| 00:01.10 | [TK]D-Fender | Oryn : it isn't the nature of * or SIP to start ringing a phone that was targeted in a current dial just because it becomes available. |
| 00:01.31 | JonR800 | would it be possible to have multiple dial statements? |
| 00:01.43 | JonR800 | in a row... with say a 2-5 second timeout.. hackish. |
| 00:02.56 | [TK]D-Fender | JonR800 : He's not trying to make a queue. He wants it so taht if he dials 5 phones at once, and one of the phone is on a call but ends it while the others are ringing, that it would BEGIN ringing again as part of the previously issued Dial command. Sort of retro-active. |
| 00:03.01 | [TK]D-Fender | And the answer to that is "no". |
| 00:04.06 | JonR800 | I understand what he was asking.. just wondering why the situation called for it I guess. |
| 00:04.09 | Oryn | JonR800: thats what I've ended up doing, but I wondered if there was a proper way |
| 00:04.34 | JonR800 | I don't think there is. |
| 00:05.00 | snewpy | isn't it called call waiting? :) |
| 00:05.01 | Oryn | hmm, customers eh, you can never please them |
| 00:05.16 | snewpy | have a queue with all the phones, all with call waiting enabled |
| 00:05.44 | Oryn | yup, but they complained about beeping and missing out on what people were saying then it beeped |
| 00:05.49 | Oryn | so I turned it off |
| 00:05.50 | snewpy | when a call comes in, ring-all style, it rings all the phones, including phones where the agent is on the phone, when that agent hangs up, his phone starts ringing again |
| 00:06.04 | snewpy | Oryn: use a decent phone with unassuming call waiting beeps? :) |
| 00:06.22 | JonR800 | im sure the customer would love that added cost :) |
| 00:06.24 | Oryn | snewpy: hehe, I'm using sipuras |
| 00:06.39 | Oryn | JonR800: you hit the nail on the head |
| 00:06.58 | snewpy | JonR800: depends on how imporant the feature is to 'em, I guess :) |
| 00:07.04 | st3v | I am trying to set up in my dialplan a way to make a 3way call on the zap channels, but how can I send a FLASH to the outside line? |
| 00:07.11 | JonR800 | very true. |
| 00:07.27 | snewpy | I think on the Polycoms you can disable the call waiting beep all together, so it's just shown on the display |
| 00:07.33 | st3v | nevermind |
| 00:08.11 | [TK]D-Fender | snewpy : Pretty sure of that here. |
| 00:08.56 | snewpy | [TK]D-Fender: yeah, I think it's set in the <callProgTones> tag |
| 00:09.00 | JonR800 | could you change the CW tone on the sipura? |
| 00:09.01 | Oryn | speaking of sending flashes, I have an isdn phone here that I'm using, its connected to a zap channel (kinda in reverse) how to you send a flash to * |
| 00:09.31 | JonR800 | I see it under the regional tab in the advanced config.. |
| 00:09.37 | *** join/#asterisk razu (n=razu@213-35-173-39-dsl.prn.estpak.ee) |
| 00:09.45 | Oryn | the R button (uk equiv of flash) does nothing |
| 00:10.31 | snewpy | JonR800: I think the prob on the Sipura phones is that they don't superimpose the call waiting tone over the audio, they just mute it, play the beep, unmute it... so it's really hard to carry on a conversation while it's beeping |
| 00:10.59 | *** join/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
| 00:11.06 | *** join/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 00:11.23 | JonR800 | ahh that puts an end to that idea. |
| 00:11.32 | darwin_35 | any one here have the latest firmware for polycom 501 |
| 00:11.41 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 00:11.49 | darwin_35 | I need it bad I have 1 phones that need updating |
| 00:11.56 | darwin_35 | that just fell in my lap |
| 00:15.55 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 00:15.56 | *** join/#asterisk twilson (n=twilson@mail.logivox.net) |
| 00:16.39 | darwin_35 | they have all diff ver of soft ware and having issues with getting them to work all the interfaces are diff in the diff ver. |
| 00:16.48 | oceanlan|dustin | for an IAX connection, what should my context be?? i think it is context=default correct? |
| 00:18.49 | *** join/#asterisk tuxinator_linux (n=Jone_Doe@70-32-106-248.ontrca.adelphia.net) |
| 00:19.22 | *** join/#asterisk kio (n=kio@ool-4577a5a5.dyn.optonline.net) |
| 00:22.34 | *** join/#asterisk nurfe (n=rgff@h24-207-70-68.dlt.dccnet.com) |
| 00:22.52 | [av]bani | OMG ILLEGAL POLYCOM WAREZ |
| 00:23.03 | [av]bani | (dunno what polycom is smoking...) |
| 00:24.05 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 00:24.06 | *** topic/#asterisk is Asterisk 1.2.3 Released (If you are running 1.2.2, this is a critical update) |
| 00:25.03 | *** join/#asterisk bn-7bc (n=bjarne@pppoecl73202.minlos.no) |
| 00:25.10 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 00:25.16 | Zodiacal | how can i get my * box to really shutdown when i tell it to shut down |
| 00:25.29 | Zodiacal | do i have to install some kind of power managment driver in linux? |
| 00:25.36 | [av]bani | ?? |
| 00:25.48 | Zodiacal | i want the box to turn off |
| 00:26.11 | [av]bani | you need acpi enabled |
| 00:26.25 | Zodiacal | know how to do that off hand? |
| 00:26.35 | [av]bani | pc bios |
| 00:26.49 | [av]bani | depends on your motherboard vendor |
| 00:26.58 | Zodiacal | its not os or software controled? |
| 00:27.03 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:27.06 | Ariel_ | hello everyone |
| 00:27.14 | [av]bani | nope, acpi is at the mercy of your motherboard vendor |
| 00:27.29 | Zodiacal | avbani okie, i will go see, but i think its set to on.. |
| 00:27.35 | Zodiacal | thanks! |
| 00:27.44 | darwin_35 | no |
| 00:27.54 | darwin_35 | I dont wants warz |
| 00:28.14 | darwin_35 | I just want to get the latest firmware and flash them and get them used |
| 00:28.45 | bn-7bc | I have the folowing line in sip conf register => xxxx:yyyy:zzzz@213.160.242.135/1051 but when I set up exstensions.conf to handle the call I must use the incomming pstn number instead of 1051 is ther a fault in my setyp or is it a problem at my provider (using astreisk 1.2.3)? |
| 00:29.47 | darwin_35 | 1.2.1 - 1.2.3 suck donkeyballs |
| 00:29.56 | *** join/#asterisk nmsclera (i=nmsclera@206-169-194-79.gen.twtelecom.net) |
| 00:30.14 | [av]bani | darwin_35: how so? |
| 00:30.14 | snewpy | darwin_35: who'd you buy them from? your reseller should be able to give you access to the firmware if they're certified resellers |
| 00:30.28 | [av]bani | http://www.freedomphones.net/polycom/files/ |
| 00:30.59 | darwin_35 | a company that was tossed out of the building 4 monhs ago . left them behind today the had a auction in the building and we bought them |
| 00:31.05 | snewpy | [av]bani: they're a couple of versions behind... 1.6.4 is the latest |
| 00:31.29 | *** join/#asterisk Grok_ (n=grok@c-71-196-75-163.hsd1.fl.comcast.net) |
| 00:31.53 | Ariel_ | [av]bani, they only have up to 1.6.2 there is a newer release 1.6.4 |
| 00:32.37 | bn-7bc | was my question unclear? |
| 00:33.06 | Ariel_ | bn-7bc, which context do you have the call going to? |
| 00:33.19 | [av]bani | snewpy: polycom should remove head from ass and make firmware available for all customers |
| 00:33.45 | snewpy | [av]bani: you're preaching to the choir, man :) |
| 00:34.02 | [av]bani | whats wrong with 1.6.1 though? |
| 00:34.39 | Ariel_ | actually that site has 1.6.2 There have been some bugs fixes in the newer one. with sound |
| 00:34.40 | [av]bani | darwin_35: how much you pay for them? :)) |
| 00:34.57 | snewpy | there's a few bugs that could likely be tickled between 1.6.2 and 1.6.4... nothing earth shattering tho |
| 00:35.10 | *** join/#asterisk vn (n=vn@modemcable184.104-203-24.mc.videotron.ca) |
| 00:35.19 | bn-7bc | Ariel_: not shore what you mean this is an incomming call form pstn , the 1051 extension is defined in the Demo contect |
| 00:36.45 | oceanlan|dustin | Where are the context= in the sip.conf and Iax.conf files generated from?? |
| 00:36.47 | darwin_35 | like 25 bucks each |
| 00:36.58 | Ariel_ | bn-7bc, your account for the provider that you register to context= should have the correct includes |
| 00:37.12 | Ariel_ | oceanlan|dustin, vi your own doing |
| 00:37.19 | nmsclera | So, I *THINK* I have everything configured properly for this PRI/Zap channels, but if someone could take a look at http://pastebin.com/526656 and tell me WTF it means, it would be greatly appreciated. |
| 00:37.20 | oceanlan|dustin | I assume extensions.conf, but I am lost in that huge file! |
| 00:37.43 | *** part/#asterisk vn (n=vn@modemcable184.104-203-24.mc.videotron.ca) |
| 00:37.45 | oceanlan|dustin | Ariel_: do i make it in extensions.conf? |
| 00:38.06 | Ariel_ | oceanlan|dustin, yes it's how you setup your dial routes and plans |
| 00:38.15 | [av]bani | darwin_35: $25 @_@ |
| 00:38.24 | Ariel_ | it does not auto generate you need to set them up. |
| 00:38.29 | nmsclera | (This occurrs when trying to make a call from a polycom SIP extension) |
| 00:38.35 | [av]bani | darwin_35: i'll buy 10 :) |
| 00:38.36 | oceanlan|dustin | Ariel_: any ideas where I can get a doc on that? |
| 00:38.52 | Ariel_ | ~docs |
| 00:38.53 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 00:39.08 | bn-7bc | Ariel_: what did you mean bu that, can you plz give an example? |
| 00:39.17 | oceanlan|dustin | Ariel_: I have the asterisk book, but I dont understand the terminology |
| 00:39.48 | oceanlan|dustin | Ariel_: I get lost in all the &,Dial,0,1, @&$%*$^ stuff... |
| 00:40.49 | oceanlan|dustin | Areil_: I am not sure how if i make a context, how will it know all the other dfault things...like the built in stuff (aka 1234 is test, #800 is for meetme, etc..) |
| 00:42.09 | st3v | I want to have asterisk wait until a call is initiated, then wait until the user presses 5, to do a Flash() and dial another number, then Flash() again. How can I do that? |
| 00:42.13 | *** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 00:42.26 | oceanlan|dustin | I guess I dont understand the theory of exten -> 100,399,Dial,x to know how to build it.. |
| 00:43.05 | Krill | anyone had success integrating google talk with asterisk? |
| 00:43.23 | oceanlan|dustin | haha, good question! I have been wondering about that |
| 00:43.33 | bn-7bc | Ariel_: sorry, yes I have got the context set in sip conf I have the provider set as friend (allso use it for calls to pstn) |
| 00:43.38 | Ariel_ | oh boy... well a context is a place in the file to do something. the exten => 100,1,dial(Sip/100,20) means extension 100 is device sip/100 |
| 00:44.18 | oceanlan|dustin | hmmm...i am going to go back and re-read that section in the book...I may have some questions later. |
| 00:44.58 | Ariel_ | bn-7bc, what do you get on the cli with verbose 9 when the call comes in. |
| 00:46.28 | bn-7bc | hold on |
| 00:47.44 | oceanlan|dustin | Ariel_: in the Asterisk book, what section should I be reading under? none of them say: Context!! |
| 00:47.49 | oceanlan|dustin | Dialplan maybe? |
| 00:49.06 | mzo | when i browse the manual i take the book and throw it across the room, it always lands open on the page i need. |
| 00:49.41 | Ariel_ | it's part of the dial plan yes. But context= is everywhere in the setups. it's also in sip.conf and iax.conf |
| 00:49.41 | bn-7bc | Ariel_: http://pastebin.com/526671 |
| 00:50.34 | bn-7bc | Ariel_: ignor the bit about congestion the client is just discconected atm |
| 00:50.58 | Ariel_ | bn-7bc, are you trying to dial yourself or a device??? |
| 00:51.30 | Ariel_ | your sip device is sip/number your trying to dial? |
| 00:52.28 | bn-7bc | Ariel_: not shore what you ask? |
| 00:52.56 | Ariel_ | what is your dial string to the divce look like? exten => blah |
| 00:52.56 | bn-7bc | sip/bjarne is an account for a softphone |
| 00:53.17 | bn-7bc | hold on |
| 00:54.01 | bn-7bc | Ariel_: Dial(SIP/bjarne,50,tTwW) |
| 00:54.34 | bn-7bc | Ariel_: bjarne is defined in sip.conf |
| 00:55.09 | *** join/#asterisk funxion (n=nunya@host-64-110-51-254.hlm.ses-americom.net) |
| 00:55.18 | *** part/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 00:55.37 | *** join/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 00:57.20 | bn-7bc | Ariel_: 3739xxxx is my incomming number (the pstn number my provider asigned to me) |
| 00:58.18 | bn-7bc | Ariel_: and it's this number that is beeng used as extension instead od 1051 |
| 00:58.30 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 00:59.02 | Ariel_ | bn-7bc, so you have exten => 1051,1,dial(sip/bjarne,50,tTwW) and you tried to dial your did to see if it would come back into the box? |
| 00:59.24 | *** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net) |
| 00:59.31 | Ariel_ | Jan 28 01:48:34 NOTICE[3227]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 00:59.37 | Underhand | when i call out via asterisk to a FWD service like the echo test, things work. when i dial another FWD user, i get: Forbidden - wrong password on authentication for INVITE to '"xxxxxx" <sip:xxxxxx@ip>;tag=.....' |
| 00:59.43 | Underhand | where should i be looking to debug this? |
| 00:59.49 | darwin_35 | ok who can explain how to setup a tftpserver for polycom update |
| 01:00.05 | darwin_35 | is it simple |
| 01:00.09 | darwin_35 | or hard |
| 01:00.10 | Ariel_ | darwin_35, yes |
| 01:00.17 | darwin_35 | ? |
| 01:00.20 | Ariel_ | but I use ftp not tftp |
| 01:00.38 | darwin_35 | so you ftp the software up to each phone |
| 01:00.44 | Ariel_ | Underhand, use canreinvite=no |
| 01:01.07 | Ariel_ | darwin_35, no you create an ftp server and point the phones to it. |
| 01:01.11 | bn-7bc | Ariel_: tha channel is geting crowded, mind if we tak the rest in private so I don't lose any important info |
| 01:01.27 | Underhand | Ariel_: where? |
| 01:01.30 | darwin_35 | ariel can you pvt me a min and explain ? |
| 01:01.40 | *** join/#asterisk SPoon_TSX (n=Administ@h24-83-96-211.sbm.shawcable.net) |
| 01:01.44 | Ariel_ | no pvt |
| 01:01.48 | darwin_35 | ok |
| 01:02.44 | Ariel_ | darwin_35, here is a good start for you. http://www.voip-info.org/wiki-Polycom+Phones |
| 01:02.44 | oceanlan|dustin | OMG! my buddie ... we both admin this one Asterisk box... and I have been geeking on it for weeks now... |
| 01:03.04 | oceanlan|dustin | and the little bastard kicked me off! and told me to go get a life! its friday nite! |
| 01:03.11 | Ariel_ | nice |
| 01:03.18 | oceanlan|dustin | and this little asshole works for me! |
| 01:03.41 | oceanlan|dustin | wow... |
| 01:03.41 | Ariel_ | oceanlan|dustin, well send him home and keep working. |
| 01:03.49 | oceanlan|dustin | hahahaah! |
| 01:03.58 | Underhand | ariel: i'm already using canreinvite=no on my connection to my sip phone. i just tried adding it to the FWD section, but that made no difference. |
| 01:04.15 | oceanlan|dustin | i think there is some conspirecy between him and my wife! |
| 01:04.48 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 01:04.54 | *** join/#asterisk QuAd|Haudrauf (n=hau@port-212-202-185-252.dynamic.qsc.de) [NETSPLIT VICTIM] |
| 01:04.54 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) [NETSPLIT VICTIM] |
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| 01:04.54 | *** join/#asterisk ptiggerdine (n=ptiggerd@c220-237-93-88.rochd1.qld.optusnet.com.au) [NETSPLIT VICTIM] |
| 01:04.54 | *** join/#asterisk artichoke (n=artichok@c220-237-68-248.kelvn1.qld.optusnet.com.au) [NETSPLIT VICTIM] |
| 01:04.54 | *** join/#asterisk angler (n=angler@gateway.digium.com) [NETSPLIT VICTIM] |
| 01:04.54 | *** join/#asterisk angler_ (n=angler@24.214.255.222) [NETSPLIT VICTIM] |
| 01:04.54 | *** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au) [NETSPLIT VICTIM] |
| 01:05.23 | darwin_35 | ok thnks |
| 01:05.28 | darwin_35 | he made me do it |
| 01:05.37 | darwin_35 | I told him not to |
| 01:05.42 | *** part/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 01:05.43 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 01:06.39 | Ariel_ | Underhand, have either setup the sip.conf section incorrect or the dial string. If you can post them on pastebin.ca without the password so we can take a look and see if we can help |
| 01:07.00 | SPoon_TSX | Hi Just wondering if anyone have any experience woth Aastra 480i? |
| 01:07.09 | Ariel_ | brb need to get my baby girl to sleep. |
| 01:07.29 | Ariel_ | SPoon_TSX, yes but I can't right now. In about 20 mintues maybe. |
| 01:08.11 | SPoon_TSX | Ariel_: Thanks, I wait until you come back. |
| 01:10.27 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 01:10.57 | Zodiacal | is there a way to have asterisk directly dial an ext. when it receives a incoming call, Other than having the trunk point to a context that then dials? is there a way to tell the trunk to immediatly transfer calls to an ext.? right now it has a delay where the caller hears two rings before my ext. hears one... im probably grasping at straws here but do you think the "handover" of the trunk to the ext. is the delay? cuz asterisk detects the c |
| 01:11.00 | Zodiacal | pstn |
| 01:11.01 | Underhand | Ariel_: http://pastebin.ca/38860 |
| 01:11.29 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 01:15.34 | *** join/#asterisk pepsiaskaya (n=Mouarfff@ip-213-49-171-205.dsl.scarlet.be) |
| 01:16.04 | *** part/#asterisk pepsiaskaya (n=Mouarfff@ip-213-49-171-205.dsl.scarlet.be) |
| 01:16.24 | *** part/#asterisk justinu (n=justin@72.18.13.34) |
| 01:16.58 | *** join/#asterisk gorauskas (n=gorauska@66-224-20-131.atgi.net) |
| 01:17.25 | *** join/#asterisk SPoon_TSX (n=Administ@h24-83-96-211.sbm.shawcable.net) |
| 01:18.48 | Ariel_ | Zodiacal, most pstn in the us waits 2 rings for the caller ID info |
| 01:18.48 | SPoon_TSX | Ariel_: I am wondering how can I gt the BLC work on Aastra 480i with Asterisk. |
| 01:18.56 | Ariel_ | blc? |
| 01:18.58 | Zodiacal | ariel i disabled that |
| 01:19.02 | Zodiacal | ariel and fax detection |
| 01:19.10 | SPoon_TSX | Busy Lamp something. |
| 01:19.23 | Zodiacal | ariel any other ideas come to mind? |
| 01:19.30 | SPoon_TSX | BLF. |
| 01:19.34 | Zodiacal | also, when dialing out theres a 10 second delay before either side hears the first ring |
| 01:19.55 | Ariel_ | argh baby crying again... just a minute |
| 01:20.32 | Zodiacal | [baby crying] |
| 01:20.37 | Zodiacal | hangup() |
| 01:20.42 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 01:20.52 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 01:23.33 | Ariel_ | SPoon_TSX, the setup on the wiki for tftp is pretty easy on the phones. The blc like on top (Red one is for message waiting) |
| 01:23.54 | Ariel_ | Zodiacal, are you running amp, normal asterisk or any special setup? |
| 01:24.23 | SPoon_TSX | Ariel_: ? Which wiki? tftp? |
| 01:24.32 | Zodiacal | ariel normal |
| 01:24.46 | Zodiacal | ariel lastest cvs checkout |
| 01:24.53 | Zodiacal | very very clean dialplan |
| 01:25.00 | Zodiacal | just answers and dials my ext. |
| 01:25.13 | SPoon_TSX | Ariel_: May I have more information which wiki you are referring to? |
| 01:25.26 | Zodiacal | not amp |
| 01:25.47 | Zodiacal | i tried asterisk@home and was getting these delays too so i thought a clean install would maybe fix it, but i still have the delays |
| 01:26.03 | Ariel_ | SPoon_TSX, yes just a sec |
| 01:26.04 | Zodiacal | my existing key meridian system doesn't do this.. |
| 01:26.22 | alx_ | cool trix for ppl using locked ATA, use iptables to forward the traffic to your asterisk instead of going out to the provider (f.ex vonage) .. working great here |
| 01:26.22 | Ariel_ | Zodiacal, humm what signal are you doing for the zap ports |
| 01:26.31 | Zodiacal | ks |
| 01:27.03 | nmsclera | new PRI, TE110P, asterisk 1.2.3, outgoing calls seem to be fine, but when an incoming calls, I get this message "Ring requested on unconfigured channel 0/1", can someone tell me what this means? |
| 01:28.17 | Ariel_ | SPoon_TSX, do you have these guides from sayson? http://www.sayson.com/support.htm#Download%20User%20Guides |
| 01:28.44 | Ariel_ | you don't have the channels setup correctly |
| 01:29.03 | Zodiacal | ariel any and all sugguestions to speed this up, no matter how small, will be greatly appreshiated. |
| 01:29.04 | SPoon_TSX | Ariel_: SIP Admin Guide? |
| 01:29.11 | Zodiacal | ariel could it be the speed of the pc? |
| 01:29.20 | Zodiacal | its a PIII 550Mhz 384MB's ram |
| 01:29.24 | nmsclera | Ariel_: What would be a good starting point to track down what the deal is? Again, outbound calls seem fine |
| 01:30.12 | Underhand | Ariel_: was the pastebin url useful? |
| 01:30.57 | Ariel_ | I am looking at 3 different people setups give me a few minutes to read please |
| 01:31.16 | Underhand | no probs |
| 01:31.30 | *** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-53-16.zoominternet.net) |
| 01:31.30 | Zodiacal | no rush for me either :) |
| 01:32.31 | iCEBrkr | Ariel_: Hurry up damnit!! |
| 01:32.33 | iCEBrkr | :P |
| 01:32.51 | Ariel_ | Underhand, yes it was just a sec. |
| 01:32.56 | Ariel_ | iCEBrkr, hahaha |
| 01:33.23 | Ariel_ | nmsclera, do you have a channel=1 setup in the zapata.conf |
| 01:36.13 | Zodiacal | i wish i could switch to a voip provider, but i have a damn verizon contract |
| 01:36.18 | Cresl1n | no! |
| 01:36.21 | Zodiacal | i have to wait that out first |
| 01:36.24 | Cresl1n | no switchy switchy! |
| 01:36.50 | SPoon_TSX | Ariel_: I did what it told me. But the icon doesn't change at all. But on Asterisk it does show the extension DOES INUSE. ANy idea? BTw, I have multipule SIP Proxy setup. would it be the problem? |
| 01:37.41 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 01:38.18 | Ariel_ | SPoon_TSX, I would start by setting up the phone as basic as you can. |
| 01:39.15 | Ariel_ | Underhand, change to iax setup for fwd and use there inkey it will work better. But put nat=yes and canreinvite=no and allow=ulaw disallow=all for starters in the sip.conf for fwd |
| 01:39.42 | Zodiacal | ariel have you used sccp before? think that would be faster? |
| 01:39.48 | Zodiacal | than sip |
| 01:39.55 | nmsclera | Ariel_: No I do not, but see, we're not using channel 1 on the PRI |
| 01:40.10 | nmsclera | Ariel_: 2-11 are phone line.. |
| 01:40.33 | Cresl1n | Ariel_: cafe? tu hablas en espanol? |
| 01:40.41 | Ariel_ | Cresl1n, si |
| 01:40.48 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 01:40.51 | Ariel_ | nmsclera, but the system says it's coming in on 0/1 |
| 01:40.58 | Ariel_ | which is telling you there setup is off then |
| 01:41.16 | nmsclera | Ariel_: What is the 0 portion of that identifer? |
| 01:41.22 | Ariel_ | Zodiacal, sccp is not any faster it's actually slower |
| 01:41.30 | Zodiacal | okie |
| 01:41.32 | Zodiacal | :/ |
| 01:42.01 | Qwell | Ariel_: I'd disagree with that statement... |
| 01:42.03 | Ariel_ | nmsclera, you have a pri located where? and are you sure they configured it correctly in getting your numbers. |
| 01:42.16 | Zodiacal | qwell you use sccp? |
| 01:42.21 | Dr-Linux | hi Qwell :) |
| 01:42.32 | Ariel_ | Qwell, it really depends on the dial rules |
| 01:42.35 | Underhand | Ariel_: i'll play with iax, but i'd like to understand why this isn't working with sip in the meantime. i put nat=yes, canreinvite=no, disallow=all, allow=ulaw into sip.conf, but no joy. |
| 01:42.53 | nmsclera | Ariel_: Here's the deal.. it's the TWTC "Versapack" deal. They come into an IAD, then the IAD has a port that comes out PRI with preconfigured channels as B and D |
| 01:43.06 | Qwell | Zodiacal: I do |
| 01:43.11 | Zodiacal | qwell is it faster? |
| 01:43.13 | Zodiacal | than sip? |
| 01:43.17 | Qwell | Ariel_: sure, but yours was a pretty blanket statement. :) |
| 01:43.28 | nmsclera | Ariel_: The Cut Sheet I have doesn't specify a B channel on 1 |
| 01:43.41 | Ariel_ | Qwell, get a new guy to use sccp correctly from the start.... humm don't think so |
| 01:43.43 | Cresl1n | nmsclera: it means that the DID it was sent to is in one of two states |
| 01:43.47 | Cresl1n | nmsclera: (the call) |
| 01:43.51 | Qwell | Zodiacal: it was designed as a very skinny client control protocol (sccp).. |
| 01:43.56 | Qwell | Ariel_: with *? It's quite easy |
| 01:44.10 | Ariel_ | Qwell, with the cisco router rules not |
| 01:44.13 | Cresl1n | nmsclera: either A.) You don't have your channels setup correctly in zapata.conf for your PRI |
| 01:44.22 | Zodiacal | qwell im trying speed up my *. outside caller hears two rings before my ext. hears one. pstn. |
| 01:44.35 | Qwell | Zodiacal: that's because you're trying to use callerid |
| 01:44.38 | Qwell | on an fxo? |
| 01:44.48 | Zodiacal | qwell, fxo, yeah, i disabled CID and fax detection |
| 01:44.50 | Zodiacal | still slow.. |
| 01:44.52 | Qwell | and I'd guess your line doesn't HAVE callerid? :) |
| 01:44.58 | Cresl1n | nmsclera: and a good way to test that is to add an "s" extension to the context that you are sending your calls into from the PRI (context=whatever in zapata.conf) |
| 01:44.59 | Zodiacal | it doesn't have callerid |
| 01:45.00 | Zodiacal | no |
| 01:45.07 | Zodiacal | i didn't sign up for it |
| 01:45.08 | Qwell | Zodiacal: pastebin your zap config |
| 01:45.13 | Zodiacal | k 1 sec |
| 01:45.15 | Qwell | it will wait for cid... |
| 01:45.48 | Cresl1n | nmsclera: or B.) Your dialplan doesn't have the DIDs setup right |
| 01:46.03 | Zodiacal | qwell i disabled it tho.. |
| 01:46.06 | Zodiacal | usercallerid=no |
| 01:46.35 | SPoon_TSX | Ariel_: Are you able to make BLF work on your installation? |
| 01:47.19 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 01:47.25 | Qwell | Zodiacal: usercallerid? |
| 01:47.32 | Zodiacal | use |
| 01:47.43 | Qwell | pastebin the file |
| 01:48.02 | Ariel_ | SPoon_TSX, your talking about the red lite on top ....yes |
| 01:49.18 | Zodiacal | qwell i can't easily cut and paste it from putty, its huge with all the ;descriptions |
| 01:49.22 | [TK]D-Fender | SPoon_TSX : What phone are yuo trying to get BLF to work on? |
| 01:49.35 | Zodiacal | qwell most everything is turned off tho |
| 01:49.41 | SPoon_TSX | [TK]D-Fender: Aastra 480i. |
| 01:49.49 | *** join/#asterisk Rowter (n=Rowter@201.145.5.26) |
| 01:49.52 | Ariel_ | Zodiacal, I would start by removing all the junk that is not for your setup. |
| 01:49.59 | [TK]D-Fender | SPoon_TSX : Ok, can't help you there. |
| 01:50.35 | Zodiacal | ariel yeah most is... |
| 01:50.52 | Zodiacal | i'll ftp it to this box 1 sec :P |
| 01:51.01 | Ariel_ | Zodiacal, I don't mean comment it out. But delete them. |
| 01:51.02 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 01:51.08 | Zodiacal | ariel oic |
| 01:51.13 | Zodiacal | u think comments could take longer to process? |
| 01:51.36 | Ariel_ | no but you might have one that is not correct un-commented |
| 01:51.39 | blkremedy | does anyone here know of any work arounds for music on hold? Some days it work and somedays it don't. |
| 01:51.54 | Ariel_ | native moh |
| 01:52.03 | blkremedy | streaming |
| 01:52.26 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 01:52.39 | blkremedy | shoutcast |
| 01:52.52 | nmsclera | Cresl1n: When it refers to channel 0/1, what does the "0/1" identify? |
| 01:54.23 | *** join/#asterisk MGSsancho (n=user@adsl-67-127-173-128.dsl.irvnca.pacbell.net) |
| 01:54.50 | *** join/#asterisk dijit0 (n=eric@adsl-68-127-10-129.dsl.pltn13.pacbell.net) |
| 01:56.18 | Zodiacal | ariel, qwell heres the ugly huge thing: http://pastebin.com/526750 |
| 01:56.26 | Zodiacal | most is commented out so i guess i'll clean that up |
| 01:56.31 | Zodiacal | search for channel => 3 |
| 01:56.34 | Zodiacal | thats the one im testing with |
| 01:56.54 | Zodiacal | would another signaling speed this up? other than ks? |
| 01:57.19 | Zodiacal | u really think all this commented sample config garbage could slow me down? |
| 01:57.35 | Zodiacal | i don't think theres anything uncommented that would cause a problem either |
| 01:57.57 | dijit0 | can someone tell me if i have a router port forwarding issue here? i can connect to asterisk with idefisk fine while in my own network, but when i try to connect from another location, it won't allow it |
| 01:58.01 | *** join/#asterisk xachen (i=justin@magnum.thisgeek.com) |
| 01:58.23 | Cresl1n | nmsclera: DS0s on the PRI |
| 01:58.25 | brc_ | ahahaha, classic http://snipurl.com/bofh_ivr |
| 02:00.52 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 02:01.20 | Ariel_ | Zodiacal, you have immediate=yes do you have a context default with an extension exten => s,1,bhah. If you dont' them put it to no. |
| 02:02.04 | Ariel_ | dijit0, it could be both your side or even theres not allowing your port outbound |
| 02:02.50 | Zodiacal | ariel i have an ext call [testing] with with s,1, but not for default |
| 02:02.57 | oceanlan|dustin | ahh hell...wife is pissed..i gotta go watch a movie |
| 02:03.01 | Zodiacal | i tried toggling immediate to no also, and it was still slow |
| 02:03.03 | oceanlan|dustin | l@3r fellas! |
| 02:03.05 | Zodiacal | i thought i would give it a try |
| 02:03.18 | Zodiacal | call = called |
| 02:03.25 | dijit0 | that really sux... what can i do to make sure my end is working ok? i set the port forwarding in my router, but do i need to set anything special in asterisk to work through the router/? |
| 02:04.11 | Ariel_ | ethereal |
| 02:04.47 | Zodiacal | qwell u still around? |
| 02:04.57 | Qwell | nope |
| 02:05.31 | Zodiacal | :P |
| 02:05.32 | rene- | ~seen jerjer |
| 02:05.44 | jbot | jerjer <n=jj@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 9d 4h 34m 50s ago, saying: 'thanks again'. |
| 02:05.44 | Ariel_ | Zodiacal, what board is this? also |
| 02:05.44 | Zodiacal | digum |
| 02:05.45 | Zodiacal | tdm400p x100 |
| 02:05.46 | Zodiacal | fxo |
| 02:07.14 | Zodiacal | ariel qwell, heres the zapata.conf cleaned up: http://pastebin.com/526761 |
| 02:07.36 | Zodiacal | think it would speed up if i didn't have it check all those things? i.e. if i let it use the defaults? |
| 02:07.45 | *** join/#asterisk sancho|laptop (n=user@adsl-67-127-173-128.dsl.irvnca.pacbell.net) |
| 02:09.40 | Ariel_ | Zodiacal, have you tried signalling=fxs_ls |
| 02:09.58 | Zodiacal | nope |
| 02:10.00 | Ariel_ | and after you make the changes you do: service zaptel restart correct |
| 02:10.36 | X-Rob | Well. |
| 02:10.38 | Zodiacal | i think i was doing more, i was reinstalling them, modprobe etc.. |
| 02:10.41 | X-Rob | It's saturday |
| 02:10.48 | X-Rob | and I've actually got time to fuck around with * again |
| 02:10.49 | X-Rob | w00t. |
| 02:10.52 | Ariel_ | X-Rob, yes for you. |
| 02:11.13 | X-Rob | Yay for me even. |
| 02:12.25 | Zodiacal | ariel trying ls, i'll let you know in a min or two |
| 02:13.07 | libila | Can someone please look over a part of my sip.conf/extentions.conf that I pasted here: http://tinyurl.com/7sagu and tell me why when I dial 1234 from user2 I get a 404 and vise versa. (both phones say they are registered correctly) |
| 02:13.50 | X-Rob | libila, you're saying that user and user2 are in the context [from-sip] |
| 02:14.01 | X-Rob | but you have your dialplan using [tutorial] |
| 02:14.36 | X-Rob | do you feel silly now? 8) |
| 02:15.03 | libila | Not really since I don't completely understand it. |
| 02:15.35 | X-Rob | user1 -- context='from-sip' |
| 02:15.36 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 02:15.46 | X-Rob | that means it's looking, in the dialplan, at the context [from-sip] |
| 02:15.49 | X-Rob | which doesn't exist. |
| 02:16.03 | X-Rob | you've called it [tutorial |
| 02:16.05 | X-Rob | ] |
| 02:16.42 | xachen | ladadadadada |
| 02:16.51 | libila | X-Rob: works now, how about that. Thanks I wasn't aware what context meant. First tutorial didn't work so I moved to the next one, heh. |
| 02:17.41 | X-Rob | libila, btw, good help-asking-for. Pasting all the required information, and the problem. |
| 02:18.20 | Zodiacal | ariel qwell after i change to ls, i get this error: WARNING[2145]: loader.c:554 load_modules: Loading module chan_zap.so failed! |
| 02:18.25 | Zodiacal | or warrning. |
| 02:18.50 | Zodiacal | oh i think i know the problem |
| 02:19.14 | X-Rob | Zodiacal, the verbose error will be in /var/log/asterisk/full (eg, 'can't load channel 1' or something like that). The odds are, your /etc/asterisk/zapata.conf file isn't set up right. |
| 02:20.30 | Zodiacal | when i run ztcfg -vv it says that its trying to use kewl start, but i told it to use ls in my conf |
| 02:20.32 | Zodiacal | :/ |
| 02:21.29 | justinu | make sure /etc/zaptel.conf matches |
| 02:21.32 | dudes | that's a bad zaptel.conf ... BAD zaptel.conf |
| 02:21.47 | justinu | no kitty, that's ma pot pie!! |
| 02:21.52 | dudes | heh |
| 02:21.52 | Ariel_ | Zodiacal, you have ks in two locations at the start of the file then around the actual channels |
| 02:22.03 | Zodiacal | yeah |
| 02:22.12 | Zodiacal | changed em both |
| 02:22.17 | Zodiacal | do i need both? |
| 02:22.25 | Ariel_ | yes |
| 02:22.54 | Ariel_ | both files |
| 02:23.02 | Ariel_ | not both in the zapata.conf |
| 02:23.04 | Zodiacal | i do'nt have a /etc/zaptel.conf |
| 02:23.15 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 02:23.18 | justinu | you need to get one |
| 02:23.24 | Zodiacal | :P |
| 02:23.28 | Zodiacal | this has worked before with out it :/ |
| 02:23.36 | dudes | hrm |
| 02:23.43 | Ariel_ | I bet it's there |
| 02:23.57 | Zodiacal | i have a /etc/asterisk/zapata.conf |
| 02:24.07 | justinu | heh |
| 02:24.10 | Ariel_ | /etc/zaptel.conf |
| 02:24.24 | Ariel_ | vi /etc/zaptel.conf |
| 02:24.29 | Ariel_ | or nano it |
| 02:24.42 | Zodiacal | not there, |
| 02:24.45 | Zodiacal | i cp'ed one tho :P |
| 02:24.52 | Zodiacal | same difference when running asterisk |
| 02:25.14 | Zodiacal | ok im going to have to do this another time, gota run.... i'll try with ls. hopefuly taht will speed things up |
| 02:25.31 | Zodiacal | Thank You Ariel! |
| 02:25.35 | Ariel_ | Zodiacal, locate the file it should be there. |
| 02:25.40 | Ariel_ | have a good night then. |
| 02:26.03 | dudes | ls /etc/zap* |
| 02:26.09 | X-Rob | (admittedly, if it's not there, it would stop ztcfg from running, which wouldn't let chan_zap.so load..) |
| 02:26.15 | Zodiacal | oh shit, zaptel, i was looking for zapata |
| 02:26.17 | Zodiacal | its there |
| 02:26.21 | justinu | lol |
| 02:26.22 | Zodiacal | sorry |
| 02:27.00 | *** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 02:27.34 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 02:27.37 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 02:28.20 | Zodiacal | ahh ha, asterisk started :P |
| 02:28.23 | Zodiacal | ok trying a call now |
| 02:30.55 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 02:31.11 | Zodiacal | ariel justinu, i think that just might have done it!! |
| 02:31.40 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 02:31.42 | Ariel_ | ok well?? |
| 02:31.44 | Zodiacal | some times it works on the first ring, some times its right after |
| 02:31.50 | Zodiacal | but its faster! |
| 02:32.09 | Ariel_ | is this with ls |
| 02:32.14 | Zodiacal | yeah |
| 02:33.01 | Zodiacal | ok now i really gota go. ariel thanks again! gnite! |
| 02:33.18 | Ariel_ | have fun |
| 02:33.29 | justinu | goodluck |
| 02:34.14 | franck | Asterisk will support jingle protocol? |
| 02:35.46 | justinu | franck: not as of yet |
| 02:36.59 | franck | justinu: I just saw the jabber/google press release... |
| 02:37.26 | franck | does asterisk has any jabber capabilities? |
| 02:37.33 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 02:37.57 | justinu | yeah, some |
| 02:39.07 | franck | justinu: like? |
| 02:39.32 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 02:40.35 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 02:41.42 | Underhand | if i use Dial(SIP/${EXTEN:3}@fwd,,r), does that pull in all the settings from the [fwd] channel, or just the host/username/password? |
| 02:41.51 | joshua_ | hmmm ... didn't get an answer a couple hours ago -- what are the capabilities of ALSA in *? |
| 02:42.22 | Underhand | in particular, will settings in [fwd] such as insecure=invite be respected? |
| 02:43.28 | justinu | franck: http://www.voip-info.org/wiki-Asterisk+Jabber |
| 02:43.47 | *** join/#asterisk flashnet[BNC] (n=flashnet@213.83.63.227) |
| 02:46.07 | justinu | there's also a SIMPLE/Jabber gateway for SER |
| 02:49.18 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 02:52.43 | *** join/#asterisk Assid (n=assid@203.115.64.14) |
| 02:52.44 | Assid | heya |
| 02:57.13 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 02:57.33 | *** join/#asterisk BugKham (n=lamer@125.24.3.163) |
| 02:58.16 | BugKham | any place to read for the app_conference apart from http://www.voip-info.org/wiki-Asterisk+app_conference |
| 02:59.37 | *** join/#asterisk Legend (n=legend@24.244.142.133) |
| 03:06.02 | *** join/#asterisk jef_ (i=fischer@p548474E8.dip.t-dialin.net) |
| 03:07.00 | *** join/#asterisk [dc] (n=dc@24-205-223-175.dhcp.slto.ca.charter.com) |
| 03:09.17 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 03:10.00 | *** join/#asterisk Cresl1n (n=matt@m495e36d0.tmodns.net) |
| 03:10.43 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 03:13.26 | *** join/#asterisk Jun_Wang (n=chatzill@pool-138-89-62-149.nwrk.east.verizon.net) |
| 03:16.41 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 03:17.46 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 03:21.04 | [dc] | anybody here on FWD care to help me test my inbound routing? |
| 03:22.13 | dogtanian | what's ur number? |
| 03:22.21 | *** join/#asterisk gopherspidey (n=spidey@12-216-165-134.client.mchsi.com) |
| 03:23.34 | [dc] | dogtanian: 725251 |
| 03:24.13 | *** join/#asterisk FastJack_ (i=fastjack@p5091EEDE.dip.t-dialin.net) |
| 03:25.01 | dogtanian | i'm getting 'hung-up' |
| 03:25.08 | [dc] | are you in .uk? |
| 03:25.11 | dogtanian | yeah |
| 03:25.22 | [dc] | i see inbound udp 4569 from you |
| 03:25.31 | dogtanian | i don't really use fwd tho |
| 03:25.43 | [dc] | or from whoever this is: 82.20.19.247 |
| 03:25.51 | [dc] | it doesn't resolve to FWD which is weird |
| 03:25.54 | [dc] | what's your fwd #? |
| 03:26.42 | dogtanian | yeah that's me |
| 03:26.58 | [dc] | can u try it again? i just tweaked an inbound setting |
| 03:27.26 | dogtanian | ringing |
| 03:29.06 | dogtanian | :) |
| 03:29.14 | [dc] | thanx mate apreciate it |
| 03:29.23 | [dc] | enjoy london... i just moved back from there at the end of dec |
| 03:29.27 | [dc] | fantastic city |
| 03:29.30 | dogtanian | np :) |
| 03:29.31 | dogtanian | well at least i know my fwd is still working |
| 03:29.38 | dogtanian | ah cool |
| 03:29.41 | dogtanian | i work here |
| 03:29.51 | dogtanian | although tbh i'd do anything to move out :) |
| 03:30.15 | dogtanian | i was born here and then went to study in teh countriside and now i've had to move back coz of work... sucks tho |
| 03:30.36 | dogtanian | it's too noisy :) |
| 03:30.44 | Dr-Linux | exten => #992,2,AGI(start.agi|${ext_pwd}) <<< whats wrong with this? |
| 03:30.51 | *** join/#asterisk Twister (n=jason@216.30.232.106) |
| 03:31.14 | Dr-Linux | Jan 28 08:33:02 NOTICE[6113]: pbx.c:1478 pbx_substitute_variables_helper_full: Error in extension logic (missing '}') |
| 03:31.37 | dogtanian | _933? |
| 03:31.39 | Twister | is there any way to get a debug of a sip register that tells you WHY a call is unauthorized? (or be able to see the information it is sending to *) |
| 03:31.43 | dogtanian | heh. dunno |
| 03:32.18 | Dr-Linux | lol |
| 03:32.27 | Dr-Linux | what i'm missing :S |
| 03:32.31 | Dr-Linux | Jan 28 08:33:02 NOTICE[6113]: pbx.c:1478 pbx_substitute_variables_helper_full: Error in extension logic (missing '}') |
| 03:32.40 | Dr-Linux | exten => #992,2,AGI(start.agi|${ext_pwd}) <<< here |
| 03:33.19 | dogtanian | should it be _#992? |
| 03:33.28 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 03:33.44 | Dr-Linux | dogtanian: read error please :) |
| 03:34.03 | Dr-Linux | pbx.c:1478 pbx_substitute_variables_helper_full: Error in extension logic (missing '}') |
| 03:34.03 | Dr-Linux | (missing '}') |
| 03:34.07 | dogtanian | ahhhh |
| 03:34.11 | dogtanian | heh |
| 03:34.15 | dogtanian | i'm up too late :) |
| 03:34.25 | Assid | umm.. does * make use of any AMD / Intel extensions sets ? or usage of hyperthreading/hyper transport ? |
| 03:34.41 | dogtanian | Assid: i don't think so |
| 03:37.43 | Assid | was just trying to figure out.. if i had to put a box together.. would an opteron help any over intels |
| 03:38.04 | Assid | i think the linux kernel does run faster though right ? on amds? |
| 03:38.30 | Nugget | It's not that simple. |
| 03:39.06 | Nugget | To begin with, "run faster" is vague enough to be a meaningless phrase. |
| 03:39.38 | Assid | hrmm.. well.. i mean being able to scale and run more applications and handle more calls on asterisk system |
| 03:39.51 | Nugget | Secondly "amds" and "intels" are pretty broad brush strokes to be painting a comparison with. |
| 03:39.52 | Assid | like a database server (pgsql) and a webserver |
| 03:40.13 | fugitivo | Assid: thank you |
| 03:40.21 | dogtanian | Assid: unless you're doing a seriously large-scale operation i'd say that the difference will be small enough to be negligable |
| 03:40.33 | Nugget | You're dangerously close to "rah rah go my team!" territory if you want to compare it that way. |
| 03:40.52 | dogtanian | lol Nugget |
| 03:41.15 | Nugget | as if buying AMD is somehow "sticking it to the man" or something. :) |
| 03:41.45 | fugitivo | i like opteron performance |
| 03:42.03 | Assid | nah.. just trying to figure out whether certain instruction sets would help one way or another |
| 03:42.08 | Nugget | I'm really, really happy with my opteron postgresql boxes. I think it's a great combination. |
| 03:42.08 | fugitivo | i didn't try it with asterisk |
| 03:42.19 | Nugget | but it's not like I bought a t-shirt and hat to go with the cpu. |
| 03:42.30 | fugitivo | but generally speaking, opteron is a nice option to go |
| 03:43.00 | fugitivo | (using a linux optimized for x86_64 obviusly) |
| 03:43.02 | Assid | like ms office for example is said to be faster with intels architecture.. but games are better on amd's |
| 03:43.07 | Nugget | I'm fond of freebsd's amd64 builds for 64-bit, so opteron is a natural choice. |
| 03:43.39 | Nugget | again, "amd's" is worthlessly vague. |
| 03:43.40 | fugitivo | Assid: you should not use the word "faster" |
| 03:43.45 | Nugget | so is "games" and "faster" |
| 03:44.05 | fugitivo | Assid: you should check if the applications are optimized for the actual architecture you have |
| 03:44.06 | Assid | nah.. more like specific instruction sets for some "home advantage" |
| 03:44.11 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 03:44.16 | rob0 | dual even |
| 03:44.39 | Assid | yeah.. thats why my first question was if * was influenced in any way by either architecture |
| 03:44.45 | fugitivo | Assid: if you have an opteron processor running an application or OS optimized for 32bit, you'll not get any performance boost |
| 03:45.27 | Nugget | Are you telling me that Freecell will kick ass on Sempron but will be sluggish and hard to play on a Pentium EE 955? |
| 03:45.53 | Nugget | that's a "game", an "amd" and an "intel" |
| 03:46.20 | fugitivo | and you're not talking about the videocard you have ;) |
| 03:46.29 | Assid | eeks |
| 03:46.42 | Nugget | and anti-arch war, I think. |
| 03:47.16 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-10-74.cybersurf.com) |
| 03:47.18 | Assid | yeah |
| 03:47.28 | Assid | okay drop what i said.. |
| 03:47.59 | Assid | i'll just try and find if linux can gain any advantage since the os could be a real key |
| 03:48.22 | franck | Is that normal that I have a CDR when I make a call from SIP to ZAP even is the person on the PSTN line has not picked up the phone? |
| 03:49.17 | *** join/#asterisk Joeymn (i=AJay@63.231.252.9) |
| 03:50.38 | *** join/#asterisk bmg505 (n=leon@dsl-146-45-90.telkomadsl.co.za) |
| 03:53.45 | gopherspidey | A Poll for everyone: Snom 360 or Polycomm 601 Which would you buy? |
| 03:56.13 | dogtanian | cisco 7960 :P |
| 03:57.13 | Assid | i dont like 7960 |
| 03:57.31 | Assid | but then i love the ftp provisioning of polycom |
| 03:57.37 | gopherspidey | Your a funny man dogtanian. I do not like the Cisco because you have to play to get the POE to work |
| 03:57.42 | Nugget | the 7960 is a fine phone, but it's probably not worth what it costs. I like mine OK, but I'd balk if I had to outfit a large office with them. |
| 03:57.47 | Nugget | I wish it had a backlit screen |
| 03:57.59 | *** join/#asterisk Suroot (n=jason@pool-141-153-22-143.char.east.verizon.net) |
| 03:58.51 | gopherspidey | That is why I think I an going with the snom, because of the Backlight. Polycomm does not have one either. That is Backlight |
| 03:58.54 | Assid | the poly601/501 support POE right? |
| 03:59.12 | [av]bani | http://bani.anime.net/phonez/ <- all your questions answered =) |
| 03:59.12 | gopherspidey | Yes, with no special cables |
| 03:59.19 | dogtanian | Nugget: yes! i totally agree about the backlit screen |
| 04:00.07 | [av]bani | gopherspidey: aastra? |
| 04:00.13 | Nugget | [av]bani: that list indicates that the cisco 7970 and 7985 can do SIP. Is that really the case? that's the first I've heard if it is. |
| 04:00.40 | *** part/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 04:00.54 | Nugget | as far as I know, the cisco sip firmware is 7912, 7940/7960 only. |
| 04:01.34 | [av]bani | hm |
| 04:01.42 | gopherspidey | I have not heard much about aastra on websites and mailing lists |
| 04:01.49 | [av]bani | are you sure you want a $3595 phone? |
| 04:01.53 | Joeymn | does anyone know why if you use a pop/smtp servers on your asterisk box, you cant use the webmail nwebmail on AMP's page because it gets a permission deined trying to read the users mail file? |
| 04:01.58 | dogtanian | <PROTECTED> |
| 04:02.08 | dogtanian | it says the 7960 only has 2 lines |
| 04:02.22 | Nugget | More than wanting a $3595 phone I'd prefer to see an inaccurate list corrected. |
| 04:02.30 | dogtanian | lol |
| 04:02.49 | Nugget | the only thing worse than spending four grand on a phone would be spending four grand on one that didn't do what you were expecting it to do. :) |
| 04:03.00 | [av]bani | dogtanian: how many does it have? |
| 04:03.00 | gopherspidey | lols |
| 04:03.01 | dogtanian | i think the clue is in the url....".../phonez" |
| 04:03.12 | dogtanian | well, mine has 6 line buttins |
| 04:03.13 | Nugget | heh, yeah, the "z" discredits all the content. for sure. |
| 04:03.16 | dogtanian | *buttons |
| 04:03.36 | [av]bani | i rename it to "s" and then the content becomes 100% reliable |
| 04:03.51 | dogtanian | heh... well at least more credible |
| 04:03.52 | Nugget | well, 50% less dubious. |
| 04:04.34 | dogtanian | which country are you in Nugget? |
| 04:04.40 | Nugget | presently? the US. |
| 04:04.45 | dogtanian | ah :) |
| 04:04.46 | dogtanian | <-uk |
| 04:04.49 | [av]bani | well, you dont have to read the page then, i can make it 404 for you if you like |
| 04:05.30 | Nugget | I presumed you'd be interested in making sure the information was accurate. |
| 04:05.37 | Nugget | Perhaps that was optimistic of me. |
| 04:06.36 | [av]bani | i tried, but cisco doesnt make all the information easily available, as you may or may not know |
| 04:06.46 | dogtanian | perhaps when you've updated it you could put it at http://bani.anime.net/a11_jour_ph0nez_are_b3long_t0_us/ |
| 04:06.49 | dogtanian | ;) |
| 04:06.52 | Nugget | heh |
| 04:07.03 | [av]bani | give me your ip's and i'll remove the page from offending your delicate sensibilities |
| 04:07.21 | dogtanian | haha |
| 04:07.35 | dogtanian | like i'm going to even consider giving you my IP :) |
| 04:07.37 | Nugget | 127.0.0.1 :) |
| 04:07.44 | dogtanian | 192.168.1.101 |
| 04:07.56 | pauldy | wow minez 127.0.0.2 |
| 04:08.04 | dogtanian | damn |
| 04:08.05 | dogtanian | :) |
| 04:08.14 | dogtanian | you must be on teh same network :) |
| 04:08.27 | gopherspidey | Thanks for your input on the phones. I was looking for opinions. |
| 04:08.41 | [av]bani | there, no longer a problem for nugget |
| 04:08.56 | dogtanian | <PROTECTED> |
| 04:08.58 | pauldy | my subnets bigger than urz 1z |
| 04:10.01 | Nugget | I never had a problem, I just thought you'd want to be informed of a potential inaccuracy. I guess the wrong data has started being correct now that you've prevented me from seeing it. |
| 04:10.26 | [av]bani | yep |
| 04:10.29 | dogtanian | yeah |
| 04:10.35 | dogtanian | schrodinger's cat right? |
| 04:10.45 | Nugget | The data is both correct and incorrect at the same time? :) |
| 04:10.54 | dogtanian | indeed :) |
| 04:11.15 | dogtanian | although tbh it's probably still slightly more incorrect |
| 04:11.27 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 04:11.45 | dogtanian | <PROTECTED> |
| 04:11.48 | dogtanian | oops |
| 04:11.52 | dogtanian | ^ignore |
| 04:11.54 | mdave | well this stupid disa thing is still boggling me |
| 04:12.06 | [av]bani | wtf is that retarded image |
| 04:12.12 | dogtanian | soz. ww |
| 04:12.31 | mdave | anyone know anything about DISA, and specifically its ability to play dialtone over a sip account to an incoming caller? |
| 04:12.34 | [av]bani | i guess its supposed to be funny |
| 04:12.48 | mdave | eg, do I need to configure anything special for it to be able to do so, depending on my sip provider? |
| 04:13.01 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 04:13.23 | pauldy | what is DISA |
| 04:13.40 | mdave | it works when the call is from (or to) a phone connected to an spa-2000 thats registered with asterisk |
| 04:14.03 | mdave | ~disa |
| 04:14.04 | jbot | it has been said that disa is direct inward system access. show application disa |
| 04:14.28 | pauldy | hrm neat reading about it already |
| 04:14.49 | pauldy | seems kinda pointless for my little setup but neat none the less |
| 04:15.00 | Dr-Linux | pauldy: DISA provides you tone |
| 04:15.09 | mdave | in any case, my end goal is to be able to tell * to dialout to a specific number, and then allow the callee to call as if they were local |
| 04:15.16 | pauldy | I see that which for me would provide someone I don't know with free long distance |
| 04:15.17 | mdave | (fyi, the callee is me, on a cell phone) |
| 04:15.20 | Dr-Linux | pauldy: you can use password as well with DISA app |
| 04:15.22 | *** join/#asterisk Jun_Wang (n=chatzill@pool-138-89-62-149.nwrk.east.verizon.net) |
| 04:15.27 | pauldy | I see that |
| 04:15.39 | gopherspidey | dogtanian: I have a Cisco 7960, but it fried it ethernet port when my switch was hit by a power surge. It will power up and load, but it never gets DHCP or a link light. :) |
| 04:15.45 | mdave | cell phone has 'free' (eg no billed for airtime) incoming calls |
| 04:16.02 | mdave | bv provides free outgoing (eg not billed for minutes) calls |
| 04:16.21 | mdave | so, if I get this working I can make unlimited calls from anywhere from cell, without getting ripped by cellco for airtime |
| 04:16.22 | Dr-Linux | :S |
| 04:16.44 | dogtanian | gopherspidey: i assume you've tried forcing an IP? |
| 04:17.14 | pauldy | mdave seems like a nice little agi script would be easier to setup to do a callback with dial tone |
| 04:17.43 | mdave | in any case, while the callout works, and while it works completely if it calls (or is called by)the phone attached to my SPA, it *doesnt* work properly when calling (or being called by) a regular pots line via the broadvoice account |
| 04:17.52 | gopherspidey | dogtanian: I lost all my nics that were connected to the switch. Not to mention the switch. :( |
| 04:17.54 | pauldy | so you call from your cell phone let it ring once then hangup your asterisk calls you back on your cell and provides you with dialtone allowing you to make an outbound call |
| 04:17.58 | mdave | mainly, the console shows disa being run, but i never hear a dialtone |
| 04:18.13 | mdave | pauldy, what is an agi script, and how would one do that? |
| 04:18.23 | dogtanian | gopherspidey :/ |
| 04:18.27 | gopherspidey | dogtanian: That is what insurance is for, right! :) |
| 04:18.31 | pauldy | ~agi |
| 04:18.34 | jbot | somebody said agi was the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
| 04:18.46 | mdave | my intention is to call from the cell, but have the call *not* be answered, but instead have * call me back |
| 04:19.02 | pauldy | right pretty much lke I just said right |
| 04:19.39 | mdave | what im doing now is using a .call file to place the call, and then attach it to an extension context that runs DISA |
| 04:19.58 | Dr-Linux | who is using AGI ? |
| 04:20.00 | mdave | where would I find more info on agi |
| 04:20.15 | mdave | im searching voipinfo now |
| 04:20.20 | mdave | nvm |
| 04:20.39 | mdave | uhm |
| 04:20.40 | mdave | hrm |
| 04:20.45 | Dr-Linux | mdave: heh i'm looking for AGI since 4 days regularly but i can't understand it yet |
| 04:21.02 | mdave | it seems it still would rely on the dialplan stuff in extensions.conf to have the callee be able to dial-out |
| 04:21.07 | pauldy | seems like you could setup an extension like exten => s/<yourcellnumber>,1,Goto(runMyCallBackAGI,s,1) |
| 04:21.17 | mdave | "If the AGI application dials outward, the script returns execution to the dialplan and loses contact with the asterisk server" |
| 04:21.17 | dogtanian | is there any way to get a free US PSTN number? |
| 04:21.24 | mdave | the callback part isnt the problem |
| 04:21.26 | mdave | i can figure that out |
| 04:21.54 | mdave | the problem is that when it makes the call to a local set on my spa, DISA provides dialtone, and I can enter the pincode and then dial |
| 04:22.03 | pauldy | AGI for the most part is just a send expect mechanism for talking to asterisk |
| 04:22.06 | mdave | when it calls something via broadvoice, i dont get the dialtone |
| 04:22.15 | mdave | even tho the console says it is executing DISA |
| 04:22.41 | pauldy | I'm going to try and setup a DISA extension here |
| 04:22.44 | pauldy | I"m running broadvoice too |
| 04:22.58 | *** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net) |
| 04:23.02 | mdave | which led to my wondering if the mechanism for providing dialtone to a phone connected via a SPA is different than that for doing so to a call connected via bv |
| 04:23.20 | Dr-Linux | pauldy: i'm already using DISA with Authenticate app |
| 04:23.21 | *** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net) |
| 04:23.33 | mdave | eg, if perhaps the spa itself generates the tone at *'s request, but * has to generate the tone itself on a bv call |
| 04:24.02 | mdave | and I have to set something special to tell it to do that |
| 04:24.28 | mdave | although, ive tried dialing assuming the tone was just missing, and it *doesnt* work either |
| 04:24.33 | Dr-Linux | pauldy: are you using AGI ? |
| 04:24.37 | Dr-Linux | ever you use it? |
| 04:25.27 | mdave | again, im not to the stage of getting * to initiate the outbound call in response to the inbound.. at this point im manually initiating the call by dropping a .call file in, but its the DISA that isnt working, but *only* when its via a bv call |
| 04:25.29 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 04:26.18 | mdave | and testing by calling *in* rather than out, gets me the same problem, if I call *in* from an outside pots line to the bv number, the asterisk console shows it getting the call, and running DISA but I never hear the disa dialtone |
| 04:27.15 | mdave | for the record 'normal' calling in between the spa phone and the outside pots works fine, both directions |
| 04:27.32 | mdave | well.. not at the moment since ive commented out the normal extensions mapping to test inbound to disa |
| 04:27.40 | mdave | but it was, and if I put it back it still does |
| 04:27.43 | mzo | using asterisk will get you laid |
| 04:28.11 | pauldy | hrm generated dial tone for me on my cell phone |
| 04:28.24 | mdave | hrm |
| 04:28.38 | mdave | what sip provider you using? |
| 04:28.43 | *** join/#asterisk jeffik (n=Jeff@CPE0050babf4cd5-CM014350000760.cpe.net.cable.rogers.com) |
| 04:28.44 | mdave | im wondering if this is a bv-specific problem |
| 04:29.03 | mdave | ive slowly come to realize that they suck in general, and ive just lived with it for now |
| 04:29.34 | pauldy | I'm using broadvoice BTW |
| 04:29.53 | mdave | im thinking of eventually getting a voicepulse-connect account, and using it only for the free incoming, and then getting a dialpad unlimited outbound acct |
| 04:30.17 | mdave | well hrm |
| 04:30.18 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 04:30.27 | mdave | share your conf? |
| 04:30.32 | pauldy | I just setup a special extension for my incomming cell number and it worked just as expected |
| 04:31.21 | mdave | perhaps theres something different in your sip.conf for bv than there is in mine |
| 04:31.43 | pauldy | running an older version of AAH so in extensions_custom.conf I added the following line in the [ext-local-custom] context |
| 04:32.00 | pauldy | err [from-pstn-custom] context |
| 04:32.21 | pauldy | exten => s/<my cell>,1,DISA(1234|ext-local) |
| 04:32.33 | pauldy | worked just as expected |
| 04:32.40 | mdave | hrm.. well.. let me try that just to see if I can get dialtone at all |
| 04:33.50 | mdave | nope.. dead silence |
| 04:34.00 | mdave | what do you have in sip.conf for bv? |
| 04:34.24 | mdave | just like my previous tries, the console shows 'executing disa', but no tone |
| 04:35.46 | pauldy | register=<mybvnumber>:<mybvpass>@sip.broadvoice.com/<mybvnumber> |
| 04:36.44 | mdave | and the peer definition? |
| 04:37.47 | mdave | mine is at http://jupiter.microwave.com/sip-conf.txt |
| 04:38.21 | pauldy | http://pastebin.com/526855 |
| 04:39.02 | mdave | only difference that seems notable is the codec allows |
| 04:39.13 | pauldy | I believe qualify needs to be off too |
| 04:39.15 | mdave | other than that you have seperate in and out |
| 04:39.18 | mdave | it was off before |
| 04:39.22 | mdave | and it still didnt work |
| 04:39.54 | pauldy | I just did that to make life easier for managing incomming DIDs vs accounts |
| 04:40.05 | pauldy | made it behave a bit more to my way of thinking |
| 04:40.30 | mdave | nod |
| 04:40.38 | mdave | added the allows, no change |
| 04:40.39 | mdave | sigh |
| 04:40.43 | pauldy | could be the codec too |
| 04:40.53 | mdave | just for grins i'll set qualify off |
| 04:41.42 | mdave | nothing, although I just noticied i pasted your context for disa, let me fix that |
| 04:41.48 | pauldy | I remember connecting with some wierd codecs that worked but I had to move back to ulaw to get my apt gate to open when someone enters my code |
| 04:42.38 | mdave | ok, that didnt help either |
| 04:42.39 | *** join/#asterisk coppice (n=chatzill@61.168.17.210.dyn.pacific.net.hk) |
| 04:42.39 | *** part/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net) |
| 04:42.43 | pauldy | also with broadvoice if one server doesn;'t work like you want I've noticed moving to another server will sometimes fix wierd problems |
| 04:43.16 | mdave | i suppose I can try it |
| 04:45.40 | pauldy | anyone else running with the new GXP-2000 firmware |
| 04:46.34 | coppice | anyone running the new grandstream video phone? :-) |
| 04:47.24 | justinu | lol |
| 04:47.33 | pauldy | don't you have some openpbx code to be working on |
| 04:48.06 | Corydon76-home | pauldy: who are you talking to? |
| 04:48.55 | mdave | bah pos bv |
| 04:49.09 | pauldy | gxv-3000 |
| 04:49.10 | mdave | i tried the chi proxy, it denies me access |
| 04:49.33 | pauldy | Corydon76-home, pretty much anyone who will listen |
| 04:50.06 | mdave | trying nyc |
| 04:50.29 | pauldy | I wonder whos DSP they are really using in that phone probably another TI setup |
| 04:51.09 | justinu | chinese TI clone? |
| 04:51.45 | coppice | pauldy: TI 6000 DSP + their own software |
| 04:52.07 | coppice | DM632, I think |
| 04:52.30 | coppice | prety much the only chinese VoIP maker that buys only the siicon |
| 04:52.54 | justinu | coppice: how're things? |
| 04:53.05 | pauldy | well they are just now getting to an acceptable firmware for the GXP-2k so I won't hold my breath on the 3k |
| 04:53.19 | coppice | ok. its a holiday, which is usually nice |
| 04:53.24 | mdave | grr.. it let me register at nyc, but while I can make calls when I try to call in it goes directly to bv vm |
| 04:53.27 | mdave | GRR |
| 04:53.46 | pauldy | mdave they propigate well huh |
| 04:53.50 | justinu | hah |
| 04:54.23 | justinu | coppice: i was in HK last september, is it always that hot? |
| 04:54.47 | coppice | no. its rather cold right now. always is for new year's holiday |
| 04:55.09 | justinu | i guess i'll have to try again in winter |
| 04:55.35 | coppice | October and November are the nicest months to visit |
| 04:56.08 | justinu | i'll probably be seeing benjk in april |
| 04:56.39 | coppice | actually, its never really hot in HK. 35 is the maximum it gets to. its the humidity that makes the climate hard to take |
| 04:57.00 | mdave | ok.. i went back to the default proxy.. calls work again.. but still not disa |
| 04:57.01 | justinu | yeah... its tough for me... even tho i love asia |
| 04:57.04 | mdave | heres a q |
| 04:57.15 | justinu | mdave: i know someone else that had a problem making DISA work on broadvoice |
| 04:57.16 | mdave | what sound file format does one use with the 'background' app |
| 04:57.43 | pauldy | I"ve had luck with gsm and wav |
| 04:57.50 | mdave | i wanna play something right before disa to confirm that the call is actually complete end to end |
| 04:58.00 | mdave | it doesnt need a specific format? |
| 04:58.04 | justinu | mdave: lemme know if you solve it |
| 04:58.04 | mdave | i suppose I should read its doc |
| 04:58.07 | pauldy | 8khz |
| 04:58.23 | pauldy | it worked fine when I tried it |
| 04:58.39 | pauldy | I dialed in with my cell phone then called my wifes cell |
| 04:59.11 | pauldy | same provider BTW |
| 04:59.11 | *** join/#asterisk comfrey (n=comfrey@dsl092-189-099.sfo1.dsl.speakeasy.net) |
| 04:59.17 | comfrey | hey all... |
| 04:59.23 | justinu | i didn't really look into his problem.... |
| 04:59.30 | justinu | wasn't worth the effort :P |
| 04:59.33 | comfrey | i am trying to get cdr_mysql setup... |
| 05:00.04 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 05:00.18 | pauldy | yea I thought that with the whole console audio and ended up spending 6 hours getting it to work on my box so I can call into an say clean up on isle 4 and give myself a rise |
| 05:00.20 | mdave | ok |
| 05:00.22 | mdave | this is stupid |
| 05:00.31 | justinu | lol |
| 05:00.31 | mdave | I added a Saydigits(1234567) before the disa, then a wait |
| 05:00.32 | pauldy | for some reason everytime I do it I think it is as funny as the first time |
| 05:00.38 | mdave | i head the digits, then I got the disa tone |
| 05:00.41 | mdave | heaRd |
| 05:00.42 | comfrey | i am trying to get cdr_mysql setup...looks like it is using cdr_odbc instead of cdr_mysql |
| 05:00.44 | mdave | wtf |
| 05:01.01 | mdave | lemme try a few other things |
| 05:01.09 | justinu | mdave: could be because of asterisk waiting for inbound rtp before sending anything |
| 05:01.14 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 05:01.20 | justinu | i dunno |
| 05:01.59 | Corydon76-home | cdr_odbc gets a lot more attention than cdr_mysql |
| 05:02.06 | mdave | yeah but saydigits isnt inbound |
| 05:02.08 | mdave | sheesh |
| 05:02.23 | comfrey | Corydon76-home: is it recomended over odbc? |
| 05:02.28 | justinu | voip can be tricky like that |
| 05:02.47 | Corydon76-home | I'd recommend using odbc |
| 05:02.50 | comfrey | err or mysql that is |
| 05:02.52 | comfrey | right |
| 05:02.58 | justinu | yikes, odbc |
| 05:03.00 | comfrey | ok, i will check it out |
| 05:03.03 | mdave | ok progress.. now I get the tone, but I cant break it |
| 05:03.08 | comfrey | justinu: you feel otherwise? |
| 05:03.14 | mdave | i can dial but nothing happens |
| 05:03.16 | justinu | i'm just not an odbc fan |
| 05:03.20 | mdave | until it times out and goes fastbusy |
| 05:03.27 | comfrey | you use cdr_mysql |
| 05:03.28 | coppice | things have to be really bad when ODBC is the best choice :-\ |
| 05:03.29 | justinu | it doesn't mean that cdr_mysql is better than cdr_odbc |
| 05:03.37 | justinu | coppice: no kidding |
| 05:03.39 | mdave | so now its something with dtmf handling with bv, i suppose |
| 05:03.55 | justinu | mdave: rfc2833? |
| 05:03.57 | comfrey | justinu: you use cdr_mysql? |
| 05:04.08 | mdave | justinu, ? |
| 05:04.10 | justinu | comfrey: no, cdr_pgsql :P |
| 05:04.18 | *** join/#asterisk bkw_ (n=bkw_@ppp-70-128-122-10.dsl.tulsok.swbell.net) |
| 05:04.28 | mdave | i assume that has something to do with dtmf since thats what I just said, but not sure what/how |
| 05:04.47 | mdave | pauldy, with your disa, after you got tone, did you actually make a call |
| 05:04.55 | justinu | ok, you need to make sure rfc2833 is set for your sip peer |
| 05:05.03 | justinu | which is probably how broadvoice sends digits |
| 05:05.06 | *** part/#asterisk dijit0 (n=eric@adsl-68-127-10-129.dsl.pltn13.pacbell.net) |
| 05:05.10 | mdave | i seem to remember something wonky about how bv handles dtmf |
| 05:05.19 | pauldy | mdave yes |
| 05:05.47 | *** join/#asterisk EriSan (n=erisan@81-174-23-205.f5.ngi.it) |
| 05:05.52 | mdave | my sip.conf entry for bv is at http://jupiter.microwave.com/sip-conf.txt |
| 05:06.11 | mdave | and except for a few insiginfigant bits, is the same as pauldy's |
| 05:06.21 | mdave | isnt that where the dtmf setting(s) would be? |
| 05:06.25 | comfrey | any cdr_mysql users? |
| 05:06.32 | mdave | and if his works, with bv, why wouldnt mine? |
| 05:07.31 | justinu | dtmfmode=rfc2833 |
| 05:08.04 | pauldy | my codecs are set to ulaw and alaw respectivly and I don't have qualify |
| 05:08.07 | justinu | maybe broadvoice uses inband on his gateway, and not yours? who the hell knows |
| 05:08.14 | mdave | hrm |
| 05:08.32 | pauldy | 147.135.4.128 sip.broadvoice.com |
| 05:08.35 | mdave | ok.. just found a posting after googling, apparently for *inbound* calls the dtmfmode is in the [general] section |
| 05:08.36 | pauldy | my host entry |
| 05:08.52 | mdave | and i have no uncommented settings for dtmf in mine |
| 05:08.57 | justinu | mdave: it depends... but go ahead and try it |
| 05:09.07 | justinu | try both inband and rfc2833 |
| 05:09.10 | justinu | maybe one will work |
| 05:09.16 | mdave | pauldy, while you go look at yours, if you would be so kind, i'll fiddle with how its set in mine |
| 05:09.59 | pauldy | ? |
| 05:10.29 | mdave | in sip.conf, do you have an entry for dtmf in [general] |
| 05:10.51 | mdave | for the record, rfc2833 didnt work |
| 05:10.54 | mdave | trying inband |
| 05:11.03 | pauldy | I devided mine up into incoming and outgoing |
| 05:11.17 | pauldy | so I don't have it in general but I do have dtmf set to inband for incomming cals |
| 05:11.27 | mdave | hrm.. neither did inband |
| 05:11.30 | mdave | ah ok |
| 05:11.41 | mdave | well apparently 'info' is an option as well, i'll try that |
| 05:11.56 | pauldy | rfc2833 only works outbound and then only outbound where the customer ont he other end isn't a voip customer |
| 05:11.57 | justinu | mdave: you'd need to get cute with ethereal to solve it at this point |
| 05:12.13 | justinu | wow, that's lame |
| 05:12.23 | pauldy | I know |
| 05:12.23 | rt | i have asterisk running on a box, and a pap2 on my local net. The pap2 registers itself to the asterisk box as a friend, and I can ring it from the asterisk box by doing a Dial(SIP) |
| 05:12.33 | mdave | ok, 'info' doesnt work either |
| 05:12.43 | rt | but when I pick up the phone, what extension does the asterisk box think ti is? |
| 05:12.58 | justinu | an extension is not a device |
| 05:13.01 | pauldy | it is going to have to be inband |
| 05:13.04 | justinu | your pap2 is a device |
| 05:13.07 | justinu | SIP/pap2 |
| 05:13.13 | justinu | an extension can point to a device |
| 05:13.46 | pauldy | wish you could monitor DND with BLF |
| 05:13.46 | rt | let's try this a differnt way: when I pick up the phone on the pap2, I want it to invoke some dialplan in extensions.conf. |
| 05:13.55 | rt | How can I make that happen? :-) |
| 05:14.01 | justinu | that's gonna be a bit tricky |
| 05:14.08 | justinu | the pap2 has to support that feature |
| 05:14.09 | pauldy | you need to set a context for the device |
| 05:14.28 | rt | what feature am I looking for? |
| 05:14.39 | justinu | they call that ringdown in the analog telephony world |
| 05:14.47 | justinu | no idea how they would describe it |
| 05:14.51 | justinu | auto dial? |
| 05:15.07 | justinu | in sip, the phone itself generates dialtone |
| 05:15.16 | justinu | and collects the digits |
| 05:15.25 | justinu | then sends out the invite/iam/setup message |
| 05:15.33 | mdave | ok inband isnt working |
| 05:15.43 | kuku5 | I need crystal clear origination - per minute - using voice pulse now and it sucks |
| 05:16.00 | justinu | yeah, voicepulse seems to be crap |
| 05:16.11 | rt | hmmm. |
| 05:16.14 | justinu | i was pretty disapointed |
| 05:16.35 | pauldy | mdave you aren't running this on some horribly underpowerefd machine are you |
| 05:16.52 | Assid | voicepulse? crap ? |
| 05:17.10 | justinu | i was having crappy sound on origination as well |
| 05:18.19 | Assid | man.. 2nd person to say that abt voicepulse today |
| 05:18.19 | kuku5 | yeh |
| 05:18.24 | mdave | dual athlon 1900 count as underpowered? |
| 05:18.27 | Assid | i thougt it was only me |
| 05:18.27 | kuku5 | its good |
| 05:18.38 | kuku5 | but the tones dont go through |
| 05:18.46 | kuku5 | people press buttons and nothing |
| 05:18.47 | Assid | dtmf? |
| 05:18.49 | kuku5 | yeh |
| 05:18.51 | Assid | weird |
| 05:18.54 | kuku5 | most of the time it works |
| 05:18.54 | Assid | incoming ? |
| 05:18.55 | pauldy | only when using notepad |
| 05:19.03 | kuku5 | i use it only for incoming |
| 05:19.10 | Assid | well.. works fine here |
| 05:19.15 | mdave | no software from redmond need apply ;P |
| 05:19.29 | kuku5 | Assid: i use like 20k minutes monthly |
| 05:19.35 | justinu | not bad |
| 05:19.41 | kuku5 | you have to seperate the tones in order for it to go through |
| 05:19.49 | kuku5 | cant press the buttons to quick |
| 05:19.51 | kuku5 | So anyone? |
| 05:19.58 | [av]bani | woohoo got snom autoprovision script working |
| 05:20.25 | mdave | Hrm : http://voxilla.com/PNphpBB2-viewtopic-t-6973.html |
| 05:20.53 | mdave | if that works i'll shit a brick, but wtf |
| 05:21.30 | Assid | yeah.. gotta be atleast 1/2 a second between numbers |
| 05:21.41 | mdave | sigh.. alright.. im gonna need a couple of tubes of prep-h |
| 05:21.47 | justinu | lol |
| 05:21.47 | pauldy | broken debounce detection |
| 05:21.50 | mdave | it worked |
| 05:21.52 | mdave | christ |
| 05:22.10 | mdave | apparently dtmf=inbound was *not* the answer, but removing it completely was |
| 05:22.12 | mdave | now wtf is that |
| 05:22.17 | mdave | er |
| 05:22.18 | mdave | inband |
| 05:22.29 | kuku5 | Assid: but thats bullshit |
| 05:22.30 | justinu | some kind of asterisk configuration issue, sounds like |
| 05:22.36 | mdave | well it works anyway |
| 05:22.38 | Assid | i know man.. i know |
| 05:22.47 | mdave | now lets go make this work the way I wanted originally |
| 05:22.48 | kuku5 | So there is no better company? |
| 05:22.55 | gopherspidey | [av]bani, Was it hard to get autoprovision working? |
| 05:23.06 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 05:23.07 | justinu | kuku5: seems not |
| 05:23.12 | pauldy | when in doubt |
| 05:23.14 | gopherspidey | [av]bani: or just never done it? |
| 05:23.15 | pauldy | get a bigger hammer |
| 05:26.30 | mdave | for the record, however, it seems call quality, when using two channels thru bv at once is shit |
| 05:26.44 | mdave | of course im sitting on a crap end consumer 384k cable modem |
| 05:26.59 | mdave | once I can get * working on my freebsd box thats sitting on a T3 maybe it will work better |
| 05:27.19 | kuku5 | Anyone know about which form to file for FCC to do internationl calling cards? |
| 05:27.34 | justinu | call the NSA |
| 05:27.38 | justinu | they'll help you out |
| 05:28.32 | SibrPhrek | anyone get musiconhold to work with streaming music? |
| 05:28.43 | mdave | well I should go to bed I suppose |
| 05:28.44 | SibrPhrek | i can only get it to work with the regular mp3's |
| 05:28.56 | [av]bani | gopherspidey: i'm working on a script that will autoconfig sipura, polycom and snom from the same db |
| 05:29.06 | *** part/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 05:29.12 | [av]bani | autogenerates sip.cfg, autoreloads it into asterisk on the fly |
| 05:29.12 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 05:29.32 | justinu | [av]bani: why do you want to support so many phones? |
| 05:29.44 | [av]bani | justinu: because we have a mix of a lot of different phones now |
| 05:29.48 | justinu | k |
| 05:29.56 | [av]bani | and itll be easier to have one unified db of a consistent format for all of them |
| 05:30.00 | gopherspidey | [av]bani: That sound like a useful tool, You could add Cisco to that list! :) |
| 05:30.05 | justinu | seems like a headache |
| 05:30.10 | Corydon76-home | Instead of autoreloading Asterisk, why not just use Realtime? |
| 05:30.12 | justinu | but yeah, a useful tool |
| 05:30.13 | [av]bani | not at all, very very very easy |
| 05:30.29 | Corydon76-home | The SIP stuff for realtime is a bit more sane than extensions.conf |
| 05:31.49 | [av]bani | the only thing it doesnt autogen yet is grandstream |
| 05:31.55 | gopherspidey | [av]bani, Is this in combination with your home grown database or something it A@H? |
| 05:31.59 | [av]bani | if someone supplies me a cisco, i'll make it work |
| 05:32.05 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
| 05:32.08 | SibrPhrek | anyone? anyone? |
| 05:32.09 | [av]bani | gopherspidey: its a flat textfile db |
| 05:32.09 | SibrPhrek | no ? |
| 05:32.18 | *** join/#asterisk YoMama (n=rewt@c-68-61-101-36.hsd1.mi.comcast.net) |
| 05:33.06 | Corydon76-home | SibrPhrek: doesn't work all that well |
| 05:33.08 | pauldy | anyone know how to make RBL monitor SIP trunk usage |
| 05:33.14 | pauldy | or BLF? |
| 05:33.22 | Corydon76-home | Streams don't take well to being paused |
| 05:33.45 | gopherspidey | [av]bani, Cisco's are just <name>: <value> pairs in a flat file located on a tftp with <MAC Address> as a file name. |
| 05:33.49 | SibrPhrek | Corydon76-home: what paused? I'm running asterisk on OS X Server, and have nice cast constantly broadcasting |
| 05:33.57 | Corydon76-home | and when you're not listening to MOH, the mp3s get paused to save on CPU |
| 05:34.24 | Corydon76-home | So you get a timeout on your stream |
| 05:34.53 | Corydon76-home | Basically the stream in Asterisk gets too far behind the server and the server cuts it off as a dead client |
| 05:34.54 | [av]bani | gopherspidey: does cisco support http for provisioning? |
| 05:35.13 | Corydon76-home | [av]bani: I think it's TFTP only |
| 05:35.19 | [av]bani | that sucks. |
| 05:35.46 | gopherspidey | [av]bani: Nope. There SIP Image works but does not have a lot of need toys |
| 05:35.59 | [av]bani | this script, you take a fresh unconfigured out of the box phone, plug it in, and it configures itself totally |
| 05:36.08 | gopherspidey | [av]bani, I had one until a power surge! :( |
| 05:36.27 | [av]bani | and automatically configures asterisk too |
| 05:36.33 | mdave | hrm.. anyway to run a shell script from within an extension entry ? |
| 05:36.34 | [av]bani | its totally PnP |
| 05:36.45 | mdave | other than using curl to call a url that has a cgi or something |
| 05:36.46 | SibrPhrek | Corydon76-home: is there no way to buffer the stream for when moh has is paused |
| 05:37.22 | gopherspidey | [av]bani, How do you tie the MAC or IP to an extension number? |
| 05:37.32 | Corydon76-home | There's always a way. Nobody has written it yet, though |
| 05:37.47 | mdave | actually it looks like agi can do that |
| 05:37.52 | [av]bani | gopherspidey: thats what the db is for. its just mac addr, list of lines, and full names if you want them (optional) |
| 05:38.03 | mdave | hrm.. no examples on the voip-info page |
| 05:38.08 | [av]bani | oh, and if you want the phone to nat or not |
| 05:38.36 | gopherspidey | [av]bani, Cool now all you have to add is a Barcode scanner to collect the MAC's. |
| 05:38.43 | gopherspidey | :) |
| 05:38.47 | [av]bani | i couldnt think of a sane way to generate extension #'s from macs |
| 05:38.56 | Corydon76-home | Streaming might work better if MOH went over UDP instead of TCP |
| 05:38.56 | Joeymn | Jan 27 23:32:28 sonic-wireless nwebmail: could not open abecker mailfile (Permission denied) |
| 05:38.57 | justinu | lol |
| 05:39.20 | Joeymn | why am i getting this error? Jan 27 23:32:28 sonic-wireless nwebmail: could not open abecker mailfile (Permission denied) |
| 05:39.37 | justinu | Joeymn: wrong channel |
| 05:39.40 | mdave | basically, if I can use a callerid-specific exen to run my script, which my script will just tell asterisk to hangup or ignore the call, then kick off the other functions i need |
| 05:40.10 | pauldy | right o |
| 05:40.15 | comfrey | is there a way to get an idea of why a mysql connection is failing? |
| 05:40.44 | gopherspidey | [av]bani, You could just do it by IP and then set the DHCP never to get up a lease. The the extensions would be numbered in the order they were first plugged into the LAN |
| 05:40.58 | gopherspidey | get = give |
| 05:41.37 | [av]bani | gopherspidey: and if someone totes a phone to another net.. it changes extensions :() |
| 05:42.02 | [av]bani | i could make it auto-gen extension if you dont give it one. but ugh... |
| 05:42.22 | gopherspidey | [av]bani, Good point. I guess it depends on the size of you enterprise! |
| 05:42.33 | gopherspidey | lols |
| 05:43.44 | gopherspidey | Well I am off to bed! I think I am going to order my SNOM 360 phones + netgear FS108 on Monday! |
| 05:44.02 | gopherspidey | Man I like !!!! marks tonight. |
| 05:45.34 | [av]bani | gopherspidey: why snom 360? |
| 05:45.39 | *** part/#asterisk YoMama (n=rewt@c-68-61-101-36.hsd1.mi.comcast.net) |
| 05:51.00 | twilson | I might be going crazy... Is there anything wrong with this: GotoIf($[${result}=1]?1|1:2|1) It keeps compalining about the '='... unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN |
| 05:51.22 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 05:54.42 | twilson | ah well, added some quotes around ${result} and 1 and got it to work. |
| 05:55.14 | gopherspidey | [av]bani, It has a backlight, POE, XML Menu interface. |
| 05:58.20 | gopherspidey | [av]bani, The short list was Cisco 79[4|6]0, Polycomm 601, Grandstream GXP-2000 |
| 05:58.39 | gopherspidey | Both Cisco and Polycomm are missing backlights |
| 05:58.49 | gopherspidey | Cisco is not standard POE |
| 05:59.07 | ptiggerdine | IMHO. the granstream looks good for the money they ask. |
| 05:59.28 | gopherspidey | Grandstream I do not trust because I have used a BudgetTone and the Feel of the phone sucks |
| 05:59.33 | [av]bani | gopherspidey: aastra? |
| 06:00.14 | [av]bani | i have a snom 360 here :) |
| 06:00.28 | gopherspidey | I could not fine much about them. Aka opinions on quality |
| 06:00.38 | gopherspidey | Do you like it? |
| 06:01.13 | [av]bani | quality wise i like the polycom 601 better |
| 06:01.21 | [av]bani | if you dont need backlight, the 601 is a better choice imo |
| 06:02.14 | gopherspidey | One of these phone's are for my nightstand. So I sort of need the backlight |
| 06:02.33 | [av]bani | you cant force it on though... |
| 06:02.39 | gopherspidey | [av]bani, have you heard much about the aastra |
| 06:03.00 | [av]bani | the backlight only comes on when you hit a key, and you cant set how long it stays on |
| 06:04.18 | pauldy | latest beta firmware for gxp2k looks like they might have finally figured out the difference between garage hack and enterprise phone |
| 06:05.17 | [av]bani | pauldy: no, the construction is still cheaps |
| 06:05.47 | [av]bani | gopherspidey: nothing much about aastra, but... http://www.o2m8.com/modules.php?name=News&file=article&sid=25 |
| 06:05.50 | [av]bani | you could ask him :)) |
| 06:07.01 | [av]bani | couple of snom concerns: its a german company, support is heavily biased to germans (german forum only for example) |
| 06:08.28 | [av]bani | the audio on the 601 is _much_ better than the snom |
| 06:08.37 | [av]bani | 601 speakerphone is excellent, for example |
| 06:09.36 | [av]bani | it would almost be worth hacking your own backlight in :) |
| 06:10.20 | gopherspidey | I am not that good at electronics. Give me software I can fix it. |
| 06:10.43 | gopherspidey | [av]bani, Are you trying to twist my hand. :) |
| 06:10.46 | [av]bani | the gxp2000 is 'almost there' as long as you dont care about cheapy construction |
| 06:11.06 | [av]bani | and dont particularly care about sound quality |
| 06:11.31 | [av]bani | grandstream is working very hard on the firmware and the phone has lots of potential |
| 06:11.31 | gopherspidey | What is funny is that I was convinced that I was going to purchase a 601 before logging on tonight |
| 06:11.50 | [av]bani | if oyu can live without a backlight, the 601 is the way to go really |
| 06:12.06 | gopherspidey | But then I really thought about th backlight |
| 06:12.08 | [av]bani | if you need a backlight but dont care about quality, gxp2000 is easy pick |
| 06:12.48 | gopherspidey | See I think the Snom fits right in the middle |
| 06:12.56 | gopherspidey | Price is in the middle |
| 06:13.07 | [av]bani | its almost as expensive as 601 |
| 06:13.14 | gopherspidey | Audio quanltity in th middle |
| 06:13.26 | gopherspidey | has a backlight |
| 06:13.47 | [av]bani | do you need the backlight on all the time? |
| 06:13.59 | gopherspidey | Nope |
| 06:14.35 | blkremedy | will ndiswrapper work with centos? |
| 06:15.02 | gopherspidey | blitzrage, Sure but you might have to recompile the kernel |
| 06:15.44 | blkremedy | that's a little too advanced for me |
| 06:16.01 | [av]bani | http://voxilla.com/forum-printview-t-4514-start-0.html |
| 06:17.09 | [av]bani | :) |
| 06:17.11 | coppice | gopherspidey: buy one of those new Dell 30" LCDs, and send it to me |
| 06:17.19 | *** join/#asterisk in-side (n=lowgitek@es-217-129-30-48.netvisao.pt) |
| 06:17.43 | [av]bani | gopherspidey: i have a snom 360. nice phone, but i wish the sound was better |
| 06:17.56 | [av]bani | oh, the lcd is large but lo-res, so it looks like a ZX81 |
| 06:17.58 | in-side | Hi |
| 06:18.15 | in-side | I'm keep getting 603 does anybody can gimme a help? |
| 06:18.16 | coppice | 2560x1600 isn't exactly low res |
| 06:18.41 | [av]bani | coppice: find me a sip hardphone with 2560x1600 display |
| 06:19.26 | coppice | phone LCDs always look mickey mouse |
| 06:19.38 | coppice | even when they use colour |
| 06:19.48 | coppice | its the industry standard (TM) |
| 06:20.04 | gopherspidey | coppice, Ha Ha What do you think I am made of money |
| 06:20.22 | in-side | the talking seems to be very usefull... anyway does any one can gimme a help herE? |
| 06:20.34 | in-side | <PROTECTED> |
| 06:20.50 | coppice | gopherspidey: you were looking for a suggestion as to what to buy. i made a suggestion in good faith. don't get pissey with me |
| 06:20.54 | gopherspidey | [av]bani, Ok do you or have you used a i480? |
| 06:21.05 | [av]bani | gopherspidey: nope |
| 06:21.45 | coppice | any product whose name begins with a small i is highly suspect |
| 06:21.50 | in-side | :| |
| 06:21.53 | gopherspidey | in-side, 603 on what |
| 06:21.57 | [av]bani | you want a nightstand phone, i'm assuming you dont particularly care about audio quality or construction |
| 06:21.58 | in-side | on a invite |
| 06:22.09 | in-side | if i piggy it back to ser |
| 06:22.13 | in-side | it gimme 603 |
| 06:22.31 | in-side | but it accepts happylly all the calls from ser without problem |
| 06:22.55 | X-Rob | w00t. |
| 06:23.00 | gopherspidey | I care about audio quality and looks |
| 06:23.29 | X-Rob | I'm teh fuxx0r1ng g00r00. I just finished patching AMP 2.0beta1 to work with asterisk-trunk. |
| 06:23.38 | gopherspidey | in-side, Never had a 603 |
| 06:23.41 | X-Rob | you may now all worship me. |
| 06:23.47 | in-side | me neither |
| 06:23.48 | in-side | :S |
| 06:24.06 | in-side | Call-ID is intact |
| 06:24.06 | coppice | X-Rob: AMP is for sinners |
| 06:24.12 | in-side | I see no reason for a 603 |
| 06:24.13 | in-side | :S |
| 06:24.15 | in-side | #$%#"$ |
| 06:24.16 | [av]bani | gopherspidey: well, i dont particularly care for the snom 360 looks, but thats subjective |
| 06:24.25 | X-Rob | coppice, heh. Indeed. but it's good for weenies who want to add and remove extensions. |
| 06:24.44 | [av]bani | gopherspidey: audio quality isnt that much better than the gxp2000 (!) |
| 06:24.56 | coppice | I don't think I've seen an IP phone that actually looks good |
| 06:24.56 | [av]bani | the 601 looks much better, the audio quality is miles better |
| 06:25.03 | [av]bani | coppice: 601 is acceptable |
| 06:25.13 | X-Rob | and, you gotta say, A@H has certainly improved the visibility of Open Source IP PBXs, which is a good thing. |
| 06:25.13 | [av]bani | even the gxp2000 doesnt look toooo bad |
| 06:25.14 | in-side | ok. so anybody don't have any clue about the 603 ?? |
| 06:25.27 | X-Rob | coppice, the snom 360's look cool. |
| 06:25.41 | X-Rob | possibly too many buttons, but feh. |
| 06:25.47 | in-side | ya ya it is very cool.. and pricey~ |
| 06:25.50 | in-side | so next.. |
| 06:25.52 | coppice | X-Rob: only in photos. when I finally saw one in real life it looked crap |
| 06:25.53 | [av]bani | it has 'euro styling', which for me is a bit ugh |
| 06:26.06 | X-Rob | gxp is a CRAP PHONE. |
| 06:26.17 | [av]bani | do you have one? |
| 06:26.18 | X-Rob | I know this for a fact. I'm ripping out 20 of them and replacing them with snoms. |
| 06:26.23 | in-side | nobody here use ser? |
| 06:26.30 | X-Rob | (so, yes, I'll be selling a whole pile cheaply soon) |
| 06:26.44 | [av]bani | you have snoms? |
| 06:26.49 | coppice | X-Rob: so you won't be ordering a pile of the new video phones? |
| 06:26.53 | X-Rob | steve, I dunno, I don't mind it at all. |
| 06:26.56 | X-Rob | ...new video phones? |
| 06:27.12 | in-side | ... |
| 06:27.18 | in-side | welll I'm really bored |
| 06:27.21 | in-side | :S |
| 06:27.24 | X-Rob | I've been busy having babys and stuff recently, so I've missed a fair bit. What new video phone? |
| 06:27.25 | coppice | X-Rob: grandstream has finally launched their $295 video phone |
| 06:27.53 | coppice | X-Rob: I javen't had any babies recently, but I have been busy practicing |
| 06:27.58 | X-Rob | hehehehe |
| 06:28.01 | [av]bani | http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-49031808000.htm |
| 06:28.13 | [av]bani | http://www.grandstream.com/GXV3000_interop.pdf |
| 06:28.19 | in-side | oh my god.. |
| 06:28.38 | X-Rob | FFS. |
| 06:28.45 | in-side | thanks very much for your atention.. iI have no words to express my happyness |
| 06:28.59 | in-side | and my gratitude |
| 06:29.11 | X-Rob | that's gotta be the crappiest thing ever. |
| 06:29.13 | X-Rob | evah! |
| 06:29.18 | gopherspidey | [av]bani, These are all very good questions. http://groups.google.com/group/Aastra-480i-Users/browse_thread/thread/cc8dafdeee93f4fc/f7cde01d34f2de30#f7cde01d34f2de30 |
| 06:29.29 | X-Rob | it's obvious that the video has been photoshopped onto it. |
| 06:30.19 | X-Rob | ok. That's crap. |
| 06:30.22 | X-Rob | crappity crap crap. |
| 06:30.30 | coppice | That's always the case with monitor pictures |
| 06:30.40 | coppice | what's so crappy? Its full H.264 |
| 06:30.56 | in-side | ok.. enough,,, what is a crap it is this conversation ...damn not usefull at all ... |
| 06:31.11 | X-Rob | because grandstream are incapable of making a product that doesn't suck. |
| 06:31.12 | [av]bani | gopherspidey: i dont know of any phone which can do 2) |
| 06:31.16 | coppice | it looks pretty ugly though. obviously this is becoming the grandstream distinctive style :-) |
| 06:31.26 | [av]bani | snom cant, neither can polycom |
| 06:32.02 | X-Rob | the hinge looks weak as. |
| 06:32.03 | coppice | X-Rob: they are the only chinese maker I know of that only buys silicon, instead of a packaged bundle. I definitely give them credit for that |
| 06:32.04 | Zodiacal | woh that video phone looks cool |
| 06:32.13 | Zodiacal | and for the price of a cisco 7960 too |
| 06:32.25 | gopherspidey | What are the USB ports for on the GXV3000? |
| 06:32.43 | X-Rob | good point, I do agree. but their stuff is designed, made and built cheap. |
| 06:32.49 | Zodiacal | are there voip providers taht support video? |
| 06:32.50 | Zodiacal | sip |
| 06:33.03 | [av]bani | gopherspidey: probably for a mouse so you can play quake3 |
| 06:33.12 | gopherspidey | lols |
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| 06:33.35 | X-Rob | hint: 'hands-free speakerphone with advanced acoustic echo cancellation' == a lie. |
| 06:33.46 | X-Rob | their echo can's _produce_ more echo than they remove. |
| 06:33.52 | [av]bani | X-Rob: .13 fixed that |
| 06:34.08 | [av]bani | or .12 did even, but that introduced more bugs :)) |
| 06:34.14 | coppice | X-Rob: at least they have an echo canceller. snom doesn't even bother |
| 06:34.14 | X-Rob | .13 also produced the sidetone whistle which makes them unusable in a noisy environment |
| 06:35.27 | X-Rob | I'm trying to remember having any issues with echo on the snom |
| 06:35.40 | X-Rob | I'm pretty sure it does, actually. |
| 06:36.13 | X-Rob | coz I was talking to someone who was hands free on their snom and it was being very egressive with the echo can (that was sip<->sip, nothing else was doing it) |
| 06:36.26 | X-Rob | or maybe it's just half duplex 8) |
| 06:36.47 | Zodiacal | so do voip providers support sip video? |
| 06:36.52 | [av]bani | what snom do you have? |
| 06:36.57 | coppice | i dunno why it is that people are so nasty about grandstream, when much more expensive phones have very serious limitations |
| 06:37.18 | X-Rob | I'm nasty because they won't fix annoying niggly problems. |
| 06:37.28 | [av]bani | X-Rob: like sipura? |
| 06:37.41 | X-Rob | like 'make the sidetone a feature that can be switched on and off' |
| 06:37.47 | coppice | X-Rob: snom, polycom and various other expensive phones do not EC the handset. they have serious problems as soon as you turn up the handset volume a little |
| 06:37.56 | X-Rob | but they'll add my BLF and **call pickup stuff. *grump* |
| 06:38.04 | gopherspidey | I called digium today and had a Bad echo! :) |
| 06:38.26 | [av]bani | coppice: and boy is that polycom loud |
| 06:39.02 | X-Rob | Um, someone asked me what snom I have - I've got a 360 here, but my customer has 4 360's and 18 GXP's |
| 06:39.27 | X-Rob | We're replacing the GXP's with 360's and 220+keypads |
| 06:39.33 | [av]bani | i wish the sound on the 360 was better, it has a lot of weird acoustic artifacts |
| 06:39.50 | X-Rob | We use alaw, don't have a problem. |
| 06:40.19 | [av]bani | its not codec related |
| 06:40.31 | X-Rob | Well. I haven't _noticed_ |
| 06:40.45 | gopherspidey | X-Rob, Why no Polycomm? Is the the $50 a phone or no experence or something else? |
| 06:40.48 | [av]bani | i think the linux kernel they compiled has scheduling issues, i get skipped frames |
| 06:40.54 | [av]bani | just on local dialtone on the phone |
| 06:40.58 | X-Rob | I'm gunna plug it in and upgrade it - I noticed they're up to version 5 with a new kernel too. |
| 06:41.10 | [av]bani | weird debounce issues with dialing |
| 06:41.21 | [av]bani | clicks when i get error tones from asterisk |
| 06:42.34 | [av]bani | actually clicks on the beginning of most rtp streams, may be some jitter buffering bugs in snom's firmware |
| 06:43.10 | [av]bani | do you have a 360 near you at the moment? |
| 06:43.59 | gopherspidey | [av]bani, Now I see why you are not recommending Snom. |
| 06:44.16 | [av]bani | i'm gonna yell at them about audio issues |
| 06:44.24 | gopherspidey | lol |
| 06:44.29 | [av]bani | i'm sure they can fix it, it seems to me just software issues |
| 06:45.04 | gopherspidey | I assume you are on the lastest firmware. |
| 06:45.21 | [av]bani | their beta xml stuff actually, dunno if thats 5.2 or not |
| 06:45.31 | [av]bani | (it doesnt say, it just says some nondescript version #) |
| 06:45.52 | X-Rob | Yes. I have my 360 near me. Just plugging my fixed switch back in |
| 06:46.43 | [av]bani | X-Rob: can you hear a weird 'sqwerp' when you hit speakerphone for dialtone then press it again to hang up? |
| 06:46.54 | [av]bani | at the tail end of the dialtone when it gets cut off |
| 06:47.38 | [av]bani | sometimes you'll hear the dialtone click as it starts playing |
| 06:47.45 | [av]bani | or skip |
| 06:49.02 | X-Rob | I hit 'speaker' and it goes 'booooooooooo'.. I hit speaker again and it goes '..oooo-i-i'. Two distinct sorta clicks. |
| 06:49.27 | [av]bani | lemme change indications to aus |
| 06:50.00 | gopherspidey | Someone in the Hard phone industry needs to standardize the phone XML interface. Most are based on Cisco's (I think) but all are a little different. |
| 06:50.30 | coppice | sqwerp? is that in the dictionary? shouldn't there be a u after the q? :-) |
| 06:50.38 | [av]bani | X-Rob: yeah, it doesnt cut off cleanly.. it rises in pitch as it hangs up... and sometimes clicks |
| 06:51.11 | [av]bani | X-Rob: and sometimes you can hear it go booooo oooooooooooo |
| 06:51.34 | coppice | or boooo hoooo hoooo |
| 06:51.36 | [av]bani | i think they've got scheduling issues in the kernel |
| 06:51.42 | [av]bani | making the audio skip |
| 06:52.27 | [av]bani | seems to happen at the beginning of any new audio stream -- dialtone, connecting to rtp stream, etc |
| 06:52.39 | X-Rob | I'm doing the upgrades now.. |
| 06:52.46 | X-Rob | 'UPGRADING LINUX' 'DO NOT SWITCH OFF' |
| 06:52.50 | [av]bani | to 5.2 ? |
| 06:52.53 | X-Rob | yeah |
| 06:53.00 | [av]bani | oh yeah, i love the confusing a/b thing |
| 06:53.01 | X-Rob | it's running 4.3 now. |
| 06:53.24 | gopherspidey | a/b thing? |
| 06:53.39 | [av]bani | use a firmware if upgrading from 4.x, but b if upgrading from 5.x |
| 06:53.46 | [av]bani | in confusing engrish on their wiki |
| 06:53.52 | gopherspidey | lol |
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| 06:55.02 | [av]bani | gopherspidey: probably the most annoying thing is that their support forums are very active, but all in german |
| 06:55.08 | [av]bani | english support seems to be almost nil |
| 06:55.25 | coppice | probably reflects where they sell well |
| 06:55.41 | [av]bani | yes, which is something to keep in mind for non german speakers |
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| 06:56.36 | [av]bani | gopherspidey: oh yeah, the phone's user interface is pretty bad. polycom's is very nice |
| 06:57.13 | [av]bani | i never liked those multidirectional pads, cant see a point for them on a phone... |
| 06:58.07 | coppice | multidirectional pads are fundamentally important. without them the phone looks like its worth $10 less :-) |
| 07:01.01 | X-Rob | *especially* if they're silver. |
| 07:01.06 | gopherspidey | lol |
| 07:01.06 | [av]bani | that tilting display has to be worth at least 10 extra HP |
| 07:01.06 | [av]bani | all it needs is a coffeecan exhaust |
| 07:01.08 | gopherspidey | As long as Polycomm has not messed up the SIP comunication. I know the Sound quality will be great. I guess I will have to think about the "need" for the backlight |
| 07:01.08 | X-Rob | ooh. that's a good sign |
| 07:01.08 | X-Rob | the phone doesn't look dead! |
| 07:01.08 | X-Rob | w00t! |
| 07:01.10 | gopherspidey | That is was a good sign |
| 07:01.14 | [av]bani | gopherspidey: yes, i really would if i were you |
| 07:02.54 | X-Rob | ooh. The buttons are a lot more programmable bow. |
| 07:02.54 | [av]bani | gopherspidey: the negatives about polycom arent as annoying, at least to me: 1) polycom are arseholes regarding firmware releases to customers 2) the phone takes 3+ minutes to boot 3) the webinterface is terrible |
| 07:02.54 | [av]bani | gopherspidey: the beautiful lcd and excellent sound more than make up for it. if it had backlight it would be slam dunk |
| 07:02.55 | gopherspidey | [av]bani, What is bad about the Web interface? |
| 07:02.59 | coppice | I haven't upgraded one of my FC4 boxes for a couple of weeks. now it is installing 178 updates. this is getting silly |
| 07:03.18 | [av]bani | gopherspidey: the polycom wants to reboot every time you make even the tiniest change. now refer to whinge 2) above |
| 07:03.55 | [av]bani | gopherspidey: "you have moved the mouse. windows must now reboot for the changes to take effect" |
| 07:04.01 | gopherspidey | [av]bani, oh the webinterface to configure the phone. |
| 07:04.22 | gopherspidey | I guess I can live with that. |
| 07:04.23 | [av]bani | the web ui on snom is excellent |
| 07:04.36 | [av]bani | gopherspidey: when they say you want to use xml to config the polycom, they arent kidding. |
| 07:04.50 | gopherspidey | Have done anything with the XML stuff? |
| 07:04.55 | [av]bani | but for home, you'd only have to config it once |
| 07:05.25 | gopherspidey | Aka xml directories. |
| 07:05.33 | [av]bani | not yet |
| 07:06.29 | [av]bani | point for snom: you can basically reconfigure every freaking key on the entire phone to do anything you want |
| 07:06.30 | [av]bani | the snom also supposedly supports ldap somehow, how i have no idea though |
| 07:06.43 | [av]bani | the polycom's xml directoty stuff is pretty well documented |
| 07:07.17 | [av]bani | snom supports srtp/sips.. too bad * doesnt :( |
| 07:07.20 | gopherspidey | All I want to do is control my HVAC Temp and lighting from the phone |
| 07:07.37 | gopherspidey | with the XML interface |
| 07:07.44 | [av]bani | well obviously the aastra 480i will let you do that :) |
| 07:07.53 | gopherspidey | I saw that. |
| 07:07.55 | [av]bani | but i dont have one so i cant tell you about audio quality or firmware |
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| 07:08.10 | coppice | <lighting>candle lit dinner for 2 style</lighting> |
| 07:08.33 | gopherspidey | Wow a couple Ipv6 addreses on the last network join. |
| 07:08.49 | gopherspidey | coppice, Sometime like that. |
| 07:08.55 | X-Rob | <lighting><device><name>Candle</name><number>10</number></device></lighting> |
| 07:09.03 | X-Rob | This is XML, get it right. |
| 07:09.20 | X-Rob | LETS MAKE IT AS VERBOSE AS POSSIBLE they said. |
| 07:09.27 | gopherspidey | coppice, More like "All house lights off" |
| 07:09.32 | [av]bani | gopherspidey: well if you do decide you cant live without backlight and get a snom, i should have a very nice autoprovisioning script done by the time you get it :) |
| 07:09.42 | X-Rob | but it's good. It's better than propritary binary formats. |
| 07:10.10 | X-Rob | Production Information:Mac:0004132305FF;Version:Standard;Hardware:snom360 (MB V10_L2,KB V10_k7);Lot: 02/05 |
| 07:10.20 | [av]bani | ?? |
| 07:10.21 | X-Rob | woo snom. woo. |
| 07:10.21 | gopherspidey | X-Rob, Have you ever look at the Polycomm config files. Talk about XML. |
| 07:10.41 | [av]bani | sipura's xml is relatively sane |
| 07:10.42 | X-Rob | gopherspidey, nah. Polycoms are too hard to buy in .au |
| 07:11.29 | [av]bani | Mac:000413232469;Version:Standard;Hardware:snom360 (MB V1.0_K7,KB V1.0_L2-NC);Lot: 08/05 |
| 07:11.35 | [av]bani | thts the xml firmware |
| 07:12.01 | gopherspidey | Try legaly purchasing a Cisco with SIP firmware any where. :) |
| 07:12.15 | [av]bani | gopherspidey: you lose functionality with sip on ciscos anyway :( |
| 07:13.16 | gopherspidey | [av]bani, You got that right. But I had trouble with sccp the last time my Cisco was working. |
| 07:13.49 | X-Rob | so, here's the thing - don't use ciscos. *gasp* |
| 07:14.53 | [av]bani | X-Rob: super annoying snom-ism: they sure like ENORMOUS FONT, especially where it makes the least sense to use it |
| 07:15.32 | [av]bani | gopherspidey: can you live without backlight [Y/N] |
| 07:16.27 | gopherspidey | I am just thinking of Caller ID at night when the phone rings. |
| 07:16.34 | [av]bani | hm |
| 07:17.02 | gopherspidey | How is the echo when you are on a call with a Snom? |
| 07:20.49 | gopherspidey | X-Rob, Have you used a 480i or 9133i from Sayson or Aastra? |
| 07:20.49 | [av]bani | no echo afaict, but thats just with upstream istp for outgoing |
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| 07:26.46 | [av]bani | hmm |
| 07:26.58 | X-Rob | indeed. |
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| 07:29.03 | X-Rob | wb |
| 07:30.11 | [av]bani | in fact, i might say the snom audio is poor |
| 07:30.42 | [av]bani | i dialed into a MOH extension, and i get weird swishing sounds in the audio |
| 07:30.49 | [av]bani | which i don't get on a sipura 3000 FXS or a polycom |
| 07:30.55 | coppice | most of the VoIP industry has poor voice quality. why do so few aspire to actually improve on the PSTN? |
| 07:32.57 | Nugget | because the early adopters of voip are the "I want unlimited voip so I can stick it to ma bell" crowd, I suspect. |
| 07:33.26 | Nugget | everyone's too busy competing on price to snag the cheapskate market to notice those of us who'd rather pay for decent service. |
| 07:33.56 | [av]bani | hmm ok better now |
| 07:34.03 | coppice | i find most are the "XML is wonderful. Bandwidth is cheap Is there anything with a lower bit rate than G.729?" crowd |
| 07:34.05 | [av]bani | apparently it was using gsm when i told it not to... |
| 07:34.14 | Nugget | heh |
| 07:34.21 | [av]bani | bad snom |
| 07:34.49 | [av]bani | muuuuch better |
| 07:36.21 | X-Rob | the fucker is not downloading it's config. |
| 07:36.36 | [av]bani | o_O |
| 07:38.26 | [av]bani | wish * did g722 |
| 07:38.56 | [av]bani | coppice: because the pstn is built on string and tin cans |
| 07:39.53 | coppice | G.722 (the original, not G.722.2) can easily be added to *, if they sort out its tie to 8000 sample/second audio |
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| 07:40.57 | gopherspidey | Is G.722.2 one of the license/patent things? |
| 07:41.25 | X-Rob | ok, it's definately not downloading it's configs. |
| 07:41.51 | coppice | G.722.2 is basically AMR wideband by another name. very encumbered |
| 07:42.00 | gopherspidey | opps |
| 07:42.01 | coppice | but also very good |
| 07:42.04 | gopherspidey | that is kicks |
| 07:42.14 | coppice | speex wideband could also be added easily |
| 07:42.27 | [av]bani | coppice: encumbered how? |
| 07:42.35 | [av]bani | omg lets patent 16kbps ? |
| 07:42.37 | coppice | patents |
| 07:42.38 | [av]bani | er 16khz |
| 07:43.03 | gopherspidey | coppice, wideband? I know nothing about codecs other then bandwidth they consume |
| 07:43.33 | coppice | wideband, as in something with an audio bandwidth greater than the usualy 4kHz |
| 07:44.26 | coppice | [av]bani: there is nothing bogus about most of the patents on codecs |
| 07:46.05 | [av]bani | coppice: including fraunhofer's claim vorbis infringes on mp3? |
| 07:46.24 | coppice | only their marketing dept claimed that :-) |
| 07:46.48 | gopherspidey | [av]bani: I had not heard of that one. |
| 07:47.18 | coppice | the vorbis people are very respectful of the scope and nature of the fraunhofer patents |
| 07:47.34 | [av]bani | ah, it was Henri Linde of Thomson who claimed vorbis infringed mp3 patents |
| 07:47.39 | [av]bani | publically |
| 07:47.39 | gopherspidey | the huffman encoding is the only thing that fraunhofer has going for them |
| 07:48.05 | [av]bani | not marketing either, he's VP of their IP licensing |
| 07:48.06 | coppice | rubbish |
| 07:48.32 | [av]bani | of course, he's pushing mp3pro ... |
| 07:48.33 | coppice | isn't that a marketing function? |
| 07:49.02 | [av]bani | no it's police unit |
| 07:49.13 | [av]bani | patent enforcement gestapo |
| 07:49.14 | gopherspidey | Good night and Thanks. Now that is is 1:50am. :( |
| 07:49.31 | [av]bani | and propaganda unit, from their behaviour :) |
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| 07:53.27 | coppice | i can't remember what is patented about MP3 - i don't take much interest in non-speech codecs - but they must have run out in most places |
| 07:53.46 | [av]bani | everyone has moved on from their very old techniques by now |
| 07:53.54 | [av]bani | mp3 was just first, so they hav elots of traction |
| 07:54.20 | [av]bani | but even the best mp3 coders sound like shit compared to vorbis |
| 07:54.27 | [av]bani | i have a lot of clips which eat mp3 alive |
| 07:54.43 | coppice | MP3 hasn't changed. it was very early, and still very competitive. i think it was a pretty wonderful design |
| 07:54.45 | [av]bani | c64 chipmusic destroys mp3 :) |
| 07:55.12 | Nugget | bullcrap. the c64 only had three voices. the atari 800 had four voice sound. THAT was kickass. :) |
| 07:55.13 | [av]bani | vorbis is able to pretty much nail it perfectly with ease |
| 07:55.25 | coppice | WMA sounds like shit most of the time. surely they could have come up with something better than that? |
| 07:55.39 | [av]bani | Nugget: c64 has filters and complex waveforms, atari 800 was only square wave... |
| 07:55.47 | [av]bani | coppice: you're talkinga bout microsoft.... |
| 07:56.13 | Nugget | bah, only if you count that goofy program that played fur elise on the 1541 floppy drive. |
| 07:56.16 | [av]bani | an intellectual powerhouse, microsoft most certainly is not |
| 07:56.25 | coppice | yeah. they have lots of money to throw at the problem, but produced rubbish. WMV 9 seems pretty good, but the audio is lousy |
| 07:56.58 | [av]bani | coppice: that defines microsoft perfectly. they just keep throwing piles of money at something, usually takes them 5 or 6 revisions to get something tolerable |
| 07:57.10 | [av]bani | they eventually get it right, even if it takes them 20 years |
| 07:57.39 | coppice | dunno, they still haven't got WinFS working after 15 :-) |
| 07:57.59 | [av]bani | some problems arent solvable :) |
| 07:59.42 | [av]bani | http://en.wikipedia.org/wiki/G.722 <- encumbered? |
| 08:00.34 | coppice | nope. G.722.1 (rubbish) and G.722.2 (excellent) are encumbered |
| 08:01.37 | coppice | G.722 is available in spandsp |
| 08:01.38 | [av]bani | snom 360 does g722, any reason * cant support it? |
| 08:01.50 | [av]bani | hm |
| 08:02.11 | coppice | * doesn't understand any codec which is not 8000 samples/second right now |
| 08:02.18 | [av]bani | :/ |
| 08:03.12 | X-Rob | and my dial regexp isn't working |
| 08:03.13 | [av]bani | in fact, doesnt appear anything does g722.2 |
| 08:03.14 | X-Rob | *kick* |
| 08:03.19 | [av]bani | they all do g722 |
| 08:03.53 | coppice | that's cos G.722 is unencumbered |
| 08:04.03 | [av]bani | yea, figured |
| 08:04.05 | coppice | but its old and inefficient |
| 08:04.09 | [av]bani | well, everyone does g729a also |
| 08:04.11 | [av]bani | :)) |
| 08:04.39 | coppice | they have to pay for G.729. Its unavoidable. G.722.2 is an option |
| 08:04.41 | [av]bani | having something for lan with better fidelity than ulaw would be nice |
| 08:04.50 | [av]bani | hence g722 |
| 08:09.39 | coppice | I agree. nobody in * land seems to care very much, though, even though wideband is a key reason people like Skype |
| 08:12.04 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 08:15.47 | |vinsik| | re... anybody knows how to dial out through a peer added to mysql database? |
| 08:16.10 | |vinsik| | +thats |
| 08:18.10 | *** join/#asterisk lorinc (n=ang@caracas-3662.adsl.interware.hu) |
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| 08:28.08 | [av]bani | hmm.. are the voicetronix cards good FXOs? |
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| 10:29.16 | |vinsik| | re... anybody knows how to dial out through a peer thats added to mysql database? |
| 10:30.46 | *** join/#asterisk ToTo (n=ToTo@host70-163.pool872.interbusiness.it) |
| 10:32.52 | *** join/#asterisk greyarea (n=adam@199.232.44.74) |
| 10:34.37 | *** join/#asterisk convergence (n=adam@199.232.44.74) |
| 10:35.12 | convergence | Im having a problem, with something Im working on, maybe someone can help. |
| 10:36.31 | *** join/#asterisk Flusher (i=flusher@filer.euroserv.com) |
| 10:36.37 | convergence | Im trying to get a dialplan to record each call out, then after each call save the audio files soxmix it, and then go back to the outdial context when pressing star. |
| 10:37.27 | convergence | I can dialout and record, but when I enable the variable to go back to the next priority, it wont stop the recording and begen a new one. |
| 10:37.59 | *** join/#asterisk dogtanian (i=dogtania@eye.ham.zee.walroos.com) |
| 10:37.59 | convergence | like Ill call in and call out to 3 different numbers and it will only spit out 1 file recorded. |
| 10:38.21 | *** join/#asterisk AndyCap (n=aoy@pdpc/supporter/sustaining/AndyCap) |
| 10:38.37 | convergence | this is my current context. |
| 10:38.39 | convergence | [dialout9999] |
| 10:38.39 | convergence | exten => s,1,Read(callto,dialtonez0r,21) |
| 10:38.39 | convergence | exten => s,2,SetVar(CALLFILENAME=${callto:1}-${TIMESTAMP}) |
| 10:38.39 | convergence | exten => s,3,Monitor(wav,${CALLFILENAME},m) |
| 10:38.39 | convergence | exten => s,4,Dial(SIP/${callto}@bv,45,Hg) |
| 10:38.40 | convergence | exten => s,5,goto(s|1) |
| 10:39.06 | RoyK | ~pb |
| 10:39.07 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 10:39.30 | convergence | Is there something I can set after s,4 to tell it to begin a new file? |
| 10:39.44 | convergence | sry RoyK |
| 10:39.51 | *** join/#asterisk j0sephi (i=j0e@biztosan.hu) |
| 10:40.09 | RoyK | what do you mean a new file? |
| 10:40.23 | RoyK | you want another file at the time the phone is hung up? |
| 10:40.26 | RoyK | or picked up? |
| 10:40.35 | RoyK | or stepped on.... |
| 10:41.14 | convergence | When I press *, and the outbound call hangs up. I would like it to save the recording then start a new one when it gets to the priority. |
| 10:41.57 | convergence | as is, its just contenuously recording. |
| 10:42.26 | convergence | and wont stop until I hangup from the asterisk server. |
| 10:43.37 | RoyK | iirc there's a dial option to allow you to do such a thing |
| 10:43.53 | convergence | http://pastebin.ca/38888 |
| 10:44.54 | RoyK | use the w option with dial |
| 10:45.42 | RoyK | prio 5 should never be reached if the call is hung up |
| 10:48.11 | convergence | I know, but thats the thing, Im using the g option, for the reason to be able to press * and make another call without myself hanging up. |
| 10:48.53 | convergence | Where lies the problem Im having. |
| 10:49.22 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 10:49.38 | *** join/#asterisk jas_williams (n=jircii@host86-136-45-178.range86-136.btcentralplus.com) |
| 10:50.15 | convergence | I can continue to place calls and it records when I press *, but its only one recording. instead of mulitable recordings on each call. |
| 11:05.41 | convergence | oh, I think this might help me. http://www.voip-info.org/wiki/view/MixMonitor |
| 11:12.19 | *** join/#asterisk Drew___ (n=foo@zux221-065-169.adsl.green.ch) |
| 11:22.53 | *** join/#asterisk jorge_ (n=jorge@54.Red-213-96-220.staticIP.rima-tde.net) |
| 11:22.54 | convergence | Oh MY GODZ0R |
| 11:23.01 | convergence | heh, wow |
| 11:23.06 | jorge_ | hi all |
| 11:23.10 | convergence | well, I forgot I wasnt on 1.2 yet. |
| 11:23.22 | convergence | so the mixmonitor was no goods to me. |
| 11:23.41 | convergence | but when I look at the registered apps, you know what I saw..? |
| 11:23.46 | convergence | your going to laugh |
| 11:23.54 | convergence | StopMonitor |
| 11:23.56 | convergence | heh |
| 11:24.02 | convergence | thats did it for me. |
| 11:24.43 | convergence | so I placed that at prio 5 then 6 went back to the dialout one. |
| 11:25.01 | convergence | thats so funny. |
| 11:26.55 | jorge_ | do any of you know if the chan_zap patch for the spanish line can be applied anyway to asterisk 1.2.3?, or it's just for 1.2.0?? |
| 11:27.13 | RoyK | convergence: i'd say upgrade |
| 11:27.17 | jorge_ | or is it applied it the 1.2.3 package yet? |
| 11:27.37 | RoyK | convergence: there really isn't much reason to stick with 1.0 unless you have custom apps not ported up to 1.2 yet |
| 11:27.45 | convergence | yeah I know, I just havent got around to it. |
| 11:28.31 | convergence | Im still on CVS-v1-0-04/22/05 |
| 11:28.32 | convergence | heh |
| 11:30.24 | jorge_ | bye |
| 11:33.49 | convergence | I feel really dumb though with the "stopmonitor" thing. |
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| 11:34.18 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 11:37.20 | burton | hello, anyone know what this error mean? where too look what is wrong ? this is what i get on incoming call > CAPI INFO 0x34e5: Message not compatible with call state ... and than == ISDN1: CAPI Hangingup |
| 11:41.21 | *** join/#asterisk ToTo (n=ToTo@host70-163.pool872.interbusiness.it) |
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| 12:00.06 | KriS83 | Hi |
| 12:02.25 | KriS83 | When using enum, and the number resolves to a sip address like: 1234@sipprovider.tld and Asterisk then dials Dial(SIP/1234@sippprovider.tld) how can I make asterisk not look for the context [sipprovider.tld] in extension.conf? Or am I doing something wrong? Missed something? Thank you for any hints |
| 12:02.27 | *** join/#asterisk afoldo (i=afoldo@c9115b58.rjo.virtua.com.br) |
| 12:02.37 | afoldo | Mornin... |
| 12:03.10 | afoldo | Can anybody shed some light on a DUNDi issue, please? |
| 12:04.20 | afoldo | I have several Asterisk boxes with ISDN trunks on them, and I'd like to use DUNDi to route calls from IAX clients in a private network to these boxes. |
| 12:04.28 | afoldo | Does this make sense? |
| 12:05.30 | burton | hello, anyone know what this error mean? where too look what is wrong ? this is what i get on incoming call > CAPI INFO 0x34e5: Message not compatible with call state ... and than == ISDN1: CAPI Hangingup |
| 12:08.53 | *** join/#asterisk chr|s_ (n=chris@54-126-66-195.spark.net.gr) |
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| 12:18.26 | X-Rob | Well. I think it's a damn good macro. |
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| 12:54.21 | krasavin | hi all! did anybody install zaphfc? |
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| 13:10.14 | *** join/#asterisk A500mg (n=x@ACaen-151-1-40-54.w86-199.abo.wanadoo.fr) |
| 13:10.26 | A500mg | hello |
| 13:10.27 | A500mg | salut à tous |
| 13:12.07 | A500mg | i want to buy a tdm400p |
| 13:12.09 | A500mg | where can i buy a tdm400p at the best price ? |
| 13:12.29 | A500mg | je suis un gros radin :) |
| 13:13.37 | X-Rob | from digium |
| 13:15.59 | A500mg | yes |
| 13:16.05 | A500mg | but a resseler ? |
| 13:19.24 | A500mg | i've found myphonecall.uk |
| 13:19.55 | A500mg | but i search a best price :) |
| 13:21.02 | coppice | I always buy from Honest Joe's VoIP Emporium :-) |
| 13:22.57 | A500mg | i'm french, i can't understand the joke :| |
| 13:24.08 | coppice | "Honest Joe's" or "Honest John's" is a standard prefix for something far from honest :-) |
| 13:25.05 | A500mg | ol |
| 13:25.08 | A500mg | lol |
| 13:28.00 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 13:29.16 | basta_ | http://www.voip-info.org/wiki/index.php?page=Asterisk+consultants+Europe |
| 13:30.01 | af_ | mhh |
| 13:30.22 | A500mg | thk :) |
| 13:32.40 | *** join/#asterisk oduke (n=oduke@cottbus.gefoekom.org) |
| 13:32.59 | oduke | hi |
| 13:33.14 | oduke | can someone help me on a problem with max_forwards? |
| 13:33.48 | oduke | I am running astereisk 1.0.10 and the max_forwads entry in the invite header is missing |
| 13:34.44 | *** join/#asterisk areski (n=areski@215.Red-83-55-96.dynamicIP.rima-tde.net) |
| 13:34.44 | oduke | therefore I get max hps errors depending on the calls |
| 13:39.29 | kink0 | A500mg: voicein@aol.com and contact Mark |
| 13:43.20 | A500mg | lool |
| 13:43.41 | A500mg | sorry, i boycott aol |
| 13:43.55 | A500mg | they have a bad mail serveur configuration |
| 13:43.55 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 13:44.15 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 13:44.23 | A500mg | and they forced to use their bad navigator |
| 13:44.33 | A500mg | (sorry for my english) |
| 13:46.04 | *** join/#asterisk kio (n=kio@195-11.customer.cloud9.net) |
| 13:46.26 | joe | A500mg: http://www.voipsupply.com |
| 13:46.28 | *** join/#asterisk coppice_ (n=chatzill@129.204.17.210.dyn.pacific.net.hk) |
| 13:47.22 | A500mg | thk |
| 13:49.54 | kink0 | aol is a disaster and nothing serious for bussines users, but these guys sold my new TE405 for about 900 Euro |
| 13:50.20 | kink0 | original Digium, no clones. |
| 13:50.26 | A500mg | mmh |
| 13:50.34 | A500mg | but it's for PRI lines |
| 13:50.45 | A500mg | for BRI there is no digium cards :( |
| 13:50.48 | kink0 | A500mg: yes , TExxx are for PRI |
| 13:51.01 | A500mg | but there is "AVM" for BRI, good :) |
| 13:51.22 | kink0 | for BRI... use "any" BRI card, Teles, Eicon, Diva, Elsa ... or so. |
| 13:51.47 | A500mg | AVM B1, 1 port for BRI |
| 13:51.52 | A500mg | (active) |
| 13:51.56 | A500mg | 300€ |
| 13:52.12 | kink0 | hmmmm , really you need active one ? |
| 13:52.24 | A500mg | not really for active |
| 13:52.27 | kink0 | Elsa would costs about less than 20 E |
| 13:52.37 | A500mg | but the driver ... |
| 13:52.51 | kink0 | isdn4linux goes fine for that |
| 13:53.00 | A500mg | i want no problem and good installation |
| 13:53.23 | kink0 | I was ussing a lot of Elsa Quick Step PCI1000 cards on one access server, not problem at all. |
| 13:53.35 | A500mg | mmh |
| 13:54.04 | kink0 | for less than about 20E , you can try it, before to spent 300 E |
| 13:55.09 | kink0 | there any way to log origin IP in standar Asterisk CDR ? |
| 13:56.08 | kink0 | i means the originate IP from where SIP/AIX is comming from. |
| 13:56.46 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 13:58.17 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 13:58.32 | A500mg | kink0, an url for buy this card ? :) |
| 13:58.41 | A500mg | it's for my compagny |
| 13:58.51 | A500mg | we can buy this, for 20€ ... |
| 13:59.02 | RoyK | festival isn't really that good |
| 13:59.04 | kink0 | A500mg: I have not the URL for Mark at hand, but I can get her phone ( I have it at home ) |
| 13:59.12 | kink0 | ahhh the Elsa... |
| 13:59.14 | kink0 | sorry... |
| 13:59.19 | A500mg | yes :)= |
| 13:59.28 | coppice | RoyK the chinese new year festival has been OK so far |
| 13:59.33 | kink0 | is an old model, but goes fine, try eBay or so, since Elsa is not longer, but excellent card. |
| 14:00.19 | kink0 | if you have not found, I have some of them, from our old access server ( when the ADSL was not very populated, and we provide services ussing ISDN ) |
| 14:00.52 | kink0 | but I live in Spain, and surely you will be able to found one nearest you. |
| 14:00.56 | coppice | RoyK: we just had the traditional last dinner of the old year - salmon sashimi, all the way from norway :-) |
| 14:01.23 | RoyK | :) |
| 14:01.27 | RoyK | sashimi is nice |
| 14:02.53 | coppice | why is it so expensive, though? Its not like the cooking consumes much energy :-) |
| 14:07.26 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 14:08.29 | A500mg | kink0, in france the isdn is very used by enterprise |
| 14:08.48 | A500mg | and all pabx use isdn |
| 14:09.18 | kink0 | here too, but we do not use individual BRI, have been replaced by PRI ports |
| 14:09.28 | kink0 | I must to go, time for lunch , see you later. |
| 14:09.44 | A500mg | the voip is a good solution for call other, but for received call, the number is 08.xx.xx.xx.xx, and all enterprise have a number in 0[1-5].xx.xx.xx.xx |
| 14:09.51 | A500mg | mouarf :) |
| 14:10.09 | A500mg | but PRI lines are very expensive :/ |
| 14:10.19 | A500mg | it's only for big enterprises |
| 14:10.26 | *** join/#asterisk ToTo (n=ToTo@host70-163.pool872.interbusiness.it) |
| 14:10.44 | A500mg | (in france, for other lands i don't know) |
| 14:13.02 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 14:15.19 | coppice | it varies. a PRI costs between maybe 6 and 60 times the price of a single analogue line |
| 14:16.52 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:17.05 | coppice | BRI only caught on in a few places, like france and germany. its rare over most of the planet |
| 14:17.51 | puzzled | afternoon all |
| 14:19.11 | puzzled | coppice: BRI is major in .nl, belgium too |
| 14:20.31 | coppice | outside .eu its hard to find. a chinese ISDN phone maker told me they sell 100% to europe. the handful that get used outside .eu are redistributed from .eu |
| 14:21.28 | coppice | you'd have to be crazy to use BRI in many places, like here |
| 14:21.41 | puzzled | coppice: sure but only because of those silly stubborn americans that insist on some ancient cellphone technology and 56k or T1 :) |
| 14:22.20 | coppice | i think its mostly because an analogue phone costs about $1 and an ISDN phone doesn't |
| 14:22.42 | coppice | however, in places like I live the pricing for BRI stops anyone using it |
| 14:22.54 | *** join/#asterisk Gourou_fou_ (n=x@ACaen-151-1-21-205.w86-195.abo.wanadoo.fr) |
| 14:22.57 | Gourou_fou_ | re |
| 14:23.05 | puzzled | it may be more expensive in parts but not that much. look at a hfc-s based card. but I agree that in many places it is cost prohibitve |
| 14:23.21 | Gourou_fou_ | erf |
| 14:23.29 | A500mg2 | strange .. |
| 14:23.42 | puzzled | coppice: and what is your alternative? ftth, 8meg adsl at $50/mo etc |
| 14:24.01 | coppice | i don't think prohiobitive is the issue. it offers nothing, so it has to be no more expensive |
| 14:24.37 | coppice | BRI doesn't really cut it for data. too slow. $15 a month for ADSL is much more reasonable |
| 14:24.44 | A500mg2 | how cost a 16Mb/8Mb adsl connection in your country ? |
| 14:24.52 | puzzled | coppice: well I worked for Lucent and I can tell you that an ISDN2 linecard was a hell of a lot more expensive than the analog stuff |
| 14:24.55 | coppice | 1.5M or above too. none of this capped crap |
| 14:24.58 | A500mg2 | yes, BRI is good for phone, not for data |
| 14:25.07 | puzzled | it beats analog |
| 14:25.20 | coppice | you mean the exchange side line card? |
| 14:26.00 | puzzled | yup, in the 5E |
| 14:27.04 | coppice | no doubt. I'm not familiar with recent pricing, but a few years ago the difference was massive. it should be less massive now |
| 14:27.17 | puzzled | let's hope so |
| 14:27.22 | coppice | we implemented the first ISDN BRI Mux in the world |
| 14:27.32 | coppice | it was built to BT requirements |
| 14:27.32 | puzzled | nice, where was that? |
| 14:27.42 | coppice | it was working out expensive |
| 14:27.55 | coppice | BT kept saying it was what they wanted |
| 14:28.05 | coppice | we kept thinking they would never buy |
| 14:28.53 | coppice | at the end of the day, they threw up their hands at the price, and didn't buy. they paid 100% NRE, but still it tied up a lot of engineers that could have designed something useful |
| 14:30.01 | coppice | the killer then was they insisted on operation up to the copper planning limit. in the 80s that required a very expensive state of the art chip just for EC |
| 14:30.05 | puzzled | silly isn't it. they could prolly have figured that out long before moving so far into the project |
| 14:31.12 | coppice | what they bought initially was something american :-) it only worked up to a couple of km, by using a cheaper EC |
| 14:31.45 | *** join/#asterisk Astar (n=astar@ANantes-154-1-36-76.w81-53.abo.wanadoo.fr) |
| 14:31.49 | puzzled | I did KPN's first ISDN30 <-> ISDN30 data install in .nl. that was fun too, working with engineers that had no clue about ISDN |
| 14:33.26 | A500mg2 | i've recompile asterisk and asterisk-addons, but i've not the cdr_addon_mysql.so, strange ... |
| 14:33.41 | A500mg2 | i try with install libmysqlclient-... and recompile ast... |
| 14:34.54 | puzzled | A500mg: you need mysql-devel too. do you have that installed? |
| 14:35.18 | A500mg2 | yes with libmysql :) |
| 14:35.23 | A500mg2 | compilation in progress.. |
| 14:35.55 | A500mg2 | ( dpkg -l mysql-server libmysqlclient*dev ) |
| 14:36.16 | puzzled | yum install mysql* :) |
| 14:36.36 | *** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net) |
| 14:36.50 | A500mg2 | no :) |
| 14:36.52 | A500mg2 | apt |
| 14:37.11 | A500mg2 | ahhhhhhhhhhhh |
| 14:37.24 | A500mg | :) |
| 14:38.00 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 14:38.09 | austinnichols101 | anyone working with SER + DD-WRT? |
| 14:38.25 | puzzled | austinnichols101: this is #asterisk :) |
| 14:38.59 | austinnichols101 | puzzled: yes - asterisk issue at the core. |
| 14:39.11 | austinnichols101 | But if there's a better place to ask I can go there |
| 14:40.08 | austinnichols101 | I'm placing an outbound call from my asterisk server to a remote phone via sip. The phone (if left unanswered) will ring three times and then I'll hear a fast busy (disconnect) |
| 14:40.12 | austinnichols101 | trying to figure out why |
| 14:40.45 | A500mg | ohh, i've a cdr_addon_mysql.c file in asterisk-addons source directory |
| 14:40.48 | A500mg | i tried ... |
| 14:41.25 | A500mg | ahhhh, it's compiling cdr_mysql |
| 14:41.29 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
| 14:48.58 | af_ | sip channel: is better define voip phones as host=dynamic or not? |
| 14:50.15 | A500mg | i define voip phone with host=dynamic and it's work :) |
| 14:50.30 | A500mg | i've tried with fixed ip and no .. |
| 14:52.48 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:06.44 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 15:13.41 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 15:17.42 | EriSan | does anyone have streaming MOH working ? |
| 15:19.08 | SibrPhrek | EriSan: i have been trying to for the last couple days |
| 15:19.26 | SibrPhrek | apparently there is an issue with MOh pauses the stream |
| 15:19.54 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 15:20.20 | *** join/#asterisk dalbjerg (n=dalbjerg@host095a.malmohus16.se) |
| 15:20.24 | EriSan | on the cli i see "started music on hold", and instantly after "stopped music ..." |
| 15:21.03 | SibrPhrek | yeah |
| 15:21.24 | SibrPhrek | b/c the stream was paused, and when moh starts it there's no music (because it's not buffered) |
| 15:21.28 | *** join/#asterisk littleball (n=littleba@cm169.epsilon169.maxonline.com.sg) |
| 15:21.29 | SibrPhrek | so it stops |
| 15:21.41 | dalbjerg | Hello, iam trying to setup my asterisk, with bristuff. So i can make call out of my ISDN, but i can't get it to work... Asterisk is started, but when i make a call in to the ISDN, nothing hapende, in asterisk cli. |
| 15:21.50 | EriSan | so no way at the moment? |
| 15:21.58 | SibrPhrek | EriSan: not really |
| 15:22.06 | SibrPhrek | EriSan: what OS are you running on? |
| 15:22.12 | EriSan | centos |
| 15:22.13 | A500mg | Jan 28 16:25:28 ERROR[1902]: cdr_addon_mysql.c:436 my_load_module: Failed to connect to mysql database asterisk on localhost. |
| 15:22.16 | *** join/#asterisk arkanis (n=test@62-2-138-202.business.cablecom.ch) |
| 15:22.17 | A500mg | ouinnn |
| 15:22.22 | littleball | hello, when i jump to another context by using GOto, if the target extension is not defined in the target context, which extension will be executed? |
| 15:22.34 | *** join/#asterisk arendjr (n=junior@dsl-083-247-031-058.solcon.nl) |
| 15:22.48 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 15:24.55 | A500mg | oh just a bad socks |
| 15:25.17 | arendjr | hi guys, for some reason DTMF tones are not working with H323 channels on my Asterisk box, they are received from the SIP phone (debug channels says: "Sending dtmf: 50 (2), at 192.168.1.226") and it also works over the Zapata channel, just not with H323. Anyone knowns what could cause this? |
| 15:25.27 | *** join/#asterisk DeadZen (n=DeadZen@adsl-153-136-41.mia.bellsouth.net) |
| 15:25.36 | DeadZen | god there's a lot of people here |
| 15:25.43 | DeadZen | hi |
| 15:26.59 | *** join/#asterisk newmember (n=newmembe@S010600036d1139bb.cg.shawcable.net) |
| 15:27.13 | DeadZen | hehe |
| 15:30.23 | *** part/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 15:31.47 | *** join/#asterisk kink0 (n=kinko@pluton.interec.com) |
| 15:31.48 | kink0 | re |
| 15:32.25 | Trazz | can * take a call in and transfer to extesion and then make a call out to do like follow me? |
| 15:32.26 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 15:32.48 | littleball | hello, when i jump to another context by using GOto, if the target extension is not defined in the target context, which extension will be executed? |
| 15:33.00 | kink0 | I have chose generic 586 g.729a.so before, but know I am planing to use up to 60 licences on a dual Xeon, any sugestion ? |
| 15:33.05 | DeadZen | littleball: i believe it just fails |
| 15:33.11 | DeadZen | seems to for me anyway |
| 15:33.31 | littleball | e.g. Goto(submenu,test,1) |
| 15:33.32 | DeadZen | do all versions of ms messenger hav sip? |
| 15:33.38 | littleball | but test extension is not defined. |
| 15:34.02 | [TK]D-Fender | littleball : try it and see |
| 15:34.03 | littleball | DeadZen, does the "i" extension will be executed? |
| 15:34.28 | DeadZen | well i use goto like this |
| 15:34.48 | DeadZen | exten => 100,1,dial(SIP/myphone) |
| 15:34.56 | DeadZen | exten => myphone,1,goto(100,1) |
| 15:35.18 | DeadZen | need to have a [myphone] section in sip.conf |
| 15:35.27 | *** part/#asterisk arkanis (n=test@62-2-138-202.business.cablecom.ch) |
| 15:36.12 | littleball | context definition is there. but the extension is not there. |
| 15:36.22 | RoyK | hi |
| 15:36.23 | [TK]D-Fender | DeadZen : What is going to dial "myphone" in that context? |
| 15:36.26 | RoyK | anyone seen this product? |
| 15:36.26 | RoyK | http://products.nortel.com/go/product_assoc_detail.jsp?segId=0&parId=0&doc_id=8&catId=null&rend_id=99pt&contOid=100176499&prod_id=25080&locale=en-US |
| 15:36.28 | kink0 | anybody know if digium g729 licences are ONLY MAC based and I will be able to change the codec_g729a.so ussing the same licences ? |
| 15:36.35 | RoyK | er |
| 15:36.35 | RoyK | http://products.nortel.com/go/product_content.jsp?segId=0&parId=0&prod_id=25080 |
| 15:36.38 | littleball | i want to define a default extension which can handle such GoTO |
| 15:36.50 | DeadZen | ok you guys have like scrolled my screen to hell hehe |
| 15:37.22 | RoyK | kink0: it's mac based and you can re-register it once after initial registering. after that, you need to contact digium |
| 15:37.30 | RoyK | or spoof the mac addr |
| 15:37.31 | RoyK | perhaps |
| 15:37.46 | [TK]D-Fender | littleball : pastebin the sample of what you'v got in mind.... typically you should plan your dialplan so that it DOESN"T get invalid entries..... |
| 15:38.16 | DeadZen | hey I can't hear any sounds when i do a Playback() |
| 15:38.18 | kink0 | RoyK: well I do not pretend to spoof the MAC... just to change the version of codec_g729a.so if needly, because I am not really sure what one to use |
| 15:38.24 | DeadZen | i installed the asterisk-sounds but no cigar |
| 15:38.34 | [TK]D-Fender | DeadZen : Have you tried putting an "Answer" first? |
| 15:38.52 | DeadZen | i believe so.. lemme try again |
| 15:38.53 | littleball | [TK]D-Fender, my case is special |
| 15:38.57 | DeadZen | what i was thinking |
| 15:39.07 | kink0 | I have used a generic 586 version for try it, and I have two licenses, but know I pretend to use about 60 licenses and I will like to use the most performanced one for my system. |
| 15:39.09 | DeadZen | is that there's no like Wait() or something and its just not allowing enough time to play |
| 15:39.09 | [TK]D-Fender | littleball : Do explain.... |
| 15:39.17 | DeadZen | does it play the duration regardless of wait? |
| 15:39.31 | littleball | I use GoTo an routing context to do PRI routing |
| 15:39.46 | Trazz | ~doc |
| 15:39.47 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 15:39.51 | [TK]D-Fender | DeadZen : pastebin the extens you you are calling, and do you se ti being called in CLI? |
| 15:39.56 | littleball | the target extension is specified by the user from WEB or other place, |
| 15:40.11 | DeadZen | yah i see it playing it in cli |
| 15:40.13 | DeadZen | just cant hear it |
| 15:40.35 | littleball | like exten=>_65X.,1,Set(path=Zap/r1/${EXTEN}|d=10) |
| 15:40.41 | [TK]D-Fender | littleball : I doubt you need to do it in a way that you can't do things explicitly or use a standard feature like "i" or "_X" as a catch-all |
| 15:41.15 | littleball | "i" doesn't work. it seems _X works. i just test "i". it doesn't work |
| 15:41.15 | kink0 | RoyK: hmmm would I able to re-register two times ? well that allow me to do a try before to select the definitive one, as I can use then the same license I have adquired yet for the develoment machine. |
| 15:41.44 | littleball | where is the pastin web? |
| 15:41.48 | kink0 | both will be not working at same time, even in develoment machine will no longer any Asterisk running, once we started the production on a new machine |
| 15:42.00 | [TK]D-Fender | littleball : And there is a testexten or similar function that allws you to see if it would work, but I don't believe Goto necessarily has any protection built-in |
| 15:42.24 | DeadZen | TKD... should it be pretty much like Answer() Playback(hello-world) Hangup() |
| 15:42.27 | [TK]D-Fender | '~pb |
| 15:42.29 | littleball | [TK]D-Fender, what is the pastin link? |
| 15:42.30 | [TK]D-Fender | ~pb |
| 15:42.31 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 15:42.45 | [TK]D-Fender | DeadZen : Assuming the sound file exists |
| 15:42.56 | littleball | thanks bot |
| 15:42.57 | DeadZen | yes it does.. all paths and files can be considered verified |
| 15:43.07 | DeadZen | just don't hear it.. |
| 15:43.08 | RoyK | kink0: you should be able to register the same license on two machines |
| 15:43.12 | DeadZen | should i paste bin the CLI response? |
| 15:43.22 | RoyK | kink0: or more if you bug digium about it |
| 15:43.24 | DeadZen | im doing asterisk -vvvgrc to read it |
| 15:43.31 | DeadZen | 4 v's |
| 15:43.40 | littleball | hi, pls see http://pastebin.com/527381 |
| 15:44.03 | *** join/#asterisk dfgas (n=dfgas@adsl-69-210-72-31.dsl.milwwi.ameritech.net) |
| 15:44.27 | littleball | [TK]D-Fender, any confliction in my extension definition? |
| 15:44.31 | DeadZen | hehe pastebin makes me want to write a bin for asterisk... with syntax hilighting ;-) |
| 15:44.54 | dfgas | ok i downloaded new version and did a make && make install and now it give me mpg123 errors and asterisk dies |
| 15:45.02 | dfgas | what did i do wrong |
| 15:45.04 | littleball | actually, this routing context works like stack |
| 15:45.12 | DeadZen | does asterisk require mpg123 ? |
| 15:45.14 | DeadZen | http://pastebin.com/527384 |
| 15:45.15 | littleball | i like it myself very much |
| 15:46.13 | [TK]D-Fender | littleball : Where do those 3 vars get set? |
| 15:46.17 | DeadZen | TK my asterisk is up think its just sjphone not playing it? |
| 15:46.57 | littleball | [TK]D-Fender, there is no problem for this 3 vars. they got set from previous context. this routing context works as stack... |
| 15:47.02 | littleball | push/pop |
| 15:47.23 | dfgas | http://pastebin.com/527389 |
| 15:47.34 | littleball | [TK]D-Fender, i want to confirm the extension matching works fine for me |
| 15:48.14 | littleball | it works for me i think |
| 15:49.08 | DeadZen | :-( |
| 15:49.44 | dfgas | and this is the error i get with asterisk -vvvvvc http://pastebin.com/527399 |
| 15:49.52 | dfgas | that is where it stops |
| 15:50.02 | dfgas | anyone have an idea why its doing this |
| 15:50.22 | dfgas | am i suppose to reinstall anything else with asterisk when upgrading? |
| 15:50.36 | dfgas | or is there a how to on upgrading |
| 15:51.50 | *** join/#asterisk razu (n=razu@217-159-187-162-dsl.prn.estpak.ee) |
| 15:53.03 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfi0l.dialup.mindspring.com) |
| 15:56.29 | arendjr | does anyone know why DMTF may not be working over a H323 channel? I'm using dtmfmode=string... |
| 15:57.47 | [TK]D-Fender | DeadZen : Where is your SJP relative to *, and do you gt audio from anything else? |
| 15:58.27 | DeadZen | thanx for gettin back to me [TK]D-Fender |
| 15:58.33 | DeadZen | here's a better pb |
| 15:58.34 | DeadZen | http://pastebin.com/527406 |
| 15:58.59 | DeadZen | servers majorcomputing.com |
| 15:59.42 | [TK]D-Fender | DeadZen : I only asked where is was RELATIVE to *. ( same local subnet, remote? ANT'd?) |
| 15:59.45 | [TK]D-Fender | NAT* |
| 15:59.59 | DeadZen | im not getting the question |
| 16:00.08 | DeadZen | its my windows computer connecting to my colocated linux server |
| 16:00.21 | [TK]D-Fender | Where in your network is SJPhone relative you your * server? |
| 16:00.25 | DeadZen | and yes im behind a nat.. |
| 16:00.59 | DeadZen | dsl router/modem -> colocated server |
| 16:01.01 | [TK]D-Fender | DeadZen : doun't just say"*I'm* behind NAT." I need to know what BETWEEN SJphone and *. |
| 16:01.37 | DeadZen | what is .* |
| 16:01.41 | DeadZen | and what do you mean by 'between' |
| 16:01.57 | [TK]D-Fender | dfgas : try "asterisk -vvvvvvvvvgc" |
| 16:02.45 | DeadZen | what network components are between my computer running sjphone and my server? |
| 16:02.52 | Trazz | what between ears |
| 16:02.55 | *** join/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 16:02.58 | [TK]D-Fender | DeadZen : Its a simple question. How far away is SJPhone from your * server?!?!?! is it a friend place on a different network? Is there a NAT router BETWEEN SJPhone and *? If so, on who's side? Both? |
| 16:03.15 | [TK]D-Fender | DeadZen : yes, that sounds like the same question. |
| 16:03.26 | DeadZen | i thought I said that |
| 16:03.36 | DeadZen | im connecting through a router to a colocated server |
| 16:03.36 | [TK]D-Fender | DeadZen : And do you get sound from ANY other application? (Echo, Dial, etc?) |
| 16:03.46 | DeadZen | the colocated server is directly connected to a 100mb pipe on cogent |
| 16:03.48 | *** join/#asterisk Soul (n=Soul@87-196-44-148.net.novis.pt) |
| 16:03.58 | dfgas | k i got i t back up and running |
| 16:04.03 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfi0l.dialup.mindspring.com) |
| 16:04.14 | [TK]D-Fender | DeadZen : So your SJphone is behind a NAT, registering to a remote PUBLIC IP * server? |
| 16:04.27 | DeadZen | yes registering to majorcomputing.com |
| 16:04.47 | dfgas | anyone know why when i startup asterisk anyone calling into my DID won't get asterisk unless i make an out going call once |
| 16:04.54 | DeadZen | if you have another configuration for testing ill be happy to try it [TK]D-Fender |
| 16:04.55 | dfgas | is there a way to fix this? |
| 16:05.04 | *** join/#asterisk dalbjerg (n=dalbjerg@host095a.malmohus16.se) |
| 16:05.31 | [TK]D-Fender | DeadZen : Ok well the * server needs to know you're behind NAT which it currently DOESN'T. Add these lines to your sip.conf entry : "nat=yes", "qualify=2000" |
| 16:06.02 | [TK]D-Fender | dfgas : How is this DID coming in? |
| 16:06.09 | DeadZen | yes i am behind a nat... 192.168.0.61 connecting to a public ip which has no firewall at ip 69.41.162.31 |
| 16:06.16 | *** join/#asterisk saftsack (n=oliver@p54A7EBF7.dip.t-dialin.net) |
| 16:06.21 | saftsack | hi |
| 16:06.24 | dfgas | [TK]D-Fender: sip |
| 16:06.28 | saftsack | some hylafax cracks here? |
| 16:06.43 | DeadZen | hylafax hehe there's an old app |
| 16:06.48 | [TK]D-Fender | DeadZen : Add the lines I told you to your server's entry for your SJPhone, and reload the config. then restart SJPhone to re-register. Should work then. |
| 16:07.08 | [TK]D-Fender | dfgas : is your * behind NAT? |
| 16:07.19 | dfgas | [TK]D-Fender: yes |
| 16:07.21 | saftsack | DeadZen, asterisk isnt good for faxing |
| 16:07.34 | DeadZen | o |
| 16:07.35 | dfgas | [TK]D-Fender: i have all ports open that i am told to have open |
| 16:07.35 | *** join/#asterisk anonymouz666 (n=lynx@200.218.193.6) |
| 16:08.09 | dfgas | [TK]D-Fender: and sip show reistry shows that i am registered |
| 16:08.18 | DeadZen | [TK]D-Fender: that did the trick good buddy |
| 16:08.22 | [TK]D-Fender | dfgas : You need to add either "EXTERNHOST" or "EXTERNIP", and "LOCALNET" entries into your sip.conf's [general] section for it to work. |
| 16:08.30 | [TK]D-Fender | DeadZen : ywc. |
| 16:08.38 | DeadZen | [TK]D-Fender: now does that work when someones NOT behind a nat? |
| 16:08.59 | [TK]D-Fender | DeadZen : What you did would have been fine if they were both "public" to each other. |
| 16:09.20 | DeadZen | I get it.. but you understand most people are behind nat |
| 16:09.54 | DeadZen | and if they aren't behind nat will it work with nat=yes and qualify=2000 ? |
| 16:09.57 | [TK]D-Fender | DeadZen : Its jsut that your router won't know where to send the UDP packets to unless it is contantly sent a "keep-alive" signal by "qualify". |
| 16:09.59 | DeadZen | as in public to public |
| 16:10.12 | DeadZen | ya makes sense |
| 16:10.18 | [TK]D-Fender | DeadZen : I think it should still work... |
| 16:10.26 | DeadZen | i never touched asterisk till 5am this morning |
| 16:10.27 | DeadZen | hehe |
| 16:10.32 | *** join/#asterisk coppice_ (n=chatzill@44.194.17.210.dyn.pacific.net.hk) |
| 16:10.51 | DeadZen | now im addicted.. |
| 16:10.58 | [TK]D-Fender | DeadZen : good start.. I haven't found a book or guide that I liked so I might jsut write my own. |
| 16:11.10 | DeadZen | is realtime any good? I was curious about real time extensions.conf |
| 16:11.36 | [TK]D-Fender | DeadZen : Some think its the second-coming, but I guess it depends if you need it for scaling.... |
| 16:11.45 | DeadZen | I'm a damn good web developer so if I have access to stuff like that I could make a whole host of open source business integrations |
| 16:11.51 | [TK]D-Fender | it IS buggy though still... (as in enough for you to want to think twice) |
| 16:12.07 | DeadZen | i think I heard the major bug is for instance |
| 16:12.16 | dfgas | [TK]D-Fender: externhost would be the domain that is coming into my box? |
| 16:12.18 | DeadZen | if the database connection is down the dialplan gets wiped |
| 16:12.24 | DeadZen | as opposed to a reload being skipped |
| 16:12.47 | anonymouz666 | With Asterisk and E1 card, A customer call to my company... Asterisk pick up the call and do a dial to technical support guy. After they start to talk, the TI manager join the conversation (like chanspy) and start to talk - he can hear both sides, but when he speaks only the technical guy can hear. Is that possible? |
| 16:13.03 | [TK]D-Fender | dfgas : if you had one, yes. Many people running a DynDNS type service would use that to keep their servers "finable" and use "EXTERNREFRESH" to set the frequency of the checks |
| 16:13.06 | *** join/#asterisk BugKham (n=lamer@125.24.29.219) |
| 16:13.31 | dfgas | k |
| 16:13.33 | DeadZen | if thats the case i dont mind... if the database is ever down.. i have to restart apache anyway to flush persistent connections so ill add that to asterisk |
| 16:13.42 | DeadZen | can you do a sighup on asterisk to simulate a reload command? |
| 16:13.52 | DeadZen | just curious |
| 16:14.19 | [TK]D-Fender | DeadZen : dunno.. why not jsut call "reload" normally? |
| 16:14.33 | dfgas | [TK]D-Fender: would would be a good number to put in there? |
| 16:14.37 | DeadZen | could you call it externally i mean |
| 16:14.56 | [TK]D-Fender | dfgas : not sure really... every 5 minutes maybe? |
| 16:15.08 | *** join/#asterisk pengyong (n=lala@218.93.153.249) |
| 16:15.19 | dfgas | is it by seconds or minutes? heh |
| 16:15.30 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
| 16:15.30 | dfgas | thats kinda what i was wondering :) |
| 16:15.50 | [TK]D-Fender | dfgas : no idea actually.. just know that the option exists |
| 16:15.58 | [TK]D-Fender | WIKI it up and confirm |
| 16:16.17 | dfgas | heh, k |
| 16:16.20 | DeadZen | haha my asterisk servers says "Nobody here but us chickens" when you call it.. |
| 16:16.22 | dfgas | i will try this quick |
| 16:16.31 | *** join/#asterisk littleball (n=littleba@cm169.epsilon169.maxonline.com.sg) |
| 16:16.56 | littleball | hello, who can help me check this: exten=>h,3,Set(t1=$[$["foo${DIALEDTIME}" != "foo"]?$[ ${DIALEDTIME} / ${d} ]:0]) |
| 16:17.14 | littleball | is there any mistake ? |
| 16:17.14 | A500mg | channel.c:784 channel_find_locked: Avoided initial deadlock for '0x815dd78', 10 retries! |
| 16:17.35 | littleball | asterisk variable is not easy to handle |
| 16:17.48 | dfgas | [TK]D-Fender: the weird thing is, is that * shows the incoming call but i get the voicemail from the voip itself unless i make a out going call |
| 16:17.58 | dfgas | guess i should have stated that, sorry |
| 16:18.35 | dfgas | i have incoming calls set to / |
| 16:18.43 | dfgas | so it handles all of them |
| 16:19.33 | [TK]D-Fender | dfgas : Maybe your SIP entry isn't quite right for your provider |
| 16:19.56 | dfgas | hmmm, for incoming or trunk? |
| 16:20.03 | *** join/#asterisk SkramX (n=mark@unaffiliated/skramx) |
| 16:20.07 | [TK]D-Fender | incoming |
| 16:20.10 | dfgas | hmmmm |
| 16:21.20 | *** join/#asterisk oceanlan|dustin (n=info@cpe-69-133-109-130.woh.res.rr.com) |
| 16:21.30 | DeadZen | [TK]D-Fender: if the sound is scratchy.. what is that an indication of |
| 16:21.47 | DeadZen | [TK]D-Fender: sjphone just has a bad decoder or something? |
| 16:22.05 | DeadZen | [TK]D-Fender: or just normal udp packets carrying sound disappearing |
| 16:22.07 | oceanlan|dustin | DeadZen: what codec are you using? |
| 16:22.17 | DeadZen | gsm i think |
| 16:22.21 | DeadZen | where do you tweak the codecs |
| 16:22.22 | [TK]D-Fender | littleball : that Set looks a LITTLE OFF.... |
| 16:22.35 | [TK]D-Fender | DeadZen : What codec is the call using? |
| 16:22.50 | DeadZen | you know im not sure |
| 16:22.56 | DeadZen | familiar with sjphone? |
| 16:23.01 | oceanlan|dustin | DeadZen: I have had quality problems with GWM also..being so compressed it gets scratchy and shoppy just like cell phones some times |
| 16:23.12 | DeadZen | yah it sounds like a cellphone |
| 16:23.14 | DeadZen | good analogy |
| 16:23.19 | [TK]D-Fender | DeadZen : not partiularly. I haven't had to try a Linux softphone to date. |
| 16:23.38 | oceanlan|dustin | DeadZen: G711 is big packets but is "toll quality" FYI |
| 16:23.39 | DeadZen | ok lemme poke around |
| 16:23.39 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 16:24.27 | oceanlan|dustin | DeadZen: I have sjphone but I have never tried GSM on that phone, The codec is specified in one of the ** config files... |
| 16:24.47 | oceanlan|dustin | ulaw = G711 = best quality = 80kbps |
| 16:24.57 | DeadZen | what other non java based sip software is there |
| 16:25.07 | Trazz | TK, when i dial an extension like a softphone thats not registered due to laptop being powered off i get the person is on the phone. i know about chanunavail, busy, no answer but I am trying to find a greate example of how to implement being able to deal with all these conditions properly |
| 16:25.46 | *** join/#asterisk whiteblue (n=whiteblu@mnch-d9ba4fc0.pool.mediaWays.net) |
| 16:25.46 | oceanlan|dustin | lots, xten is one...and many more (having a brain cramp atm) |
| 16:25.52 | DeadZen | isn't xten java? |
| 16:25.55 | [TK]D-Fender | Trazz : What does DIALSTATUS say when they are "unreachable"? |
| 16:26.16 | oceanlan|dustin | ahhh..sorry didnt catch that in the question... |
| 16:26.24 | DeadZen | yah.. java sux big time |
| 16:26.33 | *** join/#asterisk snewpy_ (n=markl@210.84.44.179) |
| 16:26.36 | DeadZen | and a cause of over 300 spyware infections on windows too |
| 16:26.40 | oceanlan|dustin | your right...i dont know of many non-java sip phones.. |
| 16:26.48 | DeadZen | i actually make everyone remove their jre's |
| 16:26.57 | oceanlan|dustin | hah |
| 16:26.57 | Trazz | Jan 28 10:39:29 NOTICE[3933]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 16:27.11 | *** join/#asterisk Igbothom (n=HiltonT@office.quarkit.com.au) |
| 16:27.26 | [TK]D-Fender | Trazz : NoOp the dialstatus variable and SEE what it says! |
| 16:27.36 | DeadZen | oceanlan|dustin: is the codec controlled by the server or the client |
| 16:27.39 | oceanlan|dustin | i use Diax and Firefly myself..i prefer IAX to sip....much easier |
| 16:27.44 | DeadZen | what if they don't have the same ones set |
| 16:28.06 | oceanlan|dustin | Server I believe...but the client has to be able to "accept" the codec... |
| 16:28.14 | whiteblue | Hi, sorry to disturb. I have a problem with compiling Asterisk 1.2.x (also the latest) on SuSE10. Make always dies with "chan_zap.c:9080: error: too few arguments to function âpri_newâ" and I cannot find any information about this problem somewhere else. Latest zaptel module is installed if this matters at all |
| 16:28.23 | DeadZen | oceanlan|dustin: where do i change the codec on the server? |
| 16:28.39 | DeadZen | just in codecs.conf? |
| 16:29.07 | oceanlan|dustin | thats 1...then I believe in sip.conf and iax.conf you have to also make it match... |
| 16:29.22 | [TK]D-Fender | DeadZen : in sip.conf entrey you add "disallow=all", and then add the codecs you want in order "allow=ulaw", 'allow=gsm", etc |
| 16:29.27 | oceanlan|dustin | like "allow=ulaw" , "disallow=gsm" |
| 16:29.36 | DeadZen | cool thanx |
| 16:29.46 | DeadZen | should i turn on preprocess in codecs ? |
| 16:29.50 | Trazz | fast busy |
| 16:29.52 | kink0 | whiteblue: install libpri |
| 16:30.23 | oceanlan|dustin | TK - do you need to change the codecs.conf or is that just giving you the ability to choose between them? |
| 16:30.36 | Trazz | i want it to say they are on the phone whey they are and then send to voicemail, then if not available to go to voicemail, et |
| 16:30.40 | oceanlan|dustin | I personally always set the rule in sip.conf.. |
| 16:30.54 | [TK]D-Fender | Don't mess with codecs.conf. make your settings PER account |
| 16:30.57 | oceanlan|dustin | DeadZen: not sure on the pre-process, never messed with it |
| 16:30.59 | kink0 | I also set codecs rules in sip.conf instead codecs.conf |
| 16:31.00 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 16:31.08 | oceanlan|dustin | TK - thanks |
| 16:31.08 | DeadZen | ok won't touch it |
| 16:31.24 | DeadZen | that ulaw change made no real difference |
| 16:31.29 | DeadZen | its just got a lot of clicks and pops it seems |
| 16:31.40 | Trazz | TK, did you see its fast busy on noop |
| 16:31.44 | [TK]D-Fender | Trazz : Did you add the NoOp like I suggested and test it? |
| 16:31.50 | Trazz | yes |
| 16:31.55 | Trazz | made it first in the list |
| 16:31.56 | oceanlan|dustin | I had read something about codecs.conf and such that you can do something in it to always allow or disallow certain ones...but i never tried it.. |
| 16:32.08 | [TK]D-Fender | Trazz : pastebin the call |
| 16:32.13 | Trazz | ok |
| 16:32.44 | oceanlan|dustin | DeadZen: do you have any way to check your sent/recieved udp packets? |
| 16:32.47 | *** part/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 16:32.47 | [TK]D-Fender | oceanlan|dustin : If you want to restrict more globally justdo it in the [general] section of sip.conf and iax.conf to suit your tastes. |
| 16:33.06 | DeadZen | hrmm.. not that i can think of off hand |
| 16:33.14 | DeadZen | im on windows |
| 16:33.18 | DeadZen | maybe ethereal? |
| 16:33.22 | oceanlan|dustin | <[TK]D-Fender> thanks for the info |
| 16:33.27 | kink0 | anyway to log originating SIP ip address ussing standard Asterisk CDR ? |
| 16:33.52 | oceanlan|dustin | hmmm...see i use a software firewall that allows me to make sure the packets are getting from point a to point b |
| 16:34.00 | oceanlan|dustin | i never tried it with etherreal.. |
| 16:34.23 | Trazz | http://pastebin.com/527452 |
| 16:34.46 | DeadZen | hey oceanlan|dustin |
| 16:34.55 | oceanlan|dustin | Can you try the call on another machine? maybe the nic or sound card is jank? |
| 16:34.56 | DeadZen | can you connect to me and tell me if you hear little clicks and pops |
| 16:35.08 | oceanlan|dustin | sure, whats the info |
| 16:35.10 | [TK]D-Fender | kink0 : put in the the "userfield" |
| 16:35.13 | DeadZen | nah i listen to music all the time |
| 16:35.19 | DeadZen | majorcomputing.com |
| 16:35.22 | DeadZen | mysjphone / blah |
| 16:35.38 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 16:35.39 | DeadZen | its not onboard sound its a sound blaster card |
| 16:35.51 | oceanlan|dustin | k..hold on a sec, i dont have a mic here with me but I can at least listen.. |
| 16:36.09 | [TK]D-Fender | Trazz : You were supposed to NoOp the variable AFTER THE DIAL. |
| 16:36.34 | [TK]D-Fender | Trazz : exten => 2000,2,NoOp(Dial status was -${DIALSTATUS-)} |
| 16:36.43 | [TK]D-Fender | Trazz : exten => 2000,2,NoOp(Dial status was -${DIALSTATUS}-) |
| 16:37.15 | [TK]D-Fender | and then pastebin the call with it |
| 16:37.24 | Trazz | ok will do |
| 16:38.00 | whiteblue | kink0: Thx, I realy forgot to update the header file in /usr/lib. Now its finaly compiling. |
| 16:38.21 | dfgas | [TK]D-Fender: it tells me to put nothing in for incoming on the trunk |
| 16:38.24 | DeadZen | ew this guy on the telly just drank a 40 year old beer |
| 16:38.25 | oceanlan|dustin | what ext. should i call? |
| 16:38.38 | DeadZen | ocean: mysjphone |
| 16:38.43 | [TK]D-Fender | dfgas : pastebin everything relevent to your connection. |
| 16:38.45 | DeadZen | or 100 same thing |
| 16:38.46 | dfgas | [TK]D-Fender: for incoming calls i have it handle everything to coming in, not just certain things |
| 16:39.08 | Trazz | TK, this is what i get with set verbose 20 on |
| 16:39.08 | Trazz | Jan 28 10:51:42 NOTICE[4108]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 16:39.11 | dfgas | [TK]D-Fender: but i can't get it to to work |
| 16:39.29 | Trazz | i added the noop |
| 16:39.32 | oceanlan|dustin | nobody here but us chickens.. |
| 16:39.42 | DeadZen | you hear clicks and pops though? |
| 16:39.43 | [TK]D-Fender | dfgas : I *know* it doesn't work. Show me what you're DOING so I can see if I can tell whats WRONG. |
| 16:39.47 | DeadZen | like a bad cell phone connection |
| 16:39.49 | oceanlan|dustin | yes |
| 16:40.07 | DeadZen | yah me too... i dont want it to sound crystal clear as itll eat processor/bandwidth |
| 16:40.07 | oceanlan|dustin | yea, that wasnt clear at all.. |
| 16:40.14 | DeadZen | but i do want about twice as good as that |
| 16:40.14 | [TK]D-Fender | trazz... where's that pastebin? |
| 16:40.21 | oceanlan|dustin | hmmmm... |
| 16:40.36 | dfgas | [TK]D-Fender: i am just trying to call in |
| 16:40.42 | oceanlan|dustin | interesting... |
| 16:40.52 | oceanlan|dustin | what type of connection is this box one? |
| 16:40.52 | DeadZen | i have no idea either |
| 16:40.56 | oceanlan|dustin | on*? |
| 16:41.01 | DeadZen | a great one 100mb burstable |
| 16:41.06 | oceanlan|dustin | wow |
| 16:41.08 | dfgas | [TK]D-Fender: not sure what else you mean :( |
| 16:41.21 | DeadZen | its burstable to 50 |
| 16:41.30 | Trazz | TK - http://pastebin.com/527479 |
| 16:41.33 | [TK]D-Fender | dfgas : Star listening please. I know that damn connection doesn't work and I want to see where you screwed up. Show me what your setup looks like ! Pastebin it! |
| 16:41.34 | oceanlan|dustin | hmmm...brb gotta get my daughter a bottle... |
| 16:41.39 | DeadZen | its a dual 3ghz xeon too |
| 16:42.03 | dfgas | k |
| 16:42.08 | [TK]D-Fender | Trazz : now pastebin a call to it. |
| 16:42.29 | DeadZen | [TK]D-Fender: are there any other quality knobs |
| 16:42.52 | [TK]D-Fender | DeadZen : What codec is being used for the calls? Any bandwidth concerns? |
| 16:43.02 | whiteblue | Did anyone try to connect asterisk to an avaya/tenovis pbx via voip? avaya/tenovis use a modified / extended h323 and an own audio codec, and it does not look like that ther (now linux based) pbx support a different one |
| 16:43.05 | Trazz | TK -http://pastebin.com/527481 |
| 16:43.06 | DeadZen | uhm just what you said |
| 16:43.10 | DeadZen | allow ulaw allow gsm |
| 16:43.19 | DeadZen | and no.. i have no bw concerns at the moment |
| 16:43.37 | [TK]D-Fender | DeadZen : Look a the CALL.. not jsut your config and what you THINK it should be doing.... |
| 16:43.52 | dfgas | http://pastebin.com/527483 |
| 16:43.56 | DeadZen | I don't get it |
| 16:44.10 | DeadZen | it should sound much better ;-) |
| 16:44.14 | dfgas | that is from sip_additional |
| 16:44.17 | DeadZen | how do i look at the call? |
| 16:44.22 | dfgas | what else would youlike |
| 16:44.28 | [TK]D-Fender | Trazz : I don't see the NoOp in that pastebin, for crying out loud don't cut out the NECESSARY info! |
| 16:44.39 | oceanlan|dustin | turn your verbosity up when you run asterisk -rvvvvvvvvvvvc |
| 16:44.41 | [TK]D-Fender | Trazz: Pastebin the ENTIRE DAMNED CALL |
| 16:44.46 | DeadZen | ok |
| 16:44.56 | Trazz | TK, that was everthing from my CLI |
| 16:45.09 | Trazz | i pasted teh window |
| 16:45.14 | Trazz | with verbose 20 |
| 16:45.17 | oceanlan|dustin | did you reload asterisk after you changed the codecs? |
| 16:45.27 | saftsack | hi |
| 16:45.27 | Trazz | i did reload after those cahnges too |
| 16:45.28 | DeadZen | of course |
| 16:45.32 | saftsack | some hylafax experts here? |
| 16:45.34 | [TK]D-Fender | trazz : Sorry, no it ISN'T. I don't see the Dial command being CALLED <---- |
| 16:45.47 | saftsack | because i have problems with the pause signal |
| 16:45.51 | Trazz | ok do i need to up the verbose ? |
| 16:46.05 | [TK]D-Fender | trazz : you are not pasting EVERYTHING. |
| 16:46.08 | oceanlan|dustin | it is sounding better than it was right now..not sure if you changed anything.. |
| 16:46.30 | Trazz | i am running 1.2.3 and maybes its issue with it |
| 16:46.42 | Trazz | i swear i am pasting all that is on my screen |
| 16:46.43 | [TK]D-Fender | I should see every damn command being called in "2000".... |
| 16:46.51 | DeadZen | http://pastebin.com/527486 |
| 16:47.12 | [TK]D-Fender | trazz : there is something ABOVE that line... |
| 16:47.39 | Trazz | do you want to see my context ? |
| 16:47.46 | dfgas | [TK]D-Fender: what exactly do you want me to paste? |
| 16:47.46 | [TK]D-Fender | Trazz : Look at DeadZen's pastebin and you see the kind of stuff I should be seeing... |
| 16:48.09 | [TK]D-Fender | dfgas : Your sip.conf entries for it and the related extensions.conf contexts |
| 16:48.45 | Trazz | TK, i am not getting that level of detail on my cli |
| 16:48.50 | Trazz | i used to before 1.2.3 |
| 16:48.58 | [TK]D-Fender | Trazz : that is a BASIC amount of info.... |
| 16:49.04 | DeadZen | trazz you need much more verbosity then |
| 16:49.13 | [TK]D-Fender | Trazz : Just set it to 7 and try again... |
| 16:49.35 | dfgas | k |
| 16:49.53 | [TK]D-Fender | Trazz : I just set mine to 11 and I see apps being called.... |
| 16:49.57 | Trazz | http://pastebin.com/527490 |
| 16:50.03 | DeadZen | Trazz: asterisk -rvvvvvvvvvvvc |
| 16:50.06 | Trazz | i am at 7 now |
| 16:50.14 | oceanlan|dustin | DeadZen: your call looks good...naybe the sound file is crap?? |
| 16:50.23 | DeadZen | oceanlan|dustin: hrmm |
| 16:50.37 | DeadZen | oceanlan|dustin: i just got the asterisk-sounds |
| 16:50.45 | DeadZen | oceanlan|dustin: I thought they were professionally done... are they crap? |
| 16:50.48 | *** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com) |
| 16:51.07 | [TK]D-Fender | Trazz : what version of * are you on? |
| 16:51.12 | Trazz | 1.2.3 |
| 16:51.25 | file[laptop] | FOOD |
| 16:51.31 | [TK]D-Fender | Trazz : You'll need to put that no-op in front of 102. |
| 16:51.32 | oceanlan|dustin | DeadZen: I donno..i have never heard the chickens before! I have only heard some of the others like tt-monkeys and the regular vmail stuff.. |
| 16:51.43 | [TK]D-Fender | Trazz : and turn off priority jumping! |
| 16:51.52 | [TK]D-Fender | file : Eat them! |
| 16:52.33 | file[laptop] | I have to locate food first... |
| 16:52.42 | DeadZen | ok |
| 16:52.52 | [TK]D-Fender | file[laptop] : NO! These people!!!!! |
| 16:52.55 | Trazz | priorityjumping=no |
| 16:53.00 | file[laptop] | I don't eat people :\ |
| 16:53.08 | DeadZen | ill play a longer one |
| 16:53.12 | oceanlan|dustin | k |
| 16:53.16 | DeadZen | a vm-instructions |
| 16:53.18 | [TK]D-Fender | file[laptop] : just think "protein"! |
| 16:53.47 | [TK]D-Fender | DeadZen : make you enten loop the playback, and then do a "sip show channels" in CLI and it will tell you the codec |
| 16:53.49 | DeadZen | restarted |
| 16:53.52 | Trazz | tk - http://pastebin.com/527495 |
| 16:54.07 | file | hrm |
| 16:54.15 | [TK]D-Fender | Trazz : pastebin the new exten.... |
| 16:54.31 | Trazz | its at the bottom of that paste |
| 16:54.49 | oceanlan|dustin | DeadZen: still crappy... |
| 16:55.01 | DeadZen | Says: Form ulaw |
| 16:55.20 | DeadZen | oceanlan|dustin: yah it sounds like a 40 year old lp |
| 16:55.27 | [TK]D-Fender | Trazz : You've skipped my requests... swap 2& 3, and insert the NoOp in 102 |
| 16:55.27 | dfgas | this is alot of crap, lol |
| 16:55.40 | Trazz | ok |
| 16:55.59 | file | gah, Swiss Chalet broke the ordering site |
| 16:56.12 | *** join/#asterisk shepherd (n=shepherd@user-0cev19a.cable.mindspring.com) |
| 16:56.15 | [TK]D-Fender | dfgas : You using AMP? |
| 16:56.16 | *** join/#asterisk ToTo (n=ToTo@host70-163.pool872.interbusiness.it) |
| 16:56.56 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 16:57.17 | dfgas | yah |
| 16:57.30 | Trazz | TK - updated output with new extension info http://pastebin.com/527505 |
| 16:57.34 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 16:57.39 | [TK]D-Fender | dfgas : try in #amportal . |
| 16:58.09 | oceanlan|dustin | DeadZen: i wonder what the hell is causing that..is asterisk the only thing running on this PC? |
| 16:58.22 | DeadZen | <PROTECTED> |
| 16:58.25 | [TK]D-Fender | Trazz : Copy the whole damn NoOp line! GEEZ!!!!!!! I need to see the stupid variable it would print on the screen!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 16:58.27 | DeadZen | dual 3ghz xeon |
| 16:58.32 | *** join/#asterisk eieiyo (n=eieiyo@68-119-68-89.dhcp.mtgm.al.charter.com) |
| 16:59.05 | oceanlan|dustin | man...i have a 333mhz that sounds better than that! |
| 16:59.06 | [TK]D-Fender | How alike do these lines look? |
| 16:59.07 | [TK]D-Fender | xten => 2000,2,NoOp(Dial status was -${DIALSTATUS}-) |
| 16:59.13 | oceanlan|dustin | do you have lame installed? |
| 16:59.14 | [TK]D-Fender | exten => 2000,102,NoOp |
| 16:59.21 | DeadZen | oceanlan|dustin: no |
| 16:59.44 | file | ah... I see |
| 16:59.44 | DeadZen | oceanlan|dustin: your 300mhz sounds better? |
| 16:59.51 | DeadZen | that's a bit depressing |
| 16:59.54 | oceanlan|dustin | yea! |
| 16:59.57 | oceanlan|dustin | i know |
| 16:59.58 | DeadZen | this should sound like a dream then |
| 17:00.15 | DeadZen | would installing lame make a difference? |
| 17:00.22 | DeadZen | or mpg123? |
| 17:00.35 | DeadZen | the 300 mhz using ulaw format? |
| 17:00.54 | oceanlan|dustin | i dont know...doesnt lame handle mp3 to wav decoding or something? |
| 17:00.56 | oceanlan|dustin | yes!! |
| 17:01.23 | oceanlan|dustin | yes, the 300 is using ulaw |
| 17:01.32 | DeadZen | nutty |
| 17:02.01 | oceanlan|dustin | yea, real nutty...i would look into the mpg123 also...maybe that has something to do with it! |
| 17:02.14 | *** join/#asterisk Paulo (n=paulos@200-168-112-132.dsl.telesp.net.br) |
| 17:02.22 | oceanlan|dustin | i am just throwing things out that have to do with audio just to see what sticks =) |
| 17:02.27 | Paulo | Hi |
| 17:02.34 | oceanlan|dustin | Lo |
| 17:03.01 | *** join/#asterisk gopherspidey (n=spidey@12-216-165-134.client.mchsi.com) |
| 17:03.48 | Paulo | Im using txfax, but the line is disconnected right after I pickup the phone. |
| 17:03.49 | Trazz | TK - http://pastebin.com/527514 that is what is on my screen |
| 17:04.11 | RoyK | Paulo: using pstn? |
| 17:04.16 | Paulo | yeps |
| 17:04.41 | RoyK | chan_tincansandstring |
| 17:05.26 | [TK]D-Fender | Trazz : I'm getting real tired of repeating myself. your second NoOp is missing all the GOD DAMNED ESSENTIAL SHIT that is in the first one! |
| 17:05.59 | [TK]D-Fender | look at the 2 stupid friggen NoOp's in yuor exten and notice you cut off all the stuff you need to see on screen!?!?!??! |
| 17:06.11 | RoyK | I repeat myself when under stress |
| 17:06.12 | Trazz | ok |
| 17:06.12 | RoyK | I repeat myself when under stress |
| 17:06.17 | [TK]D-Fender | Trazz : please compare lines 12 & 15 in your pastebin! |
| 17:07.03 | DeadZen | that might affect me if I knew what the heck an Exos Lucius was |
| 17:07.11 | RoyK | northern pike |
| 17:07.16 | RoyK | fine beast |
| 17:07.18 | dfgas | [TK]D-Fender: they are wondering why i am in there asking,lol |
| 17:07.20 | DeadZen | ahh pike |
| 17:07.29 | DeadZen | its funny how the north has different fish then the south |
| 17:07.35 | DeadZen | im in florida |
| 17:07.54 | oceanlan|dustin | haha, im in ohio... =/ |
| 17:07.59 | Trazz | TK - http://pastebin.com/527519 |
| 17:08.12 | DeadZen | cool i got friends in ohio |
| 17:08.26 | [TK]D-Fender | dfgas : Don't expect a lot of help on debugging AMP problems. It buries all the stuff so deep and in external databases in there that its a ^&%$@#ing pain to try and fix and a lot of what we'd need to know can't be pastebiin'd taht easy |
| 17:08.27 | file[laptop] | blah blah blah |
| 17:08.28 | oceanlan|dustin | but it is a really wierd 50 degrees in the middle of winter though...too bad i will miss it on account of asterisk =P |
| 17:08.53 | gopherspidey | Anyone had any experience with the 9133i or the 480i Hard phones fromSayson? |
| 17:08.59 | [TK]D-Fender | Trazz : Fine, there's you answer on how to tell if the phone is UNREACHABLE. |
| 17:09.10 | [TK]D-Fender | CHANUNAVAIL <- |
| 17:09.20 | oceanlan|dustin | gopherspidey: i use the 480i's |
| 17:09.24 | dfgas | [TK]D-Fender: heh well if it helps i pasted all of sip.conf and part of extenstions.conf http://pastebin.com/527520 |
| 17:09.33 | Trazz | thanks |
| 17:09.39 | dfgas | [TK]D-Fender: heck my fwd don't work anymore either |
| 17:09.45 | RoyK | DeadZen: esox, not exos :P |
| 17:09.54 | dfgas | i can call out but not in |
| 17:09.55 | RoyK | DeadZen: you have the esox niger |
| 17:09.59 | gopherspidey | oceanlan|dustin: How is the sound quality? |
| 17:10.00 | dfgas | it just hangs up right away |
| 17:10.12 | DeadZen | gaah i want sound quality! |
| 17:10.20 | *** join/#asterisk jeffik (n=jeffik@CPE0050babf4cd5-CM014350000760.cpe.net.cable.rogers.com) |
| 17:10.24 | [TK]D-Fender | dfgas : Yeah? And where's the [from-sip-external] context in there? |
| 17:10.33 | oceanlan|dustin | Sound quality is nealry as good as polycoms...the speakerphone is awesome |
| 17:10.36 | gopherspidey | oceanlan|dustin, I am looking to purchase a couple of phones for home |
| 17:10.39 | dfgas | i'll get that |
| 17:10.40 | [TK]D-Fender | dfgas : AMP is convoluted crap.... |
| 17:11.01 | gopherspidey | oceanlan|dustin, My Short list was Snom, Polycomm, and Sayson |
| 17:11.41 | [TK]D-Fender | Note to the * community : AMP is shit on a stick. Candy-coated with sprinkles and a friggen halo, but underneath it all, STILL SHIT. |
| 17:11.44 | DeadZen | [TK]D-Fender: you know we need |
| 17:12.05 | gopherspidey | I am trying to get opinions on with to purchase |
| 17:12.08 | DeadZen | [TK]D-Fender: a shell script that uses curl to post to a web app that runs from the /etc/asterisk directory |
| 17:12.16 | [TK]D-Fender | gopherspidey : What do you expect/need from an IP phone? |
| 17:12.17 | DeadZen | to take a config snap shot so you have all you need to check shit |
| 17:12.18 | oceanlan|dustin | gopherspidey: Saysons are very very nice...i have had some issue's with booting from tftp servers, but the web config is great |
| 17:12.50 | [TK]D-Fender | Polycom = best business IP phone for the value. |
| 17:13.00 | DeadZen | i am so dreading my upcoming wisdom teeth surgery |
| 17:13.10 | DeadZen | i gotta yank 4 pieces of my head out through my mouth |
| 17:13.11 | dfgas | [TK]D-Fender: in the sip.conf? |
| 17:13.11 | oceanlan|dustin | gopherspidey: they are very nice units...for the price, they have a better lcd than the 501 polycoms...but they lack some of the neat things that the polycoms do like im and presence |
| 17:13.12 | DeadZen | its gonna be GREAT |
| 17:13.28 | [TK]D-Fender | dfgas : No, in extensions.conf. I have no idea where you call is GOING.... |
| 17:13.35 | oceanlan|dustin | HAHA, wow...yea thats gonna suck! |
| 17:13.40 | gopherspidey | [TK]D-Fender, I would purchase a Polycomm but It does not have a backlight, and one of the Phones are going to go on my night stand. |
| 17:13.48 | DeadZen | hehe |
| 17:14.10 | Paulo | my dialplan has a TXFAX context with: Set(TIMEOUT(digit)=5), Set(TIMEOUT(response)=10), Answer, Wait(1), TxFAX(${FAXFILE}|caller|debug), Hungup |
| 17:14.13 | [TK]D-Fender | oceanlan|dustin : Polycom LCD = pixel based, Sayson = character based. I can do graphics on mine.... |
| 17:14.13 | oceanlan|dustin | gopherspidey: that is a short coming that Polycom has been getting a bunch of flak for! |
| 17:14.27 | DeadZen | why dont they have video phones? |
| 17:14.30 | [TK]D-Fender | gopherspidey : Well if thats the point for you sure. |
| 17:14.31 | DeadZen | sip can do video right? |
| 17:14.54 | Paulo | when I pick up the phone, the call dies with Unicall/X event Far end disconnected |
| 17:14.56 | oceanlan|dustin | [TK]D-Fender: that makes sense, i didnt realize that! i am just used to characters! |
| 17:14.59 | [TK]D-Fender | So far the only phones witha backlight ar the 480i, and the Grandstreams (see my comment about AMP) |
| 17:15.14 | gopherspidey | [TK]D-Fender, I am going to use these phone in a home environment for a Phone, plus an interface for my homeautomation system |
| 17:15.15 | DeadZen | im watching a man part |
| 17:15.22 | [TK]D-Fender | oceanlan|dustin : wel I'mused to a nice varilable font sized graphic display :) |
| 17:15.25 | oceanlan|dustin | a man part? |
| 17:15.55 | eieiyo | anybody familiar with app_rpt? |
| 17:16.05 | DeadZen | yah that movie with vin diesel |
| 17:16.09 | Paulo | somebody else using txfax? |
| 17:16.12 | oceanlan|dustin | [TK]D-Fender: yea, i never even thought about graphics on the LCD, duh....that would make sence! |
| 17:16.24 | [TK]D-Fender | gopherspidey : For home automation you can use any phone if its DTMF based in the dial-plan (like my X-10 setup it), or you'll need an XML capable phone (480i, 601, Cisco) |
| 17:16.25 | oceanlan|dustin | Man Apart! |
| 17:16.39 | oceanlan|dustin | omg...i thought you were watching man parts!! |
| 17:16.50 | DeadZen | no |
| 17:16.50 | dfgas | [TK]D-Fender: http://pastebin.com/527532 |
| 17:16.56 | DeadZen | i only have to look down if i wanna see those |
| 17:17.07 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 17:17.33 | [TK]D-Fender | dfgas : Now go see that this context is where you axvoice setup points to! a friggen dead-end! |
| 17:17.40 | DeadZen | see im still stuck on |
| 17:17.45 | [TK]D-Fender | dfgas : Go check your AMP settings. |
| 17:17.48 | DeadZen | why the hell your 300mhz sounds better then my dual 3ghz |
| 17:17.49 | oceanlan|dustin | gopherspidey: the 480i requires POE FYI |
| 17:17.54 | gopherspidey | [TK]D-Fender, My plan was to to do both XML and a dailpail |
| 17:18.21 | DeadZen | whats poe |
| 17:18.29 | [TK]D-Fender | gopherspidey : By the time you're done adding on the PoE adapter you could by a Polycom IP601 which would be a vastly better phone (minus the backlight) |
| 17:18.31 | gopherspidey | oceanlan|dustin, I realize that |
| 17:18.35 | oceanlan|dustin | haha, yea...its wierd...my p4 2.8ghz on a t1 sounds good too |
| 17:18.44 | [TK]D-Fender | ~poe |
| 17:18.45 | jbot | poe is, like, Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt |
| 17:18.51 | oceanlan|dustin | DeadZen: power over ethernet |
| 17:18.57 | [TK]D-Fender | stupid jbot... |
| 17:18.59 | DeadZen | ahh |
| 17:19.02 | DeadZen | those are cool too |
| 17:19.06 | gopherspidey | lol |
| 17:19.19 | oceanlan|dustin | yea, but for a home install, that can get $$$ |
| 17:19.38 | oceanlan|dustin | the 480i CT has a power pack and a cordless handset... |
| 17:19.51 | DeadZen | oh a man apart |
| 17:19.53 | DeadZen | they kill his chick |
| 17:19.54 | oceanlan|dustin | you can pair up to 4 cordless's to the one base station... |
| 17:19.57 | oceanlan|dustin | neat |
| 17:20.00 | DeadZen | and he kills half of los angeles |
| 17:20.04 | DeadZen | i think thats the plot anyway |
| 17:20.23 | *** join/#asterisk dorphalsig (n=dorphals@200.106.223.5) |
| 17:20.30 | dorphalsig | Hi |
| 17:20.31 | *** join/#asterisk ckruetze (n=ckruetze@i577A4DD1.versanet.de) |
| 17:20.35 | [TK]D-Fender | oceanlan|dustin : how "functional" is the cordless handset? Is it a completely seperate SIP device? Does it have all the standard call control features? What about range? Quality? |
| 17:20.39 | ckruetze | Hi |
| 17:20.46 | gopherspidey | oceanlan|dustin, I am planning to the POE expensive, because my last two Voip phone got fryed by a power surge. :( Cisco 7960 and BudgetTone 101 |
| 17:20.50 | dorphalsig | I'm trying to compile the mysql addon for * 1.0.10 |
| 17:21.10 | ckruetze | Are http://bugs.digium.com/ and http://lists.digium.com down? I can't reach them |
| 17:21.10 | dorphalsig | but I get funny compilation errors |
| 17:21.15 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 17:22.14 | oceanlan|dustin | [TK]D-Fender: Range = Excellent, tested up to 250ft. Quality = Good @ long range. Excellent up close. It is not a separate SIp device, but it can use any of the line appearances that you set up on the phone... |
| 17:22.18 | [TK]D-Fender | oceanlan|dustin : For a few PoE ports its not that expensive... |
| 17:22.30 | dorphalsig | can anybody help me? |
| 17:22.36 | oceanlan|dustin | Call controll is just like a stanard phone..hold, transfer, ect.. |
| 17:22.38 | [TK]D-Fender | oceanlan|dustin : The appearances is a big downer.... |
| 17:22.40 | dorphalsig | cdr_addon_mysql.o(.text+0x1674): In function `usecount': |
| 17:22.41 | dorphalsig | : undefined reference to `pthread_mutex_trylock' |
| 17:22.43 | gopherspidey | POE is not all that bad. http://www.netgear.com/products/details/FS108P.php That runs about 150 to 200 dollars |
| 17:23.14 | JonR800 | you can also pick up 3com 802.3af injectors for like $25 |
| 17:23.23 | DeadZen | whats an injector |
| 17:23.24 | oceanlan|dustin | [TK]D-Fender: POE when you could get a wall wart in the box with the phone is an added expense =/ |
| 17:23.26 | [TK]D-Fender | http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1333276&CatId=866 |
| 17:23.28 | [TK]D-Fender | 118$ |
| 17:23.33 | JonR800 | poe injector |
| 17:23.43 | DeadZen | oh really 25 is pretty cheap |
| 17:23.48 | DeadZen | cheaper then a wireless card |
| 17:24.10 | [TK]D-Fender | JonR800 : Except I believe those FORCE power on, and don't use auto-detect. If you forget that it's "live" and plug something else on it, it will FRY. |
| 17:24.20 | oceanlan|dustin | [TK]D-Fender: the line appearences is not too bad actually...we have 5 registered lines to one base and 2 cordlesses paired...we always have open lines and the phone does them all simultaneously.. |
| 17:25.05 | [TK]D-Fender | oceanlan|dustin : Can you use more than 1 line key for the same appearance on the base, and use the OTHERS for the phones, 1 each? |
| 17:25.20 | [TK]D-Fender | oceanlan|dustin : THAT would eb viable. |
| 17:25.29 | DeadZen | [TK]D-Fender: cool |
| 17:26.02 | JonR800 | [TK]D-Fender: heh.. then i've been playing with fire several times. |
| 17:26.10 | oceanlan|dustin | [TK]D-Fender: yes, you can assign any of the line keys to a registered line and when you pick up the cordless, it picks up the first available line. |
| 17:26.36 | [TK]D-Fender | JonR800 : Yes... fire is a distinct possibility |
| 17:26.38 | oceanlan|dustin | it doesnt jump into your conversation...it gets its own line |
| 17:26.56 | gopherspidey | JonR800, What is the model number of the 3com and where can you get it? |
| 17:27.01 | [TK]D-Fender | oceanlan|dustin : Hmm, I might like to see it in action.... |
| 17:27.03 | JonR800 | lemme dig. |
| 17:27.51 | *** join/#asterisk andyo (n=advorak@adsl-68-79-221-203.dsl.chcgil.ameritech.net) |
| 17:27.54 | andyo | hi |
| 17:28.15 | oceanlan|dustin | [TK]D-Fender: we thought that you would only be able to use either the base or the cordless one at a time, but when we gave more lines (linked to line keys) we found that on the cordless you can actually choose which line to pick up, or just hit "line" and it sould feel out the first available... |
| 17:28.19 | JonR800 | 3com single port POE injector http://www.pricegrabber.com/search_getprod.php/masterid=1985982/search=3cnjpse |
| 17:28.57 | oceanlan|dustin | Those 3com POE's are what we use..they are awesome. |
| 17:29.01 | gopherspidey | Thanks |
| 17:29.08 | DeadZen | oceanlan|dustin: installing lame didn't help |
| 17:29.27 | *** join/#asterisk elephantMan (n=elephant@252.205.103-84.rev.gaoland.net) |
| 17:29.32 | oceanlan|dustin | DeadZen: what about checking your mpg123? |
| 17:29.33 | Trazz | TK - http://pastebin.com/527548 |
| 17:29.39 | JonR800 | we use them as well, but for access points and actually a lone network jack.. hah |
| 17:29.44 | DeadZen | oceanlan|dustin: i found mpg321 that the same? |
| 17:29.49 | DeadZen | hehe |
| 17:30.25 | [TK]D-Fender | Trazz : You are bastardizing a MACRO without the proper formatting.... |
| 17:30.26 | oceanlan|dustin | i have read in the configs something about how some php programmers updated it to mpg321 or something? anyone know anyhing about that? |
| 17:30.28 | *** join/#asterisk roulduke_ (i=ue5agowz@p508D403B.dip0.t-ipconnect.de) |
| 17:31.03 | Trazz | TK, i pulled that from the wiki but it's not working |
| 17:31.13 | Trazz | http://www.voip-info.org/wiki-Asterisk+cmd+goto |
| 17:31.32 | oceanlan|dustin | JonR800: we started using them with AP's and found that they are perfect for the 480i's ..in fact...when you read the manual, the damn picture looks identical (and of course all the 48v 500ma specs are the same). |
| 17:31.43 | [TK]D-Fender | Trazz : Look up STDEXTEN |
| 17:32.20 | [TK]D-Fender | Trazz : Where do you think ${ARG1} and ${ARG2} were getting set? |
| 17:32.40 | RoyK | ~stdexten |
| 17:32.46 | Trazz | to be honest. not sure |
| 17:32.53 | RoyK | ~lart himself |
| 17:32.58 | [TK]D-Fender | Trazz : the WIKI is kind of like alist of SUGGESTINGS. Don't just take random code line-for-line and just expect it to work. |
| 17:33.26 | Trazz | ok |
| 17:33.48 | [TK]D-Fender | You need to lear about variables, applications, and functions from the ground-up. |
| 17:34.05 | gopherspidey | oceanlan|dustin [TK]D-Fender Are they smart? Aka If you plug in a non-POE device does it fry it? |
| 17:34.18 | [TK]D-Fender | gopherspidey : Yes, it will fry things. |
| 17:34.24 | JonR800 | oceanlan|dustin: cool, i may end up getting one for my ip600.. someday :) |
| 17:34.47 | oceanlan|dustin | yes, they are not auto detect! they just push voltage and dont care |
| 17:34.51 | [TK]D-Fender | JonR800 : I'm fine with my brink on mine at home right now. |
| 17:34.52 | RoyK | cypromis: ping |
| 17:34.55 | RoyK | ~seen cypromis |
| 17:35.00 | jbot | cypromis is currently on #asterisk-doc #asterisk, last said: 'ommmmmmmm'. |
| 17:35.01 | DeadZen | oceanlan|dustin: mpg123 doesn't help either |
| 17:35.11 | DeadZen | oceanlan|dustin: try it.. lame, lame-devel and mpg123 are installed |
| 17:35.21 | twilson | Am I going crazy or is README.variables wrong... I seem to ALWAYS have to quote variables when doing comparisons in expressions. |
| 17:35.31 | twilson | examples such as: exten => 1,2,gotoif($[${CALLERID} = 123456]?2|1:3|1) result in a syntax error (pulled directly from README.variables) |
| 17:35.50 | oceanlan|dustin | DeadZen: still shatty |
| 17:35.56 | JonR800 | [TK]D-Fender: same here |
| 17:36.07 | DeadZen | oceanlan|dustin: you sure you didnt tweak anything to make it sound better? |
| 17:36.49 | [TK]D-Fender | twilson : Keep in mind that CALLERID is depricated an may be NULL. a NULL on either side of the comparative operator will cause a syntax error <- |
| 17:36.53 | JonR800 | gopherspidey: they're basically half the price of the "smart" poe adapters. so you just have to judge how forgetful/clumsy you or your users are. :) |
| 17:36.55 | oceanlan|dustin | nope...compiled ** and it was up.. |
| 17:36.58 | [TK]D-Fender | brick* |
| 17:38.42 | oceanlan|dustin | DeadZen: i have heard of certain intel chipsets doing wierd things with audio...but it sounds like your stuff is to new to have that problem.. |
| 17:40.08 | twilson | :[TK]D-Fender: thanks. Yeah, just using it as an example, but even if I set the variable (in this case CALLERID), it complains. |
| 17:40.29 | DeadZen | sigh |
| 17:40.32 | gopherspidey | JonR800, This is want I am looking at I need to purchase aditional Ethernet ports a home. If a purchase a 8 port switch ($50)+ 2 cheap power injectors ($50) for 25 more dollars I ge tthe autosensing in the Netgear FS108p |
| 17:40.36 | robin_sz | OK, so whose idea was it to make the capi info command show the controllers as 1,2,3,4 ?? |
| 17:41.06 | [TK]D-Fender | twilson : Take a GOOD look at both side of the comparison...... |
| 17:41.19 | robin_sz | thus I called them 1,2,3,4 in the capi config ... |
| 17:41.30 | *** join/#asterisk eivindtr (n=wingnut-@062016241059.customer.alfanett.no) |
| 17:41.37 | [TK]D-Fender | gopherspidey : 18$ more :) and get 4 PoE ports, not 2 :) |
| 17:41.39 | robin_sz | thus it crashed everytime under load, it tried to dial out on controller 4 |
| 17:41.50 | twilson | :[TK]D-Fender: exten => test,1,Set(CALLERID=123) exten => test,2,GotoIf($[${CALLERID} = 1234]?3:4) |
| 17:42.31 | [TK]D-Fender | twilson: NoOp the CALLERID, and then kill the whitspace around the "=" |
| 17:42.45 | DeadZen | oceanlan|dustin: can i connect to your server to hear it? |
| 17:43.03 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 17:43.13 | gopherspidey | [TK]D-Fender, That is also true. |
| 17:43.24 | gopherspidey | [TK]D-Fender, good point |
| 17:44.22 | oceanlan|dustin | DeadZen: hang on a sec, i will make another exten. |
| 17:44.33 | DeadZen | cool |
| 17:45.03 | twilson | [TK]D-Fender: Ah, the NoOp showed empty after the set. Switched variable name to ${BOB} and it magically worked. :-] Thanks. |
| 17:45.31 | *** join/#asterisk slan (n=lba@user-12lml5g.cable.mindspring.com) |
| 17:46.08 | slan | In which file are *411 directory entries stored? |
| 17:46.25 | SkramX | eh!? |
| 17:46.47 | DeadZen | hrmm |
| 17:47.38 | slan | Probably somewhere in /etc/asterisk but I need to see the specific file |
| 17:48.36 | andyo | whose fwd can I dial to test on *? :-) |
| 17:48.41 | Aughey | ok, any GXP-2000 users here? |
| 17:48.55 | robin_sz | Aughey: yes and no ... |
| 17:48.55 | slan | Aughey: yes I have several |
| 17:49.06 | Aughey | Do you use paging? |
| 17:49.07 | robin_sz | Aughey: I have one, but it sucks so much I dont use it |
| 17:49.27 | Aughey | I just got one for "testing", and I'm trying to get it to do what I want |
| 17:49.27 | slan | Aughey: Not as yet - my installation is pretty new and paging will come later. |
| 17:49.42 | oceanlan|dustin | DeadZen, i pm'd you |
| 17:49.48 | robin_sz | it sits next to my Zyxel WiFI phone which REALLY sucks |
| 17:50.01 | *** join/#asterisk tuxinator_linux (n=tuxinato@m110e36d0.tmodns.net) |
| 17:50.25 | Aughey | Can I make it not give the busy signal when someone hangs up? |
| 17:50.51 | DeadZen | didnt see it |
| 17:50.53 | oceanlan|dustin | DeadZen: nvm i cant PM...not registered! |
| 17:51.03 | DeadZen | so register dustin ;-) |
| 17:51.08 | andyo | anybody have a fwd number I can test my asterisk setup with? |
| 17:52.41 | ravenpi | Aughey: what's your paging problem? |
| 17:52.59 | oceanlan|dustin | DeadZen: i will i will...im just lazy =) |
| 17:54.23 | Aughey | well, with paging, I can only get it to auto answer if I set it as an option in the configuration. Setting the header Call-Info: answer-after=0 doesn't seem to work. |
| 17:54.56 | Aughey | What I've done is set a separate extension (line) to be the "page extension" and configure that to auto answer |
| 17:55.00 | Aughey | it works, but not what I want |
| 17:55.12 | [TK]D-Fender | twilson : And the old for for the var you were trying was CALLERIDNUM <- |
| 17:55.40 | DeadZen | just paste a somethin temp like me |
| 17:55.51 | [TK]D-Fender | twilson : But either way depricated |
| 17:56.01 | [TK]D-Fender | ok, I'm out for a while, maybe back on later. |
| 17:56.02 | DeadZen | deprecated |
| 17:56.08 | blkremedy | is there any way to call into an asterisk box and enter a number for it to call you back? |
| 17:56.15 | blkremedy | with dial tone |
| 17:56.57 | oceanlan|dustin | DeadZen: did you get that extension? |
| 17:57.05 | DeadZen | no |
| 17:57.14 | oceanlan|dustin | i replied to your PC |
| 17:57.15 | DeadZen | join #port |
| 17:57.16 | oceanlan|dustin | PM* |
| 17:58.36 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 18:01.53 | *** join/#asterisk kippi1 (n=kippi@cpc3-hatf3-6-0-cust42.lutn.cable.ntl.com) |
| 18:01.54 | kippi1 | hey |
| 18:02.30 | kippi1 | what is the stats program that will put all you cvs files into a nice webpage for you? |
| 18:03.25 | gopherspidey | viewcvs |
| 18:04.10 | gopherspidey | or fisheye |
| 18:04.20 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 18:04.34 | gopherspidey | kippil, is that what your are looking for? |
| 18:05.22 | gopherspidey | http://www.cenqua.com/fisheye/?6qm or http://www.viewvc.org/ |
| 18:05.27 | Qwell | MikeJ[Laptop]: y0 |
| 18:08.01 | *** join/#asterisk oceanlan|dstn|di (n=info@cpe-69-133-109-130.woh.res.rr.com) |
| 18:09.20 | DeadZen | hmm |
| 18:11.12 | *** join/#asterisk BSDaemon (i=hbf@CPE00032f0d286f-CM014380004179.cpe.net.cable.rogers.com) |
| 18:11.14 | BSDaemon | Heya |
| 18:12.10 | RoyK | <PROTECTED> |
| 18:14.23 | *** join/#asterisk justme (n=justme@S0106000625828e34.ed.shawcable.net) |
| 18:14.31 | kink0 | anyway to log originating SIP ip address ussing standard Asterisk CDR ? |
| 18:17.10 | justme | <PROTECTED> |
| 18:17.28 | Qwell | justme: go play |
| 18:17.53 | BSDaemon | justme = lame |
| 18:18.30 | kippi1 | is there away I could ring my * box, get it to hang up the call and then call me back? |
| 18:19.47 | *** join/#asterisk BugKham (n=lamer@125.24.1.2) |
| 18:20.33 | tuxinator_linux | justme: why are your getting rid of it/ |
| 18:22.27 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 18:25.22 | festr__ | hello |
| 18:25.38 | festr__ | nick festr_ |
| 18:26.01 | *** part/#asterisk festr_ (n=festr@ns.regnet.cz) |
| 18:26.05 | *** join/#asterisk festr_ (n=festr@ns.regnet.cz) |
| 18:26.09 | Paulo | when I'm dialing out, the call hangs right after answered. Receiving is Ok. |
| 18:26.40 | festr_ | is it possible in queue to announce sound file as soon as call is accepted by agent? |
| 18:26.45 | festr_ | there is only announce for the agent |
| 18:26.48 | festr_ | but not for caller |
| 18:27.03 | festr_ | any trick or patch? |
| 18:27.09 | Qwell | festr_: Something like "Hello, how can I help you?" |
| 18:27.22 | Qwell | That's the agents job :p |
| 18:27.42 | Qwell | I also think MoH/ring stopping will be an obvious sign |
| 18:32.07 | festr_ | Qwell: no, annouce, that calls may be monitored |
| 18:32.11 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 18:32.22 | *** join/#asterisk oceanlan|dustin (n=info@cpe-69-133-109-130.woh.res.rr.com) |
| 18:32.25 | wunderkin | play that before you put them into the queue |
| 18:32.37 | festr_ | i'm using ring method |
| 18:32.50 | kippi1 | How can I ring a number and then hangup and get asterisk to call me back? |
| 18:33.12 | Qwell | festr_: You just need to play an announcement, then dial an agent? |
| 18:33.16 | Qwell | use Playback |
| 18:33.23 | Qwell | before you dial, that is |
| 18:33.24 | *** join/#asterisk fiber0pti (n=John@206-169-194-79.gen.twtelecom.net) |
| 18:33.30 | pifiu | morning everyone |
| 18:33.32 | pifiu | hey qwell |
| 18:33.35 | festr_ | Qwell: dialing agent is queue job |
| 18:33.45 | festr_ | Qwell: no way to insert anything |
| 18:33.55 | Qwell | festr_: Then put it as an announcement in the queue, OR, play it before sending them to the queue |
| 18:34.23 | fiber0pti | I have a question about including contexts for night and day. Do I just use "include => context|<times>|<weekdays>|<mdays>|<months> " at the begining of each context or do I need a gotoif statement? |
| 18:34.35 | *** join/#asterisk __nvrs (i=RUR@Kitchener-HSE-ppp3565498.sympatico.ca) |
| 18:34.48 | festr_ | Qwell: yes this is sollution. but for some specific reason i need to play it which announce does but for caller not to agent |
| 18:35.12 | wunderkin | erm |
| 18:35.27 | wunderkin | exten => blah,1,Playback(you-are-being-monitored) |
| 18:35.33 | wunderkin | exten => blah,2,Queue(blah) |
| 18:35.34 | wunderkin | there! |
| 18:35.46 | robin_sz | sigh, voipgate.com. what a bunch of idiots |
| 18:36.04 | robin_sz | rarely have i seen a DNS more screwed up |
| 18:36.15 | Qwell | festr_: Why would you need to tell your agent that calls might be recorded? |
| 18:36.32 | festr_ | Qwell: omg, not to agent but to the caller :) |
| 18:36.33 | *** join/#asterisk nvrs (i=RUR@HSE-Montreal-ppp3469235.sympatico.ca) |
| 18:36.43 | Qwell | yes, do what wunderkin said |
| 18:38.27 | wunderkin | man why do i always get all of the screwy problems.. |
| 18:38.55 | tuxinator_linux | you're a screw magnet |
| 18:40.03 | festr_ | i need to say this before transfer to agent (ASAP), this is customer's specification. i have to tell him, that thhis is not possible and play it before queue |
| 18:40.06 | wunderkin | WTF |
| 18:40.22 | festr_ | tahts why i need this :) |
| 18:40.42 | wunderkin | festr_, uh whut? |
| 18:41.01 | tuxinator_linux | maybe play the sound after the agent picks ups, then unmute the agent |
| 18:41.35 | festr_ | tuxinator_linux: but how? |
| 18:41.40 | tuxinator_linux | which is stupid, as you loose money having the agent wait |
| 18:41.42 | wunderkin | what is the problem with playing the file before they go into queue? the announcement is telling them they are going to be recorded? thats where it belongs! |
| 18:42.16 | tuxinator_linux | festr_: not sure, new to this stuff still |
| 18:42.18 | festr_ | you are right |
| 18:43.34 | Paulo | Hum... I know now what causes the hungup when faxing from asterisk... |
| 18:43.49 | festr_ | tell us |
| 18:43.50 | festr_ | :) |
| 18:44.08 | robin_sz | oopsy: Contr4: 2 B channels total, 3 B channels free. |
| 18:44.12 | robin_sz | umm ... |
| 18:44.12 | dfgas | anyone use axvoice? |
| 18:44.35 | Paulo | The cause is a mechanism to avoid calls paid by the receiver in Brazil... |
| 18:45.28 | Paulo | the other end has to send a hungup and reconnect in the space of 1000ms |
| 18:45.49 | dfgas | i just need to figure out how to get axvoice to wrok for incoming calls |
| 18:45.56 | dfgas | cause it works outgoing |
| 18:46.35 | Paulo | I think its called "called to charge" |
| 18:46.38 | *** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com) |
| 18:46.58 | Paulo | (my english is worst than tarzan) |
| 18:48.07 | Paulo | In Brazil its called "double pickup" |
| 18:48.40 | Paulo | how can I setup asterisk to ignore a hungup in the first 1000ms ? |
| 18:51.00 | *** join/#asterisk darren (n=darren@cpc3-neat1-3-0-cust237.swan.cable.ntl.com) |
| 18:54.16 | *** part/#asterisk darren (n=darren@cpc3-neat1-3-0-cust237.swan.cable.ntl.com) |
| 18:56.17 | *** join/#asterisk Skkip (n=Skipper@216.160.91.91) |
| 19:02.33 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 19:02.56 | *** join/#asterisk _nvrs (i=RUR@Kitchener-HSE-ppp3564687.sympatico.ca) |
| 19:03.01 | *** part/#asterisk oej (n=oej@apollo.webway.se) |
| 19:04.11 | wunderkin | weird |
| 19:05.04 | wunderkin | this is the 2nd time both of my pris have screwed up.. i have 2 pri through broadwing plus 2 others crossconnected to another machine, the 2 cross connected work.. but the 2 through broadwing keep hanging up immediately and giving me cause code 16 or 28.. last time this happened i stopped asterisk and restarted and it was fine.. wtf |
| 19:07.10 | znoG | arghhhh the random hangups are driving me nuts on this Zaptel card |
| 19:07.20 | znoG | it just hangs up all of a sudden, and busydetect is OFF |
| 19:07.53 | justinu | Cause No. 28 - Incorrect number (invalid number format, address incomplete)/Special intercept announcement |
| 19:08.26 | wunderkin | my last one that i got a debug on gave a 16 for the same number |
| 19:09.42 | *** join/#asterisk fugitivo (n=ajf@201.255.177.63) |
| 19:09.50 | justinu | oh noes!!!!!!!!1111!!!!!! broadwing!!!1!!!! |
| 19:10.07 | wunderkin | yes :( |
| 19:10.10 | wunderkin | both of them lol |
| 19:10.27 | justinu | check pri debug |
| 19:10.36 | justinu | make sure the Called Party Number IE's look good |
| 19:11.00 | wunderkin | but it worked yesterday! |
| 19:11.13 | wunderkin | hehe im going to pb it now :D |
| 19:11.17 | justinu | k |
| 19:11.47 | wunderkin | http://pastebin.ca/38918 |
| 19:12.09 | *** join/#asterisk saftsack (n=oliver@p54A7EBF7.dip.t-dialin.net) |
| 19:12.17 | wunderkin | wow thats a mess |
| 19:12.26 | justinu | is that pri intense debug? |
| 19:12.28 | justinu | or just regular |
| 19:12.47 | wunderkin | intense debug |
| 19:12.56 | justinu | just pri debug will be better for this |
| 19:12.57 | znoG | distinctive ring doesn't work, random hangups, something fishy going on here |
| 19:12.59 | wunderkin | ok |
| 19:13.02 | justinu | we don't care about the layer2 (hdlc) stuff |
| 19:13.15 | wunderkin | znoG, analog? |
| 19:13.30 | znoG | wunderkin: yep, FXO Zaptel card |
| 19:13.44 | wunderkin | there i got a 28 this time |
| 19:13.51 | justinu | cool, paste it up |
| 19:14.26 | Paulo | how can I setup asterisk to ignore a "Drop Call" event in the first 1000ms ??? |
| 19:15.47 | fiber0pti | my dial tree seams to be ignoring my digittimeout's and responsetimeout's.. any ideas? |
| 19:15.56 | Qwell | weird, features.conf has automon as *1, but when I hit *1, it tried to transfer |
| 19:16.51 | wunderkin | justinu, its missing a newline on a lot of things :/ |
| 19:17.13 | justinu | ok, lets see it anyways |
| 19:17.48 | wunderkin | http://pastebin.com/527690 |
| 19:18.50 | wunderkin | znoG, show your zapata.conf |
| 19:19.34 | justinu | wunderkin: can you try making the call without the 1 in front of the number? |
| 19:19.48 | wunderkin | can try, have always done it this way |
| 19:20.31 | wunderkin | got a 16 |
| 19:20.43 | *** join/#asterisk comfrey (n=comfrey@67.188.34.12) |
| 19:20.44 | justinu | odd |
| 19:20.55 | justinu | i'd start bitching at broadwing at this point |
| 19:21.52 | justinu | the only thing that looks odd to me is line 33/34 |
| 19:21.53 | wunderkin | this happened before and i just restart asterisk and it was fine |
| 19:22.00 | justinu | not sure what that's all about |
| 19:22.16 | DeadZen | how do you create a mailbox |
| 19:22.18 | justinu | well, if you want to restart ast, and post another pri debug of a succesful call, we can look for any differences |
| 19:22.24 | wunderkin | ok |
| 19:22.40 | fiber0pti | Anyone have problems with the dial plan ignoring the digittimeouts and responsetimeouts? |
| 19:23.34 | anonymouz666 | With Asterisk and E1 card, A customer call to my company... Asterisk pick up the call and do a dial to technical support guy. After they start to talk, the TI manager join the conversation (like chanspy) and start to talk - he can hear both sides, but when he speaks only the technical guy can hear. Is that possible? |
| 19:23.51 | *** join/#asterisk toma (i=toma@ip83.kovoks.nl) |
| 19:24.11 | Paulo | anonymouz666, there is a plead to implement this feature. |
| 19:24.41 | Paulo | digium is asking U$ 7K to do this |
| 19:24.50 | wunderkin | maybe it was the number |
| 19:24.52 | saftsack | can i receive and send faxes with the same modem in the same time? (hylafax) |
| 19:24.54 | anonymouz666 | cambada de ladroes :) |
| 19:24.58 | wunderkin | it does it still to the same number but now another one works |
| 19:25.09 | wunderkin | but i know a good number that i tested on and i had the problem with that one too.. hmm |
| 19:25.22 | anonymouz666 | Paulo, where did you see that? |
| 19:25.33 | Paulo | anonymouz666, os caras estão fazendo uma vaquina na lista de usuários. |
| 19:25.56 | DeadZen | does one need a zaptel timer |
| 19:26.00 | wunderkin | i got a 28 to that number, but i used a 1 |
| 19:26.08 | wunderkin | but i use a 1 for everything else and its fine |
| 19:26.14 | wunderkin | its a LD PRI |
| 19:26.39 | justinu | it's the number plan and number type that determine whether you need to use a 1, or what |
| 19:26.45 | justinu | i can't remember all the different combinations |
| 19:26.46 | DeadZen | how do you make /dev/zap/ctl ? |
| 19:27.00 | wunderkin | WTF now i dont use a 1 and it works but it didnt last time |
| 19:27.53 | *** part/#asterisk DeadZen (n=DeadZen@adsl-153-136-41.mia.bellsouth.net) |
| 19:28.19 | anonymouz666 | Paulo, I Think mark spencer can do it quickly |
| 19:28.21 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 19:28.27 | wunderkin | i dont have a pridialplan or prilocaldialplan specified.. |
| 19:28.38 | anonymouz666 | this is very interesting feature |
| 19:28.50 | Paulo | anonymouz666, vc é brasileiro? |
| 19:28.52 | znoG | wunderkin: http://pastebin.com/527705 |
| 19:28.57 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 19:28.59 | anonymouz666 | paulo: yeah |
| 19:29.02 | justinu | wunderkin: if you call the same number repeatedly, do you get different results? |
| 19:29.05 | znoG | wunderkin: would really appreciate it if you could take a look and see if there's anything wrong with my config? |
| 19:29.28 | Paulo | anonymouz666, estou tendo problemas com duplo atendimento |
| 19:29.30 | znoG | wunderkin: i'm pretty sure the config is OK though, I read' all the comments explaining each option and ended up with that |
| 19:29.36 | wunderkin | justinu, last time yes.. i got a 16 most of the time.. sometimes a 28 |
| 19:29.51 | justinu | that's a network issue then, talk to broadwing... |
| 19:30.00 | Paulo | anonymouz666, sabe como posso fazer para ignorar um "Drop Call" nos primeiros 1000ms de uma chamada? |
| 19:30.19 | wunderkin | justinu, sounds like i will need to send them the debugs then.. |
| 19:30.29 | justinu | lol, as if they know how to read that |
| 19:30.33 | wunderkin | lol |
| 19:30.39 | wunderkin | shit |
| 19:30.49 | justinu | if you bitch hard enough, you might get a tech that understands q931 |
| 19:30.54 | wunderkin | LOL |
| 19:31.01 | wunderkin | well maybe ill only call locally using voip right now |
| 19:31.29 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 19:31.40 | anonymouz666 | Paulo, PVT |
| 19:31.50 | jpablo | hey people, I have no money but i want to connect my asterisk to my gsm network, any suggestions ? |
| 19:32.03 | wunderkin | the colo changed the 25 pair cable that both of my pris are on (nothing else is on it) .. now i will have to wait 2 months to see if that helped that other problem.. :/ |
| 19:32.11 | tuxinator_linux | jpablo: steal? |
| 19:32.36 | wunderkin | znoG, i really dont know anything about analog, i can only go off of what i have seen here.. ill try to look at it shortly if no one else can |
| 19:32.41 | jpablo | tuxinator_linux, hehe, i want some solution using an cellphone or something i don't to buy (yet) a gateway, it is just for a demo |
| 19:33.07 | tuxinator_linux | well, when you figure it out, let me know |
| 19:33.32 | jpablo | jeje, ok. |
| 19:33.33 | justinu | wunderkin: why 2 months? |
| 19:33.47 | wunderkin | its an intermittant problem |
| 19:33.50 | justinu | ah |
| 19:33.52 | jpablo | in http://www.voip-info.org/wiki/view/Asterisk+Connecting+to+the+Cellular+Network i see there's some suggetions to use some homebrew cables. |
| 19:34.06 | jpablo | but the lists aren't working right now, so i can't follow thoses links :( |
| 19:35.00 | wunderkin | znoG, looks like all of the trouble makers are turned off.. so i dunno |
| 19:38.32 | znoG | wunderkin: no prob, thanks anyway |
| 19:38.51 | wunderkin | the formatting for the pri debug needs to be fixed too :) i think tzanger sent in a patch a little while ago regarding the formatting.. |
| 19:38.52 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 19:39.09 | justinu | heh |
| 19:39.16 | justinu | glad to see that made it into the code :P |
| 19:41.18 | wunderkin | i bet there is an easy way to strip x number of characters per line in a file :/ |
| 19:41.53 | wunderkin | cool thanks |
| 19:42.48 | Paulo | anonymouz666, are you receiving my PVT messages? |
| 19:44.11 | Paulo | anonymouz666, do you know the brazilian "duplo atendimento" trick to avoid "called to pay" calls ("chamadas a cobrar") |
| 19:45.51 | *** join/#asterisk ToTo (n=ToTo@host70-163.pool872.interbusiness.it) |
| 19:46.36 | *** join/#asterisk Jonathan_ (n=chatzill@66-168-63-104.dhcp.mdsn.wi.charter.com) |
| 19:47.44 | Jonathan_ | After upgrading to 1.2.3, the Asterisk CLI does not accept user input any longer. asterisk -r or asterisk -cr causes asterisk to boot fine but you can't type anything. Any ideas what's wrong? |
| 19:48.27 | *** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net) |
| 19:49.24 | wunderkin | it looks like asterisk defaults to national if you dont specify a dialplan.. i know people normally suggest to use unknown as a default.. i remember there being a discussion to change the default.. |
| 19:49.47 | wunderkin | but if it works.. at least most of the time.. then i dont see where that would be a problem with me.. |
| 19:49.49 | justinu | yeah, you could switch to unknown |
| 19:49.58 | wunderkin | might as well |
| 19:50.02 | justinu | that's why I'm suspecting a network issue |
| 19:50.06 | wunderkin | yeah |
| 19:50.18 | justinu | when things happen randomly, it's usually not a configuration issue |
| 19:50.25 | anonymouz666 | Paulo, voce quer bloquear chamadas a cobrar? |
| 19:50.25 | wunderkin | unless asterisk wasnt sending something right |
| 19:50.28 | fiber0pti | my dial tree seams to be ignoring my digittimeout's and responsetimeout's.. any ideas? |
| 19:50.39 | anonymouz666 | Paulo, nao recebi nenhuma mensagem tua por PVT |
| 19:50.42 | justinu | wunderkin: in that case, it would have to be some kinda bug in the pri stack |
| 19:50.48 | justinu | which many people seem to use without problems.... |
| 19:50.56 | wunderkin | yeah.. i know ;) |
| 19:51.23 | Paulo | anonymouz666, I want to call lines that use "duplo atendimento" |
| 19:51.39 | justinu | duplo! |
| 19:51.43 | justinu | my favorite kind of legos |
| 19:51.52 | anonymouz666 | Paulo, não entendi a pergunta cara. |
| 19:51.58 | mzo_ | yay portuguese! :P |
| 19:52.07 | wunderkin | i should set pridialplan and prilocaldialplan? |
| 19:52.08 | anonymouz666 | Paulo, ou talvez eu não saiba a resposta. |
| 19:52.20 | justinu | wunderkin: not sure what the difference is |
| 19:52.21 | mzo_ | asterisk is multi-language aware for config files? |
| 19:52.22 | Paulo | anonymouz666, seguinte, estou fazendo chamadas para um pabx que usa esse recurso para evitar ligações a cobrar. |
| 19:52.25 | wunderkin | not sure either |
| 19:52.31 | blitzrage | mzo_: no |
| 19:52.39 | blitzrage | stick with ASCII |
| 19:52.41 | mzo_ | oh, bummer, that would be cool to have the conf files in cyrillic :P |
| 19:52.53 | blitzrage | there is talk of making it UTF-8 compliant... but not yet |
| 19:52.57 | Paulo | anonymouz666, o que acontece é que quando o asterisk recebe o "Drop Call", ele corta a ligação. |
| 19:52.58 | mzo_ | yay |
| 19:53.01 | wunderkin | ; PRI Dialplan: Only RARELY used for PRI. ; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's numbering plan) |
| 19:53.05 | mzo_ | i have to set up more peer stuff. |
| 19:53.14 | mzo_ | and do more crazy insane asterisk hacking crap to make it to do more pointless crap :P |
| 19:53.29 | mzo_ | like why would one person need 8 sip headsets all over the house :P |
| 19:54.12 | Paulo | anonymouz666, tb estou interessado em bloquear chamadas a cobrar. |
| 19:54.29 | *** join/#asterisk jr_ewing (n=jeanmaro@d213-103-252-138.cust.tele2.fr) |
| 19:54.43 | mzo_ | who shot jr! :P |
| 19:54.47 | anonymouz666 | Paulo, FXO ou E1? |
| 19:54.52 | Paulo | anonymouz666, /join #asteriskbrasil.org |
| 19:55.02 | jr_ewing | hi there |
| 19:55.48 | jr_ewing | my first connection on irc but i seem to learn quickly, it appears of a nice chat ! |
| 19:55.59 | tuxinator_linux | I think I will go buy some floppy disks |
| 19:56.12 | justinu | jr_ewing: where you from? |
| 19:56.18 | jr_ewing | France |
| 19:56.19 | mzo_ | say dallas, texas. ;) |
| 19:56.22 | *** part/#asterisk andyo (n=advorak@adsl-68-79-221-203.dsl.chcgil.ameritech.net) |
| 19:56.24 | jr_ewing | Southwork |
| 19:56.27 | justinu | jr_ewing: bienvenue :) |
| 19:56.33 | jr_ewing | :=) |
| 19:56.37 | jr_ewing | thanks |
| 19:57.12 | jr_ewing | On digium site they explain i need to register ? someone could explain to me ? |
| 19:57.24 | jr_ewing | The Asterisk channel now requires that your nick be registered with the Freenode Nickerv in order to participate. This measure has been taken to combat spambots and the like. We apologize for the inconvenience. Please "/msg NickServ help register" in your IRC client to learn how to register your nick. |
| 19:57.30 | justinu | oh |
| 19:57.30 | Qwell | jr_ewing: /msg nickserv help |
| 19:57.41 | justinu | that kind of register, i thought it was sip register |
| 19:57.50 | mzo_ | haha, how 2 be unclear |
| 19:58.29 | jr_ewing | why to register ? i 'am currently tchatting with you ...! |
| 19:59.02 | mzo_ | so no one takes your nick away mostly |
| 19:59.23 | justinu | and so you can send privmsg's |
| 19:59.55 | jr_ewing | oki ! so this is very important...thanks to all, I'll be back in one hours, childs and wife don't like geek attitude.... |
| 20:00.05 | mzo_ | and so no one impersonates killing jr_ewing. ;) We arleady know Who Shot JR. :P |
| 20:00.25 | justinu | jr_ewing: lol |
| 20:00.33 | jr_ewing | ;=) |
| 20:03.54 | *** join/#asterisk ibob63 (n=hp@82.111.125.213) |
| 20:05.16 | [av]bani | \o/ |
| 20:10.17 | *** join/#asterisk darby_t (n=tom@abbs182.neoplus.adsl.tpnet.pl) |
| 20:16.06 | *** join/#asterisk fndude (i=sobeit@127-48.124-70.tampabay.res.rr.com) |
| 20:18.07 | Paulo | How can I configure asterisk to hold the call until the caller hungs up? |
| 20:18.21 | justinu | i don't think it's possible |
| 20:18.50 | Paulo | in Brazil the caller controls the call... |
| 20:19.01 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 20:19.11 | justinu | paulo: what do you mean? |
| 20:19.13 | *** join/#asterisk brimston1 (n=brimston@68.62.180.100) |
| 20:19.46 | Paulo | I wnat to ignore the "Drop Call" event from the called part. |
| 20:20.01 | Paulo | Is that possible? |
| 20:20.23 | justinu | perhaps if you could elaborate more on what you're trying to accomplish..... |
| 20:21.18 | Paulo | Here in Brazil, the call is not finished until the caller hungs up. |
| 20:21.59 | [av]bani | Paulo wants to hold customer lines hostage :D |
| 20:22.56 | Paulo | Yepz, this is how it works in Brazil. |
| 20:23.17 | Paulo | the caller will hold the line until hungup. |
| 20:24.54 | Jonathan_ | After upgrading to 1.2.3, I am no longer able to get asterisk to give me a CLI. During startup, it says asterisk is ready and gives the CLI> but won't accept input. If it is already running and you -r to it, it won't give you the CLI and hangs at the verbosity level line. If I CTRL + C it, it will shutdown normally. What's wrong? |
| 20:28.22 | Err | heh, the call holding behavior used to happen here in the US, although most phone switches don't do that anymore |
| 20:29.21 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 20:29.23 | kink0 | re |
| 20:29.26 | *** join/#asterisk bjerregaard (n=heste@0x57317642.hrnxx15.adsl-dhcp.tele.dk) |
| 20:29.56 | kink0 | off-topic : anybody is ussing 2N gateways ? I have a doubt about LCR and routes |
| 20:32.00 | mzo_ | heh, when i upgraded to 1.2.3 i broke the gui over apache stuff bad. Asterisk works fine but all the stats stuff is horribly broken! :) |
| 20:32.13 | bjerregaard | Has anybody experimented with YAC, YAACID or a similar program? I have managed to get YAC to work perfectly by sending the callerid-info via nc, but I am missing the ability to automatically spawn a browser on the client-side... |
| 20:33.26 | bjerregaard | YAACID connects to the asterisk manager-service, but though I'm getting lots of input, the program does not popup either a balloon or a browser |
| 20:37.03 | SibrPhrek | anyone know a way to export the Master.csv into a database program - maybe like filemaker? |
| 20:37.09 | *** join/#asterisk angom_h (n=angom@red-corp-200.38.15.43.telnor.net) |
| 20:38.29 | Err | csv values can be imported just about anywhere - it's a relatively standard file format |
| 20:41.34 | justinu | err: call holding behavior? |
| 20:41.52 | SibrPhrek | Err - yeah but i need it to be constantly importing into FMpro without creating duplicates |
| 20:42.56 | Err | justinu: what? |
| 20:43.12 | Err | SibrPhrek: oh, that's a totally different question than what you asked |
| 20:43.38 | SibrPhrek | Err - that's why i need to export it first |
| 20:43.41 | justinu | (12:28:19) Err: heh, the call holding behavior used to happen here in the US, although most phone switches don't do that anymore |
| 20:44.11 | Err | justinu: yes, that was in reponse to Paulo's comments about phone switch behavior in Brazil |
| 20:44.22 | justinu | yeah, i was wondering what that meant |
| 20:44.33 | Err | (where the call isn't terminated until the caller hangs up, regardless of the status of the callee's phone line) |
| 20:44.39 | justinu | oh |
| 20:44.42 | Paulo | Err, how can I emulate this in asterisk |
| 20:44.46 | Paulo | ? |
| 20:44.51 | justinu | i thought there was a fairly long timeout, like 10 seconds |
| 20:44.51 | Err | Paulo: I have no idea |
| 20:45.10 | Err | Paulo: it's unclear to me why you would *want* to emulate that, to be honest ;-) |
| 20:45.44 | Err | the only advantage I could *ever* see to that behavior is the ability to move between phones by simply hanging up the first, then picking up the second - but the security implications of someone being able to tie up your phone indefinitely far outweigh that "convenience" |
| 20:46.17 | justinu | so you're saying it used to be possible to tie up a person's line by calling them and never hanging up? here in the US? |
| 20:46.29 | Err | yes |
| 20:46.37 | justinu | didn't know that |
| 20:46.49 | wunderkin | DoS phone attack! |
| 20:47.14 | Err | when I was younger, my parents had an independent phone company (it had *never* been part of Ma Bell, even when the monopoly existed), and until about 2000 it still had this behavior |
| 20:47.24 | justinu | what company? |
| 20:47.27 | Err | then they finally upgraded to a switch that didn't use relays |
| 20:47.40 | Err | Germantown Independent TC |
| 20:47.45 | justinu | heh, cool |
| 20:48.30 | Err | yeah, it was a pretty interesting setup - I toured the plant once; they had rotary decoders for pulse dialing, still |
| 20:48.37 | Paulo | Err, I need to emulate this berravior, so other tricks used in Brazil will work. |
| 20:49.08 | Err | Paulo: you might have to write a different Dial |
| 20:49.11 | justinu | what tricks? |
| 20:49.12 | Err | () application |
| 20:49.45 | justinu | err: step by step switch |
| 20:51.08 | Err | justinu: yeah, they're single-step-per-pulse rotary switches that reset when the connection drops; couple that with some timers that switch the line to the next rotary decoder after some timeout, and you can decode phone dialing entirely manually :-) |
| 20:51.35 | Paulo | justinu, for example, to avoid "called to pay" calls, some lines would send a drop call event and pickup the line again. |
| 20:52.17 | Paulo | justinu, this works in Brazil because of the call holding behavior |
| 20:52.19 | *** join/#asterisk Math` (n=Math_@modemcable148.4-81-70.mc.videotron.ca) |
| 20:52.50 | justinu | doesn't sound legal to me :P |
| 20:54.20 | *** part/#asterisk toma (i=toma@ip83.kovoks.nl) |
| 20:54.35 | Paulo | justinu, "called to pay" calls are automated in Brazil, you have 5 seconds to hungup without charge (so the caller can identify himself). |
| 20:54.36 | justinu | http://en.wikipedia.org/wiki/Almon_Strowger |
| 20:54.54 | justinu | oh, you mean collect call |
| 20:54.56 | mzo_ | that's just so tempting to abuse ;) |
| 20:55.00 | justinu | lol |
| 20:55.15 | mzo_ | that or speak in 3 second coded phrases and keep redialing ;) |
| 20:55.49 | justinu | He is commonly identified as a Kansas City undertaker, (or occasionally as either a funeral parlor director or a mortician), who invented the automatic telephone exchange and has been described as the father of the automatic telephone exchange. Strowger himself would more likely have characterised his invention as the "girl-less, cuss-less" telephone system. |
| 20:56.41 | *** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 20:56.54 | Paulo | justinu, I really dont know the right english expression for when the called part is charged for the call |
| 20:57.02 | justinu | "collect" :) |
| 20:57.09 | Paulo | oh... |
| 20:57.27 | Err | mzo_: the speaking in coded phrases is a common use of collect calls in the US |
| 20:57.41 | Err | it's entirely automated now, so you can do this without the operator noticing :-) |
| 20:58.07 | Paulo | mzo_, the time limit used to be 10 seconds, and yes, we abused a lot, so they shortened it... |
| 20:58.13 | justinu | lol |
| 20:58.29 | Err | Paulo: are you using traditional phone lines coming into your switch, or are you using a digital feed? |
| 20:58.43 | Paulo | Err, E1 |
| 20:58.55 | Err | hm, that's too bad - if you used a regular phone line you could just flash it :-) |
| 20:59.30 | justinu | E1 MFCR2? |
| 20:59.32 | justinu | or PRI? |
| 20:59.46 | Paulo | justinu, mfcr2 |
| 21:00.24 | justinu | neato |
| 21:00.36 | justinu | using chan_unicall? |
| 21:01.05 | Paulo | justinu, yeps. |
| 21:01.09 | justinu | cool |
| 21:01.22 | Paulo | justinu, its a FAX number rent service. |
| 21:01.29 | justinu | ic |
| 21:01.49 | Paulo | long distance calls are very expensive in Brazil. |
| 21:02.14 | *** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net) |
| 21:02.46 | Paulo | justinu, we offer local fax numbers in some key cities, and the faxes are converted to PDF and sent by e-mail. |
| 21:02.55 | justinu | so spandsp also |
| 21:03.08 | *** join/#asterisk ddapue (i=dtolj@CPE00e0188b8c47-CM000f212fe644.cpe.net.cable.rogers.com) |
| 21:03.11 | Paulo | justinu, right on the mark. :-) |
| 21:03.23 | justinu | cool, coppice's stuff is excellent |
| 21:04.02 | Paulo | justinu, I even run the faxes trhough an OCR... |
| 21:04.21 | justinu | sounds like a nice setup |
| 21:04.39 | Paulo | justinu, its working very nice to receive FAXes |
| 21:04.49 | justinu | can you send as well? |
| 21:05.02 | *** join/#asterisk WillSip (i=WillSip@200.119.223.246) |
| 21:05.20 | justinu | have you done any work with t.38 yet? |
| 21:05.38 | Paulo | justinu, well, I'm running into trouble with lines that use de "double pickup" trick to avoid collect calls. |
| 21:05.46 | justinu | oic |
| 21:06.03 | WillSip | hi |
| 21:06.12 | WillSip | asterisk in spanish |
| 21:06.21 | WillSip | whats channels please |
| 21:06.30 | *** part/#asterisk ddapue (i=dtolj@CPE00e0188b8c47-CM000f212fe644.cpe.net.cable.rogers.com) |
| 21:06.57 | Paulo | justinu, no, we just set up an ipp printer, so the customer can fax from any application... |
| 21:07.15 | jr_ewing | hi |
| 21:07.35 | justinu | paulo: you might consider deploying t.38 gateways at customer site |
| 21:07.43 | justinu | then he won't have to pay telco for a fax line ;) |
| 21:08.29 | Paulo | Brazilian legislation are very restrictive on what kind of services one can offer without having to buy an expensive license from government telecom agency. |
| 21:08.38 | jr_ewing | Is there someone who can help with Nufone H323 channel ? there's a field i only can see in debug mode but can't see with NoOp |
| 21:08.52 | justinu | nufone does h323? |
| 21:09.10 | Qwell | justinu: I think he means the h323 channel Jeremy did |
| 21:09.12 | WillSip | alguien sabe algun canal de asterisk en español |
| 21:09.14 | justinu | ahh |
| 21:09.33 | jr_ewing | yes this jeremy Man namara did it |
| 21:10.11 | Paulo | WillSip, Hay #asteriskbrasil.org, ellos hablen portugues pero te entienden |
| 21:10.23 | jr_ewing | Up ? |
| 21:10.42 | WillSip | gracias PAulo |
| 21:11.09 | jr_ewing | My asterisk send calls to an Avaya (don't laugh ...i have a job...;=) |
| 21:11.13 | jr_ewing | through H323 |
| 21:11.13 | Math` | lol |
| 21:11.30 | jr_ewing | to a Skill (Hunt group) |
| 21:11.44 | jr_ewing | When an agent hang answer the call |
| 21:12.04 | jr_ewing | i cannot see who in cli console |
| 21:12.25 | jr_ewing | but if i monitor console in h.323 debug mode |
| 21:13.10 | jr_ewing | i can see something like 'Connection Established with < |
| 21:13.21 | jr_ewing | Laurent fournier> |
| 21:13.27 | Math` | you'll find more details in h323.log |
| 21:13.52 | jr_ewing | Laurent fournier is the name of the owner avaya phone |
| 21:14.27 | jr_ewing | So as you can understand : i want to know who answer behind the skill group |
| 21:14.32 | *** join/#asterisk dr0ck (n=dr0ck@gateway.digium.com) |
| 21:14.40 | jr_ewing | and use it in noop variable |
| 21:14.43 | *** join/#asterisk angler (n=angler@gateway.digium.com) |
| 21:14.43 | *** join/#asterisk kshumard (n=kshumard@gateway.digium.com) |
| 21:14.45 | *** join/#asterisk brookshire (n=nubb@gateway.digium.com) |
| 21:14.47 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 21:15.10 | jr_ewing | with the aim to use this variable to send a ¨Pop up to tha agent wo answer the call |
| 21:15.17 | justinu | digium's interweb comes back online |
| 21:15.25 | dpryo | jr_ewing: Btw, are you using any 4620-phones with asterisk? (sip) |
| 21:16.10 | jr_ewing | Yes, i successfully installed 4620 with asterisk , but for my current pI 'am working for Call center and i have opprotunity to use Asterisk as a Predictive dialer |
| 21:16.45 | jr_ewing | so it means : |
| 21:17.00 | jr_ewing | i 'am using 6416d + connected to Avaya S8700 |
| 21:17.06 | Math` | justinu: digium's interweb? |
| 21:17.12 | jr_ewing | Asterisk is connected trhough h323 |
| 21:17.23 | jr_ewing | what do you mean by digium interweb ? |
| 21:17.30 | justinu | looks like all the digium folks dropped off for a bit |
| 21:17.39 | jr_ewing | sorry |
| 21:17.43 | Math` | oh ok |
| 21:17.48 | dpryo | jr_ewing: Ok, I'm in the process of throwing my S8700 out the window, and replace it with asterisk.. The only problem is that my 4620s sometimes loses their registration status with sip |
| 21:17.52 | dpryo | jr_ewing: So they need a reboot |
| 21:18.17 | JunK-Y | yay ipod battery is dead, and apparently my 4 years warranty does not cover the battery, WTF! |
| 21:18.26 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
| 21:18.28 | jr_ewing | qualify should help you |
| 21:18.33 | justinu | JunK-Y: a new battery is like $30. |
| 21:18.36 | Err | of course the warranty doesn't cover the battery - you can't expect any rechargeable battery to last for 4 years |
| 21:18.41 | dpryo | JunK-Y: Solution: Step on it till it's broken, then get a new one! |
| 21:18.53 | WillSip | hi asterisk work with kernel Red HAt enterprise 2.4.21-4 |
| 21:19.04 | JunK-Y | justinu: the girl from apple said they will replace it for 300$! |
| 21:19.12 | JunK-Y | when a new one is that price. |
| 21:19.13 | dpryo | haha |
| 21:19.13 | justinu | JunK-Y: http://eshop.macsales.com/Catalog_Page.cfm?Parent=1225&Title=iPod%20Batteries&Template=1 |
| 21:19.39 | jr_ewing | Dpryo : insert qualify=yes into sip user configuration |
| 21:19.54 | dpryo | jr_ewing: Ah, will try that. Thanks |
| 21:19.56 | jr_ewing | what is your dhcp lease time for 4620 Sw |
| 21:20.04 | jr_ewing | ? |
| 21:20.30 | *** join/#asterisk rpm (n=russell@S010600111155e117.cg.shawcable.net) |
| 21:20.52 | jr_ewing | so is there someone to help me with my h323 problem |
| 21:20.57 | jr_ewing | ? |
| 21:21.12 | dpryo | jr_ewing: default lease is 600 and max 7200 |
| 21:21.50 | *** join/#asterisk Math[laptop] (n=Math_@modemcable148.4-81-70.mc.videotron.ca) |
| 21:21.56 | jr_ewing | setup it to infinite, each time dhcp lease exprired 4620 needs to reboot.... |
| 21:22.07 | dpryo | oh |
| 21:22.23 | jr_ewing | i had the problem on 300 hundred ip phone..... |
| 21:22.37 | jr_ewing | connected to s8700 |
| 21:22.39 | dpryo | Strange that it doesn't affect anything when it's connected to S8700 |
| 21:23.09 | jr_ewing | Sure ? |
| 21:23.16 | dpryo | Yeah, only problem with the sip-phones |
| 21:23.17 | justinu | jr_erwing: not sure how many people here are familiar with h323 besides JerJer |
| 21:23.54 | jr_ewing | yep but how i can reach jerjer... i think he shoul be very busy... |
| 21:24.02 | justinu | you might have to pay him, dunno |
| 21:24.24 | jr_ewing | i think it's not a h323 problem |
| 21:24.39 | justinu | maybe you could poke around the source yourself and figure it out |
| 21:24.41 | jr_ewing | just to know wich variable is used in debug mode to |
| 21:24.44 | justinu | or pay someone else to do it |
| 21:25.06 | jr_ewing | screen name of connected party phone |
| 21:25.23 | jr_ewing | I 'am not a develloper .... |
| 21:25.38 | jr_ewing | and i'am poor |
| 21:25.40 | jr_ewing | so poor |
| 21:25.40 | justinu | then you will probably have to pay |
| 21:25.48 | jr_ewing | ;=) |
| 21:25.54 | justinu | for someone's skill |
| 21:26.17 | *** part/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net) |
| 21:26.18 | Err | you have 300 IP phones and an existing phone switch, but you can't afford to pay someone some small amount to upgrade to a new switch? |
| 21:26.30 | jr_ewing | Why not to pay, but i have a collegue with strong knowledes |
| 21:26.46 | jr_ewing | sorry, My company have ... |
| 21:26.54 | jr_ewing | 3 s8700 with 900 ip phones |
| 21:27.13 | *** join/#asterisk wasim_ (n=wasim@pdpc/supporter/active/wasim) |
| 21:27.46 | jr_ewing | if i want to replace 3 brand new switch (18 month we bought it) |
| 21:28.00 | jr_ewing | i need to give serious argument and a DEMO... |
| 21:28.27 | Err | sounds like your plan doesn't really make sense :-) |
| 21:28.32 | jr_ewing | yep |
| 21:28.40 | jr_ewing | this is not my plan |
| 21:28.44 | Err | if there's nothing wrong with your existing system, why do you want to upgrade? |
| 21:29.04 | jr_ewing | i just want to introduce asterisk as an adjunct of existing assetrs |
| 21:29.32 | newmember | what is a s8700? |
| 21:29.38 | robin_sz | I always ask this .... "what problem does it solve?" |
| 21:29.44 | jr_ewing | Like voicemail, ivr, cti, fax server and Survivor processor |
| 21:29.51 | Err | google knows what it is :-) |
| 21:29.56 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 21:30.15 | jr_ewing | Avaya pbx |
| 21:30.30 | robin_sz | and it doesnt have voicemail?# |
| 21:30.49 | jr_ewing | not include with the hudge amount of money |
| 21:30.53 | *** join/#asterisk malcolmd (n=malcolmd@gateway.digium.com) |
| 21:30.55 | jr_ewing | it's an adjunt |
| 21:30.57 | *** join/#asterisk kram (n=mark@gateway.digium.com) |
| 21:30.59 | jr_ewing | under Red hat 9 |
| 21:31.03 | jr_ewing | under Red hat 9=) |
| 21:31.04 | Qwell | welcome back Digium |
| 21:31.21 | jr_ewing | S8700 work under Redhat 9 to.... |
| 21:31.44 | jr_ewing | such a shame for bigges pbx company to use open source OS |
| 21:32.01 | Err | why, exactly, is that a shame? |
| 21:32.24 | rt | well, i've got my fwd number ringing my sip phone through my asterisk/iax gateway. |
| 21:32.30 | rt | baby steps, baby steps, |
| 21:33.05 | robin_sz | my next baby step is to try and hook our fax machine back up through * to our ISDN lines |
| 21:33.25 | jr_ewing | Do you thinks it's normal to pay a pbx more than 10000 $ |
| 21:33.27 | robin_sz | probably some anaglogue to iax convertor |
| 21:33.35 | rob0 | It's a shame for important software of ANY kind to be running on broken, proprietary OS's. :) |
| 21:33.38 | jr_ewing | working on open source and community develloper still |
| 21:33.48 | jr_ewing | wait for benefits of their work... |
| 21:34.03 | robin_sz | jr_ewing: no, but now you have already paid it, moving to * sounds like a plan to make more work if it aint broke ... |
| 21:34.43 | jr_ewing | no |
| 21:34.47 | *** join/#asterisk dasuberdavid (n=dasuberd@gateway.digium.com) |
| 21:34.50 | jr_ewing | I don't want to replace |
| 21:35.00 | jr_ewing | existing avaya solution, |
| 21:35.30 | jr_ewing | My boss will sack me if i tell you one year after upgrade : Hey, i found a goog pbx totaly free |
| 21:35.40 | robin_sz | depends ... |
| 21:35.45 | jr_ewing | ...with all options you still pay |
| 21:35.57 | robin_sz | depends if you are the guy that suggested Avaya ... |
| 21:36.04 | rpm | avaya, ugh.. i remember dealing with that stuff. i love asterisk :) |
| 21:36.15 | Qwell | at least it isn't nortel |
| 21:37.11 | *** join/#asterisk DannyF (n=dannyf@c-f0aae455.24-0099-74657210.cust.bredbandsbolaget.se) |
| 21:37.13 | jr_ewing | we have avaya for ten year but i 'am the Rebel of my company .... |
| 21:37.27 | WillSip | hi i need support instalation asterisk |
| 21:37.47 | jr_ewing | So ... an idea on this variable i can't found.... |
| 21:37.51 | jr_ewing | ? |
| 21:38.09 | Err | I think the "look at the source" idea was a good one |
| 21:38.26 | jr_ewing | i tried, but i don't know how to start |
| 21:38.32 | jr_ewing | i seek into oh |
| 21:38.37 | jr_ewing | 323_channel.c |
| 21:38.45 | jr_ewing | or something like that |
| 21:39.14 | jr_ewing | and did'nt find sentence like : Connection established with '.... |
| 21:39.22 | robin_sz | anyway ... I still need to know a bit about running * and Hylafax on the same box ... 4 isdn lines, * answers them and does the ReceiveFax thing over CAPI which works perfectly ...but need to use Hylafax I think to add a fax printer for sending ... |
| 21:39.42 | robin_sz | do I need to dedicate a channel to hylafax in some random way? |
| 21:40.50 | jr_ewing | why do you try to use hylafax and not fax systeme include in asterisk ? |
| 21:41.05 | robin_sz | there is one? |
| 21:41.11 | jr_ewing | yes |
| 21:41.14 | robin_sz | for rx yes, I use that already |
| 21:41.19 | robin_sz | but for sending? |
| 21:41.27 | jr_ewing | just a minute |
| 21:42.24 | robin_sz | 30 seconds gone ... |
| 21:42.56 | jr_ewing | Sending and Receiving Faxes with Asterisk |
| 21:42.56 | jr_ewing | That means when Asterisk is the endpoint of a fax transimition. In these cases Asterisk has to simulate a fax machine and either do something with thi just received image or have received an image in some way that is latter faxed. |
| 21:42.56 | jr_ewing | To achieve this there are two Asterisk applications: app_rxfax and app_txfax which work on top of a library called spandsp. |
| 21:42.56 | jr_ewing | The ast_fax application (atand alone app) provides email-asterisk integration. To make life a lot easier, use the mail2fax and fax2mail bash scripts available from http://www.generationd.com. These 2 scripts make it easy to send and receive email (based on app_rxfax. app_txfax, and ast_fax). |
| 21:42.59 | jr_ewing | Another choice is http://wpkg.org/email2fax - it only needs spandsp/app_txfax to send faxes, and accepst e-mails with PDF and TIFF attachments. |
| 21:43.02 | jr_ewing | HylaFax and Asterisk |
| 21:43.04 | jr_ewing | Another solution is the Hylafax software. capi4hylafax and chan_capi will gladly coexist. You just tell asterisk to ignore the DIDs that are used for fax. A maximum of 1 passive card and 4 active cards are supported. RedHat users: Some useful RPM can be found here. |
| 21:43.08 | jr_ewing | hylafax-users Hylafax and Asterisk - Configuration report |
| 21:43.22 | robin_sz | of ffs |
| 21:43.24 | Err | a link would have been *way* better than pasting all of that |
| 21:43.30 | robin_sz | WAY better ... |
| 21:43.37 | jr_ewing | yep just to interrest you |
| 21:43.51 | jr_ewing | now you have to pay :=====) |
| 21:44.10 | Drew___ | how can i deactivate the internal dialplan of Xlite? |
| 21:44.10 | jr_ewing | http://www.voip-info.org/wiki/view/Asterisk+fax |
| 21:44.17 | robin_sz | ok, thats better, thanks |
| 21:44.58 | jr_ewing | don't forget |
| 21:45.23 | jebba | there are t.38 patches in bugzilla too (for faxing) |
| 21:45.30 | jr_ewing | yep |
| 21:45.40 | jr_ewing | so |
| 21:46.03 | jr_ewing | is there someone who know someone to help me with my debug trace..... |
| 21:46.14 | jr_ewing | just a little variable to find ..... |
| 21:46.29 | jr_ewing | Please.......... |
| 21:47.40 | Err | jr_ewing: just use grep and search for the string that's printed out, and see what variable it uses in the code - and then look and see if that variable is available in the dialplan |
| 21:47.45 | Err | (or wherever you need it) |
| 21:48.24 | jr_ewing | that seems a good idea, |
| 21:48.36 | Err | of course it is |
| 21:48.45 | jr_ewing | but do you mean a variable could be available in debug mode |
| 21:49.09 | jr_ewing | and not one of NoOp can display ? |
| 21:49.21 | jr_ewing | ohhh |
| 21:49.30 | libila | What does the ! do in 'while (!feof($fp)) { $line = trim(fgets($fp));' I didn't see it on the operator page so what does it do? |
| 21:49.39 | Err | no, I mean that the variable will be in C, and I'm sure that not all variables there are available via the scripting |
| 21:49.47 | Qwell | libila: not |
| 21:49.49 | libila | wrong channel |
| 21:50.03 | Qwell | That was a very, very, very basic question |
| 21:50.03 | jr_ewing | ok |
| 21:50.09 | jr_ewing | i was afraid by this idea |
| 21:50.13 | Err | yes, a C book would be good |
| 21:50.36 | jr_ewing | yep |
| 21:50.40 | jr_ewing | but |
| 21:50.46 | Err | hm, that's not really C, though - I don't know what it is - perl maybe? |
| 21:51.02 | jr_ewing | 300 hundred hours of devellopent course also required....;=) |
| 21:51.19 | Err | jr_ewing: I wasn't talking about your question - although it wouldn't hurt you, either, if you're going to use a "free" program |
| 21:51.39 | jr_ewing | i'am just self made Linux Avaya Asterisk advanced user ... |
| 21:51.41 | Err | it's stupid to consider asterisk free, though - it's clearly not, if you don't have some level of technical expertise that you clearly do not currently have |
| 21:52.10 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:52.18 | Err | (no offense intended - I'm just pointing out that asterisk isn't necessarily going to save your company any money, if nobody there knows anything about it) |
| 21:52.47 | *** join/#asterisk kletter-matze (n=kletter-@dslb-084-056-211-169.pools.arcor-ip.net) |
| 21:52.53 | jr_ewing | i quiet agree with you |
| 21:53.03 | Qwell | Err: I loudly agree |
| 21:53.23 | [TK]D-Fender | :) |
| 21:53.40 | [TK]D-Fender | Point to you this time Qwell... |
| 21:53.45 | jr_ewing | but the way of open source is to hope , for someone like me, i will obtain help from a coummunity |
| 21:53.56 | jr_ewing | may be i'had a dream... |
| 21:54.24 | *** join/#asterisk dijit0_ (n=dijit0@c-69-181-150-200.hsd1.ca.comcast.net) |
| 21:54.30 | *** join/#asterisk _blop (i=blop@openbeer.be) |
| 21:54.39 | jr_ewing | theres no other intereset in asterisk for me to inprove my own knowledges |
| 21:54.42 | *** join/#asterisk kletter-matze (n=kletter-@dslb-084-056-211-169.pools.arcor-ip.net) |
| 21:54.49 | jr_ewing | after |
| 21:54.56 | [av]bani | http://uncyclopedia.org/images/5/5b/Windowsvistamarketing.jpg |
| 21:55.53 | jr_ewing | if i can create job for C. expert in my team and use a all in one system wich improve services in my company to help her to recruit agent |
| 21:56.02 | Err | jr_ewing: it's not that people don't want to help you improve your knowledge - they certainly do - they just don't want to do your work for you |
| 21:56.22 | Err | there's sometimes a fine line between those two things, and sometimes a huge chasm, and everyone draws his/her line in a different place |
| 21:56.23 | *** join/#asterisk kletter-matze (n=kletter-@dslb-084-056-211-169.pools.arcor-ip.net) |
| 21:56.27 | [av]bani | jr_ewing: give the average person a cisco and a pile of voip phones and their results will be no better than if you gave them an asterisk pc. voip and telco in general is complex |
| 21:56.38 | *** join/#asterisk dijit0__ (n=dijit0@c-69-181-150-200.hsd1.ca.comcast.net) |
| 21:57.04 | jr_ewing | hey, i think you think i'am a dummies ..... |
| 21:57.14 | Err | it doesn't make sense for the support costs of your company's phone system to be shifted from paying your avaya dealer to costing #asterisk members time :-) |
| 21:57.39 | [av]bani | also, just because something costs lots of $$$ (eg televantage) doesnt mean you can make it do what you want |
| 21:57.44 | Err | jr_ewing: I never said that - I am saying, however, that since no one here appears to know the answer to your question off the top of his/her head, it would cost *us* time/money to find out |
| 21:57.48 | *** join/#asterisk oceanlan|dustin (n=info@cpe-69-133-109-130.woh.res.rr.com) |
| 21:58.41 | jr_ewing | Stop your attack, please, i 'am just here try to find a solution to my problem |
| 22:00.33 | jr_ewing | and if i can help, share my own knowleges (like spandsp ...) |
| 22:00.46 | Err | there's no attack intended, from me |
| 22:01.15 | *** join/#asterisk Nivex (i=kjotte@user-0ce2nsu.cable.mindspring.com) |
| 22:01.22 | Corydon76-home | jr_ewing: what's the problem? |
| 22:01.42 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 22:02.14 | jr_ewing | hi, I just want to know if someone here have ever dealed with h323 problem i 'am currentlyu encountering |
| 22:02.39 | Corydon76-home | Probably not. We tend to avoid h323 like the plague |
| 22:03.32 | jr_ewing | yeah sure i prefere IAX 2 but i work on project (Asterisk pseudo Cti for Avaya S8700 through H323) |
| 22:03.36 | WillSip | alguien que sepa como puedo actualizar kernel |
| 22:03.55 | jr_ewing | will sip : what's your distro ? |
| 22:03.58 | [av]bani | h323 is very old and limited :/ |
| 22:04.08 | jr_ewing | yes |
| 22:04.18 | WillSip | jr_ewing is Red Hat enterprise |
| 22:04.18 | [av]bani | i have h323 phone i made work with asterisk but its incredible piece of shit |
| 22:04.33 | jr_ewing | but available on my avaya without additional cost |
| 22:04.48 | WillSip | ok |
| 22:04.53 | [TK]D-Fender | [av]bani : Get a UNISTIM i2005 then! |
| 22:04.53 | *** join/#asterisk kletter-matze (n=kletter-@dslb-084-056-211-169.pools.arcor-ip.net) |
| 22:05.02 | [av]bani | [TK]D-Fender: !!! |
| 22:05.07 | jr_ewing | willsip : yum update |
| 22:05.08 | [TK]D-Fender | [av]bani: !!! |
| 22:05.28 | Corydon76-home | jr_ewing: can you get help with your Avaya without additional cost? |
| 22:05.49 | jr_ewing | no.....never find something free with avaya |
| 22:06.09 | *** join/#asterisk SERGEUS|W (n=SERGEUS@ippe-245.ippe.ru) |
| 22:06.19 | Corydon76-home | I mean, I can get dbase III for free, but I'm never going to use it, because I can't find people to work with it for free |
| 22:06.32 | jr_ewing | but it just for test, if it's works, i 'am sure my company will give money in that project |
| 22:06.36 | Err | WillSip: are you asking how to build a kernel? |
| 22:06.38 | WillSip | jr_ewing dou you speak spanish |
| 22:06.47 | jr_ewing | no |
| 22:06.50 | [av]bani | jr_ewing: h323 works with asterisk, sort of |
| 22:06.51 | jr_ewing | sorry |
| 22:06.53 | WillSip | ok |
| 22:06.56 | Qwell | he barely speaks English, heh |
| 22:06.57 | WillSip | dont worry |
| 22:07.09 | Corydon76-home | No, but I bet he speaks French |
| 22:07.16 | jr_ewing | yep but is there anyone speaking french |
| 22:07.22 | jr_ewing | here.... |
| 22:07.30 | WillSip | ok who |
| 22:07.31 | Corydon76-home | We have Quebecois in here |
| 22:07.44 | jr_ewing | nice ! |
| 22:07.59 | [TK]D-Fender | C'est pas vrais! Il faut rein croir d'eus-autres! |
| 22:08.21 | jr_ewing | but i think i would better to speak most known language.. |
| 22:08.23 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 22:08.38 | Corydon76-home | We even have people who speak Russian |
| 22:08.50 | Qwell | nice timing |
| 22:08.58 | WillSip | ok |
| 22:09.39 | jr_ewing | qwell what are you meaning ?Fender; do you have a project to play guitar with band remotely through asterisk N |
| 22:09.41 | jr_ewing | ? |
| 22:09.51 | Qwell | huh? |
| 22:10.09 | jr_ewing | you said nice , why ? |
| 22:11.57 | [Airwolf] | Is there anyone who has some experience with Realtime/Mysql and using Macro's in the extentions ? |
| 22:12.32 | *** join/#asterisk ibob61 (n=hp@82.111.125.213) |
| 22:14.09 | [Airwolf] | Because I can't get a macro to work. |
| 22:14.10 | Qwell | [Airwolf]: realtime switch? I use it with mssql, but yes, I do macros |
| 22:14.17 | [Airwolf] | Here is the output: http://pastebin.com/527934 |
| 22:15.10 | [Airwolf] | Qwell, well mssql of mysql doesn't really matter ofcourse. But do you reconize the pasted error ? |
| 22:15.12 | [TK]D-Fender | jr_ewing : It's my nickname from what I was in FPS caming playing CTF maps. And no, I play a Dean, Ibanez, and if I follow through with buying the one I was looking at today, a Yamaha. |
| 22:15.16 | Qwell | [Airwolf]: Do you have macro-dial? |
| 22:15.59 | *** join/#asterisk mutilator (i=WebChat@i.think.napoleon.dynamiteblows.com) |
| 22:16.01 | WillSip | alguien que me ayude a actualizar red hat enterprise linux 3 para multiprocesadores |
| 22:16.09 | WillSip | actualizar kernel |
| 22:16.41 | Err | WillSip: doesn't it come with a multiprocessor kernel? |
| 22:17.01 | Err | (note: I can *read* spanish some, I just can't speak/write it ;-) |
| 22:17.39 | [Airwolf] | Qwell, yes I have. This is my current configuration: http://pastebin.com/527947 |
| 22:18.05 | [av]bani | anyone use voicetronix fxo cards? |
| 22:18.34 | Qwell | [Airwolf]: in the database, replace the ,'s with |'s |
| 22:18.47 | Qwell | , can't be used to separate args in realtime, only | |
| 22:19.18 | Qwell | It's literally looking for [macro-dial,374,IAX2/374,20] |
| 22:19.36 | [Airwolf] | Qwell, ok didn't knew that |
| 22:19.44 | [Airwolf] | Thanks |
| 22:20.00 | [TK]D-Fender | Qwell : whats the deal with using "|" as a delimiter as opposed to"," now? I noticed mention of a plan to deprecate "," in extensions.conf as well. |
| 22:20.06 | [Airwolf] | Qwell, yeah well I don't have that |
| 22:20.08 | [Airwolf] | :P |
| 22:20.14 | Qwell | [TK]D-Fender: dunno |
| 22:20.28 | Qwell | but | is used far less often in strings |
| 22:21.12 | [TK]D-Fender | I personally find "|" loathsome and is harder to visually seperate from l1I and just not familiar for those programming in most languages |
| 22:21.31 | Err | WillSip: Red Hat Enterprise 3 tiene un kernel para multiprocesadores, no? |
| 22:21.53 | Err | (tener probably is incorrect, there - I do not know how to say "comes with" or "ships with") |
| 22:22.08 | [Airwolf] | Qwell, it works now |
| 22:22.19 | Err | yeah, I don't like | nearly as much as , |
| 22:22.39 | Err | of course, my vote shouldn't count much, as I run a VoIP-only asterisk to talk to a few family members ;-) |
| 22:22.44 | [Airwolf] | But I have to agree with [TK]D-Fender that visually seperating a '|' is more difficult |
| 22:23.44 | *** join/#asterisk Z-Knight (n=Z-Knight@cpe-67-10-17-120.houston.res.rr.com) |
| 22:24.00 | [TK]D-Fender | Whats the major upside to "realtime", buggy as I've heard it still is.. ? |
| 22:24.10 | Z-Knight | anyone know what happened to the asterisk forum at digium? |
| 22:25.17 | Z-Knight | ? |
| 22:25.28 | rpm | Z-Knight, asterisk is no longer. digium went out of business |
| 22:25.37 | Z-Knight | k thanks |
| 22:25.43 | X-Rob | try the mailing list. people actually read that. |
| 22:25.43 | rpm | haha, in kidding |
| 22:25.44 | Z-Knight | anyone else care to give a real answer? |
| 22:25.47 | *** join/#asterisk Rambozo (n=b@hoffmann.dls.net) |
| 22:25.58 | X-Rob | the answer is no-one knows, and no-one cares. Forums suck. |
| 22:27.33 | [av]bani | [TK]D-Fender: afaict the * developers have no idea how to design script languages |
| 22:27.54 | *** part/#asterisk Rambozo (n=b@hoffmann.dls.net) |
| 22:28.35 | [TK]D-Fender | [av]bani : Somewhat... AEL is nifty though..... |
| 22:29.14 | [TK]D-Fender | [av]bani : But as ugly as extensions.conf is, I don't mind it much really... it is very direct and you don't have to worry about matching braces 4 pages down. |
| 22:29.46 | Err | that's what real text editors are for :-) |
| 22:30.23 | [av]bani | [TK]D-Fender: 1,n suck |
| 22:30.38 | [TK]D-Fender | Err : Still.. its that our normal kludge isn't susceptable to it :) |
| 22:31.03 | [av]bani | [TK]D-Fender: i'm sorely tempted to integrate lua as a replacement, because ael is sorely lacking |
| 22:31.15 | [TK]D-Fender | [av]bani : Well I haven't had to do anything so complex that 1,n felt nasty. Anything more complex is left to AGI |
| 22:31.46 | X-Rob | ffs, it's just BASIC with more brackets and braces. |
| 22:33.46 | [TK]D-Fender | X-Rob : Hey, I wrote BBS's and plenty of other comm programs in BASIC so pipe down! |
| 22:33.59 | X-Rob | I'm saying there's nothing _wroing_ with basic |
| 22:34.08 | X-Rob | [av]bani is the one bitching about it. |
| 22:35.20 | Err | it's not really at all like basic, other than it has syntax and uses $ characters |
| 22:35.28 | [TK]D-Fender | I remember having written ANSI and AVATAR terminal emulations from the ground up.... on of my terminals started getting a scripting language for macros that went from single line nested text functions (not entirely unlike *'s) to a full mid-level language (the only to my knowledge) |
| 22:35.29 | *** part/#asterisk Z-Knight (n=Z-Knight@cpe-67-10-17-120.houston.res.rr.com) |
| 22:35.50 | [av]bani | i'd also appreciate it if * didn't sort extensions.conf, i want stuff in the order i put it in there. |
| 22:36.04 | X-Rob | Err, no, it's very similar to basic. I could throw together a perl script that would convert it to and from BASIC _Very_ easily. Go read the syntax again. |
| 22:36.16 | [TK]D-Fender | [av]bani : Yeah.. the pattern match heirarchy is a piss-off.... |
| 22:36.39 | Err | one could say that about many languages |
| 22:36.40 | [av]bani | [TK]D-Fender: its non obvious and can cause a lot of head scratching / hair loss |
| 22:37.48 | *** join/#asterisk _Soul_ (n=Soul@87-196-44-148.net.novis.pt) |
| 22:38.45 | [TK]D-Fender | [av]bani : the parts I hate most are tests in GotoIF having to have "safety chars" to protect against null values..... |
| 22:39.18 | [TK]D-Fender | Actually, thats not gotoif specifically but rather the "evaluator" for expressions...... |
| 22:39.31 | rpm | if you know basic you can write an asterisk dialplan |
| 22:40.26 | [av]bani | [TK]D-Fender: its obvious extensions.conf started out very basic and then got hack piled upon ugly hack :() |
| 22:41.16 | [av]bani | atm the sorting is the most annoying thing though. i want a switch to turn it off. |
| 22:41.29 | Err | if you know any language you can write a dialplan |
| 22:41.42 | Err | it looks more like bourne shell script to me |
| 22:41.49 | [av]bani | hardly! |
| 22:42.04 | [av]bani | dunno what bourne shell you are talking about |
| 22:42.47 | fiber0pti | I have a bunch of polycom 500's that I'm setting up. copying the same config files with the same settings 5 out of 13 of them won't register with asterisk, not even an error message, any ideas? |
| 22:43.04 | Err | I'm talking about the actual language features, like the evaluation of expressions and function calling - the other junk is just made up |
| 22:43.42 | *** join/#asterisk gambolputty2 (n=gambolpu@cblmdm72-240-116-131.buckeyecom.net) |
| 22:45.13 | gambolputty2 | Is it possible for * to do the equivalent of sip show channels within a dialplan? |
| 22:45.36 | Err | heh, what does that even mean? |
| 22:45.59 | [av]bani | Err: bourne shell has a much richer set of expressions |
| 22:46.09 | Err | [av]bani: I know - I'm sure basic does, too |
| 22:46.19 | [av]bani | ael would be closer, extensions.conf is more like eh... logo? |
| 22:46.27 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-ull-47-70.44-151.net24.it) |
| 22:46.35 | Err | my point is that it's not really like *any* language - but what you know will remind you of it, because all languages are pretty much the same :-) |
| 22:46.56 | Err | it's sort of tcl-ish in its nested evaluation syntax |
| 22:47.24 | [av]bani | Err: no... haskell is entirely unlike the ada class of languages. functional languages usually are |
| 22:47.36 | Err | obviously I meant imperative languages |
| 22:47.47 | Err | heh, "the ada class" |
| 22:47.48 | [av]bani | hence you said 'all languages' |
| 22:48.06 | [av]bani | meaning 'all languages except this entire class of them' |
| 22:48.23 | Err | meaning "all languages that most people even know exist" :-) |
| 22:52.15 | [av]bani | the digium TDM400P has no hardware AEC right? |
| 22:52.17 | [TK]D-Fender | fiber0pti : pastebin "ls -l" in your provisioning folder.. |
| 22:52.19 | [av]bani | its all software |
| 22:55.16 | *** join/#asterisk MGSsancho (n=user@adsl-67-127-173-128.dsl.irvnca.pacbell.net) |
| 22:55.28 | *** join/#asterisk funxion (n=nunya@host-64-110-51-254.hlm.ses-americom.net) |
| 22:55.44 | X-Rob | [av]bani, yes that's correct. |
| 22:55.49 | X-Rob | the 2400 can have hardware EC |
| 22:55.55 | *** join/#asterisk btoe (n=nick@adsl-71-131-185-171.dsl.sntc01.pacbell.net) |
| 22:55.57 | *** join/#asterisk dijit0 (n=dijit0@adsl-68-127-10-129.dsl.pltn13.pacbell.net) |
| 22:56.07 | funxion | Does anyone use the r option in the Dial command? |
| 22:56.42 | Err | yes - otherwise lines don't "ring" when you're waiting for the extension to be answered |
| 22:56.59 | funxion | what do you mean |
| 22:57.11 | funxion | they ring for me without the r option |
| 22:57.15 | Err | interesting |
| 22:57.23 | funxion | ur kidding ryte |
| 22:57.25 | Math` | the ringing if the other party sends our session progress with ringing status |
| 22:57.37 | funxion | yes |
| 22:57.40 | Math` | s/the ringing/they ring/ |
| 22:57.41 | fiber0pti | D-Fender: http://pastebin.com/527991 |
| 22:57.47 | Err | oh, if you're using purely IP phone stuff, you might hear ringing anyway due to signalling |
| 22:57.48 | btoe | Hi, I'm looking for a good tool to act as an answering machine for a home phone, and turn messages into email attachments. Is Asterisk what I'm looking for? I'd ideally like something easy-ish to set up |
| 22:58.03 | [av]bani | someone should just make a generic hardware EC card for * |
| 22:58.06 | X-Rob | btoe, see Asterisk@Home |
| 22:58.11 | btoe | thx much. |
| 22:58.15 | Err | if you use a FXO/FXS interface, without 'r' you won't hear a ring sound |
| 22:58.27 | Math` | yeah you will |
| 22:58.31 | funxion | I use an E1 pri and I get ringing |
| 22:58.45 | Err | during transfers, after * has answered the line? |
| 22:58.48 | Math` | in FXO you get ringing when the audio gets bridged |
| 22:58.58 | [TK]D-Fender | fiber0pti : Do you see anything in the * CLI showing a reg attempt? |
| 22:59.23 | Err | interesting - then 'r' doesn't do what I read that it does |
| 22:59.53 | *** join/#asterisk SibRhell (i=SibrPhre@user-12lccke.cable.mindspring.com) |
| 22:59.58 | SibRhell | stupid poweroutage |
| 23:00.00 | fiber0pti | D-Fender: Nothing.. that's why I'm confused by the whole thing.. but they seem to be d\l their config files because they get the wav file, and they use the parameters in the files like getting their extensions. |
| 23:00.07 | rpm | SibRhell, you in vancouver? |
| 23:00.13 | *** part/#asterisk btoe (n=nick@adsl-71-131-185-171.dsl.sntc01.pacbell.net) |
| 23:00.18 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 23:01.47 | [TK]D-Fender | you sure the reg info is right on all of them? |
| 23:02.08 | fiber0pti | D-Fender: pretty sure... I've checked all of them more than a couple of times. |
| 23:02.24 | funxion | I am sending calls to an E1 PRI that goes to a cisco whichs breaks the voice into voip then sends it over satellite. If the link goes down the cisco takes forever to timeout so * can "roll over to the next route of choice so I wanted to insert fake ringing from the time the call starts. If I do this then when the call rolls over will the ringing continue even if the second route returns a busy or will it change from ringing to bus |
| 23:02.50 | znoG | weird, i put pickupgroup=1 for 2 SIP accounts, and when I ring one of them, I press *8# from the other, and it says "nothing to pick up" ?? |
| 23:03.03 | SibRhell | rpm - vancouver?? |
| 23:03.44 | X-Rob | znoG, don't forget using 'callgroup' too. |
| 23:03.56 | [TK]D-Fender | fiber0pti : if you do them one at a time do you see each attempt? |
| 23:04.49 | [TK]D-Fender | fiber0pti : and is it the sames ones that always fail? |
| 23:04.49 | funxion | my question above was concerning the r option in the dial command |
| 23:05.25 | znoG | X-Rob: ah, why are both needed? i better check the wiki |
| 23:05.39 | X-Rob | yeah, I remember some wierdism there. |
| 23:05.39 | fiber0pti | D-Fender: don't see them register and I'm at level 5 verbose. Yes.. same 5 always fail. (In the upper left for the channels that are configed it's a little phone that's empty) |
| 23:06.01 | fiber0pti | D-Fender: and for the other ones I don't see them register either. |
| 23:06.18 | [TK]D-Fender | fiber0pti : ok, thats a reg failure. can you pastebin the phonexx.cfg for one that works, and one that fails? |
| 23:07.32 | fiber0pti | working: http://pastebin.com/527999 |
| 23:07.42 | *** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 23:08.23 | fiber0pti | D-fender: non-working: http://pastebin.com/528000 |
| 23:09.44 | [TK]D-Fender | fiber0pti : change the bad one to use the ext# for disp name, address etc... I found setting the wrong one in there killed me a few times |
| 23:09.56 | [TK]D-Fender | fiber0pti : leave naming up to *. |
| 23:10.15 | WillSip | alguien que me ayude a actualizar kernel en enterprise 3 |
| 23:10.35 | fiber0pti | The bad one is using the extension number :/ the but I also have many other ones that work with that |
| 23:12.13 | [TK]D-Fender | fiber0pti : ok, pastebin sip.conf |
| 23:12.56 | fiber0pti | D-Fender: http://pastebin.com/528011 there is another working one that is similar to the non-working |
| 23:13.22 | [TK]D-Fender | fiber0pti : Did you build these completely from scratch? |
| 23:13.40 | fiber0pti | D-Fender: No. Got the from someplace like voip-info. |
| 23:14.23 | fiber0pti | D-fender: sip.cfg is too big. Won't fit in the putty buffer |
| 23:14.32 | [TK]D-Fender | fiber0pti : missing the sip.conf pastebin... I only have a minute to look... |
| 23:14.39 | [TK]D-Fender | sip.conf, sorry |
| 23:14.41 | [TK]D-Fender | * |
| 23:14.47 | fiber0pti | ohh |
| 23:14.58 | [TK]D-Fender | :) |
| 23:15.06 | fiber0pti | it's too big too :/ |
| 23:15.07 | [TK]D-Fender | I know sip.cfg is over 100K :) |
| 23:15.12 | *** join/#asterisk saftsack (n=saftsack@p54A7CEFC.dip.t-dialin.net) |
| 23:15.14 | [TK]D-Fender | your * sip.conf? |
| 23:15.24 | fiber0pti | yeah.. cuz of all the extensions I have to register |
| 23:15.26 | fiber0pti | got 15 in it |
| 23:15.27 | [TK]D-Fender | how can it be taht bad? |
| 23:15.39 | [TK]D-Fender | 15? big deal.. paste it all |
| 23:15.56 | fiber0pti | Ok. I will try.. gotta piece it together |
| 23:16.16 | [TK]D-Fender | why in pieces? |
| 23:18.11 | SibRphrek | can you setup for a 2nd extention to be on the same outgoing phone number? or would that cause asterisk to have an anurism? |
| 23:18.29 | [TK]D-Fender | outgoing? clarify please... |
| 23:18.48 | SibRphrek | like i have a real number that calls my server. 516+***-**** |
| 23:18.54 | SibRphrek | but currently only have 1 |
| 23:19.04 | [TK]D-Fender | SibRphrek : if you mean dialing 2 zap number, then no... they are both considered "answered" as soon as you dial ...... |
| 23:19.08 | SibRphrek | can i have 2 extentions on that same number? |
| 23:19.15 | [TK]D-Fender | and most things that hit PSTN the same.... |
| 23:19.16 | SibRphrek | oh ok |
| 23:19.18 | SibRphrek | that's what i thought |
| 23:19.25 | SibRphrek | i'm waiting on a DID on monday |
| 23:19.33 | SibRphrek | i got an excel sheet coming with a bunch of numbers for me |
| 23:19.52 | [TK]D-Fender | ok I've got to go... might be back on later, but I doubt it.... |
| 23:19.55 | SibRphrek | now if i could just get the master.csv to export and get it importing into Fmpro |
| 23:19.56 | fiber0pti | D-Fender: http://pastebin.com/528017 |
| 23:19.58 | SibRphrek | later TK |
| 23:20.18 | fiber0pti | aww shit.. he left didn't he |
| 23:20.41 | [TK]D-Fender | fiber0pti : You should be setting them up as FRIEND, not PEER.... |
| 23:20.50 | fiber0pti | oh |
| 23:21.05 | SibRphrek | what's the difference between friend and peer? |
| 23:21.16 | fiber0pti | but some are working with peer |
| 23:21.23 | [TK]D-Fender | fiber0pti You also have a USERNAME clause in there for 0683 which you shouldn't in ANY of them. I believe these 2 things will fix it all |
| 23:22.11 | fiber0pti | oh |
| 23:22.12 | fiber0pti | hah |
| 23:22.13 | fiber0pti | thanks |
| 23:22.44 | fiber0pti | no user name? |
| 23:22.46 | [TK]D-Fender | if you're lucky.. I don't see anything else offhand unless you mac.cfg is pointing to the wrong phonexx.cfg file... |
| 23:22.54 | fiber0pti | don't have a mac.cfg |
| 23:22.56 | fiber0pti | what's that? |
| 23:22.57 | [TK]D-Fender | either way, triple check after the changes & reboots of phones... |
| 23:23.00 | [TK]D-Fender | later |
| 23:23.10 | dijit0 | anyone know if iax.cc/sixtel is any good? |
| 23:23.11 | X-Rob | wibble. |
| 23:23.13 | [TK]D-Fender | your <mac>.cfg in provisioning. |
| 23:23.17 | fiber0pti | ohh |
| 23:23.30 | [TK]D-Fender | triple check that you're calling the right phonexx.cfg |
| 23:23.32 | [TK]D-Fender | as well |
| 23:23.33 | [TK]D-Fender | later |
| 23:24.44 | fiber0pti | I thought that you had to have a username in sip.conf for each entry? |
| 23:25.20 | fiber0pti | didn't work.. |
| 23:29.25 | fiber0pti | how do you flash a polycom phone 100%? I've flashed it using 4,6,8,* and the line still has a name associated that I never entered |
| 23:33.24 | *** join/#asterisk hellop (n=hellop@cpe-70-95-165-136.hawaii.res.rr.com) |
| 23:33.28 | hellop | hi |
| 23:34.11 | hellop | I plugged in a USB jumpdrive to my * server, and immediately, my 100p zap card died. |
| 23:34.33 | X-Rob | good effort. |
| 23:34.41 | hellop | When I do ztcfg -vv, I get ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 23:34.45 | *** join/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au) |
| 23:34.49 | hellop | I don't understand what changed.. |
| 23:35.00 | hellop | any suggestions? |
| 23:35.01 | X-Rob | have you power cycled the machine and re-seated the board? |
| 23:35.07 | hellop | X-Rob, yes |
| 23:35.17 | X-Rob | well, go buy another x100p card then. your one just broke. |
| 23:35.27 | hellop | weird concidence... |
| 23:35.43 | hellop | maybe I ESD shocked it? |
| 23:35.45 | X-Rob | Nah. They're crap. What do you expect for $5? |
| 23:35.52 | hellop | herm.. |
| 23:36.11 | hellop | ok.. I'll try that. |
| 23:36.14 | hellop | tks X-Rob |
| 23:36.16 | *** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net) |
| 23:37.30 | *** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com) |
| 23:40.21 | *** join/#asterisk angom_h (n=angom@red-corp-201.130.171.190.telnor.net) |
| 23:42.14 | *** join/#asterisk angom (n=angom@red-corp-201.130.171.190.telnor.net) |
| 23:42.19 | *** join/#asterisk hellop (n=hellop@cpe-70-95-165-136.hawaii.res.rr.com) |
| 23:42.26 | hellop | X-Rob, you were right. |
| 23:42.32 | hellop | thanks. |
| 23:42.41 | X-Rob | heh |
| 23:42.43 | X-Rob | no probs |
| 23:42.59 | X-Rob | I've had nothing but bad luck with 'em. |
| 23:43.11 | hellop | It just couldn't compete with my massive jump drive, and commited suicide. |
| 23:43.26 | *** join/#asterisk fugitivo (n=ajf@201.255.176.5) |
| 23:44.55 | jr_ewing | hi there |
| 23:45.00 | robin_sz | why hello |
| 23:45.23 | jr_ewing | because i was not here since one hour |
| 23:45.31 | jr_ewing | ;=) |
| 23:45.42 | *** part/#asterisk angom (n=angom@red-corp-201.130.171.190.telnor.net) |
| 23:45.44 | robin_sz | ok, let me try that again |
| 23:46.03 | robin_sz | "why, hello there" |
| 23:46.18 | *** join/#asterisk Tecky` (n=jkroll@its.inevetable.com) |
| 23:46.28 | robin_sz | no question mark, see? |
| 23:46.30 | robin_sz | :) |
| 23:47.33 | jr_ewing | how to configure X101 P for france, i need fxo mode = CTR21...? i can dial but canno't receive even if ztmonitor show gain increase when it rings |
| 23:47.35 | jr_ewing | ? |
| 23:47.42 | Tecky` | got a question for anyone to answer... Could i, have a asterisk box on the inside of my nat'd network (192.168.0.13) and place a phone at my office at another location and have phone calls come in there (using some kinda external routing) ? |
| 23:47.55 | *** join/#asterisk razu (n=razu@213-35-173-39-dsl.prn.estpak.ee) |
| 23:48.22 | jr_ewing | ya with iax2 no problem with nat... |
| 23:48.35 | Tecky` | how hard would it be to setup ? |
| 23:48.38 | jr_ewing | just open 4569 port on yours both site |
| 23:49.18 | Tecky` | ahhh nice thats the only port ? |
| 23:49.37 | jr_ewing | yep |
| 23:50.07 | *** join/#asterisk The_X (i=chris@true.fiberpimp.net) |
| 23:50.11 | The_X | hi folks |
| 23:50.14 | The_X | quick Q |
| 23:50.14 | jr_ewing | hi |
| 23:50.25 | The_X | I have a 7960 behind a linksys talking to my asterisk box at work |
| 23:50.31 | Tecky` | jr_ewing: what setup do you recomend for a asterisk@home setup .... card & phones ? |
| 23:50.40 | The_X | when I talk from a cell to the 7960 it works fine |
| 23:50.46 | The_X | but voice from 7960 to cell won't work |
| 23:50.48 | jr_ewing | depend on what you need... |
| 23:50.54 | jr_ewing | if you have analog |
| 23:51.12 | Tecky` | I have a pots line from verizon feeding all the outlets in the house ... etc. |
| 23:51.13 | jr_ewing | go to order digium board to avoid lose your time |
| 23:51.34 | The_X | do I need to fwd something to get the voice from 7960 to work outside the nat |
| 23:51.39 | The_X | inbound works fine |
| 23:51.45 | The_X | can't figure it out |
| 23:52.09 | jr_ewing | what is the channel you are usinf The-x ? |
| 23:52.12 | jr_ewing | SIP ? |
| 23:52.24 | The_X | sip |
| 23:52.34 | The_X | 7960 -> linksys -> internet -> asterisk |
| 23:52.49 | jr_ewing | asterisk is on public adress ? |
| 23:52.51 | The_X | yes |
| 23:52.56 | The_X | it registers and all |
| 23:53.03 | The_X | I can talk from my cell to the 7960 and it works |
| 23:53.07 | jr_ewing | not behind nat (your company lan ) |
| 23:53.09 | The_X | but from 7960 to cell, I can't hear |
| 23:53.10 | [av]bani | hmm |
| 23:53.25 | X-Rob | The_X, welcome to the world of VoIP and NAT |
| 23:53.30 | The_X | I'm no network clueless and I read a whole lot to get it working but I can't figure out why outbound won't work |
| 23:53.31 | fiber0pti | Anyone know why my asterisk box might be ignoring the digittimeout and responsetimeout? Just hangs up when there's nothing else to do |
| 23:53.31 | dogtanian | heh |
| 23:53.32 | dogtanian | yeah |
| 23:53.35 | X-Rob | it's a world of pain and frustration |
| 23:53.36 | dogtanian | just use iax :) |
| 23:53.46 | jr_ewing | does 7960 handle IAX protocol ? |
| 23:53.49 | [av]bani | cisco VIC-FXO are they decent FXO interfaces? |
| 23:53.50 | The_X | inbound should be the problem |
| 23:53.52 | [av]bani | jr_ewing: no |
| 23:53.54 | The_X | but it works |
| 23:53.55 | jr_ewing | yea Dog |
| 23:53.58 | X-Rob | fiber0pti, Use TIMEOUT(digit) and TIMEOUT(response). |
| 23:54.02 | X-Rob | those other two are depreciated. |
| 23:54.17 | jr_ewing | you cannot expect it works well with to endpoint behind nat |
| 23:54.23 | fiber0pti | X-Rob: I am :/ |
| 23:54.25 | dogtanian | The_X: it's a NAT problem and i had it with SIP too.... I use IAX now and it works fine |
| 23:54.26 | The_X | the asterisk has a public address |
| 23:54.30 | jr_ewing | If Asterisk is realy on public adress, try a Stun server |
| 23:54.31 | fiber0pti | exten => s,1,Set(TIMEOUT(digit)=5) |
| 23:54.39 | fiber0pti | exten => s,n,Set(TIMEOUT(response)=10) |
| 23:54.45 | The_X | does it come with asterisk? |
| 23:54.57 | jr_ewing | yep |
| 23:54.59 | hardwire | no |
| 23:55.02 | *** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk) |
| 23:55.24 | The_X | still weird that inbound works |
| 23:55.27 | dogtanian | you'll need to use a voip provider that supports iax tho |
| 23:55.27 | The_X | but outbound wont |
| 23:55.37 | The_X | the asterisk server is my own at work |
| 23:56.02 | fiber0pti | X-Rob: kinda of odd? |
| 23:56.10 | jr_ewing | because asterisk don"t know where to send rtp packet behind your home gateswsay |
| 23:56.11 | X-Rob | yup. dunno. |
| 23:56.36 | fiber0pti | oh.. I do get the following message though: == Auto fallthrough, channel 'SIP/0684-a754' status is 'UNKNOWN' |
| 23:57.03 | The_X | ewing, when I call from anywhere to my 7960 it works fine |
| 23:57.11 | The_X | but it's when I talk from the 7960 that it doesn't |
| 23:57.16 | The_X | no need to fwd any ports for that |
| 23:59.58 | jr_ewing | no |
