*CLI> dahdi show channels |
| 04:55.44 | amr922 | <PROTECTED> |
| 04:55.45 | amr922 | <PROTECTED> |
| 04:55.47 | amr922 | <PROTECTED> |
| 04:55.48 | amr922 | <PROTECTED> |
| 04:55.50 | amr922 | <PROTECTED> |
| 04:55.51 | amr922 | <PROTECTED> |
| 04:55.53 | amr922 | yes i have 4 configured channels |
| 04:57.07 | ChannelZ | Then it's not able to open that channel for some other reason... |
| 04:57.40 | amr922 | i checked it and installed the driver again but it not work :( |
| 04:58.33 | ChannelZ | it might be misconfiguration of the channel, or unplugged, or something like that. Do you have verbosity turned up a little, is it saying anything else? |
| 04:59.04 | ChannelZ | AEX410, is that an analog card? |
| 04:59.49 | amr922 | yes |
| 05:00.04 | amr922 | verbos just says == Everyone is busy/congested at this time (1:0/0/1) |
| 05:00.40 | amr922 | http://www.digium.com/en/products/analog/aex410.php |
| 05:02.53 | ChannelZ | Line plugged into the right port? :) |
| 05:03.05 | pabelanger | amr922: pb your chan_dahdi.conf |
| 05:03.36 | ChannelZ | yeah, make sure it's configged with the correct signalling |
| 05:03.53 | amr922 | [channels] |
| 05:03.54 | amr922 | context=from-pstn |
| 05:03.56 | amr922 | signalling=fxs_ks |
| 05:03.57 | amr922 | rxwink=300 ; Atlas seems to use long (250ms) winks |
| 05:03.59 | amr922 | usecallerid=yes |
| 05:04.00 | amr922 | hidecallerid=no |
| 05:04.00 | *** mode/#asterisk [-v amr922] by pabelanger |
| 05:04.02 | amr922 | callwaiting=yes |
| 05:04.03 | amr922 | usecallingpres=yes |
| 05:04.05 | amr922 | callwaitingcallerid=yes |
| 05:04.06 | amr922 | threewaycalling=yes |
| 05:04.08 | amr922 | transfer=yes |
| 05:04.09 | amr922 | canpark=yes |
| 05:04.11 | amr922 | cancallforward=yes |
| 05:04.12 | amr922 | callreturn=yes |
| 05:04.14 | amr922 | echocancel=yes |
| 05:04.15 | amr922 | echocancelwhenbridged=no |
| 05:04.17 | amr922 | faxdetect=incoming |
| 05:04.18 | amr922 | echotraining=800 |
| 05:04.20 | amr922 | rxgain=0.0 |
| 05:04.22 | amr922 | txgain=0.0 |
| 05:04.23 | amr922 | callgroup=1 |
| 05:04.24 | *** kick/#asterisk [amr922!~pabelange@50.22.5.41-static.reverse.softlayer.com] by pabelanger (amr922) |
| 05:04.38 | WIMPy | pabelanger: +b |
| 05:04.39 | *** join/#asterisk amr922 (~chatzilla@41.235.138.184) |
| 05:04.47 | pabelanger | amr922: use pastebin |
| 05:05.32 | amr922 | sorry |
| 05:05.34 | amr922 | http://pastebin.com/eZMTeGJ0 |
| 05:07.13 | ChannelZ | guess we really need to see your dahdi-channels.conf |
| 05:07.22 | ChannelZ | and/or chan_dahdi_additional.conf |
| 05:08.45 | amr922 | http://pastebin.com/AGAFK6eM |
| 05:08.52 | amr922 | dahdi-channels.conf |
| 05:08.58 | amr922 | chan_dahdi_additional.conf is empty |
| 05:13.46 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 05:16.41 | ChannelZ | hmm ok well that seems alright assuming the dahdi config is right which at least part of it must be since they're showing up in * |
| 05:17.04 | ChannelZ | So again I'd first check you've got your line plugged into the right port (unless you're using all 4) and that it/they are active |
| 05:17.32 | amr922 | i use the 4 lines |
| 05:17.54 | amr922 | i tried each one of them but same error |
| 05:21.44 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 05:27.34 | amr922 | hello |
| 05:27.39 | WIMPy | Uh-oh. It's getting interesting. I actually get announcements on another line. |
| 05:27.57 | *** join/#asterisk shapr (~shapr@nat/digium/x-gepqcwmtiojuxwkh) |
| 05:29.12 | amr922 | could you help wimpy please :( |
| 05:29.39 | ChannelZ | He's beyond help |
| 05:29.43 | ChannelZ | :P |
| 05:30.01 | WIMPy | Ah, forget it, they do it wrong. But that wrong version is Asterisk compatible. |
| 05:30.15 | ChannelZ | who |
| 05:30.56 | WIMPy | My Telco. The don't disconnect the call, but tell that it's ringing while they play the anouncement. |
| 05:31.06 | ChannelZ | amr922: what country are you in? Are you sure you use kewlstart for instance? |
| 05:31.17 | WIMPy | But they use extremely shitty stuff that does many weird things. |
| 05:31.46 | WIMPy | It's one of those NGN lines. |
| 05:31.56 | amr922 | Saudia arabia |
| 05:32.22 | amr922 | i tried fxsls too now but still |
| 05:32.25 | amr922 | :( |
| 05:33.56 | ChannelZ | amr922: what does your /etc/dahdi/system.conf look like |
| 05:34.22 | amr922 | i edited it now to |
| 05:34.24 | amr922 | fxsls=1 |
| 05:34.26 | amr922 | fxsls=2 |
| 05:34.29 | amr922 | fxsls=3 |
| 05:34.31 | amr922 | fxsls=4 |
| 05:36.26 | *** join/#asterisk jblack (~jblack@pool-71-181-222-126.sctnpa.east.verizon.net) |
| 05:36.28 | ChannelZ | I'm not sure what's really proper for Saudi Arabia, but fxs is at least right |
| 05:36.50 | ChannelZ | And if you hook an analog phone up to one of those lines and dial that exact number, it works? |
| 05:37.18 | amr922 | yes |
| 05:43.37 | ChannelZ | wonders if he has the right modules |
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| 06:03.59 | *** join/#asterisk doolittlework (~d@41-134-22-10.dsl.mweb.co.za) |
| 06:05.12 | doolittlework | morning, is there a way to transfer a call and if the extension is busy, ring back to the extension that transfered the call? |
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| 06:31.03 | ChannelZ | doolittlework: well, I think that depends on a few things. |
| 06:33.15 | ChannelZ | Doesn't it do it already, assuming the transfer extension doesn't do call waiting and actually returns busy? I'm trying to remember.. |
| 06:33.23 | ChannelZ | I know something rings you back. |
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| 07:18.47 | doolittlework | nope mine just dies if user is busy |
| 07:26.42 | *** join/#asterisk admin0 (~admin0@117.20.175.236) |
| 07:27.25 | admin0 | hi .. if my sip provider is named as sipOUT, how do I forward all calls starting with 00 to this sip provider, but remove the 00 when sending to it ? exten => 00_X,1,Dial(SIP/sipOUT/1${EXTEN}) ?? |
| 07:27.53 | WIMPy | EXTEN:2 |
| 07:28.06 | kaldemar | exten => _00X.,1,Dial(SIP/sipOUT/${EXTEN:2}) |
| 07:28.17 | WIMPy | jo |
| 07:28.30 | admin0 | kaldemar, the EXTEN:2 in the end means remove the 00 prefix ? |
| 07:28.42 | kaldemar | in case of a pattern, _ needs to be first. "." matches to one or more characters. |
| 07:29.01 | *** join/#asterisk DJClean (~djclean@unaffiliated/djclean) |
| 07:29.02 | kaldemar | admin0: it means remove two first characters of the variable contents. |
| 07:29.09 | admin0 | ok |
| 07:29.54 | admin0 | if i want to send asis it will be exten => _X.,1,Dial(SIP/sipOUT/${EXTEN}) ? |
| 07:30.06 | kaldemar | yes. |
| 07:30.30 | kaldemar | but that will match all numbers, not just ones that start with 00. |
| 07:30.43 | admin0 | ok |
| 07:32.06 | admin0 | in place of EXTEN, how do I define number ? |
| 07:32.13 | admin0 | its a different peer making the call |
| 07:32.21 | admin0 | trying to use asterisk for transcoding |
| 07:34.00 | kaldemar | SIP/sipOUT/12398791279871923 for example. variables, digits, characters... they're all allowed. also concatenated, for example SIP/sipOUT/00${EXTEN}99${YOURVAR} |
| 07:34.49 | admin0 | sipIN is my peer1 calling using 011<number> .. comes to asterisk and tries to go to another peer sipOUT ... but when it sends to sipOUT, none of the number go |
| 07:35.20 | admin0 | my rule is: exten => _011X.,1,Dial(SIP/sipOUT/${$EXTEN:3}) |
| 07:37.09 | kaldemar | attach to asterisk CLI and see what happens when you make a call. |
| 07:42.39 | admin0 | <PROTECTED> |
| 07:43.17 | WIMPy | One $ too much. |
| 07:43.34 | admin0 | i see it |
| 07:45.03 | admin0 | thanks WIMPy and kaldemar ,, now worked :) |
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| 08:30.11 | *** join/#asterisk dongs (1000@l212168.ppp.asahi-net.or.jp) |
| 08:33.25 | dongs | anyone have any idea how firmware upgrade on grandstream gxw4008 supposed to work?.... the .zip has a bunch oif files in it and i cant find any info on how to actually make them upgrade. |
| 08:33.37 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 08:36.34 | kaldemar | dongs: http://www.grandstream.com/faqsfirmware.html#4 |
| 08:36.44 | dongs | clicking |
| 08:36.59 | dongs | :| |
| 08:37.38 | dongs | this is hardly easy |
| 08:38.47 | kaldemar | set up a tftp server, put the files in the root directory of the server and configure the server address in the phone and power cycle the phone. |
| 08:57.11 | *** join/#asterisk joobie (~joobie@CPE-124-176-177-112.vic.bigpond.net.au) |
| 09:09.49 | ChannelZ | life is hard |
| 09:11.58 | dongs | worked. |
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| 09:26.30 | dongs | what do people use for ipsec on lunix these days |
| 09:26.33 | dongs | is it still racoon |
| 09:34.23 | *** join/#asterisk schmidts (~schmidts@lmlo.sil.at) |
| 09:34.25 | schmidts | good morning |
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| 09:38.41 | kaldemar | dongs: strongswan, openswan... |
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| 13:12.29 | *** join/#asterisk WindBack (~quassel@kirk.capitalinasdc.com) |
| 13:16.49 | WindBack | My SIP provider sends me OPTIONS message periodically to qualify the connection. My Asterisk server is answering them with 404 since it is looking for an empty extension in default context. How I can change this behaviour to make asterisk answer with 200 OK? This is the sip message under this situation: http://pastebin.ca/2026215 |
| 13:19.12 | schmidts | windback there is allready an open issue for this regression: 18348 |
| 13:19.35 | schmidts | windback you can set pedantic=no in your sip.conf to make this work AFAIK |
| 13:20.13 | *** join/#asterisk nzw (~rychu@85.11.67.125) |
| 13:20.30 | nzw | hi |
| 13:20.31 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 13:21.04 | nzw | do anyone have working MGCP E-MTA on asterisk? |
| 13:21.19 | nzw | i'm trying to configure arris with no luck |
| 13:21.46 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:21.55 | *** join/#asterisk zplinux (~zplinux@213.8.57.217) |
| 13:21.57 | zplinux | hi all |
| 13:22.10 | zplinux | can I send a sip messege from the command line? |
| 13:22.51 | *** part/#asterisk moos3 (~rgenthner@cpe-72-224-105-166.maine.res.rr.com) |
| 13:26.21 | schmidts | zplinux not really, only some special notify messages to some type of phones (cisco, linksys....) to let them reboot |
| 13:26.48 | zplinux | I am seeking to get notified when a script finished |
| 13:26.58 | zplinux | I can't use email, but have a pbx |
| 13:27.30 | schmidts | zplinux then you should better use a callfile to start a call to your phone |
| 13:28.01 | zplinux | please explain |
| 13:28.32 | schmidts | http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
| 13:29.19 | schmidts | you can create a file in /var/spool/asterisk/outgoing and asterisk will start a call to the extension which is defined in this file. so you can start a call to your phone and playback a soundfile |
| 13:30.45 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 13:31.23 | zplinux | but I must run this from the pc running asterisk |
| 13:31.32 | zplinux | I am reffering to another pc |
| 13:32.36 | Phlogistique | zplinux: you mean you need a SIP client that runs from command line? |
| 13:32.50 | zplinux | <PROTECTED> |
| 13:32.52 | schmidts | zplinux ok then you could use sipp to start a call or sipsak |
| 13:32.54 | zplinux | but yes |
| 13:33.11 | schmidts | zplinux if your phone has a static ip you can use sipsak |
| 13:33.55 | Phlogistique | or you could allow the use of AMI from your PC's IP and use it to originate a call |
| 13:33.58 | zplinux | ok thanks |
| 13:35.18 | zplinux | GOOD IDEA |
| 13:41.09 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 13:47.27 | WindBack | schmidts: thanks, however it is not solved using pendantic=no |
| 13:50.38 | schmidts | windback could you please post this also to the issue |
| 13:51.29 | WindBack | schmidts: ok, I will do it |
| 13:53.04 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:55.52 | *** join/#asterisk i_heart_asterisk (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 13:56.02 | i_heart_asterisk | hi |
| 13:58.44 | i_heart_asterisk | hello |
| 13:58.47 | i_heart_asterisk | echo |
| 13:58.53 | i_heart_asterisk | talk to me goose |
| 14:08.37 | Katty | gooooooooood morning! |
| 14:08.47 | Katty | 2 days till christmas (= |
| 14:08.56 | WIMPy | Katty: Long time no see. |
| 14:09.11 | WIMPy | Have you been too busy with all those X-mas cards? |
| 14:10.12 | Katty | what do you mean? i'm in here every day |
| 14:10.49 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 14:10.53 | WIMPy | I didn't see you write for soe days. But maybe I've lust been here the wrong times? |
| 14:12.04 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:13.08 | Katty | possibly (= |
| 14:19.05 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 14:20.40 | i_heart_asterisk | anybody ever experience choppy sound when asterisk plays vm-options ? |
| 14:23.54 | *** join/#asterisk As001 (~IPN05@194.106.175.113) |
| 14:24.39 | *** join/#asterisk patrick^ (~patrick_@2001:470:b0ea:1:219:21ff:fe4e:f5de) |
| 14:24.57 | As001 | Hello. Can you point me to web page where I can find how to make script or dialplan to let agent login automaticly with AgentLogin (without pressing agent number and password). Is it possible at all in Asterisk 1.6.11 ? |
| 14:30.07 | Katty | As001: good morning. |
| 14:30.48 | As001 | Good afternoon here :) |
| 14:36.14 | i_heart_asterisk | good morning |
| 14:36.37 | *** join/#asterisk evangelion (~manzy_zet@212.183.172.126) |
| 14:36.41 | evangelion | hello |
| 14:36.48 | Katty | evangelion: good morning. |
| 14:36.58 | i_heart_asterisk | razzledazzle |
| 14:37.55 | evangelion | what happen to dialplan after a dialed call is answered and terminated on 1.6.2? does it jump to n+101? |
| 14:38.37 | kaldemar | evangelion: depends. generally no. |
| 14:39.02 | kaldemar | if the callee hangs up, it moves to next priority. |
| 14:39.43 | kaldemar | erm, with parameter g that is. |
| 14:40.12 | evangelion | yeah! i forgot "g" =) |
| 14:40.14 | evangelion | thank you |
| 14:53.40 | *** join/#asterisk nighty-- (~nighty@x122091.ppp.asahi-net.or.jp) |
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| 14:55.39 | X-Raimo | when both peers using TLS to connect to asterisk (SIP). Is using of SRTP is needed? |
| 14:56.20 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-lpkvkbdvtpajlhuj) |
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| 14:57.05 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:06.08 | leifmadsen | X-Raimo: yes, the TLS is just the encryption of the signalling layer |
| 15:06.16 | leifmadsen | the media layer is separate (SRTP) |
| 15:11.16 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
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| 15:12.03 | Polysics | hello |
| 15:12.11 | Polysics | still at work on reverse engineering that applet |
| 15:12.32 | Polysics | there is a closed-source java applet that is able to do a call over an HTTP proxy |
| 15:12.49 | Polysics | it apparently uses IAX over HTTP |
| 15:13.07 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 15:16.58 | Polysics | is it plausible, or even possible? or should i look for something else? |
| 15:17.14 | Polysics | next week i should get access to a firewalled network so i can wireshark the thing |
| 15:19.38 | i_heart_asterisk | Alisons recorded menu options have poor sound quality |
| 15:19.43 | i_heart_asterisk | anybody experience that ? |
| 15:28.12 | Polysics | does IAX2 work overa a regular SOCKS5 proxy? can't find a reference for that |
| 15:28.17 | Polysics | it should, i suppose |
| 15:30.43 | *** join/#asterisk Tili (~Tili@cm161.eta193.maxonline.com.sg) |
| 15:30.53 | Tili | is it possible to get REDIRECTING REASON on chan_ss7 |
| 15:34.35 | raden | Naikrovek, YO BRO |
| 15:35.45 | *** part/#asterisk ArchGT (~archgt@unaffiliated/archgt) |
| 15:41.12 | Polysics | no ideas on the IAX2 proxy thing please? |
| 15:42.38 | *** join/#asterisk pruonckk (~pruonckk@187.109.32.11) |
| 15:43.37 | pruonckk | Hello guys |
| 15:44.32 | pruonckk | im trying to make a test with a new E1 card, i want to make a master R2 node, but i do the configuration and when i make a call im getting the message below |
| 15:44.34 | pruonckk | chan_dahdi.c:1713 dahdi_r2_on_protocol_error: MFC/R2 protocol error on chan 1: Invalid R2 state |
| 15:45.08 | pruonckk | somebody know why im getting this message and how can i solve ? |
| 15:45.38 | pruonckk | when i do a dahdi show channel 1 i receive a MFC/R2 MF State: MF Engine Off |
| 15:45.46 | pruonckk | i think that can be my problem |
| 15:48.13 | Polysics | is it THAT stupid of a question? |
| 15:52.36 | pruonckk | Polysics, sorry, i dont understand the correct meaning of you answer |
| 15:52.54 | Polysics | i was referring to my own question :-) |
| 15:53.17 | Polysics | i am trying to find how an applet i know works does a IAX2 call over an HTTP or SOCKS proxy |
| 15:55.31 | _Corey_ | Polysics: It's a pretty unusual request... |
| 15:56.20 | Polysics | _Corey_: yes, but it would also be very useful in some environments |
| 15:56.41 | _Corey_ | ? |
| 15:56.53 | WIMPy | It's also useful to have internet access. |
| 15:57.21 | Polysics | please don't start the usual "people should be on open networks" thig, i know :-) |
| 15:57.24 | moy | pruonckk: what do you mean by master R2 node? |
| 15:57.36 | *** join/#asterisk giesen (giesen@204.14.17.135) |
| 15:57.53 | pruonckk | moy, hello moy, on isdn i set the pri_net on switchtype to use as provider not client |
| 15:58.12 | giesen | Is there any way to create a time condition that's valid from 3pm Friday until 5pm Tuesday |
| 15:58.31 | pruonckk | this is what im trying to do, the link sync without problem, but when i do a call, i recive the message that i say before |
| 15:58.34 | *** join/#asterisk espiceland (~erin@nat/digium/x-ytsjntvyfeqyokbt) |
| 15:58.38 | giesen | Right now, I believe the behaviour is 3pm-5pm on Friday to Tuesday |
| 15:59.05 | moy | pruonckk: that does not make sense ... MFC-R2 does not have switchtype, that is ignored for MFC-R2 .... you need to enable all debugging in asterisk and set mfcr2_logging=all ... then pastebin the debug log for a call attempt along with your chan_dahdi.conf configuration |
| 15:59.20 | WIMPy | pruonckk: I thought R2 referrs to C5/R2? What's that got to do with ISDN? |
| 15:59.21 | Polysics | i find the issue interesting because the applet DOES work, and as such, there has to be a way to get an IAX call over a proxy, that's all |
| 15:59.51 | moy | WIMPy: R2 in this context is a telephony signaling |
| 16:00.09 | pruonckk | WIMPy, what i say was that on isdn i use the switchtype to do this, i need to know if on a r2 model i need to something like this too |
| 16:00.22 | *** join/#asterisk jblack (~jblack@pool-71-181-222-126.sctnpa.east.verizon.net) |
| 16:00.35 | WIMPy | Ok, that way it makes sense :-) |
| 16:00.44 | pruonckk | :) |
| 16:00.54 | pruonckk | sorry my english |
| 16:01.06 | pruonckk | i need more training :) |
| 16:01.08 | pruonckk | ehhe |
| 16:01.17 | WIMPy | Stay online :-) |
| 16:05.09 | pruonckk | moy, http://pastebin.com/8PaxLimm |
| 16:05.43 | *** join/#asterisk m_tadeu (~quassel@89-180-98-120.net.novis.pt) |
| 16:06.54 | pruonckk | moy, , this test is call from asterisk B -> to asterisk A |
| 16:07.21 | moy | pruonckk: disable mfcr2_double_answer |
| 16:07.28 | moy | mfcr2_double_answer=no |
| 16:07.32 | pruonckk | ok |
| 16:08.11 | pruonckk | \o/ |
| 16:08.16 | pruonckk | moy, you is my master |
| 16:08.17 | pruonckk | hehe |
| 16:12.01 | pruonckk | moy, thanks for you help |
| 16:12.15 | moy | pruonckk: works now? |
| 16:13.24 | pruonckk | yes, perfect |
| 16:13.32 | *** join/#asterisk GhOnDiE (~GhOnDiE@92.29.182.59) |
| 16:15.50 | WIMPy | moy: BTW: The aswer was: yes. |
| 16:17.19 | moy | WIMPy: answer to? |
| 16:17.35 | WIMPy | ... I thought R2 referrs to C5/R2? ... |
| 16:18.24 | moy | WIMPy: what is C5/R2? |
| 16:18.48 | WIMPy | The CCITT Signalling System No. 5 Revision 2 |
| 16:19.01 | hrhrhr | skype just crashed on a few computers here |
| 16:19.05 | hrhrhr | anyone still logged in? |
| 16:20.12 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 16:24.24 | ChannelZ | my * is |
| 16:24.44 | *** join/#asterisk zplinux (~zplinux@213.8.57.217) |
| 16:24.47 | zplinux | hi all |
| 16:24.49 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 16:25.02 | zplinux | I am tring to run a simple auto-dial from shel |
| 16:25.05 | zplinux | shell |
| 16:25.18 | zplinux | what is the Action: I need to put? |
| 16:25.57 | ChannelZ | look up Originate |
| 16:26.38 | ChannelZ | depending on what version you're running |
| 16:26.47 | zplinux | Cool! |
| 16:28.36 | zplinux | I am running this |
| 16:28.38 | zplinux | http://dpaste.com/289496/ |
| 16:28.49 | zplinux | I get the call but done here anything |
| 16:29.10 | *** join/#asterisk JonnyD_work (~Jon@173.226.80.154) |
| 16:29.16 | zplinux | what I would like to hear is the IP address of the machine |
| 16:29.25 | zplinux | can asterisk do that? |
| 16:29.49 | zplinux | I run the script with |
| 16:29.52 | zplinux | mkfifo to_script && ./notify.sh < to_script |telnet |tee > to_script && /bin/rm to_script |
| 16:29.52 | Nugget | telnet is eeeeeeevil! |
| 16:29.58 | zplinux | but easy |
| 16:30.29 | zplinux | what is my next step to get asterisk to convert the IP address to a voice msg? |
| 16:30.36 | zplinux | and then read it to me? |
| 16:30.45 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 16:31.41 | *** join/#asterisk Hanumaan (~Hanumaan@132.230.17.77) |
| 16:32.10 | *** join/#asterisk sivang (~sivang@unaffiliated/sivang) |
| 16:32.14 | sivang | hi all |
| 16:32.32 | sivang | what would be a good guide to start with asterisk 1.4? |
| 16:32.40 | sivang | using it as a pbx and as a softswitch? |
| 16:35.40 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 16:35.42 | kaldemar | zplinux: SayDigit(${SHELL(ip addr ...)}) |
| 16:35.49 | zplinux | thnaks mate |
| 16:36.30 | leifmadsen | sivang: guide, as in documentation? |
| 16:39.00 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 16:39.30 | zplinux | kaldemar: do I need to write a context then send the IP address as an argument? |
| 16:39.45 | zplinux | in case I want to here the machines IP |
| 16:40.05 | sivang | leifmadsen: yeah |
| 16:42.54 | zplinux | can I run saydigits from ami? |
| 16:43.23 | kaldemar | zplinux: call the app in the originate or call an extension that executes a say app with the ip address as an argument. i already gave you a tool to run a shell command and get stdout of the command to dialplan. |
| 16:44.05 | kaldemar | your pastebin origination calls an application already. |
| 16:44.10 | zplinux | thanks, I am trying to understand I am new to ami |
| 16:44.13 | *** join/#asterisk dajhorn (~dajhorn@71.158.166.44) |
| 16:44.24 | kaldemar | so, yes you can. |
| 16:44.54 | zplinux | hi, does this mean I only need to add a line in action 2? |
| 16:45.02 | *** join/#asterisk dimm (~guest@unaffiliated/dimm) |
| 16:45.02 | zplinux | no, it can;t be that easy |
| 16:45.26 | kaldemar | it can |
| 16:48.18 | *** join/#asterisk farkus (chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 16:48.44 | sivang | what's the AMI? |
| 16:48.56 | kaldemar | set application to SayDigit and data as the address. i don't remember how SayDigit behaves with dots, but they're easy to filter out. |
| 16:49.09 | kaldemar | ~book |
| 16:49.09 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 16:49.27 | kaldemar | sivang: take a look at that book |
| 16:51.47 | zplinux | ok, I do get the call , but I can;t here 123 |
| 16:54.28 | zplinux | kaldemar: http://dpaste.com/289508/ |
| 16:54.52 | zplinux | in fact I understood I need to put it in application as you answered me |
| 16:55.08 | zplinux | I get a call but cant here 123 |
| 16:56.08 | sivang | kaldemar: yes, I knew there was somethign like that, already reading |
| 16:56.20 | zplinux | even if I add a line Data: 123 and remove the () in application |
| 16:56.22 | sivang | kaldemar: thanks! |
| 16:56.47 | sivang | kaldemar: is there somewhere to read about Rich Adamson? |
| 17:01.21 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 17:01.29 | leifmadsen | sivang: http://suzanne.supertec.com/2007/12/remember-rich-adamson-pstn-guru.html |
| 17:04.39 | *** join/#asterisk tzanger (~tzanger@gromit.mixdown.ca) |
| 17:05.00 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 17:05.11 | *** join/#asterisk wizard171 (~wizard@h47.50.20.98.dynamic.ip.windstream.net) |
| 17:09.30 | zplinux | kaldemar: awsome! |
| 17:14.13 | *** join/#asterisk Tim_Toady (~moi@77.49.3.151.dsl.dyn.forthnet.gr) |
| 17:14.30 | sivang | so I've got lots of reading to do. thanks leifmadsen again |
| 17:14.49 | leifmadsen | oh ya, asterisk is a big project :) Reading is your best friend |
| 17:15.25 | sivang | leifmadsen: I concieve it is the linux kernel for telephony actually |
| 17:15.41 | sivang | I fully appreciate it, and knew about i for years bti never had to imeplemnt one |
| 17:15.59 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 17:22.16 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:22.30 | *** join/#asterisk Scorpio2007 (~Scorpio20@jose-tc.ctc.biz) |
| 17:22.45 | Scorpio2007 | BLF and valetparking .. any help? |
| 17:23.41 | thehar | Vague party of 1 |
| 17:24.36 | citywok | roflmao |
| 17:24.45 | citywok | watch out for russellb and his ban hammer |
| 17:25.13 | thehar | I was not directing it at anyone! |
| 17:25.28 | thehar | Scorpio2007: Could you provide some more detail? We'd be happy to help :] |
| 17:25.32 | *** join/#asterisk luckman212 (~quassel@pool-96-246-172-198.nwrknj.fios.verizon.net) |
| 17:25.40 | citywok | of COURSE not. it just happened to be the only message on here in 10 minutes, and you replied 90 seconds later ;) |
| 17:26.22 | luckman212 | how can I force a SIP channel to die without killing * (1.8.1.1) is there a way? |
| 17:26.47 | thehar | luckman212: you could use soft hangup |
| 17:26.55 | citywok | channel request hangup |
| 17:26.59 | citywok | thehar: it changed! :P |
| 17:27.04 | thehar | oh bawk |
| 17:27.05 | thehar | upgrades |
| 17:27.09 | citywok | pwnt |
| 17:27.10 | thehar | 1.4 for life! |
| 17:27.13 | luckman212 | soft hangup = channel request hangup in 1.8 |
| 17:27.21 | luckman212 | yeah it doesnt kill it though :( |
| 17:27.28 | luckman212 | any other way? |
| 17:28.24 | thehar | Are you killing the right SIP channel? |
| 17:28.43 | thehar | Is it mapped to a DAHDI channel? |
| 17:28.46 | thehar | s/mapped/bridged/ |
| 17:28.47 | citywok | i've never had it fail to kill a channel |
| 17:28.49 | luckman212 | sip show channels... yep its the right one |
| 17:28.59 | luckman212 | no its not a DAHDI channel... this is a SIP channel |
| 17:29.14 | luckman212 | no DAHDI hw in this box at all |
| 17:29.24 | thehar | citywok: me either |
| 17:29.26 | citywok | what does it say when you tell it to hang up? |
| 17:29.40 | citywok | csgtacsip1*CLI> channel request hangup SIP/1593-000008f3 |
| 17:29.40 | citywok | Requested Hangup on channel 'SIP/1593-000008f3' |
| 17:29.42 | luckman212 | pbx*CLI> channel request hangup SIP/704-000000da |
| 17:29.42 | luckman212 | Requested Hangup on channel 'SIP/704-000000da' |
| 17:29.52 | luckman212 | yep |
| 17:29.55 | citywok | what happens right after that? |
| 17:30.01 | luckman212 | nuttin' |
| 17:30.24 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 17:30.37 | citywok | verbosity? |
| 17:31.08 | luckman212 | hmm let me check again with -vvvvvv |
| 17:31.21 | luckman212 | Verbosity was 0 and is now 7 |
| 17:31.27 | luckman212 | pbx*CLI> channel request hangup SIP/704-000000da |
| 17:31.27 | luckman212 | Requested Hangup on channel 'SIP/704-000000da' |
| 17:31.27 | luckman212 | pbx*CLI> |
| 17:31.32 | Scorpio2007 | thehar: i would like to be able to valetpark someone on an extension .. and my blf would light up cuz there is a call parked |
| 17:31.36 | luckman212 | that's all she wrote |
| 17:32.26 | citywok | luckman212: what exact version are you using? source? package? |
| 17:32.59 | luckman212 | 1.8.1.1 from digium source |
| 17:33.45 | *** join/#asterisk epaphus (~epaphus@ec2-50-16-226-177.compute-1.amazonaws.com) |
| 17:33.51 | Scorpio2007 | exten => _*23XXX,1,Set(PARKINGEXTEN=${EXTEN:3}) |
| 17:33.51 | Scorpio2007 | exten => _*23XXX,n,Park() |
| 17:33.51 | Scorpio2007 | exten => _*24XXX,1,ParkedCall(${EXTEN:3}) |
| 17:33.51 | Scorpio2007 | exten => _*24XXX,hint,park:${EXTEN:3}@parkedcalls |
| 17:33.51 | Scorpio2007 | this seems to work for parked calls .. my blf key set to *24555 and when i dial *23555 they get parked on 555 and my blf key is blinking |
| 17:33.57 | luckman212 | maybe a bug? |
| 17:34.02 | thehar | Scorpio2007: One second. |
| 17:34.05 | pabelanger | Scorcerer: use pb |
| 17:34.08 | epaphus | Hello all. Is it possible to make a PHP script that will interface with TrixBox so that I can place in my website and when people request a call back .. asterisk will make a conference call immediately? |
| 17:34.25 | citywok | luckman212: i'm compilining 1.8.1.1 on my dev server to try it out. |
| 17:34.32 | *** join/#asterisk ickmund (~ickmund@c-1f4be755.015-144-70697410.cust.bredbandsbolaget.se) |
| 17:34.39 | pabelanger | epaphus: yes |
| 17:34.40 | luckman212 | citywok: k, thx :D |
| 17:34.41 | citywok | epaphus: yes. you need to learn the asterisk AMI |
| 17:34.45 | pabelanger | anything is possible |
| 17:35.38 | epaphus | thanks... |
| 17:35.58 | luckman212 | I issued a 'core stop now' ... that killed it (and everything else too... ) |
| 17:36.48 | citywok | funny how that works ;) |
| 17:36.51 | pabelanger | luckman212: possible deadlock? |
| 17:36.57 | pabelanger | *CLI> core show locks |
| 17:37.13 | citywok | pabelanger: i doubt he has that compiler flag enabled |
| 17:37.15 | luckman212 | no such command in 1.8 |
| 17:37.25 | citywok | luckman212: you have to enable a compiler flag for it |
| 17:37.27 | luckman212 | well at least not on my build |
| 17:37.31 | luckman212 | ah ha... |
| 17:37.50 | luckman212 | i will look into that... but I suppose that flag isn't a good thing to leave enabled on a production system? |
| 17:37.53 | citywok | can you ~pb your dialplan for me? |
| 17:38.14 | citywok | i've run with some of the debugging stuff enabled without too much trouble. depends what options you enable, how much logging, and how much load you have. |
| 17:38.29 | epaphus | wonders if there is some kind of plugin already made for this |
| 17:38.43 | citywok | beyond 10-15 calls i started to have call quality issues with debug threads and dont optimize |
| 17:39.04 | citywok | epaphus: i'm sure plenty of people have written code to do exactly what you want. i doubt any of them made it a plugin :P |
| 17:39.31 | citywok | epaphus: http://lmgtfy.com/?q=asterisk+php+api |
| 17:39.37 | epaphus | hahah ok... |
| 17:42.10 | epaphus | The other thing i was wondering about is.. How to make asterisk listen on one FXO port for incoming calls and route extentions to ring through another FXO port (my cell phone).. and if I dont answer in 10 seconds.. ring another phone, and if not go to voicemail. Is there some code for that? how could i search this? |
| 17:42.11 | luckman212 | citywok: sorry for the noob question, but how do I ~pb the dialplan.. not sure quite what you meant |
| 17:42.13 | Scorpio2007 | thehar: okay |
| 17:42.21 | citywok | ~pb |
| 17:42.21 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 17:43.03 | citywok | epaphus: yes. you will need to create dialplan. extensions.conf |
| 17:44.09 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 17:44.19 | luckman212 | ah.. a pastebin... of course |
| 17:47.43 | citywok | luckman212: clean 1.8.1.1 build, dial in to a meetme bridge, channel requset hangup works fine. |
| 17:48.48 | epaphus | citywok, would that be something very advanced to do? |
| 17:48.56 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 17:49.02 | citywok | epaphus: no |
| 17:49.29 | luckman212 | citywok: hmm maybe it was some sort of lock, as pabelanger suggested |
| 17:49.47 | luckman212 | citywok: I'll never know, because I did a core stop anyway |
| 17:50.10 | citywok | heh. i have a nightly core restart now. my * box rarely ever crashes now with that. |
| 17:50.17 | luckman212 | thanks for the help though guys :D |
| 17:50.29 | luckman212 | you schedule it in cron? |
| 17:50.44 | citywok | 0 1 * * * /usr/sbin/asterisk -rx 'core restart now' |
| 17:52.38 | luckman212 | heh.. thats a little brutal but I guess it works |
| 17:53.10 | luckman212 | how 'bout 'core restart gracefully' |
| 17:53.17 | citywok | nobody is ever on the phone at midnight :P -- b/c if there were a lock it wouldn't restart. |
| 17:53.37 | citywok | or restart when convenient |
| 17:54.24 | luckman212 | would be cool if you could announce to any active channels that it was about to get killed, at 11:55PM |
| 17:54.30 | luckman212 | would that be possible somehow? |
| 17:54.49 | citywok | sure. use a redirect and send it to a context with a playback(about-to-restart) |
| 17:55.22 | luckman212 | and after it played that, the channel would continue? |
| 17:55.27 | citywok | Does anybody have a polycom IP550/650 with an AC adapter near by? can you tell me the ratings on the thing? |
| 17:56.01 | pabelanger | *CLI> core stop gracefully |
| 17:56.04 | pabelanger | is a little nicer |
| 17:56.05 | citywok | luckman212: i'd hang it up :), i suppose it's possible to put it back together. or send both callers in to a confbridge and use that to inject the audio. |
| 17:56.34 | citywok | b/c we are a 5am-5pm PST call center, i'm not particulalry worried about just killing calls at midnight |
| 17:56.38 | luckman212 | citywok: 24VDC 600mA, 2.1x5.5mm Center (+) Plug |
| 17:57.18 | luckman212 | citywok: sure sure, I understand.. it was just something that I was curious about |
| 17:57.46 | citywok | pretty much anything is possible, the question is how complicated |
| 17:57.56 | citywok | ty, i wasn't sure and didnt' want to find out the hard way. |
| 17:58.36 | luckman212 | I have thought about it... I remember reading about a way to inject (mix, actually) a WAV file into a channel, e.g. for playing back promotional messages "on top" of MOH. But I forgot where that was or how to do it. |
| 17:59.11 | citywok | hmm. anybody got the POE adapter? is that also 24v or 48? |
| 17:59.18 | pabelanger | heh, I thought there would have been an AMI event for DTMF transfers (feature.conf) |
| 17:59.30 | pabelanger | s/feature/features/ |
| 17:59.41 | citywok | really? there isn't? |
| 18:01.29 | pabelanger | citywok: Nope, guess I need to write a patch |
| 18:01.36 | citywok | doh |
| 18:01.41 | citywok | i'm glad i don't have to do that! :P |
| 18:01.56 | luckman212 | ^^^ what he said |
| 18:07.38 | *** join/#asterisk dlyneswork (~dlynes@216.185.79.50) |
| 18:10.45 | *** join/#asterisk TdM2 (TdM2@c-24-2-244-4.hsd1.ma.comcast.net) |
| 18:10.57 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 18:12.17 | *** join/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 18:14.35 | *** join/#asterisk epaphus (~epaphus@ec2-50-16-226-177.compute-1.amazonaws.com) |
| 18:15.10 | epaphus | Hello, so what is needed aside from a minimal install of asterisk to have speech recognition in the Ivr ? |
| 18:16.20 | *** join/#asterisk moy (~moy@CPE003048b1f1b3-CM0026f396812d.cpe.net.cable.rogers.com) |
| 18:16.38 | citywok | epaphus: google |
| 18:16.43 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:18.42 | epaphus | citywok, right.. i see many options but all of the ones i see are commercial. Is there any open source free addon ? |
| 18:19.43 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.180.137.dsl.dyn.forthnet.gr) |
| 18:20.13 | citywok | epaphus: probably not if nothing shows up in the searches :P |
| 18:20.45 | TdM2 | why do people always say google |
| 18:20.57 | TdM2 | if i wanted to google it i would have thats why they asked the question |
| 18:21.04 | TdM2 | lame |
| 18:21.49 | citywok | b/c we get tired of answering questions that are easily google-able :) |
| 18:22.01 | TdM2 | yeah well make a FAQ lol |
| 18:22.09 | TdM2 | google is becoming to bloated |
| 18:22.13 | TdM2 | too much information |
| 18:22.16 | citywok | TdM2: it's called the wiki |
| 18:22.24 | citywok | which you can search easily... using google :P |
| 18:22.42 | GhOnDiE | if i type in something i get like 20000 pages of rubbish and maybe 1 or 2 pages of actual useful content |
| 18:22.52 | TdM2 | hard to sift through especially with people doing crafty things to make nonsense appear as if it were the answer people were looking for and then it turns up to be an ad or just some jibberish |
| 18:22.58 | *** join/#asterisk ickmund (~ickmund@c-1f4be755.015-144-70697410.cust.bredbandsbolaget.se) |
| 18:22.59 | citywok | then you need to learn how to craft your searches |
| 18:23.01 | TdM2 | EXACTLY! |
| 18:23.03 | *** join/#asterisk veganadian (~andrew@206.53.59.108) |
| 18:23.18 | TdM2 | at least someone knows what im talkin about |
| 18:23.37 | citywok | most of us know what we are looking for, and how to write that down in 3 or 4 words to get what we are looking for :) |
| 18:23.47 | TdM2 | lol |
| 18:23.52 | TdM2 | i need a new car |
| 18:23.53 | pabelanger | TdM2: LumenVox or Sphinx |
| 18:23.56 | citywok | so we have no idea what your issue is |
| 18:23.59 | TdM2 | doesnt mean ill find it on google |
| 18:24.12 | TdM2 | i have no issues |
| 18:24.13 | TdM2 | lol |
| 18:24.25 | citywok | TdM2: if you don't specify what you are looking for you will get 20,000 pages of new cars. nothing related ot the new car you want |
| 18:24.33 | dlyneswork | epaphus, there's plenty of speech recognition libraries out there, that are open source: CMU Sphinx (http://cmusphinx.sourceforge.net/), Hidden Markov Model Toolkit (http://htk.eng.cam.ac.uk/), ... |
| 18:24.41 | TdM2 | just tired of smart ass people giving smart ass responses, when they know people dont know and need help |
| 18:24.45 | citywok | do you drive down to auto-row and say I NEED A NEW CAR, and expect them to hand you the exact one you wanted? |
| 18:24.58 | TdM2 | anyway |
| 18:25.03 | dlyneswork | epaphus, I believe there's an asterisk module written at one point or another that uses CMU Sphinx |
| 18:25.06 | citywok | TdM2: i'm just tired of people asking questions which i can answer with a 4 word google search |
| 18:25.09 | TdM2 | sphinx |
| 18:25.21 | epaphus | dlynes, thanks.. |
| 18:25.35 | citywok | http://lmgtfy.com/?q=asterisk+open+source+voice+recognition |
| 18:25.44 | Ad-Hoc | hi ppl |
| 18:25.45 | TdM2 | cute |
| 18:25.47 | TdM2 | lol |
| 18:26.14 | thehar | citywok: good thing that russellb isn't around |
| 18:26.37 | citywok | thehar: no kidding. don't say his name, it might trigger something from his vacation :P |
| 18:26.41 | thehar | hahahahahaha |
| 18:26.43 | dlyneswork | citywok, I like this one better: http://www.justfuckinggoogleit.com/?q=asterisk+open+source+voice+recognition |
| 18:26.57 | citywok | OH YOURS IS WAY BETTER |
| 18:27.29 | citywok | awwww but it doesn't work. Google Error: Forbidden |
| 18:27.42 | dlyneswork | citywok, funny....works for me just fine |
| 18:28.10 | citywok | Your client does not have permission to get URL /custom?q=asterisk+open+source+voice+recognition&sa=Search&client=pub-5834014132134539&forid=1&ie=UTF-8&oe=UTF-8&cof=GALT%3A%23008000%3BGL%3A1%3BDIV%3A%23336699%3BVLC%3A663399%3BAH%3Acenter%3BBGC%3AFFFFFF%3BLBGC%3A336699%3BALC%3A0000FF%3BLC%3A0000FF%3BT%3A000000%3BGFNT%3A0000FF%3BGIMP%3A0000FF%3BFORID%3A1%3B&hl=en from this server. (Client IP address |
| 18:28.16 | leifmadsen | heh I was just about to post that too :) |
| 18:28.24 | leifmadsen | same error here |
| 18:28.33 | dlyneswork | really? |
| 18:28.34 | citywok | dlyneswork: where are you located? NA? |
| 18:28.43 | dlyneswork | citywok, yes...Hamilton, ON, Canada |
| 18:29.24 | dlyneswork | citywok, why does it mention 'GIMP' in your url? |
| 18:29.26 | leifmadsen | <-- Toronto |
| 18:29.48 | citywok | i copy pated the URL you gave me in to chrome, and copy pasted the error back here |
| 18:30.02 | dlyneswork | citywok, lemme try it in chrome...I was using Firefox 4 |
| 18:30.23 | dlyneswork | citywok, it actually redirects to: http://www.google.com/custom?q=asterisk+open+source+voice+recognition&sa=Search&client=pub-5834014132134539&forid=1&ie=UTF-8&oe=UTF-8&cof=GALT%3A%23008000%3BGL%3A1%3BDIV%3A%23336699%3BVLC%3A663399%3BAH%3Acenter%3BBGC%3AFFFFFF%3BLBGC%3A336699%3BALC%3A0000FF%3BLC%3A0000FF%3BT%3A000000%3BGFNT%3A0000FF%3BGIMP%3A0000FF%3BFORID%3A1%3B&hl=en |
| 18:31.07 | dlyneswork | citywok, does the redirect url work for you? |
| 18:31.16 | citywok | It works in IE, but not in Chrome. |
| 18:31.29 | dlyneswork | Ah....so it's a chrome issue then...not a google issue |
| 18:31.42 | citywok | dlyneswork: yes, in IE. the direct URL you gave me works in chrome too. but the justfuckinggoogleit redirector sends it to the wrong place |
| 18:31.49 | p3nguin | Didn't Google make Chrome? |
| 18:31.56 | p3nguin | I guess that makes Chrome a Google issue. |
| 18:32.00 | citywok | yes, but they didn't make justfuckinggoogleit |
| 18:32.15 | wizard171 | Worked for me Firefox ... 3.6.13/Linux/Texas ... and I like the ... explanation ... heh! |
| 18:32.38 | citywok | i just grabbed the URL from the source and copy/pasted that in and it works. chrome fail! |
| 18:35.09 | dlyneswork | Yeah, but if you click 'here' to continue your search on google, it works just fine |
| 18:35.39 | p3nguin | Damn that's a lot of extra work. |
| 18:36.04 | citywok | the URL in the google links is different than what it redirects you too. |
| 18:37.05 | dlyneswork | Chrome is fscked...what can I say? :) |
| 18:37.09 | *** join/#asterisk Tim_Toady (~moi@77.49.3.151.dsl.dyn.forthnet.gr) |
| 18:37.31 | citywok | lol. yea it doesn't do perfect on the web anyways. |
| 18:37.44 | citywok | but holy god it renders pages so unbelievably fast you'll wonder why you didn't quit using firefox sooner |
| 18:37.58 | citywok | and you'll be confused as to why you no longer need 8gb of memory |
| 18:39.19 | dlyneswork | I still use firefox, so that I can actually load every page I try to :) |
| 18:40.07 | citywok | lol. i like that chrome has the IIS integrated authentication worked in to it so i can go to internal sites such as sharepoint in chrome. take THAT FF! :P |
| 18:40.19 | dlyneswork | I even tried turning off that web page resolver option that chrome has, and it doesn't seem to do anything |
| 18:40.26 | leifmadsen | woh I just found a use for MASTER_CHANNEL() :) |
| 18:40.46 | citywok | ? |
| 18:41.26 | dlyneswork | leifmadsen, I guess Digium's not giving their Canadian employees this week off? |
| 18:41.41 | leifmadsen | dlyneswork: I'm doing work for other clients :) |
| 18:41.50 | dlyneswork | leifmadsen, ah... |
| 18:42.10 | dlyneswork | leifmadsen, so you're not in oslo, anymore? |
| 18:42.11 | leifmadsen | a lot of the Digium folks do have this week off though since they took vacatoin time |
| 18:42.19 | leifmadsen | dlyneswork: I've never been to Oslo... |
| 18:42.37 | dlyneswork | * [leifmadsen] gibson.freenode.net :Oslo, Norway |
| 18:42.38 | leifmadsen | Born in Petrolia, living in Toronto for about 10 years? |
| 18:42.49 | leifmadsen | must just be the server I'm connected to? |
| 18:42.51 | dlyneswork | ewww....sarnia :p |
| 18:42.56 | leifmadsen | ewww indeed :) |
| 18:43.15 | pabelanger | Sarnia is serious business |
| 18:43.31 | dlyneswork | pabelanger, you mean for the unemployment industry? |
| 18:44.05 | pabelanger | What you talking about? Chemical Valley! |
| 18:44.22 | dlyneswork | ummmm...what valley? it's all flat |
| 18:44.39 | TdM2 | epaphus: I found a link for you for the speech recognition check out http://www.syednetworks.com/asterisk-integration-with-sphinx-voice-recognition-system |
| 18:44.54 | devmod | using asterisk branch 1.8, keeping my old config I get no debug/verbose msgs into my console, do I need to change something? |
| 18:45.08 | TdM2 | logger.conf |
| 18:45.09 | bmoraca_work | has anyone successfully used asterisk as a class 5 switch with IN and other application-level services? |
| 18:45.33 | dlyneswork | Btw, are there any incompatibilities between the latest 1.4 branch and 1.8? |
| 18:45.47 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 18:45.48 | citywok | i think | is completely deprecated now. leif? |
| 18:46.03 | dlyneswork | Either for asterisk SIP clients, or asterisk IAX2 clients? |
| 18:46.07 | pabelanger | CHANGES and UPGRADE.txt |
| 18:46.55 | dlyneswork | pabelanger, that tells me what changed and what I have to watch out for when upgrading the local box...it doesn't mention whether protocol implementations have changed, or not |
| 18:47.06 | leifmadsen | ya, no more |'s |
| 18:47.32 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 18:47.38 | dlyneswork | leifmadsen, no more |'s as of 1.6.1, right? |
| 18:47.41 | citywok | dlyneswork: it's a safe bet there have been changes to chan_sip and chan_iax2. |
| 18:47.47 | leifmadsen | something like that |
| 18:47.53 | leifmadsen | I haven't used 1.6.1 in years :) |
| 18:47.59 | citywok | but we're all using it with bandwidth.com/voicepulse/flowroute just fine :P |
| 18:48.00 | dlyneswork | citywok, i'm sure there are....i'm just wondering if 1.4 can still talk to 1.8 |
| 18:48.09 | leifmadsen | yes |
| 18:48.19 | dlyneswork | leifmadsen, sip and iax2? |
| 18:48.30 | leifmadsen | I don't use iax2 |
| 18:48.33 | leifmadsen | but it should |
| 18:48.35 | TdM2 | ive been sucessful in foip with google voice |
| 18:48.38 | dlyneswork | leifmadsen, just as long as I'm running at least 1.4.26, right? |
| 18:48.44 | leifmadsen | no idea |
| 18:48.55 | leifmadsen | I don't interoperate versions |
| 18:49.08 | dlyneswork | leifmadsen, ok...I just remember 1.4.25 and lower weren't compatible with 1.6.1 |
| 18:49.25 | leifmadsen | probably because of the security issue with call tokens |
| 18:49.38 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 18:49.43 | dlyneswork | leifmadsen, could be...I just remember it affected both iax2 and sip |
| 18:49.54 | leifmadsen | ya no idea :) |
| 18:49.59 | leifmadsen | waves to pabelanger |
| 18:50.12 | pabelanger | infobot: lick leifmadsen |
| 18:50.12 | infobot | ACTION licks leifmadsen *SHLUUURRRRPPP* |
| 18:50.20 | leifmadsen | nice |
| 18:50.25 | leifmadsen | saves me from a shower |
| 18:50.43 | *** join/#asterisk Benwa (~Schnitzel@unaffiliated/benwa) |
| 18:50.45 | pabelanger | dlyneswork: biggest difference will be IPv6 support |
| 18:50.59 | dlyneswork | pabelanger, yeah...that really has no bearing on me, currently |
| 18:51.51 | dlyneswork | pabelanger, but i guess with ipv6, I can run ipsec vpns with less overhead? |
| 18:53.31 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 18:53.44 | *** join/#asterisk cusco (~tralala@49.192.54.77.rev.vodafone.pt) |
| 18:54.21 | cusco | caller -> asterisk --dial--> dahdi |
| 18:54.34 | cusco | can caller listen to musiconhold instead of ringing tone |
| 18:55.00 | pabelanger | cusco: *CLI> core show application Dial |
| 18:55.04 | citywok | m: Provide Music on Hold to the calling party until the called channel answers. This is mutually exclusive with option 'r', obviously. Use m(class) to specify a class for the music on hold. |
| 18:55.55 | pabelanger | dlyneswork: unknown |
| 18:56.06 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 18:59.15 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 19:02.58 | *** join/#asterisk thansen (~thansen@c-174-52-84-4.hsd1.ut.comcast.net) |
| 19:04.27 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 19:25.01 | *** join/#asterisk joel_oliveira (~chatzilla@alpes.nortenet.pt) |
| 19:25.42 | cusco | ola joel_oliveira |
| 19:25.45 | cusco | lol |
| 19:26.05 | joel_oliveira | ola! portugues? |
| 19:26.08 | cusco | thanks pabelanger, citywok |
| 19:26.10 | cusco | sim |
| 19:28.16 | joel_oliveira | hi all! just to ask if anyone has a simple way to have startup scripts running on asterisk startup? or should I just change the /etc/init.d/asterisk so that I have a system call? |
| 19:31.34 | *** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f2-69-94-238-26.totalprocess.net) |
| 19:35.17 | Katty | you could use the rc.local |
| 19:35.28 | thehar | orly |
| 19:35.43 | Katty | i use rc.local for isymphony startup scripts |
| 19:35.53 | p3nguin | Start-up scripts running on asterisk? What are you trying to actually do? |
| 19:35.55 | Katty | thehar: hai sir |
| 19:36.01 | thehar | Katty: hi |
| 19:36.14 | Katty | thehar: may your day be merry |
| 19:36.16 | Katty | thehar: and bright |
| 19:36.25 | Katty | thehar: with tropical heatwave |
| 19:36.30 | thehar | it isn't lol |
| 19:36.31 | thehar | but thanks |
| 19:36.50 | Katty | :< |
| 19:36.55 | Katty | let's hug. |
| 19:36.57 | Katty | it'll be a smidgen brighter |
| 19:37.00 | Katty | hugs on thehar |
| 19:37.52 | thehar | hugs Katty |
| 19:37.56 | thehar | it's pouring outside |
| 19:37.59 | thehar | i left my jacket in the car |
| 19:38.02 | thehar | i need lunch :\ |
| 19:40.12 | Katty | :<<< |
| 19:40.15 | *** join/#asterisk JunK-Y (~junky@64.15.77.94) |
| 19:40.19 | Katty | hi junky |
| 19:40.23 | JunK-Y | hola |
| 19:43.14 | *** join/#asterisk cmendes0101 (~nn@pool-173-58-41-188.lsanca.fios.verizon.net) |
| 19:45.13 | joel_oliveira | p3nguin: I just want to sync at startup the astdb file with another instance of Asterisk running on another machine |
| 19:45.30 | joel_oliveira | but I guess I will change the /etc/init.d/asterisk file to achieve that :) |
| 19:45.32 | p3nguin | Start up of Asterisk or start up of the OS? |
| 19:45.39 | joel_oliveira | of asterisk |
| 19:47.10 | Katty | hmm. |
| 19:47.10 | p3nguin | I'd probably use the asterisk init script or another script of my own to do it. Either way, if you need to wait until asterisk is all the way up, you can use the "core waitfullybooted" command to wait for it to be up. |
| 19:47.39 | joel_oliveira | p3nguin: didn't know that last part. thanks |
| 19:48.00 | JunK-Y | maybe an option in asterisk.conf should be created, when it starts, it could execute that script |
| 19:48.20 | Katty | unless you're in russia. |
| 19:48.24 | Katty | then script would execute you |
| 19:49.13 | JunK-Y | <PROTECTED> |
| 19:49.27 | Katty | i'llb e here all week |
| 19:49.37 | drmessano | She must break yu |
| 19:49.40 | leifmadsen | claps |
| 19:51.29 | thehar | i'll teach you (*&$#es to saturate my connection |
| 19:51.32 | Letoric | Anybody able to point me to some good docs on how to properly implement *67 and any other * codes that I might want to use? I'd like to be able to have them push the feature code, then get the dial tone like other phone systems provide to continue their dialing |
| 19:51.43 | thehar | flexes his shaping muscles |
| 19:51.43 | leifmadsen | ~seen atis |
| 19:51.44 | infobot | atis <n=atis@123.236.110.207> was last seen on IRC in channel #kde, 419d 3h 33m 13s ago, saying: 'i'm from india'. |
| 19:51.44 | Katty | oh come on, thehar |
| 19:51.50 | Katty | don't tell me you don't like your connection saturated. |
| 19:51.52 | thehar | it shouldn't take 30 seconds for 'ls' to work |
| 19:51.55 | Katty | i don't believe that for a minute |
| 19:51.56 | *** join/#asterisk marksaitis (~MK@78-61-148-80.static.zebra.lt) |
| 19:51.56 | thehar | Katty: LOL |
| 19:52.16 | marksaitis | Hey. Any new guides on how to utilize TLS and SRTP in asterisk? |
| 19:52.41 | marksaitis | anybody knows any good voip clients with tls and srtp for win? |
| 19:55.07 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 19:55.55 | Katty | marksaitis: i like zoiper, but idk if it supports tls or srtp |
| 19:56.18 | p3nguin | letoric: Done it. If you'll wait, I'll get it and share with you. |
| 19:56.44 | Letoric | p3nguin: Thanks - wait until when? :) |
| 19:58.00 | marksaitis | has anybody got asterisk working with softphone, tls and srtp? |
| 19:59.03 | JunK-Y | marksaitis: look for siptls.txt |
| 20:00.01 | cmendes0101 | Q: I have asterisk running as root and I would like other users to be able to execute commands with asterisk -rx. I get the error saying unable to connect to asterisk and asks if the ctl file exists. Anyway to do this? |
| 20:00.08 | p3nguin | letoric: http://pastebin.com/ScCzgiDv |
| 20:00.35 | Letoric | Thank you |
| 20:00.42 | marksaitis | where is siptls.txt ? :) |
| 20:01.51 | leifmadsen | marksaitis: probably here: https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport |
| 20:02.19 | leifmadsen | JunK-Y: all the documentation files are imported and on the asterisk wiki now |
| 20:02.40 | leifmadsen | marksaitis: it'd have been in the doc/ subdir of your asterisk source |
| 20:03.01 | marksaitis | leifmadsen, thank you |
| 20:03.40 | JunK-Y | leifmadsen: ya, i saw all the ACF, didnt knew for the doc/ |
| 20:03.46 | JunK-Y | thanks for pointing out |
| 20:06.06 | drmessano | Anyone have a line on the MoH issue that seems to be related to dahdi timing? --> https://issues.asterisk.org/view.php?id=18262 |
| 20:06.56 | drmessano | That issue is part of it.. My streaming MoH doesn't work AT ALL.. not just for parked calls, when I have res_timing_dahdi.so loaded |
| 20:08.14 | *** join/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 20:09.33 | *** join/#asterisk jblack (~jblack@pool-71-181-222-126.sctnpa.east.verizon.net) |
| 20:10.34 | drmessano | Hmmmm guess not |
| 20:10.35 | drmessano | lol |
| 20:12.44 | *** join/#asterisk ickmund (~ickmund@c-8e4be755.015-144-70697410.cust.bredbandsbolaget.se) |
| 20:17.27 | *** join/#asterisk marksaitis (~MK@78-61-148-80.static.zebra.lt) |
| 20:23.04 | citywok | no moh even if you just send a call to musiconhold? |
| 20:23.18 | drmessano | Correct.. |
| 20:23.50 | citywok | moh show classes |
| 20:23.50 | citywok | and moh show files |
| 20:23.53 | drmessano | If I do a noload on the dahdi timer and restart, works fine for a few hours, then I can't make any calls.. and need to restart |
| 20:24.03 | drmessano | MoH isn't the problem, specifically |
| 20:24.18 | drmessano | Like I say, I dump the dahdi timer and it works fine |
| 20:25.09 | drmessano | Not sure what the secondary issue is, with the calls not working after a few hours, but it seems like #18262, what I am seeing with Streaming MoH is related, and maybe more severe |
| 20:25.30 | drmessano | if I use any of the stock flat file MoH, I only see it with parked calls, as noted in that ticket |
| 20:25.46 | drmessano | But streaming MoH doesn't work _at all_ with the dahdi timer loaded |
| 20:26.20 | drmessano | It's BEEN working on this box for a number of years now.. until recent 1.6.2.x upgrades, and through my migration to 1.8 |
| 20:27.05 | m_tadeu | why isn't the Read command reading anything from my softphone? |
| 20:30.00 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 20:32.22 | citywok | m_tadeu: DTMF issues? |
| 20:34.10 | m_tadeu | citywok: probably...and I have no ideia on where to start checking for problems |
| 20:36.12 | *** join/#asterisk devyll (~paul@thpallady.net.hostway.ro) |
| 20:36.16 | wizard171 | m_tadeu: your softphone is SIP? If yes, then what "dtmfmode=" setting for you in sip.conf? |
| 20:36.51 | devyll | is there a way to invoke from ssh asterisk pbx to make a call to X and then to continue with the dialplan ? |
| 20:37.46 | m_tadeu | wizard171: it is sip...in sip.conf I have dtmfmode=info |
| 20:37.53 | citywok | devyll: you can originate a call using the AMI, which you can telnet in to |
| 20:37.54 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 20:38.09 | p3nguin | devyll: ssh, asterisk -rx, originate, etc. |
| 20:39.08 | wizard171 | m_tadeu: and your softphone is configured for same? |
| 20:41.36 | m_tadeu | wizard171: it has 2 settings....one for rfc-2833, which is wrong, and a sip info, which stops my phone from getting any sound from asterisk |
| 20:42.25 | p3nguin | rfc-2833 would be invalid, but rfc2833 is usually preferred. |
| 20:42.56 | m_tadeu | p3nguin: so I'd better change it in sip.conf? |
| 20:43.05 | p3nguin | There's no other place. |
| 20:43.55 | m_tadeu | thanx |
| 20:44.18 | p3nguin | Be sure you set it correctly in each peer definition. |
| 20:45.37 | devyll | p3nguin, how can I originate a call from CLI ? |
| 20:46.00 | p3nguin | devyll: Use the originate command. |
| 20:46.22 | devyll | p3nguin, I can't seem to have this command ... |
| 20:46.23 | m_tadeu | p3nguin: if I set it on each peer, will it force the phone to use it? |
| 20:46.23 | p3nguin | devyll: originate, without any options or parameters, will show the usage of the command. |
| 20:46.33 | wizard171 | m_tadeu: the softphone and asterisk's "sip.conf" need to match ... for each peer ... it can be in [global] or in the definition of your phone ... as p3nguin says, better to make sure in each peer, instead of relying on [global] |
| 20:46.56 | devyll | No such command 'originate' (type 'help originate' for other possible commands) |
| 20:47.04 | p3nguin | m_tadeu: Force, no... but it's going to try like hell. Your phone will either need to be on auto or also on rfc2833. |
| 20:47.27 | m_tadeu | p3nguin: niceeee |
| 20:48.06 | p3nguin | devyll: Maybe you can google how to use originate with your version of asterisk. It works on mine exactly as I described. |
| 20:48.27 | seanbright | probably doesn't have the module loaded |
| 20:48.52 | devyll | p3nguin, thanks |
| 20:48.58 | *** join/#asterisk marksaitis (~MK@78-61-148-80.static.zebra.lt) |
| 20:49.02 | devyll | probably I need to enable a module or something |
| 20:49.43 | seanbright | what version of asterisk? |
| 20:50.33 | p3nguin | I guess res_clioriginate.so must provide it. |
| 20:51.11 | devyll | 1.4.24 |
| 20:51.13 | wizard171 | ... :) ... yeah, that would be it ... |
| 20:51.49 | p3nguin | Check to see if you have that module. If you have it, load it. If you do not have it, you'll need to go back into menuselect and enable it, then rebuild. |
| 20:52.16 | devyll | p3nguin, thanks |
| 20:52.18 | devyll | I have it |
| 20:52.22 | devyll | will enable and see if it works |
| 20:52.31 | p3nguin | module load res_clioriginate.so |
| 20:52.42 | p3nguin | Then run originate and see the usage. |
| 20:53.14 | p3nguin | If you still don't understand how to use it, explain what you don't understand and someone (or I) will help you. |
| 20:54.54 | p3nguin | For now I have to step out, so hopefully someone else will tell you what you need to know. |
| 20:55.21 | Katty | hmm. feels like naptime |
| 20:55.37 | drmessano | flips open the covers on the other side of the bed |
| 20:55.49 | Katty | gosh. |
| 20:56.08 | drmessano | I get that all the time |
| 20:56.35 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 20:59.10 | dlyneswork | Katty, how's your squirrels? |
| 20:59.57 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 21:03.12 | drmessano | That's a bit personal |
| 21:05.23 | WIMPy | What is missing whe I get "wct4xxp 0000:03:08.0: VPM450: firmware dahdi-fw-oct6114-064.bin not available from userspace"? |
| 21:09.00 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 21:09.40 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 21:10.32 | WIMPy | The firmware is availabe in /usr/lib/hotplug/firmware/dahdi-fw-oct6114-064.bin and /lib/firmware/dahdi-fw-oct6114-064.bin. |
| 21:10.53 | citywok | no idea, i avoid dahdi yay.jpg |
| 21:11.01 | citywok | pure sip ftw |
| 21:13.03 | frigidzephyr | WIMPy: i have not seen that error in a loong time. What version of DAHDI are you using? |
| 21:13.25 | WIMPy | 2.4.0 |
| 21:14.40 | devyll | originate works perfect |
| 21:14.50 | devyll | however .. I need to pass some arguments .. |
| 21:16.35 | *** part/#asterisk yonahw (~yonahw@www.mcatrack.com) |
| 21:16.42 | frigidzephyr | WIMPy: strange, file permissions? |
| 21:17.30 | WIMPy | 644 |
| 21:17.59 | WIMPy | Do I need soem other userspace stuff I might be missing. |
| 21:17.59 | WIMPy | ? |
| 21:20.06 | wizard171 | WIMPy: you get this when? (anything prior to in log?) |
| 21:20.47 | WIMPy | I get that on dahdi_cfg, wich BTW takes at least a minute and throws that error twice. |
| 21:21.27 | WIMPy | The rest looks pretty normal. |
| 21:21.47 | WIMPy | The card is working, but without HWEC. |
| 21:23.14 | wizard171 | WIMPy: is it sharing an interrupt per chance? (with something it can't, or shouldn't) |
| 21:23.46 | WIMPy | No. It's a PCI-e version. |
| 21:24.10 | WIMPy | Although it actually lists IRQ 11. Hmm. But it's not shared. |
| 21:24.27 | WIMPy | And except for HWEC it is fully working. |
| 21:31.04 | *** join/#asterisk chuckp (~chuckp@c-76-106-198-76.hsd1.fl.comcast.net) |
| 21:31.33 | *** join/#asterisk lirakis (~lirakis@ool-ad022bb1.dyn.optonline.net) |
| 21:31.46 | lirakis | hey everyone |
| 21:32.11 | lirakis | ive recently come into some where between 10-15 Polycom soundpoint 501 phones that I want to get rid of |
| 21:32.21 | lirakis | they are used and in a box. i'd prefer to unload them in reasonable bulk |
| 21:32.31 | lirakis | and for this ... im happy to accept ridiculously low pricing |
| 21:32.35 | chuckp | I am trying to add a CID name prefix, however it doesn't appear to show on any devices. In fact, if I NoOp the CIDName I get a number that is returned, however the dialparties.agi and the phone displays a real CID name... anyone able to point me in the right direction here? |
| 21:32.37 | Naikrovek | you don't know how many |
| 21:32.41 | chuckp | btw this is a trixbox setup |
| 21:32.55 | Naikrovek | chuckp did you ask in #trixbox |
| 21:33.05 | lirakis | like i dunno ... $30-40 per phone with power and cables + shipping |
| 21:33.40 | lirakis | so i thought before going to ebay id check and see if any one here had use for some polycom 501's |
| 21:33.58 | chuckp | channels dead but no, I figured this seems to be a asterisk issue as I placed the CID name NoOp right after answering the line and I don't see any commands prior to the answer that would be hijacking the CID Name |
| 21:34.35 | wizard171 | WIMPy: how many channels on the card/configured is there a mismatch of some sort ? |
| 21:34.57 | Naikrovek | chuckp: yeah that channel is dead |
| 21:35.05 | Naikrovek | trixbox is dead too, if you didn't kow |
| 21:35.07 | Naikrovek | know* |
| 21:35.09 | WIMPy | Two spans physically and two spans configured, but only one in use. |
| 21:35.38 | chuckp | Looks like it |
| 21:35.39 | *** join/#asterisk timholum1 (~timholum2@68-117-120-138.static.eucl.wi.charter.com) |
| 21:35.48 | Katty | stretches |
| 21:35.55 | wizard171 | WIMPy: is the other "plugged in" or "looped up", etc.? |
| 21:36.01 | thehar | i need today to end, Katty |
| 21:36.04 | WIMPy | open |
| 21:36.11 | Katty | thehar: whyfor? |
| 21:36.17 | Katty | shares cookies and hot cocoa with thehar |
| 21:36.17 | thehar | Katty: this week has been horrible |
| 21:36.19 | chuckp | still it seems odd that the Name would be null when I call in, but somehow as it reaches dialparty.agi it sees a CID Name |
| 21:36.27 | thehar | s/hot cocoa/whiskey/ |
| 21:36.37 | Katty | irish liquor is good in hot cocoa |
| 21:36.43 | Katty | not sure about whisky |
| 21:36.58 | thehar | it's good |
| 21:37.01 | Katty | why has this week been awful |
| 21:37.37 | thehar | my QA deployment was suppose to be done last week |
| 21:37.41 | thehar | and just tons of issues |
| 21:37.47 | thehar | with lots of people and blockers |
| 21:37.57 | Katty | i am sorry to hear this :< |
| 21:38.00 | Katty | but on the upside...christmas |
| 21:38.08 | thehar | i'm working christmas :] |
| 21:38.13 | Katty | why? |
| 21:38.28 | thehar | deadlines |
| 21:38.34 | Katty | do you have children? |
| 21:38.44 | thehar | nope |
| 21:38.47 | Katty | a wife? |
| 21:39.00 | thehar | husband |
| 21:39.05 | thehar | husband-wife |
| 21:39.08 | Katty | k |
| 21:39.14 | Katty | so...why are you working christmas, again? |
| 21:39.16 | Katty | i don't understand |
| 21:39.25 | thehar | 13:38 < thehar> deadlines |
| 21:39.31 | Katty | that's not a good enough reason |
| 21:39.33 | wizard171 | WIMPy: are in a position to try either removing the second config, or "loop up" the other connecter ? (with a plug) and see if dahdi_cfg is still slow ? |
| 21:39.34 | thehar | Welcome to Silicon Valley. |
| 21:39.46 | Katty | that's a very poor choice |
| 21:39.55 | leifmadsen | I don't think I could work in Silicon Valley |
| 21:39.58 | thehar | it's not a choice i get to make :] |
| 21:40.00 | thehar | leifmadsen: lol |
| 21:40.04 | Katty | thehar: that's a shame |
| 21:40.09 | Katty | thehar: because jobs come and go |
| 21:40.10 | Naikrovek | why not? they have air and food and stuff |
| 21:40.20 | Katty | thehar: memories do not have second chances |
| 21:40.28 | thehar | Katty: there are other times for memories |
| 21:40.30 | WIMPy | wizard171: Not at the moment. But it wasn't slow when I removed the VPMOCT64. |
| 21:40.38 | thehar | they are not determined on overbloated holidays |
| 21:40.47 | Katty | thehar: that's a poor attitude |
| 21:40.59 | thehar | not really :] |
| 21:41.06 | Katty | thehar: don't be wishing memories away |
| 21:41.12 | Katty | thehar: i fear you will regret it |
| 21:41.17 | thehar | Christmas isn't the only day you can make memories. |
| 21:41.32 | Katty | but there's only one day per year that's christmas |
| 21:41.41 | Katty | everyone is at home together |
| 21:41.45 | Katty | and your husband is... |
| 21:41.47 | Katty | single for the day |
| 21:41.49 | thehar | oh are they? |
| 21:41.51 | thehar | he is also working |
| 21:42.01 | Katty | so what day are you going to have christmas together? |
| 21:42.07 | drmessano | Katty, please stop forcing your belief system on the rest of us. I, for one, do not believe in "trees" and find this very offensive. |
| 21:42.16 | thehar | When we have time to do it. |
| 21:42.22 | Katty | thehar: good. |
| 21:42.36 | Katty | thehar: spending time together is good (= happy memories are good. |
| 21:42.41 | thehar | This is Ameriiiikaaaaaaaaa we can do whatever we want! |
| 21:42.52 | Katty | apparently not ;) |
| 21:42.55 | drmessano | lol |
| 21:43.04 | thehar | I was being facetious. |
| 21:43.08 | Katty | mhmm |
| 21:43.09 | Katty | k |
| 21:43.16 | thehar | makes cookies for Katty |
| 21:43.22 | Katty | :>>> |
| 21:44.49 | chuckp | are there any new variables handling CID in 1.6.2.x other than the CALLERID() variables? |
| 21:45.41 | chuckp | CALLERID(name) CALLERID(number) should contain the CID coming over the trunk..... right? I feel pretty dumb right now since this should be working.. |
| 21:46.46 | citywok | i use callerid(num), not number. |
| 21:48.19 | chuckp | yea the name is the one I am having issue with, but num and number both seem to function the same |
| 21:51.20 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 21:52.42 | citywok | gotcha. what's the issue? and what kind of trunk are you using? |
| 21:55.31 | *** join/#asterisk sled-dog (~luser@adsl-074-165-241-009.sip.msy.bellsouth.net) |
| 21:56.05 | sled-dog | how do i send commands to asterisk via the shell, as root? I want to issue a reload from the shell |
| 21:56.33 | WIMPy | asterisk -rx |
| 21:56.35 | JunK-Y | asterisk -rx"module reload" |
| 21:56.38 | sled-dog | ah. |
| 21:57.13 | sled-dog | smooth, thanks |
| 21:59.41 | *** join/#asterisk Tili (~Tili@cm161.eta193.maxonline.com.sg) |
| 22:01.43 | chuckp | okay well it only seems to be an issue on calls that come in from the Dahdi trunks... |
| 22:05.04 | m_tadeu | what would be the agi_accountcode var and should it be always empty? |
| 22:05.06 | citywok | ah, you may want to check your configuration there then |
| 22:05.24 | citywok | m_tadeu: unless you set it it will be empty. |
| 22:05.26 | *** join/#asterisk hfb (~hfb@pool-96-229-100-244.lsanca.dsl-w.verizon.net) |
| 22:05.33 | citywok | that's cdr(accountcode) i think |
| 22:06.29 | m_tadeu | citywok: but if a caller is logged(with a sip account) shouldn't that value be set? |
| 22:06.49 | citywok | not unless you set the accountcode value to something? |
| 22:07.21 | m_tadeu | I see....how can I check if a caller is logged? |
| 22:08.00 | citywok | look at the CDR's? |
| 22:09.03 | m_tadeu | I mean...I have to guarantie that the caller is logged, so it generates the cdr properly |
| 22:09.34 | *** join/#asterisk areay (~areay@188.220.16.92) |
| 22:09.47 | areay | hi all... i want to add a PSTN number as a queue member... is this possible? |
| 22:10.01 | *** join/#asterisk [T]ank (~T]ank@206.71.78.158) |
| 22:10.23 | citywok | areay: yes. use a local channel to do it. |
| 22:10.24 | [T]ank | so did anything ever replace astbill? looks like it was a cool project. |
| 22:10.44 | citywok | [T]ank: I wrote my own tools to parse CDR's and do everything |
| 22:11.26 | [T]ank | well... im not savy enough for actually writing my own. :-( |
| 22:12.05 | citywok | time to learn some perl! :P |
| 22:12.15 | areay | citywok: i tried using Local/<number-to-dial>@<outbound-context> but it doesn't work like a regular queue member -- as soon as a caller joins the queue it rings the first queue member's number indefinitely instead of doing a round-robin or ringall |
| 22:12.48 | citywok | then in the outbound-context where you do the dial() cmd add a timeout value. |
| 22:13.09 | areay | citywok, thanks lol... sorry i wasn't thinking there :P |
| 22:13.48 | areay | so i take it the timeout values specified in queues.conf are ignored in favor of the dial command's timeout when using local channels, yeah/ |
| 22:15.40 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-233-75-73.mia.bellsouth.net) |
| 22:16.21 | citywok | areay: if you have a queue member that is a channel, it has to dial the channel itself. so you have to tell it how long to try |
| 22:16.31 | citywok | if you use a local context you are doing your down dialplan, it can't control what you do in the dial cmd |
| 22:16.39 | citywok | s/down/own/ |
| 22:18.02 | areay | citywok, i just tried specifying a timeout value in the dial command, but to no avail.. it still dials indefinitely... also from both the caller and queue member's point of view, it is as if the caller dialled the queue member's number directly... the caller hears a ringing tone instead of hold music, and the queue member sees the original caller's caller ID |
| 22:18.47 | citywok | areay: check all of the options in queues.conf, those are all settings. |
| 22:20.17 | areay | ok i'll recheck the configuration... i'm 99% sure it's right though -- i've been using asterisk queues for years, just never with an external queue member |
| 22:21.53 | areay | citywok, yeah it's definitely set to ringall -- in theory all my members should be ringing, but only one is |
| 22:21.58 | areay | i like the name btw |
| 22:23.42 | areay | i envisaged that when copying my old config, and replacing the queue members with Local channels, that everything else would remain the same |
| 22:23.48 | areay | it's like i'm not even using the queue app |
| 22:26.15 | areay | am i experiencing standard irc delays or am i asking stupid questions? |
| 22:27.06 | [T]ank | not a delay as far as i can see |
| 22:27.34 | areay | just in responses... i accept that people's firs |
| 22:27.41 | areay | *first priority is not irc |
| 22:27.54 | areay | but if i'm asking stupid questions i'd rather be told lol |
| 22:27.58 | citywok | i just tested what you are doing, two queue members, ringall. my cellphone via local channel & my deks phone as members. works fine. |
| 22:28.11 | citywok | both ring at the same time |
| 22:28.11 | [T]ank | ;-) i was just saying there wasnt a delay.... i dont know the answer either |
| 22:28.26 | areay | ok so i'm being dumb... i'll recheck config, sorry citywok |
| 22:28.30 | citywok | areay: some questions go unanswered and unacknowledged in here. |
| 22:28.42 | areay | citywok, i know that oh too well :P |
| 22:28.48 | areay | not from today, just past experienc |
| 22:28.53 | citywok | generally people just don't know the answer to what you are asking, or you are asking something _really_ stupid. |
| 22:29.03 | areay | lol i'm guessing it's the latter |
| 22:30.56 | areay | citywok, when you tested, did you get MOH for the caller, or just a ringing tone? |
| 22:31.14 | citywok | not sure, b/c it was ringing back in to itself. lol. |
| 22:31.24 | areay | ah lol ok |
| 22:37.01 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
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| 22:56.02 | *** join/#asterisk sthon (~sean@fbx.caras.modwest.com) |
| 22:56.51 | sthon | Could anyone recommend an asterisk based hardware appliance? |
| 23:01.11 | citywok | not the digium aa50 |
| 23:05.13 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 23:07.08 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 23:07.39 | bmoraca_work | sthon: i've always just used a Supermicro 5015...great little boxes and easy to mount on a wall |
| 23:10.04 | pabelanger | How can a 9 second MixMonitor() file take 1:20mins to play back? |
| 23:11.07 | bmoraca_work | headers are fubared |
| 23:11.48 | sthon | bmoraca_work: yeah, I currently have an older 2U box but I'm looking to replace it with an appliance, one less thing we have to worry about. |
| 23:12.37 | bmoraca_work | any "appliance" you get is just going to be a computer. to my knowledge, no one's built an FPGA or ASIC with Asterisk on it. I don't even think there's an SoC with Asterisk on it yet |
| 23:12.56 | bmoraca_work | you might look at the one from Pika, but I believe it's the same thing as the aa50 |
| 23:13.14 | bmoraca_work | if you're looking for small and quiet, the Supermicro is your best bet. |
| 23:14.18 | sthon | bmoraca_work: I don't care what the hardware is, I'm looking more for something that has support. |
| 23:14.42 | sthon | most appliances in my experience are 1u supermicro's |
| 23:15.39 | bmoraca_work | do you mean hardware warranty or software support? if the former, choose anything. if the latter, there really isn't anything except pre-packaged asterisk builds (EvolutionPBX, Switchvox, Baracuda's appliance (technically freeswitch)) |
| 23:16.00 | bmoraca_work | if you've got a custom asterisk dialplan, you're not going to find something with support |
| 23:16.08 | bmoraca_work | for the asterisk component, anyway |
| 23:16.41 | citywok | yea i have the supermicro 1u, works great |
| 23:17.08 | citywok | but even a tiny little atom PC would do all the same, and take up way less space |
| 23:17.10 | sthon | bmoraca_work: software support |
| 23:17.12 | citywok | supermicro 1u + SSD |
| 23:17.25 | sthon | ssd seems a little over kill |
| 23:17.37 | sthon | even a CF card would suffice |
| 23:17.38 | citywok | no spinning disk = less likely to fail. |
| 23:17.58 | citywok | yea, CF cards are annoying, sometimes they don't always work on the right hardware, etc. a $90 X25-V SSD works great. |
| 23:18.02 | bmoraca_work | sthon: if you're looking for a toaster, check out Baracuda's appliance. it's expensive, but their shit usually works well |
| 23:18.05 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 23:18.52 | tvc123 | quick question is 1.8.1 pring run in production? |
| 23:18.56 | citywok | spending an extra $50 on a HD in a pbx isn't exactly a big deal when the whole system costs 8 grand :P |
| 23:19.02 | tvc123 | s/pring/being |
| 23:19.18 | citywok | tvc123: i imagine somebody somewhere probably uses it. also you need the trailing / or infobot won't do it. |
| 23:19.43 | citywok | i'm using it to handle my inbound phone number which just points to my cellphone, that's about it. my prod server is still 1.6.2.11 |
| 23:20.00 | tvc123 | citywok: so if I am looking at moving my setup to a production enviroment I should really be using 1.6 for now? |
| 23:20.13 | *** join/#asterisk ickmund (~ickmund@c-dd4be755.015-144-70697410.cust.bredbandsbolaget.se) |
| 23:20.18 | citywok | use whatever works for you. is there any reason you don't want to use 1.8? |
| 23:21.03 | citywok | I haven't gone to 1.8 in my production environment b/c it's a lot of testing to do before I can actually upgrade my entire call center. |
| 23:21.34 | citywok | that's the same reason i'm still on .11 and not the newest .15. too much testing, and no need as of yet. |
| 23:22.08 | tvc123 | citywok: just worries me Im setting it up for the first time. I just had an instance in testing where the asterisk process was hunk |
| 23:22.15 | tvc123 | s/hunk/hung/ |
| 23:22.25 | tvc123 | and had to do a kill -9 on it |
| 23:22.41 | citywok | see if you can reproduce the problem |
| 23:22.53 | citywok | if it's reproduceable file a bug report |
| 23:22.56 | tvc123 | although it could be a problem with my config ... I'm just wondering if 1.8 is too new to be playing with |
| 23:23.14 | citywok | if it was too new it wouldn't be released ;) |
| 23:23.19 | WIMPy | That didn't happen less on 1.6 for me. |
| 23:23.39 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 23:23.42 | citywok | yea every now and then i get a weird lock that hangs my system for 30-90 seconds. |
| 23:23.49 | citywok | fortunately it's been a couple months now |
| 23:24.21 | tvc123 | citywok: no this was a perminant lock ... it was really odd I could connect with asterisk -r but no sip phones could connect to the server |
| 23:24.30 | tvc123 | so maybe the sip process was just hung |
| 23:24.52 | tvc123 | but the fix was to do a kill -9 on the asterisk process and then restart asterisk |
| 23:26.06 | citywok | whoops |
| 23:27.28 | tvc123 | yeah ... just worries me .. but like I said I am fairly new to asterisk |
| 23:27.33 | tvc123 | so it could have been a config issue |
| 23:27.57 | tvc123 | although seems like and odd behavior for a simple config issue |
| 23:35.11 | tvc123 | how stable is the trunk? |
| 23:35.30 | tvc123 | am I asking to shoot myslef in the foot? |
| 23:35.53 | WIMPy | Works just as good as any other version so far. |
| 23:36.14 | tvc123 | I assume running off the 1.8 branch would be safer than the trunk |
| 23:39.09 | citywok | WIMPy: except for when russelb decides to break something |
| 23:39.32 | citywok | personally i wouldn't run trunk in production, it's not guaranteed that nothing will go wrong haha. |
| 23:39.51 | citywok | i do run the trunk version of meetme/confbridge b/c i added new features to it that i want. lol |
| 23:39.56 | tvc123 | citywok: but would you run the svn branch in prod? |
| 23:40.22 | WIMPy | I'd always test it, no matter where it comes from. |
| 23:40.29 | citywok | do you mean trunk, or do you mean grabbing 1.6.2.15 from svn? b/c that'd be fine. |
| 23:40.32 | tvc123 | yeah agreed there |
| 23:40.45 | tvc123 | http://svn.asterisk.org/svn/asterisk/branches/1.8/ |
| 23:40.45 | citywok | grabbing a release versus just grabbing trunk is different |
| 23:40.50 | tvc123 | is what I'm looking at grabbing |
| 23:40.53 | citywok | yea, there's nothing wrong with doing that |
| 23:41.05 | citywok | but you could just grab the tarball :P |
| 23:41.15 | tvc123 | just seeems like it would be easier to incorperate bug fixes that way |
| 23:41.22 | citywok | not really |
| 23:41.50 | citywok | i download the tarball, create a folder for each release with all the software packages i used for that one set, and then when i upgrade i create a new folder, download the new build, etc. |
| 23:42.11 | citywok | that way i can easily roll back to a prior known working version if something goes wrong. |
| 23:42.26 | tvc123 | I suppose that makes sense |
| 23:43.25 | citywok | it just makes it easier to keep my builds seperate and protect myself against regression. i KNOW the /usr/src/1.6.2.11 folder works, i just have to go in and make install it if something goes wrong in /1.6.2.15 |
| 23:44.05 | tvc123 | thanks again for the advice |
| 23:44.13 | tvc123 | I've got to leave but I'm sure I'll be back |
| 23:44.44 | citywok | later |
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| 23:53.55 | wizard171 | Why would the "STREAM FILE" via AGI(xxx.php) work ... and via AGI(agi://xxx) ... not? |
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