| 00:07.41 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:20.04 | dennis00 | Are lots of people using Voipbuster with Asterisk? |
| 00:20.12 | dennis00 | I am looking for a voip provider that does not require mobile number and accepts callerid spoofing. |
| 00:26.51 | *** join/#asterisk sjobeck (n=sjobeck@valdisere.sjobeck.com) |
| 00:27.15 | *** part/#asterisk sjobeck (n=sjobeck@valdisere.sjobeck.com) |
| 00:32.20 | p3nguin | mobile number? |
| 00:33.09 | p3nguin | Many ITSPs will allow you to set your own CID. The requirements they have in place, though, may vary from provider to provider. |
| 00:36.19 | p3nguin | VoIP.ms and Flowroute both allow setting of CALLERID(num) on a per-call basis. |
| 00:37.19 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 00:37.25 | [TK]D-Fender | dennis00: http://www.dslreports.com/forum/r17892054-VoIPBuster-Caller-ID |
| 00:37.42 | [TK]D-Fender | dennis00: they do seem to limit your ability to change your CID |
| 00:39.46 | dennis00 | I see. Thanks! |
| 00:40.50 | p3nguin | I said that hours ago. |
| 00:41.53 | p3nguin | And it isn't limited to mobile numbers. Any number you can receive a call on, you can verify and use as outgoing CID number. |
| 00:43.24 | dennis00 | Why is here no Callcenter support? |
| 00:44.14 | dennis00 | And what does callerid per-call basis mean? No limitations? |
| 00:47.01 | dennis00 | voip.ms is not very cheap and unfortunately flowroute does not accept signups. |
| 00:47.51 | dennis00 | I found voicetrading.com very cheap, but they have a sister-site who is even 20% cheaper. |
| 00:51.30 | *** join/#asterisk Mango (n=Mango@d154-20-97-118.bchsia.telus.net) |
| 00:53.08 | Mango | If I have determined that I cannot handle a particular call (misdialed extension, etc) how should I let the calling party my server refuses the call? |
| 00:54.17 | [TK]D-Fender | Mango: Missing a few verbs & nouns in there. Please rephrase, and provide some pertinent details |
| 00:54.37 | Mango | Okay. Here's an example. |
| 00:54.45 | Mango | I have three phones, extension 201, 202, and 203. |
| 00:54.51 | Mango | Someone dials 204. |
| 00:55.03 | Mango | What should happen? |
| 00:56.33 | Mango | Darn it, I was missing a few words. I'm not sure how that happened. |
| 00:57.42 | Mango | I was thinking the server could issue "503 Service Unavailable" or somesuch. Would that be appropriate? |
| 00:57.43 | *** join/#asterisk coppice (n=chatzill@238.168.17.210.dyn.pacific.net.hk) |
| 00:58.43 | [TK]D-Fender | Mango: You can send back congestion() & busy(), if you have not answered yet, but thats about it. |
| 01:01.13 | Mango | Send back both congestion() and busy()? |
| 01:01.49 | [TK]D-Fender | Mango: No, either/or |
| 01:02.20 | Mango | Right. Thank you :) |
| 01:05.19 | Katty | hmmmm. |
| 01:05.23 | Katty | chinese food sounds yummy |
| 01:06.28 | coppice | yep, chinese food 3 times a day |
| 01:08.00 | dennis00 | I registered a VOIP domain! |
| 01:08.02 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 01:08.20 | [TK]D-Fender | dennis00: ..... care to qualify that? |
| 01:08.29 | *** join/#asterisk redwizard (n=kvirc@host86-131-54-254.range86-131.btcentralplus.com) |
| 01:08.34 | coppice | avoipdomain.com? |
| 01:08.41 | dennis00 | Yes! |
| 01:08.55 | *** part/#asterisk dennis00 (n=dennis@unaffiliated/dennis00) |
| 01:09.02 | *** join/#asterisk dennis00 (n=dennis@unaffiliated/dennis00) |
| 01:09.06 | dennis00 | Hmm,, did somebody kick me? |
| 01:09.13 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 01:09.19 | dennis00 | Well, hostnames are the new thing since ipv6 ^^ |
| 01:09.41 | redwizard | hi |
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| 01:22.34 | *** join/#asterisk GameGamer43 (n=GameGame@CPE-65-27-76-78.new.res.rr.com) |
| 01:23.15 | *** join/#asterisk etfonhomey (n=etfonhom@74-131-159-160.dhcp.insightbb.com) |
| 01:23.18 | *** part/#asterisk GameGamer43 (n=GameGame@CPE-65-27-76-78.new.res.rr.com) |
| 01:24.57 | p3nguin | dennis00: voicetrading.com costs more than what I currently pay with VoIP.ms... what site is 20% cheaper? |
| 01:25.36 | dennis00 | p3nguin: Does voip.ms give away discounts? Netherlansd landline, Netherlands mobile, Turkey landline. |
| 01:26.10 | dennis00 | 1.91 ct is 2x more expensive than voicetrading. |
| 01:26.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:27.07 | *** join/#asterisk errr_ (n=errr@fedora/errr) |
| 01:28.34 | p3nguin | dennis00: For me, in the US, calling to other US numbers, I pay 1.05c/min with VoIP.ms. With VoiceTrading, I would pay at least 1.20c/min (1.48c/min on the standard route). |
| 01:29.50 | dennis00 | I see, US is cheaper. True. |
| 01:30.13 | [TK]D-Fender | p3nguin: You seem to be fixedated on US which clearly doesn't seem to matter |
| 01:30.33 | [TK]D-Fender | fixated* |
| 01:31.32 | p3nguin | [tk]d-fender: I don't know so much about fixated, but I am being very specific in what I have knowledge about. |
| 01:32.12 | p3nguin | It keeps from leading someone astray that way. |
| 01:32.33 | p3nguin | Also makes it less hard for someone to argue about it. |
| 01:32.51 | p3nguin | err, makes it less likely |
| 01:35.23 | dennis00 | p3nguin: I bought the .vg extension for voip. |
| 01:35.41 | p3nguin | what.vg? |
| 01:38.22 | dennis00 | domain: voip, extension: .vg. I just don't like posting full domain names in IRC's. |
| 01:39.00 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 01:39.38 | [TK]D-Fender | dennis00: What makes a domain "VoIP"? |
| 01:40.05 | dennis00 | never mind, I registered the domain voip.vg. |
| 01:41.10 | *** join/#asterisk pawz (n=pawz@ppp118-208-169-136.lns20.bne4.internode.on.net) |
| 01:42.42 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) [NETSPLIT VICTIM] |
| 01:46.04 | Katty | sooo full |
| 01:46.07 | Katty | sprawls |
| 01:46.27 | Mango | Heh, the Chinese food was good? |
| 01:46.34 | Katty | yeshhh |
| 01:47.36 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 01:50.00 | dennis00 | Does asterisk user exist on a Asterisk installation? |
| 01:50.22 | dennis00 | I must know because i installed freepbx previously and it also rechowned /var/lib/php/session which I want to re-chown |
| 01:51.40 | p3nguin | ~freepbx |
| 01:51.41 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
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| 01:54.22 | *** join/#asterisk kam187 (n=kam187@81-179-8-102.dsl.pipex.com) |
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| 01:54.52 | [TK]D-Fender | dennis00: * doesnt' care who its running as |
| 02:03.02 | jaytee | using the sip notify on the Asterisk CLI I can force a remote reboot of a Polycom phone. The phone has to be registered on the Asterisk server though. Anyone know of a way to do it if the phone isn't registered? I've googled around for sip notify utilities but haven't found anything. |
| 02:04.28 | *** join/#asterisk jhirley (n=jhirley@adsl-145-35-144.mia.bellsouth.net) |
| 02:08.33 | *** join/#asterisk Faiz (n=otakucon@c-24-0-92-71.hsd1.nj.comcast.net) |
| 02:08.41 | *** part/#asterisk Faiz (n=otakucon@c-24-0-92-71.hsd1.nj.comcast.net) |
| 02:13.39 | *** join/#asterisk Znuff (n=ibm86@2001:0:53aa:64c:2c26:2f82:a65a:7c98) |
| 02:19.55 | *** join/#asterisk cnu (i=cnu@75.80-203-44.nextgentel.com) |
| 02:21.49 | *** join/#asterisk random_mike (n=MrT@deathstar.corp.adam.com.au) |
| 02:22.48 | random_mike | quick question re: Asterisk v1.4.28 - is it possible to have one instance of Asterisk regsitered under mutiple usernames to a SIP server? |
| 02:23.14 | [TK]D-Fender | jaytee: Raden certainly |
| 02:23.44 | jaytee | [TK]D-Fender, huh? Raden? |
| 02:23.48 | [TK]D-Fender | oops |
| 02:23.52 | [TK]D-Fender | random_mike: Certainly |
| 02:24.27 | *** join/#asterisk encinoman (n=matt@ool-18e4157b.dyn.optonline.net) |
| 02:24.32 | random_mike | [TK]D-Fender, thankyou, I shall continue the debugging of my sip.conf file. |
| 02:25.18 | random_mike | Much obliged. |
| 02:25.39 | encinoman | I have a possible codec question |
| 02:28.19 | encinoman | A2billing doesn't detect DTMF tones. There is no codec conflict in the asterisk CLI either and I got the audio to sound pretty good. Does A2Billing use some weird codec conf file that I am not aware of? |
| 02:28.28 | *** join/#asterisk `Sauron (i=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 02:29.28 | [TK]D-Fender | encinoman: A2B doesn't sue a codec. its a set of scripts for *. |
| 02:29.31 | [TK]D-Fender | use* |
| 02:30.27 | [TK]D-Fender | encinoman: You've set the wrong mode |
| 02:30.49 | encinoman | I know its a set of scripts, the audio files that I have playing back sound poor, when I play back test recordings using AGI scripts, they sound fine. I'm pretty sure the AGI scripts aren't hearing the DTMF tones. |
| 02:31.12 | encinoman | mode? |
| 02:31.16 | *** join/#asterisk jakent (n=john@c-98-233-13-157.hsd1.va.comcast.net) |
| 02:31.20 | encinoman | DTMF mode? |
| 02:32.12 | encinoman | I'm using freepbx and have tried a couple of different modes, I guess you're refering to frc2883 and the like. |
| 02:36.01 | [TK]D-Fender | encinoman: Especially when spelled properly |
| 02:36.17 | encinoman | yea sorry rfc |
| 02:41.29 | *** join/#asterisk plut0 (n=cory@cpe-72-224-58-127.nycap.res.rr.com) |
| 02:41.48 | encinoman | I'm pretty sure there are only a few choices,i nband, rfc2833, info or auto |
| 02:45.32 | encinoman | [TK]D-Fender: None of those choices work |
| 02:48.46 | plut0 | how do you handle answering a call with asterisk when the caller is an IVR but asterisk is still ringing the extension, by the time the extension picks up the person missed what the IVR was saying |
| 02:51.23 | [TK]D-Fender | encinoman: Perhapsits not even matching the peer you're changing and thus your settings are rendered meaningless |
| 02:51.51 | [TK]D-Fender | encinoman: Hard to say since I don't see a pastebin with SIP DEBUG in it along with peer configs, etc |
| 02:52.32 | [TK]D-Fender | plut0: Don't answer first then. |
| 02:52.56 | plut0 | so the dialplan should ring the extension but not call Answer() ? |
| 02:53.13 | [TK]D-Fender | plut0: Sounds like thats what you want to do... so go do it :) |
| 02:53.19 | plut0 | thanks :) |
| 02:53.38 | encinoman | [TK]D-Fender: OK, let me get that. |
| 02:53.47 | *** join/#asterisk luckyaba (n=lucky@ip72-194-218-169.sb.sd.cox.net) |
| 02:59.22 | plut0 | what is the best quality codec? |
| 03:00.27 | p3nguin | Best quality under what conditions? |
| 03:00.38 | plut0 | no conditions |
| 03:03.36 | plut0 | if you're not worried about bandwidth whats the best? |
| 03:03.39 | [TK]D-Fender | plut0: WAv 44.1khz Stereo. Go buy a CD. |
| 03:04.12 | plut0 | ha ha , i'm talking about asterisk |
| 03:04.42 | [TK]D-Fender | plut0: ulaw/alaw dependsong on where you will termate to the PSTN |
| 03:04.49 | random_mike | Regarding registering asterisk to multiple sip servers - does anyone have an example of how to configure sip.conf ? |
| 03:05.09 | [TK]D-Fender | random_mike: add more REGISTER lines. |
| 03:05.14 | [TK]D-Fender | random_mike: The End (tm) |
| 03:05.21 | plut0 | north america |
| 03:05.28 | [TK]D-Fender | plut0: ulaw it si |
| 03:05.29 | random_mike | great - thanks |
| 03:05.39 | plut0 | how does g.729 compare? |
| 03:06.06 | [TK]D-Fender | plut0: A lot lower |
| 03:07.02 | plut0 | which is best when bandwidth is an issue? |
| 03:07.26 | *** join/#asterisk luckyaba (n=lucky@ip72-194-218-169.sb.sd.cox.net) |
| 03:08.20 | *** join/#asterisk jhirley (n=jhirley@adsl-161-131-165.mia.bellsouth.net) |
| 03:09.23 | p3nguin | It seems g.729 |
| 03:09.50 | p3nguin | You could always sink to gsm if you're really hurtin' |
| 03:10.36 | Naikrovek | that would have to be some severe hurtin' |
| 03:10.53 | plut0 | thanks |
| 03:11.02 | [TK]D-Fender | G.723.1 if you're extremely squeezed, G.729 is the better common choice |
| 03:12.19 | encinoman | member:%5BTK%5DD-Fender: http://pastebin.com/m60f81064 |
| 03:13.22 | encinoman | [TK]D-Fender: http://pastebin.com/m60f81064 |
| 03:13.53 | encinoman | Sorry screwed it up. Anyway, there is the debug code with the dtmfmode=auto |
| 03:15.05 | [TK]D-Fender | encinoman: Found no matching peer or user for '208.87.41.32:5060' <-- failure to match. Your peer settings = worthless as suspected |
| 03:15.54 | encinoman | Great, I guess thats where I am stuck. |
| 03:27.06 | encinoman | [TK]D-Fender: Thanks for the help, I'm going keep looking at this and trying to fix it. Thanks again. |
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| 03:36.58 | *** part/#asterisk encinoman (n=matt@ool-18e4157b.dyn.optonline.net) |
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| 03:49.22 | _jmcdowell | Hell all.. |
| 03:49.39 | _jmcdowell | Any experts around willing to help me figure out an issues with asterisk? |
| 03:50.05 | _jmcdowell | When I try and place a call, it fails every time, and also fails to explain why as far as I can tell. |
| 03:50.43 | scunizi | I know that gastman is outdated.. however is it only suppose to ask for the connection information ie IP address user name and password? without any other gui asterisk manipulation |
| 03:53.21 | _jmcdowell | http://pastebin.com/m3c0c2cf5 |
| 03:54.22 | [TK]D-Fender | _jmcdowell: FreePBX is not supported here. Go ask in their channel |
| 03:54.27 | [TK]D-Fender | ~freepbx |
| 03:54.28 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 03:54.52 | *** part/#asterisk _jmcdowell (n=noone@174-154-101-158.pools.spcsdns.net) |
| 03:56.20 | [TK]D-Fender | scunizi: gastman is an AMI management tool and has nothing to do with your other configs. Its loks at live status, and because development supposedly stopped so long ago may have compatibility issues with higher versions of * |
| 03:58.38 | scunizi | [TK]D-Fender: thanks.. I did run it and entered the IP of my IP-PBX at work along with user name and password.. seemed to like it but left the terminal in a state that suggests gastman is still running.. Not sure what AMI is (new to this) but sounds like it presets the entered values in a conf file someplace.. is this true? or am I way off base? |
| 03:59.42 | [TK]D-Fender | scunizi: ...~ |
| 03:59.44 | [TK]D-Fender | book |
| 03:59.48 | [TK]D-Fender | ~book |
| 03:59.49 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 03:59.50 | [TK]D-Fender | ~ami |
| 03:59.50 | infobot | somebody said ami was the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
| 03:59.57 | Raden | evening |
| 04:01.25 | scunizi | [TK]D-Fender: thanks for the link.. I've got the PDF and have read through chapt 3.. I'll keep reading.. getting use to new acronyms. |
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| 05:05.37 | erict | Hi everyone. I am interested in giving my secretary a "switchboard" phone... so she can see the status of lines in a call center envrionment... anyone have experience with this? I am not talking about a software based solution, but rather one of the sidecars for polycom phones.. with line indicators.. |
| 05:06.25 | erict | I |
| 05:06.44 | erict | I am guessing it needs to buddy watch feature in asterisk enabled. that's simple - just wondering if someone knows it works. |
| 05:06.45 | dlynes | erict, on polycom i believe the technology is called 'buddy lights', or something similar |
| 05:06.59 | dlynes | erict, in asterisk, it's called blf |
| 05:07.13 | erict | ok, but the bottom line is: it works. |
| 05:07.19 | dlynes | erict, blf means 'busy lamp field' |
| 05:07.19 | erict | right? |
| 05:07.42 | dlynes | erict, correct, but the polycom has an upward maximum number of blf's that it can monitor per phone |
| 05:08.18 | erict | I am not set on polycom |
| 05:08.32 | carrar | erict, with the polycom 601 you can have up to 3 sidecars on 1 phone, eachwith 14 buttons giving you 42 users |
| 05:08.49 | dlynes | and, if blf's in asterisk are anything like they used to be, you could be spending a bit of time troubleshooting before everything works perfectly |
| 05:09.13 | carrar | erict, the polycom work great for that sort of stuff |
| 05:09.20 | [TK]D-Fender | erict: How many do you need to monitor? |
| 05:09.26 | erict | 24 |
| 05:09.27 | dlynes | carrar, only 42 sidecar buttons maximum on a 601? |
| 05:09.36 | carrar | plus 6 on the phone |
| 05:09.42 | dlynes | erict, then the max number supported is not an issue |
| 05:09.42 | [TK]D-Fender | erict: Then 2 sidecars will do it. |
| 05:09.45 | carrar | but you need 1 or 2 for incoming lines |
| 05:10.08 | erict | I have 18 lines to monitor, and 6 incoming :) |
| 05:10.11 | [TK]D-Fender | erict: Add contacts like normal. Enable Buddy Watch on them. Set up your "hints'. The End |
| 05:10.24 | carrar | cake |
| 05:10.33 | erict | so, I can actually test and configure this without buying the sidecar first. :) |
| 05:10.34 | carrar | Cheese Cake! |
| 05:10.39 | erict | carrot cake? |
| 05:10.43 | carrar | correct |
| 05:10.48 | erict | sweet ;) |
| 05:10.48 | [TK]D-Fender | erict: What is this talk of "lines"? |
| 05:11.05 | carrar | I'm ignoring the "lines" :) |
| 05:11.05 | *** join/#asterisk jakent_ (n=john@c-98-233-13-157.hsd1.va.comcast.net) |
| 05:11.06 | erict | [TK]D-Fender, I mean channels |
| 05:11.16 | erict | or.. extensions actually |
| 05:11.19 | erict | sorry, long weekend |
| 05:11.37 | [TK]D-Fender | erict: You mean PHONES <- |
| 05:11.40 | carrar | 18 sip extensions to monitor |
| 05:11.43 | carrar | easy |
| 05:12.17 | erict | indeed |
| 05:12.33 | erict | just checking to make sure it's all gravy before placing the order :) |
| 05:13.34 | carrar | erict: http://pics.osburn.com/photo/33430/original |
| 05:13.50 | carrar | http://pics.osburn.com/photo/43548/original |
| 05:13.53 | carrar | works great |
| 05:14.16 | erict | that's pretty |
| 05:14.17 | erict | ;) |
| 05:14.34 | carrar | I have a few customers with with 3 sidecars |
| 05:14.51 | erict | my secretary has a ip550 currently |
| 05:14.56 | erict | looks like i need to go to a 601 to do this |
| 05:15.07 | carrar | I have not used the 550 yet |
| 05:15.35 | erict | either way, it works |
| 05:15.39 | erict | that's what i really needed to know :) |
| 05:15.41 | carrar | yup |
| 05:15.47 | erict | i'll figure it out from there |
| 05:15.49 | carrar | polycom phones kick ass |
| 05:15.58 | erict | yeah, i have a ton of them :) |
| 05:16.01 | carrar | and their wifi phones too |
| 05:16.14 | erict | have not tried any of those |
| 05:20.48 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 05:21.33 | erict | thanks for your help, carrar, [TK]D-Fender, and dlynes |
| 05:21.53 | [TK]D-Fender | erict: 601 = discontinued. |
| 05:21.57 | [TK]D-Fender | erict: Avoid |
| 05:22.02 | [TK]D-Fender | erict: IP 650+ |
| 05:22.29 | erict | avoid the 601, get 650? |
| 05:22.59 | erict | wow. $700 worth of a phone to buy. hah |
| 05:23.11 | [TK]D-Fender | erict: 650 is only a bit more than the 601 |
| 05:23.12 | erict | $270 for the 650, $170 each for the sidecars |
| 05:46.56 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
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| 05:55.53 | *** part/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) |
| 06:22.41 | *** part/#asterisk plut0 (n=cory@cpe-72-224-58-127.nycap.res.rr.com) |
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| 06:36.41 | donatas | if I configure Queues, how should I fix that source address would not ring ? |
| 06:43.51 | ChannelZ | doesn't understand the question |
| 06:48.14 | donatas | ChannelZ: I have a Queue: member => SIP/111, member => SIP/112, member => SIP/113. If I place a call from 111, I got a call from myself too. |
| 06:49.06 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 06:50.31 | *** join/#asterisk benngard (n=benngard@213.88.138.230) |
| 06:57.37 | ChannelZ | ringinuse=no |
| 06:57.43 | *** join/#asterisk Tim_Toady (n=moi@188.4.42.74.dsl.dyn.forthnet.gr) |
| 06:58.32 | donatas | ChannelZ: are you sure? |
| 06:58.50 | donatas | I set it, but no changes |
| 07:03.51 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
| 07:08.18 | *** part/#asterisk kam187 (n=kam187@81-179-8-102.dsl.pipex.com) |
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| 07:22.34 | ChannelZ | well it largely depends on the device being able to signify that it is 'in use' inthe first place. Your phone might not do so if it has 'call waiting' turned on. But really, do your queue members often call their own queues? |
| 07:24.30 | *** join/#asterisk cuco (n=Diego@local.xorcom.com) |
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| 07:42.15 | *** part/#asterisk ArtemMakhutov (n=ArtemMak@ip-95-223-6-41.unitymediagroup.de) |
| 07:43.30 | p3nguin | Don't forget to reload the queues, too. |
| 07:45.07 | *** join/#asterisk Ad-Hoc (n=nimbus@62.169.216.185) |
| 07:49.44 | ChannelZ | that too |
| 07:52.20 | p3nguin | With a small shop where there are only a few phones in the queue, I would want call waiting enabled and ringinuse = yes so that one person can take more than one call. So just don't ever call yourself, and there won't even be a problem. |
| 07:54.51 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 07:57.20 | ChannelZ | well I could see if you are on a call with someone and the queue keeps trying to ring through another call to you, it'd get pretty annoying with the constant beeping |
| 08:00.14 | p3nguin | "Please hold." |
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| 08:27.21 | donatas | Is it ok, that if I drop a call, agent is logged out. |
| 08:27.27 | donatas | ? |
| 08:28.06 | TommyBotten | What do you mean? |
| 08:28.50 | donatas | AgentLogin |
| 08:28.54 | af_ | donatas, what release? queue and agents? |
| 08:29.01 | donatas | af_: yes, both |
| 08:29.05 | af_ | oh agentlogin, I would prefer agentlogincallback |
| 08:29.16 | af_ | but I have not studied ebough them |
| 08:29.24 | af_ | they are ver complicated |
| 08:29.34 | af_ | say, testing them in a real env is tough |
| 08:29.56 | af_ | I think the difference is in the operationl model |
| 08:30.10 | *** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
| 08:30.35 | af_ | you know humans make a difference here, feeling, habits |
| 08:31.12 | af_ | sorry, practice |
| 08:31.23 | af_ | oi procedures |
| 08:31.30 | af_ | s/oi/or/ |
| 08:31.37 | af_ | yeah, infobot |
| 08:31.41 | af_ | you right :) |
| 08:31.45 | af_ | me too 8-) |
| 08:32.23 | donatas | who can give me an simple example of agent+queues? |
| 08:32.24 | af_ | I think agentlogin is more dumb in a sense. |
| 08:32.41 | af_ | mhh, there is a simple example of that? |
| 08:32.59 | af_ | say, three agents, and two queues? |
| 08:33.06 | donatas | one queue :) |
| 08:33.14 | af_ | hot can test them? I need three real agents? |
| 08:33.18 | af_ | plus me? |
| 08:33.26 | af_ | it cost shitload of money |
| 08:33.31 | af_ | s/cost/costs/ |
| 08:33.48 | af_ | oh, you are a real bot, I love it |
| 08:34.00 | af_ | any help on it? |
| 08:34.15 | af_ | syntax corrector? |
| 08:34.35 | af_ | looks in the empty |
| 08:34.48 | af_ | & |
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| 08:54.30 | themolester | Is there a simple way to auto-reconnect sip trunks after internet outage? |
| 08:54.44 | themolester | ie, without logging on and reloading |
| 08:55.42 | *** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com) |
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| 08:57.57 | E-bola | Hmmm there seems to be an issue with the queue on our asterisk pbx |
| 08:58.08 | E-bola | the users reported that sometimes ppl wait for ages, and i just verified it |
| 08:58.26 | E-bola | some1 had waited for 6 minutes, and i was in queue behind them, but apparently i got "picked up" before them #1 slot |
| 08:58.28 | E-bola | how can that happen? |
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| 09:06.15 | *** join/#asterisk aiksa[LV] (n=aiksa[LV@mx.fiveplus.lv) |
| 09:06.40 | aiksa[LV] | hi everyone. I have another short question regarding pre 1.6 Asterisk manager interface |
| 09:06.53 | aiksa[LV] | should hangup event be observed for every channel created? |
| 09:07.16 | aiksa[LV] | or are there exception to this. (taking into consideration that channel name might change with rename) |
| 09:07.56 | aiksa[LV] | to rephrase: should for every Newchannel event there alse be a Hangup event, or not? |
| 09:10.54 | *** part/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com) |
| 09:13.38 | *** join/#asterisk jkroon (n=jkroon@uriel.interexcel.co.za) |
| 09:14.58 | ChannelZ | aiksa[LV]: not sure about 1.4 but do you have read=call for your manager user? |
| 09:16.46 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 09:17.34 | aiksa[LV] | ChannelZ: yup, of course :) |
| 09:17.46 | *** join/#asterisk DelphiWorld (n=Miranda@41.110.0.199) |
| 09:17.48 | DelphiWorld | hi |
| 09:17.59 | DelphiWorld | dlynes: 00441992200010 |
| 09:18.06 | ChannelZ | ok.. well I don't know positively that you should see a hangup on every single channel but I would certainly assume so |
| 09:18.21 | aiksa[LV] | ChannelZ: I am experiencing thsi strange behaviour, that some channels for no obvious reason doesnt receive this event |
| 09:18.29 | aiksa[LV] | still scartching my head |
| 09:18.46 | aiksa[LV] | like some 5%. |
| 09:18.58 | aiksa[LV] | nothing particular about those calls, though. |
| 09:19.23 | aiksa[LV] | I am starting to think that maybe this is a problem of an AMI library I am using |
| 09:19.53 | ChannelZ | hmm maybe yes it's missing events |
| 09:21.01 | *** join/#asterisk retentiveboy (n=pdugas@adsl-221-170-128.pns.bellsouth.net) |
| 09:21.03 | aiksa[LV] | I will most probably fire up a nc session with an output pushed to a txt file |
| 09:21.12 | aiksa[LV] | and compare between the both |
| 09:21.45 | aiksa[LV] | I am rewriting my asterisk connection core, when I struck upon this behaviour. |
| 09:22.00 | *** join/#asterisk lftsy (n=mlr@install.deckpoint.ch) |
| 09:22.03 | *** part/#asterisk DelphiWorld (n=Miranda@41.110.0.199) |
| 09:22.18 | ChannelZ | Well good luck - I'm off to bed |
| 09:22.25 | aiksa[LV] | everything Channel, Call, Extension, Queue will be an instance of an appropriate class with event generators etc. |
| 09:22.37 | aiksa[LV] | ChannelZ: sleep tight. |
| 09:23.51 | jkroon | hi guys, i've got a client with a TDM800, 2 Quad FXO cards on it, and from time to time I need to restart the dahdi drivers to get it working again. I've got two clients with this problem, but the one is just entirely belly up today, needed a full reboot (normally unloading/reloading the modules was sufficient, dahdi 2.1.0.4 |
| 09:28.43 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) |
| 09:31.43 | *** join/#asterisk sun28 (n=light@sun28.ipfw.su) |
| 09:44.34 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 09:44.34 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.1 (2010/01/15), Asterisk 1.6.1.13 (2010/01/15), 1.6.0.21 (2010/01/15), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow #switchvox #asterisk-bugs #asteris |
| 09:44.46 | tzafrir | make sure you have wctdm24xxp there |
| 09:45.03 | Bladerunner05 | tzafrir: on this box using zaptel |
| 09:45.28 | Bladerunner05 | Zaptel Version: 1.4.12.1 and Asterisk 1.4.17 |
| 09:45.34 | tzafrir | Look at how MODULES is set in /etc/default/zaptel or /etc/sysconfig/zaptel , then |
| 09:46.16 | *** join/#asterisk DelphiWorld (n=Miranda@41.110.0.199) |
| 09:46.18 | DelphiWorld | ;) |
| 09:46.53 | DelphiWorld | #/join #asterisk-biz |
| 09:47.41 | Bladerunner05 | if I do lsmod I see ztdummy, wctdm, zaptel used by ztdummy,wctdm |
| 09:47.56 | ThoMe | tzafrir: huhu |
| 09:47.59 | tzafrir | Bladerunner05, you need wctdm24xxp loaded |
| 09:48.05 | ThoMe | how i can get only ttyIAX04 from IAX2/ttyIAX04-6128 ? |
| 09:48.15 | DelphiWorld | hi tzafrir! |
| 09:48.16 | *** join/#asterisk sun28 (n=light@sun28.ipfw.su) |
| 09:48.21 | Bladerunner05 | tzafrir: I do modprobe wctdm24xxp ? |
| 09:48.22 | tzafrir | ThoMe, Cut() |
| 09:48.25 | ThoMe | ah cut |
| 09:48.40 | *** join/#asterisk casix (n=casix@xenpbxedifici.adamvozip.es) |
| 09:48.43 | tzafrir | Bladerunner05, that's for manually loading, yes |
| 09:48.57 | Bladerunner05 | tzafrir: otherwise ? |
| 09:49.12 | tzafrir | Do you normally load it in an init.d script? |
| 09:49.12 | *** part/#asterisk DelphiWorld (n=Miranda@41.110.0.199) |
| 09:49.47 | Bladerunner05 | If I do modprobe wctdm24xxp module not found.. do U believe I have to recompile zatpel on this box ? |
| 09:49.50 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 09:51.14 | ThoMe | tzafrir: this works fine: |
| 09:51.14 | ThoMe | <PROTECTED> |
| 09:51.15 | ThoMe | <PROTECTED> |
| 09:51.15 | ThoMe | <PROTECTED> |
| 09:51.21 | ThoMe | but is it ok? |
| 09:51.34 | *** join/#asterisk DelphiWorld (n=Miranda@41.110.0.199) |
| 09:51.42 | DelphiWorld | any on e want to test algeria route? |
| 09:51.58 | DelphiWorld | anyone want to test algeria route? |
| 09:52.39 | DelphiWorld | we need a tester please |
| 09:52.46 | DelphiWorld | to call from sip to algeria (DZ) |
| 09:52.56 | sun28 | moin \o/ |
| 09:53.00 | ThoMe | sun28: tach |
| 09:54.56 | tzafrir | DelphiWorld, from algeria to where? |
| 09:55.14 | DelphiWorld | tzafrir: no, from world to algeria |
| 09:55.20 | DelphiWorld | like from your softswitch to algeria |
| 09:56.02 | themolester | is there a timeout value for a sip connection to become established? |
| 09:56.19 | aiksa[LV] | ThoMe: should work in most cases |
| 09:56.51 | themolester | I believe my asterisk is taking too long to set up, and appears to be sending BYE a second or so before the connection is up |
| 09:56.58 | aiksa[LV] | ThoMe: i suspect this will fail if you peername has a dash in it (though i am not sure if this is even allowed) |
| 09:59.23 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 10:00.30 | DelphiWorld | lol no one want to test! |
| 10:00.57 | themolester | DelphiWorld I don't know anyone in algeria |
| 10:00.58 | themolester | :) |
| 10:01.02 | Bladerunner05 | tzafrir: in /etc/default/zaptel I have to leave only MODULES="$MODULES wctdm24xxp" ? |
| 10:01.31 | DelphiWorld | themolester: not required... i give u some number to dial just test if you can |
| 10:01.44 | tzafrir | Bladerunner05, MODULES="wctdm24xxp" |
| 10:01.51 | tzafrir | will work as well |
| 10:02.01 | Bladerunner05 | tzafrir: ok I do... and try... |
| 10:02.32 | *** join/#asterisk mlarsen (n=mlarsen@212.37.141.188) |
| 10:02.52 | themolester | DelphiWorld unfortunately, I'm also having a problem with my outgoing so I am not an ideal candidate.... I just thought it would be funny :) |
| 10:04.51 | DelphiWorld | strange |
| 10:04.57 | Bladerunner05 | tzafrir: while booting it said that don't find any hardware interface and so load ztdummy as timing interface.... |
| 10:07.45 | *** part/#asterisk DelphiWorld (n=Miranda@41.110.0.199) |
| 10:14.05 | tzafrir | Bladerunner05, typo? |
| 10:14.16 | tzafrir | does a manual modprobe find the card? |
| 10:17.37 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
| 10:17.48 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 10:18.46 | Bladerunner05 | tzafrir: if I do modprobe wctdm24xxp it said: fatal module not found... |
| 10:18.53 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 10:18.55 | Gido-E | one x? |
| 10:19.31 | Gido-E | it is bluemonday today... |
| 10:20.26 | tzafrir | Bladerunner05, built it yourself? |
| 10:20.41 | tzafrir | maybe it was disabled? |
| 10:21.03 | Bladerunner05 | tzafrir: I recompile zaptel...., what can I do ? |
| 10:21.25 | Bladerunner05 | tzafrir: how may I check if was disabled ? |
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| 10:24.54 | *** join/#asterisk bzing2 (n=dr105@dhcp-194-66-208-236.canterbury.ac.uk) |
| 10:27.53 | themolester | could someone take a look at this? http://pastebin.ca/1755833 |
| 10:28.19 | themolester | I really don't know what I'm looking at, but does that appear that asterisk is hanging up before the connection is established? |
| 10:30.46 | *** join/#asterisk Cain (n=Geek@unaffiliated/cain) |
| 10:31.27 | themolester | Cain, try pastebin.ca instead of pasting on irc |
| 10:34.14 | *** join/#asterisk Cain (n=Geek@unaffiliated/cain) |
| 10:35.54 | *** join/#asterisk Cain (n=Geek@unaffiliated/cain) |
| 10:38.54 | *** join/#asterisk Cain (n=Geek@unaffiliated/cain) |
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| 10:42.44 | Zhad | Is anyone here familiar with the snom 200 ? |
| 10:51.22 | *** join/#asterisk Peste_Bubonica (n=eduardo@189.63.200.186) |
| 10:55.00 | Bladerunner05 | tzafrir: may I use zaptel 1.4.12 with tdm410p or I have to install dahdi ? |
| 10:55.44 | tzafrir | Bladerunner05, zaptel 1.4.12 should work with that card |
| 10:58.48 | Bladerunner05 | tzafrir: please take a look here http://www.pastebin.ca/1755854 |
| 11:00.09 | tzafrir | Bladerunner05, that's meaningless. ls kernel/*/*.ko |
| 11:01.01 | tzafrir | also, find /lib/modules -name wctdm24xxp.ko -o -name zaptel.ko |
| 11:01.23 | Bladerunner05 | tzafrir: /lib/modules/2.6.24-etchnhalf.1-486/misc/zaptel.ko |
| 11:01.38 | Bladerunner05 | tzafrir: what can I do to resolve this ? |
| 11:02.02 | tzafrir | Have you explicitly disabled building that module? |
| 11:02.24 | Bladerunner05 | tzafrir: I remember not, how can I check and enalbe it ? |
| 11:04.04 | tzafrir | in zaptel it could be disabled in menuselect . So I guess you can try either re-running menuselect or deleting its config file (menuselelct.makeopts ) |
| 11:05.21 | Bladerunner05 | tzafrir: thanks a lot, I try and recompile zaptel... |
| 11:06.48 | Bladerunner05 | tzafrir: make now is making a lot of device modules... |
| 11:10.55 | *** join/#asterisk cnu (i=cnu@75.80-203-44.nextgentel.com) |
| 11:11.41 | *** join/#asterisk ramindia (n=balajibh@96-10.southernonline.net) |
| 11:11.56 | ramindia | any one have idea about this error "FATAL: Error inserting zaptel (/lib/modules/2.6.26-1-686/asterisk/misc/zaptel.ko): Invalid module format" |
| 11:12.06 | ramindia | on debian |
| 11:13.26 | *** join/#asterisk cnu (i=cnu@75.80-203-44.nextgentel.com) |
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| 11:24.34 | creativx | hehe |
| 11:24.45 | creativx | uups.. porting number series in the middle of the day |
| 11:25.09 | creativx | and forgetting to uncomment playback(tt-weasels) |
| 11:25.17 | creativx | or comment it out that is |
| 11:25.44 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 11:25.44 | E-bola | hehe |
| 11:27.16 | *** join/#asterisk jbw (n=jbw@dsl-105-162.cust.imagine.ie) |
| 11:30.05 | Bladerunner05 | tzafrir: thank you, now works fine |
| 11:31.51 | tzafrir | ramindia, don't you use kernel 2.6.26-2-686 ? |
| 11:32.30 | tzafrir | ramindia, also: how have you installed it? Did you copy the module manually there? |
| 11:36.15 | ramindia | tzafrir: i have installed from SVN |
| 11:36.25 | ramindia | no i installed not copied |
| 11:36.54 | ramindia | this is my version "Linux sip 2.6.26-2-686 #1 SMP Wed Nov 4 20:45:37 UTC 2009 i686 GNU/Linux |
| 11:37.46 | tzafrir | What command did you run that emited this error message? |
| 11:37.58 | ramindia | modprobe zaptel |
| 11:38.07 | ramindia | modprobe ztdummy |
| 11:38.37 | *** join/#asterisk diatonic_afk (n=diatonic@mail.clearwater-research.com) |
| 11:39.54 | ramindia | when i try to run from /etc/init.d/zaptel start "Waiting for zap to come online...Error: missing /dev/zap!" |
| 11:44.04 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 11:48.14 | *** part/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 11:55.23 | tzafrir | ramindia, what's the output of: modinfo zaptel |
| 11:55.49 | *** join/#asterisk fors1 (n=forsen@pat-tdc.opera.com) |
| 12:00.17 | Zhad | Is anyone here familiar with the snom 200? |
| 12:03.10 | ramindia | tzafrir: http://pastebin.ca/1755904 |
| 12:05.09 | *** join/#asterisk Znuff (n=ibm86@2001:0:53aa:64c:2c26:2f82:a65a:7c98) |
| 12:12.41 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 12:16.31 | *** join/#asterisk diatonic (n=diatonic@mail.clearwater-research.com) |
| 12:17.44 | *** join/#asterisk coppice (n=chatzill@185.204.17.210.dyn.pacific.net.hk) |
| 12:19.14 | *** join/#asterisk Znuff (n=ibm86@89.165.131.103) |
| 12:27.35 | tzafrir | ramindia, nothing under /dev/zap ? Anything under /sys/class/dahdi ? |
| 12:31.02 | *** join/#asterisk italorossi (n=kvirc@189.23.15.32) |
| 12:35.05 | *** join/#asterisk sgimeno (n=santiago@226.Red-80-33-64.staticIP.rima-tde.net) |
| 12:35.16 | ramindia | i see under /dev/zap "channel ctl pseudo timer" |
| 12:35.45 | ramindia | i do not see this directory "/sys/class/dahdi" |
| 12:35.51 | tzafrir | ramindia, so the "missing /dev/zap" is false |
| 12:36.54 | ramindia | tzafrir: i can see files under /dev/zap |
| 12:38.04 | tzafrir | What happens if you run now: /etc/init.d/zaptel start |
| 12:38.13 | coppice | tzafrir: zigbee with PBXes is a strange combination |
| 12:39.29 | *** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 12:41.47 | *** join/#asterisk diatonic (n=diatonic@mail.clearwater-research.com) |
| 12:43.02 | *** join/#asterisk TheDavidFactor (n=chatzill@fw1.safedataisp.net) |
| 12:43.39 | ramindia | tzafrir: let me upgrade from 1.4.24 to 1.4.28, i see 1.4.24 buggy |
| 12:43.58 | tzafrir | ramindia, that is not related to asterisk |
| 12:44.47 | ramindia | look i have fixed zap issue, due to kernel-header package |
| 12:44.57 | ramindia | tzafrir: now i can see ztdummy loading |
| 12:45.44 | ramindia | but later i was getting "app_meetme.c:800 build_conf: Unable to open pseudo device ", so i updated to 1.4.28, now i can call conference, but i see some errors still on console |
| 12:46.22 | ramindia | my new error is " WARNING[20665]: app_meetme.c:2527 find_conf_realtime: No Zap channel available for conference, user introduction disabled" |
| 12:46.35 | ramindia | and "WARNING[20665]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info." |
| 12:49.17 | tzafrir | ramindia, ls -l /dev/zap/pseudo |
| 12:49.36 | tzafrir | Also: what user is asterisk running as? |
| 12:49.37 | ramindia | tzafrir: "crw-rw---- 1 asterisk asterisk 196, 255 2010-01-18 06:48 /dev/zap/pseudo" |
| 12:49.44 | ramindia | root |
| 12:50.43 | ramindia | tzafrir: but i can get in to conference with password, but why this is showing still that error |
| 12:52.10 | Gido-E | is 1.4.28 zaptel compatible? i thought you sould use dahdi for that release |
| 12:54.24 | *** join/#asterisk Jenna (n=JJ@unaffiliated/jenna) |
| 12:54.57 | ManxPower-work | all 1.4.x can use Zaptel |
| 12:55.43 | ManxPower-work | Since he's running Realtime I can't help him. |
| 12:56.57 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 12:58.47 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:58.47 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:58.50 | *** join/#asterisk kazaa_lite (n=eddie@94-193-98-124.zone7.bethere.co.uk) |
| 12:59.29 | kazaa_lite | When I run ./configure for asterisk-1.4.22, i get this error: http://pastebin.com/m4a19822a |
| 13:00.38 | TommyBotten | kazaa_lite: Which distro are you using? |
| 13:01.06 | kazaa_lite | CentOS |
| 13:01.24 | ramindia | tzafrir: before iam able to connect to realtime mysql, even now iam able to connect voicemail |
| 13:01.41 | ramindia | but i see the time increasing in when i give realtime mysql status |
| 13:01.46 | Jenna | kazaa_lite, why not use the yum able repository listed at asterisk.org site ? |
| 13:01.47 | ramindia | before it use to show 0seconds |
| 13:02.02 | ramindia | ManxPower-work: thanks for the help |
| 13:02.40 | TommyBotten | kazaa_lite: Are you sure you have all build tools? |
| 13:03.05 | Jenna | TommyBotten, hi there |
| 13:03.07 | kazaa_lite | yes... i can build asterisk 1.6.0.15, 1.6.0.15 etc |
| 13:03.19 | TommyBotten | Hiya, Jenna |
| 13:03.32 | dlynes | kazaa_lite, it's a very strange build error...did you download the code from asterisk.org, or somewhere else? |
| 13:04.04 | ManxPower-work | Maybe because if Jenna uses pre-packed Asterisk then nobody will help him/her |
| 13:04.12 | *** join/#asterisk etfonhomey (n=etfonhom@74-143-192-74.static.insightbb.com) |
| 13:04.19 | kazaa_lite | from http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.4.22.tar.gz |
| 13:05.11 | tzafrir | ramindia, what's the output of: zttest -c3 |
| 13:05.13 | tzafrir | hangs? |
| 13:05.53 | ManxPower-work | kazaa_lite: why are you using such an old version of Asterisk? |
| 13:05.57 | Jenna | Just wondering why would he want to do that. but anyway his errors are way too weird |
| 13:06.07 | *** join/#asterisk etfonhomey (n=etfonhom@74-143-192-74.static.insightbb.com) |
| 13:06.12 | tzafrir | kazaa_lite, can you try a later 1.4.x tarball? |
| 13:06.17 | ramindia | tzafrir: http://pastebin.ca/1755968 |
| 13:06.38 | kazaa_lite | there is just some need for this specific version. |
| 13:06.44 | tzafrir | ramindia, so you do have a working timing source |
| 13:07.00 | ramindia | tzafrir: no zap hardware, its ztdummy |
| 13:07.01 | ManxPower-work | kazaa_lite: I hope you are backporting security fixes. |
| 13:07.17 | ManxPower-work | ramindia: if you have ztdummy then you have "zaptel hardware" |
| 13:07.34 | ramindia | then why it shows "WARNING[2443]: app_meetme.c:2527 find_conf_realtime: No Zap channel available for conference, user introduction disabled" |
| 13:07.47 | ManxPower-work | ramindia: I don't know. |
| 13:07.48 | kazaa_lite | ManxPower-work: I cannot tell:) but i am not sure why asterisk has this issue |
| 13:08.11 | ManxPower-work | kazaa_lite: me neither. But nobody is going to waste their time on such an old version. |
| 13:08.16 | tzafrir | kazaa_lite, how exactly are you building it? this looks like an error from autoconf . But the tarball already includes a configure script |
| 13:09.08 | kazaa_lite | i downloaded http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.4.22.tar.gz |
| 13:09.26 | ManxPower-work | kazaa_lite: that did not answer his question |
| 13:09.27 | Jenna | kazaa_lite, btw is that centos 2, 3 4 or 5 ? |
| 13:10.09 | kazaa_lite | extracted it and issues "./bootstrap && ./configure --prefix=/home/test/ast14_22 && make && make install && make samples" |
| 13:10.14 | *** join/#asterisk RobH_ (n=robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 13:10.15 | kazaa_lite | issues == issued |
| 13:10.21 | ManxPower-work | kazaa_lite: do not run bootstrap |
| 13:10.22 | dlynes | kazaa_lite, why are you running bootstrap? |
| 13:10.24 | tzafrir | kazaa_lite, why do you run ./bootstrap ? |
| 13:10.38 | ManxPower-work | tzafrir: maybe he didn't read the install docs? |
| 13:10.43 | kazaa_lite | i always do it:( |
| 13:10.44 | Jenna | Jenna, why do u run bootstrap |
| 13:11.03 | dlynes | kazaa_lite, you always do it if you download a cvs/svn copy |
| 13:11.04 | ManxPower-work | kazaa_lite: well stop. blow away your asterisk source dir and try again |
| 13:11.08 | dlynes | kazaa_lite, not if you download a release |
| 13:11.16 | kazaa_lite | whats the purpose of bootstrap?:( |
| 13:11.24 | kazaa_lite | ahhh... right |
| 13:11.27 | tzafrir | actually, svn also includes a configure script, so there's no need for it |
| 13:11.29 | dlynes | kazaa_lite, purpose of bootstrap is to build your configure script |
| 13:11.40 | Corydon76-dig | It's for developers to rebuild the configure script |
| 13:11.43 | tzafrir | It's only needed if you patch configure.ac and the likes of it |
| 13:11.50 | kazaa_lite | cool |
| 13:12.30 | kazaa_lite | lemme rebuild the source directory again:D |
| 13:12.32 | dlynes | kazaa_lite, btw...fwiw, 1.4.22 has a whole raft of bugs |
| 13:12.33 | Corydon76-dig | wholly unnecessary if you're not changing configure.ac or one of the m4 macros |
| 13:12.54 | dlynes | kazaa_lite, most of them security bugs |
| 13:13.10 | kazaa_lite | i see |
| 13:13.17 | ManxPower-work | dlynes: I'm sure he has plenty of money to pay for the phone calls strangers will route thru his system. |
| 13:13.32 | kazaa_lite | hehehe:P |
| 13:13.42 | dlynes | kazaa_lite, most of them involve authentication |
| 13:13.53 | ManxPower-work | kazaa_lite: It won't be funny when you get a giant phone bill. |
| 13:14.17 | dlynes | ManxPower-work, he's not the one that pays it...his boss pays it, and will quickly fire him :) |
| 13:14.41 | kazaa_lite | hehehe:P I will get things upgraded soon |
| 13:14.45 | ManxPower-work | dlynes: to be fair, most "hacked" Asterisk servers are really just idiots not setting a decent password on their SIP peers. |
| 13:14.50 | *** join/#asterisk atha (n=atha@unaffiliated/athayde) |
| 13:15.01 | dlynes | ManxPower-work, and using phone numbers for peer names |
| 13:15.06 | ManxPower-work | Oh look! user 100 has password 100! |
| 13:15.12 | tzafrir | Considering he installed it to under his home directory, he might actually just want to play with it |
| 13:15.42 | kazaa_lite | and tzafrir is correct |
| 13:16.05 | dlynes | tzafrir, actually....every bsd user i've ever worked with that has installed asterisk always untars it to his home directory |
| 13:16.44 | dlynes | before installing it to /usr/local, of course |
| 13:17.04 | tzafrir | --prefix=/home/test/ast14_22 is less comon. Even with BSD |
| 13:17.20 | dlynes | ah...didn't see where he was specifying prefix |
| 13:18.07 | *** join/#asterisk freckle (n=jon@195.74.96.122) |
| 13:18.14 | kazaa_lite | it means i am playing across various versions of asterisk:D |
| 13:18.39 | kazaa_lite | 14_22, 16_1_15, 16_1_10 etc etc:P |
| 13:19.08 | *** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.163) |
| 13:21.05 | ramindia | iam using 1.4.28, realtime " Connected to asterisk@x.x.x.x, port 3306 with username asteriskuser for 17 minutes, 33 seconds." when i issue before 1.4.24, realtime mysql status, i use to see 0 seconds, now the time increasing why ? |
| 13:23.11 | tzafrir | kazaa_lite, check out svn , and you'll be able to use 'svn switch' |
| 13:23.48 | tzafrir | Or even better: get yourself a single git-svn tree with all the versions |
| 13:23.59 | tzafrir | (but the latter is more complicated) |
| 13:27.07 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 13:28.53 | kazaa_lite | ahan.... that will make my life much simple..... :) |
| 13:30.11 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 13:41.00 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
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| 13:55.39 | *** join/#asterisk e-jones (n=jkastner@nat/redhat/x-todtcujpykrreexa) |
| 13:56.39 | dlynes | kazaa_lite, 1.6.1.15 is available? |
| 13:59.21 | dlynes | ~rtcp |
| 13:59.32 | dlynes | ~spcp |
| 13:59.38 | dlynes | hrm |
| 13:59.59 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 14:00.27 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:04.42 | redwizard | anyone run anything other than AsteriskNOW? |
| 14:05.01 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 14:06.13 | [TK]D-Fender | redwizard: Very few people here USE AseriskNOW |
| 14:06.35 | redwizard | lol |
| 14:06.51 | [TK]D-Fender | redwizard: Kinda like asking if anyone has heard of a Big Mac .... in the middle of a New York McDonalds |
| 14:07.26 | [TK]D-Fender | redwizard: *NOS is a DISTRO. This is a support channel for only 1 component of it. |
| 14:07.29 | [TK]D-Fender | *NOW |
| 14:08.22 | redwizard | i've had it running on Kubuntu but i was thinking about trying AsteriskNOW is all lol |
| 14:09.04 | redwizard | although i've tried running the installer 3 times now with no success so... |
| 14:09.19 | kazaa_lite | errr....0.15 i mean |
| 14:09.28 | [TK]D-Fender | redwizard: What difference were you expecting from this transition? |
| 14:09.34 | redwizard | a gui |
| 14:09.43 | [TK]D-Fender | redwizard: just install freePBX yourself |
| 14:09.46 | redwizard | the one i downloaded with kubuntu sucked |
| 14:09.55 | tzafrir | most people |
| 14:10.07 | redwizard | yeah thats the next thing i'll try |
| 14:11.59 | redwizard | i am making the painfull transition from windows to linux though so... :P |
| 14:15.13 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 14:15.26 | dlynes | redwizard, painful would be going in the other direction |
| 14:15.50 | ariel_ | Morning folks |
| 14:16.09 | dlynes | good morning ariel_ |
| 14:17.10 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 14:17.15 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:17.15 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:17.17 | dlynes | good morning, katty |
| 14:17.24 | Katty | hi |
| 14:18.22 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 14:19.12 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.1 (2010/01/15), Asterisk 1.6.1.13 (2010/01/15), 1.6.0.21 (2010/01/15), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #freepbx #switchvox #asterisk-bugs |
| 14:19.13 | dlynes | Katty, how's all your squirrels and lewdly named birds this morning? |
| 14:19.23 | Katty | checks crittercam |
| 14:19.32 | Katty | i'd wager hungry. |
| 14:19.43 | dlynes | You should feed them, then :) |
| 14:19.47 | Katty | they're eating. |
| 14:19.52 | dlynes | ah |
| 14:19.55 | Katty | infobot: crittercam |
| 14:19.55 | infobot | well, crittercam is Katty's broadcast of The Nut House @ http://ustre.am/8H5d |
| 14:21.29 | ManxPower-work | Just remember if you are using AsteriskNOW you won't get much help for it here. |
| 14:21.56 | dlynes | Katty, oh...wow...you guys get snow there? |
| 14:22.15 | Katty | dlynes: yes, awhile back. |
| 14:22.28 | dlynes | Katty, ah...thought you were too far south to get it |
| 14:23.07 | Katty | ha, i wish. |
| 14:25.39 | tamiel | hello, asterisk versions at http://www.asterisk.org/downloads are not up to date. (sorry if someone already said that ;) ) |
| 14:26.47 | *** join/#asterisk tiav (n=tiav@mx.fr.smartjog.net) |
| 14:29.13 | ariel_ | Well I am glad that the cold weather has gone back up north. I don't think I could have gone through another cold weekend. |
| 14:30.59 | Gido-E | why not? |
| 14:32.49 | jaytee | while I was very impressed with the special effects I'm shocked that Dances With Smurfs won Best Drama at the Golden Globes Awards. |
| 14:33.51 | coppice | Dances with Smurfs? Sounds like a blue movie |
| 14:35.21 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 14:37.09 | ManxPower-work | tamiel: try the official site of downloads.digium.com? |
| 14:38.15 | tamiel | ManxPower-work: downloads.digium.com --> I got apache index :( |
| 14:38.50 | ManxPower-work | tamiel: correct. |
| 14:39.17 | dlynes | Katty, you've got the fattest squirrels around..you know that? |
| 14:39.18 | ManxPower-work | tamiel: now navigate to the correct directory and download what you need. |
| 14:39.42 | tamiel | ManxPower-work: I found latest versions without problem at http://downloads.asterisk.org/pub/telephony/asterisk/releases/ but just notice this problem. |
| 14:41.20 | ManxPower-work | tamiel: And what is "this problem"? |
| 14:42.02 | tzafrir | downloads.digium.com != downloads.asterisk.org |
| 14:42.30 | Katty | dlynes: mmhmm |
| 14:42.47 | Katty | dlynes: and soon there will be baby fat squirrels. |
| 14:44.01 | tamiel | ManxPower-work: problem is : at http://www.asterisk.org/downloads, links to asterisk packages are not up to date . |
| 14:44.37 | ManxPower-work | tamiel: the complain to the people that run asterisk.org. Digium does not run, manage, or update asterisk.org. |
| 14:44.54 | Naikrovek | they don't? |
| 14:45.07 | ManxPower-work | Naikrovek: why would they? |
| 14:45.33 | ManxPower-work | They already have a official site. |
| 14:45.36 | Naikrovek | because digium owns asterisk, they maintain it; odd that they would not maintain the website as well |
| 14:45.58 | tamiel | ManxPower-work: asterisk.org : Copyright © 2009 Digium, Inc. |
| 14:46.14 | ManxPower-work | tamiel: the official downloads for asterisk is at downloads.digium.com |
| 14:46.45 | Naikrovek | whois asterisk.org shows mark spencer as the registrant, and dotster@digium.com as the email contact |
| 14:47.02 | ManxPower-work | Naikrovek: too bad they don't update it as often as the official site |
| 14:47.09 | Naikrovek | yeah |
| 14:47.19 | Naikrovek | it's not digium's official site |
| 14:47.24 | Naikrovek | it's asterisk's official site |
| 14:47.32 | ManxPower-work | I give up. |
| 14:47.38 | Naikrovek | but it's still owned and updated by digium |
| 14:47.38 | ManxPower-work | Download it from whereever the hell you want to |
| 14:47.50 | Naikrovek | heh |
| 14:48.32 | tzafrir | ManxPower-work, downloads.digium.com will redirect you to downloads.asterisk.org for asterisk and most other things related to free software |
| 14:50.23 | *** join/#asterisk keith4 (n=keith@unaffiliated/keith4) |
| 14:52.28 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 14:52.34 | Katty | dlynes: omnomnomnom |
| 14:53.19 | ManxPower-work | tzafrir: It did not do so for me. |
| 14:55.18 | *** join/#asterisk Yedidya (n=chatzill@host86-142-22-34.range86-142.btcentralplus.com) |
| 14:55.37 | Naikrovek | it does for me... http://downloads.digium.com/pub/asterisk/ redirects to downloads.asterisk.org/pub/asterisk/ |
| 14:55.43 | Naikrovek | never noticed that before. |
| 14:56.37 | tamiel | digium.com is business part only |
| 14:58.08 | ManxPower-work | Naikrovek: Maybe tamiel is just crazy? |
| 14:58.26 | Naikrovek | i think we're all a bit crazy, at leat |
| 14:58.31 | ManxPower-work | he said there's missing files on downloads.asterisk.org, but he found them on downloads.digium.com |
| 14:58.32 | Naikrovek | especially me and you |
| 14:58.33 | *** join/#asterisk daemon (n=paul@cpc1-linc11-2-0-cust594.12-1.cable.virginmedia.com) |
| 14:58.43 | Naikrovek | huh |
| 14:58.46 | Naikrovek | i missed that |
| 14:58.48 | ManxPower-work | odd since they are the "same site" |
| 14:59.10 | ManxPower-work | (9:44:01 AM) tamiel: ManxPower-work: problem is : at http://www.asterisk.org/downloads, links to asterisk packages are not up to date . |
| 15:02.00 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 15:02.10 | *** join/#asterisk dandre (n=daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
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| 15:04.04 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:04.52 | *** join/#asterisk titter (n=titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 15:07.03 | Katty | interesting. a Large Sweet Tea from Arby's is 180 calories, but a large pepsi is 330 |
| 15:08.01 | Chainsaw | Katty: There's more to life then calories. Avoid fructose (and by extension, corn syrup). |
| 15:08.14 | Katty | Chainsaw: what does that have to do with my comment? |
| 15:08.25 | Katty | Chainsaw: I simply found it interseting that the same size Pepsi has twice the calories. |
| 15:09.20 | Chainsaw | Katty: All cokes would. It's very acidic, sweetened up until that acidic taste is gone. |
| 15:09.26 | [TK]D-Fender | Katty: One it made from syrupy goop... the other at least the basis of a normal drink. |
| 15:09.51 | *** join/#asterisk jondecker76 (n=jondecke@h139.116.96.216.dynamic.ip.windstream.net) |
| 15:10.08 | ManxPower-work | Chainsaw: What specific problems are there with corn strup? |
| 15:10.22 | [TK]D-Fender | Katty: http://1websurfer.wordpress.com/2009/05/03/nutrition-labels-should-replace-grams-with-sugar-cubes/ |
| 15:10.26 | Chainsaw | ManxPower-work: Very high fructose content. |
| 15:10.50 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733883.dsl.bell.ca) |
| 15:11.11 | Chainsaw | ManxPower-work: Fructose should be classed as a toxin really. (Like ethanol, only you don't get the buzz because it doesn't metabolise in the brain. Equally fattening though) |
| 15:11.17 | ManxPower-work | Chainsaw: If by "very high" you mean "about 50%" then yes, it does. What is wrong with Fructose? |
| 15:11.26 | ManxPower-work | Chainsaw: Cite. Your. Source. |
| 15:11.58 | Katty | i'll Site your Source in a minute. |
| 15:12.28 | Katty | went i went through arby's this morning to get a drink, i asked for unsweet tea, but got a regular one instead. |
| 15:12.30 | Chainsaw | ManxPower-work: http://today.ucsf.edu/stories/ucsfs-lustig-discusses-the-role-of-fructose-in-pediatric-obesity/ |
| 15:12.50 | Katty | i'm happy to find that it doesn't have an overwhelmingly large ammount of empty calories. |
| 15:13.16 | Katty | considers getting half sweet/half unsweet next time. |
| 15:13.23 | ManxPower-work | Chainsaw: How about some actual scientific studies. |
| 15:13.52 | ManxPower-work | Table sugar has glucose and fructose, just like corn syrup. |
| 15:13.52 | Chainsaw | ManxPower-work: I can look the talk up for you. |
| 15:13.59 | *** join/#asterisk titter (n=titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 15:14.06 | ManxPower-work | Maybe the real problem is that people should not consume so much of either kind of sugar. |
| 15:14.19 | Katty | gotta die from something. |
| 15:14.34 | Katty | i can think of worse things to stuff down me than sugar. |
| 15:14.44 | leifmadsen | Butane |
| 15:14.44 | Chainsaw | ManxPower-work: Which is hard to do if you look at the ingredient list of any processed food. |
| 15:14.54 | ManxPower-work | Chainsaw: *nod* |
| 15:15.09 | Katty | processed foods aren't real. |
| 15:15.30 | Katty | i think 50% of every processed food item is corn. |
| 15:15.44 | Katty | speaking of corn, guess what Riddick's food allergy is |
| 15:16.06 | Katty | ended up changing is food to see if it helped his allergies...and BAM |
| 15:16.07 | Katty | corn. |
| 15:16.12 | ManxPower-work | I do LIKE the taste of real sugar in soft drinks, but I'm not under any illusion it's any better for me. |
| 15:16.28 | Katty | ManxPower-work: i would think sugar is slightly better for you than high fructose corn syrup. |
| 15:16.56 | Katty | ManxPower-work: plain ole fructose would probably be best. |
| 15:17.04 | coppice | high fructose corn syrup *is* sugar |
| 15:17.17 | Katty | it's a sweetner. |
| 15:17.20 | Katty | but it's not just sugar cane |
| 15:17.24 | ManxPower-work | coppice: But it can't be! It's bad for you! |
| 15:17.42 | ManxPower-work | Katty: It is likely your table sugar didn't come from sugar cane. |
| 15:18.04 | Katty | ManxPower-work: actually mine does. |
| 15:18.15 | dlynes | most table sugar comes from sugar beets, unless it specifically says that it comes from sugar cane |
| 15:18.25 | coppice | sugar is just a family of polymers. different length polymers in the family have names like sucrose and fructose |
| 15:18.35 | Katty | uses Sugar in the Raw |
| 15:20.09 | coppice | they are stereo isomers, though. the L or D form of some might have different biological effects, like DSD is harmless, but LSD has rather interesting effects |
| 15:20.41 | Yedidya | Hey, anyone wana talk about asterisk?! |
| 15:20.53 | ManxPower-work | ~ask |
| 15:20.54 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:20.58 | [TK]D-Fender | Yedidya: Sorry, this is now #pharmacology |
| 15:21.16 | Yedidya | LOl |
| 15:21.20 | benngard | why, we are talking about real stuff like sugar and drugs here! |
| 15:21.45 | *** join/#asterisk dandre (n=daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 15:21.51 | ManxPower-work | benngard: because nobody is asking Asterisk questions? |
| 15:22.12 | Yedidya | there is a new ControlPlayback that includes multiple FFwrd and Rewind times, |
| 15:23.00 | Yedidya | does anyone know where to find it, and if it can work with 1.6.0.x |
| 15:24.06 | ManxPower-work | Yedidya: "core show applications" "core show application ControlPlayback" |
| 15:24.51 | ManxPower-work | Yedidya: DO NOT MESSAGE ME |
| 15:25.15 | Yedidya | sorry, was tring to copy your name! |
| 15:25.32 | dlynes | Anyways...someone was wanting something about high fructose corn syrup reference to health issues: http://en.wikipedia.org/wiki/High_fructose_corn_syrup#Health effects |
| 15:26.00 | Yedidya | btw, is there a quick way to copy someones name to the input field? |
| 15:26.01 | dlynes | I guess that last space should be replaced with a '%20' |
| 15:26.39 | dlynes | Yedidya, in irc? |
| 15:26.48 | *** join/#asterisk e-jones (n=jkastner@nat/redhat/x-cpcencrmvgzqaqhe) |
| 15:27.28 | Yedidya | tha built-in ControlPlayback only supports one set of FF & Rew times, I'm looking for MULTIPLE. |
| 15:28.18 | Yedidya | dlynes: I'm using chatZilla, would be happy to use other if got better / other feachures |
| 15:28.22 | *** join/#asterisk TimeRider (n=steve@78.32.26.1) |
| 15:28.33 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 15:28.54 | Naikrovek | don't copy, just type it. use TAB key completion |
| 15:29.00 | [TK]D-Fender | Yedidya: type partial name. hit tab |
| 15:29.04 | dlynes | Yedidya, you just want to put something like 'nick,' or 'nick:' when you're typing a reply? |
| 15:29.20 | Yedidya | dlynes: yes. |
| 15:29.25 | dlynes | Yedidya, just use the tab completion feature of chatzilla....it's in your preferences dialog |
| 15:29.57 | Yedidya | ALL: thanks, it works! [of-course you know it would ...] |
| 15:36.46 | Yedidya | here's a link that refers to what a seek. If anyone can give me a pointer i'd be greatful. https://issues.asterisk.org/view.php?id=8213 |
| 15:37.58 | tzafrir | Yedidya, use the tab key to complete nicks |
| 15:38.55 | tzafrir | doh, /me is lagging |
| 15:39.10 | *** join/#asterisk sun28 (n=light@78.108.73.46) |
| 15:39.29 | Yedidya | tzafrir: no harm. |
| 15:40.34 | tzafrir | Yedidya, that issue seems to be on-hold |
| 15:41.08 | tzafrir | if you're lucky, it will apply on 1.6.0 |
| 15:41.27 | tzafrir | (it's for a trunk that was "post 1.4") |
| 15:41.55 | *** join/#asterisk sun28 (n=light@78.108.73.46) |
| 15:42.31 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:42.38 | Yedidya | tzafrir: what a shame! it seems that it wouldn't be to hard to imploment. |
| 15:43.17 | ManxPower-work | Yedidya: released versions of Asterisk do not get new features |
| 15:43.46 | coppice | which is kinda silly |
| 15:43.52 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:44.00 | benngard | hmm, i wrote a small app this weekend (originating a call from a web-page) works real well, did use Action: Orginate .... CallerID: blaha<123> that caller id shows up in callers display, shouldnt i be able to get that (in some way) callerid when i "type" the number on the caller phone? |
| 15:44.40 | tzafrir | Yedidya, it seems the original contributor lost interest in it |
| 15:44.48 | Katty | wonders if many birds would like Millet seed |
| 15:45.22 | ManxPower-work | benngard: no. That isn't callerid. |
| 15:45.25 | Yedidya | tzafrir: what would it take to revive such interest |
| 15:45.49 | casix | hello, I have a problem with a Digium, Inc. Wildcard TE410P Quad-Span card. I have this error on diferents channels: chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 7. I'm using Asterisk 1.4.21.2, asterisk-addons-1.4.7, libpri-1.4.7, zaptel-1.4.12.1 Is that a hardware problem? anyone caan I solve this problem? |
| 15:46.02 | ManxPower-work | CLID (Calling Line Identification) / CPID (Called Party Identification) |
| 15:46.19 | tzafrir | Yedidya, for starters, test the patch, I guess |
| 15:46.22 | ManxPower-work | casix: that would happen when the far end caller hangs up before the near end caller |
| 15:46.33 | tzafrir | Yedidya, mostly: an active interest |
| 15:49.12 | benngard | now i am lost, i did use the app to setup a call from my siemens dect sip to my mother in law, did set her name<number> with CallerID: and that name<number> showed up on my siemen display. |
| 15:51.44 | Yedidya | tzafrir: But how can i register my interest, whith whom or what? |
| 15:51.58 | leifmadsen | casix: if you have Digium hardware, then you're entitled to support -- just call Digium. |
| 15:52.06 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 15:52.38 | casix | ManxPower-work: but I have a lot of this errors... not just one just after a call is cancelled |
| 15:52.55 | casix | leifmadsen: yes, I'll do. thanks |
| 15:53.58 | *** join/#asterisk RobH_ (n=robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 15:54.58 | tzafrir | Yedidya, if the patch happens to apply on your system - nice |
| 15:55.27 | tzafrir | apart from that - there seems to be some work (C coding) to get that patch into shape: |
| 15:55.35 | tzafrir | https://issues.asterisk.org/view.php?id=8213#76339 |
| 15:56.18 | *** join/#asterisk dmast (n=dmast@exchange.newpointe.org) |
| 15:57.23 | Yedidya | tzafrir: I would love to try to get it to work, except I can't code in C to save my wife! |
| 15:57.30 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 16:00.35 | Katty | it's very interesting. all the squirrels show up, then they all leave for awhile. all of them |
| 16:00.41 | Katty | then they all show up again. |
| 16:00.44 | *** join/#asterisk jks2 (i=jks@193.189.93.254) |
| 16:00.44 | Naikrovek | work |
| 16:00.47 | Naikrovek | lunch |
| 16:00.54 | Katty | idk |
| 16:00.56 | Katty | it's just odd... |
| 16:01.03 | Katty | there are 4 regulars. |
| 16:01.06 | Naikrovek | they're social animals like we are. they like to do things together |
| 16:01.07 | eppigy | Katty: pong |
| 16:01.11 | Katty | eppigy: ohaider. |
| 16:01.13 | Katty | eppigy: /hug |
| 16:01.14 | eppigy | herro |
| 16:01.17 | eppigy | HUGGLES |
| 16:01.29 | Katty | Naikrovek: hmm. perhaps, but they sure get into a fight if someone gets too close to their food. |
| 16:01.47 | Chainsaw | Squirrel barfight. Awesome. |
| 16:01.55 | Katty | oh it's hilarious |
| 16:01.57 | Katty | they 'bark' |
| 16:02.02 | Katty | it's not really a dog bark |
| 16:02.13 | Katty | but an irritated yipping |
| 16:02.26 | Chainsaw | *G* |
| 16:02.40 | jks2 | anyone else experienced problems with the polycom kws 300/6000 and asterisk? (missing sound) |
| 16:03.43 | *** part/#asterisk benngard (n=benngard@213.88.138.230) |
| 16:04.17 | Naikrovek | no; all my polycoms work great on asterisk |
| 16:04.32 | jks2 | Naikrovek, also the kws series? |
| 16:05.27 | Katty | http://www.youtube.com/watch?v=uFN_Yfx6fUM <- squirrel bark. |
| 16:06.07 | Katty | if someone gets to close, it sounds like that and the ears perk up |
| 16:06.10 | Katty | and the tail fluffs out |
| 16:06.22 | Katty | and shakes, much like that |
| 16:07.03 | *** join/#asterisk jakent (n=john@c-98-233-13-157.hsd1.va.comcast.net) |
| 16:07.05 | Katty | i have only seen two squirrels ont he same feeder once. |
| 16:08.33 | Naikrovek | when i was growing up, the neighbors would come to visit us; the squirrels didn't like them because they came from the house with the meanie dog. they would pick walnuts from the trees, carry them to the proper spot, then drop them on the heads of the neighbors |
| 16:08.48 | *** join/#asterisk rossand (n=aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 16:09.18 | *** join/#asterisk minotaur01 (n=minotaur@24.215.3.50) |
| 16:09.45 | Naikrovek | and they would bark loudly the entire time |
| 16:09.54 | Chainsaw | That's awesome :D |
| 16:10.07 | Naikrovek | yeah |
| 16:10.45 | Katty | lol |
| 16:10.48 | Naikrovek | it was awesome because they never ever missed |
| 16:10.53 | Katty | i didn't realize squirrels were vengful. |
| 16:10.56 | Naikrovek | not once did they hit a shoulder |
| 16:10.56 | *** join/#asterisk saxa (n=sasa@host242-95-static.223-217-b.business.telecomitalia.it) |
| 16:11.07 | saxa | hello |
| 16:11.10 | Naikrovek | not once did they hit a leg - always square on the head |
| 16:11.12 | Katty | ohaider. |
| 16:11.20 | saxa | I have a question about regiser => |
| 16:11.24 | Katty | kay |
| 16:11.45 | saxa | which is, if I register with my asterisk to the other asterisk box |
| 16:12.07 | saxa | do I need to register with the other box also to my first box, to exchange calls ? |
| 16:12.26 | Naikrovek | two asterisk servers talking to each other? |
| 16:12.27 | saxa | or is it enough that I put friend in the context ? |
| 16:12.31 | Katty | Define Exchange Calls. |
| 16:12.41 | saxa | Naikrovek: yes |
| 16:13.02 | Naikrovek | you know, i've never done sip trunking, not sure how you do it |
| 16:13.08 | Naikrovek | but that sounds right |
| 16:13.10 | saxa | in the book examples, each box is registering to th other side |
| 16:13.25 | Naikrovek | yes you want to register |
| 16:13.44 | saxa | so now, my problem is, that one box has the dyn ip the other has the static |
| 16:13.47 | *** join/#asterisk ParanoyaM (n=kvirc@93-183-242-219-dynamic.retail.datagroup.ua) |
| 16:13.48 | Naikrovek | you'll know easily that way if one box goes away |
| 16:13.59 | saxa | so the dyn ip registers ok to the static ip box |
| 16:14.00 | Naikrovek | how often does the dynamic ip change |
| 16:14.05 | saxa | but the oposite doesnt happen |
| 16:14.24 | Katty | well that's because the other box can't find you |
| 16:14.26 | Katty | your ip changed. |
| 16:14.28 | saxa | Naikrovek: every time the router gets restarted |
| 16:14.37 | Naikrovek | every time? wow |
| 16:14.49 | saxa | i mean, can be 2 or 5 days |
| 16:14.52 | saxa | depends |
| 16:15.01 | Naikrovek | interesting |
| 16:15.10 | Naikrovek | i've had the same dynamic ip for about 9 months now |
| 16:15.16 | saxa | it happened also a full week with the same ip iirc |
| 16:16.14 | Katty | have you checked the router logs? |
| 16:16.16 | saxa | Katty: my static ip box has a register of the form: register => user:pass@dynipbox |
| 16:16.38 | Katty | maybe it will give you rejection notices |
| 16:16.38 | ParanoyaM | Hi Guys, can anybody help me? i described in sip.conf 2 profiles, here is SIP.conf http://pastebin.ru/309838 , but my asterisk accepts all incoming calls |
| 16:16.39 | saxa | Katty: where dynip box is the username I use to register to the static ip |
| 16:16.46 | Katty | yes, i got that. |
| 16:16.52 | saxa | Katty: doesnt apears nothing |
| 16:16.55 | [TK]D-Fender | [11:13]<saxa>in the book examples, each box is registering to th other side <- so DON'T |
| 16:17.14 | [TK]D-Fender | saxa: You don't need to register both sides. Single peer between 2 boxes, A reg's to B |
| 16:17.15 | saxa | [TK]D-Fender: thats what i want to do |
| 16:17.17 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:17.26 | saxa | [TK]D-Fender: ok |
| 16:17.32 | [TK]D-Fender | saxa: then go do it. Its no different than a phone on one side, an ITSP on the other |
| 16:17.44 | saxa | then i need just to correct the dialplan ? |
| 16:17.56 | *** join/#asterisk Mango (n=Mango@d154-20-97-118.bchsia.telus.net) |
| 16:18.14 | saxa | [TK]D-Fender: i register from dynip to the static ip box |
| 16:18.21 | ParanoyaM | any ideas? |
| 16:18.29 | saxa | i can call from my static ip box to the dyn ip one |
| 16:19.03 | saxa | by just routing the call in the right prefix |
| 16:19.17 | Mango | Is there an easy way I can get a list of all the global variables and all the functions? |
| 16:19.24 | saxa | but i can get to ring my static ip connected phone on the dynip box |
| 16:19.29 | Mango | I was thinking of sifting through voip-info.org, but...theres a lot :P |
| 16:19.41 | [TK]D-Fender | saxa: and I see no debug for your attempts with config to match |
| 16:19.53 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 16:20.33 | ParanoyaM | why it could happen that asterisk let me call through it if no profile described for me |
| 16:20.46 | saxa | [TK]D-Fender: the question is already answered by you, i dont need to register oth boxes one to the othrer if i got it right |
| 16:21.15 | saxa | so this means i need just to rework my extensions.conf |
| 16:21.20 | saxa | correct ? |
| 16:22.06 | Gido-E | ParanoyaM what is the problem? |
| 16:22.19 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:22.28 | Naikrovek | allowguest=yes is his problem i think |
| 16:22.32 | Gido-E | ParanoyaM you probably have guest account enabled |
| 16:22.41 | Gido-E | yep, but why not :-) |
| 16:22.42 | ManxPower-work | allowguest defaults to yes |
| 16:22.42 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 16:22.50 | Gido-E | So annybody can call you on your asterisk box :-) |
| 16:22.56 | ParanoyaM | Gido-E: i've described 2 profiles http://pastebin.ru/309838 but i can call with x-lite from different ip with test user name |
| 16:23.11 | ParanoyaM | Gido-E: so question is why asterisk pass me ? |
| 16:23.35 | ManxPower-work | ParanoyaM: add allowguest=no to [general] |
| 16:23.36 | Gido-E | ParanoyaM ok, dont reinvent the wheel :-), check your guest account settings |
| 16:23.51 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 16:24.09 | ParanoyaM | Gido-E: i haven't guest accounts |
| 16:24.20 | Naikrovek | allowguest=now |
| 16:24.21 | ParanoyaM | ManxPower-work: Thanks will try right now |
| 16:24.26 | ManxPower-work | ParanoyaM: Yes you do. You always have guest accounts unless you have allowguest=no |
| 16:24.26 | Gido-E | ParanoyaM :-) |
| 16:24.27 | ParanoyaM | Naikrovek: thanx |
| 16:24.28 | Naikrovek | s/now/no/ |
| 16:25.14 | ParanoyaM | works now, whooh |
| 16:25.20 | Gido-E | ParanoyaM dheuue! |
| 16:25.33 | ParanoyaM | i worked for a month with free for all routes :D |
| 16:25.42 | ManxPower-work | I wonder how many millions of dollars of VoIP fraud has happened because allowguest= defaults to yes |
| 16:25.44 | ParanoyaM | i could be f*****d up.... |
| 16:25.55 | Naikrovek | ManxPower-work: i know |
| 16:26.02 | ManxPower-work | ParanoyaM: now go read sip.conf.sample |
| 16:26.04 | Naikrovek | if only we had the source and ability to create a patch |
| 16:26.08 | Naikrovek | :) |
| 16:26.16 | ParanoyaM | ManxPower-work: i wonder why it yes in default :) |
| 16:26.27 | Gido-E | ParanoyaM why not? |
| 16:26.30 | ManxPower-work | Naikrovek: If only we had the ability to get such a patch put in the released versions of Asterisk. |
| 16:26.39 | ParanoyaM | Gido-E: because it is unsecure |
| 16:26.43 | Gido-E | unsecure? |
| 16:26.45 | ParanoyaM | Gido-E: don't you think so? |
| 16:26.46 | *** join/#asterisk Corydon76-lap (n=Corydon7@nat/digium/x-ddusbfvtcdmeiwdo) |
| 16:26.46 | *** mode/#asterisk [+o Corydon76-lap] by ChanServ |
| 16:27.06 | Gido-E | nope :-), not knowing what you are doing is unsecure maybe. |
| 16:27.12 | ParanoyaM | :D |
| 16:27.17 | ParanoyaM | that is other question |
| 16:27.45 | Gido-E | pizza! |
| 16:28.20 | ParanoyaM | Thanks a lot guys |
| 16:28.23 | ParanoyaM | see ya |
| 16:28.32 | Naikrovek | the way my ITSP does things I have to leave allowguest=yes |
| 16:28.43 | Naikrovek | i register with one server, call audio is sent from any number of others |
| 16:28.44 | [TK]D-Fender | [11:21]<saxa>correct ? <- I don't know if your sip.conf is right. Go try things and come back if it fails |
| 16:29.00 | Naikrovek | so i have a firewall in place to restrict connections from unauthorized IPs |
| 16:29.49 | Naikrovek | and it was good and wide open there for a long time; fortunately we were never found by anyone wanting to place a lot of free calls |
| 16:30.08 | Naikrovek | and by "long time" i mean "3 years" |
| 16:30.12 | [TK]D-Fender | [11:25]<ManxPower-work>I wonder how many millions of dollars of VoIP fraud has happened because allowguest= defaults to yes <- only an idiot runs a dialplan that gives the context in [general] to do such things |
| 16:30.22 | Naikrovek | then i came to become employed here and fixed it |
| 16:31.15 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 16:32.04 | *** join/#asterisk dandre (n=daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 16:32.05 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 16:32.10 | *** join/#asterisk etfonhomey1 (n=etfonhom@74-143-192-74.static.insightbb.com) |
| 16:32.17 | *** join/#asterisk Tim_Toady (n=moi@188.4.42.74.dsl.dyn.forthnet.gr) |
| 16:32.56 | saxa | [TK]D-Fender: i use iax.conf, anyway i will try few things before |
| 16:33.26 | saxa | another question is, which is the right way to stop asterisk ? kill -9 pid ? or is there a switch or something ? |
| 16:33.53 | Naikrovek | /etc/init.d/asterisk stop ? |
| 16:34.09 | Naikrovek | ah |
| 16:34.14 | Naikrovek | asterisk -rx "stop" |
| 16:34.19 | Naikrovek | is another way |
| 16:34.23 | Naikrovek | i think |
| 16:34.29 | [TK]D-Fender | Naikrovek: Now add a parm ;) |
| 16:34.32 | Corydon76-lap | "stop now" |
| 16:39.03 | minotaur01 | is there a way to limit asterisk sip debug output to a single extension? |
| 16:39.15 | Naikrovek | you can turn on debugging per IP address I believe |
| 16:39.44 | minotaur01 | that would be good... but how? |
| 16:40.58 | [TK]D-Fender | minotaur01: help sip set |
| 16:41.45 | minotaur01 | thanks |
| 16:41.46 | ManxPower-work | Naikrovek: at my last job about once week we had panicked people calling in because they were "hacked" |
| 16:41.48 | *** join/#asterisk pawz (n=pawz@ppp118-208-178-44.lns20.bne4.internode.on.net) |
| 16:42.03 | Naikrovek | ManxPower-work: via allowguest=yes? |
| 16:42.14 | ManxPower-work | Naikrovek: no, because the customer was stupid. |
| 16:42.23 | Naikrovek | hah |
| 16:42.29 | Naikrovek | shoulda guessed |
| 16:42.38 | ManxPower-work | (usually we found at least one extension on the system with an easily guessable sip ID and password. |
| 16:43.38 | Naikrovek | ah yeah |
| 16:43.45 | Naikrovek | same as extension number or whatever |
| 16:44.05 | Mango | When an incoming call is received, is there any way to find out which peer sent the call to us? |
| 16:45.00 | [TK]D-Fender | Mango: look at the CHANNEL name |
| 16:45.50 | Mango | thx |
| 16:47.44 | minotaur01 | according to this: "sip set debug {off|on|ip addr[:port]|peer peername}" this should work: "sip set debug 172.26.18.145" but im getting an invalid command error? |
| 16:48.34 | minotaur01 | nvm i just figured it out |
| 16:50.34 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 16:50.53 | saxa | Naikrovek: [TK]D-Fender Corydon76-lap , thx all |
| 16:52.16 | Mango | [TK]D-Fender: ${CHANNEL} is SIP/peername-006625hg0. So it's everything between the first / and the last -? |
| 16:52.21 | *** join/#asterisk Diffen (n=diffen2@c-ef75e555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 16:52.23 | Mango | Or am I doing it wrong? :) |
| 16:52.39 | [TK]D-Fender | Mango: What do you think? |
| 16:53.02 | Mango | If there were a way to get simply peername that would be cool. |
| 16:53.32 | Diffen | Hello. Is it possible to do a e164 change in the asterisk? for example if someone diales +46890510 the number that are sent to the pstn gw are 00468510 |
| 16:54.18 | [TK]D-Fender | Diffen: What gets dialed out of your system is up to YOU. |
| 16:54.33 | [TK]D-Fender | Diffen: Manipulate an originating number any way you feel like |
| 16:54.58 | Diffen | d-fender ok thats nice to hear. is it at the trunk configuration i do this? |
| 16:55.15 | *** join/#asterisk smooth_penguin (n=smoove@59.95.0.181) |
| 16:55.15 | *** join/#asterisk nightrid3r (i=kvirc@41.214.154.211) |
| 16:55.24 | voipmonk | any way you feel like |
| 16:55.46 | Mango | [TK]D-Fender: Figured it out - sweet! Thanks :) |
| 16:55.48 | saxa | [Jan 18 13:54:48] ERROR[4150]: chan_iax2.c:4703 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 217.223.95.245 in the calltokenoptional list or setting user brastrak requirecalltoken=no |
| 16:55.50 | bmoraca_work | Diffen, you're using FreePBX, aren't you? |
| 16:55.53 | saxa | any ideas ? |
| 16:56.03 | saxa | what is calltoken ? |
| 16:56.04 | Diffen | bmoraca no thirdlane |
| 16:56.29 | bmoraca_work | Diffen, you'll need to ask their channel or forum for support, then. |
| 16:56.30 | [TK]D-Fender | Diffen: what "trunk configuration"? |
| 16:56.45 | [TK]D-Fender | Diffen: GUI's are not supported here, go ask in their channel |
| 16:56.53 | bmoraca_work | Diffen, people here work in vanilla asterisk without any GUIs or anything on top. they're not able to support you in this. |
| 16:57.27 | Diffen | well i dont want to do it in the gui, because you cant do it there. so i want to change it straight in the asterisk |
| 16:57.40 | bmoraca_work | Diffen, so, to answer your question: yes, ASTERISK can do what you're looking for...whether or not your GUI has been programmed to take advantage of that is a different story. |
| 16:58.11 | E-bola | I got a general question: If you got lets say 20 ip phones, which all have3 direct numbers, and need to call out via those sip accounts, how would you set that up? Normally i just have an extension that matches XXXX and everyone dials out via the same line. I guess i could give everyone a diff. context but thats a bit lame. Do i really need to write a macro to fix this? |
| 16:58.14 | bmoraca_work | Diffen, everything relating to dialplans (what is dialable, what is dialed, how numbers behave) is taken care of is extensions.conf. if you've never worked with it before, you'll want to look at the book. |
| 16:58.18 | bmoraca_work | ~book |
| 16:58.18 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:01.48 | Katty | peeks in |
| 17:02.52 | *** join/#asterisk paulc (n=paulc@unaffiliated/paulc) |
| 17:03.05 | [TK]D-Fender | E-bola: Set a var in the sip peer as to which to use |
| 17:04.15 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 17:04.40 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 17:05.46 | E-bola | [tk]D-FendeR: know of any examples of such a setup online? I do have a solution that sets a variable to the callerid when the XXXX extensions matches but it screws with my stats when i transfer to another extension because i do soemthing along the line of goto(callout,$var) |
| 17:06.08 | *** join/#asterisk minotaur01 (n=minotaur@24.215.3.50) |
| 17:06.12 | E-bola | where var is the device number, and i then have a line in the callout context which calls out with that device's pstn number |
| 17:06.14 | [TK]D-Fender | E-bola: setvar=dialoutof=providera |
| 17:06.54 | *** join/#asterisk pawz (n=pawz@ppp118-208-178-44.lns20.bne4.internode.on.net) |
| 17:06.58 | E-bola | my cdr stats then some1 show up as the called number being the caller |
| 17:07.11 | ManxPower-work | Has anyone had problems with SIPAddHeader stripping off stuff? |
| 17:07.34 | jondecker76 | can anybody think of anything that could have changed between 1.4.10 and 1.4.21.2 that would cause my asterisk to stop working? |
| 17:07.58 | jondecker76 | or is there a comprehensive list anywhere showing what changed between specific versions? |
| 17:08.04 | darkskiez | 11.2 ? |
| 17:08.06 | ManxPower-work | SIPAddHeader(Warning: 399 192.168.8.31 "Call Forward Enabled") adds Warning: 399 192.168.8.31 "Call Forward Enabled |
| 17:08.12 | ManxPower-work | Notice the missing ending " |
| 17:10.51 | ManxPower-work | Another example: exten => 5998,n,SIPAddHeader("Diversion: 4403@pbx.nyigc.net ;reason=unconditional") OR exten => 5998,n,SIPAddHeader(Diversion: 4403@pbx.nyigc.net ;reason=unconditional) sets the ACTUAL header to be Diversion: 4403@pbx.nyigc.net Notice the lack of ;reason=unconditional. |
| 17:11.13 | raden_work | how do I redirect callerid when i have my calls redirected to my cellphone or other number ? |
| 17:11.33 | Qwell | ManxPower-work: it's because the quotes are in the middle |
| 17:11.39 | Qwell | (or, really, not the first and last char) |
| 17:12.21 | Qwell | ManxPower-work: looking at the code, it's pretty obvious why it happens. it's definitely a bug |
| 17:12.27 | ManxPower-work | I have the version without the quotes |
| 17:12.36 | p3nguin | raden_work: Don't alter it before you Dial() of FollowMe() and the CID should remain on the calls. |
| 17:12.42 | p3nguin | It certainly does for me. |
| 17:12.47 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:12.48 | Qwell | basically, the code tries to strip quotes at the start and end if they exist. it doesn't care if the start quote exists when it checks the end quote |
| 17:13.00 | raden_work | p3nguin, dont use followme |
| 17:13.29 | raden_work | so then i would have to set my caller id per extension just not on the main outbound |
| 17:13.33 | jondecker76 | here is another question... I have a perl script which is creating my dialplan... In my dialplan, I have the line: |
| 17:13.42 | jondecker76 | exten => _000.,1,Meetme(${EXTEN}|q) |
| 17:13.43 | Qwell | I'm pretty sure there's an ast_stripquotes (or similar) function that should be used instead of what it's doing |
| 17:13.57 | Qwell | ManxPower-work: care to open a bug on mantis? |
| 17:14.01 | E-bola | [TK]D-Fender: This is how im using a var atm: http://pastebin.com/m26979c9d |
| 17:14.06 | jondecker76 | does this look right, or shoule ${EXTEN} actually have an extension number? |
| 17:14.08 | ManxPower-work | Qwell: This dialplan line: exten => 5998,n,Noop(Diversion: 4403@pbx.nyigc.net ;reason=unconditional) yields this: |
| 17:14.09 | ManxPower-work | <PROTECTED> |
| 17:14.15 | ManxPower-work | looks like it's a dialplan parser issue. |
| 17:14.29 | Qwell | no, it's an issue in transmit_invite |
| 17:14.45 | Qwell | I see exactly what it's doing wrong |
| 17:14.53 | *** join/#asterisk CrashHD (n=CrashHD@65.74.156.108) |
| 17:15.06 | ManxPower-work | On fact, it looks like it's the dialplan parser that is the problem. |
| 17:15.17 | Qwell | it's not |
| 17:15.49 | ManxPower-work | What would cause exten => 5998,n,Noop(Diversion: 4403@pbx.nyigc.net ;reason=unconditional) the "dialplan show" to display 3. Noop(Diversion: 4403@pbx.nyigc.net) |
| 17:16.06 | Qwell | don't know, but I can see in chan_sip where the problem is happening |
| 17:16.16 | ManxPower-work | Qwell: My examples are using NOOPS |
| 17:17.34 | ManxPower-work | In fact everything after the ; is stripped off the Noop |
| 17:17.42 | Qwell | That is a separate issue |
| 17:17.54 | ManxPower-work | Qwell: I'll have to solve that issue before the sip issue |
| 17:18.07 | Qwell | escape it |
| 17:18.23 | ManxPower-work | I just did and it seemed to work. |
| 17:20.06 | ManxPower-work | It's not working the way I *want* but at least it's not doing weird stuff. 8-) |
| 17:20.14 | Qwell | and, hrm |
| 17:20.27 | ManxPower-work | i.e. the SIP headers look like what I want. |
| 17:20.38 | Qwell | as for the quoting issue... want an easy workaround? |
| 17:20.48 | ManxPower-work | Qwell: adding \ in front of the ; ? |
| 17:20.57 | Qwell | no, the other issue |
| 17:21.09 | ManxPower-work | the quotes issue? I just added an extra quote. |
| 17:21.17 | Qwell | yeah that would do |
| 17:21.33 | Qwell | you were doing: SIPAddHeader(stuff "foo") |
| 17:21.41 | ManxPower-work | Qwell: *nod* |
| 17:21.45 | Qwell | instead; you could do: SIPAddHeader("stuff "foo"") |
| 17:22.04 | Qwell | cheesy, but it works |
| 17:22.14 | ManxPower-work | what about SIPAddHEader(stuff and more stuff "and foo"") |
| 17:22.28 | Qwell | would also work, but would be confusing to anybody reading it |
| 17:22.43 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.232.155.dsl.dyn.forthnet.gr) |
| 17:22.44 | Qwell | if you put a quote at the start, it makes it more clear what you're doing |
| 17:23.01 | ManxPower-work | Qwell: that applies to pretty much all of my dialplan. 8-) I do see your point |
| 17:23.29 | Qwell | line ~11361 of chan_sip.c in trunk, is about where the problem is |
| 17:23.54 | ManxPower-work | I just wish my Polycom would do what I want it to do when it gets a Warning: header. |
| 17:23.58 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 17:24.14 | Qwell | using ast_strip_quoted() would be much better there |
| 17:27.18 | [TK]D-Fender | jondecker76: Its whatever the exten is at that point in your dialplan. |
| 17:27.41 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
| 17:27.44 | [TK]D-Fender | jondecker76: Do not call a DEVICE (eg: sip.conf peer entry, etc) an EXTENSION |
| 17:31.07 | [TK]D-Fender | E-bola: http://pastebin.com/m8b567fd |
| 17:31.46 | E-bola | lol |
| 17:31.51 | E-bola | thats brilliant, yet simple |
| 17:32.33 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:32.40 | [TK]D-Fender | <- SMRT |
| 17:32.49 | E-bola | hmm or wait |
| 17:32.52 | E-bola | no that doesnt work |
| 17:32.58 | E-bola | ohh wait |
| 17:33.00 | E-bola | lol darn it does |
| 17:33.09 | E-bola | bows in silence :) |
| 17:33.21 | katoen | bit of a flux there :P |
| 17:34.15 | E-bola | is the syntax correct? u dont need a $ infront of the var name? |
| 17:35.11 | [TK]D-Fender | E-bola: Yeah... fix that :) |
| 17:35.34 | [TK]D-Fender | E-bola: http://pastebin.com/m180714d2 |
| 17:36.28 | E-bola | Thank you once more |
| 17:36.44 | E-bola | When opensource support works its just so much easier than commercial support hehe |
| 17:39.15 | *** join/#asterisk mgob_laptop (n=scotth@173-14-1-9-Colorado.hfc.comcastbusiness.net) |
| 17:39.18 | *** join/#asterisk HenrikJott (n=info@d83-183-134-141.cust.tele2.se) |
| 17:40.26 | mgob_laptop | Here's a quick question for a dev or someone that knows the code real well, where does the conversion of IAX2 DTMF events -> RTP happen? (we need to muck with the way DTMF is sent when converting from a IAX2 stream to SIP/RTP) |
| 17:41.18 | [TK]D-Fender | mgob_laptop: IAX2 is OOB only |
| 17:41.19 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.30) |
| 17:41.27 | [TK]D-Fender | mgob_laptop: and there is no RTP |
| 17:41.43 | HenrikJott | Hi all! I have a problem with .call-files in asterisk. Im generating call files with quite long names (to make them unique). Asterisk handles them, makes the call and everything works fine. My problem is that after the call is completed, asterisk leaves empty files in /var/spool/asterisk/outgoing/ which asterisk can´t delete for some reason and. The filenames of these files are the same as the original but with the last characters in the |
| 17:42.10 | Chainsaw | HenrikJott: That cut off at "but with the last characters in the " |
| 17:42.42 | [TK]D-Fender | HenrikJott: how are the files getting there? |
| 17:43.56 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 17:44.04 | *** join/#asterisk ChUbB (n=IceChat7@62-31-213-230.cable.ubr12.aztw.blueyonder.co.uk) |
| 17:46.14 | mgob_laptop | [TK]D-Fender, correct, but that OOB must go into RTP when that channel is bridged to SIP correct? |
| 17:46.56 | E-bola | lol ironic the problem with long files names got cut off :) |
| 17:47.06 | raden_work | OK i want are normal number to be displayed when we dial out but if someone has there phone redirected I want the inbound caller id forwarded for some reason i think im making this overly complicated |
| 17:47.19 | [TK]D-Fender | mgob_laptop: No, it goes where YOU tell it to based on the peer you dial out of |
| 17:49.16 | mgob_laptop | [TK]D-Fender, Yes, lets see if we can ASCII a bit here: Phone<-SIP->Asterisk<-IAX2->Asterisk[X]<-SIP->Dash - The X asterisk box that takes the IAX2 channel and pushes out SIP, I have to adjust DTMF there for fucked up Sonus, I know what I need to do to DTMF, I'm just wondering where it's converted/bridged from OOB->RTP. |
| 17:50.05 | [TK]D-Fender | mgob_laptop: at the point where SIP is spat out. |
| 17:50.22 | [TK]D-Fender | mgob_laptop: the peer * uses determines the mode. |
| 17:50.54 | mgob_laptop | Correct, however do you know in what file that routine is? (mode is of no concern here, everything is RFC2833, just need to add a fake leading packet to fix Sonus crap) |
| 17:51.11 | [TK]D-Fender | mgob_laptop: chan_sip.c |
| 17:51.26 | mgob_laptop | Ok, thanks. |
| 17:51.57 | *** join/#asterisk d00gster (n=doughant@77.30.19.80) |
| 17:53.34 | *** join/#asterisk hfb (n=hfb@98.112.226.53) |
| 17:54.07 | *** join/#asterisk path (i=path@server1.bshellz.net) |
| 17:54.11 | path | hello there |
| 17:54.40 | path | whats the default file for MOH (on a queue)? |
| 17:54.46 | path | cant find it |
| 17:54.59 | bmoraca_work | raden_work, what type of PSTN connectivity are you using? |
| 17:55.26 | path | Im browsing through asterisk-moh-opsound and asterisk-moh-freeplay |
| 17:55.52 | Qwell | path: anything in the moh/ directory |
| 17:56.42 | [TK]D-Fender | path: Whatever you've go where your class tells it to look |
| 17:56.46 | path | Qwell: I thought so but there isnt any moh/ directory |
| 17:56.53 | [TK]D-Fender | got* |
| 17:57.26 | *** join/#asterisk pawz (n=pawz@ppp118-208-178-44.lns20.bne4.internode.on.net) |
| 17:58.04 | path | [TK]D-Fender: there is just 'musicclass=default' on the queue |
| 17:59.35 | [TK]D-Fender | path: what does the CLASS say? |
| 17:59.53 | raden_work | bmoraca_work, SIP |
| 18:00.22 | Qwell | path: and what does musiconhold.conf say? |
| 18:00.33 | bmoraca_work | raden_work, have you made sure that your sip provider will allow you to send whatever callerid you want? not all will |
| 18:00.42 | *** join/#asterisk Cain` (n=Geek@unaffiliated/cain) |
| 18:00.48 | raden_work | bmoraca_work, yes i can :) |
| 18:00.54 | bmoraca_work | ok |
| 18:01.01 | raden_work | we are a provider |
| 18:01.13 | ManxPower-work | raden_work: make sure you have no quotes, dashes, dots, etc in the callerid number |
| 18:01.14 | raden_work | just never wanted to set a callerid per extension |
| 18:01.21 | raden_work | I dont |
| 18:01.32 | raden_work | and never redirected caller id |
| 18:01.46 | bmoraca_work | raden_work, how are your users forwarding their calls? by the phone or by some asterisk dialplan you've created? |
| 18:01.47 | ManxPower-work | "redirected callerid"? |
| 18:02.42 | raden_work | exten => 101,1,NoOp() |
| 18:02.42 | raden_work | exten => 101,n,GotoIf($[${DB_EXISTS(CFIM/${EXTEN})}]?forward:normal) |
| 18:02.42 | raden_work | exten => 101,n(forward),Dial(LOCAL/${DB(CFIM/${EXTEN})}@to-callcentric,18) |
| 18:02.42 | raden_work | exten => 101,n,Goto(vm) |
| 18:02.42 | raden_work | exten => 101,n(normal),Dial(SIP/101&SIP/120,20) |
| 18:02.49 | raden_work | ^^^ like that |
| 18:02.50 | bmoraca_work | raden_work, actually, now that I think about it...this should be the default behavior as long as you're not explicitly setting callerid |
| 18:03.05 | raden_work | out callerid is always unavailable unless i set it :( |
| 18:03.35 | p3nguin | What's the purpose of NoOp() in priority 1? |
| 18:03.50 | path | Qwell: [default] mode = files // directory = /var/lib/asterisk/moh |
| 18:03.51 | bmoraca_work | p3nguin, in what circumstance? |
| 18:03.56 | p3nguin | (1202.42) <raden_work> exten => 101,1,NoOp() |
| 18:03.59 | p3nguin | That one. |
| 18:04.00 | bmoraca_work | oh |
| 18:04.09 | raden_work | p3nguin, has todo with callcentric cant rember |
| 18:04.10 | p3nguin | Seems useless. |
| 18:04.17 | p3nguin | doubtful |
| 18:04.23 | raden_work | ok whatever |
| 18:04.56 | bmoraca_work | raden_work, callerid doesn't get touched unless you specifically touch it. are you sure your example caller doesn't have restricted callerid? |
| 18:05.12 | Qwell | path: there you go then |
| 18:05.16 | p3nguin | You should understand what you're doing instead of only doing what you read or what someone says to do. |
| 18:06.38 | *** join/#asterisk tarabuka (n=petko@78.157.4.52) |
| 18:06.59 | tarabuka | how do i tell if i'm running a 32 bit or 64 bit version of asterisk |
| 18:07.20 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:07.36 | ManxPower-work | A simple Noop(CALLERID(num)=${CALLERID(num)}) will tell you what the callerid is |
| 18:07.52 | raden_work | p3nguin, Its been there forever is there a problem does it have anything todo with caller id ? it something that was put in a long time ago for debugging something with callcentric i dont have it labeled so i dont know why it there |
| 18:08.27 | p3nguin | So will Verbose(1,${CALLERID(num)}) |
| 18:08.50 | bmoraca_work | raden_work, it doens't matter, it's not affecting anything. do what ManxPower-work did to verify that callerid exists first, and then we'll see what direction needs to be taken |
| 18:09.06 | *** join/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) |
| 18:09.46 | *** join/#asterisk verywiseman (n=khaled@unaffiliated/verywiseman) |
| 18:10.27 | raden_work | exist inbound or outbound your loosing me |
| 18:10.44 | carrar | tarabuka, you could try 'file /usr/sbin/asterisk' |
| 18:11.02 | ManxPower-work | raden_work: Both |
| 18:11.04 | bmoraca_work | raden_work, irrelevant. callerid is set per channel, whether that channel is dialing an internal phone or an external number via a PSTN provider |
| 18:11.14 | raden_work | omg :( |
| 18:11.18 | ManxPower-work | but put it right before the failing part of your dialplan |
| 18:11.28 | benngard | have u guys/girls heard about piratebay? (dont kick me i gonna show u a picture) |
| 18:11.43 | raden_work | I know i can set my caller id cause i can change it to whatever i want if i set it in my outbound context |
| 18:12.03 | tarabuka | thanks! |
| 18:12.04 | tarabuka | worked |
| 18:12.09 | ManxPower-work | raden_work: Multiple asked you to put in that Noop. Are you going to follow their advice? |
| 18:12.37 | *** join/#asterisk nny (n=scott@64.203.239.83) |
| 18:12.40 | ManxPower-work | because all you are doing is wasting our time until we see that noop CLI output |
| 18:12.40 | verywiseman | i installed asterisk 1.4 ,but when i run asterisk -cvvv , there are some error logs which indicate asterisk can't load some modules like func_sprintf.so,res_config_ldap.so,.... , what is problem? |
| 18:13.06 | ManxPower-work | verywiseman: maybe you installed from a package rather than compiling from source? |
| 18:13.09 | bmoraca_work | raden_work, we're simply trying to debug your sitation right now. nothing else. we want to verify that callerid exists on the channel in the first place. when we verify that, we can help you figure out why it's getting lost. |
| 18:13.30 | nny | 1.6 issue here, I have an issue with queues and Local, I tried to preload the chan_local.so and queues show Local/EXT@CONTEXT as invalid. I have my pseudo channel running, any advice? |
| 18:13.58 | verywiseman | ManxPower-work, i installed it from source code |
| 18:14.00 | nny | module reload chan_local.so shows no such module |
| 18:14.28 | ManxPower-work | verywiseman: delete your /etc/asterisk/modules.conf and replace it with the modules.conf.sample (rename it of course) |
| 18:14.38 | nny | module show like chan_local.so shows 1 module loaded |
| 18:14.49 | nny | reading this bug report https://issues.asterisk.org/view.php?id=14179 |
| 18:15.04 | raden_work | bmoraca_work, ok its passing through |
| 18:15.19 | ManxPower-work | raden_work: SHOW us on a pastebin |
| 18:15.29 | raden_work | how do i show you ? |
| 18:15.44 | carrar | ~pastebin |
| 18:15.45 | infobot | [~pastebin] 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 |
| 18:15.47 | raden_work | what would u like me to pastebin |
| 18:15.49 | nny | ok fixed |
| 18:15.50 | ManxPower-work | show us the CLI output of the call that includes the Noop, copy it to pastebin |
| 18:15.51 | nny | gah |
| 18:16.07 | ManxPower-work | So we should see the noop, then the dial out where you are having clid problems |
| 18:16.25 | raden_work | ManxPower-work, no problem now |
| 18:16.42 | ManxPower-work | Wow! Adding the noop really fixed it! |
| 18:16.50 | raden_work | id just like to add a name to the forwarded CID so i know it redirected from work |
| 18:17.08 | [TK]D-Fender | RadenYou have our permission. Go for it |
| 18:17.21 | raden_work | why does everyone have to be this way |
| 18:17.25 | bmoraca_work | raden_work, no can do. CNAM lookups are handled by the called party. |
| 18:17.28 | raden_work | Noop did nothing cause i did not do it |
| 18:17.29 | ManxPower-work | raden_work: when the call hits the PSTN the terminating carrier will ignore the CLID name and replace it with whatever the LDIB has on file for that number |
| 18:17.32 | [TK]D-Fender | nny: "module reload app_queue.so |
| 18:17.43 | verywiseman | ManxPower-work, problem is still happening |
| 18:17.45 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:17.54 | nny | [TK]D-Fender: actually heh I had to add a reload => pbx_config.so on top of it |
| 18:18.04 | nny | in modules.conf, a restart and queues are correct now |
| 18:18.19 | raden_work | bmoraca_work, wonderful :( |
| 18:18.28 | *** join/#asterisk t_j (n=tj@tomjudge.vm.bytemark.co.uk) |
| 18:18.28 | ManxPower-work | raden_work: I wish you the BEST of luck. But I doubt you're going to get much help since you refuse the suggestions. |
| 18:18.33 | raden_work | bmoraca_work, anything you think i can do to know where the call originating from |
| 18:18.34 | t_j | any recomendations for a single port t1 mediagateway ? |
| 18:18.51 | raden_work | ManxPower-work, wtf you want me to put nooop ? |
| 18:19.00 | raden_work | it works wtf is the point ? |
| 18:19.01 | bmoraca_work | raden_work, what I usually do is set the callerid on forwarded calls to my DID...that way I know they're forwarded from work. |
| 18:19.27 | raden_work | bmoraca_work, we have high call volume and when we miss calls we dont know who it is :( |
| 18:20.01 | bmoraca_work | raden_work, unfortunately, there isn't really anything else you can do short of using a mobile SIP client on your cellphones |
| 18:20.15 | raden_work | great :( |
| 18:20.24 | n3hxs | SMS the call info to your pocket pager. |
| 18:20.32 | n3hxs | err, cell phone. |
| 18:20.54 | p3nguin | DTMF transfers might retain the original CIDnum. |
| 18:21.29 | raden_work | n3hxs, hmmm |
| 18:21.36 | p3nguin | especially if you use the Dial() 'o' option. |
| 18:21.43 | ManxPower-work | p3nguin: the "o" option to dial controls that |
| 18:21.59 | p3nguin | Glad we're on the same page. |
| 18:22.29 | path | ringall strategy would be it possible setting a ring group? |
| 18:22.36 | path | I want to avoid the MOH |
| 18:23.37 | ManxPower-work | path: you avoid the MoH by not specifying the "m" option to the Dial. |
| 18:23.50 | bmoraca_work | raden_work, sms isn't going to be reliable |
| 18:23.55 | ManxPower-work | path: Which GUI are you using? |
| 18:24.07 | raden_work | bmoraca_work, I know ;( |
| 18:24.08 | path | Im not using any GUI ManxPower-work |
| 18:24.21 | raden_work | just use the freaking number for now :( |
| 18:24.32 | path | I have this |
| 18:24.36 | ManxPower-work | path: then were are you getting "ringall strategy" from? |
| 18:24.46 | path | ManxPower-work: queue |
| 18:24.56 | ManxPower-work | path: but a queue is not a ring group |
| 18:24.57 | path | this is the context |
| 18:25.07 | path | er thats what Im asking |
| 18:25.09 | ManxPower-work | use the correct terms == get better answers. |
| 18:25.29 | ManxPower-work | I can help with Dial, but not Queues |
| 18:25.57 | path | I just need several extensions notified |
| 18:25.58 | [TK]D-Fender | path: I can help with both... if you get your head screwed on straight |
| 18:26.41 | benngard | ManxPower-work: maybe u think i am unpolite but we are some newbeginnerers here, like me and some others, we dont always know the correct word, i do beg u apardon |
| 18:26.59 | benngard | but we try to learn ) |
| 18:27.01 | path | exten = 10001,1,Answer |
| 18:27.01 | ManxPower-work | benngard: He made up words rather than telling us what he' |
| 18:27.02 | benngard | ;) |
| 18:27.05 | ManxPower-work | s actually trying to do. |
| 18:27.14 | ManxPower-work | ~pb |
| 18:27.14 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 18:27.17 | path | exten = 10001,2,Goto(queues,45000,1) |
| 18:27.22 | path | thats it |
| 18:27.51 | path | the thing is I want to avoid the MOH whenever someone calls to 10001 |
| 18:27.57 | benngard | sec |
| 18:28.11 | [TK]D-Fender | path: And have what happen instead? |
| 18:28.30 | [TK]D-Fender | path: and that is priority TWO |
| 18:28.57 | path | should I put a Ringing before? |
| 18:28.59 | [TK]D-Fender | path: please show a complete sample that I can trust at least a little |
| 18:29.04 | bmoraca_work | path, if all you need is to ring multiple extensions, do Dial(SIP/101&SIP/102&...&SIP/n) |
| 18:29.14 | [TK]D-Fender | path: You are not giving a clear 7 complete picture of what you want |
| 18:29.21 | bmoraca_work | that will play ringing by default |
| 18:29.25 | path | that seems easier bmoraca_work |
| 18:29.31 | ManxPower-work | path: try reading the output of "core show application queue" |
| 18:29.36 | outtolunc | the priority 1 was about 12 lines above the 2 |
| 18:29.52 | [TK]D-Fender | [13:29]<ManxPower-work>path: try reading the output of "core show application queue" <--------- |
| 18:29.57 | ManxPower-work | Pay special attention to the options listed between i option and the t option |
| 18:30.04 | benngard | path: i have i queue that i call: exten => 0317998989,4,Queue(0317998989,rt) |
| 18:30.25 | bmoraca_work | r option will do it, too |
| 18:30.34 | benngard | add "r" and u get ringback instead of moh |
| 18:30.50 | path | thanks bmoraca_work will try that too :) |
| 18:30.53 | ManxPower-work | benngard: just like the documentation says |
| 18:31.08 | benngard | i know, i am learning the hard way |
| 18:31.24 | ManxPower-work | benngard: you read the documention. By doing that you avoid many common issues. |
| 18:31.36 | path | oh hell |
| 18:31.45 | path | I know how rtfm works |
| 18:31.55 | [TK]D-Fender | path: You should try it some time :) |
| 18:32.13 | benngard | and i have like 300 users that complains if my phone and "hunt.group" ( i am an avaya guy) doesnt sounds as they are used 2 |
| 18:32.32 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 18:32.35 | path | it was just a simple question, everybody is free to answer |
| 18:32.38 | path | ;) |
| 18:32.43 | benngard | did it work? |
| 18:32.47 | path | yes |
| 18:32.52 | benngard | gz ;) |
| 18:32.58 | path | thanks again ;) |
| 18:33.02 | benngard | np |
| 18:33.11 | *** part/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 18:33.15 | ManxPower-work | I wish the stuff *I* am doing actually had some documentation for it. |
| 18:33.24 | keith4 | is away: I'm busy |
| 18:33.40 | path | certainly looking at source code is the best documentation |
| 18:33.43 | carrar | ManxPower, thats part of being a CREATOR! |
| 18:33.47 | nny | well, the part about providing the answer for one Dial option is that in the process the person asking misses out on the other two dozen options for later :) |
| 18:34.06 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:34.07 | ManxPower-work | nny: Exactly |
| 18:34.38 | bmoraca_work | that's one of the great things aobut extensible software...there are many ways of accomplishing one goal |
| 18:34.49 | *** join/#asterisk Alagar (n=Administ@122.164.35.52) |
| 18:35.22 | ManxPower-work | bmoraca_work: It seems like the most common way of accomplishing the goal is come here and ask people to read the docs for you. |
| 18:36.02 | bmoraca_work | well, there is that...however, i've learned a lot myself by researching things people have come in here asking...so it cuts both ways |
| 18:36.12 | raden_work | how can I set my outbound callerid per extension ? |
| 18:36.36 | bmoraca_work | raden_work, sameway you're doing the forwarding...use an ASTDB key |
| 18:36.42 | voipmonk | TK mentioned the answer to this earlier using "setvar" in sip.conf |
| 18:36.49 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
| 18:36.52 | bmoraca_work | setvar would do it as well |
| 18:37.00 | ManxPower-work | It's pretty easy to set the callerid per DEVICE, but not per extension |
| 18:37.18 | ManxPower-work | setvar won't work for setting it per extension |
| 18:37.23 | raden_work | bmoraca_work, that the only way ? |
| 18:37.55 | raden_work | well per phone/device would work just fine |
| 18:38.06 | bmoraca_work | ManxPower-work, he's equating a device to an extension in this case...he wants to control the outgoing CID to the PSTN per device, in addition to the internal CID for that device |
| 18:38.07 | ManxPower-work | raden_work: set it in the sip.conf peer. |
| 18:38.08 | voipmonk | http://pastebin.com/m180714d2 |
| 18:38.33 | voipmonk | should give you some ideas |
| 18:38.37 | ManxPower-work | bmoraca_work: he could have said that. |
| 18:38.49 | bmoraca_work | ManxPower-work, i inferred it from his earlier questions |
| 18:39.01 | bmoraca_work | he could have been more precise, though :) |
| 18:39.04 | [TK]D-Fender | raden_work: Set(CALLERID(num0=............... |
| 18:39.23 | raden_work | [TK]D-Fender, yeah got that part want to be able to set per extension :) |
| 18:39.28 | raden_work | voipmonk, thanks :) |
| 18:39.36 | *** join/#asterisk TimeRider (n=steve@78.32.26.1) |
| 18:39.39 | *** part/#asterisk t_j (n=tj@tomjudge.vm.bytemark.co.uk) |
| 18:39.40 | ManxPower-work | raden_work: stop calling a device an extension. They are not the same thing |
| 18:39.41 | raden_work | bmoraca_work, yes i could have sorry |
| 18:39.43 | bmoraca_work | raden_work, use SETVAR in sip.conf is the easiest way...ASTDB might be a bit more extensible |
| 18:39.46 | [TK]D-Fender | raden_work: Go read the sip.conf sample config for a bit |
| 18:40.02 | bmoraca_work | depends on hwo you like to maintain your system |
| 18:40.04 | [TK]D-Fender | rade_or wait -.5 seconds for bmoraca_work to simply HAND it to you |
| 18:40.16 | ManxPower-work | Calling a device extension just makes you look like a n00b |
| 18:40.16 | *** join/#asterisk ks3 (n=ks3@74.203.195.1) |
| 18:40.23 | raden_work | [TK]D-Fender, why u have to be this way ??? |
| 18:40.37 | [TK]D-Fender | raden_work: Be what way? |
| 18:40.57 | raden_work | ManxPower-work, simple as this I dont care if i can set it per device or per extension I just want to be able to set it so it dont matter to me I got a solution |
| 18:41.12 | ManxPower-work | [TK]D-Fender: expecting the user to do work to solve their problem. how rude! 8-| |
| 18:41.13 | raden_work | [TK]D-Fender, cynical |
| 18:41.15 | bmoraca_work | ManxPower-work, we had that discussion last week...to most of the telcom world and users, a phone = extension...so that's the terminology that gets tossed around. as far as I know, asterisk is the only tech that treats its dialplan functions as extensions |
| 18:41.31 | [TK]D-Fender | raden_work: that was a jab at HIM, not you. |
| 18:41.55 | [TK]D-Fender | bmoraca_work: Not in a terribly bad way though ;) |
| 18:41.58 | benngard | raden_work: u want to be able to change calleer id per extension when u call ot? |
| 18:42.10 | bmoraca_work | [TK]D-Fender, i gave him the method...i didn't tell him how to do it :) |
| 18:42.11 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:42.14 | benngard | out* |
| 18:42.15 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:42.18 | ManxPower-work | bmoraca_work: But thinking of them of the same creates a BASIC misunderstanding about Asterisk |
| 18:42.22 | raden_work | Simple thing is each phone here only has one extension per device so being able to set it per device or per extension either would have solved my problem thats why i was not specific just wanted a solution |
| 18:42.33 | raden_work | wtf are the sample configs stored :( |
| 18:42.44 | Corydon76-lap | bmoraca_work: and most of the telecom world's users curse the system as being completely unusable, difficult to learn, unmaintainable... the list goes on |
| 18:42.47 | bmoraca_work | ManxPower-work, i'm not disagreeing with you...i'm just saying that i understand where the confusion comes from |
| 18:42.50 | ManxPower-work | raden_work: /path/to/src/asterisk/configs |
| 18:42.59 | carrar | easy to set/choose a caller id when you dial out |
| 18:43.08 | carrar | easy as PIE |
| 18:43.20 | Corydon76-lap | Sometimes you need to change the paradigm to make things easier for administrators |
| 18:43.28 | ManxPower-work | raden_work: an extension is something you dial (it is a DESTINATION). A device is something that does dialing (it is a SOURCE) they are the exact opposite from a dialplan standpoint. |
| 18:43.45 | raden_work | ok everyone settle down :( |
| 18:43.46 | ks3 | Do hashes work like standard variables? Eg.. If I use Set(HASH(__myhash)=..., is the hash inherited by spawned channels? |
| 18:44.22 | benngard | raden_work: i quick question, when u dial out is it the callee number u wanna change per extension? |
| 18:44.30 | ManxPower-work | I always used a setvar=DID=2124441212 and setvar=BTN=2125550000 and let the macro that did outbound dialing deal with it. |
| 18:44.41 | Corydon76-lap | ks3: no, hashes are not inherited |
| 18:44.43 | bmoraca_work | raden_work, we've given you two options: 1) use some dialplan code and astdb, or 2) use setvar in sip.conf...either way will accomplish what you might need to do |
| 18:45.04 | ks3 | Corydon76-lap: I was afraid of that... thanks |
| 18:45.07 | Corydon76-lap | ks3: it's a neat idea, but that's not the way it works currently |
| 18:45.33 | ks3 | Corydon76-lap: We're making heavy use of FUNC_ODBC and hashes to customize the dialplan on the fly |
| 18:46.30 | bmoraca_work | func_odbc is a lot of fun |
| 18:46.37 | ManxPower-work | Corydon76-lap: Currently SIPAddHeader adds a header to a future INVITE of a SIP device, right? Any way to add headers to other types of SIP packets? |
| 18:46.48 | leifmadsen | func_odbc was a huge way forward in distributed asterisk systems |
| 18:46.54 | Corydon76-lap | ManxPower-work: not that I know of |
| 18:47.05 | ManxPower-work | Corydon76-lap: I didn't think so. Thanks. |
| 18:47.37 | Corydon76-lap | Now that I think of it, HASH() needs to be LOCAL()-compatible. |
| 18:47.43 | Corydon76-lap | leifmadsen: add that to the list of wants |
| 18:47.52 | ManxPower-work | (the effect of this, BTW, is that Asterisk can't "support" the SIP Warning: or Reason: headers. |
| 18:47.53 | leifmadsen | :) |
| 18:48.31 | bmoraca_work | ManxPower-work, you could use AGI or System() to manually send them to the peers...but as far as reading them...dunno |
| 18:48.36 | Corydon76-lap | ManxPower-work: just in the current version. Come up with a criteria for the reason for doing it, and it'll get coded |
| 18:48.47 | ManxPower-work | bmoraca_work: I don't really need to read them, just set them. |
| 18:49.19 | ManxPower-work | bmoraca_work: I was thinking of System() + sipsak |
| 18:49.45 | ManxPower-work | bmoraca_work: I'll have to actually, really, fully, understand SIP to have any chance of success. |
| 18:49.53 | bmoraca_work | right |
| 18:50.08 | bmoraca_work | are you interfacing with another PBX or something? |
| 18:50.16 | ManxPower-work | I wonder if there's enough info in the channel variables to fake a response to the phone using sipsak. |
| 18:51.14 | ManxPower-work | bmoraca_work: trying to use the SIP Warning: header. Polycoms support displaying a popup on the display when it receives a Warning: header. For messages like "Account Balance Low" or "Call Forwarding Active" type of feedback messages to the user. |
| 18:51.34 | bmoraca_work | oh nice |
| 18:51.44 | bmoraca_work | can't you do the same by pushing XML to them? |
| 18:52.00 | ManxPower-work | The Polycom docs don't say this, but from my reading of the various SIP RFCs to seems like the phones only process that header when it's part of an invite. |
| 18:52.24 | bmoraca_work | interesting...i'd think such a feature would be more likely to be part of a NOTIFY |
| 18:52.37 | ManxPower-work | bmoraca_work: my alternative is using the microbrowser, but not all polycoms have a microbrowser. |
| 18:52.45 | bmoraca_work | true |
| 18:53.06 | redwizard | shit |
| 18:53.16 | redwizard | smallest multi touch screen this company does is 32" |
| 18:53.25 | redwizard | and its 986 quidz |
| 18:53.46 | ManxPower-work | I tried it using notify, doesn't seem to work. All the diagrams I see only show it as part of invite or bye or cancel, never notify. Maybe I'm just not using it right. 8-| |
| 18:54.41 | bmoraca_work | i can see it under all three of those circumstances, but I'd think NOTIFY would be just as useful...can't see a good reason for polycom to not support that one |
| 18:56.38 | *** join/#asterisk toppe_ (n=root@cs181001225.pp.htv.fi) |
| 18:59.29 | ManxPower-work | bmoraca_work: I'm trying to stop the "my phone doesn't work" only to find out the idiot enabled forwarding on this phone instead of on Asterisk. |
| 18:59.36 | ManxPower-work | types of problems |
| 18:59.48 | bmoraca_work | oh i'm all too familiar with those |
| 19:00.04 | bmoraca_work | if you come up with a way to fix stupid, let me know :) |
| 19:00.17 | bmoraca_work | although, with Cisco 7940s, I can disable local forwarding on the phones (thank god) |
| 19:01.04 | ManxPower-work | bmoraca_work: I disabled it on the local polycom. Seems I can still go into the menus, enable forwarding, then the phone throws it all and doesn't forward. |
| 19:01.50 | benngard | the worlds #1 pirate just got duplicated http://photos-f.ak.fbcdn.net/hphotos-ak-snc3/hs239.snc3/22649_280595509918_587744918_4449968_2854409_n.jpg (if u dont know how he is, ask me) |
| 19:01.52 | bmoraca_work | sounds like a bug...have you looked at newer versions of the SIP firmware? |
| 19:03.28 | *** join/#asterisk funtoo_nbu (i=seb@209.237.247.182) |
| 19:03.49 | funtoo_nbu | Anyone know the difference between polycom 560 and 550 phones? im about to buy 5 of them... |
| 19:04.03 | bmoraca_work | 560 has gigabit ethernet...otherwise, nothing |
| 19:04.17 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 19:04.38 | funtoo_nbu | oh snap i dont even need that :D |
| 19:04.48 | ManxPower-work | bmoraca_work: running 3.2.2 |
| 19:05.04 | bmoraca_work | i believe there is a newer version |
| 19:05.12 | ManxPower-work | bmoraca_work: I wanted to try to figure out "server based" forwarding. |
| 19:05.15 | ManxPower-work | bmoraca_work: there isn't. |
| 19:07.11 | bmoraca_work | hrm |
| 19:12.05 | [TK]D-Fender | funtoo_nbu: Why ar you aiming for the 550 at all? |
| 19:12.21 | HenrikJott | [TK]D-fender: sorry, i asked a question and just left. A situation arised here =) the files are getting there by me moving them there (mv [file] /var/spool/asterisk/outgoing/[file]). |
| 19:14.59 | bmoraca_work | HenrikJott, i'd imagine it's a permissions issue. make sure the user that asterisk is running under has access to those files. |
| 19:16.48 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:17.27 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:17.59 | redwizard | hmm anyone know the best way to get the mysql db for asterisk up and running after it failed during a kubuntu apt-get, everything else seems to be working but without the database i cant put freepbx on there |
| 19:18.42 | bmoraca_work | redwizard, you'll need to debug why mysql didn't install...we can't really help you with that |
| 19:18.54 | bmoraca_work | nor can we help you set freepbx up |
| 19:19.03 | carrar | redwizard, compile asterisk from scratch and use PostgreSQL |
| 19:19.29 | carrar | Thats your best option |
| 19:19.38 | carrar | bestest |
| 19:19.44 | carrar | bestorific |
| 19:20.05 | Kobaz | postgres ftw |
| 19:20.22 | redwizard | lol, bit beyond my expertise i'm afraid, i'm not in my comfort zone with linux yet |
| 19:20.52 | redwizard | just trying to find the settings i need to apply to mysql |
| 19:20.58 | redwizard | nevermind i'll find them somewhere |
| 19:21.10 | carrar | Might try #freepbx |
| 19:21.50 | [TK]D-Fender | redwizard: Indeed go follow their guides... their needs are specialized and documented. |
| 19:22.42 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 19:24.19 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 19:24.30 | *** join/#asterisk sun28 (n=light@78.108.73.46) |
| 19:24.38 | bmoraca_work | dreamweaver blew up on me :( QQ |
| 19:25.02 | *** join/#asterisk simplydrew_ (n=simplydr@pool-74-97-177-245.prvdri.fios.verizon.net) |
| 19:25.07 | funtoo_nbu | [TK]D-Fender: im going to get 3 ip 450s for general office use and 2 550s for receptionists |
| 19:25.20 | *** join/#asterisk Deeewayne (n=dwayne@adsl-070-148-053-233.sip.bhm.bellsouth.net) |
| 19:25.20 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:25.57 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 19:26.22 | [TK]D-Fender | funtoo_nbu: Do NOT get 550's for receptionists... 650 instead... |
| 19:26.42 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 19:26.47 | [TK]D-Fender | funtoo_nbu: 550 adds 1 line key and has no expansion. funtoo_nbu 650 adds 3 line keys, and a LOT of expansion |
| 19:27.48 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 19:29.34 | [TK]D-Fender | IP 550 = waste |
| 19:32.24 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 19:32.25 | ManxPower-work | hugs his 550 |
| 19:32.47 | hardwire | hugs a $50 |
| 19:33.01 | nny | hugs a 40, cries |
| 19:33.16 | hardwire | I'll give you two 20 for the 40 |
| 19:33.25 | nny | it's Schiltz |
| 19:33.28 | nny | but sold! |
| 19:33.30 | hardwire | mmaaaalt licker! |
| 19:33.42 | hardwire | I don't understand 40s. |
| 19:34.01 | nny | then you surely won't understand http://en.wikipedia.org/wiki/Edward_Fortyhands |
| 19:34.02 | hardwire | Whenever I go to the gas station some punk is always buying one.. of the worst beer imaginable. |
| 19:34.49 | hardwire | I guess I just don't understand if it's a fad.. because it costs less per volume to get cans of that crap. |
| 19:35.13 | nny | yeah probably more the former, I just stick to my pints |
| 19:35.19 | hardwire | idneed |
| 19:35.21 | hardwire | indeed |
| 19:35.32 | hardwire | it's monday morning and I'm already into beer.. |
| 19:35.36 | nny | heh |
| 19:35.37 | hardwire | somebody save me. |
| 19:35.50 | nny | shot? |
| 19:35.50 | hardwire | just send the reaper or a 6 pack of cider.. |
| 19:36.44 | hardwire | blech.. no shots. |
| 19:37.09 | funtoo_nbu | 550 has 4 lines? |
| 19:37.24 | funtoo_nbu | [TK]D-Fender: not sure what you mean |
| 19:37.47 | funtoo_nbu | 3 line keys? and what kind of expansion would I need? |
| 19:37.52 | [TK]D-Fender | funtoo_nbu: 4 line keys |
| 19:37.59 | [TK]D-Fender | funtoo_nbu: attendent consoles |
| 19:38.48 | funtoo_nbu | the 550 has 4 line key |
| 19:38.57 | sevv | this edward fortyhands |
| 19:38.58 | funtoo_nbu | and i wouldnt want that expansion |
| 19:39.02 | sevv | this looks awesome |
| 19:39.08 | funtoo_nbu | its already overkill |
| 19:42.21 | leifmadsen | Kobaz: stop filing issues -- your ratio is much too high |
| 19:42.42 | [TK]D-Fender | funtoo_nbu: whats the point of the 550 at all? |
| 19:43.07 | funtoo_nbu | conferencing i think |
| 19:43.26 | funtoo_nbu | i was going to just get all 450s but one of my buds told me to get 2 550s for receptionist |
| 19:44.40 | [TK]D-Fender | funtoo_nbu: they all conference..... |
| 19:44.41 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 19:45.29 | funtoo_nbu | hmm then really i see no difference between 450 and 550 :D |
| 19:45.33 | funtoo_nbu | except the line buttons |
| 19:45.47 | funtoo_nbu | and we will only have 3 lines anyway |
| 19:46.02 | ManxPower-work | funtoo_nbu: The lines on the phones have nothing to do with phone line. |
| 19:46.03 | ManxPower-work | s |
| 19:46.17 | funtoo_nbu | yea im trying to wrap my head around that |
| 19:46.45 | funtoo_nbu | the buttons on the phone will be used as separate lines here tho |
| 19:46.55 | ManxPower-work | funtoo_nbu: a Key System will have a line on the phone correspond with a physical phone line. |
| 19:47.13 | ManxPower-work | A PBX does not have a relationship between phone lines and lines on the phone. |
| 19:47.28 | ManxPower-work | your lines on the phone are "lines" Phone <-> Asterisk. |
| 19:47.48 | funtoo_nbu | we are ugprading from an old pbx to asterisk with a sip provider and polycom phones |
| 19:47.57 | funtoo_nbu | i got asterisk and 1 phone working atm |
| 19:47.59 | ManxPower-work | you dial "9" (or whatever) and Asterisk picks the phone line based on your dialplan, not based on physical lines. |
| 19:48.15 | funtoo_nbu | ye there is a key system atm |
| 19:48.29 | [TK]D-Fender | funtoo_nbu: these ideas got right out the door as of now. |
| 19:48.32 | [TK]D-Fender | go* |
| 19:48.55 | funtoo_nbu | i dig |
| 19:49.04 | funtoo_nbu | i got a lot of learning, pretty much jumped into the deep end |
| 19:49.15 | funtoo_nbu | but the local phone tolls kinda forced our hand to use voip |
| 19:50.32 | funtoo_nbu | so really now i just wana get only 450s |
| 19:50.41 | *** join/#asterisk sw_ (n=sw@unaffiliated/sw) |
| 19:50.59 | *** join/#asterisk shapr (n=shapr@c-76-29-246-122.hsd1.al.comcast.net) |
| 19:52.58 | shapr | I have a sort of topical question... is there a recommended headset/soundcard combo for using asterisk to make international calls from my desktop? |
| 19:52.59 | [TK]D-Fender | funtoo_nbu: normal users shouldn't need more than 335's or 321's |
| 19:53.18 | [TK]D-Fender | shapr: No. Softphones are not recommended PERIOD |
| 19:53.23 | sw_ | Hi, I'm trying to use the milliwatt app under asterisk 1.4 (debian lenny) but it complains about application playtones missing |
| 19:53.33 | [TK]D-Fender | shapr: Go buy a real phone |
| 19:53.36 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 19:53.50 | ManxPower-work | sw_: Did you install Asterisk from a package or a GUI? |
| 19:53.53 | [TK]D-Fender | shapr: Even an ATA + 10$ phone bought at a drugstore is a better idea |
| 19:54.00 | shapr | [TK]D-Fender: ATA? |
| 19:54.06 | [TK]D-Fender | ~ata |
| 19:54.07 | infobot | it has been said that ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 19:54.28 | sw_ | ManxPower-work: installed from debian repositories, but i've rebuilt the sources from the 1.4 branch and app_playtones.so is not built either |
| 19:54.47 | shapr | Ah, thanks. Is there perhaps a tutorial/guide for this sort of thing? That is, how to setup a home linux box as a 'real' phone line? |
| 19:55.11 | [TK]D-Fender | shapr: how is a "linux box" a "line"? |
| 19:55.13 | *** join/#asterisk mnicholson_ (n=mnichols@nat/digium/x-hrpaklvfolyzxyej) |
| 19:55.18 | [TK]D-Fender | shapr: youa ren't making much sense |
| 19:55.20 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 19:55.25 | ManxPower-work | sw_: you must have something seriously screwed up. Did you modify your /etc/asterisk/modules.conf? |
| 19:55.26 | funtoo_nbu | 450 is only 40$ more than 335 :D |
| 19:55.30 | *** join/#asterisk mnicholson (n=mnichols@nat/digium/x-mprgcblbpsbwqnpx) |
| 19:55.42 | [TK]D-Fender | funtoo_nbu: do the math, make a call. |
| 19:55.44 | *** join/#asterisk DMeloUK (n=Administ@64.129.95.226) |
| 19:55.48 | funtoo_nbu | yea :D thx |
| 19:55.58 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 19:56.05 | funtoo_nbu | do you know about headsets? one that works good with polycom phones? |
| 19:56.12 | [TK]D-Fender | funtoo_nbu: At least you'll have a chance to make it an informed one |
| 19:56.24 | sw_ | ManxPower-work: app_milliwatt.so is loaded in modules.conf... but when i'm calling the app it gives "app_milliwatt.c:160 milliwatt_exec: The Playtones application is required to run Milliwatt()" |
| 19:56.26 | [TK]D-Fender | funtoo_nbu: Plantronics M22 amp + matching headset |
| 19:56.56 | [TK]D-Fender | sw_: got an indications.conf file in your config folder? |
| 19:56.58 | ManxPower-work | sw_: Is that a "yes I modified /etc/asterisk/modules.conf" or is that a "no, I did not modify /etc/asterisk/modules.conf |
| 19:57.00 | shapr | [TK]D-Fender: I think I should read more about ATAs. |
| 19:57.24 | funtoo_nbu | phone's amp is not sufficient? |
| 19:57.31 | [TK]D-Fender | shapr: plug in normal phone, plung in ethernet. use with *. the End |
| 19:57.51 | ManxPower-work | waits for sw_'s answer |
| 19:57.55 | [TK]D-Fender | funtoo_nbu: not for call center use |
| 19:57.59 | sw_ | [TK]D-Fender: yes sir, i have indications.conf |
| 19:58.10 | sw_ | ManxPower-work: yes i did modify modules.conf to load app_milliwatt.so |
| 19:58.16 | [TK]D-Fender | sw_: pastebin your modules.conf |
| 19:58.18 | [TK]D-Fender | ~pb |
| 19:58.18 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 19:58.21 | ManxPower-work | sw_: it should load automatically. |
| 19:58.33 | sw_ | ManxPower-work: it's not the issue.. |
| 19:58.36 | [TK]D-Fender | sw_: laod it manually and test |
| 19:58.51 | funtoo_nbu | wow headsets are as expensive as the phone! |
| 19:58.52 | sw_ | ManxPower-work: the app is loaded fine.. it just requires playtones, which is not present |
| 19:59.10 | shapr | [TK]D-Fender: I have previously had asterisk installed on my desktop linux box, is a dedicated box recommended for a 'home land line' replacement? |
| 19:59.16 | sw_ | ManxPower-work: [TK]D-Fender : app_milliwatt.c:160 milliwatt_exec: The Playtones application is required to run Milliwatt() |
| 20:00.10 | [TK]D-Fender | shapr: What is a "home landline replacement"? * is not a TELEPHONE SEVICE. it is a PBX toolkit. |
| 20:00.33 | ManxPower-work | sw_: maybe because there is no app_playtones.so. It is part of something else. Sounds to me like you have a serious version mismatch in Asterisk |
| 20:00.35 | [TK]D-Fender | sw_: show us your attempt to load that other app |
| 20:01.32 | sw_ | ManxPower-work: [TK]D-Fender : exactly, there's no app_playtones.so in debian packages.. this is why I've try to rebuild asterisk 1.4 on the side.. and it's not there either |
| 20:02.17 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 20:02.22 | [TK]D-Fender | sw_: menuconfig will tell yuo why |
| 20:02.42 | ManxPower-work | sw_: That might because res_indications.so is what provides the PlayTones features. |
| 20:02.53 | ManxPower-work | there is no app_playtones |
| 20:03.49 | ManxPower-work | sw_: I can't imagine why res_indications.so isn't loading, since you didn't make ANY changes to your /etc/asterisk/modules.conf except to add the load => app_milliwatt.so, right? |
| 20:04.02 | seanbright | because indications.conf is missing? |
| 20:04.14 | ManxPower-work | seanbright: he already confirmed it exists. |
| 20:04.18 | seanbright | he's a LIAR |
| 20:04.19 | seanbright | heh |
| 20:04.24 | sw_ | ManxPower-work: thanks, I was missing res_indications.so... I upgraded my minimal asterisk from 1.2 to 1.4 this week-end |
| 20:04.35 | seanbright | little monday morning quarterbacking for ya |
| 20:04.43 | sw_ | ManxPower-work: [TK]D-Fender : i got confused by the fact there was an app_playtones.so in 1.5 |
| 20:04.45 | sw_ | 1.6 |
| 20:04.48 | ManxPower-work | sw_: why were you missing res_inidcations.so |
| 20:05.02 | ManxPower-work | or more importantly why did you mess with modules.conf |
| 20:05.22 | sw_ | ManxPower-work: because my previous asterisk was running on an embedded device with 32 MB of RAM |
| 20:06.18 | *** join/#asterisk mprime (i=c74c90e5@gateway/web/freenode/x-qcpanftrwlwtcpeq) |
| 20:06.18 | ManxPower-work | sw_: In the future answer the questions that are asked. I was trying to figure out if something was wrong with your modules.conf and you just stonewalled me. |
| 20:06.45 | sw_ | ManxPower-work: sorry for the misunderstanding, english is not my mother language |
| 20:06.50 | ManxPower-work | If you had admitted you modified that file then you would have found your answer faster. |
| 20:08.17 | Kobaz | yeah, quit breaking stuff |
| 20:11.17 | *** part/#asterisk shapr (n=shapr@c-76-29-246-122.hsd1.al.comcast.net) |
| 20:12.17 | *** part/#asterisk mprime (i=c74c90e5@gateway/web/freenode/x-qcpanftrwlwtcpeq) |
| 20:13.19 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 20:15.13 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 20:15.58 | bmoraca_work | breaking stuff is fun |
| 20:16.04 | bmoraca_work | when you don't have to answer the support calls |
| 20:17.11 | DMeloUK | what's the best way to link up a remote site to asterisk using a linux router there with the asterisk box on the public ip at the colo? I want to try and avoid sip nat issues and trying to figure out if/what kind of vpn the linux router needs for best quality or if there is a better solution? |
| 20:17.34 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:17.38 | voipmonk | vpn, avoid sip nat, public ip colo and a question mark |
| 20:17.40 | voipmonk | love it |
| 20:18.19 | titter | lol |
| 20:19.27 | [TK]D-Fender | NAT = largely irrelevant. QoS does not exist over the internet. Just configure things right |
| 20:19.28 | [TK]D-Fender | ~sipnat |
| 20:19.29 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:20.05 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 20:20.24 | DMeloUK | voipmonk, I am afraid I didn't get the 2nd part :) |
| 20:20.31 | DMeloUK | thanks for the link fender |
| 20:20.34 | funtoo_nbu | can you do intercom stuff with asterisk? |
| 20:21.13 | bmoraca_work | funtoo_nbu, yes, as long as your phone has a proper autoanswer implementation (Alert-Info) |
| 20:21.15 | *** join/#asterisk Baylink (n=jra@static-173-65-4-24.tampfl.fios.verizon.net) |
| 20:21.34 | bmoraca_work | funtoo_nbu, of course...what people mean by "intercom" varies from person to person |
| 20:21.54 | funtoo_nbu | i have not purchased them yet, but im on the brink of ordering polycoms 450s |
| 20:22.05 | bmoraca_work | polycoms work fine |
| 20:22.24 | funtoo_nbu | cool man asterisk can do anything |
| 20:24.11 | ManxPower-work | Unless, of course, you are trying to do something Asterisk can't do. 8-) |
| 20:24.13 | bmoraca_work | i wouldn't go that far...but it can do a lot |
| 20:24.23 | Kobaz | ManxPower-work: like shared lines, and bridged lines |
| 20:24.38 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 20:24.40 | ManxPower-work | Kobaz: or Custom SIP headers to the caller rather than the callee |
| 20:24.40 | *** join/#asterisk smooth_penguin (n=smoove@59.95.0.181) |
| 20:24.52 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.166.173.dsl.dyn.forthnet.gr) |
| 20:25.11 | Kobaz | hmmm |
| 20:25.16 | Kobaz | i've never needed that |
| 20:26.22 | bmoraca_work | access to custom q931 messages |
| 20:26.55 | Kobaz | hmm |
| 20:26.57 | Kobaz | yeah |
| 20:27.00 | Kobaz | never needed that either |
| 20:27.06 | Kobaz | i could imagine it's useful |
| 20:34.40 | *** join/#asterisk TimeRider (n=steve@78.32.26.1) |
| 20:41.38 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:41.39 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:56.37 | *** join/#asterisk Hemos\ (n=cyberspa@82.193.31.157) |
| 20:58.29 | *** join/#asterisk RobH (n=robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 20:58.57 | p3nguin | When someone says they have a data T1, are they talking about CAS, PRI, or something completely different? |
| 21:00.17 | p3nguin | naikrovek: You had mentioned one day: data T1 + G729 + IAX2 trunk = (about) 140 simultaneous calls |
| 21:00.53 | *** join/#asterisk Alagar (n=Administ@122.164.35.52) |
| 21:00.57 | Amorsen | p3nguin: Your guess is as good as anyones... |
| 21:01.19 | p3nguin | I'm trying to figure out how calling capacity varies between CAS and PRI. |
| 21:01.34 | *** join/#asterisk xpot-mobile (n=xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 21:01.55 | *** join/#asterisk tzafrir (n=tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 21:02.18 | Amorsen | Telephony people always find a million ways to solve the same problem, and you just have to guess what the other end picked |
| 21:03.16 | Corydon76-lap | p3nguin: usually something completely different |
| 21:03.42 | Corydon76-lap | data T1 generally means HDLC-data, all channels bonded into a single pipe |
| 21:03.53 | Amorsen | Anyway, CAS avoids the use of a dedicated channel for signalling, which gives you one more channel for voice per T1 |
| 21:04.06 | Corydon76-lap | See nethdlc |
| 21:04.17 | p3nguin | And you can cram more voip over a data T1 than a voice T1, right? |
| 21:04.38 | Corydon76-lap | However, there's about 4 different ways that you can encode data onto nethdlc, and you need to get the right one |
| 21:04.50 | Amorsen | p3nguin: You can't really cram any VoIP at all over a voice T1... |
| 21:05.14 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 21:05.22 | Corydon76-lap | p3nguin: only if you use better compression than ulaw/alaw |
| 21:05.51 | Hemos\ | Hi |
| 21:05.57 | Hemos\ | 192.168.0.9 has generated a flow coming from door 16448 and addressed to 80.80.80.80: 17658 |
| 21:06.02 | Hemos\ | the origin door has been nat with 70.70.70.70: 34413 |
| 21:06.02 | Corydon76-lap | Amorsen: sure you can... analog modem... |
| 21:06.08 | Hemos\ | return traffic turns out 80.80.80.80: 17659 - > 70.70.70.70: 34414 |
| 21:06.12 | Hemos\ | 34413 + 1 = 34414 |
| 21:06.15 | Amorsen | Heh, Corydon76-dig |
| 21:06.16 | Hemos\ | 17658 + 1 = 17659 |
| 21:06.23 | Hemos\ | why port +1? |
| 21:06.59 | Corydon76-lap | Hemos\: ask your provider |
| 21:07.20 | Amorsen | Nothing better than VoIP over modem... Except perhaps VoIP over GPRS |
| 21:07.38 | Corydon76-lap | Amorsen: no, there's one better. Fax over voip |
| 21:07.41 | *** join/#asterisk BadHorsie (n=nile@201.198.239.167) |
| 21:07.54 | Corydon76-lap | without T.38 |
| 21:07.57 | Amorsen | Don't remind me |
| 21:08.14 | Amorsen | With or without T.38, it's better just to repress the memories |
| 21:08.24 | Hemos\ | Corydon76-lap, I am the provider |
| 21:08.47 | Corydon76-lap | Hemos\: then I feel sorry for your users |
| 21:08.48 | BadHorsie | I've noticed how asteriskcdrdb has only the information on the calls already finished, is there a way to have asterisk insert the entry when the call is started rather than when it's finished? |
| 21:09.18 | Corydon76-lap | BadHorsie: Have your calls last less than a second. Then end==start |
| 21:09.23 | Amorsen | BadHorsie: Perhaps Call Event Logging will do what you want, it's a VERY new feature |
| 21:09.42 | Amorsen | I'm not sure whether it's even in a release yet, but possibly in 1.6.2.x |
| 21:10.12 | Corydon76-lap | Nope, CEL is 1.8 only |
| 21:10.50 | Hemos\ | nobody knows to say to me because rtp port +1 return ? |
| 21:11.10 | *** join/#asterisk Geminizer (n=whoami@cpe-76-180-27-4.buffalo.res.rr.com) |
| 21:11.33 | Corydon76-lap | Hemos\: generally, RTP is allocated in qty of 2, one for incoming, the other for outgoing |
| 21:12.41 | Amorsen | Sometimes you're lucky and the ports match up for the incoming and the outgoing RTP... |
| 21:13.01 | Hemos\ | Corydon76-lap, therefore it is a normal thing? |
| 21:13.23 | Corydon76-lap | Yep |
| 21:13.48 | *** join/#asterisk sp3 (n=sp3@75-144-110-147-Indianapolis.hfc.comcastbusiness.net) |
| 21:14.38 | nny | so 0 allows you to "reach an operator" during vm |
| 21:14.53 | nny | and I can control what "0" does via my context for that user |
| 21:15.04 | nny | any easy way to allow another press for, say cellphones? |
| 21:15.12 | nny | I assume features.conf |
| 21:15.20 | Hemos\ | Corydon76-lap, on the firewall I receive "event=unhandled_local action=drop rule=LocalUndelivered recvif=wan" because the port is various |
| 21:15.36 | nny | and setting the variable in the context specifically before it sends the caller to VM? |
| 21:15.47 | Amorsen | nny: Most of voicemail has traditionally been hard coded in the C file |
| 21:15.57 | Geminizer | Hello all. Question -- I have a dialplan that involves an Answer(), Wait(4), and three audio files used with a Playback()... when I pick up my phone to test out the dialplan, I hear the ending of the last audio file in the sequence... how can I tell the dialplan to wait until I have picked up the phone before doing anything? |
| 21:16.08 | nny | Amorsen: does that application ignore any key presses during? Actually I can test that |
| 21:16.11 | *** join/#asterisk dennis00 (n=user@unaffiliated/dennis00) |
| 21:16.24 | nny | Amorsen: not a huge deal, can make it one or the other for now with zero, just thinking down the road |
| 21:16.24 | Amorsen | nny: You can try the new minivm stuff, but I don't know if it solves that particular problem. Haven't tried it. |
| 21:16.35 | nny | Amorsen: k thanks |
| 21:16.55 | dennis00 | Voicechanger is sooo cool! |
| 21:17.24 | nny | Amorsen: first glance I thin it ignores anything in features.conf while the Voicemail app is running |
| 21:18.03 | *** join/#asterisk ramindia (n=balajibh@96-10.southernonline.net) |
| 21:18.40 | *** part/#asterisk BadHorsie (n=nile@201.198.239.167) |
| 21:19.16 | *** join/#asterisk ramindia (n=balajibh@96-10.southernonline.net) |
| 21:19.18 | Geminizer | slaps [TK]D-Fender around a bit with a large trout |
| 21:19.41 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 21:20.23 | *** join/#asterisk ramindia (n=balajibh@96-10.southernonline.net) |
| 21:20.54 | dennis00 | Is there a way to change your voice to robotic with Asterisk? |
| 21:21.43 | dennis00 | I also have a serious question. It seems that registering to my server takes like 5 seconds, instead of the 0.3 seconds with voipbuster. Is there an obvious explanation? |
| 21:21.55 | [TK]D-Fender | slaps a large trout around with a Geminizer |
| 21:22.05 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 21:22.35 | Geminizer | nice |
| 21:22.35 | *** join/#asterisk ramindia (n=balajibh@96-10.southernonline.net) |
| 21:23.50 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 21:23.54 | *** join/#asterisk ramindia (n=balajibh@96-10.southernonline.net) |
| 21:23.59 | *** part/#asterisk etfonhomey1 (n=etfonhom@74-143-192-74.static.insightbb.com) |
| 21:24.49 | dennis00 | Also, what software is used for iPhone/SIP by most of youj? |
| 21:26.09 | Geminizer | what channel variable holds the did used to access a given dialplan? |
| 21:26.38 | Geminizer | e.g. I call DID 18001234567 to access my pbx... is there a variable that holds 18001234567 ? |
| 21:26.57 | *** join/#asterisk jdoe (i=jdoe@falseprophet.ca) |
| 21:27.34 | carrar | Geminizer |
| 21:27.36 | carrar | ~book |
| 21:27.37 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:27.45 | carrar | Please give that a read |
| 21:28.12 | [TK]D-Fender | Geminizer: Where does the call land when it comes in? |
| 21:28.32 | Geminizer | into a dialplan context I defined as [dev_test] |
| 21:30.01 | nny | hooray! |
| 21:30.14 | nny | writing documentation is fun! |
| 21:30.19 | nny | 1.) load gun |
| 21:30.23 | nny | 2.) press to head |
| 21:30.24 | nny | 3.) ??? |
| 21:30.38 | Geminizer | 3.) realize no bullets are in it |
| 21:30.56 | nny | hmm yeah that pretty much explains the issue |
| 21:32.02 | nny | 1.) load gun. insert ".45 ACP" into magazine, and load into gun |
| 21:32.30 | nny | 2.) press to head (yours, not someone elses. See "Head" under glossary) (Head not available on all models) |
| 21:32.41 | Geminizer | nice |
| 21:33.06 | nny | back to... work! |
| 21:33.21 | bmoraca_work | nny: i put the handle on my forehead and nothing ahppened! fixitfixitifixit! |
| 21:33.26 | nny | lol |
| 21:33.28 | nny | dammit |
| 21:33.29 | nny | :D |
| 21:33.43 | bmoraca_work | i HATE end user support |
| 21:33.55 | path | agrees |
| 21:34.24 | nny | my favorite is playing dueling mouses with someone who you have a remote session with |
| 21:35.04 | path | ahahaha |
| 21:35.18 | bmoraca_work | the worst is when you have to support a user who knows just enough to be dangerous and not enough to know that he is still a moron |
| 21:35.20 | bmoraca_work | uhg |
| 21:35.29 | Geminizer | ~moron |
| 21:35.30 | infobot | hmm... moron is someone that types long lines starting with "infobot" |
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| 21:42.19 | Cliph | Is there a workaround for compiling wanpipe on a kernel > 2.6.30 now that the network device API has changed and the old compatibility configuration has been removed? |
| 21:43.00 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 21:43.20 | Cliph | I'm trying to compile on 2.6.32.3 but that doesn't work any more |
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| 21:50.53 | Cliph | Hello? Anyone here involved with wanpipe or know about it? |
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| 22:02.45 | p3nguin | Is there any circumstance where VoIP over wireless networking is okay? Such as over WiMAX? |
| 22:04.24 | Kobaz | it depends |
| 22:04.34 | Kobaz | cell phones are essentially voip over wireless |
| 22:04.52 | Kobaz | minux the ip part... it's voice over packet networks, really |
| 22:05.07 | Kobaz | *minus |
| 22:06.52 | bmoraca_work | GSM and CDMA aren't really packet networks. they're still TDM. |
| 22:07.18 | bmoraca_work | you can encapsulate packet networks over GSM and CDMA, but GSM and CDMA themselves are TDM |
| 22:07.46 | Kobaz | hmm |
| 22:08.36 | bmoraca_work | p3nguin, it depends what you want to do with it. i use VoIP over wireless bridges all the time. wireless interent, such as Clearwire or G3 cellular, has too much latency (150ms is typically the max latency you can have). and wireless (802.11b/g) phones are terrible. |
| 22:12.51 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 22:12.51 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.1 (2010/01/15), Asterisk 1.6.1.13 (2010/01/15), 1.6.0.21 (2010/01/15), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #freepbx #switchvox #asterisk-bugs |
| 22:12.51 | *** join/#asterisk ArtemMakhutov (n=ArtemMak@ip-95-223-6-41.unitymediagroup.de) |
| 22:12.55 | *** join/#asterisk jaskew (n=jdaskew@netblock-66-159-217-102.dslextreme.com) |
| 22:12.56 | p3nguin | They don't have wiring between the buildings. |
| 22:13.10 | bmoraca_work | p3nguin, they don't ahve ANY wiring? does that include copper for analog lines? |
| 22:13.27 | Kobaz | you can run dsl over the copper, if they have some |
| 22:13.28 | p3nguin | They have phone lines to the buildings. |
| 22:14.08 | hardwire | how can I get dialplan variable information like DNIDDigits |
| 22:14.09 | bmoraca_work | p3nguin, you can use "ethernet extenders" or "long-range ethernet"...which is basically VHDSL which allows you to push upwards of 100mbit over 2 wires |
| 22:14.28 | *** part/#asterisk Cliph (n=cliph@finlandia.proxus.net) |
| 22:14.36 | Kobaz | i played with some little $50 boxes that can do 1mbit over 1500 feet on one pair |
| 22:14.55 | Kobaz | bmoraca_work: what's the distance on those? |
| 22:15.10 | bmoraca_work | Kobaz, 500 feet usually for 100mbit...maybe a bit less... |
| 22:15.24 | bmoraca_work | Kobaz, the idea isn't generally to increase distance, but to get ethernet over fewer pairs |
| 22:15.40 | p3nguin | The phone lines originate from a single location, so this could be plausible. |
| 22:15.49 | Kobaz | there's a run of a half a mile or so... that i would like to stretch a network across |
| 22:15.52 | hardwire | ah.. ${DNID} is a shortcut to it |
| 22:15.58 | Kobaz | no line of sight for wireless |
| 22:16.16 | bmoraca_work | Kobaz, that would be a bit too long, likely. i wouldn't trust anything other than fiber at that distance |
| 22:17.46 | bmoraca_work | p3nguin, that might be your best bet. point-to-multipoint wireless bridges are always an option, too. as far as cost goes, you're probably about equal. |
| 22:18.11 | p3nguin | equal to the ethernet extender method? |
| 22:18.17 | bmoraca_work | yep |
| 22:18.30 | bmoraca_work | there are cheap ethernet extenders, but you're not going to want those |
| 22:19.50 | p3nguin | Is the point-to-multipoint bridge on 5 GHz good enough to use as a medium for primary phones? I know some ITSPs say their service is not to be used as the only phone service available in a location. |
| 22:20.28 | bmoraca_work | that depends on environmental factors |
| 22:20.49 | bmoraca_work | Cisco Aironets are very solid, but wireless interference can be pretty intense |
| 22:21.14 | bmoraca_work | if you have a mesh topology, combined with STP or RSTP, it can be fairly fault tolerant |
| 22:21.34 | p3nguin | okay, sounds promising. |
| 22:21.34 | bmoraca_work | and self-healing |
| 22:21.36 | bmoraca_work | but it's not cheap |
| 22:21.41 | p3nguin | yeah |
| 22:22.08 | bmoraca_work | but if they're talking about a DS3, i don't think they're too concerned |
| 22:22.44 | bmoraca_work | honestly, you might be better off just utilizing the existing copper to each location and using media gateways to provide analog service right from there...no need for an asterisk box in each location |
| 22:22.45 | p3nguin | They want the DS3 to be able to provide both data and voice services to 150+ condos. |
| 22:24.47 | bmoraca_work | it really depends on how the place is wired...if a single MPOE for all 8 buildings, just locate everything at the MPOE. if each building has it's own MPOE, that's a bit tougher. if crossconnects exist, then LRE might be an option. if crossconnects don't exist, wireless might be an option. idealy, fiber would be your best option. |
| 22:25.32 | p3nguin | The fiber idea was tossed out immediately because they can't trench or dig. |
| 22:25.57 | bmoraca_work | you can always horizontally bore. and it would probably end up much cheaper in the long run |
| 22:28.27 | jdoe | p3nguin: I'm just surprised there's no conduit, no poles they can hitch a ride on, *nothing* between the other buildings. |
| 22:29.00 | bmoraca_work | jdoe, depends on the age...the existing conduit might have too many bends or too sharp of bends to be reliably used for data cabling |
| 22:29.48 | jdoe | maybe. |
| 22:30.13 | bmoraca_work | i've seen some absolutely deplorable cable installations...nothing surprises me anymore |
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| 22:34.11 | p3nguin | With a media gateway at each location, those will communicate over the extended ethernet back to my * box at the main building? |
| 22:34.37 | bmoraca_work | p3nguin, once again, depends on the lay of the existing wiring |
| 22:34.50 | p3nguin | Best case, that is how it would work? |
| 22:34.54 | bmoraca_work | p3nguin, if all buildings terminate back to a single MPOE, there's no need to have media gateways at each location |
| 22:35.23 | bmoraca_work | if each building has its own MPOE, then, yes, your media gateways would be at each location |
| 22:35.42 | bmoraca_work | or, if that's how you preferred to do it... |
| 22:36.01 | bmoraca_work | how do you plan to deliver data to each location? |
| 22:36.25 | bmoraca_work | or, rather, each condo |
| 22:36.49 | p3nguin | That's why they were talking about the WiMAX. |
| 22:37.02 | p3nguin | Oh, to each unit. |
| 22:37.10 | p3nguin | One moment. |
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| 22:44.22 | p3nguin | He was going to give wifi access to each condo unit, but I can't figure out how it was going to be done. |
| 22:46.02 | bmoraca_work | that's expensive...and not optimal |
| 22:46.54 | p3nguin | He didn't realize that horizontal boring and laying out a fiber backbone was even an option, so we're working toward that now. |
| 22:48.52 | outtolunc | zhone has a product that some hotels like to use |
| 22:49.01 | paulc | uit |
| 22:49.10 | p3nguin | If each building has an MPOE for phone wiring, deploy a media gateway in each building. Then use fiber to connect back to the main building, where * would reside. |
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| 22:50.04 | bmoraca_work | that'd be how i'd do it |
| 22:50.13 | bmoraca_work | roughly speaking, anyway |
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| 22:52.44 | p3nguin | Now he wants to use the phone wiring to provide data to each unit in addition to the analog phone service out of the media gateway. |
| 22:53.29 | p3nguin | I guess like a DSL type of thing. |
| 22:53.30 | bmoraca_work | p3nguin, LRE can do that...though you may be better off getting a small DSLAM at that point |
| 22:55.11 | p3nguin | I have been in hotel rooms that don't have wifi, where you have to plug your ethernet cable into a jack on the side of the telephone. |
| 22:55.11 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:55.39 | bmoraca_work | yes |
| 22:56.10 | p3nguin | What are they using to do that? |
| 22:56.32 | *** join/#asterisk paulc (n=paulc@unaffiliated/paulc) |
| 22:56.38 | bmoraca_work | a variant on LRE, most of the time. or an IP phone |
| 22:56.55 | p3nguin | The ones I have used aren't IP phones. |
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| 22:57.34 | bmoraca_work | p3nguin, http://www.adtran.com/web/page/portal/Adtran/product/1179641L6/3462 |
| 22:57.42 | p3nguin | They seem to be regular bell phones, using a slightly wider-than-normal phone cord, and then it has an extra jack for your Ethernet. |
| 22:57.51 | [TK]D-Fender | p3nguin: could be cat3 single pair + 10/100 2 pair on a single RJ45 |
| 22:58.18 | bmoraca_work | p3nguin, combined with these: http://www.adtran.com/web/page/portal/Adtran/product/1179660L1/3462 |
| 22:59.27 | p3nguin | If I went with the DSLAM suggestion, what type of physical network would be needed? |
| 23:00.03 | bmoraca_work | p3nguin, with the two items that i linked (used in a pair, the 1200F supports up to 4 1248s) connect via fiber |
| 23:00.33 | bmoraca_work | depending on how the wiring is done for all 8 buildings, you might need 2 1200Fs and 8 1248s |
| 23:01.11 | bmoraca_work | the nice thing about going that route is that you now have some user authentication and what not...and can use any commercially available ADSL2+ modem |
| 23:02.12 | bmoraca_work | i should mention, that you can also connect them via copper...but fiber is always better |
| 23:02.13 | p3nguin | So each unit would need a DSL modem... and it would be exactly like the phone company giving you residential DSL? |
| 23:02.20 | bmoraca_work | yep |
| 23:04.31 | p3nguin | I'm confused. How do these appliances connect to things? |
| 23:05.07 | bmoraca_work | via ethernet to each other and to the outside world, and via copper (ADSL2+) to subscribers |
| 23:05.10 | nny | one is a DSLAM no>? |
| 23:05.34 | p3nguin | So we still need that fiber backbone to go between the buildings. |
| 23:05.51 | bmoraca_work | they would likely work over ethernet extenders, too |
| 23:06.00 | bmoraca_work | but that's just one more item to fail |
| 23:07.50 | bmoraca_work | nny: they're both part of the DSLAM, yes...one's a controll module, one's an access module |
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| 23:09.20 | p3nguin | The DSLAM only provides data to the units with modems... does the media gateway still need to be in place for phones/voice to work? |
| 23:09.45 | bmoraca_work | p3nguin, if you want to provide dialtone, yes, you'd still need media gateways |
| 23:10.14 | p3nguin | I would need both dialtone and data in each condo unit. |
| 23:10.39 | bmoraca_work | the Adtran DSLAMs would do data, and into them you would feed dialtone coming from your media gateways |
| 23:11.32 | bmoraca_work | man, why can't i be asked to do cool projects like this? |
| 23:12.02 | [TK]D-Fender | p3nguin: What wiring do you currently have to work with? |
| 23:12.10 | p3nguin | Each building will need its own 1248, but all 8 buildings can share 2 1200Fs? |
| 23:12.58 | bmoraca_work | p3nguin, yes |
| 23:13.10 | p3nguin | [tk]d-fender: Each of eight buildings has existing analog copper phone lines going to each condo unit, and each building has its own MPOE. |
| 23:13.13 | p3nguin | That is all. |
| 23:13.25 | bmoraca_work | provided you have fewer than 48 condos in each building, anyway |
| 23:13.27 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 23:13.36 | [TK]D-Fender | p3nguin: Single pair to each room? |
| 23:15.10 | p3nguin | There is a single pair to each unit, where I assume each unit is probably piggy-backing a second room/jack. |
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| 23:15.46 | [TK]D-Fender | p3nguin: Sounds messy.... maybe you should go WiFi |
| 23:16.26 | p3nguin | as a backhaul, or within each building? |
| 23:17.18 | bmoraca_work | wireless with that many users and with that much coverage has its own problems. wired is always better. i'd go the DSLAM route. |
| 23:18.25 | p3nguin | In those hotels where they give me a bell phone with an Ethernet jack on the side, they don't require me to use a DSL modem -- i just plug a patch cable between my laptop and the phone. |
| 23:18.57 | p3nguin | What type of setup would provide that? |
| 23:19.01 | [TK]D-Fender | bmoraca_work: Tahts a lot of equipment on the backend, a shit-ton of cross connecting, DSL modems in each room, pppoe auth server, etc |
| 23:19.25 | bmoraca_work | p3nguin, as [TK]D-Fender already pointed out, they probably feed you a 4-pair connection and use 2 for the phone and 2 for the ethernet |
| 23:19.56 | bmoraca_work | [TK]D-Fender, no worse than maintaining 5-10 wireless accesspoints with 802.11x RADIUS authentication and running all the cable for those |
| 23:19.56 | p3nguin | 170 total units, eight total buildings |
| 23:20.20 | jaskew | FWIW - consider that residents may have leaky microwaves, laptops & printers w/ ad-hoc WiFi and other noisy devices on 2.4Ghz. The reliability of 2.4Ghz WiFi (esp. any point to point links) is questionable. |
| 23:20.52 | bmoraca_work | per building |
| 23:21.26 | jaskew | Not quite so bad for internet connectivity, but could be a disaster for voice. |
| 23:25.01 | [TK]D-Fender | this has nothing to do with voice |
| 23:25.07 | [TK]D-Fender | Leave the existing copper for that |
| 23:25.40 | bmoraca_work | with a DSLAM, he can use the existing copper for both. it's the cleaner and more elegant solution, in my opinion, at least. |
| 23:26.37 | jaskew | I thought someone had suggested using a wireless backhaul to carry multiplexed voice & data. |
| 23:27.18 | bmoraca_work | jaskew, point-to-point wireless bridges in the 5ghz range wouldn't have a problem with that, really. |
| 23:29.36 | jaskew | True. I might have misunderstood. I thought there was a suggestion to use WiFi (2.4Ghz) for inter-building connections. |
| 23:31.22 | bmoraca_work | well, the original plan, i guess, was to use wifi to provide data to the residents, as well... |
| 23:31.46 | *** part/#asterisk DMeloUK (n=Administ@64.129.95.226) |
| 23:33.07 | jaskew | Sounds like a really fun project. Was the nature of the builds mentioned (e.g. rest-home, college dorm, mental institution)? |
| 23:33.25 | jaskew | *builds s/b "facilities" |
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| 23:33.49 | p3nguin | The thing I was talking about is an eight building condominium complex. |
| 23:34.13 | bmoraca_work | it would totally be an awesomely fun project |
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| 23:34.22 | p3nguin | They are worried about price. |
| 23:34.35 | p3nguin | How much do those Adtran appliances cost? |
| 23:34.41 | bmoraca_work | not sure |
| 23:35.44 | bmoraca_work | they don't look too expensive, though |
| 23:35.56 | bmoraca_work | er |
| 23:35.58 | bmoraca_work | nm |
| 23:36.04 | bmoraca_work | they're pretty expensive, lol |
| 23:36.27 | bmoraca_work | but if he's looking at $7000+/mo for a DS3, he can't be too concerned about price |
| 23:36.40 | p3nguin | You found the price tag? |
| 23:37.04 | jaskew | My company leased an office once in a building that offered their "own" telephone/data service. We were required to use it. In reality, it was another company that specialized in tenant services. The upshot is that the building didn't have to pay for anything. The tenant service company paid for everything and, in return, had a captive customer base. |
| 23:37.27 | bmoraca_work | the 1200Fs should run you about $2500-3000 each...the 1248s will run you about $5000 each |
| 23:37.39 | p3nguin | Verizon apparently gave them a quote of $2500/month including the loop. |
| 23:37.44 | p3nguin | for the DS3 |
| 23:37.57 | bmoraca_work | $2500? that seems way low...but maybe not if he wants it unchannelized |
| 23:38.21 | bmoraca_work | either that or it's not really a DS3 and is instead just jumping him on a SONET ring or something |
| 23:39.39 | p3nguin | For data/VoIP, unchannelized is the way to go, right? |
| 23:39.40 | bmoraca_work | if it were me, i'd go channelized so that i could have dedicated voice channels...not rely on an ITSP for that kind of traffic...then again, 150 clients isn't all that many |
| 23:39.49 | bmoraca_work | for VOIP, yes, unchannelized is what you want |
| 23:39.50 | p3nguin | They aren't necessarily needing voice channels. |
| 23:40.14 | bmoraca_work | how much is he planning on billing the customers? |
| 23:40.47 | bmoraca_work | $20/mo (voice) + $30/mo (data) * 150 = $7500/mo |
| 23:40.55 | bmoraca_work | his ROI would be huge |
| 23:41.11 | bmoraca_work | even if he were to drop $60k on DSLAMS and media gateways |
| 23:41.55 | jaskew | Now add cable TV and you are set. |
| 23:42.41 | jaskew | Don't forget to nickel & dime for all the features that Asterisk can do for free :) |
| 23:42.52 | p3nguin | ;) |
| 23:43.00 | bmoraca_work | meh, billing for that stuff is more trouble than it's worth |
| 23:43.10 | jaskew | I |
| 23:43.17 | jaskew | I'll do it 4 u. |
| 23:43.21 | bmoraca_work | lol |
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| 23:46.03 | jaskew | That's my alter-ego talking. In reality I let a lot of stuff go that I could probably bill for. |
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