| 00:01.19 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-18-80.bois.qwest.net) |
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| 00:27.43 | wonderworld | hey, i am having a problem with dead channels in asterisk. |
| 00:28.06 | wonderworld | channels just stay there forever, even if the call has ended |
| 00:28.27 | wonderworld | this leaves me with a growing number of dead channels. anything i could do about it? |
| 00:29.33 | WIMPy | What kind of channels? |
| 00:30.16 | wonderworld | SIP |
| 00:31.42 | wonderworld | http://pastebin.com/Q1DY0HYW |
| 00:31.44 | WIMPy | Some NAT thing timing out? |
| 00:32.10 | wonderworld | all these channels had calls, but they ended |
| 00:32.19 | wonderworld | server is unNATed |
| 00:32.47 | WIMPy | There must be some other issue then why the information didn't get there. |
| 00:33.06 | WIMPy | But rtptimeout should get rid of the channels. |
| 00:33.13 | WIMPy | But not the underlying issue. |
| 00:33.45 | wonderworld | is there a way to set a timeout in asterisk for such channels? |
| 00:33.54 | wonderworld | ahh ok, thanks |
| 00:34.03 | wonderworld | i will look into rtptimeout |
| 00:34.16 | WIMPy | Session timers are another one. |
| 00:40.06 | *** join/#asterisk jamicque (~jamicque@80.50.125.74) |
| 00:40.13 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 00:40.19 | wonderworld | hmm, i added rtptimeout=60 to the general section of my sip.conf and did a sip reload |
| 00:40.28 | wonderworld | the channels are still there |
| 00:40.41 | WIMPy | I wouldn;t expect it to work for existing channels. |
| 00:40.50 | wonderworld | ahh ok |
| 00:40.57 | jamicque | Hi is there any way to make asterisk accept (200 OK) from any phone. Something like guest option (accept every invite) but for regestrations? |
| 00:41.30 | jamicque | No matter what the username and password is asterisk will authenticate you |
| 00:43.23 | WIMPy | Do you just want an ok or do you also want it to keep the details? |
| 00:43.42 | WIMPy | The answer is no in both cases, just curious. |
| 00:43.45 | wonderworld | WIMPy: hey, that worked. thanks a lot |
| 00:44.08 | WIMPy | wonderworld: But it doesn't fix the issue. |
| 00:44.27 | wonderworld | nope. probably a not well behaving softphone? |
| 00:44.44 | WIMPy | Possible. |
| 00:44.52 | wonderworld | calls were made from an android sip client |
| 00:45.01 | WIMPy | It could be any sort of networking issue as well. |
| 00:45.18 | WIMPy | Lost connectivity would be another explanation. |
| 00:47.51 | wonderworld | k, as long as i can get rid of them with rtptimeout, i have no real problem. very good |
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| 02:16.51 | henningb | Hi, I'm new to asterisk - I have a small question: does the digium card TE121 have a jumper for E1/T1? |
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| 02:21.12 | *** join/#asterisk henningb (~henning@118.175.66.17) |
| 02:21.47 | henningb | Sorry I disconnected - does anyone have experience with Digium cards? |
| 02:22.40 | henningb | Does the TE121 have a jumper? |
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| 04:58.47 | ectospasm | henningb: yes, the TE121 has a T1E1 jumper |
| 04:59.35 | ectospasm | consult the manual for what each jumper state means: http://docs.digium.com/TE122/TE120_series_manual.pdf |
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| 06:02.18 | MJCS | I must be doing something wrong with my dialplan. When I answer the call, the system hangs up after roughly 6 seconds. http://p.linode.com/6902 |
| 06:05.58 | ectospasm | MJCS: do you have CLI output showing what's happening? |
| 06:06.13 | MJCS | yeah give me a second |
| 06:06.20 | MJCS | is verbose 3 fine? |
| 06:06.37 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-coyoganysvfmrxbg) |
| 06:07.30 | ectospasm | should be OK |
| 06:07.42 | MJCS | ok give me a moment then |
| 06:08.13 | ectospasm | maybe launch the remote session with "asterisk -rvvvT | tee /tmp/cli-capture.txt" |
| 06:08.18 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-xssmqrqymapecvfc) |
| 06:11.55 | MJCS | http://p.linode.com/6903 |
| 06:12.24 | MJCS | i only changed my incoming # and the SIP ID from skype |
| 06:14.19 | ectospasm | need more info. Turn on SIP debug ("sip set debug on") |
| 06:14.43 | MJCS | ok |
| 06:19.20 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:21.18 | MJCS | http://pastebin.com/jBzMWKHQ |
| 06:22.07 | MJCS | i am guessing it is something stupidly simple |
| 06:24.16 | ectospasm | you've got multiple calls here. |
| 06:24.28 | MJCS | I am calling only from 1 phone |
| 06:24.49 | ectospasm | yes, but it looks like you called multiple times in the capture. |
| 06:24.54 | MJCS | hmm |
| 06:25.07 | ectospasm | I see at least 15 INVITEs in the debug you gave. |
| 06:25.13 | MJCS | lol |
| 06:25.18 | ectospasm | Now, half of those will be for sending to your internal extension |
| 06:25.25 | ectospasm | the other half will be coming from Skype |
| 06:25.48 | ectospasm | it may be simpler to look at a packet capture |
| 06:26.05 | ectospasm | ...that corresponds to the SIP debug in the CLI capture. |
| 06:26.55 | kaldemar | phone sends a BYE with Reason: SIP;description="media stream error: 8008" |
| 06:27.01 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-mycotqbbsxkblbyq) |
| 06:27.18 | MJCS | ill try another phone |
| 06:27.58 | ectospasm | kaldemar: I see that now |
| 06:28.15 | ectospasm | not sure what that means, though |
| 06:28.19 | MJCS | samething from a different system |
| 06:29.24 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:29.29 | schmidts | good friday morning! |
| 06:29.54 | kaldemar | ectospasm: me neither. there seems to be a common codec. there are new invites though sent in both directions though. |
| 06:29.55 | ectospasm | MJCS: at least in the capture, your X-Lite phone sends the media stream error message |
| 06:30.04 | MJCS | oh |
| 06:30.13 | MJCS | so it is the phone I am recieving with that is doing it |
| 06:31.03 | ectospasm | That's what seems to be sending the initial BYE message. See packet beginning at line 667 in http://pastebin.com/jBzMWKHQ |
| 06:33.15 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 06:34.41 | MJCS | let me change a few things in the client and try again |
| 06:36.36 | MJCS | is there another free sip client I can use? |
| 06:39.07 | ectospasm | I only know Linux clients... |
| 06:39.11 | MJCS | hmm |
| 06:41.28 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 06:43.56 | kaldemar | MJCS: http://icanblink.com/ http://www.linphone.org/ |
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| 07:09.23 | *** join/#asterisk MJCS (mjcs@ip68-5-48-206.oc.oc.cox.net) |
| 07:09.32 | MJCS | the blink client works |
| 07:09.37 | MJCS | stupid POS x-lite |
| 07:09.47 | MJCS | I've been trying to figure this out for months LOL |
| 07:10.13 | MJCS | kaldemar: thank you |
| 07:15.17 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 07:15.37 | v0lZy | hi |
| 07:18.53 | *** join/#asterisk bartek123 (~bart.chwa@modemcable151.198-83-70.mc.videotron.ca) |
| 07:19.00 | bartek123 | hello, may I ask a question? |
| 07:20.40 | bartek123 | is there any way to configure a dialogic card with any free linux based pbx software? such as asterisk+freepbx+dahdi.. . or shall I just throw this card away since intel did not release driver/code since 2005? |
| 07:22.58 | phix | bartek123: instead of throwing it away you should mail it to me :) |
| 07:23.06 | phix | if it comes to that :P |
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| 07:24.30 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:25.23 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 07:29.35 | bartek123 | hahaha |
| 07:29.59 | bartek123 | good... I found it on a server I bought on ebay for 50$ + 100 ship.. lol |
| 07:30.23 | bartek123 | now I installed asterisk... freepbx... but no native support.. |
| 07:30.50 | *** join/#asterisk PhoenixMage (~Phoenix@CPE-120-146-192-94.static.vic.bigpond.net.au) |
| 07:31.04 | bartek123 | and some threads say it is compatible.. but with business version and paid drivers... however thread is 7 years old.. |
| 07:32.08 | bartek123 | So I hoped to find some pbx geeks that would advise me if possible. |
| 07:34.17 | bartek123 | I guess I will be forced to buy a cheap openvox chinese card that is supported instead and put this beast in a clear epoxy mold.. |
| 07:34.38 | bartek123 | to serve me as a geek ash tray... |
| 07:35.01 | bartek123 | what a waste |
| 08:00.22 | *** join/#asterisk Da-Geek (~Da-Geek@62.254.227.226) |
| 08:00.54 | bartek123 | does any1 know if I can config a dialogic card with freepbx? |
| 08:03.54 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 08:05.04 | bartek123 | I will gladly paypal 10$USD to ne1 who can resolve my issue... |
| 08:05.07 | bartek123 | lol |
| 08:06.07 | bartek123 | maybe that will be good incentive |
| 08:17.51 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 08:21.30 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 08:22.43 | *** join/#asterisk dax_roc (~dax_roc@pdpc/supporter/bronze/daxroc) |
| 08:36.24 | dax_roc | Morning all |
| 08:46.05 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-wmudogqxdwpulkqf) |
| 09:08.01 | beebeeep | svr-media*CLI> core show uptime C |
| 09:08.05 | beebeeep | System uptime: 3 years, 47 weeks, 3 days, 18 hours, 58 minutes, 4 seconds |
| 09:08.19 | beebeeep | wow that's really old server :) |
| 09:08.56 | beaver_rrr | nice |
| 09:11.10 | beebeeep | 1.4.21.1 is rock solid lol |
| 09:13.27 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:30f7:92ab:805e:832a) |
| 09:18.08 | rolandow | leifmadsen: is your online book updated? i think there's a little error in the recording example script http://astbook.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/IVR_id247054.html |
| 09:18.34 | rolandow | leifmadsen: the mv command doesn't include the full path, so it doesn't work .. or at least didn't work in my test :) |
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| 09:40.03 | *** join/#asterisk Chetan07 (~chetan@204.15.169.242) |
| 09:40.49 | Chetan07 | hello all, I am new to asterisk , I am trying to integrate FreeRadius Client for AAA but due to lack of documentation i am unable to integrate properly.. anybody here can please help me with this.. |
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| 10:06.23 | uskerine | hi |
| 10:06.53 | uskerine | does anyone know the best way to have multiple instances of softphones running on same server (LTSP server), all of them try to use 5060 port. |
| 10:09.42 | Roelt | uskerine, if it's a client it shoudn't be a problem? |
| 10:12.15 | uskerine | Roelt, are you sure? |
| 10:13.24 | Roelt | i'm pretty sure i'm sure. |
| 10:14.21 | Roelt | 5060 port is on the server, client port can be almost anything |
| 10:14.37 | uskerine | ok |
| 10:14.42 | uskerine | so then it is the port used ont he client |
| 10:14.49 | uskerine | as the clients are on same server, it does not work |
| 10:15.16 | uskerine | i have 7078 and 7098 as RTP ports (audio/video) |
| 10:15.21 | uskerine | are those ones server or client portS? |
| 10:16.15 | Roelt | you should have more rtp ports, because every call uses 2 ports |
| 10:16.29 | WIMPy | You need to give them different SIP ports as well. |
| 10:16.45 | Roelt | WIMPy, are you sure? no problem with running multiple voip applications here.. |
| 10:16.48 | WIMPy | You can't use the same port more than once. |
| 10:16.52 | Roelt | +on one computer |
| 10:17.10 | WIMPy | Sure if they use different ports that's fine. |
| 10:17.22 | ectospasm | they need to be different sockets |
| 10:17.31 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 10:17.36 | WIMPy | Some use any random port, some use 5060 by default. |
| 10:17.36 | ectospasm | (socket==IPaddr+port) |
| 10:19.26 | WIMPy | Is that LTSP thing lust a "marketing gag" or does it do something that wasn't already possible before Linux was even invented? |
| 10:19.52 | rolandow | when you do sip show peers, what portnumber is shown here? the port being used on the client? |
| 10:20.09 | WIMPy | yes |
| 10:21.19 | WIMPy | Or at least it should. For 'sip show registry' it doesn't seem to do so. |
| 10:21.35 | rolandow | so if i'd create a port forward for those ip/ports.. this may solve some nat issues?? |
| 10:22.24 | WIMPy | Where? Why? What NAT issue? |
| 10:23.27 | rolandow | well we seem to have a crappy modem that messes up NAT for udp ports.. anyways.. i have a dect station and a 'normal' device |
| 10:23.34 | rolandow | the dect station is working fine |
| 10:23.36 | kaldemar | rolandow: not likely to solve anything. |
| 10:23.40 | rolandow | the tiptel 280 isn't |
| 10:23.52 | rolandow | ok .. too bad :) |
| 10:24.01 | kaldemar | rolandow: unless you can fix source port that the client uses. |
| 10:24.35 | rolandow | kaldemar: that's why i asked what the port number in sip show peers is showing.. |
| 10:24.57 | kaldemar | rolandow: you can enable qualify=yes (and configure it with qualifyfreq) for the peer to help keeping the NAT hole open. |
| 10:25.51 | rolandow | yes i have qualify=yes |
| 10:25.55 | *** join/#asterisk pyther (~pyther@unaffiliated/pyther) |
| 10:26.31 | rolandow | we have a weird thomson modem that is in bridge mode, but still gives us the option to forward UDP ports.. |
| 10:26.41 | rolandow | which in my opinion is weird when it's in bridge mode right? |
| 10:26.45 | kaldemar | huh? |
| 10:26.56 | WIMPy | If both devices are behind the same NAT and appear to use the same port, there's something screwed up and definitely won't work. |
| 10:26.57 | rolandow | anyways, after forwarding udp ports to our router, we solved our one-way audio partly |
| 10:27.00 | kaldemar | that must be a UI stupidity. |
| 10:27.34 | rolandow | kaldemar: probably, but it helped fixing one way audio problem on the dect station |
| 10:27.44 | rolandow | kaldemar: the same problem you looked at earlier this week btw :) |
| 10:28.17 | rolandow | so the modem is pretty stupid .. but anyways, it seems to work now .. except for this one tiptel device.. |
| 10:28.32 | *** join/#asterisk bmg505 (~leon@196-209-101-227.dynamic.isadsl.co.za) |
| 10:29.21 | kaldemar | rolandow: oh, it was that case. your sip debug didn't seem to have anything wrong, IIRC. |
| 10:29.34 | rolandow | kaldemar: right.. that's the one :) |
| 10:31.32 | Roelt | rolandow, probably a kpn Thomson adsl modem/router? |
| 10:31.58 | Roelt | have one of those here.. hard crash when i call from phone 1 to phone 2, have to unplug the power before it comes back to life... |
| 10:39.34 | *** join/#asterisk Digweed (digweed@2a00:f10:11a:764::c0ca:c01a) |
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| 11:07.00 | rolandow | Roelt: a thomson you mean? |
| 11:07.08 | rolandow | Roelt: or a tiptel ? |
| 11:09.16 | Roelt | 4~Thomson |
| 11:09.36 | rolandow | hm.. i'm wondering if we can replace it with a better thing.. but what's supposed to be better for * ? |
| 11:14.41 | *** join/#asterisk irroot (~gregory@41.79.221.226) |
| 11:14.59 | *** join/#asterisk brian98 (~brian98@188.141.15.112) |
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| 11:22.10 | Roelt | those fritzbox modems are pretty nice |
| 11:22.38 | rolandow | Roelt: aren't those modem and routers in one? |
| 11:23.03 | rolandow | Roelt: we already have netgear routers for our vpn so we need to keep those and probably get a modem that just *really* bridges everything |
| 11:24.21 | Roelt | probably noy a problem putting it in bridge mode |
| 11:27.01 | puzzled | fritzbox no longer supports bridging mode |
| 11:27.17 | *** join/#asterisk cers (~cers@130.226.232.146) |
| 11:27.30 | puzzled | at least not my 7340 or something like that |
| 11:27.46 | Roelt | hm, the one i've installed a few months ago did that without a problem |
| 11:28.00 | Chetan07 | <PROTECTED> |
| 11:28.01 | Roelt | don't know the modelnumber |
| 11:29.14 | rolandow | i see some d-links that are just modems.. maybe that would be better |
| 11:30.26 | cers | Hi, I'm new to asterisk and I'm trying to get input from the user using this conf http://pastebin.ca/2172917, but when a user presses a button, I get WARNING[1042]: chan_sip.c:18735 handle_request_info: Unable to retrieve DTMF signal for INFO message on call ... any ideas what I've done wrong? |
| 11:31.00 | *** join/#asterisk PhoenixMage (~Phoenix@CPE-120-146-192-94.static.vic.bigpond.net.au) |
| 11:31.34 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 11:32.12 | ectospasm | cers: what DTMF mode have you selected on the phone, and in Asterisk (sip.conf) |
| 11:32.51 | ectospasm | ? |
| 11:35.38 | cers | ectospasm: this is my sip.conf http://pastebin.ca/2172919 - the phone, better question - I don't recall seeing it as an option there - it's a aastra dialog 4425 |
| 11:36.26 | cers | (which btw, could totally use a software upgrade I'm sure - if anyone happens to know the aastra software update server ip) |
| 11:37.52 | cers | ectospasm: the user I'm testing with is 1000 btw |
| 11:41.53 | ectospasm | looks like it's the phone. Check the DTMF mode on your Aastra |
| 11:42.29 | ectospasm | according to the original WARNING you posted, it appears that it's set for INFO |
| 11:42.33 | ectospasm | ...not RFC2833 |
| 11:43.15 | cers | ectospasm: I'm just looking through the settings, it's not mentioned anywhere :-S |
| 11:43.56 | ectospasm | I know zero about Aastra phones. |
| 11:44.11 | ectospasm | My Polycom knowledge is subsiding |
| 11:44.59 | cers | ectospasm: it's not getting the string "INFO" from the message - it's hardcoded in the source: http://www.asterisk.org/doxygen/trunk/chan__sip_8c-source.html#l20275 |
| 11:45.51 | cers | ectospasm: thanks for trying though :-) I'm hoping updating the phone software will magically help - as soon as I find the update server... I know they have one somewhere... |
| 11:47.46 | ectospasm | um... I assumed the phone was sending DTMF in INFO messages, but of course I haven't looked at a packet capture, nor CLI debug (dis)proving that |
| 11:49.40 | cers | ectospasm: it's entirely possible it is - as I said, I'm very new to this... |
| 11:49.52 | cers | ectospasm: here's a log: http://pastebin.ca/2172928 |
| 11:50.15 | cers | it does mention "INFO" in there a few times |
| 11:51.18 | ectospasm | yep. In there is a 9 digit, sent via INFO message from 1000@172.16.9.15... which is configured for dtmfmode=rfc2833 |
| 11:51.28 | ectospasm | ...which I think is the problem |
| 11:51.57 | ectospasm | either change sip.conf so that the dtmfmode=info, or change the setting in the 1000 phone |
| 11:52.15 | cers | thanks - I don't see a way of doing that on the phone, so I' |
| 11:52.22 | cers | 'll try in sip.conf |
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| 12:06.09 | cers | ectospasm: hmm.. setting it to =info doesn't make a difference. Thanks for the help though - I'll have to look at it more when I get back from vacation :-) |
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| 12:08.27 | ectospasm | cers: did you "sip reload"? |
| 12:08.53 | ectospasm | actually, I don't know that sip reload will change the dtmfmode... may have to restart Asterisk. |
| 12:09.01 | ectospasm | anyway, time to go home... |
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| 12:33.08 | carrar | *Y*A*W*N* |
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| 12:46.36 | LIIT | Can anyone point me in the direction of some information on how to connect two Asterisk boxes via IAX, when one of the boxes is NAT'ed ? |
| 12:48.22 | kaldemar | LIIT: just forward UDP 4569 to the asterisk box. that's it. |
| 12:48.54 | kaldemar | LIIT: actually that's not even required if you make the boxes register to each other. |
| 12:49.19 | LIIT | Heh, I should really learn to explain things better. They are both nat'ed, but I only have control over one of the firewalls. So one of them has 4569 open, and the other does not |
| 12:49.26 | LIIT | (incoming that is - outbound is all open) |
| 12:51.57 | kaldemar | LIIT: IAX has qualify options for keepalive. |
| 12:52.44 | [TK]D-Fender | Register the one whose router you don't control to the other box and enable qualify on your peer to them |
| 12:53.01 | LIIT | Thx both - I will read up on qualify :-) |
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| 13:44.37 | LIIT | Should I use qualify=yes or a specific value ? I keep getting IAX2/<public_ip_of_firewalled_box>:4569-5402 is circuit-busy |
| 13:45.08 | *** join/#asterisk grapsus (~grapsus@2a01:e35:8acb:36e0:c2cb:38ff:fe62:b4ba) |
| 13:45.22 | grapsus | hi there |
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| 13:47.35 | grapsus | is any way to route an outgoing call between et sip provider and a dahdi line, depending on which one is unused ? |
| 13:48.07 | WIMPy | LIIT: Looks like the peer is unreachable. |
| 13:48.34 | WIMPy | grapsus: Which one what? |
| 13:49.28 | grapsus | WIMPy: I have two providers for outgoing calls, one is SIP and another one is a POTS line connected via Dahdi |
| 13:49.42 | grapsus | and I want to select whatever line is available |
| 13:49.52 | n3hxs | Yawn |
| 13:49.53 | WIMPy | Yes, and how do you want to select them? |
| 13:50.09 | grapsus | each one can handle only one call at once |
| 13:50.31 | EmleyMoor | grapsus: In general, just Dial in order of preference. If not available, it falls through to the next |
| 13:51.05 | WIMPy | You can use CHANISAVAIL or you just try to dial out and look at DIALSTATUS to see if you should try again on the other route. |
| 13:51.45 | grapsus | oh I see, it as simple as that |
| 13:52.10 | EmleyMoor | You might want to do further checking but I do it that way |
| 13:52.21 | grapsus | how do I prevent the second Dial from being executed if both lines are free ? |
| 13:52.44 | EmleyMoor | grapsus: Exit status from the original prevents it automatically |
| 13:53.05 | WIMPy | If the call was answered, that is. |
| 13:53.13 | EmleyMoor | Yes... |
| 13:53.32 | grapsus | I see, I should really read some core documentation about the dialplan |
| 13:53.34 | WIMPy | If it just timed out, it would try again without checking. |
| 13:53.58 | WIMPy | Try the |
| 13:53.59 | EmleyMoor | If you don't want it to try on the other line if it's just timed out, you'd need to handle that as a special case. |
| 13:54.00 | WIMPy | ~book |
| 13:54.00 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:55.57 | grapsus | WIMPy: EmleyMoor: thank you for the info, I will try the simplest way before going into scripting etc. |
| 13:56.21 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 13:57.30 | LIIT | kaldemar & [TK]D-Fender: qualify fixed it - thanks :-) |
| 13:59.00 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 13:59.00 | EmleyMoor | Someone asked if I could take my phone number with me when I moved recently - so I told them I was taking the whole telephone exchange! |
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| 14:04.45 | EmleyMoor | Need to retest that phone I mended the other day - howls on my line. If it's OK on David's then perhaps I need a better socket... |
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| 14:11.21 | Katty | sits and stares |
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| 14:16.03 | *** join/#asterisk PhoenixMage (~Phoenix@CPE-120-146-192-94.static.vic.bigpond.net.au) |
| 14:16.06 | Katty | awfully quiet this morning. |
| 14:16.15 | Katty | is everyone trying to pick out names for leif |
| 14:16.51 | Chainsaw | Or wondering about EmleyMoor's howling werephone. |
| 14:17.20 | Katty | seems legit. |
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| 14:17.31 | Katty | hi tzafrir |
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| 14:18.40 | [TK]D-Fender | Katty, if it's a girl, "Nana" ... and for a middle name .. "Nanana" ! |
| 14:18.43 | [TK]D-Fender | \o/ |
| 14:19.00 | [TK]D-Fender | leifmadsen, ^^^ |
| 14:19.31 | Katty | facepalms. |
| 14:19.46 | Katty | that /was/ pretty good tho, fender. |
| 14:20.06 | [TK]D-Fender | Perfectly legit first name... |
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| 14:28.21 | drmessano | I liked "erik" and "elga" |
| 14:28.25 | drmessano | But whatev |
| 14:29.05 | drmessano | If I had two kids I wanted to name them "Istanbul" and "Constantinople" |
| 14:29.23 | serafie | Katty: It's going to be Þórfinn Viktor Leiffson. |
| 14:29.24 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:29.28 | drmessano | So if I called one, and the other came, "Istanbul, not Constantinople" |
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| 14:30.05 | BenC[UK] | Hi Guys |
| 14:30.39 | BenC[UK] | I'm trying to set up another voip server similar to another setup I have already but calls won't connect for outbound |
| 14:30.43 | BenC[UK] | Retransmission timeout reached on transmission |
| 14:30.46 | BenC[UK] | getting that |
| 14:31.06 | BenC[UK] | the server is behind nat, I have set externip in sip.conf and nat=yes is on for the trunks |
| 14:31.10 | Chainsaw | BenC[UK]: That would suggest that the SIP UDP traffic is going away. |
| 14:31.22 | Chainsaw | BenC[UK]: If you can't trust your ISP, and you likely can't, try SIP over TCP instead. |
| 14:31.47 | BenC[UK] | its with the same host as another one I have working |
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| 14:33.15 | Katty | needs caffeine so very badly. |
| 14:34.16 | BenC[UK] | hmm, the user I make the call from goes offline while making the call aswell |
| 14:34.59 | *** join/#asterisk luckman212 (~irc@2001:470:8abb:0:e424:e4c:eef4:a291) |
| 14:37.48 | kaldemar | BenC[UK]: you need nat=yes under [general]. |
| 14:38.05 | BenC[UK] | which file? |
| 14:38.17 | kaldemar | BenC[UK]: sip.conf, naturally. |
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| 14:47.14 | vastina | morning telephreaks |
| 14:47.29 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 14:48.30 | dhuckaby | morn'n vastina |
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| 14:49.59 | BenC[UK] | kaldemar: still times out after about 10 seconds :( |
| 14:52.53 | saysocomm | Hey guys, anyone know how to trigger a Custom devstate when a channel hangs up? |
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| 14:53.42 | Katty | need....moar....caffieine... *yawn* |
| 14:53.56 | saysocomm | I'm able to set INUSE by using macro-dialout-trunk-predial-hook but can't figure out the hangup part |
| 14:53.58 | drmessano | You need a boar and caffeine? Strange request, but ok |
| 14:54.17 | Katty | drmessano: stayed up too late |
| 14:54.21 | Katty | drmessano: FAR too late. |
| 14:54.31 | Katty | drmessano: and got woken up early....far too early |
| 14:54.50 | Katty | i'm going to go make friends with the soda machine. brb |
| 14:55.02 | kaldemar | saysocomm: where did you get macro-dialout-trunk-predial-hook? |
| 14:55.35 | saysocomm | actually a freepbx thing. |
| 14:56.00 | saysocomm | need moar caffeine |
| 14:56.02 | kaldemar | saysocomm: #freepbx is more able to help with that. |
| 14:56.09 | saysocomm | yep whoops! |
| 14:56.40 | Katty | mmm mt dew. |
| 14:56.43 | Katty | pets |
| 14:56.47 | [TK]D-Fender | BenC[UK], the server is behind nat, I have set externip in sip.conf and nat=yes is on for the trunks <--- providers are NOT behind NAT and setting them to "yes" can very likely cause problems |
| 14:57.27 | BenC[UK] | I've tried with/without |
| 14:57.31 | BenC[UK] | do I need externip set? |
| 14:57.44 | [TK]D-Fender | yes |
| 14:57.47 | [TK]D-Fender | ~sipnat |
| 14:57.47 | infobot | [~sipnat] Quick guide on configuring Asterisk + 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 . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 14:57.49 | [TK]D-Fender | ^^ |
| 14:57.51 | BenC[UK] | without it I get "Service Unavailable right away |
| 14:58.08 | [TK]D-Fender | follow the guide and replace "canreinvite" with "directmedia" for 1.6+ |
| 14:58.18 | [TK]D-Fender | BenC[UK], that message means nothing to me. |
| 14:58.23 | [TK]D-Fender | pastebin an ACTUAL call attempt |
| 14:58.30 | [TK]D-Fender | With SIP DEBUG |
| 14:58.31 | [TK]D-Fender | ~pb |
| 14:58.31 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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 install pastebinit with yum or aptitude. |
| 14:58.33 | [TK]D-Fender | ^^^ |
| 14:58.34 | BenC[UK] | Got SIP response 503 "Service Unavailable" back from 68.233.226.97:5060 |
| 14:58.50 | [TK]D-Fender | maybe your accounts isn't active |
| 14:58.57 | [TK]D-Fender | but lest see the WHOLE thing |
| 14:59.01 | [TK]D-Fender | lets* |
| 15:06.08 | BenC[UK] | http://pastebin.ca/2173013 |
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| 15:34.51 | dhuckaby | that infobot sipnat thing... that's only if using voip for external connections right? if using PSTN lines it doesn't matter if you're behind a NAT or not? |
| 15:36.00 | WIMPy | It applies whenever you have SIP connections with some sort of NAT involved. |
| 15:36.43 | WIMPy | So if you use SIP inside your LAN only, it doesn't matter. |
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| 15:48.29 | BenC[UK] | [TK]D-Fender: did you see my paste? |
| 15:49.06 | [TK]D-Fender | Yes, and you've masked IP's so I can't trust what I don't see and I do not see a COMPLETE call attemp |
| 15:49.27 | [TK]D-Fender | ONE more chance at this. verbose 10. SIP DEBUG. Complete call. |
| 15:50.23 | *** join/#asterisk echo083 (~adam@mna75-11-88-161-199-126.fbx.proxad.net) |
| 15:50.49 | BenC[UK] | I search/plreaced just my server ip with 111.111.111.111 |
| 15:51.07 | BenC[UK] | left the rest of the ips |
| 15:51.50 | echo083 | freepbx 2.10 is full of bug when trying to be installed manually executing install_amp |
| 15:52.31 | WIMPy | echo083: #freepbx |
| 15:52.45 | echo083 | WIMPy, didn't know there was a dedicated channel sorry ;) |
| 15:55.14 | BenC[UK] | [TK]D-Fender: http://pastebin.ca/2173038 |
| 15:55.33 | BenC[UK] | I have replaced the server ip with 111.111.111.111 again but thats the only thing to change |
| 15:56.38 | WIMPy | The haxx0rs will find your server anyway. |
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| 15:57.53 | *** join/#asterisk NKing (~N.King@oh-67-76-244-251.sta.embarqhsd.net) |
| 15:59.15 | BenC[UK] | yeah :( |
| 16:03.00 | [TK]D-Fender | moves on to other matters |
| 16:03.23 | dhuckaby | how ya do 'emotes' like that on IRC Fender? |
| 16:03.36 | BenC[UK] | [TK]D-Fender: so you can't help? |
| 16:04.56 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 16:07.25 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 16:08.04 | Katty | wow the knits are goign slow |
| 16:15.13 | carrar | Please assign knits.conf priority=-9 |
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| 16:34.45 | NKing | poly |
| 16:34.54 | NKing | polycom |
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| 16:53.23 | dhuckaby | with AsteriskNOW... if using only software phones... having already created an extension... do I need to setup a phone?(the quickstart guide mentions installing a digium phone only) |
| 16:58.36 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 16:59.02 | Gugge | dhuckaby: yes, you need to setup a phone (softphone) to be able to use it |
| 17:00.25 | rubberneck | I am trying to get a video conference to work in asterisk 10. I am using the jitsi client. I can connect and audio works well, but video doesn't work at all, there is no video. I can't seem to find what I am doing wrong. |
| 17:00.48 | rubberneck | confbridge.conf http://pastebin.ca/2173056 extensions.conf http://pastebin.ca/2173057 sip.conf http://pastebin.ca/2173058 |
| 17:01.10 | rubberneck | Do I have to enable video somewhere or something? |
| 17:02.12 | Gugge | i would try videosupport |
| 17:02.17 | WIMPy | I don't know what the default is, but see "videosupport". |
| 17:02.29 | WIMPy | And allowed codecs. |
| 17:03.51 | rubberneck | ...........and videosupport it is. Thanks it's working now. |
| 17:05.52 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
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| 17:11.59 | dhuckaby | Gugge: but does the smartphone need to be setup within FreePBX or just on the computer? |
| 17:12.20 | WIMPy | dhuckaby: Try in #freepbx |
| 17:12.48 | [TK]D-Fender | Softphone is it's own application |
| 17:13.01 | [TK]D-Fender | its* |
| 17:13.22 | WIMPy | has no ide what is called what in the frontend. |
| 17:13.53 | WIMPy | But if it's about the phone itself that is neither a freepbx nor an asterisk question, off course. |
| 17:15.27 | dhuckaby | it's not.. |
| 17:16.08 | dhuckaby | was just going through the AsteriskNow2.0 quick-install guide...and it mentioned setting up specific phones |
| 17:16.14 | dhuckaby | digium |
| 17:16.27 | [TK]D-Fender | No, you clearly don't need ANY phone necessarily |
| 17:16.56 | *** join/#asterisk _omer (~omer@182.178.149.235) |
| 17:17.03 | WIMPy | And you don't configure phones you don't have. |
| 17:21.09 | *** join/#asterisk Happy0 (Happy0@cpc2-broo8-2-0-cust282.14-2.cable.virginmedia.com) |
| 17:32.50 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:39.02 | _omer | I am really stucked in USer/Group thing .... |
| 17:49.33 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 18:09.21 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 18:09.48 | jaytee | anyone here ever used 3Com 2102 or 3102 phones with Asterisk? |
| 18:10.37 | [TK]D-Fender | Those are both ironically Linksys part #'s.. |
| 18:10.40 | malcolmd | negative; best of luck though |
| 18:16.04 | [TK]D-Fender | Was this a barin-fart or are 3com completely lacking in originality |
| 18:17.50 | malcolmd | i think the 3com phones there predate the linksys equipment |
| 18:18.12 | WIMPy | Like the 7950 and 7960 by AllNet? |
| 18:19.38 | dhuckaby | Fender. AsteriskNOW up and run'n like a champ :) |
| 18:20.07 | dhuckaby | with soft-phones ;)... I did have to fabricate a "microphone" for my machine tho :) |
| 18:21.02 | dhuckaby | so if I just installed a 4 port FXO(I think it's O) card and plugged in the phone lines it should be a fully functional system eh? |
| 18:21.47 | [TK]D-Fender | dhuckaby, If you configure it. |
| 18:25.24 | dhuckaby | aye ;) that configure thing... ;) |
| 18:26.35 | [TK]D-Fender | Your server might have been "fully functional" before getting a card. |
| 18:26.40 | dhuckaby | luckily(unluckily maybe).. I found the antique phone mail people are still in business and have a "modern" solution for $500 :) |
| 18:26.49 | dhuckaby | @Fender it is if we were going VOIP only |
| 18:26.57 | [TK]D-Fender | By doing so you gave it NEW "functions" to be capable of |
| 18:27.18 | [TK]D-Fender | And thus only becomes fully functional once configured. |
| 18:27.33 | [TK]D-Fender | And frankly I don't think ANYONE's system is "fully functional" by that definition. |
| 18:27.47 | [TK]D-Fender | who is using EVERY feature their system offers them? |
| 18:29.21 | *** join/#asterisk NKing (~N.King@oh-67-76-244-251.sta.embarqhsd.net) |
| 18:33.19 | dhuckaby | aye... voicemail configuration I'm work'n on now.. |
| 18:33.45 | dhuckaby | this will still be an excellent proof of concept machine(and seriously up and running and ready to be used in under 1hr) ridiculous ;) |
| 18:41.51 | jeffspeff | I'm having an issue where several phones are able to make calls (having bi-directional audio) but if you call them then there is no audio. These phones are on a remote network, connected via VPN, however, not all phones at that location are having the issue, and sometimes they don't have any problems at all. Here's the SIP debug for me (1506) calling a known issue phone (1102). Any help with this would be awesome! http://pastebin.com/1G5h7ztf |
| 18:42.17 | *** join/#asterisk pavlx (~Guest@c-76-28-147-52.hsd1.wa.comcast.net) |
| 18:42.34 | pavlx | hi |
| 18:43.04 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:43.18 | pavlx | last month i putted to disposition my ADSL to the apartment which is over mine |
| 18:44.29 | pavlx | i would like to know if there is a way to phone with asterisk and to do to the same time ANONYMOUS VOIP v |
| 18:45.38 | pavlx | i don't want to show to the TELCO's who i am calling and how it is long |
| 18:46.09 | pavlx | awaiting your reply as soon as possible, happy hacking pavlx |
| 18:47.29 | jeffspeff | [TK]D-Fender, WIMPy , any ideas to my above issue? |
| 18:48.21 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:49.20 | [TK]D-Fender | jeffspeff, Reliably Transmitting (NAT) to 192.168.251.106:5060: |
| 18:49.28 | [TK]D-Fender | jeffspeff, Contact: <sip:1506@12.49.160.154:5060> |
| 18:49.45 | [TK]D-Fender | jeffspeff, * is passing your WAN IP to the other subnet because you didn't define the localnet for it |
| 18:51.42 | jeffspeff | [TK]D-Fender, i have the localnet defined in general in sip.conf... all other locations on different subnets are working fine... could this be a NAT configuration issue? or is there a way to be more specific in how i define/use those peers? |
| 18:53.06 | [TK]D-Fender | you did not tell * that the 192.168...... network listed there was local |
| 18:54.18 | jeffspeff | [TK]D-Fender, ok... i see what you're saying, not meaning to argue, just really confused... can i specify multiple localnets? I'm not telling * that 5 other networks are local, and they seem to work. |
| 18:54.43 | [TK]D-Fender | some get lucky |
| 18:54.50 | [TK]D-Fender | typically those who IGNORE what * tells them |
| 18:55.02 | [TK]D-Fender | Yes you can, and yes, you must |
| 18:55.43 | jeffspeff | so, localnet=10.200.100.0/255.255.255.0,192.168.251.0...... etc? |
| 18:58.40 | *** join/#asterisk pyther24 (~msg31@unaffiliated/pyther) |
| 18:59.09 | pavlx | <modem>-----<switch>---port 6 <pc_pavlx> <switch>---port 12 <apartment upstair> |
| 18:59.29 | [TK]D-Fender | jeffspeff, multiple lines |
| 18:59.49 | pyther24 | So I'm lookig at setting up an asterisk server for my home. I'll only need to connect 2 maybe 3 phones to it, don't plan on doing a whole lot with it, I was wondering how well it might run on my openwrt router (128MB ram / 680MHz) |
| 19:00.01 | jeffspeff | [TK]D-Fender, awesome... i'll try this out in just a moment, and let you know how it goes.... thank you ! |
| 19:00.13 | [TK]D-Fender | jeffspeff, You're welcome |
| 19:00.35 | pavlx | <apartment upstair> ---<switch> port 3 <device to call> |
| 19:01.40 | _omer | Fender: http://www.copypastecode.com/272135/ <---- is this folder Permission issue or missing any module ? |
| 19:02.21 | pavlx | necessity of ANONYMOUS traficc to call, can i use asterisk to call ? can they use the same operator VOIPSTUNT that i got with the same phone number ? |
| 19:03.16 | pavlx | _omer is a reply to me ? |
| 19:03.55 | _omer | no, question for [TK]D-Fender |
| 19:04.06 | drmessano | _omer, this is not |
| 19:04.09 | drmessano | _omer, this is not #freepbx |
| 19:04.40 | pavlx | ok, thanks _omer, next time specify the person |
| 19:08.06 | [TK]D-Fender | pavlx, He did |
| 19:08.27 | pavlx | what i did ? |
| 19:10.07 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:11.00 | pavlx | i need t solve this problem before September, whey Stefano turn back home with his family, and guys need to use ADSL and phone to call |
| 19:12.10 | pavlx | if i could use asterisk to phone and to config to my pleasure, will be money for me |
| 19:16.00 | drmessano | Im lost.. is this for a phonesex hotline? |
| 19:16.12 | jeffspeff | [TK]D-Fender, you are a godsend! you just saved me a ver long and what would have a been a pointless trip from Arkansas to Erie, PA. |
| 19:17.07 | pavlx | nobody can help me ? |
| 19:17.16 | jeffspeff | lol |
| 19:17.41 | *** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de) |
| 19:17.44 | dhuckaby | pavlx I'd try... but dunno how to filter all the scrollback efficiently to see what ya need ;) |
| 19:17.45 | jeffspeff | i want my phone to config my pleasure too! |
| 19:18.55 | drmessano | lol |
| 19:19.35 | pavlx | that it's possible to do in a second moment, the question is: can they use the same SIP and phone number to call from their own apartment ? |
| 19:20.07 | *** join/#asterisk twanny796 (~twanny@46.11.8.126) |
| 19:20.27 | dhuckaby | the same as you use? |
| 19:20.42 | drmessano | No reason why not |
| 19:20.58 | dhuckaby | just config their phones(soft or hard) to duplicate your settings |
| 19:21.07 | drmessano | It would be like running an extension upstairs |
| 19:21.15 | dhuckaby | and you can call yourself and it will call them :) |
| 19:21.25 | pavlx | yes, possibly, without to buy a second number from VOIPSTUNT |
| 19:21.25 | dhuckaby | or they can answer your phone calls :0 |
| 19:21.37 | drmessano | pavlx: YES |
| 19:22.06 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 19:22.08 | pavlx | ok |
| 19:22.34 | pavlx | point 2 solved |
| 19:23.22 | pavlx | asterisk can be used to call between people ? |
| 19:23.23 | *** join/#asterisk gusto (~gusto@ppp-62-216-207-38.dynamic.mnet-online.de) |
| 19:23.31 | drmessano | Uh yeah |
| 19:23.44 | drmessano | ~book |
| 19:23.45 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:23.45 | pavlx | is there a manual to read about ? |
| 19:23.49 | drmessano | Right there |
| 19:23.51 | drmessano | ^^^ |
| 19:24.11 | pavlx | thanks ! |
| 19:24.54 | pavlx | point 1 solved |
| 19:25.02 | dhuckaby | anything else ? :) |
| 19:25.10 | dhuckaby | smiles |
| 19:25.52 | pavlx | the more important how to masquerade, but at moment i see that seems don't be a solution |
| 19:26.11 | dhuckaby | how to masq which? NAT? |
| 19:27.13 | pavlx | to masq all the traffic and duration about the call/s |
| 19:28.14 | Gugge | pavlx: what are you talking about? |
| 19:28.24 | Katty | takes break from knitting |
| 19:29.11 | dhuckaby | no way to disguise duration of call if you're using their service |
| 19:29.21 | dhuckaby | nor the traffic... it presumably goes through their servers |
| 19:29.36 | drmessano | pavlx: You set up the second phone as another device on your system.. Like an OFFICE PBX.. WHen they make calls, they use the SAME SIP PROVIDER. All the calls come from YOUR ASTERISK. What is this masquerade crap? |
| 19:29.52 | pavlx | i don't want to show, people need privacy, ISP's are not SHERIFFS authorized to control the traffic |
| 19:30.06 | dhuckaby | he wants to hide the call volumne from the voip provider |
| 19:30.09 | drmessano | You can hide your call going over the public internet |
| 19:30.15 | drmessano | ERR |
| 19:30.21 | drmessano | You CANT hide your call going over the public internet |
| 19:30.46 | drmessano | Oh, I know what this is |
| 19:31.35 | dhuckaby | if it walks like a duck, and talks like a duck? :) |
| 19:31.58 | Katty | then it's had some vodka. |
| 19:31.59 | [TK]D-Fender | jeffspeff, Glad it's all good now. |
| 19:32.06 | Katty | drmessano: fresh knits! |
| 19:32.15 | drmessano | I am guessing this is SIP circumvension of the govt run telco |
| 19:32.16 | coppice | put in a pancake with some hoi sin sauce |
| 19:32.53 | dhuckaby | nomnom |
| 19:32.56 | drmessano | They find you running an ITSP in your closet and chop off your hands kinda thing |
| 19:33.08 | dhuckaby | orworse |
| 19:33.23 | drmessano | What could be worse? |
| 19:33.29 | drmessano | Death > No hands |
| 19:34.08 | dhuckaby | ^ |
| 19:34.13 | pavlx | i am working from 12/13 hours near the pc without stop and without to eat |
| 19:35.10 | dhuckaby | anyone use any soft-phones with video capabilities? |
| 19:36.07 | Katty | facetime :P |
| 19:37.20 | drmessano | dhuckaby: A few of them |
| 19:37.27 | drmessano | X-Lite, Bria, Ekiga |
| 19:37.42 | dhuckaby | I'm work'n X-lite atm... |
| 19:38.01 | *** join/#asterisk kannan (~kannan@123.238.229.157) |
| 19:38.11 | *** join/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 19:38.16 | dhuckaby | does it work decently? this machine has no web-cam |
| 19:39.13 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:39.56 | kannan | hello all. in a php agi script , i am setting channel variable from an external mysql server db column. the string gets truncated after the first space. how can this be resolved? (short of cron'ing another script to update the cdr column where that variable gets stored later) |
| 19:40.17 | kannan | i am using phpagi classes |
| 19:40.27 | Katty | drmessano: http://imgur.com/e9toL <- fresh knits. |
| 19:42.01 | dhuckaby | niftyo... what is it? |
| 19:42.06 | dhuckaby | baby sweater? |
| 19:42.11 | Katty | laptop sleeve, actually |
| 19:42.21 | dhuckaby | woo static? |
| 19:42.36 | Katty | http://3.bp.blogspot.com/_QMuDihPPyPA/TNdFLgvfkbI/AAAAAAAAOvE/HppNKa3C7xU/s1600/tardis.jpg |
| 19:42.44 | Katty | it's not wool yarn. |
| 19:43.05 | dhuckaby | okay that's gonna be an awesome laptop sleeve ;) |
| 19:43.16 | dhuckaby | tardis laptop sleeve.. you might just need to market those :) |
| 19:43.33 | Katty | oh no, i don't sell things. |
| 19:43.37 | Katty | i give them away as presents |
| 19:43.43 | malcolmd | soft client w/ video capabilities, try jitsi |
| 19:43.51 | dhuckaby | thanks malcolmd |
| 19:43.55 | malcolmd | np |
| 19:44.03 | malcolmd | it's cross-platform, which is nice |
| 19:44.36 | dhuckaby | yeah just saw that... very nice |
| 19:45.03 | dhuckaby | not that teachers in a high school need web-cams and video chat with each other.. but hey! |
| 19:47.17 | pavlx | i turned back now, i take my dinner, tomatoes |
| 19:47.37 | pavlx | the question is set up the second phone as another device |
| 19:47.48 | Roelt | Katty, looking nice! |
| 19:48.07 | Roelt | (And even asterisk related, phone in the background.. :-) ) |
| 19:49.05 | kannan | i am getting every couple seconds in * CLI -- Remote UNIX connection disconnected. it was not happeningpreviously. what is the issue? |
| 19:49.35 | pavlx | the question is how to set up the second phone as another device on my system like a PB in a office ? |
| 19:50.02 | dhuckaby | 2nd phone is just an extension |
| 19:50.37 | pavlx | PBX |
| 19:50.59 | dhuckaby | phone == either hardware or software sip phone no? |
| 19:51.24 | Katty | Roelt: ty (= |
| 19:53.11 | pavlx | this is the phone that i got at home: https://gigaset.com/hq/en/product/GIGASETA58H.html |
| 19:53.42 | dhuckaby | yeah, but a PHONE can't be a PBX |
| 19:55.04 | Roelt | you need a computer or router running asterisk as pbx, and the phones connect to that |
| 19:55.05 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 19:55.09 | WIMPy | dhuckaby: What does that mean? |
| 19:55.28 | dhuckaby | he want's a 2nd phone to be a pbx |
| 19:55.50 | WIMPy | And that's a cordless phone. They always contain a mini PBX. |
| 19:56.09 | dhuckaby | they merely allow multiple sip accts no? |
| 19:56.15 | pavlx | i got a router a cisco systems 2600 series, it needs only of vick card |
| 19:56.28 | WIMPy | They usually do. |
| 19:57.42 | *** join/#asterisk mcf37821 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 19:58.42 | pavlx | if necessary i buy a new one |
| 19:59.36 | pavlx | why they put a mini pbx ? |
| 20:00.17 | pavlx | why they use a firmware based on GNU/linux and don't distribute the source code ? |
| 20:00.21 | WIMPy | Because they will support multiple handsets. |
| 20:01.18 | WIMPy | I have no idea, what kind of OS they might be using. But Linux seems less likely to me. |
| 20:02.03 | pavlx | they would, but not are so stupid as they want, HUMAN is born to asks how to and they want the control of TLC ? |
| 20:02.50 | pavlx | i am using GNU/linux just now |
| 20:02.53 | WIMPy | cannot parse that. |
| 20:03.04 | dhuckaby | got lost too |
| 20:03.28 | dhuckaby | is still pretty sure they don't behave like he wants |
| 20:03.53 | WIMPy | Don't know what he wants. |
| 20:05.19 | pavlx | :~$ uname -o GNU/linux |
| 20:06.38 | drmessano | pavlx: Setting up a SIP device and an extension is covered in the book.... It is BASIC, BEGINNER stuff that is well documented. |
| 20:06.54 | drmessano | Check the book and follow the directions. |
| 20:07.28 | drmessano | This whole discussion is circular |
| 20:07.38 | pavlx | i use GNU/linux and the maximum that i could do it's NetBSD or OpenBSD because they support Alpha processors |
| 20:08.22 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 20:09.23 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:10.54 | Roelt | can't tell if trolling or just full moon |
| 20:11.05 | drmessano | Full moon, I hope |
| 20:11.32 | Katty | why not both? |
| 20:11.35 | WIMPy | There's almost no moon. |
| 20:11.40 | *** part/#asterisk dhuckaby (~dhuckaby@74-95-39-129-Oregon.hfc.comcastbusiness.net) |
| 20:11.41 | Katty | U MAD BRO?! |
| 20:12.17 | *** join/#asterisk dhuckaby (~dhuckaby@74-95-39-129-Oregon.hfc.comcastbusiness.net) |
| 20:13.55 | chuckf | I'm so mad I'm going home! |
| 20:16.50 | WIMPy | wonders if it would be a good idea to create a custom devstate for each account and type of forwarding. |
| 20:18.16 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 20:18.33 | WIMPy | Or per extension rather. |
| 20:20.31 | [TK]D-Fender | Can only be a good thing IMO |
| 20:23.04 | drmessano | exten => _X.,1,Dial(SIP/${EXTEN}) |
| 20:23.05 | drmessano | Done |
| 20:23.14 | pavlx | Debian moved the support to Alpha processors, and iam using a laptop ThinkPad T23 with GNU/linux gNewSense, an i686, a distribution 100% Free Software but is available at moment only for i686, other possbility could be GNU/linux Ututo, but anyway there is not suppoer for powerpc, for example support for MIPSEL is offered by GNU/linux Parabola |
| 20:23.44 | drmessano | Is that an advertisement? |
| 20:25.50 | pavlx | only the real problem that i find with other architectures, if i want i can use MIPSEL or if iwant more i have to use distribution GNU/linux which put disposition mirrors with restricted packages or i don't find what i want |
| 20:27.21 | pavlx | in the future i could be interested to MIPSEL, POWERPC, ALPHA and i don't want limits, FREEDOM ok, but this is "ghetto", i want to use more than one i686 |
| 20:28.08 | pavlx | for the fact that will be donated a i686 or ppc apple computer cupertino.inc |
| 20:29.26 | WIMPy | pavlx: What are you talking about? And is that related to anything? |
| 20:29.41 | coppice | I think its a bot |
| 20:30.00 | pavlx | coppice, me a bot ? |
| 20:36.46 | jpsharp | Someone mixed RMS and the Time Cube guy into a markov chainer. |
| 20:41.10 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 20:41.10 | drmessano | It could be a bot |
| 20:41.25 | drmessano | I never thought of that |
| 20:41.30 | drmessano | WHO IS YOUR MASTER? |
| 20:43.07 | drmessano | Isnt there interstellar law that states if you ask a robot for it's master it MUST tell you? |
| 20:44.53 | drmessano | ~whoisyourmaster |
| 21:05.03 | dhuckaby | laughs @drmessano |
| 21:07.05 | *** join/#asterisk BenC[UK]_ (~bcummins@cpc2-lock3-2-0-cust77.6-1.cable.virginmedia.com) |
| 21:17.01 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 21:18.59 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-157-214.dsl.stlsmo.sbcglobal.net) |
| 21:19.11 | LemensTS | what is the asterisk mysql realtime in yum for 1.6? |
| 21:19.28 | LemensTS | asterisk16-addons-mysql.x86_64 ? |
| 21:19.53 | LemensTS | or asterisk16-addons-core.x86_64 |
| 21:21.17 | LemensTS | nm it insalls core with it ... im a debian guy |
| 21:22.59 | WIMPy | Have you thought about using a more recent version? |
| 21:27.28 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 21:27.30 | ChrisInSydney | Mornin' all |
| 21:27.59 | WIMPy | In about half an hour :-) |
| 21:28.04 | blizzow | Is there a way to set the outbound callerID to be a number that's in the same area code my agents are calling to? |
| 21:28.07 | ChrisInSydney | :) |
| 21:28.30 | WIMPy | blizzow: Just do it. If your provider cares is another matter. |
| 21:28.49 | ChrisInSydney | blizzow: WIMPy beat me to it |
| 21:29.09 | blizzow | WIMPy: I'm asking how I'd do it. |
| 21:29.19 | ChrisInSydney | Set(CALLERID(num)=3215551234) |
| 21:30.05 | blizzow | okay, and the logic to make sure it's set to the area code of the number I'm calling? |
| 21:30.09 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:30.10 | ChrisInSydney | Some of our providers in Aus need an E164 number eg 61299998888 |
| 21:30.50 | WIMPy | blizzow: Look at what you're dialling and what numbers you've got. I'd do it with an AGI. |
| 21:31.06 | LemensTS | WIMPy: on a client system that doesn't care lol |
| 21:31.45 | ChrisInSydney | blizzow: Some providers need it on another header, P-Asserted-Identity, etc |
| 21:31.47 | WIMPy | ?? |
| 21:32.30 | ChrisInSydney | Had that issue with some wierd carrier route |
| 21:34.43 | ChrisInSydney | blizzow: Have a play with the dialplan and see if you can get it to work. After that, the logic can be done any number of ways, dial plan contexts, AGI, astDB |
| 21:35.37 | ChrisInSydney | just try from a single extension. Make sure your overstamp works |
| 21:36.07 | ChrisInSydney | if you cant, then anythng else is simply academic |
| 21:36.21 | WIMPy | What does "overstamp" mean? |
| 21:37.30 | ChrisInSydney | WIMPy: thats what we call changing the outbound CLID from the original number provided by the carrier |
| 21:38.16 | WIMPy | Always this incompatible terminology. :-( |
| 21:38.54 | ChrisInSydney | I had incompatible bass strings once |
| 21:39.01 | ChrisInSydney | not just limited to this stuff |
| 21:39.09 | ChrisInSydney | had an incompatable wife too |
| 21:39.28 | ChrisInSydney | that was disasterous |
| 21:39.57 | WIMPy | Ok, you win. |
| 21:40.09 | ChrisInSydney | She used to say to me, "you only ever want to have sex with me when you're drunk" |
| 21:40.28 | WIMPy | What did she look like? |
| 21:40.30 | ChrisInSydney | I replied "Sometimes I want a kebab when I am drunk" |
| 21:40.33 | WIMPy | scnr |
| 21:41.10 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 21:42.09 | ChrisInSydney | ive finished with that one. (last two comments), they are there to kae and adopt as your own for anyone who wants them |
| 21:42.22 | ChrisInSydney | take |
| 21:43.04 | ChrisInSydney | On a more serious note: Asterisk->ODBC->MySQL.... |
| 21:43.26 | WIMPy | I'm out |
| 21:44.16 | ChrisInSydney | ...to get some stuff from multiple tables am I better off using AGI, a stored procedure, or just crafting some SQL in func_odbc.conf ? |
| 21:45.03 | ChrisInSydney | Aim is to scale to 1000 calls / hr |
| 21:45.53 | WIMPy | Normal measure is per second |
| 21:46.15 | ChrisInSydney | Im in Australia, we are a little slower paced here ;-) |
| 21:46.55 | ChrisInSydney | not as lay back as Kiwis though |
| 21:47.32 | ChrisInSydney | does thet bot on this channel have some basiic math ? |
| 21:47.44 | ChrisInSydney | damn I key is playing up |
| 21:47.52 | ChrisInSydney | drag out the soldering iron again |
| 21:48.00 | drmessano | ?1000/60 |
| 21:48.04 | drmessano | Nope |
| 21:48.28 | ChrisInSydney | I could cut and paste from google |
| 21:48.32 | ChrisInSydney | use calc.exe |
| 21:48.47 | drmessano | Exe? |
| 21:48.50 | drmessano | :( |
| 21:49.10 | ChrisInSydney | thats only so I can get a shell |
| 21:49.21 | Roelt | hooray for /calc |
| 21:49.24 | Roelt | 23:49:14 -!- Irssi: 1000/60 = 16.6666666666667 |
| 21:49.35 | ChrisInSydney | yay |
| 21:50.01 | drmessano | Colloquy doesnt have a /calc |
| 21:50.05 | ChrisInSydney | still life hasnt sped up any here |
| 21:50.10 | ChrisInSydney | not that I've noticed |
| 21:50.55 | ChrisInSydney | 7:50am and no coffee, no wonder life is slow |
| 21:51.07 | Roelt | oh, tha's bad |
| 21:51.20 | ChrisInSydney | brb |
| 21:51.35 | Roelt | almost midnight here, and wine and solderfumes |
| 21:52.15 | drmessano | Mmmm solder fumes |
| 21:52.53 | drmessano | That and the smell of rubbing alcohol brings back memories |
| 21:53.17 | Roelt | for masking circuits probably? |
| 21:55.05 | drmessano | Cooling down RF connectors. |
| 21:55.53 | Roelt | ah, you talked about that 30 kW fm transmiter a few days ago :) |
| 21:56.08 | drmessano | 240 watt weller iron heats a connector VERY fast |
| 21:56.31 | drmessano | But you have to cool it before the dialetric melts |
| 21:56.31 | pyther | Is the obi100 a good adapter for an analog phone, or is there something better? Looking to integrate it into asterisks |
| 21:59.07 | dhuckaby | woo fm transmitter! |
| 22:02.05 | drmessano | 120 watt, not 240 watt. Had to look it up.. the SP120 |
| 22:02.15 | drmessano | Havent used it some time |
| 22:04.08 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-cebwsihxboonlxdh) |
| 22:04.48 | Roelt | hihi, nice :) |
| 22:05.17 | Roelt | also usable for soldering coper pipes :) |
| 22:07.26 | Roelt | weller wecp-20 soldering station here... don't know the model of the iron itself |
| 22:08.14 | drmessano | You know, I tried 1/2 inch copper pipes and it was taking way too long. Sort of surprised me. Ended up using a torch |
| 22:08.26 | drmessano | I _wanted_ it to work |
| 22:10.59 | drmessano | Nice soldering station. I cant say I have one with any sort of adjustment on it |
| 22:12.40 | ChrisInSydney | Roelt: Cool you can fix my <i> key |
| 22:13.42 | ChrisInSydney | Ive got an old '90s http://www.paceworldwide.com/products/conductive-rework-systems/mbt-250/mbt-250-sd- |
| 22:14.14 | *** join/#asterisk BenC[UK] (~bcummins@cpc2-lock3-2-0-cust77.6-1.cable.virginmedia.com) |
| 22:14.38 | ChrisInSydney | used to use it to swap out intel 80386sx with Cyrix 80486sl |
| 22:14.39 | drmessano | I have a couple WTCPT stations.. For the little bit of bench component work I do. I have my trusty Weller 8200 soldering gun that gets all the work |
| 22:14.58 | ChrisInSydney | the wellers just go and go and go |
| 22:15.27 | drmessano | Yep |
| 22:15.51 | drmessano | Sadly, I dont have one of the older style 8200's |
| 22:15.51 | ChrisInSydney | the Pace is a damn Ferrari. Unbelievable, but costs a bomb to run in bits, tips, filters etc |
| 22:16.04 | coppice | I'd rather use a Hakko than a Weller |
| 22:16.54 | Roelt | ChrisInSydney, i can fix you I key.. just bring it overhere when you're in the neighbourhood :) |
| 22:17.04 | ChrisInSydney | :D |
| 22:17.07 | WIMPy | This is getting historic. |
| 22:17.21 | WIMPy | Ersa vs Weller used to be the same as emacs vs vi. |
| 22:17.37 | drmessano | The newer weller guns dont have a brass nut that holds the tip down, they have this steel ring with a set screw in it. The screws strip out in about a week and need to be tapped and retapped |
| 22:18.56 | ChrisInSydney | does anyone bother these days with surface mount rework ? |
| 22:19.16 | WIMPy | I ordered some el cheapo iron for on the way at dx for IIRC 7 bucks. Pretty impressive thing. |
| 22:19.21 | coppice | strange question. of course they do |
| 22:19.28 | ChrisInSydney | I use mine for keeping my old amps and music gear working |
| 22:19.34 | ChrisInSydney | all through hold |
| 22:19.36 | ChrisInSydney | hole |
| 22:19.40 | WIMPy | Temperature controlled and the tip was finer than that of my station. |
| 22:20.38 | WIMPy | That reminds me that maybe I should check the capacitors in my amp again. |
| 22:20.38 | drmessano | I cant do surface mount work. Dont have the eyes or patience for it |
| 22:21.18 | Roelt | i've done smd.. but after 4 resistors my patience ends and a restistor and cap are the same.. |
| 22:21.35 | ChrisInSydney | coppice: I guess I asked as the cost / complexity ratio has got to the point where I Iusually couldnt be bothered |
| 22:21.43 | WIMPy | You always need to be prepared for a total loss. |
| 22:22.28 | coppice | ChrisInSydney: lots of stuff needs customising, and development work always needs rework |
| 22:23.38 | WIMPy | Ok, this is probably the best time to ask: Have any of you ever managed to repair flexband connections on LCDs? |
| 22:24.13 | WIMPy | The one where you have a band instead of the zebra rubber beteen the PCB and the glass. The gule seems to fail after some time :-( |
| 22:24.18 | coppice | I've seen people so that with a heat sealer device |
| 22:24.50 | WIMPy | I read about that, but the stuff I have here doesn't seem to react to heat. |
| 22:26.11 | WIMPy | I read about filing down pencils to get graphite dust and mixing them with silicone. But I'm not yet convinced I should try that. |
| 22:29.19 | WIMPy | Maybe I should just try to connect some other LCD, but I have doubts they a Hitachi compatible. |
| 22:32.28 | ChrisInSydney | WIMPy: I used to fix them on Casio and Yamaha keyboards. Had an old temp controlled soldering iron that was good town to 60 Celcuis |
| 22:32.42 | dhuckaby | fear.. |
| 22:32.57 | ChrisInSydney | I had a chisel tip that was almost the correct spacing |
| 22:33.44 | ChrisInSydney | I have seen the heat strips for sale |
| 22:39.42 | pyther | Can anyone recommend an ATA? |
| 22:44.09 | WIMPy | Well, as I said: It didn't react to heat. the heat seemed to have cleaned the contacts, as applying pressure helped where it didn't before, but it doesn't stick any more. |
| 22:45.19 | ChrisInSydney | pyther: The Sipura/Linksys/Cisco McSPA3102 |
| 22:45.59 | ChrisInSydney | Like McDonalds, there are much better ATAs but when youve been travelling for hours and the kids are screaming in the back of the car..... |
| 22:46.35 | ChrisInSydney | has anyone played with the newer Cisco branded 118s etc ? |
| 22:46.39 | WIMPy | :-) |
| 22:46.52 | pyther | I just need something very simple that can connect one analog phone |
| 22:47.00 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-18-80.bois.qwest.net) |
| 22:47.07 | ChrisInSydney | did you need a line as well ? |
| 22:47.29 | ChrisInSydney | PAP2T are < $30 on feeBay and have 2 x FXS |
| 22:47.29 | WIMPy | And you don't have any plastic router with ATA? |
| 22:47.50 | WIMPy | has got a lot of them, even tho I never needed one. |
| 22:47.55 | pyther | I'm using callcentric as my voip provider, if I understand what you mean by line, that I don't need one |
| 22:47.59 | WIMPy | They are like AOL CDs used to be. |
| 22:48.04 | ChrisInSydney | SPA3102 have FXS and FXO |
| 22:48.13 | ChrisInSydney | :-) |
| 22:49.40 | pyther | Would the cisco SPA112 work? |
| 22:49.52 | ChrisInSydney | yup |
| 22:50.04 | pyther | awesome |
| 22:50.46 | pyther | So if I want to have 2-3 extensions and no more than one call taking place at at time, would 128MB and 600MHz be enough to run asterisk? |
| 22:50.53 | pyther | basically on an openwrt router |
| 22:51.05 | ChrisInSydney | no reg between Ast and the SPA, just IP based entry in sip.conf |
| 22:51.17 | ChrisInSydney | yup, no transcoding |
| 22:51.21 | ChrisInSydney | please |
| 22:51.35 | pyther | I don't want to transcode anything |
| 22:51.43 | WIMPy | pyther: Easily |
| 22:51.57 | pyther | great now I do need a sip account for each device I'm going to use right? |
| 22:52.03 | *** join/#asterisk Alex_Bkash (b4eac135@gateway/web/freenode/ip.180.234.193.53) |
| 22:52.07 | WIMPy | yes |
| 22:52.22 | ChrisInSydney | on the ast box not the carrier |
| 22:52.38 | WIMPy | yes |
| 22:52.54 | WIMPy | Or ou leave out Asterisk. |
| 22:52.56 | pyther | and then I can setup a hunt group (is that the right term) to ring all the devices? |
| 22:53.21 | WIMPy | You can do whatever you like. |
| 22:53.52 | pyther | well I'd like to ring all devices, is that the general idea? |
| 22:53.56 | WIMPy | We don't really have groups, just call a list of devices. |
| 22:54.28 | WIMPy | Then list them all. |
| 22:56.29 | ChrisInSydney | pyther: You'll need Asterisk, as I couldn't get the old Milkfish OpenSER implmentatoin to do a ring many |
| 22:57.20 | ChrisInSydney | pyther: Is it a non intel chip you are compiling against ? |
| 22:57.27 | ChrisInSydney | (for ?) |
| 22:57.39 | pyther | ChrisInSydney: yes, but the binaries are already built for openwrt |
| 22:57.46 | ChrisInSydney | ahh |
| 22:57.57 | pyther | 1.8.7 |
| 22:58.22 | ChrisInSydney | cool |
| 22:59.05 | ChrisInSydney | havent got too much experience with OpenWRT, but done a bit with DD |
| 23:00.10 | ChrisInSydney | Was looking at a small Asterisk implimentation as a B2BUA for some SIP trunks to a 3rd party PBX |
| 23:00.36 | ChrisInSydney | on broadcom K26 compatable systems |
| 23:00.53 | WIMPy | Not the RPi? |
| 23:01.40 | pyther | WIMPy: if that is directed to me, because I have the openwrt router (it is main router) and the rpi doen't have serial |
| 23:01.53 | ChrisInSydney | ahh, must go. The 2.75 Year old has arisen and Yummy Mummy has gone away for a weekend with the girls |
| 23:01.54 | WIMPy | No to ChrisInSydney |
| 23:02.17 | pyther | ChrisInSydney: thanks for the help |
| 23:02.21 | ChrisInSydney | cool |
| 23:02.38 | ChrisInSydney | WIMPy: rpi ? |
| 23:02.53 | WIMPy | Raspberry Pi |
| 23:02.57 | ChrisInSydney | ahh |
| 23:03.01 | ChrisInSydney | no, unfortunately |
| 23:03.21 | WIMPy | Thought you have to use that for everythign now :-) |
| 23:03.21 | ChrisInSydney | It is sort of a must have that |
| 23:03.35 | ChrisInSydney | Almost |
| 23:03.46 | WIMPy | Well the value for money is excellent. |
| 23:03.59 | ChrisInSydney | cluster them |
| 23:04.30 | WIMPy | But It's too big for embedded applications and a little thin as a PC replacement. |
| 23:04.34 | ChrisInSydney | ahh, he is starting to climb the bookshelf |
| 23:04.49 | ChrisInSydney | 2.75 year old |
| 23:05.43 | TomCat2 | I'm considering renting an Asterisk server instead of hosting my own in my home office - I've found several - rentpbx.com has good pricing for example - before I commit to using them, does anyone have a solid recommendation? |
| 23:06.15 | TomCat2 | not looking for much - just a box I can reliably run asterisk on and manage it myself |
| 23:06.42 | TomCat2 | and of course, solid uptime and no oddball network issues |
| 23:07.31 | ChrisInSydney | TomCat2: I grabbed a linode.com and stuck asterisk on it for a lark and it worked a treat |
| 23:07.36 | ChrisInSydney | $20/mo |
| 23:07.43 | ChrisInSydney | 200Gb xfer |
| 23:08.23 | pyther | I don't use lindoe for asterisk, but I do have a linode instance for email and it has been very reliable (granted I've only had it 2 months) |
| 23:08.38 | ChrisInSydney | time to feed the family |
| 23:08.41 | ChrisInSydney | must go |
| 23:08.45 | ChrisInSydney | c 'yall |
| 23:11.38 | TomCat2 | nice - avg. 29ms ping to linode |
| 23:13.36 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 23:25.02 | *** join/#asterisk Happy0 (~Happy0@cpc2-broo8-2-0-cust282.14-2.cable.virginmedia.com) |
| 23:38.31 | *** join/#asterisk bmg505 (~leon@196-209-101-227.dynamic.isadsl.co.za) |
| 23:46.25 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 23:53.41 | jpsharp | I used linode for Asterisk for quite sometime. It worked well. |
