| 00:00.40 | Strom_M | it takes about three months |
| 00:00.44 | Strom_M | sit tight |
| 00:00.48 | Strom_M | grab a beer |
| 00:00.54 | hardwire | mmbeer |
| 00:00.57 | hardwire | and a brat |
| 00:01.00 | teknoprep | lol |
| 00:01.04 | teknoprep | yeah 3months |
| 00:01.06 | teknoprep | not too long |
| 00:01.35 | teknoprep | i have it inside of vmware ... the vmware client is running 2cpu 2ghz opteron with 1gb ram |
| 00:01.43 | hardwire | distcc anyone? |
| 00:02.28 | *** join/#asterisk moy (n=moy@189.169.88.136) |
| 00:03.50 | teknoprep | sooooooo |
| 00:03.59 | teknoprep | any idea on time to complete lol? |
| 00:04.41 | teknoprep | oh wow |
| 00:04.43 | teknoprep | already done |
| 00:04.59 | TJNII | teknoprep: It should complete around now. |
| 00:05.07 | teknoprep | HAHAHA |
| 00:06.50 | *** part/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 00:07.10 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 00:08.58 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 00:11.28 | mandd | I set callerid="test" in sip.conf in [general], but it is not working |
| 00:11.31 | mandd | using 2 x sip phones |
| 00:11.33 | mandd | anything i am missing? |
| 00:13.26 | teknoprep | lol |
| 00:13.32 | teknoprep | getting seg faults on 1.6-beta9 |
| 00:13.33 | teknoprep | w0ot |
| 00:19.27 | *** join/#asterisk moy (n=moy@189.169.88.136) |
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| 01:17.14 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-238.hsd1.ut.comcast.net) |
| 01:17.47 | [T]ank | anyone looking for a t1 card? I have a few i am looking to unload. $500 |
| 01:18.00 | [T]ank | used sangoma a104d's |
| 01:18.02 | JT | but i can buy new ones for less than that |
| 01:18.05 | JT | ooh |
| 01:18.08 | JT | 4 ports |
| 01:18.16 | [T]ank | yes 4 port |
| 01:18.49 | [T]ank | works fine... just gone to an all SIP system |
| 01:19.12 | [T]ank | its a few months old, but have always been in a data center. so they have been clean and cool the whole time |
| 01:19.42 | [T]ank | interested? |
| 01:19.58 | JT | i could be |
| 01:20.16 | [T]ank | move to pm? |
| 01:20.45 | lucky|aba | Why would one computer be able to call me but I not be able to call him? |
| 01:22.54 | TJNII | represses the urge to answer "because you touch yourself at night" |
| 01:23.02 | TJNII | What do you mean "can't call him" |
| 01:27.52 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 01:29.07 | obnauticus | Err i have an MP3 track, and it's a VBR. How do I tell if it's a V0, V1, V2, or V3 ? |
| 01:29.17 | obnauticus | I'm trying to get it working with this * plugin :\ |
| 01:30.18 | obnauticus | JT don't buy it |
| 01:30.20 | obnauticus | it's stolen! |
| 01:30.23 | obnauticus | and make sure it's not only the box |
| 01:30.30 | obnauticus | I did that to a couple on ebay with the xbox 360 |
| 01:32.07 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:38.11 | Strom_M | http://www.theonion.com/content/news_briefs/u_s_intelligence_burundi |
| 01:40.10 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 01:44.42 | *** join/#asterisk emiller (n=ed@65.208.79.2) |
| 01:46.16 | jblack | In light of Iran, that's pretty nice, strom |
| 01:46.30 | jaytee | lol |
| 01:46.42 | emiller | anyone have any luck installing asterisk on os x? :) |
| 01:46.57 | [T]ank | emiller: have seen it done successfully. |
| 01:47.14 | jblack | osx is for ipods, not phone systems. |
| 01:47.21 | emiller | is taking a stab at it |
| 01:47.36 | emiller | ha, well, i just want to get some practice setting it up |
| 01:47.56 | emiller | downloading xcode 3.0 right now |
| 01:48.05 | Strom_M | osx is there to take perfectly good UNIX and add a whole new layer of smarmy and smug on top |
| 01:48.13 | jaytee | I did but it insisted on calling itself iAsterisk and wearing black turtlenecks, the only MOH that would play was Rick Astley's Never gonna give you up! |
| 01:48.19 | jblack | We'll let you know when there's an iAsterisk |
| 01:48.31 | Strom_M | we're no strangers to phones |
| 01:48.37 | jaytee | hehe |
| 01:48.37 | emiller | or love? |
| 01:48.38 | Strom_M | you know the dialplan, and so do I |
| 01:48.38 | Strom_M | etc |
| 01:48.43 | jaytee | hahaha |
| 01:48.47 | emiller | hahaa |
| 01:48.56 | Strom_M | never gonna hang you up / never gonna tear you down |
| 01:49.00 | emiller | im taking the dcap exam tomorrow :\ |
| 01:49.05 | Strom_M | never gonna dial around |
| 01:49.12 | jaytee | emiller, best of luck |
| 01:49.16 | emiller | gracias |
| 01:49.39 | TJNII | ~rickroll |
| 01:49.40 | jbot | i guess rickroll is http://www.xkcd.com/396/ |
| 01:49.53 | jaytee | I'd love to take the exam but my boss is so frigging cheap he won't let me take the bootcamp class. |
| 01:50.02 | TJNII | Oh.... I thought that pointed to some nastyp page with the lyrics in javascript popups. |
| 01:50.03 | emiller | ahh, its actually really good |
| 01:50.04 | Strom_M | I've got all of you beat. I have two copies of the 12" single of that song |
| 01:50.12 | emiller | jaytee: im on my last day tomorrow |
| 01:52.38 | emiller | the digium headquarters is actually very nice |
| 01:58.11 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 02:01.09 | *** join/#asterisk emiller (n=ed@65.208.79.2) |
| 02:01.20 | emiller | ugh. stupid hotel wireless. |
| 02:08.03 | jaytee | so I've tried all kinds of things and I still can't find a way to prevent a specific sip phone or extension from being able to transfer a call externally. |
| 02:09.03 | *** join/#asterisk edwin_quijada (n=macaruch@tdev253-154.codetel.net.do) |
| 02:09.09 | edwin_quijada | hi |
| 02:09.20 | edwin_quijada | asterisk can play .wav files into AGI? |
| 02:09.21 | jaytee | hey |
| 02:11.23 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 02:11.26 | UnixDog | agi is a script lang |
| 02:11.28 | emiller | jaytee: can't you try something like exten => s,n,GotoIf($${EXTEN} = 6002?t,disconnect) |
| 02:11.39 | UnixDog | based on php or perl or python |
| 02:11.40 | emiller | sans the extra $ |
| 02:11.40 | edwin_quijada | UnixDog:perl |
| 02:12.11 | UnixDog | but you can use it to play a wav/gsm/ulaw/ what ever file |
| 02:12.18 | edwin_quijada | i am using $AGI->stream-file($file) |
| 02:12.30 | edwin_quijada | but cant play anything |
| 02:12.35 | jeev | has 20 polycom 330's coming in... |
| 02:12.37 | jeev | can't wait! |
| 02:12.57 | jaytee | emiller, I'm using Polycom phones and the softkey feature in the phone doesn't use *'s transfer so I can't intercept the redirect. |
| 02:13.07 | UnixDog | 330 are nice |
| 02:13.14 | jeev | i'm just freaked out with my config |
| 02:13.14 | UnixDog | cheap and easy to configure |
| 02:13.19 | jeev | i need to be able to transfer calls and i dunno how yet |
| 02:13.20 | UnixDog | why |
| 02:13.22 | jeev | i'm only using 1 to test |
| 02:13.32 | emiller | jaytee: ahh, i see what you mean now. that can be a bit tricky. |
| 02:13.35 | jaytee | jeev, transfer is easy with the Polycom |
| 02:13.39 | jeev | really ? |
| 02:13.42 | jaytee | blocking it is near impossible |
| 02:13.45 | edwin_quijada | i need to use a speific code to play |
| 02:13.46 | edwin_quijada | ? |
| 02:13.53 | jaytee | jeev, when you're on a call press the transfer softkey |
| 02:13.54 | UnixDog | I love my 550 |
| 02:14.00 | jeev | oh that's it ? |
| 02:14.04 | jeev | i have nothing to test with |
| 02:14.06 | jaytee | I have a 501 and a 550 I haven't setup yet |
| 02:14.08 | emiller | jeev: did you take a look at the sample config on voip-info |
| 02:14.08 | jeev | jaytee, does it need extra config ? |
| 02:14.13 | jeev | not yet emiller. |
| 02:14.13 | jaytee | jeev, no |
| 02:14.17 | jeev | damn dood |
| 02:14.26 | jaytee | not for simple transfers or conferencing |
| 02:14.27 | emiller | http://www.voip-info.org/wiki/index.php?page_id=1056&tk=4cf8c58bf7f14452a0f5&comments_page=1 |
| 02:14.29 | jeev | i love the fact, i can create a conference room on the fly and password protect it. |
| 02:14.31 | UnixDog | 330 is a very basic polycom |
| 02:14.42 | UnixDog | yeah |
| 02:14.54 | jaytee | the audio quality of the 330 is superb. |
| 02:14.55 | emiller | that is the exact config that we built in the asterisk boot camp training |
| 02:15.28 | UnixDog | get a 550 and stand back |
| 02:15.42 | jeev | eh, i'm fine with 330's. |
| 02:16.02 | jeev | i cant freaky dicky wait |
| 02:16.08 | emiller | haha |
| 02:17.36 | jeev | i'm gonna put asterisk on an E8300 box.. |
| 02:17.46 | jeev | should be fast enough to provide routing, asterisk and dansguardian with squid |
| 02:17.52 | jeev | with dual wan |
| 02:19.31 | hardwire | wanfu |
| 02:19.34 | hardwire | you haz it |
| 02:20.02 | hardwire | that's a lot of high priority processes for just about any box however |
| 02:20.12 | jeev | wanfu ? |
| 02:20.14 | hardwire | you should run the newer linux-rt kernels on it and prioritize the hell out of asterisk |
| 02:20.16 | hardwire | wan fu |
| 02:20.18 | hardwire | like kung fu |
| 02:20.22 | jeev | haha |
| 02:20.25 | jeev | i'm gonna run pfsense with bsd |
| 02:20.59 | hardwire | the asterisk box won't have two wan interfaces? |
| 02:21.05 | jeev | yea it will |
| 02:21.11 | hardwire | you're running asterisk on BSD? |
| 02:21.14 | jeev | yea |
| 02:21.21 | hardwire | oh |
| 02:21.24 | hardwire | you haz a good day now.. |
| 02:21.28 | jeev | lol |
| 02:21.30 | hardwire | has to go home anyways. |
| 02:21.34 | hardwire | tootles |
| 02:21.39 | jeev | toodaloo |
| 02:21.47 | hardwire | toodly tiddly tot |
| 02:21.51 | jeev | lol |
| 02:22.10 | tristanbob | is the TE122B the same thing as the TE122P + echo cancellation? |
| 02:23.21 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:23.22 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:25.47 | tristanbob | russellb, is the TE122B the same thing as the TE122P + echo cancellation? |
| 02:26.18 | edwin_quijada | to use audio files should be gsm? |
| 02:28.02 | russellb | tristanbob: i think it's just called the TE122 |
| 02:28.08 | russellb | so yeah, same thing i guess |
| 02:29.13 | tristanbob | russellb, thanks - I'm creating a quote for a customer |
| 02:30.36 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-86-23.vif.net) |
| 02:31.28 | pcrane | emiller: which version of asterisk? I've tried 1.6, but all I get is something about popping blanks |
| 02:32.03 | emiller | pcrane: im sorry, are you referring to my os x question? |
| 02:32.10 | pcrane | yeop |
| 02:32.38 | emiller | i actually havent tried it yet. i am/was trying to download xcode for the gcc compiler, but this hotel connection is too slow |
| 02:33.00 | pcrane | lol that part is quite big ;) |
| 02:33.10 | flush | yo people i know its kinda off topic but any of you knows about very silent fans for pc's |
| 02:33.13 | emiller | :( i wasnt aware it was so huge |
| 02:33.28 | emiller | i wanted to set up a mini pbx in my room prior to my dcap exam tomorrow |
| 02:35.10 | russellb | emiller: what version of 1.6? is it older code? |
| 02:36.10 | russellb | emiller: that problem should be fixed in the latest code .... |
| 02:36.37 | emiller | well, im not really having a problem; unless you consider my slow bandwidth |
| 02:36.41 | russellb | but yeah, downloading vmware and a linux ISO to install in a virtual machine is a smaller download than downloding xtools :) |
| 02:36.48 | emiller | haha |
| 02:36.50 | russellb | i meant the "popping blanks" thing |
| 02:37.00 | emiller | ohh, that was pcrane |
| 02:37.08 | russellb | oh, sorry |
| 02:37.11 | emiller | np |
| 02:37.15 | russellb | pcrane: the popping blanks thing should be fixed. |
| 02:37.24 | russellb | in the latest code that is |
| 02:37.27 | pcrane | ok |
| 02:37.35 | pcrane | I'll have a look at that |
| 02:37.53 | russellb | you may have to try just from svn |
| 02:37.54 | emiller | can you guys point me to a centos iso? i have parallels on my mac. |
| 02:38.05 | russellb | google :-p |
| 02:38.12 | emiller | :P |
| 02:38.28 | russellb | if all you want to do is compile asterisk and run it, then you might want to download something more minimal |
| 02:40.38 | emiller | im sorry, ive only ever installed and run asterisk on centos/red hat distros |
| 02:40.41 | emiller | what do you suggest |
| 02:40.47 | mmlj4 | phoniceq T1 cards... garbage? |
| 02:41.20 | russellb | shrugs |
| 02:41.24 | russellb | then that's fine |
| 02:41.31 | russellb | was just thinking since you had download pipe issues |
| 02:41.33 | pcrane | emiller: we use debian here |
| 02:41.44 | russellb | a debian minimal install would get you going |
| 02:41.51 | pcrane | mmm |
| 02:41.53 | emiller | thanks for the info guys |
| 02:41.55 | pcrane | easy as pie |
| 02:42.20 | russellb | how do you like parallels? i use vmware fusion .. |
| 02:43.02 | emiller | its not bad. i rarely use it. |
| 02:43.20 | emiller | i was thinking about checking out vmware fusion |
| 02:44.23 | russellb | i like it a lot, but it's all i have used |
| 02:44.43 | russellb | i like doing all of my development in linux, so i have it constantly running, heh |
| 02:45.02 | emiller | heh, i hate trying to get web stuff to work for ie |
| 02:45.06 | pcrane | parallels is cool with the coherence mode |
| 02:45.12 | UnixDog | 1.6 is a pain in the ass |
| 02:46.23 | russellb | UnixDog: why? |
| 02:47.59 | russellb | falls asleep waiting for an answer |
| 02:48.08 | UnixDog | because it does no play nice on bsd |
| 02:48.21 | russellb | rolls eyes |
| 02:48.22 | UnixDog | it fails to compile properly and crashes |
| 02:48.28 | russellb | sure it does, plenty of people use it on BSD |
| 02:48.36 | russellb | and a core asterisk contributor is also a FreeBSD developer |
| 02:48.43 | russellb | PEBKAC, i guess :) |
| 02:49.07 | UnixDog | well I compiled it today on a duron and it crashes after 3 min of running |
| 02:49.12 | pcrane | russellb: the popping blanks is solved ;) |
| 02:49.14 | pcrane | ta russellb |
| 02:49.20 | russellb | pcrane: you're welcome! |
| 02:51.42 | tzanger | pcrane: infertility problems? |
| 02:51.56 | pcrane | heh |
| 02:52.04 | russellb | tzanger: it was a debug message that we thought would never actually show up :-p |
| 02:52.22 | russellb | but, bug in the code, and people's macs were poppin' blanks all over the place |
| 02:52.27 | UnixDog | russellb: I have freebsd 7.0 fully updated and everything built fromports all the deps |
| 02:52.38 | UnixDog | aI am not a newb at this |
| 02:52.39 | tzanger | russellb: heh |
| 02:52.53 | russellb | UnixDog: then please file a bug report. |
| 02:54.12 | *** part/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 02:55.51 | pcrane | it doesn't seem to respond to commands |
| 02:56.47 | pcrane | core set verbose 100 doesn't give me any output... in 1.4 I'd expect: Verbosity was 0 and is now 100 |
| 02:56.49 | pcrane | hmm... |
| 02:56.50 | pcrane | odd |
| 02:59.09 | ManxPower | pcrane: check /etc/asterisk/logger.conf (or is it logging.conf?) |
| 02:59.19 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 02:59.50 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:02.47 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 03:03.06 | pcrane | ManxPower: console is set to notice,war, and error, messages is set to the same |
| 03:03.10 | pcrane | stuff appears in the message |
| 03:03.15 | pcrane | but not in the console... |
| 03:03.29 | pcrane | it's not important at the moment, more than enough stuff to get on with ;) |
| 03:03.40 | ManxPower | but not verbost or debug |
| 03:03.45 | pcrane | no |
| 03:03.51 | pcrane | not verbose nor debug |
| 03:04.09 | ManxPower | perhaps since you are setting the verbosity higher, you might want to send verbose messages to the console? |
| 03:16.41 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:22.47 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-238.hsd1.ut.comcast.net) |
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| 03:33.38 | mmlj4 | fwiw, i just bought 2 of the old-style tormenta T1 cards, $180 each |
| 03:33.57 | mmlj4 | we'll see how much I regret it |
| 03:40.25 | x86 | a lot ;) |
| 03:47.00 | *** join/#asterisk dmz (n=dmz@216.16.220.180.dyn-cm-pool48.pool.hargray.net) |
| 03:53.47 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 03:54.15 | mmlj4 | they're for my lab, anyhow |
| 04:03.32 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
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| 04:36.20 | [T]ank | anyone know a good place besides ebay to sell used voip equipment? |
| 04:37.14 | outtolunc | astricon <G> |
| 04:38.10 | [T]ank | is it just that nobody uses T1 cards anymore? |
| 04:38.17 | [T]ank | I am having a hard time getting rid of them. |
| 04:38.46 | outtolunc | what kind? |
| 04:38.59 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 04:39.01 | [T]ank | sangoma a104d |
| 04:40.03 | outtolunc | you would think someone would want that, what you asking for it? |
| 04:40.14 | [T]ank | $500 |
| 04:40.45 | outtolunc | reasonable price |
| 04:40.57 | outtolunc | if i had the $ i'd buy it |
| 04:41.17 | outtolunc | still kickin myself for letting go of my t400p |
| 04:41.42 | [T]ank | may have one person who wants it. he... got one of those too. want it? |
| 04:41.56 | outtolunc | how much? |
| 04:42.18 | [T]ank | eh... well, the go for what, $1400? its actually never been used |
| 04:42.44 | outtolunc | hehe well if i can't afford $500, $1400 is way out of the question <G> |
| 04:42.56 | [T]ank | hehe, yeah |
| 04:43.07 | outtolunc | i'll let you know if i hear of anyone looking |
| 04:51.22 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
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| 04:56.03 | Kobaz | df |
| 04:59.27 | *** join/#asterisk MCooper (n=kmoore@64.110.169.173) |
| 05:02.11 | *** part/#asterisk moy (n=moy@189.169.88.136) |
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| 05:04.32 | NeoSkedar | I have users not on my lan connecting to my asterisk server, they can make and recieve calls fine, but if they try to check their voicemail, it always says invalid password |
| 05:05.24 | NeoSkedar | any ideas |
| 05:05.47 | *** join/#asterisk lucky|aba (n=lucky@ip68-6-98-146.sb.sd.cox.net) |
| 05:08.19 | jeev | so people on the lan |
| 05:08.22 | jeev | checking their voicemail, it works ? |
| 05:08.50 | [T]ank | NeoSkedar: share your extensions.conf line where you call VoicemailMain() |
| 05:14.32 | [T]ank | NeoSkedar: guess you got it figured out. you stopped responding |
| 05:14.39 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-238.hsd1.ut.comcast.net) |
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| 05:18.31 | Swabby | Is there a mp3 to wav converter that does u law wav files for free? |
| 05:21.38 | NeoSkedar | sorry was afk |
| 05:21.45 | NeoSkedar | yes all lan users can check vm fine |
| 05:28.19 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 05:28.23 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
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| 05:38.45 | NeoSkedar | sighs |
| 05:42.18 | NeoSkedar | missed my chance for help i guess |
| 05:46.18 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
| 05:47.09 | j0 | when i buy say 4 fxo ports, do i then need to get the telco to split their wires into 4 different jacks? |
| 05:58.08 | NeoSkedar | jeev, are u there |
| 05:59.09 | pcrane | NeoSkedar: check to make sure they can do DTMF properly, or that features aren't conflicting with IVR options |
| 06:00.17 | NeoSkedar | idk what any of that means |
| 06:00.41 | pcrane | DTMF is the tones that the buttons generate when you press them... |
| 06:01.16 | pcrane | if the DTMF isn't getting to the asterisk server (or something happens to corrupt them along the way) then * will not know how to process them (and say invalid password) |
| 06:02.13 | pcrane | if the features (e.g. # used to bxfer) is also used in the IVR system, * will interpret that as a transfer, not as a # to pass to the IVR |
| 06:02.37 | NeoSkedar | pcrane, http://slexy.org/view/s20mtGvOd1 |
| 06:02.47 | NeoSkedar | thats what * says when they try to get vm |
| 06:03.36 | pcrane | have a look online for vm_authenticate: Unable to read password |
| 06:03.58 | *** join/#asterisk Swat2 (n=bler@218-214-169-112.people.net.au) |
| 06:04.05 | Swat2 | Can anyone provide some kind of guidance with a voicemail problem. I have 2 messages waiting for me at the moment, so i dial *97 enter my password then it reads out the menu. I press 1 for recieving my mail and it starts with "Message received on" then it hangs up and doenst give me the message... Any ideas? |
| 06:04.34 | Swat2 | I can recieve my messages via the webpage. |
| 06:06.08 | *** join/#asterisk {sean} (n=sean@17.sub-75-214-40.myvzw.com) |
| 06:06.26 | {sean} | hey |
| 06:06.48 | NeoSkedar | pcrane, would the dtmf affect remote users and not lan users? |
| 06:07.22 | pcrane | possibly |
| 06:07.54 | pcrane | I'm thinking that there might be something odd happening with the connection |
| 06:08.01 | NeoSkedar | they can call fine |
| 06:08.13 | pcrane | hmm... |
| 06:09.08 | j0 | if a customer couldn't afford polycom phones, what would you recommend to them? |
| 06:09.14 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:11.11 | {sean} | anybody ever used a zultys pbx? |
| 06:11.33 | NeoSkedar | softphones |
| 06:11.36 | NeoSkedar | @ j0 |
| 06:11.42 | Kyoshi | does TDS and asterisk play nice together for realtime using MS SQL or will I need to use unixODBC? |
| 06:12.16 | j0 | NeoSkedar: yikes.. is that sarcasm or do softphones actually have a place in a business phone system? |
| 06:12.48 | pcrane | j0: we've used Linksys phones here, they're good |
| 06:13.01 | JT | j0: a lot of callcentres use cisco softphones |
| 06:13.16 | pcrane | never seen a physical polycom phone (have tried to remotely configure things -- that was fun) |
| 06:13.19 | pcrane | right |
| 06:13.22 | pcrane | hometime |
| 06:13.23 | pcrane | night all |
| 06:13.35 | Kyoshi | polycoms are very nice |
| 06:13.50 | Kyoshi | aastra's are nice too |
| 06:14.04 | j0 | thanks.. yeah the linksys ones don't look bad |
| 06:14.16 | j0 | after learning how to configure polycom's i'm not sure if i want to learn another. =D |
| 06:14.27 | Kyoshi | poly's aint hard to config |
| 06:14.42 | j0 | too many darn options |
| 06:14.50 | Kyoshi | snom's are nice too |
| 06:14.51 | j0 | i guess that's a good thing, i use most of them |
| 06:15.10 | Kyoshi | well you dont configure each phone individually really |
| 06:15.20 | j0 | Kyoshi: well the only brand u havn't recommended now is grandstream. lol |
| 06:15.23 | j0 | yeah |
| 06:15.33 | Kyoshi | via freepbx' interface, use the endpoint manager, make a group config and enter the mac's of each phone into the group for extensions |
| 06:15.35 | j0 | does polycom have a configuration editor? |
| 06:15.40 | Kyoshi | let it propagate |
| 06:15.51 | j0 | yeah, that's what i do |
| 06:15.54 | Kyoshi | jo: grandstream aint worth it |
| 06:16.11 | Kyoshi | i got the handytones and the gxp's, bah |
| 06:16.15 | NeoSkedar | j0 theres no issues with using softphone in a biz |
| 06:16.16 | j0 | Kyoshi: i know.. i bought one for my personal voip setup to learn on |
| 06:16.17 | Kyoshi | handytones crash too much |
| 06:16.34 | Kyoshi | gxp's sound quality is like a 99 cent special |
| 06:16.47 | j0 | yup, i only have 501s in my office now |
| 06:17.07 | Kyoshi | poly's? |
| 06:17.09 | j0 | NeoSkedar: what about connecting a proper handset to it? |
| 06:17.10 | j0 | Kyoshi: yes |
| 06:17.11 | JT | NeoSkedar: no issues, that's a big optimistic |
| 06:17.23 | JT | s/big/bit/ |
| 06:17.23 | Kyoshi | those are kinda old |
| 06:17.33 | NeoSkedar | not realluy |
| 06:17.36 | NeoSkedar | whats the big deal |
| 06:17.37 | j0 | Kyoshi: old? |
| 06:17.39 | NeoSkedar | depending on the biz |
| 06:17.52 | JT | the audio quality on a softphone is nowhere near as good as on a polycom |
| 06:17.57 | Kyoshi | using a softphone in a company? |
| 06:18.06 | JT | most softphones have shit echo cancellation and noise reduction |
| 06:18.07 | j0 | i can see a softphone working on a locked down computer.. but your typical office computer running vista might not cut it |
| 06:18.09 | Kyoshi | not sure if id trust it |
| 06:18.11 | JT | and clumbsy call handling |
| 06:18.27 | NeoSkedar | my softphones have no issues |
| 06:18.37 | NeoSkedar | [23:08] <j0> if a customer couldn't afford polycom phones, what would you recommend to them? |
| 06:18.40 | Kyoshi | how many are we talking? |
| 06:18.40 | j0 | NeoSkedar: what sort of headset do you use? |
| 06:18.41 | NeoSkedar | inother words funds are tight |
| 06:18.42 | JT | everything has no issues until you try something better. |
| 06:18.55 | j0 | NeoSkedar: yes.. "afford" is a loose word |
| 06:19.01 | *** join/#asterisk MaliutaLap (n=nikolai@kiev.lusan.id.au) |
| 06:19.12 | NeoSkedar | u wont find many phones under $100 |
| 06:19.17 | Kyoshi | neo: how many softphone users are we talking? |
| 06:19.17 | NeoSkedar | at least new |
| 06:19.25 | NeoSkedar | i said depends on teh buisness didnt i |
| 06:19.27 | j0 | "but my analog phones only cost $120" (they suck and that's another reason they're upgrading) |
| 06:19.34 | JT | IP320s are under $100 |
| 06:19.52 | Kyoshi | polycom ip330's are about $120 |
| 06:19.52 | NeoSkedar | i said "many" |
| 06:19.55 | j0 | ah that's right.. the 320 even has speakerphone |
| 06:19.59 | JT | Kyoshi: $89 |
| 06:20.02 | JT | USD |
| 06:20.04 | Kyoshi | hahaha |
| 06:20.05 | Kyoshi | there ya go |
| 06:20.06 | j0 | $83.99 |
| 06:20.07 | JT | err |
| 06:20.09 | Kyoshi | even better |
| 06:20.10 | JT | 320 sorry |
| 06:20.12 | NeoSkedar | lol |
| 06:20.15 | JT | you said 330, oops |
| 06:20.18 | NeoSkedar | get those j0 |
| 06:20.19 | Kyoshi | yea |
| 06:20.20 | JT | yeah |
| 06:20.24 | JT | 330s are slightly more |
| 06:20.43 | Kyoshi | i just dont think i could in good conciousness suggest a customer uses all softphones in their company |
| 06:20.45 | JT | you do need a PoE switch though |
| 06:20.51 | Kyoshi | what if they need to reboot their pc uring a call? |
| 06:20.52 | j0 | what does the 330 add over the 320? |
| 06:20.52 | NeoSkedar | depends on teh company |
| 06:20.54 | NeoSkedar | and size |
| 06:21.03 | NeoSkedar | if its a small biz has like 5 users |
| 06:21.04 | Kyoshi | jo: 40 bucks? |
| 06:21.06 | NeoSkedar | then no big deal |
| 06:21.07 | JT | j0: ethernet switch and port for pc |
| 06:21.11 | NeoSkedar | but yeah |
| 06:21.14 | NeoSkedar | hard phones are better |
| 06:21.17 | JT | j0: that's all |
| 06:21.27 | j0 | JT: ah.. i'd need that here .. thanks -- i missed that in the comparision chart |
| 06:23.24 | *** join/#asterisk gnorbert (n=gnorbert@228-177.static.ew.hu) |
| 06:23.31 | gnorbert | Good morning |
| 06:23.40 | NeoSkedar | looks at clock |
| 06:24.40 | gnorbert | Does somebody know, how can I use speex in asterisk? I mean how can I configure Asterisk to |
| 06:24.54 | gnorbert | accept speex? |
| 06:25.41 | gnorbert | I've already configured codecs.conf, sip.conf, iax.conf... |
| 06:26.20 | gnorbert | But it still doesn't work... |
| 06:26.23 | j0 | gnorbert: so it's enabled everywhere.. what exactly isn't happening? |
| 06:27.33 | j0 | gnorbert: have you checked your sip debug? |
| 06:27.42 | gnorbert | j0: When I try join to a meetme conference via asterisk with ekiga (with speex wideband codec), it writes: |
| 06:27.46 | gnorbert | <PROTECTED> |
| 06:28.32 | gnorbert | With GSM codec, this works, just so noisy, that nothing can be heard... |
| 06:28.44 | j0 | noisy? |
| 06:29.10 | *** join/#asterisk seanmh (n=johndoe@c-68-35-21-64.hsd1.nm.comcast.net) |
| 06:29.15 | gnorbert | The only thing, that can be heard is some noise and clicking... |
| 06:29.25 | j0 | gnorbert: what about your other codec? |
| 06:29.26 | j0 | s |
| 06:30.08 | j0 | look at your sip debug too.. maybe ekiga isn't advertising that speex is available |
| 06:30.08 | gnorbert | GSM works, but I should use speex wideband, or something, that's good at least like that. |
| 06:31.37 | NeoSkedar | bah |
| 06:31.45 | NeoSkedar | i cant get this stuipid vm thing to work |
| 06:32.27 | j0 | speaking of vm's.. does * have proper timing in a vm yet? |
| 06:32.43 | j0 | it's the last box that isn't in a vm |
| 06:33.48 | NeoSkedar | i have someone connected in another location, they cant retieve voicemail, always says vm password is invalid |
| 06:33.54 | NeoSkedar | yet all local users can access it fine |
| 06:34.28 | j0 | NeoSkedar: dtmf problem? |
| 06:34.45 | NeoSkedar | dont think so |
| 06:34.51 | NeoSkedar | why would that affect only remote users |
| 06:35.13 | j0 | more jitter issues |
| 06:35.27 | NeoSkedar | the remote users can place calls fine |
| 06:36.11 | j0 | the vm password is entered with dtmf, the placing of the call isn't |
| 06:36.29 | NeoSkedar | wouldnt dtmf be system wide? |
| 06:36.47 | j0 | u mean a problem with it would be system wide? |
| 06:36.53 | NeoSkedar | yeah |
| 06:37.08 | j0 | could be any number of reasons why it works someplaces and not others |
| 06:37.30 | NeoSkedar | well it affects all non lan users |
| 06:37.45 | j0 | worth a try to look at anyway |
| 06:37.52 | NeoSkedar | what should i look at |
| 06:38.04 | j0 | check the debug to see if the tones are interpreted properly |
| 06:38.37 | NeoSkedar | how do i do that |
| 06:40.10 | j0 | NeoSkedar: google |
| 06:40.18 | NeoSkedar | theres a ton of debugs |
| 06:40.21 | j0 | asterisk dtmf debug |
| 06:40.36 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:40.58 | pputman | You know what's really annoying? All these third party versions of asterisk that by default turn off verbose for the console. |
| 06:41.36 | pputman | in logger.conf |
| 06:45.03 | JT | otherwise you'd see all the noise from their horrible dialplans |
| 06:52.25 | NeoSkedar | maybe latency of teh remote connection causes the vm password to not register? |
| 06:53.50 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-f9f0357dcb16f0b7) |
| 07:05.03 | gnorbert | I still can't use speex codec. Any idea? (The fault message is: "NOTICE[7270]: chan_sip.c:5426 process_sdp: No compatible codecs, not accepting this offer!") |
| 07:05.34 | JT | what does show translations give you |
| 07:06.07 | JT | does it show any translation path to speex? |
| 07:06.42 | gnorbert | There is speex... |
| 07:07.06 | JT | does it show any translation paths? |
| 07:07.12 | JT | no numbers, no path |
| 07:07.22 | gnorbert | It has (in the row): - 9 3 3 1 3 2 4 - 199 - - - |
| 07:08.07 | gnorbert | In column it has - 198 198 198 199 198 197 208 - - - 199 - |
| 07:08.07 | *** join/#asterisk svenna_ (n=svenna@p548D0AFD.dip0.t-ipconnect.de) |
| 07:08.11 | JT | bah |
| 07:08.17 | JT | that's all useless to me |
| 07:08.20 | JT | it's just a table |
| 07:08.22 | JT | look it up |
| 07:08.53 | JT | check if the codec you're trying to transcode to has a number that corresponds with speex |
| 07:08.57 | gnorbert | That table is the only thing, it writes for show translation. |
| 07:09.05 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1176000517.dsl.bell.ca) |
| 07:09.12 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 07:09.32 | JT | yes. |
| 07:09.48 | JT | what codec are you translating to/from |
| 07:09.51 | *** join/#asterisk bmg505 (n=leon@196-209-79-182-tbnb-esr-2.dynamic.isadsl.co.za) |
| 07:10.05 | gnorbert | I would like to use only speex... Now I use GSM, because that works, but it has a number with speex |
| 07:10.45 | JT | the number is the latency estimate in milliseconds |
| 07:10.58 | *** join/#asterisk MaliutaLap (n=nikolai@kiev.lusan.id.au) |
| 07:11.11 | gnorbert | That's 198 and 9... |
| 07:11.29 | gnorbert | gsm to speex 198, speex to gsm 9 |
| 07:12.11 | JT | hmm |
| 07:12.16 | JT | should work in theory |
| 07:12.47 | *** join/#asterisk pootle (n=pootle@wchurches.plus.com) |
| 07:13.06 | gnorbert | But I wouldn't like to use gsm, if speex would work.. |
| 07:14.07 | gnorbert | I mean in ekiga I could select speex wideband as the only codec, just then asterisk doesn't accept the call even. |
| 07:14.52 | gnorbert | If asterisk accepted it, I wouldn't have to use gsm, I guess. |
| 07:16.04 | gnorbert | I made sip.conf, aix.conf, extensions.conf and codecs.conf... |
| 07:16.13 | gnorbert | Could I miss something?:S |
| 07:16.31 | NeoSkedar | [Jul 11 00:15:20] WARNING[11624]: app_voicemail.c:6907 vm_authenticate: Unable to read password |
| 07:16.37 | NeoSkedar | that prolly is a dtmf issue right? |
| 07:18.10 | gnorbert | I've also installed speex already. |
| 07:19.25 | gnorbert | NeoSkedar: I know, in Zoiper you have to register for a server first and you have to give the password in the options... |
| 07:19.55 | NeoSkedar | gnorbert, all my computers work fine on my lan |
| 07:19.59 | NeoSkedar | its a remote connection |
| 07:20.01 | NeoSkedar | that doesnt |
| 07:20.37 | gnorbert | Hmm, interesting one.:) |
| 07:22.08 | gnorbert | Sorry, I'm not like an expert, it was just a try...:) |
| 07:22.41 | NeoSkedar | =] |
| 07:22.45 | NeoSkedar | no worries |
| 07:24.09 | bkw_ | NeoSkedar: what phone are you using? |
| 07:24.47 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 07:25.39 | NeoSkedar | x-lite |
| 07:25.55 | gnorbert | So I guess, Ther isn't so much tip about the speex-problem.:) |
| 07:26.09 | bkw_ | gnorbert: noise on speex? |
| 07:26.16 | gnorbert | Yes. |
| 07:26.19 | gnorbert | I mean no.:D |
| 07:26.21 | bkw_ | NeoSkedar: the best bet is to get a pcap and look at it on wireshark |
| 07:26.28 | bkw_ | gnorbert: static? or something? |
| 07:26.32 | bkw_ | I know there is a bug open on that issu |
| 07:26.33 | bkw_ | e |
| 07:26.40 | gnorbert | Noise on GSM and asterisk doesn't accept speex. |
| 07:26.49 | bkw_ | did you allow speex? |
| 07:26.55 | bkw_ | on the allow= line for the user or peer? |
| 07:27.05 | NeoSkedar | bkw_, my lan pohnes work |
| 07:27.06 | gnorbert | I allowed all... |
| 07:27.12 | bkw_ | whats this 198 and 8 business? |
| 07:27.14 | NeoSkedar | but it doesnt work from a wan ip |
| 07:27.20 | gnorbert | And I tried allow=speex. |
| 07:27.31 | gnorbert | *also |
| 07:27.47 | gnorbert | JT asked about show translation |
| 07:27.48 | bkw_ | oh translation costs |
| 07:28.03 | gnorbert | Between GSM and speex. |
| 07:28.04 | bkw_ | took me a second.. those numbers mean ZERP |
| 07:28.06 | bkw_ | er ZERO |
| 07:28.17 | bkw_ | they are not a real gauge of the transcoding performance |
| 07:28.18 | gnorbert | But I should use only speex. |
| 07:28.33 | bkw_ | which speex did you allow on the phone and what does asterisk say? |
| 07:29.12 | gnorbert | I used Ekiga and allowed 16 khz speex |
| 07:29.22 | bkw_ | well asterisk can't do 16k speex |
| 07:29.27 | bkw_ | so that might be a huge part of your problem |
| 07:29.35 | gnorbert | (The wideband version, I guess...) |
| 07:29.39 | bkw_ | yes |
| 07:32.54 | *** join/#asterisk ReD-MaN (i=rox-ur-s@172-220.static.golden.net) |
| 07:35.56 | NeoSkedar | sighs |
| 07:36.25 | NeoSkedar | why would lan users send dtmf tones and not remote users? |
| 07:36.31 | bkw_ | you have three options.. are you doing 2833, info or inband? |
| 07:36.41 | bkw_ | what codec are the remote users? |
| 07:36.45 | NeoSkedar | same as the lan |
| 07:36.49 | NeoSkedar | same phones |
| 07:36.54 | bkw_ | weird |
| 07:36.57 | bkw_ | I would double check the settings |
| 07:37.05 | NeoSkedar | i have a computer on lan, dialed it worked, connected to my evdo adn remoted, doesnt work |
| 07:37.12 | NeoSkedar | no settings changed on phone |
| 07:37.21 | bkw_ | famous last words |
| 07:37.27 | NeoSkedar | just the ip that registered to teh network |
| 07:37.52 | NeoSkedar | i know they arent sending tones |
| 07:38.17 | NeoSkedar | cuz if i call and push a key on the remote phone i cant hear it on the lan phone, but if i push a key on teh lan phone u can hear the tone on teh remote phone |
| 07:38.22 | bkw_ | you have wireshark? |
| 07:38.37 | NeoSkedar | not on here |
| 07:39.01 | bkw_ | install it and try this command line |
| 07:39.02 | bkw_ | tshark -o "rtp.heuristic_rtp: TRUE" -R sip\|\|rtp\|\|rtcp |
| 07:39.08 | bkw_ | start it before you make the call |
| 07:39.14 | bkw_ | you'll see the digits if its sending them |
| 07:39.25 | NeoSkedar | or not sending |
| 07:39.26 | NeoSkedar | lol |
| 07:39.30 | bkw_ | yah |
| 07:39.36 | bkw_ | it'll make sure you're not insane |
| 07:39.44 | NeoSkedar | im pretty sure im not |
| 07:39.49 | bkw_ | well check it to be sure |
| 07:39.54 | NeoSkedar | but no ideas why that would happen? |
| 07:39.55 | bkw_ | I know asterisk does some funky stuff in 2833 |
| 07:40.01 | bkw_ | are the phones all running the same firmware? |
| 07:40.05 | NeoSkedar | yeah |
| 07:40.17 | NeoSkedar | told u i had the phone on my wifi, then switched to evdo |
| 07:40.20 | NeoSkedar | exact same phone |
| 07:40.24 | NeoSkedar | 2 diff connections |
| 07:40.27 | NeoSkedar | lan works |
| 07:40.28 | NeoSkedar | wan doesnt |
| 07:40.40 | NeoSkedar | call work fine |
| 07:40.45 | NeoSkedar | but calls dont use tones |
| 07:40.51 | *** join/#asterisk bboschman (n=bboschma@p50997436.dip0.t-ipconnect.de) |
| 07:41.31 | NeoSkedar | [Jul 11 00:41:03] WARNING[11624]: chan_sip.c:2752 retrans_pkt: Hanging up call MWYwMzAyMjI4MDIzMGE0Njg4YzcwODcwNGZhYjEzNWE. - no reply to our critical packet. |
| 07:41.35 | NeoSkedar | any idea what that means? |
| 07:42.26 | bkw_ | NeoSkedar: you're having nat drama? |
| 07:42.45 | bkw_ | there are so many broken sip devices out there.. I refuse to bow down and hack around them |
| 07:42.51 | bkw_ | the manufacture fixes them or I talk shit about em |
| 07:42.56 | NeoSkedar | lol |
| 07:42.59 | *** join/#asterisk ^shark_ (n=jochieng@41.222.2.65) |
| 07:43.14 | NeoSkedar | well ive used two different sip softphones both have exact same issue |
| 07:43.24 | bkw_ | you said it was x-lite? |
| 07:43.26 | bkw_ | or hard phones? |
| 07:43.27 | NeoSkedar | aye |
| 07:43.30 | NeoSkedar | xlite |
| 07:43.34 | bkw_ | x-lite is the biggest pile of shit |
| 07:43.39 | bkw_ | I can crash it in two sceonds |
| 07:43.43 | NeoSkedar | lol |
| 07:43.45 | bkw_ | I think if you look at it wrong it crashes |
| 07:43.56 | ^shark_ | roflol |
| 07:44.00 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 07:44.01 | bkw_ | when I was writing mod_voipcodecs for FreeSWITCH .. I would call the delay echo app.. it crashes EVERY time |
| 07:44.03 | bkw_ | without fail |
| 07:44.10 | bkw_ | you feed it bad audio data.. KABOOM |
| 07:44.19 | bkw_ | you click the wrong box at the right time.. KABOOM |
| 07:44.26 | bkw_ | man that thing crashed on me more times than I care to know about |
| 07:44.29 | NeoSkedar | well i havent had it crash yet |
| 07:44.30 | NeoSkedar | lol |
| 07:44.37 | bkw_ | you don't really use it then :P |
| 07:44.46 | NeoSkedar | im just learning asterisk |
| 07:44.52 | NeoSkedar | softphones free way to test |
| 07:46.57 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:50.11 | mvanbaak | x-lite is unstable ??? |
| 07:50.13 | mvanbaak | no way ! |
| 07:50.15 | mvanbaak | hides |
| 07:50.51 | mvanbaak | start it, select your usb headset as audiodevice, unplug the usb headset, KABOOM |
| 07:51.21 | mvanbaak | and yes, if you look at it the wrong way it goes beserk on the cpu |
| 07:52.11 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 07:53.53 | ^shark_ | there are better softphones that x-lite, who would continue using x-lite anyways!? |
| 07:55.27 | *** join/#asterisk SanityIO__ (n=SanityIO@77.242.105.20) |
| 07:55.31 | mvanbaak | I try x-lite from time to time to see if they fixed their stuff |
| 07:55.54 | mvanbaak | for day-to-day use, we are happy with ekiga |
| 07:56.31 | mvanbaak | hhmm, anyone know what the time for a ring is ? |
| 07:56.43 | ^shark_ | bria is my favourite |
| 07:57.17 | ^shark_ | mvanbaak: i didnt understand your question. |
| 07:59.04 | *** join/#asterisk bkw__ (n=brian@70-3-114-27.area5.spcsdns.net) |
| 08:00.17 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:00.20 | mvanbaak | well, a lot of customers ask me: 'go to voicemail after 5 rings' |
| 08:00.28 | mvanbaak | they dont think in seconds, but in rings |
| 08:00.39 | *** join/#asterisk xenonex (n=xenonex@89.218.235.41) |
| 08:01.16 | *** join/#asterisk LND (n=Lee@89.192.116.132) |
| 08:01.18 | *** join/#asterisk SanityIO__ (n=SanityIO@77.242.105.20) |
| 08:01.19 | *** join/#asterisk SanityIO (n=SanityIO@77.242.105.20) |
| 08:01.32 | mort_gib | mvanbaak: Tell them that mostly you start hearing rings even before the remote handset starts ringing, so the "5 rings" is relative anyway |
| 08:01.33 | NeoSkedar | just time it |
| 08:01.51 | NeoSkedar | 30s is a long time |
| 08:01.58 | bkw__ | yep it is |
| 08:02.03 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-4aa989f1cbb895a1) |
| 08:02.14 | NeoSkedar | ur back |
| 08:02.49 | NeoSkedar | bkw_, still have pm =] |
| 08:03.03 | *** join/#asterisk SanityIO__ (n=SanityIO@77.242.105.20) |
| 08:05.17 | bkw__ | on this nick |
| 08:08.38 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 08:26.28 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 08:30.28 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 08:30.39 | Rico29 | hi |
| 08:37.12 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 08:43.37 | *** join/#asterisk Fuzix (n=fuzix@82.201.19.192) |
| 08:46.18 | *** join/#asterisk BipBip (n=BipBip@bl8-139-112.dsl.telepac.pt) |
| 08:50.04 | Fuzix | Hello there, is there anyone that could help me analyse a gdb backtrace? It's the first one I made ;-) |
| 08:53.02 | Rico29 | what is gdb ? |
| 08:53.38 | Fuzix | GNU Debugger |
| 08:54.49 | Rico29 | ok |
| 09:00.27 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:02.37 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-aa8a8023d04b6dc7) |
| 09:02.52 | Great_Anta_Baka | morning |
| 09:03.37 | Great_Anta_Baka | any reason why voice mail isnt picking up my key presses on a snom but when i call into another company's ivr it detects my button presses? |
| 09:04.29 | Great_Anta_Baka | i have set the phone to send dtmf info via sip but then when i call into an ivr it doesnt detect the keypresses |
| 09:04.40 | Great_Anta_Baka | but then the voice mail works |
| 09:05.54 | *** join/#asterisk Neil_mccarthy2 (n=neil@dsl-p5-222.gibconnect.com) |
| 09:06.47 | Neil_mccarthy2 | hi there, I am having a problem that hopefully someone else has seen and can point me in the right direction |
| 09:07.30 | *** join/#asterisk af_ (n=getsmart@88-149-230-202.dynamic.ngi.it) |
| 09:08.38 | Neil_mccarthy2 | I have a service provider who is directing incoming calls to me, but he is directing the RTP stream to another address. I wish to bridge through asterisk to my extensions so that I can do call recording (it is a call centre). The problem seems to be that asterisk is not honoring this change. I do a tshark on the interface and do not see any packets go back to the RTP server |
| 09:09.23 | Neil_mccarthy2 | that they specify. |
| 09:11.10 | Great_Anta_Baka | mm seems like everyone is busy |
| 09:11.26 | Neil_mccarthy2 | It does |
| 09:11.58 | TheH | Great_Anta : what codecs did you allow in your sip.conf |
| 09:12.13 | Neil_mccarthy2 | disallow=all, allow=g729 on the trunk |
| 09:12.23 | mort_gib | Great_Anta_Baka: No, but your problem has to do with what codecs you use, so that locally it wont work because you are using the wrong codecs/dmft combination |
| 09:12.34 | Neil_mccarthy2 | on the extensions disallow=all, allow=g729,ulaw, alaw |
| 09:13.22 | mort_gib | g729 needs rtf2833 |
| 09:14.17 | Neil_mccarthy2 | Sorry I dont understand what you mean |
| 09:14.18 | Great_Anta_Baka | hey |
| 09:14.37 | Great_Anta_Baka | its not my box but i am going to the site now to check it out |
| 09:14.47 | Great_Anta_Baka | i see mort_gib |
| 09:15.11 | Neil_mccarthy2 | Sorry, wrong conversation |
| 09:15.38 | Great_Anta_Baka | mort_gib: rtf2833 == rfc2833? |
| 09:15.49 | mort_gib | Well yes :-) |
| 09:15.53 | mort_gib | -Sorry |
| 09:15.53 | Great_Anta_Baka | kk :P |
| 09:16.47 | Great_Anta_Baka | thanks for the help .. going to the site now and will probably be back on here if it still fails :/ |
| 09:18.28 | mort_gib | Well, I'll be hanging around for a while, doing my account, so interruptions are welcome :-) |
| 09:18.52 | Neil_mccarthy2 | can anyone help me with an RTP problem |
| 09:19.12 | mort_gib | btw, I would dissalow=all, allow=g729,allow=alaw,allow=ulaw |
| 09:21.25 | mort_gib | Neil_mccarthy2: I didn't understand your problem |
| 09:23.20 | Neil_mccarthy2 | mort_gib: Thanks for replying. The VOIP provider sends us a RTP server/port to connect to on the INVITE |
| 09:23.31 | Neil_mccarthy2 | which is different from the SIP server |
| 09:24.10 | mort_gib | Hmm, I'm not sure I would be able to help, would that not be automatic then? |
| 09:24.27 | gnorbert | If asterisk and the softphone both write, that connection is made, then how can be, that the line is still so noisy, that nothing can be heard? |
| 09:24.42 | gnorbert | It's in a meetme conference and with zoiper sotphone. |
| 09:25.26 | Neil_mccarthy2 | mort_gib: asterisk is picking it up, but not sending any packets to teh address |
| 09:26.10 | mort_gib | Is the traffic sent to the SIP server?? |
| 09:26.25 | gnorbert | It's sip. |
| 09:33.39 | *** join/#asterisk Great_anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-b23a09f46bc8ad19) |
| 09:34.12 | Delvar | quick question... how to change the behavior of asterisk when a call comes in without callerid, asterisk replaces the name and number with 'Asterisk', is there a setting for this? we would prefer 'Private' or 'Widthheld' ... |
| 09:36.04 | Great_anta_Baka | when i make a call it says "starting tones - we have inband data" is that because my side is set up to send dtmf info inband? |
| 09:36.46 | Neil_mccarthy2 | mort_gib: no I cannot see any RTP traffic |
| 09:39.10 | TheH | Great_ Do you have allow=gsm anywhere there ? |
| 09:39.35 | Great_anta_Baka | let me check |
| 09:39.54 | Great_anta_Baka | well for this extension i have set allow=alaw |
| 09:40.17 | TheH | ok but in the sip.conf you have the "master" part of codecs to allow and disallow right |
| 09:40.17 | Great_anta_Baka | and the dtmf for this extension is set to rfc2833 |
| 09:41.10 | TheH | i had the same problem when in my sip.conf i had disallow=all and then allow=gsm allow=ulaw allow=alaw (after putting gsm in 3d place instead of first it resolved the issue with the tones) |
| 09:41.13 | Great_anta_Baka | eish thats why i hate trixbox looking through so many config files at the moment |
| 09:41.25 | Great_anta_Baka | i see |
| 09:41.28 | TheH | vi /etc/asterisk/sip.conf |
| 09:41.45 | Great_anta_Baka | thats a very small files.. its including all these other files |
| 09:41.51 | Great_anta_Baka | looking through them now |
| 09:42.11 | TheH | grep disallow=all * |
| 09:43.35 | Great_anta_Baka | i did a more /etc/asterisk/sip* |grep allow and nothing comes up with gsm |
| 09:45.19 | *** join/#asterisk Ast001 (n=Administ@cable-89-216-185-58.dynamic.sbb.rs) |
| 09:46.32 | Great_anta_Baka | my dtmf info is definately not getting sent out and is asterisk supposed to show key presses on the consoles? |
| 09:47.01 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.140) |
| 09:48.00 | Great_anta_Baka | i do a sip debug on my extension and i can see the key presses being recognised |
| 09:51.49 | Great_anta_Baka | if i deselect the option DTMF via SIP INFO:then the key presses are recognized in the ivr i am calling |
| 09:52.13 | Great_anta_Baka | but then calling into voice mail doesnt work o_0 |
| 09:52.44 | *** join/#asterisk funkyhippy (n=funkyhip@78-32-92-91.no-dns-yet.enta.net) |
| 09:53.27 | funkyhippy | Hi does anyone know if mysql support is configured in asterisk Busines Edition? |
| 09:53.34 | TheH | Great : mmm you might want to do a dailplan trick then to allow it when needed |
| 09:54.01 | *** part/#asterisk ^shark_ (n=jochieng@41.222.2.65) |
| 09:54.39 | Great_anta_Baka | mmm dont know how to do that but will shout if i cant figure it out |
| 09:55.18 | *** join/#asterisk mkl1525 (n=some@89.246.177.250) |
| 09:55.19 | TheH | Great : http://lists.digium.com/pipermail/asterisk-dev/2005-January/008797.html |
| 09:55.43 | Ast001 | hi I still have that annoying problem with pri isdn and openvox d110p card i can not receive any call |
| 09:55.46 | Ast001 | here are my files |
| 09:56.09 | TheH | Ast001: We need a trace in order to help you |
| 09:56.09 | Ast001 | /proc/zaptel/1 http://pastebin.com/d578df5df |
| 09:56.37 | Ast001 | /proc/interrupts |
| 09:56.38 | Ast001 | http://pastebin.com/d2a4b160a |
| 09:57.04 | Ast001 | zapata.conf http://pastebin.com/d366af96a |
| 09:57.19 | Ast001 | zaptel.conf http://pastebin.com/d7f90e770 |
| 09:57.42 | Ast001 | extensions.conf http://pastebin.com/d89a8a3b |
| 09:57.54 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 09:58.02 | Ast001 | pri debug span does not show anything |
| 09:58.39 | Ast001 | when I call I just hear telco message and after few secs signal like i put down phone |
| 09:59.30 | TheH | try running ./ genzaptelconf |
| 09:59.43 | TheH | and then restart zaptel service and asterisk |
| 10:00.25 | Ast001 | ok |
| 10:00.49 | TheH | and then add to your extensions.conf in the bottom : exten => _0X.,1,Dial(Zap/g1/0${EXTEN:1}) |
| 10:01.02 | TheH | and do a module reload in the CLI |
| 10:01.26 | TheH | then type: dial 0 + phone number of your mobile for example |
| 10:01.44 | TheH | also in the CLI |
| 10:01.50 | Ast001 | TheH i have only In service i can not dial out |
| 10:01.54 | Ast001 | only they can dial me |
| 10:02.11 | Ast001 | thats telco politics |
| 10:03.02 | TheH | Ast001: are you not allowed to make outgoing calls ?? If you do those steps I mentiod above and give me the trace of that call then i can help you |
| 10:03.22 | Ast001 | yes I am not allowed to make outgoing calls |
| 10:03.43 | Ast001 | this is IN platform they can only call me i can not call anyone |
| 10:05.04 | Ast001 | if I remember correct there is some zaptel command to check interupts |
| 10:05.33 | TheH | ok then configure your inbound trunk on the zapata ( [channels] context=ibtrunk ) |
| 10:05.53 | TheH | and configure [ibtrunk] in your extensions.conf to go to a sip |
| 10:05.59 | TheH | extension |
| 10:09.11 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 10:09.57 | Ast001 | i think i did it |
| 10:13.27 | Ast001 | genzaptel made zaptel conf and now i have rec/blue alarm on zttool |
| 10:13.43 | Ast001 | because it add crc4 at the end |
| 10:18.35 | *** join/#asterisk matrix1233 (n=Administ@41.227.249.128) |
| 10:18.44 | matrix1233 | hi |
| 10:18.51 | Ast001 | Can you help me with configuration ? span=1,1,0,ccs,hdb3 I think 0 on 3rd place is problem I guess i need to receive timing from post so 2nd number is 1 but 3rd what means 3rd number ? Manual said distance betweet card and telecom gateway.When I put 0 yesterday isdn was yellow (not ready) when i change it to 1 it become green so I guesss 1 is ok , but now i can put 0 1 or what else and it is always green |
| 10:19.09 | Great_anta_Baka | ok i cant figure out how to send dtmf tones via sip to asterisk and not when dialing out |
| 10:20.28 | Ast001 | zttest gave me result of 99.994 is worst result |
| 10:23.04 | matrix1233 | hello |
| 10:23.25 | matrix1233 | any one can help me with video on asterisk |
| 10:23.40 | matrix1233 | Thxxxxxxxxxxxxxxxxxx |
| 10:23.44 | matrix1233 | no idea ? |
| 10:28.19 | TheH | matrix: allow h232 |
| 10:28.27 | TheH | videosupport=yes |
| 10:28.34 | TheH | and your up and running :) |
| 10:30.55 | matrix1233 | thx TheH |
| 10:31.15 | matrix1233 | the code is installed by default |
| 10:31.47 | *** join/#asterisk dream_th (n=dream_th@91.187.96.48) |
| 10:32.15 | *** join/#asterisk XnOSX (n=XnOSX@212.145.170.219) |
| 10:32.55 | dream_th | hi |
| 10:33.07 | dream_th | i need some help with inbound routes please |
| 10:33.09 | gnorbert | Is here somebody, who already made a working asterisk server with an at least 16 khz codec? |
| 10:33.17 | gnorbert | Maybe with speex. |
| 10:33.19 | dream_th | here what is going on http://pastebin.com/de006b9a |
| 10:34.30 | Ast001 | is service capi important for running asterisk or zaptel ? |
| 10:34.52 | mort_gib | Is there ANY way of running a verify on the syntax on extensions.conf ?? |
| 10:35.05 | gnorbert | Then with any kind of codec?:) |
| 10:36.06 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 10:36.50 | eXistenZ | hallo |
| 10:40.45 | TheH | hoi |
| 10:41.54 | *** part/#asterisk dream_th (n=dream_th@91.187.96.48) |
| 10:44.37 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 10:44.39 | casix | hello |
| 10:45.39 | TheH | ast001 : zap tel you need for your PRI |
| 10:46.22 | TheH | gnorbert : vi sip.conf , disallow=all , allow=speex (but be sure you have installed that codecs if not do a make menuconfig » codecs » select speex and rebuild asterisk) |
| 10:46.25 | Great_anta_Baka | sigh* anyone know where i can get the latest snom firmware the phones on this network arent allowed to access the internet |
| 10:46.37 | kamanashisroy | where is defender ! |
| 10:47.01 | casix | I have a problem. one * cuts some calls. I have found this in the logs but I don't know how to repair it: channel.c: Didn't get a frame from channel |
| 10:47.03 | mvanbaak | Great_anta_Baka: did you try http://www.snom.com ? |
| 10:47.09 | gnorbert | TheH: I already tried this, but with no result. |
| 10:47.28 | kamanashisroy | casix: what do you mean ? |
| 10:47.46 | Ast001 | so I don't need capi service ? |
| 10:47.49 | kamanashisroy | acting like TKd-fender :) |
| 10:48.32 | casix | I don't know why the calls are interrups and I don't know why. All that I have seen in logs are this error |
| 10:48.49 | Great_anta_Baka | mvanbaak: i get to the snom wiki pages but then it only lets me do an automatic update from version 6 to version 7 |
| 10:49.00 | TheH | ast001: welly ou have never showed me a trace of the incoming call so i wont help you until you do so |
| 10:49.19 | Great_anta_Baka | mvanbaak do you know why i cant get my voicemail to pick up my dtmf tones why i call into it |
| 10:49.29 | gnorbert | TheH: I get the same message. |
| 10:49.30 | TheH | gnorbert= Are you sure its compiled in the asterisk cause by default it will not be installed |
| 10:49.35 | Ast001 | I have not trace |
| 10:49.41 | Ast001 | i can not see anything in CLI |
| 10:50.23 | gnorbert | TheH: Now, that you ask, I'm getting less sure.:D |
| 10:50.32 | TheH | Ast001 :ok but did you configure a inbound trunk for all your PRI lines |
| 10:50.43 | casix | I have googled but I didn't find any solution or explanation of this error |
| 10:50.51 | Ast001 | periodicly I can see NOTICE chan_zap.c:8535 pri_dchannel: PRI got event: HDLC Abort(6) on Primary D-channel of span 1 |
| 10:52.21 | Ast001 | i wrote context=dolaz in zapata.conf and made [dolaz] in extension.conf and there exten => s,1,Answer ... etc... so call should come to Callqueue wheere operater is waiting |
| 10:53.22 | gnorbert | TheH: How can I rebuild it from the menu, or save the settings of the menu and then rebuild it? |
| 10:53.27 | Strom_M | Wow. #asterisk at this hour of the morning is like watching the blind leading the blind in some crazy circus act. |
| 10:53.43 | TheH | gnorbert : ./configure , make menuconfig , Codecs » select there |
| 10:53.52 | gnorbert | done |
| 10:54.14 | kamanashisroy | casix: what is your error ? |
| 10:54.17 | TheH | Strom_M : soo who is blind here then ? |
| 10:54.47 | gnorbert | But then how can I exit without ctrl+z and rebuild it? |
| 10:54.54 | Ast001 | also I can see this mesage : WARNING [3410]: chan_zap.c:2402 pri_find_dchan: NO D-channels avaliable! Using primary channel 16 as D-channel anyway! |
| 10:55.09 | TheH | s |
| 10:55.18 | TheH | gnorbert s or escape |
| 10:55.38 | gnorbert | And the rebuild? |
| 10:55.42 | kamanashisroy | gnorbert: you need to go back to the main menu for that , use arrow keys .. |
| 10:55.47 | casix | kamanashisroy: that the calls are interrupted, cutted, dropped I'm not shure how to say in english |
| 10:55.50 | gnorbert | Simply make install? |
| 10:55.53 | kamanashisroy | gnorbert: make; make install; |
| 10:56.00 | gnorbert | Ok, thanks. |
| 10:56.00 | kamanashisroy | gnorbert: sure |
| 10:56.09 | XnOSX | anybody have a text list about english voices? |
| 10:56.20 | casix | kamanashisroy: and I have this error in the logs. I have googled it but not find anything |
| 10:56.24 | kamanashisroy | casix: pastebin the errors ! |
| 10:57.37 | Great_anta_Baka | TheH: i cant use your option cos its trixbox and it rewrites the config files everytime i apply a setting |
| 10:58.10 | Great_anta_Baka | i have tried setting the dtmf options to info rfc2833 and inband but voicemail still cant detect my key presses |
| 10:58.45 | TheH | Great : Sorry but rm -rf * and reinstall with just asterisk... and set fire to trixbox :) |
| 10:59.18 | Great_anta_Baka | i know i know i told the administrator of this box that it is a fail but sometimes people dont listen |
| 10:59.20 | troubled | hey guys, anyone know if skype can take sip calls from asterisk? or do I need a recompile of asterisk with skype patch, a skype account and some voodoo to bridge the call? |
| 10:59.42 | Great_anta_Baka | if it wasnt trixbox the only way to be able to do it was the way you told me? |
| 11:00.37 | gnorbert | It already doesn't work... |
| 11:00.39 | casix | kamanashisroy: http://pastebin.com/m306206e1 |
| 11:00.50 | gnorbert | I mean I can't even enter asterisk.:D |
| 11:00.59 | gnorbert | I write sudo asterisk -vvvc |
| 11:01.06 | troubled | hilight me if you have something to add or help, thanks *switches windows* |
| 11:01.17 | kamanashisroy | gnorbert: asterisk -vvvr ! |
| 11:01.19 | TheH | gnorbert= /etc/init.d/zaptel stop and then /etc/init.d/asterisk stop and then asterisk start |
| 11:01.21 | gnorbert | It writes a lot of stuff, but after all it quits. |
| 11:01.43 | kamanashisroy | gnorbert: pastebin ! |
| 11:02.36 | kamanashisroy | casix: what are you trying to do ? |
| 11:02.45 | *** join/#asterisk lbenzo (n=lbenzo@89.140.19.226.static.user.ono.com) |
| 11:02.52 | kamanashisroy | casix: are you testing call from asterisk cli ? |
| 11:02.53 | casix | to know why the call is dropped |
| 11:03.16 | gnorbert | Still the same fault. |
| 11:04.01 | casix | kamanashisroy: is a vpbx from one client. It is in production an some calls are dropped and I have to know why for repair it |
| 11:04.37 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 11:04.51 | kamanashisroy | casix: it seems that frames are missing from sip channel |
| 11:04.56 | kamanashisroy | you need to enable sip debug |
| 11:05.19 | kamanashisroy | casix: as it is production you can enable debug on a single channel .. |
| 11:05.19 | casix | but frames are from sip or from rtp?? |
| 11:05.42 | casix | the problem is from sip packets or from rtp packets? |
| 11:06.44 | tzafrir_laptop | TheH, no use trying to unload the zaptel modules while asterisk is running . At least zaptel itself would fail to unload |
| 11:06.53 | kamanashisroy | casix: that is why we need to see sip debug |
| 11:06.56 | tzafrir_laptop | stop asterisk first, and then zaptel |
| 11:07.06 | casix | because in the middle of a call there are no sip traffic, no? |
| 11:07.34 | kamanashisroy | casix: it seems there is no rtp packet .. |
| 11:07.38 | kamanashisroy | casix: guessing .. |
| 11:07.46 | Great_anta_Baka | ok when asterisk recieves DTMF-RELAY event recieved: 7 what does that mean and why cant the other end recieve that relay? |
| 11:07.46 | casix | ok, I'll try to find a call that have this issue but its no easy because it happens some times only |
| 11:07.55 | kamanashisroy | casix: if sip packets are missing then the call will drop in other ways I believe .. |
| 11:08.06 | kamanashisroy | casix: I see |
| 11:09.11 | kamanashisroy | casix: that is big problem .. you do not know when it happens :-P .. happy debugging :) |
| 11:09.51 | casix | kamanashisroy: yes :) thx for help I'll try to find it! |
| 11:10.42 | *** join/#asterisk SparFux (n=raoul@e182026166.adsl.alicedsl.de) |
| 11:10.50 | SparFux | Hey, is fwdout gone? |
| 11:11.28 | Great_anta_Baka | ok how do i set my outbound dtmf then? |
| 11:12.00 | Great_anta_Baka | its clearly getting it since its picking it up in my voice mail |
| 11:15.32 | pputman | Is there any known reason why a telco would reject the display information element in a setup message on a pri, and a possible fix in zapata.conf? I have already tried setting facilityenable=no and overwriting the callerid, however it's still being rejected. |
| 11:16.02 | TheH | tza: if i cannot connect to my asterisk using the -rv switch i normally see its caused with zaptel / but ofcourse tail /var/log/asterisk/messages would tell the real reason |
| 11:16.13 | pputman | However I |
| 11:16.29 | pputman | I'm not entirely sure what the nsf option does, maybe that might change something? |
| 11:19.07 | gnorbert | "WARNING[15793]: pbx.c:2981 ast_register_application: Already have an application 'Pickup'" |
| 11:19.14 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:19.47 | gnorbert | ASterisk writes this and then I don't get the CLI back, but unix console. |
| 11:20.34 | gnorbert | TheH: Any idea? |
| 11:20.46 | TheH | gnorbert try a reboot :) |
| 11:21.09 | *** join/#asterisk ccesario (n=ccesario@mailserver.damata.ind.br) |
| 11:21.09 | gnorbert | Will that help?:) |
| 11:21.35 | TheH | gnorbert: it might , it think you have multiple instances running |
| 11:21.59 | gnorbert | Ok, then I end some things and then try it.:) |
| 11:22.53 | *** join/#asterisk zonech (n=vinc@zoneech.fr.nf) |
| 11:23.02 | Great_anta_Baka | arrrrg why does this box have two bri cards |
| 11:24.50 | zonech | Bonjour, est-ce qu'une personne parlant francais pourrais m'aider concernant l'installation d'asterisk? svp |
| 11:33.10 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 11:34.05 | *** join/#asterisk gnorbert (n=gnorbert@228-177.static.ew.hu) |
| 11:34.08 | gnorbert | Re. |
| 11:34.12 | gnorbert | The same problem.>D |
| 11:34.15 | gnorbert | :D |
| 11:34.58 | TheH | can you pastebin your /var/log/asterisk/messages |
| 11:38.32 | Ast001 | zap restart gave me errors |
| 11:39.21 | Ast001 | ERROR[3161]: chan_zap.c:905 zt_open: Unable to specify channel 1: Device or resource busy |
| 11:39.38 | *** join/#asterisk Blinkiz (n=Blinkiz@213-65-195-201-no96.business.telia.com) |
| 11:40.01 | Ast001 | chan_zap.c:10582 build_channels: Unable to register channel '1-15' |
| 11:40.08 | *** part/#asterisk Blinkiz (n=Blinkiz@213-65-195-201-no96.business.telia.com) |
| 11:40.40 | Ast001 | and at the end Reload channels from zap config failed |
| 11:40.55 | Ast001 | thats why I get always busy signal |
| 11:41.32 | Ast001 | how can I found who is using channels ? |
| 11:41.32 | pputman | Ast001, Pastebin your zaptel and zapata.conf. and what type of card is it? |
| 11:41.41 | Ast001 | openvox d110p |
| 11:42.13 | Ast001 | zaptel http://pastebin.com/d7f90e770 |
| 11:42.23 | *** join/#asterisk emiller (n=ed@65.208.79.2) |
| 11:42.34 | Ast001 | zapata http://pastebin.com/d366af96a |
| 11:42.54 | Ast001 | /proc/zaptel/ |
| 11:42.55 | Ast001 | http://pastebin.com/d578df5df |
| 11:46.19 | kamanashisroy | hi, in asterisk 1.2 .. does it support ael include ? |
| 11:46.20 | Ast001 | when I did zap destroy channel 1 2 3 4 .. and zap restart i got error message Unable to open D-channel 16 (Device or resource busy) |
| 11:47.01 | Ast001 | would one fine recompiling of zaptel solve the isue ? |
| 11:47.06 | *** join/#asterisk viraptor (n=viraptor@awh178.internetdsl.tpnet.pl) |
| 11:47.14 | Ast001 | how can i see wich thing is using zap channels |
| 11:48.21 | viraptor | hi all; does asterisk use rtcp for anything else than reporting statistics? I mean does rtcp affect it's behaviour in any way? |
| 11:48.28 | Ast001 | can it be notorios infamous capi service |
| 11:51.17 | Ast001 | here is dmesg http://pastebin.com/d1a215cb4 |
| 11:51.52 | Ast001 | HDLC Receiver overrun on channel WCT1/0/16 (master=WCT1/0/16) |
| 11:55.22 | Great_anta_Baka | how does asterisk detect the dtmf tones of sip calls if the dtmf info is not sent via sip info? |
| 11:55.36 | Ast001 | I think unable to open D Channe 16 iis key here |
| 11:55.55 | Ast001 | device or resouce busy |
| 11:57.26 | Ast001 | how can i chek what thing is using that channel |
| 11:57.43 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 11:59.25 | Ast001 | ok see you later |
| 12:01.23 | MCooper | I am looking to configure a couple of 7912g CIscos to work with SIP - has anyone gotten this to work? |
| 12:02.51 | *** join/#asterisk ccesario_ (n=ccesario@mailserver.damata.ind.br) |
| 12:04.04 | *** join/#asterisk rreck (n=rreck@wsip-70-169-164-165.dc.dc.cox.net) |
| 12:04.38 | rreck | sorry for the newbie question but im brand new and getting " 127.0.0.1 tried to authenticate with nonexistent user 'user'" |
| 12:05.07 | rreck | i see the answer here http://ubuntuforums.org/showthread.php?t=609803 but dont understand |
| 12:05.30 | rreck | please help me stop the hollerin |
| 12:10.08 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:10.08 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:10.52 | *** part/#asterisk my007ms (i=master@botmaster.x86.be) |
| 12:10.55 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177581608.dsl.bell.ca) |
| 12:11.44 | emiller | rreck: did you add the [user] context |
| 12:13.21 | rreck | no, i am sorry i dont know what that is. is that a section in the conf file? |
| 12:13.53 | rreck | i am completely brand new but entirely commited. the polycoms will be here within hours |
| 12:14.36 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 12:14.53 | dominic1 | Any guys from amooma here? |
| 12:14.57 | mort_gib | rreck: and someone who knows what they are doing will too I hope :-) |
| 12:15.40 | rreck | no, i just need a little help. i have been reading. i could run trixbox but im trying the install myself just because |
| 12:15.57 | mort_gib | So what's your question?? |
| 12:16.11 | rreck | sorry for the newbie question but im brand new and getting " 127.0.0.1 tried to authenticate with nonexistent user 'user'" |
| 12:16.15 | rreck | i see the answer here http://ubuntuforums.org/showthread.php?t=609803 but dont understand |
| 12:16.31 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:16.31 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:16.35 | rreck | emiller told me to add a context but im figuring out what that is |
| 12:17.00 | rreck | apparently its a user |
| 12:18.06 | mort_gib | Trixbox is like bad language in here... |
| 12:18.11 | mort_gib | ~thebook |
| 12:18.11 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 12:18.24 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:18.26 | rreck | yeah i got it and read most |
| 12:18.41 | mort_gib | In which case a context should be well known to you |
| 12:18.50 | rreck | ok, well its not |
| 12:19.04 | rreck | i asked if its a section in a config file |
| 12:19.13 | emiller | yes |
| 12:19.14 | emiller | it is. |
| 12:19.35 | mort_gib | The message you get is not really about a context though, the manager is trying to log in but user/passwd has not been set in /etc/asterisk/manager.conf |
| 12:19.51 | rreck | thanks. |
| 12:19.59 | mort_gib | More options apply though, so it's not enough to just set a username and password... |
| 12:20.14 | emiller | that is one part of your error... |
| 12:20.27 | mort_gib | A context is a logical separation of entries in a dialplan |
| 12:20.42 | rreck | ok, i understand. |
| 12:21.03 | rreck | /etc/asterisk/manager.conf is null |
| 12:21.10 | emiller | is null? |
| 12:21.18 | rreck | empty but exists |
| 12:21.23 | [TK]D-Fender | rreck: Then you have noone who CAN login. |
| 12:21.35 | mort_gib | Like you have two BRI interfaces, one for company acme and one for atom. You don't want Acme picking up Atom calls and vice versa |
| 12:21.43 | mort_gib | More to it though |
| 12:22.49 | dominic1 | is it possible to use hints with extstates instead of devstate? |
| 12:23.14 | [TK]D-Fender | dominic1: huh? |
| 12:23.58 | dominic1 | I saw a entry about a function called extstate, thought it would match better to my system with virtual numbers... |
| 12:25.21 | rreck | so i make a file in /etc/asterisk/manager.d called 'user' that contains 'secret = password' , but that isnt it |
| 12:25.59 | rreck | ill figure it out, thanks for the nudge |
| 12:26.44 | [TK]D-Fender | rreck: no, you fill in /etc/asterisk/manager.conf with the general connection (port, etc) settings, and the user entry with pass and PRIVILEGES <- |
| 12:26.56 | [TK]D-Fender | rreck: Go look at the SAMPLE config * came with |
| 12:27.01 | mort_gib | rreck: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf |
| 12:27.04 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 12:27.10 | mort_gib | TK he didn't install from source.... |
| 12:27.15 | [TK]D-Fender | .... \o/ |
| 12:27.34 | [TK]D-Fender | mort_gib: then he should DOWNLOAD it, because it comes bundled with DOCS <- |
| 12:27.44 | rreck | yeah, debian has a way of "helping" like that |
| 12:27.55 | mort_gib | :-) Good point, incidently that is what I did... |
| 12:27.57 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:27.57 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:28.45 | mort_gib | If you go through the trouble of installing Debian over more smooth installs, then using apt-get to install * makes no sense |
| 12:29.00 | mort_gib | -My twopence |
| 12:30.01 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 12:33.21 | [TK]D-Fender | mort_gib: ... none the richer ;) |
| 12:34.13 | mort_gib | :-) I know |
| 12:40.09 | jblack | That's what you get for linking the pound to the dollar. |
| 12:40.35 | *** join/#asterisk coppice (n=chatzill@6.196.17.210.dyn.pacific.net.hk) |
| 12:43.00 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 12:43.11 | gnorbert | What can be the problem, if the line works with GSM codec, but nothing can be heard, just for like 0.1 sec/sec |
| 12:43.56 | gnorbert | I mean when you have sound like: sound-no sound-sound-no sound |
| 12:44.27 | [TK]D-Fender | gnorbert: constantly throughout the entire call? |
| 12:44.35 | *** join/#asterisk AdamWest (n=Leif@CPE001d7e2f9574-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
| 12:44.36 | jblack | I had a problem with dropouts not long ago. The problem was cpu starvation under Xen. |
| 12:45.24 | gnorbert | [TK]D-Fender: Yes. |
| 12:45.52 | [TK]D-Fender | gnorbert: And if you use another codec? |
| 12:46.01 | gnorbert | But with linphonec, the welcome message can be perfectly heard. |
| 12:46.16 | [TK]D-Fender | gnorbert: that made no sense |
| 12:46.19 | gnorbert | It doesn!t work with speex and the same with alaw,ulaw |
| 12:46.21 | Great_anta_Baka | what could be possible reasons for voice mail not detecting dtmf but remote ivr's do? |
| 12:46.25 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:47.28 | jblack | gnorbert: ALso, I used to have a problem playing back sounds before I answered. I fixed that by doing an answer and a wait(1) before playing audio |
| 12:47.45 | [TK]D-Fender | Great_anta_Baka: 1st guess, your phone is set to inband which * ignores. Local voicemail never sees it. Outbound call however its jsut audio to them so yes, its "there". |
| 12:48.03 | [TK]D-Fender | jblack: he said its throughout the entire call |
| 12:48.05 | gnorbert | [TK]D-Fender:It starts with Answer and then WaitExten |
| 12:48.22 | jblack | That's right, he did. |
| 12:48.25 | [TK]D-Fender | gnorbert: So basically NO call works well with any codec? |
| 12:48.47 | gnorbert | That's right. |
| 12:49.05 | [TK]D-Fender | gnorbert: Where is your phone located relative to your server? |
| 12:49.13 | gnorbert | The same computer.. |
| 12:49.29 | gnorbert | But the same fault from the same lan, but not same computer. |
| 12:49.34 | Great_anta_Baka | [TK]D-Fender: how do i check what the phone is using becuase the sip.conf file is set to rfc2833 |
| 12:49.52 | [TK]D-Fender | Great_anta_Baka: on your PHONE. |
| 12:50.28 | Great_anta_Baka | polycom FTL.. really hard to navigate these menus |
| 12:50.49 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 12:50.58 | [TK]D-Fender | Great_anta_Baka: if its a Polycom and you never changed anything on this it should already be set to rfc2833. |
| 12:51.07 | Great_anta_Baka | ok sweet |
| 12:51.32 | [TK]D-Fender | Great_anta_Baka: pastebin your sip.conf masking only passwords. Then include the complete CLI output of a failed call to VM, and a "successful" one to an outside IVR |
| 12:51.34 | Great_anta_Baka | thats what i have set up but then that doesnt explain why i am not getting key tones registering with asterisk |
| 12:52.02 | Great_anta_Baka | ok do you want me to turn on sip debug for the peer i am testing on? |
| 12:52.11 | [TK]D-Fender | Great_anta_Baka: surprisingly, not yet |
| 12:53.09 | gnorbert | [TK]D-Fender: So the problem is that I should use speex, but I can't, even after changing iax.conf,sip.conf,extensions.conf,meetme.conf and installed speex and asterisk many times too.. |
| 12:53.32 | Great_anta_Baka | cool gimme a sec please |
| 12:53.54 | [TK]D-Fender | gnorbert: no, the problem is that apparently NO codecs are working. |
| 12:54.16 | gnorbert | That's the other problem.:) |
| 12:54.30 | [TK]D-Fender | gnorbert: Use a different computer for your testing. |
| 12:54.46 | gnorbert | But with GSM, I can connect to the server, but with speex I can't even do this... |
| 12:54.50 | gnorbert | I tried that too. |
| 12:55.28 | [TK]D-Fender | gnorbert: then pastebin a complete call attempt with SIP debug enabled from this OTHER compluter. |
| 12:56.16 | gnorbert | sip debug enabled? |
| 12:56.36 | [TK]D-Fender | gnorbert: yes |
| 12:57.03 | gnorbert | Where can I change that? |
| 12:57.07 | [TK]D-Fender | gnorbert: * CLI |
| 12:57.57 | gnorbert | Minute... |
| 12:58.09 | gnorbert | I have to look for an other computer.:D |
| 13:01.50 | Great_anta_Baka | ok [TK]D-Fender here is the config http://pastebin.com/m2f74c0c2 and here is first the completed call successfull to voicemail http://pastebin.com/m7cc88608 and here is one to a remote ivr and failing to recognize the dtmf http://pastebin.com/d94dc894 |
| 13:04.04 | gnorbert | [TK]D-Fender: With speex: |
| 13:04.13 | gnorbert | [Jul 11 15:03:04] NOTICE[11466]: chan_sip.c:5426 process_sdp: No compatible codecs, not accepting this offer! |
| 13:04.24 | [TK]D-Fender | Great_anta_Baka: [08:46]<Great_anta_Baka>what could be possible reasons for voice mail not detecting dtmf but remote ivr's do? |
| 13:04.33 | [TK]D-Fender | Great_anta_Baka: You are now telling me the REVERSE is true! |
| 13:04.40 | [TK]D-Fender | Great_anta_Baka: WTF? |
| 13:04.45 | Great_anta_Baka | sorry i got the first one confused |
| 13:04.58 | Great_anta_Baka | but it has been changing back and forth the whole day with every setting i change |
| 13:05.42 | [TK]D-Fender | Great_anta_Baka: [ 6] --> * Unknown Indication:20 pid:4 P[ 6] * IND : Digit 1 |
| 13:05.51 | [TK]D-Fender | Great_anta_Baka: looks like an misnd issue. |
| 13:06.02 | [TK]D-Fender | Great_anta_Baka: and you are using FreePBX which is NOT supported here. |
| 13:06.36 | Great_anta_Baka | i know i would go to the freepbx channel but no one answers there :( |
| 13:06.47 | [TK]D-Fender | Great_anta_Baka: Not our problem. |
| 13:07.07 | Great_anta_Baka | i guess but if its an misdn issue then its not a freepbx issue right? |
| 13:07.10 | [TK]D-Fender | Great_anta_Baka: Anyways get googling. You have error messages to look up now. |
| 13:07.41 | Great_anta_Baka | kk thanks |
| 13:08.08 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:09.22 | Great_anta_Baka | hey [TK]D-Fender lastly what does this mean "Starting Tones, we have inband Data" |
| 13:10.05 | [TK]D-Fender | Great_anta_Baka: http://www.google.ca/search?hl=en&q=%22Starting+Tones%2C+we+have+inband+Data%22&btnG=Google+Search&meta= |
| 13:10.38 | gnorbert | [TK]D-Fender: "Found no matching peer or user for '172.18.1.142:5061' |
| 13:10.39 | gnorbert | Found RTP audio format 110 |
| 13:10.39 | gnorbert | Found RTP audio format 101 |
| 13:10.39 | gnorbert | Peer audio RTP is at port 172.18.1.142:8000 |
| 13:10.39 | gnorbert | Found audio description format speex for ID 110 |
| 13:10.39 | gnorbert | Found audio description format telephone-event for ID 101 |
| 13:10.41 | gnorbert | Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x200 (speex)/video=0x0 (nothing), combined - 0x0 (nothing) |
| 13:10.44 | gnorbert | Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) |
| 13:10.47 | gnorbert | [Jul 11 15:08:58] NOTICE[11607]: chan_sip.c:5426 process_sdp: No compatible codecs, not accepting this offer!" |
| 13:11.01 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 13:11.23 | Great_anta_Baka | thank you once more |
| 13:11.27 | [TK]D-Fender | gnorbert: do NOT spam like that again. |
| 13:11.51 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.228.111) |
| 13:12.04 | gnorbert | Sorry, you told not to write in private, but I can, if you want.:S |
| 13:12.19 | mvanbaak | ~pb |
| 13:12.20 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:12.44 | *** join/#asterisk hi365_m (n=hi365@213.151.63.92) |
| 13:12.51 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 13:12.55 | [TK]D-Fender | gnorbert: And it clearly shows taht you did not configure * to ALLOW speex |
| 13:13.03 | gnorbert | Sorry, I'm new here, I'm going to do it this way next time. |
| 13:13.04 | mvanbaak | gnorbert: and the reason why it's not working is there loud and clear |
| 13:13.11 | [TK]D-Fender | gnorbert: indeed, PASTEBIN. |
| 13:13.30 | glaz | wow, gotta love the zaptel driver for freebsd, asterisk works just great now. |
| 13:13.44 | [TK]D-Fender | gnorbert: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x200 (speex)/video=0x0 (nothing), combined - 0x0 (nothing) <- "us" doesn't allow speex |
| 13:13.52 | mvanbaak | glaz: asterisk works great without zaptel as well |
| 13:14.07 | gnorbert | Where can I do that then? |
| 13:14.07 | *** join/#asterisk Peri (n=redanti@CPE0012171ab654-CM00080d9b7243.cpe.net.cable.rogers.com) |
| 13:14.16 | mvanbaak | gnorbert: in sip.conf |
| 13:14.18 | gnorbert | I already made it in sip.conf. |
| 13:14.28 | glaz | mvanbaak: yeah but I needed zaptel. |
| 13:14.35 | mvanbaak | then you have to make sure you compiled asterisk with speex support |
| 13:14.35 | [TK]D-Fender | gnorbert: then you did it wrong |
| 13:15.02 | [TK]D-Fender | gnorbert: go prove the codec module is compiled and loaded. |
| 13:15.22 | [TK]D-Fender | gnorbert: And then PASTEBIN your config masking only passwords |
| 13:15.33 | gnorbert | stop now |
| 13:15.40 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 13:15.45 | gnorbert | Sorry, wrong window. |
| 13:18.19 | gnorbert | I got the same message. |
| 13:18.35 | [TK]D-Fender | gnorbert: and you're showing us NOTHING. |
| 13:18.37 | gnorbert | Now in sip.conf there is not allow=all, but allow=speex |
| 13:18.58 | gnorbert | Minute, I try pastebin. |
| 13:20.42 | gnorbert | I've put it on pastebin.com |
| 13:21.41 | [TK]D-Fender | gnorbert: and it gave you a specific LINK you should be iving us |
| 13:22.03 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 13:22.06 | gnorbert | http://pastebin.com/m79881f99 |
| 13:22.19 | gnorbert | Then I think it must be that. |
| 13:23.19 | [TK]D-Fender | gnorbert: And AGAIN, you aren't showing your CONFIGS |
| 13:23.25 | mvanbaak | your asterisk is not supporting speex |
| 13:23.29 | [TK]D-Fender | gnorbert: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x200 (speex)/video=0x0 (nothing), combined - 0x0 (nothing) <-- same old error... |
| 13:23.42 | mvanbaak | like I said before, make sure you compiled asterisk with speex support |
| 13:23.55 | [TK]D-Fender | gnorbert: pastebin your config along with "show modules like codec" |
| 13:25.34 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 13:26.16 | gnorbert | http://pastebin.com/m7562508e |
| 13:26.55 | mvanbaak | gnorbert: that is not your whole sip.conf is it ? |
| 13:27.04 | gnorbert | It is. |
| 13:27.32 | mvanbaak | "Found no matching peer or user for '172.18.1.142:5061'" <--- means it is not using the [sip_user] section, but the global section |
| 13:27.54 | [TK]D-Fender | gnorbert: congratulations. Fix your phone's auth |
| 13:28.16 | [TK]D-Fender | gnorbert: and you should "disallow=all", and THEN "allow=" ony the 1 codec it should be using. |
| 13:28.24 | [TK]D-Fender | mvanbaak: Good catch. |
| 13:28.38 | mvanbaak | thank you |
| 13:28.50 | *** join/#asterisk ManxPower (n=manxpowe@61.sub-75-248-146.myvzw.com) |
| 13:29.16 | *** join/#asterisk moy (n=moy@nat/ibm/x-71a198f701fa6f3d) |
| 13:29.19 | mvanbaak | hhmm, is speex a good codec ? |
| 13:30.09 | [TK]D-Fender | mvanbaak: Largely a waste |
| 13:30.28 | *** join/#asterisk freezey (i=hidden-u@gw.mypublisher.com) |
| 13:30.34 | mvanbaak | k. I never played with it, and was wondering if it was worth my time to try it |
| 13:30.59 | *** join/#asterisk ccesario (n=ccesario@mailserver.damata.ind.br) |
| 13:31.03 | anonymouz666 | [TK]D-Fender: what about iLBC? |
| 13:31.13 | [TK]D-Fender | anonymouz666: in *'s hands... not so hot |
| 13:31.37 | mvanbaak | then I'll stick to G711 |
| 13:32.32 | [TK]D-Fender | mvanbaak: Should never have thought twice on it :) |
| 13:34.49 | mvanbaak | problem is that G711 takes a lot of bandwidth. Maybe it's time to look at G729 |
| 13:35.38 | [TK]D-Fender | mvanbaak: What are you connecting to * where BW's a concern? |
| 13:37.56 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:38.54 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 13:39.10 | *** join/#asterisk great_anta_baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-0602cdd98d6e98c4) |
| 13:39.38 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:40.49 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 13:40.52 | great_anta_baka | [TK]D-Fender: now that i have changed the dtmf to info it has switched around.. i can interact with the ivr but not the voicemail |
| 13:41.52 | Zeeek | Ladies and gentlemen... and [TK]D-Fender... good [morning|afternoon|evening|*] |
| 13:42.14 | *** join/#asterisk BipBip (n=BipBip@static-b5-253-32.telepac.pt) |
| 13:42.30 | [TK]D-Fender | \o/ --- (Yay others!) |
| 13:43.10 | Zeeek | had this been an actual alert, you would be instructed to tune to #asterix on your radio dial |
| 13:44.42 | Zeeek | Friday July 11 is SIP DAY on the VoIP Users Conference at http://bit.ly/voip in about 2.25 hours |
| 13:45.36 | [TK]D-Fender | "If this had been an actual emergency, the initial shockwave would have vaporized you and your entire neighbourhood. Those in outlaying regions are advised to flee in an organized panic to escape the radioactive fallout and mutant zombie it brings. |
| 13:46.49 | *** join/#asterisk emiller (n=ed@216.207.245.1) |
| 13:47.55 | *** join/#asterisk Mike8861 (i=Mike8861@221.124.81.21) |
| 13:48.02 | Mike8861 | hello all |
| 13:48.07 | Mike8861 | hows everyone doing |
| 13:48.16 | dvdevel | hey all, do the aastra (480i) support a jitter buffer setting? |
| 13:48.43 | ManxPower | dvdevel: virtially all phones do. |
| 13:48.58 | dvdevel | i agree, but i certainly don't find it in the web ui or in their admin guide... |
| 13:49.12 | ManxPower | dvdevel: then you'll have to contact their support. |
| 13:49.13 | Zeeek | Does anyone still own a radio with little triangles over 640 and 1240 Khz? |
| 13:49.23 | dvdevel | breaks out laughing |
| 13:49.37 | Zeeek | that was pretty funny, actually |
| 13:50.00 | dvdevel | oh mercy, that was a good one. seriously though, i've found their support to be slightly less than helpful. |
| 13:50.01 | ManxPower | dvdevel: I suppose we could wait for the Psychic Asterisk Squad to pull the information out of the ether..... |
| 13:50.14 | ManxPower | ~phones |
| 13:50.14 | jbot | i guess phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 13:50.32 | tzanger | that is a little long |
| 13:50.36 | Zeeek | the butracher scrapes an infinite number of local phone web interfaces every night |
| 13:50.40 | ManxPower | Any decent phone will have a jitter buffer. Keep in mind that what sounds like jitter could be some other issue. |
| 13:50.43 | tzanger | jbot gets cut off after "places like such as" |
| 13:50.50 | dvdevel | hey, i know ManxPower, and i agree. quality on the aastra is rock solid until it's not. |
| 13:50.55 | Mike8861 | I have a problem, while i am dial out using SIP outbound trunks, i hear "all circuits are busy now" (but they are not!!!). is there log i can check ? or does asterisk do no log by default ? |
| 13:51.14 | ManxPower | Mike8861: analog or digital or voip? |
| 13:51.22 | Zeeek | Mike8861 watch the CLI output |
| 13:51.34 | tzanger | actually a friend of mine has a polycom or aastra phone (I can't remember which offhand) -- when someone calls him, he gets 5s to say "I'll call you right back" before the call gets disconnected. He can call out and talk all day though |
| 13:51.40 | Mike8861 | ManxPower: a VOIp SIP trunk provide by VOIP provider |
| 13:52.13 | ManxPower | Mike8861: then as with everything the CLI output is your friend. |
| 13:52.15 | [TK]D-Fender | Mike8861: pastebin is your friend... |
| 13:52.17 | Mike8861 | Zeeek: sorry, this does not always happens, is there Log beside asterisk -r realtime output for futher trace back ? |
| 13:52.41 | dvdevel | well, i've "reproduced" the problem with a packet capture and replaying with the jitter buffer down to 5ms. even if i tweak it up to 10ms it's good enough... |
| 13:52.47 | [TK]D-Fender | Mike8861: show us the problem or we can't help you. |
| 13:53.00 | ManxPower | Mike8861: Unless you tell asterisk to do so it won't play that message. I suspect the issue is with your ITSP/carrier, not Asterisk |
| 13:53.05 | dvdevel | i'll have "someone" contact aastra support and report back. |
| 13:53.30 | ManxPower | dvdevel: all phones that I've heard of default to at least 20ms jitter buffer |
| 13:53.46 | [TK]D-Fender | dvdevel: A quick Google search for it showed up Aastra's in a list with a ton of other models where they all list jitter buffer, but Aastra DOESN'T strangely enough... |
| 13:53.59 | ManxPower | actually 40ms, no point in having a 20ms jitter buffer on VoIP |
| 13:54.19 | Zeeek | don't they have a site or forum or sthing? |
| 13:54.25 | dvdevel | yes, ManxPower, i initially thought the customer was crazy, because ethereal defaults to 50ms, and the call was perfect! |
| 13:54.29 | Zeeek | Twitter them! |
| 13:54.30 | Mike8861 | ManxPower: i understand without Output theres no help, in case it happens in future, what should i do to log it down ? |
| 13:54.46 | Mike8861 | asterisk -r is for realtime only, theres no record.... |
| 13:54.53 | ManxPower | Mike8861: So if you do an "asterisk -rvvv" and make a call nothing is printed on the screen? |
| 13:55.05 | ManxPower | No, -r IS NOT FOR REALTIME. It is for REMOTE CONSOLE> |
| 13:55.19 | Zeeek | and all these years I thought it was asshat!!!!! |
| 13:55.27 | Mike8861 | can i enable all asterisk -r with timestamp to a file ? |
| 13:55.30 | Zeeek | no, that asterisk -a |
| 13:55.43 | Mike8861 | Zeeek: ohh!!!!! |
| 13:55.48 | ManxPower | Mike8861: never use -r, always use at least 3 v's in addition to -r |
| 13:56.10 | [TK]D-Fender | ManxPower: ummm... -r" ... |
| 13:56.18 | *** join/#asterisk masus (i=masus@88.248.14.186) |
| 13:56.21 | [TK]D-Fender | ManxPower: nvm |
| 13:56.21 | masus | hi all , [Jul 11 16:30:29] ERROR[19715]: chan_agent.c:1053 agent_new: A blocker exists after agent channel ownership acquired. Get this message and one of the agents cant login . Does aynone know why ? |
| 13:56.27 | [TK]D-Fender | ManxPower: jsut got the context of that. |
| 13:56.27 | *** join/#asterisk coppice (n=chatzill@6.196.17.210.dyn.pacific.net.hk) |
| 13:56.37 | ManxPower | [TK]D-Fender: only on 2nd cup of coffee |
| 13:56.44 | [TK]D-Fender | ManxPower: mind you I always just do "-r" and follow "set verbose 10" :) |
| 13:56.52 | Mike8861 | ManxPower: so it should be Asterisk -rrr FILENAME.TXT ?? |
| 13:57.03 | ManxPower | [TK]D-Fender: mike is so new he still has the dirt from the turnip field |
| 13:57.07 | [TK]D-Fender | Mike8861: no "filename' |
| 13:57.17 | ManxPower | Mike8861: I can't help you if you don't listen. I said "asterisk -rvvv" |
| 13:57.18 | [TK]D-Fender | Mike8861: for simplicity's sake : asterisk -rvvvvvvvvvv |
| 13:57.21 | Mike8861 | how can i log the output |
| 13:57.24 | Zeeek | asterisk -h ; hang up all calls after 10 seconds of bad quality audio |
| 13:57.30 | [TK]D-Fender | Mike8861: Copy & f'n PASTE |
| 13:57.33 | ManxPower | Mike8861: IT LOGGS BY DEFAULT |
| 13:58.02 | mvanbaak | module unload logger.so |
| 13:58.05 | mvanbaak | ;) |
| 13:58.12 | russellb | it doesn't log _all_ of the console output by default ... |
| 13:58.13 | Zeeek | isn't there like a logging.conf ? |
| 13:58.15 | ManxPower | Mike8861: you can either argue and keep asking silly questions and someone else can help you OR you can start following our advice and I can help you. Choose. |
| 13:58.17 | mvanbaak | 'I see nothing in the logfile' |
| 13:58.29 | mvanbaak | Zeeek: yeah. logger.conf |
| 13:58.38 | Mike8861 | ManxPower: sorry, but i am confused, dont meant to be offense |
| 13:58.40 | Zeeek | LIKE %log%.conf |
| 13:58.52 | russellb | asterisk -rvvvvvvvvv | tee logthisconsole.txt |
| 13:59.07 | Mike8861 | @russellb: thanaks |
| 13:59.08 | Zeeek | Mike8861 look at logger.conf and then google on that for examples |
| 13:59.25 | ManxPower | Mike8861: OK, do an "asterisk -rvvv" that will connect you to the Asterisk console, then do a call. take the screen output and put it on pastebin.ca. If you get no screen output when using "asterisk -rvvv" then your install is seriously screwed up and we need to fix that first. |
| 13:59.39 | Mike8861 | Zeeek: thanks |
| 14:00.07 | Mike8861 | ManxPower: I will open asterisk -rvvv at all time, and when problem occues, I will post the screen! thanks for everyone helped |
| 14:00.12 | Zeeek | and here the first: http://bit.ly/logger |
| 14:00.48 | ManxPower | Mike8861: I did not say "make a problem call" We can learn much by seeing a call without a problem. |
| 14:01.09 | masus | hi all , [Jul 11 16:30:29] ERROR[19715]: chan_agent.c:1053 agent_new: A blocker exists after agent channel ownership acquired. Get this message and one of the agents cant login . Does aynone know why ? |
| 14:01.42 | [TK]D-Fender | Mike8861: No... Show us the problem :) |
| 14:01.56 | Mike8861 | [TK]D-Fender: i am so noob >_< |
| 14:01.59 | [TK]D-Fender | masus: No more than we did 5 minutes ago. |
| 14:02.05 | ManxPower | Mike8861: logging to FILES is set in /etc/asterisk/logger.conf and the files it logs are normally in /var/log/asterisk |
| 14:02.13 | *** join/#asterisk Rem| (n=remi@bg-fw2out.monmouth.com) |
| 14:02.15 | masus | [TK]D-Fender: :/ ok |
| 14:02.24 | Mike8861 | ManxPower: thx |
| 14:03.03 | ManxPower | Mike8861: but in my 5 years of using Asterisk I've only actually had to look at the log files a few times, every other time I just used the CLI logging to the screen. |
| 14:03.58 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:04.10 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 14:04.15 | jaytee | I just find it annoying when I'm using putty and I'm trying to select some of the console output and then a call in progress makes it jump around |
| 14:04.43 | Mike8861 | ManxPower: oh, so CLI output can resolve most issue |
| 14:04.59 | Zeeek | The CLI tells the story |
| 14:05.01 | ManxPower | Mike8861: Correct |
| 14:05.08 | *** join/#asterisk afink (n=afink@asa.redglaze.com) |
| 14:05.18 | ManxPower | I use the Asterisk Command Line Interface Terminal all the time. |
| 14:05.26 | Zeeek | except when the message "we should never get here" comes up. |
| 14:05.27 | jaytee | lotsa lotsa stuff available for debuggin thru the console |
| 14:05.34 | Mike8861 | well, I was on a trip last night to china |
| 14:05.35 | ManxPower | wonders if anyone gets the acronym |
| 14:05.38 | Zeeek | russellb where is that one? |
| 14:05.38 | Kobaz | okay, so i have a rhino rcbfxx card with some fxo's.... every so often the volume levels on the line drop off randomly, and then come back |
| 14:05.50 | Zeeek | It's like DOS |
| 14:05.50 | Kobaz | what's the more likly source of the problem... the telco or the line card? |
| 14:05.58 | ManxPower | Kobaz: almost nobody uses Rhino cards here. |
| 14:06.01 | Mike8861 | the hotel bathroom got a see though glass, like those in sex scene |
| 14:06.02 | Kobaz | i know |
| 14:06.13 | ManxPower | but even then I've never heard of volume randomly changing |
| 14:06.14 | Kobaz | but we have a stack of 4 of these things and we tried to get rid of them |
| 14:06.26 | Kobaz | so now their on a customer site and they might be flaking out |
| 14:06.57 | Mike8861 | however, china blocks some domain, so we cannot connect to our asterisk server |
| 14:07.00 | ManxPower | Kobaz: I have no idea of Rhino makes good cards or not, but not many people use them, so just because of support issues, I can't recommend them. |
| 14:07.14 | Kobaz | ManxPower: oh i know, i dont recommend them either |
| 14:07.16 | Zeeek | So I talked to Olle Johannesen and he had the nerve to be flying over Europe during the conference |
| 14:07.25 | ManxPower | Kobaz: any OTHER errors like IRQ misses or HDLC messages? |
| 14:07.29 | Kobaz | ManxPower: their hardware/drivers are pretty terrible |
| 14:07.32 | Mike8861 | Kobaz: RHino card got good pricese |
| 14:07.57 | Kobaz | ManxPower: i'll check irq misses if i can find it... this is analog not digital |
| 14:08.20 | ManxPower | Kobaz: analog would not have HDLC aborts, but could have IRQ misses, if you are getting IRQ misses, that could be one of the issues. |
| 14:08.32 | Kobaz | irq misses: 0 |
| 14:08.41 | Kobaz | yeah i know, analog doesnt use hdlc |
| 14:08.50 | ManxPower | Kobaz: It sucks to be you. |
| 14:10.57 | Zeeek | OT but may be iof interest: N95 acting as modem for linux: http://linux.sgms-centre.com/nokiafaq/mobile_broadband/ |
| 14:11.21 | errr | Zeeek: sweet =) |
| 14:11.49 | errr | I love my n95 8gb |
| 14:11.59 | errr | I wouldnt trade it for 3 iphones |
| 14:12.13 | *** join/#asterisk kc2tnk (n=fskrotzk@host.textwise.com) |
| 14:12.24 | Mike8861 | Zeeek: maybe you should try it with your n95 |
| 14:12.36 | Mike8861 | Zeeek: pls post vid to youtube |
| 14:12.39 | *** part/#asterisk afink (n=afink@asa.redglaze.com) |
| 14:13.03 | Kobaz | manxpower: i'll populate up some fxo's on one of the extra rhino cards and test for volume randomness |
| 14:13.35 | Zeeek | more on that site: http://howto-pages.org/ |
| 14:14.09 | Zeeek | Mike8861 maybe you should read this: http://howto-pages.org/asterisk/ |
| 14:15.33 | Mike8861 | Zeeek: thank you, I am so happy, that guide might help me a lot |
| 14:15.45 | Mike8861 | Zeeek: bookmark it, and share with everyone!!! |
| 14:16.07 | Mike8861 | Zeeek: jerjer guide are also good one! |
| 14:16.11 | Zeeek | there's another great bookmark: http://google.com |
| 14:16.30 | Mike8861 | ~jerjer |
| 14:16.31 | jbot | somebody said jerjer was the guy who runs nufone |
| 14:16.32 | *** join/#asterisk kc2tnk (n=fskrotzk@host198.textwise.com) |
| 14:16.38 | [TK]D-Fender | Zeeek: exten => _NXXXXXX,1,Set(CALLERID(ALL),Your Name <11011234567>) <- broken. Why you should take all "guides" with a grain of salt |
| 14:16.39 | *** join/#asterisk spokra (n=spokra@host093-179-153.sea0.speakeasy.net) |
| 14:16.51 | Mike8861 | =_=!!! why jbot dont respose to ~jerjer |
| 14:16.51 | Zeeek | Try it: http://www.google.com/search?q=Mike8861 |
| 14:16.56 | *** join/#asterisk korihor (n=korihor@190.39.163.45) |
| 14:16.57 | Mike8861 | well, anyway, heres the link http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 14:17.08 | [TK]D-Fender | Mike8861: if jsut did. |
| 14:17.23 | [TK]D-Fender | Mike8861: and you didn't get what you expected because your keyword was wrong. |
| 14:17.27 | [TK]D-Fender | ~jerjerguide |
| 14:17.28 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 14:17.39 | Zeeek | ~seen JerJer |
| 14:17.43 | jbot | jerjer <n=PhatJ@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #asterisk-dev, 2h 38m 5s ago, saying: 'mvanbaak: at least your Monkey's Uncle is still there for ya :)'. |
| 14:18.06 | Mike8861 | [TK]D-Fender: hmm....jbot dont have fuzzy logic yet ? |
| 14:18.31 | Zeeek | I complained about comcast moh on twitter and they got mack to me! Still can't fix problem tho. |
| 14:18.40 | [TK]D-Fender | Mike8861: Yes, it was very fuzzy and had to be thrown out as a biohazard. |
| 14:18.42 | Zeeek | s/mack/back/ |
| 14:20.23 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:22.34 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 14:22.34 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:29.36 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 14:30.29 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 14:34.12 | Mike8861 | any one knows gray audin ? |
| 14:36.16 | *** join/#asterisk ccesario (n=ccesario@mailserver.damata.ind.br) |
| 14:38.17 | viraptor | is there any way to force standard rtp forwarding instead of p2p bridging? |
| 14:40.01 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 14:40.35 | [TK]D-Fender | viraptor: Where P2P = reinvite? |
| 14:41.25 | viraptor | no... packet2packet -> forwaring rtp without repacking contents |
| 14:41.28 | coppice | P2P == a ferry route |
| 14:42.02 | viraptor | I want to force asterisk to go through ast_frame-s |
| 14:46.40 | *** join/#asterisk queuetue (n=scott@MTRLPQ02-1279391519.sdsl.bell.ca) |
| 14:47.32 | Kyoshi | is there a native freetds driver that asterisk can use for realtime or does it have to be odbc? |
| 14:48.06 | queuetue | Hi. I just had 4 grandstream 100's just stop working on me, after over a year of service, simultaneously. Softphones are still working and sipura 1000's I dug out of the closet to replace them are working fine, but the grandstreams just died. Is there some time-sensitive firmware problem? |
| 14:49.33 | queuetue | The phones seem to be working fine in every capacity except actually registering. The built-in webserver is fine, they save config changes, negotiate STUN, etc. I'll try some internal tests later, but this was a very weird problem. |
| 14:49.47 | tristanbob | queuetue, lighting strike? power surge? |
| 14:50.03 | queuetue | tristanbob, I'd be right with you, but one wasn't plugged in at the time. |
| 14:50.21 | [TK]D-Fender | queuetue: Well, something changed... either your * settings, the phone's, or they just flaked out. They seem to have a time-bomb-warranty :) |
| 14:50.56 | queuetue | [TK]D-Fender, That's how it feels to me, as well, which would really tick me off. (Oh, well, everyone hated those phones anyway.) |
| 14:51.27 | [TK]D-Fender | queuetue: And its not like well tell everyone to avoid them or anything... |
| 14:51.38 | queuetue | Oh, do you? :) |
| 14:51.40 | Mike8861 | queuetue: save ur money, and buy snom or aastra if you are budget concern |
| 14:52.01 | Mike8861 | queuetue: if u have resources, look for polycom |
| 14:52.10 | *** join/#asterisk badcfe (i=christia@peter.mindslice.net) |
| 14:52.19 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 14:52.29 | badcfe | i get messages like this |
| 14:52.30 | badcfe | NOTICE[29006]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/2539047903-00a863f0' not posted |
| 14:52.44 | badcfe | and i dont have any cdr*.conf activated |
| 14:52.54 | queuetue | Just to add insult to injury, "Firefox can't find the server at www.google.com." - is google down? :) |
| 14:52.57 | [TK]D-Fender | queuetue: ... |
| 14:52.59 | badcfe | 1.4.13 |
| 14:53.00 | [TK]D-Fender | ~gs |
| 14:53.01 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 14:53.04 | [TK]D-Fender | ~grandstream |
| 14:53.05 | jbot | [grandstream] the Yugo of VoIP hardware. Run. Run away now. |
| 14:53.17 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 14:53.35 | badcfe | i do Transfer. thats when i get the CDR not posted NOTICE |
| 14:53.57 | badcfe | wonder if there is a way to shut up those NOTICE messages |
| 14:54.26 | Mike8861 | ~jbot |
| 14:54.27 | jbot | [jbot] a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch, or suck, or a pain in the ass |
| 14:54.32 | *** join/#asterisk bored_kid (n=bored_ki@70.89.49.188) |
| 14:54.46 | Mike8861 | [TK]D-Fender: 0.0 jbot knows yourname ??? |
| 14:55.35 | [TK]D-Fender | Mike8861: My bitch! |
| 14:55.49 | Mike8861 | [TK]D-Fender: how to man jbot ? |
| 14:56.14 | Mike8861 | ~man |
| 14:56.15 | jbot | Tem and Tain are the man. |
| 14:57.10 | Mike8861 | ~Mike8861 |
| 14:57.26 | Mike8861 | come on, say something! am I talking to a dead bot |
| 14:57.39 | Mike8861 | ~[TK]D-Fender |
| 14:57.40 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 14:57.53 | Mike8861 | ~\ManxPower |
| 14:57.57 | Mike8861 | ~ManxPower |
| 14:57.58 | jbot | manxpower is probably Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. Contact eric@fnords.org |
| 14:58.19 | Mike8861 | ~Kerrg |
| 14:58.23 | Mike8861 | ~kerryG |
| 14:58.39 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:58.41 | Mike8861 | ~@Qwell |
| 14:59.08 | ManxPower | I'm me! |
| 14:59.14 | [TK]D-Fender | Mike8861: open a private chat with job and stop spamming in here. |
| 14:59.37 | [TK]D-Fender | ManxPower: You can't be "me, **I"m Me**, You're "YOU"! Sheesh! |
| 14:59.41 | Mike8861 | * /msg: insufficient parameters <-- my irc dun let me message with jbot... |
| 15:00.11 | [TK]D-Fender | Mike8861: open a private chat window. Go learn how to use your client. |
| 15:00.45 | Mike8861 | [TK]D-Fender: okay |
| 15:01.01 | Mike8861 | I did find the read me, but jbot not talk with me |
| 15:01.14 | Mike8861 | Waiting for acknowledgement... <--- |
| 15:01.21 | ManxPower | Mike8861: in MY irc client "/msg jbot hello world!" |
| 15:01.56 | Mike8861 | ManxPower: thanks, it works |
| 15:02.50 | ManxPower | Mike8861: learn to use your IRC client |
| 15:03.16 | spokra | hehehe loading iphone 2.0 firmware now!! |
| 15:06.45 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 15:17.22 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 15:18.14 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 15:19.00 | *** join/#asterisk grEvenX (n=even@85.221.107.130) |
| 15:21.14 | Kobaz | hmmmmmm |
| 15:21.25 | Kobaz | how do you do custom ring tones on a polycom, it's nowhere in the docs |
| 15:22.09 | [TK]D-Fender | Kobaz: Yes, it is. |
| 15:22.19 | [TK]D-Fender | Kobaz: in sip.cfg you set your alertType. |
| 15:22.25 | Kobaz | hmm |
| 15:22.33 | [TK]D-Fender | Kobaz: And you can see a sample of this in the "paging" WIKI page |
| 15:22.50 | Kobaz | ah |
| 15:23.07 | [TK]D-Fender | Kobaz: Now go read the stock firmware's sip.cfg and the admin guide. |
| 15:24.37 | Kobaz | heh |
| 15:24.47 | Kobaz | i have the admin guide, i've been searching a while |
| 15:25.19 | Kobaz | i'm finding, i'm finding |
| 15:26.27 | Zeeek | what do you hate about SIP? Come and rant at http://bit.ly/sip in about 30 m call sip:123@ts.x2z.eu 22622# 1# |
| 15:27.51 | Qwell | Zeeek: you reaaaaallllyyyy want people to come troll on SIP? |
| 15:28.00 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 15:28.01 | Qwell | Seems futile. :) |
| 15:28.18 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 15:28.44 | [TK]D-Fender | Zeeek: So you want to hear waht people hate about SIP... by having them CALL YOU with it? http://www.ratemyeverything.net/post/4065/Illiterate.aspx |
| 15:29.12 | Qwell | [TK]D-Fender: nice |
| 15:29.23 | Zeeek | no |
| 15:29.29 | Qwell | that's like the old Hooked on Phonics phone number.. |
| 15:29.33 | Zeeek | what do you like about SIP? Come and praise it at http://bit.ly/sip in about 30 m call sip:123@ts.x2z.eu 22622# 1# |
| 15:29.43 | Qwell | 800-abcdefg |
| 15:30.04 | Zeeek | like my driving? Call 800 fsckyou |
| 15:30.45 | Zeeek | At VoIP Users COnference, your call is important to us. For quality control, your call may be recorded. |
| 15:32.03 | *** join/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 15:34.19 | Zeeek | I can't believe I'm talking to comcast on Twitter and it's still taking forever to get someone |
| 15:36.50 | *** join/#asterisk ManxPower (n=manxpowe@61.sub-75-248-146.myvzw.com) |
| 15:38.09 | Qwell | Zeeek: Comcast? Aren't you in France? |
| 15:38.12 | jaytee | illiteracy is a seriousl problem in this country. Someone in the Dept of Education wrote an important memo for the present administration, unfortunately the head of the Executive branch couldn't read it because he's illiterate. "Our childrens is learning!" |
| 15:38.24 | jaytee | serious -l |
| 15:38.29 | *** join/#asterisk oilinki (n=oil@ppp-124-120-16-135.revip2.asianet.co.th) |
| 15:38.33 | Zeeek | Qwell yes but the issue is an email blocklist for my customer |
| 15:38.58 | Qwell | people use comcast mail? |
| 15:39.16 | Zeeek | I dunno, I'm just here to serve the needs of my customer |
| 15:39.37 | Zeeek | First we had her change to port 587 because Orange now blocks 25 |
| 15:39.52 | jaytee | I have Comcraptastic as an ISP but I use Gmail because they don't have such a low limit on email attachment size. I'm not sure what Gmail's attachment limit is. |
| 15:40.00 | Zeeek | when she finally got that working there's a problem with her assistant receiving mail |
| 15:40.39 | [TK]D-Fender | jaytee: Go run your own MTA, no limit you don't set :) |
| 15:41.04 | Zeeek | when you run your own, you definitely WILL set limits |
| 15:41.23 | Zeeek | remember the mailbombs of 1998 |
| 15:41.37 | jaytee | [TK]D-Fender, yeah and I can always override the limit on our Exchange box for my account but I have no control over the other end until I perfect my mind control ray and become Emperor of Earth. |
| 15:41.58 | ManxPower | our mail server limits are usually about 30MB per message, 250GB quota |
| 15:42.16 | ManxPower | ..e.r. 300MB quota |
| 15:42.38 | jaytee | Comcrapcast is under 10MB, I think it's around 5MB actually |
| 15:42.49 | Zeeek | mine is at 10M. Anyone needs to send more there are servral excellent services that do this via the web. |
| 15:42.51 | ManxPower | The president of the company got an automated "your' using 80% or more of your quota" message today and he called the helpdesk all in a panic |
| 15:42.53 | Qwell | Zeeek: if somebody can send me a 100gb message, to use up my HD... |
| 15:43.01 | Qwell | I'll be more impressed, than anything |
| 15:46.29 | Zeeek | <PROTECTED> |
| 15:48.29 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:48.51 | [TK]D-Fender | Qwell: ... isn't there a chan_skinny botnet plugin for that? ;) |
| 15:51.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:51.08 | *** join/#asterisk emiller (n=ed@216.207.245.1) |
| 15:51.59 | emiller | is this where we find all the answers to the dcap? :) |
| 15:52.43 | ManxPower | The answers you seek are *within* you, grasshopper. |
| 15:52.50 | *** join/#asterisk IamTux (n=kman@64.5.9.121) |
| 15:52.57 | emiller | looks within |
| 15:53.32 | *** part/#asterisk emiller (n=ed@216.207.245.1) |
| 15:57.14 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 15:57.51 | Kobaz | hmmm |
| 15:58.01 | Kobaz | now i need a regular phone ring ringtone |
| 15:58.10 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 15:58.31 | IamTux | what phone are you using ? |
| 15:58.53 | Kobaz | polycom |
| 15:58.55 | Kobaz | 320 |
| 15:59.11 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 15:59.19 | Kobaz | this customer gets annoyed by the built in ringtones, and they all sound the same |
| 15:59.25 | Kobaz | i dont blame her |
| 15:59.26 | Kobaz | heh |
| 16:00.03 | IamTux | Mmm I think those phone you actually define the ringtones in the config, the pitch of the ring and duration that kind of stuff |
| 16:02.12 | *** join/#asterisk af_ (n=getsmart@88-149-230-202.dynamic.ngi.it) |
| 16:03.20 | [TK]D-Fender | Kobaz: Then go record one and use it. |
| 16:03.46 | [TK]D-Fender | Kobaz: I use a Cisco "24" warble myself. |
| 16:04.04 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:04.49 | *** join/#asterisk geek_cl (n=geek@190.54.42.62) |
| 16:05.44 | Kobaz | ah |
| 16:06.03 | *** part/#asterisk routerweasel (n=4stfed@core.spokanecomputing.com) |
| 16:06.27 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:07.22 | IamTux | here is a how-to for the 500s they have basically the same configs |
| 16:07.24 | IamTux | http://www.voipphreak.ca/archives/78 |
| 16:10.09 | *** join/#asterisk Ast001 (n=Administ@cable-89-216-185-58.dynamic.sbb.rs) |
| 16:10.58 | Ast001 | I have just seen another dimension of my problem.In /var/log/messages i can see dozens of messages Dobule miss (190,54) It show up 10x in second |
| 16:11.36 | Ast001 | and it stops when i unplug isdn-openvoc d110p cable |
| 16:11.44 | [TK]D-Fender | Ast001: And what is your "problem"? |
| 16:11.58 | Ast001 | can not receive any calls |
| 16:12.15 | Ast001 | ERROR[19612]: chan_zap.c:9470 start_pri: Unable to open D-channel 16 (Devuce or resource busy) |
| 16:12.19 | ManxPower | Ast001: The ACTUAL error message is helpful |
| 16:12.26 | Ast001 | ERROR[19612]: chan_zap.c:10582 build_channels: Unable to register channel '1-15' |
| 16:12.36 | [TK]D-Fender | Ast001: And I recall asking you for a PASTEBIIN of the complete CLI output with intense PRI debug which you never provided |
| 16:12.40 | ManxPower | Ast001: you have a config problem |
| 16:12.53 | Ast001 | i provided |
| 16:13.13 | [TK]D-Fender | Ast001: pastebin your configs, and the output of ztcfg -vvvv |
| 16:13.22 | Ast001 | zaptel.conf http://pastebin.com/d578df5df |
| 16:13.31 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 16:14.09 | Ast001 | /proc/zaptel/1 http://pastebin.com/d578df5df |
| 16:14.36 | [TK]D-Fender | Ast001: try again.... |
| 16:15.18 | Ast001 | ok just a sec |
| 16:16.42 | Ast001 | here it is http://pastebin.com/d6c74fa73 |
| 16:16.46 | Ast001 | ztcfg -vvvv |
| 16:17.24 | [TK]D-Fender | Ast001: And your zaptel & zapata confs please. and then reload chan_zap in * CLI. |
| 16:18.11 | Ast001 | zaptel http://pastebin.com/d7f90e770 |
| 16:18.23 | Ast001 | zapata http://pastebin.com/d366af96a |
| 16:19.08 | Ast001 | zaptel and zapata is from another call center where digium card is running and everything works fine |
| 16:19.09 | [TK]D-Fender | Ast001: 14-20 in your zapata can cause problems... |
| 16:19.28 | Ast001 | what to do with them ? |
| 16:19.38 | [TK]D-Fender | Ast001: show us THIS one. reload chan_zap and enable intense debug and place a a call to it |
| 16:19.56 | [TK]D-Fender | Ast001: leave them for this test. then comment them all out and put "pridialplan=national" |
| 16:20.03 | [TK]D-Fender | (for the next test as needed) |
| 16:20.10 | Ast001 | ok |
| 16:21.18 | *** join/#asterisk bool2 (n=Bool@176-65-19-212-hildon-t15-z1.wireless.as15758.net) |
| 16:21.54 | Ast001 | can you give me stright command to reload chan_zap as well for intense debug please ? |
| 16:22.16 | pputman- | module reload chan_zap.so pri intense debug span 1 |
| 16:22.57 | M1s3ry | pputman-, go to sleep! |
| 16:23.15 | pputman- | I never sleep |
| 16:23.56 | [TK]D-Fender | slams the toothpick propping pputman's eyelids open back in place. |
| 16:24.01 | Ast001 | a lot of warnings ignoring swithtype pridialplan prilocal dialplan Ignoring signalling etc... and then Reconfigured channel 1, ISDN PRI signalling |
| 16:24.03 | pputman- | ponders exactly who M1s3ry is. |
| 16:24.14 | Qwell | pputman-: some nub |
| 16:24.38 | M1s3ry | walks upstairs to knock Qwell in the forehead |
| 16:24.39 | [TK]D-Fender | Ast001: Guess those options were all bad.. |
| 16:25.12 | jeev | damn, i posted a $399 e8300 dell inspiron on slickdeals and got a thumbs down, what morons |
| 16:25.31 | macros73 | Having an issue with calling out on Asterisk. Phone || NAT || Asterisk <---> ITSP. Phone can access voicemail on the server without problem. When I try to call out, though, it sits there on Standby. Invite goes out, but no return from the ITSP. |
| 16:26.00 | [TK]D-Fender | macros73: pastebin the complete call with SIP debug. |
| 16:26.47 | IamTux | macros73: what is the nat device ? |
| 16:27.06 | [TK]D-Fender | IamTux: Highly likely irrelevant, |
| 16:27.49 | IamTux | [TK]D-Fender: I have had lots of issues with nat in the past and it was all the firewall device |
| 16:29.34 | [TK]D-Fender | IamTux: Then again he just said the phont > * is FINE. that means NAT isn't an issue between them. |
| 16:30.06 | [TK]D-Fender | IamTux: his ITSP settings and likely reinvites specifically are probablyt he problem. |
| 16:30.38 | Ast001 | I've seen a lot of these http://pastebin.com/d5d14f105 |
| 16:31.01 | [TK]D-Fender | Ast001: Yeah, its still sharing IRQs, isn't it? |
| 16:31.14 | macros73 | Looking this over again before I paste. Vitelity may be doing something dumb. They told me to set my inbound and outbound to both use inbound16.vitelity.net as part of troubleshooting earlier voice quality issues. But I am seeing responses from inbound6. |
| 16:31.32 | Ast001 | no it is not |
| 16:31.49 | Ast001 | /proc/interrupts http://pastebin.com/d2a4b160a |
| 16:32.08 | Ast001 | its alone on 12 |
| 16:32.32 | [TK]D-Fender | Ast001: 20: 66668 66865 67562 65643 IO-APIC-fasteoi wcte11xp <- not sure on the "fastoi" part.. |
| 16:32.36 | Ast001 | buf is filling /var/log/messages with Double miss messages |
| 16:32.53 | Ast001 | whats that fastoi ? |
| 16:33.35 | [TK]D-Fender | Ast001: Either way your card looks to be having issues. |
| 16:34.12 | [TK]D-Fender | Ast001: Now please stop looking at all of this distracting junk, and PLACE THE DAMN CALL. |
| 16:34.45 | Ast001 | I tryed few secs ago I got nothing and after few secs busy signal |
| 16:34.59 | Ast001 | and on cli nothing |
| 16:35.51 | Ast001 | Cli is filiing with these http://pastebin.com/d5d14f105 |
| 16:35.52 | [TK]D-Fender | Ast001: if you get NOTHING with debug then your telco isn't even sending anything to you. |
| 16:36.11 | [TK]D-Fender | Ast001: Go call them and ask them to monitor asw you place the call |
| 16:36.17 | [TK]D-Fender | as* |
| 16:36.31 | Ast001 | i called them and few guys come here with tester connect tester to isdn and received a call |
| 16:36.42 | Ast001 | and talked |
| 16:36.54 | Ast001 | and said it is not our problem it is your equipment |
| 16:36.59 | Ast001 | or setting |
| 16:37.10 | pputman- | Ast001, Is your card over 133 feet away from the isdn box? |
| 16:37.26 | Ast001 | isdn pri is on 1 m from me |
| 16:37.32 | Ast001 | from card |
| 16:37.35 | pputman- | Because if not, change your zaptel.conf span=1,1,0 instead of 1,1,1 |
| 16:37.37 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 16:37.41 | Ast001 | i i tryed that too |
| 16:37.54 | Ast001 | but got the same results |
| 16:38.16 | [TK]D-Fender | Ast001: well if he said its not your equipment then it IS your telco. |
| 16:38.29 | [TK]D-Fender | Ast001: Get off your ass and call them. |
| 16:39.11 | Ast001 | I called them today and they said we can come again with tester |
| 16:39.14 | pputman- | Ast001, then verify the line encoding and framing they told you, make sure you're on the latest zaptel, and ask them if you need crc4 |
| 16:39.24 | *** join/#asterisk shtoom (n=shtoom@122.167.47.25) |
| 16:39.35 | Ast001 | i dont need crc4 they told me that |
| 16:39.44 | pputman- | okay |
| 16:39.47 | pputman- | goes back to reading |
| 16:40.00 | [TK]D-Fender | macros73: CSeq: 104 REGISTER SIP/2.0 401 Unauthorized <- your register is bad. But thats going to be an inbound call issue |
| 16:40.22 | shtoom | Hi, what is the audio resolution that is available on E1 line ( 16-bit or 8-bit)? |
| 16:40.27 | Ast001 | it is strange that the same configuration works in other call center |
| 16:40.39 | [TK]D-Fender | macros73: pastebin your sip.conf entries for your phone, and your itsp masking only the passwords. |
| 16:40.44 | Ast001 | but there is digium card inside there |
| 16:40.48 | Ast001 | not dam openvox |
| 16:41.18 | Ast001 | I feel it is my motherboard losing interrupts or bad openvox card :( |
| 16:41.21 | [TK]D-Fender | Ast001: Your card may be messed up. Then again.. you haven't had your telco monitor an incoming call attempt. |
| 16:41.42 | Ast001 | yes they monitor that |
| 16:41.51 | Ast001 | they connect tester to isdn and call and talk |
| 16:42.57 | Ast001 | and experiment with another machine and card would be to expensive |
| 16:43.28 | [TK]D-Fender | Ast001: I didn't say with a tester at YOUR SITE. I said let the telco see what they THINK they are transmitting to you from the SOUREC |
| 16:43.59 | macros73 | [TK]D-Fender: http://pastebin.com/d2a3f09fb |
| 16:44.50 | [TK]D-Fender | macros73: you should ahve "nat=no" and "canreinvite=no" for all of your ITSP entries |
| 16:45.16 | macros73 | [TK]D-Fender: I'll add that and see what happens. |
| 16:45.34 | Ast001 | you know telco here is monopolist and they don't care when you call them they just say we can come with tester and test again but we did it yesterday. |
| 16:45.47 | macros73 | canreinvite=no is already there, adding nat=no |
| 16:46.04 | Ast001 | and attack you you don't know to setup pbx, card etc... |
| 16:46.30 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 16:46.49 | Ast001 | ok thank you for your help i'll se what i can do |
| 16:46.54 | j0 | does anyone here receive faxes with a sangoma a101d card? i'm having at least 20% of incoming faxes fail |
| 16:46.55 | *** part/#asterisk Ast001 (n=Administ@cable-89-216-185-58.dynamic.sbb.rs) |
| 16:47.25 | [TK]D-Fender | j0: early wanpipes & firmwares had issues. Check with Sangoma for the best combo. |
| 16:47.40 | macros73 | [TK]D-Fender: Same behavior. * is sitting there "Trying." |
| 16:47.56 | j0 | [TK]D-Fender: thanks, i'm using latest wanpipe. i'll check firmware.. would using hylafax make any difference? |
| 16:48.05 | hsv-al | heh, this iphone is badass |
| 16:48.11 | coppice | j0: check your clock sync |
| 16:48.13 | hsv-al | the cisco vpn client has good throughput |
| 16:48.32 | [TK]D-Fender | j0: well.. it can be an additional factor. is it on the same server? |
| 16:48.42 | macros73 | [TK]D-Fender: Aha, a call goes through...after setting up for 30 seconds |
| 16:48.43 | j0 | [TK]D-Fender: yes |
| 16:49.00 | j0 | coppice: TE_CLOCK in the config? i have mine set to normal |
| 16:49.16 | [TK]D-Fender | j0: thats fine. |
| 16:49.56 | macros73 | [TK]D-Fender: Okay, so this works, it just takes 30+ seconds for the call to start. Do I yell at Vitelity or is that my fault? |
| 16:51.01 | [TK]D-Fender | macros73: Uncertain. |
| 16:51.02 | j0 | already have the latest firmware.. hmm |
| 16:51.15 | macros73 | [TK]D-Fender: Km, really getting food now. bbiab. |
| 16:58.52 | shtoom | Hi, what is the audio resolution that is available on E1 line ( 16-bit or 8-bit)? |
| 16:59.18 | *** join/#asterisk joshaidan (n=joshaida@S0106001c1023e838.tb.shawcable.net) |
| 17:00.00 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:02.05 | [TK]D-Fender | shtoom: 8bit |
| 17:08.40 | *** join/#asterisk sack (n=sack@70.Red-88-24-156.staticIP.rima-tde.net) |
| 17:14.28 | *** join/#asterisk Tako-san (n=jmkiffia@24.108.192.144) |
| 17:14.59 | *** join/#asterisk Mawkee (n=Mawkee@200.152.178.136) |
| 17:15.39 | jblack | I just saw ny representative say that "We have contigency plans to invade England" on national TV. |
| 17:16.53 | [TK]D-Fender | jblack: Manifest Destiny 2009! |
| 17:17.04 | jaytee | we even have contingency plans to invade Canada |
| 17:17.42 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 17:18.03 | jaytee | which means that if you're a US citizen you don't have to ingest massive doses of LSD to become acutely paranoid, just join the military as an officer. |
| 17:18.08 | jameswf | who lives in a pineapple undet the sea ? |
| 17:18.19 | [TK]D-Fender | fires up the orchestra for another 1812 Overture. |
| 17:18.20 | jaytee | Sponge Bob Squarepants!!!! |
| 17:18.55 | jameswf | did someone say revolution? |
| 17:19.11 | jaytee | Tracey Chapman did but it sounds like a whisper |
| 17:20.18 | jameswf | I get all my news from CNN so I am totaly in the dark on wtf is happening in the real world.... but I did hear hilliry clintons hair grew 1mm over the last week |
| 17:22.32 | jaytee | she voted no on FISA and Obama voted yes. That's an unexpected twist. |
| 17:22.56 | [TK]D-Fender | jaytee: serious piss-off. |
| 17:23.38 | jaytee | [TK]D-Fender, um, not sure what you mean. You're pissed at Obama? |
| 17:23.39 | [TK]D-Fender | jaytee: Unless it is being "set up" to be smashed by the supreme court |
| 17:23.50 | [TK]D-Fender | jaytee: On this point, yes |
| 17:24.00 | jaytee | so am I |
| 17:24.15 | jaytee | on both the legislation and Obama's vote. |
| 17:24.41 | jaytee | as Ben Franklin said, "Those who are willing to sacrifice liberty for security deserve neither." |
| 17:24.51 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 17:25.00 | CanWood | jamesswf, switch to FOX, then you're hear that Obama forgot to wash his hands ten minutes ago instead. more real news |
| 17:25.03 | [TK]D-Fender | jaytee: I have to wonder if they really caved on it, or it has an inherent flaw someone is going to rip open to trash it later with. |
| 17:25.15 | jaytee | and Old Ben knew of where he spoke. The dude was a major playa with 27 illegitimate children. |
| 17:25.23 | [TK]D-Fender | jaytee: Obama for my #4 pick... this solidifies it. |
| 17:25.38 | jaytee | [TK]D-Fender, we can only hope the latter. |
| 17:26.11 | *** join/#asterisk Mawkee (n=Mawkee@200.152.178.136) |
| 17:26.20 | [TK]D-Fender | jaytee: Yup... in the mean-time... no more US trips for me... strictly Commonweath / free-world travel. |
| 17:26.24 | [TK]D-Fender | (DIRECT) |
| 17:26.42 | rwaite | i have a normal pots line i am connecting with a zaptel card ... it shows up with callerid of asterisk(asterisk) on my phone |
| 17:26.58 | rwaite | is there something i have to do to get callerid to work on a zap channel? |
| 17:27.11 | [TK]D-Fender | rwaite: "usecallerid=yes" , "callerid=asreceived" <- zapata.conf |
| 17:27.20 | rwaite | they're in there? |
| 17:27.28 | [TK]D-Fender | rwthey SHOULD BE |
| 17:27.28 | jaytee | right now I'm wavering between McCain and Nader. I figure if McCain gets in we'll attack Iran, gas will hit 30 bucks a gallon and the economy will collapse. After the rioting and the burning maybe we can build a new better nation to replace the one those bastards in D.C. stole out from under our complacent asses. |
| 17:27.49 | *** part/#asterisk Mawkee (n=Mawkee@200.152.178.136) |
| 17:27.51 | rwaite | i know, but it still is showing up as asterisk(asterisk) |
| 17:27.55 | *** join/#asterisk Mawkee (n=Mawkee@200.152.178.136) |
| 17:28.30 | rwaite | and there is nothing listed under src in cdr, totally blank |
| 17:28.33 | [TK]D-Fender | jaytee: McCain is a massive liar, ignorant on foreign policy and it implications, the economy, and a general militant. |
| 17:28.42 | [TK]D-Fender | rwaite: What card? |
| 17:29.04 | rwaite | TDM400P |
| 17:29.22 | [TK]D-Fender | rwaite: pastebin your zapata.conf |
| 17:29.26 | rwaite | ok |
| 17:29.40 | [TK]D-Fender | rwaite: and blocked # calls my show up as "asterisk" anyways on occasion. |
| 17:29.44 | jaytee | [TK]D-Fender, I agree and he also wears Depends although his campaign people have tried very hard to cover that up. |
| 17:29.53 | [TK]D-Fender | jaytee: lol. |
| 17:30.00 | [TK]D-Fender | jaytee: Diapergate! |
| 17:30.22 | rwaite | http://pastebin.com/m4f0745a4 |
| 17:30.25 | jaytee | [TK]D-Fender, hahahaha |
| 17:30.52 | rwaite | [TK]D-Fender: but if i make this call from same number to my voip did, it shows the callerid correctly |
| 17:31.14 | [TK]D-Fender | rwaite: maybe your line itself doesn't have the service. |
| 17:31.28 | rwaite | hmm. maybe. this is a fax line after all. |
| 17:31.51 | rwaite | (testing, the "real" analog lines come in monday) |
| 17:32.10 | [TK]D-Fender | jaytee: http://www.alternet.org/election08/90956/?page=entire "John McCain -- 61 Flip-Flops and Counting" <- A litany of BS |
| 17:32.30 | jaytee | is that Eric Alterman's site? I love that guy |
| 17:32.34 | [TK]D-Fender | rwaite: your fax line is real analog... jsut one you cheaped out on :) |
| 17:32.52 | rwaite | i dunno, i wasnt here when it was put in |
| 17:32.52 | [TK]D-Fender | jaytee: dunno... but it jsut made Digg's front page :) |
| 17:33.18 | jaytee | ah, I've heard of Steve Benen before |
| 17:33.21 | rwaite | [TK]D-Fender: thanks for your help |
| 17:33.23 | [TK]D-Fender | rwaite: Don't ask why a service isn't working... when you're not sure you've even GOT it ;) |
| 17:33.41 | rwaite | will keep that in mind |
| 17:34.41 | shtoom | [TK]D-Fender: Thank you ! |
| 17:35.22 | [TK]D-Fender | shtoom: You're welcome. |
| 17:35.51 | shtoom | [TK]D-Fender: will zapata hardware change the resolution and to and pass it to asterisk ? or just pass the same resolution i.e 8-bit? |
| 17:36.16 | jaytee | cripes, I've been in a Dell queue for 18 minutes now. So much for their dreams of getting #1 in customer service. |
| 17:38.59 | [TK]D-Fender | jaytee: You are getting #1.... you just don't know how low the standard is ;) |
| 17:39.20 | [TK]D-Fender | shtoom: this is the LCD of telecom. |
| 17:39.50 | [TK]D-Fender | shtoom: It is effectively G.711a/u with minor companding differences. |
| 17:41.37 | jaytee | once again [TK]D-Fender manages to reaffirm my deepest fears.......and no, I wasn't referring to Toto getting back together :-) |
| 17:42.37 | [TK]D-Fender | "I bless the rains down in Africaaaaaaaaaaaaaaaaaaaa!!!!" |
| 17:43.31 | hsv-al | d-fender |
| 17:43.38 | hsv-al | native sip client on 3G iphone |
| 17:43.40 | hsv-al | = win |
| 17:43.47 | [TK]D-Fender | jaytee: and actually good rendition (solo acoustic) : http://www.youtube.com/watch?v=dt1fB62cGbo |
| 17:43.54 | jaytee | ok, the onhold RAN announcements after 20 minutes have now changed to indicate heavy call volumes with a wait time that "may exceed ten minutes" |
| 17:44.22 | jaytee | was it you or Qwell that got me hooked on Bela Fleck a week or so ago? |
| 17:45.19 | [TK]D-Fender | jaytee: Neither.. can't recall just yet who... |
| 17:45.57 | macros73 | returns, and finds it still takes 30+ seconds to setup a call via the Vitelity trunks. |
| 17:46.02 | [TK]D-Fender | hsv-al: unlimited eve/wk + full PRI that doesn't cost me a penny > SIP client on iPhone :) |
| 17:47.01 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 17:47.50 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 17:48.13 | [TK]D-Fender | hsv-al: Quick AT&T plan Q : you get unlimited data for $60 right? X minutes + data. |
| 17:48.15 | shtoom | [TK]D-Fender:can you tell me if zaptel hardware changes that resolution of audio or passes the exact resolution it receives on line? |
| 17:48.25 | hsv-al | y3z |
| 17:48.45 | [TK]D-Fender | shtoom: * translates one side to the other. Each sees the bes of what it can from the other. |
| 17:48.55 | [TK]D-Fender | hsv-al: Any limitations on the kind of traffic? |
| 17:50.14 | hsv-al | im gonna hook it into usb to cat5 when I get home |
| 17:50.19 | hsv-al | and run wireshark |
| 17:50.27 | hsv-al | see what goes through, what doesnt |
| 17:51.41 | [TK]D-Fender | hsv-al: Yeah, if you can bridge with it, do FTP/SSH, etc all over Cell, that would be interesting. |
| 17:52.05 | [TK]D-Fender | hsv-al: my unlimited mobile browser is limited. No ftp, etc. Sorta like proxied HTTP-only. |
| 17:52.19 | hsv-al | im at work with it now, and im capturing logs on it |
| 17:52.22 | hsv-al | with websense daemon |
| 17:52.30 | hsv-al | going to kiwi syslog trapping |
| 17:52.39 | hsv-al | so ill bring this data home too, |
| 17:54.28 | *** join/#asterisk crudpuppy (n=someone@71-14-97-085.dhcp.gnvl.sc.charter.com) |
| 17:55.13 | crudpuppy | Can a vonage ata and a asterisk system operate on the same network? |
| 17:55.38 | *** join/#asterisk implicit (n=bayan@ip72-211-213-26.oc.oc.cox.net) |
| 17:55.51 | hardwire | you'd have to set up your firewall so that 5060 points in to the asterisk server |
| 17:55.59 | hardwire | that's about it |
| 17:56.24 | macros73 | Ron Paul / Cthulu for 2008. |
| 17:56.39 | crudpuppy | hardwire, that won't interfere with the vonage at all? |
| 17:57.18 | hardwire | the vonage phone will register with vonage, and if it uses sip ports it will negotiate for 5061 |
| 17:57.18 | hardwire | or 5062 |
| 17:57.18 | hardwire | etc.. |
| 17:57.29 | hardwire | if it's smart. |
| 17:57.41 | crudpuppy | vonage + smart = oximoron? |
| 17:57.52 | hardwire | you own it, I don't. |
| 17:57.56 | hardwire | uamoron? |
| 17:57.57 | crudpuppy | hehe, I'm moving off it |
| 17:58.06 | crudpuppy | but need it in place for the time being |
| 17:58.10 | hardwire | yar |
| 17:58.20 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 17:58.20 | hardwire | business ? |
| 17:58.23 | crudpuppy | yeah |
| 17:58.43 | crudpuppy | wait, business line, but not vonages business servie |
| 17:58.46 | crudpuppy | bbl |
| 17:58.50 | hardwire | how long you gonna keep using vonage for? |
| 17:58.53 | hardwire | darnit |
| 18:00.05 | macros73 | Heh, now it's all CHANUNAVAIL |
| 18:03.59 | *** join/#asterisk JenniferAkemi (n=akemi@76-10-148-105.dsl.teksavvy.com) |
| 18:04.42 | *** join/#asterisk andreadb7474 (n=andrea@195.94.142.68) |
| 18:05.57 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.static.dsl.tele.dk) |
| 18:06.25 | andreadb7474 | I've provider that use silence suppression and i've problem with moh when i've upgaded asterisk from 1.2 to 1.4... can help me? |
| 18:10.15 | [TK]D-Fender | andreadb7474: details please.... |
| 18:10.32 | [TK]D-Fender | andreadb7474: And * does not support silence suppression. |
| 18:11.23 | unpaidbill | audiomode:0x00000010 :/ |
| 18:12.48 | andreadb7474 | I know that, but last for 1 year i've used asterisk 1.2 without problem, when i've upgraded to 1.4 i've encountered problem with moh |
| 18:13.06 | andreadb7474 | and ring generation |
| 18:13.40 | [TK]D-Fender | andreadb7474: Well, get on with the deatils... |
| 18:14.06 | andreadb7474 | I use ztdummy driver because i'vent no zap hardware |
| 18:14.33 | hardwire | andreadb7474: heh.. :) |
| 18:15.44 | andreadb7474 | Do you think that can i resolv the problen if i mount a span-E1 ? |
| 18:19.28 | [TK]D-Fender | andreadb7474: I think you need to provide some real backup isf you want opinions. |
| 18:19.38 | [TK]D-Fender | andreadb7474: PASTEBIN is your friend... |
| 18:19.40 | [TK]D-Fender | ~pb |
| 18:19.40 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:19.42 | [TK]D-Fender | ^^^^^^^^^^^ |
| 18:21.00 | hardwire | hmm.. I need to do ackcall on a per-agent basis |
| 18:21.19 | hardwire | and I hate forking to local from queues, because it freaks out CDR in 1.2 |
| 18:21.20 | hardwire | hmmm |
| 18:22.42 | hardwire | it seems like chan_agent.c supports it |
| 18:26.51 | jaytee | [TK]D-Fender, that youtube is pretty cool. |
| 18:26.57 | macros73 | Overall, this thing performed better when the Asterisk server was inside the firewall, not outside. For me, at least. |
| 18:27.23 | jaytee | just finally got time to listen to it |
| 18:29.49 | jaytee | my fav is still Antoine Dufour and Tommy Gauthier with the 4 hands guitar one |
| 18:29.51 | andreadb7474 | http://pastebin.ca/1069375 |
| 18:35.29 | [TK]D-Fender | jaytee: Yeah, that one's funny.. |
| 18:36.03 | jaytee | I like Antoine's free style for acoustic. |
| 18:36.20 | [TK]D-Fender | jaytee: Erik Mongain is a lot more fun though :) |
| 18:36.43 | hardwire | nice |
| 18:36.48 | [TK]D-Fender | jaytee: http://youtube.com/watch?v=AbndgwfG22k |
| 18:36.54 | hardwire | setting the ack call variable before callback login worked fine |
| 18:41.20 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 18:42.33 | iratik | I've got a crazy question .... I don't know exactly how audio streams work in asterisk... but is it possible to combine all inbound audio streams into one stream and send to a channel? |
| 18:43.01 | *** join/#asterisk Yourname`` (i=chatzill@unaffiliated/yourname/x-837320) |
| 18:43.19 | [TK]D-Fender | iratik: Meetme <- |
| 18:43.28 | iratik | in other words... i have 24 agents... need to hear everyone at the same time on a channel |
| 18:43.31 | [TK]D-Fender | iratik: and "all inbound strems" is a no-go. |
| 18:43.39 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 18:43.45 | iratik | without them knowing |
| 18:43.51 | [TK]D-Fender | iratik: Meetme is what you've got to work with. |
| 18:43.55 | iratik | hmm.. |
| 18:44.06 | [TK]D-Fender | iratik: to hear? have everyone chanspy onto the main. |
| 18:44.12 | iratik | ? |
| 18:44.23 | [TK]D-Fender | iratik: "core show application chanspy" |
| 18:44.24 | *** part/#asterisk Mawkee (n=Mawkee@200.152.178.136) |
| 18:44.38 | iratik | yeah i know what chanspy does .. i've made quite a few custom apps with it |
| 18:44.45 | iratik | it spies on one channel from another channel |
| 18:44.49 | iratik | it can spy on multiple channels? |
| 18:45.10 | iratik | in parallel as opposed to round robin sequential cycling through channels? |
| 18:45.19 | [TK]D-Fender | iratik: You want to spy on multiple calls simultaneously? |
| 18:45.23 | iratik | yes |
| 18:45.34 | [TK]D-Fender | iratik: Not enough voices in your head already? |
| 18:45.38 | iratik | lol |
| 18:45.57 | iratik | Well... its hard to get a good feel for the volume of the sales calls in the room without being in there |
| 18:46.03 | [TK]D-Fender | iratik: this would normally class as "I'm a psycho" type demand.. |
| 18:46.18 | iratik | and even then... the room is so big .... can't get a good feel for total "effort" on the floor from any one spot |
| 18:46.30 | [TK]D-Fender | iratik: And you can't tell that a call is for sales without listening to them all... SIMULTANEOUSLY? |
| 18:46.42 | [TK]D-Fender | iratik: a channel list won't do? |
| 18:46.47 | iratik | no ... all calls are for sales |
| 18:47.04 | [TK]D-Fender | iratik: Then a raw call-list should answer that, no? |
| 18:47.09 | iratik | i don't need to know who is on sales calls... I need to get a gauge of how loud the floor as a whole is ... |
| 18:47.12 | iratik | not any specific agent |
| 18:47.58 | iratik | also ... if an agent happens to say anything misrepresenting us ... we can't capture by listening to agents one by one ... but maybe if we had one place to listen to them... as if we were standing in the same physical room as them |
| 18:48.02 | [TK]D-Fender | iratik: how... loud... |
| 18:48.03 | iratik | just louder |
| 18:48.25 | [TK]D-Fender | iratik: and you don't want to just review recordings? |
| 18:48.32 | iratik | lol... thats not live |
| 18:48.52 | [TK]D-Fender | iratik: ok, then your solution is Meetme + chanspy. |
| 18:49.23 | [TK]D-Fender | iratik: pump Originate-d chanspy local channels into a meetme and join that insane room. |
| 18:49.34 | iratik | that would work |
| 18:50.00 | iratik | thats an excellent idea |
| 18:50.05 | iratik | kudos [TK]D-Fender |
| 18:50.51 | iratik | Can I enter commands like Originate etc ... directly from the asterisk> prompt? |
| 18:53.30 | [TK]D-Fender | iratik: So make a script that will snag all of the calls you want (identifiable somehow I'm sure), and then issue them via call-files, Originate, etc. |
| 18:53.44 | [TK]D-Fender | iratik: go read to CLI reference to see what you can do from CLI. |
| 18:53.57 | iratik | Thanks TK D-Fender ... the meetme is going to work |
| 18:53.58 | [TK]D-Fender | iratik: But this sounds to eb something you should be doing from a script. |
| 18:54.11 | *** join/#asterisk Gh0sty (n=ghosty@ip-81-11-169-225.dsl.scarlet.be) |
| 18:54.18 | Gh0sty | hello |
| 18:54.34 | Gh0sty | anyone here from belgium who can help me with some country specific settings? :s |
| 18:54.34 | iratik | [TK]D-Fender: of course... but I need to figure out what to write first |
| 18:55.16 | [TK]D-Fender | iratik: Figure out how to pick out the channels you want. Then just issue the call-out for each |
| 18:55.35 | Gh0sty | have some problems with the buildup and the hangup of zap trunk |
| 18:55.42 | iratik | using AMI originate for this and a script to pull the extensions of all the sales agents from the DB |
| 18:56.03 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 19:01.58 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 19:03.31 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:08.37 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 19:13.25 | iratik | I'm a little confused .... the Meetme channel .. what channel is it? |
| 19:13.37 | iratik | Its just Meetme(confno |
| 19:14.04 | iratik | i think i might have just asked a dumb question |
| 19:14.08 | iratik | let me think for a bit more |
| 19:14.49 | Strom_M | it's....not a channel. |
| 19:15.47 | iratik | I need to use originate to join several chanspy extensions into one meetme room |
| 19:16.11 | iratik | I've setup _556X. to chanspy with no prompts .... Extension: _556512 in the AMI originate command |
| 19:16.20 | iratik | What do i put for Channel: ? |
| 19:16.39 | iratik | to join the chanspy extension _556512 into Meetme(123 |
| 19:16.46 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 19:18.32 | iratik | Its not SIP/123 |
| 19:19.10 | magic_hat | hey everyone. I'm certain I'm doing something dumb here, but I can't figure out what. I have a dialplan that has an s extension. Incoming calls are being rejected because extension not found. I'm not following why the 's' extension isn't being activated. |
| 19:20.04 | Strom_M | magic_hat: what kind of entrance facilities are you using? |
| 19:20.43 | Strom_M | 's' is only for situations where the inbound call contains NO dialed number information (i.e. an analog line ringing) |
| 19:21.10 | magic_hat | ah, it's from Teliax. I thought it defaulted to s if there was no other valid ext. |
| 19:21.16 | Strom_M | nope |
| 19:23.21 | *** join/#asterisk moy (n=moy@nat/ibm/x-c41353b4b50ed6bf) |
| 19:23.29 | unpaidbill | that would be i |
| 19:25.32 | iratik | this is far more difficult than i thought |
| 19:25.59 | Strom_M | why are you trying to merge chanspy extensions into a meetme room? |
| 19:26.01 | Strom_M | that makes no sesne |
| 19:26.02 | Strom_M | sense |
| 19:26.26 | iratik | yeah it does |
| 19:26.31 | iratik | Need to listen to a room full of calls |
| 19:26.36 | iratik | without being in the room |
| 19:26.50 | iratik | Not sequentially.. but in parallel |
| 19:27.00 | unpaidbill | does Data work for you |
| 19:29.13 | iratik | any ideas? |
| 19:32.08 | magic_hat | okay, i got that prob. solved. here's another: I have an AGI app that dials a bunch of numbers. My teliax account has 10 available channels. I want to make sure that I only have 10 outbound calls going at once, and I don't want to use AMI to get there... possible? |
| 19:34.15 | iratik | I keep getting originate failed ....... |
| 19:36.15 | iratik | The asterisk documentation seems to indcate that the channel for a Meetme would be local/123 for a Meetme started with Meetme(123 |
| 19:36.52 | *** join/#asterisk DiegoFerreira (n=webirc_u@mail.grupoabv.com.br) |
| 19:37.39 | macros73 | If my zttest results are an average 99.955, could that be a factor in poor g729 voice quality and some crackling in system messages? |
| 19:37.47 | Strom_M | iratik: how about just joining the meetme conference using the "listen only" option |
| 19:38.27 | iratik | I can join the conference fine ... i just dial 123 ... or i can create a custom extension that joins in quiet |
| 19:38.44 | iratik | Its figuring out the command to join the extension 556512 into the meetme room 123 |
| 19:38.57 | iratik | 556512 is a chanspy extension that spies on channel 512 |
| 19:38.59 | *** join/#asterisk bl4q (i=Bl@dslb-088-066-224-181.pools.arcor-ip.net) |
| 19:41.39 | *** join/#asterisk bl4q (i=Bl@dslb-088-066-224-181.pools.arcor-ip.net) |
| 19:44.00 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 19:47.27 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 19:47.48 | *** join/#asterisk Dr-Linux|home (n=Nothing@221.132.117.17) |
| 19:49.13 | Dr-Linux|home | How can i enable pound key options after the number? |
| 19:50.18 | Dr-Linux|home | like the messeage is "please dial your destinationi number followed by the pound key, so when i dial number and press pound, my system takes # key as a part of number :S |
| 19:50.33 | Dr-Linux|home | but i want this # key for input termination |
| 19:51.01 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 19:51.44 | [TK]D-Fender | Dr-Linux|home: "core show application read" |
| 19:51.59 | [TK]D-Fender | macros73: No. |
| 19:52.07 | [TK]D-Fender | magic_hat: Yes |
| 19:52.17 | [TK]D-Fender | NEXT!@!!@ (c) BKW |
| 19:52.59 | macros73 | lol. So if I hear crackling on a voicemail greeting that doesn't leave the LAN, how do I resolve? ("Be as smart as [TK]D-Fender" doesn't help me..) |
| 19:53.40 | [TK]D-Fender | macros73: Whats crackling, recordings YOU made, or stock system sounds? |
| 19:53.47 | jameswf | ummm be as smart as russellb |
| 19:53.56 | [TK]D-Fender | jameswf: UNPOSSIBLE! |
| 19:53.59 | macros73 | [TK]D-Fender: Stock |
| 19:54.08 | Dr-Linux|home | [TK]D-Fender: where Read application involved in my question? |
| 19:54.09 | [TK]D-Fender | macros73: Try another codec. |
| 19:54.20 | [TK]D-Fender | Dr-Linux|home: To get input, terminated by a "#" |
| 19:54.38 | [TK]D-Fender | macros73: And why on earth are you using G.729 in a LAN? |
| 19:54.50 | macros73 | [TK]D-Fender: I'm not, using G711 in the LAN. |
| 19:55.09 | [TK]D-Fender | macros73>If my zttest results are an average 99.955, could that be a factor in poor g729 voice quality and some crackling in system messages? |
| 19:55.24 | [TK]D-Fender | macros73: Not sounding very consistent.. |
| 19:55.26 | macros73 | [TK]D-Fender: So I can't tell 29, from 11, okay? |
| 19:55.44 | [TK]D-Fender | macros73: Ok, what are you using for phones? |
| 19:55.53 | *** join/#asterisk znoG (n=gs@host88.190-31-90.telecom.net.ar) |
| 19:56.01 | jameswf | the diff of 729 and 711 is 18 |
| 19:56.15 | jameswf | grandsuck budgettones |
| 19:56.20 | [TK]D-Fender | macros73: Actaully... doesn't matter so much... its your phone's fault in all likelyhood. |
| 19:56.23 | macros73 | Ooops, I probably shouldn't admit that. The conjunction joined two different issues. g729 is being used on external trunks. g711 internal. No, I junked the grandsuck. |
| 19:56.29 | macros73 | Currently using Ekiga as a softphone. |
| 19:56.46 | jameswf | isnt ekiga a womans exersize |
| 19:56.50 | [TK]D-Fender | macros73: Ekiga itself has sound quality issues. I've tried it, and in most cases sucked when it had no reason to. |
| 19:57.12 | macros73 | [TK]D-Fender: Thanks, I'll retest with something else. How's X-Lite for a soft phone? |
| 19:57.13 | [TK]D-Fender | macros73: using ULAW local LAN as well. |
| 19:57.17 | drako | i've tried ekiga and i discarted it due its sounds problems. |
| 19:57.29 | [TK]D-Fender | macros73: More solid. More limited, but will sound better most likely. |
| 19:57.35 | drako | macros73, xlite is bad |
| 19:57.36 | jameswf | sticks to his aastra |
| 19:57.44 | drako | macros73, i recommend you Twinkle |
| 19:57.51 | iratik | i've tried almost every softphone out there |
| 19:57.54 | iratik | Twinkle? |
| 19:57.56 | iratik | lol |
| 19:58.02 | jameswf | i use moziax |
| 19:58.04 | drako | Twinkle is pretty good |
| 19:58.06 | macros73 | um....i don't like boys? |
| 19:58.19 | drako | and stable |
| 19:58.38 | drako | almost 2 years using twinkle and im happy with it. |
| 19:58.43 | iratik | not bad |
| 19:58.45 | *** join/#asterisk wasabi (n=wasabi@ubuntu/member/wasabi) |
| 19:58.47 | jameswf | KIAX loked like they are trying a comeback but the software still sucks |
| 19:58.54 | Corydon76-dig | macros73: what about men? |
| 19:58.59 | wasabi | Hey... I'm looking for some sort of recommended practice on what to number my extensions. |
| 19:59.10 | iratik | wasabi: I use the id column of the users table |
| 19:59.14 | wasabi | Right now I've been using 1XX for sip phones, 500 for ivr stuff, 600 for call queues, and various segmentation like that |
| 19:59.20 | iratik | that way their user_id is their extension |
| 19:59.22 | [TK]D-Fender | wasabi: Only even numbers if you know whats good for you. |
| 19:59.27 | wasabi | But I realized that if I want to dial long distance, while off hook. |
| 19:59.36 | wasabi | that it won't work right |
| 19:59.40 | wasabi | it'll dial soem local thing first. |
| 19:59.53 | macros73 | I'll try out Twinkle and X-Lite |
| 20:00.06 | magic_hat | [TK]D: tell me more... how would I go about that? |
| 20:00.13 | [TK]D-Fender | wasabi: No, you can use 1XX for internal stuff jsut fine. |
| 20:00.37 | wasabi | What about dialing area code 368? |
| 20:00.42 | wasabi | Where you type 13680000000 |
| 20:00.48 | [TK]D-Fender | magic_hat: parse out : asterisk -rx "show channels concise" |
| 20:00.50 | wasabi | 136 would get picked up first. |
| 20:01.00 | [TK]D-Fender | wasabi: says who? |
| 20:01.01 | wasabi | and it would go to extension 136, which is wrong. |
| 20:01.09 | wasabi | says who? i'm doing it right here. |
| 20:01.11 | [TK]D-Fender | wasabi: not in MY world it doesn't |
| 20:01.18 | [TK]D-Fender | was then you're doing it wrong. |
| 20:01.25 | wasabi | and what would i be doing wrong? |
| 20:01.36 | wasabi | if i pick up the phone and dial 136, i get extension 136. |
| 20:01.41 | [TK]D-Fender | wasabi: I allow NANPA 7-10-11 digit diaing alonside 4 digit extensions. |
| 20:01.44 | wasabi | it does not wait for the next digit. |
| 20:01.51 | [TK]D-Fender | wasthen your PHONE is making decisions for you |
| 20:02.04 | [TK]D-Fender | wasabi: and you need to go configure IT |
| 20:02.04 | wasabi | oh. |
| 20:02.12 | wasabi | i see. |
| 20:02.18 | *** join/#asterisk Dr-Linux|work (n=Nothing@221.132.117.17) |
| 20:02.34 | wasabi | oh... yeah. i see. so the phone is realizing that that's a completed number, and making the call |
| 20:02.45 | wasabi | so there's something programmable about the phone... |
| 20:02.57 | jaytee | must be talking about the dialplan.digitmap on Polycoms |
| 20:03.03 | wasabi | yeah, i probably am |
| 20:03.08 | wasabi | i didn't put the two together. |
| 20:03.08 | magic_hat | [TK]D: that is sweet. I owe ya. |
| 20:03.10 | [TK]D-Fender | jaytee: No implication of make/model yet. |
| 20:03.17 | wasabi | they are all polycoms |
| 20:03.24 | [TK]D-Fender | jaytee: NOW there is :) |
| 20:03.37 | [TK]D-Fender | wasabi: Yes, go make a PROPER digitmap for them. |
| 20:03.59 | wasabi | so the digit map is a set of things seperated by xx and |? |
| 20:04.06 | wasabi | and any matches are dialed? |
| 20:04.09 | jaytee | yeah, you need to edit the dialplan.digitmap |
| 20:04.15 | jaytee | yeah, the | is the seperator |
| 20:04.22 | [TK]D-Fender | wasabi: yes. |
| 20:04.32 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 20:04.43 | wasabi | soo there is probably some way to make it wait a few seconds before it matches? |
| 20:04.47 | jaytee | if you put a T in front of each | it will enable the default timeout |
| 20:05.01 | [TK]D-Fender | wasabi: "x.T|#.T!*.T" = salvation. |
| 20:05.11 | [TK]D-Fender | wasabi: "x.T|#.T|*.T" = salvation. |
| 20:05.18 | jaytee | and you can increase each section's timeout interval for slow, fat-fingered people |
| 20:05.18 | [TK]D-Fender | better... 2nd bar fixed |
| 20:05.21 | wasabi | So, anything, 1 or more times, default timeout. |
| 20:05.40 | [TK]D-Fender | wasabi: anything + wait = just do it |
| 20:06.05 | [TK]D-Fender | wasabi: Phones shouldn't get delusions like thinking they know whats possible. |
| 20:06.14 | wasabi | yeah |
| 20:06.22 | wasabi | i agree. |
| 20:06.24 | [TK]D-Fender | wasabi: tell the phone to STFU and let * run the show like it should. |
| 20:06.34 | wasabi | i was not making the mental connection that is obvious in sip: the phone makes a call to a single number |
| 20:06.37 | wasabi | no open channel thing |
| 20:07.06 | [TK]D-Fender | wasabi: it isn't an "open channel thing" |
| 20:07.18 | wasabi | i know |
| 20:07.20 | [TK]D-Fender | wasabi: and indeed it DOES make a call to a "single number". |
| 20:07.22 | wasabi | that's what i just said |
| 20:07.45 | jaytee | [TK]D-Fender, speaking of which Polycom confirmed that I can't block external transfers by phone so my only option for that is to disable the softkeys for transfer and call forward and use * to do it by setting the TRANSFER_CONTEXT variable. |
| 20:07.49 | wasabi | too much ingrained phone stuff in my brain |
| 20:08.18 | jaytee | wish they had a digit mask that you could set for transfers in the phone |
| 20:08.38 | [TK]D-Fender | jaytee: did you test the blindtransfer var I told you to look for? and to look at the header? |
| 20:09.18 | jaytee | yeah, it was coming up blank. the softkey just does the 502 redirect for the call so there's no re-entry in the dialplan to trap it. |
| 20:09.30 | [TK]D-Fender | jaytee: and the header? |
| 20:09.53 | *** part/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 20:10.15 | jaytee | no info in the header that looked like anything I could use to divert based on extension. |
| 20:10.31 | jaytee | but I haven't entirely given up yet. |
| 20:10.39 | *** join/#asterisk TrentCreek (n=kvirc@red1.cs.panam.edu) |
| 20:10.49 | wasabi | thanks! |
| 20:10.51 | *** part/#asterisk wasabi (n=wasabi@ubuntu/member/wasabi) |
| 20:11.07 | errr | when I make a call from ext -> ext and they are both sip and both local to the box, is the context used what I have in my peer section for context=?? |
| 20:11.08 | jaytee | just not a guru at this so I have to dig and dig and read and then pester you with questions :-) |
| 20:11.38 | [TK]D-Fender | jaytee: heres a thought : when it redirects, it'll hit the dialplan. That new channel will INHERIT vars from the first. You could then test for a var you'd set before calling that phone in the first place. |
| 20:12.25 | [TK]D-Fender | errr: if its mathes up the right peer. |
| 20:12.30 | [TK]D-Fender | matches* |
| 20:13.00 | jaytee | [TK]D-Fender, maybe! I can use the underscore with TRANSFER_CONTEXT to force inheritance but where does it reenter the dialplan? in the context set for that sip device? |
| 20:13.09 | TrentCreek | And with IAX2 debug on...who can tell me where to look to find this problem? NOTICE[5758] chan_iax2.c: Rejected connect attempt from 64.34.181.47, who was trying to reach |
| 20:13.23 | [TK]D-Fender | jaytee: yes |
| 20:13.33 | errr | [TK]D-Fender: so if 100 and 101 are both in context=internal if I want to make it so when 100 calls 101 it records all 100's outbound calls I would need to setup a mixmonitor in the internal context? |
| 20:13.40 | [TK]D-Fender | jaytee: When a phone transfers/redirects, its pased on its privileges |
| 20:14.10 | [TK]D-Fender | errr: You don't setup monitor in a "context", you call it in an EXTEN. |
| 20:14.33 | errr | [TK]D-Fender: Im not sure I understand. I better go read up on the wiki |
| 20:14.40 | jaytee | aha!!! my brain is percolating!!! I'm thinking I can make it work. I'll have to setup a separate context for users I want to block outbound though because of the way I'm doing outbound calls. I have two contexts I include for local and long-distance so I'll create another 2 that are restricted. |
| 20:15.07 | [TK]D-Fender | errr: "monior" is not some global things to a "context". Its an application you specifically call as a step in processing an EXTENSION. |
| 20:15.17 | [TK]D-Fender | jaytee: Now go play :) |
| 20:15.28 | jaytee | If I can pull this off it'll have the same level of granularity in controlling user calls that Nortel or Avaya has! |
| 20:16.07 | [TK]D-Fender | jaytee: I still suspect there is an extra flag in the SIP header you should be able to snag to better ID this... |
| 20:16.17 | errr | [TK]D-Fender: yes I have that working like youre saying when 100 gets an incoming call but I dont know how to make it work when 100 dials 101.. |
| 20:16.53 | [TK]D-Fender | errr: exten => 101,1,MixMonitor(myfile.wav) |
| 20:17.48 | errr | hmm |
| 20:19.19 | ManxPower | jaytee: Isn't that what I told you 3 days ago? |
| 20:21.09 | TrentCreek | And with IAX2 debug on...who can tell me where to look to find this problem? NOTICE[5758] chan_iax2.c: Rejected connect attempt from 64.34.181.47, who was trying to reach "numnumnumb" |
| 20:23.38 | jaytee | ManxPower, I honestly don't recall. I remember discussing it with unpaidbill and russellb the other night and [TK]D-Fender during the day. |
| 20:24.07 | *** join/#asterisk mandd (n=moo@bas1-toronto61-1279436026.dsl.bell.ca) |
| 20:25.13 | mandd | hello, if I want to add a second line, FAX to >pap2 linksys to >asterisk |
| 20:25.21 | mandd | can that work? |
| 20:26.08 | hsv-al | Freedom From Hemoroids? ....... FreedHEM Hemoroid Cream .......... Freedom From Hemoroids? FreedHEM Hemoroid Cream |
| 20:27.17 | russellb | hsv-al: wtf? |
| 20:27.44 | *** mode/#asterisk [+b %hsv-al!*@*] by russellb |
| 20:27.53 | russellb | don't say random crap like that for no reason ... |
| 20:28.16 | *** mode/#asterisk [-b %hsv-al!*@*] by russellb |
| 20:28.55 | mandd | is there an SIP.conf exsample you can point me to? |
| 20:28.57 | mandd | for FAX |
| 20:29.34 | geek_cl | <PROTECTED> |
| 20:29.43 | ManxPower | mandd: fax is just a voice call |
| 20:29.52 | *** join/#asterisk emiller (n=ed@216.207.245.1) |
| 20:29.53 | ManxPower | the voice is just REALLY REALLY REALLY fast. |
| 20:30.19 | mandd | good, ManxPower you know of any info on how to add a second account in a same sip.conf |
| 20:30.28 | ManxPower | mandd: um add it |
| 20:30.39 | mandd | I only have one line, and a second one is registered, just gotta figure out how to add it to configs |
| 20:30.44 | ManxPower | just like you added the first account. You must use different user ID's of cousre |
| 20:31.44 | mandd | well, in [genral] I have register=> my first line info |
| 20:31.56 | mandd | do I jsut add a second account right uneder it? |
| 20:32.10 | mandd | and how will I be able to reffer to it from extensions.conf |
| 20:32.37 | mandd | just usign different Context ? |
| 20:32.47 | ManxPower | mandd: register => registers Asterisk to a REMOTE SERVER |
| 20:32.54 | ManxPower | mandd: perhaps you should step back and read The Good Book |
| 20:32.56 | ManxPower | ~book |
| 20:32.57 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 20:33.05 | mandd | okay! |
| 20:33.07 | mandd | thanks |
| 20:33.08 | ManxPower | not perhaps, you must. |
| 20:33.18 | mandd | i got everytihng working from the tutorials on google |
| 20:33.36 | jaytee | it's quittin time. be back later from the homefront |
| 20:33.40 | ManxPower | you should have read the book first |
| 20:33.43 | *** part/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 20:33.53 | *** join/#asterisk jeremy_g (n=a@c213-100-17-48.swipnet.se) |
| 20:33.56 | jeremy_g | halo |
| 20:34.00 | jeremy_g | sexy boys |
| 20:34.07 | jeremy_g | your daddy is here |
| 20:35.15 | macros73 | Wow. Twinkle's call quality sounds better, at least locally, than Ekiga. |
| 20:36.12 | macros73 | Call quality to an external number sucks just as bad as the rest via g729. |
| 20:36.39 | macros73 | twinkle -> g711 -> * -> g729 -> extern # |
| 20:37.12 | hsv-al | macros |
| 20:37.24 | hsv-al | there was a glitch in twinkle, that allowed certain packets to bypass mpls boundaries |
| 20:37.29 | hsv-al | was a secunia vuln posted 2 days ago :) |
| 20:38.48 | macros73 | hsv-al: Would that impact call quality? |
| 20:39.33 | ManxPower | bypassing MPLS boundaries is a whole lot more serious than just some call quality issues. |
| 20:39.49 | hsv-al | yep |
| 20:39.59 | ManxPower | i.e. the whole "With MPLS you get your own private network" goes out the window. |
| 20:41.38 | *** join/#asterisk neverbart (n=nevermor@78-105-116-210.zone3.bethere.co.uk) |
| 20:41.55 | hsv-al | manx |
| 20:42.08 | hsv-al | this gentleman shows techniques, that sort of supercede the secunia posting |
| 20:42.11 | hsv-al | http://www.youtube.com/watch?v=BLxa8K0cIkg |
| 20:42.16 | ManxPower | hsv-al: no questions, I'm off duity |
| 20:42.49 | neverbart | hi all. i'm trying to configure my new X100P fxo card. When calls come in, if I reject the call it keeps ringing at the dialling party's end, then asterisk tries again to call my local device |
| 20:42.57 | neverbart | and I get this: |
| 20:42.58 | neverbart | [Jul 11 21:40:11] WARNING[3040]: chan_zap.c:4131 zt_handle_event: Ring/Off-hook in strange state 7 on channel 1 |
| 20:43.05 | neverbart | but I can't find anything online as to what's causing it |
| 20:43.07 | neverbart | any ideas? |
| 20:43.19 | ManxPower | neverbart: you can't reject a call on analog |
| 20:43.42 | neverbart | so when it does == Everyone is busy/congested at this time (3:1/0/2) |
| 20:43.42 | neverbart | <PROTECTED> |
| 20:43.44 | neverbart | the fxo just ignores it? |
| 20:43.49 | ManxPower | Your best two options is WaitForRing with a large value or answer and hangup |
| 20:44.14 | ManxPower | neverbart: BUSY is the internal asterisk thing. You CANNOT because of the nature of analog "reject" a call. |
| 20:44.26 | neverbart | well, that answers that one :) |
| 20:44.45 | ManxPower | neverbart: you can fake it pretty easily. |
| 20:44.55 | neverbart | and the call? ;) |
| 20:45.14 | neverbart | sorry, couldn't resist. pls go on. this is explaining a lot :) |
| 20:45.41 | ManxPower | um, you are wanting to fake rejecting the call, right? Either don't answer the call in your dialplan or in your dialplan answer the call, play a "You're rejected! Next!" audio message then hangup. |
| 20:45.52 | ManxPower | you could also run Zapatateller to generate the beep tones |
| 20:46.06 | ManxPower | but the caller will still be billed for the call because the call was "answered" |
| 20:46.48 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 20:47.29 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.58.30) |
| 20:47.33 | ManxPower | neverbart: and there has not been an X100P manufactured for like 5 years. |
| 20:48.18 | neverbart | it's an old card. linux is quite forgiving about these things, ManxPower |
| 20:48.44 | ManxPower | (3:42:53 PM) neverbart: hi all. i'm trying to configure my new X100P fxo card.... |
| 20:48.47 | ManxPower | hence my confusion |
| 20:48.56 | neverbart | new in the sense that "i've just got it" :) |
| 20:49.02 | neverbart | bad choice of words, i realise :) |
| 20:51.21 | TrentCreek | And with IAX2 debug on...who can tell me where to look to find this problem? NOTICE[5758] chan_iax2.c: Rejected connect attempt from 64.34.181.47, who was trying to reach "numnumnumb" |
| 20:51.34 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:53.11 | jeremy_g | macros73:which codec should i have my clients on in order to have best voip quality |
| 20:53.49 | jeremy_g | g.729 was so heavy |
| 20:53.59 | macros73 | jeremy_g: Depends. For me, right now, in my current test environment, g711 gives the best call quality. |
| 20:54.11 | jeremy_g | we ran a client on HTC Touch with g.729 support and its CPU usage was reportedly at 98% |
| 20:54.34 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:54.34 | jeremy_g | g.729 is nice for bw constrained channels i think |
| 20:54.49 | jeremy_g | i have an interesting issue here |
| 20:55.35 | jeremy_g | how do i get my asterisk REGISTER message that it sends to some remote proxy of the format: REGISTER sip:chat.brain.net.pk:8891 SIP/2.0 |
| 20:55.43 | jeremy_g | this 8891 port |
| 20:55.59 | jeremy_g | how can i get it in there |
| 20:56.27 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:57.27 | neverbart | i'm going to go and have a play. thanks for the pointers ManxPower |
| 20:59.45 | TrentCreek | Anyone? Anyone? |
| 20:59.57 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 21:03.50 | TrentCreek | Anyone? Anyone? |
| 21:04.36 | TrentCreek | Bueller? Bueller? |
| 21:05.52 | jeremy_g | my is box sending OPTION messages to the remote proxy |
| 21:06.02 | jeremy_g | is there any place where enable/disable the option |
| 21:06.07 | jeremy_g | i dont have qualify set |
| 21:06.08 | jeremy_g | wtf |
| 21:14.37 | TrentCreek | Bueller? Bueller? |
| 21:16.39 | *** join/#asterisk WhiteWolf (i=whitewol@i-am.whitew0lf.info) |
| 21:16.52 | mandd | http://pastebin.com/m58a1cb4c |
| 21:16.52 | mandd | <PROTECTED> |
| 21:17.09 | mandd | and route is there, cant figure out what I did wrong |
| 21:17.17 | mandd | checked over and over again |
| 21:22.00 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:28.39 | *** part/#asterisk Rem| (n=remi@bg-fw2out.monmouth.com) |
| 21:30.05 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 21:33.14 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 21:33.36 | MatBoy | mhh, an incoming sip trunk is whick I can' t get working. The call comes in, but gets a busy tone... what should I check ? |
| 21:35.14 | TJNII | ~trunk |
| 21:35.15 | jbot | from memory, trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 21:35.27 | TJNII | What does sip show peers show? |
| 21:35.38 | [TK]D-Fender | MatBoy: SIP DEBUG at CLI for the failed call. |
| 21:36.42 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 21:38.00 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.58.30) |
| 21:43.54 | *** join/#asterisk edwin_quijada (n=m@25.116.88.200.m.sta.codetel.net.do) |
| 21:43.58 | edwin_quijada | Hi! |
| 21:44.15 | edwin_quijada | Somebody here has used Cepstral for IVRS |
| 21:44.16 | edwin_quijada | ? |
| 21:44.23 | MatBoy | [TK]D-Fender: yes I tried that, but I was not able to find any solution using it |
| 21:44.59 | TJNII | MatBoy: What does sip show peers show? |
| 21:45.14 | MatBoy | TJNII: peers are OK |
| 21:49.12 | [TK]D-Fender | MatBoy: And the reason you're not showing US the debug is....? |
| 21:49.48 | MatBoy | [TK]D-Fender: oh, I can show it, but I thought I better ask for better search options first :) |
| 21:49.54 | *** part/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 21:50.02 | MatBoy | as the debug didn' t gave me what I wanted to see |
| 21:50.06 | [TK]D-Fender | MatBoy: Do you think we're psychic? |
| 21:50.14 | *** join/#asterisk xenonex (n=xenonex@88.204.243.150) |
| 21:50.19 | [TK]D-Fender | MatBoy: Whats wrong with my car... it doesn't work! |
| 21:50.43 | MatBoy | [TK]D-Fender: how do you mean ? I mean, maybe you have tips to get a better debug, I didn' t ask for a solution, I asked where to look ;) |
| 21:51.07 | [TK]D-Fender | MatBoy: and I told you where to look. * CLI with SIP DEBUG. |
| 21:51.19 | MatBoy | ok, I will use that again than :) |
| 21:51.42 | MatBoy | I'm not a solution asker... I wanna know what happens :) |
| 21:51.55 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:53.11 | *** join/#asterisk deeperror (n=deeperro@d149-67-253-63.try.wideopenwest.com) |
| 21:53.15 | lesouvage | I use ${STRFTIME(${EPOCH},GMT+1,%C%y-%m-%d %r} in a line and for some reason the time is 3 hours to early (20:00 when it is 23:00) When I do date on linux terminal the time is ok. What can be the reason for this wrong representation of the actual time? |
| 21:54.45 | deeperror | If i have a wholesale account/contract setup for outbound termination with a minimum amount of minute commitment is there any reason a provider would bill based on the caller id being sent? |
| 21:55.44 | deeperror | I make a month of calls using 1 area code i get my contracted rate....I make a month of calls using another area code on caller id and i get billed at a different rate? Does this make sense to anyone? |
| 21:57.40 | [TK]D-Fender | Deethis is up to THEM and their terms. |
| 21:58.22 | TJNII | Perhaps you shoule read said contract? |
| 22:06.50 | *** join/#asterisk `paul (n=aldee@125.252.68.126) |
| 22:07.22 | `paul | can i change the incoming call caller id in asterisk??? |
| 22:10.41 | [TK]D-Fender | `paul: "core show function CALLERID" |
| 22:12.57 | TrentCreek | Bueller? Bueller? |
| 22:13.14 | jpcansa | how can i easyly disable a zap channel on asterisk? |
| 22:13.35 | [TK]D-Fender | jpcansa: define "disable" |
| 22:15.14 | TrentCreek | With IAX2 debug on...who can tell me where to look to find this problem? NOTICE[5758] chan_iax2.c: Rejected connect attempt from 64.34.181.47, who was trying to reach "numnumnumb" |
| 22:16.02 | deeperror | TJNII, yea no mention that cid has anything to do with outbound cost |
| 22:16.23 | TJNII | Then you should call them, contract in habnd, and ask "WTF?" |
| 22:16.55 | edwin_quijada | I recorded a few files usiong cepstral but I cant hearing from my AGI but if I can this file with PlayBack in extensions this works fine |
| 22:18.27 | jpcansa | [TK]D-fender: i have a pstn line with no dial tone, so i want that line not available to dial |
| 22:18.42 | jpcansa | momentarily |
| 22:18.50 | [TK]D-Fender | jpcansa: change zapata, reload chan_zap |
| 22:21.27 | *** part/#asterisk moy (n=moy@nat/ibm/x-c41353b4b50ed6bf) |
| 22:22.34 | *** join/#asterisk derelm (n=derelm@p5B23FD25.dip.t-dialin.net) |
| 22:26.28 | TrentCreek | Bueller? Bueller? |
| 22:27.21 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:27.31 | *** join/#asterisk eross (n=drarem@6532142hfc81.tampabay.res.rr.com) |
| 22:28.08 | eross | is there a cheap virtual call queue service available, perhaps web-based? |
| 22:28.25 | *** join/#asterisk Bananaskin (n=mike@78-105-247-227.zone3.bethere.co.uk) |
| 22:28.35 | eross | but using the regular land lines |
| 22:28.40 | jpcansa | thx Fender |
| 22:31.27 | [TK]D-Fender | eross: What does "web" have to do with "land lines"? |
| 22:31.44 | *** join/#asterisk dlynes (n=daniel@d206-116-189-12.bchsia.telus.net) |
| 22:31.44 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 22:31.48 | TrentCreek | Dial Up |
| 22:32.26 | dlynes | Does the DISA command not take a pin code any more? It doesn't seem to prompt me for a pin code...or does that happen after I dial my number? |
| 22:32.50 | [TK]D-Fender | dlynes: Either a buggy version, or you're calling it wrong |
| 22:33.23 | eross | in other words, work from a virtually hosted website, the queued calls are handled at the service provider's servers |
| 22:33.37 | eross | then select which one you want to receive next in the list |
| 22:33.44 | derelm | hi i am trying to take a call with zoiper, but it continues to play the ringing sound and the call is terminated some 10 seconds later. is that an asterisk config issue? |
| 22:33.45 | tzanger | [TK]D-Fender: ever run across a polycom phone that had low audio even with the volume cranked? |
| 22:34.10 | dlynes | [TK]D-Fender: exten => 1,n(auth),DISA(/etc/asterisk/file.auth) |
| 22:34.38 | dlynes | [TK]D-Fender: and then say first line of DISA file is: 762538|outbound|"CLIDNAME"<6041234567> |
| 22:35.07 | dlynes | [TK]D-Fender: I get the dialtone, but no prompt for the pin number |
| 22:35.34 | [TK]D-Fender | dlynes: "they say"? |
| 22:35.41 | dlynes | [TK]D-Fender: hrm? |
| 22:36.42 | [TK]D-Fender | dlynes: nvm, misread |
| 22:37.01 | *** join/#asterisk angryuser (n=sdfsdf@88.140.144.209) |
| 22:37.09 | dlynes | [TK]D-Fender: I've also tried it as DISA(762538|outbound|"CLIDNAME" <6041234567>), too |
| 22:37.13 | dlynes | [TK]D-Fender: that doesn't work, either |
| 22:37.27 | dlynes | [TK]D-Fender: both give me dialtone, and no prompt for password |
| 22:37.54 | angryuser | hi all |
| 22:38.43 | angryuser | what do you think about ael, is is stable, has someone used it in hight load/production environement ? |
| 22:40.12 | [TK]D-Fender | angryuser: No impact. |
| 22:40.26 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 22:40.37 | [TK]D-Fender | angryuser: and no real point. It only makes your coding things look more like "C" |
| 22:40.56 | [TK]D-Fender | angryuser: Can't do anything more, probably does LESS for special stuff |
| 22:41.20 | ManxPower | angryuser: It CANNOT have much impact, as AEL is converted to regular dialplan stuff on startup. |
| 22:41.27 | angryuser | i see |
| 22:41.32 | ManxPower | there's even some standalone utility with asterisk that takes ael and converts it for you in batch mode |
| 22:41.57 | angryuser | thank for info |
| 22:42.02 | ManxPower | But AEL does allow you a more "programming" style of syntax, and I like that idea. |
| 22:43.43 | angryuser | i like that style too |
| 22:46.15 | dlynes | [TK]D-Fender: oh well...guess I'll just use authenticate first, then |
| 22:50.28 | *** join/#asterisk zapp-branigan (n=malebolg@9.218.216.87.static.jazztel.es) |
| 22:50.46 | magic_hat | [TK]D: your asterisk -rx approach works great. I'm wrestling a bit with the log output though, in terms of which entries are actually taking up one of my 10 available teliax channels. http://pastie.org/232401 |
| 22:52.06 | [TK]D-Fender | magic_hat: You can see 2 "outbound" channels there. |
| 22:52.35 | [TK]D-Fender | magic_hat: due to your macro |
| 22:52.44 | [TK]D-Fender | magic_hat: but clearly that is the same call |
| 22:52.54 | [TK]D-Fender | magic_hat: so 1 call out to teliax right there. |
| 22:53.21 | magic_hat | Yeah. Is there a way to crank all that code into a single process/context that just occupies a single chan? That'd be easier. |
| 22:53.42 | [TK]D-Fender | magic_hat: only thing you really have to count are the teliax channels |
| 22:54.13 | magic_hat | And that's basically the ones that have the macro-outbound-connect context, yes? |
| 22:55.31 | magic_hat | or is it outbound-handler, since some of the calls might actually get dialed to teliax but not connect. |
| 22:56.08 | [TK]D-Fender | magic_hat: Your call. Depends on wheich end starts things |
| 22:57.13 | magic_hat | this is someone else's code that I'm admittedly not grasping all that well. it seems like it'd be easier to just have a single context that answers, dials the call to teliax and then connects it to the agi server. But is there some reason not to do that? |
| 22:57.45 | [TK]D-Fender | magic_hat: Time to take some ownership and understanding of the code. |
| 22:58.00 | [TK]D-Fender | magic_hat: You should already have all the answers you need for this process.... |
| 22:58.51 | magic_hat | [TK]D, yeah, that's what I'm working on right now... I tried changing it, which is mostly how I learn, but it blew up. So I'm wondering if perhaps there's a reason why it's set up that way. |
| 22:59.49 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:00.32 | [TK]D-Fender | magic_hat: Sorry, you're just going to have to figure that out for yourself.. |
| 23:02.37 | magic_hat | lol. I'll monkey with it. But am I at least right in thinking you could do the same thing but cleaner in one context? |
| 23:04.45 | [TK]D-Fender | magic_hat: You need to understand when your channels are created and come up with the best place to put restrictions on. |
| 23:06.50 | magic_hat | Well, yeah. That was my original thought. But of course I'm also wondering whether it wouldn't be easier to just make it one call, one channel, if that's feasible. |
| 23:07.16 | magic_hat | I've only been mucking with AGI for about 3 weeks now, so it's pretty new for me. |
| 23:08.34 | [TK]D-Fender | magic_hat: I still have no idea why this has anything to do with AGI at all... |
| 23:29.58 | magic_hat | Well, it's called from an AGI script, and it hands the call back to it. |
| 23:36.37 | *** part/#asterisk Peri (n=redanti@CPE0012171ab654-CM00080d9b7243.cpe.net.cable.rogers.com) |
| 23:36.54 | *** join/#asterisk Servergod (i=Servergo@70.97.157.6) |
| 23:40.12 | *** join/#asterisk Alpha_AI (n=Ben@210.11.97.57) |
| 23:40.56 | *** join/#asterisk mandd (n=moo@bas1-toronto61-1279436026.dsl.bell.ca) |
| 23:47.58 | mandd | http://pastebin.com/m58a1cb4c |
| 23:48.02 | mandd | <PROTECTED> |
| 23:48.06 | mandd | but it is there |
| 23:49.32 | *** part/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 23:53.47 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 23:54.31 | [TK]D-Fender | mandd: Now pastebin the complete call at verbose 10, SIP DEBUG enabled. |
| 23:54.37 | jblack | mandd: You're not in the context you think you are. |
| 23:54.52 | angryuser | mandd: context |
| 23:55.02 | angryuser | netout is in 'office" |
| 23:57.26 | angryuser | hm but still he got s in there |
| 23:58.45 | Servergod | hey all, got a rtp problem, sip_nat configs and testing results here http://pastebin.com/d3ba4de60 any help much TIA |