| 00:00.02 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 00:00.04 | snadge | <3 |
| 00:00.47 | WIMPy | A gret opportunity for just splitting out one priority. |
| 00:01.11 | WIMPy | Although that can arguably make reading the dialplan harder. |
| 00:01.16 | WIMPy | great |
| 00:03.14 | runfromnowhere | Thanks for the help and support all :) |
| 00:07.28 | danfromuk | WIMPy: no, i'd disconnected because i left the client's office. |
| 00:07.42 | danfromuk | WIMPy: what did you say about mlayer3? |
| 00:08.33 | WIMPy | I asked if maybe you have multiple versions of that installed. |
| 00:08.59 | danfromuk | I think if i get time, i'll wipe it and start again without freepbx. |
| 00:09.10 | danfromuk | however now, the client's decided to stick with asterisk 1.4 |
| 00:09.18 | WIMPy | You obviousely had some v1 stuff on there. So my guess is that it found a version of that file belonging to v1. |
| 00:10.25 | danfromuk | I agree. |
| 00:12.55 | danfromuk | Its been a long day. Thanks for your help. I'm signing off. |
| 00:13.04 | danfromuk | It was greatly appreciated. |
| 00:16.06 | gartral | my brain hurts. I have a user who's on a different network, and she can hear me when I call, but she can't initiate a call herself nor can I hear her if I call her.. when she tries to call her client says "Loop detected" without further info |
| 00:17.49 | gartral | as a side note, if I want to detect a specific phone number and do something different with it, how would I do this? |
| 00:18.48 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 00:18.57 | WIMPy | What number? |
| 00:19.52 | gartral | WIMPy: it's a PSTN number 216 area code, I kinda don't want to give you the rest of it :P |
| 00:21.59 | WIMPy | What kind of number? |
| 00:22.16 | WIMPy | The dialled one? The caller ID? Something else? |
| 00:24.04 | WIMPy | Or was that part of the change caller ID one? |
| 00:24.20 | *** join/#asterisk s7r (~s7r@openvpn/user/s7r) |
| 00:24.22 | gartral | WIMPy: it's actually my door of my building.. when someone dials my unit number it initiates a phone call to my gvoice, I want to pick out just the door's phone number and have it bypass my "standard" dialplan.. make it ring 2 extensions, as well as call out of another trunk to my cell phone |
| 00:25.24 | *** join/#asterisk newtonr (~newtonr@173-17-135-67.client.mchsi.com) |
| 00:25.24 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 00:25.25 | WIMPy | Ok, then look at the /callerID option for extensions. |
| 00:26.27 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 00:26.27 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:31.00 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 00:48.12 | *** join/#asterisk suneye (~atcmmi@119.139.62.75) |
| 01:06.44 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 01:10.19 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.125.117) |
| 01:11.44 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 01:16.19 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 01:18.08 | *** join/#asterisk xzarth_ (~krikkit@dh207-25-60.xnet.hr) |
| 01:19.59 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 01:22.53 | *** join/#asterisk SGjunior (~sgjunior@modemcable066.0-70-69.static.videotron.ca) |
| 01:35.56 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 01:51.36 | *** join/#asterisk gartral (~gartral@unaffiliated/gartral) |
| 02:04.04 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 02:11.53 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 02:19.12 | *** join/#asterisk felipealmeida (~user@177.159.41.94) |
| 02:26.04 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 02:28.04 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:31.31 | gartral | ok, is there any issues with trying to have a client connect to my asterisk server if they run one from their local network? |
| 02:46.31 | newtonr | gartral, a lot of info lacking there for us to be of much help. Short answer is no. Long answer probably involves a few yes'es, depending on what you mean by connect, what each asterisk instance is doing, what ports and IPs things are running on, how the networks are setup, on and on and on. |
| 02:47.05 | newtonr | I'm unfortunately logging off to go to sleep. If you provide more information , perhaps someone else will be able to help you out! |
| 02:47.07 | newtonr | later! |
| 03:04.51 | *** part/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 03:22.52 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.195) |
| 03:38.42 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.127.199) |
| 03:44.38 | ChannelZ | gartral: no |
| 04:04.33 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 04:04.36 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 04:05.42 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 04:12.13 | *** join/#asterisk Defraz (~Defraz@24-116-129-19.cpe.cableone.net) |
| 04:35.50 | *** join/#asterisk the_5th_wheel (~edd@105-237-91-106.access.mtnbusiness.co.za) |
| 04:43.06 | *** join/#asterisk atcmmi (~atcmmi@116.7.102.29) |
| 04:54.46 | *** join/#asterisk Agrajag- (~filip@c211-30-195-162.artrmn3.nsw.optusnet.com.au) |
| 04:58.03 | Penguin | seri: Are you around today? |
| 05:00.11 | Agrajag- | g'day, i'm having a problem where i can't see why the command after Dial() isn't being executed. very simple example: http://pastebin.com/a378AnW0 - it doesn't get to the 2nd NoOp. i just see "Spawn extension (sipincoming, 101, 2) exited non-zero on 'SIP/foo-0000000a'" as the last thing in debug |
| 05:00.26 | Penguin | The caller hung up. |
| 05:00.46 | Agrajag- | ok yeah - i thought i would get NOANSWER from that because that's what i'm seeing in cdr |
| 05:00.57 | Agrajag- | is it possible to catch that? |
| 05:01.23 | Penguin | If the caller hangs up, the call ends. No questions asked. |
| 05:01.50 | Penguin | You can see the DIALSTATUS in the h extenstion after the call ends. |
| 05:02.04 | Agrajag- | ok, i'll check that out. thanks |
| 05:02.10 | Penguin | That's probably what you want. |
| 05:02.27 | Agrajag- | yep sounds like it, thanks |
| 05:02.39 | Penguin | In order for that second NoOp() to run, the caller would have to stay on the line longer than the Dial() timeout value. |
| 05:06.46 | [TK]D-Fender | exten => 101,1,NoOp(Call from ${CALLERID}) |
| 05:06.56 | [TK]D-Fender | and that variable hasn't been proper to use since 1.0 |
| 05:07.57 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 05:08.27 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 05:18.59 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 05:24.15 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.94) |
| 05:35.16 | *** join/#asterisk hos7ein (~chatzilla@91.98.33.208) |
| 05:37.03 | *** join/#asterisk s7r (~s7r@openvpn/user/s7r) |
| 05:58.50 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-wfttiohlbedgihmw) |
| 05:59.49 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.126.118) |
| 06:17.53 | *** join/#asterisk ivan` (~ivan@unaffiliated/ivan/x-000001) |
| 06:19.10 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 06:21.29 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.118) |
| 06:25.27 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.166.212.65) |
| 06:41.17 | *** join/#asterisk CeBe (~CeBe@port-92-206-87-240.dynamic.qsc.de) |
| 06:45.41 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:21b:63ff:fe31:8426) |
| 06:47.03 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:21b:63ff:fe31:8426) |
| 06:58.51 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.118) |
| 07:00.50 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 07:15.28 | *** join/#asterisk magespawn (~Eames@105-236-71-217.access.mtnbusiness.co.za) |
| 07:15.34 | magespawn | good day |
| 07:16.31 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 07:16.57 | bkruse | ello |
| 07:26.00 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 07:45.50 | *** join/#asterisk mirela666 (~mirko.bra@93-87-217-142.dynamic.isp.telekom.rs) |
| 07:52.03 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 08:02.37 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 08:05.03 | *** join/#asterisk gerritfromsa (~gerritfro@8ta-229-161-156.telkomadsl.co.za) |
| 08:05.21 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 08:06.57 | *** join/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 08:23.25 | *** join/#asterisk kresp0 (~kresp0@gateway/tor-sasl/kresp0) |
| 08:33.33 | *** join/#asterisk v0lZy (~Thunderbi@84-255-194-41.static.t-2.net) |
| 08:36.59 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:38.31 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:49.15 | *** join/#asterisk Tokeiito (~quassel@main.kbi.lt) |
| 08:59.50 | *** join/#asterisk danjenkins (~danjenkin@80.1.94.250) |
| 09:05.17 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 09:13.41 | *** join/#asterisk atha (~athayde@unaffiliated/athayde) |
| 09:22.49 | *** join/#asterisk suneye (~atcmmi@121.34.42.101) |
| 09:26.25 | *** join/#asterisk ipalmer (~IceChat77@host81-133-140-79.in-addr.btopenworld.com) |
| 09:27.10 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.60) |
| 09:28.22 | ipalmer | Hi all, I'm having a problem where chanspy keeps locking the SIP channel which used chanspy anyone know how to drop the channel 'channel request hangup' doesn't work |
| 09:31.02 | gerritfromsa | I think I saw something like this last night - if I remember correctly is was a version problem - try up/download your version to fix |
| 09:32.37 | ipalmer | gerritfromsa: apologies were you talking to me |
| 09:36.05 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 09:38.14 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:38.16 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:41.40 | *** join/#asterisk yang (yang@freenode/sponsor/fsf.member.yang) |
| 09:49.06 | *** join/#asterisk danjenkins (~danjenkin@80.1.94.250) |
| 09:51.34 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 09:52.35 | gerritfromsa | ipalmer, yes |
| 10:00.07 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 10:02.30 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 10:06.44 | ipalmer | gerritfromsa: thanks, I'm currently running 1.8.5 read some changelogs for 11.6 and there does appear to be some stuff added for chanspy so will look at updating cheers |
| 10:07.37 | bulkorok | hi |
| 10:21.09 | *** join/#asterisk barbosa2 (~juliano.b@177-069-248-123.static.ctbctelecom.com.br) |
| 10:29.32 | gerritfromsa | hi |
| 10:31.40 | *** join/#asterisk atha (~athayde@unaffiliated/athayde) |
| 10:34.24 | SeRi | ipalmer: you are way behind. You dont need to jump to 11.6 to fix your issue. You can stay on the 1.8.x branch if you want too. the latest is 1.8.24.0 |
| 10:35.03 | gerritfromsa | SeRi, I agree with this 100% |
| 10:35.26 | SeRi | 1.8.x is an LTS branch. I am sticking with it... |
| 10:36.36 | ipalmer | SeRi: really, this was a common issue before that version then? |
| 10:37.09 | gerritfromsa | Early 1.8's got quite a few issues |
| 10:37.40 | gerritfromsa | I also found Queue issues in 1.8.4 and 1.8.7 |
| 10:37.54 | SeRi | I use chanspy and I have no problems so far. Though I didnt use chan spy back than. |
| 10:39.18 | gerritfromsa | Using ChanSpy() via the dialplan or AMI? |
| 10:39.36 | ipalmer | Excellent, an upgrade from 1.8.5 to 1.8.24 doesn't seem so drastic. Think I'll do as advised. |
| 10:40.05 | SeRi | ipalmer: Good luck. |
| 10:55.43 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 10:56.25 | eduzimrs | Hi guys, anyone here use Khomp E1 devices ? |
| 10:58.38 | gerritfromsa | Never heard of Khomp - are they PCI cards / Gateways ? |
| 10:58.41 | eduzimrs | Guys im getting an issue with * 1.6.2.13 related to using of memory, after about 15 days of uptime, * consumes all free memory from linux e stop delivering calls |
| 10:59.05 | Greenlight | ~upgrade |
| 10:59.05 | infobot | Upgrading is easy! Go that way, really fast. If something gets in your way, turn. |
| 10:59.10 | Greenlight | ~update |
| 10:59.10 | infobot | hmm... update is http://conversations.nokia.com/2012/07/11/nokia-n9-software-update-pr1-3/ |
| 10:59.26 | Greenlight | Hmm |
| 10:59.40 | Greenlight | eduzimrs: Update to a current, and supported build. |
| 11:00.50 | eduzimrs | Greenlight, its quite a problem, all my hole system is based on that version, i cant upgrade right now, instead i must find an workaround |
| 11:01.40 | *** join/#asterisk danjenkins (~danjenkin@80.1.94.250) |
| 11:01.43 | gerritfromsa | eduzimrs, even so an upgrade wont affect it negatively |
| 11:02.16 | Greenlight | It's very difficult for people to help you when you're using an outdated version, and your issue has likely been fixed in recent builds. |
| 11:02.51 | Greenlight | It's like me asking for help getting my new usb 3g dopngle working on windows 3.1 |
| 11:03.27 | eduzimrs | gerritfromsa, ive been reading about changelog, i notice many changes in dialplan syntax, wich would cause some problems at CRM |
| 11:04.05 | eduzimrs | Greenlight, do u know in what version it has been correct ? |
| 11:04.12 | eduzimrs | Greenlight, do u know in what version it has been corrected ? |
| 11:04.54 | Greenlight | No, i've not looked through the thousands of bug fixes made in the last 4 years to see which might apply. |
| 11:05.05 | gerritfromsa | eduzimrs, I understand , can you give me an example ? |
| 11:11.33 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 11:11.35 | *** join/#asterisk Sjors (~sgielen@foo.kassala.de) |
| 11:13.30 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 11:33.11 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 11:35.33 | *** join/#asterisk vlad_starkov (~vlad_star@83.69.245.162) |
| 11:51.36 | davlefouAMD | Bonjour, i have an problem with ovh and asterisk, my sip is ok with Asterisk 1.8.8.0~rc4-1digium0+1~lucid but not with Asterisk 1.8.10.1~dfsg-1ubuntu1 built by buildd @ yellow on a x86_64 |
| 11:51.55 | davlefouAMD | i use the same configuration, |
| 11:52.13 | Greenlight | If you can elaborate on "ok" |
| 11:52.29 | Greenlight | Specifically what make the second build, "not ok" |
| 11:52.34 | davlefouAMD | i have that message : timed out, trying again |
| 11:52.57 | Greenlight | WHere do you see this message ? |
| 11:57.23 | *** join/#asterisk ivan` (~ivan@unaffiliated/ivan/x-000001) |
| 11:57.52 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 11:58.42 | davlefouAMD | in the cli |
| 12:00.11 | davlefouAMD | and when i sip reload i have that : No valid transports available, falling back to 'udp'. |
| 12:15.54 | *** join/#asterisk martinfletcher_ (~martinfle@87.237.70.109) |
| 12:24.17 | *** join/#asterisk zigg (~matt@unaffiliated/zigg) |
| 12:26.57 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:28.34 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 12:31.24 | *** join/#asterisk Pullphinger (~Pullphing@12.40.23.68) |
| 12:34.38 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 12:49.37 | *** join/#asterisk wdoekes (~walter@wjd.osso.nl) |
| 12:49.49 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 12:49.55 | *** join/#asterisk danjenkins (~danjenkin@80.1.94.250) |
| 12:57.39 | davlefouAMD | hi, |
| 12:58.35 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 12:58.37 | davlefouAMD | i a use hardware 729 like spa 112 sound a very good, but with asterisk g729, it an very very bad. Why? |
| 12:59.52 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 13:00.48 | [TK]D-Fender | davlefouAMD: your description is too vague |
| 13:06.01 | davlefouAMD | [TK]D-Fender, sflphone <-ulaw local->asterik/g729<-->ovh or spa/g729<->asterisk<->ovh, in the first case, sound cut all the time, it is difficulte to undestant, in the second case, it ok |
| 13:06.17 | Penguin | davlefouamd: Set "transport=udp" in sip.conf to avoid that notice. |
| 13:06.54 | [TK]D-Fender | davlefouAMD: maybe your SFLphone is bad |
| 13:09.00 | cusco | hi.. can ChanSpy(<prefix>) be a pattern? |
| 13:09.12 | cusco | say.. SIP/[0-9] |
| 13:09.21 | [TK]D-Fender | no |
| 13:09.26 | cusco | ok |
| 13:10.14 | Greenlight | Although you can use a variable, and then limit to a prefix that way |
| 13:10.42 | Greenlight | exten => _XXX,1,ChanSpy(SIP/${EXTEN}) |
| 13:11.40 | [TK]D-Fender | not what he's asking for... |
| 13:11.58 | [TK]D-Fender | Greenlight: he wants the targets to accept a range, not that he'd have to specify |
| 13:12.07 | [TK]D-Fender | (by what he dials) |
| 13:12.42 | Greenlight | Yea, but say he wanted the SIP/1XX range... |
| 13:13.08 | Greenlight | exten => _X,1,ChanSpy(SIP/${EXTEN}) |
| 13:13.52 | Greenlight | That would allow him to use a pattern. |
| 13:14.01 | Penguin | That would accept 10 or 199999999999999999999. Not really a good pattern for using 1XX extens. |
| 13:14.42 | [TK]D-Fender | Greenlight: no, he only asked to spy on a range... not accept a specific item IN a range. |
| 13:15.32 | Greenlight | ChanSpy takes a prefix, doesn't it. So if he had 3 dig extensions, and wanted to spy on any starting with 1, eg 100,101, 102, then what I suggested would work. |
| 13:15.50 | [TK]D-Fender | Greenlight: I want to spy across all SIP/1XX is not the same as I want to spy one THIS one specific guy that I dial specifically |
| 13:15.56 | Penguin | But by your suggesteion, he'd have to dial those numbers. |
| 13:16.17 | Penguin | He's not asking to dial each exact thing to spy on. He asked to spy on a range. |
| 13:17.08 | Greenlight | What I suggested would spy on a RANGE not a specific extension. |
| 13:17.16 | Penguin | If you use a prefix of SIP/1, then you wouldn't get anything other than numbers starting with 1. |
| 13:17.24 | Penguin | He asked for starting to 0 through 9. |
| 13:17.26 | Greenlight | Yea, the 101,102,103 range. |
| 13:18.13 | *** join/#asterisk redotis (~redotis@nc-76-5-176-123.dhcp.embarqhsd.net) |
| 13:18.50 | cusco | ah thanks Greenlight |
| 13:18.55 | cusco | err |
| 13:18.55 | Greenlight | It may not be exactly what he's looking for, but it's a suggestion that may fit his needs. |
| 13:19.04 | Greenlight | It may not, but that's life :) |
| 13:19.06 | Penguin | A prefix of SIP/1 would exclude all the ones starting with 0, 2, 3, 4, 5, 6, 7, 8, and 9. |
| 13:19.15 | cusco | I would like to spy channels starting with sip/digits |
| 13:19.40 | Greenlight | Ahh, then yea, what I suggested wouldn't help. |
| 13:20.00 | cusco | when I spy all SIP it includes teh gateway (sip/gateway) |
| 13:20.13 | [TK]D-Fender | [09:17]GreenlightWhat I suggested would spy on a RANGE not a specific extension. <- no, he has to dial that specific number, instead of pressing some generic feature and getting a Chanspy-limited choice of range |
| 13:20.14 | cusco | so when I cycle channels, I will get twice the same channel (when bridged) |
| 13:20.40 | Greenlight | You can use a group can't you |
| 13:20.46 | Greenlight | If memory serves |
| 13:20.52 | cusco | yes |
| 13:20.53 | cusco | sure |
| 13:21.02 | cusco | yes thing is, I'm already definyng spygroup .. well I'll have a look at that |
| 13:21.05 | *** join/#asterisk BorjaGVO (d51beb92@gateway/web/freenode/ip.213.27.235.146) |
| 13:21.07 | BorjaGVO | Hi. I'm having issues trying to distribute device_states. I'm using Asterisk 1.8.23.0, Openfire 3.8.2. This is the error I'm getting: [2013-10-31 13:48:28] ERROR[25319]: res_jabber.c:3520 aji_handle_pubsub_error: Error performing operation on PubSub node device_state, 403.. You can find my config files in http://pastebin.com/v6Sb7tba |
| 13:21.33 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 13:22.13 | Greenlight | You're passing the "g" option to ChanSpy ? |
| 13:24.07 | Penguin | That could work. |
| 13:24.21 | Greenlight | thinks he just accidentally stumbled on a way to listen to anyones voicemail messages on the UK network three |
| 13:24.27 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 13:25.07 | Penguin | Set the SPYGROUP=something on every device that you want to spy on, don't set it on peers that you don't want to spy on. Then ChanSpy(SIP,g(something)). |
| 13:25.54 | Greenlight | It seems that if you call a Three mobile number, and you are *presenting that number as your outbound CLI* and it goes to voicemail, you get directed to the main voicemail menu with no security... |
| 13:26.14 | Greenlight | I can change PIN, listen to messages, everything. |
| 13:26.47 | Penguin | That's pretty common. |
| 13:26.59 | Penguin | At least it used to be with Cingular/AT&T in the US. |
| 13:27.03 | Greenlight | Seems quite a security flaw, no ? |
| 13:27.21 | Penguin | Verizon always had passwd security enabled by default. |
| 13:27.39 | Greenlight | It seems the system is thinking that it's the user calling, based purely on the CLI |
| 13:27.45 | Penguin | Oh yeah it is. It caused a lot of people to lose their privacy. |
| 13:28.18 | Greenlight | I know we had a big fiasco in the press over here, but that was to do with people not setting a default PIN. This completely circumvents that. |
| 13:28.45 | cusco | yes thing is I already use spygroup for another extension where chanspy is ran |
| 13:28.56 | cusco | thanks for the feedback |
| 13:30.57 | *** join/#asterisk makmak78 (~makmak78@195-67-63-194.customer.telia.com) |
| 13:31.54 | Greenlight | I wonder what other networks are afflicated by similar lax security. |
| 13:45.38 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:45.38 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:47.05 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 13:48.04 | *** join/#asterisk eloycoto (~eloycoto@159.179.60.213.static.mundo-r.com) |
| 13:55.51 | *** join/#asterisk CeBe (~CeBe@port-92-206-87-240.dynamic.qsc.de) |
| 13:59.07 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-tsxnufynmuxfdaxj) |
| 13:59.07 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:03.58 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-qkrvidkswvirxgvg) |
| 14:03.58 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:09.30 | *** join/#asterisk felipealmeida (~user@139.82.86.17) |
| 14:10.20 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:10.21 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:12.59 | BorjaGVO | Hi. I'm having issues trying to distribute device_states. I'm using Asterisk 1.8.23.0, Openfire 3.8.2. This is the error I'm getting: [2013-10-31 13:48:28] ERROR[25319]: res_jabber.c:3520 aji_handle_pubsub_error: Error performing operation on PubSub node device_state, 403.. You can find my config files in http://pastebin.com/v6Sb7tba |
| 14:13.34 | *** join/#asterisk gerritfromsa (~gerritfro@8ta-229-161-156.telkomadsl.co.za) |
| 14:14.42 | *** join/#asterisk danjenkins (~danjenkin@80.1.94.250) |
| 14:23.34 | *** part/#asterisk danjenkins (~danjenkin@80.1.94.250) |
| 14:32.26 | *** join/#asterisk MrQuist (~Peter@83.232.96.217) |
| 14:39.33 | *** join/#asterisk kresp0 (~kresp0@gateway/tor-sasl/kresp0) |
| 14:41.15 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 14:44.00 | *** join/#asterisk gtjoseph (~gtj@unaffiliated/gtj) |
| 14:47.41 | *** join/#asterisk Elleni (3ec00582@gateway/web/freenode/ip.62.192.5.130) |
| 14:48.47 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 14:49.15 | Elleni | Hi all, my asterisk server 11-5-1 configured with freepbx 2.11.0.6 works as expected but after ~48 hours suddenly no calls are possible anymore with following error: http://pastebin.com/hYnQi7GH after a reboot its ok again for another 48h. anyone an idea from what that could come or how I could find out how to fix? |
| 14:49.56 | *** join/#asterisk Cloin (~Colin@pool-173-79-237-246.washdc.fios.verizon.net) |
| 14:50.02 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:1e8:347f:e2b1:bedc) |
| 14:51.21 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 14:52.06 | ipengineer | Hello! Does anyone have any ideas why I would be getting this message when starting Asterisk? The file exist.. ERROR[31334] media_index.c: Failed to stat /var/lib/asterisk/sounds/en/conf-onlyperson.gsm |
| 14:54.53 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 14:55.08 | mjordan | ipengineer: most likely permissions. Asterisk may not have the proper permissions to execute stat on that file. |
| 14:56.14 | mjordan | Elleni: that error would not be time based. It wouldn't work for some amount of time and then fail. Some phone is sending you an INVITE request and indicating that they want secure audio, but is not providing any crypto keys. |
| 14:56.58 | ipengineer | mjordan: I was thinking that but I am starting Asterisk from root and safe_asterisk is running as root. Cant tell about asterisk itself because it dies. Root has all permissions on that directory |
| 14:57.48 | Elleni | The thing is all softphones are configured with srtp and work and all of a sudden after about two days uptime of the server every phone / every call is rejected until I reboot the server, thats why I cannot find any infos about this on net and suggested to ask here |
| 14:59.00 | mjordan | ipengineer: brb, but in the meantime, do you feel comfortable patching Asterisk? :-) |
| 14:59.27 | Elleni | how can it be that server after 2 days would not provide crypto keys anymore? |
| 14:59.42 | Elleni | until reboot... |
| 14:59.59 | ipengineer | mjordan: Ok.. Sure this is a new installation of Asterisk 12. I am just trying to get it up and running right now |
| 15:01.05 | zafu | why is + illegal in a channel variable name? |
| 15:01.22 | zafu | I have SIP/+33145454545 |
| 15:01.43 | zafu | : ast_yyerror(): syntax error: syntax error, unexpected '+', expecting '-' or '!' or '(' or '<token>'; Input: SIP/+ = Local |
| 15:01.56 | Chainsaw | zafu: Because it is used as a control character. |
| 15:02.03 | Chainsaw | zafu: You could replace the + with 00. |
| 15:02.17 | zafu | what is a control char? |
| 15:02.29 | Chainsaw | zafu: A character that has special meaning, like the & and the +. |
| 15:02.38 | ipengineer | mjordan: I created a new user called asterisk and set that in asterisk.conf and the error goes away.. Now it just dies.. Very strange. Must have been a permission issue running under root |
| 15:02.50 | zafu | Chainsaw: what does + do in that context? |
| 15:03.21 | Chainsaw | ipengineer: Permission "issues" for root generally suggest you are running SELinux, GrSec or other protection mechanisms that go beyond UNIX permissions. |
| 15:03.21 | Chainsaw | ipengineer: Does dmesg not have any hints? |
| 15:03.47 | Chainsaw | zafu: I would have to look that up and my connectivity here is disappointingly slow. |
| 15:03.55 | ipengineer | Chainsaw: Good thoughts I always forget about disabling that! I am not seeing anything in dmesg of value |
| 15:03.55 | Chainsaw | zafu: But rest assured that it has special meaning, or it would have been allowed. |
| 15:04.02 | zafu | ok, thanks |
| 15:04.26 | Chainsaw | ipengineer: It could be logged elsewhere I suppose. See if anything in syslog lines up. |
| 15:04.42 | zafu | also, is it absolutely necessary to quote these constructs GoSubIf($["${CHANNEL:0:5}" = "Local"]? ..) ? |
| 15:04.53 | zafu | I mean both sides of the = |
| 15:05.31 | Cloin | for dialing purposes + has international dialing connotations, not sure if that is relevant or not zafu/Chainsaw |
| 15:05.49 | Chainsaw | zafu: My bash scripting background makes me quote what I can. The Asterisk parser is relatively helpful if you underquote, it tends to be obvious from the warning. |
| 15:05.54 | ipengineer | Chainsaw: looks like it is running now.. SElinux seems to have did the trick |
| 15:06.07 | ipengineer | Chainsaw: Thanks You |
| 15:06.11 | Chainsaw | Cloin: Well yes, that is why I suggested 00. |
| 15:06.21 | Cloin | :) |
| 15:06.25 | Chainsaw | Cloin: Because in most dialling schemas that is a synonym for +. |
| 15:07.07 | Chainsaw | ipengineer: Very good. I wouldn't just go "*click* off" on SELinux though. Best see what policy you violated and extend/modify the policy or your behaviour. |
| 15:07.42 | ipengineer | Chainsaw: Will do. |
| 15:09.04 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 15:10.27 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 15:16.32 | *** part/#asterisk eloycoto (~eloycoto@159.179.60.213.static.mundo-r.com) |
| 15:18.53 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.40) |
| 15:23.24 | gerritfromsa | zafu, the quotes are necessary yes |
| 15:23.43 | zafu | ack |
| 15:24.48 | gerritfromsa | zafu, how does the + get in SIP/+33145454545 ? shouldn't it be SIP/TRUNKNAME/+33145454545 ? |
| 15:24.59 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 15:25.19 | zafu | I named my sip client +<country_code><number> |
| 15:25.55 | gerritfromsa | zafu, it doesnt seem as if asterisk like it |
| 15:26.05 | [TK]D-Fender | I don't recall it being legal to have a peer with a "+" in it |
| 15:26.07 | zafu | actually after quoting the warning goes away |
| 15:26.08 | *** part/#asterisk Elleni (3ec00582@gateway/web/freenode/ip.62.192.5.130) |
| 15:26.12 | wdoekes | I use pluses all the time |
| 15:26.20 | wdoekes | although not in context names |
| 15:26.27 | [TK]D-Fender | Which is what i just said |
| 15:26.45 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 15:27.06 | gerritfromsa | Yes but the + goes in at the number position : SIP/TRUNKNAME/+33145454545 |
| 15:27.07 | *** join/#asterisk fantastic (~rainmaker@gateway/tor-sasl/fantastic) |
| 15:27.23 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:27.37 | Greenlight | No, I think he's named his peer with the actual number |
| 15:27.37 | gerritfromsa | That works only if your upstream provider allows it |
| 15:27.48 | gerritfromsa | yes i see that |
| 15:27.48 | zafu | Greenlight: yes |
| 15:27.56 | Penguin | zafu: No, the quotes are not required. You could just as easily compare it without the quotes. The quotes ensure than there will be something to compare in the case of the value being null. |
| 15:27.59 | Greenlight | Although I must say that's going to confuse |
| 15:28.12 | gerritfromsa | zafu, maybe change it too 00 instead |
| 15:28.13 | zafu | would sip providers generally accepts + as 00 ? |
| 15:28.35 | Greenlight | zafu: Depends |
| 15:28.38 | gerritfromsa | generally is a dangerous word |
| 15:28.39 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 15:28.47 | Chainsaw | Look at the consultants hedging their bets... |
| 15:28.57 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 15:28.59 | zafu | whereas they will _always_ accept 00 ? |
| 15:29.09 | Greenlight | Nope |
| 15:29.16 | Greenlight | Always is another dangeorous word |
| 15:29.27 | Penguin | You'll have to see what number formats they accept. |
| 15:29.29 | Greenlight | Don't you love SIP |
| 15:29.34 | gerritfromsa | sometimes it must start with the country code without 00 |
| 15:29.57 | Penguin | For example, for international dialing, I'll accept 00 or 011 at the international prefix. |
| 15:30.19 | gerritfromsa | Proprietary platforms is not exactly flexible |
| 15:30.21 | Greenlight | The international prefix differs between countries |
| 15:30.42 | Penguin | I'm talking about the country where I am, as opposed to a country where I'm not. |
| 15:31.07 | Chainsaw | You'd have to specifically exclude a country that is proud to be different... in order to say 00 is universal. |
| 15:31.30 | zafu | I see :) |
| 15:31.44 | Chainsaw | It would be like saying mains electricity is always ~230V. Yes. Would work in +33 and most other places. |
| 15:31.46 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 15:31.46 | Chainsaw | But not in +1. |
| 15:32.17 | Chainsaw | (As in the country code, not the silly google plus "like") |
| 15:32.23 | Greenlight | wonders what country could possibly like to be so different.. |
| 15:35.40 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.126.216) |
| 15:40.12 | gerritfromsa | Chainsaw, we only change 2 years ago , so now we're also using 00 - have to follow int std at some stage ... |
| 15:40.53 | Chainsaw | gerritfromsa: Yes, one day we'll *all* be sensible and use 00. |
| 15:41.38 | Greenlight | You're an idealist :) |
| 15:41.52 | gerritfromsa | one day we'll get rid of digits entirely and dial you@url.com |
| 15:42.02 | gerritfromsa | ENUM or whatever |
| 15:42.06 | Penguin | Some of us already do that. |
| 15:42.17 | Chainsaw | Greenlight: Definitely. |
| 15:43.01 | gerritfromsa | use DUNDI but thats it |
| 15:43.44 | gerritfromsa | only problem is bandwidth is a luxury is Africa |
| 16:03.18 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 16:03.30 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 16:03.52 | *** join/#asterisk Ice_Strike (~Ice_Black@cpc1-oldh7-0-0-cust772.10-1.cable.virginm.net) |
| 16:04.21 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 16:04.54 | Ice_Strike | I am getting chan_sip.c:8109 sip_reg_timeout: -- Registration for 'xxx@xxxx.sip.xxx.eu' timed out, trying again (Attempt #2) |
| 16:05.18 | Ice_Strike | After changing the resolver.conf |
| 16:05.23 | Ice_Strike | but ping google.com work |
| 16:05.39 | Greenlight | Sure it was resolver.conf and not resolv.conf ? |
| 16:07.28 | [TK]D-Fender | I'm also sure I don't trust a single masked line of debug. |
| 16:07.53 | Ice_Strike | /etc/resolv.conf i meant |
| 16:08.04 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.216) |
| 16:11.12 | Greenlight | Well if you changed DNS servers, and no other changes were made, then I think the problem is looking like DNS .. |
| 16:11.18 | Greenlight | Why not change back to the working server ? |
| 16:12.02 | [TK]D-Fender | Why not... look at all of the actual debug.... |
| 16:15.28 | *** join/#asterisk serafie (~erin@nat/digium/x-savrvofqglshytgb) |
| 16:16.39 | Penguin | It always amuses me when people hide the server names of their providers, which are PUBLIC companies. |
| 16:32.38 | *** join/#asterisk dant (~dan@180.191.120.253) |
| 16:37.48 | *** join/#asterisk kresp0 (~kresp0@gateway/tor-sasl/kresp0) |
| 16:42.57 | gartral | gerritfromsa: might I recomend opus or speex codecs if bandwidth is so limited? |
| 16:54.49 | *** join/#asterisk jpoz (~jpoz@107-1-105-37-ip-static.hfc.comcastbusiness.net) |
| 16:58.30 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 17:28.19 | gerritfromsa | gartral, use g729 - works ok |
| 17:39.01 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:39.39 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 17:40.43 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 17:43.10 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 17:50.00 | davlefouAMD | [TK]D-Fender, sound is chopped with my asterisk, but not with my ata. |
| 17:50.39 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 17:52.33 | [TK]D-Fender | davlefouAMD: I don't us having prven that yet. |
| 17:52.41 | [TK]D-Fender | davlefouAMD: and is something I highly doubt... |
| 17:57.49 | davlefouAMD | For me, it is nologique, i have by g729 liscence |
| 17:59.22 | [TK]D-Fender | davlefouAMD: There is no reason I'd trust that your softphone headset, etc is not responsible . |
| 17:59.43 | [TK]D-Fender | davlefouAMD: The codec processes what it is given and it isn't mysteriously bad for anyone else that we've heard of. |
| 18:03.51 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 18:03.51 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 18:15.38 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 18:22.41 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 18:22.41 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 18:23.34 | davlefouAMD | [TK]D-Fender, i can teste over softphone, but i use with alaw/ulaw codec. In local call, we don't have any problem. |
| 18:24.17 | [TK]D-Fender | davlefouAMD: Do you have actual calls to show us from each? |
| 18:25.01 | *** join/#asterisk petris_ (~petris@192.184.93.7) |
| 18:25.33 | davlefouAMD | i can send you but i is in french |
| 18:25.52 | [TK]D-Fender | Fine |
| 18:26.02 | [TK]D-Fender | I mean DEBUG btw... |
| 18:26.13 | davlefouAMD | Do you think there are an problem with ovh and asterisk g729? |
| 18:26.23 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 18:26.49 | [TK]D-Fender | What is "ovh"? |
| 18:27.00 | davlefouAMD | how can i send sound message? |
| 18:27.03 | [TK]D-Fender | And Asterisk doesn't care about any particular device. Codecs are codecs |
| 18:27.13 | [TK]D-Fender | What is a "sound message"? |
| 18:27.17 | [TK]D-Fender | You aren't making any sense |
| 18:27.19 | davlefouAMD | ovh is europeen voip |
| 18:27.29 | davlefouAMD | www.ovh.com/fr/ |
| 18:27.31 | Greenlight | OVH is presumably the large French data centre companyu |
| 18:27.35 | Greenlight | They are awful |
| 18:27.43 | davlefouAMD | Greenlight, ? |
| 18:27.47 | Greenlight | Cheap, but awful |
| 18:28.59 | Greenlight | From their UK forums their branching into voip was plauged with problems, and then discontinued. |
| 18:29.03 | *** join/#asterisk mic_ (~mic@0305ds4-vby.0.fullrate.dk) |
| 18:29.11 | mic_ | hello, I had a dirty meeting today |
| 18:29.22 | mic_ | did you ever encounter "Lync certification required"? |
| 18:29.38 | Greenlight | "dirty meeting"... sounds fun |
| 18:30.05 | mic_ | Greenlight: not really. 2 MS guys forcing a customer to buy something you can do in asterisk in 15 minutes... |
| 18:30.08 | davlefouAMD | France ovh is ok, but my problem is from my town where internet is bad. |
| 18:30.24 | mic_ | Greenlight: but "you cannot do it in asterisk because it's not Lync certified and then they withdraw all support" |
| 18:32.08 | davlefouAMD | I can't use more ulaw, i need g729 solution |
| 18:32.35 | jmetro | Why not |
| 18:34.12 | *** join/#asterisk gerritfromsa (~gerritfro@8ta-229-161-156.telkomadsl.co.za) |
| 18:34.31 | gerritfromsa | \join #beer |
| 18:35.12 | *** join/#asterisk Xylitool (~XyliBox@208.94.110.41) |
| 18:35.44 | Xylitool | hello all |
| 18:37.23 | Xylitool | does anyone now a way to execute a play message that can be stopped by keying dtmf; my app require dtmf identification, so waitdigit is next instruction after play; |
| 18:37.39 | Penguin | xylitool: BackGround() |
| 18:37.45 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:21b:63ff:fe31:8426) |
| 18:37.47 | Xylitool | Penguin niet |
| 18:38.08 | jmetro | background() is what youre looking for |
| 18:38.16 | Penguin | I usually use BackGround() followed by WaitExten(). |
| 18:38.22 | Xylitool | Background require having the extension defined so dtmf will point to |
| 18:38.32 | Xylitool | so my exten is not only one dtmf |
| 18:38.36 | Penguin | That way the caller can hit the key before or after the playback finishes. |
| 18:38.39 | jmetro | ok, so pattern match it. |
| 18:38.43 | Xylitool | it's an auth code 3-6 digits |
| 18:39.00 | jmetro | exten => _XXXXXX to match 6 digit codes of any digit. |
| 18:39.03 | Penguin | There's also Read(). |
| 18:39.44 | Xylitool | jmetro goess u re right, i haven't thinked about _XXXXX |
| 18:39.54 | Xylitool | that was missing :) |
| 18:39.56 | gerritfromsa | Or you simply use Authenticate() |
| 18:40.02 | jmetro | you should read the book about basic dialplan |
| 18:40.05 | jmetro | it might help a lot. |
| 18:40.27 | Xylitool | i do auth external with perl script, asterisk I need only basic |
| 18:40.43 | mic_ | Greenlight: ok, turns out ms publishes even a list of switches that "work with Lync" |
| 18:40.53 | mic_ | Greenlight: I will go back to my laundry. |
| 18:41.42 | Xylitool | jmetro -> exten => _XXXXXX won't work also :( because i have to define each code in astetisk extension.conf |
| 18:41.48 | *** join/#asterisk yano (yano@freenode/staff/yano) |
| 18:42.01 | Xylitool | i have a simple perl script to do database checking, and dtmf reading |
| 18:42.17 | Xylitool | wonder if there is any way to play message, and go to next instruction |
| 18:42.21 | Xylitool | (my script) |
| 18:42.23 | jmetro | i dont get it. You play the sound file, and get their code. |
| 18:42.30 | Xylitool | yep |
| 18:42.42 | jmetro | Exten => _XXXXXX,1,[Code-to-pass-it-to-your-script] |
| 18:42.43 | Xylitool | but sometimes message won't end, and caller will key in dtmf |
| 18:43.12 | jmetro | you pass ${EXTEN} to the script and thats the digits they dialed |
| 18:43.39 | Xylitool | so I can pass ${EXTEN} as an argument to my script in _XXXXX extension |
| 18:43.46 | gerritfromsa | Then BackGround() with WaitExten() will work |
| 18:44.01 | gerritfromsa | But you said 3-6 digits? |
| 18:44.05 | Xylitool | yes |
| 18:44.12 | jmetro | i know theres something you can limit it to 3-6 digits too |
| 18:44.34 | jmetro | http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 18:44.36 | gerritfromsa | Then the single _XXXXXX match will not work |
| 18:44.56 | Xylitool | need to put twice |
| 18:45.01 | Xylitool | one time _XXX |
| 18:45.03 | gerritfromsa | You need _XXX,_XXXX,_XXXX and _XXXXX |
| 18:45.04 | Xylitool | then _XXXXXX |
| 18:45.06 | Xylitool | yep |
| 18:45.12 | jmetro | No |
| 18:45.15 | jmetro | _XXX!!! |
| 18:45.15 | gerritfromsa | Just 3 or six ? |
| 18:45.21 | jmetro | will match 3 - 6 |
| 18:45.31 | gerritfromsa | not 3,4,5 or 6? |
| 18:45.38 | jmetro | it will match 3 4 5 and 6 |
| 18:45.44 | jmetro | not 2 not 7 |
| 18:45.53 | Xylitool | aha |
| 18:46.08 | jmetro | http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 18:46.30 | Xylitool | but this will make my life harder, because background function will play message on and on |
| 18:46.31 | WIMPy | Since whan can you have multiple ! ? That makes as little sense as multiple . . |
| 18:46.49 | gerritfromsa | Problem is when users enter 6 the dialplan will carry on without delay , but if they enter less digits , * will wait and that often created a delay |
| 18:46.51 | jmetro | I thought ! functioned as a placeholder that didnt have to be filled? |
| 18:46.53 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 18:47.08 | Penguin | BackGround() stops playing when DTMF is received. |
| 18:47.15 | gerritfromsa | that causes users to think theres something wrong ... |
| 18:47.20 | _Corey_ | ! matches 0 or more digits. |
| 18:47.23 | Xylitool | my extension is smarter than that; can count if no digit has been keyed in and let user try again, 3 times |
| 18:47.37 | Penguin | Not even digits, but characters. |
| 18:47.38 | jmetro | gerritfromsa: if you set the timeout shorter it wil only wait an additional 1/2 seconds between digit input and matching |
| 18:47.43 | Xylitool | so if will use background, there is no control |
| 18:47.48 | *** join/#asterisk admin0 (~admin0@5356416B.cm-6-7b.dynamic.ziggo.nl) |
| 18:47.50 | Penguin | Could be letters. |
| 18:47.52 | _Corey_ | Penguin: indeed, you're correct |
| 18:48.04 | WIMPy | And ! matches as soon as no other extension can match . waits until the number is complete. |
| 18:48.11 | admin0 | hi all .. anyone using a2biilling ? does the trunks and users need to be in the same call plan for the rate to work ? |
| 18:48.28 | jmetro | admin0: i think there is an #a2billing channel |
| 18:48.36 | admin0 | oh |
| 18:48.38 | admin0 | thanks |
| 18:48.49 | admin0 | chanserv and me :D |
| 18:49.04 | Penguin | ask alis |
| 18:49.05 | gerritfromsa | jmetro, how do you adjust the timing in BackGround() ? |
| 18:49.30 | jmetro | gerritfromsa: beforehand, you do like a Set(timeout thingy i forget what) |
| 18:49.48 | jmetro | http://www.voip-info.org/wiki/view/Asterisk+func+timeout |
| 18:49.59 | Penguin | TIMEOUT(digit)? |
| 18:50.08 | jmetro | ^ |
| 18:50.30 | phix | hmmmm |
| 18:51.06 | gerritfromsa | how does one disable the adds in voip-info.org ? |
| 18:51.14 | gerritfromsa | its in my way |
| 18:51.25 | jmetro | adblock+ probably |
| 18:51.25 | Xylitool | exten => s,n,Background then Exten => _XXXXXX,1, then how to get back to my menu with => s,n (it doesn;t work labeling s type ) |
| 18:51.27 | phix | throw money at them |
| 18:51.32 | gerritfromsa | TIMEOUT(response) it is |
| 18:51.37 | phix | then they won't need to put ads up there |
| 18:51.40 | jmetro | Xylitool: what do you mean? |
| 18:51.46 | jmetro | gerritfromsa: its timeout(digit) |
| 18:51.51 | Xylitool | my menu is for answering |
| 18:52.02 | gerritfromsa | jmetro, shot |
| 18:52.10 | Xylitool | so it's writtien with => s,1 ... s,n |
| 18:52.37 | *** join/#asterisk drjfreeze (~Jim@rrcs-67-78-64-218.sw.biz.rr.com) |
| 18:52.54 | jmetro | Xylitool: you need to read the book honestly, this is basic stuff |
| 18:53.06 | jmetro | Xylitool: you set a label, like same => n(goherenow),[code] |
| 18:53.14 | drjfreeze | Anyone know about replacing a TE121 Digium PRI card with at TE133 or TE134 card? |
| 18:53.15 | Xylitool | yes |
| 18:53.36 | gerritfromsa | The O'reilly books are best |
| 18:53.48 | jmetro | anything that leif madsen has touched. |
| 18:54.37 | Penguin | ~book |
| 18:54.38 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:55.48 | Penguin | Seriously, you could just use Read() to accept your digits. It will allow for interrupting the playback. |
| 18:55.52 | Xylitool | jmetro, my whole menu is made with labeling with - exten => s,n(retry1) ... but if will put Exten => _XXXXXX,n,Goto(retry1) it won't work :( it will search in _XXXXXX for label |
| 18:56.09 | Penguin | Goto a different place! |
| 18:56.19 | Penguin | Goto(s,1) |
| 18:56.26 | Penguin | or s,2 |
| 18:56.30 | Penguin | Or whatever you want. |
| 18:56.51 | Xylitool | Penguin, thanks |
| 18:57.03 | Penguin | But if I am not accepting DTMF to be used for an extension, I wouldn't go to the extension. |
| 18:57.11 | Penguin | I would read the data in and process it. |
| 18:57.18 | Penguin | That is standard IVR stuff. |
| 18:58.07 | Penguin | You use Read(), store the user's input into a variable, and process it later. |
| 18:58.13 | Xylitool | Penguin, i did waitdigit stuff inside perl script, was easier for me to manage how many digits and database query |
| 18:58.20 | gerritfromsa | talking about ivrs , I created a web-interface for an ivr and its works like a bomb |
| 18:58.37 | Penguin | I don't even know what waitdigit is. Never heard of it, never used it. |
| 18:58.42 | Penguin | gerritfromsa: It blew up? |
| 18:58.58 | gerritfromsa | I then recorded a walk-through from screen and send it to the client - he loved it |
| 18:59.14 | gerritfromsa | Linux command to record screen |
| 18:59.31 | Penguin | If you make the user key in some digits and you're going to feed that to an external script, it sounds like a regular IVR using Read() is the right tool for the job. |
| 19:00.11 | gerritfromsa | ffmpeg -f alsa -f x111grab -threads 0 filename.avi/mkv |
| 19:00.41 | gerritfromsa | can recommend it for presentations or/and demos |
| 19:02.52 | gerritfromsa | Xylitool, exten => _XXXXXX,1,System(external script ${EXTEN}) ;WHERE EXTEN is your 1st argument for the external script |
| 19:03.17 | gerritfromsa | or AGI be better |
| 19:03.24 | gerritfromsa | or Macro() |
| 19:04.00 | gerritfromsa | so many options really ... |
| 19:04.46 | *** join/#asterisk Cloin (Colin@pool-173-79-237-246.washdc.fios.verizon.net) |
| 19:05.34 | Xylitool | gerritfromsa, what if there are many steps of auth, using same number of digits ? how do you know which step of menu u are ? |
| 19:05.46 | Xylitool | it will always go to exten => _XXXXXX,1,System(external script ${EXTEN}) |
| 19:06.29 | Xylitool | it will work perfect for one layer |
| 19:06.50 | Cloin | Struggling a bit to get my first installation working with the first phone I've tried to set up. Is it typical to just spam the channel here with a detailed description or ask for responses in PM? |
| 19:07.29 | gerritfromsa | use SET() as you go , the CHANNEL VARIABLES get passed along as the dialpan gets processed |
| 19:09.45 | gerritfromsa | spam away |
| 19:12.19 | [TK]D-Fender | Cloin: you can DESCRIBE your issue here... for actual debug & configs --->> PASTEBIN |
| 19:12.22 | [TK]D-Fender | ~pb |
| 19:12.23 | 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. |
| 19:12.24 | [TK]D-Fender | ^^^ |
| 19:13.31 | Cloin | I have an AsteriskNOW installation working at the data center. The host is on private address space (10.15.20.5/24) and resides behind a firewall over which I have full control. I've provisioned a public IP (8.8.8.8) to translate on all TCP/UDP ports to 10.15.20.5. The phone resides at a second location and is on private address space (172.16.10.5/24) and NATs to a public address (7.7.7.7) |
| 19:13.31 | Cloin | in order to reach the Asterisk server at 8.8.8.8. |
| 19:13.52 | Cloin | It's a Cisco 7960 phone and I managed to use the TFTPD on the Asterisk server to get the firmware updated on the phone to be SIP compatible. |
| 19:14.46 | Cloin | The phone boots with the new binary from the TFTPD and pulls its configuration but immediately goes to "Phone Unprovisioned" |
| 19:15.20 | Cloin | I initially supposed I had NAT configuration wrong in either one of the .cnf files or on the Asterisk SIP configuration itself but after trying variuos combinations with no luck I'm not so sure now. |
| 19:15.44 | Cloin | I've permitted only 7.7.7.7 to connect to 8.8.8.8 on what I thought are the relevant ports and have no denied connections thus far for that traffic. |
| 19:15.57 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 19:16.07 | Cloin | (5060/5061 10000 = 20000) |
| 19:16.15 | Cloin | and TFTP |
| 19:16.48 | Cloin | Can't stress enough this is my first time doing anything with a PBX or using SIP |
| 19:17.34 | Cloin | Any suggestions or advice would be welcome, and witty criticism certainly encouraged |
| 19:17.43 | gartral | Cloin: and you're sure that AsteriskNOW's firewall isn't blocking the tfpt port? |
| 19:17.51 | gerritfromsa | Cloin, have you set up the provisioning server |
| 19:18.15 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 19:18.15 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 19:18.15 | *** join/#asterisk Sjors (~sgielen@foo.kassala.de) |
| 19:18.15 | *** join/#asterisk anonymouz666 (~anonymouz@186-241-66-19.user.veloxzone.com.br) |
| 19:18.15 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 19:18.15 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 19:18.15 | *** join/#asterisk TBryant (uid12962@gateway/web/irccloud.com/x-takpnccmbdkgnwju) |
| 19:18.15 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 19:18.15 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 19:18.34 | Cloin | gartral: I believe the phone updated its firmware from that TFTP server so I don't think so, but I'm also not certain as I don't recall modifying or even looking at it |
| 19:18.41 | Cloin | gerritfromsa: I'm not certain, where is that located? |
| 19:18.42 | _Corey_ | Cloin: It sounds like more of a provisioning issue with the phone. If it's pulling firmware from the remote server you likely don't have a good config file for the phone itself, hence the unprovisioned msg. |
| 19:19.02 | gerritfromsa | Cloin, you need to set that up yourself |
| 19:19.14 | Cloin | _Corey_: entirely possible as I pieced one together from what I could find online |
| 19:19.14 | gerritfromsa | It gets this info via the DHCP server |
| 19:19.20 | gerritfromsa | Option 66 and 67 |
| 19:19.33 | Cloin | Yes those are set to 8.8.8.8 for the DHCP server that responds to the phone |
| 19:19.40 | _Corey_ | Cloin: I could probably dig you up a working file if you want... |
| 19:20.04 | Cloin | _Corey_: if you're willing I would appreciate looking at one that is known to be in good working order |
| 19:20.07 | gerritfromsa | I agree with _Corey_ |
| 19:20.11 | Cloin | I will pastebin mine in the meantime |
| 19:20.22 | _Corey_ | yeah, gimme a few moments |
| 19:20.36 | gerritfromsa | Cloin, confirmed all this with Wireshark? |
| 19:20.57 | Cloin | Haven't done any packet sniffing, was looking at builds and teardowns in the firewall logs so far |
| 19:21.14 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 19:21.26 | gerritfromsa | Cloin, you can also program the Cisco manually |
| 19:22.08 | _Corey_ | Cloin: Here's a SIP<MACADDRESS>.cnf file: http://pastebin.com/sVNZP1Fj |
| 19:22.41 | _Corey_ | You may want to enable some logging on your tftp so you can verify the phone is requesting the correct file also |
| 19:23.05 | Cloin | http://pastebin.com/nfEpeQk4 |
| 19:23.18 | Cloin | That is the output from SIPDefault.cnf and SIPmac_address.cnf |
| 19:23.32 | Cloin | ok _Corey_ |
| 19:24.32 | Cloin | Thanks, I'll look through your configuration file now |
| 19:24.55 | _Corey_ | Cloin: Here's a SIPDefault.cnf to go with the other one: http://pastebin.com/k0C0Wf5d |
| 19:26.34 | _Corey_ | Cloin: You shouldn't need to do any NAT on the phone side. Just nat_enable=1 in the config. Then tell the phone to register to the public IP on the server side. |
| 19:27.22 | Penguin | Setting nat_enable to 1 does do nat traversal on the phone side, doesn't it? |
| 19:28.59 | _Corey_ | It's been a few years since I abandoned these Cisco phones and moved on to greener pastures, but as I recall it may not be strictly required |
| 19:29.11 | _Corey_ | these phones are very touchy with NAT though, IIRC |
| 19:29.13 | Cloin | Herein lies my confusion. I understand nat conceptually and practically as it relates to what I do regularly on network equipment, but the terminology has me confused both on the Asterisk GUI as well as in the configuration files |
| 19:29.33 | Cloin | s/nat/NAT/ |
| 19:29.45 | Cloin | hah! wonderful |
| 19:30.04 | Cloin | what a nice little bot |
| 19:30.07 | _Corey_ | Cloin: Well, Cisco quirks aside, you're basically going to configure an external IP address on the Asterisk SIP side of things to match up with your public IP |
| 19:30.24 | _Corey_ | then make sure NAT is enabled on your FreePBX "extension" |
| 19:30.33 | Cloin | _Corey_ despite the Asterisk host system not _really_ having a public address on it? |
| 19:30.54 | Cloin | but rather having one translated to it by an auxiliary device |
| 19:31.01 | _Corey_ | Yeah, it's going to use the external address when it's talking to a subnet that's not defined as local |
| 19:31.25 | _Corey_ | (see the section of the sample sip.conf pertaining to this as it explains) |
| 19:32.02 | _Corey_ | and when I say "use the external address" I should be more specific, as that sounds confusing |
| 19:32.23 | _Corey_ | It will use that within the SIP messages in place of its private IP |
| 19:32.29 | Cloin | As I understand it I need to modify the Asterisk SIP Settings portion of the GUI, the SIPmac_address.cnf file, and the SIPDefault.cnf file. Are there any other locations I will need to put NAT information? |
| 19:32.32 | Penguin | Usually, I would set nat_enable to 0 and enable nat for the device in asterisk. |
| 19:33.18 | _Corey_ | Penguin: Yeah, you're probably right. I don't think enabling it would do anything useful on the phone side |
| 19:35.19 | Cloin | Right now the Asterisk SIP Settings, NAT Settings are defined as: NAT - yes, IP Configuration - Static IP, External IP - 8.8.8.8, Local Network 10.15.20.5/24 |
| 19:36.03 | Cloin | And the only mention of NAT in my cnf file(s) is in SIPmac_address.cnf as: |
| 19:36.06 | Cloin | http://pastebin.com/sVNZP1Fj |
| 19:36.09 | Cloin | oops |
| 19:36.27 | Cloin | nat_enable: "1" |
| 19:36.27 | Cloin | nat_address: "7.7.7.7" |
| 19:36.33 | Cloin | nat_received_processing: "1" |
| 19:36.56 | Penguin | In most cases, enabling the nat stuff on the phone causes the phone to try to be smart and rewrite the addresses in the packets. |
| 19:36.57 | Cloin | 7.7.7.7 being the address that the phone would receive when it NATs in order to reach the Asterisk server at 8.8.8.8 |
| 19:37.08 | Penguin | That breaks asterisk's ability to handle it properly. |
| 19:37.41 | Cloin | I see. I'll start with setting nat_enable to 0 and removing the nat address as well as setting nat_received_processing to 0 |
| 19:37.46 | _Corey_ | Yeah, and you can remove that NAT translation on your router. You don't need to expose the phone to the Internet. |
| 19:38.02 | Cloin | Sorry I phrased that poorly |
| 19:38.10 | Cloin | It's actually PAT for the phone not NAT |
| 19:38.27 | Cloin | So only for outbound traffic leaving the interface, not the other way around |
| 19:38.48 | _Corey_ | Shouldn't be required unless you're filtering outbound traffic |
| 19:39.02 | Cloin | I am |
| 19:39.47 | Cloin | So SIPmac_address.cnf 's NAT parameters are now: |
| 19:39.50 | Cloin | nat_enable: "0" |
| 19:39.50 | Cloin | nat_address: "" |
| 19:39.50 | Cloin | nat_received_processing: "0" |
| 19:40.19 | Cloin | Should I need to modify Asterisk SIP Settings NAT Settings before proceeding? |
| 19:41.24 | gerritfromsa | Cloin, the SIP[MACADDRESS}.cnf file - the mac must be in HIGHER CASE |
| 19:43.59 | Cloin | You're referring to the file name right? |
| 19:44.28 | Cloin | It is all upper case on my system, I've only been using the lower case when obscurring its name to 'SIPmac_address' |
| 19:45.43 | gerritfromsa | just making sure , I remember it was case sensitive |
| 19:46.24 | gerritfromsa | Cloin, can you share the firmware please? |
| 19:46.55 | Cloin | SIP0013807818D8 |
| 19:46.59 | Cloin | woops |
| 19:47.09 | Cloin | so much for mac redaction |
| 19:47.14 | Cloin | P003-08-11-00.bin |
| 19:47.37 | gerritfromsa | I mean the file , the bin file - where did you download it? |
| 19:48.49 | gerritfromsa | THX |
| 19:49.38 | *** join/#asterisk bkruse (~Adium@24.42.229.8) |
| 19:52.33 | Cloin | So my TFTPD log is showing RRQ from 7.7.7.7 filename SIPmac_address.cnf over and over |
| 19:52.44 | Cloin | And the phone is still at Unprovisioned |
| 19:52.53 | Cloin | After changing the NAT parameters in that .cnf file |
| 19:53.56 | _Corey_ | Cloin: Well, that parameter has nothing to do with provisioning |
| 19:54.16 | Cloin | The RRQ messages indicate that TFTP is working right? |
| 19:54.26 | Cloin | Sorry, bit rusty with TFTP |
| 19:54.48 | gerritfromsa | request |
| 19:55.00 | _Corey_ | The face that it's looping would suggest not |
| 19:55.03 | Cloin | Oct 31 15:40:36 asterisk1 in.tftpd[19783]: RRQ from 7.7.7.7 filename SIPmac_address.cnf |
| 19:55.03 | gerritfromsa | Cloin, check permissions |
| 19:56.05 | _Corey_ | Cloin: Use a tftp client to make sure you can pull the file like the phone would... |
| 19:56.13 | gerritfromsa | +1 |
| 19:56.17 | _Corey_ | If that works, then you know it's the content of the file |
| 19:56.27 | _Corey_ | i.e. something mangled |
| 19:56.36 | Cloin | The perms are quite open, 777 |
| 19:56.44 | gerritfromsa | no sjit |
| 19:57.26 | gerritfromsa | agree with Corey , try it with a tftp client 1st |
| 20:00.01 | *** join/#asterisk dpeloquin (uid13057@gateway/web/irccloud.com/x-yssoznocgcqmtggc) |
| 20:00.45 | gerritfromsa | have2go ... thanks for everybody's free time! |
| 20:01.06 | Cloin | Later, thanks for your help so far |
| 20:01.17 | gerritfromsa | Enjoy Holland |
| 20:01.37 | *** part/#asterisk gerritfromsa (~gerritfro@8ta-229-161-156.telkomadsl.co.za) |
| 20:08.53 | Cloin | _Corey_ I'm noticing the two files you linked me have conflicting values for nat |
| 20:09.01 | Cloin | The first has it set to 0 while the second has it set to 1 |
| 20:11.28 | Penguin | The SIPDefault.cnf is a default value. SIP<MAC>.cnf will override the default if it is different. |
| 20:12.00 | Penguin | So if default says 0 and the phone-specific one says 1, it will be 1 (enabled). |
| 20:12.10 | Cloin | and the second link he shared was meant to be the Default? |
| 20:12.11 | _Corey_ | Cloin: I would try first with it disabled, as Penguin has said it shouldn't be needed |
| 20:12.18 | Cloin | Okay |
| 20:13.03 | Penguin | I don't override the nat value in the phone-specific confs. I set it to 0 in the SIPDefault.cnf and let asterisk do its nat stuff. |
| 20:14.08 | Cloin | Oh cool that new config file got it online |
| 20:14.18 | Cloin | Thank you everyone |
| 20:14.46 | Cloin | Now to find out why no dialtone, haha |
| 20:15.41 | *** join/#asterisk dant (~dan@180.191.120.253) |
| 20:15.43 | Penguin | The phone should provide the dial tone once a "call manager" is registered in the phone. |
| 20:15.57 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 20:17.08 | Cloin | I see. The testing extension I set up in Asterisk has a little 'x' by it so I'm sure something is not quite right |
| 20:17.43 | Penguin | I'm not sure what that means. |
| 20:18.01 | Penguin | Did you configure the phone's peer entry in sip.conf? |
| 20:18.26 | _Corey_ | On the SIP firmware I believe that just means it's not registered if I remember correctly |
| 20:18.46 | _Corey_ | so you could have an authentication or config issue |
| 20:18.58 | Penguin | It's been a while since I used SIP on my 7900 phones. |
| 20:19.49 | _Corey_ | You're not missing anything enjoyable ;) |
| 20:19.58 | Cloin | Hah |
| 20:20.08 | Cloin | Sounds like the overwhelming consensus is that I should use non Cisco phones |
| 20:20.29 | Cloin | _Corey_: I haven't been manually editing sip.conf as I was under the assumption the GUI was using it, and that I should do everything through there |
| 20:20.34 | Penguin | I use Cisco 7900 series phones... |
| 20:20.41 | _Corey_ | Well, asking for a phone preference in here is opening a big can of worms |
| 20:20.46 | Penguin | But I use SCCP, not SIP. |
| 20:21.18 | _Corey_ | The GUI, as in FreePBX? Unless you're using the "endpoint manager" -- which I have never used -- it would not |
| 20:21.34 | paulc | I had a guy the other day cursing the 79xx series because the phones he bought were SCCP and Cisco wanted $40/phone to make them SIP.. He didn't want to spent $200 (did he need to? Is the firmware licensed per phone (technically?)) |
| 20:22.01 | _Corey_ | Not that I recall |
| 20:22.11 | _Corey_ | If you have SmartNet you're entitled to it I believe |
| 20:22.18 | Penguin | Yes, but there are ways to get it, if you're into that sort of thing. You have to have a SmartNET contract to get the firmware. |
| 20:22.54 | paulc | Yeah - I figured "Pay for it on one phone and you'd be ok".. Then I told him I'd had great success with the SPA508G and would tend to lean towards those if you wanted the Cisco name (they're smaller/nicer on the desk than the 79xx's too, no?) |
| 20:22.59 | Penguin | I would even tell you the file names if you needed them. |
| 20:23.10 | Penguin | I won't search for them or give them to you, but I'd tell you the file names. |
| 20:23.53 | Cloin | I am referring to FreePBX yes, _Corey_ |
| 20:23.54 | paulc | He'd be appreciative I'm sure. I didn't care to get that involved (as he'd reached out to me for selling a Digium D70 by asking "Know anything about these Cisco phones I've got?".. uh, no.) |
| 20:23.56 | *** join/#asterisk CeBe (~CeBe@port-92-206-87-240.dynamic.qsc.de) |
| 20:25.04 | Penguin | The 7900s are old, but they are still okay phones... as long as you don't need HD audio. |
| 20:25.17 | Cloin | ;--------------------------------------------------------------------------------; |
| 20:25.17 | Cloin | ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; |
| 20:25.17 | Cloin | ; this file must be done via the web gui. There are alternative files to make ; |
| 20:25.17 | Cloin | ; custom modifications, details at: http://freepbx.org/configuration_files ; |
| 20:25.17 | Cloin | ;--------------------------------------------------------------------------------; |
| 20:25.23 | Cloin | which is why i hadn't touched it |
| 20:25.32 | Penguin | You can use SIP on them, but I use SCCP on mine (with asterisk and chan_sccp-b). |
| 20:25.53 | _Corey_ | Cloin: You pulled that from something in your tftp folder? |
| 20:26.12 | Cloin | that's from the heading of /etc/asterisk/spi.conf |
| 20:26.13 | Cloin | sip* |
| 20:26.32 | Penguin | cloin: That's why we don't support FreePBX here. You ask asterisk questions, I'll do my best to give asterisk answers. If you use FreePBX, consider most of the answers to be things that will ruin your FreePBX configuration. |
| 20:26.48 | Cloin | I see |
| 20:27.07 | Penguin | We'll help you all day long with asterisk. When it comes to FreePBX, there's another channel for that. |
| 20:27.13 | Cloin | Well I'm not attached to it I just didn't know which avenue to begin with and chose that one somewhat arbitrarily |
| 20:27.58 | _Corey_ | nothing wrong with it for many scenarios, but you may find it limiting if you progress beyond a basic setup |
| 20:29.04 | Penguin | For all my purposes, the biggest problem with it is that I can't have full control over the configs. For that reason, I will probably never use FreePBX. |
| 20:29.43 | Penguin | For others, the "problem" I described may be their only reason to use it. |
| 20:29.44 | Cloin | Yeah that doesn't sound ideal to me. Wonder how easily I can back out of FreePBX |
| 20:30.27 | Penguin | If you're just starting out, it wouldn't be too difficult to get out of it. You might spend a couple hours at it. |
| 20:30.51 | _Corey_ | Cloin: If your goal is to learn Asterisk, you will want to avoid FreePBX until you've mastered the basics |
| 20:30.53 | Penguin | I'm totally comfortable with ssh and vim -- I don't need a web interface to configure things. |
| 20:31.36 | _Corey_ | once you understand Asterisk, you may find FreePBX useful... I use it on many deployments. |
| 20:31.37 | jalewis | <PROTECTED> |
| 20:31.44 | jalewis | oops |
| 20:32.01 | Penguin | I use Alt+w for that. :) |
| 20:35.46 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 20:43.01 | Cloin | Perhaps I'll just start with a new installation. My sysadmin capabilities are a bit limited and I don't want to waste time trying to distinguish what effects FreePBX has had on the installation thus far |
| 20:43.38 | Penguin | It's not too difficult to get away from it. |
| 20:44.44 | Penguin | sip.conf and extensions.conf are your main files you'll be using. FreePBX takes them over and creates some additional and custom files for you to play with manually. |
| 20:44.51 | Penguin | I think it also uses users.conf, which you'll want to delete. |
| 20:45.01 | Penguin | Phones are configured in sip.conf. |
| 20:45.13 | Penguin | Extensions (which are not phones) are configured in extensions.conf. |
| 20:45.47 | raub | Can I configure asterisknow completely from command line? |
| 20:46.01 | Penguin | That depends on what you installed. |
| 20:46.04 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 20:46.22 | Penguin | I use AsteriskNOW for the OS and asterisk. I configure it totally from the command line. |
| 20:46.34 | raub | Excellent |
| 20:46.44 | Penguin | If you installed FreePBX, too, don't use the command line to configure things. |
| 20:46.46 | raub | I am using the asterisk now ISO, whcih is centos |
| 20:47.03 | Penguin | There should still be an option to install asterisk only with no GUI. |
| 20:47.41 | raub | Penguin: others might use the gui but I just want to have it so I can create a puppet thingie for it |
| 20:48.09 | Penguin | Generally speaking, you use FreePBX only if you have and use FreePBX at all. |
| 20:48.24 | Penguin | If you're going to do command line administration, don't use FreePBX. |
| 20:48.44 | Penguin | BUT... there are some custom files that FreePBX will allow you to use for certain things. |
| 20:49.31 | Penguin | My advice is to do command line administration OR FreePBX configuration. Don't combine them. |
| 20:49.37 | Cloin | Would you suggest AsteriskNOW but trying to opt-out of the GUI during the installation then? |
| 20:49.45 | Penguin | Although some people have success with those custom files. |
| 20:49.59 | Penguin | cloin: Yes. That's what I used to do when I needed an asterisk box. |
| 20:50.30 | Penguin | I would get the AsteriskNOW CD, start it up, select the NO GUI option, and away I go. |
| 20:50.36 | Cloin | I'll give that some consideration then before I go too much further |
| 20:50.45 | Penguin | In 20 minutes or less, I have a running system with asterisk waiting to be configured. |
| 20:51.09 | raub | Penguin: I see what you mean. Lemme try doing that then. It is just a vm anyway |
| 20:51.44 | Penguin | Now it has been a couple years since the last time I used it, and there have been changes to AsteriskNOW... but I assume there is still a "no GUI" option. |
| 20:52.44 | raub | K |
| 20:54.48 | raub | On garden-variety asterisk, has anyone successfully run it as a vm + pci passthrough (for the card: in my case sangoma a101 and a400)? |
| 20:55.16 | raub | I am getting a lot of messages like these: |
| 20:55.17 | raub | [1763710.859734] wanpipe1:w1g1: Error: TxDMA Length not equal 0 (reg=0x60000801) |
| 20:55.17 | raub | [1763710.860486] wanpipe1:w1g1: Tx Error: Abort from Master: pci fatal error! |
| 20:55.23 | raub | in dmesg |
| 20:55.46 | raub | Inthis setup (not an asterisknow one), I am not running freepbx |
| 20:55.52 | raub | (if this matters) |
| 20:57.47 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 21:10.42 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 21:24.15 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 21:27.25 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 21:28.34 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 21:35.29 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.76) |
| 21:36.57 | *** join/#asterisk bchamberlain (~brian98@unaffiliated/brian98) |
| 21:37.15 | bchamberlain | Hello - is there any way to make asterisk not respond to options pings from a peer? |
| 21:37.47 | Penguin | Why would you want to? |
| 21:38.47 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.76) |
| 21:39.34 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.76) |
| 21:41.52 | *** join/#asterisk ghost75 (~quassel@dslb-088-066-167-067.pools.arcor-ip.net) |
| 21:41.55 | *** part/#asterisk ghost75 (~quassel@dslb-088-066-167-067.pools.arcor-ip.net) |
| 21:41.58 | *** join/#asterisk ghost75 (~quassel@dslb-088-066-167-067.pools.arcor-ip.net) |
| 21:42.26 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 21:44.06 | bchamberlain | penguin: we have a carrier that uses options to tell if a box is alive to send calls. |
| 21:44.28 | Penguin | Yep, that's the typical use of OPTIONS packets with Asterisk. |
| 21:44.57 | bchamberlain | Penguin I'd like to stop responding to pings so the carrier knows the box is off/going offline |
| 21:45.27 | Penguin | If the box is off, it will not respond. Problem solved. |
| 21:45.29 | bchamberlain | Penguin otherwise we can shutdown asterisk and we can have upto 60 seconds where the carrier continues to send calls. |
| 21:45.40 | Penguin | Oh, I see. |
| 21:45.55 | bchamberlain | before the options fails and then it moves onto next destination :) |
| 21:45.59 | Penguin | You want them to stop early. |
| 21:47.23 | Penguin | I don't know of a good way to stop response. |
| 21:47.51 | Penguin | I suppose you could do it in iptables. |
| 21:48.13 | Penguin | You'd have to do string matching and block the packets based on the string OPTIONS. |
| 21:48.29 | bchamberlain | Penguin that would work of course! |
| 21:49.54 | Penguin | Let me know if that works like you want. |
| 21:51.15 | bchamberlain | Penguin iptables -I INPUT -j DROP -p udp --dport 5060 -m string --string "OPTIONS sip" --algo bm should work.... I'll give it a go! Thanks for help.. |
| 21:52.13 | Penguin | It looks like it would work. |
| 22:01.31 | skrusty | has anyone here used s3 as a virtual drive for storing/feteching audio files? |
| 22:06.43 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 22:10.28 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-tsxnufynmuxfdaxj) |
| 22:20.36 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:22.26 | *** join/#asterisk wasanzy (~wasanzy@41-66-239-57-dedicated.4u.com.gh) |
| 22:22.29 | wasanzy | hi |
| 22:22.35 | wasanzy | does asterisk do sms? |
| 22:26.59 | *** join/#asterisk jhlavacek (~jirka@jix.nextradsl.cz) |
| 22:39.07 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.124.113) |
| 22:39.43 | *** join/#asterisk barbosa2 (~juliano.b@177.158.1.247) |
| 22:55.19 | *** join/#asterisk dongola7 (~dongola7@unaffiliated/blair/x-0911782) |
| 22:58.38 | bchamberlain | Penguin worked like a treat! Thanks again for great idea. |
| 23:05.20 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 23:12.08 | Penguin | bchamberlain: Great! |
| 23:18.14 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.105) |
| 23:24.15 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.127.237) |
| 23:28.18 | *** part/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 23:31.32 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 23:45.55 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 23:59.30 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.194) |