| 00:20.10 | Micc | http://pastebin.com/G0TaKchX |
| 00:20.32 | Micc | I wonder if the problem is because I'm parking a call that was made from my phone instead of a call that came into it. |
| 00:23.19 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 00:23.34 | Micc | Its also playing the parking spot number to the parked call instead of to my phone that parked the call. |
| 00:26.01 | *** join/#asterisk troyt (~troyt@c-24-10-222-127.hsd1.ut.comcast.net) |
| 00:26.20 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 00:27.19 | Micc | I added the peer config from sip.conf and features.conf to the pastebin |
| 00:37.18 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:37.30 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:41.25 | *** join/#asterisk jploh (~jploh@121.58.248.162) |
| 00:42.28 | jploh | i'm trying to connect a PSTN phone line to asterisk. i bought an x100m module for a tdm410p |
| 00:42.51 | jploh | dahdi-channels.conf says signalling=fxs_ks, is that correct? |
| 00:43.45 | jploh | dahdi_scan says port=1,FXO (until port=4) though |
| 00:53.04 | p3nguin | micc: Is this some kind of crap FreePBX created for you? |
| 00:54.06 | p3nguin | signalling is actually spelled signaling. I don't know if that's relevant, though. |
| 01:00.09 | jploh | that's what's spelled by dahdi_genconf |
| 01:00.15 | jploh | i'll try it with a single L |
| 01:01.01 | p3nguin | If that's the way it is, someone made an error. |
| 01:01.59 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 01:02.15 | jploh | that was from dahdi 2.5.0.1, changed the configuration by dahdi_genconf but channels are still not detected |
| 01:02.38 | jploh | dahdi show channels only returns pseudo |
| 01:05.48 | jploh | okay i had it working now |
| 01:06.09 | jploh | but i manually pasted the contents of dahdi-channels.conf to chan_dahdi.conf |
| 01:07.00 | jploh | include => chan_dahdi.conf must be wrong |
| 01:08.09 | *** part/#asterisk LostyJai (~blah@202.171.190.130) |
| 01:08.20 | jploh | d'oh! it should be #include dahdi-channels.conf |
| 01:08.22 | jploh | thanks! |
| 01:10.38 | *** join/#asterisk sixohquad (~sixoh@184.65.142.249) |
| 01:13.11 | *** join/#asterisk mindCrime (~chatzilla@cpe-076-182-089-009.nc.res.rr.com) |
| 01:14.55 | *** join/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 01:15.27 | asteriskmonkey | is it normal for asterisk to take up nearl 300megs in memory on a quite system? |
| 01:17.06 | KavanS | no |
| 01:17.15 | KavanS | not in my limited experience... |
| 01:23.12 | *** join/#asterisk SupYoshi (SupYoshi@ip51cc8577.speed.planet.nl) |
| 01:23.32 | SupYoshi | Hi can some suggest me a Dutch language voice for asterisk |
| 01:38.31 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 01:45.10 | *** join/#asterisk dj_hamsta (~werwer@unaffiliated/dj-hamsta/x-2342346) |
| 01:46.09 | dj_hamsta | i set up two extentions. I set one up outside of my network and forwarded the proper ports (5060). I am unable to make calls, what else would i need to set up? |
| 02:03.19 | p3nguin | dj_hamsta: Phones. You must have phones. Got any phones? |
| 02:07.45 | Micc | p3nguin, that is stock 1.8.8.1 vanilla asterisk. |
| 02:09.51 | p3nguin | I've seen someone else talking about it playing the parking lot to the parked leg... it has something to do with the phone doing the parking being hung up before the announcement plays. |
| 02:10.50 | Micc | p3nguin, thats a good place to look. |
| 02:10.52 | p3nguin | As for calling back, that seems to be working correctly. Parking returns the call to the PEER which parked it. It was parked by the peer used for the ITSP, so it tries to call back to the ITSP's peer. Completely correct and normal. |
| 02:11.28 | Micc | The peer that parked the call was the phone though. |
| 02:11.35 | p3nguin | What phone? |
| 02:11.56 | Micc | I guess I left that part out. |
| 02:13.35 | p3nguin | What I saw in the debug was a call from an a specific peer was being parked by a phone using the same peer. |
| 02:14.01 | Micc | heres the full call http://pastebin.com/EK1EpxS3 |
| 02:14.28 | Micc | nwdmic1 is the peer that parked the call. |
| 02:14.58 | p3nguin | What is it? |
| 02:15.18 | p3nguin | What kind of device? |
| 02:15.18 | Micc | Yealink T38G |
| 02:15.32 | Micc | I've got a few other phones here I can try. |
| 02:16.37 | p3nguin | Who created this dial plan? |
| 02:16.47 | Micc | me |
| 02:17.36 | p3nguin | What is the purpose of the added overhead? |
| 02:17.46 | dj_hamsta | p3nguin, lol yes.. phones have been configured, i fear it may be the router on the other side. will check that first before asking again |
| 02:18.23 | p3nguin | dj_hamsta: Do the phones register successfully? |
| 02:19.04 | Micc | what added overhead? It sets callerid because we use different callerid for internal. It makes sure they aren't using too many lines, then checks the dialout route in the database. |
| 02:19.21 | dj_hamsta | p3nguin, yes |
| 02:19.31 | Micc | seems like the only things that need to be there to make it work correctly. |
| 02:19.39 | p3nguin | I'm talking about the added overhead of this macro. All those things can be done without a macro. |
| 02:20.12 | p3nguin | That was what made it appear to me as being a FreePBX problem. |
| 02:20.24 | p3nguin | Most sane people wouldn't use a macro in a case like that. |
| 02:21.14 | Micc | ok, I could have used a gosub instead or just a regular context. |
| 02:21.18 | p3nguin | My concern with the macro was that the call was going to be returned to extension 's' in whatever context it said. |
| 02:21.43 | p3nguin | All of those things can be done in the existing extension. No need for a subroutine of any kind. |
| 02:21.51 | Micc | It says it will return to nwd-sip, which is the correct context for the phone. |
| 02:22.06 | p3nguin | What extension does it say it will call back? |
| 02:22.19 | p3nguin | What I read said extension s priority 1. |
| 02:22.34 | Micc | Will timeout back to extension [nwd-sip] s, 1 in 200 seconds |
| 02:22.43 | Micc | oh, which doesn't exist. |
| 02:22.49 | p3nguin | And there's no phone at that extension. |
| 02:23.00 | p3nguin | Your macro has it all messed up. |
| 02:23.08 | p3nguin | The unnecessary macro, that is. |
| 02:23.35 | Micc | I'd be happy to learn how to do those things without this macro. |
| 02:24.03 | p3nguin | So when the thing goes to s, it reads the channel information into the macro and calls back the existing channel rather than the one where it really should be going. |
| 02:25.01 | Micc | it would be NPANXXXXXX extension in most normal cases, but I don't see hwo that would solve the problem. |
| 02:25.18 | p3nguin | It's the macro that is the problem. |
| 02:25.53 | p3nguin | When the call returns TO the macro, the macro's apps/functions take the wrong channel information and process a new call. |
| 02:26.15 | Micc | I'll hard code it and see if that fixes it then. |
| 02:26.25 | p3nguin | At least that was my interpretation of what I read in your first pastebin. |
| 02:27.03 | p3nguin | Lose the macro or you won't be gaining anything. |
| 02:28.09 | *** join/#asterisk emhs (~emhs@204-16-153-102-static.ipnetworksinc.net) |
| 02:28.28 | p3nguin | A basic extension is capable of setting CALLERID(num), checking GROUP_COUNT(), and looking up a peer entry from a database. |
| 02:29.32 | Micc | sure, but its a lot more code. I was trying to simplify things by putting it in a macro. |
| 02:29.46 | Micc | I'd rather put it in a sub now, but I haven't gotten around to that. |
| 02:30.04 | p3nguin | How is it more code? Move the subroutine out of the macro and into the main routine. |
| 02:30.26 | p3nguin | It's the exact same amount of code. |
| 02:30.46 | p3nguin | Actually less... |
| 02:30.58 | p3nguin | one less line to execute the macro, and one less for the macro context name. |
| 02:31.12 | p3nguin | All the rest will be exactly the same. |
| 02:31.32 | emhs | Howdy folks. I'm a devout Google Voice user, but I'm in the process of brainstorming a transition to secure space. I'm hoping y'all might be able to help with brainstorming a way to use asterisk to replace GV's ability to route from a central number to either my cell or a geographically static sip-phone as needed, centrally receive and forward text messages, and record calls and voicemails, while that central number is displayed as C |
| 02:31.32 | emhs | ID for all outbound calls from either my cell or static sip-phone. |
| 02:31.41 | Micc | I call that macro from a few different places. Now I'll have that code in all those places. |
| 02:31.42 | emhs | Anyone have any thoughts on such a concept? |
| 02:32.05 | p3nguin | Why do you need to put it in more than one place? Variables are variable for a reason. |
| 02:32.06 | Micc | I can set some channel variables and just use goto. |
| 02:32.40 | Micc | so what is the point of a macro then? |
| 02:32.43 | p3nguin | I know you don't want to change your ways, but it is because of your way that the thing failed. |
| 02:33.00 | p3nguin | Macros have been deprecated, anyway. |
| 02:33.19 | emhs | ponders. |
| 02:33.30 | Micc | I'm fine changing my ways but I'd like to change it to something clean. Can I use a Gosub then? |
| 02:33.43 | p3nguin | Subroutines are often useful for things... but I haven't seen anything warranting a subroutine in your case. |
| 02:33.56 | p3nguin | YOu don't need a subroutine at all. |
| 02:34.27 | resist0r | has anyone compiled iaxclient? |
| 02:34.45 | Micc | Its mostly for managability. I have each customer's dial code all in one place and I can call it from many other places. |
| 02:35.02 | p3nguin | You could do that without a subroutine. |
| 02:35.15 | Micc | If they want to have a special prefix like 9 that dials out a different number or something, its easy, I just call it with a different parameter. |
| 02:35.18 | *** join/#asterisk kactusotp (~chatzilla@203.59.226.41) |
| 02:35.49 | Micc | Goto doesn't take parameters, so you have to do a bunch of set's before you goto |
| 02:35.51 | p3nguin | If they want to dial a 9 before a phone number, I'd tell them that 1986 called and they want their phone system back. |
| 02:36.07 | resist0r | heh |
| 02:36.49 | Micc | p3nguin, true thats not the default, but its nice in cases where they have multiple numbers and they want to choose which caller id number is used. |
| 02:37.18 | p3nguin | All of that is still easily performed without sub routines. |
| 02:37.24 | kactusotp | Hi Everyone, just wondering if there was anyone on hand to help with a major issue I'm seeing in asterisk 1.8? "Too much delay in IAX2 calltoken timestamp from address XX.XX.XX.XX" |
| 02:37.46 | p3nguin | kactusotp: Check your system time on both peers. |
| 02:39.43 | kactusotp | that can do it? I'm pretty sure ntp is setup on both but I'll double check. Seems to be intermittent happens after anywhere from 6 - 24 hours of calls. |
| 02:40.37 | p3nguin | I don't know that the time *is* the problem, but the fact of a timestamp delay made me think of a system time sync issue. |
| 02:43.13 | p3nguin | I recommend three ntp peers to keep accurate time. |
| 02:43.35 | emhs | Anyone know any good, basic, free sip services compatible with an asterisk setup? |
| 02:43.39 | Micc | I use all kinds of subs and macros for inbound call routing too. Does that mean it will break parking for inbound calls too? |
| 02:43.50 | resist0r | p3nguin: what peers do you recomend |
| 02:43.51 | resist0r | ? |
| 02:43.57 | resist0r | ntp |
| 02:44.26 | p3nguin | resist0r: It is dependent on your location. I usually use geographically relevant peers from pool.ntp.org. |
| 02:44.37 | resist0r | gotcha |
| 02:44.55 | p3nguin | such as {0,1,2}.us.pool.ntp.org |
| 02:47.09 | resist0r | okay cool, I was having a discussion the other day with someone about use'n more than one ntp server. Thanks |
| 02:47.48 | kactusotp | p3nguin: the distressing thing about this is that when it happens it drops all the iax peers (unreachable), and still processes sip but fails on iax. Have to restart asterisk to solve :/ time seems more or less a match though |
| 02:47.50 | p3nguin | Depending on your ntpd, you might be able to specify a single "servers" line and it will use multiple servers. |
| 02:48.06 | p3nguin | servers us.pool.ntp.org |
| 02:48.24 | p3nguin | As opposed to server 0.us.pool.ntp.org |
| 02:51.48 | p3nguin | If you aren't in the US, check http://www.pool.ntp.org/zone/@ for other geographical pools. |
| 02:51.59 | emhs | Okay. Even simpler question: Does anyone know a good SIP dialout service? |
| 02:52.11 | emhs | For reaching POTS from within sip-land? |
| 02:52.19 | p3nguin | You want an ITSP. |
| 02:52.19 | p3nguin | ~itsp |
| 02:52.19 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 02:52.22 | Micc | p3nguin, if I can't even use a goto then how am I supposed to make this work without copying my code to 4 different places in the dialplan? |
| 02:52.47 | p3nguin | Use variables in a single routine. |
| 02:53.03 | Micc | I would have to match on everything then _. |
| 02:53.09 | Micc | that doesn't seem secure. |
| 02:53.19 | p3nguin | That is a TERRIBLE pattern. |
| 02:53.49 | Micc | I have different logic for if they dial 1 or use an npa prefix or not. |
| 02:54.00 | kactusotp | _X. is better, sorry Micc i missed what you were doing but would a gotoif not work? |
| 02:54.25 | Micc | kactusotp, apparently not even a goto will work with parking in 1.8. |
| 02:54.55 | Micc | I've tried them all. onlything that works is keeping it all in the device's sip context. |
| 02:54.59 | emhs | ~itsplist-us |
| 02:55.00 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 02:55.00 | emhs | Yep, fail. |
| 02:55.00 | emhs | p3nguin: Can't see the list. Any ideas why? |
| 02:55.01 | emhs | p3nguin: Am I not authorized? |
| 02:55.22 | *** join/#asterisk emhs (~emhs@204-16-153-102-static.ipnetworksinc.net) |
| 02:55.25 | emhs | ~itsplist-us |
| 02:55.26 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 02:56.16 | kactusotp | Micc: have not played with parking in 1.8 sorry but push comes to shove if you want to do alot of processing but keep your patterns simple there is func_odbc. We do most of our heavy lifting there |
| 02:56.38 | Micc | kactusotp, yes I use that a lot too. |
| 02:57.38 | Micc | I just don't want to have 10 lines of dialplan duplicated 3 times for every single sip device context. |
| 02:58.05 | p3nguin | Each device does not need to have its own context. |
| 02:58.30 | Micc | p3nguin, each customer has a context for its devices. |
| 02:58.44 | p3nguin | Okay, then use a gosub. |
| 02:58.51 | p3nguin | Now you're making more sense to me. |
| 02:58.55 | Micc | it doesn't work either. I already tried gosub and goto. |
| 02:59.01 | p3nguin | I'm glad we finally arrived here. |
| 02:59.03 | Micc | and I moved the extension from s to the extension. |
| 03:00.49 | p3nguin | It is possible for a routine to start, run a Gosub(), do stuff common to all companies, Return() back to the main routine, then Park() the call, and when it times out it should go back to the correct place (if the other parts of the dial plan are correct). |
| 03:01.29 | Micc | p3nguin, thats not a bad idea at all. |
| 03:01.42 | Micc | I'll see if that works. |
| 03:02.30 | p3nguin | Just pay attention to what your verbose output says for where the call will go when it times out. |
| 03:02.51 | Micc | p3nguin, its still wrong, but its working. |
| 03:03.03 | Micc | the output says my default context s,1 |
| 03:03.22 | emhs | So, can asterisk take a call from an ITSP and route it to a modem (which in turn is connected to a basic telephone)? |
| 03:03.24 | Micc | but it goes back to park-dial with the peer |
| 03:03.28 | p3nguin | That should be the standard fall back in the event the other does not exist. |
| 03:03.42 | p3nguin | emhs: Modem? No, this is VoIP. |
| 03:03.46 | emhs | ... |
| 03:03.57 | p3nguin | You want an ATA. |
| 03:03.58 | p3nguin | ~ata |
| 03:03.59 | infobot | methinks ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 03:04.05 | Micc | it was using the parkee's peer before. |
| 03:04.19 | p3nguin | because of the macro. |
| 03:04.21 | Micc | so it would go back to park-dial with SIP_360-inbound |
| 03:05.05 | Micc | I still don't understand why that would matter if its just looking at the peer. but it doesn't seem to work when its in any other context than the sip peer's context. |
| 03:05.29 | emhs | p3nguin: And if I want have a VoIP number that rings both my cell and the ATA-connected device and connects the call to whichever I answer? |
| 03:05.31 | p3nguin | The macro worked correctly, taking the values of the call... which were inaccurate. |
| 03:06.10 | p3nguin | emhs: Yes, you can Dial() multiple devices at one time. |
| 03:06.11 | emhs | s/want/want to/ |
| 03:06.46 | Micc | what values of the call were inaccurate? |
| 03:07.19 | p3nguin | The data that the macro read was the data from the parked leg. That wasn't what you wanted it to read. |
| 03:07.22 | p3nguin | The parked leg ran the macro. |
| 03:07.47 | Micc | the macro is run way before the park |
| 03:08.01 | p3nguin | But when the park tried to return the call, it ran the macro again. |
| 03:08.04 | emhs | Right... So, can asterisk see the ATA device and both route calls to it and route its attempted call-outs to the ITSP? |
| 03:08.29 | Micc | p3nguin, no it didn't. |
| 03:08.50 | p3nguin | emhs: I'm having trouble with your sentence... but asterisk can Dial() multiple devices at one time. |
| 03:08.51 | Micc | p3nguin, when the park returned it returned to park-dial with the wrong peer name. |
| 03:09.38 | emhs | p3nguin: I'll make an infographic that explains what I'm trying to do. If I did that would you mind telling me what devices and services are necessary to make it happen? |
| 03:10.03 | p3nguin | emhs: I don't need a picture. I understand that you want to call an ATA and a cell phone. |
| 03:10.10 | Micc | p3nguin, the value of the peer to return to is wrong at the time the park happens. And it ends up playing the park spot digits to the call that is being parked. |
| 03:10.20 | p3nguin | And for the third time, asterisk can Dial() multiple devices at the same time. |
| 03:11.36 | Micc | p3nguing, I agree that the macro was causing a problem, but not for the reason you think. |
| 03:11.47 | Micc | I just want to understand what is really happening. |
| 03:12.09 | p3nguin | What I saw said it was running the macro on returning the call to the phone who parked it. |
| 03:12.10 | emhs | ... That part's easy enough. The real question is can the ATA and the cell phone, via asterisk (but at different times), dial out through the same ITSP number (the one whose received calls are being routed to the ATA and cell). |
| 03:12.21 | Micc | p3nguin, no that never happened. |
| 03:12.50 | emhs | And can asterisk arbitrarily (at a keypress) record those calls, both in from the ITSP and out to it? |
| 03:13.10 | emhs | s/keypress/keypress on either phone/ |
| 03:13.13 | p3nguin | Yes it can do that. That is called automon. |
| 03:13.34 | p3nguin | How will you interface your cell phone with asterisk? |
| 03:14.00 | emhs | There-in lies the question. I'm kinda making this up as I go along. |
| 03:14.20 | emhs | Basically I'm trying to replace google voice service. |
| 03:14.44 | p3nguin | I don't directly interface my cell with asterisk... I simply dial into my asterisk system and then I can dial back out as needed. |
| 03:14.59 | emhs | That's pretty much what I was thinking. |
| 03:15.13 | emhs | Added bonus: put it in your friends and family list to get free calls |
| 03:15.13 | p3nguin | But there surely is something to do direct interface. Maybe something like chan_mobile. |
| 03:15.21 | p3nguin | exactly |
| 03:15.23 | p3nguin | yes |
| 03:15.45 | p3nguin | But you'll pay the rate of the ITSP for calls you make through them going out to the PSTN. |
| 03:16.03 | emhs | Well, voicepulse connect has an unlimited-america plan. |
| 03:16.07 | emhs | Covers canada too |
| 03:16.13 | emhs | So that's trivial. |
| 03:16.27 | p3nguin | For termination service? |
| 03:16.40 | emhs | http://www.voicepulse.com/residential-plans-pricing.aspx |
| 03:16.42 | p3nguin | Unlimited is often only for origination, and termination will be per minute. |
| 03:17.26 | emhs | Looks all-inclusive to me. |
| 03:17.32 | p3nguin | okay |
| 03:18.24 | emhs | So the last question is this: Can ITSPs typically handle making two calls at once? |
| 03:18.32 | p3nguin | yes |
| 03:18.36 | emhs | IE, how many lines do I need to buy? |
| 03:19.08 | p3nguin | Well, there are no lines, because it is VoIP... but you can ask them how many channels you get. |
| 03:19.36 | p3nguin | On my per minute plan, I have some 25 outbound channels, which will support 25 calls at one time. |
| 03:20.10 | p3nguin | On my inbound flat rate plan, I get two channels, which will support only two simultaneous inbound calls. |
| 03:20.24 | emhs | Channels is the term for how many connections I can have going at once for that line/number? |
| 03:20.36 | *** join/#asterisk emhs (~emhs@204-16-153-102-static.ipnetworksinc.net) |
| 03:20.39 | emhs | Channels is the term for how many connections I can have going at once for that line/number? |
| 03:20.53 | p3nguin | For each call, there is a channel to carry the call. |
| 03:21.10 | p3nguin | Like a channel on your TV. |
| 03:21.17 | emhs | Thought so. |
| 03:21.19 | emhs | Right |
| 03:21.36 | emhs | So I just need to see how many channels the plan comes with and if it's >=2, I'm golden. |
| 03:21.52 | p3nguin | So if they provide you with 5 channels, that will support 5 calls at a time. |
| 03:22.07 | p3nguin | Companies will also sell you additional channels. |
| 03:22.18 | Micc | http://pastebin.com/ggjzvrf1 I made some notes in this one. The macro is not being executed when it returns. The macro is only to dial the phone we want to park, which is my cell phone in this case. |
| 03:22.22 | emhs | p3nguin: Thank you so much. This is exceptionally helpful. |
| 03:22.59 | emhs | When I get the devices and services ready to config, I'll probably be back for more. |
| 03:23.50 | emhs | Last question: If I expose the right ports at the physical location of the ATA(s), can asterisk route at them remotely? Like can a server in the middle of nowhere receive the ITSP call and point it at an ATA in my living room? |
| 03:24.19 | p3nguin | yes |
| 03:24.30 | emhs | Shiny. |
| 03:24.35 | emhs | Y'all take care now. |
| 03:25.37 | Micc | p3nguin, in that one its not even a macro its just a goto. |
| 03:28.11 | Micc | p3nguin, call flow here is. I dial my cell phone from the peer nwdmic1, it sends call out 360, I answer on my cell phone. Then I place the call on park. You can see right at the time I park it is placing the correct call on park, but it is trying to play the park spot to the parked call. So when it returns its not rerunning anything, its just trying to dial the wrong peer. |
| 03:29.46 | Micc | p3nguin, have you given up on me? |
| 03:30.03 | p3nguin | Not exactly... I'm trying to eat supper. |
| 03:30.41 | Micc | no problem. :) |
| 03:31.09 | p3nguin | corned beef, potatoes, and carrots |
| 03:31.41 | Micc | oh nice, thats my favorite. |
| 03:32.07 | Micc | with a little sourcrout |
| 03:32.34 | Micc | I'm gonna try your suggestion, gosub then return. |
| 03:32.36 | *** join/#asterisk corretico (~luis@190.211.93.11) |
| 03:36.42 | Micc | that works. So it has to be in the context of the sip peer when it dials. I don't understand why. I haven't looked at the parkinglot code in 1.8 much yet though. But it never used to matter. |
| 03:37.43 | Micc | maybe its using the call stack to store where to return the parked call to. |
| 03:38.34 | SeRi | p3nguin: I am porting a did to voip.ms for one of my clients.... I will let you know how that goes as a reseller.... maybe more bugs or maybe is not possible.... |
| 03:39.08 | p3nguin | That part is easy. |
| 03:39.35 | p3nguin | Take their last invoice, have them sign it, email/fax it to voip.ms and wait. |
| 03:40.15 | SeRi | o sool! |
| 03:40.23 | SeRi | s/sool/cool/ |
| 03:40.42 | p3nguin | Did you already start the port request? |
| 03:41.10 | *** join/#asterisk radic (~radic@dslb-094-216-234-162.pools.arcor-ip.net) |
| 03:42.48 | SeRi | p3nguin: No. |
| 03:42.53 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 03:42.58 | *** join/#asterisk Cubber (~ronny@cpe-24-58-133-224.twcny.res.rr.com) |
| 03:43.08 | p3nguin | It's fairly painless. |
| 03:43.23 | SeRi | I open up a ticket to see what I need it but you answer my question! lol :) |
| 03:43.33 | p3nguin | It's in the wiki. |
| 03:43.48 | Cubber | I compiled asterisk 10 from source and am working on getting it setup with Google Voice. I have outbound calling working fine, however inbound calls fail with the following error: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 03:44.23 | Cubber | after that displays in the CLI i get a busy signal on the phone I am calling from, it never rings the asterisk extension that is mapped to the gvoice account for incoming calls |
| 03:44.32 | p3nguin | It looks like your SIP phone is not available. |
| 03:45.20 | Cubber | p3nguin odd I can use it to access voice mail and make outbound calls. it is a polycom ip335 gonna try to restart it |
| 03:46.09 | p3nguin | seri: http://wiki.voip.ms/article/Porting_a_Number |
| 03:46.22 | SeRi | p3nguin: Thanks! |
| 03:46.51 | Cubber | p3nguin phone restart fixed it! |
| 03:46.59 | p3nguin | easy |
| 03:47.20 | Cubber | what is this warning I get whenever an extension connects? db.c:295 ast_db_put: Couldn't execute statment: SQL logic error or missing database |
| 03:47.32 | Cubber | is it because I do not have mysql support in use? |
| 03:47.34 | p3nguin | Extensions do not connect. |
| 03:47.40 | p3nguin | Extensions are your dial plan rules. |
| 03:48.01 | Cubber | let me rephrase, whenever a sip device registers |
| 03:48.11 | p3nguin | Are you using real time SIP peers? |
| 03:48.26 | Cubber | I have not set any up |
| 03:48.36 | p3nguin | Your phones are configured in sip.conf? |
| 03:48.47 | Cubber | yes |
| 03:49.57 | p3nguin | If that is showing up when your phone registers, maybe you need to disable your realtime SIP stuff. |
| 03:50.22 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 03:51.19 | Cubber | i never set any up. I just installed asterisk with the required gvoice modules, and setup a basic sip.conf and extensions.conf to get things going. Was going to do more after I got it running correctly |
| 03:51.20 | *** join/#asterisk vinhdizzo (~vinh@pool-74-100-182-10.lsanca.fios.verizon.net) |
| 03:51.48 | p3nguin | What other confs do you have in /etc/asterisk/ ? |
| 03:52.41 | Cubber | http://pastebin.com/2pH1c25G |
| 03:52.58 | Cubber | I have only edited sip.conf, extensions.conf, gtalk.conf, jabber.conf and voicemail.conf |
| 03:53.23 | p3nguin | You see all those confs? |
| 03:53.30 | Cubber | yes |
| 03:53.31 | p3nguin | If you aren't using them, get rid of them. |
| 03:53.45 | Cubber | they are just the samples |
| 03:54.10 | p3nguin | The samples are NOT FOR PRODUCTION files. |
| 03:54.20 | p3nguin | You'll need a few for minimal operation, but many of them need to go. |
| 03:54.37 | Cubber | ok so is there a list of the ones that are required somewhere? |
| 03:54.53 | *** join/#asterisk sixohquad (~sixoh@184.65.142.249) |
| 03:55.02 | Cubber | I have the oreilly book Asterisk the definitive guide. Just got it, and need to start digging in. |
| 03:56.20 | p3nguin | I would keep: asterisk.conf, modules.conf, sip.conf, extensions.conf, indications.conf, voicemail.conf, codecs.conf, features.conf, rtp.conf, queues.conf |
| 03:56.43 | p3nguin | Only create and configure files that you need. |
| 03:56.59 | Cubber | thanks for the tip |
| 03:59.14 | sixohquad | hey guys, am i able to include other .conf files in sip.conf and extensions.conf and if so, how do i do it? |
| 03:59.30 | sixohquad | i think i asked this earlier but i was working on the tablet and missed the answer |
| 03:59.37 | p3nguin | #Include other_file.conf |
| 03:59.43 | sixohquad | great |
| 03:59.44 | sixohquad | lol |
| 03:59.49 | sixohquad | thanks. |
| 04:00.29 | sixohquad | im reading the section on includes in the book and it says nothing about including outside files, just other contexts. maybe it talks about it later in the book |
| 04:02.39 | p3nguin | Are you reading it online? |
| 04:02.46 | sixohquad | no |
| 04:02.52 | sixohquad | is it available online? |
| 04:02.55 | p3nguin | Yes. |
| 04:02.56 | sixohquad | im reading it in ebook |
| 04:02.58 | p3nguin | ~book |
| 04:02.59 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 04:03.07 | sixohquad | oh shite |
| 04:03.38 | sixohquad | well, then i feel less bad about downloading torrent. excellent. |
| 04:03.45 | p3nguin | woleium: I wish you'd knock it off. |
| 04:03.49 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 04:05.25 | Cubber | p3nguin seems to be working with the minimal confs thanks again. |
| 04:05.49 | p3nguin | Add more as needed. |
| 04:06.21 | Cubber | I just moved the old /etc/asterisk folder to a backup folder and copied in the ones I needed to /etc/asterisk |
| 04:06.43 | p3nguin | Good plan. |
| 04:06.57 | p3nguin | The sample files are for reference. |
| 04:07.22 | p3nguin | Some of the samples, however, are suitable to use because they have bare minimum configuration in them. |
| 04:07.39 | *** join/#asterisk blognewb (~blognewb_@unaffiliated/blognewb) |
| 04:08.14 | blognewb | hey guys im new to obi |
| 04:09.26 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 04:09.50 | p3nguin | Obi-Wan Kenobi? |
| 04:10.10 | [TK]D-Fender | You're our only hope... |
| 04:10.18 | woleium | p3nguin: what am i doing? |
| 04:10.22 | sixohquad | p3nguin, and [TK]D-Fender are asterisk jedi |
| 04:10.50 | blognewb | p3nguin obi110 |
| 04:10.55 | p3nguin | woleium: All the goddamned nick changing is really annoying. You're the only one in the channel doing it, so you kind of stand out like a sore thumb. |
| 04:10.58 | blognewb | i don't get what a sip gateway is |
| 04:11.14 | blognewb | do you guys have a layman explanation for it? |
| 04:11.32 | [TK]D-Fender | Gateway = going from one kind of thing to another |
| 04:11.35 | p3nguin | On, obihai. |
| 04:11.40 | blognewb | i was watching this http://www.youtube.com/watch?v=k5BwQ0E6fmQ then i got lost halfway :( |
| 04:11.41 | blognewb | yes sir |
| 04:11.58 | [TK]D-Fender | SIP is one telephony statndard. In the case of the OBI its either FXO or FXS |
| 04:12.07 | blognewb | [TK]D-Fender ] how is a gateway different from a router? |
| 04:12.17 | [TK]D-Fender | So that lets you tun an analog phone into a SIP phone |
| 04:12.26 | p3nguin | A gateway could be a router with a specific purpose. |
| 04:12.35 | [TK]D-Fender | A router just moves packets. Analog phone != VOIP |
| 04:12.40 | blognewb | [TK]D-Fender ] what is a sip though? :((( |
| 04:12.50 | p3nguin | But a gateway can be other types of devices with a similar purpose. |
| 04:12.51 | woleium | aaah, sorry p3nguin - I installed znc the other day. still tweaking settings |
| 04:12.51 | [TK]D-Fender | ~sip |
| 04:12.51 | infobot | well, sip is Session Initiation Protocol, http://www.cs.columbia.edu/sip/ (see RFC 3261) It's HIP to be SIP! |
| 04:12.56 | [TK]D-Fender | blognewb: ^ |
| 04:12.58 | woleium | wanted off to change his settings |
| 04:13.00 | sixohquad | sip is a protocol |
| 04:13.00 | blognewb | p3nguin so strictly speaking are all routers gateways technically? |
| 04:13.05 | [TK]D-Fender | SIP is a VoIP protocol |
| 04:13.07 | p3nguin | no |
| 04:13.16 | blognewb | oh its a voip protocol |
| 04:13.22 | p3nguin | But a router with the purpose of being a gateway is a gateway. |
| 04:13.42 | blognewb | p3nguin how else are routers classified as other than being a gateway? |
| 04:13.46 | p3nguin | Such as your router between your LAN and your DSL modem... that's a gateway. |
| 04:14.19 | sixohquad | a router is not a gateway if its not delivering internet to you |
| 04:14.28 | sixohquad | if its only acting as a LAN hub, its not a gateway |
| 04:14.33 | p3nguin | I guess in a loose sense, a router between two networks of any kind could be considered a gateway, but it's not common usage of the term. |
| 04:14.36 | blognewb | LAN Hub |
| 04:14.42 | blognewb | sixohquad like intranet? |
| 04:14.46 | [TK]D-Fender | DSL modem = gateway. It traverses mediums |
| 04:14.48 | sixohquad | blognewb, yes |
| 04:14.57 | blognewb | phew it gets so confusing :'( |
| 04:15.14 | p3nguin | woleium: Thank you! Less people will hate you once you have that fixed. |
| 04:15.21 | sixohquad | a sip gateway is just something that turns a sip line into something else for you, typically an analog line...dare i say? |
| 04:15.22 | sixohquad | lol |
| 04:15.26 | p3nguin | woleium: They should have had some sane defaults, though. |
| 04:15.44 | blognewb | sixohquad so if a router acts as a LAN hub for instance but not as a gateway, what does it "route" as the name suggests? |
| 04:16.01 | p3nguin | A router is never a "LAN hub." |
| 04:16.09 | sixohquad | it still assigns dhcp addresses to each device |
| 04:16.13 | blognewb | p3nguin i thought sixohquad said it is |
| 04:16.16 | woleium | /me is sad that he has been upsetting people :-( |
| 04:16.17 | sixohquad | lan hub may have been the wrong terminaology |
| 04:16.19 | sixohquad | lol |
| 04:16.21 | p3nguin | A dhcpd does that, not a router. |
| 04:16.21 | sixohquad | i was comparing |
| 04:16.33 | sixohquad | router's have dhcpd's in them tho |
| 04:16.34 | sixohquad | lol |
| 04:16.34 | p3nguin | But a router appliance can also contain a dhcpd. |
| 04:16.49 | blognewb | what is the last d in dhcpd |
| 04:16.53 | sixohquad | daemon |
| 04:16.53 | p3nguin | A router appliance can also have a switch. |
| 04:17.00 | p3nguin | but the router just routes. |
| 04:17.04 | sixohquad | yeah |
| 04:17.28 | blognewb | but don't routers have dhcpd in them |
| 04:17.33 | blognewb | i just checked mine |
| 04:17.34 | p3nguin | Sometimes, yes. |
| 04:17.36 | [TK]D-Fender | Many do |
| 04:17.48 | p3nguin | But that doesn't make it a gateway in all cases. |
| 04:17.50 | sixohquad | most in consumer applications do |
| 04:18.08 | blognewb | so a dsl modem is also a gateway |
| 04:18.16 | sixohquad | yes |
| 04:18.24 | p3nguin | For example, I have a router with a dhdp server which is not my internet gateway. |
| 04:18.31 | blognewb | so if there's a router in between, is it defined as a gateway to the gateway |
| 04:18.54 | sixohquad | you shouldn't really do that lol |
| 04:18.56 | blognewb | in a dsl modem - router - nodes setup, are the modem and router both gateways |
| 04:19.12 | sixohquad | well not if your dsl modem has a builtin router |
| 04:19.21 | p3nguin | If your modem is a bridge, I wouldn't call it a gateway. |
| 04:19.29 | sixohquad | IF your DSL modem is a gateway, then you should be able to plug a switch into it |
| 04:19.32 | sixohquad | you shouldn't plug another router |
| 04:19.33 | blognewb | p3nguin oh wow what is a bridge this time? |
| 04:19.47 | blognewb | gateway and bridge sound synonymous |
| 04:19.52 | p3nguin | A modem in bridged mode does not behave as a node on the network. |
| 04:19.59 | blognewb | oh |
| 04:20.00 | p3nguin | A router does, though. |
| 04:20.04 | sixohquad | it just passes traffic through |
| 04:20.08 | sixohquad | a router will actually route traffic |
| 04:20.09 | p3nguin | And the router would be the gateway. |
| 04:20.12 | sixohquad | yes |
| 04:20.20 | blognewb | so p3nguin what does that imply if you set the modem in bridged mode while using a router |
| 04:20.38 | p3nguin | When a modem is not in a bridged mode, it is probably acting as a gateway. |
| 04:20.46 | sixohquad | your modem will authenticate to the outside network, and your router will take care of passing traffic from internally to externally through the modem |
| 04:21.03 | p3nguin | If your modem is in bridged mode, your connected gateway device (your router) is the first node on your premises. |
| 04:21.37 | p3nguin | If you traceroute through it, your bridged modem will not appear as a hop. |
| 04:21.51 | blognewb | so if the modem is not in bridged mode it's always a node? |
| 04:21.56 | p3nguin | Yes. |
| 04:22.02 | blognewb | this is very educational |
| 04:22.10 | p3nguin | If your modem is not bridged, it will be a hop in a traceroute. |
| 04:22.29 | p3nguin | And in that case, it is a gateway. |
| 04:22.40 | blognewb | why would one want to set the modem bridged? |
| 04:22.41 | sixohquad | yeah i guess if its bridged it probably DOESN"T do authentication, you would have to set your router to PPPOE eh? |
| 04:23.29 | sixohquad | hm, im not really sure about that concept, its been so long since i've really used ADSL |
| 04:23.44 | p3nguin | What does any of this have to do with an Obihai SIP gateway? |
| 04:23.57 | sixohquad | hah, he's just asking questions. and now i had one too :) |
| 04:24.11 | blognewb | p3nguin i was just confused since im gonna connect the obi into the router |
| 04:24.28 | blognewb | which is a gateway |
| 04:24.32 | p3nguin | Go for it... you're really connecting it to a switch in the "router" appliance. |
| 04:24.50 | blognewb | p3nguin i still have to buyh one though lol |
| 04:24.59 | p3nguin | Plastic routers usually have at least a 4-port switch built in. |
| 04:25.00 | blognewb | hoping i could find a coupon code somewhere |
| 04:25.29 | blognewb | p3nguin so this guy in the video keeps on using the term "sip account" does that pertain to getting a phone number? |
| 04:25.47 | p3nguin | Probably not. |
| 04:26.05 | p3nguin | He's probably talking about the account that Obihai will give you. |
| 04:26.09 | blognewb | i dont get the concept of sip |
| 04:26.20 | blognewb | so it's a protocol followed by a hardware...? |
| 04:26.37 | p3nguin | SIP will be the protocol used for the VoIP calls. |
| 04:26.48 | blognewb | i see |
| 04:26.59 | p3nguin | The SIP account will be the account used on their servers to provide those calls to your device. |
| 04:27.53 | p3nguin | Keep in mind that I have never used an Obihai device for Google Voice calling... because I use Asterisk and it isn't necessary. |
| 04:28.10 | blognewb | p3nguin so what would you call that thing which would provide you the phone number if not google voice |
| 04:28.27 | p3nguin | I would call it Google Voice. |
| 04:28.57 | blognewb | p3nguin are you able to use your landline phone device via asterisk |
| 04:29.07 | p3nguin | Yes. |
| 04:29.14 | p3nguin | ~ata |
| 04:29.14 | infobot | ata is, like, Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 04:29.42 | blognewb | p3nguin is there anything i need to buy to implement asterisk? is there like a for idiots/dummies guide? |
| 04:29.45 | p3nguin | An ATA allows the connection of a bell telephone to your Ethernet LAN. |
| 04:30.01 | p3nguin | The ATA turns it into SIP (in most cases). |
| 04:31.25 | blognewb | information overload. i wanna cry |
| 04:32.35 | [TK]D-Fender | ~book |
| 04:32.35 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 04:32.38 | blognewb | p3nguin is there an idiot friendly guide in the wiki page i can link my dad to |
| 04:32.38 | [TK]D-Fender | blognewb: ^ |
| 04:32.40 | blognewb | oh |
| 04:32.49 | [TK]D-Fender | blognewb: Ok, lets try this. Ever used Skype? |
| 04:32.53 | blognewb | yes sir |
| 04:33.00 | blognewb | is it easy to setup? |
| 04:33.11 | [TK]D-Fender | blognewb: that is VOIP. Skype is a PROTOCOL. It also has an APPLICATION you install to use it |
| 04:33.12 | blognewb | i mean do i need to buy any hardware or something? or everything is software |
| 04:33.15 | p3nguin | There's an entire book on most aspects of asterisk. |
| 04:33.25 | [TK]D-Fender | blognewb: Ever used SkypeOut? Or at least familiar with? |
| 04:33.45 | p3nguin | To most people, skype is an application they execute on their computer. |
| 04:34.06 | blognewb | [TK]D-Fender ive used google voice to call out |
| 04:34.13 | [TK]D-Fender | blognewb: An ATA lets you plug a boring analog phone and use it to speak that VoIP protocol. |
| 04:34.15 | blognewb | is that what skypeout essentially is |
| 04:34.21 | [TK]D-Fender | blognewb: So that is what your device is |
| 04:34.40 | [TK]D-Fender | blognewb: Just something to plug a phone into instead of using software on a PC w/ headset |
| 04:35.13 | [TK]D-Fender | and a gateway is typically the other end of the equation. A device that you plug a LINE into so that you can send those voip calls out onto the real world phone system' |
| 04:35.48 | [TK]D-Fender | Sample : you want to use your PC to place local telephone calls out your cousin's physical phone line. |
| 04:35.57 | blognewb | i really want to teach senior citizens with this thing |
| 04:36.14 | [TK]D-Fender | You install a SIP Gateway at his place. You send sip calls to it and it dials out onto the physical telco line you plug into it |
| 04:36.14 | p3nguin | What do you intend to teach? |
| 04:36.31 | blognewb | p3nguin my dad how to setup this asterisk thing |
| 04:36.46 | blognewb | hmm so i need to setup an asterisk voip server??? |
| 04:36.52 | blognewb | is that what we are all getting into |
| 04:36.53 | [TK]D-Fender | blognewb: You don't understand what it is let along the owrkings... Physician.. heal thyself :p |
| 04:37.04 | blognewb | i know! |
| 04:37.07 | blognewb | lol |
| 04:37.14 | [TK]D-Fender | blognewb: Don't get aahead of yourself. |
| 04:37.20 | p3nguin | You're probably not going to be teaching anyone to set up asterisk for a little while. |
| 04:37.35 | blognewb | lol yeah i know |
| 04:37.57 | [TK]D-Fender | blognewb: Asterisk is a PBX a telephony toolkit. It gives you means of bringing various telephony technologies together. What you do with it is up to you |
| 04:38.25 | [TK]D-Fender | blognewb: You could use it to build a PBX. Or you can do what I do and use it as a coffee timer and a jukebox. |
| 04:38.38 | blognewb | this is like the evolution theory to a creationist for me |
| 04:39.00 | blognewb | yes it's getting less blurry |
| 04:39.41 | p3nguin | I think drmessano uses asterisk to turn on and off the lights in his house. |
| 04:40.24 | blognewb | that's insane |
| 04:40.45 | [TK]D-Fender | Nope, same harware I use for my coffee.... |
| 04:41.00 | blognewb | you guys are so smart |
| 04:41.00 | [TK]D-Fender | perfectly sane |
| 04:41.42 | p3nguin | What is your coffee maker's extension? :) |
| 04:42.20 | [TK]D-Fender | p3nguin: h4X0r |
| 04:42.25 | p3nguin | Does the module have a way to poll for the current state, or do you have to use one extension for power on and another for power off? |
| 04:42.41 | [TK]D-Fender | p3nguin: Time delayed on/off |
| 04:42.43 | [TK]D-Fender | cycle |
| 04:42.49 | p3nguin | oh |
| 04:44.15 | p3nguin | "Hello, Mr. Coffee speaking..." |
| 04:44.38 | blognewb | the concept is amazing |
| 04:45.09 | p3nguin | "Hi, brew me four cups of dark roast at zero six-hundred, please." |
| 04:47.32 | [TK]D-Fender | p3nguin: Prior to starting with * I was working on a VB + Dragon ASR engine base for an AI to teach my home what i have and how to use it. You program it by talkig to it. |
| 04:47.43 | [TK]D-Fender | p3nguin: So I literally explain my home to itself |
| 04:47.44 | blognewb | [TK]D-Fender http://www.freepbx.org/ <--? |
| 04:47.54 | p3nguin | ~freepbx |
| 04:47.55 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 04:48.05 | blognewb | oh |
| 04:48.10 | blognewb | im sorry |
| 04:48.20 | [TK]D-Fender | blognewb: That is a WEB Gui front-end for configuring Asterisk within the limits of the structures they came up with |
| 04:48.21 | p3nguin | You'll never learn how to use asterisk if you let FreePBX run asterisk for you. |
| 04:48.43 | p3nguin | s/run /use / |
| 04:48.47 | [TK]D-Fender | blognewb: FreePBX builds * into a realtively common set of structures for an SMB PBX |
| 04:49.29 | blognewb | [TK]D-Fender i see. i was just trying to search a guide how to get it started like what to buy or do i have to buy anything at all etc |
| 04:49.50 | p3nguin | But if you have no use for learning asterisk, FreePBX can do the work for you. |
| 04:49.58 | [TK]D-Fender | blognewb: Depends what you want to do. |
| 04:50.25 | [TK]D-Fender | blognewb: First you need to get a better grasp on what * can do, and then start seeing how it may help you accomplish your goals |
| 04:50.48 | blognewb | [TK]D-Fender well right now i jsut want to help my mom save on her landline bills and set her up on using google voice or something similar to cancel her hefty at&t monthly expenses |
| 04:50.49 | p3nguin | A 10 year old computer, a compatible Linux distro, and asterisk ... make a pretty good PBX. |
| 04:51.16 | blognewb | p3nguin i got my mom a lenovo laptop... does it have to be a separate comuter? |
| 04:51.17 | [TK]D-Fender | blognewb: Asterisk doesn't inherently do any of that. |
| 04:51.32 | [TK]D-Fender | blognewb: The key for you is "get her on cheaper service |
| 04:51.34 | [TK]D-Fender | " |
| 04:51.47 | p3nguin | You'd have to leave the laptop on and online in order to make and get phone calls. |
| 04:51.57 | blognewb | i see |
| 04:52.06 | blognewb | like a server |
| 04:52.08 | [TK]D-Fender | blognewb: GoogleVoice to call out is a cheapr service perhaps, as are probably many ITSP's |
| 04:52.11 | [TK]D-Fender | ~itsp |
| 04:52.11 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 04:52.13 | [TK]D-Fender | ^^ |
| 04:52.15 | p3nguin | Exactly like a server. |
| 04:52.32 | [TK]D-Fender | So that is the service she would use instead of her current expensive one |
| 04:53.10 | [TK]D-Fender | Then the question is how will she use that service? She could use software on a PC & headset just like GoogleVoice & Skype offer. |
| 04:53.24 | sixohquad | where does asterisk install all of its sounds? |
| 04:53.31 | blognewb | so i guess in the mean time ill get her an obi and use her existing google voice number and existing landline devices |
| 04:53.51 | p3nguin | I typically reduce a $50/mo landline phone service down to $15/mo or less by switching to VoIP over an internet service which is already being paid for anyway. |
| 04:54.05 | [TK]D-Fender | Or if she already has regular phone you could get a SIP ATA, plug the phone in abnd have that talk directly to the ITSP. Presto, she has a real analog phone that places calls over that service |
| 04:54.12 | p3nguin | /var/lib/asterisk/sounds/ |
| 04:54.59 | sixohquad | nice thanks |
| 04:55.03 | blognewb | p3nguin but google voice is free at the moment right.. i was planning to buy the obi 50$ and get free calls via google voice using the same phone devices |
| 04:55.07 | [TK]D-Fender | blognewb: You should note that I have not introduced Asterisk into this solution at all. Your defined need can be solved without it. |
| 04:55.57 | blognewb | [TK]D-Fender i got the itsp part but a "sip ata" is what, like a real-life example.. |
| 04:55.59 | *** join/#asterisk vinhdizzo (~vinh@pool-74-100-182-10.lsanca.fios.verizon.net) |
| 04:56.08 | blognewb | obi = sip gateway right |
| 04:56.12 | p3nguin | So $50 for phone + $30 for internet = $80. Or $15 for VoIP + $30 for internet = $45. |
| 04:56.21 | [TK]D-Fender | yes |
| 04:56.28 | blognewb | phew we never learned this in networking class. |
| 04:56.31 | [TK]D-Fender | You plug dumb phone in, it talks SIP out the other side |
| 04:56.42 | p3nguin | The obi device is like an ATA, but it is for the Obihai Google Voice service. |
| 04:56.57 | [TK]D-Fender | because a dumb analog phone != LAN networking |
| 04:57.08 | [TK]D-Fender | that's what the ATA is for |
| 04:57.20 | p3nguin | I guess it's not like an ATA... it *is* an ATA, but one with a specific purpose predefined. |
| 04:57.43 | blognewb | so it's a sip ata gateway? |
| 04:58.10 | p3nguin | It's an ATA preprogrammed for Obihai's service. |
| 04:58.16 | blognewb | ok |
| 04:58.38 | [TK]D-Fender | "OBi100 — The OBi100 has one phone port. It supports Google Voice, SIP & OBiTALK VoIP services. Use the OBi100 when you do not need an analog line to a traditional telephone network or service. The OBi100 is perfect for customers who do not have a traditional telco phone service and want the savings and simplicity of using a VoIP service for all their calls" |
| 04:58.49 | [TK]D-Fender | Helps when you read their product page |
| 04:58.59 | [TK]D-Fender | it can talk multiple protocols <- |
| 04:59.12 | [TK]D-Fender | It supports Google Voice, SIP & OBiTALK VoIP services |
| 04:59.14 | [TK]D-Fender | ^ |
| 04:59.21 | blognewb | ill get the 110 |
| 05:00.26 | blognewb | [TK]D-Fender do you have a clip of your * coffeemaker |
| 05:00.38 | [TK]D-Fender | blognewb: Video? No |
| 05:01.38 | [TK]D-Fender | blognewb: blognewb I use the HEYU2 script to talk to my X-10 CM11A computer interface which remotely triggers the wall socket it is connected to |
| 05:01.54 | [TK]D-Fender | Coommodity gear that costs very little |
| 05:02.02 | blognewb | lol |
| 05:02.14 | blognewb | [TK]D-Fender you know those are all jargon to me |
| 05:02.23 | p3nguin | google it |
| 05:02.28 | blognewb | i will |
| 05:02.35 | blognewb | that's like talking to a grandma what a twitter is |
| 05:02.36 | [TK]D-Fender | blognewb: Where are you from? |
| 05:02.45 | blognewb | bay area |
| 05:02.47 | blognewb | the irony |
| 05:02.48 | *** join/#asterisk coreyf1513 (~cfarrell@75-130-93-234.dhcp.wlmn.ct.charter.com) |
| 05:02.55 | [TK]D-Fender | blognewb: http://www.x10.com/homepage.htm |
| 05:03.13 | [TK]D-Fender | blognewb: Well RadioScrap has been selling this stuff for 30 YEARS |
| 05:03.36 | [TK]D-Fender | blognewb: And you'll likely have been bombarded with their adds for years |
| 05:03.45 | [TK]D-Fender | ads* |
| 05:04.13 | blognewb | im only 26.. half of which i spent day dreaming |
| 05:04.26 | blognewb | googles radioscrap |
| 05:04.33 | [TK]D-Fender | RadioShack <- |
| 05:04.48 | blognewb | oh |
| 05:04.51 | blognewb | jeez |
| 05:04.58 | blognewb | winblows |
| 05:05.32 | blognewb | im not a hardware guy im clueless about everything networking and radioshack |
| 05:05.59 | blognewb | i bought replacement regular phone batteries at radioshack once |
| 05:06.02 | blognewb | doesn't count |
| 05:06.46 | [TK]D-Fender | Well they sold this home automation gear going back to before you were born.... |
| 05:06.55 | blognewb | this should be taught in school |
| 05:07.04 | blognewb | along with financial planning |
| 05:08.21 | [TK]D-Fender | Agreed |
| 05:09.13 | blognewb | all i hear around me is jersey shore or kim kardashian so the next generation might be pretty phckd |
| 05:09.24 | resist0r | ratshack was good for buying CB radio crystals as a kid heh |
| 05:09.52 | [TK]D-Fender | blognewb: Go talk to that kid who it bulding HAM radios, blowing things up, playing with lasers, etc. |
| 05:10.01 | [TK]D-Fender | These are people who are seeing how things work. |
| 05:10.14 | blognewb | [TK]D-Fender that's like big foot these days. |
| 05:10.18 | blognewb | at least in my area |
| 05:10.31 | resist0r | heh a myth ? |
| 05:10.39 | resist0r | often seen yet never confirmed |
| 05:10.50 | blognewb | wait there's this 16 yo girl in cupertino who cured cancer lol |
| 05:10.58 | [TK]D-Fender | Nope, they're all over the place and most people are too dumb to truly begin to comprehend just how dumb they are |
| 05:11.00 | blognewb | so might not be far out |
| 05:11.25 | resist0r | haha she didnt cure it... and besides she was working under a well known 'mad scientist' |
| 05:11.44 | blognewb | ill put asterisk in my to-do list |
| 05:11.48 | [TK]D-Fender | Canada already found a cure for cancer and it doesn't seem it'll ever hit the market |
| 05:11.49 | blognewb | his dad |
| 05:12.02 | blognewb | his dad i learnt is some molecular biologist i think |
| 05:12.20 | blognewb | the DCA something in canada? |
| 05:12.28 | [TK]D-Fender | drmessano: I should seriously write a book for all of these "what is * and what are the bits it's used to talk to. |
| 05:12.42 | [TK]D-Fender | blognewb: Unfamiliar acronym.... |
| 05:12.52 | blognewb | yeah that had side effects. but there's plenty more like beta glucan, egcg, liposomal curcumin |
| 05:13.02 | blognewb | even vit c IV |
| 05:13.07 | sixohquad | [TK]D-Fender, university of alberta cured cancer, you famillar with their program there? |
| 05:13.31 | resist0r | heh vit c IV |
| 05:13.36 | [TK]D-Fender | blognewb: http://www.youtube.com/watch?v=z1ifXxbxhZc |
| 05:13.45 | blognewb | sodium ascorbate but not all the time |
| 05:14.07 | blognewb | [TK]D-Fender yes i think they used dca in that |
| 05:14.14 | sixohquad | yeah DCA |
| 05:14.25 | blognewb | no dca has side effects |
| 05:14.34 | blognewb | might have worked in vitro but nah. |
| 05:15.05 | blognewb | [TK]D-Fender this one works: http://www.youtube.com/watch?v=widz9zM53y0 |
| 05:15.27 | blognewb | disregard the company. beta 1,3->6 glucans is the molecule |
| 05:15.35 | resist0r | well I suppose one could claim nearly anything with antioxidant properties is a cancer cure |
| 05:16.04 | blognewb | um not really. like say egcg robs you of iron, while vit c rusts the cancer cells with it |
| 05:16.17 | resist0r | he oxidation |
| 05:16.17 | blognewb | so if you use both concurrently, you would not get treated |
| 05:16.19 | resist0r | gotcha |
| 05:16.46 | blognewb | but they would work separately... although i think say if it's prostate cancer, might not want to use vit c |
| 05:17.28 | resist0r | in the form of sodium ascorbate or at all ? |
| 05:17.33 | *** join/#asterisk irroot (~gregory@196-215-57-38.dynamic.isadsl.co.za) |
| 05:18.26 | blognewb | resist0r at all. ascorbate is only used to alleviate the usual side effect of diarrhea |
| 05:19.07 | blognewb | egcg is the molecule from green tea |
| 05:19.25 | blognewb | 8bux cancer treatment |
| 05:19.43 | resist0r | hmm I dunno, I'm lean'n toward the cytotoxic impact |
| 05:20.06 | resist0r | haha 8bux cancer treatment |
| 05:20.08 | resist0r | cool |
| 05:20.14 | resist0r | way to put it |
| 05:21.07 | blognewb | resist0r lol well it's 8$ on iherb |
| 05:21.11 | blognewb | ,com |
| 05:23.04 | resist0r | what corner of the world do you currently reside blognewb ? |
| 05:23.12 | blognewb | cali |
| 05:23.16 | resist0r | ah okay |
| 05:23.29 | resist0r | I'm on the east coast of that very same country |
| 05:23.30 | resist0r | heh |
| 05:24.22 | resist0r | s/that/the |
| 05:24.25 | blognewb | lol |
| 05:24.34 | resist0r | not imply'n that cali is a country heh |
| 05:24.41 | blognewb | yay area |
| 05:24.45 | resist0r | haha |
| 05:24.55 | resist0r | typo I assume yet still works |
| 05:25.26 | blognewb | yay area just take a trip downtown |
| 05:25.34 | resist0r | heh I bet |
| 05:25.43 | blognewb | ebay hp sun |
| 05:25.45 | blognewb | i mean oracle |
| 05:27.34 | blognewb | I can't wait for nanotech though |
| 05:28.39 | blognewb | nanorobots pbx |
| 05:34.50 | p3nguin | scratches his head over this $36/hr thing. |
| 05:37.18 | *** join/#asterisk hipitihop (~denis@122.150.61.85) |
| 05:38.27 | blognewb | p3nguin what was it |
| 05:38.53 | p3nguin | I need to deinstall an old Toshiba KSU. |
| 05:39.24 | p3nguin | I was just puzzled by the pay rate to do it. |
| 05:39.46 | p3nguin | I guess that is the negotiated rate or something odd. |
| 05:59.58 | blognewb | [TK]D-Fender you familiar with soursop |
| 06:00.09 | [TK]D-Fender | nope |
| 06:00.16 | [TK]D-Fender | aside from sweet& |
| 06:00.20 | [TK]D-Fender | and hot& |
| 06:00.22 | blognewb | DAMN |
| 06:00.22 | [TK]D-Fender | :) |
| 06:00.27 | blognewb | yeah that IS the cancer cure |
| 06:00.42 | blognewb | http://www.iherb.com/Rain-Tree-Nutrition-Graviola-Max-600-mg-120-Capsules/17438?at=0 |
| 06:01.24 | blognewb | it's been buggin me all night so now i finally remembered |
| 06:02.07 | blognewb | it's a mitochondrial complex I inhibitor, ok shutting up now |
| 06:04.38 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-qfikfvbbqgbbcitr) |
| 06:05.33 | blognewb | you guys want me to use your amazon referral link? anybody |
| 06:17.12 | resist0r | ah, the mighty mitochondria. The power house |
| 06:21.16 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
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| 07:03.27 | *** join/#asterisk kactusotp (~chatzilla@203.59.226.41) |
| 07:04.08 | kactusotp | Hi Everyone, just wondering if there is a way to force disable calltoken between peers on IAX? |
| 07:05.03 | kactusotp | I've set call token optional on both sides the peer is showing that calltoken req is no, but |
| 07:07.03 | kactusotp | I still get "Too much delay in IAX2 calltoken timestamp from address" which searching through chan_iax is only called during in case 2 of handle_call_token |
| 07:09.30 | kactusotp | This occurs between an asterisk 1.8.8.0 and an asterisk (trixbox CE) updated to 1.6.0.26 |
| 07:11.07 | kactusotp | The 1.8 box was built over Christmas to replace a 1.6 server and I just need something to get me by until we can replace the trixboxes with PIAF |
| 07:12.01 | kactusotp | Problem crops up after 6-24 hours and causes all peers to go unreachable and stops processing all iax channels. |
| 07:14.14 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
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| 09:29.18 | moobius | hey there, it seems that there is a serious bug with video parsing on process_sdp crashing asterisk on 1.6.2.22. Has anyone seen this before? |
| 09:31.50 | kaldemar | moobius: even if there is, it won't be fixed. 1.6.2 branch has been in security fix only state since 2011-04-21. |
| 09:33.57 | moobius | kaldemar: i think that it is a improperly constructed c= on the sdp received by asterisk |
| 09:34.13 | moobius | if so it could possible lead to dos attacks |
| 09:44.54 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 09:45.04 | IsUp | hello |
| 09:45.12 | *** join/#asterisk hetii (~lew@194.181.154.25) |
| 09:45.13 | hetii | Hello :) |
| 09:46.10 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 09:46.55 | hetii | I have trouble with hylafax. I got response on AT command when i send it via minicom or cu -l /dev/ttyS0 but the probemodem cannot deduce DTE-DCE speed. Any clue ? |
| 09:47.37 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 09:50.01 | c4rg | hetii: what's connected to hylafax? iaxmodem? |
| 09:51.28 | hetii | also but now i talk about serial modem connected to ttyS0 |
| 09:53.11 | *** join/#asterisk irroot (~gregory@196-215-57-38.dynamic.isadsl.co.za) |
| 09:54.51 | hetii | hmm here i found some interesting stuff http://ftp.ntua.gr/mirror/hylafax/archive/html/1999-01/msg00263.html |
| 09:55.18 | hetii | <PROTECTED> |
| 09:56.07 | hetii | on my manual of cat the -u is (ignored) |
| 09:56.20 | hetii | so i wondering if this can be a reason |
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| 10:55.04 | schmidts | hello |
| 11:26.09 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 11:26.31 | joobie | hey guys.. anyone know of an auto-attendant style phone that has 50+ extensions that support presence? |
| 11:26.45 | joobie | i had a look at the polycom 650, btu that has max 42 extensions |
| 11:26.57 | joobie | want something wiht 50+ that's a decent phone |
| 11:29.53 | schmidts | joobie the cisco spa 5xx series can use two side bars with 64 extensions at all |
| 11:30.35 | hipitihop | can someone tell me how to fix following error: ERROR[31552]: res_config_sqlite.c:842 cdr_handler: database is full |
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| 11:31.13 | joobie | schmidts, thanks.. |
| 11:31.16 | joobie | what do you think of snom? |
| 11:31.41 | joobie | i need to get more basic phones for ordinary users.. but need some speed-dials on there.. the snom 320 has 12 speed dials |
| 11:31.44 | joobie | but duno about quality |
| 11:32.31 | schmidts | joobie forget about snom, they are EVIL |
| 11:33.00 | joobie | ahh |
| 11:33.01 | joobie | why? |
| 11:33.45 | schmidts | joobie trust me, they are just evil ;) |
| 11:34.14 | joobie | poor quality audio? |
| 11:34.22 | joobie | .. just out of curiosity :P |
| 11:34.39 | joobie | do you know of any decent phones with 10-12 line speed dial? |
| 11:36.14 | sekil | you can buy a sidecar for any phone out there |
| 11:36.32 | joobie | im trying to find a <$200 solution |
| 11:36.37 | joobie | as it's for the ordinary user |
| 11:37.01 | hipitihop | is there an easy way to disable cdr_handler ? |
| 11:37.07 | schmidts | joobie a cisco spa 502 should be available for around 100$ and the 532 side panel is around the same price |
| 11:41.13 | joobie | schmidts, need a 2 line phone min |
| 11:41.34 | joobie | who is decent in the US to buy these phones from btw? |
| 11:41.36 | joobie | like a good price |
| 11:41.42 | joobie | cos in AU, they are more expensive than 100$ |
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| 12:29.37 | skrusty | does anyone know what: "Received invalid event that had no device IE" means? I keep getting it when doing an Originate from AMI |
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| 13:01.51 | saxa | hi, I have a strange situation, I use Grandstream GXP285 phones here. They have a LAN and a PC port RJ45. If I connect my laptop to the PC port by the cable, when I receive a call from asterisk, every time after hangup it rings back. On the console I saw it starts MOH. Any idea, on what is wrong ? |
| 13:04.35 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 13:08.01 | skrusty | i had a ploblem like that once before, might be worth looking at the dialplan |
| 13:09.06 | skrusty | i dont remember the specifics, but when one party hung up, they would be called again almost right away... |
| 13:14.40 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:18.25 | *** join/#asterisk Kernel_Core (~IceChat7@h-213-136-53-142.na.cust.bahnhof.se) |
| 13:18.27 | Kernel_Core | hi all |
| 13:18.53 | Kernel_Core | is it possible to use Silk codec with IAX2 ? or IAX2 Trunk ? |
| 13:19.18 | saxa | skrusty: yeah, it rings back imediately |
| 13:20.14 | Kernel_Core | what about trunk ? |
| 13:22.39 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 13:22.46 | [TK]D-Fender | Kernel_Core, Yes |
| 13:24.15 | Kernel_Core | [TK]D-Fender: Thanks! |
| 13:24.59 | saxa | skrusty: strangely this happens only on one phone, and only when the notebook is connected to the PC port of the phone. |
| 13:25.58 | saxa | afaik in the dialplan all the phones (I have 5 of them) are listed in the same way. |
| 13:29.42 | *** join/#asterisk cadey (5a980315@gateway/web/freenode/ip.90.152.3.21) |
| 13:30.13 | cadey | Hi guys, anyone got or can point me in the direction of a rather detailed Asterisk+CentOS hardening guide |
| 13:38.51 | *** join/#asterisk celord (~celord@201.195.243.194) |
| 13:41.04 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 13:41.06 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 13:41.54 | as001 | Hi is it possible to see manager events without telneting to manager port. I have problem a lot of manager events which fly on the screen so I can't find what I need. |
| 13:41.55 | Nugget | telnet is eeeeeeevil! |
| 13:44.25 | cadey | as001 : use the manager through the web proxy |
| 13:45.20 | cadey | as001 : Example would be - http://localhost:8088/asterisk/manager?action=login&username=foo&secret=bar |
| 13:45.25 | as001 | ok thanks |
| 13:45.33 | cadey | you will then see the manager output in yoru browser |
| 13:46.01 | *** part/#asterisk as001 (~uros@82.117.198.142) |
| 13:46.40 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 13:49.31 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 13:50.12 | *** join/#asterisk batfastad (~benb@93-97-161-95.zone5.bethere.co.uk) |
| 13:51.31 | batfastad | Hi there. Not strictly Asterisk but anyone got any experience/recommendations for UK-based companies that can provide PSTN call forwarding for a pair of US/Canada toll free 866 numbers? |
| 13:51.52 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 13:52.35 | batfastad | I've been trying to port these numbers away from their current provider but it's a nightmare trying to go through 4 levels of reseller to change the RespOrg |
| 13:53.52 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 13:54.41 | batfastad | Or US-based call forwarding companies would be fine as well. Forwarding a pair of 866 numbers to our UK geographic numbers. I'd like the forwarding to happen over PSTN, not VoIP, as I'm concerned about the latency of the transatlantic hop over VoIP. |
| 13:54.47 | Defraz | I have a PRI coming into an asterisk server and I have two other asterisk servers, one as a primary and one as a secondary. I have my phones registered to both and I want some type of failover if server1 is offline. Would this dial plan work? |
| 13:55.00 | Defraz | exten=> 2085551234,1,Dial(SIP/2085551234@server1.domain.com) exten=> 2085551234,2,Dial(SIP/2085551234@server2.domain.com) |
| 13:55.12 | Defraz | sorry that was supposed to be two lines |
| 13:55.46 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 13:55.52 | *** part/#asterisk jaxon007_ (~jayesh@123.252.144.92) |
| 13:56.10 | [TK]D-Fender | dfrSure, but you should set up proper perrs between your servers and not se sending un-authed calls across like that |
| 13:56.21 | [TK]D-Fender | Defraz, ^ |
| 13:56.33 | [TK]D-Fender | peers* |
| 13:57.53 | Defraz | Oh I was trying to be as simple as possible but I could do peers. I allow only the ips of the phones and the Gateway PRI asterisk box to evenpass traffic to the server1 and server2 |
| 13:57.59 | Defraz | Do I still need to authenticate? |
| 13:59.55 | Defraz | For some reason my little two line dial plan doesn't work. |
| 14:00.16 | [TK]D-Fender | Defraz, As soons as IP's get spoofed or other traffic undesirably forwarded through your server... yes.. don't start getting lazy |
| 14:01.03 | Defraz | True |
| 14:01.31 | Defraz | Can you point me in the direction of an example of a peered failover dialplan? |
| 14:02.11 | Defraz | If you are authenticating would you go IAX or keep it SIP? |
| 14:02.45 | WIMPy | Just dialling one after the other was the right way. |
| 14:03.11 | WIMPy | You could check status before, but for the simple case I don't see any benefit. |
| 14:03.38 | [TK]D-Fender | Defraz, SIP if you aren't choked for BW |
| 14:04.21 | Defraz | Weird, I will have to find out why it didn't work then. Server1 was actually off and it just hung there passing the call. I will have to check into that, but I might peer them anyhow. |
| 14:04.38 | Defraz | Nope this is mostly internal. |
| 14:04.50 | WIMPy | Turn on qualify to get rid of the timeout. |
| 14:05.06 | WIMPy | You can also modify the timeouts. |
| 14:05.09 | Defraz | just in the sip.conf under general. |
| 14:05.19 | Defraz | just a qualify=yes |
| 14:05.33 | WIMPy | yes |
| 14:05.39 | *** join/#asterisk serafie (~erin@nat/digium/x-zldejnqbxjwqygdf) |
| 14:06.12 | Defraz | THanks for the info [TK]D-Fender and WIMPy. |
| 14:06.26 | WIMPy | You can even tune qualifyfreq if you want. |
| 14:06.45 | Defraz | Haven't every done that. |
| 14:06.53 | Defraz | is that just another option? |
| 14:07.07 | Defraz | Has that always been around? |
| 14:07.55 | Defraz | or do you just put qualify=100 |
| 14:07.57 | Defraz | or somethinglikethat |
| 14:08.16 | [TK]D-Fender | Defraz, qualify always has been, the freq I think is somewhat newish |
| 14:08.30 | Defraz | yea that is what I was thinking. |
| 14:11.28 | Defraz | WIMPy do you have a check status example? |
| 14:12.39 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:12.39 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:14.22 | WIMPy | qualify= gives the timeout, qualifyfreq how often to check. |
| 14:14.26 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 14:15.35 | *** join/#asterisk dwayne (~dwayne@c-76-29-230-45.hsd1.al.comcast.net) |
| 14:16.11 | *** join/#asterisk Deeewayne (~dwayne@c-76-29-230-45.hsd1.al.comcast.net) |
| 14:16.11 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:16.26 | WIMPy | And you may want to take a look at the timer values as well. |
| 14:19.18 | Defraz | autofallthrough=yes would that affect things. |
| 14:19.31 | *** join/#asterisk tamiel (~tamiel@85-171-170-252.rev.numericable.fr) |
| 14:20.02 | Defraz | I actually have never seen that setting until now. Just saw it while I was looking for how to set timer values |
| 14:20.20 | Defraz | *timeout |
| 14:20.47 | WIMPy | That's the dialplan. |
| 14:21.26 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-gzntpadneguoaaht) |
| 14:21.26 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:22.32 | *** part/#asterisk batfastad (~benb@93-97-161-95.zone5.bethere.co.uk) |
| 14:24.07 | *** join/#asterisk fireman_biff (~biff@65.48.133.103) |
| 14:24.08 | Defraz | Well thanks for all the help! Off to work I go! Have a great day! |
| 14:34.08 | *** part/#asterisk minaguib (~mina@modemcable132.139-59-74.mc.videotron.ca) |
| 14:35.52 | *** join/#asterisk Assalino (53f49842@gateway/web/freenode/ip.83.244.152.66) |
| 14:39.02 | Assalino | hello :) |
| 14:39.29 | Assalino | are any of you Asterisk-experienced developers, based in the UK? |
| 14:40.14 | WIMPy | ~ask |
| 14:40.14 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:40.28 | *** join/#asterisk JamesJRH (~james@vennington.plus.com) |
| 14:41.51 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 14:41.59 | Assalino | WIMPy, was that for me? |
| 14:42.35 | WIMPy | Yes |
| 14:42.48 | *** join/#asterisk woleium (~woleium@208.53.145.169) |
| 14:43.18 | Assalino | right |
| 14:44.05 | Assalino | My agency is looking for an Asterisk-experienced developer, based in the UK, to work on a project for a few weeks. Is anyone interested? |
| 14:44.13 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 14:44.16 | Assalino | @WIMPy, is that better? |
| 14:45.18 | WIMPy | First a short cut: th asterisk-bix mailing list may be the right place. |
| 14:45.21 | schmidts | Assalino imho you will have more luck on the asterisk-biz list then in here but i might be wrong ;) maybe you could provide some information what exactly you will need |
| 14:45.25 | schmidts | Wimpy :P |
| 14:45.25 | WIMPy | But you should tell more about the task. |
| 14:45.48 | Assalino | I see |
| 14:46.02 | wdoekes2 | why does he/she have to be in the uk? |
| 14:46.02 | WIMPy | Asterisk is far too komplex to know all of it so you should be more specific. |
| 14:47.57 | Assalino | My agency is looking for an Asterisk-experienced developer, based in the UK, to create an Asterisk solution that would allow a user to input a number on a website, get a phone call with an IVR which listens to the answers and processes them through Sphinx or any other speech recognition engine, returning the results as text and changing the website based on the result. |
| 14:48.14 | Assalino | Ideally they'd be in the UK so they could freelance from our office and work closely with our developers |
| 14:48.30 | Assalino | (I'll check the asterix-biz list though, thanks) |
| 14:48.52 | asteriskmonkey | Assalino, sphinx is sucking the big one for tranlating most of the time |
| 14:49.03 | WIMPy | Yes, that description shold give decent answers. |
| 14:49.33 | Assalino | I'm not too fussed about Sphinx. Could be LumenVox or Nuance :) |
| 14:49.39 | Assalino | whatever works better |
| 14:50.59 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:50.59 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:53.06 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:53.09 | Assalino | so I don't suppose any of you lot are in the UK? :) |
| 14:53.44 | WIMPy | I've been there a few times. Does that count? ;-) |
| 14:54.26 | [TK]D-Fender | I've watched Dr. Who for several decades... does that count? |
| 14:54.40 | saxa | http://pastebin.com/0cHMTGy9 |
| 14:54.48 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 14:54.51 | WIMPy | can't keep up with that. |
| 14:54.53 | jaytee | I had some English Breakfast tea once...does that count? |
| 14:55.04 | saxa | this is a sample call, why do I see the starting of MOH ? |
| 14:55.08 | rgsteele | Can you configure zaptel to handle calls from handsets out to the phone company if your card supports fxo and fxs (e.g., a tdm400p)? I'm trying to decipher the zaptel docs, but it's not clear whether I can use a single POTS line to do fxs (to interface with the PTSN) and fxo (to accept calls from handsets). |
| 14:56.01 | jaytee | zaptel? sounds jurassic |
| 14:56.02 | WIMPy | rgsteele: Yes, but zaptel has been replaced by dahdi some years ago. |
| 14:56.22 | [TK]D-Fender | rgsteele, A single line isn't FXO and FXS, it's one or the other |
| 14:56.23 | rgsteele | Yeah, I know, this happens to be a location with a very old server that they don't want to have upgraded right now. |
| 14:56.26 | rgsteele | Just trying to make it work. |
| 14:56.29 | p3nguin | rgsteele: The FXO port is the one you connect to your phone line. |
| 14:56.33 | [TK]D-Fender | rgsteele, And your description is somewhat circular |
| 14:56.37 | [TK]D-Fender | rgsteele, Try again. |
| 14:56.42 | rgsteele | [TK]D-Fender: K, let me rephrase |
| 14:56.44 | p3nguin | rgsteele: The FXS port is the one that you connect phones to. |
| 14:57.42 | p3nguin | Hook it up backward, and it will not work. It might even ruin your module. |
| 14:57.59 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 14:59.07 | rgsteele | [TK]D-Fender: I have a bunch of VOIP phones that all connect to an Asterisk server. SIP calls work fine. However, I need to be able to dial 911 via the POTS lines in an emergency if the SIP trunk is, for example, down. I have a TDM400P card in the Asterisk box, with a single POTS line connected to it from the phone company. I'm not quite sure what the right mixture of config options for... |
| 14:59.09 | rgsteele | ...zaptel.conf, zapata.conf, and zapata-channels.conf are to get it to work, or if one POTS line connected to the TDM400P is even sufficient. |
| 15:00.17 | [TK]D-Fender | rgsteele, How many lines do you think you need? |
| 15:00.32 | [TK]D-Fender | rgsteele, And forget the config files. You need to confirm your hardware requirement |
| 15:00.37 | rgsteele | I've been looking at the docs on voip-info.org, but I'm still a little fuzzy on whether I need more than 1 line (wasn't sure if a single line could do fxs and fxo simultaneously; I know the card supports both) |
| 15:01.07 | [TK]D-Fender | rgsteele, I don't see you stating any need for FXS |
| 15:01.19 | rgsteele | It looks like you can configure the card to do fxsks so that it interfaces with the PSTN |
| 15:01.26 | rgsteele | And then configure each module to do fxo |
| 15:01.27 | [TK]D-Fender | rgsteele, And that card supports both kinds of modules. What you have on your card is another matter |
| 15:01.51 | [TK]D-Fender | You don't configure the module to do FXS or FXO... it simply IS. |
| 15:01.56 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 15:01.59 | rgsteele | http://www.voip-info.org/wiki/view/Asterisk+config+zaptel.conf is what I'm looking at (the section called "Example zaptel.conf file") |
| 15:02.08 | [TK]D-Fender | rgsteele, Stop. |
| 15:02.16 | [TK]D-Fender | rgsteele, Forget the configs. |
| 15:02.24 | rgsteele | [TK]D-Fender: Okay. |
| 15:02.42 | [TK]D-Fender | rgsteele, You have what looks like a requirement to plug ONE phone line in. Not a phone. a LINE. From the telco. |
| 15:02.52 | [TK]D-Fender | rgsteele, Is this all? |
| 15:03.01 | rgsteele | Correct. |
| 15:03.11 | [TK]D-Fender | rgsteele, Then you need an FXO device. |
| 15:03.30 | [TK]D-Fender | rgsteele, single red FXO module on a TDM4XX will do. |
| 15:04.10 | rgsteele | [TK]D-Fender: Oh, I was assuming each module on the TDM400P could do either FXO or FXS. |
| 15:04.33 | [TK]D-Fender | rgsteele, the signalling mode in zaptel.conf / system.conf is "fxsks", in zapata.conf / chan_dahdi.conf is "fxs_ks" |
| 15:05.02 | [TK]D-Fender | <rgsteele> [TK]D-Fender: Oh, I was assuming each module on the TDM400P could do either FXO or FXS. <- no. that is why they are modules. So you buy and plug in the kind you need. |
| 15:06.00 | rgsteele | [TK]D-Fender: Understood. We never modified it - just used the standard configuration OOB |
| 15:06.04 | *** join/#asterisk irroot (~gregory@196-215-57-38.dynamic.isadsl.co.za) |
| 15:06.14 | rgsteele | (The TDM400P, that is) |
| 15:06.26 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 15:06.26 | [TK]D-Fender | <rgsteele> [TK]D-Fender: Understood. We never modified it - just used the standard configuration OOB <--- huh? |
| 15:06.39 | [TK]D-Fender | rgsteele, Well what has it got on it? |
| 15:07.22 | rgsteele | Hm... ztscan says they're all FXO... |
| 15:07.36 | [TK]D-Fender | rgsteele, Get a screwdriver and look at it |
| 15:08.18 | rgsteele | [TK]D-Fender: So, the FXO's are red and the FXS's are blue? |
| 15:08.42 | rgsteele | Er, green |
| 15:11.19 | [TK]D-Fender | yes |
| 15:19.41 | *** join/#asterisk CGMChris (~chatzilla@74.143.228.142) |
| 15:20.41 | CGMChris | Does anyone know of an option or different command that works similarly to ChanSpy(), but would effectively require you to use your mute button as you spy, and turn off mute to turn the call into a full blown forced conference? (SIP, not ZAP) |
| 15:21.45 | *** join/#asterisk gego_ (~quassel@b238085.customer.hansenet.de) |
| 15:24.34 | WIMPy | core show application ChanSpy |
| 15:24.36 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:24.36 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:25.10 | rgsteele | [TK]D-Fender: Hm, both had 4 Red FXO's |
| 15:25.20 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 15:25.54 | rgsteele | [TK]D-Fender: I thought that the FXS's were what interfaced with the PSTN, though? |
| 15:27.02 | rgsteele | I know you said only an FXO was needed - I just want to verify because the docs indicate that the FXS's are what's needed to talk to the telco. |
| 15:27.33 | CGMChris | WIMPy: I've already read that, and the application reference in "The Book". I do my best not to ask stupid questions in here. Are you implying whisper mode accomplishes what I'm trying to achieve? |
| 15:28.05 | WIMPy | The later. |
| 15:28.29 | WIMPy | You can use it on either or both channels. |
| 15:28.49 | rgsteele | WIMPy: You talking to me? |
| 15:28.56 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 15:29.17 | WIMPy | rgsteele: No, that was for CGMChris |
| 15:29.46 | rgsteele | WIMPy: Thought as much, just wanted to make sure so as not to ignore you if I was wrong :) |
| 15:29.53 | [TK]D-Fender | rgsteele, An FXO module is for connecting you to the Foreign eXchange Office. |
| 15:30.17 | [TK]D-Fender | rgsteele, However in the configs you can see the signalling says "fxs" all over it. It ACTS like a station TO the telco |
| 15:31.11 | rgsteele | Ah, had it backwards, my apologies. |
| 15:31.12 | [TK]D-Fender | <CGMChris> Does anyone know of an option or different command that works similarly to ChanSpy(), but would effectively require you to use your mute button as you spy, and turn off mute to turn the call into a full blown forced conference? (SIP, not ZAP) <-- there is no "mut button" in SIP. |
| 15:31.34 | rgsteele | [TK]D-Fender: The docs were a little ambiguous - thanks for clearing that upl |
| 15:32.01 | [TK]D-Fender | rgsteele, There are a few docs that also explain this, but might not be always the first thigns you come across. |
| 15:32.13 | [TK]D-Fender | rgsteele, It does confuse plenty of people the first time through. |
| 15:33.19 | rgsteele | [TK]D-Fender: Gotcha. Yeah, the whole "configure the FXO module with FXS statements" certainly did that. :) |
| 15:33.37 | *** join/#asterisk sled-dog (~luser@65-124-95-55.dia.static.qwest.net) |
| 15:34.01 | [TK]D-Fender | rgsteele, When talking to the telcoyou should not be talking as though you are the telco. Keep that logic in mind and you're good to go... |
| 15:34.27 | CGMChris | WIMPy: ChanSpy(,w) and ChanSpy(,qw) and ChanSpy()...on an inbound call, no matter how many * asterisk toggles through channels, only the inbound caller...not the person using my PBX, can hear the spy-er. |
| 15:34.38 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:34.58 | sled-dog | SIP clients behind NAT I've gotten to work.... but now they're telling me my server also has to be behind NAT (the morons...). Can this too be worked around? |
| 15:35.24 | [TK]D-Fender | ChanSpy only spys on 1 legs of the call. So when it sends audio, its to one side only. |
| 15:35.24 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:35.37 | [TK]D-Fender | CGMChris, If you want to talk to both ---> Bridge() |
| 15:36.05 | [TK]D-Fender | sled-dog, It isn't a problem (unless your router is crap) |
| 15:36.14 | [TK]D-Fender | ~sipnat |
| 15:36.14 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 15:36.16 | [TK]D-Fender | ^^ |
| 15:36.43 | [TK]D-Fender | sled-dog, 1.6+ uses "directmedia" instead of "canreinvite" |
| 15:36.48 | [TK]D-Fender | sled-dog, The rest applies. |
| 15:36.53 | sled-dog | thanks |
| 15:37.11 | CGMChris | [TK]D-Fender: That's new in 1.6, I'm using 1.4. I guess that means I'd have to get really elaborate...or upgrade. Thanks for your help. |
| 15:37.39 | WIMPy | CGMChris: See barge mode. |
| 15:37.51 | [TK]D-Fender | CGMChris, No, it means you can follow that guides as-is |
| 15:38.18 | [TK]D-Fender | CGMChris, sorry, coss-conversation. |
| 15:38.33 | [TK]D-Fender | CGMChris, Stuck on 1.4? Yes, time to think about upgrading |
| 15:38.34 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:38.34 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:39.58 | p3nguin | ChanSpy() has the ability to spy on both channels. See option B. |
| 15:40.15 | *** join/#asterisk e-fon_patrick (~e-fon_pat@fwj00.e-fon.ch) |
| 15:40.19 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 15:42.09 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 15:42.57 | rgsteele | bleh, channel congestion. |
| 15:43.27 | rgsteele | So, it's not connecting anymore, but I guess it's not really a step back since I was only getting silence prior to using all fxs config options |
| 15:45.25 | schmidts | p3nguin to spy on both yes, but you can only talk to one side |
| 15:45.32 | tzanger | http://www.voip-info.org/wiki/view/Asterisk+consultants+USA#COLORADO <-- has anyone used any of the guys listed for Colorado? any recommendations or other? |
| 15:46.25 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:47.22 | WIMPy | schmidts: What about option B? |
| 15:49.39 | schmidts | sorry my fault ;) |
| 15:51.59 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:54.17 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:56.08 | *** join/#asterisk felimwhiteley_ (~quassel@089-101-203026.ntlworld.ie) |
| 15:57.05 | _Corey_ | Anyone else have one of these new Polycom VVX500 phones? We're experiencing a lot of them "crashing"... curious if anyone else has had the same experience |
| 15:57.53 | leifmadsen | _Corey_: I know michael graves uses his quite a bit and hasn't run into that -- latest firmware I presume? |
| 15:58.38 | _Corey_ | yeah, I think so... I've triggered it a couple ways so far. Once adding a contact and now with a particular dial string... phone just locks and reboots |
| 15:59.16 | _Corey_ | I've only had it on my desk a couple days so far... still need to poke it a bit |
| 16:00.26 | leifmadsen | gotcha, well that's certainly interesting; those are the kinds of things that, if you can show how to reproduce it, that sending information upstream to Polycom would be very beneficial |
| 16:01.30 | _Corey_ | Yeah, we normally will submit a bug report after appropriate sanity checks... :) |
| 16:01.33 | *** join/#asterisk eZz (~ez@195.114.6.134) |
| 16:01.50 | eZz | hi |
| 16:02.05 | *** part/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 16:02.25 | leifmadsen | _Corey_: :) unfortunately I don't have the resources to get something quite so fancy :) |
| 16:02.27 | leifmadsen | nice phone though |
| 16:02.38 | leifmadsen | better than my Cisco 7970 which was a nightmare to get setup |
| 16:02.44 | leifmadsen | glad I sold that when I did |
| 16:03.16 | _Corey_ | leifmadsen: They're cheap actually... like $240 retail or something in that neighborhood |
| 16:03.40 | eZz | I have the following rule: exten => s,n,Set(Threshold=global=0.9;HUMAN=0.75;MACHINE=0.85;FAX=0.85) |
| 16:03.43 | eZz | somebody knows why the char '=' in value is causing to: [Jan 10 07:59:24] WARNING[12874]: pbx_config.c:1546 pbx_load_config: No closing parenthesis found? 'Set(Threshold=global=0.9' at line 864 of extensions.conf |
| 16:03.48 | leifmadsen | oh wait, you said VVX500, the new smaller version, I was thinking the VVX1500 or whatever it is |
| 16:04.00 | _Corey_ | Yeah, that thing is ridiculous |
| 16:04.11 | _Corey_ | I had one on my desk for 6 months... hated it |
| 16:04.17 | leifmadsen | eZz: because you have semi colons in the string |
| 16:04.21 | [TK]D-Fender | eZz, Looks like a missing ")" to me... |
| 16:04.22 | leifmadsen | which are comments for asterisk |
| 16:04.26 | leifmadsen | [TK]D-Fender: no he's not |
| 16:04.30 | eZz | leifmadsen: yes but i need to pass '=' char |
| 16:04.40 | eZz | [TK]D-Fender: no |
| 16:04.41 | leifmadsen | eZz: you need to escape the semi-colons |
| 16:04.41 | [TK]D-Fender | <PROTECTED> |
| 16:04.46 | eZz | leifmadsen: by \ ? |
| 16:04.47 | leifmadsen | [TK]D-Fender: no, you're wrong |
| 16:04.58 | leifmadsen | eZz: yes, in 1.8 it is \; in 1.4 \\\; |
| 16:04.59 | [TK]D-Fender | leifmadsen, Where do you see it? |
| 16:05.07 | leifmadsen | [TK]D-Fender: at the very end, he has it |
| 16:05.09 | eZz | leifmadsen: No closing parenthesis found? 'Set(Threshold=global\=0.9' at line 864 of extensions.conf |
| 16:05.11 | leifmadsen | his problem is the semi-colons |
| 16:05.12 | eZz | the same here |
| 16:05.22 | leifmadsen | eZz: that is an equals sign, not a semi colon |
| 16:05.29 | leifmadsen | it's not the equal that is the problem |
| 16:05.54 | eZz | ok, what do to in this case ? |
| 16:05.56 | leifmadsen | you have semi-colons in your string, which means everything after 0.9 is commented out |
| 16:05.57 | [TK]D-Fender | leifmadsen, Never seen a need to escape those before... ok... |
| 16:06.02 | leifmadsen | I just told you |
| 16:06.05 | [TK]D-Fender | leifmadsen, Silly * :p |
| 16:06.12 | leifmadsen | [TK]D-Fender: of course you would, why wouldn't you? |
| 16:06.15 | leifmadsen | it's a comment string |
| 16:06.17 | leifmadsen | ; this is a comment |
| 16:06.26 | eZz | hm |
| 16:06.30 | leifmadsen | Set(Threshold=foo;this is now a comment) <-- not closed |
| 16:06.38 | eZz | i see |
| 16:06.39 | [TK]D-Fender | leifmadsen, because *'s parser should know that the ( is open |
| 16:06.44 | *** join/#asterisk corretico (~luis@190.211.94.6) |
| 16:06.46 | corretico | hi |
| 16:07.00 | leifmadsen | ; Set(This is a perfectly valid comment and nothing was opened or closed) |
| 16:07.02 | eZz | hm, looks like \; shows no error |
| 16:07.04 | [TK]D-Fender | leifmadsen, Bracket counting dangnammit :) |
| 16:07.09 | leifmadsen | eZz: yes, like I said |
| 16:07.23 | leifmadsen | [TK]D-Fender: you continue to be wrong |
| 16:07.26 | corretico | is possible to make a call directly to any voicemail?? on my own asterisk? |
| 16:07.40 | [TK]D-Fender | leifmadsen, How so? |
| 16:07.40 | eZz | ok thank you very much |
| 16:07.41 | eZz | I just never had this before :-D |
| 16:07.59 | leifmadsen | eZz: np, just remember that a semi-colon is a comment, so if you use it where you want it to be literal, you need to escape it |
| 16:08.09 | eZz | leifmadsen: ok will note, thanks |
| 16:08.46 | [TK]D-Fender | <leifmadsen> Set(Threshold=foo;this is now a comment) <-- not closed <--- yes * bitches... but it shouldn't. It should be smart enough to know that the app call was missing its ) and not start counting it as a comment. As I said this is a dumb move on the parser's part. |
| 16:11.04 | [TK]D-Fender | <[TK]D-Fender> leifmadsen, Bracket counting dangnammit :) <- this was in reference to the parser's lack of intelligence. Not that this was the user's fault. |
| 16:12.29 | [TK]D-Fender | corretico, it's your dialplan. You can do whatever you want withit |
| 16:13.44 | leifmadsen | [TK]D-Fender: well that's how it has worked since the beginning of time, and it hasn't bothered anyone enough in over a decade to change, so it seems like a non-issue to me |
| 16:14.50 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 16:14.55 | [TK]D-Fender | leifmadsen, certainly not critical... I suppose if we were to look into that we may as well shred the whole thing and get a proper language in there with typed variables, etc ;) |
| 16:15.19 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v004-093.mobile.uci.edu) |
| 16:27.36 | *** part/#asterisk gego (~gego@b238085.customer.hansenet.de) |
| 16:29.27 | corretico | <[TK]D-Fender> this is for other asterisk that I need to call to specific voicemail. he ask me, if exist any number to dial my voicemail directly |
| 16:30.17 | [TK]D-Fender | corretico, And again, you can dial whatever you want outwards, and inward it's your dialplan, you can do whatever you want with it |
| 16:31.34 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 16:33.15 | *** join/#asterisk celord (~celord@201.195.243.194) |
| 16:40.06 | Assalino | guys, I've setup a server with ASterisk 1.8 and was trying to make a SIP call with 2 softphones |
| 16:40.25 | Assalino | I can never hear sound and after 30 seconds the call drops |
| 16:40.36 | Assalino | the asterisk CLI shows: Retransmission timeout reached on transmission |
| 16:40.43 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 16:40.48 | [TK]D-Fender | Common of improper NAT handling. |
| 16:40.55 | Assalino | and then chan_sip.c:3658 retrans_pkt: Hanging up call YmJjMjQyYzA0OGI0MGFhYjU3ZGE4NTQ0YmU5YWQxZDg. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). |
| 16:40.58 | [TK]D-Fender | Or other similar networking SNAFU's |
| 16:41.06 | Assalino | is it a port thing? |
| 16:41.09 | Assalino | or a config i missed? |
| 16:41.17 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 16:41.20 | Assalino | I've followed so many different tutorials, I wouldn't be surprised |
| 16:41.26 | Assalino | they're all incomplete, to some extent |
| 16:41.52 | [TK]D-Fender | ~sipnat |
| 16:41.52 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 16:41.55 | [TK]D-Fender | ^ |
| 16:41.59 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-sdniywfctjfxbwzb) |
| 16:42.09 | Assalino | checking :) |
| 16:42.22 | [TK]D-Fender | 1.6+ uses "directmedia" instead of "canreinvite". The rest applies. |
| 16:43.44 | Assalino | cheers |
| 16:48.42 | Assalino | [TK]D-Fender, do you know if Digium/Asterisk themselves recommend any Speech Recog engine? |
| 16:49.40 | [TK]D-Fender | Assalino, Nothing formal that I know of. Lumenvox is very reasonably priced and considerably better than Sphinx |
| 16:52.29 | Assalino | thanks :) |
| 17:00.31 | Assalino | [TK]D-Fender, that NAT tutorial you showed me is odd |
| 17:00.39 | Assalino | does it assume that we'll always know the caller's IPs? |
| 17:02.38 | [TK]D-Fender | no |
| 17:03.14 | [TK]D-Fender | Assalino, You tell your peer whether you can trust the IP their INVITE will give you and how to handle the media. |
| 17:03.44 | Assalino | geez, this is so over my head :D |
| 17:07.00 | *** join/#asterisk codatory (~codatory@IP-216-37-19-2.nframe.net) |
| 17:07.30 | p3nguin | assalino: Did you forward all the necessary ports? |
| 17:07.40 | p3nguin | Many people forward their SIP port but forget RTP. |
| 17:08.16 | Assalino | I added all the ones that were in a tutorial |
| 17:08.17 | Assalino | let's see |
| 17:10.04 | Assalino | i've got these: http://dl.dropbox.com/u/1977230/Screen%20shot%202012-01-10%20at%2017.09.18.png |
| 17:10.08 | Assalino | anything missing? |
| 17:10.40 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 17:12.17 | WIMPy | Usually, SIP uses UDP. |
| 17:14.55 | *** part/#asterisk codatory (~codatory@IP-216-37-19-2.nframe.net) |
| 17:16.07 | Assalino | so I've got 4569 for UDP |
| 17:16.34 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 17:17.25 | Assalino | actually, there was a scroll and I didn't include all ports |
| 17:17.35 | Assalino | for UDP I also have 5060 and 10000-20000 |
| 17:18.09 | *** join/#asterisk mattsqz (~luser@67-61-162-124.cpe.cableone.net) |
| 17:19.54 | mattsqz | what is the best prefab asterisk distro to start with? im setting up a VM to get my bearings and make sure this will work before i start building a server to replace the fonality box |
| 17:20.39 | mattsqz | i have asteriskNOW installed in a VM, should i stick with that for learning on? |
| 17:25.30 | *** join/#asterisk moy (~moy@216.172.42.74) |
| 17:27.36 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000:1def:bacd:1428:792b) |
| 17:30.06 | [TK]D-Fender | mattsqz, Depends on what you are "learning" |
| 17:30.56 | mattsqz | how asterisk works, familiarity with the .conf files, etc |
| 17:31.40 | [TK]D-Fender | mattsqz, Then forget pre-fab and compile it yourself over something like debian, centos, etc |
| 17:33.19 | mattsqz | production server will likely be gentoo or debian. i really just want to spend a few days getting familiar with a running asterisk system before i dive in |
| 17:33.44 | [TK]D-Fender | mattsqz, no bare install is "running" |
| 17:34.26 | mattsqz | first task on my list will be to take all the fonality .conf and xml files and rework them for *>1.2 |
| 17:34.40 | [TK]D-Fender | mattsqz, sample sip & extensions configs are usable only for documentation and as some loose syntax guide. Not usable as-is |
| 17:34.59 | *** join/#asterisk Cubber (~ronny@cpe-24-58-133-224.twcny.res.rr.com) |
| 17:35.05 | [TK]D-Fender | mattsqz, What is your end deployment supposed to look like? |
| 17:35.23 | [TK]D-Fender | mattsqz, And * knows nothing of XML |
| 17:35.28 | *** join/#asterisk Da-Geek (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 17:35.57 | mattsqz | i know, but i have to take what they do in xml and do similar things with * |
| 17:35.58 | [TK]D-Fender | mattsqz, And the only conf info that is of any use are things like DAHDI & SIP. dialplan is worthless |
| 17:36.25 | [TK]D-Fender | mattsqz, so you are going completely DIY on this? |
| 17:36.27 | mattsqz | ACD 10 seats, an additional 10 phones for regular incoming/outgoing |
| 17:36.34 | Cubber | I have setup a google voice inbound section in my dial plan, but for some reason I cannot get the call to pass to the asterisk voicemail system if the call is not answered. It just terminates. Here is my gv-incoming dial plan: http://pastebin.com/mm6vt07Z |
| 17:36.53 | mattsqz | the grandstream and avaya phones we have should work fine. |
| 17:37.01 | Cubber | I can get the call to go to GV voicemail if I change it a bit, but would rather use asterisk vm |
| 17:37.32 | mattsqz | pretty much |
| 17:37.50 | [TK]D-Fender | Cubber, You aren't calling Voicemail() in there at all. |
| 17:38.11 | mattsqz | i dont think it should be terribly difficult to get an asterisk box up and running that i can swap the fonality system out for over a weekend |
| 17:38.32 | Cubber | so do I need to add a line like exten => username@gmail.com,2,Voicemail(234) |
| 17:38.51 | [TK]D-Fender | Cubber, You need to call Voicemail at the point where you want to call it. |
| 17:39.48 | Cubber | [TK]D-Fender I want it to pass to VM after say 5 rings |
| 17:40.00 | [TK]D-Fender | Cubber, then call it after |
| 17:40.23 | Cubber | I added that line at the end and reloaded the dialplan but it did not work |
| 17:40.48 | [TK]D-Fender | Cubber, ok... |
| 17:41.00 | Cubber | probably a matter of syntax I will keep messing with it |
| 17:41.08 | Cubber | got it |
| 17:41.09 | [TK]D-Fender | Cubber, Or you could try showing us |
| 17:41.14 | Cubber | I had a 2 and needed a n |
| 17:41.31 | [TK]D-Fender | Cubber, Because you already had a "2" |
| 17:41.58 | Cubber | so what does n actually mean, the next number? |
| 17:42.02 | [TK]D-Fender | yes |
| 17:42.06 | Cubber | ahh makes total sense now |
| 17:42.07 | Cubber | thanks |
| 17:42.08 | [TK]D-Fender | Cubber, exten => username@gmail.com,n,Set(crazygooglecid=${CALLERID(name)}) <- this was "2" |
| 17:42.16 | [TK]D-Fender | "N"ext |
| 17:42.17 | Cubber | yup I get it |
| 17:44.11 | *** part/#asterisk nny (~Scott@174.107.223.14) |
| 17:45.07 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 17:46.46 | *** join/#asterisk min3r (1000@173.81.252.114) |
| 17:58.55 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 18:00.41 | *** join/#asterisk ikevin (~kevin@2a01:240:fe76:1::1) |
| 18:05.38 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 18:12.41 | Assalino | ~voip-ul |
| 18:12.43 | Assalino | ~voip-uk |
| 18:17.51 | Assalino | hello again |
| 18:18.01 | Assalino | so I setup an asterisk server |
| 18:18.10 | Assalino | and I'd like it to make a call to a phone number |
| 18:18.15 | Assalino | and play an mp3 (for example) |
| 18:18.26 | Assalino | I'll need to register with a VOIP service 1st, right? |
| 18:20.08 | *** join/#asterisk tamiel (~tamiel@85-171-170-252.rev.numericable.fr) |
| 18:23.56 | tzafrir | You need some sort of PSTN connectivity |
| 18:24.10 | tzafrir | A voip service is one way |
| 18:24.13 | [TK]D-Fender | Assalino, What you choose for that is up to you. |
| 18:28.02 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 18:29.23 | Assalino | [TK]D-Fender, thanks |
| 18:29.44 | Assalino | [TK]D-Fender, I'm trying to find a tutorial that will explain how to set up a VOIP service with ASterisk and make a 1st phone call |
| 18:29.53 | [TK]D-Fender | ~book |
| 18:29.54 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:30.21 | Assalino | tutorials are usually quicker, but i'll try my best |
| 18:30.22 | Assalino | thanks |
| 18:31.29 | Cubber | the book is deffinitly worth getting |
| 18:31.51 | Assalino | do you know of any VOIP service that will allow me to trial it with ASterisk 1st? |
| 18:32.05 | Cubber | google voice is free |
| 18:32.10 | Assalino | so I can see if it works before registering to a random plan |
| 18:32.12 | Cubber | and works fine |
| 18:32.20 | Assalino | can you call a landline from google voice? |
| 18:32.46 | Cubber | yes I have spent the past 2 days messing with it and got it setup so I can call out and in to pstn |
| 18:33.22 | Cubber | https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 18:33.25 | Assalino | but it's US & CAnada only, right? |
| 18:33.28 | Assalino | I'm in the UK :( |
| 18:33.31 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-ipzvihmjwyjbimtc) |
| 18:33.36 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 18:33.49 | Cubber | I also needed to reference the book to get it to work correctly, the wiki did not work for me but got me going in the right direction |
| 18:34.05 | Cubber | I think you can use it in UK but may have to pay, not sure |
| 18:34.18 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 18:34.57 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:34.57 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:35.55 | Cubber | I am trying to get it so I can dial *97 from my phone and access voicemail without typing in my user/pass. I know there is a way to do it. |
| 18:36.13 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 18:36.33 | Cubber | currently I have this for voicemail in extensions.conf http://pastebin.com/xP0ZCNjW |
| 18:36.35 | p3nguin | You can do it based on caller id number. |
| 18:36.46 | p3nguin | Or you can do it based on account code. |
| 18:37.00 | Cubber | I think what I have pasted is attempting caller id |
| 18:37.09 | p3nguin | I do it with both. If the account code is present, use it; if not, use the caller id number. |
| 18:37.10 | Cubber | but it does not work atm the way it is |
| 18:37.21 | Cubber | do you have an example? |
| 18:37.32 | p3nguin | Stop using numbered priorities. |
| 18:37.49 | Cubber | sorry changed to n's I got that from a site. |
| 18:38.01 | [TK]D-Fender | Cubber, ${CALLERIDNUM} <-- this variable was deprecated in * 1.2 |
| 18:38.13 | [TK]D-Fender | Cubber, "core show function CALLERID" |
| 18:38.38 | p3nguin | http://pastebin.com/K7d3FvYK |
| 18:38.52 | [TK]D-Fender | Cubber, Which if you look at your CLI outpu from your failed attempt, you'll probably see coming back blank |
| 18:38.55 | Cubber | so I can use CALLERID instead of what I am using or CID from what the dump shows? |
| 18:39.14 | [TK]D-Fender | Cubber, yes |
| 18:39.21 | p3nguin | ${CALLERID(num)} is the correct way to get the caller id number. See my example. |
| 18:40.42 | Cubber | changing to CALLERID(num) prompts just for the user password, thanks. |
| 18:40.52 | Cubber | p3nguin is the ? in line2 doing an or? |
| 18:40.58 | p3nguin | no |
| 18:41.08 | p3nguin | core show application ExecIf |
| 18:41.32 | Cubber | ahh thanks |
| 18:41.44 | p3nguin | If the account code is null, set the vmbox to the callerid number; otherwise, set it to the account code. |
| 18:41.51 | p3nguin | That's what line 2 says. |
| 18:41.55 | Cubber | so an if else statement baseically |
| 18:42.35 | p3nguin | All of the if applications use this concept. |
| 18:42.46 | p3nguin | zerohalo: You can fix that at any time. Thanks. |
| 18:43.35 | p3nguin | AppName(condition?what-to-do-if-true:what-to-do-if-false) |
| 18:43.57 | p3nguin | Some times it runs another application, such as in the case of ExecIf. |
| 18:44.14 | leifmadsen | p3nguin: you could also have done: Set(VMBOX=${IF($[${ISNULL(${CDR(accountcode)})}]?${CALLERID(num)}:${CDR(accountcode)})}) |
| 18:44.29 | leifmadsen | same thing, different sauce |
| 18:44.42 | p3nguin | Some times it goes to a priority label, as in the case of GotoIf and GosubIf. |
| 18:44.45 | Qwell | Now that's a Leifism if I ever saw one. |
| 18:44.56 | leifmadsen | Qwell: <3 the IF() function |
| 18:45.07 | Qwell | IF() nested with $[] FTW |
| 18:45.21 | Qwell | bonus points for the ISNULL() |
| 18:45.27 | Cubber | so by using: exten => *98,n,VoicemailMain() |
| 18:45.35 | p3nguin | leifmadsen: I'll mull that over for a while. I may change the behavior if I determine I like it better. |
| 18:45.36 | Cubber | they would be prmopted for everything correct? |
| 18:46.13 | p3nguin | It will throw the caller into the default vm context, and ask for the mail box number. |
| 18:46.21 | leifmadsen | Qwell: I just took the ExecIf() from p3nguin and tweaked it for my own style :) |
| 18:46.34 | Cubber | ok thanks |
| 18:46.50 | Qwell | leifmadsen: we need a S_OR() |
| 18:47.09 | leifmadsen | <3 braces, brackets, and parenthesis |
| 18:47.11 | Qwell | We have one in the Asterisk code. It does exactly what you just did above. You'd use like... |
| 18:47.16 | leifmadsen | Qwell: for some reason I almost thought we did... |
| 18:47.30 | Qwell | Set(foo=${S_OR(${VAR1}, ${VAR2})} |
| 18:47.33 | Qwell | ) |
| 18:47.35 | [TK]D-Fender | ISNULL() takes more chars than "" = "". Friends don't let friends waste characters ;0 |
| 18:48.17 | Qwell | leifmadsen: in Asterisk, it's something like.. !ast_strlen_zero(var1) ? var1 : var2 |
| 18:48.27 | leifmadsen | Qwell: is it done yet? |
| 18:48.51 | Qwell | meh. |
| 18:49.43 | *** join/#asterisk Sean-Der (~Sean-Der@NW1-DSL-74-215-64-154.fuse.net) |
| 18:51.47 | rgsteele | [TK]D-Fender: Hm, still having some problems - getting instant busy signal with the FXS-based configs. Anything jump out at you here? http://pastie.org/3161710 |
| 18:52.49 | [TK]D-Fender | rgsteele, Executing [s@macro-place-outbound-call:7] Dial("SIP/ryans-007c2980", "Zap/g1/411||TW") in new stack |
| 18:53.06 | [TK]D-Fender | rgsteele, you are dialing Group 1 .....and all of your channels are in Group 0 |
| 18:53.14 | [TK]D-Fender | rgsteele, There are no channels to dial from |
| 18:53.43 | rgsteele | [TK]D-Fender: Yeah, I tried using group 1, but then asterisk reports that reloading the chan_zap.so module is unsuccessful |
| 18:53.49 | [TK]D-Fender | rgsteele, Also curious you have 8 channels, but only mentioned one TDM400 |
| 18:54.02 | [TK]D-Fender | rgsteele, Not the reason |
| 18:54.39 | [TK]D-Fender | rgsteele, Also, what ver of * are you on? |
| 18:55.28 | [TK]D-Fender | rgsteele, And simpler logic say just change your Dial() to match.. that doesn't require chan_zap to be reloaded |
| 18:55.36 | rgsteele | [TK]D-Fender: Yeah, I tried putting channel => 1-8 in zapata.conf, but I also got the chan_zap.so unsuccessful reload with that set. This box is running 1.4.17 with some updates from newer versions. |
| 18:55.50 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:56.07 | [TK]D-Fender | rgsteele, Well you can look at that after... fix the group first |
| 18:57.22 | [TK]D-Fender | rgsteele, BTW, zapata-channels.conf is not a regular * config file and isn't being INCLUDE'd anywhere so it isn't being used for anything |
| 18:57.41 | troyt | How does one make a voicemail box have no password (ie. no password prompt, etc.) |
| 18:57.42 | [TK]D-Fender | rgsteele, I guess that may be "leftovers", because your zapata.conf only lists 4 |
| 18:58.57 | [TK]D-Fender | rgsteele, note you have also only configured 4 of your 8 channels. across those 2 cards... make sure your line(s) are connected to the one * will actually end up using... |
| 18:59.41 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:00.26 | [TK]D-Fender | troyt, "core show application VoiceMailMain" |
| 19:00.56 | *** join/#asterisk bmg505 (~leon@196-209-10-111.dynamic.isadsl.co.za) |
| 19:03.26 | rgsteele | [TK]D-Fender: Ok, back to the original behavior of nothing but silence: http://pastie.org/3161710 |
| 19:03.37 | rgsteele | [TK]D-Fender: I also updated it so all 8 channels are in zapata.conf |
| 19:03.40 | troyt | Thanks |
| 19:04.12 | rgsteele | [TK]D-Fender: There are two cards, so yeah, 8 actual modules |
| 19:06.07 | rgsteele | [TK]D-Fender: FWIW, I do see this when doing a 'reload' or 'reload chan_zap.so': [Jan 10 14:05:36] WARNING[28115] chan_zap.c: Ignoring signalling |
| 19:06.33 | rgsteele | Might not be relevant though. |
| 19:07.38 | [TK]D-Fender | rgsteele, And are all 8 ports connected to lines that should be providing dialtome? |
| 19:08.15 | rgsteele | [TK]D-Fender: Nope, only one. |
| 19:08.24 | rgsteele | port number 1 on the first card |
| 19:08.29 | [TK]D-Fender | rgsteele, Excellent odds it isn't on Zap/1 |
| 19:08.54 | [TK]D-Fender | rgsteele, I'd just start jacking it in to each one after the next |
| 19:09.22 | rgsteele | [TK]D-Fender: Hm, okay |
| 19:22.21 | saxa | hi, anybody can help me understand why it starts moh by its own ? |
| 19:22.25 | saxa | http://pastebin.com/vCwhGFz8 |
| 19:23.56 | p3nguin | zerohalo: Knock it off already. |
| 19:24.06 | zerohalo | eek |
| 19:24.08 | zerohalo | sorry |
| 19:25.14 | [TK]D-Fender | saxa, try looking at what SIP/ludmila is doing |
| 19:27.59 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:30.23 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:528:6b19:5ff:e5a9) |
| 19:32.54 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 19:35.17 | akrohn | i have an auto attendant with no phones attached to it. it just forwards calls using Dial() based on menu selections. Problem is, some of the calls drop off after a minute of connection. does anyone know what settings I can check to see which this happens? |
| 19:39.59 | rgsteele | [TK]D-Fender: I'm pretty sure it was just telling me which channel it was connecting to |
| 19:40.24 | rgsteele | If I switched it to the slot marked '1' on the other card, it hits Zap2/1 |
| 19:41.10 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 19:41.21 | [TK]D-Fender | rgsteele, that isn't what you think it is, nor the test I suggested |
| 19:42.20 | rgsteele | [TK]D-Fender: I did switch the port it was plugged in to |
| 19:43.17 | [TK]D-Fender | rgsteele, through all 8 until you found dialtone on one? |
| 19:44.24 | rgsteele | I had to stop switching after four though - apparently that line is used more frequently than I thought for inbound international callers that can't use the toll-free's (which get routed through the SIP trunk). |
| 19:45.27 | rgsteele | [TK]D-Fender: But, on card 1, port 1, it was Zap/1-1, on card 2, port 1, it was Zap/2-1, etc. etc. |
| 19:45.44 | [TK]D-Fender | Maybe card 1 .. isn't card 1 |
| 19:46.07 | [TK]D-Fender | You stopped your test early and are trunning on an assumption of which one is which |
| 19:47.47 | *** join/#asterisk SerajewelKS (devnull@wikipedia/Crazycomputers) |
| 19:48.04 | rgsteele | Right, which is why I tested on the other card, on the port marked '1' (it's etched into the metal next to the port), and it changed to 2-1, and so on. I will run through all of them tomorrow morning though, before 8am. Besides, I won't hear a dialtone for it. I mean, if I hook an analog phone up to the line, I do, but the SIP phones get a dialtone once they connect to the * server |
| 19:48.54 | Micc | is there any way to log all sip debug data to a database? |
| 19:48.57 | rgsteele | [TK]D-Fender: So, the two numbers after the Zap/... Are they supposed to be <card>-<channel>? |
| 19:49.21 | SerajewelKS | i've got a rather bizarre problem with my asterisk setup. i don't think asterisk is the problem here, but hopefully someone knowledgeable can point me in the right direction. i'm trying to dial in to an extension on my asterisk install from the telephony stuff we use at work, and the connection gets established but the client at work immediately disconnects and reports that RFC 2833 support is required. |
| 19:50.00 | SerajewelKS | a DID number i got works just fine (i can dial in from my phone) which leads me to believe that this is not an asterisk-related problem, but more likely something with my softphone. however, the softphone appears to support RFC 2833. |
| 19:50.08 | rgsteele | SerajewelKS: in your sip.conf, do you have rfc2833compensate=yes? |
| 19:50.15 | rgsteele | Is your SIP provider requiring it? |
| 19:50.41 | SerajewelKS | the SIP service i use at work appears to require that callees support it |
| 19:50.50 | rgsteele | There are also other rfc2833 settings you may want to look at, depending on your provider's requirement. |
| 19:51.41 | SerajewelKS | rgsteele: in my sip.conf i have dtmfmode unset (rfc2833 is the default) and i have dtmfmode=rfc2833 in the config for this particular phone |
| 19:52.08 | SerajewelKS | rgsteele: would there be a way to tell from the sip debug logs whether the phone supports rfc2833? |
| 19:52.38 | [TK]D-Fender | rgsteele, No, it's channel-sbcall |
| 19:52.41 | [TK]D-Fender | subcall |
| 19:52.49 | SerajewelKS | at the point when the connection is established, i see this: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) |
| 19:52.52 | [TK]D-Fender | rgsteele, Which on FXO is pointless. |
| 19:52.56 | rgsteele | SerajewelKS: Better would be to look at the docs for the phone |
| 19:53.02 | rgsteele | [TK]D-Fender: Ah |
| 19:53.08 | rgsteele | [TK]D-Fender: Thanks for the clarification |
| 19:53.48 | SerajewelKS | rgsteele: the phone's documentation doesn't provide any indication one way or the other (http://microsip.org.ua/) |
| 19:54.09 | SerajewelKS | rgsteele: is the telephone-event capability related to rfc2833 at all? |
| 19:54.11 | rgsteele | I guess maybe call the company. Also, do a tcpdump on the traffic |
| 19:54.39 | rgsteele | SerajewelKS: My inclination would be to think that it's your * config, but hard to say without debugging. |
| 19:54.52 | rgsteele | [TK]D-Fender: Gotta run, but I'll run thru all the lines tomorrow. |
| 19:55.10 | rgsteele | [TK]D-Fender: Thanks for the help |
| 19:55.13 | [TK]D-Fender | rgsteele, Alrighty... |
| 19:55.17 | [TK]D-Fender | rgsteele, np |
| 19:56.12 | SerajewelKS | hmm, i can't find any documentation in asterisk itself about rfc2833compensate. all the google hits are vague and don't specify exactly what it does. |
| 19:57.25 | SerajewelKS | would turning on RTP debugging help identify if RFC2833 is supported by my softphone? |
| 19:58.33 | Qwell | If your phone doesn't support RFC2833, you need to set it on fire. |
| 19:59.40 | SerajewelKS | it's a softphone, that's not really possible without damaging other hardware ;) |
| 20:00.02 | SerajewelKS | i suspect that the softphone does support it though. i'm trying to deduce why our telephony system here can't figure that out. |
| 20:01.57 | SerajewelKS | it does appear to be the telephone-event capability |
| 20:03.33 | SerajewelKS | what exactly am i looking for in the SIP and/or RTP sessions to determine if my phone correctly advertises this capability? |
| 20:04.24 | lauris | how to make asterisk 1.6 to register as a client to a remote sip proxy? i did everything what was written in the manual but asterisk only sends OPTIONS packet |
| 20:04.27 | lauris | and no REGISTER |
| 20:04.30 | saxa | [TK]D-Fender: thx, any hint on how to do that ? sip set debug on ? |
| 20:04.56 | lauris | is there any magick in this case? |
| 20:05.16 | *** join/#asterisk its_jeremy_ (~omghax@24-119-28-208.cpe.cableone.net) |
| 20:05.50 | Qwell | lauris: see register under the [general] section |
| 20:06.29 | lauris | ooops, my bad :) |
| 20:06.33 | lauris | thanks, now it works |
| 20:06.44 | SerajewelKS | my phone advertises in the OK response to the INVITE: a=rtpmap:96 telephone-event/8000 |
| 20:07.08 | Qwell | ~sip debug |
| 20:07.12 | Qwell | ~sipdebug |
| 20:07.14 | Qwell | stupid bot. |
| 20:07.32 | Qwell | SerajewelKS: pastebin the SIP debug for the full call, including logs where Asterisk is failing it |
| 20:07.33 | SerajewelKS | Qwell: are you directing that to me or to lauris? |
| 20:07.57 | SerajewelKS | Qwell: asterisk is not failing, our telephony system is hanging up the call because it wants RFC 2833 support |
| 20:08.07 | Qwell | Set it on fire. |
| 20:08.25 | Qwell | alternatively, pastebin the SIP debug etc etc |
| 20:08.56 | [TK]D-Fender | saxa, Yes |
| 20:12.36 | SerajewelKS | Qwell: any particular part of the debug log you are interested in? the full log contains several of my phone numbers many times and i'm a bit leery of posting those in public, yet i don't want to sanitize the entire log if only part of it would be helpful. |
| 20:13.57 | citywok | SerajewelKS: Find & Replace is your friend :) |
| 20:15.42 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 20:18.00 | Qwell | SerajewelKS: sanitize phone numbers and/or IPs as needed, just make sure you replace them all with the same values. |
| 20:18.09 | SerajewelKS | right |
| 20:18.36 | SerajewelKS | i'm taking a more careful read of the log right now and i may have found the problem. gimme a sec to research. |
| 20:20.08 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 20:20.19 | [sr] | hellou my friends |
| 20:20.49 | SerajewelKS | phone->* has telephone-event, *->phone has telephone-event. corp->* has telephone-event, *->corp does not have telephone-event. |
| 20:22.01 | SerajewelKS | which, if i'm not mistaken, means that * is not opening an rfc2833-capable channel to our telephony system |
| 20:24.29 | SerajewelKS | this phone has directmedia=no (since it is behind NAT) and dtmfmode=rfc2833. it appears that * is not proxying the telephone-event RTP channel from the phone. |
| 20:26.14 | [TK]D-Fender | SerajewelKS, * isn't a "proxy", it will send what you put in your peer |
| 20:27.05 | SerajewelKS | [TK]D-Fender: with directmedia=no it will effectively proxy the RTP stream, yes? (it won't reinvite to get an optimal path between the peers) |
| 20:27.22 | SerajewelKS | not proxy in the sense of blindly forwarding packets, of course |
| 20:28.28 | SerajewelKS | but the RTP stream will in effect be going through * instead of directly between the two callers |
| 20:28.43 | [TK]D-Fender | SerajewelKS, I wouldn't bet on it... perhaps it does or perhaps the packets are filtered... |
| 20:28.58 | [TK]D-Fender | SerajewelKS, make sure your peer explicitly states what it should be using every time. |
| 20:29.09 | SerajewelKS | well the incoming packets are interpreted and repackaged before sending them to the other peer |
| 20:29.59 | SerajewelKS | what i see is that my phone advertises the telephone-event channel in the OK response to *, then * turns around and does not advertise telephone-event to the corp in its OK |
| 20:31.58 | SerajewelKS | can the SIP debug log be directed to a file so that i can edit it without a huge copy/paste? (running inside screen, so copying pages of data is not fun) |
| 20:32.56 | [TK]D-Fender | SerajewelKS, Firs tset the dtmfmode for your peers |
| 20:33.30 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 20:40.11 | SerajewelKS | [TK]D-Fender: hmm? |
| 20:40.21 | SerajewelKS | [TK]D-Fender: the peer defined in sip.conf for my phone has dtmfmode=rfc2833 |
| 20:40.36 | SerajewelKS | [TK]D-Fender: the other peer is an incoming call, so it should use the [global] dtmfmode=rfc2833 |
| 20:40.43 | [TK]D-Fender | SerajewelKS, that is only HALF of the picture |
| 20:41.27 | SerajewelKS | where is the other half then? |
| 20:43.08 | [TK]D-Fender | the other end of the call. |
| 20:46.41 | SerajewelKS | the other end of the call is not being sent from asterisk |
| 20:46.51 | SerajewelKS | however, i can see the traffic in a tcpdump on my server |
| 20:47.23 | SerajewelKS | i get the sense that i haven't accurately communicated the entire scenario |
| 20:47.37 | [TK]D-Fender | SerajewelKS, You have phone & corp. You've describes PHONE, and have avoided CORP solidly |
| 20:47.54 | SerajewelKS | "avoided" in what way? i can see the traffic corp<->* |
| 20:48.32 | *** join/#asterisk moos3 (~rgenthner@cpe-72-224-121-41.maine.res.rr.com) |
| 20:48.49 | [TK]D-Fender | <SerajewelKS> [TK]D-Fender: the peer defined in sip.conf for my phone has dtmfmode=rfc2833 <-- no mention of corp, only phone |
| 20:49.04 | SerajewelKS | the traffic is the only information i have. i do not have control over the PBX at corp. |
| 20:49.15 | [TK]D-Fender | [TK]D-Fender: the other peer is an incoming call, so it should use the [global] dtmfmode=rfc2833 <- a peer is a peer, a peer is not a call |
| 20:49.18 | SerajewelKS | the INVITE comes from the corp PBX |
| 20:49.19 | moos3 | what would cause a SIP trunk from bandwidth.com claim to be unreachable even tho the traffic is leaving my network ? |
| 20:49.36 | [TK]D-Fender | SerajewelKS, You aren't showing how you're matching the call. |
| 20:50.01 | [TK]D-Fender | SerajewelKS, And made some assumed claim that [general] was responsible for it. We never got to see the call |
| 20:50.41 | SerajewelKS | matching in what sense, the dialplan? |
| 20:50.49 | [TK]D-Fender | no, SIP.CONF |
| 20:51.30 | SerajewelKS | considering that my phone and [general] are the only two sections in use, should [general] not apply to incoming calls? |
| 20:51.53 | [TK]D-Fender | SerajewelKS, Make a proper peer for them |
| 20:53.12 | *** part/#asterisk Da-Geek (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 20:55.14 | SerajewelKS | ok, how can i verify that incoming calls are properly associate with this new peer, just to make sure this config works? |
| 20:55.42 | *** join/#asterisk MrNemus (~lbuttars@c-76-27-99-15.hsd1.ut.comcast.net) |
| 20:55.47 | [TK]D-Fender | watch the SIP debug from * CLI |
| 20:57.31 | SerajewelKS | hah well i guess that answers that question. forgot to define that context in the dialplan, so the call failed completely. |
| 20:58.14 | SerajewelKS | and now the call goes through. hmm, fun. |
| 20:58.34 | SerajewelKS | would it be safe to assume that some setting in [general] was interfering? |
| 21:01.41 | *** part/#asterisk lauris (~la@unaffiliated/lauris) |
| 21:02.10 | SerajewelKS | i'm a bit confused since i thought that settings in [general] are inherited by peer/device sections. the only thing in this peer section is type=peer, host/port, and context. |
| 21:02.19 | SerajewelKS | i do not change dtmfmode, for example |
| 21:02.33 | SerajewelKS | i'm baffled how the existence of a peer could change how the RTP stream is negotiated |
| 21:03.29 | moos3 | anyone know anyone i can hire to look at a issue on by asterisk box, we use to hire leif but hes not avaliable |
| 21:04.12 | SerajewelKS | in fact, after commenting out that section and reloading the config, the call still goes through |
| 21:05.00 | WIMPy | moos3: The asterisk-biz mailing list is supposed to be a good place to ask, but you should be a lot more specific. |
| 21:05.07 | [TK]D-Fender | SerajewelKS, Well rejected for dialplan SNAFU's is pretty blatant and serious |
| 21:05.42 | SerajewelKS | [TK]D-Fender: you mean how our PBX hangs up when the peer doesn't support rfc2833? |
| 21:05.56 | [TK]D-Fender | No, that would be something else... |
| 21:06.06 | SerajewelKS | then what do you mean by "rejected" |
| 21:06.10 | [TK]D-Fender | SerajewelKS, But I'm not going to go around guessing without seeing calls at this point. |
| 21:06.36 | [TK]D-Fender | "forgot to define that context in the dialplan, so the call failed completely." <- rejected |
| 21:07.13 | SerajewelKS | oh, yes. that was a misconfiguration that i quickly fixed. that was after creating the new peer definition. i only said that to indicate that obviously the incoming call was being matched to that peer. |
| 21:08.10 | SerajewelKS | that wasn't the issue i've been wrestling with, that only indicated to me that i had the sip.conf configuration correct after adding the peer definition, just forgot to add a dialplan context |
| 21:09.33 | SerajewelKS | i have reverted my config to where it was when i started having this problem and now i can't reproduce the problem. so i'm happy that it works, but really confused about why it wasn't. |
| 21:12.34 | SerajewelKS | mmkay, so restarting asterisk brought the problem back. here's what i've figured out: defining the peer in sip.conf resolves the negotiation problem, but i haven't the faintest clue why. |
| 21:17.17 | [TK]D-Fender | SerajewelKS, Possible that applicable parameters are peer-only. |
| 21:17.44 | SerajewelKS | right, i'm now trying to isolate which one is causing this problem |
| 21:18.26 | SerajewelKS | it's not dtmfmode. i can set dtmfmode to rfc2833 or auto in the peer definition and the call goes through. |
| 21:18.35 | SerajewelKS | and those are the only values i've ever put in [global] |
| 21:22.16 | SerajewelKS | i just copied everything in [general] into this peer definition. and it still works |
| 21:22.22 | SerajewelKS | i am extremely confused |
| 21:22.34 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 21:23.35 | SerajewelKS | i can't define a peer for the entire internet. so i'm unsure how i can expect this config to work with other internet hosts that want rfc2833. |
| 21:25.56 | SerajewelKS | and i'm not even sure why this config works at all :( |
| 21:26.10 | *** join/#asterisk Troyblackman_ (~samuelmat@65-36-42-216.static.grandenetworks.net) |
| 21:26.50 | *** part/#asterisk Troyblackman_ (~samuelmat@65-36-42-216.static.grandenetworks.net) |
| 21:26.58 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 21:28.53 | [TK]D-Fender | Checkout time, BBIAB |
| 21:29.17 | [TK]D-Fender | SerajewelKS, You aren't often dealing with "every ranom person on the internet". That's what peers are for... |
| 21:29.18 | [TK]D-Fender | BBL |
| 21:30.40 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 21:30.46 | paulc | <PROTECTED> |
| 21:37.03 | [sr] | hi WIMPy |
| 21:39.39 | WIMPy | Hi [sr] |
| 21:48.55 | SerajewelKS | boggles |
| 21:49.11 | SerajewelKS | i've no idea wtf is going on with this config |
| 21:52.29 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:55.03 | SerajewelKS | anyway, can sip debug info be traced into a file instead of the console? |
| 21:55.57 | SerajewelKS | i want to collect some traces for a mailing list post |
| 21:56.17 | p3nguin | How are you connected to the asterisk cli? |
| 21:56.51 | SerajewelKS | invoked from a shell inside a screen session attached from an ssh session |
| 21:58.24 | SerajewelKS | i could invoke with |tee but that seems a bit dirty, and prompts will be mixed in with the output |
| 21:58.35 | p3nguin | You aren't using PuTTY? |
| 21:58.44 | SerajewelKS | i am |
| 21:58.57 | p3nguin | Good. Use PuTTY's logging feature. |
| 21:59.12 | SerajewelKS | that will log the prompts as well, yes? |
| 21:59.35 | p3nguin | It will log everything that you see on the console as you see it. |
| 21:59.48 | SerajewelKS | and i've had some pretty weird behavior from putty's built-in logging when using screen, since screen does some unconventional things. |
| 22:00.05 | p3nguin | So get your thing like you want it, then enable logging, then do your tests, then turn off logging. |
| 22:00.35 | p3nguin | If it gives you trouble, detach from screen, then run asterisk -r from outside of the screen session. |
| 22:04.25 | jim_ec2 | tmux > * |
| 22:05.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:05.11 | jim_ec2 | lol not asterisk, just all other terminal multiplexers out there |
| 22:05.12 | p3nguin | I missed the part where we started a debate about that. |
| 22:06.32 | *** part/#asterisk MrNemus (~lbuttars@c-76-27-99-15.hsd1.ut.comcast.net) |
| 22:06.34 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 22:08.37 | jim_ec2 | sounded like you were contemplating shells/terminal multiplexers |
| 22:08.40 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 22:09.50 | SerajewelKS | jim_ec2: i use multiplexers, but that doesn't mean that i'm going to drop everything and switch to a different one because some guy on IRC claims that his is better |
| 22:10.13 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:10.13 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 22:11.01 | jim_ec2 | w/e just a suggestion |
| 22:12.19 | jim_ec2 | actually checked tmux's manual and i don't see any logging options so maybe it's not an alternative for this use case |
| 22:12.37 | SerajewelKS | screen has logging, it just logs control characters as well |
| 22:13.17 | jim_ec2 | except for the capture-pane function might accomplish that |
| 22:13.49 | SerajewelKS | p3nguin: and putty's logging feature does not handle escape characters properly either |
| 22:14.21 | *** join/#asterisk libryder (~david@209.33.214.243) |
| 22:15.08 | libryder | any suggestions on how to test a sip trunk without making a call via the agi? |
| 22:15.21 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 22:16.18 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 22:18.08 | [TK]D-Fender | libryder: First, AGI doesn't "make calls" |
| 22:18.55 | [TK]D-Fender | libryder: And "test trunk" = "place a call". |
| 22:25.44 | *** join/#asterisk dj_hamsta (~werwer@unaffiliated/dj-hamsta/x-2342346) |
| 22:26.36 | dj_hamsta | any one encountered problems with the spa942 not working with more than one extention ? |
| 22:36.47 | libryder | sorry, i meant test a sip trunk using the agi |
| 22:37.00 | libryder | without physically placing a call. |
| 22:37.11 | [TK]D-Fender | Again, AGI doesn't place calls... |
| 22:37.14 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 22:37.20 | [TK]D-Fender | AGI is a a dialplan app issued FROM a call |
| 22:37.38 | p3nguin | You can't make a call without making a call. |
| 22:38.00 | libryder | you can't test connectivity to the sip server from asterisk? |
| 22:38.53 | p3nguin | First of all, SIP is UDP, so there is no connection to be had. |
| 22:39.06 | [TK]D-Fender | AGI isn't the tool <- |
| 22:39.19 | [TK]D-Fender | You aren't talking about the right thing |
| 22:39.24 | p3nguin | You can enable qualify for the peer to see that there is communication, but you can't actually test your service without making a call. |
| 22:39.37 | p3nguin | And you don't have to use a phone to initiate the call, either. |
| 22:39.46 | WIMPy | p3nguin: Are you sure, that "connectivity" implies a connection? |
| 22:39.50 | [TK]D-Fender | It's like asking where the gas cap is on a bicycle. |
| 22:39.58 | libryder | no, it isn't |
| 22:40.00 | p3nguin | wimpy: What do you think connectivity means? |
| 22:40.38 | WIMPy | p3nguin: I's understand it as the possibility fpr (successfull) cummunication. |
| 22:41.20 | libryder | 3. Writing a small app that simply interrogates those interfaces that are important to the operation (iax2/udp, sip/udp, etc, send a crafted pkt and interpret the returned result. Port not open is obvious, no response is obvious, incorrect response is not so obvious) |
| 22:41.21 | p3nguin | I covered a way to ensure communication. I'm sure he still hasn't done it, though. |
| 22:41.23 | libryder | http://www.voip-info.org/wiki/view/Asterisk+monitoring |
| 22:42.04 | libryder | i'm using adhearsion to go through our inventory to initiate a call and log it's successful path back to adhearsion |
| 22:42.25 | libryder | i'm also looking for a basic way to ensure that our sip trunk is up |
| 22:43.00 | [TK]D-Fender | libryder: "sip show registry", "sip show peers". There. |
| 22:43.09 | WIMPy | libryder: Do as p3nguin suggested and turn on qualify. |
| 22:43.22 | libryder | WIMPy: thanks i'll look at that too |
| 22:43.33 | [TK]D-Fender | "sip show peers" lists the qualify time |
| 22:43.52 | p3nguin | The qualify sends an OPTIONS packet to the peer. The peer then responds to the packet, often with a negative response. |
| 22:44.03 | p3nguin | If you get a response, there is communication. |
| 22:44.32 | libryder | 19 sip peers [Monitored: 0 online, 0 offline Unmonitored: 19 online, 0 offline] |
| 22:44.45 | p3nguin | You have zero with qualify enabled. |
| 22:44.58 | p3nguin | But you have 19 out of 19 online. That's good. |
| 22:45.15 | libryder | i'll enable qualify on those, thanks |
| 22:45.20 | [TK]D-Fender | 19 defined that is. |
| 22:45.26 | WIMPy | which doesn;t mean much without qualify. |
| 22:45.27 | [TK]D-Fender | "online" is debatable |
| 22:45.36 | p3nguin | You really don't need to qualify your ITSP, though. |
| 22:45.52 | libryder | without qualify on, what does online mean? |
| 22:45.57 | [TK]D-Fender | nothing |
| 22:46.00 | WIMPy | Defined? They must have registered. But it may have been any time. |
| 22:46.00 | [TK]D-Fender | ^ |
| 22:46.02 | p3nguin | If you qualify them, that doesn't force it to be available. |
| 22:46.13 | [TK]D-Fender | WIMPy: Peers do not register. |
| 22:46.18 | p3nguin | Mine do. |
| 22:47.02 | [TK]D-Fender | That's a neat trick... |
| 22:47.13 | p3nguin | It's pretty standard for most phones. |
| 22:47.24 | [TK]D-Fender | No, I mean * sip.conf peers <- |
| 22:47.28 | p3nguin | Me too. |
| 22:47.32 | [TK]D-Fender | [iamapeer] |
| 22:47.41 | p3nguin | Right. My phone is named iamapeer. |
| 22:47.46 | p3nguin | It registers. |
| 22:47.53 | [TK]D-Fender | other way. |
| 22:47.59 | [TK]D-Fender | I'm alking peer to ITSP |
| 22:48.04 | p3nguin | Asterisk never registers to my phones. |
| 22:48.09 | [TK]D-Fender | where * REGSISTERS' |
| 22:48.22 | [TK]D-Fender | Ok, if you're stuck on phones, I'm on the other side... |
| 22:49.03 | p3nguin | You're saying peers don't register, and I'm saying my peers do register. In this case, the peers which are registering are phones. |
| 22:49.40 | [TK]D-Fender | and asterisk SIP peer does not register OUT. Please realize the direction I'm talking about |
| 22:49.53 | [TK]D-Fender | I'm not talking about a PHONE registering IN to an * peer. |
| 22:49.55 | libryder | what's the downfall of using qualify? |
| 22:49.56 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 22:50.03 | [TK]D-Fender | I'm talking about a peer as you'd use for an ITSP. |
| 22:50.12 | p3nguin | Extra packets, I guess. |
| 22:50.29 | [TK]D-Fender | libryder: If qualify fails * won't even try to send calls to that peer. |
| 22:50.43 | [TK]D-Fender | And you'd better have your failover ready. |
| 22:50.49 | p3nguin | That could be used for failover. |
| 22:50.56 | [TK]D-Fender | Used to speed it up anyway |
| 22:50.59 | libryder | yeah it seems that would be preferred |
| 22:51.10 | [TK]D-Fender | depends on what you have |
| 22:51.10 | p3nguin | But ChanIsAvail could be used for failover just the same without using qualify, I'd guess. |
| 22:51.22 | [TK]D-Fender | Um.. not really |
| 22:51.26 | p3nguin | Why not? |
| 22:51.36 | WIMPy | How could ChanIsAvail work without qualify? |
| 22:51.46 | [TK]D-Fender | Doesn't prove any actual call will go through. |
| 22:51.48 | p3nguin | If the channel is not available, don't try to dial it. |
| 22:52.12 | WIMPy | How do you find out if it is available? |
| 22:52.17 | p3nguin | ChanIsAvail() |
| 22:52.41 | WIMPy | Do you think ChanIsAvail makes a test call? |
| 22:52.46 | WIMPy | I don't. |
| 22:52.50 | p3nguin | Of course not. |
| 22:52.59 | p3nguin | But it does poll the channel. |
| 22:53.08 | libryder | if qualify fails then would it matter if * didn't try to send the call to that peer? unless it was wrong? |
| 22:53.11 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-gzntpadneguoaaht) |
| 22:53.36 | WIMPy | libryder: No, that's exactely the point. |
| 22:53.58 | WIMPy | p3nguin: And how would chan_sip know unless you enabled qualify? |
| 22:53.58 | p3nguin | If you are using qualify and a peer becomes unreachable for any reason, Dial() will fail if you try to dial that unreachable peer. |
| 22:54.12 | WIMPy | (or it is a dynamic peer that has never been seen) |
| 22:54.36 | p3nguin | wimpy: I don't know. Test it and get back to me. |
| 22:56.02 | WIMPy | I'm sure it won't find out. |
| 22:56.32 | libryder | i LOVE it! thanks guys |
| 22:59.43 | libryder | i'm *starting* to get the hang of asterisk |
| 23:00.07 | WIMPy | You're doomed. |
| 23:00.16 | libryder | doomed for failure? lol |
| 23:00.37 | *** join/#asterisk dj_hamsta (~werwer@unaffiliated/dj-hamsta/x-2342346) |
| 23:00.42 | dj_hamsta | any one encountered problems with the spa942 not working with more than one extention ? |
| 23:02.56 | libryder | 19 sip peers [Monitored: 14 online, 4 offline Unmonitored: 1 online, 0 offline] woot |
| 23:03.17 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:03.17 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 23:05.00 | p3nguin | I don't know how it is doing it, but when the phone is available, ChanIsAvail returns information on the available channel; when the phone is not available, ChanIsAvail returns a null value for the available channel. |
| 23:06.09 | p3nguin | The phone is not using qualify, by the way. |
| 23:07.41 | WIMPy | You are saying that ChanIsAvail will tell you is someone pulled the plug? |
| 23:08.33 | *** join/#asterisk F2Knight (~ben@c-98-246-210-98.hsd1.or.comcast.net) |
| 23:11.47 | Micc | WIMPy, if your expire time is small, then within that window it will know. |
| 23:12.01 | Micc | WIMPy, if the phone doesn't reregister then its unavailable. |
| 23:12.19 | p3nguin | I tested it by pulling the network cable and immediately executed ChanIsAvail. It reported that the channel was available. |
| 23:12.31 | Micc | WIMPy, but remember the cable could unplug right after registering, then you gotta wait for it to expire before it knows. |
| 23:12.41 | WIMPy | Registration may have been a long timer ago or unavailable. |
| 23:13.06 | Micc | WIMPy, I always use 120 seconds for expire time if the device supports it. |
| 23:13.11 | p3nguin | Now I'll enable qualify and see if it responds within 1 second. |
| 23:13.52 | Micc | p3nguin, then its based on qualify frequency. |
| 23:14.00 | WIMPy | within qualifyfreq. |
| 23:14.11 | Micc | Nothing will tell you if a cable is unplugged right away. |
| 23:14.17 | p3nguin | Okay, 2 seconds. |
| 23:14.26 | p3nguin | qualify is 2000ms. |
| 23:14.46 | WIMPy | qualifyfreq, not qualify. |
| 23:14.49 | Micc | unless your qualify frequency is low, which seems like a waste of resources to me, but on a small number of devices it should be fine. |
| 23:15.00 | WIMPy | That's the maximum allowed latency (timeout). |
| 23:15.33 | bmoraca_work | qualifyfreq = 10...only way to roll! |
| 23:17.51 | p3nguin | So what is the purpose of the qualify time, then? |
| 23:18.07 | [TK]D-Fender | "how long to wait for an answer" |
| 23:18.09 | [TK]D-Fender | ^ |
| 23:18.22 | WIMPy | Somehow. |
| 23:18.26 | [TK]D-Fender | tack that on to the tail of total time since last check-in |
| 23:18.36 | WIMPy | It will wait longer, but the peer would becom "lagged". |
| 23:18.36 | p3nguin | Okay, I got it. |
| 23:18.49 | bmoraca_work | yes, qualify time is not how often to send, it's how long to wait...and it expires after some multiple of qualify |
| 23:18.59 | bmoraca_work | qualifyfreq is how often to send qualifies |
| 23:19.07 | bmoraca_work | i have a backport for 1.4.36 if you need it |
| 23:19.31 | WIMPy | looks like I don;t get ChanIsAvail to work at all. I always get AVAILSTATUS=0. |
| 23:19.47 | *** part/#asterisk libryder (~david@209.33.214.243) |
| 23:20.02 | p3nguin | Check AVAILCHAN. |
| 23:22.05 | WIMPy | Ok, that is either set or not. |
| 23:22.23 | bmoraca_work | it's a list of avaialble channels from your ChanIsAvail call |
| 23:22.31 | WIMPy | But shouldn't AVAILSTATUS contain something that changes? |
| 23:23.17 | SerajewelKS | [TK]D-Fender: should you be interested, i've posted on the mailing list about this problem, and i attached the scrubbed sip debug log: http://lists.digium.com/pipermail/asterisk-users/2012-January/269348.html |
| 23:23.56 | p3nguin | If the channel is not available, you'll get a different AVAILSTATUS. |
| 23:24.01 | bmoraca_work | WIMPy: check "core show application chanisavail" for a description of the variables |
| 23:24.09 | p3nguin | For example, I got a 27 on one test. |
| 23:24.27 | *** join/#asterisk Sean-Der (~Sean-Der@NW1-DSL-74-215-64-154.fuse.net) |
| 23:24.48 | moos3 | anyone have pointers for running asterisk 1.8 in ec2 ? |
| 23:25.12 | WIMPy | No, AVAILSTATUS always gives 0. The other variables change. |
| 23:25.13 | [TK]D-Fender | SerajewelKS: Well none of us got to see complete SIP debug or your actual ful configs masking only PW's and that post is missing most of that as well... |
| 23:25.26 | Sean-Der | What is the function that echo's to stdout? |
| 23:25.38 | SerajewelKS | [TK]D-Fender: the post does contain SIP debug logs |
| 23:25.41 | [TK]D-Fender | SerajewelKS: As well intentioned as it is you come off looking like a "secret squirrel" as they say.... |
| 23:26.24 | [TK]D-Fender | SerajewelKS: Where is it? |
| 23:26.25 | p3nguin | sean-der: There's no function that I know of, but there is an application that prints on the CLI. See: Verbose(). |
| 23:26.31 | SerajewelKS | [TK]D-Fender: attached to the message |
| 23:26.35 | [TK]D-Fender | SerajewelKS: I don't see it in the raw text, or some link yet... |
| 23:26.51 | [TK]D-Fender | Ah I think I may have just found it... |
| 23:27.07 | [TK]D-Fender | .bin? what kind of capture is this? |
| 23:27.30 | SerajewelKS | [TK]D-Fender: mailman does whatever the hell it wants to with attachments |
| 23:27.35 | Sean-Der | printf to console, exactly what I was looking for. Thanks p3nguin |
| 23:27.36 | p3nguin | An annoying one, I'd imagine. |
| 23:27.53 | SerajewelKS | [TK]D-Fender: see where it says "Name: sip-debug.log" on the message page? that's what i called it. |
| 23:28.18 | [TK]D-Fender | Ok, just a temp name, all good.. |
| 23:28.28 | SerajewelKS | [TK]D-Fender: attachment.bin is the filename in the URL, and presumably the mailing list web interface isn't setting a content-disposition header, so your browser goes with the .bin name. :) |
| 23:28.32 | [TK]D-Fender | Just how the browser interpreted the link. |
| 23:29.00 | [TK]D-Fender | Unfortunately still don't see the [general] section as a point of reference, but this is a still a good start... |
| 23:29.31 | [TK]D-Fender | SerajewelKS: and rather than just hit the mailing list I'd have submitted this on the presise of the bug it might appear to be. This would be on the tracker |
| 23:29.49 | SerajewelKS | [TK]D-Fender: i may add that later. i will have to trim down all the comments and whatnot, and i didn't want to spend the time just yet. if asked on the thread i'll post my config. |
| 23:30.51 | [TK]D-Fender | SerajewelKS: Well you've hit a better spot on the "thorough" scal so the odds of something coming out of this are decent |
| 23:30.58 | SerajewelKS | [TK]D-Fender: hmm, usually i don't file bugs unless i'm pretty certain it's a bug. i find projects are happier if i ask about it on the mailing list first, then go to the tracker if they indicate that it's probably a bug. people seem to be happer to direct you to the bug tracker from the list than to close your bug NOTABUG. |
| 23:31.25 | SerajewelKS | (in other words, i don't like using bug trackers for support) |
| 23:31.44 | [TK]D-Fender | SerajewelKS: So far its not a bad thing with this project |
| 23:31.48 | SerajewelKS | and especially i don't like using bug trackers the first day i use some software as mature as asterisk :) |
| 23:32.01 | citywok | SerajewelKS: just be careful or leifmadsen will body slam you if you submit too many non-bugs to the tracker |
| 23:32.20 | leifmadsen | citywok: not my job anymore :) |
| 23:32.28 | SerajewelKS | citywok: exactly. "hey, this is a bug and should be reported" is a better way to proceed in my opinion than "this isn't a bug, go ask the mailing list." |
| 23:32.33 | citywok | haha i see the @ on your name still ;) |
| 23:32.36 | WIMPy | did I read "mature"? |
| 23:32.49 | leifmadsen | citywok: well I'm still kind of a big deal |
| 23:32.54 | [TK]D-Fender | citywok: Not all ops are employed by Digium |
| 23:33.12 | SerajewelKS | WIMPy: in terms of stability and age, not the character of the community ;) |
| 23:33.14 | citywok | leifmadsen: that implies you ever WERE a big deal... *BURN**** |
| 23:33.26 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 23:33.43 | WIMPy | Ok, the age may fit. |
| 23:34.11 | SerajewelKS | anyway, time to run home. highlight me or reply on the thread if you have further input. |
| 23:34.44 | leifmadsen | citywok: I AM A PILLAR OF THE COMMUNITY! NOW OBEY MY COMMANDS! |
| 23:34.52 | SerajewelKS | thanks for all the assistance and suggestions so far |
| 23:35.39 | citywok | lol whatever helps you sleep at night you cute little canuck you! |
| 23:36.13 | *** join/#asterisk schreiber1337 (b838b7e6@gateway/web/freenode/ip.184.56.183.230) |
| 23:37.32 | [TK]D-Fender | leifmadsen: Pillars are OUT! Suspension ALL the way.... FaB-U-LOUS! |
| 23:38.07 | WIMPy | Open plan |
| 23:38.32 | leifmadsen | citywok: <3 |
| 23:39.00 | citywok | :heart: just please don't hurt me... again |
| 23:39.12 | citywok | how's life post bugmarshal? |
| 23:47.48 | *** join/#asterisk k-man (~k-man@unaffiliated/k-man) |
| 23:54.25 | *** join/#asterisk kaushal (~kaushal@14.97.196.245) |
| 23:59.51 | leifmadsen | citywok: good so far! I'm working on a lot of things now :) |
