| 00:00.28 | JT | well yes as long as the company accepts your account, you may have to put down a norwegian address when signing up |
| 00:00.58 | The_Ball | yes, that's probable, but that's not a problem |
| 00:01.18 | *** part/#asterisk gJon (n=gjon@ein.cr.aptaculo.us) |
| 00:01.31 | JT | only other major concern i can think of is the latency and in general the quality of the Internet connection to norway |
| 00:02.09 | The_Ball | what network does the voip providers use between countries ? |
| 00:02.26 | JT | err, it's voip, so, the Internet |
| 00:02.27 | hmmhesays | arpanet |
| 00:03.20 | The_Ball | so it would be the same problem using an australian voip provider or a norwegian provider when making calls to norway |
| 00:03.36 | JT | maybe, maybe not |
| 00:03.44 | JT | likely a voip provider has better connectvity |
| 00:03.46 | Un1x | hey is there a command to log all numbers dailied into a TEXT file or sometihng similar? |
| 00:03.58 | JT | they may even have dedicated bandwidth, but unlikely |
| 00:04.11 | JT | Un1x: you mean like CDRs? |
| 00:04.14 | The_Ball | i see, btw, does engin support iax? |
| 00:04.18 | JT | no |
| 00:04.32 | Un1x | CDR? no not record the call i ment just record the number dailed... |
| 00:04.37 | JT | they use cisco call manager at the border |
| 00:04.48 | Un1x | heh |
| 00:04.48 | JT | yes that's what a bloody CDR is |
| 00:04.56 | JT | call detail record |
| 00:05.02 | Un1x | I See |
| 00:05.23 | JT | please look it up next time before correcting me, i was on the right track :P |
| 00:05.43 | JT | most asterisk installs write CSV CDRs by default |
| 00:05.47 | The_Ball | JT, engin has a good norwegian rate, 4c/min |
| 00:05.53 | JT | The_Ball: cool |
| 00:06.01 | JT | is the rate the same to landlines and mobiles |
| 00:06.12 | The_Ball | no, 39c/min to mobile |
| 00:06.17 | JT | ah |
| 00:06.28 | Un1x | Hey JT, where are th soundfiles for asterisk |
| 00:06.33 | JT | about the same price as telstra charge for a call from a landline to a non-telstra mobile |
| 00:06.40 | Un1x | exten => s,3,Playback(vm-isunavail) |
| 00:06.48 | Un1x | like this for example where would that soundfile be? |
| 00:06.57 | JT | <PROTECTED> |
| 00:06.57 | Corydon-w | Un1x: /var/lib/asterisk/sounds/ |
| 00:07.00 | Un1x | thanks :) |
| 00:07.03 | JT | all this info is in the book |
| 00:07.06 | JT | you should read it |
| 00:07.10 | JT | ~thebook |
| 00:07.17 | jbot | well, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 00:07.17 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 00:07.17 | The_Ball | Un1x, try the command locate if installed on your system: "locate isunavail" |
| 00:07.22 | EmleyMoor | Are there any docs on using a British caller display with *? |
| 00:07.45 | JT | i've seen stuff on voip info about it, EmleyMoor |
| 00:08.29 | Un1x | JT, i am going to read it, last and simple question so if i have .gsm files the recordings basicly i just shove whatever i want into /var/lib/asterisk/sounds/ and then i can use it correct like any othe recording? |
| 00:08.51 | JT | yes as long as it's a format asterisk recognises |
| 00:09.04 | EmleyMoor | Also, is there a way I can get a Strowger-like dialtone? |
| 00:09.29 | Un1x | Yea, it is i donloaded asterisk sounds from asterisk.org :) |
| 00:09.33 | JT | i'm not sure what a strowger-like dialtone is, but there is a us-old tonezone or similar |
| 00:09.55 | JT | Un1x: the best quality is slin, ulaw or alaw files |
| 00:10.03 | JT | gsm isn't very good quality |
| 00:10.10 | EmleyMoor | JT: A purring sound... |
| 00:10.20 | JT | no idea |
| 00:10.23 | JT | try us-old :P |
| 00:10.40 | *** join/#asterisk Aximas (n=Aximas@gw-100.selfnet.de) |
| 00:10.50 | JT | you could always make your own tonezone |
| 00:10.58 | JT | call it "the-undertaker" |
| 00:11.00 | hmmhesays | you can make whatever tones you want |
| 00:11.08 | *** join/#asterisk ToyMan (n=stuq@static-74-41-52-30.dsl1.mdl.ny.frontiernet.net) |
| 00:11.31 | JT | yeah, not always easy to get it right though |
| 00:12.32 | hmmhesays | if you know what frequencies you need it is |
| 00:13.39 | robin__sz | EmleyMoor, I failed to getUK CID to work with a X100P card or a clone, using ISDN2e and a HFC/Cologne card and mISDN, it worked right out of the box ... but then again thats ISDN, not the weird ass UK CID schema |
| 00:13.57 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 00:14.07 | Un1x | agent-loginok.gsm |
| 00:14.13 | Un1x | this file was in /var/lib/asterisk/sounds |
| 00:14.17 | Un1x | im trying to use it but it says |
| 00:14.23 | EmleyMoor | I have the caller ID coming in from BT on a genuine TDM400P - just wondering how I can send it out to displays |
| 00:14.23 | Un1x | Dec 8 07:17:02 WARNING[21272]: file.c:512 ast_openstream_full: File agent-loginok.gsm does not exist in any format |
| 00:14.23 | Un1x | Dec 8 07:17:02 WARNING[21272]: file.c:824 ast_streamfile: Unable to open agent-loginok.gsm (format g729): No such file or directory |
| 00:14.23 | Un1x | Dec 8 07:17:02 WARNING[21272]: app_playback.c:133 playback_exec: ast_streamfile failed on SIP/perlninja-b6c03450 for agent-loginok.gsm |
| 00:15.00 | robin__sz | EmleyMoor, for genuine Strowger tones, you need a tonewheel set :) |
| 00:15.26 | *** join/#asterisk stuq (n=stuq@static-74-41-53-4.dsl1.mdl.ny.frontiernet.net) |
| 00:15.58 | robin__sz | basically, some large steel discs on a motor ... with some magnets :) |
| 00:15.58 | Un1x | nevermind i entered extension hehe :P |
| 00:16.13 | EmleyMoor | I remember when they modernised the exchange where my dad lives - a new TXE4RD for the 5-digit numbers, and a change of dialtone on the 4-digit, which remained on the Strowger |
| 00:17.01 | *** part/#asterisk hoobastoob1 (n=ckwall@63.149.122.93) |
| 00:17.13 | EmleyMoor | One of our phones back then was a Yeoman |
| 00:17.20 | EmleyMoor | So glad I have one now |
| 00:19.32 | robin__sz | I still have a problem with one of my real Snom 190s, runing the same firmware as another one that works fine |
| 00:19.48 | EmleyMoor | Sounds like a hardware fault |
| 00:19.54 | robin__sz | when I go to do a transfer, it clears down the incoming call |
| 00:20.14 | robin__sz | ive got a SIP debug soemwhere on pastebin |
| 00:20.18 | JT | hmmhesays: in theory it's easy |
| 00:20.29 | JT | the au tonezone's busy tones is mostly right |
| 00:20.45 | JT | but it's still wrong |
| 00:21.01 | Un1x | hey JT i beleive youre familiar with DISA you know when it asks you to enter pass then it if you enter wrong pass it says Password InCorrect file wich is auth-incorrect |
| 00:21.01 | EmleyMoor | How can I implement a means of transfer that will work on a phone (a) with and (b) without TBR? |
| 00:21.10 | Un1x | can i change it instead o playing auth-incorrect to something else? |
| 00:21.18 | JT | between the noises there's some weird effect that sounds like aliasing |
| 00:21.41 | robin__sz | Un1x, yes |
| 00:21.58 | Un1x | robin__sz wich file do i change it in |
| 00:22.03 | Un1x | so it plays a differnt soundfile |
| 00:23.17 | robin__sz | app_authenticate.c |
| 00:23.40 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 00:24.03 | robin__sz | or. for a quick and dirty solution, move auth-incorrect.gsm out of the way and symlink it to something else |
| 00:24.10 | Un1x | oh crap that means recompiling and such nevermind :P |
| 00:24.16 | Un1x | heh yea i could rename the file :P |
| 00:24.17 | Un1x | hehe |
| 00:24.20 | Un1x | good idea thanks |
| 00:24.25 | Dovid | anyone know whent he next astricon will be ? |
| 00:24.28 | EmleyMoor | My partner is usually but not universally known by his middle name |
| 00:24.49 | *** join/#asterisk wundaboy (n=asdf@c-24-21-109-26.hsd1.or.comcast.net) |
| 00:24.52 | *** join/#asterisk axisys (n=axisys@69.143.190.152) |
| 00:25.09 | *** join/#asterisk Newbie___ (n=Newbie__@219.95.205.172) |
| 00:25.19 | *** part/#asterisk axisys (n=axisys@69.143.190.152) |
| 00:25.21 | EmleyMoor | When I realised that his first name was needed, I had to re-record |
| 00:26.00 | robin__sz | IVR? |
| 00:27.08 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:27.10 | blitzrage | anyone happen to have an example config set for a 7970? |
| 00:27.31 | blitzrage | the phone configs, not the sip.conf :) |
| 00:27.38 | EmleyMoor | Interactive Virtual Receptionist |
| 00:27.51 | EmleyMoor | Give it a try if you like |
| 00:28.16 | robin__sz | mmm, no ta :) |
| 00:29.02 | robin__sz | sounds fun, but we;re running a business ... |
| 00:29.15 | robin__sz | gotta try and stay focussed :) |
| 00:29.35 | robin__sz | blitzrage, tried the info on voip-info.org? |
| 00:29.43 | robin__sz | http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP |
| 00:29.54 | robin__sz | quite a bit of phone config on there |
| 00:30.00 | bkw_ | its Interafcive voice Response |
| 00:30.19 | EmleyMoor | Well, whatever it is... |
| 00:31.41 | JT | yeah that's what i thought |
| 00:32.15 | EmleyMoor | Now I have a working FXO port it's great |
| 00:32.41 | JT | robin__sz: what hardware do you use with mISDN? |
| 00:32.56 | robin__sz | some random hfc/cologne card |
| 00:33.04 | robin__sz | Asus I think |
| 00:33.10 | JT | ah a small thing |
| 00:33.13 | JT | like 1 or 2 ports/ |
| 00:33.31 | robin__sz | 0000:02:09.0 Network controller: Asustek Computer, Inc. ISDNLink P-IN100-ST-D (rev 02) |
| 00:33.34 | robin__sz | 1 BRI |
| 00:33.38 | JT | ah |
| 00:33.46 | JT | right |
| 00:33.56 | JT | i'm curious how well it works with an 8 port card :) |
| 00:34.07 | robin__sz | dunno |
| 00:34.28 | robin__sz | Ive got 8 port cards from Eicon .. but I use CAPI with those |
| 00:34.43 | JT | chan_capi |
| 00:34.50 | JT | the bristuff version, or pre-bristuff? |
| 00:34.51 | robin__sz | yeah |
| 00:34.58 | robin__sz | pre I think |
| 00:35.07 | robin__sz | long time since I looked at those boxes |
| 00:35.10 | JT | bristuff has chan_capi |
| 00:35.16 | JT | but hmm |
| 00:35.28 | robin__sz | istr having to add it specially |
| 00:35.42 | robin__sz | from a rep run by ... mmm |
| 00:36.45 | robin__sz | melware.org |
| 00:37.01 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 00:37.20 | *** join/#asterisk stuq_ (n=stuq@74-32-0-95.dsl1.mdl.ny.frontiernet.net) |
| 00:37.52 | robin__sz | so pre-bri I guess |
| 00:38.22 | JT | is the eicon card cologne hfc chipset? |
| 00:39.09 | robin__sz | nah |
| 00:39.26 | robin__sz | not as far I I remember |
| 00:39.33 | robin__sz | custom glue chips |
| 00:39.56 | danp | mog: would it be ok if i had the pastie IRC bot join this channel? you can read about its usage here: http://pastie.caboo.se/usage/ |
| 00:40.04 | JT | oh, what does it use for drivers/ |
| 00:40.06 | JT | ? |
| 00:41.18 | robin__sz | mmm ... something from eicon .. these a CAPI interface to it once its up and running, and a open source utility to control it |
| 00:41.26 | mog | why not just use pastebin danp |
| 00:41.27 | JT | oh ok |
| 00:42.04 | robin__sz | its an "active" card ... makes faxing low trouble |
| 00:42.34 | JT | hfc cologne i think is active |
| 00:42.48 | danp | pastie has better IRC integration...not as much work to get and give out the paste URL |
| 00:43.05 | robin__sz | nah, passive AFAIK ... it can do NT or TE mode though |
| 00:43.09 | *** join/#asterisk RoyKa (n=roy@217-175-222.100710.adsl.tele2.no) |
| 00:43.44 | robin__sz | but, I could be worng |
| 00:44.45 | JT | it's listed as active acording to http://www.asteriskguru.com/tutorials/chan_misdn.html |
| 00:45.43 | Un1x | <PROTECTED> |
| 00:45.43 | Un1x | Dec 8 07:48:48 WARNING[17902]: chan_zap.c:10874 setup_zap: Ignoring signalling |
| 00:45.43 | Un1x | <PROTECTED> |
| 00:45.44 | Un1x | Dec 8 07:48:48 WARNING[17902]: chan_zap.c:10874 setup_zap: Ignoring signalling |
| 00:45.44 | Un1x | <PROTECTED> |
| 00:45.44 | Un1x | Dec 8 07:48:48 WARNING[17902]: chan_zap.c:10874 setup_zap: Ignoring signalling |
| 00:45.48 | Un1x | i wonder why get those :S |
| 00:46.21 | Un1x | let me try something |
| 00:46.45 | robin__sz | JT, oh, OK, thats good ... especially as you can buy Cologne cards for $trivial if you shop around |
| 00:47.06 | JT | robin__sz: i have not seen any cologne cards for $trivial with 4 or 8 ports |
| 00:47.13 | robin__sz | true |
| 00:47.23 | robin__sz | but for single BRI they are plentiful |
| 00:48.12 | JT | which is not very useful for anywhere but a house or the smallest of businesses |
| 00:48.16 | Un1x | Somone please tell me why i get those Warnings? |
| 00:48.40 | JT | Un1x: you haven't even told us what you were doing |
| 00:48.54 | Un1x | nothing when just starting asterisk like 'asterisk -vvvvc' |
| 00:49.00 | Un1x | i just see those when asterisk is startiung uo |
| 00:49.22 | JT | probably something wrong in either /etc/zaptel.conf or /etc/asterisk/zapata.conf |
| 00:49.54 | Un1x | well i moved around zaptel.conf nothing fixed it i'll try zapata |
| 00:50.07 | JT | "moved around"? |
| 00:51.19 | JT | do you have any idea what you are doing? |
| 00:54.38 | Un1x | err jt moved around meaning the signalling |
| 00:54.39 | Un1x | let me show u |
| 00:54.49 | *** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net) |
| 00:54.58 | JT | you better use pastebin |
| 00:55.01 | Un1x | fxoks=1-2 |
| 00:55.01 | Un1x | fxsks=3-4 |
| 00:55.01 | Un1x | loadzone = us |
| 00:55.01 | Un1x | defaultzone = us |
| 00:55.04 | JT | NO |
| 00:55.05 | Un1x | zaptel.conf.... |
| 00:55.07 | JT | do not paste here |
| 00:55.10 | Un1x | i think its wrong |
| 00:55.13 | JT | ~pb |
| 00:55.16 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 00:55.21 | Un1x | its not working well .ca isn't |
| 00:55.56 | JT | that doesn't make it ok to spam 300 people |
| 00:56.17 | aptura | Anyone have specs on the ip501 and how many watts it consumes? |
| 00:56.19 | JT | pastebin.ca works fine for me |
| 00:56.40 | aptura | Also, is there such thing as a POE blade based switch that would fit in a pci slot? |
| 00:57.03 | aptura | Or just a nic that supplies poe. |
| 00:57.29 | JT | iirc the limit for PoE is around 15W |
| 00:57.47 | aptura | iirc? |
| 00:58.02 | JT | ~iirc |
| 00:58.04 | jbot | from memory, iirc is "if I recall correctly" |
| 00:59.09 | aptura | Trying to tally the total watts a low power mobo with flash and possibly a switch that while it may not exist can snap into a pci slot and have everyone enclosed in one case. Called the only company that sells an tough 37 bls pbx like case. |
| 00:59.22 | aptura | and want to put a ups inside it. |
| 00:59.36 | JT | hmm |
| 00:59.37 | JT | why ups |
| 00:59.49 | JT | just use a 12vDC power supply |
| 00:59.52 | JT | makes more sense |
| 01:00.10 | aptura | not when power is down. |
| 01:00.39 | JT | yes it does |
| 01:00.49 | JT | 12VDC sead lead acid battery |
| 01:00.51 | aptura | But what ever is best. Going to use a Antex ps in the case. One of the most reliable units. |
| 01:00.53 | JT | sealed |
| 01:01.15 | JT | if you're going the lowpower compact route |
| 01:01.20 | aptura | Im not asking for sugestions. |
| 01:01.31 | JT | 12VDC power supply + battery makes the most sense |
| 01:01.36 | *** join/#asterisk xnon (i=xnon@200.8.5.123) |
| 01:01.37 | aptura | Just watt req for the phone. |
| 01:01.51 | JT | no need to get all anal |
| 01:01.54 | JT | just trying to help |
| 01:01.58 | *** join/#asterisk jusse (n=jusse@190.41.135.115) |
| 01:02.09 | JT | why mention something if you don't want it commented upon |
| 01:02.58 | aptura | I know |
| 01:03.14 | JT | my suggestion would allow you to put it into a smaller case, i assume you want it to appear to be a normal pbx from the outside |
| 01:04.11 | aptura | Well there really is no atx case that looks like a pbx. The one I am looking at is mil spec. pricy but its can accomidate 115 volt conduit. |
| 01:04.28 | JT | then why use an ATX case? |
| 01:04.37 | JT | mount it in something else |
| 01:05.03 | aptura | I have a case in mind. |
| 01:05.06 | JT | i thought you might've been using something smaller than ATX though |
| 01:06.40 | aptura | 17x22x8 is the size of this box. |
| 01:10.14 | Un1x | aptura got a picture of the case? |
| 01:11.06 | aptura | Since thay are very low volume I would hesitate to give it out as it can take up to three weeks to order it. |
| 01:12.22 | Un1x | heh save the image of the bo |
| 01:12.25 | Un1x | *case |
| 01:12.27 | Un1x | and upload it to |
| 01:12.29 | Un1x | imageshack |
| 01:12.37 | Un1x | i just wanna see what it looks like man i like rachmounts |
| 01:12.44 | Un1x | i dont like other cases :p |
| 01:12.50 | aptura | its not a rack mount |
| 01:12.51 | aptura | :) |
| 01:12.59 | Un1x | yea i know i just wanna see it |
| 01:13.04 | Un1x | its why i said upload image |
| 01:13.08 | Un1x | i dont care where its from :p |
| 01:13.32 | *** part/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net) |
| 01:18.52 | *** join/#asterisk Dustyservers (n=eric@S01060060979872cb.ed.shawcable.net) |
| 01:19.26 | Dustyservers | can anyone tell me where I can get an fxo card? |
| 01:20.31 | Supaplex | well yea. digium, ebay |
| 01:20.43 | Dustyservers | umm |
| 01:20.47 | Dustyservers | ok |
| 01:20.58 | Dustyservers | is there a candian digium site |
| 01:21.02 | Dustyservers | as am from canada |
| 01:22.35 | *** join/#asterisk ManxPower (n=manxpowe@stirprop-s4-0-0-21.ndcr2.datasync.net) |
| 01:23.14 | Un1x | Dustyservers Sangoma |
| 01:23.16 | Un1x | there canadian |
| 01:23.22 | Un1x | there in Markham |
| 01:23.26 | Un1x | ontario |
| 01:24.06 | Dustyservers | ok so that would be www.digium.com then? |
| 01:24.45 | blitzrage | Dustyservers: www.voipsupply.ca |
| 01:25.06 | Un1x | Dustyservers no |
| 01:25.18 | blitzrage | Sangoma != Digium :) |
| 01:25.22 | Un1x | no \ |
| 01:26.05 | Un1x | Sangoma cards are way better |
| 01:26.08 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net) |
| 01:26.21 | Un1x | and alot of people have said same thing due to quality and less interrupts and shit |
| 01:26.33 | Un1x | i dont think i would have spend 2k on a A200 if it wasn't good |
| 01:26.54 | JT | how much is the A200 board alone? |
| 01:27.25 | Un1x | like 300 |
| 01:27.25 | Un1x | or something |
| 01:27.25 | *** join/#asterisk SimoAmi (n=SimoAmi@ip67-91-253-242.z253-91-67.customer.algx.net) |
| 01:27.25 | Un1x | im not exactly sure but i bought mine with 24 FXS ports |
| 01:27.25 | SimoAmi | hi there |
| 01:27.25 | ManxPower | Sangoma and Digium cards both work. More people use Digium so there are more people around to help. Sangoma's install procedure is a little weird. However, many people think that Sangoma cards are compatible with a wider range of motherboards and that is a plus. Both company's products are priced similar. |
| 01:27.25 | Un1x | with the addition daughter boards |
| 01:27.45 | Un1x | Yes. but Sangoma tends to bend over backwards for there customers |
| 01:27.50 | Un1x | if you have problems with installs or anytihng |
| 01:27.51 | blitzrage | not in my experience |
| 01:27.55 | Un1x | they will help you with all of it |
| 01:27.59 | blitzrage | Sent an email, had no reply for 3 days |
| 01:28.00 | Un1x | and even do it for you if you want |
| 01:28.11 | ManxPower | My one tech support experience with Sangoma was good. |
| 01:28.11 | Un1x | blitzrage they dont like americans :P |
| 01:28.22 | orlock | i use Traverse PCI ADSL cards |
| 01:28.25 | Un1x | i dont know man i ususaly call |
| 01:28.26 | Un1x | i dont email |
| 01:28.27 | blitzrage | Un1x: who said I was an American -- I live in Toronto |
| 01:28.30 | SimoAmi | I get "All circuits are busy now" when using a boadvoice sip trunk. any idea/help? |
| 01:28.30 | orlock | which are essentialy Sangoma cards |
| 01:28.35 | blitzrage | Un1x: AND they know who I am there :) |
| 01:28.35 | ManxPower | blitzrage: I thought I had that experience too until I checked my spam folder. For some reason their response got tossed in there. |
| 01:28.37 | Un1x | and they help solve the problem within like 6 hours usualy |
| 01:28.39 | orlock | well, theyare rebadged sangoma cards |
| 01:28.40 | Un1x | all the times ive called |
| 01:28.43 | blitzrage | ManxPower: nah, it wasn't that :) |
| 01:28.44 | Un1x | gotten help witihn 6 hours |
| 01:28.47 | ManxPower | SimoAmi: You want #freepbx |
| 01:29.31 | Un1x | anywya i prefer sangoma over digium cards |
| 01:29.33 | blitzrage | well either way, hardware sucks :) |
| 01:29.40 | orlock | blitzrage: yup |
| 01:29.48 | Un1x | heh the money part |
| 01:29.50 | Un1x | or all hardware :P |
| 01:29.50 | blitzrage | totally unnecessary except for phones |
| 01:29.55 | Un1x | i hate when u gotta give money lol |
| 01:29.56 | orlock | MAC addresses vanishing from the arp cach every 30 seconds |
| 01:30.02 | blitzrage | money doesn't bother me, I make enough |
| 01:30.10 | ManxPower | orlock: not a duplex mispatch? |
| 01:30.16 | Un1x | i prrefer money coming in rather then going out |
| 01:30.22 | Dustyservers | lol |
| 01:30.23 | Un1x | and anyonew with a business mind will tell you same thing |
| 01:30.26 | Un1x | its all more about profit |
| 01:30.28 | Dustyservers | anyways thanks for all the help |
| 01:30.29 | Un1x | then revenues :) |
| 01:30.32 | blitzrage | well duh |
| 01:30.33 | Un1x | heh cya dusty |
| 01:30.37 | blitzrage | you go to school for that? |
| 01:30.39 | Un1x | No |
| 01:30.40 | Un1x | lol |
| 01:30.47 | Un1x | im 18 f00 |
| 01:30.57 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 01:31.01 | blitzrage | ahhhh, it all makes sense now :) |
| 01:31.08 | Un1x | what makes sense? |
| 01:31.52 | Un1x | that i know about business at this age |
| 01:31.53 | blitzrage | no |
| 01:31.53 | ManxPower | Un1x: Not quite as bitter, jaded, and hateful as the rest of us, eh? |
| 01:31.53 | Un1x | Nopes |
| 01:31.53 | Un1x | im a peoples person |
| 01:32.02 | *** join/#asterisk dr0ne (n=fn@S01060016b6b541d2.va.shawcable.net) |
| 01:32.03 | Un1x | i gotta go warm my stupid gell packs i got drom doctor |
| 01:32.05 | Un1x | brb |
| 01:32.10 | SimoAmi | thanks |
| 01:32.31 | ManxPower | Some asshole consultant from the Accounting department wants to put the company on Vonage. A USD$ 550 Million/year company with 400 employees, and 18 offices. On Vonage. |
| 01:32.46 | Qwell | ManxPower: congrats |
| 01:33.02 | ManxPower | I'm really glad I was not at the meeting. I would have wet my self laughing. |
| 01:33.26 | Qwell | ManxPower: $20 says you would've wet the consultant |
| 01:33.57 | orlock | ManxPower: hahahah |
| 01:34.01 | ManxPower | Qwell: They don't let me go to those sorts of meetings anymore. I tend to make people mad at me. |
| 01:34.08 | blitzrage | ManxPower: wow... that's amazing |
| 01:34.12 | orlock | ManxPower: heh, i do thatto |
| 01:34.14 | blitzrage | I would have laughed HARD |
| 01:35.02 | ManxPower | I used to spend 1/2 of my time fending off those fly by night consultants, then I realized they are all idiots and could not do anything cheaper than we are already doing things so why not just ignore them. |
| 01:35.34 | blitzrage | yep |
| 01:36.01 | blitzrage | in fact, if a business is going after the cheapest possible solution, I don't want them |
| 01:36.13 | blitzrage | they are usually the worst as they expect the moon for $1 |
| 01:36.20 | ManxPower | blitzrage: someone got mad at me for saying that person X not knowing what the RELEASE button on a Nortel phone (she is the dept receptionist) is, is like a 30 yr old now knowing how to tie their shoes. |
| 01:36.25 | blitzrage | and tend to have very little idea what they really want |
| 01:36.40 | ManxPower | blitzrage: the ACCOUNTING department wants to take over IT. |
| 01:36.51 | blitzrage | oh yah, that makes a lot of sense |
| 01:37.07 | SimoAmi | #freepbx is very quiet |
| 01:37.27 | ManxPower | SimoAmi: well that is the place to ask questions about FreePBX/Trixbox |
| 01:37.37 | SimoAmi | maybe someone here can help |
| 01:37.38 | *** join/#asterisk marlow (n=marlow@87.198.132.2) |
| 01:37.51 | ManxPower | SimoAmi: Asterisk does not play the message you are receiving |
| 01:38.00 | ManxPower | Asterisk does not have "SIP trunks" |
| 01:38.07 | ManxPower | All that stuff is specific to FreePBX/Trixbox |
| 01:38.08 | blitzrage | I can't help with GUI stuff since I only know the CLI |
| 01:38.10 | JT | freepbx users are probably too busy 0wning others in online games atm to help with freepbx |
| 01:38.17 | JT | or whatever it is that kiddies do |
| 01:38.58 | SimoAmi | ok, well, I'll have to dig in through the different configs |
| 01:39.13 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 01:39.39 | blitzrage | JT: lol |
| 01:39.49 | SimoAmi | also, I want to setup a pure asterisk application that interacts with php |
| 01:39.50 | Supaplex | it's for kids |
| 01:40.03 | blitzrage | SimoAmi: learn Asterisk then |
| 01:40.04 | JT | agi would do that interaction |
| 01:40.22 | Supaplex | sh -c "rm -f `which php`" |
| 01:40.27 | Supaplex | ;) |
| 01:40.52 | SimoAmi | how easy is it to have a service call a specific client and ask for a confimation code for a specific transaction? |
| 01:41.07 | blitzrage | using a GUI tends to just delay the inevitable. I only recommend GUIs for managers, and those who already know asterisk and only use the GUI for repetitive/management type things |
| 01:41.22 | blitzrage | SimoAmi: that's reasonably easy |
| 01:41.26 | SimoAmi | sort of "Hello, Please enter the code for transaction 14678" |
| 01:41.40 | blitzrage | yah... use a callfile to dump the caller into a context which does that |
| 01:41.46 | blitzrage | callfile + Read() |
| 01:41.55 | JT | umm if it's inbound |
| 01:41.58 | JT | why use callfile |
| 01:42.05 | *** join/#asterisk kyo (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 01:42.06 | JT | use diaplan + agi |
| 01:42.10 | blitzrage | oh... for some reason I thought he was polling out to an end point |
| 01:42.17 | blitzrage | agi for what? |
| 01:42.26 | blitzrage | if you say the DB lookup, use func_odbc |
| 01:42.27 | JT | he wanted to connect with php |
| 01:42.42 | SimoAmi | yes |
| 01:42.52 | blitzrage | for that situation, I don't see the advantage of using AGI, unless you're doing something really funky |
| 01:42.57 | blitzrage | too much overhead |
| 01:43.12 | SimoAmi | it could be through xmlrcp instead of connecting remotly to mysql server |
| 01:43.39 | *** join/#asterisk Dustyservers (n=eric@S01060060979872cb.ed.shawcable.net) |
| 01:43.41 | blitzrage | still, calling a parser is still going to be more memory than calling the DB from the dialplan directly |
| 01:43.47 | JT | yeah it depends on the application |
| 01:43.57 | blitzrage | right -- I'm just a func_odbc fanboy :) |
| 01:44.09 | blitzrage | I still use AGI for certain tasks, but rarely anymore |
| 01:44.30 | SimoAmi | the main program will be a cron job that inquires for new unconfirmed transactions and get the list of numbers to dial, the trxn # and confirmation code |
| 01:44.30 | Dustyservers | hi I notice that the fxo card is an rj45 connector how would I connect my regular phone compnay to that when the regular phone connector is a rj11? |
| 01:44.49 | blitzrage | Dustyservers: you just plug it in, it fits |
| 01:44.56 | Dustyservers | oh |
| 01:44.56 | JT | SimoAmi: you can use call files or the manager interface to make outgoing calls |
| 01:45.01 | blitzrage | rj11 is just narrower (2 pair instead of 4 pair) |
| 01:45.15 | JT | rj45 are 4 pair btw |
| 01:45.17 | Dustyservers | it will fit even if it not the right size? |
| 01:45.23 | JT | yes it will fit |
| 01:45.27 | Dustyservers | werid |
| 01:45.31 | Dustyservers | that new to me |
| 01:45.33 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:45.35 | blitzrage | Dustyservers: not really if you look at the connectors |
| 01:45.56 | Dustyservers | ic |
| 01:46.01 | SimoAmi | JT: manager interface of "FreePBX" ? |
| 01:46.01 | Dustyservers | one outher question |
| 01:46.09 | JT | of asterisk |
| 01:46.31 | JT | you wouldn't want to be running a business system on freepbx |
| 01:46.32 | Dustyservers | how to I do an auto attence message thingy |
| 01:46.49 | SimoAmi | ok, does asterisk automatically dial out call files? |
| 01:46.51 | blitzrage | ~thebook |
| 01:46.59 | jbot | thebook is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:46.59 | JT | Playback() plays sound files |
| 01:47.00 | blitzrage | ~book |
| 01:47.01 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:47.02 | blitzrage | shoot |
| 01:47.17 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 01:47.23 | JT | SimoAmi: yes, also covered in the book |
| 01:48.05 | shmaltz | anybody seen this? Polycom 501 with asterisk 1.2.13, asterisk keeps losing the registrations fo the Polycom phones. THey are all on the same network, no NAT |
| 01:48.34 | Dustyservers | would you recomend to use freepbx or to configure asterisk your self? |
| 01:48.36 | blitzrage | try dropping the registration timeout |
| 01:48.43 | blitzrage | so it re-reg's more often |
| 01:48.44 | Dustyservers | just trying to figure out what woudl be best for me |
| 01:49.05 | blitzrage | Dustyservers: depends what you're trying to do -- if for business, learn Asterisk. Play hobby, you can start with FreePBX |
| 01:49.33 | Dustyservers | aww |
| 01:49.35 | blitzrage | some things you have no choice but to learn asterisk if you want to take full advantage |
| 01:49.52 | Dustyservers | can you have auto attence with freepbx? |
| 01:50.02 | blitzrage | Dustyservers: go look at the freepbx website |
| 01:50.02 | JT | wtf is attence? |
| 01:50.07 | JT | do you mean auto attendant? |
| 01:50.11 | Dustyservers | liek prince 1 sale 2 for support |
| 01:50.11 | Dustyservers | etc |
| 01:50.19 | Dustyservers | *press |
| 01:50.20 | blitzrage | Dustyservers: yes, of course |
| 01:50.24 | Dustyservers | ok |
| 01:50.24 | JT | never heard of the word "attence" |
| 01:50.29 | Supaplex | ivr |
| 01:50.36 | Dustyservers | aww ic |
| 01:50.37 | Dustyservers | ok |
| 01:50.38 | JT | i know what an ivr is |
| 01:50.41 | Dustyservers | lol |
| 01:50.44 | blitzrage | well... ivr is more than just an auto attendent |
| 01:50.45 | Dustyservers | sorry am new to this |
| 01:50.50 | Dustyservers | ok |
| 01:50.57 | blitzrage | Dustyservers: yah, you should read some docs, |
| 01:51.10 | Dustyservers | why is ivr more then just an auto attendent jw? |
| 01:51.16 | Supaplex | ~wiki |
| 01:51.46 | shmaltz | anybody seen this? Polycom 501 with asterisk 1.2.13, asterisk keeps losing the registrations fo the Polycom phones. THey are all on the same network, no NAT |
| 01:51.47 | *** join/#asterisk topping (n=topping@64.212.181.67) |
| 01:52.19 | blitzrage | yah.. try dropping the registration timeout so it re-reg's faster |
| 01:53.05 | shmaltz | blitzrage, done that :( still no help |
| 01:53.17 | blitzrage | hrmmmm.... thats weird |
| 01:53.20 | shmaltz | maxexpiry=1800 |
| 01:53.21 | shmaltz | defaultexpiry=1200 |
| 01:53.33 | *** part/#asterisk ming_zym (n=zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 01:53.42 | blitzrage | try like.... 60 |
| 01:53.44 | shmaltz | in fact the Polycoms think that they are registerd |
| 01:54.58 | shmaltz | I changed it to this, lets see if it will help, but I doubt it will |
| 01:55.00 | shmaltz | maxexpiry=600 |
| 01:55.01 | shmaltz | defaultexpiry=120 |
| 01:55.16 | SimoAmi | don't you think using some kind of api is better than creating call files? |
| 01:56.05 | blitzrage | sure... that's what the manager is for |
| 01:57.44 | *** join/#asterisk hoobastoob1 (n=ckwall@63.149.122.93) |
| 01:58.12 | SimoAmi | ah ok |
| 01:58.43 | blitzrage | agi is for asterisk to interact with programs, and manager is for programs to interact with asterisk (generally) |
| 01:58.49 | *** join/#asterisk bkrus1 (n=root@69.73.127.92) |
| 01:59.26 | SimoAmi | is manager an api, a source file within the asterisk file structure? |
| 01:59.48 | blitzrage | you connect to it over a network socket (like telnet) |
| 01:59.48 | Nugget | telnet is eeeeeeevil! |
| 01:59.52 | blitzrage | infact, you can telnet into it :) |
| 02:00.31 | SimoAmi | ah, I think I remember something about the manager now |
| 02:01.34 | hoobastoob1 | when i am dialing a meetme conference number and then hanging up, the channel is not being removed. what could be causing that? |
| 02:01.44 | SimoAmi | I think there's a php api that does the low level connection |
| 02:01.59 | blitzrage | that'd be useful |
| 02:02.25 | blitzrage | luckily someone I work with wrote a class that did it |
| 02:03.08 | SimoAmi | also, do I need to write a file for the asterisk logic? |
| 02:03.08 | hoobastoob1 | and if i do a soft hangup and the channel, it does not hang it up |
| 02:03.17 | blitzrage | what version? |
| 02:03.28 | blitzrage | is it only in the meetme? |
| 02:03.32 | hoobastoob1 | Asterisk SVN-branch-1.2-r48356 |
| 02:03.34 | blitzrage | I've never experienced that... |
| 02:03.34 | hoobastoob1 | yes |
| 02:03.43 | blitzrage | does it do it in 1.2.12.1? |
| 02:03.53 | hoobastoob1 | have not tried it |
| 02:04.02 | blitzrage | give that a quick shot to make sure its not a new bug introduced |
| 02:04.09 | hoobastoob1 | ok, will do |
| 02:05.25 | hoobastoob1 | i just tested it on another Asterisk SVN-branch-1.2-r48356 server and it does not do it |
| 02:05.30 | hoobastoob1 | it works correctly there |
| 02:05.36 | hoobastoob1 | so it shouldnt be a bug |
| 02:05.41 | blitzrage | agreed |
| 02:05.44 | hoobastoob1 | this is one i am trying to set up with vicidial |
| 02:05.46 | blitzrage | whats different in the network? |
| 02:05.52 | hoobastoob1 | vicidial |
| 02:05.55 | bkrus1 | someone mention php class? :D |
| 02:06.10 | SimoAmi | yes |
| 02:06.21 | blitzrage | no |
| 02:06.25 | SimoAmi | :) |
| 02:06.27 | bkrus1 | blitzrage: whats up <3 |
| 02:06.35 | bkrus1 | :P |
| 02:06.45 | blitzrage | leafs game |
| 02:06.55 | bkrus1 | Lol! |
| 02:06.55 | bkrus1 | i need to post my 1.2 and 1.4 class for php, it does anything you can do at the command line |
| 02:07.09 | bkrus1 | s/command line/manager interface |
| 02:07.17 | SimoAmi | not sure it's a php class but it's an api to interface with asterisk's manager interface |
| 02:07.18 | blitzrage | bkrus1: that'd be useful |
| 02:07.30 | bkrus1 | SimoAmi: yes, class |
| 02:07.40 | bkrus1 | so when you include('whatever.php'); |
| 02:07.42 | bkrus1 | no |
| 02:07.46 | bkrus1 | how do you do that in php again(thinks) |
| 02:07.53 | bkrus1 | require_once |
| 02:08.16 | bkrus1 | then new asterisk....whatever i forget, its been a couple weeks and my mind isnt in php mode |
| 02:08.17 | SimoAmi | require_once('classfile.php'); |
| 02:08.24 | bkrus1 | correct, thanks |
| 02:08.46 | blitzrage | yah, I go work on php, then I go back into asterisk mode and forget about the php :) |
| 02:09.01 | bkrus1 | then $asterisk=new('class...i forgot |
| 02:09.04 | bkrus1 | ill get it to you though :] |
| 02:09.06 | blitzrage | anyways... GO LEAFS GO |
| 02:09.11 | bkrus1 | :D |
| 02:09.26 | JT | leafs? |
| 02:09.39 | bkrus1 | JT: its blitzrage, who knows :] |
| 02:09.46 | JT | heh |
| 02:09.56 | bkrus1 | but then, you can just do $asterisk->login_agents("agent_array etc etc"); and $asterisk->blah blah blah |
| 02:10.27 | bkrus1 | $asterisk->raw_command("$server1", "command", "args"); etc etc, ill try to get that to somewhere you guys can get to it |
| 02:10.48 | shmaltz | anybody seen this? Polycom 501 with asterisk 1.2.13, asterisk keeps losing the registrations fo the Polycom phones. THey are all on the same network, no NAT |
| 02:10.55 | Qwell | bkrus1: Where's your e? |
| 02:11.05 | partition | Qwell: he didn't pay his bill |
| 02:11.08 | Qwell | ahh |
| 02:11.13 | Qwell | nub ;/ |
| 02:11.15 | bkrus1 | at&T cut it off ;[ |
| 02:11.27 | bkrus1 | Qwell: Im signed on irc at work, and this was my secondary name :P |
| 02:11.31 | Qwell | ahh |
| 02:11.33 | bkruse_home | :] |
| 02:11.44 | bkruse_home | whats up qwell |
| 02:13.16 | bkruse_home | SimoAmi: what would you be interfacing the manager interface in php for? |
| 02:14.16 | SimoAmi | good question |
| 02:15.03 | bkruse_home | SimoAmi: :P I will get them to you and you can go from therE :] |
| 02:15.22 | SimoAmi | sure |
| 02:15.28 | SimoAmi | let me explain |
| 02:15.30 | bkruse_home | k |
| 02:15.53 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:16.22 | SimoAmi | I want to create a simple application that queries a db for new client transactions |
| 02:16.48 | shmaltz | anybody seen this? Polycom 501 with asterisk 1.2.13, asterisk keeps losing the registrations fo the Polycom phones. THey are all on the same network, no NAT |
| 02:16.57 | Un1x | ~FXSFXO |
| 02:17.00 | jbot | well, fxsfxo is an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 02:17.00 | SimoAmi | then call each client and ask them to confirm the transaction via a confirmation code |
| 02:17.04 | shmaltz | anybody seen this? Polycom 501sip 1.6.7 with asterisk 1.2.13, asterisk keeps losing the registrations fo the Polycom phones. THey are all on the same network, no NAT |
| 02:17.37 | bkruse_home | SimoAmi: that will be easily done with this class |
| 02:17.41 | hoobastoob1 | anyone here done vicidial? |
| 02:17.41 | bkruse_home | you can call on the db with |
| 02:17.52 | bkruse_home | $asterisk->raw_command("blahb lah"): |
| 02:17.53 | SimoAmi | so the voice prompt is "Hi, please enter the confirmation code for the transaction 14256" |
| 02:18.27 | Un1x | SimoAmi even if you do thatthere is a way around those applications |
| 02:18.47 | SimoAmi | good, like what? |
| 02:18.52 | Un1x | no not a good way for you |
| 02:18.57 | bkruse_home | SimoAmi: you could probably do something like that......by handling your variables in php and then making a call with whatever your var names are.......... |
| 02:18.57 | Un1x | meaning a way around for clients |
| 02:19.09 | Un1x | if youre trying to prevent fraud etc on ur site with that app there is a way around |
| 02:20.07 | SimoAmi | it's not what you think. We need to confirm over the phone. It sets the transaction status to a higher level "Confirmed" |
| 02:20.35 | bkruse_home | SimoAmi: I think you could do that, i could help you with it if you need it |
| 02:20.51 | bkruse_home | Un1x: I agree........to some extent..........but for a basic confirmation i dont think its a problem |
| 02:20.56 | SimoAmi | bkruse_home: what do you suggest? |
| 02:20.59 | bkruse_home | it depends how big security plays a role |
| 02:21.15 | bkruse_home | SimoAmi: we could just design our own php based system to handle everything, as long as the call volume isnt huge |
| 02:21.39 | SimoAmi | not much of a security issue. because you don't provide any info |
| 02:21.41 | bkruse_home | well.....it could still be high call volume we are just reading db numbers and originating calls in the manager interface |
| 02:21.54 | SimoAmi | true |
| 02:21.58 | bkruse_home | SimoAmi: anything could be hacked, just always remember |
| 02:22.33 | bkruse_home | so my suggestion is to keep all (most) your vars that you can in php, so you can directly reference them instead of $callerid = $asterisk->db_call("blahblahblah"); |
| 02:22.41 | bkruse_home | or whatever. |
| 02:22.53 | SimoAmi | I know how to inquire for the list of pending transactions. I would need help initiating the calls, speaking the transaction number and getting the code back |
| 02:23.49 | SimoAmi | the db query can be done through a mysql_query() |
| 02:23.53 | bkruse_home | ya, i dont think its a problem, we could do it |
| 02:23.56 | bkruse_home | correct. |
| 02:24.20 | bkruse_home | so you just want them to punch in a confirmation number, match that to their number/username whatever authentication we want etc etc |
| 02:24.54 | SimoAmi | now the details would be something like how to repeat the message after 10 seconds of non response |
| 02:24.57 | Un1x | bkruse_home well at least you agree it doesn't help prevent fraud |
| 02:25.05 | Un1x | i mean if the guy is doing fraud he probably has a bunch of dids anyway |
| 02:25.17 | Un1x | so he could just use anyone of them for his server will just call he will answer get code |
| 02:25.20 | Un1x | and hes verified |
| 02:25.26 | Un1x | it really doesn't protect anythin |
| 02:25.29 | bkruse_home | Un1x: deffinitly not.... |
| 02:25.32 | Un1x | but just basic confirmation |
| 02:25.39 | Un1x | that the phone #asterisk is active |
| 02:25.40 | Un1x | lol |
| 02:25.41 | Un1x | err |
| 02:25.44 | Un1x | phone number |
| 02:25.49 | bkruse_home | right. |
| 02:25.52 | bkruse_home | or w/e |
| 02:25.56 | Un1x | Yep |
| 02:26.01 | bkruse_home | i would NOT implement this system in some type of ordering system |
| 02:26.09 | Un1x | neither would i |
| 02:26.11 | bkruse_home | callerid spoofing = easy, imagine what you could do with that!!! |
| 02:26.12 | *** join/#asterisk rtcg (n=chatzill@mail.richardthecomputerguy.com) |
| 02:26.27 | Un1x | i would rather, a Verified by Visa and mastercard secure code along with bg credential verification |
| 02:26.30 | Un1x | its alot harder process |
| 02:26.30 | bkruse_home | authentication by DID's is never a good idea :P |
| 02:26.35 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 02:26.35 | bkruse_home | right |
| 02:26.40 | Un1x | but as long as you do it thoroughly bank cant chargeback |
| 02:26.44 | bkruse_home | anything is crackable, but this method is a couple of numbers |
| 02:26.48 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 02:26.53 | SimoAmi | just remember, the call will target subscribed clients, so they know they're getting these calls |
| 02:26.53 | Un1x | yep |
| 02:27.12 | bkruse_home | SimoAmi: i dont see a problem with this then, looks like a good project |
| 02:27.33 | Un1x | tbh i hate people/telemarketers who call and ask if i would like new services or woudl like to renew or would like to buy something |
| 02:27.37 | Un1x | i prefer if i dont phone u |
| 02:27.43 | *** join/#asterisk Qwell_ (n=north@unaffiliated/qwell) |
| 02:27.43 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 02:27.45 | Un1x | dont call me |
| 02:27.49 | bkruse_home | haha |
| 02:27.55 | bkruse_home | +v Qwell |
| 02:27.57 | Un1x | because if i wanted to keep youre service i would have called you to extend |
| 02:28.00 | Un1x | not you calling me |
| 02:28.03 | bkruse_home | right. |
| 02:28.05 | SimoAmi | ok, now is the ivr script something to code in php as well or no? |
| 02:28.09 | bkruse_home | is this a telemarkter this SimoAmi? |
| 02:28.16 | Un1x | i dont know... |
| 02:28.21 | Un1x | but somethingh with calling clients |
| 02:28.28 | bkruse_home | SimoAmi: um.....maybe? |
| 02:28.29 | Un1x | i bitch at my insurance company once for calling me |
| 02:28.33 | Un1x | never receieved calls from them again |
| 02:28.36 | bkruse_home | SimoAmi: the COOLEST thing about the manager interface is tihs |
| 02:28.52 | partition | Qwell: stay! |
| 02:28.54 | bkruse_home | SimoAmi: you can do this with my manager interface |
| 02:28.55 | Qwell | no |
| 02:28.58 | Qwell | you |
| 02:29.07 | bkruse_home | SimoAmi: $asterisk->addvar("varname", "varvalue"); |
| 02:29.12 | bkruse_home | so in your dialplan you could do |
| 02:29.24 | *** join/#asterisk wundaboy (n=asdf@c-24-21-109-26.hsd1.mn.comcast.net) |
| 02:29.28 | bkruse_home | saydigits($custom_digits_from_manager_interface); or w/e |
| 02:29.37 | bkruse_home | well ${}* not $() |
| 02:30.40 | SimoAmi | bkruse_home, no it's a form of quality of service improvement. I know you're trying to understand the concept but trust me this is legitimate |
| 02:30.50 | bkruse_home | SimoAmi: kk, cool |
| 02:31.03 | bkruse_home | SimoAmi: the cool thing is, you can update your vars from php into your dialplan |
| 02:31.03 | bkruse_home | :] |
| 02:31.12 | bkruse_home | so you make your dialplan SUPER dynamic |
| 02:31.22 | Qwell | I'm not a huge fan of AGI. :p |
| 02:31.40 | bkruse_home | Qwell: me either! |
| 02:31.47 | Qwell | </random admission> |
| 02:31.54 | Qwell | erm..(sp) |
| 02:32.13 | bkruse_home | :] |
| 02:32.17 | bkruse_home | +v Qwell |
| 02:32.19 | SimoAmi | wow that's cool |
| 02:32.22 | bkruse_home | exten => dial(Iax2/zomgtrunk/${allthiscrap}|30) |
| 02:32.30 | SimoAmi | and is the dial plan in a seperate file? |
| 02:32.35 | bkruse_home | so SimoAmi we can work on that later on if you want |
| 02:32.41 | bkruse_home | uh...how much asterisk experience do you have? |
| 02:32.53 | *** mode/#asterisk [+v Qwell] by Qwell |
| 02:32.56 | Qwell | bkruse_home: good idea |
| 02:33.08 | Un1x | heh |
| 02:33.15 | Un1x | man hes more lost then i am at times |
| 02:33.34 | bkruse_home | :D |
| 02:33.50 | bkruse_home | SimoAmi: vi /etc/asterisk/extensions.conf |
| 02:33.56 | Un1x | Qwell; you know all those previous issues i used to have with extensions.conf i got some time yestorday to take a good look at it |
| 02:34.04 | bkruse_home | set aside some time tomorrow night SimoAmi and we can work on it |
| 02:34.06 | Un1x | i found out the dumbass freind of mine duplicated many rules |
| 02:34.10 | Un1x | thats why everything was a mess |
| 02:34.15 | SimoAmi | In terms of asterisk experience, I just setup a trixbox server for a company |
| 02:34.17 | Un1x | now its soooooo clean :) |
| 02:34.24 | Un1x | trixbox lol |
| 02:34.31 | SimoAmi | sure |
| 02:34.39 | SimoAmi | or maybe even better |
| 02:34.41 | docelmo | SimoAmi ok great you figured out how to be dumb.. No go join their channel |
| 02:34.50 | JT | no need to be so rude docelmo |
| 02:35.10 | bkruse_home | SimoAmi: trixbox=lame |
| 02:35.12 | docelmo | I said it everyone else was thinking it |
| 02:35.31 | bkruse_home | SimoAmi: you wana know where its at? what all the best of the best use? |
| 02:35.35 | bkruse_home | its not trixbox.... but http://asterisknow.org/ |
| 02:35.43 | bkruse_home | Qwell: Laugh |
| 02:36.03 | SimoAmi | ok, thx |
| 02:36.11 | bkruse_home | SimoAmi: no problem, you will LOVE it |
| 02:36.18 | JT | docelmo: no, they weren't, it's a starting point for some people, there really is no need to put things that way |
| 02:36.28 | SimoAmi | ok, do you have an email |
| 02:36.37 | bkruse_home | SimoAmi: we will have to get you caught up to speed on asterisk before we dive into the managers interface, but your idea is DEffinitily do-able |
| 02:36.42 | bkruse_home | SimoAmi: bkruse@digium.com |
| 02:36.54 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
| 02:37.19 | SimoAmi | ok, thanks bkruse_home |
| 02:37.28 | SimoAmi | mine simoami@hotmail.com |
| 02:37.40 | bkruse_home | no problem, let me know how you like asterisknow, it has a nice pretty web interface any everything |
| 02:37.59 | bkruse_home | and if you like that, you will love Business Edition Asterisk :] |
| 02:38.42 | bkruse_home | SimoAmi: send me an email one day when you think your ready :] |
| 02:39.03 | bkruse_home | SimoAmi: http://asterisknow.org/images/gui |
| 02:39.37 | Un1x | Okayguys i'll be back in a bit |
| 02:39.46 | SimoAmi | the gui looks neat |
| 02:39.50 | bkruse_home | Un1x: have fun :] |
| 02:39.54 | Un1x | I'm going to try and look for more toll free did providers |
| 02:40.02 | SimoAmi | thx Unlx |
| 02:40.02 | Un1x | that provide only DID, not other crap services with it :p |
| 02:40.12 | Un1x | SimoAmi for what? |
| 02:40.18 | bkruse_home | Un1x: tell me how that goes :P |
| 02:40.19 | bkruse_home | SimoAmi: oh its awesome, very very helpful for getting down asterisk in a quick, and getting the conceptual in your head |
| 02:40.35 | SimoAmi | for sharing this conversation |
| 02:40.37 | SimoAmi | ;) |
| 02:41.00 | Un1x | bkruse_home i found one that charges 50 cents per month for the did and .017 cents a minute |
| 02:41.06 | Un1x | its pretty cheap compared to most |
| 02:41.10 | Un1x | and doesn't even charge setup fees |
| 02:41.17 | Un1x | its where i got one from |
| 02:41.24 | bkruse_home | Un1x: WHAT! |
| 02:41.28 | bkruse_home | Un1x: thats awesome! |
| 02:41.40 | bkruse_home | Un1x: thats the best ive seen, who is it, link meh! plz |
| 02:41.45 | Un1x | yep 50 cents per month and .017 cents a minute its better then all |
| 02:42.07 | bkruse_home | thats the best ive seen, by a good bit, i wonder how good the service is |
| 02:42.10 | Un1x | you will love them quality is good too |
| 02:42.13 | Un1x | they support g729 as well |
| 02:42.19 | Un1x | its good i like it |
| 02:42.22 | Un1x | especialy for that price |
| 02:42.24 | Un1x | its awesome |
| 02:42.24 | bkruse_home | nice! |
| 02:42.38 | Un1x | i mean even if you dont use the did or anytihn u only pay 50 cents per month to keep |
| 02:42.43 | Un1x | not like 5-20$ per month |
| 02:42.43 | bkruse_home | well if your running more than a couple g729 channels licensed, then your not worried about DID costs :P |
| 02:42.45 | Un1x | from other carriers |
| 02:42.56 | Un1x | heh im running 4 licenced channels ;) |
| 02:43.04 | Un1x | i would have got more but i doubt i will have the need :) |
| 02:43.22 | Un1x | N/ADebit$0.05N/A11-06-2006 to 12-07-2006 Usage Charges (2.8 minutes) |
| 02:43.22 | Un1x | 12-07-2006Credit$501398z73454z7789Initial Credit Card Payment |
| 02:43.22 | Un1x | 12-07-2006Debit$.50naqvujwuflucqyrjajToll Free Number: 8772329080 |
| 02:43.24 | Un1x | see |
| 02:43.29 | Un1x | only charged 50 cents :P |
| 02:43.51 | bkruse_home | dang |
| 02:43.59 | Un1x | bkruse_home where do you get youre dids from |
| 02:44.00 | bkruse_home | thats so cool! |
| 02:44.08 | Un1x | i got a asian did from didww.com btw it costs 10 per month |
| 02:44.17 | Un1x | but worth it since it obsoletes the long distance call |
| 02:44.25 | Un1x | and they dont charge per min either :) |
| 02:45.18 | bkruse_home | Un1x: dids from which place i have asterisk at :X |
| 02:45.27 | bkruse_home | Un1x: im currently looking for a good iax provider |
| 02:45.31 | bkruse_home | err. iax termination |
| 02:46.21 | Un1x | yea same here |
| 02:46.27 | Un1x | i heard Nufone is ok |
| 02:46.32 | bkruse_home | thats what i heard to |
| 02:46.34 | Un1x | and they allow you to spoof callerid |
| 02:46.38 | bkruse_home | i wonder what the price is, i want a fixed rate. |
| 02:46.39 | Un1x | i might sign up with them |
| 02:46.40 | bkruse_home | woot! |
| 02:47.01 | Un1x | well if you dont care about digital phone type high quality |
| 02:47.05 | Un1x | you can go with decent quality |
| 02:47.07 | Un1x | broadvoice |
| 02:47.12 | Un1x | offers like for 30$ a month |
| 02:47.16 | Un1x | call in 32 countrys |
| 02:47.26 | *** join/#asterisk Laggy_McGee (n=jchadwic@pool-71-245-124-113.cmdnnj.fios.verizon.net) |
| 02:47.32 | bkruse_home | 30 dollars flat rate eh? |
| 02:47.39 | Laggy_McGee | Don't mean to start a flame war, but... GSM better than ulaw? |
| 02:48.00 | Qwell | better in what way? |
| 02:48.01 | bkruse_home | Laggy_McGee: depends on what you compare it to, what are you looking for, call quality err what |
| 02:48.10 | Laggy_McGee | Better quality |
| 02:48.16 | Qwell | then no |
| 02:48.20 | Laggy_McGee | I just used ulaw and it was kind of choppy |
| 02:48.28 | bkruse_home | absolutly not. |
| 02:48.31 | Laggy_McGee | will gsm be more or less choppy? |
| 02:48.34 | bkruse_home | Laggy_McGee: thats not ulaw's fault. |
| 02:48.40 | bkruse_home | its your fault for not settings things up right :] |
| 02:48.45 | bkruse_home | Laggy_McGee: over a PRI? |
| 02:48.59 | *** join/#asterisk mceGEEK (n=mceGEEK@70-89-196-165-inkleaf-mn.hfc.comcastbusiness.net) |
| 02:50.01 | Laggy_McGee | bkruse_home: no, an IAX phone connecting directly to my server over FiOS |
| 02:50.23 | *** join/#asterisk waverly360 (n=waverly@c-68-52-128-176.hsd1.tn.comcast.net) |
| 02:50.24 | Un1x | heh i liked ulaw but i needed the bandwidth so went to g729 its not bad its like high class cell phone quality calls, without the dropped calls :P |
| 02:51.35 | bkruse_home | Un1x: i agree. thats a good way to describe it |
| 02:51.36 | bkruse_home | Laggy_McGee: i would look into your network traffic being to great |
| 02:51.50 | bkruse_home | gota remember its 64kbps per call, not to mention overhead, its about 80kbps per call |
| 02:52.13 | Laggy_McGee | bkruse_home: Well, it was overseas, so the latency was probably not great either |
| 02:52.31 | mceGEEK | when we look at SIPDEBUG information what is the f: tag and what is the m: tag? |
| 02:52.53 | bkruse_home | Laggy_McGee: that would be it.....watch ping between the machines, thats a first easy thing to do |
| 02:56.11 | JT | bkruse_home: it's closer to 85kbit/s for g.711 with sip ovethead |
| 02:56.16 | JT | overhead |
| 02:57.58 | JT | Laggy_McGee: ulaw and alaw are base codecs (g.711) and are uncompressed, but companded, they will always be better quality than gsm unless you have insufficient bandwidth, then you may experience breakups :) |
| 02:58.39 | bkruse_home | JT: i would beg to differ its almost exactly 80 kbps. |
| 02:58.54 | bkruse_home | not to start a flame, but i would deffinitly argue that |
| 02:59.01 | JT | bkruse_home: i found in real world usage it is close to exactly 85kbit/s continuous |
| 02:59.17 | JT | watching traffic over a dedicated connection in realtime |
| 02:59.34 | bkruse_home | JT: me to, and thats what spec sheets tell me to ;] |
| 02:59.46 | bkruse_home | either way.......you should be able to tell if u are overloading ur network |
| 02:59.51 | JT | this would include any overhead introduced by asterisk |
| 02:59.55 | *** join/#asterisk _DAW (n=_DAW@adsl-156-78-145.msy.bellsouth.net) |
| 03:00.07 | JT | i think qualify=yes adds some overhead |
| 03:00.11 | bkruse_home | JT: interesting, ima checkl that out tomorrow ;] |
| 03:00.19 | bkruse_home | but i gota go, ill see you guys lata :D |
| 03:00.30 | _DAW | snap |
| 03:00.42 | JT | qualify sends sip messages for OPTIONS the way i understand it |
| 03:00.46 | JT | bkruse_home: alright |
| 03:00.48 | *** part/#asterisk _DAW (n=_DAW@adsl-156-78-145.msy.bellsouth.net) |
| 03:00.57 | bkruse_home | JT: Up |
| 03:01.07 | bkruse_home | JT: it does, depending on your quality time also |
| 03:01.29 | bkruse_home | qualify=1000 ; seconds or milisecs or w/e |
| 03:01.35 | JT | bkruse_home: i've noticed 80kbit/s bandied about the place, but i found it to be an almost completely constant 85kbit/s |
| 03:01.44 | bkruse_home | wow |
| 03:01.46 | bkruse_home | interesting..... |
| 03:01.57 | bkruse_home | it doesnt seem like much, but on a system pushing 3 quad span cards......... |
| 03:01.58 | JT | a 512kbit/s connection could not handle 6 concurrent g.711 calls |
| 03:02.03 | bkruse_home | :P |
| 03:02.07 | bkruse_home | hmm.......... |
| 03:02.14 | bkruse_home | thats kinda cool |
| 03:02.39 | JT | a tiny bit of that would be dsl overhead, and not being able to get full speed out of provider |
| 03:02.42 | JT | cool? :P |
| 03:02.50 | bkruse_home | haha, ya :D |
| 03:02.56 | bkruse_home | i love people that wana run 100+ call centers with a "5 meg line" |
| 03:03.04 | bkruse_home | optical cable/t1 build up? |
| 03:03.05 | bkruse_home | no comcast |
| 03:03.07 | bkruse_home | lol! |
| 03:03.17 | JT | do they just do cable/ |
| 03:03.19 | [TK]D-Fender | <PROTECTED> |
| 03:03.29 | JT | ilibc :P |
| 03:03.32 | bkruse_home | AH! |
| 03:04.00 | bkruse_home | [TK]D-Fender: agreed. but they wanted ulaw :X |
| 03:04.24 | JT | as if not use pri for inbound calls |
| 03:04.31 | bkruse_home | right........ |
| 03:04.36 | JT | i'd have a pri at home if i could afford it |
| 03:04.37 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-44.mtco.com) |
| 03:04.37 | mceGEEK | howdy kruz! |
| 03:04.38 | JT | :P |
| 03:04.41 | Qwell | bah, lpc10 |
| 03:04.42 | bkruse_home | same here, |
| 03:04.44 | bkruse_home | whats up mceGEEK! |
| 03:04.51 | *** join/#asterisk mhnoyes_ (n=mhnoyes@dialup-4.246.6.173.Dial1.SanJose1.Level3.net) |
| 03:05.04 | bkruse_home | Qwell: bah = cant find or bah = dangit i hate it |
| 03:05.07 | JT | i have a t1 at home, but it's only 3metres long and connects straight to the channel bank |
| 03:05.10 | [TK]D-Fender | bkruse_home : Ask them if they want fries with that.... |
| 03:05.16 | mceGEEK | not much .. SR is fishy .. heh |
| 03:05.19 | bkruse_home | [TK]D-Fender: ha! i need to :] |
| 03:05.21 | [TK]D-Fender | Qwell : Domo arigato... |
| 03:05.23 | Qwell | bah = lpc10 > g729 |
| 03:05.27 | Qwell | [TK]D-Fender: exactly |
| 03:05.38 | bkruse_home | mceGEEK: </3 SR |
| 03:05.44 | bkruse_home | Qwell Dang!! |
| 03:06.08 | JT | IMBE over VoIP would sound... interesting |
| 03:06.15 | JT | IMBE is used for digital 2-way radio |
| 03:06.52 | bkruse_home | JT: like techno |
| 03:07.01 | JT | heh yeah |
| 03:07.16 | JT | they use it to fit a voice channel in 10kHz of RF spectrum |
| 03:07.16 | JT | so yeah |
| 03:07.19 | JT | it's low bandwidth |
| 03:07.26 | JT | less thank 10kbit/s |
| 03:07.27 | JT | than |
| 03:07.37 | bkruse_home | dang |
| 03:07.42 | Qwell | JT: see lpc10 |
| 03:07.51 | JT | heh |
| 03:07.52 | bkruse_home | Qwell: HAHA was just typing that! |
| 03:08.09 | bkruse_home | some describe it as "a bit like a robot sound..." |
| 03:08.26 | JT | was lpc10 tested by public safety committies by using it whilst driving a pursuit cars or with gunshots in the bacground? :P |
| 03:08.39 | Qwell | You can get like 12 lpc10 channels on a dialup line |
| 03:08.43 | Qwell | (not really) |
| 03:08.48 | bkruse_home | :P |
| 03:08.52 | bkruse_home | 56k? |
| 03:08.55 | bkruse_home | pssh thats like a call center! |
| 03:09.08 | bkruse_home | the earth on a 3 meg comcast line. |
| 03:10.10 | JT | does comcast do just cable? |
| 03:10.35 | Juggie | no |
| 03:10.55 | JT | you know what we call comcast in australia? |
| 03:10.58 | JT | spamcast |
| 03:11.09 | JT | because we only see spam from people with comcast addresses :P |
| 03:11.13 | JT | on irc and in emails |
| 03:11.23 | Juggie | oh yes |
| 03:11.25 | JT | owned machines i'd expect |
| 03:11.27 | Juggie | because bigpond is a brilliant isp |
| 03:11.38 | JT | heh |
| 03:11.47 | Juggie | check mate my friend :) |
| 03:11.49 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 03:11.50 | JT | has it's good point(s?) |
| 03:11.53 | JT | has bad points |
| 03:12.23 | JT | i haven't seen much spam from the bigpond domain |
| 03:12.25 | Qwell | damnit, I opened a shell...and I don't remember why |
| 03:12.27 | Juggie | gotta love how all the wifi/cable/adsl routers support real tcpip, pppoe, and bigpond |
| 03:12.33 | JT | Qwell: rm -rf / |
| 03:12.49 | JT | hah, bpalogin ftw |
| 03:12.51 | Qwell | I don't have +r on / |
| 03:13.02 | Qwell | or +x for that matter |
| 03:13.04 | JT | +w |
| 03:13.06 | Juggie | Qwell: emacs ~/asterisk/res/res_agi.c find fgets, fix. |
| 03:13.24 | Qwell | svn rm res/res_agi -m "Juggie is a nub" |
| 03:13.26 | Qwell | :D |
| 03:13.47 | Juggie | hah. |
| 03:16.05 | bkruse_home | cya guys. |
| 03:16.06 | *** part/#asterisk bkruse_home (n=root@69.73.127.92) |
| 03:17.04 | Juggie | i'd like to know who these nubs are on the bug tracker who think agi should be async |
| 03:17.40 | Juggie | Qwell, you could also fix app_dial |
| 03:17.50 | Qwell | eh? |
| 03:18.00 | Juggie | 1.2 svn its broken |
| 03:18.07 | Juggie | no longer does proper ring indication |
| 03:18.29 | Juggie | eg, lets say i have a did on a pri, for arguments sake, 1111. |
| 03:18.39 | Juggie | i have exten => 1111,1,Dial(SIP/qwell) |
| 03:18.40 | Qwell | that the thing Damin was talking about earlier? |
| 03:18.44 | Juggie | the far end wont hear ringing |
| 03:18.45 | Juggie | yeah. |
| 03:19.05 | Juggie | i can call you and i'll hear dead silence until you answer. |
| 03:19.24 | Qwell | only from PRI? |
| 03:20.14 | Juggie | so it seems so far yes. |
| 03:20.37 | Qwell | not just sip, right? |
| 03:20.40 | Juggie | no |
| 03:20.44 | Juggie | damin tested it on iax2 as well |
| 03:20.59 | Juggie | but it seems to only happen when zap is involved |
| 03:21.04 | Juggie | i havnt had time to test it myself. |
| 03:21.10 | Juggie | i dont have a box w/ a pri i can flatten right now |
| 03:21.44 | Juggie | but as the bug says if you do Ringing() |
| 03:21.46 | Juggie | before the dial |
| 03:21.47 | JT | funny that |
| 03:21.47 | Juggie | it works |
| 03:22.05 | Juggie | Qwell, who generates the ringing on a zap channel? |
| 03:22.12 | Juggie | asterisk via audio from the core or chan_zap? |
| 03:22.17 | *** join/#asterisk nvicf (n=vincent@201.250.169.175) |
| 03:22.19 | nvicf | hello |
| 03:22.32 | Qwell | core, I think? via a signal from chan_zap |
| 03:22.48 | Juggie | i mean on a zap channel. |
| 03:23.01 | Juggie | is the ringing sound audio? or is it generated in zap. |
| 03:23.30 | docelmo | Anyone know how to pipe shit into a php script? |
| 03:23.53 | Juggie | docelmo, your going to have to be more specific then 'pipe shit' |
| 03:24.10 | docelmo | cat textfile | phpscript.php |
| 03:24.24 | docelmo | how would I read what cat is sending me? |
| 03:24.47 | Juggie | ummm... good question. |
| 03:25.03 | Juggie | why not just read the textfile in the php code? |
| 03:25.08 | docelmo | I have been all over the web and the only function I could find would be popen |
| 03:25.12 | mceGEEK | i think i may be able to help someone who wants to configure asterisk with SR .. |
| 03:25.21 | SimoAmi | I get the following message on asterisk console: |
| 03:25.30 | docelmo | Cause I am gonna pipe emails from sendmail to a php script |
| 03:25.30 | Juggie | docelmo, why not just use php to read the file. |
| 03:25.36 | Juggie | ah. |
| 03:25.50 | SimoAmi | REGISTED attempt 7 to 2122021437@sip.broadvoice.com@sip.broadvoice.com |
| 03:25.50 | Juggie | well, i dont know if php could read it from stdin maybe? |
| 03:25.52 | *** join/#asterisk DrRighteous (n=DrRighte@ool-457843d1.dyn.optonline.net) |
| 03:25.55 | docelmo | basically I am making my own mail2fax gateway |
| 03:26.04 | Juggie | look at the source code for more |
| 03:26.06 | Juggie | find out how they do it. |
| 03:26.12 | Juggie | but that doesnt mean php will be able to do it. |
| 03:26.13 | docelmo | I cant find any.. |
| 03:26.20 | docelmo | yes it can do stdin |
| 03:26.33 | JT | almost anything can do stdin |
| 03:26.41 | Juggie | yeah i know php can, but i dont know if cat blah | php -q myphp.php |
| 03:26.50 | Juggie | i dontk now if the 'cat blah' will end up in stdin or not |
| 03:27.09 | JT | there must be a way to do so |
| 03:27.11 | docelmo | ah.. |
| 03:27.14 | JT | not familiar with php |
| 03:28.28 | docelmo | I have a php guru on staff.. Im buggin him right now |
| 03:28.35 | docelmo | thought someone here might be able to throw me a bone |
| 03:29.12 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:29.13 | Juggie | your only chance is if you can read of stdin probally. |
| 03:29.17 | Juggie | that would be how i'd try. |
| 03:29.38 | docelmo | ok thanks! |
| 03:29.45 | Juggie | or alternatively you could pipe to a temp file first, then use php to read that file. |
| 03:30.49 | *** join/#asterisk _cleric_ (n=dacleric@p54822565.dip0.t-ipconnect.de) |
| 03:40.57 | *** join/#asterisk stuq (n=stuq@cpe-24-161-103-133.hvc.res.rr.com) |
| 03:42.03 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 03:43.26 | Juggie | mog, fix appdial :P |
| 03:43.52 | Qwell | mog, fix mythtv |
| 03:44.01 | Juggie | mog, buy me jack in the box. |
| 03:44.23 | [TK]D-Fender | "Error in reality.sys, press any key to reboot Universe..." |
| 03:44.25 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:44.27 | Qwell | ! |
| 03:44.41 | Qwell | I get to go to JitB next month :P |
| 03:44.46 | mog | lol |
| 03:44.50 | mog | whats wrong with mythtv |
| 03:44.57 | Qwell | it doesn't record, heh |
| 03:45.15 | Qwell | unless I'm sitting there watching it - then it records fine |
| 03:47.50 | nvicf | is there a way to share an E1 line between asterisk and a digital panasonic? |
| 03:47.58 | blitzrage | zup |
| 03:49.51 | JT | nvicf: not really, best to act as a gateway between the telco line and the pabx |
| 03:50.09 | JT | in theory you could but an add-drop multiplexer in the middle, but yeah... :P |
| 03:50.38 | nvicf | add drop multiplexer?what's that? |
| 03:51.30 | JT | a piece of telco hardware, you probably don't need to worry about that option |
| 03:51.50 | JT | allows you to take out or insert timeslots/channels |
| 03:51.59 | nvicf | so how can I do then?the E1 enters the digium and from that I should go to the panasonic? |
| 03:52.35 | *** join/#asterisk bmg505 (n=leon@c1-153-8.rndf.isadsl.co.za) |
| 03:53.10 | JT | yeah, so a 2 port PRI card |
| 03:54.58 | Un1x | OKay, is there a way i can setup other secondary security measures for disa other then the password? |
| 03:55.12 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 03:57.32 | JT | yes |
| 03:57.42 | JT | you've got a whole dialplan at your disposal |
| 03:57.56 | JT | you can do whatever you like before letting someone get to the DISA application |
| 04:00.51 | mceGEEK | what is regcontext used for? |
| 04:01.12 | *** join/#asterisk kashmish_ (n=kashmish@m1.ince.net) |
| 04:01.55 | mceGEEK | brb |
| 04:02.34 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 04:03.25 | *** part/#asterisk kashmish_ (n=kashmish@m1.ince.net) |
| 04:08.44 | nvicf | JT: any digium 2 port pri card works to act as a gw?do I need to configure something else? |
| 04:11.10 | Kumbang | if im going to use a T1 gateway, what do you guys prefer to connect to *? |
| 04:11.44 | *** join/#asterisk tim27 (n=tim27@97-70.dr.cgocable.ca) |
| 04:11.46 | JT | nvicf: one will need to be set as pri_net the other as pri_cpe |
| 04:11.55 | JT | it's mainly a software issue |
| 04:13.13 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 04:13.33 | joelsolanki | dlynes_laptop: hello |
| 04:18.07 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 04:20.07 | dlynes_laptop | joelsolanki: goodbye |
| 04:20.38 | *** join/#asterisk TheCops (n=henri@got.securebinary.com) |
| 04:20.44 | TheCops | Hi |
| 04:21.33 | *** join/#asterisk mceGEEK (n=mceGEEK@70-89-196-165-inkleaf-mn.hfc.comcastbusiness.net) |
| 04:21.37 | TheCops | Someone know how to change the incoming caller ID "asterisk" when the caller ID is unknown? |
| 04:22.01 | rob0 | TheCops are here ... RUN! |
| 04:22.04 | TheCops | :p |
| 04:22.12 | rob0 | Depends what kind of channel of course. |
| 04:22.29 | rob0 | well, then there's SetCallerID() |
| 04:22.31 | TheCops | zap <-> SIP |
| 04:23.07 | JT | rob0: that's deprecated |
| 04:23.16 | TheCops | Set(Caller(num)=) |
| 04:23.19 | TheCops | Callerid |
| 04:23.20 | TheCops | sorry |
| 04:23.45 | TheCops | but, how can I set it if Bell is not sending it |
| 04:24.15 | russellb | TheCops: it's near the top of chan_sip.c |
| 04:24.21 | russellb | just search for "asterisk" |
| 04:24.22 | TheCops | good |
| 04:24.25 | TheCops | thanks russellb |
| 04:24.28 | russellb | you're welcome |
| 04:24.29 | rob0 | On my Zap FXO and FXS lines I set a default caller ID for each channel. |
| 04:24.58 | TheCops | #define DEFAULT_CALLERID "asterisk" |
| 04:25.00 | rob0 | Since I don't have caller ID service on the POTS line (!) that's what I see for FXO calls. |
| 04:25.02 | TheCops | perfect |
| 04:25.03 | russellb | TheCops: that's it |
| 04:25.44 | TheCops | russellb, does it is accepting accent (french) like é ? |
| 04:25.53 | TheCops | I'll test it anyway hehe |
| 04:26.14 | russellb | TheCops: i doubt it will work, to be honest |
| 04:26.21 | russellb | you can try, though. :) |
| 04:26.49 | mceGEEK | if i use register => phone:auth:secret@provider.com the extensions context s: rings and if i say register => phone:auth:secret@provider.com/localextension .. the call goes to my providers voicemail .. any ideas? |
| 04:29.22 | TheCops | russellb wow this is pretty well, all default value of the config file are there, useful. I'm not really a coder but I understand hehe that's the important part. |
| 04:29.58 | *** join/#asterisk wundaboy (n=asdf@c-24-21-109-26.hsd1.or.comcast.net) |
| 04:32.15 | russellb | TheCops: ;) |
| 04:32.19 | russellb | we're not *that* evil |
| 04:32.32 | partition | or... are we? |
| 04:32.36 | russellb | partition: !!! |
| 04:32.37 | *** join/#asterisk Entrophy (n=ben_Da_h@static24-72-82-139.regina.accesscomm.ca) |
| 04:32.38 | Entrophy | this is amazing! --> http://knightsdivine.net/weapon.php?uid=y20w75 |
| 04:32.40 | *** part/#asterisk Entrophy (n=ben_Da_h@static24-72-82-139.regina.accesscomm.ca) |
| 04:32.45 | partition | russellb: who are YOU! |
| 04:32.59 | TheCops | seem to be a asterisk coder |
| 04:33.14 | TheCops | russellb, you did chan_sip module? |
| 04:33.39 | russellb | i've made small changes ... but I did not write it. |
| 04:33.45 | TheCops | ok |
| 04:34.14 | russellb | i review a lot more of other people's code than actually write code myself |
| 04:34.30 | russellb | that's just what ends up being needed the most ... |
| 04:34.39 | partition | russellb: everyone's coming to get me! |
| 04:34.40 | russellb | i suppose i end up making changes to contributed changes |
| 04:34.42 | russellb | partition: eep |
| 04:34.48 | partition | hear the voices in my head... |
| 04:35.10 | partition | eeeeeeeep |
| 04:37.10 | Qwell | fdisk /dev/partition |
| 04:37.11 | Qwell | d |
| 04:37.11 | Qwell | 1 |
| 04:37.11 | Qwell | w |
| 04:37.21 | partition | uh oh |
| 04:37.54 | Supaplex | it's rude to point |
| 04:39.29 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 04:40.39 | rob0 | So, since symlink points to russellb ... who's the rude one? |
| 04:41.40 | *** join/#asterisk JamesDotCom (i=jamesdot@creep.bur.st) |
| 04:43.45 | *** part/#asterisk james (i=jamesdot@creep.bur.st) |
| 04:52.02 | *** join/#asterisk Mark55505 (n=Mark@c-68-62-100-66.hsd1.mi.comcast.net) |
| 04:57.37 | *** join/#asterisk lwizardl (n=1@69.51.144.65) |
| 04:57.40 | lwizardl | hi |
| 05:00.11 | lwizardl | I currently use vonage and want to have my own voip phone system (home use). and was told about asterisk. will any pc work as the asterisk pbx? |
| 05:01.20 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 05:02.08 | *** join/#asterisk peter21 (n=Peter@203.6.132.1) |
| 05:05.06 | *** join/#asterisk hterag (n=chatzill@gw.irrational.com.au) |
| 05:05.27 | peter21 | kb |
| 05:06.19 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:07.01 | hterag | G'day all |
| 05:08.33 | hterag | <PROTECTED> |
| 05:10.41 | joelsolanki | anybody using sangoma |
| 05:10.49 | joelsolanki | i m having problem with configuring sangoma |
| 05:11.02 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 05:11.08 | *** part/#asterisk lwizardl (n=1@69.51.144.65) |
| 05:11.12 | leprechau | i've used several sangoma cards |
| 05:11.30 | leprechau | never for VoIP though...just data connections :/ |
| 05:11.31 | joelsolanki | i have just bought sangoma A200 card with 8 ports |
| 05:12.00 | joelsolanki | i have installed wanpipe sucessfully and recompiled zaptel sucessfully. |
| 05:12.14 | joelsolanki | sangoma seems to be installed corectly. |
| 05:12.17 | leprechau | what's the problem? |
| 05:12.23 | leprechau | is the wanpipe coming up? |
| 05:12.23 | joelsolanki | but how do i configure it. |
| 05:12.26 | leprechau | and reporting status? |
| 05:12.44 | leprechau | well there is a configure directory /etc/wanpipe by default i believe |
| 05:12.57 | joelsolanki | yes it is thre |
| 05:13.06 | leprechau | did you create a config file? |
| 05:13.15 | leprechau | or use the wizard to make one? |
| 05:13.30 | leprechau | the install source comes with a config generator |
| 05:13.31 | joelsolanki | which wizard |
| 05:13.40 | leprechau | check out the README |
| 05:13.49 | leprechau | it's just an interactive shell script type deal |
| 05:13.53 | joelsolanki | i tried with wancfg zaptel but it gives error |
| 05:14.01 | leprechau | what error? |
| 05:14.07 | joelsolanki | let me give u |
| 05:14.43 | joelsolanki | card detected is AFT -A200-SH |
| 05:14.52 | joelsolanki | wancfg: Error in File: menu_hardware_probe.cpp, Function: run(), Line: 174. Text: |
| 05:14.59 | joelsolanki | Failed to get Card Type from selected card line!! line: AFT-A200-SH SLOT=8 BUS=2 IRQ=193 CPU=A PORT=PRI |
| 05:15.51 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 05:17.37 | joelsolanki | any hints / |
| 05:17.37 | joelsolanki | ? |
| 05:21.11 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 05:27.46 | *** join/#asterisk aadilismail (n=aadilism@2-237-154-202.wol.net.pk) |
| 05:27.54 | aadilismail | hi |
| 05:28.10 | saam | hi |
| 05:28.38 | xain | hi |
| 05:29.58 | germy_g | hi |
| 05:30.41 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 05:30.41 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:30.41 | xain | ok |
| 05:35.07 | *** part/#asterisk denon (i=denon@sassinak.net) |
| 05:36.00 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 05:36.00 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:44.32 | *** join/#asterisk dlucas (n=root@Broadband-Dynamic-Western354.connect.com.fj) |
| 05:44.56 | dlucas | ok, lots of people here |
| 05:45.12 | dlucas | anyone able to answer a question on dtmf timing on outgoing calls? |
| 05:45.30 | monsted | shush, we're sleeping |
| 05:45.36 | dlucas | asterisk 1.4-beta3 |
| 05:45.47 | dlucas | you might be sleeping ... ;) |
| 05:46.02 | dlucas | still daylight here! |
| 05:48.30 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 05:48.32 | dlucas | no takers? |
| 05:49.34 | Sed[PCT] | anyone here ever configure a 7912G with sip.. when it has a sccp image on it? |
| 05:50.21 | dlucas | not I |
| 05:50.45 | dlucas | still looking for help on the DTMF tones outgoing while on call through ZAP |
| 05:53.10 | rob0 | dlucas: You probably should ask it, maybe someone who knows will see it. |
| 05:53.27 | rob0 | Fiji? |
| 05:56.10 | *** join/#asterisk Math` (n=privmath@modemcable186.59-131-66.mc.videotron.ca) |
| 05:57.17 | *** join/#asterisk dlucas (n=root@Broadband-Dynamic-Western354.connect.com.fj) |
| 05:57.31 | dlucas | sorry I missed the last few minutes |
| 05:58.10 | dlucas | rob, did you respond ... I just saw something before my comp crashed |
| 06:00.28 | dlucas | hullo? |
| 06:09.15 | *** part/#asterisk dlucas (n=root@Broadband-Dynamic-Western354.connect.com.fj) |
| 06:10.40 | *** join/#asterisk peter21 (n=Peter@203.6.132.1) |
| 06:26.02 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 06:29.51 | *** join/#asterisk yardB (n=yardie@c-68-44-44-42.hsd1.nj.comcast.net) |
| 06:33.23 | peter21 | anyone online ? |
| 06:35.29 | yardB | i am peter ..half asleep anyway |
| 06:38.15 | yardB | peter do u have a sip phone? |
| 06:42.13 | *** join/#asterisk ComPuTeR (n=DeviL_Of@85.108.151.230) |
| 06:44.54 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:50.11 | joelsolanki | any body using sangoma |
| 06:50.23 | joelsolanki | i have problem with it configuring |
| 06:51.38 | joelsolanki | dlynes_laptop: u tre |
| 06:51.39 | yardB | not i |
| 06:51.54 | joelsolanki | yardB: do u use sangoma |
| 06:52.15 | yardB | no, what is it? |
| 06:52.48 | joelsolanki | oh ok. it is pstn card. |
| 06:53.05 | yardB | sorry |
| 06:53.14 | joelsolanki | :( |
| 06:53.15 | yardB | do u have a sip phone |
| 06:53.31 | joelsolanki | no not right now. |
| 06:53.35 | joelsolanki | have it at office |
| 06:54.08 | yardB | ok, i needed to see if mine work by having someone call me or me calling someone |
| 06:54.37 | yardB | <PROTECTED> |
| 06:54.42 | joelsolanki | oh ok. right now i dont have else |
| 06:54.43 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 06:54.46 | joelsolanki | no i m not. |
| 06:54.53 | joelsolanki | brb |
| 06:55.09 | yardB | ok |
| 06:56.08 | *** join/#asterisk spunz (n=spunz@h081217096236.dyn.cm.kabsi.at) |
| 07:01.57 | *** join/#asterisk spunz_ (n=spunz@h081217096236.dyn.cm.kabsi.at) |
| 07:02.27 | *** join/#asterisk Un1x|laptop (i=Sean@CPE0051c5534c1f-CM001371145406.cpe.net.cable.rogers.com) |
| 07:02.36 | Un1x|laptop | ~forward |
| 07:04.12 | Un1x|laptop | is there a way to forwards incoming calls to a phone number or something |
| 07:04.37 | DrCron | yup |
| 07:04.52 | DrCron | it takes another chan though, and an outgoing line/service |
| 07:05.43 | Un1x|laptop | DrCron what is the command or how is it done so i can read up |
| 07:06.08 | Un1x|laptop | and by the way when im on my phone and someone else trys to call asterisk denies the call because call waiting isn't set, whats is the cmd for that as well |
| 07:06.32 | Un1x|laptop | ~cmd |
| 07:06.33 | jbot | somebody said cmd was the ultimate Microsoft IIS remote administration tool. |
| 07:06.38 | Un1x|laptop | :O |
| 07:06.41 | Un1x|laptop | lol |
| 07:06.52 | Un1x|laptop | Answer() |
| 07:07.34 | hads | show application Dial |
| 07:07.43 | Un1x|laptop | hads is that for forwarding? |
| 07:09.03 | Un1x|laptop | sigh, no help in here at all lol |
| 07:09.10 | dlynes_laptop | ? |
| 07:09.14 | Un1x|laptop | dynes wusup man |
| 07:09.19 | Un1x|laptop | dlynes_laptop |
| 07:09.20 | dlynes_laptop | not much |
| 07:09.26 | Un1x|laptop | long time since i've spoken to you |
| 07:09.35 | dlynes_laptop | yeah, maybe |
| 07:09.37 | dlynes_laptop | been busy with work |
| 07:09.56 | Un1x|laptop | hey dlynes_laptop can you help me out a bit here trying to find out how call waiting works on asterisk when im on the phone no one else can call me |
| 07:10.02 | Un1x|laptop | asterisk denies the call :( |
| 07:10.20 | DrCron | Un1x, you know how to dial a call from asterisk? |
| 07:10.31 | Un1x|laptop | from console you mean? |
| 07:10.36 | DrCron | you take your incoming call, and you dial out |
| 07:10.45 | DrCron | in the extensions.conf |
| 07:10.52 | DrCron | just as an example |
| 07:10.53 | dlynes_laptop | Un1x|laptop: you must have a call-limit of 1 on that line |
| 07:11.25 | Un1x|laptop | hmm dlynes how do i fix it so i can get call waiting so when someone calls my phone knows there is a call waiting so i can switch |
| 07:11.40 | dlynes_laptop | Un1x|laptop: or you might have an incoming-limit of 1, or outgoing-limit of 1, or something equally incompatible |
| 07:11.55 | dlynes_laptop | Un1x|laptop: what technology is the line using? |
| 07:12.05 | *** join/#asterisk LoneShadow (n=duh@c-67-180-81-124.hsd1.ca.comcast.net) |
| 07:12.15 | Un1x|laptop | dlynes.. technology? |
| 07:12.25 | Un1x|laptop | what do you mean by that? as in encoding? |
| 07:12.37 | dlynes_laptop | Un1x|laptop: zaptel, sip, skinny, h323, iax2, iax, ... |
| 07:12.47 | Un1x|laptop | oh zaptel |
| 07:14.12 | dlynes_laptop | Un1x|laptop: make sure you have callwaiting=yes in your zapata.conf file |
| 07:15.14 | dlynes_laptop | Un1x|laptop: after it's added, do an unload chan_zap.so and then a load chan_zap.so |
| 07:15.33 | Un1x|laptop | dlynes_laptop where exactly should it be http://pastebin.ca/271501 |
| 07:15.39 | Un1x|laptop | at the top right? |
| 07:15.46 | Un1x|laptop | under the other stuff |
| 07:17.06 | dlynes_laptop | Un1x|laptop: under each of the lines that say fxo_ks, add it |
| 07:17.21 | Un1x|laptop | same with fxs_ks? |
| 07:17.30 | dlynes_laptop | No |
| 07:17.36 | dlynes_laptop | It's only an option for fxs ports |
| 07:17.45 | dlynes_laptop | It's not an option for the phone lines |
| 07:18.01 | Un1x|laptop | so its not a option if i dont use a incoming pstn line then... |
| 07:18.12 | Un1x|laptop | i thought maybe it was if i had a sip account with 2 channels |
| 07:18.14 | dlynes_laptop | It's not an option if you don't use analog phones |
| 07:18.23 | Un1x|laptop | im using a analog phone... |
| 07:18.25 | dlynes_laptop | Un1x|laptop: No, it's an option for that, too |
| 07:18.29 | Un1x|laptop | but phones plug into a FXS port |
| 07:18.40 | Un1x|laptop | ok so add it under fxs_ks too then correct? |
| 07:18.52 | dlynes_laptop | Un1x|laptop: what part of no, did you not understand? |
| 07:19.07 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 07:19.19 | JT | no |
| 07:19.22 | dlynes_laptop | Un1x|laptop: fxs_ks is for fxo ports; fxo_ks is for fxs parts |
| 07:19.25 | JT | fko_ks for FXS ports |
| 07:19.27 | dlynes_laptop | s/parts/ports/ |
| 07:19.38 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-231-237.dc.res.rr.com) |
| 07:19.42 | JT | how many times have we said to read the book, Un1x|laptop ? |
| 07:19.56 | JT | every single question you've asked is easily answered by the book |
| 07:19.59 | Un1x|laptop | dude i am reading it just got it today :P |
| 07:20.03 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:20.09 | JT | you can use it as a reference as well |
| 07:20.12 | JT | just text search it |
| 07:20.12 | Un1x|laptop | so fxs_ks is fxo and fxo_ks is for fxs now i got ya thanks |
| 07:20.33 | Un1x|laptop | ok adding callwaiting=yes under fxo_ks |
| 07:20.33 | dlynes_laptop | Un1x|laptop: Yes. fxo ports use fxs signalling and fxs ports use fxo signalling |
| 07:20.40 | dlynes_laptop | Un1x|laptop: now you got it |
| 07:20.54 | hads | Don't forget, options are inherited in zapata.conf |
| 07:21.21 | Un1x|laptop | heh i always though FXO uses FXO_KS |
| 07:21.28 | Un1x|laptop | i dind't know they used each other :P |
| 07:22.18 | *** join/#asterisk psiforce (i=psiforce@marksnb.eng.unimelb.edu.au) |
| 07:22.58 | DrCron | Un1x, so, do you want to have your phone company forward the phone? or do you want to do it |
| 07:23.29 | Un1x|laptop | i want to do it |
| 07:24.22 | Un1x|laptop | basicly, about forwarding its simple as this.. i want to have someone call in by dailing my DID wich reaches my server from there i want to put a command into the answering context that answers plays a file with the playback cmd and then i want to forward it to another number |
| 07:24.31 | Un1x|laptop | but i dont know the command for forward |
| 07:24.54 | DrCron | dial() |
| 07:25.05 | DrCron | how many lines do you have? |
| 07:25.11 | Un1x|laptop | will it bridge the calls automaticly? |
| 07:25.19 | Un1x|laptop | well the 2 lines |
| 07:25.24 | Un1x|laptop | each account comes with 2 channels |
| 07:25.31 | Un1x|laptop | the one channel will be used when the person calls |
| 07:25.37 | Un1x|laptop | and then one more when the call is forwarded |
| 07:25.51 | Un1x|laptop | or if i can use the conference command and have asterisk phone the person |
| 07:25.58 | Un1x|laptop | and him automaticly enter the conference |
| 07:26.05 | Un1x|laptop | that would be same too but prefer forward |
| 07:26.43 | psiforce | hi all, I have a dual xeon 3ghz with a 4 port pri card installed (sangoma) and terminating 90 calls via the pstn (40 of which are g729), but asterisk is maxing out the cpu at 99% (or so top tells me). People are then complaining about drop outs.. Suggestions? Digium say you can terminate 150 |
| 07:26.53 | DrCron | ok, what exactly do you mean by forward |
| 07:27.26 | DrCron | you want someone to call in, and for it to connect to a diffrent number, right? |
| 07:27.26 | psiforce | calls. perhaps this is because asterisk is only using one cpu? so is there away to get asterisk to use both? |
| 07:27.30 | Un1x|laptop | DrCron corrrect |
| 07:28.07 | Un1x|laptop | psiforce did you compile with SMP support ur kernel |
| 07:28.24 | Un1x|laptop | oh wait nvm |
| 07:28.24 | *** join/#asterisk DaeJeon (i=Singh@124.62.150.38) |
| 07:30.39 | DaeJeon | Hello guys, I recently installed Solaris 10 on x86(intel)machine. Now, I want to install an asterisk. can I get some help? |
| 07:30.57 | DaeJeon | is it possible? |
| 07:31.01 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:31.04 | psiforce | uni1x|laptop: yep, I assume so as "cat /proc/cpuinfo" states that there are 2 cpus |
| 07:31.06 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:31.31 | DrCron | DaeJeon, i think so |
| 07:31.53 | DaeJeon | DrCron: can I get link? |
| 07:32.00 | DaeJeon | any documentation? |
| 07:32.29 | DrCron | just a standard install |
| 07:32.59 | DaeJeon | u mean, whatever we do on linux? |
| 07:33.13 | DrCron | yup |
| 07:33.34 | DrCron | solaris is very much a *nix system |
| 07:33.38 | dlynes_laptop | DaeJeon: might want to make sure you've got a gnu development environment set up, though |
| 07:33.57 | dlynes_laptop | DaeJeon: and have some sane CFLAGS and LDFLAGS environment variables |
| 07:34.09 | joelsolanki | dlynes_laptop: u r up again |
| 07:34.10 | joelsolanki | hehe |
| 07:34.42 | DaeJeon | i am new to solaris |
| 07:35.41 | dlynes_laptop | DaeJeon: good way to get your feet wet :0 |
| 07:35.51 | dlynes_laptop | DaeJeon: wade in up to your shoulders :) |
| 07:35.56 | DaeJeon | is there anyweb site i could read more how to install asterisk on soloris 10 |
| 07:36.27 | *** join/#asterisk lorinc (n=ang@213.178.121.201) |
| 07:36.43 | dlynes_laptop | DaeJeon: nope |
| 07:36.52 | dlynes_laptop | DaeJeon: and certainly not on intel |
| 07:37.08 | dlynes_laptop | DaeJeon: anyone that I know of that's running Asterisk on Solaris 10 is running it on a SPARC |
| 07:37.20 | DaeJeon | got it man |
| 07:37.29 | dlynes_laptop | DaeJeon: are you wanting to use the zaptel stuff as well? |
| 07:37.35 | DaeJeon | no no |
| 07:37.38 | DaeJeon | nothing |
| 07:37.40 | DaeJeon | just sip |
| 07:37.46 | dlynes_laptop | Ok, I was going to say, that would definitely be a no-starter |
| 07:38.03 | DaeJeon | pc to pc |
| 07:38.09 | *** join/#asterisk mithraen_ (n=mithraen@87.228.121.245) |
| 07:38.13 | dlynes_laptop | spandsp has a few issues building on solaris, too |
| 07:38.26 | dlynes_laptop | but it has been tested on 64-bit |
| 07:38.33 | DaeJeon | amd? |
| 07:38.36 | *** join/#asterisk X-Rob (n=rob-x@dsl-124-150-115-171.vic.westnet.com.au) |
| 07:38.43 | dlynes_laptop | Probably |
| 07:38.53 | dlynes_laptop | It hasn't been tested on sparc yet |
| 07:38.58 | dlynes_laptop | I'm in the process of doing that, myself |
| 07:39.07 | DaeJeon | well i found one site |
| 07:39.18 | DaeJeon | kind of benchmark |
| 07:39.26 | dlynes_laptop | Got a url? |
| 07:39.37 | DaeJeon | yes |
| 07:39.41 | stephane_ | jour |
| 07:39.44 | DaeJeon | hold on plz |
| 07:41.10 | DaeJeon | http://www.thrallingpenguin.com/articles/asterisk-solaris.htm |
| 07:41.41 | DaeJeon | is it true? |
| 07:42.05 | DaeJeon | thats why I wanted to play with solaris |
| 07:43.17 | dlynes_laptop | DaeJeon: if you notice, the solaris machine is running on a sunfire |
| 07:43.26 | DaeJeon | yes |
| 07:43.31 | DaeJeon | I saw that |
| 07:43.36 | dlynes_laptop | DaeJeon: Solaris is extremely good for telecommunications on UltraSPARC |
| 07:44.00 | dlynes_laptop | DaeJeon: not necessarily on Intel, although they've been getting really good performance on Opteron-based smp systems |
| 07:44.33 | DaeJeon | actually I want to learn more abt it |
| 07:44.34 | dlynes_laptop | DaeJeon: traditionally, telco software has run on Solaris on UltraSPARC and SPARC |
| 07:45.04 | DaeJeon | i do not have sun hardwares |
| 07:45.16 | DaeJeon | right now |
| 07:45.36 | dlynes_laptop | DaeJeon: You can achieve more simultaneous calls on a Solaris on UltraSPARC than you can on Linux running on Intel chips |
| 07:46.03 | DaeJeon | is it really true? |
| 07:47.04 | dlynes_laptop | Yes |
| 07:47.15 | DaeJeon | sun is providing systems for 60 days on trail |
| 07:47.25 | DaeJeon | i ordered one |
| 07:47.33 | dlynes_laptop | If you want some benchmark results, talk to qwell[] some time about how many simultaneous calls he's able to get on his sunfire |
| 07:47.45 | Un1x|laptop | hey anyone know did providers from asian countrys such as japan |
| 07:47.58 | *** join/#asterisk unfo (n=QSECADM@CPE0004e2d9f884-CM0013718690da.cpe.net.cable.rogers.com) |
| 07:48.01 | *** join/#asterisk RichiH (i=richih@freenode/staff/richih) |
| 07:48.29 | RichiH | unfo: thanks |
| 07:48.38 | *** part/#asterisk RichiH (i=richih@freenode/staff/richih) |
| 07:48.46 | benjk | Un1x, how many DIDs do you need? |
| 07:48.50 | *** part/#asterisk unfo (n=QSECADM@CPE0004e2d9f884-CM0013718690da.cpe.net.cable.rogers.com) |
| 07:49.38 | Un1x|laptop | benjk one probably depending on this clients request... |
| 07:49.42 | Un1x|laptop | he only wants one yea :( |
| 07:50.10 | benjk | one is going to be expenisve, you can only get them in blocks of 100 |
| 07:50.16 | Un1x|laptop | Really from where? |
| 07:50.26 | Un1x|laptop | call up the phone exchanges in that country or something? |
| 07:50.33 | benjk | and Japanese companies are extreeeeeeeeemely suspicous of foreigners who want a Japanese phone number |
| 07:51.10 | benjk | I can get you DIDs for Tokyo |
| 07:51.10 | Un1x|laptop | but only in block of 100's huh |
| 07:51.16 | benjk | but a single one will be expensive cause I have to order it retail on our own ISDN circuit |
| 07:51.40 | Un1x|laptop | ~ISDN |
| 07:51.44 | jbot | from memory, isdn is (Integrated Services Digital Network) This is a digital line that is often used to connect to the Internet. It generally come in two flavors: one is a 56 Kbps version, which in actuality only uses half of the ISDN line's bandwidth; the other is the 128 Kbps version, which uses both the 56 Kbps channels on the line. However, that's only 112 ... |
| 07:52.02 | benjk | jbot is talking garbage |
| 07:52.07 | Un1x|laptop | heh i figured |
| 07:52.12 | benjk | ISDN is no longer of any interest for internet dialup |
| 07:52.36 | hads | That should be moved to BRI |
| 07:52.36 | benjk | nobody in their right mind uses ISDN for internet dialup in places where you have other options |
| 07:53.02 | Un1x|laptop | heh |
| 07:53.08 | benjk | also the description doesn't take into consideration that ISDN = BRI + PRI |
| 07:53.08 | Un1x|laptop | i would die and have a heart attack |
| 07:53.11 | Un1x|laptop | if i was on dailup |
| 07:53.23 | Un1x|laptop | i can barely handle free wlans that are around 50kB/s |
| 07:53.37 | benjk | well, ISDN BRI dialup is fast and not too slow as you get 128K full duplex |
| 07:53.42 | benjk | but its expensive |
| 07:53.55 | Un1x|laptop | benjk is a ISDN just like a asterisk/openbx server with bunch o FXO cards? |
| 07:54.17 | benjk | in our case ISDN means a digital telephony circuit |
| 07:54.54 | nibbler_de | Un1x|laptop: think of isdn like a digital version of your phone line |
| 07:55.19 | Un1x|laptop | so when you buy lets say block of 1000 numbers from a specific exchange do they transfer those over internet lines , or they sent to the ISDN via traditional copper lines |
| 07:55.27 | Un1x|laptop | eh |
| 07:55.27 | benjk | so you would get a DID number, calls coming in on which would arrive via ISDN on our server and then forwarded via SIP or IAX to you |
| 07:55.31 | JT | jbot is also talking rubbish because who the hell has 56kbit/s isdn anymore |
| 07:55.40 | JT | most places on the planet are 64kbit/s per B channel |
| 07:55.49 | benjk | Japanese telcos deliver DIDs via ISDN voice |
| 07:56.08 | Un1x|laptop | is that the case wit all asian countrys or just japan? |
| 07:56.10 | benjk | no SIP, no H323, no MGCP, no IAX, not other VOIP |
| 07:56.25 | nibbler_de | JT: was that 56kbit/s isdn anywhere deployed outside of the us? |
| 07:56.33 | benjk | for VOIP there are 050 DIDs, special numbers for VOIP service |
| 07:56.42 | benjk | thos are only avaolable to Japanese residents |
| 07:56.56 | JT | nibbler_de: maybe us |
| 07:57.00 | JT | doubful anywhere else |
| 07:57.02 | nibbler_de | Un1x|laptop: in germany it's very common that you get an isdn line where your numbers terminate |
| 07:57.02 | JT | i mean canada |
| 07:57.17 | Un1x|laptop | :o |
| 07:57.18 | benjk | so you have two options to get Japanese DIDs delivered over VoIP to you ... |
| 07:57.22 | nibbler_de | Un1x|laptop: almost nobody here has analog phone lines |
| 07:57.25 | JT | bri is common in most worst world countried outside north america, Un1x|laptop |
| 07:57.34 | JT | it's available in north america, just very expensive |
| 07:57.39 | JT | gar |
| 07:57.43 | JT | my engrish is bad |
| 07:57.46 | Un1x|laptop | so what kinda equipment is needed so you can terminate for other people |
| 07:57.48 | benjk | 1) you can colocate a server in Japan, have an ISDN circuit delivered by a telco to the colo and get your block of 100 numbers |
| 07:57.59 | JT | bri is common in most first world countries outside north america, Un1x|laptop |
| 07:58.04 | benjk | then deliver the calls over VOIP yourself from your own colocated server |
| 07:58.05 | JT | what i meant to say |
| 07:58.07 | Un1x|laptop | oh ok so ISDN is Circut connected to the server :) |
| 07:58.09 | DaeJeon | Unix/laptop: try http://www.didww.com/ |
| 07:58.17 | nibbler_de | Un1x|laptop: mostly BRI for end users (featuring 2 lines of 64kbit/s each and a signalling channel of 16k) or PRI for larger enterprises that has 31 64k lines, one of them for signalling and one for sync |
| 07:58.22 | DaeJeon | u might good deal |
| 07:58.23 | Un1x|laptop | Daejeon i have one with them cant have another for disa |
| 07:58.30 | benjk | 2) find a provider who does this for you, mostly wholesale though |
| 07:58.45 | Un1x|laptop | Ah |
| 07:59.01 | benjk | nibbler_de only 30 voice channels |
| 07:59.04 | benjk | with E1 |
| 07:59.15 | Un1x|laptop | i was thinking of terminating in countrys where voip is still emerging because once VOIP is really stable like how pstn is at the moment it will be superseded by voip |
| 07:59.20 | Un1x|laptop | just like how isdn was by DSL |
| 07:59.23 | nibbler_de | ah, ok - i'm always unsure about the exact numbers there |
| 07:59.28 | benjk | Japan uses J1 though which is T1 with different framing but also 23 voice channels like T1 |
| 07:59.48 | Un1x|laptop | hey benjk mind if i pm you for a sec? |
| 07:59.50 | JT | Un1x|laptop: illegal in a lot of those countries, btw |
| 08:00.06 | nibbler_de | with 30 we're talking about channelized E1, right? |
| 08:00.17 | Un1x|laptop | hmm yea its only illegal because local telcos pay the goverments money to force people to use local telcos |
| 08:00.24 | Un1x|laptop | rather then cut costs and ssave money via Voip |
| 08:00.28 | benjk | nibbler_de, in general E1 = 32 timeslots => 30 payload channels, 1 signaling channel, 1 slot used as a timing reference |
| 08:00.46 | JT | Un1x|laptop: often the government is the local telco |
| 08:00.53 | Un1x|laptop | yea or that :P |
| 08:00.56 | nibbler_de | a-ha! ;) ok. then i recall it rather correctly - was unsure about the timing reference |
| 08:01.04 | nibbler_de | benjk: is the timing reference mandatory? |
| 08:01.11 | Un1x|laptop | JT so what kinda of equipment do i need seeing how the ISDN Circut comes from the telco there must be other things i need no |
| 08:01.15 | benjk | its part of the protocol |
| 08:01.27 | benjk | you cant use that timeslot for anything else |
| 08:01.28 | JT | i've hearof the timeslot referenced as for framing purposes |
| 08:01.59 | benjk | well, you could, but that it wouldn't be compatible with the standard anymore |
| 08:02.46 | nibbler_de | benjk: ah, ok - that's why i wondered - on the systems i configured in the past you can set the channel where your signalling is taking place but not the timing reference |
| 08:03.57 | psiforce | anyone know why asterisk would only be using 1 cpu ? |
| 08:04.57 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 08:05.01 | DrCron | i dont know how asterisks threading works |
| 08:05.47 | *** join/#asterisk xtr-II (n=94752345@S0106000c41ed11e1.vf.shawcable.net) |
| 08:06.31 | Un1x|laptop | psiforce youd have to talk to the devs :P or someone who nows... |
| 08:06.47 | Un1x|laptop | benjk so what kinda equipment do you use for youre server? |
| 08:06.56 | Un1x|laptop | that the ISDN circuit connects too |
| 08:07.02 | *** join/#asterisk SimoAmi (n=SimoAmi@ip67-91-253-242.z253-91-67.customer.algx.net) |
| 08:07.26 | benjk | asterisk threading is not exactly a story of glamour and fame |
| 08:08.10 | benjk | Un1x, Sangoma for PRI and Billion on smaller BRI based ones |
| 08:08.35 | Un1x|laptop | o benjk btw |
| 08:08.38 | Un1x|laptop | i got the A200 |
| 08:08.43 | Un1x|laptop | from sangoma i forgot to tell you |
| 08:09.33 | benjk | A200 is analog, not ISDN |
| 08:09.37 | Un1x|laptop | yea i know |
| 08:09.40 | Un1x|laptop | its for home :) |
| 08:09.48 | benjk | fari enough |
| 08:09.51 | Un1x|laptop | for testing etc just to play around u know :) |
| 08:09.56 | Un1x|laptop | grasp the technology learn a bit |
| 08:10.02 | Un1x|laptop | then i will get the ISDN one :) |
| 08:10.09 | Un1x|laptop | benjk wich card do you use for ISDN |
| 08:10.09 | Un1x|laptop | ? |
| 08:10.13 | benjk | it depends on where you are |
| 08:10.23 | benjk | North America is not exactly very BRI friendly |
| 08:10.38 | Un1x|laptop | no if i get a ISDN i will use it in asia |
| 08:11.02 | hads | Are you in Asia? |
| 08:11.30 | Un1x|laptop | hads i will use it in asia when i go... |
| 08:11.35 | Un1x|laptop | wich will be few weeks probably |
| 08:11.41 | Un1x|laptop | i go often with dad |
| 08:11.47 | Un1x|laptop | he goes business meetings to tokyo |
| 08:11.54 | Un1x|laptop | i end up going elsewhere |
| 08:15.21 | benjk | you can't just go to Asia on a trip and plug in to some PRI or BRI somwhere |
| 08:15.28 | Un1x|laptop | i know that |
| 08:15.33 | Un1x|laptop | im just saying i go |
| 08:15.38 | benjk | you'd have to have a server colocated |
| 08:15.38 | Un1x|laptop | [3:10am] <hads> Are you in Asia? |
| 08:15.43 | Un1x|laptop | hes asked that :p |
| 08:16.08 | benjk | I can get you colocation in Tokyo, in the building where all the big telcos have their biggest pipes go through |
| 08:16.17 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 08:16.26 | Un1x|laptop | really? |
| 08:16.48 | SimoAmi | hi again |
| 08:16.57 | benjk | yes |
| 08:17.15 | Un1x|laptop | how much |
| 08:17.33 | SimoAmi | I have 2 broadvoice sip accounts that can't be registered |
| 08:17.38 | SimoAmi | can anyone look at the debug message and see if something is odd? |
| 08:17.39 | Un1x|laptop | lmao |
| 08:17.44 | Un1x|laptop | sigh |
| 08:18.00 | Un1x|laptop | benjk have a quick look at youre pm please. |
| 08:18.18 | *** join/#asterisk shinux__ (n=shinux@196.220.26.174) |
| 08:18.55 | DrCron | SimoAmi, um, pastebin em? |
| 08:20.05 | SimoAmi | yep, one sec |
| 08:22.52 | SimoAmi | this damn java ssh cannot copy text |
| 08:23.02 | SimoAmi | getting putty in a moment |
| 08:29.41 | SimoAmi | ok, finally |
| 08:29.48 | SimoAmi | here's the log |
| 08:29.50 | SimoAmi | http://pastebin.ca/271569 |
| 08:31.39 | DrCron | what does the err mesage say |
| 08:32.23 | SimoAmi | have you looked at the log I just pasted? |
| 08:32.32 | mendol | morning guys, CAUSE CODE : 50, no authority found, its becouse of wrong context (incoming calls got busy signal) |
| 08:32.59 | *** join/#asterisk zumbush (n=ztriver@62.209.179.131) |
| 08:34.31 | DrCron | um, unless i'm reading something totaly diffrent thats a log of exactly what your machine sent out |
| 08:35.06 | SimoAmi | yep |
| 08:35.31 | SimoAmi | after a "sip debug" command |
| 08:35.58 | SimoAmi | the status is unregistered |
| 08:36.34 | DrCron | have you tried running asterisk -rddvv then doing a reload? see if any err messages popup? |
| 08:37.34 | SimoAmi | let me try |
| 08:39.40 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 08:40.07 | SimoAmi | nothing that I could spot |
| 08:41.03 | mendol | CAUSE : No authority found, and my iax2 trunk registered just fine, just have no idea how to fix that error :-/ |
| 08:41.19 | badcfe | are there any techniques that could kill a sip call where incoming RTP stream is dead. as * apparently does not do this by itself. known "work-arounds"? |
| 08:42.16 | *** join/#asterisk reber (n=reber@cl-157.dub-01.ie.sixxs.net) |
| 08:42.32 | DrCron | SimoAmi, sip debug isnt going to help much for me |
| 08:43.10 | SimoAmi | ok, check this log |
| 08:43.11 | SimoAmi | http://pastebin.ca/271579 |
| 08:43.15 | *** join/#asterisk shinux__ (n=shinux@196.220.26.174) |
| 08:43.41 | SimoAmi | it's the result of "sip show registry" and "sip show peers" |
| 08:43.51 | DrCron | you have a firewall? |
| 08:44.17 | SimoAmi | a router with firewall |
| 08:44.42 | DrCron | anything between your asterisk server and the internet |
| 08:45.08 | DrCron | ie: are you sure packets incoming to port 5060 get to the asterisk server |
| 08:45.27 | SimoAmi | not sure, how can I? |
| 08:46.02 | SimoAmi | all I know is port forwarding is active on 5060-5080 on udp for the * server |
| 08:46.11 | SimoAmi | so is 10000-20000 |
| 08:47.17 | DrCron | i think its a NAT issue |
| 08:47.44 | SimoAmi | would you suggest trying DMZ ? |
| 08:48.03 | SimoAmi | for * server |
| 08:48.51 | *** join/#asterisk KermitTheFragger (n=ktf@118-197.bbned.dsl.internl.net) |
| 08:50.39 | DrCron | i have no idea how your router/firewall is set up |
| 08:50.48 | DrCron | try asking back here later |
| 08:50.52 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 08:51.10 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 08:51.17 | *** join/#asterisk Mark-Man22 (n=Mark@c-68-62-100-66.hsd1.mi.comcast.net) |
| 08:51.46 | SimoAmi | ok, thx |
| 08:52.38 | badcfe | are there any techniques that could kill a sip call where incoming RTP stream is dead. as * apparently does not do this by itself. known "work-arounds"? |
| 08:53.54 | badcfe | my problem is that a sip to sip bridge just hangs when one of the friends dies by network disconnection (no BYE). this is not good for the CDRs... |
| 08:55.13 | hads | How about rtptimeout |
| 08:55.13 | badcfe | and, no, i cannot do re-INVITE, wich would "fix" it, in this case. |
| 08:55.51 | badcfe | hads: does that live in rtp.conf? ill check my current config. |
| 08:56.13 | hads | <PROTECTED> |
| 08:56.37 | mosty | are the echo cancellation settings in zapata.conf for software or hardware echo cancellation? |
| 08:57.02 | hads | Both, sort of. |
| 08:57.35 | hads | echocancel=yes/no will turn on/off either the hardware or software EC depending on what you have. |
| 08:57.46 | hads | The other settings only apply to software EC. |
| 08:58.09 | mosty | hads: how can i verify that echo cancellation is being done entirely in hardware? |
| 08:58.51 | hads | mosty: I don't have a hardware EC system so I can't tell you for sure. |
| 08:59.48 | benjk | mosty, you can't |
| 09:02.36 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 09:07.21 | zumbush | In the asterisk dialplan.. I want to trigger an action when an incoming call is answered. Anyone how u catch the pickup of a call in an AGI-script ??? |
| 09:07.36 | zumbush | Anyone know* |
| 09:08.24 | zumbush | The CDR doesnt write to mysql until the call ended |
| 09:08.45 | mosty | because it doesn't know all the details until the call is ended. what exactly do you want to trigger? |
| 09:09.21 | zumbush | write to mysql the id of the call and the answering ext |
| 09:09.56 | EmleyMoor | I continually get the warning Dec 8 09:07:27 WARNING[2961]: db.c:67 dbinit: Unable to open Asterisk database - probably permissions related, and I intend to do some things shortly that need the database - what do I need to check permissions on? |
| 09:10.57 | zumbush | im writing to a custom table in mysql not the CDR |
| 09:11.21 | dlynes_laptop | Anyone on that uses sangoma cards? |
| 09:12.43 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 09:13.07 | dlynes_laptop | qdk: you use sangoma, right? |
| 09:14.05 | zumbush | want the person to enter a couple of digits on the phone, save these together with the extension that picked up the phone and retrieve these from a webapp (check what the person I'm talking to entered) |
| 09:15.35 | qdk | dlynes_laptop: no, not yet anyway. still only have digium in production and in stock. |
| 09:15.42 | *** join/#asterisk phonetalk (n=phonetal@host210-2-164-29.isb.dancom.net.pk) |
| 09:15.52 | dlynes_laptop | ah |
| 09:16.24 | dlynes_laptop | phonetalk: ceshia galee |
| 09:16.50 | qdk | dlynes_laptop: but i plan to use sangoma E1 PRI/SS7 for my new/own system. |
| 09:17.03 | dlynes_laptop | qdk: yeah...they're great little cards |
| 09:17.11 | dlynes_laptop | qdk: just a huge pain in the ass to configure |
| 09:17.19 | mosty | EmleyMoor: are you using debian? |
| 09:17.31 | EmleyMoor | Yes |
| 09:17.38 | qdk | dlynes_laptop: that pain at that time. .-) |
| 09:17.45 | dlynes_laptop | qdk: they've got some new cards now, too |
| 09:17.51 | dlynes_laptop | qdk: with a db25 connector |
| 09:18.07 | dlynes_laptop | qdk: why they're not using an amphenol connector is beyond me, though |
| 09:18.22 | mosty | EmleyMoor: it's because asterisk doesn't run as root in debian. try chown'ing /var/lib/asterisk to asterisk:asterisk |
| 09:18.59 | qdk | dlynes_laptop: oh, so I dont have to convert to RJ45? |
| 09:19.11 | dlynes_laptop | never did |
| 09:19.20 | dlynes_laptop | sangoma a200's use rj9 |
| 09:19.42 | EmleyMoor | OK - chowned the whole thing |
| 09:20.23 | dlynes_laptop | qdk: only their pri cards use rj45 |
| 09:20.26 | EmleyMoor | I am planning to set up time-dependent black/whitelisting - presumably that's easy |
| 09:20.31 | qdk | dlynes_laptop: I have BNC-something to RJ45 converters to use with my digium to hookup to my E1/SS7 connection. |
| 09:20.43 | qdk | dlynes_laptop: yes, what else is there to use? |
| 09:21.01 | dlynes_laptop | qdk: oh...you want it for the pri |
| 09:21.09 | dlynes_laptop | qdk: rj45 is pretty standard for pris |
| 09:21.47 | dlynes_laptop | I donm't think i've ever seen bnc connections |
| 09:21.49 | qdk | dlynes_laptop: yes, i though so too... but E1 uses BNC connectors. |
| 09:21.55 | dlynes_laptop | ah |
| 09:22.16 | dlynes_laptop | Sangoma might have a version available that uses bnc, for all I know |
| 09:22.33 | dlynes_laptop | But t1's use rj45 |
| 09:22.54 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 09:23.32 | qdk | dlynes_laptop: are you use? or is it just what your supplier provides you with? |
| 09:23.32 | hads | THe E1's that I've seen use RJ45 too. |
| 09:23.34 | dlynes_laptop | qdk: http://www.sangoma.com/datasheets/p_aft-et3-specs?PHPSESSID=7 |
| 09:23.44 | qdk | dlynes_laptop: ISDN-30 for PRI uses RJ45 too, but E1 doesnt. |
| 09:23.46 | dlynes_laptop | qdk: that one supports dual channel bnc connectors |
| 09:24.24 | dlynes_laptop | qdk: You're using e3, not e1? |
| 09:25.26 | yebo | some E1s use BNC |
| 09:25.31 | yebo | http://www3.shopping.com/xPO-Cisco-16-ft-Network-Cable-CAB-E1-BNC~r-1~CLT-INTR~RFR-www.google.com |
| 09:25.35 | qdk | dlynes_laptop: Well i pay for E1. :-) |
| 09:26.03 | dlynes_laptop | qdk: ah |
| 09:26.19 | dlynes_laptop | qdk: well, anyways...the one with bnc connectors is in cvs, so you wouldn't want that one :) |
| 09:26.52 | qdk | dlynes_laptop: but that card might be a good option when i go from multi ISDN-30 to SS7/E1. |
| 09:27.07 | qdk | dlynes_laptop: cvs? |
| 09:27.12 | dlynes_laptop | qdk: yeah, but then you'd have to switch to yate or freeswitch |
| 09:27.18 | dlynes_laptop | qdk: I don't think it's supported by zaptel |
| 09:27.51 | dlynes_laptop | qdk: or Opal for that matter, or write your own driver for bayonne |
| 09:28.02 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 09:28.41 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 09:29.01 | qdk | dlynes_laptop: well im surely not gonna write anything like that. :-) |
| 09:29.48 | qdk | dlynes_laptop: but i need to do more research and get more VoIP<-> PSTN minutes. ;-) |
| 09:30.08 | *** join/#asterisk rtcg (n=chatzill@mail.richardthecomputerguy.com) |
| 09:31.19 | benjk | bayonne is usually the driver-availability champion, what kind of drivers do you think it lacks? |
| 09:31.44 | qdk | dlynes_laptop: http://www.sangoma.com/datasheets/p_aft-104-specs?PHPSESSID=7 <- ill probably get a few of those. |
| 09:31.45 | dlynes_laptop | benjk: ah...just assumed it didn't have drivers for sangoma, because I don't recall it ever being mentioned on their wiki |
| 09:32.05 | dlynes_laptop | qdk: yeah..it comes standard with hardware echo canceller, too |
| 09:32.24 | benjk | they don't list the hardware they support in great detail, but from talking to David Sugar I know that there is more than what's on their website |
| 09:33.00 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 09:33.01 | dlynes_laptop | benjk: david sugar is a sangoma employee? |
| 09:33.02 | benjk | with a bit of luck (timezone matching wise) you might be able to catch david on #bayonne |
| 09:33.14 | benjk | David Sugar is Mr.Bayonne :) |
| 09:33.17 | dlynes_laptop | ah |
| 09:33.30 | dlynes_laptop | Yeah...I was talking about sangoma's wiki, not bayonne's website |
| 09:33.54 | benjk | Sangoma don't list all the software that uses their cards though |
| 09:33.56 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 09:34.05 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 09:34.18 | benjk | it took us at least one month of asking again and again before they added OpenPBX.org to some of their pages |
| 09:35.00 | benjk | there's also http://www.voip-info.org/wiki/view/Bayonne |
| 09:35.13 | dlynes_laptop | yeah...they support Opal PBX too, whatever that is |
| 09:35.19 | benjk | dont recall ecactly how complete the list of hardware on there is though |
| 09:35.34 | benjk | Opal is the new name of what used to be OH323 |
| 09:36.08 | benjk | they have embraced SIP now I think |
| 09:36.22 | benjk | and I hear they have their own IAX2 stack |
| 09:36.32 | benjk | actually I was looking that it last week |
| 09:36.59 | dlynes_laptop | ah |
| 09:37.04 | benjk | the Freeswitch folks evaluated it against Digium's IAX stack and they claim that the Opal IAX stack is superior |
| 09:37.16 | dlynes_laptop | means nothing to me |
| 09:37.22 | benjk | something to look into in the not so distant future |
| 09:37.28 | dlynes_laptop | I had no interest in open h323 |
| 09:37.41 | dlynes_laptop | It's a huge pia to get installed |
| 09:37.47 | benjk | that's what I am telling you, they are no longer H323 only |
| 09:37.55 | benjk | that's why their renamed to Opal |
| 09:37.58 | dlynes_laptop | f'in thing requires you to have X windows installed on your server |
| 09:38.33 | benjk | Opal is a multi-protocol softswitch/PBX thing, just like Asterisk, Bayonne, FreeSwitch, OpenPBX.org and Yate |
| 09:38.41 | benjk | its been transformed |
| 09:38.46 | qdk | benjk: you know if the EC of sangoma is working on a lower level than the software? |
| 09:39.13 | qdk | benjk: or does the software have to support it directly to make use for the hardware EC? |
| 09:39.17 | benjk | qdk, if it is hardware EC, then it would have to |
| 09:39.19 | dlynes_laptop | Yeah, but coming from open h323, I would imagine it still requires the h323 component to be compiled and linked, right? |
| 09:39.48 | benjk | dlynes, I haven't used it myself and never built it, so I don't know |
| 09:39.51 | dlynes_laptop | qdk: zaptel software echo canceller is used unless you tell wanrouter to use a hwec |
| 09:40.00 | qdk | benjk: yes, but without a software driverthingy then the hardware is useless. |
| 09:40.17 | qdk | dlynes_laptop: wanrouter? |
| 09:40.23 | benjk | qdk, I don't know the details of the hardware EC architecture |
| 09:40.34 | dlynes_laptop | qdk: then all the code within zaptel to use the software echo canceller will call the hardware echo canceller on the sangoma instead |
| 09:40.37 | qdk | benjk: Ok. |
| 09:40.56 | dlynes_laptop | qdk: wanrouter is the tool used to administer your sangoma card |
| 09:41.13 | benjk | I am still looking forward to Sangoma completing libsangoma, so we don't have to use zaptel anymore |
| 09:41.33 | qdk | dlynes_laptop: ok, i would love not to be bound to zaptel drivers. |
| 09:42.10 | dlynes_laptop | qdk: it's still using the zaptel drivers, but if you have the hwec on the sangoma card, it's used instead of the zaptel software echo canceller |
| 09:42.29 | dlynes_laptop | qdk: you still configure the hardware echo canceller through the use of the echo canceller options in zapata.conf |
| 09:43.10 | dlynes_laptop | qdk: so, to asterisk, it doesn't know any different between the software echo canceller and the hardware echo canceller |
| 09:43.19 | qdk | dlynes_laptop: ok, so the hardware EC have the same issues with JB and PL as the software or can it compensate better for that? |
| 09:43.33 | dlynes_laptop | JB and PL? |
| 09:43.50 | dlynes_laptop | jitterbuffer and packetloss? |
| 09:43.54 | qdk | ups, just J... as in jitter and pack... yes |
| 09:43.59 | dlynes_laptop | the hardware echo canceller has nothing to do with iax |
| 09:44.07 | dlynes_laptop | nor does the software echo canceller |
| 09:44.09 | qdk | dlynes_laptop: huh+ |
| 09:44.11 | qdk | huh? |
| 09:44.14 | dlynes_laptop | it only controls zaptel channels |
| 09:44.25 | dlynes_laptop | zaptel channels don't have packets |
| 09:44.33 | dlynes_laptop | so therefore there's no packet loss, anyways |
| 09:45.21 | qdk | perhaps the fact is still that either, high latency, jitter and/or pl will case the software EC to be interrupted/reset. |
| 09:45.48 | dlynes_laptop | Yeah, if you're talking about a heavy machine load affecting the echo canceller, the hwec is not affected by that |
| 09:45.58 | dlynes_laptop | Only the software echo canceller |
| 09:46.29 | qdk | Im not sure whats really causing it to get disabled thereby introducing echo in the call, but i have customers complaining about it. |
| 09:46.50 | qdk | dlynes_laptop: the machine isnt doing that much. |
| 09:47.18 | dlynes_laptop | qdk: it's caused by the remote analog end usually |
| 09:47.31 | dlynes_laptop | qdk: and yes, the hwec does a much better job of controlling it |
| 09:47.50 | dlynes_laptop | qdk: the echo canceller sangoma is using, is a carrier grade echo canceller |
| 09:48.01 | dlynes_laptop | qdk: digium has some new cards that use the octasic chipset as well |
| 09:48.10 | qdk | dlynes_laptop: wel i know its the VoIP. |
| 09:48.21 | dlynes_laptop | voip doesn't cause echo |
| 09:48.23 | dlynes_laptop | analog does |
| 09:48.42 | qdk | dlynes_laptop: pay attention or this is pointless. |
| 09:49.00 | *** join/#asterisk asterisk_baby (n=shady@202.61.50.75) |
| 09:49.38 | asterisk_baby | i am not able to register sip users on my asterisk |
| 09:49.49 | asterisk_baby | Connected to Asterisk 1.2.13 |
| 09:50.08 | qdk | dlynes_laptop: there IS echo from the analog, but thats NOT what we are talking about. |
| 09:50.46 | dlynes_laptop | qdk: so you're getting call quality issues on your sip or iax channels then? |
| 09:50.46 | asterisk_baby | dlynes_laptop: hiiii.. i was having problems with my asterisk. could you help me again plz |
| 09:51.02 | qdk | dlynes_laptop: yes |
| 09:51.38 | dlynes_laptop | qdk: and how would a hardware or software echo canceller for a zaptel channel have anything to do with your voip channels? |
| 09:51.55 | dlynes_laptop | asterisk_baby: pastebin your log file |
| 09:52.07 | asterisk_baby | http://pastebin.ca/270452 |
| 09:52.12 | asterisk_baby | one moment |
| 09:52.41 | dlynes_laptop | that's your config file |
| 09:53.07 | asterisk_baby | there's nothing in the logs.. its a fresh system |
| 09:54.15 | asterisk_baby | '/var/log/asterisk/event_log' is empty |
| 09:54.18 | qdk | dlynes_laptop: the echo is too close to the real voice in analog to be an issue. |
| 09:55.06 | dlynes_laptop | asterisk_baby: Not event_log |
| 09:55.12 | dlynes_laptop | asterisk_baby: /var/log/asterisk/full |
| 09:58.34 | asterisk_baby | http://pastebin.ca/271643 |
| 09:59.12 | *** join/#asterisk merbanan (n=banan@136.240.13.217.in-addr.dgcsystems.net) |
| 09:59.48 | asterisk_baby | i dont see a 'full' file in /var/log/asterisk.. there was 'messages' so i copied the stuff from that |
| 10:01.29 | asterisk_baby | i was using register => 100000:blah@myserver.com/100000 |
| 10:01.30 | qdk | asterisk_baby: /etc/asterisk/logger.conf |
| 10:01.32 | asterisk_baby | in my sip.conf.. |
| 10:01.45 | qdk | asterisk_baby: and then activate full-logging. |
| 10:02.28 | dlynes_laptop | asterisk_baby: well, that server address has port 5060 open and filtered |
| 10:03.04 | dlynes_laptop | asterisk_baby: however wherever port 5060 is forwarded to or listened on by is probably not running |
| 10:03.14 | dlynes_laptop | asterisk_baby: because you're timing out, trying to register |
| 10:03.58 | asterisk_baby | that makes sense |
| 10:04.12 | asterisk_baby | let me copy the logger.conf thing too |
| 10:04.39 | asterisk_baby | ;debug => debug |
| 10:04.39 | asterisk_baby | console => notice,warning,error |
| 10:04.39 | asterisk_baby | ;console => notice,warning,error,debug |
| 10:04.39 | asterisk_baby | messages => notice,warning,error |
| 10:04.39 | asterisk_baby | ;full => notice,warning,error,debug,verbose |
| 10:05.19 | mendol | i have problem with IAX2 incoming settings/ context :-/ |
| 10:07.19 | qdk | mendol: probably a typo. |
| 10:07.28 | qdk | asterisk_baby: remove ; in ;full => ... and restart * |
| 10:07.47 | asterisk_baby | http://pastebin.ca/271651 |
| 10:08.39 | asterisk_baby | done |
| 10:08.52 | asterisk_baby | updated logger.conf & restarted asterisk |
| 10:09.34 | dlynes_laptop | asterisk_baby: check line 438 of extensions.conf |
| 10:09.43 | dlynes_laptop | asterisk_baby: you'll find you've got an error in your dialplan |
| 10:10.07 | asterisk_baby | wow its registering on 5060 |
| 10:10.11 | asterisk_baby | <PROTECTED> |
| 10:10.11 | asterisk_baby | <PROTECTED> |
| 10:10.29 | asterisk_baby | dlynes_laptop.. just checking line 438 |
| 10:11.16 | asterisk_baby | dlynes_laptop: yyes there was an error on 438.. its fixed now. thank u |
| 10:12.17 | *** join/#asterisk seva (i=seva@sevatech.com) |
| 10:13.16 | seva | it's really late and i am getting confused, i just want to test * out and want a simple extension to dial out via Zap/g1 shouldn't the following work for dialing from the console? |
| 10:13.17 | seva | exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) |
| 10:13.17 | seva | exten => _91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) |
| 10:13.40 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 10:13.47 | dlynes_laptop | seva: it depends |
| 10:14.02 | asterisk_baby | its not registering for 8891 though |
| 10:14.14 | dlynes_laptop | seva: do you have chan_alsa.so and/or chan_oss.so loaded? |
| 10:14.26 | seva | yes, chan_oss is loaded |
| 10:14.28 | asterisk_baby | how do i fix it? i need it on 8891 and not 5060 |
| 10:14.42 | dlynes_laptop | asterisk_baby: port 8891? |
| 10:15.01 | seva | oh wait, it is working! |
| 10:15.11 | asterisk_baby | http://pastebin.ca/271656 |
| 10:15.20 | asterisk_baby | dlynes_laptop: yes |
| 10:15.38 | asterisk_baby | dlynes_laptop: it just registered on 5060.. bt i need it for port 8891 |
| 10:15.55 | asterisk_baby | http://pastebin.ca/271656 has the sip debug result for u |
| 10:16.15 | dlynes_laptop | asterisk_baby: you're wanting to register elsewhere on port 5060? |
| 10:16.27 | dlynes_laptop | asterisk_baby: or you want someone else to register to you on port 8891? |
| 10:16.31 | dlynes_laptop | asterisk_baby: you're wanting to register elsewhere on port 8891 |
| 10:16.37 | dlynes_laptop | ? |
| 10:17.12 | asterisk_baby | yes |
| 10:17.35 | asterisk_baby | basically i want it on port 8891.. for myself and for anybody who needs it |
| 10:18.38 | seva | dlynes_laptop: hrm, it seems to be working sporadically |
| 10:18.43 | asterisk_baby | i have another machine working just fine on 8891.. ive changed the port (bindport = 8891) but its still not registering |
| 10:19.17 | asterisk_baby | have a look at this.. Retransmitting #2 (NAT) to 208.109.119.199:5060: |
| 10:19.25 | asterisk_baby | http://pastebin.ca/271656 |
| 10:19.59 | dlynes_laptop | and? |
| 10:20.44 | dlynes_laptop | asterisk_baby: after you did bindport=8891 |
| 10:20.53 | dlynes_laptop | asterisk_baby: did you do unload chan_sip.so |
| 10:21.00 | *** part/#asterisk seva (i=seva@sevatech.com) |
| 10:21.01 | dlynes_laptop | asterisk_baby: and then load chan_sip.so? |
| 10:21.01 | asterisk_baby | no.. i reloaded it |
| 10:21.11 | dlynes_laptop | Yeah...try unloading and then loading instead |
| 10:21.20 | dlynes_laptop | I don't think reload unbinds and binds ports |
| 10:21.28 | asterisk_baby | hmm k im stopping asterisk and starting it again |
| 10:21.35 | asterisk_baby | is that ok dlynes_laptop? |
| 10:21.54 | dlynes_laptop | yeah...that works, too |
| 10:22.01 | qdk | asterisk_baby: restart * is the safe bet. |
| 10:23.01 | asterisk_baby | still the same :( |
| 10:23.19 | dlynes_laptop | asterisk_baby: pastebin the output of "netstat -anp | grep asterisk" |
| 10:23.20 | asterisk_baby | http://pastebin.ca/271662 sip debug |
| 10:24.00 | asterisk_baby | http://pastebin.ca/271665 |
| 10:24.12 | asterisk_baby | http://pastebin.ca/271665 "netstat -anp | grep asterisk" |
| 10:27.08 | dlynes_laptop | asterisk_baby: your sip peer is binding to port 8891, not asterisk |
| 10:27.21 | dlynes_laptop | asterisk_baby: and your sip peer is still trying to register on port 5060 |
| 10:27.28 | asterisk_baby | correct |
| 10:28.10 | asterisk_baby | dlynes_laptop: how do i fix it then |
| 10:28.34 | dlynes_laptop | Read up on sip.conf |
| 10:28.58 | dlynes_laptop | It seems you don't understand a lot of the options...if you read up on sip.conf on the wiki, you'll be aware of all the options available for sip channels |
| 10:29.20 | asterisk_baby | that's true.. im new to asterisk |
| 10:29.57 | asterisk_baby | i'll check voip-info.org |
| 10:30.14 | dlynes_laptop | ok |
| 10:30.22 | *** join/#asterisk af_ (n=af@ip-156-221.sn2.eutelia.it) |
| 10:30.41 | *** join/#asterisk clive- (n=pirch@dsl-243-110-143.telkomadsl.co.za) |
| 10:30.59 | asterisk_baby | dlynes_laptop: thanks for ur help |
| 10:31.14 | clive- | Hi, anyone got any pointers as top why a tdm fxo module doesnt answer the phone |
| 10:31.15 | asterisk_baby | dlynes_laptop: u the best |
| 10:31.37 | DrCron | clive-, yhea, you didnt post any info |
| 10:31.41 | DrCron | :P |
| 10:32.12 | DrCron | does full debugging show anything? |
| 10:32.12 | clive- | drcron hi |
| 10:32.59 | clive- | well, it seems to load up all fine, the modprobes and stuff, but when the phoen rings, the CLI says "ringing" and exectuing answer, but it just doesnt answer |
| 10:33.27 | DrCron | does the phone keep ringing? |
| 10:34.00 | *** join/#asterisk henrique (n=henrique@200-161-80-249.dsl.telesp.net.br) |
| 10:34.05 | dlynes_laptop | asterisk_baby: i know |
| 10:34.11 | *** join/#asterisk adker (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net) |
| 10:36.12 | clive- | drCron, yup, the phone just rings |
| 10:36.42 | DrCron | how familiar are you with * |
| 10:36.55 | *** join/#asterisk Gr1ncheux (n=devine@AToulouse-257-1-50-134.w90-5.abo.wanadoo.fr) |
| 10:37.10 | clive- | fairly |
| 10:37.44 | clive- | usually use isdn, this analogue is a whole new bag of tricks |
| 10:39.16 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 10:39.22 | mendol | sorry anybody could give me www adress for some good iax context tutorial? cant get mine to work :-/ |
| 10:39.41 | clive- | I have this on my screen: Executing Answer("Zap/3-1", "") in new stack |
| 10:39.42 | clive- | ...but no luck in actually doing it |
| 10:39.59 | clive- | mendol try the wiki |
| 10:41.45 | mendol | yeh was looking around, just to stupid to understand all this :-/ and cant get my incoming calls working |
| 10:42.18 | clive- | lol..same as me, ...my calls dont get answered |
| 10:42.23 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 10:42.52 | mendol | first i had no authority found, now it cant ring |
| 10:43.17 | qdk | mendol: just look at context in general... the context of ANY channels it the context which a call with start at coming "in" that way. |
| 10:44.20 | mendol | now you are just confusing me. i have incoming call from somewhere, and "IAX2/home-3", "No DID or CID Match" |
| 10:44.33 | mendol | from debug :-/ |
| 10:44.47 | clive- | mendol, you should use exten =>s,1,..... |
| 10:45.03 | qdk | mendol: as in context=im_the_context_coming_form_IAX_users/peer_X_make_me_do_something_in_your_extensions_please |
| 10:45.06 | *** join/#asterisk newbie1 (n=Main@203.208.196.140) |
| 10:45.21 | *** join/#asterisk CleanerX (n=nix@p54A395A0.dip0.t-ipconnect.de) |
| 10:46.15 | newbie1 | Hello every body |
| 10:46.22 | newbie1 | I'm new in asterisk |
| 10:46.51 | newbie1 | I want to catch from which IP a call is coming in my AGI |
| 10:46.56 | newbie1 | how can do that? |
| 10:47.15 | clive- | newbie you will have to do that from teh dialplan |
| 10:47.23 | qdk | mendol: and in extensions: [im_the_context_coming_form_IAX_users/peer_X_make_me_do_something_in_your_extensions_please] exten => s,1,NoOp(Well hello there mr. im_the_context_coming_form_IAX_users/peer_X_make_me_do_something_in_your_extensions_please, what can i do for you today?) |
| 10:47.28 | clive- | and transferthe ip address from the dialplan to your agi |
| 10:47.58 | newbie1 | clive: Can you show me an example? |
| 10:48.24 | qdk | clive-: or just "ask" the dialplan for it. |
| 10:48.27 | clive- | perl agi? |
| 10:48.41 | newbie1 | clive: How can I have the IP in my dial plan? |
| 10:48.48 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:49.03 | newbie1 | clive: Is there any asterisk variables that holds the IP? |
| 10:49.29 | clive- | not too sure |
| 10:49.35 | newbie1 | clive: PHP agi |
| 10:50.04 | qdk | newbie1: what do you need it for? |
| 10:50.49 | qdk | newbie1: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sippeer |
| 10:51.47 | *** join/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk) |
| 10:52.14 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 10:52.15 | Stephnie | hi |
| 10:52.43 | mendol | o thanks qdk |
| 10:52.54 | newbie1 | qdk: I want to include origination IP in my cdr |
| 10:52.55 | Stephnie | Sometimes I get "== Forcing Marker bit, because SSRC has changed" ..after Attempting native bridge....may I know wht is this msg for ? |
| 10:53.08 | mendol | clive yea i tried that :-/ |
| 10:53.40 | clive- | mendol any success? |
| 10:54.09 | qdk | newbie1: show function SIPCHANINFO perhaps. |
| 10:54.30 | qdk | newbie1: as an extra info or for billing purpose? |
| 10:54.51 | clive- | qdk any suggesstions to help get a fxo module to pick up an incomming call , , the CLI shows: Executing Answer("Zap/3-1", "") in new stack |
| 10:54.57 | mendol | nope still the same error in debug |
| 10:54.59 | newbie1 | qdk: Yes |
| 10:55.05 | Stephnie | anubody?? |
| 10:55.36 | newbie1 | qdk: also i want to setup a firewall by observing IP |
| 10:55.45 | *** join/#asterisk te_lo_meto_mami (n=te_lo_me@8.10.2.50) |
| 10:56.08 | newbie1 | qdk: I want to pass only some particulars IP |
| 10:56.22 | newbie1 | qdk: Isn't better doing in AGI? |
| 10:56.47 | mattfletcher | is it possible to maintain the original incoming caller id after using a phone's transfer button to transfer a call |
| 10:59.21 | qdk | clive-: well thats seem correct, whats the problem? |
| 10:59.28 | newbie1 | qdk: I got it. THanks |
| 10:59.48 | newbie1 | qdk: It is http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sipchaninfo |
| 10:59.49 | qdk | newbie1: i put an or in my question, so yes is a rather irritating answer. ;-) |
| 10:59.52 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 11:00.34 | kaldemar | mattfletcher: what phones are you using? |
| 11:00.47 | qdk | newbie1: imho you should have some firewall thingy doing the IP accounting and not your * |
| 11:00.58 | newbie1 | qdk: I missed your question. For billing purpose. |
| 11:01.44 | mattfletcher | aastra 480i's mainly |
| 11:02.18 | qdk | newbie1: ok, I dont really see how to IP is relevant to the billing. IPs tend to change, so it will probably be a pain to maintain. |
| 11:02.26 | qdk | newbie1: ok, thx. |
| 11:03.12 | *** join/#asterisk lilalinux (n=plasma@80.69.41.2) |
| 11:06.48 | newbie1 | qdk: Can you refer me an open source SIP phone? |
| 11:07.24 | mosty | there's lots on google |
| 11:07.53 | clive- | qdk the trouble is that the line is not picked up |
| 11:08.06 | newbie1 | mosty: ya... I'm not sure which one would be good |
| 11:08.14 | mosty | newbie1: try a few, see what you like |
| 11:09.35 | newbie1 | mosty: Thanks, I will try |
| 11:11.07 | qdk | newbie1: i use sjphone, but the source isnt open. |
| 11:11.49 | qdk | clive-: ok, i dont do much ZAP-TECH so i cant help you much. |
| 11:11.57 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 11:12.41 | clive- | qdk analogue sucks, I must prefer isdn |
| 11:12.49 | clive- | much prefer |
| 11:14.04 | qdk | clive-: yes, me2... and I want "big" ISDNs. :-) |
| 11:15.49 | mendol | baahh, kill me ;-p |
| 11:16.52 | mendol | <PROTECTED> |
| 11:16.56 | mendol | :-/ |
| 11:17.56 | qdk | mendol: ? |
| 11:19.00 | *** join/#asterisk topping (n=topping@64.212.181.67) |
| 11:19.14 | *** join/#asterisk evandro (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 11:19.40 | mendol | still got this error on 2nd asterisk,on 1st one all is fine -- Executing Set("IAX2/etel-3", "FROM_DID=224346287") in new stack |
| 11:19.50 | mendol | dont know how to fix it |
| 11:20.37 | mosty | mendol: what is the error? |
| 11:21.07 | *** join/#asterisk evandro (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 11:21.37 | mendol | "IAX2/home-3", "No DID or CID Match" |
| 11:21.41 | mendol | from incoming call |
| 11:22.28 | *** join/#asterisk beyond (n=beyond@200-155-185-1.static.spo.ifx.net.br) |
| 11:22.41 | mendol | <PROTECTED> |
| 11:22.53 | mosty | who are they trying to call? |
| 11:23.39 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 11:24.18 | mendol | have to test iax2 trunk if its working, outbound is ok, inbound doesnt work |
| 11:24.30 | mendol | and dont know how to edit context for it |
| 11:24.34 | key2 | Do I need to authentify before connecting to a stun ? |
| 11:24.56 | mosty | mendol: well it's not an error per se, the dialplan is doing exactly what you've told it to |
| 11:25.47 | mendol | so its dialplan fault? |
| 11:26.58 | mosty | mendol: it's the fault of whoever wrote the dialplan |
| 11:27.18 | mendol | hah, means me ;-) |
| 11:28.07 | mosty | paste that context on a paste site somewhere |
| 11:28.19 | mosty | and explain what you want it to do |
| 11:31.32 | EmleyMoor | If I have an extension that rings all of a SIP, IAX and Zap channel, should the unanswered channels stop ringing when one is answered? |
| 11:31.50 | mattfletcher | emleymoor: yes |
| 11:31.57 | EmleyMoor | Hmmm |
| 11:32.15 | EmleyMoor | It's not happening when I answer my Zap phone - SJphone still rings |
| 11:32.37 | mattfletcher | emleymoor: assuming you're using for example dial(sip/201&IAX/202&Zap/1) |
| 11:33.04 | EmleyMoor | Yes - does including a timeout and options make any difference? |
| 11:33.20 | mattfletcher | emleymoor: i haven't found it o |
| 11:33.31 | EmleyMoor | Maybe it's SJphone's fault |
| 11:33.40 | zoa | use zoiper! |
| 11:33.41 | zoa | :p |
| 11:33.49 | mattfletcher | i think so |
| 11:33.56 | zoa | the unanswered calls should NOT keep ringing |
| 11:34.11 | zoa | www.zoiper.com (shameless plug) |
| 11:34.20 | *** part/#asterisk newbie1 (n=Main@203.208.196.140) |
| 11:34.29 | kaldemar | when can we get zoiper free? ;) |
| 11:35.11 | zoa | its there already |
| 11:35.16 | zoa | in beta |
| 11:37.03 | *** part/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 11:37.33 | kaldemar | well that's logic. |
| 11:37.40 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:38.16 | kaldemar | isn't it only biz beta that expires dec 20th? |
| 11:38.48 | *** part/#asterisk beyond (n=beyond@200-155-185-1.static.spo.ifx.net.br) |
| 11:40.05 | *** join/#asterisk ambriento (n=ambrient@200.192.160.100) |
| 11:40.06 | EmleyMoor | Is there a good way of looking at the cdr.db? |
| 11:40.58 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 11:46.32 | mattfletcher | zoa: the version i just downloaded says it expires - hardly free |
| 11:47.30 | benjk | free is an illusion anyway |
| 11:48.40 | zoa | yeah |
| 11:48.44 | zoa | its the beta that expires |
| 11:48.49 | zoa | there will be a new beta by that date |
| 11:49.06 | benjk | that's a great feature then |
| 11:49.12 | benjk | automatically expiring bugs |
| 11:49.32 | zoa | that was the idea |
| 11:49.40 | zoa | i dont want the people to run something buggy for too long |
| 11:49.47 | zoa | its bad for the name :) |
| 11:49.54 | zoa | reputation |
| 11:50.22 | *** join/#asterisk _mh (n=largo@202.5.145.13) |
| 11:50.59 | zoa | btw, you can bypass it just by changing the date before you startup the phone |
| 11:51.01 | kaldemar | hope the gui gets improvements before an official version is released. |
| 11:51.09 | zoa | send me all your comments |
| 11:51.14 | zoa | info@attractel.com |
| 11:51.23 | zoa | on gui and all other stuff |
| 11:51.46 | qdk | ${SIP_CODEC}: Used to set the SIP codec for a call <- is there a similar way to get the codec? |
| 11:52.00 | kaldemar | will do. |
| 11:53.34 | mattfletcher | zoa: cunning! u ought to make that obvious on your site. it looks great and works fantastically, but i almost didn't download it coz of the expiry - i immediately thought "shareware" |
| 11:53.56 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 11:56.19 | *** join/#asterisk Skarmeth (n=Skarmeth@201009008073.user.veloxzone.com.br) |
| 11:57.19 | *** join/#asterisk ToyMan (n=stuq@74-32-63-190.dsl1.mdl.ny.frontiernet.net) |
| 11:59.25 | *** join/#asterisk ToTo (n=ToTo@host150-83-dynamic.60-82-r.retail.telecomitalia.it) |
| 12:00.15 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 12:04.44 | *** join/#asterisk vgster` (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 12:08.34 | *** join/#asterisk stuq (n=stuq@74-32-63-190.dsl1.mdl.ny.frontiernet.net) |
| 12:14.02 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-231.range81-152.btcentralplus.com) |
| 12:17.54 | xain | hi |
| 12:19.25 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 12:22.10 | vooduhal | Does anyone know of a manual way to turn on the MWI on a SIP phone? |
| 12:22.30 | Nibbier | hi |
| 12:23.23 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) |
| 12:27.37 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 12:30.12 | *** join/#asterisk SoftIce (n=softice@vc-196-207-45-253.3g.vodacom.co.za) |
| 12:30.41 | SoftIce | hi, what is the best troubleshooting steps to identify why asterisk wont accept a call, I can phone out through the line just wont get any calls in |
| 12:31.34 | EmleyMoor | What happens on the console (set verbose 10 if you haven't already done so) when a call is attempted? |
| 12:32.00 | *** join/#asterisk lilalinux (n=plasma@80.69.41.2) |
| 12:32.35 | SoftIce | k, hard as its all done remotly |
| 12:32.40 | SoftIce | will have let you know now |
| 12:34.01 | SoftIce | <PROTECTED> |
| 12:34.05 | SoftIce | good sign |
| 12:34.35 | *** join/#asterisk rwa (n=rwa@213.211.189.168) |
| 12:34.51 | rwa | hi all |
| 12:35.04 | SoftIce | what is a fast pastebin |
| 12:35.17 | rwa | can anybody tell me how i can remove spaces out of a dialstring ? |
| 12:35.52 | EmleyMoor | How is your dialstring getting spaces in it in the first place? |
| 12:36.03 | SoftIce | EmleyMoor: where can I paste this info to you |
| 12:36.07 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 12:36.10 | rwa | i had to add them cause of a fixed field length |
| 12:36.14 | EmleyMoor | www.pastebin.ca |
| 12:36.25 | EmleyMoor | rwa: Are they at the end? |
| 12:36.30 | SoftIce | I just get the number u have dialed is not in service |
| 12:36.42 | rwa | Yes, and there can be 3 til 5 of them |
| 12:36.54 | rwa | lets say 5 till none |
| 12:37.03 | SoftIce | EmleyMoor http://pastebin.ca/271742 |
| 12:37.06 | SoftIce | thank you ;) |
| 12:37.13 | EmleyMoor | SoftIce: I probably can't help you but I'll look and I'm sure someone will |
| 12:37.43 | *** join/#asterisk coppice (n=chatzill@40.168.17.210.dyn.pacific.net.hk) |
| 12:37.55 | EmleyMoor | rwa: Well, I don't know how * handles strings, but maybe you could find the position of the first and slice it to there - 1? |
| 12:38.17 | rwa | I tried, * is limited in this |
| 12:38.39 | rwa | also i can't find FILTER function, its not inthere |
| 12:39.10 | EmleyMoor | SoftIce: No idea - maybe someone else can advise |
| 12:40.01 | SoftIce | anyone here, might be able to help ? |
| 12:40.31 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 12:43.09 | kaldemar | SoftIce: where does your dialplan come from? |
| 12:44.36 | kaldemar | your channels are set immediate in zapata.conf so asterisk tries to find extension s in the incoming context for the channel. |
| 12:45.00 | *** join/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 12:45.01 | vooduhal | Does anyone know of a good way to manually turn on an MWI indicator on a SIP device? I've used a pcap dump of a NOTIFY message before and just resent it but that no longer seems to work to turn it off. All of our users are agents and not tied to a phone but they still want the MWI to come on when they are logged into that phone. |
| 12:45.24 | SoftIce | kaldemar: so what do you sugest, as i'm not smart enough to give you all the answers ;) |
| 12:45.42 | SoftIce | with out people like you I would be in deep water.. |
| 12:45.43 | kaldemar | after all that checking and nooping it goes to context ext-did extension s priority 1 that gives you the announcement ss-noservice and hangs up the channel. |
| 12:46.36 | kaldemar | SoftIce: have you configured your * box yourself? |
| 12:47.10 | SoftIce | kaldemar: no, i'm a sys admin, I just got the hfc drivers, etc working |
| 12:47.24 | SoftIce | not of this system either, I was just able to get bristuff, etc working |
| 12:47.31 | SoftIce | I don't know enough about asterisk to comment |
| 12:47.54 | SoftIce | and the person that configured it doesn't know why it isn't working |
| 12:48.03 | SoftIce | maybe you could have a peek at the config? |
| 12:48.32 | kaldemar | umm, set the immeadiate=yes to immediate=no in your zapata.conf for these channels. |
| 12:49.04 | kaldemar | so you at least have a chance of getting the B-number from the signaling. |
| 12:49.36 | *** join/#asterisk cstextiles (n=SpShah@59.184.1.24) |
| 12:49.52 | SoftIce | ok, thats done |
| 12:50.14 | SoftIce | kaldemar: should I pastbin my zapata.conf file? |
| 12:50.52 | kaldemar | no need to, give a try like that, if it fails, you could pastebin your extensions.conf. |
| 12:51.06 | SoftIce | k thanks. |
| 12:51.11 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:52.15 | *** join/#asterisk stuq_ (n=stuq@user-12lcqia.cable.mindspring.com) |
| 12:52.30 | vooduhal | Does any know if it is possible to change a SIP phones mailbox= entry on the fly? |
| 12:52.41 | vooduhal | And without making config file changes. |
| 12:53.17 | mosty | vooduhal: realtime |
| 12:54.06 | SoftIce | kaldemar: heh, remote box and they have a busy phone system, so getting them to plug the isdn line back into this box |
| 12:54.11 | SoftIce | then will see if it works,. |
| 12:54.20 | SoftIce | kaldemar: how long have you been working with * ? |
| 12:54.24 | vooduhal | mosty, good idea. |
| 12:54.30 | *** join/#asterisk [Airwolf] (n=airwolf@84.241.219.185) |
| 12:54.32 | kaldemar | SoftIce: a bit over two years. |
| 12:54.51 | SoftIce | kaldemar: what sort of scale? |
| 12:54.52 | JT | http://cgi.ebay.com/MTI-TCS-9200-Transportable-Commincation-System_W0QQitemZ110063724735QQihZ001QQcategoryZ97143QQrdZ1QQcmdZViewItem |
| 12:54.55 | JT | man |
| 12:55.08 | JT | if that doesn't make you look like a formidable geek |
| 12:55.11 | JT | i dunno what will :P |
| 12:56.37 | kaldemar | SoftIce: not full time, but i've become familiar with most parts. |
| 12:57.24 | kaldemar | SoftIce: here's some good reading regarding asterisk's dialplan: http://www.voip-info.org/wiki-Asterisk+config+extensions.conf |
| 12:57.47 | vooduhal | Message waiting. |
| 12:58.53 | vooduhal | mosty, with realtime, can I change the table structure (by adding a field) without breaking realtime? |
| 12:59.29 | SoftIce | kaldemar: well that sorted that service error |
| 12:59.34 | *** part/#asterisk [Airwolf] (n=airwolf@84.241.219.185) |
| 12:59.35 | SoftIce | but still not working |
| 13:00.12 | SoftIce | http://pastebin.ca/271768 |
| 13:00.17 | SoftIce | that is the asterisk log |
| 13:00.25 | SoftIce | and let me show you my extentions |
| 13:00.41 | mosty | vooduhal: i doubt it. the standard method is to add/delete/modify rows of a table, not change the table format |
| 13:00.52 | *** join/#asterisk Kigh (i=kai@unixuni.org) |
| 13:01.43 | *** join/#asterisk M_at (n=matt@dsl092-214-175.atl1.dsl.speakeasy.net) |
| 13:02.52 | M_at | Can someone help me diagnose a problem with config for a Sangoma PRI card. I have two Sangoma cards n one box, Analog (FXS) and PRI. The analog works fine but when the PRI channels are activated in zapata.conf asterisk fails to start |
| 13:03.11 | mosty | m_at: what error in the logs? |
| 13:03.51 | M_at | ztcfg -vvvvv output : http://pastie.caboo.se/26454 |
| 13:03.52 | M_at | zaptel.conf : http://pastie.caboo.se/26455 |
| 13:03.52 | M_at | zapata-channels.conf : http://pastie.caboo.se/26457 |
| 13:03.52 | M_at | ./var/log/asterisk/full : http://pastie.caboo.se/26458 |
| 13:04.45 | mosty | fix the unknown directive bits |
| 13:05.18 | M_at | Where are those? Can yo give me the file and line number? |
| 13:05.33 | mosty | it's in the asterisk log you pasted |
| 13:05.54 | mosty | but that's not the fatal error, that is probably the bit about the unknown D channel |
| 13:06.27 | mosty | i'd check that your zaptel.conf is configured correctly for your country |
| 13:07.12 | M_at | It's a # on a line on it's own - they're removed now - same errors in full just without the unknown directives |
| 13:07.39 | M_at | I'm in the US right now - the card is not plugged into anything yet |
| 13:07.44 | *** join/#asterisk psiforce (n=foo@c210-49-175-128.mckinn1.vic.optusnet.com.au) |
| 13:08.11 | psiforce | does anyone know why asterisk seems to only run on 1 of my cpus on a smp machine |
| 13:08.23 | cy3o3 | hmm |
| 13:08.40 | cy3o3 | anyone on FWD that'd be down to give me a ring and see if I'm up and running correctly? |
| 13:09.02 | psiforce | cat /proc/cpuinfo shows 2 cpus |
| 13:09.34 | JT | psiforce: what makes you think it should run on both? |
| 13:10.57 | kaldemar | SoftIce: is it working? |
| 13:13.04 | mosty | jt: asterisk is supposed to be "heavily threaded" |
| 13:13.28 | *** join/#asterisk Tili (n=tili@202.133.67.207) |
| 13:16.21 | *** join/#asterisk bitwise_ (n=bitwise@84.254.39.169) |
| 13:19.06 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 13:19.06 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Zaptel 1.2.11 released (Nov. 9, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 13:20.00 | *** join/#asterisk ming_zym (n=zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 13:21.03 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 13:26.47 | cy3o3 | no one on FWD that can just give me a ring? :P |
| 13:26.56 | *** part/#asterisk clive- (n=pirch@dsl-243-110-143.telkomadsl.co.za) |
| 13:27.13 | phonetalk | hello |
| 13:27.15 | phonetalk | i have a question |
| 13:27.32 | phonetalk | how can i change default law = ulaw in zaptel ? |
| 13:28.39 | kaldemar | disallow=all |
| 13:28.43 | kaldemar | allow=ulaw |
| 13:29.20 | phonetalk | no |
| 13:29.20 | phonetalk | i m asking in zapata |
| 13:29.20 | phonetalk | its easy to use it in sip |
| 13:29.20 | phonetalk | but when i type this |
| 13:29.20 | phonetalk | in CLI |
| 13:29.20 | phonetalk | zap show channel x |
| 13:29.35 | phonetalk | its output is Default law: alaw |
| 13:29.38 | phonetalk | Fax Handled: no |
| 13:29.56 | phonetalk | how it can be changed there ? |
| 13:30.42 | kaldemar | write those lines in zapata.conf... |
| 13:30.49 | phonetalk | i did |
| 13:30.54 | phonetalk | i add defaultlaw=ulaw |
| 13:30.58 | phonetalk | but no use |
| 13:33.01 | *** join/#asterisk af_ (n=af@ip-156-221.sn2.eutelia.it) |
| 13:33.32 | M_at | did you add disallow and allow lines like kaldemar said or just defaultlaw? |
| 13:34.27 | phonetalk | just defaultlaw |
| 13:34.38 | phonetalk | cz i think zapata.conf doesnt allow these entries |
| 13:34.41 | M_at | do as kaldemar said AND remove defaultlaw=ulaw |
| 13:34.52 | *** join/#asterisk mithraen (n=mithraen@87.228.121.245) |
| 13:34.57 | phonetalk | ok |
| 13:35.34 | phonetalk | checking |
| 13:36.04 | phonetalk | same |
| 13:36.06 | phonetalk | Default law: alaw |
| 13:37.08 | phonetalk | i want my e1 channels on ulaw |
| 13:37.15 | phonetalk | cz i m not having voice on ss7 e1 link |
| 13:37.16 | M_at | Why? |
| 13:37.28 | M_at | what is on it then? |
| 13:37.35 | phonetalk | alaw |
| 13:39.24 | M_at | what is it linked to? |
| 13:39.24 | phonetalk | pstn |
| 13:39.24 | phonetalk | on ss7 link |
| 13:41.46 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
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| 13:46.19 | M_at | Can anyone diagnose why * is only showing the first 15 channels on a T1 card and bitching about the 16th? |
| 13:47.19 | docelmo | You have something screwed up? or your provider does |
| 13:47.48 | M_at | It's not connected to a provider yet - it's me at fault. Or the card. |
| 13:47.57 | M_at | ztcfg -vvvvv output : http://pastie.caboo.se/26454 |
| 13:47.58 | M_at | zaptel.conf : http://pastie.caboo.se/26455 |
| 13:47.58 | M_at | zapata-channels.conf : http://pastie.caboo.se/26457 |
| 13:47.58 | M_at | ./var/log/asterisk/full : http://pastie.caboo.se/26458 |
| 13:48.21 | zumbush | CISCO SUCKS!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! Worst ever when it comes to customer support. Atleast when ure just an end user. |
| 13:48.34 | zumbush | just had to get it off my chest :-P |
| 13:48.48 | docelmo | zumbush cisco rules.. :) |
| 13:49.02 | zumbush | NOT |
| 13:49.05 | zumbush | :-) |
| 13:49.10 | mosty | m_at: because the 16th is the D channel |
| 13:49.24 | docelmo | My whole network is cisco.. But then again I know cisco very well so issues dont bother me |
| 13:49.35 | zumbush | tried to get support for my Cisco phone and get firmware updates supporting SIP now for like 1 month |
| 13:49.39 | JT | it's connected to anything, you really shouldn't expect it to work |
| 13:49.43 | benjk | mosty, looks ike your setup is for an E1, not T1 |
| 13:49.47 | zumbush | their support phone numbers and emails dont work |
| 13:49.56 | zumbush | no one is there to help me |
| 13:49.59 | benjk | m_at, I meant |
| 13:49.59 | zumbush | GRRRR |
| 13:50.03 | mosty | benjk: correct |
| 13:50.04 | docelmo | ohh ya.. ok phones. They suck |
| 13:50.11 | M_at | mosty: Even on a T1? I know it is on an E1 |
| 13:50.18 | docelmo | Im talking about hardware that is > $10,000 USD |
| 13:50.25 | mosty | m_at: i guess not, i've never seen a t1 before |
| 13:50.28 | JT | isn't the D chhan TS24 on a T1? |
| 13:50.44 | zumbush | gtg |
| 13:50.47 | docelmo | The D chanel on a E1 is 15 on a T1 is usually 24 |
| 13:51.03 | benjk | m_at, you can configure the d channel wherever you want it, but 16 is the default for E1 not for T1 |
| 13:51.32 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:51.46 | M_at | I haven't configured it - got genzaptelconf to do it |
| 13:51.59 | JT | docelmo: wrong, TS16 is the C chan on an E1 |
| 13:52.02 | JT | D chan even |
| 13:52.08 | docelmo | my bad.. |
| 13:52.17 | docelmo | Its been about 6 months since I have had to deal with one |
| 13:52.21 | JT | heh |
| 13:52.36 | M_at | I have an E1 working fine at home. bchan=1-5, dchan=16 |
| 13:52.40 | *** join/#asterisk merbanan (n=banan@136.240.13.217.in-addr.dgcsystems.net) |
| 13:54.13 | JT | you can 5 chan fractional E1? |
| 13:56.05 | M_at | 1-15 rather |
| 13:56.05 | *** part/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 13:56.41 | JT | that's a few chans for at home :) |
| 13:57.02 | M_at | Home = UK, I'm in our US office right now. |
| 13:57.21 | docelmo | M_at where in the US? |
| 13:57.23 | M_at | phone talk reckons I must have the PRI as Span1 and the analog card as 2 |
| 13:57.25 | M_at | Miami |
| 13:57.30 | docelmo | ahh nice.. |
| 13:57.40 | docelmo | I used to live in Tampa.. Now I live in the fricken cold |
| 13:58.26 | M_at | It's just about the right temperature for me this morning - 68 :) |
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| 14:13.33 | npc105 | Anyone ever had a problem with MONITOR_EXEC not being recognized by Monitor()? |
| 14:13.36 | npc105 | I am trying to get sox to join the two legs and compress the single file into a ogg, however it never calls the MONITOR_EXEC commmand |
| 14:13.54 | ashish_ | 15 i need help in configuring PRI line ussing TE110p01 |
| 14:14.24 | *** part/#asterisk ashish_ (n=ashish@59.181.110.97) |
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| 14:32.43 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 14:45.12 | mosty | what does this warning mean? WARNING[17008] pbx.c: Requested contexts didn't get merged |
| 14:45.57 | *** part/#asterisk andreasmk2 (n=andreas@88.218.44.171) |
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| 14:53.02 | ashish_ | 15 i need help in configuring PRI line ussing TE110p01 |
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| 14:58.55 | key2 | Is there any open source TURN server ? |
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| 15:02.55 | *** join/#asterisk zmef420_ (n=zmef420@metarb3-pool4-44.mtco.com) |
| 15:03.40 | tRSS | my mitel 5220 phone was working fine until yesterday. I installed a T1 (TE110P) card yesterday, and now my mitel dtmf is not getting detected by Asterisk? How can I troubleshoot dtmf problems? |
| 15:05.34 | *** join/#asterisk PjEr (n=pjer@74.12.211.96) |
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| 15:10.12 | vooduhal | If I'm using realtime for my sip.conf and I update the mailbox entry in the database, when exactly does that mailbox setting get retrieved? On registration, or is there a way for it to update dynamicly? |
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| 15:12.08 | sloth | Good morning.....Has anyone played with t.38 passthtru on 1.4b3? |
| 15:13.51 | *** join/#asterisk Teeli (n=tili@202.133.67.188) |
| 15:13.52 | PjEr | Is it normal that asterisk send this to a server "NVITE sip:8006226232@vbuzzer.com:80 SIP/2.0" (notice le lack of 'I' in INVITE) ... cause, after this, each time the server respond "SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL)" ? |
| 15:16.48 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 15:17.10 | phonetalk | hello |
| 15:17.17 | phonetalk | i am having problem |
| 15:17.21 | phonetalk | no audio on asterisk-ss7 |
| 15:20.13 | *** part/#asterisk PjEr (n=pjer@74.12.211.96) |
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| 15:32.11 | hoobastooba | i did a new install of an asterisk server last night using current svn asterisk on centos. I am trying to set up vicidial. testing asterisk, I get no sound. has this ever happened to anyone else? if so what can I check on and fix? |
| 15:33.26 | *** join/#asterisk nibbler_de (n=nibbler@some.host.name) |
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| 15:34.43 | mosty | what is vicidial? |
| 15:35.04 | benjk | predicitive dialer |
| 15:35.08 | hoobastooba | yeah |
| 15:35.18 | benjk | telemarketers use that |
| 15:35.23 | benjk | to dial out |
| 15:35.33 | benjk | the agents who call you don't actually dial |
| 15:35.46 | benjk | the software picks the victims for them and dials automatically |
| 15:35.58 | benjk | that evil magic is called a predictive dialer |
| 15:37.34 | blitzrage | its even better when you answer and it drops you into a queue |
| 15:37.41 | blitzrage | I just hangup |
| 15:37.46 | blitzrage | or leave the phone offhook |
| 15:39.13 | mosty | i want a little answering machine sized device that prompts the caller to dial some random pin if they're not telemarketers or pollsters, and only ringing through to the phone if they do that |
| 15:40.20 | Nugget | that's what asterisk is for, mosty. |
| 15:40.30 | hoobastooba | just use authenticate in your dialplan |
| 15:40.30 | hoobastooba | works really nice |
| 15:40.36 | mosty | i want something smaller than a pc |
| 15:40.52 | mosty | something the size of say an adsl line filter, ideally |
| 15:41.07 | blitzrage | mosty: look at gumstix and astlinux |
| 15:41.39 | *** join/#asterisk shinux_ (n=shinux@196.220.29.150) |
| 15:42.24 | mosty | i'd also like to buy a premade unit, for under $40USD ;) |
| 15:44.43 | *** join/#asterisk supers (i=supers@Sia.AnimeNfo.com) |
| 15:45.06 | mut | heh |
| 15:46.42 | mosty | hrm, and it should draw power from the phone line too |
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| 15:57.26 | cy3o3 | any recommendations for PSTN -> FWD access? None of these seem to work on notaduck.com |
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| 16:03.00 | Katty | morning! |
| 16:03.13 | mishehu | ugh morning :-/ |
| 16:03.17 | blitzrage | mosty: does not exist afaik |
| 16:04.35 | blitzrage | hehehe |
| 16:04.45 | blitzrage | darn... I broke the chain of action |
| 16:04.56 | blitzrage | that'd be a good band name... |
| 16:04.59 | blitzrage | "Chain of Action" |
| 16:05.13 | blitzrage | bah! |
| 16:05.18 | Katty | blitzrage: we love you!!! |
| 16:05.20 | symlink | Katty: how are 'chu? |
| 16:05.21 | blitzrage | the love is stifling |
| 16:05.23 | blitzrage | lol |
| 16:05.27 | blitzrage | Katty: good to know :D |
| 16:05.29 | Katty | symlink: wishing i was drinking, actually |
| 16:05.33 | blitzrage | oh great idea |
| 16:05.46 | blitzrage | hrmmm..... maybe I should wait till noon... |
| 16:05.56 | blitzrage | breakfast would be good about now |
| 16:06.08 | symlink | blitzrage: PIZZA?!? |
| 16:06.29 | Katty | i'm thinking a shot of chocolate cake. |
| 16:06.38 | Katty | that'll help me make it to lunch, anyway |
| 16:06.40 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 16:06.58 | blitzrage | symlink: actually yah, other than that, I only have canned tuna. But last night I got a free pizza, so I have like 1.5 pizza's in the fridge |
| 16:07.04 | symlink | ooh |
| 16:07.34 | blitzrage | chicken, bacon, feta, green pepper, mushroom |
| 16:08.19 | *** join/#asterisk Gr1ncheux (n=devine@AToulouse-257-1-50-134.w90-5.abo.wanadoo.fr) |
| 16:09.07 | Katty | skip the meat, and that's sounds pretty good |
| 16:09.17 | Katty | four cheese souffle! mm! |
| 16:09.48 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 16:11.15 | *** join/#asterisk spyda (n=scott@hera.copi-rite.com) |
| 16:11.20 | spyda | hiya |
| 16:11.38 | mut | cheese ftw! |
| 16:13.41 | mishehu | Katty: you don't drink guiness beer do you? |
| 16:14.14 | Katty | mishehu: ugah, no. i hate bear. |
| 16:14.17 | Katty | i mean beer. |
| 16:14.21 | Katty | bears are awesome. |
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| 16:14.38 | mishehu | Katty: ah, even if you did I wouldn't think you'd drink it because I hear they put beef broth in it for flavor. |
| 16:14.43 | Katty | mew :< |
| 16:16.28 | mishehu | I won't tell you what flavor they are trying to simulate |
| 16:17.41 | Katty | that'd be just...swell. |
| 16:19.29 | blitzrage | ok... pizza time |
| 16:19.35 | *** join/#asterisk shinux__ (n=shinux@196.207.13.202) |
| 16:19.46 | Katty | blitzrage: you could be my new best friend :> |
| 16:20.18 | blitzrage | w00t |
| 16:20.27 | blitzrage | you just need to get to Toronto |
| 16:20.32 | *** join/#asterisk af_ (n=af@ip-156-221.sn2.eutelia.it) |
| 16:20.39 | Katty | butbut toronto is forever away. |
| 16:20.57 | blitzrage | but at least you'd be in the centre of the universe |
| 16:21.17 | blitzrage | and I have a great view.. AND pizza |
| 16:21.31 | blitzrage | and I've got a selection of wine and liquor :) |
| 16:21.31 | mut | pizza? |
| 16:21.46 | blitzrage | the Za |
| 16:21.51 | zoa | ZoA ? |
| 16:21.57 | blitzrage | not so much :) |
| 16:22.05 | zoa | TSsssSS |
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| 16:40.34 | BSDTech | .clear |
| 16:44.44 | cy3o3 | anyone on FWD? heh |
| 16:44.47 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:55.01 | *** join/#asterisk BSDTech (n=RNeese@ppp-71-128-112-135.dsl.irvnca.pacbell.net) |
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| 16:56.11 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 16:56.48 | skirmisha | hello |
| 16:56.56 | skirmisha | have a quick question |
| 16:57.12 | skirmisha | if asterisk have 2 interface one public and one private |
| 16:57.27 | skirmisha | will asterisk do nat of calls that come from private interface |
| 16:57.33 | skirmisha | or i need to configure it? |
| 16:59.32 | BSDTech | If you are behind a NAT you probably need to create an /etc/asterisk/sip.conf file with AT LEAST these two lines: 1) externip=your.external.dotted.IPaddess 2) localnet=192.168.0.0/255.255.255.0 (assuming your local network uses 192.168.0.x addresses) |
| 16:59.37 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 17:00.09 | BSDTech | and on the exten that is external you have to have nat=yes |
| 17:00.29 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 17:01.08 | skirmisha | no no |
| 17:01.17 | skirmisha | asterisk server has 2 interfaces |
| 17:01.28 | skirmisha | one with public ip and one with private ip |
| 17:01.36 | skirmisha | there is no nat at all on both interfaces |
| 17:01.41 | cy3o3 | NO one is on FWD? heh... jeeze |
| 17:01.50 | skirmisha | but calls are coming from private ip |
| 17:02.06 | skirmisha | so will asterisk do nat and forward calls to pubplic interface auto |
| 17:02.10 | skirmisha | that is my question |
| 17:02.34 | Nivex | cy3o3: I am connected to FWD via IAX2 |
| 17:02.45 | BSDTech | your asking if the inbound call will get routed toth right phone ? |
| 17:02.57 | Nivex | cy3o3: I haven't done much with it lately as the connection was rather unstable for awhile. |
| 17:03.08 | BSDTech | it should |
| 17:03.43 | skirmisha | ok lets make it clear |
| 17:03.57 | skirmisha | private ip -->asterisk-->term provider |
| 17:04.08 | skirmisha | asterisk has 2 ip assigned |
| 17:04.15 | skirmisha | one public and one private |
| 17:04.19 | cy3o3 | Nivex: awesome! All I want is someone to call my FWD extension to see if it works, heh |
| 17:04.30 | skirmisha | call comes from private ip |
| 17:04.38 | cy3o3 | can you do that? heh |
| 17:04.48 | skirmisha | also phone is registered with private ip with asterisk but it's not behind nat |
| 17:04.54 | skirmisha | ast listen on 0.0.0.0 |
| 17:04.59 | IPmonger | skirmisha: make sure canreinivte=no |
| 17:05.08 | Nivex | cy3o3: I'm not in front of the thing at the moment. Lemme see if I can remote in and bridge someting to you. |
| 17:05.19 | skirmisha | that's default i think |
| 17:05.22 | cy3o3 | that'd be sweet dude thanks |
| 17:05.38 | skirmisha | IPmonger is that enough for call getting thru? |
| 17:06.00 | skirmisha | anyway i need to test and see |
| 17:06.01 | skirmisha | thanks |
| 17:06.10 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.219.160.108) |
| 17:06.11 | Nivex | cy3o3: what # ? |
| 17:06.25 | *** part/#asterisk DarKnesS_WolF (n=wolf@196.219.160.108) |
| 17:07.12 | cy3o3 | Nivex: 788998 |
| 17:07.51 | Nivex | ringing |
| 17:08.01 | cy3o3 | yay! it wored |
| 17:08.01 | Nivex | should be hearing music |
| 17:08.01 | cy3o3 | worked even |
| 17:08.01 | *** join/#asterisk DroopZ (n=Droopy@0x5551903d.adsl.cybercity.dk) |
| 17:08.03 | cy3o3 | thanks man... I really appreciate it |
| 17:08.06 | Nivex | You're welcome. |
| 17:08.35 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:08.55 | *** join/#asterisk topping_ (n=topping@207.47.6.182.static.nextweb.net) |
| 17:08.59 | cy3o3 | now here's an error I've never seen.. heh |
| 17:08.59 | cy3o3 | Dec 8 10:08:27 WARNING[9800]: interface.c:215 decodeMP3: Junk at the beginning of frame 00000000 |
| 17:11.58 | *** join/#asterisk Arno[Slack] (i=100@master.infinityperl.org) |
| 17:14.27 | *** join/#asterisk _lanman_ (n=lanman@pool-71-253-247-137.nrflva.east.verizon.net) |
| 17:15.28 | *** join/#asterisk |Franky| (n=Franky@host75-233-static.28-87-b.business.telecomitalia.it) |
| 17:15.33 | *** part/#asterisk |Franky| (n=Franky@host75-233-static.28-87-b.business.telecomitalia.it) |
| 17:25.04 | *** join/#asterisk LordBacon (n=kvirc@unaffiliated/frb) |
| 17:25.07 | LordBacon | moo |
| 17:30.43 | LordBacon | does the asterisk system let a user change their extension password, and their voicemail password? |
| 17:31.36 | Crescendo_ | If I'm in a queue, and I'm on a call, how can I tell how many people are waiting? Or if there's anyone waiting at all? |
| 17:32.33 | CunningPike | LordBacon: No, and yes |
| 17:34.48 | LordBacon | is there a way to say #include foo/*.conf ? |
| 17:35.26 | CunningPike | LordBacon: You just did |
| 17:35.28 | *** join/#asterisk alamantia (i=alamanti@nat/digium/x-c5e040dec81a336e) |
| 17:35.43 | LordBacon | ok, so wildcards work in #include... good |
| 17:35.55 | mercestes | lol |
| 17:35.55 | MrChimpy | in what? |
| 17:35.58 | CunningPike | LordBacon: Yes, they do |
| 17:36.05 | MrChimpy | they certainly don't in C |
| 17:36.15 | LordBacon | MrChimpy: I'm not on #C am I? |
| 17:36.33 | CunningPike | LordBacon: In almost every .conf file - voicemail.conf is an exception, as people won't be able to change their passwords |
| 17:37.15 | LordBacon | looks like I can't use wchars in the caller ID |
| 17:37.22 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 17:38.03 | *** join/#asterisk GuriSK (n=10335069@chello085216171092.chello.sk) |
| 17:44.40 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-231.range81-152.btcentralplus.com) |
| 17:45.30 | CunningPike | LordBacon: wchars? |
| 17:45.49 | LordBacon | my office is mostly asian |
| 17:46.11 | LordBacon | I wanted caller ID to show both english and korean |
| 17:46.32 | *** join/#asterisk Aximas (n=Aximas@gw-100.selfnet.de) |
| 17:47.21 | *** join/#asterisk coppice (n=chatzill@40.168.17.210.dyn.pacific.net.hk) |
| 17:48.53 | CunningPike | LordBacon: Ah |
| 17:55.27 | *** join/#asterisk shinux_ (n=shinux@196.207.13.202) |
| 17:55.48 | LordBacon | do snom phones auto-configure the same way grandstream do? |
| 17:56.00 | *** join/#asterisk shinux__ (n=shinux@196.207.13.202) |
| 17:57.08 | wunderkin | ... auto configure? |
| 17:57.30 | LordBacon | http://www.trixbox.org/modules/smartsection/item.php?itemid=18 |
| 17:59.26 | wunderkin | well everyone here doesn't like trixbox.. they just have scripts setup to generate the phone config files... it only lists support for cisco and grandstream, so the answer would be no i guess, but do you mean do snoms do mass provisioning? probably.. |
| 17:59.56 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 18:00.31 | LordBacon | I'm just using trixbox in a vm for now until I know what I'm doing |
| 18:00.50 | LordBacon | then I'll put a real asterisk server on the gateway (native linux, not vm) |
| 18:01.01 | wunderkin | asterisk isnt that hard, you can figure it out |
| 18:01.39 | *** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
| 18:01.47 | LordBacon | I still prefer to experiment in vms than on production machines :) |
| 18:01.52 | jmls | just installed the latest 1.4 svn in a production environment, 75 agents on queues, 150 people in total (Cisco 79xx). Using Jabber in the dialplan, running like a champ. rock solid. Kudos to the development team. Good Job! |
| 18:02.08 | wunderkin | right, jmlsnmnopqyz? |
| 18:02.21 | jmls | que? |
| 18:02.24 | wunderkin | :D |
| 18:02.56 | jmls | I know nathing, I from Barcelona .. |
| 18:03.51 | *** join/#asterisk afrosheen (n=cj@txprotoa2.august.net) |
| 18:04.10 | afrosheen | anyone have problems with missing audio during calls on polycom ip501's? |
| 18:04.18 | CunningPike | jmls: On Spanish TV, Manuel was Portuguese |
| 18:04.27 | CunningPike | afrosheen: NAT? |
| 18:04.41 | afrosheen | CunningPike, nope, everything on a local network |
| 18:04.57 | CunningPike | afrosheen: Codecs? |
| 18:05.05 | afrosheen | CunningPike, ulaw from phones to asterisk, out through a PRI |
| 18:05.09 | wunderkin | is the power plugged in? |
| 18:05.17 | afrosheen | lol |
| 18:05.19 | *** join/#asterisk queuetwo (n=scott@MTRLPQ02-1177745854.sdsl.bell.ca) |
| 18:05.41 | afrosheen | it's like some calls turn into bad cellphone calls, the audio never gets dirty, just drops out from time to time |
| 18:05.44 | CunningPike | afrosheen: Something else to check - make sure that the cable into the bottom of the handset is pushed in all the way. They click twice on the way in |
| 18:06.14 | afrosheen | CunningPike, haha..yeah, I noticed that awhile back on one person's phone, but all these are on new phones I personally double clicked myself |
| 18:06.30 | wunderkin | how hard was it to double click yourself? |
| 18:06.37 | afrosheen | surprisingly simple |
| 18:06.39 | wunderkin | thats kinda kinky |
| 18:06.55 | CunningPike | afrosheen: OK - just checking the obvious. Hmmmm, does it drop in and out, or stay gone? |
| 18:07.03 | afrosheen | CunningPike, it's dropping in and out |
| 18:07.15 | afrosheen | the call just keeps on trucking though, it's not being dropped |
| 18:07.15 | jmls | CunningPike: no way! didn't know that. |
| 18:07.18 | CunningPike | afrosheen: And nothing on the CLI, I take it |
| 18:07.26 | CunningPike | jmls: It's true :) |
| 18:07.37 | afrosheen | nothing on the cli, nothing in the full logs...the pri links isn't throwing any alarms |
| 18:07.42 | jmls | man that it so funny. Bit like the English and Irish ;) |
| 18:07.54 | jmls | (or could we agree on a common foe, the Welsh ?) |
| 18:08.16 | CunningPike | jmls: No - the common foe of the Irish, Welsh and Scots is.....? |
| 18:08.21 | CunningPike | ;) |
| 18:08.29 | afrosheen | liquor shortages |
| 18:08.31 | jmls | heh. |
| 18:09.02 | afrosheen | brb gotta attend a benefits meeting |
| 18:09.13 | *** join/#asterisk Weezey (n=ohno@lan6.LO.iasl.com) |
| 18:09.19 | jmls | There's a bridge between England and Wales. You gotta pay to get into Wales. Kinda like a fine for even thinking of going there |
| 18:09.26 | Weezey | how do I set my QoS to 5 with asterisk? |
| 18:14.16 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 18:14.56 | vooduhal | Does anyone know of a solution to turn on MWI manually for a SIP device? |
| 18:15.02 | vooduhal | Specifically Polycom phones. |
| 18:15.19 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-44.mtco.com) |
| 18:15.21 | jmls | yeah - leave a voicemail ;) |
| 18:15.32 | jmls | sorry. had to say that ;) |
| 18:15.49 | Weezey | vooduhal: mailbox=voicemailcontext ? |
| 18:15.54 | vooduhal | Lol. Can't use mailbox in sip.conf because the voicemails are being left for the agents, not the sip extensions. |
| 18:16.21 | Weezey | I believe you can manually send a NOTIFY |
| 18:16.48 | vooduhal | We've used sipsak to send them in the past but now the phones are no longer responding to it. |
| 18:17.04 | Weezey | hmm |
| 18:18.07 | vooduhal | I wonder if it is ignoring the NOTIFY becuase the phone never registered to receive the notifications. |
| 18:18.26 | vooduhal | I wonder if I had it register to the phones exten (which no box exists) would allow me to send them. |
| 18:22.22 | Weezey | can you make it light on more than one mailbox? |
| 18:22.47 | Weezey | never mind. |
| 18:23.12 | *** join/#asterisk zmef420_ (n=zmef420@lugop.org) |
| 18:23.14 | Weezey | my friggin' asterisk box has narcolepsy. Need a new power supply. |
| 18:24.13 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 18:26.29 | *** join/#asterisk zmef420 (n=zmef420@lugop.org) |
| 18:29.01 | *** join/#asterisk Dobaj (n=root@avonstreet.plus.com) |
| 18:29.24 | Dobaj | hello, anyone got a Siemens Gigaset phone talking to Asterisk? |
| 18:29.24 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-44.mtco.com) |
| 18:31.20 | Dobaj | better still anyone got WMI working on the Gigaset phone? |
| 18:32.36 | *** join/#asterisk mut (n=ana@65.111.222.120) |
| 18:32.44 | hoobastooba | so i have tried both xlite and sjphone... I can connect the to my old server and my new server. my new server however I get no sound. I did playback and some different files, the cli shows they are playing, but i dont here them, the old server works fine. so i know its not my softphone or laptop, it has to be something with asterisk. any help would be appreciated. |
| 18:32.52 | hoobastooba | i am using the latest svn version of asterisk |
| 18:33.29 | hoobastooba | i get no warnings or errors regarding sound or anything. |
| 18:36.16 | *** join/#asterisk DavoFrom818 (n=Vito310@cpe-76-173-43-80.socal.res.rr.com) |
| 18:36.17 | DavoFrom818 | hi |
| 18:36.30 | jmls | lo |
| 18:36.44 | DavoFrom818 | is there any module or addon that i can install to do Email 2 Fax? |
| 18:36.44 | vooduhal | Nm. Got sipsak working again. |
| 18:41.38 | Supaplex | sakit2em |
| 18:41.47 | DavoFrom818 | ? |
| 18:43.57 | Supaplex | DavoFrom818: there's a few options in the wiki |
| 18:46.26 | *** join/#asterisk Block (n=sa@c-2121e353.1257-1-64736c12.cust.bredbandsbolaget.se) |
| 18:48.06 | Block | I use the Sipura-3000 (ATA) with asterisk. Now to the problem. For some reason the Sipura "eats" the tones dialed from the caller analog->sipura->asterisk, thus asterisk does not respond to them. |
| 18:49.11 | *** join/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
| 18:50.51 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 18:51.30 | *** part/#asterisk Dobaj (n=root@avonstreet.plus.com) |
| 18:51.51 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:53.55 | Block | anyone? :P |
| 18:57.07 | rob0 | Block: I don't have a Sipura, but I bet you need to reconfigure the onboard dialplan, read the docs for it? |
| 18:58.08 | Block | sorry for my ignorance. I have passed through numerous of docs but I seem to keep getting it wrong. |
| 18:58.42 | Block | this is my current dial plan "(S0<:192.168.0.24>)" |
| 18:58.49 | Block | 24 is the ast server |
| 19:00.28 | npc105 | Anyone ever had a problem with MONITOR_EXEC not being recognized or honored by Monitor()? |
| 19:00.31 | npc105 | I am trying to get sox to join the two legs and compress the single file into a ogg, however it never calls the MONITOR_EXEC commmand |
| 19:00.59 | *** join/#asterisk Dobaj (n=root@avonstreet.plus.com) |
| 19:01.29 | Dobaj | anyone got zaptel knowledge to help me with an issue? |
| 19:01.46 | mog | whats wrong dobaj |
| 19:02.06 | *** join/#asterisk Ant0n1zz (n=Ant0n1z7@adsl45-209static.access.acn.gr) |
| 19:02.54 | hoobastooba | if I missed a package durring os installation would that affect the sound on asterisk? for example is there a dependency in asterisk for some sort of a sound processing engine? i have arts installed... but i am lost as to why i have no sound. |
| 19:03.24 | Dobaj | I'd like to get asterisk to wait 30secs and if no one picks up the incoming call then got to the unavailable voicemail. However I need to detect if the call has been picked up outside of asterisk so I do get the VM announcements appearing mid chat |
| 19:03.40 | Block | Is there anything wrong with my dial plan? |
| 19:04.30 | mog | Dobaj, waitforring |
| 19:04.41 | mog | its an application that allows you to do exactly |
| 19:04.42 | mog | <PROTECTED> |
| 19:07.03 | Dobaj | mog, I've exten => s,1,wait(30) |
| 19:07.03 | Dobaj | exten => s,2,Voicemail,u3000 |
| 19:07.23 | mog | no there is an application waitforring |
| 19:07.33 | mog | if someone picks up it fails and asterisk never answers line |
| 19:07.43 | mog | that will answer in 30 seconds no matter what |
| 19:07.45 | mog | unless call is over |
| 19:08.05 | Dobaj | but zaptel will get the ring at the sametime fixed line phones ring |
| 19:09.00 | mog | and? |
| 19:09.00 | Dobaj | Asterisk and other phones are on the same line so it'll not ring one then the other |
| 19:10.24 | Block | Can anyone just throw me a generic DP that throws all calls to an asterisk server? |
| 19:10.25 | mog | Dobaj, if you use the application waitforring, asterisk will only pickup the line if the phone is still ringing at the time of time |
| 19:10.26 | mog | out |
| 19:10.41 | mog | instead of wait |
| 19:11.43 | Dobaj | so I could have exten => s,1,waitforring(7) exten=> s,2,voicemail,u3003 and if someone has picked up the call then it'll stop at point one |
| 19:12.12 | mog | yes |
| 19:12.23 | Dobaj | cheers mog |
| 19:12.34 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.229) |
| 19:12.36 | mog | show application waitforring for more info |
| 19:13.56 | npc105 | does anyone use asterisk with broadvoice? I am trying to figure out how to set my broadvoice call forwording (*72/*73) from withon my DP so I can set the forwarding from my phone instead of having to login to their portal... |
| 19:14.54 | Block | anyone with a spa-3000 or any other spa with a pstn chan? |
| 19:16.10 | *** part/#asterisk Dobaj (n=root@avonstreet.plus.com) |
| 19:19.04 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 19:20.43 | Ant0n1zz | hello all, anyone can help me with this error ? I get this when I try to make the latest Zaptel drivers You do not appear to have the sources for the 2.6.9-34.0.2.ELsmp kernel installed. |
| 19:22.56 | rob0 | Ant0n1zz: Install the sources for the 2.6.9-34.0.2.ELsmp kernel. Check your distro documentation for help. |
| 19:24.27 | Ant0n1zz | will do so rob0 :D |
| 19:26.45 | *** join/#asterisk loadsysinc (n=loadsysi@ip67-95-66-69.z66-95-67.customer.algx.net) |
| 19:28.36 | *** join/#asterisk yardB (n=yardie@c-68-44-44-42.hsd1.nj.comcast.net) |
| 19:28.42 | *** part/#asterisk loadsysinc (n=loadsysi@ip67-95-66-69.z66-95-67.customer.algx.net) |
| 19:30.11 | *** join/#asterisk loadsysinc (n=loadsysi@ip67-95-66-69.z66-95-67.customer.algx.net) |
| 19:30.14 | *** join/#asterisk xtr (n=01897268@staff-nat.netnation.com) |
| 19:32.41 | *** join/#asterisk CleanerX (n=nix@p54A395A0.dip0.t-ipconnect.de) |
| 19:35.40 | *** join/#asterisk Alowishus (n=jpenix@ip72-199-253-51.sd.sd.cox.net) |
| 19:36.02 | [hC] | anyone here doing full g729? (aka g729 pass thru) |
| 19:36.14 | [hC] | without needing codec licenses? |
| 19:38.24 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:41.19 | Block | I got it verified, nothing is wrong with my DP or the ast server. Its the settings in the SPA-3000. |
| 19:41.40 | *** join/#asterisk tsurk0 (n=tsurko@145-226.go.evo.bg) |
| 19:43.09 | rpm | how do i allow Playback() to be interrupted by a user dialing an extension and jumping to the next priority? |
| 19:43.41 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 19:46.44 | *** join/#asterisk xnon (n=xnon@200.8.5.123) |
| 19:49.21 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 19:50.12 | champster | Has anyone here used Vonage as a trunk? |
| 19:51.36 | brad_mssw | champster: do you have one of the 'business plus' plans ? we had vonage a couple years back, and it was terrible |
| 19:52.22 | champster | Yes, the business Plus BYOD plan |
| 19:52.43 | champster | But I always get a 404 error when dialing out. |
| 19:52.52 | champster | I am also not seeing it in sip show registry |
| 19:53.32 | brad_mssw | did you use the wiki: http://voip-info.org/tiki-index.php?page=Asterisk%20and%20Vonage |
| 19:55.19 | heh_v_water | so for a little home setup do i go with a linksys sipura or a grandstream or other.. any expericences woul dbe greatly appreciated |
| 19:56.00 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:56.25 | *** join/#asterisk Curus (n=Curus@kbhn-vbrg-sr0-vl209-213-185-8-10.perspektivbredband.net) |
| 19:56.42 | champster | Yes, I did and other examples I found as well. |
| 19:56.44 | Curus | What is the English word for the three tones which signify number error? |
| 19:56.46 | Katty | file: i just put another video on youtube :> |
| 19:57.16 | Crescendo_ | If I'm in a queue, and I'm on a call, how can I tell how many people are waiting? Or if there's anyone waiting at all? |
| 19:57.20 | spyda | oh yeah, it's great. too. *moan* |
| 19:57.28 | Curus | (And how do I make Asterisk play that sound) |
| 19:57.37 | *** join/#asterisk X-Rob (n=rob-x@dsl-124-150-115-171.vic.westnet.com.au) |
| 19:57.47 | Katty | spyda: shut up, you. |
| 19:57.56 | file | Katty: ooh where |
| 19:57.56 | spyda | ha |
| 19:58.00 | Katty | spyda: or i'll transfer all my calls to you :P |
| 19:58.18 | Katty | file: see /query |
| 19:59.34 | IPmonger | Curus: the tone is called reorder |
| 19:59.45 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.89) |
| 20:00.00 | yardB | heh_v linksys sipura works find for me |
| 20:00.23 | spyda | Katty: aren't you already doing that anyway? |
| 20:00.47 | *** join/#asterisk ToTo (n=ToTo@host150-83-dynamic.60-82-r.retail.telecomitalia.it) |
| 20:00.49 | Katty | spyda: you know you love me. |
| 20:01.30 | hmmhesays | I want my mtV |
| 20:02.00 | spyda | I want one of these usb missle launchers! http://www.thinkgeek.com/geektoys/warfare/86b8/ |
| 20:02.42 | Curus | Thanks IPmonger, turns out that wasn't the one I wanted, but "special information tone" is |
| 20:03.41 | Curus | Now I just need a sound file for it |
| 20:04.20 | champster | Call a bunch of number until you find it, then record it. lol |
| 20:05.04 | champster | I want a modem connecting sequence as my ringtone, I think I will have to record it myself. |
| 20:05.12 | Curus | Heh, I guess that'd work. Apparently Zapateller can do it |
| 20:06.48 | peter21 | hi everyone, what is the best snom 320 firmware to use ? |
| 20:08.06 | IPmonger | Curus: did you look at http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Playtones |
| 20:08.12 | champster | Does anyone know if you can force a re-registration to a SIP service? (without restarting asterisk) |
| 20:10.25 | Katty | hmmhesays: and a few hot chicks. |
| 20:10.29 | Katty | hmmhesays: and a few beers. |
| 20:10.48 | Katty | hmmhesays: i'm tellin ya, this august we need to find a few to share. |
| 20:10.49 | Curus | IPmonger: Zapataller does it perfectly, but thanks |
| 20:10.51 | hoobastooba | does sip reload do that? |
| 20:10.56 | hoobastooba | reregister? |
| 20:13.20 | [hC] | anyone here doing full g729? (aka g729 pass thru) , without g729 licenses? |
| 20:13.33 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 20:15.04 | [hC] | ive run into some issues |
| 20:15.54 | *** part/#asterisk hoobastooba (n=ckwall@c-67-182-209-145.hsd1.ut.comcast.net) |
| 20:16.22 | mog | ? |
| 20:16.24 | mog | what issues |
| 20:16.29 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 20:16.33 | brad_mssw | [hC]: you'd run into issues if you had to put someone on hold, or do voicemail, etc ... |
| 20:16.58 | brad_mssw | [hC]: either buy licenses or don't use g729 ... for the most part |
| 20:17.15 | [hC] | well, thats the easy part |
| 20:17.28 | [hC] | the hard part is when meetme wants to do slin for dtmf/silence detection |
| 20:17.32 | SwK | [hc] it should work just fine in passthru mode |
| 20:17.40 | [hC] | or voicemail wants to do it for the same reason. |
| 20:17.43 | mog | no hc |
| 20:17.48 | [hC] | meetme plays the welcome tone in slin |
| 20:17.51 | mog | meetme mixes audio in slin |
| 20:18.01 | mog | you cant mix 729 streams in 729 |
| 20:18.07 | mog | they need to be converted to slin |
| 20:18.14 | [hC] | right so i need g729 licenses for meetme |
| 20:18.15 | mog | you cant do it without transcoding |
| 20:18.29 | mog | all the other files you can get 729 versions of |
| 20:18.33 | SwK | doing meetme, voicemail etc (ie: interacting w/ the media streams) you must transcode thats not passthru |
| 20:18.36 | mog | but you cant have a meetme with slin |
| 20:18.50 | mog | voicemail can be done without licenses swk |
| 20:18.56 | SwK | well voicemail you can store and retrieve like mog says |
| 20:18.58 | mog | you just record files to 729 |
| 20:19.03 | *** join/#asterisk reber (n=reber@cl-157.dub-01.ie.sixxs.net) |
| 20:19.05 | mog | and have 729 prompts |
| 20:19.05 | SwK | og: :P was correcting myself ;) |
| 20:19.11 | [hC] | it seems that the limitations are (that ive found): meetme, no wav file attachments in voicemail ( i could be wrong) no silence detection in voicemail, |
| 20:19.21 | *** part/#asterisk rtcg (n=rtcg@cust-216-59-192-52.t-speed.net) |
| 20:19.40 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.89) |
| 20:19.45 | SwK | [hc] if you intend to do all that stuff, your best bet is just to call up digium and order some licenses |
| 20:19.48 | [hC] | wav file attachments n email i mean. |
| 20:19.59 | *** join/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 20:20.07 | mog | converting 729 to wav would require a license |
| 20:20.23 | [hC] | SwK: have already, just making sure i hadnt missed a hack somewhere. apparently app_conference doesnt use slin, but the other limitations are enough to warrant licenses |
| 20:20.36 | [hC] | customers would be upset if they got g729 email vm attachments/ |
| 20:20.38 | [hC] | :) |
| 20:20.45 | [hC] | man its hard typing with one hand in a cast |
| 20:20.54 | mog | app_conference doesnt mix audio is my understanding |
| 20:20.55 | [hC] | i broke a bone in my hand last friday |
| 20:21.04 | brodiem | [hC] you seem to be typing fairly fast for one hand |
| 20:21.05 | mog | should just get some licenses |
| 20:21.07 | [hC] | app_conference also seems kinda,, alpha. |
| 20:21.23 | [hC] | brodiem: im holding a pen in one hand |
| 20:21.27 | [hC] | and pecking |
| 20:21.27 | brodiem | [hC] lol |
| 20:21.41 | [hC] | its pretty quick, but it can be error prone |
| 20:21.43 | [hC] | :) |
| 20:21.44 | rob0 | The pen is mightier than the sword. |
| 20:22.05 | brodiem | [hC] is half your keyboard blue? haha |
| 20:23.19 | Block | sipura anyone? *cries* |
| 20:23.21 | [hC] | hah the lid's on "_ |
| 20:23.23 | [hC] | :) |
| 20:24.48 | [hC] | mog: hows the new house? |
| 20:24.57 | mog | i love it |
| 20:25.11 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-3e0937e2645eba96) |
| 20:25.24 | rob0 | New house? How can you afford such a thing? |
| 20:25.25 | mog | my first mortgage payment cleared today |
| 20:25.31 | bkruse | mog: woot! |
| 20:25.31 | mog | its my first house |
| 20:25.31 | Qwell[] | yuck, already? |
| 20:25.35 | mog | yeah |
| 20:25.39 | mog | fealt good |
| 20:25.44 | Qwell[] | they usually wait for 45 days |
| 20:25.51 | Qwell[] | with some products anyhow |
| 20:25.55 | mog | its been 45 days |
| 20:25.58 | bkruse | :] |
| 20:26.00 | Qwell[] | seriously? |
| 20:26.02 | mog | yeah |
| 20:26.09 | Qwell[] | well then |
| 20:26.09 | mog | have you payed your first one? |
| 20:26.17 | Qwell[] | yeah, couple weeks ago |
| 20:27.20 | *** join/#asterisk tsurk0 (n=tsurko@145-226.go.evo.bg) |
| 20:29.01 | BSDTech | ok where is 1.4 -r |
| 20:29.09 | Qwell[] | -r what? |
| 20:29.26 | BSDTech | -r =-Release |
| 20:29.32 | mog | well i bought my house a few weeks after you did |
| 20:29.35 | mog | just seemed longer |
| 20:29.41 | [hC] | wow |
| 20:29.43 | [hC] | 45 days al;ready |
| 20:29.48 | Qwell[] | mog: guess so.. doesn't really feel like I've been here 2 months yet |
| 20:30.01 | Qwell[] | BSDTech: doesn't exist |
| 20:30.03 | mog | i have nothing but boxes on first floor still |
| 20:30.04 | [hC] | since astricon |
| 20:30.09 | mog | but i do have my white board wall |
| 20:30.14 | Qwell[] | heh |
| 20:30.23 | rob0 | I happen to like boxes. ;) |
| 20:30.26 | mog | still dont have my projecter |
| 20:30.39 | BSDTech | thats what I asking for . For them to get off thier butts and get it out |
| 20:30.40 | Qwell[] | I was thinking yesterday - I have the perfect place for a projector |
| 20:30.52 | Qwell[] | BSDTech: submit patches to fix bugs |
| 20:31.02 | mog | waiting for asterisk community to send me a wedding present |
| 20:31.06 | mog | hint i want a projector ^_^ |
| 20:31.08 | BSDTech | I do on BSD |
| 20:31.27 | BSDTech | and I need 1.4-r so we can find and patch all the breaks |
| 20:31.41 | Qwell[] | BSDTech: there is a beta3, and you can get svn |
| 20:32.28 | BSDTech | just have to svn and patch daily |
| 20:35.40 | danp | 1.4 was looking pretty cool until was bitten by this bug: http://bugs.digium.com/view.php?id=8416 |
| 20:36.32 | danp | +i |
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| 20:42.43 | *** part/#asterisk BSDTech (n=RNeese@ppp-71-128-112-135.dsl.irvnca.pacbell.net) |
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| 20:47.26 | jarrod | anyone have problems using cisco pstn as gateway and receiving short intervals of 'hiss' |
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| 21:10.38 | Block | meh, what are the options to suppress DTMF tones in asterisk? |
| 21:13.50 | *** join/#asterisk SimoAmi (n=SimoAmi@ip67-91-253-242.z253-91-67.customer.algx.net) |
| 21:13.58 | SimoAmi | hi there |
| 21:16.14 | SimoAmi | how can i change the sip registration timeout |
| 21:16.31 | SimoAmi | qualify=yes sets the time between 2 retries |
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| 21:25.03 | *** join/#asterisk GrabrielA (n=metfan@189.136.84.9) |
| 21:25.07 | GrabrielA | hello all!!! |
| 21:25.53 | Metfan | I need help to use Asterisk with a Norstar MICS |
| 21:26.03 | Metfan | please help |
| 21:26.41 | Nugget | we can't help you unless you actually ask a question. |
| 21:26.53 | Metfan | ok ok |
| 21:27.22 | LordBacon | WHat is the airspeed velocity of an unladen swallow? |
| 21:27.27 | Metfan | What is the cost of telephone suppor per hour in Mexico????? hehehe |
| 21:27.35 | Nugget | african or european swallow? |
| 21:28.36 | joe | hehe |
| 21:28.58 | joe | Metfan: what is the problem would be a good place to start |
| 21:30.13 | Metfan | Ok, I want to place Asterisk between a Norstar MICS and the Telco using T1 links |
| 21:31.04 | Metfan | We have found bugs regardin this in various forums, and I think we have the same problem |
| 21:31.56 | Metfan | Astersik does not pass the correct signal from Nortel span to Telco span T1 link |
| 21:32.41 | *** join/#asterisk peter21 (n=Peter@203.6.132.1) |
| 21:33.04 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 21:33.09 | peter21 | hi everyone. does anyone know when ABE 1.3 will be ready for download ? |
| 21:33.56 | Metfan | does anybody has done a succesfull integration with Norstars??? |
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| 21:48.29 | afrosheen | I'm getting " interface.c: Junk at the beginning of frame" in some spots, is this something to be concerned about? |
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| 21:58.17 | robin_sz | meep? |
| 21:59.45 | afrosheen | pong |
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| 22:06.56 | LordBacon | ok, I have a snom phone, and I can't figure out how to get in to the menu without rebooting |
| 22:07.58 | *** join/#asterisk te_lo_meto_mami (n=te_lo_me@adsl-11-92-66.mia.bellsouth.net) |
| 22:09.45 | psiforce | does anyone know why asterisk seems to only run on 1 of my cpus on a smp machine |
| 22:09.46 | psiforce | cat /proc/cpuinfo shows 2 cpus |
| 22:10.09 | Supaplex | how many asterisk threads are there? |
| 22:12.42 | *** part/#asterisk pdunkel (n=pdunkel@213.235.192.27) |
| 22:17.14 | *** join/#asterisk aao_pwner (i=asd@c-24-21-91-140.hsd1.mn.comcast.net) |
| 22:19.09 | ChkDigit | Hey all, |
| 22:19.36 | ChkDigit | I've got a Mediatrix 1204 gateway that is calling into *, but using the Caller-ID as the From: address. |
| 22:20.25 | ChkDigit | How do you change the From address in the SIP session, or just tell * to send it to a default context. |
| 22:20.42 | ChkDigit | Presently, I'm just getting a SIP authentication failure. |
| 22:21.49 | *** join/#asterisk RoyK (n=roy@ti211310a080-14732.bb.online.no) |
| 22:23.31 | *** join/#asterisk lTtLsNk (n=ltasd@59.60.88.210) |
| 22:23.59 | LordBacon | wtf is going on |
| 22:24.50 | LordBacon | I setup like 30 extensions in freepbx, and I don't see them in /etc/asterisk |
| 22:27.51 | LordBacon | there. found them |
| 22:28.14 | robin_sz | LordBacon, there is a special command to type for help on that very subject |
| 22:28.17 | LordBacon | now to get asterisk to allow NxxNxxxx |
| 22:29.32 | robin_sz | on your IRC client .. type "/join #freepbx" |
| 22:29.38 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 22:34.23 | hmmhesays | google spreadsheets rocks |
| 22:35.03 | psiforce | supaplex: only 2 or so... and I have over 30 g729 calls being transcoded to pstn |
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| 23:06.08 | peter21 | anyone still online ? i'm confused with the context matching. I want outgoing calls to match contexts like [international] - currently cdr shows default |
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| 23:16.56 | *** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com) |
| 23:17.08 | e-milio | hello all |
| 23:17.41 | CunningPike | peter21: Are you placing outgoing calls in a context called 'international'? |
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| 23:45.48 | LordBacon | can I set this grandstream to have a password that isn't just numbers? |
| 23:53.19 | *** part/#asterisk LordBacon (n=kvirc@unaffiliated/frb) |
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