| 00:54.46 | *** join/#asterisk gomez1 (~kvirc@bzq-79-181-222-179.red.bezeqint.net) |
| 00:54.58 | gomez1 | hi |
| 00:56.20 | gomez1 | do you know usage of this dialplan function: Set(SIP_CODEC=g726) |
| 00:56.23 | gomez1 | ? |
| 00:56.53 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.81) |
| 00:56.54 | gomez1 | does it matter which priority I put this command? |
| 00:58.14 | *** join/#asterisk deo (~deo@112.198.79.21) |
| 00:59.56 | [TK]D-Fender | gomez1: Clearly before the call gets answered or any call-out is placed |
| 01:00.10 | [TK]D-Fender | gomez1: Why are you even looking to do that in the dialplan? |
| 01:02.50 | gomez1 | because I find it a very flexible function.. |
| 01:03.58 | [TK]D-Fender | If it's hard-coded it typically may as well be in the peers themselves |
| 01:05.11 | gomez1 | what happens if I use that function AFTER playing early-media? |
| 01:09.00 | *** join/#asterisk chazzam (~cmoye@nat/digium/x-teukvwldxilzzmxt) |
| 01:15.52 | gomez1 | bump |
| 01:17.20 | [TK]D-Fender | There is no point in "bump". This isn't a message board. |
| 01:17.30 | [TK]D-Fender | If we go AFK then you'll just have to wait... |
| 01:17.49 | [TK]D-Fender | any audio setup is probably the end of the road |
| 01:18.06 | [TK]D-Fender | So I wouldn't make plans on trying to change that afterwards. |
| 01:23.02 | gomez1 | OK thanks. How can I check the codec being used during a call? |
| 01:23.05 | gomez1 | is there a command for that? |
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| 01:56.18 | Tim_Toady | ${CHANNEL(audionativeformat)} variable |
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| 05:50.09 | joako | 3 days ago was released a CentOS kernel security update. When will the corresponding Asterisk packages be updated and why does this process always lag behind so much? |
| 06:08.45 | jeev | joako, get a real distribution then and do it yourself, i prefer slackware. |
| 06:09.40 | wdoekes | joako: because of limited resources |
| 06:09.46 | wdoekes | you get what you're paying for |
| 06:16.31 | *** join/#asterisk bombev (258fc08d@gateway/web/freenode/ip.37.143.192.141) |
| 06:16.40 | joako | Automated build system? I´m using the AsteriskNOW distro... |
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| 07:10.52 | ChannelZ | Does rfc2833 dtmf specify how long the tone should be? |
| 07:12.16 | coppice | the duration of the tones is conveyed in the rfc2833 packets |
| 07:12.34 | *** join/#asterisk Defraz (~Defraz@209.141.122.71) |
| 07:13.42 | ChannelZ | Can it be changed on the asterisk side? |
| 07:13.46 | Defraz | has anyone played with the volume function and turning up the volume on a sip call with * and #. I have the set Volume in my dial plan and it is being set but I hear no difference in the call volume. and * and # don't seem to affect it. |
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| 07:15.55 | [TK]D-Fender | Defraz: Show us |
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| 07:18.17 | Defraz | [macro-dial] exten => s,1,Set(VOLUME(TX,p)=8) exten => s,n,Set(VOLUME(RX,p)=8) |
| 07:22.11 | Defraz | kinda going off what i see here http://www.withsupport.co.uk/wiki/increasing-sip-volume-freepbxasteriskelastixtrixbox |
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| 07:23.04 | [TK]D-Fender | Defraz: full call with SIP debug... |
| 07:28.18 | ChannelZ | The DTMF length seems to be 100ms but that doesnt seem to be quite long enough for my damn bank's automated thingy. |
| 07:30.21 | coppice | 100ms is far too long. The specified minimum length for a DTMF digit is 40ms |
| 07:32.01 | ChannelZ | Specified by what and where? |
| 07:35.26 | coppice | http://www.itu.int/rec/T-REC-Q.24-198811-I/en |
| 07:41.01 | ChannelZ | Hmmm. |
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| 09:41.21 | ghost75 | is pulseaudio required on * or alsa? |
| 09:42.00 | ghost75 | pulseaudio is installed, not sure if i need it |
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| 11:46.06 | Quest | is it possible to have 2 or more extension numbers for one phone in freepbx? |
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| 12:47.43 | din3sh | hello all |
| 12:48.07 | din3sh | 503 Service Unavailable on asterisk.org & digium.com!? |
| 12:48.07 | din3sh | :o |
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| 12:53.24 | w9sh | interesting way to start the day. weekend maint? |
| 12:54.24 | w9sh | i've got a group meeting at 11am EDT. have to remember not to browse to those sites unless they are back up. |
| 12:56.35 | din3sh | personally i've never hit a 503 error on asterisk.org for the pas few years |
| 12:56.38 | din3sh | :o |
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| 14:28.49 | HolTech | Getting a 404 from an extension which is an analog phone in an ATA, any help is appreciated |
| 14:31.10 | [TK]D-Fender | HolTech: pastebin the complete call with sip debug enabled. |
| 14:31.12 | [TK]D-Fender | ~pb |
| 14:31.12 | 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. |
| 14:31.14 | [TK]D-Fender | ^^^ |
| 14:31.15 | [TK]D-Fender | ^^^ |
| 14:34.26 | HolTech | D-Fender: http://pastebin.com/Rw3CmMLJ |
| 14:35.04 | HolTech | extension holtech is the analog phone, soft is a softphone... no nat involved here |
| 14:35.54 | HolTech | the 404 is line 88... if I am understanding correctly the analog phone ( moreso the ATA which is a sipura 3000) is not accepting the call.. but 404 means not found |
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| 14:46.14 | [TK]D-Fender | HolTech: Complete call.. evrything leading up to it. |
| 14:46.21 | [TK]D-Fender | and SIP peer config... |
| 14:51.06 | HolTech | Sorry, thought i copied all... this should be everything . http://pastebin.com/NvgUJtui |
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| 14:58.39 | [TK]D-Fender | HolTech: those are not the default ports that model uses (5060/5061), and typically it should register to *. Also * can't transcode G.723 without an expensive card. You should not be allowing that codec unless you really know what you are doing ... and happen to be desperate AND crazy.... |
| 14:59.33 | [TK]D-Fender | HolTech: Also, kill the "auth" ,"fromdomain", and "canreinvite" lines and add "directmedia=no" |
| 15:00.23 | [TK]D-Fender | HolTech: Fix all of these , retest them provide the same things in the next PB if it fails |
| 15:02.08 | HolTech | will do, thanks. is it neccessary to use default ports ? on the ATA I had the codec as G711u, but * complained when I put the line allow=g711u in my sip.conf. |
| 15:03.04 | [TK]D-Fender | HolTech: Because that's "ulaw" which is what you need to put. |
| 15:03.21 | [TK]D-Fender | HolTech: And I highly advise as it's on a local LAN |
| 15:13.19 | HolTech | [TK]D-Fender: thanks... updated http://pastebin.com/Cht0tiTv |
| 15:19.55 | *** join/#asterisk obalci (~Orkun_BAL@92.44.153.50) |
| 15:21.00 | Kobaz | is it okay to manually delete msg0000 and etc |
| 15:21.02 | obalci | Hello Everybody. I have a big problem. I cannot run my asterisk server behind openvpn. Do you know any tutorial on the internet about this issue. I want to form a private PBX over openvpn encryption. |
| 15:21.08 | Kobaz | will the voicemail system be smart enough to start at msg0001 |
| 15:21.11 | Kobaz | and not die |
| 15:21.35 | [TK]D-Fender | HolTech: I'd suggested to just let the device register and NOT specify a hard host... |
| 15:22.02 | [TK]D-Fender | HolTech: It might refuse calls if it expecting to have registered and because of not having done so would consider it "foreighn" |
| 15:22.50 | [TK]D-Fender | HolTech: Also be very sure you got the ports right for the FXS side... |
| 15:24.01 | [TK]D-Fender | HolTech: While you're at it you have an extra "allow=all", "canreinvite=no" that should be replaced by directmedia, and no codecs locked down for [soft] |
| 15:24.28 | [TK]D-Fender | HolTech: Check all again including the web status screen on the SPA, retest then pastebin again |
| 15:25.58 | HolTech | [TK]D-Fender: on the device I have configured it to accept calls to answ without registering.. but i will change it all to register... the ports are correct on the fxs side... set to 5060, and the fxo is 5061, ill take out the allow=all and canreinvite and redo |
| 15:25.58 | [TK]D-Fender | obalci: There is no "guide" for this. Packets are packets. They are either making it to your PBX or they are not. |
| 15:27.07 | [TK]D-Fender | obalci: Go enable SIP DEBUG and see if anything even arrives. |
| 15:27.23 | obalci | D-Fende: When I am using it without openvpn it works correctly. But when I close all the ports but openvpn it doesnt work correctly. |
| 15:27.30 | obalci | ok I will try SIP DEBUG thanks |
| 15:34.35 | HolTech | Having some trouble with device registering now that it is host=dynamic.... might be firewall on ast box... registration occurs on udp 5060 ? |
| 15:38.23 | *** part/#asterisk obalci (~Orkun_BAL@92.44.153.50) |
| 15:39.34 | [TK]D-Fender | HolTech: Clarify "trouble" |
| 15:39.48 | [TK]D-Fender | HolTech: if htere was a firewall issue you shouldn't have gotten a 404 back... |
| 15:41.01 | HolTech | device web admin page says failed, not seeing any attempts in * CLI, which makes me wonder if my * box is dropping packets... so im curious what protocol/port registration occurs on... udp 5060 is open on my * box |
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| 15:43.35 | HolTech | D-Fender: on your advice I'm changing to require registration, the 404s were when the ATA was set NOT to register, but in sip.conf I defined the ATA's host and port.. now in sip.conf host=dynamic and on the ATA web admin page it is set to register, but failing |
| 15:46.27 | [TK]D-Fender | something has been misconfigured on it then.... |
| 15:49.51 | HolTech | nope. it was the firewall... turned iptables off on the asterisk box and now both devices are registered... still wondering what port though |
| 15:50.10 | [TK]D-Fender | Odd... |
| 15:50.20 | HolTech | gree.. eveything works not... |
| 15:50.21 | [TK]D-Fender | firewall should have block the response to your previous call-outs... |
| 15:50.22 | HolTech | * now |
| 15:50.29 | [TK]D-Fender | HolTech: Glad to hear... |
| 15:50.47 | HolTech | right, so now everything is working, thank you so much for your help D-Fender, sorry it was something so simple.. I should have tried that already |
| 15:51.04 | HolTech | I've googled and googled looking for what ports * needs, and have never found. |
| 15:52.35 | [TK]D-Fender | HolTech: * listens on the port you bind it to in [general]. * can call OUT to whatever port your peers says if it is different than the default |
| 15:52.43 | [TK]D-Fender | HolTech: RTP... is rtp.conf |
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| 20:23.36 | talntid2 | whats the recommended install path for ubuntu machines? from packages, or source? production environment.. |
| 20:23.44 | talntid2 | 1.8.10 is what is in repos |
| 20:24.10 | [TK]D-Fender | And what does the topic say? |
| 20:25.26 | talntid2 | says it's 11.3.0. I know that. I am asking more specifically if there is another repo to use, or just to do it from sources |
| 20:25.47 | [TK]D-Fender | Source |
| 20:25.52 | talntid2 | roger that |
| 20:25.54 | [TK]D-Fender | Nother for Ubuntu/Debian yet |
| 20:25.59 | [TK]D-Fender | Or ... change your OS |
| 20:27.13 | talntid2 | are there up to date packages in another OS's repo? |
| 20:27.24 | [TK]D-Fender | CentOS only |
| 20:27.53 | talntid2 | roger that |
| 20:27.54 | slav3_kitten | i'd not run it on a ubuntu install personally |
| 20:28.08 | talntid2 | I am not married to Ubuntu, so I'll run it on CentOS :) |
| 20:28.17 | slav3_kitten | i did it on debian from source |
| 20:29.16 | slav3_kitten | compiling from source is pretty quick and simple |
| 20:29.47 | talntid2 | yeah |
| 20:30.15 | talntid2 | I would like to have a package manager in place for security updates and such though |
| 20:30.39 | slav3_kitten | just compile it every update |
| 20:30.42 | slav3_kitten | pretty simple |
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| 21:04.52 | *** join/#asterisk Quest (~syncsys@175.110.31.35) |
| 21:05.26 | Quest | is it possible to have 2 or more extension numbers for one phone in freepbx? |
| 21:07.23 | Quest | how long does it takes to install asterisk and configure 30 cisco phones? |
| 21:08.01 | *** join/#asterisk Quest (~syncsys@175.110.31.35) |
| 21:08.18 | Quest | is it possible to have 2 or more extension numbers for one phone in freepbx? |
| 21:08.20 | Quest | how long does it takes to install asterisk and configure 30 cisco phones? |
| 21:13.46 | *** join/#asterisk Quest (~syncsys@175.110.31.35) |
| 21:17.18 | talntid2 | Quest: yes, 1 hour. |
| 21:17.23 | talntid2 | *rolleyes* |
| 21:19.51 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 21:20.11 | Quest | and what freepbx is not installed but askterisk alone is installed. what time theN? |
| 21:21.43 | talntid2 | 60 minutes. |
| 21:23.57 | Quest | one other question. what time difference is expected for freepbx install with configs and asterisk install alone with configs? I heard freepbx takes less time? |
| 21:24.16 | talntid2 | 11 minutes. |
| 21:25.14 | Quest | ! |
| 21:25.28 | talntid2 | i know, right? |
| 21:32.01 | *** join/#asterisk dfighter (~dfighter@arcemu/staff/dfighter) |
| 21:47.07 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 21:53.21 | Quest | <PROTECTED> |
| 21:56.35 | ChannelZ | and then you spend the next 11 days trying to get freepbx to do what you want |
| 21:58.00 | talntid2 | yup |
| 21:58.06 | *** join/#asterisk ujjain2 (~ujjain@unaffiliated/ujjain) |
| 21:58.29 | talntid2 | Quest, yes, it is possible to have 2 or more extension numbers for one phone in freepbx. but this is not a freepbx channel - maybe try in there? :) |
| 21:58.49 | [TK]D-Fender | Nothing sinks in with this one |
| 22:00.11 | igcewieling | "Welcome to Generic Ford Dealership, how can I help you?" "I need help with my Mazda." |
| 22:02.14 | [TK]D-Fender | “Everybody knows you never do a full retard.” - Lazarus |
| 22:08.01 | igcewieling | [TK]D-Fender: I feel that people who repeatedly ask FreePBX questions here should be banned. Since that won't happen, I tend to put them on my /ignore list. |
| 22:08.16 | igcewieling | *** Quest has been added to Ignore List |
| 22:09.37 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 22:11.37 | ChannelZ | Hmm. Did Frys change their logo? |
| 22:13.33 | *** join/#asterisk nix8n82 (~AndChat27@24.143.11.81) |
| 22:19.52 | *** join/#asterisk iprouteth0 (~james@unaffiliated/iprouteth0) |
| 22:20.16 | iprouteth0 | happy saturday |
| 22:27.51 | ChannelZ | Thrilling. |
| 22:37.27 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 22:53.46 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 23:54.45 | *** join/#asterisk ledoktre (~ledoktre@199-102-208-169.static.osage.net) |
| 23:56.07 | ledoktre | Greetings. Trying to track down a weird one. I have two SPA3102 boxes. When I dial out on PSTN directly (analog phone) it dials in exactly 4 seconds. No echo. When dialling through SPA3102, it is taking about 8 seconds. I tried with and without appending a # to the end of the dial string. Where do you look to see why its delaying? |
| 23:57.04 | ledoktre | FWIW, on the asterisk console, it indicates almost immediately (less than a half second) that the spa is ringing, and then the next line, spa answered |
| 23:57.18 | ledoktre | but there is dead air for about another 7 or 8 seconds until it starts ringing |