IRC log for #asterisk on 20121116

00:00.45SeRidijib: I replyed to you
00:00.51*** join/#asterisk itamarjp (~itamar@fedora/itamarjp)
00:01.54itamarjpanyone have setup a polycom music on hold using uri ?  voIpProt.SIP.musicOnHold.uri ?
00:06.13ChainsawI just leave that to my regular Asterisk MOH.
00:06.31Chainsaw(I do have Polycom handsets, yes)
00:08.00itamarjpChainsaw, the phone is not using asterisk, I just need music on hold from asterisk or other place.
00:09.42[TK]D-Fenderitamarjp: We don't do that.  Guess you'll just have to try for yourself
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00:24.15paulctosses ghost75 a loonie
00:28.45*** join/#asterisk jraddin (~jraddin@72-48-177-116.static.grandenetworks.net)
00:31.32jraddinI am using ALSA to tie into a ConfBridge.  I can dial in to the bridge via a SIP trunk and merge the ALSA in with "console dial 6300@conferences".  Over time, however, the audio quality degrades and begins to motorboat.  After about 30 minutes, the audio is unusable.  If I do a console hangup and redial, everything is fine.  Any ideas?
00:36.32ChrisInSydneyjraddin: Ouch ! :-/
00:37.47ChrisInSydneyjraddin: Small question from someone who really has no idea, but, are you using an internal sound card ? do you have a USB one you could try ?
00:37.54WIMPyI haven't tried for a while, but when I did, i had the issue that I got an ever increasing delay.
00:38.32jraddinI have tried the onboard soundcard and 2 different PCI cards
00:39.12ChrisInSydney:-/
00:39.18jraddinhavent tried USB, but a lot of the USB cards don't support multiple sample rates which I have found asterisk requires since it outputs at 8khz
00:39.26ChrisInSydneyahh
00:39.41WIMPyKnown issue :-(
00:40.12ChrisInSydneyno specific reason for USB, just wonderin' if you had tried a different device
00:40.19jraddinsample rate issue or motorboating?
00:40.29ChrisInSydneybut if WIMPy is right :-/
00:41.28ChrisInSydneyMotorboating usuallly power caps, but probably not in this instance
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00:46.30jraddinwould meetme have the same issue?
00:48.32WIMPyI wouldn't be surprised if just a normal call would have it.
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00:56.35slav3_kittenis 100ms of jitter correctable on iax?
00:58.39*** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu)
00:58.47jraddinI'm also getting a chan_alsa.c:479 alsa_read: Read error: Resource temporarily unavailable when I dial the call from the console
00:59.32WIMPyslav3_kitten: Sure, but it might be a bit hard for the conversation.
01:00.52slav3_kittenhmmm
01:01.30slav3_kittenthis wisp sucks dick... i seem to get jitter and such on my IAX2 voip.ms incoming calls. sip out to flowroute is supposedly nice an solid
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01:14.31jraddinWIMPy: Is there any timeline for a fix on this issue?
01:15.15WIMPyI don;t know if anyone is interested in fixing it.
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01:15.39jraddinSo ALSA just doesn't work?
01:16.30WIMPyI don't know. As I said, I heven;t tried it for quite a while.
01:16.42jraddinGotcha
01:16.43jraddinsorry
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01:33.22slav3_kittenhuzzah i have a UK number
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02:10.49dijibanybody in here know if asterisk has the ability to accept collect calls?
02:10.53dijibautomaticall?
02:10.54dijiby
02:11.23MaliutaWould depend on how the call is to be accepted
02:11.38dijibhmm ok then
02:12.09MaliutaIf you get an operator asking a question you could use a small IVR (Collect calls accepted, press 1 to connect)
02:13.05MaliutaOtherwise you'd have to have something recognise a particular phrase and then have it send the DTFM response
02:13.30Maliutado you know what happens when you receive a collect call?
02:13.48dijibno
02:13.51dijibi should try it
02:14.38MaliutaAnother option would be to set up a dialback (assuming the other end is sending CID) ... if the rings X times and hangs up then you dial the number back and connect it to a local SIP/IAX/DAHDI channel at the same time
02:14.59dijiband i could be different from a jail. im just wanting to setup a system in case one would need to call out from a cell block through asterisk
02:15.11Maliutathere are many options, you need to know exactly what you want/need
02:15.13dijibcallback is not an option
02:15.29dijibok then thanks Maliuta
02:15.47Maliutaunless you know the procedure that occurs when the call comes in, you're screwed
02:15.49Maliuta:)
02:16.19dijibthat cant be that hard to attain
02:16.34dijibjust mixmonitor a call
02:21.10ChrisInSydneyhi all
02:21.43ChrisInSydneyquick q?? Does anyone have a list of SIP addresses that "do things"
02:21.51ChrisInSydneymonkeys,
02:22.00ChrisInSydneytell me has etc
02:22.04ChrisInSydneytell me has gone
02:22.22ChrisInSydneyjust want to do some quick tests / demos for trainnig
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02:27.46WIMPyWhat do you need?
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02:32.52ChrisInSydneyWIMPy: Maybe something like tellme or something where we can use a Dial() command to connect directly to a SIP URI and have it do somthing like tell the time, play some file, echo test
02:33.10ChrisInSydneygot a list with ekiga I'm testing but they are all a bit crap
02:35.34MaliutaChrisInSydney: with which provider? most of them have echo test numbers and talking clocks
02:36.26Maliutaif you want to do it on your own machine then just create extensions that play sound files
02:36.39Maliutaor do echo
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02:46.44WIMPyIs there a way to set the language for sip guest calls only?
02:46.44*** join/#asterisk deo (~deo@222.127.13.226)
02:48.17WIMPyIf I get it right, you can only set defaults and not anything for only guests.
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02:49.41linociscohi all, I have three PSTN lines and want to share it among VOIP phones extensions, which grandstream devices should I buy?
02:50.02linociscoor any cheaper products?
02:50.36WIMPywaits for someone to argue that this isn't possible.
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02:57.13*** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com)
02:57.30dfgas-cr48trying to compile asterisk 11.0.1
02:58.04dfgas-cr48and its complaining about missing ccart
02:58.07dfgas-cr48ccar
02:58.12dfgas-cr48not ccart
02:58.34dfgas-cr48what is that? i put in apt-get install ccar and no
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03:04.52ChrisInSydneyMaliuta: Nahh. Looking for an anon SIP URI to call
03:04.59ChrisInSydneyWIMPy fixed me up offline
03:05.02ChrisInSydney:-)
03:07.14dijibdfgas-cr48: i got that when i tried to compile the rc1 or two whatever it was
03:07.52dfgas-cr48dijib, i am downloading what ever is on the asterisk page
03:07.56ChrisInSydneyWIMPy: Re Guests, I'm pretty sure or at least i had vivid haloucinations of, a way of setting the language from within the dialplan
03:08.16dfgas-cr48dijib, how did you get past it?
03:08.34WIMPyChrisInSydney: yes. That's possible.
03:08.36slav3_kittenanyone in here good with cisco ios routers?
03:09.09WIMPyActually that would work as guest have their own context.
03:09.16ChrisInSydneyCorrect
03:09.33ChrisInSydneyI always have a separate context for guest connects
03:09.40slav3_kittenhttp://pastie.org/5385221 < which is better and will they both work?
03:10.02ChrisInSydneyOther SIP services get their own too
03:10.03dijibi got the offical release and not the review candidate.
03:10.05dijibdfgas-cr48:
03:11.05WIMPyOk. That should help.
03:11.42ChrisInSydneyslav3_kitten: The one that requires less typing ;-)
03:11.52ChrisInSydneyunless you need to fix it,
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03:11.56slav3_kittenChrisInSydney, but will they both accomplish the same thing?
03:12.00ChrisInSydneythen the one that is most descriptive
03:12.24dfgas-cr48dijib, hmmmm
03:12.24ChrisInSydneynot too good on Cisco, I only wear their shirts so  I can charge more
03:12.29ChrisInSydneybut
03:12.52slav3_kittenChrisInSydney, i'm not great myself but i think they'd both work an option B is prettier
03:13.34dfgas-cr48dijib, this is the link for the source i am using http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11-current.tar.gz
03:14.11ChrisInSydneylooking at it, if you're just doing IAX stuff, B. Less ACLs and probably easier to read
03:14.16ChrisInSydneyassuming it works
03:14.32ChrisInSydneyWIMPy. Glad to return the favour
03:14.43slav3_kittenChrisInSydney, as far as i can test it appears to function
03:14.55ChrisInSydneycool
03:15.18ChrisInSydneycopy run start. Then reload. Just to check :-)
03:15.24WIMPyChrisInSydney: So you should be able to understand the talking clock now :-)
03:15.34dijibthat should do it
03:15.44ChrisInSydneyWIMPy: I'll give it a go
03:16.54ChrisInSydney1191 Zeitansage ?
03:17.08WIMPygoogle fail?
03:17.10WIMPyyes
03:17.34WIMPyOops. Typo.
03:17.49ChrisInSydney:-)
03:18.24WIMPyI guess CHANNEl works better than CAHNNEL.
03:18.42dijibdfgas-cr48: try 'make distclean' from /asterisk-11.0.xxx/res/pjproject then go back to root and remake
03:18.53dijibit was something to do with that res/pjproject
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03:20.33dfgas-cr48dijib, alrighty
03:21.18slav3_kittenany easy cli to show the version on 1.8?
03:21.20dijibdfgas-cr48: http://lists.digium.com/pipermail/asterisk-dev/2012-September/056899.html is where i got the res/pjproject
03:21.33slav3_kittenChrisInSydney, it's been running and i get the iax in from what i can tell
03:21.46WIMPycore show version
03:21.52dijibi think it was through a make clean , ./configure , make menuselect , make , make install
03:21.56dijibpossibly
03:22.06dijibafter the pjprojects
03:22.07slav3_kittenthanks WIMPy
03:22.10WIMPyInteresting how google changes the numbers.
03:23.12WIMPyAfter translation **61*xxx**sek# becomes **61*xxx#sec**
03:24.31ChrisInSydneyWIMPy: Im getting a mixture of Alison Smith and some German chick
03:24.34slav3_kittensvn checkout http://svn.asterisk.org/svn/asterisk/trunk asterisk < should get version 11 right?
03:24.55WIMPySounds interesting.
03:25.01ChrisInSydneyanyway WIMPy. Thanks heaps. I'll keep going
03:25.15WIMPyslav3_kitten: More than that.
03:25.26ChrisInSydneyWIMPy. Yep. I get the "Im sorry thats not a valid extension"
03:25.30ChrisInSydneyanyway
03:25.36ChrisInSydneyThanks heaps
03:25.38slav3_kittenWIMPy, more then that?
03:25.38ChrisInSydney:-)
03:26.03WIMPyI can see that SayDigits() won't succeed with your caller ID.
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03:49.26*** join/#asterisk rue_mohr (~rue@24-207-103-226.eastlink.ca)
03:50.18rue_mohrif I needed a 6 or 8 channel "channelbank" of mixed fxo fxs what would you suggest
03:50.26rue_mohrbox or card
04:00.11slav3_kittenWIMPy, upgrading from 1.8 to 11, just make and make install everything right?
04:11.49ChrisInSydneyWIMPy: It skips the non numeric characters
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04:13.47slav3_kitteni think oyu killed him ChrisInSydney
04:14.28ChrisInSydneyoops
04:15.06rue_mohris there a way to check the voip audio levels?
04:15.17rue_mohrsay, in the rtp stream?
04:15.51rue_mohr_realtime_
04:16.24rue_mohrinstead of hours of recording 1mw tones, loading them into an app, and readjusting gains
04:26.24Nivexrue_mohr: http://www.voip-info.org/wiki/view/Asterisk+cmd+Milliwatt  ?
04:26.39dijibdfgas-cr48: whats going on over there
04:28.45rue_mohrI know I c an generate a 1mw tone
04:29.06rue_mohrI use to use the providers tone and check the levels between the gain amps on the rtp stream
04:29.14rue_mohrI have evil polycom phones
04:33.38ChrisInSydneyc yaz all later
04:33.47dijibgday
04:38.46dijibi think i killed you slav3_kitten
04:39.07slav3_kitteni am dead
04:40.07dijibi knew it
04:40.18dijibare you good in linux?
04:40.25dijibor have you just been toying with asterisk
04:40.31dijib+ asterisk related issue
04:40.46dijibISP blocks port 25
04:41.05dijibhave a intranet smtp server and allow relay.
04:41.14dijibthe relay does not allow .wav
04:41.22dijibto be send as attachment
04:41.26dijibsent
04:41.31slav3_kittendijib, i'm decent with linux but i'm no good with smtp server crap
04:41.38dijib:D
04:41.55dijibwelcoe to the world of home asterisk
04:51.35dfgas-cr48dijib, well i got it farther, i am now installing freepbx
04:52.07dijibDO NOT INSTALL FREEBOX
04:52.11dijibeffin hell
04:52.26dfgas-cr48how do i know if asterisk compiled with sip support, because typing core show help does not show anything with sip
04:52.27dijibwait or is that pbx in a flash
04:52.37dijibsip show registry
04:52.52dfgas-cr48sip show registery does not work either
04:53.00dijibhere people if you want my help join the conf if you can, & teaviewer me
04:53.39dfgas-cr48Laptop*CLI> sip show registry
04:53.41dfgas-cr48No such command 'sip show registry' (type 'core show help sip show' for other possible commands)
04:53.41dfgas-cr48Laptop*CLI>
04:57.14dijibwell then youve got.... possibly permission isues
04:57.43dfgas-cr48ughhhh
04:58.02dijibwhat did you do with make menuselect
04:58.15dijibdo anything funny in there or just eep the basic + mp3?
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05:03.45dfgas-cr48basic with mp3
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05:23.43slav3_kittenis there anyway to forbid dialing out on an interface?
05:23.58slav3_kittenerr channel
05:24.01slav3_kittenthing
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05:40.11Kobazso is it bad when ast_db_put deadlocks
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05:52.52navaismo~nat
05:52.52infobotsomebody said nat was Network Address Translation  Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly.  See docs.
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06:57.23slav3_kittenLucent 8411D anyone ever use?
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07:18.57bulkorokhi
07:20.45v0lZyHi.
07:20.56dijibno
07:21.08dijibim working on dfgas-cr48 asterisk deployment
07:21.22dijiband i think im going to barf
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07:32.14ectospasmlooks like you already did
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08:08.27dijiball over you ectospasm
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08:51.56unicrondijib: still awake?
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09:04.32mirela666Can a channel be hanged if i know SIPCALLID or BRIDGEDPEER or any other var than CHANNEL(name)
09:04.34mirela666?
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09:16.01frecklemirela666: CHANNEL REQUEST HANGUP <SIP CHANNEL>
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09:21.52hwtgood morning. if I were to implement a failover scheme (to the outgoing partner) using an AGI script, what would be the best way to proceed?
09:22.04hwtfetching dialstatus and dialing over, or is there a smarter way to do this?
09:22.19hwt(basically this, but in AGI: http://mikepultz.com/2010/05/automatic-dial-resource-fail-over-in-asterisk/)
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09:24.14luneffhey guys. i'm trying to use ARA here with mysql and i can't get calls to ARA SIP users. I can make a call as a ARA user, but when i try to dial ARA number, i get bad-number message. i guess, the trouble lies within nonconfigured realtime switch, but i can't get the thing right :-(
09:27.14R1ckhey. I've used Trixbox CE in the past to get an "Out of the Box" Asterisk PBX running but it seems that Trixbox CE won't be maintained anymore. Can anybody recommend a different Asterisk PBX solution?
09:28.28ghost75gemeinschaft
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09:34.05kaldemarR1ck: try asterisknow
09:35.02hwtis it perhaps easier to just dump the call into a macro from AGI and let that one sort it out? i would of course need to supply it with the two (or more) SIP peers it should try
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09:38.35kaldemarhwt: why are you using AGI for that?
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09:44.34WIMPyghost75: That's dead as well. At least as far as Asterisk is concerned.
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09:48.25ghost75dead?
09:50.13hwtkaldemar: it is a generic SIP router for SIP trunking. all routing decisions are fetched from a RESTful API.
09:50.20hwtkaldemar: also call state, actually
09:50.59hwtkaldemar: 1. get incoming call, 2. check if source is authenticated and tell where to route it, 3. try primary IP address, and if it fails 4. try secondary IP address
09:51.03hwtsomething like that
09:53.52kaldemarhwt: still, why are you using AGI?
09:54.38bulkorokghost75: Macro() is depricated; Use GoSub() instead with RETURN()
09:55.06kaldemarchecking DIALSTATUS is the way to approach this.
09:55.23ghost75dont tell me
09:56.06kaldemarghost75: anything that still uses old versions like 1.6.X should not be recommended to anyone.
09:57.10ghost75what is freepbx using
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10:02.45kaldemarfreepbx is originally just the GUI, but they do have a full linux distro with asterisk and freepbx nowadays.
10:03.50ghost75i see gemeinschaft changed to freeswitch
10:04.09bulkorokdon't use gemeinschaft
10:04.42bulkorokthere is only one main programmer...
10:05.04kaldemarthe freepbx distro has a version of 1.8 branch, i guess.
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10:44.37WIMPyInteresting to see how many people are looking for hfcs-usb support in dahdi.
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11:04.19hwtkaldemar: because i don't want to embed scripts and stuff in the dialplan?
11:04.27hwtkaldemar: i need to talk to this restful api
11:04.35hwtkaldemar: and you can't do that from the dialplan
11:05.14hwtkaldemar: from this IP address i get the request uri user, the hostname, transport type, port, etc.
11:05.27hwtkaldemar: so i don't have any peers defined (allowguest=yes)
11:05.38hwtkaldemar: if there is an uknown IP or user trying to call, i just deny the call
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11:09.12asteriskATmarmuDany idea on how to reducing the volume of the ringtone heard by the caller?
11:09.43ghost75turn volume down on phone?
11:10.01kaldemarasteriskATmarmuD: what kind of a phone?
11:10.10asteriskATmarmuD:) this is not an option, since it would reduce the overall volume
11:10.20WIMPyAssuming you're talking SIP it's probably coming from the phone itself.
11:10.39WIMPySo that's the only option.
11:10.45asteriskATmarmuD<PROTECTED>
11:10.59freckleanyone know if it is possible for another channel to PauseMonitor a different monitored channel?
11:11.05WIMPyOk, the gateway then.
11:11.18kaldemarasteriskATmarmuD: configuring the gateway is pretty much your only choice.
11:11.49asteriskATmarmuDWIMPy: ok, I guess I can't do anything about it, since the gateway only allows to tune the general volume
11:12.01asteriskATmarmuDthx guys for reassuring
11:12.34WIMPyYou can do the hack of answering the call before passing it on.
11:13.32kaldemarfreckle: you'd have to do that via AMI.
11:14.09hwtkaldemar: let me turn the question around, how would you design it?
11:15.54kaldemarhwt: depends on how your API is used.
11:16.45hwtkaldemar: it's an HTTP service with JSON data. i query it when an INVITE comes in, and Dial() based on data I get in the response to the API
11:17.07hwtkaldemar: if it's a known IP and user, I forward it to the PSTN. if not I deny it. if it comes from PSTN, I forward it to the correct user.
11:17.12kaldemarnothing stops your from using dialplan for that.
11:17.23WIMPyhwt: How many calls/time?
11:17.26frecklekaldemar: thanks
11:18.27kaldemarusing func CURL is one possibility.
11:18.37hwtWIMPy: nothing frightening. perhaps 5-10 calls per second
11:19.18WIMPyI wouldn't use anything that forks anyway.
11:19.49hwtkaldemar: yes, i know it's possible. it would just be very complicated to change the logic. i need a lot of if/elses, and might need to add more advanced options later on. also, i'm more comfortable in python. you haven't really convinced me it's a problem using AGI for this. ;)
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11:20.06hwtWIMPy: agree, it's ugly, but i don't have that many options if i want to use ast.
11:20.24hwtWIMPy: i wanted to use sippy for this, actually. but the decision wasn't mine.
11:20.37WIMPySo you don't get an result from your http request but have to do lots of processing still?
11:20.57kaldemarhwt: i'm not convincing you that it's a problem using AGI for that. it's just that AGI is often used even when there is no need whatsoever for it.
11:21.19hwtWIMPy: if the API is down, i handle this in a different way.
11:21.51hwtWIMPy: advantage here is that i can use an absolute timeout (from urllib2), which i guess will be hard with func CURL.
11:22.13WIMPyis AMI fan
11:22.26hwtkaldemar: ok.
11:22.33hwtWIMPy: would AMI fit here?
11:22.51WIMPyYou can do anything with it.
11:23.19hwt*reading*
11:23.33hwtWIMPy: because i too am worried about the scaling of forking a script for every call
11:23.47hwtWIMPy: plus, i have experience with scripts not terminating properly
11:25.56hwtWIMPy: so basically you can set it up to send an event over the manager interface to my program when a call enters, then pass back Command statements to forward that particular call?
11:26.44WIMPyYou don;t even have to send the event yourself. You just have to enable them.
11:27.00kaldemarhwt: btw, you can configure curl usage with CURLOPT.
11:27.20WIMPyYou just have to make sure your call waits in the dialplan for your application to direct it elsewhere.
11:27.59hwtkaldemar: okay, cool. however, i think the logic will be to complex to make it maintainable in extensions.conf. AEL could possibly work, but then I need to learn that first. ;)
11:28.30kaldemarhwt: that is most likely just a matter of opinion. :)
11:28.47hwtWIMPy: and you can also add headers, change transport, port, host, etc, and catch dial status through it?
11:29.08hwtif anyone has some good literature on getting started with AMI (voip-info sucks) it would be appreciated
11:29.25WIMPyI haved never tried to change transport in realtime.
11:29.47gustohi WIMPy
11:30.01hwtWIMPy: you can do that with Dial(). if you get an incoming call over UDP, it's not a problem to forward it using TCP or even TLS
11:30.04WIMPyBut getting info about dialstatus works very well that way.
11:30.13frecklekaldemar: I am sending this "Action: PauseMonitor  SIP/mychan-0002475f" but it says invalid command. Any ideas?
11:30.22kaldemarhwt: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-AMI.html
11:30.25WIMPyhwt: Then you're fine.
11:30.32WIMPyMoin gusto
11:30.46hwtWIMPy: any limitations I should be aware of?
11:30.51kaldemarfreckle: that is not a valid command. :P
11:31.19kaldemarfreckle: see what "manager show command PauseMonitor" tells you.
11:31.20WIMPyhwt: Don't think so.
11:31.50frecklekaldemar: ty
11:32.13hwtthanks, guys!
11:42.06gustowell
11:42.13gustoWIMPy: what's new?
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11:58.10mirela666Can I hangup channel if I don't know CHANNEL(name), if i have something else valuble, maybe bridgedpeer,...
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12:01.19kaldemarmirela666: how are you trying to perform the hangup? in dialplan?
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12:05.48fabsofthi all
12:07.59fabsoftis there a way to fix called number in a macro routine? i need to call this macro with EXTEN which has some pattern and return to caller stack at new fixed EXTEN
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12:08.23WIMPyGoto
12:12.06fabsoftWIMPy: how i'll able to return to the caller context ?
12:12.34fabsofti'd pass it as argument ?
12:13.13WIMPyGoto(${CONTEXT},newexten,1)
12:13.16fabsofteg: Macro(fixDID,thiscontextname)
12:14.31fabsofthe calling extension, context, and priority are stored in ${MACRO_EXTEN}, ${MACRO_CONTEXT} and ${MACRO_PRIORITY} respectively. .. ok :)
12:14.38mirela666kaldemar: yes
12:15.59mirela666kaldemar: basicly idea is that i create channel to another pbx with originate, and i would like to hang up that Originated call on my hangup
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12:17.24mirela666depending on time of my hangup, that outgoing channel might not yet be created, so  i guess in that case it's impossible
12:17.40mirela666well created but not bridged
12:19.58mirela666SoftHangup(SIP/<resource>,a) is good but hangs all calls to that pbx
12:20.39kaldemarmirela666: func CHANNELS() might help
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12:22.54mirela666kaldemar: thanks, that might just help :)
12:23.58kaldemarwould be nice if Originate set a variable with the channel for the originated call name.
12:24.23mirela666yes I agree, to know who created it
12:25.04mirela666or whos child  is it :)
12:26.37mirela666I tried listening with AMI Events but I get Channel name only when originate gets response
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12:51.24hwtok, so i have the basic manager daemon running. it's currently just displaying all events.
12:51.48hwthow should my extensions.conf look like if i want to leave all routing decisions to AMI?
12:53.11WIMPyYou need to keep the call somewhere until your app reacts.
12:54.33hwtWIMPy: allright, so just do an _X.,1,Wait(10) or whatever
12:55.21WIMPyFor example.
12:55.29hwtWIMPy: and then issue Command statements?
12:55.55hwtWIMPy: or call out, then bridge?
12:56.07hwtsorry for all the stupid questions
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12:58.04WIMPyThe easiest way might be to set up some variables and then redirect to an extension that dials to those variables.
12:59.11hwtWIMPy: so i set some ${DIALSTRING} from AMI, then in extensions i have 1: if $dialstring, dial $dialstring, else goto 1?
13:02.02WIMPyI thought more like doing Dial({destvar},${time},${opts})
13:02.20hwtWIMPy: yes, but i can't know when this string is set.
13:02.50WIMPyYou set it befor sending the call to that exten.
13:06.27hwtWIMPy: yes, but i don't want to have an unnecessary wait(n)
13:06.49hwti want it to trigger the Dial somehow after the {destvar} contains anything meaningful
13:07.02WIMPyYou don;t have to wait.
13:07.13WIMPyYou set up the vars and then do a redirect.
13:08.09WIMPySo basically what I do is a Wait() and then somethig to execute in case the script fails.
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13:08.31hwtWIMPy: oh, so Wait will just wait until you "pick it up" with a redirect?
13:08.45WIMPyThat's the idea.
13:08.52WIMPyJust a timeout for the app.
13:08.59hwtaha
13:11.09hwtwell, that leaves for a very compact dialplan. ;)
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13:17.15hwthm, docs says you can't run Wait without arguments
13:18.23WIMPyJust put as meny seconds there as you think your app will need at maximum.
13:19.04hwtWIMPy: aha, then we were talking around each other. i don't find it acceptable even with a 1 second wait, if it takes 0.002 seconds for the API query to run
13:19.45hwtWIMPy: maybe a Ringing() will be slightly better, but i don't like it really
13:20.09WIMPyIt doesn't matter. When you tell it to redirect, it will do so immediately.
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13:20.26hwtWIMPy: oh, it does. then it's good.
13:20.30WIMPyNo, you need something that waits.
13:21.18hwtthat seemed to work. i guess Uniqueid is the best to use as an identifer?
13:21.41hwtper channel
13:21.45WIMPyOr the channel name.
13:21.54hwtthe uniqueid is shorter :)
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13:52.16WIMPyYou need the channel name anyway.
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13:53.42hwtWIMPy: okay, good point
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13:55.00Sanongood morning all
13:55.31SanonI'm having a bit of an issue with setting up softphones
13:55.58SanonI have some settup just fine on my smart phones
13:56.18Sanonbut the pc's are acting strange
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14:01.34SanonI downloaded sflphone, put in all the right info
14:01.40Sanonbut nothing
14:02.38Sanoni have 3cx phone and zoiper on my android devices working just fine
14:04.15*** join/#asterisk serafie (~erin@76.73.167.231)
14:06.22Sanonis anyone here?
14:07.50WIMPyYou need to be more specific.
14:10.01jmetroi am neither here nor there
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14:12.21ghost75some whee
14:12.26ghost75where
14:12.40jmetroover the rainbow?
14:13.26Sanonok. can anyone tell me how to set this sflphone thing up
14:13.59WIMPyThe manual?
14:15.40[TK]D-FenderBAI BAI
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14:21.43jayteeI'm at a client site with a new Asterisk installation to replace an old 3Com VCX system that died. They have a T1 PRI and inbound and outbound calls to a landline sound clear and work fine but the system seems to have issues with inbound calls from cell phones breaking up and sometimes dropping or sometimes not accepting the DTMF tones when the caller dials an extension from the IVR menu.
14:21.44jayteeThis is a very rural area but there is a Verizon 4G cell tower nearby. Not sure what I can check or test at this time to pinpoint the problem. Anyone have any suggestions?
14:24.07[TK]D-Fenderjaytee, Make sure your EC is stable and that your gains aren't distorting things.  Then also try "relaxdtmf=yes" for your channels and see if that loosens it up.
14:24.46jmetroi have a client who gets called on cell all the time and always complains about their inbound calls but outbound are perfect- cell calls are just awful imo and for the most part you cant be responsible if the inbound caller sets up a terrible connection.
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14:38.56jaytee[TK]D-Fender, I have the TE122B with the hardware echo cancel module. I don't have relaxdtmf set though. IIRC, that will require a restart, not just a reload.
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14:40.57[TK]D-Fenderjaytee, A reload of the channel module, yes
14:41.12[TK]D-FenderWhichw ill kill calls over it
14:41.17jayteeBoth txgain and rxgain are set to 0.0, wonder if kicking up the rxgain would help?
14:41.45WIMPyIt surely would make things worse.
14:42.36[TK]D-Fenderjaytee, Shouldn't have to.  Kicking up gains tends to cause distortions...
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14:49.54ghost75i can replace macro(bla) straight with gosub(bla) ?
14:50.54[TK]D-Fenderghost75, Depending hwhat you use to return, in general, yes
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14:52.43P-NuTHi all, has anyone had any luck getting cisco 7940 xml configuration working? I've been googling for days but can only find references to the old configs or patchy documentation on the current examples.
14:52.58P-NuTDoes anyone have a real and actual xml config example for one?
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14:56.04ghost75does it make sense to use g.711 from ata to asterisk and g.729 on trunk ?
14:56.47[TK]D-Fenderghost75, Could be.
14:57.13ghost75but then it would be translated
15:01.42[TK]D-Fenderghost75, Clearly.
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15:31.31xoverukhi
15:31.40xoverukI am having issues with my FXO card, I cannot dial out, but can dial in
15:31.48xoverukHow can I diagnose the problem?
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15:32.03[TK]D-Fenderxoveruk, Like everything else... * CLI
15:32.12[TK]D-Fenderxoveruk, And I dunno ... looking at your configs.
15:32.17xoverukIf I run dahdi show channel 1 I get offhook
15:32.24xoverukconfig should be ok, it normally works
15:37.57xoverukcan you advise?
15:38.50drmessanoLook at the CLI output of a failed call, for starters
15:42.24xoveruk-- Called g0/xxxxxxxxxxxxxx
15:42.24xoveruk<PROTECTED>
15:42.27xoverukthat is it
15:42.31xoveruknothing happens
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15:45.58and7eyhi, is something wrong with this dial plan - http://pastebin.com/uLrViPhC ? I expect 4 digit calls to be routed within network and all others - to `beeline`
15:50.32[TK]D-Fenderand7ey, exten => _X.,n,Dial(SIP/beeline/${EXTEN}) <-- you have no priority *1*
15:50.41[TK]D-Fenderand7ey, You can't just start with "n"
15:51.59[TK]D-Fender(which would imply being related to a previous extensions priority which was a different pattern anyway)
15:58.38and7ey[TK]D-Fender: aah, priority shhould be 1 for new pattern.. ok, thanks!
16:00.41kaldemarand7ey: note that XXXX is a better match than X. for 4 digits.
16:02.20[TK]D-Fenderkaldemar, He laready has that
16:02.23[TK]D-Fenderalready*
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16:05.20and7ey[TK]D-Fender: is that correct? http://pastebin.com/mqMvhkk8
16:06.04and7eylooks like it tries to dial both SIP/${EXTEN} and SIP/beeline/${EXTEN}
16:08.40[TK]D-Fenderand7ey, those are both priority 1.  That can't both execute.
16:09.08jmetrowell, anything that matches 4 digits will hit the first, and every other call with match the second, so they are independent yeah.
16:09.09[TK]D-Fenderand7ey, And only one of those patterns leads to an explicit Hangup.
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16:11.55and7ey[TK]D-Fender: here is new version - http://pastebin.com/xVFzhFqn
16:12.19[TK]D-Fenderand7ey, What do ouside phone #'s really look like?
16:12.50and7ey[TK]D-Fender: for ex., 79169801234
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16:13.11and7ey[TK]D-Fender: or +79169801234
16:13.46*** join/#asterisk Sanon (47c8d864@gateway/web/freenode/ip.71.200.216.100)
16:13.54[TK]D-Fenderand7ey, perhaps you should make the pattern that lets you dial those out be a little more specific than the pattern you're using....
16:17.56*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
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16:48.53Sanonnot much activity going on here.
16:49.18Sanonas you all know, i'm new to this and have a lot of questions.
16:50.04wdoekes~ask
16:50.04infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
16:50.05Sanonhere is one. how can i get my system accept any sip or iax2 softphone
16:50.44Sanoni've downloaded many diff. ones, none work
16:50.58Sanonbut all my android devices work
16:51.27jmetroif you register them properly they all should work. maybe 2 or 3 lines of code and I got my 3cx softphone online
16:52.53Sanonok. i have Ekiga on linuxmint right now. i put the information from my sip.conf in, but it's not registering
16:53.41jmetrodid you sip reload
16:54.16Sanonyes. i reload
16:54.39jmetrois the phone pointing to the correct IP address [your asterisk box]
16:54.47*** join/#asterisk cklimos (~Claude@209.5.121.227)
16:54.58jmetroala "outbound proxy" "sip registrar" etc
16:55.14*** join/#asterisk jblack (~jblack@93.sub-70-192-150.myvzw.com)
16:55.17Sanonthey all are correct.
16:56.40navaismoshow us the cli output with the sip debug
16:56.43jmetroset your console verbose and debug levels high enough to see any incoming requests? make sure you can ping the * box from the linuxmint box
16:56.56jmetrocore set verbose 9999 core set debug 9999
16:57.05jmetrosip set debug on
16:57.14navaismo?_? level 3 is enough
16:58.07Sanonok
16:59.56*** join/#asterisk jblack_ (~jblack@93.sub-70-192-150.myvzw.com)
17:01.58navaismouse pastebin to show us the debug ~pb
17:02.02Sanonok, i tried to register the information for my computer on my android device and it rejected it
17:04.25[TK]D-Fender"sip set debug on" <---- if you aren't looking at SIP debug then you are only fooling yourself.
17:07.17*** part/#asterisk freckle (~jon@firebox.enta.net)
17:13.58SnivetsIs there an easy way to boost volume across a sip channel?
17:14.05SnivetsAre certain codecs louder than others?
17:14.16leifmadsenSnivets: VOLUME()
17:14.25Snivetshmm
17:14.54leifmadsencodecs should not be "louder" than others no
17:15.03leifmadsenthe amplitude that is recorded should be the same throughout
17:15.07Snivetswell, wait a second. So I installed a snom m9 handset yesterday, and it is really damn quiet, even with the unit's volume up and the other party's mic set higher too
17:15.13leifmadsensince it's just a digital representation of analog data
17:15.21Snivetsk, that makes sense
17:15.29leifmadsenSnivets: then that's a problem with the device itself driving the analog data
17:15.48SnivetsWouldn't boosting the volume to that particular client's channel be an option?
17:15.56leifmadsenhence VOLUME()
17:16.08SnivetsFair enough, now where does that go? lol
17:16.15leifmadsenin the dialplan
17:16.22leifmadsenVOLUME() is a dialplan function
17:16.24Snivetsyes, but wherein?
17:16.26leifmadsen'core show function VOLUME'
17:16.35leifmadsenSnivets: depends what you're trying to do and where it should go
17:16.40QwellIt's too bad there isn't some sort of book that has this stuff in it.
17:16.41leifmadsensuggests before Dial() at the very least
17:16.47leifmadsenQwell: IKR?!
17:17.04QwellI bet it would sell a HUNDRED copies.
17:17.16QwellThink of how many nickels you'd get.
17:17.20Snivetswell, realistically, I should be looking at voip info
17:17.27QwellNo you shouldn't.
17:17.30Snivetsoh ya?
17:17.31Qwell~asterisk wiki
17:17.31infobotsomebody said asterisk wiki was http://wiki.asterisk.org/
17:17.48Snivetsoh snap, I see,using the set function
17:18.00Qwellapplication*
17:18.20Snivets"Search Confluence"
17:18.40SnivetsRusty Newton looks kinda like a dog, guys.
17:18.58mjordanSnivets: we'll let him know
17:19.04Qwellnewtonr: ^
17:19.05mjordannewtonr: you look like a dog.
17:19.16Snivetsaww, yeah, we look out for each other.
17:20.05*** join/#asterisk blee (~blee@72.188.117.219)
17:20.52SnivetsI'm getting a Windows phone today
17:20.55Snivetswhat are our thoughts on that?
17:21.00QwellWhy?
17:21.06SnivetsI think that about sums it up, yeah
17:21.12coppicemy sympathies
17:21.13Robotman321Congratz Snivets :p
17:21.14jmetrothey look interesting.
17:21.23SnivetsI'm getting the 920
17:21.27Robotman321I have a HTC Trophy, never had an issue
17:21.54QwellTrophy is a fantastic name for a paperweight.
17:22.02jmetrosamsungs make the best mobile devices on the market
17:22.07Snivetsso, I've been telling my nerd friends that the Nokias, pre-MS takeover, were intended to run the full on linux, you know? And I keep messing up the name and calling it "sybian" instead of "symbian"
17:22.11coppiceI wonder if nokia will be able to sell phones for more than $200 now
17:22.13jmetrobut i dont think there are any samsung windows pones.
17:22.19leifmadsenI'm not sure what this flametopic has to do with asterisk
17:22.27SanonERROR[6255]: chan_iax2.c:5003 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address "0.0.0.0" in the calltokenoptional list or setting user "USER" requirecalltoken=no
17:22.31leifmadsenpeople use phones. it's true.
17:22.32coppicejmetro: there are some
17:22.33leifmadsenend of story.
17:22.43*** join/#asterisk serafie (~erin@nat/digium/x-ygaosnwfmkayrixs)
17:22.48SanonTHIS IS WHAT I GOT THIS TIME
17:23.03SanonERROR[6255]: chan_iax2.c:5003 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address "0.0.0.0" in the calltokenoptional list or setting user "USER" requirecalltoken=no
17:23.14QwellSanon: okay, and?
17:23.20SnivetsActually, it was going to be related:
17:23.26SnivetsI was wondering if there are any good mobile sip clients
17:23.27jmetroyoure using iax instead of sip?
17:23.39Sanonusing iax
17:23.51Sanonsflphone
17:24.09QwellSanon: And what happened when you did what the error message told you to do?
17:24.31*** join/#asterisk oej_ (~olle@2001:16d8:cc57:1000::42:1005)
17:24.47jmetroneed to edit iax.conf, yes? not sip.conf?
17:24.52SanonI tried to register my sflphone to one of my users in ixa.conf
17:24.57Sanonand i got that
17:25.00QwellSanon: And what happened when you did what the error message told you to do?
17:25.16[TK]D-Fender<Qwell> Sanon: And what happened when you did what the error message told you to do?
17:25.59Sanonhow do i do that
17:26.05Sanonis what i'm asking
17:26.19navaismo<PROTECTED>
17:27.25[TK]D-FenderSanon, it just GAVE you the exact parameters to put in your iax.conf peer entry.  What do you mean "how"?
17:27.33[TK]D-FenderSanon, PUT THOSE INTO YOUR PEER
17:27.47Sanongot it.
17:29.01Sanonthanks guys, for being patient and helping me
17:29.07Sanonit's working
17:30.30slav3_kittenson of a bitch...
17:31.05slav3_kittengot a filling a week ago, eating cold pizza pulled it out.
17:31.14*** join/#asterisk chris_n (~Chris@184.7.21.42)
17:31.46slav3_kitteni'm going to be getting a free filling
17:34.32n3hxsSounds like you set the first one free, slav3_kitten
17:35.18[TK]D-Fenderslav3_kitten, Go for Boston creme.....
17:36.38Snivetsthat is unfortunate
17:37.07*** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1005)
17:38.16jmetroget the ceramic fillings, they dont come out. I cant even remember what tooth I got that in.
17:38.26Qwelljmetro: maybe it came out
17:39.30[TK]D-FenderDon't ask, don't tell
17:39.38Qwell[TK]D-Fender: was abolished
17:39.45slav3_kittenjmetro, i'll check itout
17:39.58[TK]D-FenderQwell, Not for FILLINGS yet... progress is a slow wheel
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17:53.01*** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net)
17:53.57jeffspeffanybody having any issues with confbridge causing an error making asterisk restart?
17:53.59*** join/#asterisk navaismo (~navaismo@189.144.200.37)
17:54.22jeffspeffjust a moment ago, i did confbridge list
17:54.33jeffspeffit showed 2 bridges open
17:54.50jeffspeffi did confbridge list bridge1 and it showed a few users in the bridge
17:55.05jeffspeffi did confbridge list bridge2 and asterisk died
17:55.37QwellDid you search JIRA?
17:56.12jeffspeffno
17:56.14jeffspeffi'll go there now
17:57.22mjordanjeffspeff: what version?
17:57.27jeffspeff11.0
17:57.50mjordankk.  If you can reproduce it and get a backtrace, please open a new issue.  I'm pretty sure I haven't seen any crashes reported from CLI commands interacting with ConfBridge
17:58.00mjordana quick peek in Jira would be appreciated however :-)
17:58.52ghost75i had once zement filling for like 15 years
17:58.53jeffspeffyes, looking in jira now
17:59.04jeffspeffit's a production box, so i'll try and replicate it on a test box
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18:38.09oneadventhi i'm trying to do a phpagi script but my log says: Starting SIP/vitel-outbound-00000023 at makeCall,s,1 failed so falling back to exten 's'
18:38.23oneadventdoes that mean it cannot find makeCall context?
18:38.32oneadvent(generated by call files)
18:38.50[TK]D-FenderIt cannot find a match for that exten in that context at priority 1.
18:38.53[TK]D-Fenderjsut like it says
18:40.12oneadvent[groupCall]
18:40.12oneadventexten => s,1,Answer
18:40.12oneadventexten => s,2,AGI(makeCall.php)
18:40.13oneadventexten => s,3,Hangup
18:40.19oneadventis at the bottom of extensions.conf
18:40.29oneadventi'm not sure what else i need for it :(
18:40.48*** part/#asterisk ipiera (~Paul@ipiera.plus.com)
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18:46.56oneadventanywhere i should read up on this [TK]D-Fender ? I'm lost, i have everything in the agi-php directory and the call file calls out...it is like it can't find my script but the error says it can't find the context, and i'm not sure how to make it find it anymore than putting it in extensions.conf.
18:47.09slav3_kittenok maybe i missed it in the book, but when having all phones dial *7 to get to voicemail main. how do i pull the caller id number out to use for automatically throwing them into the correct mailbox?
18:47.19[TK]D-Fenderoneadvent, PASTEBIN your dialplan and call file....
18:47.35slav3_kittenwait...
18:47.39[TK]D-Fender<oneadvent> hi i'm trying to do a phpagi script but my log says: Starting SIP/vitel-outbound-00000023 at makeCall,s,1 failed so falling back to exten 's'
18:47.43slav3_kittennvrmind
18:47.44[TK]D-Fender<oneadvent> [groupCall]
18:47.59[TK]D-FendermakeCall != groupCall
18:48.15slav3_kitteni had it right, apparently my saving of my voicemail password didn't work
18:48.57oneadventomg if that was it [TK]D-Fender....
18:49.05[TK]D-Fenderif?
18:49.09oneadventlol
18:49.20oneadventwell i fixed that but i still just get hung up on :(
18:49.34[TK]D-Fenderok/fine/sure
18:51.07oneadventhttp://paste2.org/p/2486964
18:51.08oneadventthere
18:51.11oneadventthat is the pastebin
18:51.24oneadventprobably me doing something stupid with phpagi
18:51.31*** join/#asterisk Galen (~Galen@rrcs-24-43-17-235.west.biz.rr.com)
18:53.46slav3_kittenhelps to have proper file owner/group
18:55.21[TK]D-FenderAlso helps to enable AGI debug
18:55.41[TK]D-FenderAlso helps to show us that the files are even where * is trying to call them from...
18:55.54[TK]D-Fender...with the right permissions...
18:57.27oneadventslav3_kitten: that was the problem
18:57.33oneadventthanks [TK]D-Fender and slav3_kitten
18:57.48*** join/#asterisk luckman212 (~luckman21@2001:470:8abb:0:211:32ff:fe10:cdc1)
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18:58.39slav3_kittenwait it was file permissions oneadvent ?
18:59.04oneadventyes sir slav3_kitten i had it owned by root and it should have been asterisk:asterisk, it didn't even occur to me to check
18:59.59slav3_kittenholy crap i inadvertently solved that
19:00.11slav3_kittenthat was the solution to my voicemail password not getting changed
19:00.12oneadventlol slav3_kitten good job doing it though!
19:00.45slav3_kittenhuzzah
19:04.50*** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e)
19:06.35*** join/#asterisk anonymouz666 (~anonymouz@189-25-41-221.user.veloxzone.com.br)
19:21.29*** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net)
19:22.15AkkerKidAnyone know how to pass channel variables to hylafax?
19:22.41AkkerKidOr at least where Hylafax sources it's callerID info...
19:27.49*** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net)
19:32.55p3nguinits
19:33.59AkkerKidCorrention: Or at least where Hylafax sources it is callerID info...
19:34.05AkkerKid:)
19:34.21jmetrocorrection
19:34.52AkkerKidCorrectshun: Correction
19:35.10AkkerKidSo i guess thats a no then?
19:35.30*** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell)
19:35.30*** mode/#asterisk [+o Qwell] by ChanServ
19:38.52ghost75hmm gosub is not a own context?
19:39.11slav3_kittenhuzzah, voicemail setup and mwi correctly triggered
19:41.19[TK]D-FenderAkkerKid, you channel has its callerid
19:41.35[TK]D-Fenderyour*
19:42.27*** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch)
19:44.35ghost75is this valid under 1.6: Gosub(bla,s,1)
19:47.25Qwell~upgrade asterisk
19:47.25infobotBefore requesting assistance, you should be running the latest version of a supported release branch.  See the channel topic for the latest versions available in currently supported branches.
19:47.42ghost75haha
19:48.28jmetroslav3_kitten the hard part is getting blf's to work.
19:48.39dijibslav3_kitten: now tell me how to disable mwi
19:49.11dijibhey p3nguin you would be proud of me I pretty much did an entire * deployment for a guy last night
19:49.25dijib5/4 in the bag to boot
19:50.25ghost75in incoming context i use gosub and when sub is finished i get: Auto fallthrough, channel 'SIP/arcor_in-00000004' status is 'UNKNOWN'
19:50.25[TK]D-Fenderghost75, Does it work?
19:50.40ghost75wtf
19:50.49[TK]D-Fenderghost75, Why would we assume that has anything to do with your Gosub?
19:51.10ghost75i changed from macro to gosub
19:51.35[TK]D-Fenderghost75, You are asking for an autopsy without providing the body.....
19:51.41ghost75one sec
19:54.35ghost75http://pastebin.com/hTvaiFvC
19:54.46ghost75do i need another gosub from spamcheck to incoming arcor?
20:03.36*** join/#asterisk k610 (~Instantbi@host-78-129-3-116.brutele.be)
20:11.06[TK]D-Fenderghost75, .... and the full call....
20:12.07[TK]D-Fenderghost75, And you are Goto-ing OUT of that context.  I also see no use of the retun app
20:12.14[TK]D-Fenderreturn
20:15.28ghost75how i return to incoming-arcor
20:16.04[TK]D-Fenderghost75, You are goto-ing into no-mans-land, we do not see the full picture.
20:16.15[TK]D-Fenderghost75, Your layout is messay and warrants a redesign
20:16.25ghost75ok one sec
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20:18.59ghost75http://pastebin.com/8QKHqy80
20:20.18ghost75http://pastebin.com/a7bdaEL5
20:21.03[TK]D-Fenderhttp://www.voip-info.org/wiki/view/Asterisk+cmd+Return
20:23.25ghost75thx
20:29.28AkkerKidIf I move a fax call through asterisk into hylafax and change the callerid variables, the hylafax still sees the original CID number...   How can I inject MY CID info?
20:30.53[TK]D-FenderAkkerKid, Show us
20:34.14AkkerKidRight before I put a fax call into a ring group full of virtual fax machines, I Set(CallerID(number)=${CALLERID(number)}.${LocationCode})
20:35.06AkkerKidbasically, this should addend the LocationCode  (which is a variable I set based on the did a call comes in on) to the end of the callerid number.
20:35.39[TK]D-FenderAkkerKid,  Set(CallerID(number)=${CALLERID(number)}.${LocationCode}) <-- you should be setting the function here.....
20:35.44AkkerKidbut when faxrcvd.conf gets the callerid variables on the other side of hylafax, I get the original CID number
20:35.56[TK]D-FenderAkkerKid, But seem to have forgotten that it has to be UPPERCASE
20:36.13AkkerKidfunction?
20:36.24[TK]D-FenderSet(CallerID(number) <-------------------------
20:36.30[TK]D-FenderYour set is bad.
20:36.35AkkerKidhow so?
20:36.41[TK]D-FenderHAS TO BE UPPERCASE
20:36.51AkkerKidI disagree.
20:37.33AkkerKidI have success with CallerID(name) in lowercase.  Why should CallerID(number) be any different?
20:37.55[TK]D-Fender.....
20:38.45AkkerKidI'm certain my case is not the issue.  I have a few instances of the variable name formatted exactly like that and it works fine.
20:38.47[TK]D-FenderYou are wrong.
20:39.05AkkerKidI'm just thinking the hylafax gets its CID info from some other variable.
20:39.15AkkerKidI'll try it.
20:39.33[TK]D-FenderThere is no fact to back up your "thought".
20:40.14AkkerKidnone beside the fact that other instances of that variable work fine.
20:40.25[TK]D-FenderI'm not seeing backup.
20:40.46AkkerKidBRB testing.
20:45.21*** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net)
20:45.57ghost75there is an predefined extension "fax" in asterisk?
20:48.36*** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net)
20:50.50AkkerKidthats another thing i'm wondering about...  sometimes i'll get a call into my main call sorting code from exten "fax"...
20:51.00AkkerKidno other CID info
20:51.24ghost75then maybe someone calls you with a t.38 fax
20:52.28AkkerKidi can't even trace the call to see who it may have come from
20:52.58AkkerKidi was worried that that maybe some rogue function was intercepting my outbound faxes and redirecting them back into the dialplan...
20:53.12AkkerKidbut i haven't seen hard evidence of that
20:53.31ghost75no CALLERID ?
20:53.37AkkerKidnil.
20:54.04AkkerKid[TK]D-fender: no change between cases.
20:56.12*** join/#asterisk vfabi (~fabi@host-static-89-41-121-42.moldtelecom.md)
20:57.43AkkerKidthe no callerid on inbound call from "fax" thing may have something to do with elastix rearing it's ugly head.
21:03.07ghost75exten => fax,n,System('/usr/bin/fax2mail ${CALLERIDNUM} "${CALLERIDNAME}" FaxNum RecipName email@address.com ${FAXFILENOEXT} p')
21:03.14ghost75i couldnt find any fax2mail on linux
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21:09.52*** join/#asterisk mdg (6c5df7d9@gateway/web/freenode/ip.108.93.247.217)
21:10.25mdgGreetings fellow telephony enthusiasts, is there a way to have a pattern matching extension of variable length ?
21:10.57mdgI know you can use !, but that would not work in coordination with "enter the amount followed by the pound sign"
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21:12.57artyxmdg . i think
21:13.25artyxX. for a dialplan could be any # of digits .. but its generally a bad idea
21:14.25mdgartyx: ah yea, forgot about that from the book.  thanks for the reminder
21:14.57artyxWorsst case you could try a fixed 5 digit #, and say in the script "prefix with 0's , for 100 use 00100" tacky but
21:16.17mdgtrue
21:16.32Qwellerr
21:16.37QwellWhy not use Read?
21:17.09mdgQwell: because Cepstral 6 wants 200 bucks to generate sound files :-/
21:18.07QwellHow is using Read any different from what you're doing?
21:18.33artyxi didnt realize he was talking ivr until i scrolle ddown
21:19.03mdgartyx: I wasnt clear enough in my original question, sorry
21:19.16mdgQwell: Im not sure I understand?
21:19.52mdgOn Cepstral 5, I could generate the wav files and use Read, but my hand has been forced in upgrading an IVR to use Cepstral 6
21:22.16AkkerKidwhat's wrong with _X.#
21:22.40AkkerKidthat would at least require a single digit followed by a pound, wouldn't it?
21:23.42mdgAkkerKid: yea.. I just tested it and it is working the same as _X. - Both work fine, but dont seem to go to the extension untill the swift command times out
21:23.57mdgwhich im about to confirm (was just counting in my head)
21:24.28ghost75does $NumberCalled still exist in h context ?
21:24.31*** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se)
21:24.47Qwellghost75: What?  Such a thing never existed.
21:25.21ghost75CALLERID(num)
21:25.31ghost75no wait
21:25.47ghost75forget it
21:26.43AkkerKidwhat about _X.# followed by something the takes the pount off the end like ${Exten:0:-1}
21:29.34mdgAkkerKid: thats not the issue (although I still need to do that), I just confirmed that the dialplan wont goto _X.# extension until my swift call times out (i have it set for 10 seconds)
21:29.52mdgSo it does work.. but not ideal having a 10 second delay after hitting #
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21:31.28AkkerKidoh i see.
21:33.22mdgnot sure... but it looks like i can modify https://github.com/awayment/app_swift/blob/master/Asterisk_1.8/app_swift.c#L261 to return if '#' is encountered.. since the '#' is always terminating on this ivr
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21:34.20mdgwonders if cepstral did that on purpose
21:35.26artyxHow do you turn a phone number into an address?
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21:38.38slav3_kittenif i'm not mistaken if i have a variable with 04 in it an i go VAR:0:1 i'll get 0 an :1:1 i'll get 4 right?
21:39.58mdgha... it works
21:45.38*** join/#asterisk crienzo (~crienzo@66-87-30-15.pools.spcsdns.net)
21:45.39leifmadsenslav3_kitten: VAR:<offset>:<length>
21:45.51mdghttps://github.com/awayment/app_swift/blob/master/Asterisk_1.8/app_swift.c#L278 right after the digit gets concatenated, check if it == '#' then break
21:45.54leifmadsenyou don't even need the second :1 in your 2nd example
21:46.08leifmadsensince your length is only 2, and an offset of 1 with no length specified will go to the end
21:49.19slav3_kittenisn't it better to specify a length however even if it's not needed in in the event somehow there is extra data?
21:52.47*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
21:54.32ghost75what is stored in /var/spool/asterisk/fax/${CALLERIDNUM}
21:54.40ghost75header from fax ?
21:56.49*** join/#asterisk Sean-Der (~sean@NW1-DSL-74-215-64-154.fuse.net)
21:56.54slav3_kittenghos i think calleridnum is depreciated a should be CALLERID(num)
21:57.01slav3_kittenghost75, *
21:57.10ghost75yes, i just copypasted it
21:57.17Sean-DerDoes anyone know of a service that does 'area code -> current time OR zip code'
21:57.37Sean-DerI know it won't be perfect, but seems like a cool thing to do
21:58.37slav3_kittenSean-Der, you could make a database with area code, and local time . then search it?
21:59.43ghost75http://www.voip-info.org/wiki/view/Asterisk+fax#SpanDSPSendingandReceivingFaxeswithAster
22:00.08Sean-Derslav3_kitten: Yea that would be a lot smarter, just store it in SQLite. Hell it would be small enough I could put it the stack in what ever lang I use to process the data
22:01.23slav3_kittenstore local time as an offset to UTC
22:01.29slav3_kittenit'd work well
22:04.49ghost75dahdi is for t.38 ?
22:05.34*** join/#asterisk navaismo (~navaismo@187.187.96.51)
22:05.46Sean-DerOk thanks for the advice! Is there an easy way to change the time in Asterisk per call thread? Or can I only go off server time
22:08.35slav3_kittennow that's a question i have no idea about
22:14.43ghost75why so many commands are deprecated i dont get it
22:15.05Chainsawghost75: This is to generate demand for consultancy and sustain the Asterisk eco-system.
22:15.52Sean-DerJust wait till 11 comes out.... Dialplan can only be written in whitespace
22:16.46ghost75lol
22:16.55slav3_kittenSean-Der, you could set the caller name to be like LocalTime: hh:mm
22:18.21slav3_kitteni have it set the name of my friends based on their number
22:19.00Sean-Derslav3_kitten: What I am thinking of doing is pipeing out to a program to returns a var of the users current 24 hour time, then (if $time < 700 Good morning)
22:19.20slav3_kittenah, that's cool
22:19.32slav3_kitteni have a shit auto attendant for my system
22:20.07slav3_kitteni might recode it so the extensions like parents room an such are disabled past certain hours
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22:23.43Sean-DerI actually have something like you are describing! Teachers set 'no-disturb' hours in a web
22:23.48Sean-Derinterface
22:24.22slav3_kittenmy interface is the old busted ass network hardware interface
22:24.38slav3_kittenand i'm the admin of it so only i change things
22:25.01Sean-DerHahha keep it that way, supreme ruler is the best way to be
22:25.45slav3_kittenyea my equipment is all old cico
22:25.46slav3_kittencisco even
22:26.59Sean-DerAt least you have nice hardware! I know everyone makes duds, but you always hear good things about Cisco. I am more of a software developer than a networking guy though :( I am proud of myself that I got my routing set up for OpenVPN
22:27.43slav3_kitten:D
22:27.54slav3_kittenyea it's decent stuff but all very old
22:28.03jmetrocisco stuff is okay.
22:28.07slav3_kittenlike i have a 7911 as the living room phone, it has 1 way speaker phone :|
22:28.34slav3_kittenno mic so yea that makes it useless, still got no idea how to transfer calls
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22:29.13slav3_kitteni have no idea how to do transfers actually
22:29.36Sean-DerAsterisk CLI only :D
22:30.07slav3_kittenyea don't even knowhow to do that
22:30.20slav3_kitteni'm sure it's in my giant book i've had a hard time getting through
22:31.02*** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone)
22:32.14Sean-Derthe 'Asterisk: The Definitive Guide' is a great book
22:33.07Sean-DerI enjoy flipping around in it, and learn random stuff from it all the time
22:33.17slav3_kittenit's like getting through Ulysses though
22:34.36Sean-DerIts not that big! I have actually never read Ulysses :/ I really should though
22:35.26slav3_kittenit's not bad, just not engaging
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22:48.54ghost75i need to allow anything in trunk if i want t.38 over sip?
22:49.44Alex25I'm an new here, so I have a newbie general question about codecs
22:50.31Alex25What's the point of allowing some codecs and disallowing others, or putting codecs in specific order?
22:50.35Alex25Won't it be better to allow all codecs by default for all channels, and let Asterisk chose the best one for each call?
22:50.54ghost75you tell asterisk the order
22:51.01Alex25Maybe you knwo some link which explains this point
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22:52.21jpsharpAsterisk doesn't know the "best" for a call.
22:52.35[TK]D-FenderAlex25: because maybe you want to prioritize ULAW for internal calls, but allow G.729 to be native bridged for outbound
22:52.51[TK]D-FenderAlex25: "best" is based on YOUR priorities.
22:52.54Alex25yea
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22:54.03Alex25so what if I allow all codecs?
22:54.40Alex25won't asterisk chose a good codec by itself?
22:54.52Alex25will the call fail?
22:55.10jpsharpIf you allow all codecs, Asterisk will choose the first one that it and the client can agree upon.
22:55.35Alex25and what's wrong with that?
22:56.05ghost75depends which codecs you want to use
22:56.13[TK]D-FenderAlex25: You keep assuming "good".  It will pick the FIRST thing it agrees on in order, which may not be good for YOU.
22:56.20jpsharpNothing, assuming you don't have a codec preference.
22:56.23[TK]D-Fender[17:52][TK]D-FenderAlex25: "best" is based on YOUR priorities.
22:56.33*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
22:58.02[TK]D-FenderAlex25: If you pick your owder wrong * could negotiate each side differently and in the case of licenced codecs like G.729, if you don't have any, or are full, then your call dies.
22:58.25jpsharpBut you can allow all and then someone comes along with a client that decides that it wants to present a SIP call and LPC10 is the first codec in its list, even though it supports ULAW, and your calls will sound like crap.
22:58.32[TK]D-Fender'aleStop thinking you can get away without configuring your system and that it'll all just work out.  You are asking for your system to bite you in the ass.
22:58.37[TK]D-FenderAlex25: ^
22:58.57Alex25so why people list a 'priority' for a peer, instead of chosing the BEST for them??
22:59.14[TK]D-FenderAlex25: best is BASED on your priorities.
22:59.15Alex25the best one, i mean
22:59.18[TK]D-FenderAlex25: YOUR <-------------
22:59.21Alex25instead of a list
23:00.57ghost75the best is the first you list
23:01.18Alex25ok
23:01.28Alex25i'm reading your comments
23:01.40Alex25thanks very much
23:02.11Alex25i wonder if you know some link which puts it simple
23:02.19[TK]D-FenderThis IS simple.
23:02.24[TK]D-FenderWhat's so hard to understand?
23:02.34[TK]D-FenderYOUR circumstances determine what is "best"
23:02.35ghost75just dont put codecs which you dont want, easy as that
23:02.40[TK]D-FenderThere is no formula.
23:02.43[TK]D-FenderThere is no magic.
23:02.47[TK]D-FenderIT DEPENDS ON YOU
23:02.54*** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir)
23:03.10Alex25sure
23:03.23[TK]D-FenderAlex25: If you have shitty bandwidth then if your calls all use high-BW codecs they'll CHOKE out and your calls will sound like crap and may just drop entirely.
23:03.33[TK]D-FenderSo the best quality will SCREW YOU
23:04.03[TK]D-FenderIf you have lots of BW, then maybe you can afford to use higher quality codecs.
23:04.38Alex25is g722 the best codec today for high bw?
23:04.41[TK]D-FenderIt also matters for things like CPU load when recording depending on the format you choose to write in.
23:04.56[TK]D-FenderAlex25: There are times when "best quality" don't even matter.
23:05.40[TK]D-FenderAlex25: Each codec also has a transcoding performane hit.  This will really add up when recording, or in a conference, etc
23:05.53[TK]D-FenderAlex25: So time to really look at what you're doing.
23:06.49*** part/#asterisk mjordan (~mjordan@nat/digium/x-spdhnzgaqhrkvxwe)
23:06.53Alex25sure. My system is up and working, I just want to optimize it with best codecs. so i must learn this terminilogy..
23:06.56ghost75t.38 goes over alaw/ulaw ?
23:07.20Chainsawghost75: No, normally you switch away *to* T.38 from alaw/ulaw.
23:07.46Chainsawghost75: You do need something that's clear enough to hear the fax CNG tone so you can renegotiate for T.38
23:07.58ghost75but i need to have alaw/ulaw in my trunk enabled then
23:08.49*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
23:09.21ghost75will this passthrough work also over nat ?
23:10.23[TK]D-Fenderghost75: Same as voice
23:12.45ghost75crap if i need to enable g711 in trunk then all calls go over g711 instead g729
23:21.05p3nguinDoesn't T.38 still use the exact same RTP stream that a voice call will use?
23:24.03ghost75http://what-when-how.com/voip/fax-over-ip-overview-voip/
23:24.36ghost75so no g.711 at all i guess
23:25.39ghost75asterisk has udptl
23:28.01*** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch)
23:35.12jpsharpT38 does not use a voice codec.
23:35.24p3nguinWe covered that.
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