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| 05:29.57 | jaxxan | hi |
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| 05:45.07 | *** join/#asterisk cleb (sysop@cleb.org) |
| 05:48.12 | cleb | Hello all, I am using Asterisk with a DCS SIP Trunk, and have 3 DID's. I would like the Call ID on outgoing calls to be set as a certain DID for certain extensions. Some are local extensions on this PBX, and some are coming over an IAX trunk from another Asterisk box. I've done a bit of research and haven't seen a clean and simple way to implement this. |
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| 05:54.58 | [TK]D-Fender | cleb, have the dialplan they use set it. Either hard-coded from their specific dialplan, or based on looking at who the caller is before dialing out |
| 05:55.13 | [TK]D-Fender | Or based on a variable set if the caller is a sip peer |
| 05:56.03 | jaxxan | exten => _9NXXNXXXXXX,1,Set(CALLERID(all)=XXXXXXXXXX) |
| 05:56.30 | cleb | [TK]D-Fender: OK, thats what I'm doing in the outbound context for the first/default DID, how do I scale that to multiple DID's? |
| 05:56.52 | [TK]D-Fender | Up to you |
| 05:57.07 | [TK]D-Fender | Depends how much you need to scale, how you intend to update them,etc |
| 05:57.18 | [TK]D-Fender | YOU have look at the options and judge your own scale |
| 05:58.03 | jaxxan | Span 1: Channel 1/1 got hangup, cause 100 |
| 05:58.11 | jaxxan | dahdi |
| 05:58.21 | jaxxan | i get that for inbound and outbound calls |
| 05:58.43 | jaxxan | i see the calls coming and leaving asterisk over a PRI |
| 05:58.57 | jaxxan | but they just immediately hangup |
| 05:59.50 | cleb | [TK]D-Fender: OK. And for my other IAX trunks with more extensions hanging off of them, for the outbound do I set the call ID on the seconday PBX, or the primary with the DCS trunk |
| 06:00.35 | [TK]D-Fender | you should be setting it on the system actually dialing out |
| 06:03.06 | cleb | OK. |
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| 06:25.55 | [TK]D-Fender | heads to bed |
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| 07:10.46 | [Consultant]C-As | does asterisk 13 still use res_config_mysql.conf for realtime |
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| 09:28.30 | yun1989 | hello all |
| 09:28.40 | yun1989 | good morning :) |
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| 09:29.25 | yun1989 | I receive this packect in asterisk but i don't understand your format |
| 09:29.25 | yun1989 | http://pastebin.com/eHDZTQzg |
| 09:30.34 | yun1989 | because i receive this "anonymous@anonymous.invalid" ? |
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| 10:04.22 | *** join/#asterisk rox (~rox@BSN-142-168-6.static.siol.net) |
| 10:05.38 | rox | I have a set of PRI lines connected to Asterisk through ZAP, and for inbound calls those lines are not sending out the ALERTING signal, so callers don't get to hear the ringing tone |
| 10:05.52 | rox | does anybody have an idea, what configuration I am missing? |
| 10:07.13 | rox | Calls go (in PRI signaling): |
| 10:07.14 | rox | < SETUP |
| 10:07.14 | rox | > CALL PROCEEDING |
| 10:07.14 | rox | > CONNECT |
| 10:07.14 | rox | < CONNECT ACKNOWLEDGE |
| 10:07.14 | rox | ... DISCONNECT |
| 10:07.47 | rox | there should be an ALERTING signal between CALL PROCEEDING and CONNECT |
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| 10:55.32 | WIMPy | rox: Do you really still use zaptel? If so a simple upgrade should help. |
| 11:09.31 | *** join/#asterisk casdude (6def5752@gateway/web/freenode/ip.109.239.87.82) |
| 11:11.28 | casdude | hi all, I have written a macro that is activated when caller hits *4, the macro playsback a sound file. This works well however it only playsback on one side of the call. Is it possible to playback on both sides of the call at the same time? |
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| 11:36.05 | fireglow | Hello. Asterisk 11.16.0. I use DAHDI to receive calls. When the caller hangs up, my phones continue to ring for about 5 seconds afterwards. How can I shorten this time, and improve detection of the hangup event? |
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| 12:13.56 | *** join/#asterisk xpheres (~bodhi64@dslb-088-073-052-099.088.073.pools.vodafone-ip.de) |
| 12:13.59 | xpheres | hi |
| 12:14.23 | xpheres | hi guys |
| 12:14.24 | xpheres | could someone explain me the deny and permit mask? |
| 12:14.24 | xpheres | <PROTECTED> |
| 12:14.24 | xpheres | my extension is in the local address 192.168.178.23 |
| 12:14.24 | xpheres | I wrote permit 192.168.178.0/255.255.255.0 and deny 0.0.0.0/0.0.0.0 |
| 12:14.24 | xpheres | is that correct? |
| 12:16.01 | Demon_VoIP | acl.conf? |
| 12:17.22 | xpheres | what? |
| 12:17.49 | xpheres | I set up everything with freepbx gui |
| 12:18.04 | xpheres | but I can also modify files if needed |
| 12:18.52 | Demon_VoIP | xpheres, your mask is correct. But first will be better "deny", second is "permit" |
| 12:19.22 | xpheres | yes |
| 12:19.24 | xpheres | deny is first |
| 12:19.42 | xpheres | I just reinstalled asterisk, I'm going to test if it works now |
| 12:20.18 | xpheres | but it seems that my local phone tried to connect from outside, the rejected acl is my wan |
| 12:22.12 | Demon_VoIP | xpheres, delete acl, when phone connected type in console "sip show peer ???" and look IP address.. |
| 12:22.26 | xpheres | ok |
| 12:22.58 | xpheres | I forgot the command to connect to asterisk sorry |
| 12:23.07 | xpheres | I dont use much the console |
| 12:24.05 | Demon_VoIP | xpheres, try "asterisk -r" |
| 12:24.45 | xpheres | ok |
| 12:24.56 | xpheres | it keeps trying to register but it always reject it |
| 12:25.37 | Demon_VoIP | acl exists? What the reason of rejection? |
| 12:25.44 | xpheres | how can I stop the show peer ? |
| 12:25.48 | xpheres | I can not copy |
| 12:25.58 | xpheres | because it moves and it does not allow me to copy |
| 12:26.07 | Demon_VoIP | exit :) |
| 12:26.12 | xpheres | [2015-03-11 12:26:05] NOTICE[1425]: acl.c:748 ast_apply_acl: SIP Peer ACL: Rejecting '88.73.52.99' due to a failure to pass ACL '(BASELINE)' |
| 12:26.12 | xpheres | [2015-03-11 12:26:05] NOTICE[1425]: chan_sip.c:28104 handle_request_register: Registration from '<sip:100@xpheresserver.hopto.org>' failed for '88.73.52.99:53008' - Device does not match ACL |
| 12:26.56 | Demon_VoIP | ok. IP Address is: 88.73.52.99. External :) |
| 12:27.13 | yun1989 | hello |
| 12:27.32 | yun1989 | it's possible when one user leave the conference close the conference ? |
| 12:27.36 | yun1989 | in confbridge ? |
| 12:27.38 | xpheres | yes demon |
| 12:27.42 | xpheres | why is this happening? |
| 12:27.56 | xpheres | why a local cisco phone with extension 100 does not connect from inside the network? |
| 12:28.10 | xpheres | it's my phone and is here |
| 12:28.46 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:29.01 | Demon_VoIP | yun1989, you can look toward "marked" |
| 12:30.06 | Demon_VoIP | xpheres, what address of sip server in configuration of phone? Local? WAN? |
| 12:30.47 | xpheres | ah maybe that's the problem |
| 12:30.49 | xpheres | the wan |
| 12:31.32 | xpheres | yes you are right |
| 12:31.34 | xpheres | that's it |
| 12:32.16 | yun1989 | @Demon_VoIP but if you have two users with "marked = yes" |
| 12:32.25 | yun1989 | it not works correctly |
| 12:32.34 | xpheres | I changed this to the local address: <natAddress>192.168.178.25</natAddress> |
| 12:32.39 | xpheres | in my cisco |
| 12:32.48 | xpheres | let's see if it works |
| 12:33.33 | Demon_VoIP | yun1989, yes. I don't know any other way. Except of intercept AMI events of ConfBridge and... |
| 12:34.55 | yun1989 | @Demon_VoIP how intercept AMI events in confbridge ? |
| 12:34.57 | yun1989 | ? |
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| 12:37.08 | *** join/#asterisk noko (pavel@gate6.zhovner.com) |
| 12:38.29 | noko | what the variable have user sip number? I wanna use GotoIF(user from 11X to 12X...) |
| 12:38.57 | noko | to match users from 110, 111, 112.. |
| 12:39.06 | noko | i just need variable name |
| 12:39.12 | [TK]D-Fender | noko: WHAT "SIP NUMBER"? |
| 12:39.51 | noko | [TK]D-Fender, local pbx users numbers that stored in sip.conf |
| 12:40.51 | WIMPy | Users shouldn't be numbers. that's impolite. |
| 12:40.53 | WIMPy | And insecure. |
| 12:43.13 | [TK]D-Fender | noko: "core show application SIPPEER" |
| 12:45.19 | casdude | hey |
| 12:45.38 | casdude | does anyone know if its possible to playback to callee rather than a caller? |
| 12:46.19 | WIMPy | When? |
| 12:46.56 | [TK]D-Fender | Several ways. |
| 12:47.05 | wjeanneau_ | Hi, anyone has experience with ARI ? I mean I use ARI in order to make an outgoing call and start a specific AGI (like call file). I'm looking a way to pass external parameter into my AGI but I didn't find any option in the ARI doc except the option related to spasis app but I don't use it. I use asterisk 13. |
| 12:48.46 | *** join/#asterisk LooserOuting (~LooserOut@dsdf-5d82f684.pool.mediaWays.net) |
| 12:49.06 | casdude | I am running a macro to playback a sound based from received DTMF |
| 12:49.37 | [TK]D-Fender | casdude: You're now bringing the term "macro" into this which also doesn't answer WIMPy 's question. WHEN? |
| 12:49.51 | [TK]D-Fender | Ah, DTMF |
| 12:49.55 | [TK]D-Fender | . features.conf <- |
| 12:50.07 | casdude | however the playback only plays to the caller rather while on inbound calls i would like it to play to the callee |
| 12:51.04 | casdude | http://pastebin.com/1i79Ba9v |
| 12:51.12 | casdude | her are my settings |
| 12:53.33 | casdude | *here |
| 12:54.22 | casdude | does that help at all |
| 12:54.29 | casdude | let me know if you need more info |
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| 12:56.16 | [TK]D-Fender | casdude: Whole features.conf.... |
| 12:56.42 | [TK]D-Fender | casdude: And a failed call attempt |
| 12:56.51 | casdude | the call does not fail |
| 12:57.03 | casdude | it is the playback |
| 12:57.11 | casdude | of the de-activated |
| 12:57.45 | casdude | is it possible to playback to the callee rather than the caller |
| 12:58.14 | [TK]D-Fender | Show us the call |
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| 13:00.49 | casdude | http://pastebin.com/yhJTFJ3P |
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| 13:02.14 | [TK]D-Fender | I don't seee it executing at all |
| 13:02.18 | [TK]D-Fender | For either channel |
| 13:02.35 | [TK]D-Fender | 68-69. Answer then hangup |
| 13:03.11 | casdude | http://pastebin.com/Edf1Fd1t |
| 13:03.22 | casdude | ok |
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| 13:04.44 | casdude | i'm sorry i must have gotten the wrong call |
| 13:05.04 | casdude | the site is very busy so is hard for me to grab the entire call |
| 13:06.20 | [TK]D-Fender | As long as we get something good to follow up on. |
| 13:07.50 | casdude | http://pastebin.com/Xg5V862g at line 84 the call is paused |
| 13:12.35 | [TK]D-Fender | casdude: Yup, that looks like it's executing against the wrong channel... |
| 13:13.04 | [TK]D-Fender | casdude: If you've done mods to the filee I don't thinkg a simple reload would apply them. IIRC you have to reload the module entirely. |
| 13:13.14 | [TK]D-Fender | which I'm not certain if it would disrupt calls |
| 13:19.06 | [TK]D-Fender | casdude: Verify "features show" |
| 13:20.01 | [TK]D-Fender | in case you flipped these around during testing |
| 13:25.46 | casdude | kk |
| 13:26.31 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 13:26.31 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:28.13 | casdude | http://pastebin.com/2HAe0qbX |
| 13:28.53 | casdude | ummm |
| 13:28.55 | casdude | ok |
| 13:29.23 | casdude | i think the features look the same |
| 13:32.21 | [TK]D-Fender | Yeah, it doesn't list the activateon/activateby though... |
| 13:32.32 | [TK]D-Fender | Think you might have done changes sice last restart? |
| 13:32.50 | [TK]D-Fender | Because based on the name se see executed it should be running on the othere channel... |
| 13:34.32 | casdude | http://pastebin.com/1i79Ba9v |
| 13:34.48 | casdude | this is the original settings posted, which have not been changed |
| 13:35.11 | [TK]D-Fender | This LOOKS like a bug based on that.... |
| 13:35.30 | casdude | yeh |
| 13:35.31 | [TK]D-Fender | What versi0on are you running? |
| 13:35.52 | casdude | 1.8.20.0 |
| 13:36.27 | casdude | it works its just the playback does not go to the callee |
| 13:36.35 | casdude | it only plays to the caller |
| 13:37.07 | [TK]D-Fender | Look at the channel it's executing on directly though.. that's what seems to be the issue |
| 13:37.20 | [TK]D-Fender | <PROTECTED> |
| 13:37.36 | [TK]D-Fender | -- Feature Found: in-pauseMonitor exten: in-pauseMonitor |
| 13:37.37 | casdude | umm |
| 13:37.38 | [TK]D-Fender | -- Executing [s@macro-pause-record:1] Playback("DAHDI/i1/01633508781-9441", "de-activated") in new stack |
| 13:37.52 | casdude | yeh |
| 13:38.10 | [TK]D-Fender | in-pauseMonitor is supposed to be ActivateOn = peer. |
| 13:38.20 | [TK]D-Fender | but it's runnign against the caller channel.... |
| 13:38.34 | casdude | yeh |
| 13:38.41 | [TK]D-Fender | upgrade to the latest in your branch and retest. If it continues, post it on the tracker |
| 13:38.49 | casdude | damn it |
| 13:38.51 | casdude | ok |
| 13:38.52 | file | 1.8 is no longer supported |
| 13:38.59 | file | it's security fixes only |
| 13:39.03 | casdude | sure |
| 13:39.06 | [TK]D-Fender | file: When did that happen? |
| 13:39.13 | file | [TK]D-Fender, end of last year? |
| 13:39.16 | [TK]D-Fender | file: isn't that early for "LTS"? |
| 13:39.28 | file | nope |
| 13:39.51 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 13:40.09 | casdude | thanks for your help |
| 13:40.23 | casdude | TK |
| 13:42.18 | [TK]D-Fender | file: ah, 4 years and just crossed it |
| 13:42.24 | file | yup |
| 13:43.02 | [TK]D-Fender | file: I thought 12 was EOL'd completely already... |
| 13:43.37 | [TK]D-Fender | I remember 10 got shit-canned faster than usual |
| 13:45.57 | malcolmd | 10 and 12 had the same schedule. 10 went away a little faster than 1.6.2 did, but it was the same as 1.6.1 and 1.6.0. 1.6.2 drug a little bit as we transitioned back away form the 1.6.x experiment. |
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| 14:07.14 | pEYEd | what is a decent gui to use with raw asterisk? nothing modular freepbx |
| 14:07.56 | pEYEd | like freepbx o.0 |
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| 14:16.15 | [TK]D-Fender | no |
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| 15:01.37 | ThomasKeller | can somebody please advise how to get rid of the following error? |
| 15:01.38 | ThomasKeller | ERROR: config_options.c: Unable to load config file 'udptl.conf' |
| 15:01.55 | ThomasKeller | I am not using fax |
| 15:02.09 | ThomasKeller | and i am not loading any of the fax-related modules |
| 15:02.43 | ThomasKeller | why does asterisk still keep looking for the file ? |
| 15:03.10 | file | because UDPTL support is core |
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| 15:04.11 | ThomasKeller | is UDPTL used for anything other than fax ? |
| 15:06.23 | coppice | thankfully, no |
| 15:08.03 | ThomasKeller | can I disable UDPTL ? |
| 15:10.51 | file | lack of that file is not fatal, it's safe to ignore it if you aren't using UDPTL |
| 15:13.36 | *** join/#asterisk superscrat (asanders@nat/digium/x-uqsiesautueersez) |
| 15:14.04 | ThomasKeller | but how can I get rid of the errors in my log ? |
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| 15:43.34 | mjordan | provide a udptl.conf |
| 15:47.07 | ThomasKeller | I have created an empty file udptl.conf and the errors went away |
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| 16:24.45 | roler | I am running Elastix (asterisk) on an NLX4000. We have been using this phone system for a few weeks now and everything is great, however we tried to dial out to a 3rd party conference line (hosted elsewhere) and when it says to enter the conference code, it is not registeering our key presses. Is there a setting on our end to make it send DTMF (or whatever it is) a different way? |
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| 17:27.42 | jaxxan | hello |
| 17:27.47 | jaxxan | anyone awake ? |
| 17:28.10 | linocisco | jaxxan, yes. We are GMT+ |
| 17:28.23 | linocisco | jaxxan, but still awake |
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| 17:29.16 | jaxxan | do you have experience passing calls via PRI through a wcte43x+ Wildcard TE435/235 to an adtran 908? |
| 17:29.32 | jaxxan | (= |
| 17:29.45 | jaxxan | i'm getting error 100 on the PRI side |
| 17:29.56 | jaxxan | whenever i pass calls in either direction |
| 17:30.04 | jaxxan | it's a pretty vague telco error code |
| 17:30.32 | linocisco | jaxxan, I am sorry . it is beyond my experience |
| 17:30.56 | jaxxan | no worries |
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| 17:31.21 | jaxxan | i havent' really used PRI's in asterisk since it used to be zaptel |
| 17:31.40 | jaxxan | i think i got the dahdi thing figured out though |
| 17:32.05 | jaxxan | i'm just missing some pertinent configuration option somewhere |
| 17:33.19 | jaxxan | you guys still use pastebin a lot in here ? |
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| 17:35.05 | linocisco | jaxxan, for long paste, IRC users love to see pastebin or something like that |
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| 17:49.00 | [Consultant]C-As | hi, incomming calls are comming with a plus.. is there a way freepbx can change that in trunk or routes before sending to a destination app? |
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| 18:22.14 | WIMPy | jaxxan: What is vague about that error? |
| 18:23.02 | WIMPy | [Consultant]C-As: You need to ask in #freepbx |
| 18:23.29 | jaxxan | the fact that i have no clue as to how to fix it. so i'm reinstalling asterisk and dahdi from scratch |
| 18:23.29 | [Consultant]C-As | opps , thought i was in there lol |
| 18:23.46 | jaxxan | just trying to ensure i didn't miss something |
| 18:24.01 | jaxxan | before i call someone for help (= |
| 18:24.38 | WIMPy | Reinstalling is not going to help. Reconfiguring might. |
| 18:25.21 | WIMPy | But sounds interesting that you get the same in both directions. |
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| 19:01.23 | jaxxan | in chan_dahdi.conf i dont need to worry about trunk groups or spanmaps if i'm only connecting a single PRI right? i can just leave that stuff commented out? |
| 19:01.59 | WIMPy | You will need one group. |
| 19:02.32 | jaxxan | so just uncomment ;spanmap => 1,1,1 ? |
| 19:02.46 | jaxxan | and trunkgroup => 1 |
| 19:03.59 | WIMPy | can't remember all those parameters... |
| 19:04.49 | WIMPy | They can all be left commented out. |
| 19:06.37 | jaxxan | k |
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| 19:24.05 | jaxxan | ok |
| 19:24.27 | jaxxan | outbound calls are working |
| 19:24.32 | jaxxan | inbound, i have no audio |
| 19:24.38 | jaxxan | yay. progress (= |
| 19:29.49 | jaxxan | hrm |
| 19:46.08 | jaxxan | hrm |
| 19:46.34 | jaxxan | i'm not seeing any errors for why i'm not hearing audio from my cell phone calling into the PRI |
| 19:47.25 | WIMPy | Well, you can't, that kind of error doesn't exist. |
| 19:47.38 | jaxxan | i dial in, i see the call hit asterisk over the dahdi/i1 and transfer to my x-lite extension. my x-lite rings and I can answer the call, but no audio |
| 19:47.45 | WIMPy | Unless the call was never answered. |
| 19:47.50 | jaxxan | oh |
| 19:48.17 | jaxxan | i had it going to: |
| 19:48.28 | WIMPy | Missing audio is a SIP thing. |
| 19:48.28 | jaxxan | exten => 4848,1,Answer() |
| 19:48.28 | jaxxan | exten => 4848,n,Wait(1) |
| 19:48.28 | jaxxan | exten => 4848,n,Playback(tt-monkeys) |
| 19:48.30 | jaxxan | exten => 4848,n,Hangup |
| 19:50.03 | jaxxan | it shouldn't be a nat issue since it's coming through the PRI |
| 19:50.21 | Synthase_ | You can use dahdi_monitor to visually confirm you have audio at the PRI leg. But yeah, missing audio is usually a RTP issue. |
| 19:50.28 | WIMPy | No IP, no NAT. |
| 19:50.50 | jaxxan | [Mar 11 15:47:21] NOTICE[18648][C-00000006]: res_rtp_asterisk.c:4367 ast_rtp_read: Unknown RTP codec 126 received from '172.22.10.122:53566' |
| 19:51.03 | jaxxan | that's probably it there then |
| 19:51.33 | file | that's a keep alive from X-Lite |
| 19:52.25 | WIMPy | So theobvious question is: Did you enable your microphone? |
| 19:55.35 | Synthase_ | You may need Progress() before Answer(), if I am remembering correctly, when using Playback(). Someone correct me if wrong. |
| 19:56.06 | WIMPy | No need. |
| 19:56.23 | WIMPy | Unless you want to use Playback with the ,noanswer option. |
| 19:56.47 | Synthase_ | That's it, dealing with early media on PRI. |
| 20:00.57 | jaxxan | my x-lite client can call via sip to asterisk and out the PRI to my cell phone and I get audio in both directions. the call is clean. |
| 20:01.21 | jaxxan | the other directions, from my cell phone down the PRI to asterisk to the SIP client results in no audio in either direction |
| 20:09.54 | *** join/#asterisk bkruse (~Adium@64.89.97.113) |
| 20:10.31 | bkruse | Hey guys - is there a way to check if a sound file exists, through Asterisk? Was thinking of adding this functionality in ARI, as it's important for me to know that information before I originate a call to playback a file. I could be overthinking, but I think it'd be neat |
| 20:12.26 | jaxxan | got it |
| 20:12.54 | jaxxan | it helps when you dont fat finger: siganling=pri_cpe |
| 20:14.34 | mjordan | bkruse: there is a sounds resource |
| 20:14.59 | SamDaMan | jaxxan, I know how you feel I make stupid typos that cause me headaches all the time |
| 20:15.24 | jaxxan | if you tried to play it with dialplan and it resulted in a non-zero, then you could go to different dialplan? |
| 20:19.12 | jaxxan | well i'm all excited now |
| 20:19.33 | jaxxan | time to prank call people @ work |
| 20:23.05 | SamDaMan | use pitch_shift and disguise your voice |
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| 20:53.40 | [Consultant]C-As | is this the correct way AGI(a2billing.php,1,callback) then this AGI(a2billing.php|1|callback) ? |
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| 21:29.36 | pjensen00 | Is there a transcript or something for all the core asterisk sounds? |
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| 21:48.01 | malcolmd | there's a .txt file in the package that has a transcript |
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| 22:04.04 | pjensen00 | ok, I'll look for that |
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| 22:04.55 | pjensen00 | malcolmd: found it thanks |
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| 23:28.47 | oli-work | hey guys/gals |
| 23:28.58 | oli-work | i've got a pbxiaf install running in deviceanduser mode |
| 23:29.10 | oli-work | trying to work out if there's a way to set the device displayname |
| 23:29.15 | oli-work | to be the currently logged in extension |
| 23:29.27 | oli-work | rather than the static device name |
| 23:29.34 | oli-work | (polycom 330ip handsets) |
| 23:29.37 | oli-work | any ideas? |
| 23:35.20 | jaxxan | no. i just gave them physical extensions and let users log in and out. each user had their own extension that followed them to whatever phone they happened to be logged into. |
| 23:35.59 | jaxxan | if it was permanent station then i put it in the phone tftp config |
| 23:36.25 | jaxxan | i haven't used pbxiaf |
| 23:37.00 | [TK]D-Fender | <oli-work> rather than the static device name <- no, but you could work out a MicroBrowser script to show it on the Idle page instead. You'd then leave the line-keys ambiguous as to "ID" |
| 23:37.54 | [TK]D-Fender | oli-work, This is all on you to create of course |
| 23:38.12 | jaxxan | yeah, he's right. it's doable, it just depends on how much time you have to develop it yourself |
| 23:38.27 | jaxxan | there is nothing in the world like custom tailored bro |
| 23:38.52 | oli-work | :) |
| 23:38.53 | oli-work | thanks |
| 23:40.28 | [TK]D-Fender | not the easiest, but far from challenging |
| 23:40.58 | jaxxan | i developed an asterisk based pbx for the business office of a telecom quite a few years ago. it was fantastic. |
| 23:42.10 | jaxxan | when i left though. no one else currently employed by them knew anything about asterisk and they couldn't maintain it. and then voip hackers became predominant and they apparently didn't know how to firewall either. |
| 23:43.37 | jaxxan | i see voip hackers trying to pass international calls day in and day out. all dey evry dey |
| 23:45.35 | jaxxan | today i find myself delivering PRI's to businesses with traditional PBX's using Adtrans. i love that i can use asterisk to test all of these service delivery technologies |
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| 23:50.30 | jaxxan | you know what i dont like? dealing with those nortel admins that sit there doing nothing except ticking the clock, getting paid by the hour. |
| 23:51.25 | jaxxan | the most unhelpful bunch of <explicative> |
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