| 00:00.20 | danfromuk | Extensions.conf is here http://pastebin.com/P7AiwWk7 |
| 00:02.56 | danfromuk | Sip.conf http://pastebin.com/WLA47pzu |
| 00:04.56 | *** join/#asterisk FuriousGeorge (45745b30@gateway/web/freenode/ip.69.116.91.48) |
| 00:05.19 | FuriousGeorge | hey all. I recently upgraded from 1.4 to 1.8 and now I cant take incoming calls: http://pastebin.ca/2198870 |
| 00:05.30 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 00:05.58 | FuriousGeorge | the dialplan is dead simple, and I see no problem with syntax of dial command, yet for some reason it jumps over to the s extension and claims carmen answered when her phone is not even plugged in |
| 00:06.10 | FuriousGeorge | im starting to think the problem might be with * and not my dialplan |
| 00:06.56 | WIMPy | The voicemail syntax isn't valid any more. |
| 00:07.21 | FuriousGeorge | WIMPy: i can fix that, but we aren't even getting there |
| 00:07.35 | WIMPy | 2. Why are you answering the call? |
| 00:07.57 | *** join/#asterisk mjordan (~mjordan@adsl-98-81-86-236.hsv.bellsouth.net) |
| 00:07.58 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:08.32 | FuriousGeorge | WIMPy: I'm not answering it at all. that users phone is not registered at all as a matter of fact |
| 00:08.53 | FuriousGeorge | WIMPy: so it is a total mystery to me how in the world sip/carmen is anwering |
| 00:08.54 | WIMPy | What are you receiving the call from? |
| 00:09.04 | FuriousGeorge | voicepulse sip service |
| 00:09.22 | WIMPy | The first step in your Dialplan is an Answer. |
| 00:09.46 | WIMPy | Do you have insecure=very for them? |
| 00:09.55 | FuriousGeorge | WIMPy: oh i see what you mean |
| 00:09.58 | WIMPy | That needs to be changed to insecure=port,invite. |
| 00:10.04 | FuriousGeorge | WIMPy: ill check |
| 00:10.04 | danfromuk | [TK]D-Fender: it seems that what i'm trying to do isnt possible. When the phone thinks that line is busy, pressing the button has no effect. |
| 00:11.01 | WIMPy | danfromuk: Maybe it has some special syntax to dial different things based on the status? |
| 00:12.16 | danfromuk | WIMPy: i'll have to see if I can get some support from Panasonic. SIP Debug shows that nothing is sent when the key is pressed in a ringing state. |
| 00:12.28 | FuriousGeorge | WIMPy: i don't have insecure set at all for any of the phones |
| 00:13.39 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 00:14.55 | FuriousGeorge | WIMPy: let me update the PB. This thing is going nuts! I just took out the answer and wait. Now the call is DC after a few seconds |
| 00:15.30 | WIMPy | Is any of the phones reachable? |
| 00:15.59 | [TK]D-Fender | danfromuk: So when it isn't "busy" it can dial? |
| 00:17.50 | *** join/#asterisk coppice (~chatzilla@m121-202-56-67.smartone.com) |
| 00:18.52 | FuriousGeorge | WIMPy: actually now none are so I'm waiting for them to register before trying again (i restarted server) |
| 00:19.13 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:19.13 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 00:20.57 | WIMPy | FuriousGeorge: Tehn fix the voicemail to use ,u in the meantime. And make sure you've got an Answer before VoiceMail. |
| 00:21.27 | danfromuk | [TK]D-Fender & WIMPy: thanks for all your help. I managed to find a settings called Directed Call Pickup that allows BLF keys to send a call pickup request. Its working perfectly now. |
| 00:43.15 | FuriousGeorge | WIMPy: thanks for all the help thus far. My inbound extension now contains nothing else. The phones don't ring but I successfully make it to voicemail after timeout, so I guess that's progress |
| 00:43.19 | FuriousGeorge | http://pastebin.ca/2198874 |
| 00:44.29 | FuriousGeorge | i see now why it was going to exension s before. for some reason it is trying to call extension 'carmen' as well as 'sip/carmen' |
| 00:46.11 | FuriousGeorge | and if anyone else can explain why this simple seeming bit of dialplan fails to ring phones I would very much appreciate it |
| 00:46.11 | WIMPy | No, it's trying to call sip.carmen, but it looks like it thinks carmen is itself. |
| 00:46.13 | FuriousGeorge | http://pastebin.ca/2198874 |
| 00:46.41 | FuriousGeorge | WIMPy: i know why, carmen is my hostname, i can rename the phone |
| 00:47.09 | WIMPy | Looks like it doesn't find a peer named carmen then. |
| 00:47.16 | WIMPy | For whatever reason. |
| 00:47.40 | WIMPy | And yes, it would tahe it as hostname then. |
| 00:47.57 | WIMPy | take |
| 00:48.26 | WIMPy | Or seemingly as hostname AND username. |
| 00:48.41 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 00:50.51 | *** join/#asterisk Nemus (~Nemus@c-76-23-30-50.hsd1.ut.comcast.net) |
| 00:52.50 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 00:52.54 | FuriousGeorge | WIMPy: we are making great strides. sip/carmen is now sip/manny. the only remaining problem is that once again, inexplicably to me sip/manny is the only one that rings |
| 00:53.04 | FuriousGeorge | WIMPy: ill make a pastebin |
| 00:53.37 | WIMPy | sip show peers |
| 00:56.28 | *** join/#asterisk coppice (~chatzilla@m121-202-56-67.smartone.com) |
| 00:56.52 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 00:58.54 | FuriousGeorge | WIMPy: http://pastebin.ca/2198880 |
| 00:59.08 | FuriousGeorge | WIMPy: once again I have no freaking clue what the problem is |
| 00:59.46 | FuriousGeorge | it appears to only dial the first tech/resource and ignore the ones after it, and before the timeout option (it gets the timeout right) |
| 01:04.01 | WIMPy | See line 38. You have a path or permission issue. |
| 01:04.25 | WIMPy | Without astdb it won't keep registrations. |
| 01:04.37 | WIMPy | Same with your CDR. |
| 01:04.47 | FuriousGeorge | WIMPy: didnt think that was related, but i'll fix |
| 01:05.00 | FuriousGeorge | this is a new install |
| 01:05.39 | WIMPy | Without AstDB all registrations are gone when you restart. |
| 01:06.48 | WIMPy | And I see a lot of called <user>, but only Manny is ringing. |
| 01:10.29 | FuriousGeorge | WIMPy: no change now that astdb is writeable |
| 01:10.35 | FuriousGeorge | as well as the cdr |
| 01:11.25 | FuriousGeorge | WIMPy: i had asterisk 1.6 installed briefly then i removed it using yum erase... perhaps i needed to do more to get rid of it? |
| 01:11.42 | WIMPy | To me it looks like the phones are called, but don;t answer. |
| 01:11.48 | WIMPy | NFI |
| 01:15.35 | FuriousGeorge | upgrading asterisk, i followed outdated docs (from asterisk.org no less) to install asterisk16 and then i just changed the 16 to 18. probably should not have done it that way |
| 01:16.50 | WIMPy | No idea abot yum or packages. |
| 01:23.00 | FuriousGeorge | WIMPy: thanks anyway for all your help |
| 01:23.20 | FuriousGeorge | uninstall reinstall not fixing either, i could either downgrade to 1.4 or go with something else |
| 01:23.24 | FuriousGeorge | or install from source |
| 01:24.22 | FuriousGeorge | before I try any of that, does anyone else have any clue about this: http://pastebin.ca/2198880 |
| 01:24.27 | FuriousGeorge | only sip/manny is dialed |
| 01:24.33 | WIMPy | You can enable sip debug or use wireshark or something to find out wheat's going on between Asterisk and the phones. |
| 01:25.09 | WIMPy | No, look at the -- Called lines. |
| 01:25.34 | WIMPy | But only manny seems to respond. |
| 01:30.56 | FuriousGeorge | http://pastebin.ca/2198882 |
| 01:31.15 | FuriousGeorge | WIMPy: i never learned to read sip unfortunately, so this does very little for me |
| 01:40.24 | *** join/#asterisk GameGamer43 (users.5533@id-5533.hampstead.irccloud.com) |
| 01:42.29 | WIMPy | See all the retransmits. It can't communicate with those peers. |
| 01:42.46 | tzanger | what? I can't hear you |
| 01:42.57 | FuriousGeorge | WIMPy: not sure what you're referring to, but peers can call eachother |
| 01:45.50 | WIMPy | I don't see any (successfull) communication with the other phones. |
| 01:45.53 | FuriousGeorge | WIMPy: i appreciate your help, but i gotta give up for today |
| 01:46.07 | FuriousGeorge | WIMPy: or at ;least for an hour or so |
| 01:47.14 | FuriousGeorge | g2g thanks again |
| 01:56.44 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:56.45 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 02:14.26 | *** join/#asterisk coppice (~chatzilla@m121-202-56-67.smartone.com) |
| 02:18.52 | SeRi | yay! I just got my savi in.... |
| 02:36.27 | raden | is it normal on a daily basus for someone to try to hack asterisk |
| 02:39.02 | jpsharp | yEP. |
| 02:39.08 | SeRi | raden: yes |
| 02:39.16 | jpsharp | My asterisk box gets dozens upon dozens attacks a day. |
| 02:39.23 | jpsharp | Or more. |
| 02:39.28 | SeRi | raden: mostly bot's... |
| 02:39.48 | SeRi | scanning subnets, etc... |
| 02:40.13 | *** join/#asterisk Nemus (~Nemus@c-76-23-30-50.hsd1.ut.comcast.net) |
| 02:40.27 | jpsharp | The instant the bots detect open port 5060, they start wailing away. |
| 02:42.01 | SeRi | yeap... |
| 02:42.45 | raden | Gah |
| 02:42.55 | raden | suggestions to protect crap ? |
| 02:43.02 | raden | besides manually blocking it in the router |
| 02:43.27 | jpsharp | Some people run an instance of fail2ban. |
| 02:43.37 | raden | i gotcha |
| 02:44.07 | raden | would a ASA help ? |
| 02:46.42 | raden | i suppose I could just allow IP addresses in wisconsin inbound on that box ..... . |
| 02:46.43 | jpsharp | Not really. Either completely block port 5060 or make sure your SIP accounts have strong usernames/passwords. |
| 02:49.41 | raden | ill have to set a allow range for ISPS around here and then block all from there |
| 02:49.57 | *** part/#asterisk mjordan (~mjordan@adsl-98-81-86-236.hsv.bellsouth.net) |
| 02:49.58 | raden | so like accept IP range accept established deny all type deal |
| 02:50.02 | jpsharp | If you can get all those ranges. |
| 02:52.08 | jpsharp | Though I suppose you could to some fancypants BGP query through a lookingglass site. |
| 02:52.21 | raden | 95% of our voip inside |
| 02:52.35 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.171) |
| 02:52.35 | raden | so besides like 5 ranges of addresses im good |
| 02:52.41 | raden | may as well just lock her down |
| 02:54.58 | *** join/#asterisk luckman212 (~luckman21@2001:470:8abb:0:211:32ff:fe10:cdc1) |
| 03:00.08 | SeRi | I only allow my ITSP in and thats it |
| 03:00.25 | SeRi | everything else gets silently droped |
| 03:01.57 | jpsharp | If you know your acceptable IP ranges, that's awesome. |
| 03:04.44 | SeRi | how wouldnt you know? |
| 03:05.03 | SeRi | you have bigger problem if you dont know your ITSP ranges.... |
| 03:05.57 | jpsharp | Well, yeah. I was thinking about clients, rather than the ITSP. |
| 03:06.10 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 03:07.20 | SeRi | even than you should know your clients IP.... if they ar dynamic than use a dynamic service and allow the host in. |
| 03:07.37 | SeRi | I do it for family around the world |
| 03:07.43 | SeRi | well PR. |
| 03:08.29 | drmessano | If your usernames and passwords are strong, theres little to worry about |
| 03:08.45 | jpsharp | I had to manage it for a handful of sales droids who travelled around the world. How would you map them to something you know? |
| 03:09.17 | SeRi | using dynamic hosts... |
| 03:09.28 | SeRi | drmessano: yeap... true indeed |
| 03:09.51 | SeRi | I am paranoid though.. I hate seen my logs getting full of crap :) |
| 03:10.13 | drmessano | People pop in to here and #freepbx all the time paranoid that they're getting scanned/probed by some bot. Keep your crap difficult and the bots will move along. |
| 03:10.27 | jpsharp | I don't quite follow what you mean by "dynamic hosts". |
| 03:10.51 | SeRi | ie: dyndns.com |
| 03:10.57 | jpsharp | Oh. |
| 03:11.00 | SeRi | :) |
| 03:12.32 | jpsharp | The one time someone made unauthorized calls through * is because they rooted the box and swiped SIP credentials. |
| 03:13.20 | carrar | jpsharp, make them VPN in |
| 03:13.46 | carrar | viA laptop or phone |
| 03:13.51 | SeRi | carrar: +1 |
| 03:13.52 | jpsharp | I could barely get the sales droids to not use their yahoo accounts to handle sales contacts. |
| 03:14.07 | SeRi | lol |
| 03:14.21 | jpsharp | "Outlook was hard to use!" |
| 03:14.30 | carrar | then cut off access if they can't protect the company |
| 03:14.41 | jpsharp | Oh, I tried. |
| 03:14.53 | carrar | switch jobs and work for company that cares |
| 03:14.56 | carrar | heh |
| 03:14.58 | SeRi | LOL! |
| 03:15.00 | SeRi | nice |
| 03:15.03 | carrar | or enjoy the rooting |
| 03:15.05 | jpsharp | That's what I ended up doing. |
| 03:15.24 | carrar | make it a game |
| 03:15.30 | carrar | how amny times in a day |
| 03:15.33 | jpsharp | sent the CEO and supervisor an email that started off "Take this job and shove it..." |
| 03:15.53 | carrar | OPEN ROOT POLICY NOW ENFORCED |
| 03:15.57 | *** part/#asterisk Bullmoose (~Bullmoose@65-129-23-220.bois.qwest.net) |
| 03:16.21 | carrar | everyone from here on how will not be logging in as root |
| 03:16.27 | carrar | how=out |
| 03:16.31 | carrar | err |
| 03:16.32 | SeRi | todat at the astlinux user list somebody sent an email reminding how important is to secure your pbx.... He pasted his logs where his asterisk box got hacked... not nice... |
| 03:16.33 | carrar | man |
| 03:16.38 | carrar | time for bed I can't type |
| 03:16.51 | carrar | heh |
| 03:16.55 | jpsharp | The security there suck so badly that they're still using the same root passwords and SIP credentials as I set up. I haven't worked there in 4 years. |
| 03:17.08 | SeRi | thats sad |
| 03:17.30 | jpsharp | I fired up an old IP phone that I had from there and it registered without a problem. |
| 03:18.07 | carrar | they probably never replaced you |
| 03:18.23 | SeRi | lol... tehy probably think you will come back one day |
| 03:18.24 | carrar | sales people are doing IT now |
| 03:18.31 | drmessano | Maybe you could set up an Asterisk box that registers to that extension, then let all of us make anonymous calls to only the extension range on their system. |
| 03:18.41 | carrar | "Oh he's on a extended vacation" |
| 03:18.45 | SeRi | LOL ^^ |
| 03:18.45 | drmessano | We could all call and say hello |
| 03:18.52 | SeRi | drmessano: lol |
| 03:19.17 | jpsharp | SeRi: Not after I sued them and cleaned out their bank account. |
| 03:19.18 | drmessano | I *LOVE* just calling to say hello |
| 03:19.57 | SeRi | lol |
| 03:20.04 | SeRi | jpsharp: wow it was that bad |
| 03:20.32 | jpsharp | SeRi: Yep. One of the reasons I left was an extended period of bouncing paychecks. |
| 03:21.06 | jpsharp | They stiffed me on 2 pay checks, many hours of vacation pay, and the invoice for some equipment I sold them. |
| 03:21.35 | SeRi | jpsharp: back when I was consulting I had a company who's check bounce.... It only happen once... I made sure they pay me and quit on the spot |
| 03:21.50 | SeRi | I do not have time to stick around and "hope" the check does not bounce |
| 03:21.58 | carrar | OH YEAH |
| 03:22.01 | jpsharp | Filed suit, they never showed up in court, I got a default judgement, and since I knew when payday was and when money would be in the bank, I scheduled the asset garnishment for that day. |
| 03:22.02 | carrar | Well back in band camp |
| 03:22.06 | SeRi | LOL |
| 03:22.08 | SeRi | ROFL |
| 03:22.39 | carrar | I just have boring stories |
| 03:22.46 | SeRi | hahahahaha |
| 03:22.48 | jpsharp | But now it is "Cash upfront, boyo" before I do work. |
| 03:22.49 | carrar | Today I did rate index conversions |
| 03:23.04 | SeRi | carrar: fun fun fun.... |
| 03:23.15 | carrar | band camp would be more fun |
| 03:23.26 | jpsharp | Especially with Allyson Hannigan. |
| 03:23.29 | jpsharp | Ahem. |
| 03:23.49 | carrar | yeah |
| 03:23.53 | SeRi | I just got my navi in and been playing around with it |
| 03:23.59 | SeRi | I like it so far... |
| 03:24.13 | SeRi | for got to order the EHS :/ so waiting for that piece to come in |
| 03:24.57 | SeRi | jpsharp: I am over those days. I now work for reputable companys and not worrie about it... sometimes is a pain but it works out |
| 03:25.17 | SeRi | I was working for a univerity some time a go but didnt wor out for me so I bounced... |
| 03:26.54 | jpsharp | I retired on medical disability after I quit, but that's an entirely different story. |
| 03:28.29 | SeRi | almost *time* to bed |
| 03:28.40 | SeRi | conf tomorrow |
| 03:29.12 | SeRi | by the way raspberry pi makes an excellent low power server |
| 03:29.19 | SeRi | :) |
| 03:29.41 | jpsharp | I have one on order. |
| 03:29.55 | SeRi | jpsharp: you will have fun for sure |
| 03:30.08 | SeRi | just make sure you get a compatible sd card |
| 03:30.12 | SeRi | thats a bitch |
| 03:30.26 | SeRi | Thats my only real complaint |
| 03:30.56 | SeRi | right now is running my ssh/nx/gallery3/squeezebox server |
| 03:30.58 | jpsharp | I want to see if I can turn it into a low-power echolink system. |
| 03:31.40 | SeRi | I am sure you can. I am working on my second one as a zigbee server |
| 03:31.51 | SeRi | home automation |
| 03:33.05 | jpsharp | What are you using as the "brains"? I've played with Mr House, but just never could get it working. |
| 03:33.44 | SeRi | Th raspberry will controll everything |
| 03:33.57 | jpsharp | What software will drive it? |
| 03:34.16 | SeRi | http://www.ebay.com/itm/150856345817?ssPageName=STRK:MEWAX:IT&_trksid=p3984.m1423.l2649 |
| 03:34.21 | SeRi | ALL DETAILS THERE ^^ |
| 03:34.24 | SeRi | ops |
| 03:34.25 | SeRi | caps |
| 03:34.28 | drmessano | I've been pretty happy with X10, except that the price of the hardware DOUBLED this year due to supply issues |
| 03:35.31 | SeRi | drmessano: I looked in to X10. |
| 03:35.41 | jpsharp | I've got noisy power here, so X10 kinda sucks. |
| 03:35.45 | SeRi | just decided zigbee to start a project |
| 03:37.27 | jpsharp | Hmm. I didnt know there were so many zigbee home automation things. |
| 03:37.41 | SeRi | yeap |
| 03:37.49 | drmessano | I've strategically placed multiple transcievers and worked around the noisy spots in the place. I love the heyu application for running the show. Was VERY happy with my total X10 investment until this year. Now I can't justify $20 for an appliance module |
| 03:39.18 | carrar | Is it setup to call you on events? |
| 03:39.49 | carrar | "Hi dave, the bay pod access door is open" |
| 03:40.00 | drmessano | One thing I liked about the X10 was just how many oddball devices they had out there. The keychain remotes, panic buttons, the array of alarm hardware, wristwatch and pendant buttons, and all sorts of interfaces |
| 03:40.50 | drmessano | Apparently they have cut most of those oddball devices out and are only going to manufacture the core modules |
| 03:41.09 | drmessano | I don't have it call me, but I can call it and control it remotely |
| 03:41.28 | drmessano | I mostly do it from the web, but the Asterisk-makes-my-coffee thing is pretty near |
| 03:41.30 | drmessano | neat |
| 03:42.48 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 03:44.52 | SeRi | lol |
| 03:45.32 | jpsharp | No recording of Allison Smith asking "Coffee, Tea, or me?" |
| 03:45.43 | SeRi | ok g/n all |
| 03:46.05 | drmessano | Allison would playback "Coffee Maker On" thanks to Cepstral |
| 03:46.11 | drmessano | Though that hasn't worked in ages |
| 03:46.48 | drmessano | Cepstral can lick my left.. uh, nevermind |
| 03:49.17 | *** join/#asterisk mihamina (~mihamina@41.190.237.66) |
| 04:11.40 | *** join/#asterisk roe (~roe@unaffiliated/roe) |
| 04:12.05 | roe | I must say, I am finally setting up my own asterisk server from scratch. It is a bit overwhelming how complete the configuration is |
| 04:24.36 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 04:29.03 | jpsharp | roe: But by the same nature, it is extremely powerful. |
| 04:34.21 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 04:35.34 | roe | jpsharp, yes it is |
| 04:35.44 | roe | I'm missing something here |
| 04:36.51 | roe | I am running 1.8 and I am trying to make use of the built in contexts in extensions.conf |
| 04:36.59 | roe | specifically the stdexten |
| 04:42.16 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-zjcjriegmxfvnglz) |
| 04:44.52 | ChannelZ | That doesn't say a whole lot |
| 04:45.02 | roe | sorry, got distracted by reading docks |
| 04:45.04 | roe | docs* |
| 04:46.47 | ChannelZ | No sorrow necessary, we like people who read them :) |
| 04:52.20 | roe | what does the '#' sign in this mean: exten => #,1,Goto(menu),s,1)? |
| 04:52.34 | roe | I don't see it in the extension description |
| 04:53.33 | roe | oh is that the actual pound sign on the phone? |
| 04:54.36 | roe | sometimes I"m just thinking too hard |
| 05:17.02 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1004) |
| 05:18.28 | [TK]D-Fender | Roeyes, just # |
| 05:22.42 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 05:22.42 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 05:22.58 | *** join/#asterisk luckman212 (~luckman21@2001:470:8abb:0:211:32ff:fe10:cdc1) |
| 05:25.01 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 05:27.45 | raden | messing with gotoiftime |
| 05:27.51 | raden | where does asterisk get its time from |
| 05:27.56 | raden | how do i see it |
| 05:27.59 | [TK]D-Fender | System clock |
| 05:28.00 | raden | and how can i adjust it |
| 05:29.47 | raden | why are cdr logs always so off ? |
| 05:30.06 | [TK]D-Fender | because your system clock is so off |
| 05:30.21 | raden | its right on :( |
| 05:31.26 | [TK]D-Fender | compare to the HWclock |
| 05:31.37 | [TK]D-Fender | there are 2 timers |
| 05:31.39 | kaldemar | raden: [,timezone] |
| 05:33.29 | raden | hmmmmmm |
| 05:42.29 | raden | [Aug 29 00:41:51] WARNING[27809]: chan_sip.c:5467 create_addr: No such host: inbound |
| 05:47.29 | [TK]D-Fender | checkout time, later all |
| 05:47.59 | *** join/#asterisk josephtikva (~colloquyu@12.sub-70-192-132.myvzw.com) |
| 06:01.29 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1004) |
| 06:12.49 | *** part/#asterisk josephtikva (~colloquyu@12.sub-70-192-132.myvzw.com) |
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| 06:26.02 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 06:26.02 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 06:26.35 | *** join/#asterisk josephtikva (~josephtik@12.sub-70-192-132.myvzw.com) |
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| 06:34.44 | *** join/#asterisk FuriousGeorge (~chatzilla@ool-435655d6.dyn.optonline.net) |
| 06:36.08 | FuriousGeorge | hey all. i have a new server on centos, installed * using yum, and inbound calling is broken inexplicably. whereas before i could dial(sip/moe&sip/larry&sip/curley) and all would ring, now only larry rings |
| 06:36.22 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:36.33 | FuriousGeorge | i see nothing in the console to explain why, and i can't read sip debug info |
| 06:36.54 | josephtikva | Before when? |
| 06:36.56 | FuriousGeorge | i figure it must be something obvious, but I can't wrap my mind around what that might be |
| 06:37.16 | ChannelZ | show actual console output |
| 06:37.18 | ChannelZ | ~pb |
| 06:37.18 | 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. |
| 06:37.44 | *** join/#asterisk Assid (~assid@unaffiliated/assid) |
| 06:37.44 | FuriousGeorge | josephtikva: before when I was running 1.4.? on a different box. |
| 06:37.53 | Assid | how do i know if the card has echo cancellation or not? |
| 06:38.01 | Assid | i have a tormento pci card |
| 06:38.25 | FuriousGeorge | ChannelZ: i should do I will do it with sip debug, ok? console output tells us nothing |
| 06:38.50 | josephtikva | Core set verbose 10 |
| 06:39.26 | josephtikva | Tail /var/log/asterisk/full should have info |
| 06:39.47 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 06:39.54 | FuriousGeorge | josephtikva: sip debug on or no? |
| 06:40.16 | josephtikva | Yes. |
| 06:40.41 | FuriousGeorge | k thanks for helping me out. gimme a sec |
| 06:42.43 | FuriousGeorge | josephtikva: i have no /var/log/asterisk/full |
| 06:42.53 | FuriousGeorge | u want console in a pb |
| 06:42.55 | FuriousGeorge | ? |
| 06:43.07 | josephtikva | Yes |
| 06:44.48 | kaldemar | Assid: afaik, no tormenta card had on-board EC. |
| 06:45.47 | josephtikva | Maybe /var/log/asterisk/messages? Check /etc/asterisk/asterisk.conf for the astlogdir |
| 06:46.36 | ChannelZ | full isn't on by default |
| 06:47.20 | ChannelZ | just pb the console output (copy from the window) |
| 06:47.39 | ChannelZ | My guess is the console (not sip debug) is telling you what is wrong or will reveal a typeo |
| 06:48.52 | *** join/#asterisk vlad_starkov (~vlad_star@ppp91-79-65-225.pppoe.mtu-net.ru) |
| 06:49.05 | josephtikva | Maybe pb dialplan.conf. |
| 06:49.32 | ChannelZ | if that existed |
| 06:50.22 | FuriousGeorge | http://pastebin.ca/2198927 |
| 06:50.46 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 06:51.22 | FuriousGeorge | ChannelZ: josephtikva what's up with that address 67.18.187.245 i recognize that Ip and it should nto have anything to do with anything |
| 06:51.29 | FuriousGeorge | that's my xen slice |
| 06:51.47 | kaldemar | FuriousGeorge: those were ALL dialed. |
| 06:51.48 | ChannelZ | Looks like you've got some communications problems |
| 06:51.52 | ChannelZ | SIP retransmissions |
| 06:53.16 | ChannelZ | aron, maria, remotebrian all seem to have issues |
| 06:53.32 | kaldemar | asterisk has an address for them. maybe qualify=yes is not set and their NAT has closed the port. |
| 06:53.39 | ChannelZ | register too |
| 06:54.11 | ChannelZ | (guess Manny is the only one ringing?) |
| 06:54.22 | kaldemar | also, asterisk needs some NAT related configuration because there's a private address in the SDP (externaddr, localnet and nat=yes under [general]). |
| 06:55.55 | FuriousGeorge | ChannelZ: yup you guessed it |
| 06:56.07 | FuriousGeorge | kaldemar: i set externhost |
| 06:57.04 | ChannelZ | and localnet? |
| 06:57.30 | FuriousGeorge | ChannelZ: had it in 1.4, not sure what i did in 1.8 |
| 06:57.33 | FuriousGeorge | lemme check |
| 06:57.57 | ChannelZ | It tells it what is LAN and what isn't |
| 06:58.10 | FuriousGeorge | i know at least that much |
| 06:58.12 | FuriousGeorge | ;) |
| 06:58.29 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 06:58.29 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 06:58.30 | ChannelZ | sorry I thought you said "what IT did", misread |
| 06:58.52 | FuriousGeorge | lol, np, and thanks again for your help |
| 06:58.54 | FuriousGeorge | one sec |
| 06:58.57 | ChannelZ | Is Asterisk behind a firewall (or is the firewall)? |
| 06:59.04 | FuriousGeorge | behind |
| 06:59.32 | ChannelZ | And all these devices (except Manny) are outside? |
| 06:59.51 | FuriousGeorge | all inside |
| 07:00.09 | FuriousGeorge | looking at sip.conf and my dumbass forgot to uncomment the localnet setting |
| 07:00.13 | ChannelZ | is 67.18.187.245 your real external IP? |
| 07:00.36 | FuriousGeorge | no, its my linnode xen slice and should be totally unrelated. very mysterious |
| 07:01.14 | ChannelZ | Well its confused about how to represent its self based on your network setup |
| 07:01.17 | FuriousGeorge | im gonna try again with localnet uncommented |
| 07:01.47 | josephtikva | And reload chan_sip.so |
| 07:03.18 | ChannelZ | and click your heels |
| 07:03.25 | FuriousGeorge | ack cant remember how to turn off sip debug in post 1.4 * |
| 07:03.32 | FuriousGeorge | ChannelZ: no place like home |
| 07:03.33 | ChannelZ | sip set debug off |
| 07:03.42 | ChannelZ | ...same as 1.4... |
| 07:04.29 | FuriousGeorge | then i cant remember how to set it in any version ;) |
| 07:04.36 | FuriousGeorge | no change with localnet setting |
| 07:04.44 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 07:04.44 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 07:05.00 | ChannelZ | what did you actually set them to? |
| 07:05.17 | FuriousGeorge | 10.0.0.0/255.255.255.0 |
| 07:05.28 | josephtikva | 255.0.0.0 |
| 07:05.34 | ChannelZ | And for whatever reason you say all the phones are behind the firewall but shouldn't they have 10.x IPs and not that 67.* one? |
| 07:06.12 | FuriousGeorge | ChannelZ: that's my mystery too. where is that 67 address coming from? all phones have 10.0.0.X addresses incluiding manny |
| 07:06.20 | ChannelZ | sip show peers |
| 07:06.37 | ChannelZ | It's where Asterisk thinks they are. Why, that's a mystery of your network topology |
| 07:07.11 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:07.18 | FuriousGeorge | ChannelZ: sip show peers says nothing about 67.* address shows all phones on the 10.0.blah localnet |
| 07:07.50 | FuriousGeorge | Maria/Maria 10.0.0.120 D 2051 OK (14 ms) |
| 07:08.59 | FuriousGeorge | maybe a few years ago i was doing something with my xen slice, but that ip does not show up in sip.conf.... i have no idea why it's showing up at all |
| 07:09.23 | FuriousGeorge | i took extensions.conf, sip.conf, and voicemail.conf from the old server |
| 07:10.39 | ChannelZ | I don't know, what does a sip debug look like when one of the broken peers initially registers? |
| 07:10.59 | ChannelZ | Was your old setup doing realtime peers or something and you have a frankenconfig? |
| 07:11.21 | FuriousGeorge | ChannelZ: I'll check. no realtime before |
| 07:13.05 | ChannelZ | is your externaddr a hostname or IP? |
| 07:13.19 | FuriousGeorge | externhost => url |
| 07:13.27 | ChannelZ | uhm.. url? |
| 07:13.34 | ChannelZ | externaddr |
| 07:14.40 | ChannelZ | regardless if you are using externhost is that where that 67 IP is coming from? (but still it seems to not understand what is internal and what is external.. my guess is your sip.conf is still hosed) |
| 07:15.21 | josephtikva | Externip |
| 07:15.21 | FuriousGeorge | i used externhost and that is not where that url is coming from |
| 07:15.44 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:3cc3:afff:d6f6:d2bf) |
| 07:16.01 | FuriousGeorge | err that is not where that ip is coming form (the 67 one) |
| 07:16.26 | FuriousGeorge | josephtikva: it's dynamic |
| 07:17.43 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 07:19.18 | FuriousGeorge | ChannelZ: note that the broken phones can all call eachother and call out,. u still wanna see the sip debug from when one of them log in |
| 07:20.07 | ChannelZ | Yes |
| 07:20.12 | FuriousGeorge | k |
| 07:21.32 | ChannelZ | actually - a different question is, does it work if you only dial one of those peers? like just SIP/aron and remove all the rest |
| 07:28.40 | ChannelZ | well I gotta go to bed. Basically Asterisk is trying to send the calls to that 67...245 IP for all those devices for a reason I don't know based on what we have(nt) seen of your overall config. I don't have any guesses based on you saying those devices can call each other, if a sip debug reveals Asterisk transmitting the INVITEs to the proper 10.0.x.x IPs for those devices. |
| 07:29.44 | FuriousGeorge | ChannelZ: thanks im gonna hit the sack too |
| 07:29.46 | ChannelZ | Since all those devices are on the same side of your firewall however, it seems like they shouldn't be set for NAT in the first place |
| 07:30.02 | FuriousGeorge | they all nat=no |
| 07:30.17 | ChannelZ | then I'm confused as well why it thinks they are: |
| 07:30.34 | ChannelZ | Reliably Transmitting (NAT) to 67.18.187.245:5060 |
| 07:31.18 | FuriousGeorge | thats a gentoo httpd with no asterisk or sip stuff installed. i cant figure out where that comes from, lemme look in sip.conf again |
| 07:31.21 | ChannelZ | although come to think of it that might be an oddity of the output. I can't remember. |
| 07:32.16 | FuriousGeorge | dont sweat it, i can just rebuild it from scratch and that's the answer for everything |
| 07:34.24 | ChannelZ | pastebin 'sip show settings' |
| 07:42.37 | *** join/#asterisk irroot (~gregory@2002:294f:dde2:1:49f:62b2:7a25:b4ba) |
| 07:43.56 | FuriousGeorge | ChannelZ: k |
| 07:48.23 | FuriousGeorge | ChannelZ: ill pm it to u |
| 07:53.10 | fling | exten => ChannelZ,1,Hello() |
| 07:53.23 | fling | how can I reload extensions? |
| 07:53.24 | ChannelZ | yesh |
| 07:55.30 | *** join/#asterisk longst (~longst@46-22-127-214.bredband.alvsjo.qonet.se) |
| 07:57.56 | ChannelZ | dialplan reload |
| 07:58.03 | ChannelZ | sorry |
| 07:58.58 | fling | exten => ChannelZ,n,ThankYou() |
| 08:14.26 | *** join/#asterisk fredericve (~fes@wan.gw.belgacom.office.wavre.escaux.com) |
| 08:14.30 | Aelia | well I tried finding where my problem comes from, and I found out that Using ast_request_and_dial("Local", ast_best_codec(chan->nativeformats), chan, ext_and_context, TIMEOUT, &reason, chan->called.id.number.str, chan->chan.id.number.str) doesn't manage at all the 183 progress and I don't get any audio after the channel is UP. while instead of calling my application I put a Dial(Local/ext_and_context) it works ... |
| 08:15.13 | Aelia | should I setup manually the channel instead of using ast_request_and_dial ? if yes how should I set it up ? |
| 08:16.05 | fredericve | Hi all |
| 08:16.43 | fredericve | I think I found a bug in asterisk 1.8 concerning reinvites and I'd like to make sure before filing an issue on the bugtracker. |
| 08:17.57 | fredericve | I have 2 SIP peers, each of them with canreinvite=yes. One peer is only allow=alaw, the other one is only allow=ulaw |
| 08:19.16 | fredericve | when the call setup happens, asterisk tries reinvite, which fails, after which the channels to both peers are hung up |
| 08:19.36 | ChannelZ | well canreinvite is now called directmedia but I don't remember if it was totally deprecated in 1.8 |
| 08:22.26 | *** join/#asterisk wonderworld (~ww@dsdf-4db53c81.pool.mediaWays.net) |
| 08:22.58 | fredericve | oh, well the behaviour is in any case different if I change canreinvite to no so I guess it is not completely removed from 1.8 |
| 08:23.12 | fredericve | let me try playing around with directmedia |
| 08:23.55 | kaldemar | canreinvite still exists as a synonym to directmedia in 11. |
| 08:24.19 | ChannelZ | well, but I guess what you want to happen is that if the two endpoints can't talk directly via reinvite (because of incompatible codecs) Asterisk should fall back to being in the media path in order to do a transcode? |
| 08:24.35 | fredericve | correct |
| 08:25.10 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 08:25.15 | fredericve | either asterisk should not try the reinvite, or try the reinvite but not hang up the channels if it fails. |
| 08:25.39 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:27.17 | fling | ChannelZ: how to select exact moh from a directory? how to playback different moh if caller pressed '1'? |
| 08:28.12 | ChannelZ | well you can define different MOH contexts |
| 08:29.14 | fredericve | changing canreinvite to directmedia has the same behaviour so it indeed looks like it's a synonym to directmedia |
| 08:29.15 | ChannelZ | and set what MOH context the current channel is in via the CHANNEL(musicclass) function |
| 08:29.51 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 08:30.06 | fling | ChannelZ: can you please give me an example? or a link to read one |
| 08:30.40 | *** join/#asterisk anebi (~anebi@91.207.191.17) |
| 08:30.40 | ChannelZ | fredericve: either way.. I'm not actually sure what the defined behavior is _supposed_ to be, to be honest. I always thought it was as you desired, especially since * already knows the codecs aren't compatible because both peers are configured that way |
| 08:32.09 | ChannelZ | fling: well the MOH contexts are defined in musiconhold.conf - if you want specific songs for specific operations, you'd just have to make different directories and put those songs in them and set each context to the separate directories as desired |
| 08:32.42 | anebi | hi hi. I installed asterisk 1.8.4.4 on ubuntu 11.10 and i have problems on this installation with modules not loaded. Can you please tell me which are the modules responsible for ring groups and time conditions/time groups as i need to load them manually? |
| 08:33.34 | ChannelZ | Then if you want one set of songs to be played if someone gets put on hold having gone through some particular phone menu, you can Set(CHANNEL(musicclass)=easylistening) or whatever somewhere in your dialplan when they get to that menu |
| 08:33.39 | fling | ChannelZ: how to read and use digit from caller for changing moh? |
| 08:33.42 | fling | oh |
| 08:33.48 | ChannelZ | If they get put on hold sometime after that, they get that MOH context. |
| 08:34.16 | kaldemar | anebi: depends. just use autoload=yes in /etc/asterisk/modules.conf and then start to remove modules when you know what they do. |
| 08:35.30 | anebi | kaldemar: asterisk is set with autoload=yes, but this version has problems as i get errors like this one: Error loading module 'app_stack.so': /usr/lib/asterisk/modules/app_stack.so: undefined symbol: ast_agi_unregister |
| 08:35.49 | anebi | <PROTECTED> |
| 08:36.08 | fredericve | ChannelZ: Thanks. I'll try a bug report then. It looks like a bug to me |
| 08:36.47 | kaldemar | anebi: did you upgrade from an earlier version without cleaning the modules directory first? how did you install in the first place? |
| 08:38.02 | anebi | <PROTECTED> |
| 08:40.56 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:41.42 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:51.08 | fling | ChannelZ: somebody is calling and asterisk answers, playbacks welcome, starts moh and dialing sip client ; now I need to read '1' from caller somehow |
| 08:52.41 | fling | ChannelZ: http://dpaste.com/793018/ |
| 08:53.29 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 08:55.25 | fling | how to read a number while asterisk is dialing? |
| 08:57.07 | *** join/#asterisk longst (~longst@46-22-127-214.bredband.alvsjo.qonet.se) |
| 09:00.09 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 09:07.51 | fling | ChannelZ: looks like I need WaitExten() |
| 09:10.07 | *** join/#asterisk rhce7320 (~rhce7320@59.167.200.141) |
| 09:18.26 | fling | kaldemar: hey :p |
| 09:18.56 | fling | kaldemar: how to read numbers from caller while dialing and playing moh? |
| 09:19.20 | *** part/#asterisk anebi (~anebi@91.207.191.17) |
| 09:19.32 | fling | I want to change moh if caller press '1' |
| 09:22.58 | kaldemar | Background could do it if you ditch MOH. |
| 09:23.47 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 09:24.06 | fling | kaldemar: give me an example please |
| 09:24.37 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 09:25.01 | kaldemar | fling: http://pastebin.com/CS4txKF4 |
| 09:26.25 | kaldemar | there's lots of extra but you should get the idea. during background, if the caller presses *, the GoSub returns. if something else is pressed, it gets a new track to play with background. |
| 09:29.06 | fling | kaldemar: how to dial same time? here is the plan: somebody calls, asterisk says welcome, dials sip phone, starts moh |
| 09:29.26 | fling | and while noone answers caller may press 1 to listen to the commercial |
| 09:30.29 | fling | so if sip answers, asterisk just stops moh/commercial |
| 09:31.01 | fling | don't I need moh at all? do I need Bacground only if I want to read numbers? |
| 09:39.47 | fling | don't I need queue |
| 09:40.30 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 09:40.57 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 09:44.53 | kaldemar | fling: what's the point of all this? have a caller listen to music while the destination is dialed and let the caller change the music? that's all? |
| 09:45.21 | *** join/#asterisk TobSnyder (~schneider@146-52-60-185-dynip.superkabel.de) |
| 09:45.31 | fling | kaldemar: yes, change the music to commercial :p thats all! |
| 09:45.36 | TobSnyder | can I have the same username/password for multiple sip accounts? |
| 09:46.07 | TobSnyder | e.g. in sip.conf [20] [21] [22] but the softphones should be able to register with the same user/pass instead of 20 21 22 as username? |
| 09:46.25 | kaldemar | TobSnyder: you can do whatever you want. that being smart or safe is another matter. |
| 09:46.48 | TobSnyder | it's just for local tetsing purposes |
| 09:47.55 | TobSnyder | so would it be correct to use such a config: |
| 09:47.55 | TobSnyder | [20] |
| 09:47.55 | TobSnyder | username=test |
| 09:47.55 | TobSnyder | secret=1234 |
| 09:47.55 | TobSnyder | [21] |
| 09:47.55 | TobSnyder | username=test |
| 09:47.56 | TobSnyder | secret=1234 |
| 09:48.40 | kaldemar | fling: i'd try if features can be used first. add something to applicationmap that changes what the caller hears. |
| 09:50.13 | fling | hmm hmm |
| 09:57.09 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 09:58.51 | *** join/#asterisk rjvvliet (~rjvvliet@2001:610:600:86a5:595d:dacf:b55f:f0c6) |
| 10:01.19 | rjvvliet | Hi, anyone has any experience with Asterisk hints en Mitel 5220 SIP Firmware, i seem to have a problem when using about 8 BLF keys that the phone ignores calls on 'Hight' BLF load. |
| 10:02.19 | rjvvliet | Just brought is down to arond 5 and i now seems to have no problems. |
| 10:06.51 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 10:21.12 | *** join/#asterisk philfry28_ (~aj@h222.191.101.208.static.ip.windstream.net) |
| 10:23.47 | *** join/#asterisk jmls (~julian@host217-36-208-155.in-addr.btopenworld.com) |
| 10:23.56 | jmls | morning all |
| 10:25.12 | jmls | have a question about custom sounds. I have a customer that wants to record over some of the standard sounds files (e.g. demo-thanks) . I don't want to override the standard asterisk files, nor do I want to copy the entire sounds folder to a custom folder. |
| 10:25.40 | jmls | is there a way of saying to asterisk, "if you can't find demo-thanks in the foo folder, use the standard demo-thanks instead" ? |
| 10:26.32 | rjvvliet | hi jmls, you could play with the Language setting, when you set language=en-cust is will look in the durectory sounds\en-cust end is not exists uses sound\en |
| 10:26.49 | jmls | rjvvliet: I *hoped* that you were going to say that ;) |
| 10:27.24 | rjvvliet | remember, have not yet doen that mi self... i'am using language=nl ;-) |
| 10:27.28 | *** join/#asterisk mealstrom (~mealstrom@46.63.71.254) |
| 10:27.42 | jmls | so asterisk defaults to \nl if a file doesn't exist in the \nl-cust folder ? |
| 10:27.59 | rjvvliet | ni defaults to en |
| 10:28.19 | jmls | right. That's cool for me, as I use en ;) |
| 10:28.20 | rjvvliet | the default language is always en |
| 10:28.25 | rjvvliet | yeap |
| 10:28.41 | fling | kaldemar: this is my test setup > http://dpaste.com/793045/ |
| 10:28.54 | fling | it reads numbers when I call from softphone |
| 10:29.05 | fling | but not from pstn. How to fix this? |
| 10:29.09 | jmls | Â dank je wel |
| 10:29.22 | rjvvliet | jmls: graag gedaan ;-) |
| 10:29.25 | jmls | :) |
| 10:31.29 | mealstrom | hi, im using asterisk 1.8 and ive got problem with say.conf and language. If i put fullpath to directory at say.conf -- all is ok . if just simple digists/${SAY} - -this doesnt work. How to check in what directory it makes search? |
| 10:32.05 | mealstrom | i've putted Set(CHANNEL(language)=ua) before playback -- no help |
| 10:33.34 | *** join/#asterisk pyther (~pyther@unaffiliated/pyther) |
| 10:33.51 | mealstrom | asterisk.conf:languageprefix = yes, sip.conf:language=ua |
| 10:34.04 | kaldemar | fling: depends on how you connect to PSTN. |
| 10:34.15 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 10:36.56 | fling | kaldemar: over sip |
| 10:37.18 | kaldemar | fling: dtmfmode might be wrong. |
| 10:37.35 | fling | should I set it in [general] ? |
| 10:37.46 | kaldemar | fling: add "dtmf" to the console line in logger.conf and do "logger reload". you should see incoming recognized DTMF in CLI after that. |
| 10:38.23 | kaldemar | fling: no, under the peer that matches incoming calls from your provider. |
| 10:39.09 | fling | ok |
| 10:39.35 | *** join/#asterisk irroot (~gregory@2002:294f:dde2:1:4d3b:bcec:471:7312) |
| 10:41.05 | jmls | ok, another sounds question |
| 10:41.11 | fling | kaldemar: nothing in the console |
| 10:41.42 | jmls | is it possible to have the language directories somewhere other than /var/lib/asterisk/sounds/[language] |
| 10:42.04 | rjvvliet | jmls: Yep, look in asterisk.conf |
| 10:42.10 | jmls | meh |
| 10:42.12 | jmls | sorry |
| 10:42.47 | *** part/#asterisk irroot (~gregory@2002:294f:dde2:1:4d3b:bcec:471:7312) |
| 10:43.01 | rjvvliet | jmls: be ware that by default this section is a template look at the (!) behind it, when you do not remove it it just displays the defaults. |
| 10:43.36 | jmls | hmm. can't see anything relating to custom sounds |
| 10:44.08 | zamba | how can i make asterisk bind to both udp and tcp? |
| 10:44.20 | rjvvliet | jmls: mmm let me check... |
| 10:44.37 | Aelia | does someone know what's the meaning when you read a frame (with ast_read(chan)) of type AST_FRAME_CONTROL and the subclass is -1 ? it's not defined in frame.h so I'm kind of lost here... |
| 10:45.16 | kaldemar | fling: then DTMF is not recognized. rfc2833 is the most common way to send DTMF, it that mode first. |
| 10:45.42 | rjvvliet | jmls: mmm my mind is playing tricks on me.... i meant the [directories] section but sorry.... |
| 10:45.52 | rjvvliet | jmls: you could create a Sym link. |
| 10:46.13 | fling | kaldemar: exten => 2,1,Playback(transfer) <- so if I perss 2 on a softphone it playbacks |
| 10:46.17 | jmls | I didn't want to pollute the standard install directories with custom folders |
| 10:46.31 | fling | kaldemar: am I doing it wrong? do I need to use Read or something? |
| 10:46.41 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 10:47.04 | rjvvliet | jmls: i think its the only way. |
| 10:47.10 | kaldemar | fling: if Background is executing when 2 is pressed and that's in the same context, then that looks right. |
| 10:48.35 | jmls | wonders what would happen if the language was called "custom/foo" ... |
| 10:48.52 | fling | kaldemar: do I need background? I do not have it yet > http://dpaste.com/793045/ |
| 10:48.56 | jmls | would it read from sounds/custom/foo |
| 10:49.17 | coppice | jmls: the foo fighters would object |
| 10:49.30 | kaldemar | fling: app playback does not react to anything a user presses. |
| 10:49.50 | kaldemar | on the other hand, you have WaitExten which does. |
| 10:50.12 | rjvvliet | jmls: what you fill in the language= parameter wil be used as the language directory. |
| 10:50.34 | kaldemar | but the user would have to wait until playback has played it all, so... you need to change that. |
| 10:52.43 | *** join/#asterisk reisi (koivunej@mustatilhi.cs.tut.fi) |
| 10:56.30 | fling | kaldemar: right, I will use Background, thanks. I will try to fix dtmf tomorrow |
| 10:57.13 | fling | exten => kaldemar,n,Cya(gtg) |
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| 11:03.21 | *** part/#asterisk mealstrom (~mealstrom@46.63.71.254) |
| 11:06.24 | reisi | if i want to list all chan_*.so modules, what is the correct [like keyword] in 1.8? |
| 11:07.07 | kaldemar | ... "module show like chan_" |
| 11:07.14 | reisi | aah :D |
| 11:08.48 | reisi | ok, still with 1.8 (beginner here); i seem to have chan_sip.so loaded, but sip show peers (like in chapter 5 of asteriskdocs.org) fails; there's no command "sip", how do i find out the correct command? |
| 11:08.53 | *** join/#asterisk iulhk (~iulhk@116.71.190.126) |
| 11:10.06 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 11:11.45 | reisi | oh, i cannot unload or load it, i guess that's the reason.. |
| 11:12.23 | kaldemar | reisi: "sip show peers" is a correct command. there's something wrong with loading the module. enable verbosity with "core set verbose 10", core debug with "core set debug 10" and do "module unload chan_sip.so" and "module load chan_sip.so". see if you get some useful output. |
| 11:12.40 | kaldemar | otherwise, pastebin your sip.conf and someone will surely take a look at it. |
| 11:13.36 | reisi | yeah, with 'core set verbose 3' before 'module load chan_sip.so' i get that there's a permission problem |
| 11:15.38 | *** join/#asterisk d0pe_ (~d0pe_@31-16-49-111-dynip.superkabel.de) |
| 11:16.17 | reisi | nice, and we have a registration |
| 11:34.23 | reisi | in http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceConfig_id292056.html there's this example "same => n,Playback(hello-world)"; is this "hello-world" a file? exits non-zero, i might be missing an ubuntu package if it's a file? |
| 11:34.58 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:36.57 | reisi | pastebin of the call, from the answer http://pastebin.ca/2198980 |
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| 11:40.59 | kaldemar | reisi: it's a sound file that is expected to be found somewhere under /var/lib/asterisk/sounds. |
| 11:42.04 | *** join/#asterisk deo (~dnepangue@203.177.214.75) |
| 11:46.39 | zamba | anyone got any experience with tandberg c40 and if it's able to use just sip and voice? |
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| 11:57.07 | reisi | nice, hello-world is working |
| 12:02.31 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-23-220.bois.qwest.net) |
| 12:03.31 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:09.16 | *** join/#asterisk danfromuk (~IceChat77@2.27.38.64) |
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| 12:10.46 | danfromuk | Hi, has anyone set up shared line appearance with asterisk on a Panasonic KX-UT133/6? Specifically, is there any way to make the Shared Line Key go green when in use by the local user? Currently, the shared line key goes red and a DN key shows green. |
| 12:13.03 | [TK]D-Fender | danfromuk, You have 1 actualt line on that phone, and BLF's. BLF's will light up with their colour. The actual "line" key your call is on is going to light up in its colour. in yours (and most) people cases this is going to be different because the phone probably wants BLF to look different from an actual line in use |
| 12:14.54 | [TK]D-Fender | danAnd Your odds are like one in a million of finding someone else using *'s "SLA" with those phones specifically. |
| 12:18.16 | danfromuk | Thinking of taking a look at the * source code to see if I can get it to work how i need it. For example, when a call is placed on hold, the BLF key remains solid so theres no way to tell which calls are on hold and which are active. |
| 12:20.42 | [TK]D-Fender | danfromuk, just enable SIP debug, put a call on hold and lookk for a presence notification state change being sent out. If there isn't (and I doubt there), then there is nothing to do for this. If it does send it out (to the WATCHING phone) then its up to the phone to show the indication differently. |
| 12:21.16 | [TK]D-Fender | danfromuk, Polycom's for instance don't visually indicate "ringing" vs "in use" differently |
| 12:21.19 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 12:21.29 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 12:25.33 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 12:26.58 | danfromuk | [TK]D-Fender: do you know of any software PBXs that I could put in front of our asterisk boxes that would support BLF on shared lines like a key system does? |
| 12:27.22 | danfromuk | [TK]D-Fender: i really need to make this work as its a large sale |
| 12:28.00 | [TK]D-Fender | danfromuk, I would fail to see how * even plays into this solution you're thinking of. That other system would do EVERYTHING |
| 12:28.35 | danfromuk | We have a hosted pbx platform using asterisk. |
| 12:29.23 | [TK]D-Fender | Your system would not figure into this in any way I could figure as being meaningful. |
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| 12:31.55 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 12:35.33 | zamba | can someone help me figuring out how to register a tandberg c40 codec with asterisk? |
| 12:35.41 | zamba | it's supposed to support sip |
| 12:35.55 | cusco | hi... where can I find the sql table structure for cel ? |
| 12:36.19 | cusco | in 1.8 |
| 12:36.52 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
| 12:36.57 | TheCompWiz | mornin' folks. |
| 12:37.22 | *** join/#asterisk danfromuk (~IceChat77@2.27.38.64) |
| 12:38.04 | rjvvliet | cusco: https://wiki.asterisk.org/wiki/display/AST/CEL+Function |
| 12:38.25 | cusco | rjvvliet: I meant distributed in asterisk tarball |
| 12:38.35 | cusco | rjvvliet: I copied that and it is incorrect |
| 12:39.31 | rjvvliet | cusco: mmm |
| 12:41.22 | [TK]D-Fender | zamba, unless you've got an * cedec module, all you'd get is passthrough. No transcoding, indications, etc..... |
| 12:41.32 | [TK]D-Fender | codec* |
| 12:45.38 | zamba | [TK]D-Fender: what do you mean? |
| 12:46.09 | [TK]D-Fender | zamba, what is unclear there? |
| 12:46.11 | zamba | G.711, G.722, G.722.1, 64 bit & 128 bit MPEG4 AAC-LD, |
| 12:46.13 | zamba | <PROTECTED> |
| 12:46.25 | zamba | the c40 codec supports the audio codecs listed above |
| 12:46.48 | rjvvliet | cusco: does this help : http://files.rm-f.nl/upload/upload/CEL-table_structure_mysql.txt |
| 12:47.16 | zamba | i manage to get it to register, but it deregisters immediately |
| 12:47.33 | [TK]D-Fender | zamba, * doesn't know about that codec. Unless you have a complete codec module made for * then the best * could do is let the media pass through between 2 endpoints. * could not transcode it to play things like dialtone, use conferencing, record in any other format, no Chanspy to listen in, etc. |
| 12:47.47 | zamba | http://pastie.org/4609938 |
| 12:48.06 | zamba | we basically want it to work for voice only |
| 12:48.11 | zamba | as a telephone |
| 12:49.01 | [TK]D-Fender | Ah, I see the term "codec" is part of the PRODUCT MANE |
| 12:49.03 | [TK]D-Fender | NAME* |
| 12:49.12 | zamba | yup :) |
| 12:49.27 | zamba | https://www.google.com/search?q=c40+tandberg&ie=utf-8&oe=utf-8&client=ubuntu&channel=fs |
| 12:49.30 | zamba | first hit is the box |
| 12:49.36 | [TK]D-Fender | This is a horrible source of confusion. The marketing people who chose it should be SHOT |
| 12:49.55 | zamba | hehe |
| 12:49.58 | zamba | yeah |
| 12:50.06 | [TK]D-Fender | -- Registered SIP 'radarsat' at <ip of codec>:49261 > Saved useragent "TANDBERG/516 (TC5.1.2.289652)" for peer radarsat |
| 12:50.11 | [TK]D-Fender | ^^ lokos like it IS registered... |
| 12:50.31 | zamba | yeah, but then the codec (hehe) deregisters immediately afterwards |
| 12:51.53 | zamba | hold on.. giving you a screenshot, maybe you can figure better out what to fill in and how to set up the stanca in sip.conf |
| 12:52.15 | [TK]D-Fender | well .... go ask them for support.. * said "yes" ... they can't seem to accept yes for an answer ;) |
| 12:52.24 | tomodachi | zamba: dont screenshot the confg, use pastebin |
| 12:53.01 | josephtikva | ~pb |
| 12:53.01 | 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. |
| 12:53.05 | [TK]D-Fender | zamba, No need. * ACCEPTS the registration. |
| 12:53.13 | [TK]D-Fender | zamba, Like I said, it's their problem. |
| 12:53.27 | zamba | :) |
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| 13:03.32 | *** join/#asterisk deo (~dnepangue@122.53.72.218) |
| 13:03.56 | deo | good day all! |
| 13:04.55 | *** join/#asterisk josephtikva (~josephtik@12.sub-70-192-132.myvzw.com) |
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| 13:07.15 | rjvvliet | Hi,anyone has any experience with Asterisk hints en Mitel 5220 SIP Firmware, the Mitel seems to have a problem when using about 10 BLF keys than the phone refuses to take calls on 'High' BLF load. |
| 13:07.33 | leifmadsen | ya that sounds like a mitel phone... |
| 13:07.37 | rjvvliet | I now brought it down to around 5 and i now seems to have no problems, wondering if someone else noticed this. |
| 13:07.48 | Katty | hi |
| 13:07.59 | carrar | hi |
| 13:08.23 | rolandow | hi |
| 13:08.30 | rjvvliet | leifmadsen: mm oke great, your quick response on it tells me i will not solve this. |
| 13:08.57 | leifmadsen | never had good luck with mitel phones. Seem to be underpowered |
| 13:10.29 | rjvvliet | leifmadsen: Well, customer was not prepared yet to replace the phone, only the PBX, so i wil have to work with them.( and the SIP FW is so lacking features) |
| 13:10.42 | leifmadsen | yep, just saying |
| 13:10.52 | leifmadsen | it has no effect on me what phone you use :) |
| 13:10.59 | rjvvliet | leifmadsen: thanks for sharing ;-) |
| 13:13.26 | specing | Uhm |
| 13:13.54 | specing | Can I suggest improvements to the http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html here? |
| 13:14.36 | leifmadsen | specing: please use the errata page at http://oreilly.com/catalog/errata.csp?isbn=9780596517342 |
| 13:14.40 | leifmadsen | otherwise it won't get evaluated |
| 13:15.24 | danfromuk | leifmadsen: Not sure if you recall, I was looking for a phone with around 30 buttons that supports shared line appearance. You recommended I take a look at the panasonic KX-UT136. I've managed to get hold of 2 demo phones from my suppliers to try it out on. I'm wondering if you've ever seen an implementation that has the SLA key turn green when the local user is active on it? Rather than the DN key turning green. |
| 13:15.53 | leifmadsen | danfromuk: yes, I've made that work |
| 13:16.24 | danfromuk | leifmadsen: any idea if its documented anywhere? |
| 13:17.08 | leifmadsen | honestly not sure... the panasonic docs aren't very good. I had some help from a Panasonic engineer to get it working. |
| 13:17.32 | danfromuk | Ok, thanks. i'll see if i can get access to panasonic support on this. |
| 13:17.46 | leifmadsen | it will indeed work though with the correct settings |
| 13:17.52 | leifmadsen | I can say that much :) |
| 13:18.18 | leifmadsen | used the SLA apps in Asterisk along with several manufacturers, one of them being Panasonic KX-UT products |
| 13:18.31 | specing | leifmadsen: Im not sure where to get the page number since Im reading the online version... |
| 13:18.45 | leifmadsen | specing: just say the section name |
| 13:19.01 | leifmadsen | put in some bogus page number |
| 13:20.17 | kaldemar | |
| 13:20.45 | rolandow | <PROTECTED> |
| 13:21.35 | carrar | page 2469 |
| 13:23.20 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 13:26.57 | *** join/#asterisk jgreig (jgreig@89.21.227.66) |
| 13:29.50 | jgreig | Not sure if anyone can help on this but i'll ask anyway: We've got an asterisk PBX setup and we use a VOIP provider (registered trunks). When we call a number that no longer exists the provider sends back a '403 forbidden' which our PBX then plays the 'all circuits are busy' macro. Is there anyway that we can stop our PBX from playing the macro and instead just playing whatever the provider is saying on the other end? |
| 13:30.00 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
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| 13:30.30 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:31.15 | jgreig | I should add we use freepbx. |
| 13:31.51 | rjvvliet | jgreig: i don't think you get a sound message from the ISP, you get a SIP 403 message in dialog so the call is not yet setup. |
| 13:31.56 | danfromuk | jgreig: if the provider sends 403 Forbidden, it means that they aren't 'playing' anything |
| 13:32.24 | rjvvliet | danfromuk: oeps.... my bad, 403 it Auth required ... |
| 13:32.39 | jgreig | chan_sip.c: Received response: "Forbidden" from Everyone is busy/congested at this time |
| 13:32.56 | danfromuk | If you are using freepbx, i recommend finding the 'all circuits are busy' sound file and replace it with a custom one. |
| 13:33.03 | jgreig | I guess, what I'm trying to achieve is to avoid thinking that the trunks are actually busy or there's a bigger issue than what there actually is |
| 13:33.05 | danfromuk | there may be a better way of doing this |
| 13:33.43 | jgreig | is it normal for providers to just send a 403? |
| 13:34.53 | rjvvliet | jgreig: Are you sure that the 403 is not the result of a problemn with a NAT setup? |
| 13:35.56 | jgreig | we do only get it on numbers that are actually non-existent (tested on mobile) - we don't use NAT, though, it's possible that an option could be in there somewhere that shouldn't be RE NAT |
| 13:36.38 | [TK]D-Fender | 403 is not an appropriate response to an invalid number being dialed. |
| 13:36.44 | rjvvliet | jgreig: i have found that some NAT solutions are not working perfectly and the PBX never receives the 403 and reinvites a second time without Auth. |
| 13:38.11 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-qgnhpduvllxftdbg) |
| 13:38.11 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:38.51 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-94-17.user.veloxzone.com.br) |
| 13:39.10 | [TK]D-Fender | danfromuk, Reminder that NO-ONE uses those phones here. Time to dive deep into whatever admisitrators guide you've got to search for this. You very likely don't havee an option at all. |
| 13:39.33 | kaldemar | neither is 403 appropriate for requiring authentication. |
| 13:41.07 | deo | hello everyone, need some help here.. |
| 13:42.07 | [TK]D-Fender | 401 can be a challenge. 403 = GTFO |
| 13:42.11 | deo | we have an 8 analog lines coming from our ISP |
| 13:42.25 | deo | it is connected to a old panasonic pabx |
| 13:42.35 | [TK]D-Fender | deo, Neat trick .... ISP's tend to deliver "IP", not analog copper |
| 13:43.01 | deo | i want to get rid of our old pbx and want to try asterisk pbx |
| 13:43.17 | deo | [TK]D-Fender: we have an analog lines |
| 13:43.35 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 13:43.56 | [TK]D-Fender | deo, Well, Asterisk can be used to make a pretty nice and flexible phone system... |
| 13:44.07 | deo | yeah right |
| 13:44.08 | [TK]D-Fender | deo, We highly recommend it :) |
| 13:44.19 | deo | i want to try asterisk [TK]D-Fender |
| 13:44.29 | deo | but the problem is i dont know where to start |
| 13:44.37 | [TK]D-Fender | deo, Then today's bonus : you have our permission to start! |
| 13:44.42 | [TK]D-Fender | ~book |
| 13:44.42 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:44.45 | deo | i want to transfer all the 8 lines into asterisk |
| 13:44.46 | [TK]D-Fender | deo, ^^ |
| 13:44.51 | deo | hmmnnn |
| 13:45.13 | deo | one question [TK]D-Fender |
| 13:45.45 | [TK]D-Fender | deo, First, if your ISP is offering you direct copper analog lines right off the pole then it sounds more like they are a basic telco that just happens to provide your internet connectivity as well (by whatever means that is delivered). |
| 13:45.47 | deo | do i need some additional hardware except for the computer |
| 13:46.13 | deo | yeah [TK]D-Fender those analog lines do have internet as well |
| 13:46.33 | [TK]D-Fender | deo, If you are intending to use those lines, then you clearly do. Signals don't tend to magically flow over the air into your typical computer to let your server control those lines... |
| 13:46.37 | deo | ive heard of some channel bank stuff.. but i dnot know what are they |
| 13:46.40 | danfromuk | [TK]D-Fender: thanks for your last message to me, however i was responding to jgreig. |
| 13:47.01 | [TK]D-Fender | deo, there are lots of devices to let you use analog lines. Which one's we'd recommend depend on certain scenarios. |
| 13:47.12 | deo | hmm what is it [TK]D-Fender ?? |
| 13:47.14 | [TK]D-Fender | deo, but it's best to take a step back and see if you have other options first |
| 13:47.14 | rolandow | deo: you could transfer your phonenumbers to a sip provider, and have all telephony through sip |
| 13:47.17 | jgreig | danfromuk: thanks for the advice might do as you said and change the sound file |
| 13:47.33 | deo | how do i connect all those 8 lines to my asterisk pbx? |
| 13:47.48 | [TK]D-Fender | deo, rolandow has mentioned possibly porting number to an ITSP so that you don't need special hardware and your calls arrive over an internet connection. |
| 13:47.49 | deo | i dont want to have a provider [TK]D-Fender |
| 13:48.00 | deo | i want to install asterisk pbx |
| 13:48.05 | [TK]D-Fender | deo, what are some of your reasons? |
| 13:48.10 | [TK]D-Fender | deo, just for contrast... |
| 13:48.28 | deo | hmmn first, i want to learn how to configure asterisk from scratch |
| 13:48.42 | deo | which i believe that book you sent will help |
| 13:48.49 | [TK]D-Fender | It should. |
| 13:49.00 | [TK]D-Fender | and you don't need special hardware to learn *. |
| 13:49.08 | deo | secondly, if i had to go on a provider, that is an addtional cost for us |
| 13:49.19 | [TK]D-Fender | deo, Not if they replace those lines. |
| 13:49.34 | rolandow | deo: usually voip providers have better rates than your analog line |
| 13:49.35 | [TK]D-Fender | deo, How much do you pay for them? |
| 13:49.40 | rolandow | deo: and you don't need additional hardware |
| 13:49.43 | [TK]D-Fender | ^^ |
| 13:50.03 | deo | i dont know exactly the exact price. it should be on the accounting dept. |
| 13:50.27 | [TK]D-Fender | deo, get that info. Really. You should understand the big picture before really settling on a "solution" as a complete package. |
| 13:50.28 | danfromuk | jgreig: maybe ask in #freepbx before you do |
| 13:50.55 | [TK]D-Fender | jgreig, There is a FreePBX module that lets you change the failure announcements, etc. |
| 13:51.19 | deo | so maybe [TK]D-Fender you can tell me what are the requirements in setting asterisk pbx... |
| 13:51.35 | [TK]D-Fender | deo, A computer to run it on. That's all. |
| 13:51.40 | deo | including hardware.. |
| 13:51.52 | [TK]D-Fender | deo, Some people run them on crappy linksys routers you can get for $30 |
| 13:52.01 | deo | analog lines > asterisk pbx |
| 13:52.15 | deo | how do I connect all those 8 lines into a computer? |
| 13:52.16 | [TK]D-Fender | deo, I'm sure your needs will be greater though. So any sheap computer will do probably |
| 13:52.27 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-23-220.bois.qwest.net) |
| 13:52.50 | deo | how do I connect all those 8 lines into a computer? |
| 13:52.52 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 13:52.54 | deo | ^ |
| 13:53.02 | rjvvliet | deo: i build my PBX's on Atom systemboards will do for about 50 to 75 Users. |
| 13:53.20 | deo | yeah thats good |
| 13:53.23 | [TK]D-Fender | deo, Here are 2 options for conencting the lines : http://www.telephonydepot.com/Catalog/Digium-TDM800P/Digium-AEX800P-Blank-Board |
| 13:53.25 | drmessano | I couldnt imagine putting 8 analog lines into a box. You get a simple card for it |
| 13:53.27 | jgreig | ok, cheers guys will take a look |
| 13:53.28 | *** join/#asterisk Johnny- (~John@23-24-48-118-static.hfc.comcastbusiness.net) |
| 13:53.42 | carrar | What about some old extra sounds cards? |
| 13:53.42 | [TK]D-Fender | deo, that card with 2 X100M modules listed below |
| 13:53.47 | deo | drmessano: that is what im looking for |
| 13:53.52 | [TK]D-Fender | deo, as an "internal" solution. |
| 13:53.53 | deo | so I need a card. |
| 13:54.13 | [TK]D-Fender | deo for external : http://www.telephonydepot.com/Catalog/Analog-Gateways/AudioCodes-MediaPack-MP-118-FXO |
| 13:54.27 | drmessano | 8 analog lines from AT&T is about half of what I pay for 32 SIP channels from my provider |
| 13:54.27 | [TK]D-Fender | deo, No, a card is ONE way. Clearly there are external devices as well. |
| 13:54.32 | drmessano | No way I would ever go back |
| 13:54.56 | [TK]D-Fender | deo, So there you go. 2 hardware options to let your server use your lines. |
| 13:55.11 | [TK]D-Fender | deo, there are many makers of both cards and external gateways. |
| 13:55.20 | deo | hmmmmnn maybe i need that 8 port card [TK]D-Fender |
| 13:55.27 | carrar | Adtran works nicely too |
| 13:55.32 | [TK]D-Fender | carrar, Sure.... |
| 13:55.55 | carrar | ADIT600!! |
| 13:56.11 | carrar | heh |
| 13:56.18 | [TK]D-Fender | deo, What you need ... is now to fully understand what solution you have now, it's costs, and what other options are available to you in your area. |
| 13:56.49 | [TK]D-Fender | deo, Analog sucks. Hard. BOTH WAYS. It's what you use when you're screwed beyond any realistic hope of an alternative. |
| 13:56.51 | deo | hmmnn |
| 13:57.08 | deo | thats what i want to know |
| 13:57.34 | rolandow | maybe for a backup if you don't have a proper internet connection |
| 13:57.36 | [TK]D-Fender | deo, But I gave you 2 sample hardware options so you can put that on the table for consideration. But I highly recommend you look at the big picture and not just jump at them. |
| 13:57.51 | deo | what u mean [TK]D-Fender ?? |
| 13:58.07 | [TK]D-Fender | deo, What part of what I've told you are you unsure about? |
| 13:58.22 | [TK]D-Fender | deo, Nothing I've said should be hard to follow... |
| 13:59.47 | deo | [TK]D-Fender: some one told me before that i need to use channel banks like Rhino for me to connect all those 8 analog lines.. and that channel bank will be connected to the asterisk pbx |
| 13:59.51 | deo | is that also right?? |
| 13:59.55 | deo | [TK]D-Fender: ^ |
| 14:00.04 | [TK]D-Fender | No. You don't NEED one. They are another option however |
| 14:00.23 | [TK]D-Fender | deo, However you'd also need a T1 card to plug that channel-back into. |
| 14:00.25 | deo | but that 8 port card could replace channel bank right? |
| 14:00.27 | [TK]D-Fender | bank* |
| 14:00.33 | deo | yup.. |
| 14:00.59 | [TK]D-Fender | deo, http://www.telephonydepot.com/Catalog/Rhino-Channel-Banks/Rhino-24-Channel-Bank-FXS-FXO-Combination |
| 14:01.13 | [TK]D-Fender | deo, http://www.telephonydepot.com/Catalog/Digium-Digital-T1-E1-J1-Cards/Digium-TE121 |
| 14:01.20 | [TK]D-Fender | deo, You'd need TWO pieces for that solution. |
| 14:01.28 | [TK]D-Fender | deAnd as yuo can see, FAR more expensive. |
| 14:01.29 | deo | and it need a wiring block also to connect those lines to channel bank right? |
| 14:01.46 | [TK]D-Fender | deo, Yes. No pretty RJ11 jacks on the back. |
| 14:02.20 | [TK]D-Fender | deo, Channel Banks are things you use if you already have a T1 card available and happen to get a good deal |
| 14:02.37 | deo | hhmm i see.. so that is expensive. because i need a TE121 card to connect channel bank to my asterisk pbx |
| 14:02.39 | deo | hmmmnn |
| 14:02.41 | [TK]D-Fender | deo, Never something I recommend to someone starting from scratch |
| 14:02.48 | rolandow | the lcd screen on the rhino looks impressive though |
| 14:02.56 | rolandow | could be something that managers fancy |
| 14:03.03 | [TK]D-Fender | rolandow, I have one. I occasionally notice it's on my rack. |
| 14:03.17 | deo | then what can you recommend to me [TK]D-Fender ?"?? |
| 14:03.18 | [TK]D-Fender | rolandow, To which I say "Do I care?". You acn guess what the answer is... |
| 14:03.22 | deo | based from my scenario.. |
| 14:03.33 | jgreig | heh just found a similar thread: http://www.elastix.org/component/kunena/3-help/58937-all-circuits-are-busy-now-vs-busy-signal.html |
| 14:03.39 | deo | 8 analog lines, almost 100 user in a buliding |
| 14:03.40 | rolandow | :) |
| 14:03.58 | specing | holy**** digium hardware is expensive |
| 14:04.04 | [TK]D-Fender | deo, I already gave you my recommendation. Find out what you're paying for those lines. See how much you could get a PRI from your telco for. full or fractional down to 8 channels, etc. Then we can look at ITSP's |
| 14:04.14 | [TK]D-Fender | specing, As compared to ...? |
| 14:04.21 | carrar | sound cards |
| 14:04.26 | [TK]D-Fender | ~savemoney |
| 14:04.26 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 14:04.29 | [TK]D-Fender | \o/ |
| 14:04.42 | deo | hmmnn a bit confused by PRI [TK]D-Fender |
| 14:04.45 | [TK]D-Fender | Rockin' it OLD SKOOL Y0! |
| 14:04.47 | specing | [TK]D-Fender: I don't know |
| 14:04.55 | [TK]D-Fender | deo, PRI is a digital voice circuit |
| 14:05.07 | deo | can i get PRI from our ISP? |
| 14:05.12 | [TK]D-Fender | specing, How acn you say X is expensiev without Y as a point of comparison? |
| 14:05.12 | deo | [TK]D-Fender: ^ |
| 14:05.13 | specing | [TK]D-Fender: I just went there to see how much this (analog) costs in real life |
| 14:05.20 | [TK]D-Fender | deo, maybe |
| 14:05.25 | [TK]D-Fender | deo, Ask them. |
| 14:05.36 | deo | hmmnn what if not? [TK]D-Fender ?? |
| 14:05.37 | specing | [TK]D-Fender: Im a student, $1000 isn't exactly cheap for me ;P |
| 14:05.55 | deo | should i go to 8 port card? |
| 14:05.58 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:05.59 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:06.25 | [TK]D-Fender | specing, Expensive is relative. You probably also couldn't afford a Ferrari at $10,000 even if it was brand new typically selling for 10 times as much. |
| 14:06.40 | rolandow | specing: if you're a student, what do you need that hardware for anyways? |
| 14:06.48 | [TK]D-Fender | specing, so saying a card is expensive just because your broke is retarded :) |
| 14:06.57 | [TK]D-Fender | you're* |
| 14:07.06 | deo | [TK]D-Fender: im sorry but what is ITSP?? |
| 14:07.09 | carrar | WE TODD DID |
| 14:07.10 | [TK]D-Fender | ~itsp |
| 14:07.10 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 14:07.13 | [TK]D-Fender | ^^^ |
| 14:07.36 | deo | hhmmn ok i got that.. |
| 14:07.38 | *** join/#asterisk italorossi (~Adium@189.124.200.92) |
| 14:07.45 | rolandow | deo: basically they have all the expensive hardware for you to convert from a "real" phone number to your asterisk box. |
| 14:07.59 | deo | what provider can u recommend [TK]D-Fender rolandow ?? |
| 14:08.05 | [TK]D-Fender | deo, go find out what all that copper is costing you. |
| 14:08.06 | rolandow | depends on your location |
| 14:08.09 | carrar | a college student can use a free softphone |
| 14:08.11 | zamba | [TK]D-Fender: i managed to register to call TO the codec, but i'm not able to dial out, because of this: Aug 29 13:50:30 (none) main: 523927.15 SipReg W: Outbound is enabled, but it was not supported all the way to the registrar. Stopping registrations. |
| 14:08.21 | zamba | [TK]D-Fender: do you have any guess to what that might mean? |
| 14:08.27 | [TK]D-Fender | deo, Stop. Go find out what you pay NOW. Then come back to us. |
| 14:08.40 | [TK]D-Fender | deo, We should not be making recommendations blind. |
| 14:08.59 | deo | hmnn ok ok.... ill ask from accounting dept, how much did those 8 lines cost monthly |
| 14:09.04 | rolandow | well you may be able to say that voipbuster isn't as good as ... whatever other company :) |
| 14:09.16 | [TK]D-Fender | zamba, Nope and as those are messages from your other hardware... you've got to check with them... |
| 14:09.18 | *** join/#asterisk pbxMan (c335d9a2@gateway/web/freenode/ip.195.53.217.162) |
| 14:09.21 | rolandow | deo: also check the rates .. what do you pay per second for a call? |
| 14:09.25 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:09.26 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:09.33 | [TK]D-Fender | rolandow, Yeah well I'd rther not waste time before seeing the big picture |
| 14:09.36 | deo | hmmm ok ok.. |
| 14:09.37 | rolandow | deo: also check if you call international a lot, and check those rates |
| 14:09.46 | pbxMan | hello |
| 14:09.47 | deo | we cant call international |
| 14:09.51 | [TK]D-Fender | Yes, do check your LD expenses on top of basic service. |
| 14:09.52 | deo | rolandow: ^ |
| 14:10.23 | specing | rolandow: I just wanted to see how much it costs |
| 14:10.46 | specing | [TK]D-Fender: It looks expensive nevertheless |
| 14:10.51 | rolandow | specing: how much what costs? asterisk is free .. hardware is almost free |
| 14:11.04 | carrar | Whats expensive? |
| 14:11.12 | rolandow | specing: you could start with voipbuster or something like that which is almost free.. i registered at 12connect and got a free incoming phonenumber.. |
| 14:11.15 | carrar | a Sonus SBC is expensive |
| 14:11.30 | rolandow | so .. i guess it doesn't have to be all that expensive :) |
| 14:11.40 | deo | thank you so much for you time [TK]D-Fender |
| 14:11.44 | [TK]D-Fender | carrar, No it's not... Bill Gates could but the whole company with his hourly allowance ;) |
| 14:11.48 | deo | ill be back with the rates |
| 14:11.52 | carrar | haha true |
| 14:11.59 | carrar | he's my neighbor too |
| 14:12.03 | rolandow | deo: what country are you in? |
| 14:12.09 | [TK]D-Fender | Philipines... |
| 14:12.10 | carrar | by about 4 miles |
| 14:12.11 | deo | im from Phlippines |
| 14:12.16 | rolandow | ok |
| 14:12.25 | deo | PLDT is our telco |
| 14:12.49 | carrar | deo |
| 14:12.50 | [TK]D-Fender | deo, Analog may be the worthwhile thing in the end, but you should make that decision once you have all the #'s |
| 14:12.56 | carrar | deo, isn't voip illegal in the Phlippines |
| 14:13.01 | [TK]D-Fender | carrar, no |
| 14:13.10 | deo | nope its not.. |
| 14:13.10 | [TK]D-Fender | carrar, I've had clients out there |
| 14:13.15 | carrar | whasn't there someplcae that it is? |
| 14:13.20 | *** join/#asterisk longst (~longst@46-22-127-214.bredband.alvsjo.qonet.se) |
| 14:13.25 | deo | maybe on some places carrar |
| 14:13.31 | [TK]D-Fender | carrar, India is pretty locked down, Pakistan as well IIRC |
| 14:13.35 | deo | i thnk there is an issue with that |
| 14:13.48 | deo | but voip in here is still legal :D |
| 14:14.01 | deo | and im trying to do it in our company :) |
| 14:14.25 | carrar | maybe it was just voip going over seas or something |
| 14:14.35 | deo | to save expenses isn't it [TK]D-Fender ?? |
| 14:14.37 | carrar | international |
| 14:14.39 | deo | :) |
| 14:15.05 | [TK]D-Fender | deo, what you pay for your lines you have now VS what another solution will look like could be very different |
| 14:15.15 | [TK]D-Fender | deo, So go egt all the #'s |
| 14:15.18 | [TK]D-Fender | get* |
| 14:16.27 | deo | ok ok.. thank s [TK]D-Fender |
| 14:16.39 | deo | ill be back for that #s |
| 14:16.49 | deo | im out for now.. :) |
| 14:17.03 | deo | thanks for all the help..really appreciate it.. |
| 14:17.09 | deo | bye.. :) |
| 14:17.11 | *** part/#asterisk deo (~dnepangue@122.53.72.218) |
| 14:17.51 | carrar | http://www.ekantipur.com/the-kathmandu-post/2010/08/12/top-story/cops-bust-major-voip-racket-in-capital/211500/ |
| 14:18.27 | carrar | who knows what the "real" story is |
| 14:19.16 | *** join/#asterisk navaismo (~navaismo@189.144.212.238) |
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| 14:29.41 | fredericve | Anyone knows what the expected behaviour is when reinvites are enabled and 2 channels have incompatible codecs? |
| 14:32.53 | fredericve | so one SIP friend has allow=alaw, the other one allow=ulaw e.g. |
| 14:34.13 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1005) |
| 14:34.44 | [TK]D-Fender | reinvite = renegotiate |
| 14:35.30 | fredericve | Ok, I mean when directmedia=yes |
| 14:35.41 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 14:36.31 | [TK]D-Fender | same thing. |
| 14:36.45 | [TK]D-Fender | that is a renamed parameter, not something separate |
| 14:38.10 | *** join/#asterisk chris_n (~Chris@nc-63-162-197-228.sta.embarqhsd.net) |
| 14:38.20 | fredericve | In the setup I explained above, asterisk tries reinvite, fails, and then hangs up both channels |
| 14:39.16 | chris_n | are deb packages available for asterisk 10 and 11beta? |
| 14:39.38 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 14:40.20 | specing | My country is not in indications.conf :( |
| 14:40.47 | [TK]D-Fender | specing, Get writing... |
| 14:40.58 | leifmadsen | it's probably just easier to change countries |
| 14:40.58 | fredericve | chris_n: I briefly looked for them this morning but could not find any. I don't know if they are out there somewhere |
| 14:41.29 | leifmadsen | fredericve: no, the deb files are not currently being updated |
| 14:41.33 | specing | [TK]D-Fender: But, But.. I don't know what to write! |
| 14:41.39 | chris_n | fredericve: I could not find them either; I was hoping to avoid building from source, but it may be better that way |
| 14:41.52 | [TK]D-Fender | specing, Go look up what your area's standards are |
| 14:42.41 | specing | Where? |
| 14:43.08 | specing | I don't even remember having a real telephone line... |
| 14:43.27 | chris_n | any real reason not to put 11beta into production? |
| 14:44.22 | fredericve | chris_n: you could try apt-get source and replace it with the * 10 or 11 source. Then try debuild and see where it gets you. I doubt you'll get very far though |
| 14:45.24 | *** join/#asterisk teff (~teff@client-80-1-164-210.bsh-bng-011.adsl.virginmedia.net) |
| 14:45.25 | fredericve | chris_n: because it's a beta release and not widely tested I guess |
| 14:45.38 | leifmadsen | isn't that the point of betas? :) |
| 14:46.02 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 14:46.11 | chris_n | is bad for running betas in production, but maybe should not do that with the phone system |
| 14:47.15 | [TK]D-Fender | specing, Ask your telcos. GOOGLE. Go to a library. Perform some ritual to whatever deity(s/y/ies) you may worship. |
| 14:48.11 | specing | Meh Im just going to leave it as "us" |
| 14:51.37 | [TK]D-Fender | specing, So cheap AND lazy. You just need to add "sense of entitlement" for the trifecta ;) |
| 14:54.01 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 14:56.00 | specing | [TK]D-Fender: :) |
| 14:56.57 | [TK]D-Fender | specing, Actually.. you are "whiney", so I think that's a fair substitute. You can now claim a small plushie from the bottom shelf... |
| 14:57.09 | [TK]D-Fender | :p |
| 14:57.58 | specing | I inputed what the book told me to put into modules.conf; I get 156 modules loaded while the book tells me there shouldn't be any ;_; |
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| 15:00.38 | specing | Is res_adsi.so internal? |
| 15:00.46 | [TK]D-Fender | specing, And we might have something to say if you've got something to show... |
| 15:00.48 | [TK]D-Fender | ~pb |
| 15:00.48 | 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. |
| 15:00.50 | [TK]D-Fender | ^^^ |
| 15:01.07 | specing | I put it in noload => explicitly and it is still listed in "module show" |
| 15:01.19 | [TK]D-Fender | specing, Show us all of it.. |
| 15:01.19 | specing | autoload is set to no |
| 15:01.25 | specing | Okok |
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| 15:05.18 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 15:05.41 | specing | [TK]D-Fender: https://gist.github.com/3513984 |
| 15:07.12 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 15:07.25 | specing | [TK]D-Fender: call me whiney all you want, the fact is that it is not working as it should |
| 15:09.52 | specing | *grin* |
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| 15:10.28 | zamba | [TK]D-Fender: i've figured it out.. the video conferencing solution uses h323 by default.. so if i want to call using my asterisk, i have to prefix the number with 'sip:'.. the problem with this, however, is that i have to learn my users to prefix "sip:" or "h323:" (depending on what i've set as default protocol).. can asterisk handle this somehow? let all calls go on sip and then just "send back, and try |
| 15:10.31 | zamba | yourself" for all calls that should use h323 (ip addresses) |
| 15:11.40 | zamba | if that makes any sense.. |
| 15:12.16 | TheCompWiz | you'd have to treat them as two separate extensions... as they're handled through two separate channels. |
| 15:12.28 | WIMPy | Not to me. Maybe I missed an earlier part of the story. |
| 15:12.34 | zamba | TheCompWiz: talking to me? |
| 15:12.38 | TheCompWiz | yup. |
| 15:12.43 | TheCompWiz | but sure... with enough dialplan... just about anything can be done. |
| 15:12.51 | zamba | TheCompWiz: well, basically i don't want the h323 traffic |
| 15:13.12 | TheCompWiz | who does? |
| 15:13.25 | zamba | TheCompWiz: the backstory: i'm trying to set up audio calls through asterisk using sip on our tandberg/cisco c40 videoconferencing solution |
| 15:13.41 | TheCompWiz | I'm waiting for the problem. |
| 15:13.43 | zamba | it's not registered to any MCU (or whatever it's called), so it's a standalone system |
| 15:13.46 | zamba | it's coming, it's coming :) |
| 15:13.56 | zamba | i'm able to register the system to asterisk over sip and i'm also able to receive calls.. |
| 15:14.09 | TheCompWiz | yawns. |
| 15:14.17 | zamba | but to dial out from the system i have to prefix all calls with 'sip:', since h323 is the default protocol for outgoing calls |
| 15:14.36 | [TK]D-Fender | zamba, No. If your app defaults to H323, that's its problem... |
| 15:14.47 | zamba | if i change the default protocol to sip, i have to prefix all video calls with "h323:" |
| 15:15.02 | WIMPy | Not sure if "problem" is the word. |
| 15:15.09 | zamba | me neither |
| 15:15.15 | zamba | it's just a default setting |
| 15:15.20 | zamba | since it can handle both |
| 15:15.35 | WIMPy | Why don't you enable H323 on Asterisk? |
| 15:15.39 | zamba | but the problem here is that i don't really want to teach my users how to prefix their calls, depending on where and how they dial |
| 15:15.46 | zamba | WIMPy: that's what i'm considering |
| 15:16.08 | *** join/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 15:16.09 | zamba | WIMPy: can asterisk just do the signaling and then leave the other streams alone? |
| 15:16.18 | [TK]D-Fender | .... |
| 15:16.30 | WIMPy | I have no idea, how good the H323 channels are, but that seems to be the obvious way to go. |
| 15:16.42 | zamba | [TK]D-Fender: have i offended you in some way? :) |
| 15:16.45 | [TK]D-Fender | zamba, If your device starts speaking a protocol you didn't configre * to accept calls under... what are you expecting? |
| 15:17.03 | WIMPy | H323 uses RTP just like SIP, otherwise see above. |
| 15:17.07 | zamba | [TK]D-Fender: huh? |
| 15:17.13 | zamba | [TK]D-Fender: you're not really paying attention, are you? |
| 15:17.15 | [TK]D-Fender | zamba, if you want * to handle H.3232 from it, then set them up for H.323 |
| 15:17.42 | zamba | i didn't really -want- to handle h323 from it, but i'm starting to believe that's the only solution here |
| 15:17.48 | zamba | i don't want h323 through asterisk |
| 15:17.51 | [TK]D-Fender | zamba, There is no such concept as "auto-negotiation", nor does * configure your device for you. |
| 15:18.14 | [TK]D-Fender | zamba, Or apparently you can tell you device to use the right protocol and you jsut don't like the way you have to do it. |
| 15:18.29 | [TK]D-Fender | zamba, None of those has anything to do with Asterisk, and everything to do with your other device |
| 15:18.36 | *** join/#asterisk italorossi (~Adium@189.124.200.92) |
| 15:18.58 | [TK]D-Fender | And SIP supports video. |
| 15:19.30 | zamba | yeah |
| 15:19.56 | zamba | the best solution would be to have some sort of logic in the c40 that looks at what you're trying to call (ip vs number) and selects the correct protocol based on that |
| 15:20.09 | zamba | all h323 connections are end-to-end |
| 15:25.20 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:30.57 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 15:35.05 | *** join/#asterisk _Corey_ (~chatzilla@pool-72-78-178-17.phlapa.fios.verizon.net) |
| 15:35.59 | *** join/#asterisk rbowles (~bb@75-147-63-201-NewEngland.hfc.comcastbusiness.net) |
| 15:37.05 | rbowles | rolled my 11.0-beta back to 10.8.0-RC1 and all is well with my REGISTERs again :-) |
| 15:37.50 | pabelanger | rbowles: try disabling ICE support in chan_sip for asterisk 11 |
| 15:37.56 | pabelanger | and open an issue on the tracker |
| 15:39.12 | rbowles | ack, and it's been a while, where do I open issues these days? |
| 15:41.18 | roe | how do the 10.x and 11.x version numbers correlate with the 1.x version numbers? |
| 15:42.17 | *** join/#asterisk Galen (~Galen@rrcs-24-43-17-243.west.biz.rr.com) |
| 15:44.10 | leifmadsen | rbowles: issues.asterisk.org |
| 15:44.23 | leifmadsen | oops, add /jira on the end |
| 15:44.29 | leifmadsen | looks like mantis isn't "down" yet :) |
| 15:45.22 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 15:51.18 | *** join/#asterisk keith4 (~keith@unaffiliated/keith4) |
| 15:53.20 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 15:54.04 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 15:55.56 | rbowles | thanks Leif! |
| 15:57.04 | adeel|work | is there a reliable method to adding the bridged call-id to a SIP header? i've realized the ${BRIDGEPVTCALLID} is only populated after the call has successfully been bridged (which is logical)...would an inherited variable work? |
| 15:58.22 | specing | ~sip |
| 15:58.22 | infobot | it has been said that sip is Session Initiation Protocol, http://www.cs.columbia.edu/sip/ (see RFC 3261) It's HIP to be SIP! |
| 16:02.09 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:12.06 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 16:16.38 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:17.37 | specing | Uhm |
| 16:17.44 | specing | Where is [UserServices] supposed to be found? |
| 16:17.52 | specing | "To test that your music is working correctly, add the following to the [UserServices] context in your dialplan:" |
| 16:18.03 | specing | grep can't find it in /etc/asterisk/ |
| 16:18.07 | Qwell | specing: where are you seeing this? |
| 16:19.20 | specing | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#additional_configuration_tasks-musiconhold-conf-file |
| 16:19.28 | specing | bottom, near Conclusion |
| 16:21.45 | Qwell | leifmadsen: ? |
| 16:22.10 | [TK]D-Fender | specing, Shove it anywhere in yoru dialplan your phones will reach |
| 16:22.11 | Qwell | specing: It's not a literal context. It just means put it where the rest of that stuff is |
| 16:22.41 | TheCompWiz | adds more figurative in your literals. |
| 16:23.50 | specing | The concept of a dialplan isn't explained either... |
| 16:23.58 | Qwell | it most certainly is |
| 16:24.07 | specing | Did I miss something? |
| 16:24.21 | Qwell | the section titled "The Dialplan" |
| 16:24.33 | *** join/#asterisk bipul (~vipul@unaffiliated/bipul/x-4918593) |
| 16:24.36 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-133.mobile.uci.edu) |
| 16:24.39 | bipul | join #sii-dev |
| 16:24.45 | [TK]D-Fender | You completely the dialplan.... |
| 16:24.48 | Qwell | and the chapter "Dialplan Basics" |
| 16:25.00 | Qwell | (and lots more...) |
| 16:25.56 | specing | Oh my bad, it is there |
| 16:29.53 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:29.53 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:34.31 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:41.51 | *** join/#asterisk irroot (~gregory@2002:c5ac:a5d6::1) |
| 16:42.36 | leifmadsen | Qwell: ya, was a typo that I've fixed for 4th edition |
| 16:42.39 | leifmadsen | should be LocalSets |
| 16:42.43 | leifmadsen | specing: ^^^ |
| 16:43.40 | [TK]D-Fender | There is another reference in there to that context. Might want to look at that as well |
| 16:47.04 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 16:50.06 | *** join/#asterisk sexton (~sexton@net-216-37-64-130.in-addr.worldspice.net) |
| 16:51.46 | anonymouz666 | irroot! |
| 16:52.11 | irroot | yo yo |
| 16:52.47 | Qwell | [TK]D-Fender: I noticed that one too - some log somewhere |
| 16:52.51 | Qwell | no [] though |
| 16:55.57 | *** join/#asterisk wonderworld (~wonderwor@46.115.76.127) |
| 16:58.35 | *** join/#asterisk Devon_ (~chatzilla@63.214.236.169) |
| 16:59.55 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-lxrfvpwqcjrlnsby) |
| 17:04.27 | specing | leifmadsen: Ah so Im supposed to put context=LocalSets in sip.conf and then [LocalSets] in extensions.conf? |
| 17:04.49 | leifmadsen | yes, UserContext is probably a left over from our dialplan examples from 1.4 |
| 17:04.55 | leifmadsen | s/UserContext/LocalSets |
| 17:05.03 | leifmadsen | and follow instructions per normal |
| 17:05.31 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:07.38 | specing | I assume I can name it [HoweverIWant] |
| 17:07.48 | leifmadsen | of course |
| 17:08.02 | shido6 | Hi all |
| 17:09.20 | *** join/#asterisk Olobola (~OloBola@c-76-102-4-89.hsd1.ca.comcast.net) |
| 17:10.54 | [TK]D-Fender | specing, You can make your dialplan however you want. |
| 17:13.27 | *** join/#asterisk jsjc (~Adium@213.37.155.15) |
| 17:16.25 | *** join/#asterisk mtbf (~ewilded@static.124.15.9.5.clients.your-server.de) |
| 17:17.02 | mtbf | Hi guys, I just noticed my duration value in CDR is really a half of the real recording duration, why is that? |
| 17:17.22 | mtbf | Recordings are of course mixed, not concatenated. |
| 17:18.07 | navaismo | hi, finally i get the voicemail email alert in html but using a script, natively only using <520 words, any hints? |
| 17:18.59 | Qwell | navaismo: is that a question? O.o |
| 17:19.21 | navaismo | only the part of the hints |
| 17:19.33 | navaismo | or should I use suggestion instead? |
| 17:19.42 | Qwell | Your sentence does not parse. |
| 17:19.50 | navaismo | ok let me see. |
| 17:21.40 | navaismo | finally I can send the voicemail alert to the mailbox email in html format using a script. Natively(only using the voicemail.conf) I can send the html body less than 520 words. Any suggestion to make a bigger body natively? |
| 17:21.43 | *** join/#asterisk aossama (~ao@41.232.123.201) |
| 17:21.55 | navaismo | that make sense now? |
| 17:22.16 | Qwell | patch app_voicemail. |
| 17:22.20 | Qwell | I doubt it's configurable. |
| 17:23.43 | leifmadsen | likely a character limit, not a word limit :) |
| 17:23.46 | navaismo | yes I patched for use html but i cant find where is the size? of emailbody |
| 17:24.01 | *** join/#asterisk wonderworld (~ww@dsdf-4db53c81.pool.mediaWays.net) |
| 17:24.05 | leifmadsen | guesses 2048 or 4192 |
| 17:24.28 | *** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-pkesodibvnbpsmmv) |
| 17:24.52 | Qwell | that too |
| 17:24.59 | navaismo | sorry yes characters not word |
| 17:32.15 | navaismo | thanks i will look for that |
| 17:33.55 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 17:40.02 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:40.37 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-orrkkclatmopmuxz) |
| 17:41.40 | mtbf | I just discovered this topic http://forum.asterisk2billing.org/viewtopic.php?f=13&t=8821 and in matter of fact my asterisk runs on a VM, but migrating it to physical version is not acceptable, I'd like to know the core reason of this disfunction. |
| 17:42.20 | *** join/#asterisk YoMomma (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
| 17:45.15 | [TK]D-Fender | mtbf, because the VM screwss with your system timer |
| 17:52.43 | *** join/#asterisk albertoandrade (~albertoan@200.175.4.108) |
| 17:52.55 | leifmadsen | don't know what it being on a VM would have anything to do with |
| 17:53.24 | leifmadsen | honestly seems like a fluke that he "fixed something" when he restored to a physical machine |
| 17:53.46 | leifmadsen | could always just pop up a development physical machine to test the theory |
| 17:54.17 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 18:06.22 | raden | how well do grandstream ata's work ? |
| 18:10.50 | rbowles | and not all VMs are created equal ... I wouldn't run production with any amount of load on esxi or other emulation style VM, OpenVZ probably works a lot better for scale |
| 18:10.50 | *** join/#asterisk Neptu (~Neptu@c-af90e255.113-1-64736c14.cust.bredbandsbolaget.se) |
| 18:14.22 | jpsharp | raden: I've used a lot of grandstream atas without a problem. |
| 18:15.29 | [TK]D-Fender | ~gs |
| 18:15.29 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 18:15.50 | SeRi | lol |
| 18:16.23 | specing | Can I nest templates in sip.conf? |
| 18:16.37 | specing | [SipPhone](!) |
| 18:16.50 | specing | [OfficeHardphone](SipPhone)? |
| 18:17.08 | specing | [actualyPhone](OfficeHardphone)? |
| 18:18.01 | drmessano | http://www.voip-info.org/wiki/view/Asterisk+config+template |
| 18:19.21 | *** join/#asterisk lorsungcu (~anonymous@pat-address.martinwilliams.com) |
| 18:20.25 | specing | Thanks |
| 18:20.33 | Katty | rages |
| 18:21.41 | drmessano | against the appliance? |
| 18:34.13 | *** join/#asterisk Goldwing (~Goldwing@84.245.47.128) |
| 18:35.55 | eduzimrs | hi, im using SRTP with polycom phone, and im getting a warning error on * when i dial: "chan_sip.c: Multiple audio streams are not supported" |
| 18:36.54 | *** join/#asterisk chasingSol (~ao@41.232.120.114) |
| 18:38.49 | eduzimrs | is that about codec issue? |
| 18:44.49 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 18:48.03 | *** join/#asterisk soulslayer (~soulslaye@212.73.146.222) |
| 18:52.07 | soulslayer | hey guys does anyone have a chunk of free time and hint me about how to debug one simple AGI case i have exten => _X.,1,AGI(call-processor.sh) under [incoming] in extensions.conf |
| 18:52.28 | soulslayer | and call-processor.sh in /var/lib/asterisk/agi-bin/ |
| 18:52.35 | soulslayer | chmoded to 777 |
| 18:52.58 | lorsungcu | is it +x'ed? |
| 18:53.03 | soulslayer | but the script is not executed when i enable agi debug in the cli there is alot of ... nothing |
| 18:53.16 | soulslayer | lorsungcu, -rwxrwxrwx 1 root root 42 Aug 29 21:24 call-processor.sh |
| 18:53.29 | lorsungcu | can you execute it outside of *? |
| 18:53.34 | soulslayer | yep |
| 18:53.37 | soulslayer | for sure |
| 18:53.55 | [TK]D-Fender | soulslayer, show us this "alot of nothing" |
| 18:53.55 | lorsungcu | by a lot of nothing |
| 18:53.58 | lorsungcu | you mean nothing |
| 18:53.59 | lorsungcu | ? |
| 18:54.01 | soulslayer | :D yep |
| 18:54.03 | soulslayer | exactly nothing |
| 18:54.08 | lorsungcu | pb the whole call |
| 18:54.11 | [TK]D-Fender | ^ |
| 18:54.14 | soulslayer | i recieve only the regular debug data provided buy -vvvvv |
| 18:54.20 | soulslayer | okay |
| 18:54.23 | soulslayer | give me a second |
| 18:54.43 | [TK]D-Fender | and AGI debug. "ls" file dumps, etc. |
| 18:55.41 | lorsungcu | also |
| 18:55.48 | lorsungcu | first few lines of the file |
| 18:56.11 | soulslayer | http://pastebin.com/SZSJvzi7 |
| 18:56.28 | soulslayer | thats what i get from the cli |
| 18:57.40 | soulslayer | in /var/log/asterisk there is almost nothing excepts some warning about retransmission timeout ... but nothing agi related |
| 18:57.56 | soulslayer | it beahves like there is no call to that script |
| 18:58.01 | lorsungcu | first few lines of the file? |
| 18:58.14 | navaismo | amm nat issues |
| 18:58.30 | navaismo | actually your cal is hangip up due to retransmission issues -i guess- |
| 18:59.00 | soulslayer | actualy that's the script |
| 18:59.01 | soulslayer | #!/bin/bash |
| 18:59.01 | soulslayer | echo "WORK DAMN IT !!!" >> tmp |
| 19:00.00 | soulslayer | navaismo, i can answer from the other side |
| 19:00.24 | soulslayer | but nevertheless the script must be executed on each incoming call no matter is it accepted or not ... and the other side rings so ... |
| 19:00.54 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.37) |
| 19:01.40 | soulslayer | do i need to include/enable agi somehow ... |
| 19:01.48 | zerohalo | soulslayer: do you have duplicate priorities? I see prio 1 is Answer, but in the above you said 1 is the AGI |
| 19:02.15 | *** join/#asterisk leedm777 (~Adium@nat/digium/x-loqqqpafkchmfahq) |
| 19:03.08 | soulslayer | oh |
| 19:04.10 | soulslayer | the second param after_X. is the priority ? |
| 19:04.21 | zerohalo | yes |
| 19:04.56 | zerohalo | Needs to start with 1 and continue incrementing, or use 'n' |
| 19:05.03 | soulslayer | oh jesus |
| 19:05.08 | zerohalo | ;) |
| 19:05.14 | soulslayer | i am a dumpass :D |
| 19:05.20 | soulslayer | just a sec to test that |
| 19:06.20 | reisi | is there a way to activate an extension without actually making a call (from anywhere)? |
| 19:07.16 | soulslayer | zerohalo, now i put priority = 2 to the answers |
| 19:07.23 | soulslayer | but i am not capable to make a call ? |
| 19:07.35 | zerohalo | can you pastebin your dialplan? |
| 19:08.04 | soulslayer | but of course ! :D |
| 19:08.22 | soulslayer | http://pastebin.com/dxD3ySDe |
| 19:08.44 | soulslayer | that's the whole content of extensions.conf .. its my first first frist asterisk try |
| 19:09.04 | zerohalo | ok... |
| 19:09.34 | zerohalo | and the peer that you're calling from has context=incoming, right? |
| 19:10.09 | zerohalo | from what I can tell, it's going to the default context instead |
| 19:10.13 | soulslayer | i must set this in sip.conf right ? |
| 19:10.23 | bitglue | say i have a queue, and one of the members is a handset that has call forwarding enabled. Is there some way I can require the party that was called (because of the forwarding on the handset) to accept the call before bridging them to the caller from the queu? |
| 19:10.29 | zerohalo | correct |
| 19:10.52 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 19:11.04 | soulslayer | nope they dont |
| 19:11.11 | zerohalo | Since I see an IP address as the peer, I expect you're using allowguest=yes |
| 19:11.22 | bitglue | i had a thing where I set a channel variable on outgoing calls, and if later a call was outgoing again, i'd see the variable was already set, and so prompted for confirmation. But, that also requires transfers to be confirmed, which isn't what i want. |
| 19:11.56 | soulslayer | zerohalo, http://pastebin.com/Uh1aUe9i thats sip.conf |
| 19:12.15 | zerohalo | ok |
| 19:12.20 | soulslayer | if i put context=incoming in general ... i will add the context to all the peers right ? |
| 19:12.24 | zerohalo | so in your [general] entry, the context=default |
| 19:12.38 | zerohalo | meaning any call not coming from a specified peer will go to that context |
| 19:12.55 | zerohalo | you would need to set it to context=incoming to get it to go to your context with the AGI call |
| 19:13.04 | [TK]D-Fender | soulslayer, never ever use a context named [default] . pick another name. Also you should not allow unauthed calls unless absolutely necessary |
| 19:13.20 | bitglue | what's wrong with the name default? |
| 19:13.25 | zerohalo | And what, [TK]D-Fender said |
| 19:13.36 | [TK]D-Fender | bitglue, there are certain modules that can dump calls there as a failover |
| 19:13.59 | [TK]D-Fender | if you point chan_DAHDI to an invalid context it will failover there for example |
| 19:14.05 | [TK]D-Fender | BAD idea |
| 19:14.09 | soulslayer | [TK]D-Fender, i setup that instance of asterisk for testing purposes i have to integrate it with third party system and because i am the only guy that "love to do things" ... they give it to me |
| 19:14.10 | bitglue | hm...well my [default] context rejects all calls, so i suppose that's not bad |
| 19:14.12 | [TK]D-Fender | Do not ever do that |
| 19:14.26 | [TK]D-Fender | bitglue, contexts don't reject calls. |
| 19:14.33 | soulslayer | zerohalo, is there a way to put multiple contexts to the peers |
| 19:14.42 | [TK]D-Fender | soulslayer, No, nor should you need to |
| 19:14.44 | soulslayer | i mean what is the best way to execute my AGI in production enviroment |
| 19:14.59 | bitglue | not even if the context just has exten => s,1,Busy()? |
| 19:15.07 | zerohalo | no. You would need to either goto another context, or include => the other context, depending on what you want to accomplish |
| 19:15.08 | soulslayer | .. or the painless will be more correct word |
| 19:15.09 | [TK]D-Fender | soulslayer, Point your peers to to context they need to be pointed to |
| 19:15.30 | soulslayer | [TK]D-Fender, im do not understand how to point them |
| 19:15.40 | [TK]D-Fender | there is little point in going 2 places when one is a massive pattern that is practically a "match-all" |
| 19:15.49 | [TK]D-Fender | soulslayer, context=blah <-------- |
| 19:15.51 | [TK]D-Fender | POINT |
| 19:15.57 | soulslayer | huh :) |
| 19:16.01 | navaismo | ~book |
| 19:16.01 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:17.54 | soulslayer | :D |
| 19:17.57 | soulslayer | <PROTECTED> |
| 19:18.37 | soulslayer | the script does not execute correctly and do not append nothing to the file but .... i think thats a proggress |
| 19:18.48 | [TK]D-Fender | soulslayer, Now that you doing something at least vaguely more correct, that isn't an actual AGI. You are not dumping the input devices, etc or anything else one is supposed to do. |
| 19:20.12 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 19:20.45 | soulslayer | [TK]D-Fender, i am sure that you are right but all i need to do is get ${CALLERID(num)} in the script |
| 19:20.50 | soulslayer | nothig more nothing less |
| 19:20.51 | eduzimrs | hi, im using SRTP with polycom phone, and im getting a warning error on * when i dial: "chan_sip.c: Multiple audio streams are not supported" |
| 19:20.57 | eduzimrs | is that about codec issue? |
| 19:20.59 | [TK]D-Fender | soulslayer, And do what with it? |
| 19:21.50 | soulslayer | send it via tcp socket to nodejs extension that parse it and redirect it to the core of a custom erp/bpm java based system |
| 19:22.11 | [TK]D-Fender | soulslayer, then you don't need AGI at all... |
| 19:22.16 | soulslayer | ... |
| 19:22.28 | [TK]D-Fender | System(/path/to/script.sh ${CALLERID(num)}) |
| 19:22.31 | [TK]D-Fender | ^ |
| 19:22.35 | soulslayer | oookay :D i think now a beam of revelation will shine upon me |
| 19:22.37 | lorsungcu | TK u r blowin my mynd. |
| 19:22.38 | [TK]D-Fender | yuo aren't doing anything interactive |
| 19:22.48 | soulslayer | actually yes |
| 19:23.08 | soulslayer | let me try this one |
| 19:23.12 | *** part/#asterisk leedm777 (~Adium@nat/digium/x-loqqqpafkchmfahq) |
| 19:24.31 | specing | Im having problems registering a device into asterisk |
| 19:25.01 | specing | - Is it ok if the server and client machines are the same? (same physical machine) |
| 19:25.28 | specing | - connection method is SIP and the config is as per the book |
| 19:25.35 | [TK]D-Fender | specing, You need to force your client to use a different port to bind to |
| 19:25.46 | specing | Ok |
| 19:26.16 | leifmadsen | specing: yes, but you need to make your client (or asterisk) listen on a different port |
| 19:26.20 | leifmadsen | otherwise you get port conflicts |
| 19:27.37 | specing | netstat shows ekiga(client) on 127.0.0.1 and 10.0.0.8 while asterisk is on 0.0.0.0 and ::: |
| 19:27.44 | specing | mess :) |
| 19:28.13 | leifmadsen | not really |
| 19:28.36 | leifmadsen | anyways, just change the source port on ekiga to 5061 or something |
| 19:28.49 | soulslayer | [TK]D-Fender, -- Executing [1002@default:1] System("SIP/192.168.0.102-00000002", "/home/asterisk/call-processor.sh User One") in new stack |
| 19:28.58 | soulslayer | :) |
| 19:29.04 | [TK]D-Fender | You MIGHT want to put that in quotes.... |
| 19:29.16 | soulslayer | yep yep :D |
| 19:29.26 | soulslayer | this will not be bash for sure |
| 19:29.27 | [TK]D-Fender | Since your "number" seemed to have turned into TEXT and included a space while they were at it |
| 19:29.51 | soulslayer | i just want to get around beacause i didnt understand nothing from asterisk |
| 19:30.13 | soulslayer | in our company i am the only guy that .... "love to do things..." whatever that means so thay said .. come on it will be fun |
| 19:30.19 | leifmadsen | has a double negative overload notification |
| 19:30.28 | [TK]D-Fender | soulslayer, Well your script has nothign to do with * besides taking that call's callerID and letting you do what you want externally with it |
| 19:30.31 | soulslayer | ... but it's not fun because it has a deadline ... and ... thay are dump |
| 19:30.37 | soulslayer | yep i know |
| 19:30.38 | soulslayer | :D |
| 19:31.22 | soulslayer | that's why i think i can continue by myself thank you about the help it was just in time :D if you was a girl i will fall in love probably (*JOKING*) |
| 19:31.52 | specing | kewl, Im stuck with editing the friggin gnome registry |
| 19:32.05 | soulslayer | gnome sux use fluxbox |
| 19:32.13 | specing | I use Awesome |
| 19:32.22 | specing | Ekiga is made with gnome in mind though |
| 19:32.23 | navaismo | awkward |
| 19:32.31 | specing | I'll just emerge linphone |
| 19:32.31 | soulslayer | its the best allpowerfull and allmighty registerd by the church of emacs because of the keybindings |
| 19:32.40 | specing | fuck ekiga |
| 19:32.49 | leifmadsen | use jitsi |
| 19:32.56 | SeRi | +1 |
| 19:33.57 | soulslayer | :D |
| 19:34.08 | specing | jitsi is not in the package manager |
| 19:34.37 | leifmadsen | so? |
| 19:34.45 | leifmadsen | grab it off their website |
| 19:34.49 | drmessano | I download and install Jitsi. True story. |
| 19:34.51 | specing | where is Chainsaw!!! |
| 19:34.59 | leifmadsen | drmessano: omg you must be some kind of pro! |
| 19:35.30 | SeRi | lol |
| 19:36.18 | specing | Im going to install linphone for the time being |
| 19:36.32 | specing | and bug Chainsaw to add jitsi to portage |
| 19:36.40 | drmessano | If by "pro" you mean "full of awesome", then yes. |
| 19:36.44 | SeRi | linphone is ok.. still buggy as hell. jitsi by far is better |
| 19:37.39 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-cgnesjyoyhccwatp) |
| 19:39.35 | specing | grabs a coffee while linphone compiles |
| 19:40.01 | leifmadsen | facepalms |
| 19:41.28 | rbowles | linphone and ekiga just aren't reliable enough for me |
| 19:41.49 | specing | I like mumble ;P |
| 19:42.07 | leifmadsen | aye |
| 19:42.10 | leifmadsen | jitsi ftw |
| 19:42.34 | SeRi | leifmadsen: lol |
| 19:42.41 | rbowles | I love[d] Blink but it uses so much CPU |
| 19:42.45 | bitglue | is there some way to control the context that will be used to interpret SIP redirections received when calling a queue member? |
| 19:42.54 | rbowles | I'm download jitsi now for Ubuntu 12.4 |
| 19:43.09 | bitglue | or, is there some way in the dialplan to know that such a thing is happening, and conditionally do some stuff? |
| 19:43.34 | rbowles | on Windows, my favorites are Bria/Xlite and 3CXphone :-) |
| 19:44.41 | [TK]D-Fender | bitglue, sorry, could you be a little more vague? |
| 19:45.29 | [TK]D-Fender | bitglue, I think I've parsed it.... if your call it 302'd you MIGHT get "BLINDTRANSFER filled in.. unsure.... otherwise... probably nothing |
| 19:46.08 | bitglue | well, the problem is handsets forwarding queue calls to external numbers that have robots that answer, like voicemail. |
| 19:47.05 | bitglue | i can ignore 302s with the I option to Queue, but then people just can't take queue calls away from their desk |
| 19:48.08 | bitglue | and i can set a __channel_variable and know that a call is being made because of another call, but then this applies to transfers and non-queue calls too. So for example, if a receptionist transfers a call, then the called party gets the confirmation prompt. Especially bad if the intent was to send it to voicemail. |
| 19:48.53 | bitglue | and I read some things about a FORWARD_CONTEXT context variable, but it doesn't work. |
| 19:49.40 | bitglue | and i can configure queues.conf to always prompt the agent, but then they must confirm calls for *all* calls, even when they are taking them at their handset, and i know it can't be going to voicemail. |
| 19:50.27 | bitglue | it does seem that REDIRECTING gets mangled somehow, but I can't find any clear documentation on how or when, so i'm reluctant to rely on that behavior. |
| 19:52.29 | *** join/#asterisk F2Knight (~Adium@70-89-188-5-or.portland.hfc.comcastbusiness.net) |
| 19:53.32 | specing | <PROTECTED> |
| 19:54.39 | specing | Woooho I get music on hold |
| 19:55.54 | specing | lol I don't know how to make linphone hang up ;_; |
| 19:56.50 | specing | the pause and stop buttons are all greyed out |
| 19:57.02 | specing | as it says the remote is ringing... |
| 19:57.10 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-133.mobile.uci.edu) |
| 19:58.06 | specing | Gtk-WARNING **: Attempting to store changes into `/home/specing/.local/share/recently-used.xbel', but failed: Failed to create file '/home/specing/.local/share/recently-used.xbel.1IKOJW': Permission denied |
| 19:58.11 | specing | Muahahaha |
| 19:58.19 | specing | fu gnome crap |
| 19:58.29 | specing | ;P |
| 20:01.24 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 20:08.35 | specing | Hmm |
| 20:09.00 | specing | If I can reload sip by doing "sip reload", How can I reload extensions? |
| 20:09.18 | mjordan | specing: dialplan reload |
| 20:11.30 | *** join/#asterisk ipengineer (~ipenginee@static-71-252-134-63.dllstx.fios.verizon.net) |
| 20:11.56 | ipengineer | I am running asterisk 1.4.29 how do I reset queue stats? I know in 1.6 there is a flush command for this |
| 20:13.14 | [TK]D-Fender | ipengineer, Step 1 : Upgrade Asterisk. Step 2 : flush |
| 20:13.35 | [TK]D-Fender | Or restarting * should probably do it |
| 20:13.45 | [TK]D-Fender | but I highly recommend the former |
| 20:14.38 | ipengineer | [TK]D-Fender: unfortunately upgrading is not an option for me. I wish it were for several other reasons.. Will reloading the queue module cause the stats to reset? |
| 20:16.29 | specing | ipengineer: What kind of an ancient system are ya running? |
| 20:16.40 | specing | even debian stablee ships with 1.6.1... |
| 20:17.37 | bitglue | why would asterisk decline a transfer but not log anything? verbose is at 3, and with sip debug on i can see asterisk send a 603 declined. I have no idea why. |
| 20:18.02 | ipengineer | specing: It is an old centos box with a lot of OLD custom work done to it.. At some point we need to move away from it. Sooner than later preferably. Today is just not that day though with all the custom work.. |
| 20:18.16 | bitglue | [TK]D-Fender: wouldn't i also get BLINDTRANFSER if the call were a blind transfer, though? |
| 20:19.08 | reisi | should Playback application work with absolute paths? |
| 20:19.24 | [TK]D-Fender | bitglue, I don't think you're getting a happy ending nere no matter what... |
| 20:19.29 | [TK]D-Fender | reisi, yes |
| 20:19.40 | bitglue | here's the problem, a call comes in, and rings the "reception" queue. The receptionist has forwarded her handset to her cell phone. Also, someone else logged in to the queue to answer the phones while she's out. Now the receptionist's cell phone voicemail answers, and the other people in the queue have no chance to answer |
| 20:20.03 | [TK]D-Fender | bitglue, Solution : stop using phone-based forwards. |
| 20:20.21 | [TK]D-Fender | bitglue, Kill idiocy at the source :p |
| 20:20.56 | SeRi | [TK]D-Fender: you on a roll...boom head shoot! :) |
| 20:21.00 | bitglue | here's the other problem: I set a channel variable just before Queue() that triggers some call screening logic in the dialplan. The receptionist answers a call. Now she transfers it to george. Geroge doesn't answer, it goes to his voicemail. So far so good. But, the channel variable is set to require the call to be confirmed. It's not confirmed, so the caller gets disconnected. |
| 20:21.22 | bitglue | [TK]D-Fender: and what, implement a site-wide deep packet inspection firewall that kills every 302 response everywhere? |
| 20:22.25 | [TK]D-Fender | bitglue, etll them to stop doing it on the phone. If they continue, disable the button on the phone. If they re-enable it, install electodes to the button and connect to a high-voltage power source |
| 20:23.14 | bitglue | how will i know they enabled it? |
| 20:23.33 | reisi | [TK]D-Fender: should the absolute filename still be without any suffices? |
| 20:23.39 | [TK]D-Fender | walk over to their phone and stare at it |
| 20:23.51 | [TK]D-Fender | reisi, Correct |
| 20:24.00 | bitglue | [TK]D-Fender: so i'm going to spend all my time now walking all over the building, checking people's phones? |
| 20:24.34 | [TK]D-Fender | just the once. Then you start tasing people. |
| 20:25.06 | bitglue | unfortunately, our nonleather weapon budget has been exhausted this year. |
| 20:25.15 | [TK]D-Fender | bitglue, http://www.youtube.com/watch?v=6bVa6jn4rpE |
| 20:25.17 | lorsungcu | you get leather weapons |
| 20:25.18 | bitglue | also the nonleathal budget |
| 20:25.19 | lorsungcu | wat |
| 20:25.24 | bitglue | heh :) |
| 20:25.31 | bitglue | yes, leather weapons -> separate budget |
| 20:25.35 | lorsungcu | nice. |
| 20:25.42 | [TK]D-Fender | bitglue, Fine, then just shoot the without aiming around vital organs :p |
| 20:25.44 | lorsungcu | you will need to send me whatever catalog you order those from. |
| 20:25.50 | bitglue | so perhaps i could bullwhip them. |
| 20:26.11 | [TK]D-Fender | You kniw it takes a lot time to kill someone with a thumb-tack. |
| 20:26.27 | [TK]D-Fender | I'm sure they'll comlpy before it becomes "lethal" |
| 20:26.30 | [TK]D-Fender | asjdhadghdf |
| 20:26.37 | [TK]D-Fender | typing skills dying fast today... |
| 20:26.39 | lorsungcu | typing IS hard. |
| 20:27.00 | bitglue | bleh |
| 20:27.19 | bitglue | this is all complicated by this strange refusal of my test server to transfer calls |
| 20:27.26 | [TK]D-Fender | bitglue, but yeah... fix your employees... this isn't worth the programming nightmare a little social engineering should fix |
| 20:27.47 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 20:28.19 | bitglue | [TK]D-Fender: "you can't fix stupid" |
| 20:28.43 | bitglue | so any idea what i'm doing to solicit this declined transfer? http://paste.debian.net/186406/ |
| 20:29.14 | [TK]D-Fender | bitglue, But you can provide an excruciating environment for motivate self-change ;) |
| 20:29.18 | *** part/#asterisk ipengineer (~ipenginee@static-71-252-134-63.dllstx.fios.verizon.net) |
| 20:29.34 | [TK]D-Fender | checkout time, BBIAB |
| 20:34.44 | *** join/#asterisk HolTech (~IceChat77@pool-74-107-106-223.bltmmd.fios.verizon.net) |
| 20:35.40 | bitglue | is there an easy way to make asterisk call me and play music on hold or something equally mundane? |
| 20:36.44 | kresp0 | yes bitglue, using asterisk call files |
| 20:37.23 | Qwell | bitglue: There are at least 5 different ways to go about that. |
| 20:38.02 | bitglue | ok, what's the easiest? I just need to generate some test calls on a test server to my handset, one at a time, interactively. |
| 20:38.49 | kresp0 | bitglue, you edit a file called whatever.call and put this: |
| 20:38.50 | kresp0 | Channel: SIP/trunkname/18882223333 |
| 20:38.50 | kresp0 | Application: Playback |
| 20:38.50 | kresp0 | Data: hello-world |
| 20:39.03 | kresp0 | then mv it to /var/spool/asterisk/outgoing/ and you get your call |
| 20:39.51 | HolTech | Anyone know of a free/open source softphone that works with asterisk and does not have ads or trial time period ? |
| 20:40.16 | bitglue | hah, no. |
| 20:40.34 | HolTech | was afraid of that |
| 20:40.41 | reisi | how does one GotoIf if a variable has not been set? |
| 20:40.42 | bitglue | actually, i don't know of a working softphone. |
| 20:40.44 | TheCompWiz | CSipSimple works pretty nicely. |
| 20:40.44 | Qwell | ~softphones |
| 20:40.53 | Qwell | huh |
| 20:40.54 | mjordan | Qwell: ha ha |
| 20:41.05 | bitglue | HolTech: actually, jitsi almost works. Buggy, but it works. |
| 20:41.08 | mjordan | jitsi/ekiga/zoiper (classic) |
| 20:41.08 | Qwell | ~soft phones |
| 20:41.33 | bitglue | HolTech: it was months ago when i tested it. Maybe it's improved. |
| 20:41.46 | kresp0 | HolTech, SFLPhone works ok too |
| 20:42.06 | TheCompWiz | I'd throw csipsimple into production... honestly. It is pretty impressive. |
| 20:42.12 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 20:42.30 | HolTech | csipsimple is for android only ? or is there a windows ver |
| 20:42.38 | HolTech | bitglue: looks like a winner, thanks |
| 20:43.46 | TheCompWiz | you never specified an OS. |
| 20:44.04 | kresp0 | I dont recomend jitsi. Today I receive a test call from a client using jitsi and I cannot hear the voice, only glibberish noise. Then tried with twinkle (an old one) and works ok |
| 20:44.16 | HolTech | TheCompWiz: I did not, my apologies |
| 20:44.37 | TheCompWiz | was being a smart-ass... but sure.... go w/ that. |
| 20:45.21 | bitglue | kresp0: well, wait a month, and then jitsi will work and twinkle won't. like i said, there really aren't any working softphones. |
| 20:45.42 | kresp0 | sad to hear that bitglue :/ |
| 20:45.59 | bitglue | everyone knows software doesn't actually work |
| 20:46.18 | ryan42 | hrm. Does anyone know how I can show exactaly what a trunk is connected to or why it is in use? |
| 20:46.23 | TheCompWiz | it can work... but nobody wants to spend the time/effort to make a proper software-phone. |
| 20:46.28 | bitglue | actually, isn't there a way to use /dev/dsp as an endpoint in asterisk? does that qualify as a softphone? |
| 20:46.29 | TheCompWiz | (time/effort/money) |
| 20:46.56 | TheCompWiz | it's not a trivial task. |
| 20:47.07 | ryan42 | I've got this issue where a line on a TDM410 keeps showing offhook but nobody is using the line and I can' tfigure out why it is offhook |
| 20:47.35 | kresp0 | HolTech, on windows you may also try blink |
| 20:48.33 | bitglue | <PROTECTED> |
| 20:48.42 | bitglue | this is a stupid problem. |
| 20:49.11 | kresp0 | what problem bitglue |
| 20:50.07 | bitglue | cell phone voicemails answering queue calls when people have their number forwarded |
| 20:52.22 | WIMPy | bitglue: chan_alsa, chan_oss, chan_console. |
| 20:52.24 | kresp0 | bitglue, chan_alsa |
| 20:53.25 | bitglue | HolTech: there you go. chan_alsa works with (is) asterisk. |
| 20:53.51 | WIMPy | wonders if Dial(,,i) will do anything on dahdi. |
| 20:54.37 | rbowles | blah, jitsi .deb package failed install on a java dependency ... gotta go chase that |
| 20:54.43 | bitglue | there isn't some way i can signal to cell providers that i'd not like voicemail to answer, is there? |
| 20:54.56 | jpsharp | Nope. |
| 20:55.27 | ryan42 | bitglue: I have had that issue as well |
| 20:55.36 | bitglue | ryan42: how did you solve it? |
| 20:55.36 | WIMPy | bitglue: There is. But I think you need SS7 for that. |
| 20:55.45 | ryan42 | on AT&T there is a way to make the cell phone take longer to go to Voicemail |
| 20:56.04 | ryan42 | but ... here's the stupid thing... when a cell has no service or is turned off it *always* goes to voicemail immediately :( |
| 20:56.23 | bitglue | exactly |
| 20:56.27 | WIMPy | You can turn that off. |
| 20:56.36 | WIMPy | Unless your provider want's to rip off your callers. |
| 20:56.46 | jpsharp | what provider doesn't? |
| 20:56.58 | ryan42 | I haven't come up wiht a solution. I am limited to the extent I can mess with the cells because they need to work for personal calls and voicemails for their owners |
| 20:57.12 | WIMPy | the legitimate ones. |
| 20:57.12 | bitglue | so i'm trying to find a way to have the queue agent confirm the call. Easy enough to implement, but I only want to do it for queue calls that have been forwarded, and I'm not finding a way to determine when that's the case. |
| 20:57.17 | ryan42 | but we don't want the business line afterhours forwarding to be answered by personal voicemail boxes |
| 20:57.55 | WIMPy | On some providers that won't allow you to cancel VM, you can instead set a CF to an invalid number as a workaround. |
| 20:58.24 | bitglue | i don't want to disable cell phone's VM. I just don't want queue calls to end up there. |
| 20:58.35 | specing | Calls are not coming through :( |
| 20:58.40 | jpsharp | How are they forwarding calls? By pressing the forward button on their IP phone? |
| 20:58.46 | bitglue | jpsharp: yes |
| 20:58.58 | WIMPy | How do you call the cell phones? |
| 20:59.23 | bitglue | 9-1-NXX-XXX-XXXX |
| 20:59.41 | WIMPy | What technology. |
| 20:59.43 | jpsharp | I remember seeing some way to detect the SIP 302 that most phones send when they're set to forward and get a call. |
| 20:59.58 | bitglue | jpsharp: that would be awesome. |
| 21:00.08 | bitglue | that's exactly what my phones do. I can't find any way to detect it. |
| 21:00.15 | WIMPy | Yes, that's the i I was wondering about. |
| 21:00.23 | bitglue | There's a FORWARD_CONTEXT channel variable, but it doesn't seem to actually do anything. |
| 21:01.03 | WIMPy | That a one you set. |
| 21:01.19 | WIMPy | Where do you want to detect diversion? |
| 21:02.19 | bitglue | for SIP queue agents that respond with 302 |
| 21:02.49 | bitglue | so i tried setting FORWARD_CONTEXT just before Queue, but the Refer-To number went into the context configured for the handset in sip.conf anyway. |
| 21:03.07 | specing | oops, they are coming through |
| 21:03.08 | specing | Yay |
| 21:03.22 | WIMPy | Didn't work for me, either. |
| 21:03.41 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:04.00 | WIMPy | Disable deflection and use forwarding instead. Or in Asterisk terms, don;t do it on the phone, use a feature code instead and let Asterisk do it. |
| 21:04.29 | bitglue | i've been avoiding that solution because i don't want to deal with configuring the phones. But I guess i'm out of options. |
| 21:05.03 | bitglue | i could tell myself the advantage would be implementing maybe alternate interfaces to configure forwarding. maybe a web page or something. as if i'll have time to implement that some day. |
| 21:05.18 | jpsharp | Can you see a channel variable that you'd set before a call lands in the queue in the outbound dial sequence? |
| 21:06.18 | bitglue | jpsharp: yeah, that's what i'm doing now. And it worked really well, until I made the "receptionist" a queue. Now, when the receptionist transfers a call, that variable is set. So if she transfers to someone, they don't answer, then their voicemail answers (as intended), the voicemail can't confirm the call, and the caller is disconnected. |
| 21:06.31 | WIMPy | web based forwarding isn't that bad. |
| 21:06.50 | bitglue | really it's been a problem all along, but until now, probably no one had a reason to transfer a queue call. |
| 21:07.21 | bitglue | WIMPy: well, i see the setting on my phone to make it call a number to enable forwarding. It's just ....ugh....so much work. |
| 21:07.29 | WIMPy | Doesn't look like an obvious thing to do to me. |
| 21:07.43 | bitglue | and, what if the phone UI is out of sync with asterisk? what if people's phones don't take the new settings? |
| 21:07.51 | WIMPy | No provisioning? |
| 21:07.54 | bitglue | WIMPy: what's not an obvious thing? |
| 21:08.11 | bitglue | WIMPy: i have provisioning, but i have to reboot all the phones, and then I don't have a good way to confirm that they got the setting. |
| 21:08.19 | WIMPy | deflection of calls from a queue. |
| 21:08.36 | bitglue | WIMPy: oh, right. For a normal, call center use of a queue, it's not. |
| 21:08.48 | WIMPy | I'n using custom devstates for CF. |
| 21:09.25 | bitglue | I replaced our static ring groups (Dial(SIP/bob&SIP/alice&...) with queues so that users can easily join and leave the group |
| 21:09.45 | bitglue | also i get more robust statistics with some reporting i already had in place for queues |
| 21:16.05 | jpsharp | I wonder if a patch to chan_sip to make it return congestion if it gets a 302 would be applicable. |
| 21:17.00 | bitglue | there are Dial() and Queue() options to ignore 302 |
| 21:17.03 | specing | Aww jitsi is java |
| 21:17.04 | specing | blegh |
| 21:17.47 | jpsharp | Oh. Hadn't needed those, so didn't notice they were there. |
| 21:18.32 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:18.48 | WIMPy | jpsharp: Dial(,,i) |
| 21:19.33 | jpsharp | Shows that even after 10+ years with Asterisk, you can still learn something :) |
| 21:26.32 | bitglue | i have another problem which would seem different, but has many things similar. Currently, I prepend "queuename>" to CALLERID so the called party can distinguish queue calls from personal calls, or answer the phone differently, depending on the queue. But, that doesn't get removed when the call is answered. So, if a call goes to reception, isn't answered, falls though to the auto-attendant, and is then sent to sales, the callerid ends up being "sales>recep |
| 21:27.01 | bitglue | i guess in that case i can remodify CALLERID right after the Queue() application. But what if the call is transferred? |
| 21:27.31 | bitglue | so receptionist answers, then does a blind transfer to an employee. The employee should see "JOHN DOE", not "reception>JOHN DOE". |
| 21:28.04 | bitglue | where would i remove the prefix from CALLERID? Or, any better way to indicate the queue to the called party before they answer? |
| 21:28.24 | jpsharp | Can you do a distinctive ring on the phone? |
| 21:29.00 | bitglue | already doing that, but I like the additional visual indication. Sometimes its loud, also good to have the name displayed on the phone if the caller is on hold, etc. |
| 21:29.19 | bitglue | i wonder if the distinctive rings are also sticky in this underirable way :( |
| 21:29.25 | TheCompWiz | at the risk of sounding like a jerk... you add it... you remove it. |
| 21:29.36 | jpsharp | Probably, since they're setting a channel variable. |
| 21:29.36 | bitglue | obviously. Where? |
| 21:29.48 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 21:30.43 | bitglue | maybe i make a context for queue agents, and use one of those Dial() options to run a macro before bridging the call, and switch all my agents from SIP/ to Local/ channels, using this fancy context. |
| 21:30.57 | bitglue | and in that macro, i strip the prefix from CALLERID |
| 21:31.19 | bitglue | ugh. |
| 21:34.22 | TheCompWiz | I actually have a custom context for my agents... |
| 21:34.41 | bitglue | and your agents are local channels? |
| 21:34.44 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
| 21:34.49 | bitglue | s/agents/members |
| 21:34.58 | TheCompWiz | nope. |
| 21:35.00 | bitglue | or are you using Agents? |
| 21:35.03 | TheCompWiz | neither |
| 21:35.07 | TheCompWiz | queue members. |
| 21:35.12 | TheCompWiz | with a bit o' custom-magic. |
| 21:35.41 | TheCompWiz | queue members are actually connected to other PBXes in our network. |
| 21:36.13 | TheCompWiz | but I have a php script running that monitors device-states & sets a custom-state on the "queue" box. |
| 21:37.19 | TheCompWiz | when an agent "logs in" ... it just cheats & sets a database entry to their caller-id for call-back. |
| 21:37.30 | TheCompWiz | then does a queueadmember... |
| 21:37.33 | TheCompWiz | *add |
| 21:37.45 | TheCompWiz | with the custom-device state as one of the parameters |
| 21:38.23 | bitglue | makes sense |
| 21:38.40 | bitglue | what do you do if they give you a call-back number that has voicemail on it? |
| 21:39.00 | TheCompWiz | and because they're not local... I basically add AGENTID@agents-context to the queue... and in the "agents-context" it does the translation back to their extension. |
| 21:39.16 | TheCompWiz | the queue timeout isn't long-enough to trigger the VM |
| 21:39.35 | bitglue | i guess with a pbx you control you can rely on that |
| 21:39.36 | TheCompWiz | s/queue timeout/agent timeout/ |
| 21:39.59 | TheCompWiz | agent timeouts shouldn't be that high.... |
| 21:40.14 | bitglue | if your agent is a cell phone with no service, the VM timeout is 0. |
| 21:40.29 | TheCompWiz | my "agents" are not on cells :D |
| 21:40.40 | bitglue | not a luxury i have |
| 21:40.49 | bitglue | anyway, tomorrow i'll be fresh and ready to tackle this mess |
| 21:41.10 | TheCompWiz | well.... you're F*d then.... because I can't think of any way to get around the "voicemail" issue. |
| 21:41.23 | bitglue | you require the agent to accept the call with a dtmf response |
| 21:41.39 | TheCompWiz | you could do that with a custom-context still. |
| 21:42.08 | bitglue | not if Dial(SIP/alice) can return 302 and forward to a cell phone with voicemail |
| 21:42.12 | TheCompWiz | on the dial-cmd... just put a read in there looking for something. |
| 21:42.37 | bitglue | i can't know if Dial(SIP/alice) is calling Alice's handset, or if it's being forwarded to her cell phone. |
| 21:43.33 | bitglue | unless i set a channel variable right before Dial so i can know if i'm looped back into the dialplan a second time; then I can't know if I'm being looped back because alice redicted me, or if Alice answered and is tranfering the call. |
| 21:44.17 | *** join/#asterisk bintut (~bintut@cm38.sigma15.maxonline.com.sg) |
| 21:45.15 | TheCompWiz | could Hangup(21) or some such... I think that'l throw 'em back into the queue... |
| 21:45.30 | bitglue | Hangup when? |
| 21:45.53 | TheCompWiz | after calling the "cell-phone" and they don't press your magic digit. |
| 21:45.58 | bitglue | that's not the problem |
| 21:46.06 | bitglue | that part already works |
| 21:46.11 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 21:46.15 | TheCompWiz | so what doesn't? |
| 21:46.36 | bitglue | the problem is requiring the response only when it's a queue call and it was forwarded to an external extension, and not when it was a queue call that was answered then transferred, and not when it was a queue call that was not forwarded. |
| 21:48.35 | TheCompWiz | it's do-able... but a helluva lot more work. |
| 21:49.19 | TheCompWiz | you could put a clause in your devices' dial cmd that looks to see if it came from a queue... and if it did... just reject the call. |
| 21:49.29 | bitglue | and how do i know if it came from a queue? |
| 21:49.43 | TheCompWiz | set the var before dialing the "agent" |
| 21:49.49 | TheCompWiz | i.e. like in my custom-context |
| 21:50.00 | bitglue | how do i prevent it from being set after the agent answers? |
| 21:50.12 | bitglue | ie, the "answered by an agent, then transfered" case |
| 21:50.18 | TheCompWiz | it should always be set... why wouldn't it? |
| 21:50.39 | TheCompWiz | transfer would be in a separate context as well.... iirc. |
| 21:50.46 | bitglue | because the agent answered. It's not coming from the queue. |
| 21:53.01 | TheCompWiz | could set a transfer context... maybe. |
| 21:53.57 | TheCompWiz | unset it in the transfer-context if you really need to... |
| 21:54.10 | *** part/#asterisk rhce7320 (~rhce7320@59.167.200.141) |
| 21:55.39 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-qgnhpduvllxftdbg) |
| 21:57.09 | TheCompWiz | or set the M(x) parameter on the dial cmd for the agent. |
| 22:00.45 | TheCompWiz | shrugs... it's quittin' time. |
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| 22:04.51 | rbowles | okay, jitsi on Ubuntu 12.04 is pretty cool ... |
| 22:06.05 | *** join/#asterisk danfromuk (~IceChat77@2.27.38.64) |
| 22:06.31 | *** join/#asterisk autofsckk (~que@unaffiliated/autofsckk) |
| 22:07.30 | autofsckk | hi all |
| 22:07.34 | lorsungcu | hi |
| 22:07.52 | danfromuk | Hi all, I need to give advice to a client regarding a door phone. Should they be looking at sip-based doorphones or should they be getting a normal doorphone with a voip adapter? Does it make a difference? What are people doing nowadays? |
| 22:08.29 | autofsckk | p3nguin: have you installed openvox cards on any * server? do you recommend me to use arch with a local server with an openvox card? |
| 22:08.34 | lorsungcu | danfromuk: http://algosolutions.com |
| 22:09.09 | lorsungcu | i'd just get a SIP one |
| 22:10.04 | danfromuk | Interesting. I was worried about exposing the network cabling to the outside of the building but that solution looks ok |
| 22:10.19 | lorsungcu | goes much further, too |
| 22:13.16 | danfromuk | do you know any stockists in the UK? |
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| 23:03.03 | xSmurf | so I have playbacks that get cut in the middle and I yet I don't see any errors in the logs |
| 23:03.11 | xSmurf | (running in xen) |
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| 23:06.43 | Galen | Hey all. I seem to have a strange intermittent issue with outgoing calls not connecting properly. I eventually get a hangup cause of 127 but when the line initially connects I get the "We are sorry the number you have dialed has been disconnected or is no longer in service" message. The strange thing is that if I call the same number from my old POTS lines or a cell phone the call will go |
| 23:06.43 | Galen | through. Also if I try to make the call 5 or 6 times with my main system it will suddenly go through. Can anyone suggest where to start troubleshooting this or has anyone see something like this that would be able to point me in the right direction? |
| 23:11.42 | WIMPy | How are you calling? |
| 23:13.55 | Galen | WIMPy: Is exten => _91NXXNXXXXXX,n,Dial(DAHDI/g1/${EXTEN:1},,T) what you are looking for? |
| 23:14.18 | WIMPy | And what kind of dahdi interface is that? |
| 23:15.38 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 23:19.03 | Galen | WIMPy:We have a PRI coming into our Xorcom unit |
| 23:19.37 | WIMPy | Do you see any differences? |
| 23:20.29 | Galen | The only difference I see is this message "PROGRESS with cause code 1 received" |
| 23:20.35 | WIMPy | Are the failed calls really going out at all? |
| 23:21.08 | Galen | The CLI shows they are |
| 23:21.13 | WIMPy | Have you looked at a pri debug? |
| 23:21.54 | WIMPy | progress with cause 1 sounds a littel strange anyway. |
| 23:23.07 | *** join/#asterisk darkdrgn2k (~darkdrgn@CPE001310998982-CM00137189ced4.cpe.net.cable.rogers.com) |
| 23:23.17 | Galen | I haven't looked at a pri debug yet, how would I go about doing that? |
| 23:23.36 | darkdrgn2k | hey all, can any one recommend a IP intercom device for a front door of a house? |
| 23:24.03 | WIMPy | pri set debug 2 span 1 |
| 23:24.09 | *** part/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 23:24.34 | WIMPy | Or if you want to pastebin, pri set debug hex span 1 is enough. |
| 23:26.10 | Galen | The first command worked but the second one doesn't regognize the hex |
| 23:26.35 | WIMPy | How old is your Asterisk? |
| 23:26.59 | Galen | Installed it about a month ago running 1.8 |
| 23:28.06 | WIMPy | Strange |
| 23:28.38 | Galen | for pri set debug tab shows 0 1 2 off on file as the options |
| 23:29.13 | WIMPy | 1.8.what? |
| 23:29.47 | WIMPy | I've got it on a version from February. |
| 23:29.56 | Galen | 1.8.14.0 |
| 23:32.06 | WIMPy | Maybe it wasn't added to 1.8. |
| 23:33.11 | Galen | I can pastebin a one of the passes the pri set debug 2 span 1 returned if that would be helpful |
| 23:33.47 | WIMPy | A failed ad a successfull attempt to compare would be usefull. |
| 23:33.56 | WIMPy | s/ad/and/ |
| 23:34.22 | Galen | Okay, let me get that together |
| 23:42.12 | Galen | Here is the pastebin of the calls. 2 failed and 1 successful. http://pastebin.com/DVEMwQeH |
| 23:45.11 | WIMPy | But no pri debug. |
| 23:46.06 | WIMPy | But if you get an announcement it looks like your provider has some sort of issue not related to your connection. |
| 23:46.28 | Galen | Yeah, I am getting an announcement. |
| 23:46.50 | Galen | Did you want me to send a pri debug or just skip that and get in contact with my provider? |
| 23:47.25 | WIMPy | I think you can blame them :-) |
| 23:47.37 | WIMPy | But the signalling is strange anyway. |
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| 23:48.20 | WIMPy | You usually get that sort of announcement with a disconect, not a progress. |
| 23:49.13 | Galen | Yeah, that is what is seemed like to me. Thank you for the help. I will contact my provider and go from there. |
| 23:50.35 | WIMPy | Maybe they sent you a SIP gateway instead of a NT? |
| 23:54.46 | Galen | I will check on that |