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| 01:23.50 | koffel | hello |
| 01:25.04 | ChannelZ | hi |
| 01:37.45 | koffel | on 11.11.0 for realtime sip device how would i find how to |
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| 02:06.53 | overyander | I want to be able to receive sms from any of my DID's and optionally reply. all of the sms gateways i've found online seem to be a web hosted solution and offer/charge for a lot of features that I would code myself in asterisk |
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| 08:17.37 | XATRIX | Hi, i still have a problem with bruteforcing my server... Any idea what can i do to fix it out ? |
| 08:17.52 | XATRIX | [2014-08-11 11:07:34] NOTICE[2781][C-00000473] chan_sip.c: Failed to authenticate device 111<sip:111@176.111.63.xxx>;tag=db659da8 |
| 08:18.40 | XATRIX | [2014-08-11 10:49:16] NOTICE[2781][C-0000046d] chan_sip.c: Failed to authenticate device 2014<sip:2014@176.111.63.xxx>;tag=daeabaae |
| 08:19.02 | XATRIX | 176.111.63.xxx - it's my own IP. How does he hijack it ? |
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| 08:23.48 | XATRIX | How can i make my asterisk to show the real ip of the registering peer ? |
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| 08:39.02 | kleszcz | XATRIX: asterisk 1.8 ? |
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| 08:43.15 | XATRIX | kleszcz: 11.11.0 |
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| 08:53.26 | kleszcz | XATRIX: http://wklej.to/LATQs |
| 08:53.46 | kleszcz | use this patch |
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| 09:14.05 | linocisco | hi al |
| 09:14.33 | linocisco | why should we use "Asterisk" ? to explain to non-IT guys |
| 09:15.16 | sekil | to not have to use avaya or cisco or similar crap |
| 09:31.57 | Edilberto_EXD124 | Cisco: 500 phone + 2 callmanagers = 400.000⬠+ maintenance ... in 5 years out of live ... in 7 years out of services |
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| 09:42.22 | fling | What is the right way to connect to skype? |
| 09:44.52 | sekil | use skype client |
| 09:46.55 | linocisco | why should we use "Asterisk" ? to explain to non-IT guys |
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| 10:01.43 | kleszcz | linocisco: http://www.asterisk.org/get-started |
| 10:02.49 | fling | sekil: and how to connect to skype client? Do I need siptosis? |
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| 10:33.35 | SirLouen | gm |
| 10:33.40 | fling | gm |
| 10:34.07 | r00f | ga |
| 10:34.15 | SirLouen | anyone knows if its possible to playback a file to 2 bridged channels at the same time? |
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| 11:43.14 | XATRIX | kleszcz: damn, i have asterisk from rpms. not from the sources |
| 11:47.08 | paolo_ | how can i sue multiple transport in chan_pjsip for one endpoint ? |
| 11:49.43 | paolo_ | if a sip packet is greater then 1300 bytes pjsip tries to send it per TCP. byz there is no transport |
| 11:53.49 | file | you configure multiple transports in pjsip.conf |
| 11:54.02 | file | add a tcp one |
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| 11:55.08 | paolo_ | in endpoint section ? transport=udp-one, tcp-one ? |
| 11:55.33 | file | if you explicitly specify a transport in an endpoint then it can't switch over |
| 11:55.49 | file | don't specify a transport in the endpoint and it'll choose |
| 11:56.12 | file | but you need to configure a TCP transport (type=transport, protocol=tcp) |
| 11:56.42 | paolo_ | file: ok. thanks |
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| 12:16.10 | paolo_ | file: without tranport it works. thank you again. |
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| 12:31.47 | keiths_ | I asked this last week, but want to try again. Anyone know of a way I can make a cisco SPA50X, ring on call waiting instead of the beep? |
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| 12:35.41 | [TK]D-Fender | keiths_: Go read their manual |
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| 13:36.23 | youjelly | hi TK |
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| 13:58.15 | lorenzo | hi! how do I handle a CHANUNAVAIL condition after WaitExten() ? |
| 13:59.49 | WIMPy | That question doesn't really make sense. Try again. |
| 13:59.58 | lorenzo | okay :) |
| 14:00.18 | lorenzo | let's say I have an extension, 123, which waits for a user to dial an extension with WaitExten() |
| 14:00.49 | lorenzo | sometimes the extension they dial is invalid, or the user is not registered, so how do I handle that by having a voice message |
| 14:00.59 | lorenzo | instead of the tone? |
| 14:01.18 | WIMPy | "the tone"? What tone? |
| 14:01.39 | WIMPy | For invalid extensions there's the special i extension. |
| 14:01.42 | lorenzo | the default one for subscriber absent |
| 14:01.58 | WIMPy | There isn't any. |
| 14:02.06 | [TK]D-Fender | There is no tone for that.... |
| 14:02.23 | lorenzo | hm, so what am I hearing must be done in the client |
| 14:02.35 | WIMPy | And WaitExten doesn produce a CHANUNAVAIL. Dial might. So that's where you handle that. |
| 14:02.44 | WIMPy | Possibly. |
| 14:03.12 | [TK]D-Fender | That is a DIALSTATUS perhaps.... |
| 14:03.18 | lorenzo | yeah |
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| 14:03.48 | [TK]D-Fender | You should probably check the validity BEFORE trying to actually dial. |
| 14:04.08 | WIMPy | HOw does that help? |
| 14:04.56 | WIMPy | It can still fail. Just wasted effort IMHO. |
| 14:05.12 | [TK]D-Fender | General best practices. Prevent failure rather that try to clean up after it. |
| 14:05.44 | [TK]D-Fender | Either way though... he's in the dialplan. chech what you have to check and do what you want to do with it... |
| 14:05.52 | [TK]D-Fender | check* |
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| 14:25.33 | Katty | good morning, [TK]D-Fender |
| 14:25.56 | [TK]D-Fender | Katty: Mew. |
| 14:27.28 | lorenzo | uhmm I know it's very basic |
| 14:27.47 | lorenzo | but Dial SIP/ doesn't allow extensions to be called directly by their number |
| 14:27.49 | lorenzo | is that right? |
| 14:28.30 | [TK]D-Fender | lorenzo: Never call a device an "extension" |
| 14:28.50 | [TK]D-Fender | lorenzo: And device names are nor restricted to being numeric. |
| 14:29.10 | [TK]D-Fender | not* |
| 14:29.32 | lorenzo | I see many people use digits in the sip.conf for identifying users |
| 14:29.45 | lorenzo | e.g. [101] [102] |
| 14:29.51 | lorenzo | but I'm using usernames |
| 14:30.06 | lorenzo | then in extensions.conf I have them "mapped" so that 101 points to lorenzo for example |
| 14:30.28 | lorenzo | does that sound okay? |
| 14:32.10 | [TK]D-Fender | lorenzo: That isn't "mapped". That you made an extension to dial a named device isn't really an association except logically to you. It means nothing special to Asterisk. |
| 14:32.46 | [TK]D-Fender | lorenzo: Three is no lookup you can do to treat this as an "association" or "mapping" |
| 14:33.13 | lorenzo | ah okay, so there's nothing like a lookup table to perform search from extension ID to username in sip.conf |
| 14:33.21 | [TK]D-Fender | no. |
| 14:33.32 | SirLouen | i'm having an issue, for example, it is possible to run the Dial property M() to execute a macro when call is bridged and in that macro start executing scripts while the call is in progress? |
| 14:33.45 | lorenzo | and how do you handle incoming extension selection? you must have extension numbers as usernames in sip.conf? |
| 14:34.08 | [TK]D-Fender | lorenzo: No, you ahve to actually make extensions to match what they dial. |
| 14:34.49 | lorenzo | but, I do have extensions |
| 14:34.55 | [TK]D-Fender | SirLouen: Nothing will execute against the background of that channel. M() needs to complete for bridging to happen |
| 14:35.00 | lorenzo | but Dial refuses them as they're numeric only |
| 14:35.12 | lorenzo | maybe I should use GoTo users, number |
| 14:35.16 | [TK]D-Fender | lorenzo: You are not making any sens. SHOW us what you've made |
| 14:35.20 | [TK]D-Fender | ~pb |
| 14:35.21 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:35.56 | lorenzo | ok let me paste it |
| 14:36.12 | [TK]D-Fender | SirLouen: You could Originate a new channel in your macro and that will be processed along-side, but it will not be technically "bound" the the channel that launches it. |
| 14:37.02 | lorenzo | http://pastebin.com/aNYRFaS4 |
| 14:37.25 | lorenzo | (696 is just a test, then it will be the default ivr) |
| 14:37.41 | [TK]D-Fender | lorenzo: I see you've already stopped using "waitexten" like you had told us you were using... |
| 14:38.09 | lorenzo | yeah you told me it didn't return status codes basically |
| 14:38.13 | [TK]D-Fender | lorenzo: in [demo-menu] you shoudl INCLUDE your other context where you already have extens to dial your devices. |
| 14:38.18 | lorenzo | so I replaced it with Dial. WaitExten worked perfectly :P |
| 14:38.22 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 14:38.45 | [TK]D-Fender | You don't "replace" it with dial... |
| 14:38.58 | [TK]D-Fender | lorenzo: You don't seem to understand the basics of these dialplan apps. |
| 14:39.08 | lorenzo | I don't :> installed this yesterday |
| 14:39.38 | [TK]D-Fender | lorenzo: "Dial" .... DIALS. You don't get a status for WAITING for input from a user. DIALSTSTUS is a return result from actually trying to call a device. This has nothing to do with getting the input from the user as to what they want to do. |
| 14:40.14 | [TK]D-Fender | lorenzo: Get input. Match it. Take action. Check result. |
| 14:40.29 | [TK]D-Fender | lorenzo: Waitexten = Get input. Dial = Take Action. |
| 14:41.12 | [TK]D-Fender | lorenzo: As for what you should do for this : |
| 14:41.13 | [TK]D-Fender | [10:38][TK]D-Fenderlorenzo: in [demo-menu] you shoudl INCLUDE your other context where you already have extens to dial your devices. |
| 14:41.30 | [TK]D-Fender | lorenzo: And go back to using WaitExten in there. |
| 14:41.35 | lorenzo | okay, I've added include => users |
| 14:41.54 | [TK]D-Fender | and get exten=>696,1,GoTo(demo-menu,s,1) out of [users] |
| 14:42.01 | [TK]D-Fender | or basically just split them up |
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| 14:42.40 | [TK]D-Fender | your phones should point to a context that includes the context where you have extens to dial between your devices, and other contexts that have extens for extra features like testing your menu |
| 14:42.55 | [TK]D-Fender | lorenzo: Your IVR should include the context that has just the extens to dial your devices. |
| 14:43.19 | [TK]D-Fender | lorenzo: So that you can't go in cirecles dialing 696 from the IVR |
| 14:43.28 | lorenzo | I see, so in my [users] I should include another test-menu context |
| 14:43.34 | lorenzo | ? |
| 14:43.54 | lorenzo | is that what you mean by splitting them? |
| 14:45.08 | lorenzo | with exten=>696 in it |
| 14:45.30 | [TK]D-Fender | lorenzo: http://pastebin.com/CrhASq98 |
| 14:45.53 | [TK]D-Fender | lorenzo: And you're welcome... this is today's free hand-out.... |
| 14:46.00 | lorenzo | :* |
| 14:46.19 | lorenzo | thanks |
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| 14:53.15 | lorenzo | this works perfectly |
| 14:53.23 | lorenzo | I wonder how they manage huuuge setups |
| 14:53.31 | lorenzo | I mean you have to calculate every possible path from every other path etc |
| 14:53.34 | lorenzo | to avoid loops |
| 14:56.14 | [TK]D-Fender | lorenzo: How many paths should you really have? In most scenarios that is a pretty small number. |
| 14:57.05 | [TK]D-Fender | lorenzo: I may have 3 groups of users, 3 classes of user-features, 4 classes of rights to dialing out.... that's pretty basic stuff... |
| 14:57.29 | lorenzo | yeah I'm just curious :) |
| 14:57.41 | lorenzo | but think of a phone company helpdesk, menus and submenus |
| 14:57.42 | [TK]D-Fender | lorenzo: It's as simple as the structure you need to implement. |
| 14:58.14 | *** join/#asterisk ThatDamnRanga (~wiretap@unaffiliated/wiretap) |
| 14:58.22 | [TK]D-Fender | Sub-menus usually just lead another level deeper. Then have options to do specific things that don't need to be anywhere but in that context itself. |
| 14:59.19 | [TK]D-Fender | lorenzo: Like the [users] I gave you. I just included the raw match for your IVR test there. I could have put it in another context named [testing] or somesuch, but since you only had one thing I just left it there directly. |
| 15:00.35 | lorenzo | yeah I modified it |
| 15:02.03 | lorenzo | http://pastebin.com/WqY0uRdK |
| 15:02.29 | lorenzo | should basically be a tree |
| 15:02.39 | [TK]D-Fender | exten => s,n,Read(Digits,,3) <-- get rid of... don't need it anymore |
| 15:02.48 | lorenzo | ah yeah |
| 15:03.33 | [TK]D-Fender | exten=>666,1,GoTo(demo-hold,s,1) ; music on hold demo <- this could be just an include. |
| 15:04.01 | lorenzo | yeah or I should replace 666,1 with s,1 |
| 15:04.09 | [TK]D-Fender | lorenzo: You could. |
| 15:04.19 | lorenzo | yay I'm starting to learn ^_^ |
| 15:04.35 | [TK]D-Fender | lorenzo: That would make it a little mor logical if you made another pattern somewhere else to lead to the same functionality |
| 15:04.39 | [TK]D-Fender | more* |
| 15:05.04 | [TK]D-Fender | So your phone users would dial it like a "code", but you could get to it with a simpler pattern from an IVR, etc. |
| 15:05.24 | lorenzo | another thing I was concerned about was security, sometimes I see random users |
| 15:05.29 | lorenzo | trying to log in with 1004, 2000000 |
| 15:06.06 | lorenzo | I've set alwaysauthreject=yes |
| 15:06.15 | [TK]D-Fender | lorenzo: "fail2ban" <- logger.conf "security". Read up on it in the sample config for * 10+ |
| 15:06.27 | lorenzo | I don't think there are any guest users enabled by default |
| 15:06.38 | [TK]D-Fender | lorenzo: Because they'll keep trying until they do find something that seems valid and then wear it down. |
| 15:07.02 | lorenzo | I've also noticed that it opened up many ports, including one for Cisco compatibility (?) |
| 15:07.17 | lorenzo | I've set iptables to allow in input only 22, 5060 tcp/udp, 10000-20000 udp |
| 15:07.22 | lorenzo | and reject everything else |
| 15:08.10 | [TK]D-Fender | What port? |
| 15:08.21 | lorenzo | 2000 |
| 15:08.32 | lorenzo | Skinny |
| 15:08.34 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 15:08.56 | [TK]D-Fender | yup. If you don't plan on using it add "noload => chan_skinny.so" to modules.conf |
| 15:10.12 | lorenzo | I've tried but it didn't work |
| 15:10.17 | lorenzo | maybe it should be before the autoload=yes ? |
| 15:10.32 | [TK]D-Fender | Shouldn't really matter... |
| 15:10.40 | [TK]D-Fender | perhaps you're using chan_sccp.so |
| 15:10.53 | [TK]D-Fender | I'm not sure which one is standard, but they are both for the same protocol |
| 15:11.00 | [TK]D-Fender | check your modules folder to be sure |
| 15:12.37 | lorenzo | yeah the other name fixed it |
| 15:12.44 | lorenzo | I see I still have IAX going |
| 15:12.44 | *** join/#asterisk FlashDel (~benedict@static-87-79-94-28.netcologne.de) |
| 15:13.00 | lorenzo | not really using it :/ |
| 15:13.37 | [TK]D-Fender | kill that off too then... |
| 15:14.23 | FlashDel | hi folks! I got a problem: I have an asterisk 11.5.0 installed which works fine so far, but today there occured an error: "Call to peer '42' rejected due to usage limit of 1" here is the whole output: http://pastebin.de/129074 Can somebody help me with that? |
| 15:14.28 | lorenzo | done |
| 15:15.18 | *** join/#asterisk u0m3 (~u0m3@92.80.93.41) |
| 15:15.46 | file | FlashDel, have you configured 42 in sip.conf with call limits? |
| 15:17.05 | lorenzo | this one is odd [Aug 11 11:15:02] NOTICE[1251][C-00000001]: chan_sip.c:25679 handle_request_invite: Failed to authenticate device 2000000<sip:2000000@81.4.109.xx>;tag=e727854a |
| 15:17.07 | FlashDel | file: oh, yes :S |
| 15:17.12 | lorenzo | the IP address is the same as the asterisk server :/ |
| 15:17.25 | FlashDel | file: let me guess, i have to delete this entry?^^ |
| 15:17.39 | file | FlashDel, if you tell it to limit it to 1... then it'll limit it to 1 ^_^ |
| 15:18.10 | lorenzo | maybe someone spoofing the address? |
| 15:18.24 | [TK]D-Fender | lorenzo: that is not the IP it's coming from. They are trying to say "I'm 2000000 on your server". |
| 15:18.39 | [TK]D-Fender | lorenzo: when you look at "sip set debug on" you'll see the source packets. |
| 15:18.44 | lorenzo | ah, guess I should look in /var/log/ then |
| 15:19.04 | FlashDel | file: i see :-) ... it works now, thank you! |
| 15:19.06 | [TK]D-Fender | lorenzo: And when you enable proper security logging you'll be able to use it with things like fail2ban to firewall them off automatically, etc |
| 15:19.07 | *** join/#asterisk Nopal (~Gnu@177.237.248.56) |
| 15:19.11 | lorenzo | yes |
| 15:21.05 | lorenzo | http://highsecurity.blogspot.it/2013/09/upgrade-to-asterisk-11-and-fail2ban-088.html |
| 15:21.08 | lorenzo | I'm following this |
| 15:23.54 | [TK]D-Fender | lorenzo: You should be on your way then. Make sure to whitelist enough to prevent locking yourself out accidentally. |
| 15:24.17 | lorenzo | yeah I'll whitelist the whole openvpn class |
| 15:34.16 | *** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 15:39.42 | hexanol | I'm experimenting with cel_odbc, using a postgresql database, and a cel table with an "eventtime" column of type timestamp |
| 15:39.57 | hexanol | my problem is that when you look at cel_odbc.conf, it seems like it supports microseconds precision for the eventtime |
| 15:40.10 | hexanol | but then, when I test, I don't have such precision (i.e. I only have second precision) |
| 15:40.26 | hexanol | which is a bit unfortunate |
| 15:40.29 | hexanol | so I was wondering |
| 15:41.02 | hexanol | is that a kinda a bug, or a limitation of cel_odbc when using a column of type "timestamp", that it can't store the microseconds ? |
| 15:44.34 | hexanol | if I'm using a varchar type for my column eventtime, than I get... milliseconds precision |
| 15:44.58 | hexanol | (but I don't want to use a varchar type for the eventtime column) |
| 15:45.00 | *** join/#asterisk timahvo1 (~rogue@197.237.134.227) |
| 15:46.31 | *** join/#asterisk init_1045 (~init_1045@193.158.213.219) |
| 15:49.01 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:a42:224:1dff:fecd:234c) |
| 15:49.43 | *** join/#asterisk karlh626 (~karlh626@addr-199.21.194.73.nptpop-cmts-cable-sub.rdns-bnin.net) |
| 15:56.15 | init_1045 | hi, i'd like to implement an answering machine with asterisk, where i can have different answers depending on the time of the call. Also in any case the machine should first take the call and welcome the caller. I have a ISDN Line. Do I need a HFC-Card for that purpose? |
| 15:57.35 | wdoekes | hexanol: try this http://fpaste.org/124696/ |
| 15:58.20 | [TK]D-Fender | init_1045: Those are supported.... |
| 15:59.46 | hexanol | wdoekes: k, I'm recompiling the module |
| 16:00.03 | WIMPy | init_1045: record them yourself. |
| 16:00.59 | WIMPy | And from what we saw last week, it looks like you need to use LCR to use the bog standard cards. |
| 16:01.03 | init_1045 | [TK]D-Fenderis there already a solution where such a purpose can be taken by a raspberry PI? |
| 16:01.34 | hexanol | wdoekes: hum... it doesn't seem to work, I'll double check |
| 16:01.44 | WIMPy | There are even Asterisk Distors for the Pi. |
| 16:01.49 | hexanol | my mistake, sorry |
| 16:02.53 | init_1045 | what about the hardware then, wimpy? Do i need something spezial for ISDN and the Pi? |
| 16:03.25 | WIMPy | An USB adaptor, obviousely. |
| 16:03.38 | hexanol | wdoekes: hum, now I'm getting something in the CEL but with still only second precision |
| 16:03.49 | WIMPy | Ant they definitely only work with either mISDN version. |
| 16:04.29 | [TK]D-Fender | Xorcom also makes USB ISDN interfaces taht work with DAHDI. |
| 16:04.44 | WIMPy | True. |
| 16:04.46 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.hsi01.unitymediagroup.de) |
| 16:04.56 | WIMPy | But a very different price range. |
| 16:05.02 | [TK]D-Fender | Indeed. |
| 16:05.18 | [TK]D-Fender | But parger scale as well... which I suspect is not the need. |
| 16:05.25 | WIMPy | And it didn't work that well for me, either. |
| 16:05.35 | [TK]D-Fender | WIMPy: I know you know the BRI market quite well.... |
| 16:05.50 | hexanol | hum, I do see "{ts '2014-08-11 12:05:36.000'}" |
| 16:06.00 | hexanol | when I put verbose level to 11, to have more output from cel_odbc |
| 16:06.13 | hexanol | ok... so there's milliseconds, but it's 000 |
| 16:06.22 | init_1045 | so what would you guys recommend? |
| 16:07.11 | WIMPy | A nortmal off the shelf USB dongle with an HFC S USB. |
| 16:07.18 | SirLouen | one question: it is possible to force launching a dtmf feature in call for example, after 10 seconds ?¿ |
| 16:07.35 | wdoekes | hexanol: ok, I didn't really test that or anything. just a quick stab |
| 16:07.50 | [TK]D-Fender | SirLTo do what? |
| 16:08.12 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-kuxuictpmdgvmhgz) |
| 16:08.18 | SirLouen | [TK]D-Fender launch something within the first 10 seconds of the call started like a feature |
| 16:08.38 | [TK]D-Fender | SirLouen: To do what? |
| 16:08.40 | SirLouen | program an execution |
| 16:09.31 | SirLouen | [TK]D-Fender the only thing i've been able to program is the "warning message" with the L flag |
| 16:09.40 | [TK]D-Fender | SirLouen: I already told you you can Originate another channe from M() and then let the call continue... |
| 16:09.56 | SirLouen | you did not mention me before? |
| 16:10.10 | SirLouen | you can originate from a macro you mean? |
| 16:10.33 | [TK]D-Fender | [10:36][TK]D-FenderSirLouen: You could Originate a new channel in your macro and that will be processed along-side, but it will not be technically "bound" the the channel that launches it. |
| 16:10.35 | [TK]D-Fender | ^^^^^^^^^ |
| 16:10.49 | SirLouen | [TK]D-Fender ok |
| 16:11.11 | SirLouen | [TK]D-Fender you mean something like M(newevent) [macro-newevent] exten.... Originate(....? |
| 16:11.32 | [TK]D-Fender | That is exactly the sort of thing I just said/ |
| 16:11.47 | WIMPy | Sounds like you should look into AMI. |
| 16:12.01 | SirLouen | but how you can postpone it like 10 seconds after the call has been bridged? |
| 16:12.27 | WIMPy | Wait(10) |
| 16:12.42 | SirLouen | WIMPy then the call wont bridge until the 10 seconds has passed |
| 16:12.59 | WIMPy | in the new call, off course. |
| 16:13.32 | SirLouen | I mean, if you do the Dial(...M(newevent)) [macro-newevent] exten => ..... Wait(10) |
| 16:13.42 | hexanol | wdoekes: but do you think it's a bug that cel_odbc doesn't store at least microseconds precision ? |
| 16:13.44 | SirLouen | then the Dial wont bridge util the macro has ended |
| 16:13.49 | hexanol | hum, i Mean, millisecond precision |
| 16:13.59 | WIMPy | correct. |
| 16:14.01 | [TK]D-Fender | SirLouen: You will wait in the ORIGINATED channel. |
| 16:14.12 | [TK]D-Fender | SirLouen: NOT in the Macro you call from Dial |
| 16:15.12 | SirLouen | [TK]D-Fender i dont understand it |
| 16:15.13 | hexanol | wdoekes: ok, I got milliseconds precision by modifying your patch |
| 16:15.51 | [TK]D-Fender | SirLouen: ORIGINATE a LOCAL CHANNEL... then in THERE add your delay, etc.... |
| 16:16.04 | hexanol | seems with a problem with ast_strftime |
| 16:16.14 | hexanol | that is only outputting millisecond precision |
| 16:16.31 | hexanol | with %q |
| 16:16.39 | hexanol | anyway, I got to go, I'll be back later... |
| 16:17.08 | SirLouen | [TK]D-Fender http://pastebin.com/dHe1N7SP |
| 16:17.20 | SirLouen | if you do this, then the dial wont be bridged |
| 16:17.54 | [TK]D-Fender | SirLouen: What par of "don't put the wait BEFORE the Originate" are you having trouble with? |
| 16:18.01 | SirLouen | ok |
| 16:18.08 | SirLouen | i think i undesrtand it |
| 16:18.19 | [TK]D-Fender | SirLouen: same => n,Originate(......) <- IN the processing of this NEW CHANNEL |
| 16:19.49 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 16:21.17 | init_1045 | WIMPy: the Draytek minivigor128 is the only one i could find. Do you know others? |
| 16:22.32 | WIMPy | Trust or Billion. |
| 16:22.49 | SirLouen | [TK]D-Fender i think i have it now |
| 16:23.29 | SirLouen | [TK]D-Fender how can i get inside the macro the peer name? |
| 16:23.43 | SirLouen | the peer name of the call |
| 16:24.18 | [TK]D-Fender | SirLouen: Use AMI or a call-file to lauch the call, not the Originate dialplan app. Then you can pass it as a variable. |
| 16:25.26 | SirLouen | you mean not the Dial app ? |
| 16:26.12 | [TK]D-Fender | "not the Originate dialplan app" |
| 16:26.27 | SirLouen | but i need the peername before the originate lauches |
| 16:26.28 | *** join/#asterisk kaziklu-bey (0c0adbdf@gateway/web/freenode/ip.12.10.219.223) |
| 16:26.30 | SirLouen | launches |
| 16:26.54 | SirLouen | the peername of the Dial |
| 16:27.02 | [TK]D-Fender | SirLouen: PASS IT when you issue the call-file or AMI Originate. You cannot pass variables with the DIALPLAN APP, or CLI COMMAND versions.... |
| 16:29.40 | init_1045 | Wimpy: Thank you, i just read that i could also use an old fritz!box, so i guess i'll go with that plan first. |
| 16:31.22 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 16:31.32 | WIMPy | init_1045: Yes, but the only reliable way is as a SIP gateway. |
| 16:31.56 | SirLouen | [TK]D-Fender thx im going to try it |
| 16:45.04 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 16:51.19 | *** join/#asterisk c|oneman (cloneman@1337.montrealdark.com) |
| 16:54.30 | ipengineer | Does anyone have any thoughts as to why my /var/log/asterisk/messages file keeps getting set back to root? Asterisk cant write to it at that point. |
| 17:01.41 | [TK]D-Fender | ipengineer: Look at your logrotate..... |
| 17:02.46 | ipengineer | [TK]D-Fender: create 640 root root I guess is the problem. |
| 17:04.19 | *** join/#asterisk JerJer (~Adium@asterisk/original-h323-guy/JerJer) |
| 17:09.38 | *** join/#asterisk metakungfu (~textual@209.49.226.194) |
| 17:24.56 | *** join/#asterisk W0rmDr1nk (~wormdrink@unaffiliated/wormdrink) |
| 17:25.13 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 17:46.26 | *** join/#asterisk SirLouen (~pickupdog@95.63.247.9) |
| 17:46.45 | SirLouen | [TK]D-Fender everything great thx |
| 17:48.09 | SirLouen | it is possible to set the language for the callee in the dialplan? |
| 17:49.06 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 17:50.23 | mjordan | SirLouen: CHANNEL function, plus a pre-dial handler. |
| 17:50.42 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers |
| 17:50.42 | WIMPy | CHANNEL(language) |
| 17:50.49 | *** join/#asterisk Defraz (~Defraz@24-117-69-71.cpe.cableone.net) |
| 17:53.22 | SirLouen | mjordan interesting thx! |
| 17:55.11 | SirLouen | mjordan but you set the language in the "callee handler exten" the same? Set(CHANNEL(language)=fr) ?? |
| 18:00.54 | mjordan | yes |
| 18:01.23 | mjordan | it just executes a subroutine on either the callee or the caller (depending on which option you use) prior to dialling the channel. For setting up things like languages on a channel, it's a good time to do it. |
| 18:02.15 | *** join/#asterisk metakungfu (~textual@209.49.226.194) |
| 18:04.52 | SirLouen | mjordan perfect it works awesome |
| 18:13.48 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 18:15.44 | *** join/#asterisk areski (~areski@80.174.128.86.dyn.user.ono.com) |
| 18:23.07 | *** join/#asterisk JerJer (~Adium@asterisk/original-h323-guy/JerJer) |
| 18:24.06 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 18:43.18 | lorenzo | <PROTECTED> |
| 18:43.22 | lorenzo | just wrote this if anyone needs it :) |
| 18:43.35 | lorenzo | prints username and ip of succcessful logins |
| 18:44.08 | WIMPy | I get them from AMI in realtime. |
| 18:45.25 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 18:46.06 | lorenzo | WIMPy: I still have to fiddle with that |
| 18:46.55 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 18:47.33 | rrittgarn | So my experience with Dahdi is limited. I just installed a digium A8B in my server and show it in dahdi_hardware, but not in dahdi_genconf or dahdi_span_assignments auto. What information do you need to help troubleshoot? I had this working at one point in a different server, but seem to have lost whatever configuration changes I made. |
| 18:47.56 | WIMPy | lorenzo: you should. It makes Asteriskso much better. |
| 18:48.40 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 18:49.16 | lorenzo | WIMPy: but do you use a web interface? |
| 18:49.42 | WIMPy | lorenzo: For what? |
| 18:50.49 | rrittgarn | also, is /etc/init.d/dahdi status supposed to show anything? |
| 18:52.11 | lorenzo | WIMPy: for checking out this stuff |
| 18:52.15 | lorenzo | like logged in users |
| 18:52.18 | lorenzo | ongoing calls etc |
| 18:52.47 | WIMPy | lorenzo: Yes. |
| 18:52.56 | WIMPy | Well, account status that is. |
| 18:53.13 | WIMPy | No ongoing calls. |
| 18:54.47 | [TK]D-Fender | WIMPy: You don't use AMI to check on active channels? |
| 18:55.08 | WIMPy | no |
| 19:00.20 | [TK]D-Fender | WIMPy: Guess if you don't ahve a need.. it is good for that though. I usually parse "core show channels concise" rather than the more native channel dumps |
| 19:03.04 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:03.27 | WIMPy | If I wanted it, I'd listen for the events. But I don't find that information too interesting. |
| 19:09.48 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 19:11.36 | [TK]D-Fender | WIMPy: I use if for phone-browser BLF sometimes. |
| 19:11.42 | lorenzo | hmm |
| 19:11.51 | lorenzo | I have 3 devices with the same public ip address under nat |
| 19:11.55 | [TK]D-Fender | I think I may have subbed that for a "core show hints" dump.... |
| 19:11.58 | lorenzo | but there doesn't seem to be a way for them to pass audio |
| 19:12.00 | lorenzo | SIP works though |
| 19:12.03 | lorenzo | nat=yes is set |
| 19:19.29 | lorenzo | looks like only one of the three devices works |
| 19:25.01 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 19:28.08 | *** join/#asterisk felimwhiteley_ (~quassel@89.101.203.26) |
| 19:31.43 | *** join/#asterisk Defraz (~Defraz@205.185.92.236) |
| 19:32.31 | *** join/#asterisk usnavi (~DevWork2@static-100-32-93-142.lsanca.fios.verizon.net) |
| 19:33.31 | usnavi | Hi, I'm getting some intermittent issues with some of my hard phones.. i know its/nat firewall related, but it works 99% of the time so trying to figure out whats the deal. In my sip peer for outbound calls, I have nat=no, but I do use a nat, whats the significance of nat=XX? |
| 19:35.18 | [TK]D-Fender | usnavi: Virtually no provider is NAT'd so the peer you use to them should be "nat=no" |
| 19:35.32 | usnavi | ok so its to them, gotcha. |
| 19:35.38 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 19:35.40 | VultureZ | usnavi, if you can describe the "issues"? |
| 19:35.49 | VultureZ | *remove the if you |
| 19:36.15 | VultureZ | typically nat issues I see 1 way audio problems crop up |
| 19:36.46 | usnavi | on some devices if you receive or dial a call, you will hear no audio, but alledgedly sometimes that can hear you fine. Also when dialing a call, it will take forever to ring, like nothing happens. |
| 19:36.58 | usnavi | That is intermittent however. |
| 19:37.24 | usnavi | Our firewall rule table was filling up, I adjusted it, not sure if it filled up since. |
| 19:38.19 | usnavi | We were having reinvite issues, so I tried disabling all reinvites, there was a patch in the asterisk tree related to it being on despite having it disabled. |
| 19:38.36 | usnavi | Just to narrow the cases of nat to solve down to the asterisks server itself. |
| 19:45.02 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 19:52.54 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:55.11 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 19:56.27 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:a42:224:1dff:fecd:234c) |
| 19:58.19 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 20:04.55 | VultureZ | usnavi, check your rtp.conf file and see what your port range is set to |
| 20:05.20 | usnavi | 20-30k |
| 20:05.22 | VultureZ | also, I would make sure your router is using latest firmware, and any speed boost/voip shaping is turned off |
| 20:05.33 | usnavi | Our router is a freebsd box. |
| 20:05.55 | usnavi | I think its hitting its state limit again |
| 20:06.47 | VultureZ | and unable to dynamically assign those RTP ports... that might make sense. But your symptoms are similar with when an RTP isn't translated properly through the nat |
| 20:07.09 | usnavi | yea i agree. |
| 20:07.53 | usnavi | what about third parties using ports <20000 and >30000 |
| 20:07.56 | usnavi | I've seen that before. |
| 20:10.36 | lorenzo | fixed my no audio issue behind nat with canreinvite=no |
| 20:10.37 | lorenzo | qualify=yes |
| 20:11.22 | WIMPy | thinks the enhanced cross compiling stuff in 11.11 is too good. I just got a segfault when trying to start it. |
| 20:12.09 | VultureZ | usnavi, just to be clear are these SIP clients local or remote? |
| 20:12.23 | usnavi | local, but on a different subnet |
| 20:12.32 | VultureZ | oh rly... |
| 20:12.39 | VultureZ | localnet is specified? |
| 20:12.41 | usnavi | the calls are always remote ones. |
| 20:12.43 | usnavi | yes |
| 20:13.03 | VultureZ | does the server have non-nat access to the sip trunk? |
| 20:13.04 | usnavi | localnet=10.42.0.0/21 |
| 20:13.26 | usnavi | let me verify, its been awhile since I have looked at how they tied that in |
| 20:15.31 | VultureZ | sounds good, also, do you have recordings of the calls? You might want to check to see if the recorded audio is there since it would record at the server level. If there is audio it is the local network, if there is not then it is likley the trunk |
| 20:17.30 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 20:17.31 | *** join/#asterisk SuPeRSoAKeR (~supersoak@users.757.org) |
| 20:18.38 | usnavi | We have a wireless sip provider. They have some wifi beam on the rough, and a router board in our server room, our router has an ip 10.149.248.91 and we route all sip traffic over that interface, * has an externaddr with whatever the router boards public IP is |
| 20:18.44 | usnavi | roof* |
| 20:18.46 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 20:19.17 | SuPeRSoAKeR | Hello #asterisk, I was woundering if Queue starts a new channel when dialing the members and if channel scope variables are defined, otherwise, is the a vairable set that i can see that the context was invocked by the Queue app? |
| 20:19.40 | usnavi | its either nat isn't 100% setup right, or the rules, or the PF state table is spilling, leaving some NAT connections dropped. |
| 20:20.56 | VultureZ | usnavi, you have wireless SIP trunking? How is the latency on that? |
| 20:21.09 | usnavi | Fine. |
| 20:21.20 | usnavi | They have always provided us with wireless VOIP |
| 20:21.27 | usnavi | We switched to SIP only over it. |
| 20:21.34 | usnavi | And we got a fibre data in the building. |
| 20:21.48 | VultureZ | hey if it works! |
| 20:21.50 | usnavi | So we route just phone traffic from our gateway, to the microwave link. |
| 20:22.16 | VultureZ | have you noticed that it happens during high load? |
| 20:22.21 | usnavi | The latency is basically non existant, I've had two ex veteran phone company line men types use our phones and can't even tell its voip OR wireless. |
| 20:22.47 | VultureZ | usnavi, yea I am not knocking it, just never tried it :) |
| 20:22.58 | usnavi | didnt think you where. |
| 20:23.00 | usnavi | Just describing. |
| 20:23.29 | usnavi | We wouldn't do it normally, but we had a large remainder of a pay it all or keep it type contract with them |
| 20:23.54 | VultureZ | ah yes... |
| 20:23.55 | usnavi | ( we signed 5 years on a symmetric 6/6 connection.... verizon laid 200/200 lines a month later!! ) |
| 20:24.23 | usnavi | So we told them, we don't want the 6/6, we just want SIP now. They didn't normally do that kind of thing ( they have sip backend, but definitely not a sip only account with a customer etc... ) |
| 20:24.43 | usnavi | So they dropped our bill considerably, but we will fulfill the remaining time with this. |
| 20:24.50 | VultureZ | did you see that Q about high volume during issues? |
| 20:25.03 | usnavi | uhh |
| 20:25.08 | usnavi | No I didnt. |
| 20:25.16 | VultureZ | <PROTECTED> |
| 20:25.24 | usnavi | Whats high load? My no audio issues? |
| 20:25.27 | VultureZ | like high activity |
| 20:25.34 | usnavi | Define high activity :) |
| 20:25.37 | VultureZ | when X+ NAT connections are open |
| 20:25.44 | VultureZ | high for your local network |
| 20:25.55 | usnavi | Well I've seen my network start dropping states in the last two months. |
| 20:25.57 | VultureZ | basically do you notice any trends to when it occurs |
| 20:26.02 | usnavi | I doubled it 4 weeks ago. |
| 20:26.17 | usnavi | I checked today after this was brought to my attention, it was at 18k. |
| 20:26.25 | usnavi | Which is dangerously close to the 20000 i doubled it to |
| 20:26.26 | VultureZ | well it is very likley the issue with the states, a dropped state will lead to a call hanging like that |
| 20:26.33 | usnavi | yea thats what I thought. |
| 20:26.47 | usnavi | Anyways, I got the states back down to 750~ |
| 20:26.56 | VultureZ | any reported issues since? |
| 20:26.58 | usnavi | So I'm going to wait it out to see if this happens. |
| 20:27.03 | usnavi | I just did it 30 minutes ago |
| 20:27.13 | usnavi | Going to have to wait a week or so to be sure. |
| 20:27.53 | VultureZ | yea, but you know how it is... one person starts saying "I have been having this or that issue..." next thing you know everyone is claiming it lol |
| 20:28.22 | usnavi | Yea. I know who to trust :) |
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| 20:48.47 | usnavi | yea this is looking much cleaner. 730 entries still. |
| 20:51.28 | SuPeRSoAKeR | VultureZ: any advice for me? |
| 20:52.56 | VultureZ | supersoaker, that is a very specific question and I could not tell you without digging in the code or running tests myself |
| 20:53.06 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:53.37 | VultureZ | supersoaker, but I am pretty sure there is a way to access them, because I know we track the call through the queue and back |
| 20:58.16 | supersoaker | VultureZ: Looks like the scope of the __Vars to not carry over to the channel queue makes |
| 20:58.19 | supersoaker | from my test :/ |
| 20:58.32 | supersoaker | but i need to way to see if a context was invoked by Queue |
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| 21:18.36 | supersoaker | VultureZ: do you know the difference between MixMonitor and Record? |
| 21:24.56 | [TK]D-Fender | apples & orangutans..... |
| 21:30.39 | WIMPy | If you're talking apples as in Apple Computers, the orangutans might not be far off. |
| 21:31.12 | kaziklu-bey | like cars and beds uh? |
| 21:31.21 | kaziklu-bey | wait there are cars with beds... |
| 21:32.30 | WIMPy | And cars as beds. |
| 21:36.38 | mbowie | Beer & Mayonnaise |
| 21:37.47 | [TK]D-Fender | mbowie: Those are at least standard foodstuffs. |
| 21:38.19 | mbowie | Point well made. |
| 21:40.27 | WIMPy | And if you add some fries they fit quite well. |
| 21:41.12 | mbowie | But sans fries, not so much. |
| 21:41.17 | [TK]D-Fender | SO HAPPY TOGETHER!! |
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| 22:27.56 | overyander | I want to be able to receive sms from any of my DID's and optionally reply. all of the sms gateways i've found online seem to be a web hosted solution and offer/charge for a lot of features that I would code myself in asterisk. Is it typical in the U.S. to go through those types of gateways or is that something that my provider (Intelepeer) can provide? |
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| 22:30.23 | *** join/#asterisk jaredkipe (~jaredkipe@50.248.197.213) |
| 22:30.35 | jaredkipe | hello all |
| 22:30.48 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 22:31.33 | overyander | hello |
| 22:32.41 | jaredkipe | I've got an odd problem, if someone would like to help a little |
| 22:33.03 | Chainsaw | As long as it involves Asterisk. |
| 22:33.08 | jaredkipe | of course |
| 22:33.26 | jaredkipe | so Iâve got a handful of extensions I use to shortcut into queues, as well as sign in and out of them |
| 22:34.07 | jaredkipe | I went and added a new one today, and for some reason, my phones are not waiting long enough for the âslightly longerâ extension to sign into queue, instead just going to the queue short cut |
| 22:34.19 | jaredkipe | e.g. 311 goes to queue, 3110 signs out of queue |
| 22:34.46 | jaredkipe | you go to dial 3110 and immediately upon finishing 311, it launches into that extenâs flow |
| 22:35.01 | Chainsaw | jaredkipe: I imagine [TK]D-Fender, the dialplan whisperer, is going to want to see extensions.conf |
| 22:35.32 | jaredkipe | for reference, 304 goes to a different queue, 3040 signs out |
| 22:35.34 | jaredkipe | and works properly |
| 22:35.38 | overyander | jaredkipe, never messed with queues, sorry |
| 22:35.48 | jaredkipe | I donât think its queue related |
| 22:36.00 | jaredkipe | Iâm primarily a SW developer |
| 22:36.36 | overyander | a phone signing in and out like that is queue based. a regular sip phone just registers when it boots up |
| 22:37.46 | jaredkipe | Iâm pretty sure if I replaced the âqueueâ related logic with just goto statements it would behave the same |
| 22:37.53 | jaredkipe | or dial, whatever |
| 22:38.17 | overyander | when you run "sip show peers" does it show all of the phones? |
| 22:39.20 | jaredkipe | yes, but these are not sip peers |
| 22:39.29 | overyander | what are they? |
| 22:39.29 | jaredkipe | these are extensions |
| 22:39.49 | jaredkipe | part of extensions.conf, and in the context of from sip |
| 22:40.38 | jaredkipe | I donât know asterisk that well, but Iâve been able to hack together some stuff like macros to turn on and off forwarding to cell phones etc. |
| 22:41.04 | overyander | contexts are completely unique on all systems. in your dialplan you point exten 1234 to dial sip/blahblah |
| 22:41.08 | jaredkipe | I feel like asterisk has âcachedâ this, or the phone have, and will not wait for the extra digits |
| 22:41.50 | jaredkipe | for sake of argument, exten 304 dials x 3040 dials y |
| 22:42.23 | jaredkipe | 311 dials x, 3110 SHOULD dial y, but you cannot physically dial it because after you press the second 1 in 311 it goes straight to x |
| 22:42.48 | jaredkipe | 311 and 3110 are unique in the dialplan |
| 22:43.20 | cusco | <PROTECTED> |
| 22:43.52 | overyander | oh, i think i see, it's in the order the patterns are matched. you have 3110 listed lower than 311 don't you? |
| 22:44.06 | jaredkipe | hrm |
| 22:44.35 | jaredkipe | hard to say because of the âinclude => macroâ stuff |
| 22:44.53 | jaredkipe | what I do know is that they are organized exactly like the 304, 3040 stuff |
| 22:45.01 | jaredkipe | namely, they should have the same âorderâ |
| 22:45.05 | overyander | includes are used in the order they are listed prior the extens in the context |
| 22:46.31 | cusco | jaredkipe: in asterisk cli: dialplan show <context> |
| 22:46.35 | cusco | go from there |
| 22:46.38 | cusco | paste |
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| 22:48.53 | jaredkipe | https://gist.github.com/jaredkipe/eeb30761d34d11f7f73e |
| 22:50.13 | jaredkipe | â¦. some of that junk was not me and Iâve simply extended the madness |
| 22:52.46 | jaredkipe | the 301X - 304X stuff works so well, no idea why 311X stuff doesnât |
| 22:53.04 | overyander | move your includes for utilities above the internals includes, then do a "dialplan reload" and I bet that'll solve your problem |
| 22:53.21 | [TK]D-Fender | [18:42]jaredkipe311 and 3110 are unique in the dialplan <- they are in separate contexts |
| 22:53.21 | overyander | it's because of the matching order i mentioned |
| 22:53.34 | [TK]D-Fender | jaredkipe: THAT is what's happening. |
| 22:53.41 | jaredkipe | but SO is 304 and 3040 |
| 22:53.59 | [TK]D-Fender | * will lok in the the contexts in the order listed |
| 22:54.19 | jaredkipe | they are in the same ârelativeâ order |
| 22:54.22 | overyander | yes, it's matching 304 before 3040 because 304 is found first |
| 22:54.32 | overyander | nothing is 'relative' |
| 22:54.34 | [TK]D-Fender | it lawys look in-context first |
| 22:54.34 | jaredkipe | BUT thats not what happens |
| 22:54.40 | [TK]D-Fender | Show us the calls |
| 22:54.42 | jaredkipe | 3040 dials correctly |
| 22:54.44 | overyander | move those and reload and watch what happens |
| 22:54.45 | [TK]D-Fender | looks* |
| 22:55.26 | [TK]D-Fender | your phone's dialplan is another matter. |
| 22:56.08 | [TK]D-Fender | Nowhere in the dilplan you have shown is * waiting for input. |
| 22:56.19 | jaredkipe | https://gist.github.com/jaredkipe/2948f7ac5aa053dea10b |
| 22:56.22 | [TK]D-Fender | It's expecting to receive whole numbers from your endpoitns... |
| 22:56.33 | jaredkipe | phone dial plan? sounds promising |
| 22:56.46 | [TK]D-Fender | <PROTECTED> |
| 22:56.51 | [TK]D-Fender | it got 311 from your phone |
| 22:56.59 | [TK]D-Fender | This has nothing to do with Asterisk at that point |
| 22:57.05 | jaredkipe | thank you |
| 22:57.23 | jaredkipe | so this is new... |
| 22:57.25 | jaredkipe | to me anyway |
| 22:57.34 | [TK]D-Fender | Phones is king. |
| 22:57.53 | [TK]D-Fender | It chooses when to stop you from pushing digits all on its own... |
| 22:58.08 | jaredkipe | I kinda assumed that because of how fast it was |
| 22:58.10 | [TK]D-Fender | NEXT!@!!@@!@ (c) BKW |
| 22:58.14 | [TK]D-Fender | gavels |
| 22:58.19 | jaredkipe | ideas on how to fix? |
| 22:58.20 | [TK]D-Fender | ADJOURNED! |
| 22:58.28 | jaredkipe | lol |
| 22:58.29 | [TK]D-Fender | jaredkipe: Go change the dialplan on your phones |
| 22:58.49 | [TK]D-Fender | jaredkipe: that is entirely dependent on model and means of configuring |
| 23:00.46 | jaredkipe | yeah, soft phone works |
| 23:00.50 | jaredkipe | stupid polycomâ¦. |
| 23:01.01 | *** join/#asterisk petris (~petris@198.251.81.123) |
| 23:03.23 | Chainsaw | jaredkipe: The Polycom uses whatever internal dialplan you give it. |
| 23:03.35 | Chainsaw | jaredkipe: I'm afraid that's still your fault. |
| 23:03.41 | jaredkipe | not my fault |
| 23:03.43 | jaredkipe | inherited! |
| 23:03.53 | jaredkipe | my Cisco ATAâs Iâve deployed do NOT have the problem |
| 23:04.08 | [TK]D-Fender | Polycom > All |
| 23:04.32 | [TK]D-Fender | Polycom's dialplan are highly configurable. It's your job to set it to your needs |
| 23:05.12 | jaredkipe | like I said, I inherited this problem |
| 23:05.21 | [TK]D-Fender | 311 is a North Amrican STANDAND service code. Phones would cone matching this stock.... |
| 23:05.33 | jaredkipe | service code? |
| 23:05.49 | [TK]D-Fender | [34569]11 .... |
| 23:05.53 | [TK]D-Fender | 911 = POLICE |
| 23:06.14 | [TK]D-Fender | 311 = for example |
| 23:06.18 | [TK]D-Fender | 411 = directory assitance... |
| 23:06.35 | jaredkipe | ha, thats why weâre using 311 for internal helpdesk |
| 23:06.45 | [TK]D-Fender | regardless go figure out how the phones were configured, and go in to change the dilplan |
| 23:06.51 | jaredkipe | Iâm looking |
| 23:06.59 | jaredkipe | they receive the configuration from tftp |
| 23:07.12 | [TK]D-Fender | Warning to these guys : don't use national standard for your internal features |
| 23:07.30 | jaredkipe | what does 311 supposedly dial? |
| 23:07.33 | [TK]D-Fender | Or get sued when someone trying to use your phone to call emergency services can't get through because you are hijacking standard #'s |
| 23:07.53 | jaredkipe | bah, I have congestion on remote 911 ;) |
| 23:07.58 | [TK]D-Fender | 311: Non-Emergency Police, Municipal and Other Governmental |
| 23:10.08 | jaredkipe | found it! |
| 23:10.18 | jaredkipe | [2-9]11|⦠|
| 23:10.35 | jaredkipe | huzzah, [4-9]11 |
| 23:10.48 | jaredkipe | how power cycling phone auto configures ;) |
| 23:10.52 | jaredkipe | hope* |
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| 23:24.24 | jaredkipe | yep |
| 23:24.30 | jaredkipe | sweet sweet override blis |
| 23:25.05 | jaredkipe | thanks everyone for your support |
| 23:29.11 | *** part/#asterisk jaredkipe (~jaredkipe@50.248.197.213) |
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| 23:59.54 | supersoaker | Hello #asterisk, I was woundering if Queue starts a new channel when dialing the members and if channel scope variables are defined, otherwise, is the a vairable set that i can see that the context was invocked by the Queue app? |