| 00:01.38 | WIMPy | The RPi can dewfinitely do it. |
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| 00:18.15 | julius | have to check that again tomorrow, my bouncer will watch you ;) |
| 00:18.17 | julius | good night |
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| 00:55.43 | ZX81 | so nickserv must remember usernames and passwords for a while :-) |
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| 02:40.18 | volga629 | Hello Everyone, I need some help with messages |
| 02:40.38 | volga629 | function MessageSend() |
| 02:40.49 | volga629 | here are dialplan so far |
| 02:40.50 | volga629 | http://fpaste.org/331466/14567999/ |
| 02:42.00 | volga629 | So in use devices users model dial string SIP/100/200/3300 that mean one user with multiply devices |
| 02:42.26 | [TK]D-Fender | no |
| 02:42.54 | volga629 | I am trying break the string on devices without & then check peer status and then send message to one which not in use |
| 02:42.56 | [TK]D-Fender | there is no "multiple" |
| 02:43.29 | volga629 | yes, it like one dial string |
| 02:43.48 | [TK]D-Fender | nvm, I'm getting what you're saying... |
| 02:43.58 | [TK]D-Fender | the DB key has multiple valiues to split on |
| 02:44.07 | volga629 | yes correct |
| 02:44.56 | volga629 | in my paste where empty space not sure how to proceed if need one more loop or possible avoid it |
| 02:45.27 | [TK]D-Fender | You should be nooping this so you can see what it evaluates throughout the process. |
| 02:46.35 | volga629 | ok let me add NoOp first to see what is the return |
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| 02:59.25 | volga629 | I added NoOp(Checking Status for peer ${PEERS} and device state ${PEER_STATE}) |
| 03:01.47 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-dlqiazcdwzwtjjge) |
| 03:02.09 | volga629 | here are what I see when invoked on cli |
| 03:02.10 | volga629 | http://fpaste.org/331468/14568012/ |
| 03:02.46 | volga629 | what is reliable way determine if extension online or not |
| 03:04.20 | [TK]D-Fender | "core show function DEVICE_STATE" |
| 03:06.36 | volga629 | it report both extensions for same user INVALID |
| 03:07.26 | [TK]D-Fender | And what do you see on the peer dump? |
| 03:07.55 | [TK]D-Fender | Where do we see what you even queried? |
| 03:08.53 | volga629 | I added |
| 03:09.01 | volga629 | exten => _X.,n,Set(PEER_STATE=${DEVICE_STATE(SIP/$PEERS)}) |
| 03:09.02 | volga629 | exten => _X.,n,NoOp(Peer: ${PEERS} and Peer State: ${PEER_STATE}) |
| 03:10.50 | volga629 | here are updated version when I send message |
| 03:10.50 | volga629 | http://fpaste.org/331469/14568018/ |
| 03:12.03 | volga629 | I want get which extension online then send send message online to available one |
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| 03:19.27 | [TK]D-Fender | no good |
| 03:19.34 | [TK]D-Fender | (SIP/$PEERS)} <- clearly wrong |
| 03:20.07 | [TK]D-Fender | You're supposed to NoOp the what you are PASSING to that function to prove you aren't passing garbage |
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| 03:22.34 | volga629 | peers come from exten => _X.,n,While($["${SET(PEERS=${SHIFT(PEER_GRP,&)})}" != ""]) |
| 03:24.14 | volga629 | or SIP/peer |
| 03:24.24 | volga629 | should be just extension |
| 03:24.38 | volga629 | DEVICE_STATE(102) |
| 03:25.05 | [TK]D-Fender | no good |
| 03:25.10 | [TK]D-Fender | there is no TECH in there |
| 03:26.53 | volga629 | I see |
| 03:27.42 | volga629 | if I do core show hints |
| 03:28.09 | volga629 | 102@ext-local : SIP/102&SIP/10102&Cu State:Idle Presence:available Watchers 0 |
| 03:28.27 | volga629 | do I need query ext-local |
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| 03:31.23 | [TK]D-Fender | taht is a dialplan dcontext |
| 03:31.28 | [TK]D-Fender | and doesn't mean anything |
| 03:31.34 | [TK]D-Fender | DEVICE_STATE(102) <- BAD |
| 03:31.41 | [TK]D-Fender | DEVICE_STATE(SIP/102) <- GOOD |
| 03:31.49 | [TK]D-Fender | 102 is not a device. |
| 03:32.00 | [TK]D-Fender | device needs the tech |
| 03:32.10 | volga629 | ok |
| 03:35.55 | volga629 | let look on asterisk db might be I can find how to determine if extension online or not. DEVICE_STATE return PEER_STATE=INVALID |
| 03:38.49 | [TK]D-Fender | I don't see you calling it properly yet |
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| 03:40.37 | volga629 | that updated version http://fpaste.org/331474/14568036/ |
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| 03:42.50 | vrtigo1 | Hi! I'm trying to implement an HTTP lookup in a dialplan. Specifically, I'm trying to have asterisk fetch the phone number to be dialed from an external website at runtime. Can anyone point me in the right direction toward docs or an example of how I might go about this? |
| 03:45.41 | vrtigo1 | Nevermind, I seem to have found info on using cURL, just had to adjust my search terms. |
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| 04:03.05 | volga629 | what is criteria that device state will return correct state ? |
| 04:03.25 | volga629 | I am not sure what is not right |
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| 04:26.53 | [TK]D-Fender | volga629: You are still not showing what you're aaactually passing. |
| 04:26.59 | [TK]D-Fender | CLI OUTPUT |
| 04:27.17 | [TK]D-Fender | NoOp what you are passing to the function to prove that it is sane |
| 04:28.28 | volga629 | http://fpaste.org/331488/45680649/ |
| 04:30.49 | [TK]D-Fender | <PROTECTED> |
| 04:30.51 | [TK]D-Fender | 102 is NOT a peer |
| 04:30.55 | [TK]D-Fender | SIP/102 is |
| 04:31.05 | [TK]D-Fender | What are you having trouble understanding?\ |
| 04:31.14 | [TK]D-Fender | You don't do Dial(100). |
| 04:31.19 | [TK]D-Fender | S100 is not a device |
| 04:31.22 | [TK]D-Fender | SIP?100 is a device |
| 04:31.26 | [TK]D-Fender | SIP/100 is a device |
| 04:33.42 | volga629 | yes I tried SIP/100 and it always state is INVALID |
| 04:35.02 | [TK]D-Fender | Show that the peer is there as well |
| 04:35.10 | [TK]D-Fender | so far all I see is invlaid things |
| 04:35.14 | [TK]D-Fender | invalid |
| 04:35.20 | [TK]D-Fender | by that I mean WRONG |
| 04:38.37 | volga629 | I set exten => _X.,n,Set(PEER_STATE=${DEVICE_STATE(SIP/$PEERS)}) |
| 04:39.32 | volga629 | this output |
| 04:39.33 | volga629 | http://fpaste.org/331492/14568071/ |
| 04:45.24 | [TK]D-Fender | BAD |
| 04:45.34 | [TK]D-Fender | $PEERS <_ NOT VALID |
| 04:45.51 | [TK]D-Fender | PAY ATTENTION |
| 04:45.56 | [TK]D-Fender | i TOLD YOU TO FIX THIS ONCE ALREADY |
| 04:46.07 | [TK]D-Fender | Caps failure../m |
| 04:48.31 | volga629 | you mean this SET exten => _X.,n,While($["${SET(PEERS=${SHIFT(PEER_GRP,&)})}" != ""]) |
| 04:51.52 | volga629 | or my finally |
| 04:51.53 | volga629 | exten => _X.,n,Set(PEER_STATE=${DEVICE_STATE(SIP/${PEERS})}) |
| 04:51.58 | volga629 | long date |
| 04:54.15 | volga629 | send actual message this is good state NOT_INUSE correct ? |
| 04:54.34 | [TK]D-Fender | Only meant I' |
| 04:54.46 | [TK]D-Fender | I'm still not seeing what it is getting PASSED |
| 04:55.05 | [TK]D-Fender | NOT_INUSE <- tthis is a VALID state at least |
| 04:55.12 | [TK]D-Fender | so taht looks good |
| 04:57.40 | volga629 | here output right now |
| 04:57.41 | volga629 | http://fpaste.org/331494/80824814/ |
| 04:59.26 | volga629 | I just need fix Send statement I want check state send only one which state NOT_INUSE and if all extensions fail then send to fail message |
| 04:59.53 | volga629 | not sure if I can fit in one statement |
| 05:02.05 | [TK]D-Fender | packs up to head home... |
| 05:03.12 | volga629 | [TK]D-Fender Thanks for the help |
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| 06:09.41 | Merlin | I have an asterisk 1.6.1.24 server peered with a asterisk 1.6.0.28, and when the asterisk 1.6.0.28 server restarts, the asterisk 1.6.1.24 server doesn't automatically reconnect-- i have to restart it. what sip setting would I adjust to fix this? |
| 06:15.59 | [TK]D-Fender | None of these are supported |
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| 06:19.33 | Merlin | well i suppose that is one answer |
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| 06:20.42 | volga629 | <PROTECTED> |
| 06:21.50 | [TK]D-Fender | Neither of those ... or the 3 branches that followed. |
| 06:21.54 | [TK]D-Fender | Or the 5th |
| 06:22.12 | Merlin | i'm aware |
| 06:22.46 | Merlin | I have the misfortune of having to work with Fonality products |
| 06:22.53 | volga629 | I will finish tomorrow the dialplan, not sure yet about Send command. How to do it that evaluate first all peers and send with specific state first and if all fail then send fail message |
| 06:30.15 | [TK]D-Fender | Merlin, May the almighty have pity upon thee.... |
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| 06:36.04 | Merlin | haha thank you :) |
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| 06:41.36 | elvisthedj | Hi - I have an agi script that I used back in 2012 (don't remember what * version was then), but I'm wondering if something changed with AGI.. If an AGI script calls a perl module that also uses AGI, does it recieve the variables passed to the original script? |
| 06:42.00 | elvisthedj | the module is able to execute commands via agi, but the following has null variables: |
| 06:42.26 | elvisthedj | my $recording = "/tmp/recording-" . $AGI->{AGI_PARAMS}->{calleridnum} . "-" . $AGI->{AGI_PARAMS}->{uniqueid}; |
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| 10:44.31 | julius | hi |
| 10:44.45 | julius | is it possible to run asterisk on a fritzbox? |
| 10:56.29 | Kunsi | julius: it is. personally i'd recommend usind a second device (even a raspberry pi has enough power) to run asterisk |
| 10:57.26 | Kunsi | quick google-ing returns http://www.asterisk-kompakt.de/asterisk/45-asterisk-auf-fritzbox-phone.html (german) |
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| 10:58.37 | julius | Kunsi, why would you prefer a second device? |
| 10:59.39 | Kunsi | because I only have a rented box (provider stuff), so if something happens to it (had id replaced six times in last two years) I don't have to reinstall/reconfigure asterisk |
| 11:00.58 | julius | ok, makes sense |
| 11:01.35 | julius | Kunsi, can a asterisk on a pi still surpress the phone from ringing? |
| 11:02.04 | julius | i mean, to block anonymous callers i would like them to be redirected to a voice message but i dont want my phone to ring |
| 11:02.36 | Gugge | a pi is just a computer, asterisk on it has all the features. |
| 11:02.55 | Gugge | But dont expect it to have enough power to handle many calls at once |
| 11:03.03 | julius | only got one phone |
| 11:04.02 | WIMPy | Why do you want to try Asterisk at all for only one phone? |
| 11:04.29 | Kunsi | julius: also, recent fritzbox firmware shuld have an option to block anonymous calls at all |
| 11:04.48 | julius | just getting started with the "problem" |
| 11:04.55 | julius | maybe a fritzbox will do the trick |
| 11:05.47 | WIMPy | The phone features of the FB aren't bad. |
| 11:07.28 | julius | ok |
| 11:08.02 | julius | can i ask a isdn specific question here that has nothing todo with asterix or is there a better channel? |
| 11:08.43 | WIMPy | Just ask. |
| 11:10.34 | julius | we got isdn at home, the 2 telephone wires comming from the wall to into some kind of isdn box which allows to attach isdn devices. can i just attach any fritzbox directly to the 2 wires in the wall? |
| 11:10.57 | WIMPy | No. |
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| 11:11.40 | WIMPy | The NT is always required. (Except for some special cases in the US as usual). |
| 11:13.06 | julius | the fritzbox 7272 for example got some kind of isdn connector, do i connect the isdn box there? |
| 11:13.13 | Kunsi | no |
| 11:13.21 | julius | because the connector is labelled S0 with a phone picture |
| 11:13.43 | WIMPy | The same way as any ISDN device: On the S0 bus. |
| 11:14.04 | Kunsi | http://service.avm.de/support/media/filter/l/transfer/img/4c52ce91-41dc-4124-9307-37aeac100096/anschluss_ntba_tae_y_kabel.png - that's how you connect your stuff (left side of picture) |
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| 11:14.33 | WIMPy | The S0 interface is a definite standard, while there are at least 3 common versions of the U interface. |
| 11:14.39 | Kunsi | the box's S0 connector is for connecting phones |
| 11:14.56 | Kunsi | s/phones/ISDN devices/ |
| 11:14.56 | julius | ah ok |
| 11:15.16 | WIMPy | The other S0 connector, if present. |
| 11:15.29 | julius | so the fritzbox will probably come with the right Y-cable to attach it to both devices |
| 11:15.33 | WIMPy | On the current models they are both missing, I think. |
| 11:15.40 | WIMPy | Yes |
| 11:15.58 | julius | so basically any fritzbox will do? |
| 11:15.59 | WIMPy | But make sure you get a model that still supports ISDN. |
| 11:16.06 | julius | ah, there it is |
| 11:16.40 | WIMPy | I think the most recent version with full ISDN support is still the 7490. |
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| 11:17.32 | Kunsi | even 7050 supports ISDN, which is very old. older models simply don't have new Y-cable, istead they have two separate cables to two ports on fritzbox. but everything should be described in box manual (if you don't have one, you can download them at AVM website |
| 11:18.03 | julius | https://en.wikipedia.org/wiki/Fritz!Box |
| 11:18.13 | julius | theres a nice list |
| 11:18.37 | julius | according to that the 7570 also got isdn |
| 11:19.08 | WIMPy | No. They always had the Y-cable. |
| 11:20.09 | Kunsi | WIMPy: I _have_ an 7050, which does NOT have the Y-Cable. |
| 11:21.08 | julius | any idea when they started supporting the redirect anonymous caller functionality? because i already got a wlan router and this would only be for one phone...so i would rather go for cheap |
| 11:21.20 | WIMPy | I do also have an 7050. It has the usual shared port. |
| 11:21.38 | WIMPy | has NFI |
| 11:22.24 | Kunsi | WIMPy: mine has a grey DSL port and a blue labelled "ISDN/analog". maybe provider specific stuff (it's 1&1-branded |
| 11:23.58 | julius | or would it be possible to use a really old fritzbox and have asterix manage the call redirecting? |
| 11:24.14 | julius | i could get a used fritzbox fpr ~15⬠|
| 11:24.28 | WIMPy | Possible. Some of the branded ones were non standard hardware. |
| 11:25.07 | Kunsi | julius: maybe, don't really know which firmware version introduced anonymous blocking |
| 11:25.16 | Kunsi | but what's you current setup? |
| 11:25.16 | WIMPy | You could use both. But one of them would be kind of senseless then. |
| 11:25.58 | julius | sure...but when i get a rpi2 for <50â¬, and a fritzbox for 15⬠im still cheaper than a fritzbox with isdn >100⬠|
| 11:25.59 | WIMPy | What else do you want to do, apart from sending anonymous callers to VM? |
| 11:26.31 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 11:26.36 | julius | im kind of a linux nerd, i like it very much. having access to a call listing for plotting graphs would be nice |
| 11:27.05 | julius | but that would be extra, just getting rid of unwanted calls is all i need now |
| 11:27.19 | WIMPy | You don't have to handle the calls to do that :-) |
| 11:27.39 | julius | for what? |
| 11:28.06 | *** join/#asterisk caller (~caller@85.27.175.143) |
| 11:28.09 | WIMPy | You can just set up voicemail on any device and connect that in parallel. You just can't do internal calls to listen to your messages then. |
| 11:28.14 | WIMPy | To get statistics. |
| 11:28.30 | julius | ah |
| 11:28.57 | julius | how do you get the "voicemail" count from a phone? |
| 11:29.19 | julius | i would rather prefer them in a database reachable via network or a text file |
| 11:29.58 | carrar | in a sip notify message |
| 11:30.00 | carrar | Voice-Message: 2/7 (0/0) |
| 11:30.08 | julius | uh nice |
| 11:30.20 | WIMPy | Great. If you want to listen on the PC. You don't need more than just a VM application. |
| 11:30.24 | julius | still, i think im gonna go the raspberry + old fritzbox way |
| 11:31.35 | WIMPy | You still want a way to connect the RPi to the line, unless you want to ring the phone in parallel. |
| 11:32.07 | julius | yes, how do i do that? |
| 11:32.15 | julius | with a ata? |
| 11:32.19 | WIMPy | Or a more recent FB where you redirect the calls via SIP to the RPi first and then back to the FB and then the phone. |
| 11:32.35 | WIMPy | That would the the USB part again. |
| 11:34.05 | *** join/#asterisk arg (~smuxi@159-253-77-232.static.kc.net.uk) |
| 11:35.15 | caller | How do i set up caller ID on a Digium TE820 card? |
| 11:35.45 | WIMPy | Caller IDs have nothing to do with hardware. |
| 11:35.46 | julius | WIMPy, why usb? will this do? LINKSYS SPA1001 |
| 11:36.31 | WIMPy | julius: And where do you connect that? To your analog adapter? That's going to be pretty crappy. |
| 11:42.51 | julius | ok |
| 11:43.00 | julius | so im missing a hardware part here, usb to ? |
| 11:43.10 | caller | Okay thanks, Then where would i set the caller ID? |
| 11:43.57 | WIMPy | julius: Yes, you really want some ISDN interface. And as te RPi doesn't have PCI, teher's only USB. |
| 11:44.45 | WIMPy | Or maybe you can still find a D-Link HorstBox Professional. They came with Asterisk (1.2), there you got hardware, but you will have to build your own firmware. |
| 11:45.34 | julius | http://www.raspberry-asterisk.org/faq/#isdn says here that a fritzbox can be used to interface a rpi with isdn |
| 11:45.38 | julius | now im confused |
| 11:45.47 | WIMPy | caller: You have to set up the parameters in your chan_dahdi.conf (*dialplan) and then you set it from your dialplan with the CALLERID function. |
| 11:46.43 | *** join/#asterisk arg (~smuxi@159-253-77-232.static.kc.net.uk) |
| 11:46.57 | WIMPy | julius: Yes. There are two ways: 1. Via CAPI. Don't know how CAPI support in Asterisk is these days. or 2. via SIP, which requires a not too old FB. |
| 11:47.43 | julius | sip is probably the better way |
| 11:47.50 | WIMPy | 2 is also what you'd do if installing Asterisk on the FB. |
| 11:48.06 | WIMPy | No, but definitely still supported. |
| 11:48.14 | caller | Great! So I buy a local phone number and set that as a CALLERID and then I can have multiple agents calling simultaneously and not pay buy the minute like a do now with my voip provider? |
| 11:48.25 | julius | WIMPy, oh sip is outdated? so capi is the way to go? |
| 11:48.47 | WIMPy | caller: huh? |
| 11:49.31 | WIMPy | julius: No. The other way round. But CAPI is less of a diversion. Less loss of features. |
| 11:51.07 | julius | ok |
| 11:52.28 | julius | WIMPy, thanks for the great support btw |
| 11:54.40 | caller | I am trying to set up a call center with goautodial/vicidial running asterisk 1.8, right now I am using a sip trunk provider that charges buy the minute and I adds up quickly so I am exploring other possibilities |
| 11:55.31 | *** join/#asterisk arg (~smuxi@159-253-77-232.static.kc.net.uk) |
| 11:55.40 | caller | it* |
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| 12:10.38 | retentiveboy | After looking through the code, I'm pretty sure the new pjsip doesn't support users.conf, right? Is there another scheme for creating endpoints along with phoneprov, voicemail, etc. other than manually? |
| 12:15.36 | file | no |
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| 13:00.17 | saltsa | is it possible to compile asterisk without menuconfig? |
| 13:00.23 | saltsa | for doing the rpm package? |
| 13:07.12 | *** join/#asterisk marceloamorim (~marcelo@189-90-206-161.isimples.com.br) |
| 13:11.40 | retentiveboy | saltsa, https://blog.rhodestech.ca/building-asterisk-13-rpms/ |
| 13:11.43 | *** join/#asterisk keithf (~keithf@ool-2f151dc2.static.optonline.net) |
| 13:12.14 | retentiveboy | and https://wiki.asterisk.org/wiki/display/AST/Using+Menuselect+to+Select+Asterisk+Options |
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| 14:57.24 | *** mode/#asterisk [+o cresl1n] by ChanServ |
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| 15:12.21 | *** join/#asterisk markl (~markl@d54C2C766.access.telenet.be) |
| 15:12.49 | markl | hi all |
| 15:13.00 | markl | we're having a problem with our asterisk server |
| 15:13.21 | markl | occasionally after a reload asterisk seems to freeze |
| 15:13.41 | markl | still can input cli commands but there is no output |
| 15:13.49 | markl | phones etc. don't work |
| 15:13.55 | markl | anyone an idea? |
| 15:13.55 | *** join/#asterisk doome_ (~doome@82.150.48.146) |
| 15:16.09 | acidfoo_ | markl what version |
| 15:16.25 | markl | 11.20.0 |
| 15:16.51 | markl | with freepbx 2.11.0.43 |
| 15:16.59 | voip | how much ram? |
| 15:17.25 | markl | <PROTECTED> |
| 15:17.25 | markl | Mem: 1869 1673 196 0 203 869 |
| 15:17.25 | markl | -/+ buffers/cache: 601 1268 |
| 15:17.25 | markl | Swap: 1697 26 1671 |
| 15:17.30 | voip | ah ha |
| 15:17.32 | voip | mem running out |
| 15:17.34 | voip | try this |
| 15:17.48 | voip | sync; echo 3 > /proc/sys/vm/drop_caches |
| 15:17.58 | voip | then |
| 15:17.59 | voip | free -m |
| 15:18.20 | markl | <PROTECTED> |
| 15:18.21 | markl | Mem: 1869 533 1336 0 2 22 |
| 15:18.21 | markl | -/+ buffers/cache: 508 1361 |
| 15:18.21 | markl | Swap: 1697 26 1671 |
| 15:18.25 | voip | :) |
| 15:18.31 | voip | crontab that puppy. |
| 15:18.37 | markl | is it good to clear cache once a day? |
| 15:18.45 | voip | i clear it every couple minutes |
| 15:18.47 | voip | up to you |
| 15:18.59 | markl | ok, 'll try that |
| 15:19.01 | markl | thanks |
| 15:19.13 | voip | get the most out of these little vps / vm hosting |
| 15:20.09 | acidfoo_ | voip good idea |
| 15:20.25 | markl | should we increase the memory or is clearing the cache ok? |
| 15:23.14 | *** join/#asterisk puzzola (~puzzola@unaffiliated/puzzola) |
| 15:28.51 | voip | i take donations :P |
| 15:29.11 | voip | 10% of money saved from increasing ram to paypal@phonesupport.org <3 haha |
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| 15:34.26 | *** join/#asterisk anonymouz666 (bd193818@gateway/web/freenode/ip.189.25.56.24) |
| 15:34.59 | anonymouz666 | hi |
| 15:35.38 | anonymouz666 | anyone already set up a SRTP exchange between MS LYNC 2013/Skype for Business and Asterisk? |
| 15:36.22 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 15:37.07 | anonymouz666 | the libpri master is in da house |
| 15:38.18 | cresl1n | howdy all! |
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| 15:48.28 | julius | WIMPy, sorry to bother you again, but im still not sure what to buy. going the CAPI route for example: isdn connector for the pc (maybe raspberry) that connects via the isdn bus to the fritzbox? |
| 15:52.01 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-zswscghfotcuxfkc) |
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| 15:56.13 | *** join/#asterisk fbnts (~fbnts@s100.vectes.co.uk) |
| 15:56.38 | WIMPy | julius: That doesn't make sense. You need some hardware to connect to the S0 bus. |
| 15:57.15 | WIMPy | And in case of the RPi the only direct route is USB. |
| 15:57.54 | fbnts | Hi, I'm looking at a better way to handle invalid outbound calls. I usually just answer(), play a recorded message then hangup() however this shows in the CDR disposition as ANSWERED but I would like this to register as the available FAILED - is there a hangup() code that would force this? |
| 15:58.10 | samgoody | Hello. Am completely newbie, so please bear with me: |
| 15:58.19 | WIMPy | Maybe yu should find a board with PCI for the card you've got. |
| 15:58.20 | samgoody | I created a IVR on Plivo. |
| 15:58.25 | samgoody | It worked OK, but I need non-English text, and they don't support the language I need. |
| 15:58.30 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-tqxjlscjqxntpdxi) |
| 15:58.43 | samgoody | So, I figured I can just use my home computer and install â gee I have no idea. |
| 15:58.58 | WIMPy | fbnts: Too late if you answered. Try to playback with ',noanswer'. |
| 15:58.58 | samgoody | Is this something Asterisk does? If so, whats the best tutorial for beginners? |
| 15:59.22 | samgoody | I need it to connect to a dataabase and lookup values and return them to the user |
| 15:59.47 | samgoody | Which in Plivo was done using a simple node app that was queried each call |
| 16:00.08 | julius | WIMPy, you can see that i dont know what im talking about |
| 16:00.20 | julius | WIMPy, do you have a picture maybe how it all could look like connected? |
| 16:00.34 | samgoody | Postgres database. Please forgive me if this is totally the wrong type of tool, but I don't know, and a few hours of googling has left me more confused than I started |
| 16:00.39 | julius | ths s0 bus comes out of the ntba? |
| 16:00.48 | WIMPy | julius: Unfortunately there are many possibilities. |
| 16:00.57 | julius | just one |
| 16:01.04 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 16:01.05 | WIMPy | Yes, The NTBA provides te S0 bus. |
| 16:01.29 | julius | so ntba to my pc, and then the pc somehow to the phone? |
| 16:02.12 | WIMPy | So you need something to connect the computer to the NT. Either a PCI card, a USB adapter or some IP gateway. |
| 16:02.26 | julius | lets keep it simple, i will just use a old pc with the fritzcard |
| 16:02.41 | WIMPy | That's the next thing. *IF* you need to connect your phone to the computer at all. |
| 16:04.00 | WIMPy | You can connect your phone to a second S0 interface of your computer via the analog adapter you're using now or you can use an ATA to connect it via ethernet. |
| 16:04.14 | WIMPy | Does France have an open or a closed numbering plan? |
| 16:04.48 | WIMPy | Do you call abroad? |
| 16:04.51 | julius | no |
| 16:05.04 | julius | well, maybe my parents once in a while |
| 16:05.36 | julius | was that "..france...numbering plan..."? for me? |
| 16:05.38 | WIMPy | Then you might want to only handle incomming calls. |
| 16:05.48 | WIMPy | yes |
| 16:05.56 | fbnts | WIMPy: I was going to change it so it doesn't answer at all. I tried just doing Hangup() but that doesn't put any entry in the CDR at all |
| 16:05.58 | julius | im german...why did you ask about fr? |
| 16:06.31 | WIMPy | Oh, sorry, I thought I remembered somethign about France. That was probably someone else then. |
| 16:06.33 | julius | yes for now incomming will do just fine, i dont want to go into voip for now until it is absolutely needed |
| 16:06.40 | julius | no problem |
| 16:07.07 | julius | so if my phone is connected to the pc via a isdn card - how does it call to the outside? |
| 16:07.33 | WIMPy | Then I'd go for any old Fritzbox and a computer with some sort of ISDN interface. |
| 16:08.10 | julius | asterisk has to handle that |
| 16:08.19 | julius | is that maybe setup by default? |
| 16:08.31 | WIMPy | You can then configure the computer as additional SIP provider to connect both. |
| 16:08.57 | julius | both? |
| 16:08.57 | WIMPy | Outgoing calls with Asterisk are a bit of a pain if the length of the dialled number isn't known in advance. |
| 16:09.17 | WIMPy | The computer and teh FB. |
| 16:09.19 | acidfoo_ | we have an old instance of Asterisk 1.8 that haven't been migrated yet, and sometime the berkeley DB (astdb) gets corrupted and Asterisk crash, for the time being while we work on migrating it I was thinking to simply remove the astdb file everytime we start Asterisk in case it's corrupted |
| 16:09.24 | julius | ah |
| 16:09.33 | acidfoo_ | so I was thinking adding it to /etc/init.d/asterisk startup script |
| 16:09.38 | acidfoo_ | any better Idea ? |
| 16:10.17 | WIMPy | acidfoo_: *If* you don't need any data in there, go for it. |
| 16:10.43 | WIMPy | But remember that you shouldn't use it for permanent storeage later on. |
| 16:10.51 | acidfoo_ | well, the only data inside the DB is the cached data |
| 16:11.00 | acidfoo_ | from sip registering, etc |
| 16:12.05 | julius | would be nice if the fritzbox could send all calls to the pc by default and let the pc decide what todo with them. either throw away or send to the phone |
| 16:13.19 | WIMPy | That is possible, but not if the box is too old. |
| 16:13.33 | julius | yes |
| 16:13.36 | julius | makes sense |
| 16:13.39 | julius | that would be sip? |
| 16:14.17 | WIMPy | You need a version that supports SIP phones. And then you'd make the computer a SIP phone. |
| 16:14.35 | julius | ok, im gonna check the capi way first |
| 16:14.55 | WIMPy | Old versions only support SIP "lines", not SIP phones. |
| 16:15.55 | WIMPy | CAPI works with old models as well. But might also only work with older versions of Asterisk. Not sure about the state. |
| 16:17.26 | julius | ok, gonna check that out |
| 16:19.01 | *** join/#asterisk busymind (~textual@c-24-21-79-235.hsd1.wa.comcast.net) |
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| 16:31.59 | *** join/#asterisk juned (~juned@103.240.209.29) |
| 16:34.29 | juned | I'm using call files to originate the call but the problem is even if I'm rejecting the call asterisk is giving a status as NO ANSWER instead of BUSY |
| 16:35.25 | julius | WIMPy, http://postimg.org/image/pkecsxzkj/ <- does this look right? the "fritzbox" in this case would only do internet access? |
| 16:36.26 | craigify | juned, is the number you are calling via Originate signaling BUSY? |
| 16:36.52 | craigify | or conjestion |
| 16:36.53 | WIMPy | julius: Yes |
| 16:37.36 | julius | so i dont need to buy a fritzbox for this at all |
| 16:38.09 | WIMPy | No, you can do with a 2nd ISDN interface. |
| 16:38.33 | juned | <PROTECTED> |
| 16:38.38 | WIMPy | the 2nd card cannot be a FritzCard, however. You need a cheap one to be able to connect a phone. |
| 16:38.41 | juned | - Auto fallthrough, channel 'Local/s@playback_1-00000008;2' status is 'NOANSWER' |
| 16:38.42 | juned | [Mar 1 17:37:49] NOTICE[21468]: pbx_spool.c:413 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) |
| 16:38.56 | Kunsi | you don't even need two isdn cards, just configure fritzbox to connect to asterisk (as phone line), then connect your phones to fritzbox like you do now |
| 16:39.09 | WIMPy | juned: that looks like a timeout. |
| 16:39.10 | craigify | juned, Orginate usually has a time limit, even by default if you don't specify it |
| 16:39.25 | craigify | one possible scenario is that the time is exceeded before it answers |
| 16:39.36 | craigify | which will result in a disposition value of no answer |
| 16:39.51 | julius | Kunsi, i didnt kill the "router/fritzbox" in the picture. but my router does not know isdn |
| 16:39.58 | craigify | the only way it will get busy is if the channel reports a busy or conjestion state explicitly |
| 16:40.06 | juned | I'm dialing call using gateway |
| 16:40.30 | juned | but as soon as my mobile is ringing I'm simply rejecting the call |
| 16:40.39 | juned | at the moment it rings I just disconnect it |
| 16:40.47 | julius | Kunsi, this feature is supported in older fritzboxes like the 5140? |
| 16:41.05 | craigify | juned, how do you know that your mobile returns a busy signal when you reject a call? |
| 16:41.08 | *** part/#asterisk Merlin (merlin@omni.gcinfotech.com) |
| 16:41.12 | craigify | normally a rejection sends a call to voicemail |
| 16:41.22 | craigify | which will be an answered call to asterisk |
| 16:41.24 | julius | Kunsi, or is that SIP? |
| 16:41.30 | craigify | but that line you posted looks like originate is timing out, like I said earlier |
| 16:41.47 | WIMPy | julius: Don't know that model, but SIP "lines" have been supported for at least 15 years. |
| 16:42.17 | julius | ok |
| 16:42.45 | WIMPy | julius: Maybe you should re-read the whole conversation. Maybe it makes more sense now with more knowledge. |
| 16:42.46 | julius | having a fritzbox inbetween would get rid of the "calling outboung" problem |
| 16:42.56 | Kunsi | julius: the internet⢠says "So wird für die Internettelefonie der weit verbreitete SIP-Standard unterstützt und VoIP-Telefonate lassen sich auch auch die angeschlossenen Festnetztelefone leiten", so yes |
| 16:43.27 | juned | craigify: there is no way it will go to voicemail but as I;m rejecting the call myself I expect status as BUSY |
| 16:43.40 | WIMPy | julius: Yes, but you can fix that in software as well. I handle the ISDN interfaces with Linux-Call-Router, not with Asterisk. |
| 16:43.44 | juned | any configuration or something to achieve this ? |
| 16:44.04 | craigify | well it's not returning busy |
| 16:44.09 | craigify | or else asterisk would report it |
| 16:44.12 | WIMPy | juned: Rejected is not the same as busy. |
| 16:44.19 | julius | WIMPy, well...looking how easy it is to redirect connections/tunnel connections and do traffic shaping on them with linux. im not sure i really want to understand isdn...could cause brain damage at prolonged exposure |
| 16:44.45 | WIMPy | Ah, it's pretty simple, really. |
| 16:45.03 | juned | When I'm checking extension to extension calling and if I'm rejecting the call its giving me BUSY status , the only difference is I;m sending call to my cell using trunk |
| 16:45.46 | julius | im sure after you have dont it all, it will look that way |
| 16:45.54 | craigify | juned, I think Originate is timing out before it gets an answer |
| 16:45.55 | WIMPy | juned: Maybe the provider you're using has issues with the situation. |
| 16:46.16 | julius | any idea how the SIP option in the fritzbox interface might be called? |
| 16:46.18 | juned | Its Voxbeam |
| 16:46.27 | julius | to redirect calls to my pc |
| 16:46.34 | craigify | juned, regardless if whether you press reject, or whatever you press, you need to debug what's going on at the PSTN level and find out what happens with the channel |
| 16:47.12 | WIMPy | julius: You need to create a new phone of type SIP. |
| 16:47.23 | juned | craigify: All right I'm going to enable debugging and check |
| 16:47.31 | julius | ok, im gonna order one and see how it goes |
| 16:47.49 | Kunsi | no, not a hardware phone |
| 16:48.59 | Kunsi | you need to add a new phone of type "Telefon" - "LAN/WLAN" |
| 16:49.00 | craigify | juned, Originate has a 30 second time out value by default. If you call your cell phone via a some technology that gets you to the PSTN, that channel is not reporting BUSY before 30 seconds is up |
| 16:49.13 | julius | Kunsi, for asterisk? |
| 16:49.17 | Kunsi | yes |
| 16:49.18 | craigify | juned, I can see that just by the 1 line you posted here. Now why that is happenneing is unknown |
| 16:49.22 | julius | Kunsi, ah ok |
| 16:49.24 | *** join/#asterisk Garibaldo (~smuxi@191.250.94.63) |
| 16:49.35 | craigify | juned, that is assuming you didn't set a different timeout value for Originate of course |
| 16:49.49 | craigify | juned, call files use the Originate function, by the way.... |
| 16:50.02 | Kunsi | then, have your asterisk register itself to your fritzbox. setup your box to send all calls to this sip phone, then, on asterisk side, you can handle the calls |
| 16:50.33 | juned | craigify: so If i set some timeout in my call file then that will resolve my issue ? |
| 16:50.46 | julius | thank you guys so much for your time, will take some days before the fritzbox gets here. |
| 16:51.10 | julius | Kunsi, sounds logical |
| 16:51.46 | juned | btw provider is sending SIP/2.0 487 Request cancelled when I reject the cal |
| 16:51.46 | julius | you have my permission to use that artistic picture i posted as a reference ;) |
| 16:52.19 | WIMPy | julius: One of the cars is cut off. :-) |
| 16:53.55 | julius | thats art |
| 16:54.01 | julius | ;) |
| 16:54.22 | julius | i thought maybe someone would say: that phone does not support isdn |
| 16:55.21 | WIMPy | Doesn't look like one. But you said you have a TA a/b, so I just project that in there. |
| 16:55.40 | WIMPy | Wel, THAT was you, I hope :-) |
| 16:57.08 | julius | just grabbed the picture online |
| 16:57.25 | julius | the isdn -> analog converter isnt in it, but the rest looks like at home |
| 16:57.58 | julius | what was "TA" again? |
| 16:58.12 | WIMPy | Termonal Adapter |
| 16:59.06 | juned | craigify: how do I set timeout in call file ? |
| 16:59.34 | juned | Unknown keyword 'Timeout ' |
| 17:00.36 | juned | I can do that in dialplan i guess |
| 17:01.43 | WIMPy | WaitTime |
| 17:02.53 | *** join/#asterisk sekil (~sekil@78.24.111.22) |
| 17:02.55 | julius | WIMPy, if this is right: http://www.thenetworkencyclopedia.com/imagens/I16.jpg then the TA in my picture would be the box in the upper left |
| 17:03.33 | WIMPy | Nope. Thats the NT. |
| 17:03.41 | WIMPy | Err |
| 17:03.53 | WIMPy | No. That picture makes no sense from a technical point. |
| 17:04.11 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-97-110.hsd1.or.comcast.net) |
| 17:04.14 | *** join/#asterisk woose (~woose@unaffiliated/woose) |
| 17:04.29 | WIMPy | That box there would be multiple boxes, NT and TAs. |
| 17:05.41 | julius | well...im not at home in the moment, but if memory serves right that setup supports isdn/dsl/telephone |
| 17:06.07 | julius | i just read that my telephone provider will kill all analog connections till 2018...maybe earlier |
| 17:06.15 | julius | then its voip |
| 17:06.17 | WIMPy | That picture also has RS-232 and V.35. |
| 17:06.25 | *** join/#asterisk djiboutiii (~chrisp@c-67-171-9-140.hsd1.wa.comcast.net) |
| 17:06.30 | julius | yeah...hard to find good pictures |
| 17:06.34 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 17:06.38 | WIMPy | Change to another one. |
| 17:07.16 | julius | in germany they call the new line of contracts "magenta" - its a color near red |
| 17:07.20 | julius | thats how creative they are |
| 17:07.24 | julius | or purple |
| 17:07.47 | djiboutiii | Hi guys. Hoping for some help. I have a very basic asterisk server that can receive calls successfully, except for forwarded calls. There is only 1 way audio from asterisk to the forwarded #, the return RTP doesn't come in. I know it's a NAT/routing issue, but does anyone have any tips? This setup worked until a reboot so I think its related to my iptables |
| 17:08.00 | WIMPy | Maybe one day they will find out how to make them work as well... |
| 17:08.57 | julius | its all ethernet by that time |
| 17:09.04 | julius | so it will probably be easy |
| 17:09.19 | WIMPy | No. |
| 17:09.43 | julius | well, it is not now...but in the future |
| 17:09.45 | julius | why not? |
| 17:09.57 | WIMPy | In what future? |
| 17:10.05 | julius | ah, there was the range of ethernet |
| 17:10.15 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 17:11.24 | WIMPy | That stuff has been in use for at least 15 years and is still suffering from all sorts of infants deseases. How long do you take to realze it just won't work? |
| 17:12.23 | *** part/#asterisk samgoody (~Adium@93-172-239-109.bb.netvision.net.il) |
| 17:12.33 | julius | you mean dsl and isdn? |
| 17:12.48 | WIMPy | No, SIP. |
| 17:13.06 | julius | oh i wasnt thinking about sip, more like pure tcp |
| 17:13.12 | julius | some protocol over it |
| 17:13.43 | WIMPy | Doesn't really work for VOIP. |
| 17:14.19 | julius | voip isnt working over tcp? |
| 17:14.24 | julius | oh, yes |
| 17:14.26 | julius | it was udp |
| 17:14.26 | WIMPy | No |
| 17:14.50 | WIMPy | The signalling part can use TCP. But usually that's UDP as well. |
| 17:15.07 | julius | ive been using programs like teamspeak for some time, it works pretty well. sound quality went way up |
| 17:15.42 | julius | i always thought i was talking to robots from another planet while using teamspeak 2.x (the old one)...until they release version 3 |
| 17:21.01 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-97-110.hsd1.or.comcast.net) |
| 17:21.55 | julius | now theres just the question to get a second pc in case the first fries. the asrock c70m1 looks nice, with case ~70⬠|
| 17:22.42 | *** join/#asterisk bkruse (~Adium@64.89.97.100) |
| 17:26.14 | Kunsi | raspi 2 handles 5 concurrent calls here with 20 phones registered |
| 17:28.58 | julius | sure, what did you pay for the usb _> isdn connector? |
| 17:30.19 | WIMPy | Anything from 2 to 8 EUR. |
| 17:30.43 | WIMPy | (inc P&P that is) |
| 17:31.08 | *** join/#asterisk sekil (~sekil@78.24.111.22) |
| 17:31.14 | *** join/#asterisk juned (~juned@212.174.104.44) |
| 17:31.37 | julius | hm, i looked at the amazon listing but the first page was <60⬠|
| 17:31.41 | julius | > i mean |
| 17:31.53 | WIMPy | ebay |
| 17:34.38 | julius | no luck today |
| 17:35.18 | julius | draytek mini for 15 would be available, but i read that those come with different chipsets...and of course only one is supported |
| 17:36.22 | WIMPy | The ones that have a mouse like shape are good. They come with all kinds of logos. |
| 17:37.01 | *** join/#asterisk resist0r (~resist0r@unaffiliated/mrresist0r) |
| 17:39.41 | *** join/#asterisk craigify (~craigify@ip184-186-18-123.no.no.cox.net) |
| 17:40.06 | julius | yes, looks like the chinse copycats found it ;) |
| 17:41.44 | julius | a fritzcard i can get for 4⬠+ postage |
| 17:42.03 | WIMPy | Don't. Get a cheap one. |
| 17:47.16 | julius | fritzcard was the one and only isdn card...until isdn was replaced |
| 17:47.20 | *** join/#asterisk jeev (~j@unaffiliated/jeev) |
| 17:47.26 | julius | of course, its second hand |
| 17:48.00 | WIMPy | No. Unlike the cheap ones it can't do NT mode. |
| 17:48.16 | julius | oh, i misread your comment |
| 17:48.34 | julius | i read "dont get a cheap one" |
| 17:48.43 | julius | do i need NT mode? |
| 17:49.05 | WIMPy | Only if you want to connect terminals (like phones). |
| 17:49.19 | *** join/#asterisk jab416171 (~jab416171@c-76-27-96-12.hsd1.ut.comcast.net) |
| 17:49.39 | WIMPy | But if you think you will go without a line in the future, that would be the only use then. |
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| 17:50.51 | julius | if the fritzbox sends the call to asterisk(isdn card), can asterisk redirect that call back to a analog phone connected to the fritzbox? |
| 17:51.10 | WIMPy | Yes. |
| 17:51.42 | *** join/#asterisk Get_The_Fish (~get_the_f@2601:281:8301:94c2:4191:73c3:6021:298e) |
| 17:51.54 | julius | nice |
| 17:52.12 | julius | so i will start with the fritzcard that i have and get one just in case the first one kills itself |
| 17:52.36 | julius | that thing is probably over 10 years old |
| 17:52.36 | Get_The_Fish | Is there any way to dial while a recording is playing? I've tried using background then dial, however background plays the entire message before executing dial, and then will play call progress to the channel (which I dont want)( |
| 17:52.55 | julius | gonna take a break, cu |
| 17:53.47 | [TK]D-Fender | Get_The_Fish: "core show application dial" <- |
| 17:54.19 | Get_The_Fish | [TK]D-Fender: Uh, yes, are you referering to the hold music option? |
| 17:54.31 | [TK]D-Fender | yes, unless you see something else |
| 17:55.36 | Get_The_Fish | Not exactly what I was looking for |
| 17:56.29 | Get_The_Fish | I would have thought background would have done it. |
| 17:56.52 | *** join/#asterisk sekil (~sekil@78.24.111.22) |
| 17:56.58 | WIMPy | Background works with WaitExten. |
| 17:57.24 | Get_The_Fish | I suppose technically I can achieve the same result with a queue though, as the queue will play a recording while the caller waits? |
| 17:58.18 | Get_The_Fish | WIMPy: the docs seem to suggest that it will return control to the dialplan immediately after execution (while the file is playing), thus allowing additional processing to occur while the file is playing |
| 17:58.23 | Get_The_Fish | that is not the case |
| 17:59.39 | WIMPy | I think WaitExten is the only combination where it works. |
| 17:59.59 | WIMPy | And a queue would do the same, i.e. play MOH. |
| 18:01.11 | resist0r | Get_The_Fish: what is it you want to move on to in the dialplan while the file is playing? |
| 18:02.33 | Get_The_Fish | resist0r: I want to play a greeting message to the caller, however I want to ring a UA while the message is playing. If the UA answers, the caller is bridged to the UA. If not, the dialplan moves on to the directory. |
| 18:03.11 | Get_The_Fish | resist0r: this is the customer's afterhours schema for their office - they want to know if someone is calling and have the option to pick up if they want. |
| 18:04.05 | [TK]D-Fender | us an MoH with a single defined file |
| 18:05.56 | resist0r | yeah I dont see why moh couldnt do it, if background and waitexten dont cut it |
| 18:06.26 | [TK]D-Fender | they can't |
| 18:06.33 | [TK]D-Fender | they do NOT background behind DIAL |
| 18:06.41 | resist0r | I suppose you dont want any caller input/selection you just want it to inform them they have dialed the correct number and to standby |
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| 18:32.55 | Get_The_Fish | [TK]D-Fender: the reason that wont work is that the length of the file and the dial timeout are not the same. |
| 18:33.11 | [TK]D-Fender | change the file |
| 18:33.19 | WIMPy | Make the file longer. |
| 18:33.38 | Boogieman_JS | Hello everyone |
| 18:34.42 | Boogieman_JS | I have a question, I recently installed AsteriskNow and when trying to update via System Admin pro...I get "Update currently in progress, please wait until complete" But it has been that way for days! |
| 18:35.08 | Boogieman_JS | "update already in progress (locked) and there are no logs of locked files, and I have rebooted several times. |
| 18:35.12 | Boogieman_JS | Suggestions? |
| 18:36.17 | *** join/#asterisk arg (~smuxi@159-253-77-232.static.kc.net.uk) |
| 18:39.15 | Boogieman_JS | Anyone....Beuller..... |
| 18:39.38 | WIMPy | Maybe someone in #centos? |
| 18:40.11 | Boogieman_JS | lol...Asterisk website suggested here. I'll look around, thank you. |
| 18:48.57 | *** join/#asterisk infobot (ibot@rikers.org) |
| 18:48.57 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.7.2 (2016/02/05), 11.21.2 (2016/02/11); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 19:56.11 | wonderworld | is SSL2 enabled in asterisk by default? |
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| 20:06.03 | shudon | hi all :) i'm on ubuntu 14.04. what softphones do you recommend? |
| 20:06.22 | shudon | can i use webrtc to talk directly to asterisk? |
| 20:07.52 | [TK]D-Fender | You could |
| 20:07.55 | shudon | i'm trying qutecom (dunno if this is a defunct project or not; seems like it) but it just tries connecting for a couple of minutes and then gives up |
| 20:08.27 | [TK]D-Fender | Then your networking or other config is bad |
| 20:09.22 | *** join/#asterisk crocodilehunter (~Thunderbi@CPE-121-211-223-68.hhui7.cht.bigpond.net.au) |
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| 20:09.32 | shudon | yeah :( |
| 20:10.06 | shudon | if i have another phone already connected using a given extension, will i be able to connect and use it just like normal? i would assume so |
| 20:11.31 | [TK]D-Fender | SIP is SIP |
| 20:11.42 | [TK]D-Fender | if the credentials are right it'll do what it's supposed to |
| 20:12.27 | shudon | tcpdump on the machine running asterisk doesn't see any packets from me except broadcast crap |
| 20:14.12 | shudon | i see no packets leaving my machine bound for it either |
| 20:14.17 | shudon | so i must have this client configured wrong |
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| 20:22.15 | [TK]D-Fender | Entirely possible |
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| 20:27.44 | shudon | hmm... #ubuntu recommended ekiga... i was able to register with the asterisk server but when i place any call it seems to h immediately |
| 20:29.48 | [TK]D-Fender | Which is something you should look at directly |
| 20:30.45 | shudon | oh i see. i was entering the dst addr incorrectly |
| 20:31.06 | shudon | seems that i need to include the "sip:" at the beginning and @sipserver at the end |
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| 20:31.25 | shudon | [TK]D-Fender: yes i watched the asterisk log to see what was happening and noticed the calls wern't being sent to the server |
| 20:31.28 | shudon | thanks :) |
| 20:31.46 | shudon | now how can i listen in on a call? |
| 20:31.51 | shudon | i usually use isymphony to "barge" but |
| 20:31.55 | shudon | it's too cludgy |
| 20:32.11 | gbkersey | I'm having problems forwarding an inbound call (dahdi pri) to an external number (same dahdi pri). Looks like I'm running into a context issue where the incoming call is in the inbound context but to make outbound calls I need to be in the internal context. Anyone? |
| 20:32.16 | shudon | and i've already written a server that can send commands to asterisk |
| 20:32.26 | shudon | using the AMI |
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| 20:34.24 | pjensen00 | bhkersey: You say forwarding, do you mean SIP 302 redirect or simply make an outbound dial for that incoming call |
| 20:34.57 | [TK]D-Fender | shudon: AMI doesn't magically give you audio. Typically you'd call an extension that issues chanspy, etc |
| 20:35.08 | gbkersey | Not SIP.... incoming call is from a PRI via dahdi. outbound call will be through the same PRI via dahdi |
| 20:35.38 | [TK]D-Fender | gbkersey: To send you call out you just Dial(). |
| 20:35.47 | [TK]D-Fender | So whatever you're doing in the dialplan .. just Dial() |
| 20:35.50 | gbkersey | yes... |
| 20:36.49 | gbkersey | something like exten => INCOMINGDID,1,Dial(Dahadi/g1/NUMBERTOFORWARDTO,25) |
| 20:37.00 | [TK]D-Fender | Yes. |
| 20:38.53 | gbkersey | and I get.... |
| 20:39.46 | gbkersey | -- Called g1/NUMBERTOFORWARDTO |
| 20:40.02 | gbkersey | --DAHDI/5-1 is proceeding passing it to DAHDI/4-1 |
| 20:40.14 | gbkersey | -- Channel 0/5, span 1 got hangup request, cause 21 |
| 20:40.36 | [TK]D-Fender | http://www.trinityos.com/ISDN/isdn-causecodes.txt |
| 20:40.50 | gbkersey | yes, I'm getting CONGESTION |
| 20:40.56 | [TK]D-Fender | Verify with PRI debug enabled |
| 20:41.05 | [TK]D-Fender | "congestion" emans nothing really |
| 20:41.08 | [TK]D-Fender | means |
| 20:41.21 | gbkersey | right.... |
| 20:43.01 | [TK]D-Fender | ~pb |
| 20:43.01 | 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. |
| 20:43.03 | [TK]D-Fender | ^^^^ |
| 20:43.09 | [TK]D-Fender | Show us a call with PRI debug enabled |
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| 20:45.47 | gbkersey | I think I see the problem. It may be sending the originating caller's caller ID |
| 20:47.52 | gbkersey | nope... debug on the way |
| 20:49.32 | [TK]D-Fender | It should send their CID out (which they may not like) |
| 20:49.44 | [TK]D-Fender | if you send the call out without changing anything |
| 20:49.52 | jfindley | Exciting... The pilot for a new product I just developed was installed a little while ago and is actually working! |
| 20:49.56 | gbkersey | that's what I want to do |
| 20:51.07 | jfindley | Not surprising that it's working, but always tense the first time I go from the lab to some company several states over where I have no control of anything. |
| 20:52.04 | *** join/#asterisk arg (~smuxi@159-253-77-232.static.kc.net.uk) |
| 20:54.07 | jfindley | Asterisk is truly a revolutionary piece of software. |
| 20:54.39 | gbkersey | http://pastebin.com/8UHgj75M |
| 20:56.51 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 20:57.18 | [TK]D-Fender | Can't judge the # validity, etc there |
| 20:57.24 | [TK]D-Fender | and you're missing basic verbose |
| 20:58.12 | gbkersey | verb is 4 |
| 20:58.13 | *** join/#asterisk klow (~klong@c-73-53-31-109.hsd1.wa.comcast.net) |
| 20:59.49 | gbkersey | I've got another with verb 9 |
| 21:00.18 | [TK]D-Fender | And I don't see the dial in there |
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| 21:01.36 | gbkersey | [TK]D-Fender: I appreciate your help. let me wait until the pbx is a bit less busy and I'll paste another. |
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| 21:10.32 | gbkersey | [TK]D-Fender: So I setup an internal extension that does the same thing exten => INTERNALEXT,1,Dial(Dahadi/g1/NUMBERTOFORWARDTO,25) |
| 21:11.01 | gbkersey | that works... And I'm seeing the presentation of the PBX's main number in the PRI Debug |
| 21:11.35 | gbkersey | However, when I try calling this extension from the outside, exten => INCOMINGDID,1,Dial(Dahadi/g1/NUMBERTOFORWARDTO,25) |
| 21:12.18 | *** join/#asterisk slav3_sergal (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 21:12.26 | gbkersey | I'm seeing that the PBX is not sending the correct caller ID and the provider is rejecting the call. (The outgoing DID is the same is the inbound CALLERID |
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| 21:13.06 | gbkersey | does that make any sense? |
| 21:14.19 | [TK]D-Fender | It should always call out with whatever CID it inherits from the calling channel |
| 21:15.36 | gbkersey | yes, it should, but it does not appear to be doing to. |
| 21:15.43 | gbkersey | s/to/so/ |
| 21:20.41 | gbkersey | [TK]D-Fender: http://pastebin.com/Va5Darex |
| 21:21.41 | gbkersey | The outdial starts @ line 57 |
| 21:22.48 | [TK]D-Fender | Ext: 1 Cause: Call Rejected (21), class = Normal Event (1) ] |
| 21:23.01 | [TK]D-Fender | Yup, everything looks legit. Not too evident here. Call your telco... |
| 21:23.21 | gbkersey | looks like I'm presenting the wrong callerID |
| 21:23.23 | *** join/#asterisk arg (~smuxi@159-253-77-232.static.kc.net.uk) |
| 21:23.44 | WIMPy | Where is that? |
| 21:23.51 | gbkersey | I'm think that line 77 |
| 21:23.53 | [TK]D-Fender | well you're presenting the CALLER's callerid. |
| 21:23.59 | [TK]D-Fender | which is normal behavioir |
| 21:24.08 | [TK]D-Fender | if you want to fix that then set it before you dial |
| 21:24.39 | gbkersey | I don't want to do that because the telcos here PIN the PRIs so that I can only set a number that is assigned to that PRI |
| 21:25.18 | WIMPy | That's the default. |
| 21:26.43 | gbkersey | so can I just set callerid to what I want it to in the the first line of the extension def? |
| 21:26.48 | [TK]D-Fender | yes |
| 21:27.17 | gbkersey | f*ck me |
| 21:28.09 | gbkersey | I had already tried that because that is what I expected.... I guess I forgot to do a dialplan reload after making the change to extensions.conf |
| 21:28.13 | gbkersey | !#head |
| 21:28.18 | WIMPy | Thanks, but thanks no. |
| 21:29.57 | gbkersey | WIMPy: I'm just banging my head against the desk. |
| 21:31.42 | gbkersey | [TK]D-Fender: Thanks so much for your help.... I'm a dumbass. |
| 21:31.57 | [TK]D-Fender | 11 steps tto go! |
| 21:32.08 | gbkersey | LOL |
| 21:36.19 | [TK]D-Fender | heads home for the day |
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| 21:46.25 | *** join/#asterisk asdfasdafas (~user@ip68-3-30-126.ph.ph.cox.net) |
| 21:47.05 | asdfasdafas | is ther a way to tell if a call is using SRTP? I think I enabled all the SRTP stuff correctly, as well as checking that the module is loaded with rasterisk, but how can I see if an in-progress call is encrypted? |
| 21:50.48 | klow | asdfasdafas, I would like to know too, but a few things Ive noted - some soft phones show if encryption is working (I use Bria and it says right on the call) , also many phones you can put SRTP to be required - also, in chan_sip at least, if you have the peer/eextension configured to use SRTP required, the call will fail and the asterisk log will say the call was not using SRTP . |
| 21:51.42 | asdfasdafas | I set the phones (polycom 335) to enable SRTP, offer SRTP, and require SRTP, but I just wasn't sure if it worked or not :/ I can't tell from the phone itself if it's enabled or not |
| 21:52.27 | asdfasdafas | and it's chan_sip, I set each extension in sip.conf to "encryption=yes". Is there anything else I need to do? |
| 21:56.35 | asdfasdafas | I tried looking at the tcpdump too, which just looks like R2D2 vomit. I was hoping it would tell me if it was SRTP or RTP, so I guess the fact it can't parse it is a good thing...? I was using 'tcpdump -nn -v -i ethX' etc |
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| 22:09.13 | *** join/#asterisk qakhan (~qakhan@50-204-254-12-static.hfc.comcastbusiness.net) |
| 22:10.03 | qakhan | is there any variable which give us call holdtime info in queue |
| 22:11.56 | qakhan | if want to check if CALL holdtime is exceed above 5 min then do something with this call or send it to supervisor |
| 22:14.25 | newtonr | qakhan, https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_QUEUE_VARIABLES |
| 22:14.29 | newtonr | queueholdtime |
| 22:14.47 | newtonr | well, that is the avg hold time for the queue apparently |
| 22:15.01 | newtonr | maybe that is what you want |
| 22:15.06 | qakhan | does this variable give realtime info? |
| 22:15.41 | newtonr | I don't know |
| 22:15.57 | newtonr | Give it a test |
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| 22:42.14 | qakhan | queueholdtime gives time in seconds? |
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