| 00:02.48 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 00:03.56 | *** join/#asterisk deo (~dnepangue@112.198.79.33) |
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| 01:18.03 | *** join/#asterisk grim_biker (~grimm@99-103-166-215.lightspeed.milwwi.sbcglobal.net) |
| 01:18.20 | grim_biker | yo guys and gals!!! ;-) Happy Monday! |
| 01:18.21 | grim_biker | anyone know if its possible to send a command via ssh or api to authenticate a user, not bypass bind a user by actually authenticate a user as if they accepted your splash page? |
| 01:18.36 | grim_biker | awe damn... sorry wrong message... |
| 01:18.50 | grim_biker | anyone know mixmonitor real well? |
| 01:19.44 | grim_biker | I have a problem with the recorded channels loosing time. 100% SIP system. using pseudo dahdi for timing... |
| 01:21.30 | grim_biker | hello????? anyone home?????? |
| 01:24.11 | hachque | how can I set up asterisk to register with another SIP provider, but only allow certain SIP users to exit via that peer? |
| 01:25.36 | *** part/#asterisk grim_biker (~grimm@99-103-166-215.lightspeed.milwwi.sbcglobal.net) |
| 01:25.44 | *** join/#asterisk Linuturk_ (~linuturk@unaffiliated/linuturk) |
| 01:25.55 | *** join/#asterisk grim_biker (~grimm@99-103-166-215.lightspeed.milwwi.sbcglobal.net) |
| 01:26.49 | grim_biker | still here hachque? |
| 01:26.56 | hachque | yes |
| 01:27.21 | grim_biker | ok |
| 01:27.30 | grim_biker | users are assigned to context. |
| 01:27.40 | grim_biker | context control what they can and can not do. |
| 01:27.56 | grim_biker | i.e. user at extension 501 would look like this? |
| 01:28.01 | grim_biker | [501] |
| 01:28.10 | grim_biker | fullname = john smith |
| 01:28.18 | hachque | I have my users set up in sip.conf |
| 01:28.28 | grim_biker | context = limited_user |
| 01:28.34 | grim_biker | right, this would be in sip.conf |
| 01:28.41 | hachque | oh okay |
| 01:28.49 | hachque | i know there's another file users.conf which has those other options in it |
| 01:28.56 | hachque | http://hastebin.com/gofeqahiye.ini |
| 01:29.34 | grim_biker | ok, just checked your file. |
| 01:29.47 | grim_biker | right, user is in "users" context |
| 01:30.07 | grim_biker | wait a moment. let me put together an example. |
| 01:33.19 | grim_biker | in extensions.conf |
| 01:33.19 | grim_biker | [users] |
| 01:33.19 | grim_biker | include = default |
| 01:33.19 | grim_biker | include = speed_dials |
| 01:33.19 | grim_biker | include = park_context |
| 01:33.19 | grim_biker | include = features |
| 01:33.21 | grim_biker | include = dialout_provider_b |
| 01:33.23 | grim_biker | . |
| 01:33.31 | grim_biker | [dialout_provider_a] |
| 01:33.33 | grim_biker | exten = NXXXXXXX,1,Dial(SIP/voip-stream/${EXTEN}) |
| 01:33.35 | grim_biker | [dialout_provider_b] |
| 01:33.36 | hachque | uh |
| 01:33.37 | grim_biker | exten = NXXXXXXX,1,Dial(SIP/voip-logic/${EXTEN}) |
| 01:33.38 | hachque | pastebin it :P |
| 01:33.39 | grim_biker | . |
| 01:33.45 | hachque | hastebin.com |
| 01:35.17 | grim_biker | now as you see here, your user is in the users context, so the user can only dial out over provider b. not provider a. since provider a dialout is not included in the users context |
| 01:35.21 | grim_biker | make sense? |
| 01:35.32 | hachque | yeah |
| 01:35.37 | hachque | but how do i connect that up to the register declaration? |
| 01:35.47 | grim_biker | you dont |
| 01:36.11 | hachque | well then how do i prevent the users from dialing out through there? |
| 01:36.13 | grim_biker | the register declaration is simply to register with the sip provider, not to control who uses the channe. |
| 01:36.23 | hachque | the other SIP server can call any number since it's connected to the telephone system |
| 01:36.47 | hachque | okay, then what connects dialout_provider_b to the register? |
| 01:37.48 | hachque | say I dial 01234678 which we'll pretend is a valid phone number |
| 01:37.50 | grim_biker | in my example voip-stream and voip-logic are registration identifiers. |
| 01:37.50 | hachque | from my voip client |
| 01:37.55 | hachque | okay okay |
| 01:37.59 | hachque | *oh okay |
| 01:38.09 | hachque | so it would be |
| 01:38.10 | grim_biker | so if you run sip show peers |
| 01:38.30 | grim_biker | it will show the registration identifier, use that instead of voip-logic |
| 01:38.31 | hachque | register => user:pass@voip-stream/ |
| 01:38.40 | hachque | oh okay |
| 01:38.43 | hachque | thanks |
| 01:38.46 | hachque | i'll give this a shot |
| 01:38.50 | grim_biker | k |
| 01:39.04 | grim_biker | good luck. if it doesn't work, you can email me |
| 01:41.02 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 01:47.23 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 01:49.15 | [TK]D-Fender | Registration has nothing to do with placing calls out |
| 01:49.45 | [TK]D-Fender | ~sipregister |
| 01:49.46 | infobot | [~sipregister] SIP registration is to tell your provider what IP address & EXTEN to send INCOMING calls to. Some ITSPs let you use a fixed address or host rather than registering. Registration is NOT normally needed to PLACE calls, as those are typically auth'ed independently. Others accept unauth'ed calls once you are registered (saves on negotiation BW). |
| 01:49.49 | [TK]D-Fender | ^^ |
| 01:50.18 | [TK]D-Fender | And once a provider does send you call... it has nothing to do with processing the call that comes in. That is done by your peer entries |
| 01:50.30 | [TK]D-Fender | Which also specify what context the calls will fall in. |
| 01:53.58 | hachque | is there a way to tell asterisk that all calls and operations should be forwarded onto another SIP server? |
| 01:54.24 | hachque | when I dial any number on the client, I need it to go through the asterisk server to another SIP server that it's registered with |
| 01:56.24 | jpsharp | Dialplan. |
| 01:56.36 | hachque | so just XXXXXXXX,1, Dial(blah) right? |
| 01:56.38 | hachque | to match everything? |
| 01:56.40 | jpsharp | Yep. |
| 01:56.45 | hachque | awesome |
| 01:56.53 | hachque | thanks |
| 01:57.01 | jpsharp | exten => _X.,1,Dial(SIP/${EXTEN}@your.other.sip.server.com) |
| 01:58.41 | hachque | will that match **1 ? |
| 01:59.06 | [TK]D-Fender | Of course not |
| 01:59.30 | hachque | well I don't know the matching rules for dialplan |
| 01:59.37 | hachque | so I'm unsure what exactly N and X match |
| 01:59.44 | [TK]D-Fender | Go read the Dialplan basics chapter.... |
| 01:59.46 | [TK]D-Fender | ~book |
| 01:59.46 | 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 |
| 02:00.15 | hachque | oh that's handy |
| 02:00.23 | hachque | i've just been using the asterisk wiki which is rather incomplete |
| 02:01.36 | hachque | and the website is down :P |
| 02:03.09 | hachque | it seems to have the full thing on google books anyway |
| 02:03.10 | hachque | thanks :) |
| 02:06.39 | leifmadsen | ya, host is down |
| 02:06.58 | leifmadsen | my own website and such is down with it; thankfully google caches that stuff |
| 02:07.42 | hachque | hmm |
| 02:07.51 | hachque | is there any special value that will match *? |
| 02:08.01 | hachque | other than entering **X. as well? |
| 02:08.37 | leifmadsen | nope |
| 02:08.44 | hachque | oh okay |
| 02:08.47 | leifmadsen | only X, N, Z, . and ! are special |
| 02:08.56 | leifmadsen | well.... ! and . match * as well |
| 02:09.11 | leifmadsen | but that's 0 or more and 1 or more characters (respectively) |
| 02:09.26 | leifmadsen | doing _.X for example would make no sense |
| 02:09.31 | hachque | so _. would match everything then? |
| 02:09.32 | leifmadsen | the X would be superfluous |
| 02:09.41 | leifmadsen | of 1 or more characters in length, yes |
| 02:09.46 | leifmadsen | _. is not recommended |
| 02:10.01 | hachque | because it matches everything? |
| 02:10.09 | leifmadsen | you should do something like _[*0-9]. instead |
| 02:10.10 | leifmadsen | yes |
| 02:10.24 | jpsharp | _. matches "h", doesn't it? |
| 02:10.28 | leifmadsen | yes that too |
| 02:10.30 | leifmadsen | everything |
| 02:10.30 | hachque | hmm well in this case I do want to match everything |
| 02:10.37 | leifmadsen | the main reason being, someone could perform a string injection |
| 02:10.39 | hachque | since the asterisk server doesn't know about anything |
| 02:10.49 | leifmadsen | you should always filter |
| 02:10.57 | leifmadsen | match what you expect, not match everything |
| 02:11.01 | hachque | yeah but the asterisk server can only be used by another asterisk server, permitted by IP address |
| 02:11.07 | leifmadsen | otherwise, be very very very careful about how you construct your dialplan |
| 02:11.20 | jpsharp | Rule #1: Never trust external input. |
| 02:11.20 | leifmadsen | until that box is compromised |
| 02:11.30 | leifmadsen | Rule #2: DON'T TALK ABOUT FIGHT CLUB |
| 02:11.36 | leifmadsen | errr... see Rule #1 |
| 02:11.38 | hachque | if that box is compromised, there are much bigger problems :P |
| 02:11.50 | leifmadsen | ya, like 2 compromised boxes now |
| 02:11.53 | leifmadsen | anyways, don't use _. |
| 02:11.54 | leifmadsen | it's wrong |
| 02:12.59 | leifmadsen | bare minimum is: _[0-9A-Za-z].,1,Set(thisRequest=${FILTER(0-9A-Za-z,${EXTEN})}) |
| 02:13.48 | leifmadsen | if you don't care about letters, then only allow 0-9 characters |
| 02:13.55 | hachque | hmm okay |
| 02:14.19 | hachque | I don't need letters; this asterisk server is simply acting as a gateway to the actual SIP server that dials out to telephony systems |
| 02:14.26 | leifmadsen | http://svn.asterisk.org/svn/asterisk/branches/11/README-SERIOUSLY.bestpractices.txt |
| 02:14.29 | leifmadsen | then filter out letters |
| 02:14.32 | leifmadsen | only allow 0-9 |
| 02:14.35 | leifmadsen | runs off to bed |
| 02:16.12 | *** join/#asterisk deo (~dnepangue@203.177.214.75) |
| 02:18.37 | grim_biker | hachque - voip-info.org |
| 02:31.49 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 03:35.45 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 03:43.02 | *** join/#asterisk dandate2 (~dan@180.190.212.189) |
| 03:43.35 | dandate2 | would going wideband codec improve audio quality at all when talking to US PSTN customers? |
| 03:46.37 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
| 03:47.15 | wasanzy | hi guys |
| 03:47.50 | [TK]D-Fender | dandate2: no, because the PSTN is NOT wideband |
| 03:48.21 | wasanzy | please I want to build VOIP using asterisk and ip phone, do I need a service provider? |
| 03:48.43 | jpsharp | Only if you want to talk to the outside world. |
| 03:48.50 | [TK]D-Fender | wasanzy: Need? No. Want? You tell us. |
| 03:49.18 | [TK]D-Fender | wasanzy: Asterisk can talk to a wide variety of interfaces & protocols |
| 03:49.27 | [TK]D-Fender | wasanzy: What you use if for is up to you. |
| 03:49.40 | [TK]D-Fender | wasanzy: I use mine as a jukebox and coffee-timer. |
| 03:50.07 | dandate2 | what about wide-band headsets? do these provide any benefit over PSTN? |
| 03:50.39 | [TK]D-Fender | ... |
| 03:50.57 | dandate2 | =) |
| 03:51.04 | wasanzy | I just want an IP Phone and asterisk, to be able to make both internal and external calls with out a service provider (I mean no telephone number from telecom) |
| 03:51.09 | [TK]D-Fender | dandate2: What part of PSTN != Wideband are you having trouble with? |
| 03:51.27 | [TK]D-Fender | wasanzy: Well you now say you WANT to use a service provider.... |
| 03:51.29 | dandate2 | yeah i get that, i guess i lack understanding of why they want more money for wideband headsets |
| 03:51.39 | [TK]D-Fender | wasanzy: And you walked in asking if you NEED one. |
| 03:51.57 | [TK]D-Fender | dandate2: Because the other models are so much WORSE |
| 03:52.10 | wasanzy | may be my question is not clear |
| 03:52.16 | [TK]D-Fender | dandate2: and that is just audio TO the phone. |
| 03:52.37 | jpsharp | If you're not talking to your spouse across the world in 192Khz wideband stereo, you just don't love them. |
| 03:52.40 | [TK]D-Fender | wasanzy: You asked if you need one. YOU apparently do. Others don't. |
| 03:53.17 | [TK]D-Fender | jpsharp: If you've put half a planet between you, whats a few more kilobits? ;) |
| 03:53.22 | dandate2 | so you guys think wideband headsets would be a significant improvement despite using ulaw |
| 03:53.44 | [TK]D-Fender | dandate2: And where do you get that the heatset is speaking ulaw? |
| 03:54.02 | jpsharp | ulaw = so not widebang. |
| 03:54.05 | jpsharp | or wideband either. |
| 03:54.06 | [TK]D-Fender | !!! |
| 03:54.23 | dandate2 | well the phone is transmitting ulaw, we would consider replacing our narrowband headsets with wideband headsets if it would improve things |
| 03:54.50 | [TK]D-Fender | dandate2: If you are using shitty headset then yes, better headset should clearly make a difference |
| 03:54.56 | wasanzy | what I mean is, I wanted to know if it is possible with out acquiring a telephone number and all that from a service provider. so what I want to do is build a telephone system with IP Phone and asterisk, well if just asterisk will be fine then no problem but I don't want to deal with any service provider. |
| 03:55.27 | jpsharp | YOu can build yourself a telephone network where you can talk to just yourself or anyone else connected directly to your Asterisk server. |
| 03:55.34 | dandate2 | well they aren't necessarily shitty, theres more expensive models however but I can't figure out why except that they are "wideband" |
| 03:55.42 | [TK]D-Fender | wasanzy: Yuo can use * with whatever you want. If you just want to use phones in a building to call each other then you don't anything more |
| 03:55.46 | jpsharp | If you want to talk to the world, you need a service provider. |
| 03:55.53 | jpsharp | dandate2: Marketing |
| 03:56.02 | [TK]D-Fender | jpsharp: no, not just mnarketing |
| 03:56.40 | wasanzy | ok thank you all. |
| 03:57.01 | [TK]D-Fender | jpsharp: When you consider the companding & other compression certain wireless headsets use and then add the A/D conversion on the phone itself the impact can be very noticable |
| 03:57.15 | jpsharp | Oh, I missed the part about wireless. |
| 03:57.21 | jpsharp | dines on crow. |
| 03:57.54 | jpsharp | With a side of size 13 foot. |
| 03:58.34 | [TK]D-Fender | jpsharp: I would recommend a light honey-dijon for that. |
| 03:58.53 | jpsharp | mmmm honey |
| 04:00.17 | wasanzy | and sorry, what type of card will u recommend for the machine going to be connected to the ip phone? |
| 04:01.08 | [TK]D-Fender | wasanzy: a NETWORK card |
| 04:01.17 | [TK]D-Fender | wasanzy: I'd say at least 100bt |
| 04:01.54 | wasanzy | you mean the normal network card that accepts RJ45? |
| 04:02.42 | [TK]D-Fender | wasanzy: I'm getting the impression the term "IP phone" hasn't quite registered with you yet |
| 04:03.04 | [TK]D-Fender | wasanzy: IP phone ... as in TCP/IPP as in boring network device jsut like any other PC, printer, etc. |
| 04:03.26 | [TK]D-Fender | Asterisk will talk to it over your network interface |
| 04:03.40 | wasanzy | ok |
| 04:03.49 | [TK]D-Fender | wasanzy: Your server WILL have a network card won't it? Hard to find a motherboard that doesn't .... |
| 04:04.19 | wasanzy | you are right |
| 04:04.35 | [TK]D-Fender | wasanzy: The point of IP phones is not needing special cards. Get a big swith. Plug in 500 phones. |
| 04:05.01 | [TK]D-Fender | wasanzy: Or if youre crazy, get phones with a passthrough port and daisy-chain them all without a switch |
| 04:05.16 | jpsharp | twitches. |
| 04:06.27 | wasanzy | so with this two things, IP Phone and asterisk, I can build an IVR system as well right? or I need more than just IP Phone and asterisk? |
| 04:06.42 | [TK]D-Fender | wasanzy: Yes you can make IVRs in Asterisk |
| 04:07.07 | [TK]D-Fender | wasanzy: IP phones are not required. But they are nice and flexible. |
| 04:07.56 | wasanzy | but with out the IP Phones, how do the outside worlds reach the asterisk server? |
| 04:08.15 | jpsharp | Through a service provider. |
| 04:08.23 | [TK]D-Fender | wasanzy: First, IP phones have nothing to do with the outside world |
| 04:08.37 | [TK]D-Fender | wasanzy: You IP phone is just an IP phone. It talks to whatever you set it up with |
| 04:08.56 | [TK]D-Fender | wasanzy: It does not magically have anything to do with the outside world by design. |
| 04:09.15 | [TK]D-Fender | wasanzy: Asterisk can talk with your IP phones. |
| 04:09.37 | [TK]D-Fender | wasanzy: Asterisk can talk with PHYSICAL phone lines via a variety of hardware interfaces. |
| 04:09.46 | [TK]D-Fender | wasanzy: Asterisk can talk to ITSP's |
| 04:09.50 | [TK]D-Fender | ~itsp |
| 04:09.50 | 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. |
| 04:10.02 | [TK]D-Fender | wasanzy: the combination YOU want to use it up to you |
| 04:10.15 | [TK]D-Fender | wasanzy: None of the pieces has any dependency on the other |
| 04:10.52 | [TK]D-Fender | wasanzy: I can use ANALOG phones and a ITSP. I could use IP phones and ANALOG POTS phone lines. Or whatever. |
| 04:13.06 | wasanzy | "Asterisk can talk with PHYSICAL phone lines via a variety of hardware interfaces" is my point. because considering this you can see IP phone is more flexible than the physical phone line which will require a special card for the asterisk server |
| 04:13.30 | [TK]D-Fender | wasanzy: You just compared an IP phone... to a an analog phone LINE |
| 04:13.49 | [TK]D-Fender | wasanzy: a phone is not a line You just compared a car to the ROAD you drive it on |
| 04:15.15 | [TK]D-Fender | wasanzy: as for analog phones and lines, there are also gatway boxes that convert them to VoIP and can then send to your Asterisk server so that you dont a special card in the server. |
| 04:15.23 | [TK]D-Fender | don't need* |
| 04:15.42 | [TK]D-Fender | ~ata |
| 04:15.42 | infobot | well, ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 04:16.26 | wasanzy | I think I need a lot of reading here. but the line is from the service provider to the phone right? |
| 04:16.30 | [TK]D-Fender | wasanzy: sample : http://www.telephonydepot.com/Catalog/Cisco-Analog-Adapters/Cisco-SPA112 |
| 04:16.47 | [TK]D-Fender | wasanzy: Depends how you use the term "line" |
| 04:17.10 | [TK]D-Fender | wasanzy: ITSP = INTERNET telephony service provider. Comes in over the INTERNET using a VoIP protocol. |
| 04:17.45 | [TK]D-Fender | wasanzy: a PHYSICAL line comes from a traditional telco and is a copper JACK sitting on your wall like we've had for the past century. |
| 04:19.55 | wasanzy | right, but both are good to choose between depending on myself right? |
| 04:20.37 | jpsharp | Depends on what you want to do. Asterisk is the Swiss Army Knife of telephony. |
| 04:21.47 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 04:21.51 | [TK]D-Fender | wasanzy: Everything depends on you. What option you have in your area. What serivece you already have. Budget. Expansion requirements. |
| 04:22.25 | [TK]D-Fender | wasanzy: Reliability & quality concerns. |
| 04:23.52 | wasanzy | I think I have already stated what I really seek to achieve but I will think of this in more details and get back with questions, because as at now two things are involve which am getting confuse about. |
| 04:24.07 | [TK]D-Fender | wasanzy: Which 2 things? |
| 04:24.58 | [TK]D-Fender | wasanzy: There are typically 2 sides to this: A: The phone you want to hold in your hands. and B: What kind of line or service you want to use to connect to the outside world. |
| 04:26.14 | [TK]D-Fender | wasanzy: IP phones are typically a good choice.. If you have netowrking then you're already ready to go on that side. If you're in an old building with lots of single-pair boring analog wiring then maybe IP phones aren't the most realistic option for you. Like I said, it depends. |
| 04:27.12 | wasanzy | this part is getting me confuse a little, but I think am clear now. |
| 04:32.28 | wasanzy | 1. With VOIP for internal use, I don't need service provider nor any phone but just an asterisk 2. for external calls etc, I need a service provider. is that all? |
| 04:33.51 | [TK]D-Fender | wasanzy: If you jsut want a pile of IP phones to call between each other inside a building then no, you don't need some outside provider |
| 04:35.12 | wasanzy | I initially thought just one IP phone can do the job? do you have any idea of ss7? |
| 04:36.07 | [TK]D-Fender | wasanzy: wasanzy You only need 1 phone? We have no idea what you ar actually trying to accomplish yet... |
| 04:37.00 | [TK]D-Fender | wasanzy: SS7 is just another signalling protocol to the PSTN. Typically delivered over T1/E1 and larger links |
| 04:38.58 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 04:51.51 | *** join/#asterisk bipul (~vipul@unaffiliated/bipul) |
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| 05:25.55 | hariom | I was wondering what is the difference between AGI and EAGI? What is that one can easily do in EAGI but will be little efforts in AGI? What type of work will suit to EAGI compared to AGI? |
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| 05:45.02 | hariom | I was wondering what is the difference between AGI and EAGI? What is that one can easily do in EAGI but will be little efforts in AGI? What type of work will suit to EAGI compared to AGI? |
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| 06:01.50 | hariom | Has anybody worked on EAGI? I was wondering what is the difference between AGI and EAGI? What is that one can easily do in EAGI but will be little efforts in AGI? What type of work will suit to EAGI compared to AGI? |
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| 06:11.59 | nunne | hariom: change the audio stream is the short and easy answer i think ;) |
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| 06:18.45 | hariom | nunne: Didn't get. Could you pls explain in little detail? |
| 06:19.49 | hariom | nunne: What you could do with EAGI which would be not possible or not straight forward to do using AGI? As I read, it provides access to audio stream on fd 3. But that audio we can also access in AGI after record |
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| 06:57.46 | hariom | Looks like no body wants to talk about EAGI |
| 06:58.05 | hariom | What you could do with EAGI which would be not possible or not straight forward to do using AGI? As I read, it provides access to audio stream on fd 3. But that audio we can also access in AGI after record |
| 07:12.04 | ectospasm | you do realize that most folks dealing with Asterisk are in the US, and are thus asleep... << hariom |
| 07:12.32 | ectospasm | hariom: you should wait longer than 40min to ask your question again. |
| 07:12.49 | ectospasm | ...if I knew the answer, I'd answer (-; |
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| 07:31.36 | AliRezaTaleghani | hi buddies |
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| 07:32.09 | AliRezaTaleghani | i need a guide for kernel optimizeing |
| 07:32.21 | AliRezaTaleghani | I use Ubuntu as server for asterisk |
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| 07:48.49 | hachque | if I'm behind a firewall that permits only port-80 traffic outbound, there's practically no way to run VoIP through that correct? |
| 07:49.08 | hachque | since I need both traffic on 5060 for SIP and 10000-20000 for voice |
| 07:50.19 | mirela666 | hachque: exacly |
| 07:50.28 | hachque | thought so |
| 07:50.36 | hachque | stupid university IT department >_> |
| 07:51.01 | hachque | they block all non-80 port traffic outbound |
| 07:51.10 | hachque | flips a table or two |
| 07:56.26 | kaldemar | hachque: you can use IAX, it only needs a single port. |
| 07:56.57 | hachque | interesting; what does IAX do? Is that the protocol for transmission between VoIP servers? |
| 07:57.37 | WIMPy | Yes. |
| 07:57.44 | WIMPy | But it's still UDP. |
| 07:57.57 | AliRezaTaleghani | hachque: yep, not only that, but also client registeratoin |
| 07:58.02 | AliRezaTaleghani | registration* |
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| 08:01.35 | kaldemar | hachque: it's not limited to traffic between servers. you'll just find less phones that support it. |
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| 09:10.56 | TimeRider | Benefits of ISDN in UK for running an Asterisk based PBX?? Anyone? |
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| 09:49.42 | shaiton | hi there, what could prevent a call file being sent from /var/spool/asterisk/outgoing/ ? (It was working but now my call files are stuck there). Could it be caused by a wrong system clock? (there is nothing on the logs) |
| 09:50.01 | shaiton | (good ownership, good chmod) |
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| 10:10.46 | shaiton | hum ok, now that I correctly set up the clock it works again.. |
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| 10:20.27 | mcolombo | anybody say, because asterisk send me this message ? |
| 10:20.29 | mcolombo | Invalid cause given to Hangup(): "NOANSWER" |
| 10:20.32 | mcolombo | Thanks |
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| 10:26.48 | AliRezaTaleghani | can some body give me a hint about asterisk11 |
| 10:27.01 | AliRezaTaleghani | is there any new feature for Realtime? |
| 10:31.27 | WIMPy | mcolombo: How did you call Hangup()? |
| 10:32.26 | mcolombo | WIMPy : http://pastebin.com/GqJWsRvG |
| 10:32.41 | mcolombo | this is my Dial, Macro and Hangup |
| 10:32.55 | mcolombo | exten |
| 10:33.22 | WIMPy | That's not valid. |
| 10:33.32 | WIMPy | Hangup() takes a numeric parameter. |
| 10:34.08 | WIMPy | Hangup(${HANGUPCAUSE}) would work. But no parameter should do the same. |
| 10:35.53 | mcolombo | mmm |
| 10:35.59 | mcolombo | Thanks WIMPy |
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| 11:32.32 | hariom | What you could do with EAGI which would be not possible or not straight forward to do using AGI? As I read, it provides access to audio stream on fd 3. But that audio we can also access in AGI after record |
| 11:33.35 | WIMPy | But not in realtime. |
| 11:34.08 | wdoekes | mcolombo: Hangup takes one of the causes from causes.h. Hangup(NO_ANSWER) would work |
| 11:35.23 | WIMPy | You can use names? o.O |
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| 11:39.04 | wdoekes | cause = ast_str2cause(data) |
| 11:40.06 | WIMPy | Never seen that. |
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| 12:09.22 | gavimobile | question, I know I can use the cdr options, but for testing purposes im trying to add a custom field to my cdr. I have a field in my mysql table called callerid and I added same => n,Set(CDR(callerid)=zombie pancakes) for my incoming dialplan rule, however after I initiate the call and hangup, the mysql field for callerid is blank. what am I missing here please |
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| 12:16.24 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:17.47 | oquidave | hello ...anybody used asterisk-java framework? |
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| 12:18.32 | oquidave | first am failing to decide which of these files i should download https://nexus.reucon.com/content/repositories/opensource/org/asteriskjava/asterisk-java/1.0.0-m2/ |
| 12:34.18 | AliRezaTaleghani | L-) |
| 12:34.44 | mcolombo | Wdoekes, Good! I will apply in the dialplan |
| 12:34.45 | mcolombo | Thanks |
| 12:34.56 | AliRezaTaleghani | is there any guide for kernel recompiling? sufficient/optimize it for asterisk? |
| 12:38.20 | [TK]D-Fender | AliRezaTaleghani, Only thing is to set 1000hz timer. |
| 12:39.15 | AliRezaTaleghani | [TK]D-Fender: are u sure? is it all? |
| 12:41.03 | [TK]D-Fender | AliRezaTaleghani, https://www.google.ca/#hl=en&sclient=psy-ab&q=asterisk+kernel+tuning&oq=asterisk+kernel+tuning&gs_l=hp.3..33i21.2465.11434.0.11550.26.22.2.2.2.1.436.3712.0j16j3j0j1.20.0.les%3B..0.0...1c.1.bs-eWhbMEdc&pbx=1&bav=on.2,or.r_gc.r_pw.r_cp.r_qf.&fp=bdeee7373c0f7079&biw=1600&bih=883 |
| 12:41.47 | oquidave | between Manager API and fastagi protocols, which is the most prefered for developing or scripting asterisk apps using other languages |
| 12:43.44 | WIMPy | That really depends on *what* you want to do. |
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| 12:45.56 | gavimobile | [TK]D-Fender: hey man.. could I trouble you to look at my question? |
| 12:47.45 | oquidave | WIMPy: what i want to do? what do u mean? i want to integrate my application with asterisk and i also want to do asterisk scripting using my favourite languages(java, python) |
| 12:48.30 | WIMPy | What kind of functionality you need. |
| 12:49.36 | oquidave | WIMPy: things like click-to-call..IVR for various companies/users that sort of thing |
| 12:50.32 | WIMPy | If you want to do what otherwise would be done in the dialplan, use AGI, if you need more, use AMI. |
| 12:52.26 | oquidave | WIMPy: so AMI is more powerfull and does what AGI does? is that correct |
| 12:52.46 | cusco | hi oquidave, we use asteriskjava .. im not the java programmer but we use the 'event' class to read events |
| 12:53.08 | WIMPy | No, they do not do the same, but if you can;t do it via AMI, you can't do it at all. |
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| 12:55.35 | [TK]D-Fender | oquidave, IVR & click to call are completely different. One is a thing you do IN a call.. the otehr is triggering Asterisk to do somthing while not in a call. |
| 12:55.44 | oquidave | cusco: i've been playing with asterisk diaplan mainly doing my scripting via the extension.conf file...but now since am a java and python programmer, i want to script the dialplan from there. |
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| 12:56.05 | [TK]D-Fender | oquidave, Go actually read the feature-set of each. They are not options against each other. They are conceptually nothing alike. |
| 12:56.46 | gavimobile | question, I know I can use the cdr options, but for testing purposes im trying to add a custom field to my cdr. I have a field in my mysql table called callerid and I added same => n,Set(CDR(callerid)=zombie pancakes) for my incoming dialplan rule, however after I initiate the call and hangup, the mysql field for callerid is blank. what am I missing here please |
| 12:57.03 | [TK]D-Fender | oquidave, external scripting also puts a greater load on your server and should be kept to a minimum. Go validate the actual requirement for things the dialplan doesn't do. |
| 12:59.29 | oquidave | TK]D-Fender> interesting so i've to keep external dialplan scripting to the minimum...i just read from asterisk-java that since the connection is via tcp/ip and the agi-server is running. the connection is only setup once instead of restarting the process each time, hence reducing the load on the server/asterisk |
| 13:00.13 | oquidave | through some sort of threading |
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| 13:00.40 | WIMPy | that's AGI vs fast AGI. |
| 13:01.11 | WIMPy | AMI is not call realted at all and will even only need one connect at all. |
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| 13:02.52 | WIMPy | But via AMI you can't just start an application on a channel. |
| 13:03.06 | WIMPy | You can toss your calls around in the dialplan, however. |
| 13:03.44 | [TK]D-Fender | After identifying it first... and maybe having a reason to do so |
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| 13:04.56 | WIMPy | That's the way to do AGI like stuff via AMI, if you have or want to. |
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| 13:07.02 | oquidave | WIMPy: thanks... i've aleady downloaded asterisk-java framework and it supports AMI and AGI ...reading the tutorial and trying it out to(http://www.asterisk-java.org/development/tutorial.html). |
| 13:09.31 | [TK]D-Fender | oquidave, You're reading the wrong tutorial. |
| 13:10.07 | [TK]D-Fender | oquidave, you need to lean how ASTERISK works first, not some bolt on library to interface with API's it supports. |
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| 13:10.20 | WIMPy | indeed |
| 13:10.56 | [TK]D-Fender | oquidave, You are walking in with this idea that "java is the solution" and seem to be trying to do everything you can to make the facts fit the reality you want. |
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| 13:13.36 | oquidave | TK]D-Fender i've read the asterisk book and am well conversant with the asterisk dialplan syntax...just trying something different. But am talking your advice am i'll continue raw asterisk dialplan scripting which i enjoy by the way |
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| 14:18.28 | oquidave | hello again...i want to create an interface to connect to a GSM network....ordinarily am trying to create a GSM-VOIP gateway using a bluetooth dongle and a phone via bluetooth...i've however, had problems with getting my usb bluetooth dongle running with my asterisk server. Now am wondering for development purposes, is there anyother way i can pull this off? thanks |
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| 14:22.58 | oquidave | i can see that i can buy one of these http://shop.eudesignbg.com/ ...but is there anyother setup for my development environment? |
| 14:24.20 | [TK]D-Fender | Maybe you should try working out the bluetooth problems... because you already SEE other products that can do this and don't want to.... |
| 14:24.29 | [TK]D-Fender | oquidave, So you already have the answer..... |
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| 14:27.08 | oquidave | <[TK]D-Fender> i suppose i i've to do it the hard way! but that's for my devp enviroment |
| 14:30.13 | WIMPy | There are many ways to do it. |
| 14:30.40 | WIMPy | YOu could also use an USB dongle or an old Motorola phone via osmocom-bb. |
| 14:32.54 | oquidave | WIMPy: osmocom-bb...first time to hear about it. am already googling it |
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| 14:35.12 | pguima | Good Morning, i am trying to deal with problem I have between 2 asterisk boxes, the voice through the IAX trunk disappear for a few seconds. Anyone ever seen this? |
| 14:35.42 | Faustov | networking issues? |
| 14:38.36 | pguima | i cant really find any issues, i do have a sonicwall firewall on the location. |
| 14:39.27 | oquidave | WIMPy: do you have any link for such an implimentation |
| 14:41.19 | pguima | oquidave, what do you mean by that? |
| 14:43.43 | oquidave | pguima: mean by what? |
| 14:44.19 | pguima | by do i have any link for such an implementation? |
| 14:48.44 | oquidave | pguima: i meant a tutorial on the net showing how that can be done...but nevermind am already doing some googling |
| 14:50.14 | pguima | The setup that I have works, I have multiple boxes linked like this one, but on of my servers is having the audio issue |
| 14:50.54 | WIMPy | oquidave: For osmocom? |
| 14:52.19 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:55.48 | oquidave | WIMPy: something to do with using osmcom to create a gsm-voip trunk/gateway with my asterisk box...remember i started off a bluetooth usb dongle with a mobile phone using chan_mobile.so asterisk module. but as i explained, my OS has issues with the bluetooth dongle, so i was looking at different options for which you suggested "an old Motorola phone via osmocom-bb" |
| 14:56.27 | oquidave | so am reading about this osmocom software now |
| 14:56.55 | WIMPy | I planned to do it, but didn;t get round to do it, yet :-( |
| 14:57.31 | WIMPy | I hooked up a phone and loaded the software on to it, but then I got distracted again. |
| 14:57.33 | WIMPy | (as always :-( ) |
| 14:58.32 | *** join/#asterisk last1 (~last1@ns4.top-consulting.net) |
| 14:59.00 | *** join/#asterisk grohne (~grohne@vortex.stw-bonn.de) |
| 14:59.41 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:59.58 | oquidave | WIMPy: oh..i see |
| 15:00.59 | oquidave | WIMPy: am also reading about OpenBTS |
| 15:01.22 | oquidave | "OpenBTS is a Unix application that uses a software radio to present a GSM air interface to standard 2G GSM handset and uses a SIP softswitch or PBX to connect calls. " http://wush.net/trac/rangepublic |
| 15:01.38 | WIMPy | Many exciting things... |
| 15:02.35 | grohne | hi. why do we have to lock the newly created channel in channels/sig_pri.c:pri_dchannel event PRI_EVENT_RING to setvar CALLINGSUBADDR, but there is no need to lock it for setvar ANI2 27 lines later? |
| 15:03.35 | *** join/#asterisk last1 (~last1@ns4.top-consulting.net) |
| 15:04.00 | oquidave | WIMPy: it seems to be an over-kill for what i what to achieve though |
| 15:04.44 | *** join/#asterisk rcaskey (~Rob@dumbledore.athenshousing.org) |
| 15:04.49 | rcaskey | anyone here using the SNOM 821? |
| 15:07.48 | grohne | hmm. the first lock seems useless as pbx_builtin_setvar_helper acuiqres a channel lock by itself!? |
| 15:08.04 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-43-209.user.veloxzone.com.br) |
| 15:08.14 | grohne | (possible with recursive locks, but confusing) |
| 15:08.30 | WIMPy | grohne: I think you should take that to #asterisk-dev |
| 15:08.59 | grohne | they said tier 3 support was over here... |
| 15:09.24 | last1 | I am having a problem with IAX2 registration |
| 15:09.30 | WIMPy | That's not support, that's source code discussion and that's what -dev is for. |
| 15:09.58 | last1 | I am defining some peers on the asterisk server and then I try to connect to them on 127.0.0.1 |
| 15:10.00 | anonymouz666 | anyone knows what could cause asterisk to send "asterisk" in SIP FROM field and in the RPID header, the correct callerid? |
| 15:10.42 | WIMPy | anonymouz666: No valid callerid(num)? |
| 15:10.47 | anonymouz666 | event with SetCallerPres(allowed) |
| 15:11.02 | anonymouz666 | WIMPy: wrong. it is valid, as I said in RPID. |
| 15:11.12 | anonymouz666 | if I print in dialplan it also correct. |
| 15:11.24 | anonymouz666 | but when building the SIP message, it is "asterisk" |
| 15:11.36 | anonymouz666 | and RPID goes correct |
| 15:11.40 | WIMPy | That might be callerid(ani) |
| 15:12.22 | anonymouz666 | WIMPy: ? |
| 15:12.27 | last1 | for example, this is a sample of attempted IAX2 registration |
| 15:12.27 | last1 | http://pastebin.com/zbMpj7Ws |
| 15:12.39 | last1 | is it in fact trying to register to 127.0.0.1 but asterisk responds using the public IP ? |
| 15:12.46 | anonymouz666 | WIMPy: I should set ani to num, is that what you mean? |
| 15:12.54 | WIMPy | anonymouz666: There can be multiple IDs. Where is that call coming from? |
| 15:13.00 | anonymouz666 | PSTN |
| 15:13.23 | WIMPy | No my idea was that the callerid is set in (ani), but not in (num). |
| 15:13.35 | anonymouz666 | but callerid(num) IS printed in dialplan when asterisk originates "asterisk" in SIP INVITE. |
| 15:13.39 | WIMPy | How? Do you have a log? |
| 15:14.03 | anonymouz666 | there is must be a privacy stuff I am not aware of that is causing this |
| 15:14.07 | WIMPy | Ok, but the valid flag might not be set for some reason. |
| 15:14.35 | WIMPy | That's the direction I thought. |
| 15:15.00 | *** join/#asterisk j4m3s_ (~j4m3s_@c-71-196-88-32.hsd1.fl.comcast.net) |
| 15:15.13 | anonymouz666 | I will back on this after lunch. |
| 15:16.14 | grohne | as far as I can tell that privacy striping should happen inside libpri |
| 15:16.45 | WIMPy | I think there is some confusion about all the caller ID fields an their meaning. |
| 15:16.47 | grohne | but that would contradict your statement that a callerid(num) would be printed |
| 15:17.21 | WIMPy | I planned to write up some questtion to help someone to clarify the situation. |
| 15:17.27 | akshayms | Hey guys |
| 15:17.42 | akshayms | I enabled AJAM on my Asterisk server |
| 15:17.54 | akshayms | and I have FreePBX installed |
| 15:18.23 | akshayms | Now the webinterface for FreePBX is screwed up |
| 15:18.40 | WIMPy | It makes sense to me that (num) is valid even if is withheld. |
| 15:18.47 | WIMPy | ~freepbx |
| 15:18.47 | 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 |
| 15:18.50 | akshayms | and Asterisk status on FreePBX says "ERROR" and Web server status is also error |
| 15:18.52 | WIMPy | akshayms: ^^ |
| 15:19.11 | akshayms | WIMPy, my bad. thanks! |
| 15:19.40 | pguima | anyone using Sonicwall firewalls? Do you have any issues with it/ |
| 15:20.09 | *** join/#asterisk j4m3s_ (~j4m3s_@c-71-196-88-32.hsd1.fl.comcast.net) |
| 15:21.56 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 15:22.05 | akshayms | WIMPy, in Asterisk command line, how do I get the status of Trunks? |
| 15:23.05 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 15:24.11 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 15:27.59 | *** join/#asterisk GameGamer43 (uid5533@gateway/web/irccloud.com/x-vjsnmhuavbwwybpg) |
| 15:29.22 | grohne | ugh. the callerid seems to get copied no less than 5 times while being transferred from the wire to the asterisk channel. libpri copies it from a protocol buffer, so a q931_call and then to a pri_event_ring which is copied to a sig_pri_chan, then copied to a dahdi_pvt and then finally lands in a proper ast_channel. |
| 15:29.54 | grohne | now this makes data flow very easy to understand. :-( |
| 15:34.30 | grohne | bonus points for indirecting the two copies via a member function table. |
| 15:40.09 | *** join/#asterisk aboven (~aboven@host-85-201-12-229.brutele.be) |
| 15:40.39 | aboven | How can I have my own "sip" domain ? |
| 15:41.06 | grohne | basically all you need is a dns a record pointing to your asterisk |
| 15:42.59 | pabelanger | aboven: setup DNS |
| 15:43.49 | WIMPy | grohne: Did I mention that I don't trust libpri too much? |
| 15:44.12 | grohne | WIMPy: don't worry. the bad party are in asterisk itself |
| 15:45.21 | WIMPy | grohne: Are you talking baout cahn_dahdi or the core? |
| 15:45.42 | WIMPy | I think the core has gotten ok-ish. |
| 15:46.03 | grohne | chan_dahdi.c + sig_pri.c |
| 15:47.50 | aboven | OK, I'll try |
| 15:47.51 | aboven | thx |
| 15:48.48 | grohne | WIMPy: also sig_pri is quite copy&past-ish |
| 15:50.06 | WIMPy | If I had more time to spare, I'd try to ask Karsten Keil for his chan_misdn and try to make it work with current Asterisk versions. |
| 15:51.13 | *** join/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 15:57.28 | *** join/#asterisk bradleyprice (~bradleypr@74.254.239.1) |
| 16:01.45 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 16:02.40 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 16:03.46 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:03.46 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:04.03 | *** join/#asterisk nny (~Scott@174.107.223.14) |
| 16:10.07 | anonymouz666 | grohne: yeah ${CALLERID(num)} is never empty |
| 16:10.47 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 16:10.51 | anonymouz666 | I was getting as from "Anonymous@invalid" stuff, then then I set sendrpid=yes and trustrpid=yes and setcallerpres(allowed) |
| 16:10.53 | grohne | anonymouz666: could you elaborate a bit more on your general setup? |
| 16:11.02 | anonymouz666 | sure |
| 16:12.22 | anonymouz666 | there is AMI originate that dials through DAHDI. DAHDI could be MFCR2 or ISDN PRI. When the dst answers, we call the exten that simple will do Dial(SIP/${EXTEN}) and then establish the bridge. |
| 16:12.23 | WIMPy | anonymouz666: I'd try to look at the -valid and -pres values. |
| 16:13.11 | anonymouz666 | in this DIAL (looking at INVITE) and ${EXTEN} got as from CallerID (from header) "asterisk" but RPID header there's the correct value. |
| 16:13.23 | WIMPy | Err. That seems perfectely normal now. |
| 16:13.41 | anonymouz666 | before sendrpid=yes and trustrpid=yes, I was getting anonymous@invalid in the SIP FROM header. |
| 16:13.57 | anonymouz666 | I also set SetCallerPres(allowed) in the path before dials using DAHDI. |
| 16:14.14 | *** part/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 16:14.16 | WIMPy | Yes, that call doesn't have any caller ID, but you get the connectedline info in the RPID. |
| 16:14.19 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 16:14.25 | anonymouz666 | now I got, "asterisk" as from, and the correct value in SIP RPID going to SIP/${EXTEN} |
| 16:15.25 | anonymouz666 | I didn't understand why I have this behaviour |
| 16:15.40 | WIMPy | I just tried to tell you. |
| 16:15.54 | WIMPy | If you originate, the calls are always anonymous. |
| 16:16.09 | WIMPy | If you want a callerid you have to expicitely set it. |
| 16:16.30 | WIMPy | For the 2nd call led that means via a local channel and a piece of dialplan. |
| 16:16.39 | anonymouz666 | I set it before the DIAL(SIP/${EXTEN}) |
| 16:16.49 | anonymouz666 | but doesn't seem to work. |
| 16:16.57 | WIMPy | Show us the context. |
| 16:16.59 | anonymouz666 | ok |
| 16:17.30 | WIMPy | Or better the whole thing including the originate. |
| 16:21.32 | NovceGuru | has faxing over voip gotten any better in the past ~2 years? As in, hooking an adapter to an old fashioned fax macine, and successfully sending 99% of the time? |
| 16:22.01 | NovceGuru | I know it would probbaly take a special provider/codec/etc |
| 16:22.22 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 16:22.24 | aboven | What can I do if I've "Duplicate DTMF" ? |
| 16:22.29 | WIMPy | You want T.38 and that has not changed in the last couple of years. |
| 16:22.41 | WIMPy | aboven: From where? |
| 16:23.03 | aboven | I use asterisk 1.8 |
| 16:23.23 | WIMPy | *from* where? |
| 16:23.31 | aboven | http://pastie.org/4897650 |
| 16:23.34 | coppice | NoviceGuru: it works as well or as badly as your setup lets it. broken setups tend to give <99% success rates |
| 16:23.49 | aboven | call from a sip provider |
| 16:23.52 | aboven | cell phone |
| 16:24.28 | WIMPy | SIP was the answer I was looking for. |
| 16:24.44 | aboven | ok |
| 16:24.49 | WIMPy | It might be an issue of your ITSP. |
| 16:25.03 | NovceGuru | WIMPy: thanks. As of a few years ago I don't think there were that many providers (area code wise) |
| 16:25.21 | aboven | What could I do ? |
| 16:25.53 | anonymouz666 | WIMPy: http://pastebin.ca/2238528 |
| 16:26.45 | aboven | http://pastie.org/4897667 |
| 16:26.53 | aboven | that's my config |
| 16:27.04 | anonymouz666 | WIMPy: I have very few calls with anonymous (now asterisk) it's not happens all the time. |
| 16:28.37 | WIMPy | anonymouz666: Hmm. Interesting. I had that situation way back when 1.4 was new that the PI of outgoing calls was set to the PI of the last incomming call that used the same channel. |
| 16:29.00 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 16:30.25 | aboven | WIMPy: what can I do in my sip.conf configuration ? |
| 16:30.46 | WIMPy | anonymouz666: YOu should set CALLERID(num-pres) instead of SetCallerPres. |
| 16:31.01 | anonymouz666 | already tried to use CALLERID(all) |
| 16:31.07 | WIMPy | aboven: You'd have to do some extensive debugging of the stream. |
| 16:31.39 | anonymouz666 | indeed |
| 16:32.04 | WIMPy | anonymouz666: And you could try to explicitely set num-valid. |
| 16:32.42 | aboven | WIMPy: What do you mean by extensive debugging ? |
| 16:33.11 | WIMPy | aboven: look at what exactely you receive from your ITSP. |
| 16:34.00 | aboven | WIMPy: by recording the stream ? |
| 16:34.36 | WIMPy | The easiest way would probably be to use wireshark. |
| 16:36.48 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 16:36.49 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:37.23 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:37.47 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-omuzjyxykddpfmcw) |
| 16:37.47 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 16:37.51 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:37.51 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:38.57 | aboven | ok |
| 16:40.01 | anonymouz666 | WIMPy: set everything now |
| 16:40.07 | anonymouz666 | num-pres, num-valid, all |
| 16:40.55 | anonymouz666 | From: "Anonymous" <sip:Anonymous@anonymous.invalid>;tag=as230efb31. Remote-Party-ID: "0171936571810" <sip:0171936571810@172.16.0.205>;party=calling;privacy=full;screen=no. |
| 16:40.58 | anonymouz666 | dammit |
| 16:41.04 | anonymouz666 | still failing |
| 16:42.11 | WIMPy | wonders if it might be the local channel screwing something. |
| 16:42.32 | anonymouz666 | i have exactly this setup working fine in other customers |
| 16:42.40 | anonymouz666 | same version, same code, same everything |
| 16:42.53 | anonymouz666 | the telco is different |
| 16:42.53 | anonymouz666 | only |
| 16:43.06 | WIMPy | Well, there must be a difference somewhere. |
| 16:43.22 | anonymouz666 | telco and gsm gateways in this customers. |
| 16:43.29 | anonymouz666 | customer |
| 16:43.45 | WIMPy | Not sure if that would make sense, but is it the only line offering COLP perhaps? |
| 16:43.59 | Qwell | offers file instead |
| 16:45.43 | anonymouz666 | WIMPy: From what I understoon I can't do much in this case |
| 16:46.11 | WIMPy | You might be able to ignore it. |
| 16:46.21 | WIMPy | But the first thing is to identify the difference. |
| 16:46.31 | anonymouz666 | wait |
| 16:46.38 | WIMPy | Then maybe we can find out why it's acting silly. |
| 16:46.39 | anonymouz666 | what if I sett the CONNECTEDLINE ? |
| 16:46.59 | WIMPy | Try it. |
| 16:47.01 | carrar | to the main frame? |
| 16:47.42 | WIMPy | wonders if that can end up in the from? |
| 16:52.09 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:53.50 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 16:54.50 | anonymouz666 | WIMPy: I wonder that too, that's the CALLERID(num) job |
| 16:55.18 | WIMPy | It should be. |
| 16:55.45 | WIMPy | I think you should try to find out what the difference between the systems is. |
| 16:58.03 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 16:58.09 | Kobaz | hmmm, trying to build dahdi |
| 16:58.19 | Kobaz | /root/download/dahdi-linux-2.6.0/drivers/dahdi/wcb4xxp/base.c:38:46: error: asm/system.h: No such file or directory |
| 16:58.29 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 16:59.29 | *** join/#asterisk serafie (~erin@nat/digium/x-tzzekbrkgxwyoqqp) |
| 16:59.33 | WIMPy | Missing kernel source or headers. |
| 16:59.33 | Kobaz | there is no asm/system.h in the linux kernel in 3.5.x |
| 16:59.57 | WIMPy | It has to be configured. |
| 17:00.09 | WIMPy | Just untaring it is not enough. |
| 17:00.20 | Kobaz | i know how to build a kernel |
| 17:00.21 | WIMPy | And the configuration must match what you have running. |
| 17:00.41 | Kobaz | 3.5.x doesn't have an asm/system.h for x86 anymore |
| 17:00.54 | Kobaz | i guess dahdi needs to be updated |
| 17:01.00 | Kobaz | i wonder if there's a newer one |
| 17:01.14 | WIMPy | Ah, the file not the dir is missing. |
| 17:01.23 | WIMPy | Well, yes, I guess you have to wait for an update then. |
| 17:01.54 | last1 | can anyone help with an IAX2 dialog debugging ? |
| 17:01.59 | Kobaz | ah okay |
| 17:02.03 | Kobaz | dahdi 2.6.1 fixed it |
| 17:02.18 | last1 | I'm trying to register an IAX2 host defined on 127.0.0.1 |
| 17:02.29 | last1 | but it never completes |
| 17:02.32 | last1 | the logs show this: http://pastebin.com/zbMpj7Ws |
| 17:02.50 | WIMPy | Are you using diffent ports? |
| 17:03.14 | last1 | yes, those ports point to 4569 |
| 17:03.31 | last1 | what I was more concerned is that it seems it tries to connect to 127.0.0.1 |
| 17:03.42 | WIMPy | For client and server that is. |
| 17:03.45 | last1 | but the reply appears to come from the public IP: 12.34.56.78 |
| 17:03.56 | last1 | am I reading that correctly ? |
| 17:04.01 | WIMPy | But you are using the same machine for both? |
| 17:04.06 | last1 | yes |
| 17:04.12 | last1 | all this is local |
| 17:04.28 | WIMPy | Yes, that's what I read in there. |
| 17:04.49 | last1 | so is that ok ? |
| 17:05.09 | WIMPy | But I also see two pairs of ports. |
| 17:05.18 | WIMPy | Is it more than one client? |
| 17:05.23 | last1 | yes |
| 17:05.38 | last1 | 40569 and 41569 |
| 17:05.47 | WIMPy | ok |
| 17:06.09 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:06.52 | WIMPy | Maybe the client binds to the external IP explicitely. |
| 17:06.58 | Katty | HI LADS. |
| 17:07.23 | WIMPy | That would explain both why you receive that IP and why it wouldn't listen. |
| 17:07.41 | last1 | well, it does listen on 127.0.0.1 40569 for example |
| 17:07.47 | WIMPy | Katty: lo ladette. |
| 17:13.20 | gavimobile | excuese me for reposting, I asked my question several hours ago when the room was a bit less quit. question, I know I can use the cdr options, but for testing purposes im trying to add a custom field to my cdr. I have a field in my mysql table called callerid and I added same => n,Set(CDR(callerid)=zombie pancakes) for my incoming dialplan rule, however after I initiate the call and hangup, the mysql field for callerid is |
| 17:14.41 | Katty | waves at WIMPy |
| 17:15.26 | *** join/#asterisk tomaw_ (tom@freenode/staff/tomaw) |
| 17:16.23 | Katty | dances with Qwell |
| 17:16.34 | Qwell | trips over his own feet |
| 17:16.40 | Katty | now with added moose ears! |
| 17:16.46 | Katty | applies moose ear headband to Qwell |
| 17:17.15 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 17:17.21 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 17:18.37 | jpsharp | Watch the moose teeth. They can be nasty. |
| 17:20.04 | WIMPy | last1: Try to look with tcpdump how far packets are going. |
| 17:21.56 | *** join/#asterisk jrgill (~jrgill@unaffiliated/jrgill) |
| 17:23.12 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 17:25.44 | *** join/#asterisk thecardsmith (~quassel@pdpc/supporter/student/thecardsmith) |
| 17:26.28 | thecardsmith | Will Asterisk support it if my ITSP sends the SDP in the "200 OK" message instead of the "INVITE"? |
| 17:26.49 | thecardsmith | I'm reading some specs from my ITSP, and, I just hadn't seen the SDP in anything but the INVITE previously |
| 17:27.03 | Qwell | thecardsmith: What happened when you tried it? |
| 17:27.07 | thecardsmith | s/from my ITSP/from a potential ITSP/ |
| 17:27.53 | thecardsmith | Haven't the opportunity |
| 17:31.45 | *** join/#asterisk tuxq (~steven@24.172.84.82) |
| 17:32.05 | tuxq | so. what's the usual method of question asking in here? :) |
| 17:32.21 | carrar | ~ask |
| 17:32.21 | 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. |
| 17:33.25 | tuxq | right on. Is there a way to disable recording of extensions system-wide on an Asterisk (1.4) server without doing it per-extension. There are a lot of extensions and we're having choppy audio. I'm hoping to isolate the cause, and I'm guessing that the uLaw->GSM transcode that's taking place may have something to do with it. Then again, Asterisk is not my forte. |
| 17:34.08 | seanbright | asterisk doesn't record by default |
| 17:34.15 | carrar | disable recording |
| 17:34.17 | seanbright | maybe you're looking for #freepbx |
| 17:34.24 | carrar | does the probably go away? |
| 17:34.29 | carrar | err problem |
| 17:34.43 | tuxq | actually, trixbox. dang, I was hoping it would be a generic asterisk thing |
| 17:35.49 | tuxq | carrar: that's what I was hoping to figure out. But the phone system is production and in use with all 40 extensions going, so I didn't want to go in one-by-one to disable them, and some are even active. If there were a way to disable recording for everyone no matter the extensions' preference, I was going to use that to test |
| 17:36.08 | carrar | Then how you gonna find out? |
| 17:36.17 | tuxq | hmm. maybe I included too much information |
| 17:36.35 | tuxq | Is there a way to disable system recordings no matter the extension preference of "Always On" |
| 17:36.55 | carrar | no clue, your in the wrong channel to ask trixbox questions |
| 17:37.02 | carrar | but genreal issues |
| 17:37.07 | carrar | like too much recording |
| 17:37.11 | carrar | disable it |
| 17:37.14 | carrar | or use a RAM Drive |
| 17:37.41 | AkkerKid | tuxq: you can go into the dialplan that freepbx makes and disable it entirely |
| 17:38.09 | [TK]D-Fender | Trixbox has been dead for YEARS |
| 17:38.14 | carrar | transcoding will chew up resources also, off loard that to hardware if it's g729 |
| 17:38.19 | carrar | load |
| 17:38.21 | Katty | pokes carrar |
| 17:38.26 | tuxq | lol tell me about it. but our sales team decided to sell it to a client because ... well ... fill in the blank |
| 17:38.33 | carrar | pinkypokes Katty back!! |
| 17:38.39 | Katty | carrar: how'rechu dear |
| 17:38.53 | AkkerKid | put something like this into your extensions_custom.conf file. http://pastebin.com/c7sPaa0a |
| 17:39.10 | carrar | doing good, sippin espresso |
| 17:39.15 | Katty | :> |
| 17:39.18 | carrar | listening to tunes |
| 17:39.24 | Katty | espresso++ |
| 17:39.26 | carrar | workin! |
| 17:39.26 | AkkerKid | if trixbox uses a similar freepbx backend, it should work or at least get you close |
| 17:40.01 | tuxq | AkkerKid: thank you -- i will take that and run w/ it |
| 17:40.45 | AkkerKid | that should hijack the code that starts the recording. I have something like that in my Elastix box right now. |
| 17:42.08 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:47.05 | *** join/#asterisk italorossi (~textual@189.124.163.167) |
| 17:55.19 | *** join/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210) |
| 17:55.50 | *** join/#asterisk gboudreau (~gboudreau@mail2.danslereseau.com) |
| 17:56.45 | gboudreau | hi guys. having some problems with my asterisk server. sometimes, calls would come in, and when answering, I would not hear the other party, and I think the other party wouldn't hear me (but I'm not sure about that part). |
| 17:57.12 | gboudreau | I tried hitting *3 to record both parties when this happened, and I think it only recorded the other party audio, not mine. |
| 17:57.14 | Katty | hey i just ${AGENTSTATUS}ed you, and this is crazy, but here's my ${CALLERID(num)} so Dial(SIP/me) maybe. it's hard to ${BLINDTRANSFER} to you baby, so here's my ${EXTEN} so ${MEETME} maybe! |
| 17:57.31 | [TK]D-Fender | NAT & other networking screwups are the usualy suspects. Are any funny subnets, or NAT involved? |
| 17:58.19 | gboudreau | nothing too funky. cable modem plugged into wan port of router. phone plugged in lan port of same router. so NAT, yes. |
| 17:58.28 | gboudreau | but this doesn't happen every time. |
| 17:58.38 | gboudreau | only some times. other times, it works just fine |
| 17:59.00 | gboudreau | if NAT was the problem, wouldn't it always happen ? |
| 18:04.04 | WIMPy | Depends on the natting device. |
| 18:04.25 | WIMPy | It could hapen per direction or even pretty randomly. |
| 18:04.47 | WIMPy | Alsywas and never are the easiest options. |
| 18:05.48 | AkkerKid | Katty: I'm not even safe in here from that song?!¿!? |
| 18:05.52 | *** join/#asterisk Vince-0 (~androirc@41-135-186-247.dsl.mweb.co.za) |
| 18:07.37 | rcaskey | so any feelings on the SNOM 821 vs the SNOM 760? |
| 18:08.28 | carrar | move it to the left a little |
| 18:08.53 | [TK]D-Fender | gboudreau, You need your server set up to work from behind NATY, and * needs to know that certain devices are themselves behind their own NAT's |
| 18:10.00 | [TK]D-Fender | gboudreau, if your * is NAT'd : nat=yes , directmedia=no , externaddr=ip.of.your.server , localnet=network/mask (for each local subnet) |
| 18:10.32 | [TK]D-Fender | gboudreau, For your peers, ITSP's should always be "nat=no". Remote phones should be "nat=yes", "qualify=yes" |
| 18:10.38 | gboudreau | * isn't natted. it's accessible on the internet |
| 18:10.50 | gboudreau | it's not inside our office network. |
| 18:10.54 | [TK]D-Fender | gboudreau, It has a PUBLIC IP right on it? |
| 18:10.58 | gboudreau | yes |
| 18:11.04 | gboudreau | which is used by the phones to connect to it |
| 18:11.18 | [TK]D-Fender | gboudreau, Then just disable directmedia & set the device to nat=yes, qualify=yes |
| 18:12.07 | gboudreau | I have nat=auto_force_rport,auto_comedia |
| 18:12.38 | gboudreau | directmedia=no and qualify=yes |
| 18:12.44 | [TK]D-Fender | nat=yes <- |
| 18:12.48 | gboudreau | ok |
| 18:12.57 | gboudreau | i didn't see that as an option in the conf |
| 18:14.22 | gboudreau | I have nat defined globally, not per device. is that ok ? |
| 18:15.41 | [TK]D-Fender | gboudreau, Explicit. the only way to fly |
| 18:15.57 | *** join/#asterisk netman (netman@247.48.78.188.dynamic.jazztel.es) |
| 18:16.11 | [TK]D-Fender | Also may sure your firewall allows the full rtp.conf range, plus SIP |
| 18:36.36 | gboudreau | ok. will try that and see if it helps. |
| 18:36.39 | gboudreau | thanks for the pointers |
| 18:41.36 | *** part/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 18:43.24 | *** join/#asterisk jaxon007_ (~jaxon007_@121.245.159.59) |
| 18:46.32 | nny | any switchvox gurus here? |
| 18:47.05 | carrar | ~ask |
| 18:47.05 | 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. |
| 18:50.21 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:50.21 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:51.18 | jaxon007_ | <PROTECTED> |
| 18:52.59 | *** join/#asterisk angryuser_laptop (~angryuser@143.86.112.78.rev.sfr.net) |
| 18:53.25 | leifmadsen | jaxon007_: test on some standard commodity hardware and then determine what it peaks at, then scale accordingly |
| 18:53.44 | leifmadsen | there are no hard and fast rules, so no one is going to be able to guide you beyond that |
| 18:53.56 | leifmadsen | there are so many factors involved, that only your own implementation testing can tell you what you need |
| 18:57.09 | jaxon007_ | @leifmadsen: thanks i will do as per you say..Even can give me some idea about the server requirements |
| 18:57.10 | *** join/#asterisk airshock (~Adium@143.66.77.131) |
| 18:57.54 | leifmadsen | as stated, I have no idea |
| 18:58.01 | leifmadsen | start with a 4 core CPU and 4 GB of RAM |
| 18:58.13 | leifmadsen | which is pretty standard nowadays |
| 18:58.17 | cusco | Can Asterisk 1.8 realtime (mysql) use/insert column time (in queue_log) as unixtimestamp just like 1.6 did ? |
| 18:58.21 | airshock | Hi all. New to Asterisk, setting it up along with FreePBX on a CentOS 6 x64 server. Got Asterisk and FreePBX installed successfully and set up a few extensions, but whenever I try to call any extension on the system from my own the call always gets dropped, and the logs say that the extension cannot be reached because it can't be found in the "from-internal" context. When I log into Asterisk and do "dial plan show 1001@from-internal" it says the "from-inte |
| 18:59.05 | cusco | airshock: seems that your message was trunkated ? |
| 18:59.11 | jaxon007_ | @leifmadsen: thanks..I was Planning 6 * Quad core processor and 16 GB RAM.. |
| 19:00.04 | airshock | cusco: Odd. What part are ou missing? |
| 19:00.38 | leifmadsen | airshock: you'll need to obtain freepbx support from #freepbx |
| 19:00.39 | cusco | it says the "from-inte |
| 19:00.42 | cusco | no more after that |
| 19:01.26 | *** part/#asterisk gboudreau (~gboudreau@mail2.danslereseau.com) |
| 19:02.31 | airshock | It says the "from-internal" context does not exist. |
| 19:02.36 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 19:05.07 | airshock | cusco: Did that come through? |
| 19:05.26 | rrittgarn | Having an issue creating an IVR when using background. I have a series of options for a user to choose from after calling in. The options are all 1 digit options less 1, the one option that I have that is 3 digits is the "Dial the person's extension" and it does a match on _XXX. This is all fully functional, however there is about a 2 second delay before it actually takes any action. If I remove the _XXX as an option |
| 19:05.29 | *** join/#asterisk mathi (~Matthew@78.129.54.191) |
| 19:05.32 | mathi | hi |
| 19:05.58 | mathi | The Read() application hangs up only if there is no response for about 20 seconds. I would like to shorten that delay to 10 seconds for example, that was my attempt: |
| 19:05.59 | mathi | same => n,Read(langNum,select-language,1,,,10) |
| 19:06.07 | mathi | But still, the call will only hang up after 20 seconds when nothing has been entered. |
| 19:08.36 | *** join/#asterisk jsjc (~Adium@188.78.152.54) |
| 19:09.32 | [TK]D-Fender | mathi, "core show application read". Pastebin that, and your call. |
| 19:10.22 | airshock | Dors anyone know how I can fix this? Asterisk keeps rejecting calls because it says "from-internal" context does not exist. |
| 19:10.41 | ChannelZ | Either make it exist, or put the phone in the correct context for your actual dialplan |
| 19:10.43 | [TK]D-Fender | rrittgarn, Well if you're expecting single digit options to be faster... you shouldn't. * doesn't know if you are going to be dialing a 3 digit number starting with that. |
| 19:10.55 | [TK]D-Fender | rrittgarn, that's what you get for using a pattern like _XXX. |
| 19:11.19 | [TK]D-Fender | airshock, Well apparently it doesn't and you told it to go there. You should look into that.... |
| 19:11.56 | rrittgarn | i've also tried _3XX as all the extensions start with 3, and 3 is not an option elsewhere in the context |
| 19:12.09 | airshock | But I se the context in the files in /etc/asterisk. I installed Asterisk and FreePBX withou making any custom changes or etc., so it's supposed to exist. |
| 19:12.38 | mathi | [TK]D-Fender, http://fr.pastebin.ca/2238605 |
| 19:13.12 | [TK]D-Fender | rrittgarn, and ....? |
| 19:13.28 | airshock | see* without* |
| 19:13.53 | ChannelZ | well note that FBX (AFAIK) is a realtime config and most of that stuff exists in a database, not your files (or do they not put extensions in a DB? I dunno...) |
| 19:14.16 | [TK]D-Fender | airshock, "dialplan show" <- |
| 19:14.24 | rrittgarn | Fender: And I'm going to stop trusting what my co-workers tell me... |
| 19:14.29 | ChannelZ | Either way something is configged wrong; Your device's context is set for "from-internal" but there is no dialplan context with that name. |
| 19:14.36 | [TK]D-Fender | rrittgarn, You started? Shame on you. |
| 19:14.52 | rrittgarn | I know... I had a dream that I wouldn't be the only one supporting this system... |
| 19:15.13 | rrittgarn | _3XX works fine... im guessing he forgot to reload the dialplan |
| 19:17.08 | airshock | D-Fender: "dial plan show 1001@from-internal" returns error "There is no existence of from-internal context" |
| 19:17.27 | [TK]D-Fender | airshock, that is not what I just told you to do. |
| 19:17.30 | mathi | [TK]D-Fender, do you have any idea for that timeout problem? |
| 19:17.41 | [TK]D-Fender | mathi, And that isn't all I told YOU to do. |
| 19:17.53 | [TK]D-Fender | wonders when people are giong to actually read & pay attention. |
| 19:18.23 | mathi | [TK]D-Fender, I paste bin the documentation for Read, and the instruction with Read(), I'm sorry if I missed something? |
| 19:18.25 | airshock | D-Fender: I'm visually impaired so am using a screen reader that reads the screen aloud to me. Maybe it misread your message. |
| 19:18.25 | last1 | where can I see a full explanation of what the IAX debug output means ? |
| 19:18.32 | last1 | for example, what is the DCall field ? |
| 19:20.56 | ChannelZ | airshock: just 'dialplan show' will show all the contexts and extensions Asterisk knows about. |
| 19:21.59 | airshock | ChannelZ: And that list is very, very large. Is there a way to parse it so I can just find info about the from-internal context, which it still says does not exist? |
| 19:22.20 | ChannelZ | If it doesn't exist, it doesn't exist. That was just to prove to you it doesn't. |
| 19:23.04 | ChannelZ | (or perhaps show a type-o) |
| 19:23.26 | airshock | ChannelZ: Yet I see dozens of references to it in the files in /etc/asterisk. So it's there I think, just Asterisk isn't seeing it. |
| 19:23.38 | *** join/#asterisk ideaman55 (~ideaman55@173-10-29-218-BusName-utah.ut.hfc.comcastbusiness.net) |
| 19:24.44 | ChannelZ | The only reference that matters would be in extensions.conf |
| 19:25.17 | airshock | ChannelZ: What exactly in extensions.conf am I looking for? I'm very new to Asterisk. Have been using FreePBX for a while on other systems but have never ran into this problem. |
| 19:25.35 | mathi | [TK]D-Fender, maybe by call you meant the sip debug logs ? |
| 19:25.36 | ChannelZ | [from-internal] |
| 19:26.20 | [TK]D-Fender | <[TK]D-Fender> mathi, "core show application read". Pastebin that, and your call. <- And. Your. Call. |
| 19:26.40 | ChannelZ | But again FreePBX does whatever it does, which I don't know. They do lots of includes of multiple files |
| 19:27.08 | mathi | ... :-( |
| 19:27.14 | [TK]D-Fender | <[TK]D-Fender> airshock, "dialplan show" <- |
| 19:27.23 | *** join/#asterisk bitglue (~pfrost@207.8.235.242) |
| 19:27.24 | mathi | I don't get it |
| 19:27.35 | [TK]D-Fender | We didn't get it either. |
| 19:27.39 | [TK]D-Fender | We keep asking for it. |
| 19:27.44 | [TK]D-Fender | But we never get it |
| 19:28.25 | bitglue | i'm trying to implement a means that Queue agents can hang up on the caller, but then continue in the dialplan to do their after-call-work, and then enter a ticket number. There's a bewildering number of options to Queue and Dial, and I'm not sure which I'd need. Any suggestions? |
| 19:28.28 | ChannelZ | And now we have blue balls. |
| 19:29.24 | Kobaz | the biggest, balls of them all |
| 19:29.30 | bitglue | the idea is that if i'm a helpdesk agent, when i'm done with the call, i send some dtmf instead of hanging up, and that hangs up on the caller, but then asterisk prompts me for a ticket number |
| 19:29.59 | bitglue | this way time spent after call is captured, i don't get new calls until i'm done, and i can associate tickets with calls. |
| 19:31.20 | [TK]D-Fender | bitglue, You would still have to deal with the caller hanging up first. This is best dealt with AFTER the call during wrap-up |
| 19:32.00 | bitglue | [TK]D-Fender: ok, how exactly? |
| 19:32.23 | bitglue | there's the F option to Dial(), which I thought would be useful maybe |
| 19:32.27 | [TK]D-Fender | Track the last call they were on and add it afterwards |
| 19:32.52 | bitglue | how do agents indicate when they are done with wrap-up, and which ticket they worked on? |
| 19:34.17 | [TK]D-Fender | Whre is this "ticket" known"? |
| 19:34.27 | bitglue | the agent knows it |
| 19:34.35 | [TK]D-Fender | And there is no indication for "wrap up". Its just a time limit in the queue |
| 19:34.56 | [TK]D-Fender | bitglue, well then make some dialplan to prompt them for it and tag it to the last queue call they answered |
| 19:34.59 | bitglue | i know that option, and it's not what i want, because 1) it's not interactive, and 1b) it's a fixed amount of time |
| 19:35.17 | last1 | in iax config, does: permit=0.0.0.0/0.0.0.0 mean that everyone is allowed ? |
| 19:35.28 | bitglue | [TK]D-Fender: yes, that's exactly what i'm trying to do. The problem is how to get them to that part of the dialplan after their call. |
| 19:35.49 | *** part/#asterisk jsjc (~Adium@188.78.152.54) |
| 19:35.50 | [TK]D-Fender | A separate call. After the fact. |
| 19:36.46 | bitglue | that's no good. 1) it's more work, 2) what if another call comes in before they have called to wrap up? |
| 19:37.01 | bitglue | or perhaps most importantly: 3) what if they don't bother to call? |
| 19:39.13 | [TK]D-Fender | bitglue, You have to deal with it anyway if you get hung up on. |
| 19:39.25 | [TK]D-Fender | Or the reverse |
| 19:39.27 | *** join/#asterisk jmpiloq (~jmpiloq@static-87-79-94-28.netcologne.de) |
| 19:39.58 | bitglue | isn't that what F does for Dial? |
| 19:40.02 | [TK]D-Fender | Agent is the callee.... Nowhere do I see a way that the caller (the queue) can end the call and the callee gets to continue on. |
| 19:40.46 | [TK]D-Fender | bitglue, Hrm ... actually ... that MIGHT do... |
| 19:41.06 | [TK]D-Fender | bitglue, Now you still need to be able to trigger a hangup yourself... AND conitune. that';s the half I'm less sure about |
| 19:41.11 | bitglue | yes, me too |
| 19:41.28 | bitglue | unless maybe we just say "goodbye" and wait for the caller to hang up |
| 19:41.48 | bitglue | i thought maybe some "h" option somewhere, or something. |
| 19:42.01 | airshock | ChannelZ: I was wrong, from-internal is in extensions_additional.conf, not extensions.conf |
| 19:42.06 | bitglue | anyway, is there a way i can have a ")" as part of a parameter to a Gosub? |
| 19:42.11 | [TK]D-Fender | Doesn't seem to be the case and you could be stuck floating around... |
| 19:42.33 | [TK]D-Fender | bitglue, Not pretty. There does come a point where an agent should take responsibility for classifying their calls |
| 19:42.51 | mathi | [TK]D-Fender, I understood that 10 seconds timeout isn't really 10 seconds from the moment Read() is executed |
| 19:43.15 | mathi | if I put 50 seconds for example, the timeout will occur after 58 seconds or so |
| 19:43.44 | [TK]D-Fender | last1, Yes |
| 19:43.50 | bitglue | [TK]D-Fender: well they do, by creating a ticket. I just need a reasonable way for them to associate the call with the ticket. |
| 19:43.54 | mathi | I guess it is from the moment the audio files have finished playing |
| 19:45.41 | bitglue | so this seems like a silly problem: same => n,Gosub(dial-gosubs,dialsipopts,1(${EXTEN},cF(agents,acw,1)) |
| 19:45.51 | [TK]D-Fender | bitglue, follow up call to classify. Tell them "that's what your wrapup time is for" |
| 19:46.27 | airshock | Can anyone answer this: is it a problem that the from-internal context exists in /etc/asterisk/extensions_additional.conf but not extensions.conf? |
| 19:46.27 | bitglue | but the first ) is parsed as the end of the Gosub parameters, so there's no closing ) for the F option to Dial |
| 19:47.42 | bitglue | [TK]D-Fender: the harder i make this for them, the less likely they are to do it. |
| 19:48.02 | bitglue | my goal is to make my employees job's easier, not harder. |
| 19:48.30 | [TK]D-Fender | airshock, I am not seeing the dump of your dialplan from * CLI like I asked for over half an hour ago. |
| 19:48.34 | bitglue | realizes he put his ' in the wrong place both time :( |
| 19:49.17 | [TK]D-Fender | bitglue, you still have to deal with the other ende. Better off accepting this limitation and just making the process uniform. |
| 19:49.40 | [TK]D-Fender | bitglue, People are always going to shirk responsibility. Log it and deal with the human problem. |
| 19:49.44 | last1 | D-Fender: how can I find out why my asterisk is not responding to an IAX2 REGREQ packet ? |
| 19:49.55 | last1 | I have iax2 debug enabled, I see the packet coming in |
| 19:50.00 | last1 | but no response going out |
| 19:50.07 | last1 | from what I gathered, it should send at least an ACQ |
| 19:50.10 | last1 | ACK |
| 19:50.20 | [TK]D-Fender | I'd start by actually showing the full peer and attempt |
| 19:50.20 | bitglue | [TK]D-Fender: i don't see how requiring them to make a separate call is "better", except in that maybe it works around a lack of reasonable functionality that could exist in asterisk. |
| 19:50.41 | [TK]D-Fender | bitglue, then by better you mean "the only thing that'll do the job"? :) |
| 19:50.59 | bitglue | [TK]D-Fender: it's a kludge, not a better solution. |
| 19:51.16 | [TK]D-Fender | bitglue, Kludge is what we HAVE. It came bundled with the solution. |
| 19:51.23 | bitglue | whatever |
| 19:51.28 | [TK]D-Fender | bitglue, aside from that : vi app_queue <---- |
| 19:51.39 | Katty | with The Solution(tm) |
| 19:51.58 | [TK]D-Fender | bitglue, As you attest, the functionality does not exist. I can't tell you settings for things that don't exist. So yes, you'll have to do something else.... |
| 19:52.08 | bitglue | fair enough |
| 19:52.13 | [TK]D-Fender | Sorry ... vi app_queue.c |
| 19:52.22 | bitglue | anyway, any ideas about this: Gosub(dial-gosubs,dialsipopts,1(${EXTEN},cF(agents,acw,1)) |
| 19:52.24 | Katty | are you being nice today, fender? |
| 19:52.24 | [TK]D-Fender | Should at least get the full filename |
| 19:52.27 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 19:52.52 | [TK]D-Fender | Katty, in selected geography and capacity, yes |
| 19:53.08 | Katty | that means no. |
| 19:53.12 | bitglue | dialsipopts, if it's not obvious, is a subroutine that takes two parameters: the sip endpoint to dial, and the options to give to Dial, and it also implements handling forwarding, dnd, etc. |
| 19:53.17 | Katty | you should be nice, dear. |
| 19:53.21 | *** join/#asterisk anfecora (~IceChat77@108.60.132.170) |
| 19:53.28 | anfecora | Hi alll |
| 19:53.29 | *** join/#asterisk mbrit (~mbrit@186.120.97.194) |
| 19:53.30 | Katty | ohai. |
| 19:53.31 | airshock | [TK]D-Fender: The dial plan show output is way too long for me to be able to copy. |
| 19:53.48 | bitglue | the problem is it's called with $ARG2 set to "cF(agents,acw,1" because the first ) is parsed as matching Gosub(, not F( |
| 19:53.48 | [TK]D-Fender | airshock, Get a bigger buffer |
| 19:54.27 | anfecora | i have an isse trying to make guest calls, i have avpf=yes on the general context but when i make a call it states that i have not set it on for the peer have anyone had this issue or should i reported as a bug |
| 19:54.33 | anfecora | thanks in advance |
| 19:55.33 | Qwell | anfecora: That is not a valid general setting. |
| 19:55.58 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:55.58 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:56.04 | anfecora | No matching peer for '1622' from '11.1.1.1:50830' |
| 19:56.04 | anfecora | [Oct 2 15:26:55] WARNING[8800][C-0000000c]: chan_sip.c:9639 process_sdp: Received SAVPF profle in audio offer but AVPF is not enabled: audio 45518 RTP/SAVPF 103 104 9 102 0 8 106 105 13 127 126 |
| 19:56.04 | anfecora | [Oct 2 15:26:55] WARNING[8800][C-0000000c]: chan_sip.c:9974 process_sdp: Insufficient information in SDP (c=)... |
| 19:56.15 | Qwell | looks at file |
| 19:56.16 | *** join/#asterisk SeRi (~wtf@c-98-200-53-71.hsd1.tx.comcast.net) |
| 19:56.22 | anfecora | how can i set it so any guest can take it |
| 19:56.32 | anfecora | i need it for a conference server |
| 19:56.55 | anfecora | and i don't want to create a peer for each call |
| 19:57.07 | mathi | so how to handle a timeout from a Read() ? I have a special "t" extension, but it is not being executed after a Read()-timeout |
| 19:57.33 | anfecora | how can i make it available for everyone, i am doing a webrtc setup on asterisk 11 |
| 19:57.40 | anfecora | thank you |
| 19:58.28 | mathi | I should always check ${READSTATUS} to OK ? |
| 19:59.37 | [TK]D-Fender | mathi, READ() already tells you what it returns... I'm sure you can see what to check out of those... |
| 20:01.31 | mathi | [TK]D-Fender, I could just check whether the return value of Read() is empty or not |
| 20:01.55 | [TK]D-Fender | mathi, A plan! Or something vaguely like one! Run with it! |
| 20:03.28 | mathi | [TK]D-Fender, but the documentation (core show application ...) doesn't specify what it returns. Who knows, it might return a value for some specific error, so I prefer to check ${READSTATUS} |
| 20:04.10 | [TK]D-Fender | mathi, Feel free to check that too.. |
| 20:04.34 | *** join/#asterisk Nemus_ (~Nemus@108-166-98-33.static.cloud-ips.com) |
| 20:04.35 | Qwell | anfecora: Please don't message people without their permission. My fee is $299.99/hour, 4 hour minimum. DCC chat is extra. |
| 20:04.40 | anfecora | sorry |
| 20:04.52 | [TK]D-Fender | mathi, So far you're reading a single digit, and not specifying any as illegal. |
| 20:04.54 | anfecora | it was by mistake |
| 20:06.01 | mathi | [TK]D-Fender, hah, I should handle that too ... I don't know a good way to for example accept only 1 2 3, 0 and 9 |
| 20:06.12 | mathi | a lot of if conditions ? |
| 20:06.27 | [TK]D-Fender | mathi, check the instructions for it again |
| 20:06.32 | mathi | ok |
| 20:06.54 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:07.24 | *** join/#asterisk Nemus_ (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 20:08.07 | [TK]D-Fender | mathi, I just did... was pretty sure that one had a list of valid entires... doesn't appear to. So yup... gotta check it yourself. |
| 20:08.37 | mathi | [TK]D-Fender, it would be nice if I could specify an ARRAY with allowed digits, but unfortunately it doesn't have that feature, that would be a very nice feature to have, don't you think so? |
| 20:09.21 | [TK]D-Fender | mathi, I might say "yes", but then... that's less useful for those entering a value of more than 1 digit.... basically you're going to have to validate at some point. |
| 20:09.30 | [TK]D-Fender | mathi, And you can do that in the dialplan as it is. |
| 20:09.48 | [TK]D-Fender | mathi, Then again you could also have made an auto-attendant out of this instead of using Read. |
| 20:09.56 | [TK]D-Fender | mathi, You have options on both sides. |
| 20:10.08 | *** join/#asterisk vlad_starkov (~vlad_star@77.41.88.210) |
| 20:10.36 | mathi | I got to go, will think about this on my way home:) |
| 20:10.38 | mathi | thank you! |
| 20:13.03 | jaxon007_ | I am runnin asterisk server with 200 concurrent calls with 4*Quad Core Xeon 2.4 GH Processor ..Can we use the same configuration for 100 concurrent conference calls with 4 participants .. |
| 20:13.29 | Qwell | jaxon007_: try it |
| 20:14.34 | drmessano | jaxon007_: I would use ConfBridge in Asterisk 10 and not MeetMe |
| 20:15.09 | anfecora | anyone can please point me on how to made avpf available for all calls |
| 20:15.25 | anfecora | if it is not a general setting why is it available on the general context |
| 20:15.56 | Qwell | anfecora: It's not. |
| 20:16.11 | airshock | [TK]D-Fender: It's no that, it's due to my screen reader. |
| 20:16.47 | jaxon007_ | drmessano@: i have to use meetme because i ave not used ast10 anywhere for production... |
| 20:17.32 | last1 | jaxon: are you using vicidial ? |
| 20:18.28 | jaxon007_ | last1: no , for handling bulk conference |
| 20:19.34 | last1 | what kind of cpu's you have ? |
| 20:19.49 | last1 | we have a dual xeon e5620 ( 6 cores ) I believe |
| 20:20.03 | last1 | and it runs vmware with about 7-8 vicidial instances |
| 20:20.18 | last1 | 30 users each, no problem |
| 20:20.19 | jaxon007_ | @4 GB RAM |
| 20:20.21 | jaxon007_ | model name: Intel(R) Xeon(R) CPU X3430 @ 2.40GHz |
| 20:20.23 | jaxon007_ | 4 * QUAD core processor |
| 20:20.37 | last1 | we have 64GB of ram though and very fast disks |
| 20:21.05 | last1 | look here |
| 20:21.06 | last1 | http://www.cpubenchmark.net/cpu_lookup.php?cpu=Intel+Xeon+X3430+%40+2.40GHz |
| 20:21.19 | jaxon007_ | @last: thanks let me check |
| 20:21.32 | last1 | you have 3852 'points' for your cpu |
| 20:21.55 | last1 | 4000x4 = roughly 16 000 |
| 20:22.21 | last1 | our e5620 has 9600 |
| 20:22.29 | last1 | http://www.cpubenchmark.net/cpu_lookup.php?cpu=%5BDual+CPU%5D+Intel+Xeon+E5620+%40+2.40GHz |
| 20:22.41 | Qwell | How is any of this useful for Asterisk dimensioning? |
| 20:22.41 | last1 | so we have 19 200 vs your 16 000 with only 2 cpus |
| 20:22.49 | last1 | which means your hardware is quite old |
| 20:24.09 | last1 | Qwell: would you happen to know how I can troubleshoot why asterisk won't respond to an IAX2 request ? |
| 20:24.19 | last1 | I have iax debug enabled, I see it Rx's the packet |
| 20:24.21 | Qwell | You probably need faster CPUs. |
| 20:24.24 | last1 | but nothing after that |
| 20:25.11 | drmessano | I can't imagine running 100 conferences with MeetMe. I know its been done, but my god.. All of the Bogomips that are wasted. |
| 20:25.36 | Katty | glomps drmessano |
| 20:26.04 | drmessano | Bogoglomps Katty |
| 20:26.14 | Katty | ^_^ |
| 20:26.19 | Katty | bogoglomps++ |
| 20:26.42 | drmessano | :) |
| 20:26.54 | jaxon007_ | |
| 20:26.56 | jaxon007_ | drmessano: u r saying that 100 conference is not passible using meetme |
| 20:28.55 | drmessano | No, I am just saying MeetMe is a bunch of wasted wasted wasted resources. |
| 20:31.55 | jaxon007_ | |
| 20:31.58 | jaxon007_ | drmessano: suppose i am using confbridge , then ntel(R) Xeon(R) CPU X3430 @ 2.40GHz 01:50 |
| 20:31.59 | jaxon007_ | 4 * QUAD core processor 01:50 |
| 20:32.01 | jaxon007_ | <PROTECTED> |
| 20:32.12 | Qwell | try it. |
| 20:32.24 | drmessano | ..... |
| 20:33.38 | drmessano | jaxon007_ NOBODY is going to commit to telling you something will work based on CPU specs, unless you are suggesting something silly like 1000 conferences or simple like 10 conferences. IT MAY WORK.. then again.. IT MAY NOT |
| 20:34.14 | drmessano | All I am saying is that I hate MeetMe, its a waste of CPU, and I am so glad its days are numbered |
| 20:34.53 | jaxon007_ | |
| 20:34.55 | jaxon007_ | drmessano: thank you.. |
| 20:35.32 | last1 | it depends on many things |
| 20:35.36 | last1 | clock source for example |
| 20:35.42 | jaxon007_ | |
| 20:35.44 | jaxon007_ | drmessano: any one used asterisk10 on production ..how is ur expereince |
| 20:35.52 | drmessano | Asterisk 10 is badass |
| 20:36.02 | Qwell | drmessano: So's your face. |
| 20:36.05 | Qwell | crap. I'm so bad at that. |
| 20:36.10 | drmessano | LOL |
| 20:36.13 | cmendes0101 | lol |
| 20:36.17 | drmessano | XKCD Fail |
| 20:36.21 | Qwell | hmm? |
| 20:36.39 | drmessano | Sounds like something from an XKCD |
| 20:36.54 | Qwell | shrugs |
| 20:37.01 | drmessano | My hobby: Incorrectly judging the need for an insult |
| 20:37.09 | drmessano | Steve: I got a new blender |
| 20:37.20 | drmessano | Me: That's what your MOMMA said |
| 20:38.25 | Katty | your momma don't got nothin on my blender. |
| 20:38.28 | Katty | prettyboy. |
| 20:39.08 | leifmadsen | wow, didn't I answer that question about load testing like... 2 hrs ago? |
| 20:39.30 | leifmadsen | oh only 1.5 hrs ago |
| 20:39.32 | leifmadsen | my bad |
| 20:39.39 | Katty | yes. shame on you for losing 30 minutes. |
| 20:39.45 | Katty | we can't have people just losing 30 minute blocks all day. |
| 20:40.02 | Katty | you start those shenanigans and soon EVERYONE is losing 30 minutes. all the time. everytime. |
| 20:40.09 | Katty | before you know it POOF. time itself is GONE MISSING. |
| 20:40.14 | drmessano | leifmadsen: I think he said something about what the heck do you know about Asterisk.. then something about your momma |
| 20:40.24 | leifmadsen | drmessano: sounds right |
| 20:40.38 | Katty | leifmadsen's baby's momma has a pretty face. |
| 20:40.46 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 20:40.50 | leifmadsen | Katty: stop hitting on my wife! |
| 20:40.56 | leifmadsen | and my baby's momma too |
| 20:40.58 | leifmadsen | heyo! |
| 20:40.59 | Qwell | She didn't say your wife |
| 20:41.00 | Qwell | :D |
| 20:41.03 | Katty | i'd never hit a preggy lady. |
| 20:41.04 | drmessano | leifmadsen: Need to go all Bill Nye on them. "My name is one front of THE BOOK. Can I eat my sandwich now?" |
| 20:41.10 | drmessano | on* |
| 20:41.10 | Katty | that's just not polite. |
| 20:41.11 | drmessano | Gah |
| 20:41.14 | Katty | think of the children. |
| 20:41.19 | leifmadsen | heh |
| 20:41.50 | drmessano | http://xkcd.com/200/ <--- WIN |
| 20:43.59 | *** join/#asterisk LittleFool (~LittleFoo@95.129.212.174) |
| 20:45.14 | Katty | thermodynamics. |
| 20:46.52 | drmessano | I have a buddy on the phone with Cisco tech support. I told him "When they ask you for the IOS version, tell them you just upgrade from IOS5 to IOS6.. I bet that never gets old" |
| 20:47.05 | Katty | teehee |
| 20:47.47 | Katty | drmessano: imma have to do that next time. |
| 20:47.50 | Katty | ^_^ |
| 20:47.55 | drmessano | hells yesh |
| 20:49.55 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 20:52.49 | Katty | o/~ thermodynamics in the air o/~ |
| 20:52.56 | jaxon007_ | CAn use MeetMeCount to count number of participants in confbridge ? |
| 20:54.39 | *** join/#asterisk darkskiez (~mhb@fsf/member/darkskiez) |
| 20:55.27 | *** join/#asterisk mobile_gordita (~Robert@66-87-92-159.pools.spcsdns.net) |
| 20:56.38 | mjordan | jaxon007_: no |
| 20:57.23 | mjordan | jaxon007_: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CONFBRIDGE_INFO |
| 20:57.44 | mjordan | also available in 10: https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_CONFBRIDGE_INFO |
| 20:58.54 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 21:02.48 | *** join/#asterisk droemel (~droemel@p5B03D36C.dip.t-dialin.net) |
| 21:06.22 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:08.11 | bitglue | [TK]D-Fender: so, it looks like adding a local channel as a queue member, and using the hF(...) options to Dial can do what I want. |
| 21:09.01 | bitglue | [TK]D-Fender: so an agent can either wait for the caller to hang up, or press * to hang up on them, without actually hanging up. Then F sends them to an agent-acw context that prompts them for a ticket number, and adds to the queue-log with QueueLog and some __ prefixed channel variables that were set earlier. |
| 21:12.30 | *** join/#asterisk airshock (~Adium@143.66.77.131) |
| 21:13.22 | [TK]D-Fender | bitglue: Didn't thing that * would let them pass though... tested it? |
| 21:13.27 | [TK]D-Fender | think* |
| 21:14.07 | bitglue | [TK]D-Fender: tested and working. |
| 21:14.17 | bitglue | [TK]D-Fender: i didn't think it would work either, based on the documentation, but it does. |
| 21:14.48 | [TK]D-Fender | bitglue: if you were to do the post-call method I was suggesting I'd have taken those vars in M() and pushed them on a stack to retrieve as the "last call" data for the classification dialplan |
| 21:15.05 | [TK]D-Fender | bitglue: Keep an eye out for timing gotcha's, etc |
| 21:15.33 | bitglue | yeah, this way just has less state to track all over |
| 21:15.33 | [TK]D-Fender | bitglue: Like of course... if the agent hangs up the call without doing"*", etc |
| 21:16.25 | bitglue | in that case i'm back to hitting them with small hammers i guess |
| 21:16.57 | bitglue | since the phone stays off-hook, i'm hoping ringinuse=no will prevent that agent from getting another call until they are done wrapping |
| 21:17.18 | bitglue | that should provide sufficient incentive for them to use * |
| 21:19.14 | *** join/#asterisk campee (~campee@ec2-107-21-209-242.compute-1.amazonaws.com) |
| 21:19.29 | bitglue | it seems essential to use Local/..../n and not just Local/... to get * to work, though. |
| 21:20.16 | campee | kind of an asterisk newbie here, so please be gentle :) if i run /usr/sbin/lsdahdi i see the following for every line on my PRI: PRI Clear (In use) (EC: MG2 - INACTIVE) RED but none of these lines should be "in use", is there a way to remove the call from that particular line or do i have a bigger problem? |
| 21:20.48 | [TK]D-Fender | bitglue: When using AQM you can also specify the state device more directly if its a more standard SIP peer for a phone, etc |
| 21:23.19 | WIMPy | campee: (In use) only means that Asterisk is bound to the channels. |
| 21:26.35 | campee | ah, ok. we can't seem to make outgoing calls after a reboot. i'm a sysadmin trying to help our phone guy. |
| 21:26.45 | WIMPy | And als long as you see a RED condition, you won;t get any calls anyway. |
| 21:27.02 | WIMPy | Sounds fun. |
| 21:27.20 | campee | i tried making an outgoing call and saw this in /var/log/asterisk/messages: TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to other trunks |
| 21:27.24 | campee | i thought that meant we were out of lines |
| 21:27.32 | *** part/#asterisk keycruncher (~keycrunch@c-174-59-233-118.hsd1.pa.comcast.net) |
| 21:27.59 | campee | are there any basic troubleshooting suggestions you can make? :) |
| 21:28.02 | WIMPy | No, that's just Asterisk giving misleading messages. |
| 21:28.21 | WIMPy | Yes, get rid of the RED alarm. |
| 21:28.29 | WIMPy | That is make the line work. |
| 21:28.48 | WIMPy | Where do you come from? Has it ever worked? New install? |
| 21:29.18 | campee | it was working fine and the system was rebooted last night by our phone guy. since that time it has stopped working |
| 21:29.41 | WIMPy | Maybe the other end has shut down the line. |
| 21:30.03 | campee | because we rebooted our server? |
| 21:30.17 | campee | like our end goes down so they shut down the line automatically and don't bring it back up? |
| 21:30.24 | WIMPy | Red alarm is a 'no link' type of situation. |
| 21:30.29 | anfecora | campee have you checked the crossconnect cable |
| 21:30.29 | *** part/#asterisk jaxon007_ (~jaxon007_@121.245.159.59) |
| 21:30.45 | WIMPy | I don't know. It can happen. |
| 21:30.49 | campee | no. what should i check for? |
| 21:30.49 | anfecora | did the phone guy made changes physically or remotelly |
| 21:31.04 | anfecora | maybe it got disconnected from your smartjack |
| 21:31.10 | anfecora | it happens alot |
| 21:31.17 | campee | he rebooted it and after it failed to work correctly after the reboot he swapped our PRI card for an older one we had lying around. when the older one failed to work he swapped it back to the original PRI card |
| 21:31.28 | campee | so both remotely and physically |
| 21:31.50 | WIMPy | Ask your telco if they shut the line. |
| 21:31.58 | campee | smartjack = the port in the wall that connects to the phone company? |
| 21:32.09 | WIMPy | Some switches disable lines if there are too many errors. |
| 21:32.18 | anfecora | if you loopback the connection does still shown as down |
| 21:32.32 | campee | based on the output of the above command is it reasonable to assume that the card is configured correctly? |
| 21:32.42 | campee | anfecora: what do you mean? like plug it in to a switch that i have lying around? |
| 21:33.00 | WIMPy | Unless the line parameters are wrong, yes. But if it used to work... |
| 21:33.09 | WIMPy | NO! |
| 21:33.19 | WIMPy | But a loopback plug would help. |
| 21:33.37 | campee | i don't know what that is.. is it some specialized piece of equipment? |
| 21:33.40 | anfecora | check if you see the card with dmesg |
| 21:33.57 | anfecora | a lookback usually comes with the card |
| 21:34.05 | WIMPy | A modular plug and two wires. |
| 21:34.09 | anfecora | but you can make une with a new rj connector |
| 21:34.09 | campee | i see this when i run dmesg, and it looks interesting: Status Change on Span 0 from 8900 to 8a00 |
| 21:34.12 | campee | Status Change on Span 0 from 8a00 to 8800 |
| 21:34.35 | campee | useful? |
| 21:34.48 | WIMPy | Not to me. |
| 21:35.09 | WIMPy | Any LEDs in the NT? |
| 21:35.12 | anfecora | this may help you build a loopback plug http://www.askemergent.com/how-to-make-a-t1pri-loopback-plug |
| 21:35.36 | anfecora | you need one to test the card an verify everything is ok on your side before troubleshooting with carrier |
| 21:35.39 | campee | interesting. the red alarms don't show anymore if i run "lsdahdi": PRI Clear (In use) (EC: MG2 - INACTIVE) |
| 21:36.05 | anfecora | do you see a digium card or tiger driver or extra network card on your dmesg output |
| 21:36.12 | WIMPy | Good. |
| 21:36.16 | anfecora | what about lspci |
| 21:36.25 | anfecora | which os do you have |
| 21:36.29 | campee | this is the kind of card i have from lspci: 01:01.0 Network controller: Rhino Equipment Corp. Rhino R2T1 |
| 21:36.41 | WIMPy | Oh, it might just have taken some time to get the line up again. |
| 21:36.45 | anfecora | so it is a rhino card no digium |
| 21:36.48 | WIMPy | Does it still not work? |
| 21:36.50 | campee | it's running some kind of modified version of CentOS release 5.7 (Final) |
| 21:36.53 | anfecora | you must check that the drivers are ok |
| 21:36.57 | anfecora | first |
| 21:37.45 | anfecora | if you restart dahdi what ouput do you get |
| 21:37.52 | anfecora | if you do service dahdi restart |
| 21:37.55 | campee | hm, apparently the phone consulting company that he contacted was able to get the lines up again, that's why the alarm went away |
| 21:38.06 | campee | waiting to hear what they did to get it working |
| 21:38.33 | WIMPy | the equivalent of ' no shut'? |
| 21:40.23 | *** join/#asterisk angryuser_laptop (~angryuser@2a02-8422-1230-bb00-04a6-2b75-e80a-c2a5.rev.sfr.net) |
| 21:42.34 | anfecora | anyone has good experiences with webrtc and asterisk 11 |
| 21:45.03 | anfecora | especially with beta2 |
| 21:45.18 | Qwell | ~poll |
| 21:45.18 | infobot | Script for automating Fidonet polls. URL: http://www.drmach.demon.co.uk/vashti/software/index.html |
| 21:45.21 | Qwell | ~polls |
| 21:45.21 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 21:46.00 | WIMPy | That account/domain doesn't even exist any more. |
| 21:46.37 | leifmadsen | infobot: forget poll |
| 21:48.28 | *** join/#asterisk lvlolvlo_ (~lvlolvlo@unaffiliated/lvlolvlo) |
| 21:48.30 | tzanger | oh man fidonet |
| 21:48.45 | WIMPy | Those were the days... |
| 21:53.43 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 21:54.04 | jpsharp | A friend of mine still runs a fidonet node on a dialup BBS. |
| 21:54.21 | jpsharp | Can't quite drag him out of the 1980s. |
| 21:54.33 | *** join/#asterisk jsjc (~Adium@54.152.78.188.dynamic.jazztel.es) |
| 21:55.00 | WIMPy | I don't, but it's all still in place. Would take just one doubleclick to start it again :-) |
| 21:55.38 | WIMPy | Well, I'd have to connect the TA again. |
| 21:55.43 | carrar | That 80's was fun |
| 21:55.49 | carrar | Big Hair!! |
| 21:56.02 | WIMPy | Yes. So many different networks. |
| 21:56.53 | WIMPy | But the old mailbox thing is still my active smtp server. |
| 21:58.51 | anfecora | there are lots of extenal usrobotics really cheap now if you want to connect again |
| 21:59.19 | WIMPy | I still have 3 ZyXEL 2864, two of them working. |
| 21:59.29 | anfecora | i couldn't afford any on those days |
| 22:00.29 | WIMPy | I got one from the very beginning as it was classified as PBX and you got a big refund from the telco. |
| 22:00.47 | tzanger | jpsharp: I used to be a point in 1:519 but that was a long long time ago |
| 22:00.53 | WIMPy | The 2nd came to me from a sysop friend who sut down his mailbox. |
| 22:01.09 | WIMPy | shut |
| 22:02.17 | WIMPy | The third I got from ebay not long agao because one of the PSUs broke and I got the whole thing for one buck. |
| 22:02.47 | tzanger | heh what do you use them for now? |
| 22:03.07 | WIMPy | Funny experiments :-) |
| 22:03.43 | WIMPy | I dug it out after I found that my phone was able to pull a software update, even tho I've got an NGN line. |
| 22:04.05 | WIMPy | Turns out that X.75 is stable enough to be used over VOIP. Even if it's dead slow then. |
| 22:05.58 | WIMPy | In the BBS years it also doubled as my voicemail. |
| 22:06.31 | *** join/#asterisk pcAngel (~yoink@S0106602ad07d2b78.vc.shawcable.net) |
| 22:12.24 | pcAngel | Hi guys, does anybody know how to get the active channel's SIP peer name in a dialplan? ie ${PEERNAME} or ${CHANNEL(PEER)} ? (I made those two up.... voipinfo.org's list of built in channel variables hasn't been much help) |
| 22:12.31 | WIMPy | Looking back, I can't believe how much I did way back then and how little I get done today :-( |
| 22:12.33 | pcAngel | It's to do a dbget based on the peer name |
| 22:13.07 | WIMPy | The answer is in 'core show function CHANNEL'. |
| 22:13.11 | *** join/#asterisk airshock (~Adium@user-10bj23e.cable.mindspring.com) |
| 22:13.21 | WIMPy | That's the documentation you should be looking at. |
| 22:14.55 | WIMPy | Hmm didn't we have a |
| 22:15.00 | WIMPy | ~voip-info |
| 22:15.00 | infobot | well, voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 22:15.31 | WIMPy | Hmm. No mention of outdated stuff? |
| 22:18.18 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 22:22.55 | pcAngel | Thank you WIMPy, I didn't know about that console command |
| 22:23.10 | pcAngel | btw the bug we were talking about last week got assigned to a developer for resolution already ^_- |
| 22:23.24 | pcAngel | Where an atxfer couldn't complete to any extension containing a # |
| 22:29.33 | WIMPy | Yes, I saw that. |
| 22:29.55 | WIMPy | Actually I commented on it. |
| 22:30.00 | *** join/#asterisk hardwire (~spencersr@199.167.193.204) |
| 22:31.15 | mjordan | pcAngel: just to set expectation, it isn't assigned. It's been acknowledged as an issue and moved out of Triage |
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| 22:57.58 | airshock | Hi all. Still having the problem I posted about here earlier today, where I've got FreePBX installed with Asterisk, have a few extensions, but nobody can call anyone else because the "from-internal" context does not exist. But it's referenced in /etc/asterisk/extensions_additional.conf. |
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| 23:02.33 | [TK]D-Fender | airshock: and is that file even included by extensions.conf? |
| 23:02.46 | [TK]D-Fender | airshock: because by itself that file is not inherently an Asterisk config file |
| 23:05.05 | airshock | [TK]D-Fender: At the bottom of extensions.conf there is an include line that says "#include extensions_additional.conf" so I'm assuming it's being included OK, though if Asterisk is reporting that the "from-internal" context cannot be found maybe there's an inclusion error. |
| 23:05.30 | [TK]D-Fender | And I told you PASTEBIN the dump from CLI |
| 23:09.42 | cmendes0101 | 'dialplan show' might help? |
| 23:10.54 | airshock | [TK]D-Fender: I can't paste the CLI output, my screen reader won't let me copy it because it's too large (too much information to highlight). |
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| 23:15.25 | [TK]D-Fender | "copty all to buffer" |
| 23:15.48 | [TK]D-Fender | This is 2 clicks in PuTTY for example |
| 23:16.29 | airshock | [TK]D-Fender: Actually using the Terminal app on a Mac. And there's more in my Terminal window then just the Asterisk console output. |
| 23:17.44 | [TK]D-Fender | If you're crippled from showing us anything I guess we can't help |
| 23:18.16 | [TK]D-Fender | Somethings not getting loaded. Syntax error, wrong file location, wrong permissions, SOMETHING... |
| 23:19.22 | airshock | [TK]D-Fender: Figured it out, try this: http://pastebin.com/gtQ4CPfm |
| 23:20.58 | [TK]D-Fender | Clearly no from-internal there |
| 23:23.11 | [TK]D-Fender | You are running a broken bastardized config. I see the huge mess that is the SAMPLE extensions.conf that * comes with along with the FreePBX bits. |
| 23:24.37 | Kobaz | borken |
| 23:24.38 | Kobaz | all borken |
| 23:25.39 | airshock | [TK]D-Fender: Here is extensions.conf (extensions_additional.conf to follow): http://pastesite.com/45031 |
| 23:26.16 | [TK]D-Fender | airshock: 800 lines of pure junk. That is not a basic FreePBX extensions.conf |
| 23:26.38 | [TK]D-Fender | airshock: Go take it up in #freepbx and maybe someone can walk you through rebuilding a proper one. |
| 23:26.47 | airshock | [TK]D-Fender: extensions_additional.conf: http://pastesite.com/45032 |
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