| 00:00.17 | Katty | maxime1986: bmoraca_work is leaving, not me |
| 00:00.37 | maxime1986 | it's not a big problem since iax can easily be filter by port |
| 00:01.02 | *** part/#asterisk cguerrero (~cuauhtemo@200.79.231.94) |
| 00:01.29 | maxime1986 | Katty: oups .. working .. configuring asterisk .. follow the chat .. maybe to much things in parallel .. sorry |
| 00:02.24 | Katty | maxime1986: no big (= |
| 00:03.05 | *** join/#asterisk jks (jks@193.189.93.254) |
| 00:07.20 | *** join/#asterisk micols (~mio@rlogin.dk) |
| 00:10.12 | ryanlin | can anyone assit me on getting the asterisk autoattendent to work with cme? |
| 00:10.27 | ryanlin | i configured the dial peer on the cme |
| 00:11.09 | ryanlin | made a phone call..the call goes throughn to the asterisk, asterisk does not accept the calls |
| 00:15.46 | ryanlin | ManxPower-work: ah i see...well..we are actually using sip..but it didn't work |
| 00:16.59 | *** join/#asterisk Godfather_ (~Godfather@79.109.251.13.dyn.user.ono.com) |
| 00:17.27 | *** join/#asterisk ltd (~z@pat.transact.net.au) |
| 00:23.47 | *** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net) |
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| 00:34.05 | *** join/#asterisk jhirley (~jhirley@adsl-3-198-18.mia.bellsouth.net) |
| 00:37.01 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 00:37.50 | *** join/#asterisk Godfather_ (~Godfather@157.Red-88-11-88.dynamicIP.rima-tde.net) |
| 00:42.46 | *** part/#asterisk maxime1986 (~maxime198@anonymous-132-207-233-94.broker.freenet6.net) |
| 00:45.22 | *** join/#asterisk Slugs_ (Slugs_@c-76-97-205-31.hsd1.ga.comcast.net) |
| 00:51.56 | *** join/#asterisk nickaugust (~anonymous@167.83.189.72.cfl.res.rr.com) |
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| 00:54.10 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 01:03.19 | *** join/#asterisk seanjohn (~john@173.50.101.10) |
| 01:03.51 | seanjohn | which version of asterisk is the latest and most stable? 1.6.0? |
| 01:05.44 | *** join/#asterisk Slugs_ (Slugs_@c-76-97-205-31.hsd1.ga.comcast.net) |
| 01:09.42 | ruben23 | hi |
| 01:10.31 | Slugs_ | . |
| 01:11.04 | ruben23 | hi, dahdi is compatible with asterisk 1.4.22 |
| 01:20.20 | *** join/#asterisk wdbl (~not@ool-44c0668f.dyn.optonline.net) |
| 01:29.47 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 01:32.22 | *** join/#asterisk Flametail (~chatzilla@dynamic-acs-72-23-74-123.zoominternet.net) |
| 01:32.31 | Flametail | hello? |
| 01:33.21 | Flametail | I was hoping to use asterisk as kind of a private phone service.... where all my friends can communicate with each other |
| 01:33.48 | Flametail | simply by enter ip and port number of the server... and th eextension they want to call |
| 01:34.33 | Flametail | aka 76.56.45.456 for the ip 8650 for the port and call extension 5786 for Robert.... is this possible with the asterisk server? |
| 01:37.37 | *** join/#asterisk wdbl (~not@ool-44c0668f.dyn.optonline.net) |
| 01:40.42 | *** join/#asterisk pentanol (~pentanol@77-35-6-037.pppoe.primorye.net.ru) |
| 01:41.29 | *** join/#asterisk seanjohn (~john@173.50.101.10) |
| 01:42.28 | jaytee | Flametail, yes but * uses the SIP protocol which normally uses port 5060. You can learn more how to setup what you want by reading the book |
| 01:42.31 | jaytee | ~book |
| 01:42.32 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 01:43.17 | TJNII | Flametail: You'll want to set up accuonts for your users and have them log in. It is the easiest way and, if done properly, secure. |
| 01:43.57 | TJNII | Portscanning bots will try and exploit your install if you put it on the web, you don't want unauthenticated access. |
| 01:44.06 | Flametail | is new to sip and asterisk... perhaps more specific please... also note Flametail uses Asterisk fr windows |
| 01:44.27 | ManxPower-work | Flametail, "asterisk for windows" is not supported in any shape or form |
| 01:44.30 | jaytee | Asterisk for Windows is a joke. dump it |
| 01:44.45 | Flametail | well I havent a clue how to use linux |
| 01:44.51 | ManxPower-work | jaytee, I believe it *actually* was released on April 1st. |
| 01:44.59 | jaytee | it originally started as an April fools joke |
| 01:45.08 | ManxPower-work | Flametail, Ah! Sorry, I just recognized you for the Troll you are. |
| 01:45.12 | jaytee | another flaming turd that refuses to be extinguished |
| 01:47.12 | Flametail | ima just have to find another server..... |
| 01:47.14 | *** part/#asterisk Flametail (~chatzilla@dynamic-acs-72-23-74-123.zoominternet.net) |
| 01:47.52 | seanjohn | webenabled = yes is this legal in manager.conf for each user? |
| 01:48.45 | seanjohn | flame, linux uses itself |
| 01:49.05 | seanjohn | the basic commands aren't much different from windows command shell |
| 01:49.29 | seanjohn | instead of \ linux uses the correct / |
| 01:49.44 | TJNII | seanjohn: He left |
| 01:49.50 | seanjohn | webenabled = yes : is this legal in manager.conf for each user? |
| 01:50.18 | seanjohn | like can i enable web for specific people? |
| 01:51.32 | seanjohn | i do know that using linux as my router is a lot better than windows using a regular router |
| 01:51.44 | seanjohn | very low latency with iptables |
| 01:51.45 | *** join/#asterisk simcop2387-lap (~simcop238@p3m/member/simcop2387) |
| 01:51.59 | simcop2387-lap | how do i reduce the verbosity of the asterisk console? (its at 3 right now i want it at 0) |
| 01:52.31 | Godfather_ | core set verbosity 0 |
| 01:52.32 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 01:52.52 | simcop2387-lap | thanks |
| 01:53.08 | simcop2387-lap | so much was going by i couldn't see anything in tab compeltion |
| 01:53.21 | seanjohn | no one knows about webenabled? |
| 01:53.27 | seanjohn | i'm using 1.6.2 |
| 01:54.24 | simcop2387-lap | i'm on 1.6.1 here |
| 01:58.42 | *** join/#asterisk Slugs_ (Slugs_@c-76-97-205-31.hsd1.ga.comcast.net) |
| 02:01.21 | jaytee | I'm on 1.6.arglebargle.whoosh |
| 02:04.28 | Katty | hi |
| 02:06.45 | *** join/#asterisk Arsenick (~y@modemcable022.82-21-96.mc.videotron.ca) |
| 02:07.38 | jaytee | hi Katty |
| 02:08.13 | Katty | hugs jaytee |
| 02:08.20 | Slugs_ | hola |
| 02:08.21 | jaytee | hugs Katty |
| 02:08.36 | Katty | hey slugs |
| 02:09.05 | Slugs_ | how long have you guys been using asterisk? |
| 02:09.28 | Katty | bout 5 years |
| 02:10.31 | Slugs_ | how long did it take for you to get a 'decent' handle on things? |
| 02:10.43 | Katty | probably a year |
| 02:11.21 | Slugs_ | were you a complete newb like me in the beg? |
| 02:11.22 | Slugs_ | ;0 |
| 02:11.27 | Katty | probably worse |
| 02:11.32 | Katty | i didn't know anything about linux |
| 02:11.59 | Slugs_ | that's uplifting ;) |
| 02:12.51 | Slugs_ | i get so frustrated, because i know im asking ignorant questoins and feel bad because im sure people are thinking 'wtf is he talking about' ;/ |
| 02:14.13 | Katty | :< |
| 02:14.29 | jaytee | Slugs_, you've been in here often enough that you should know that reading the book will make you 1) less ignorant and 2) enable you to ask better questions. |
| 02:14.37 | Slugs_ | i think in the past 2 weeks ive learned so much, and the deeper i get into it the more i realize i don't knoe but the more i love it |
| 02:14.52 | Katty | put the asterisk book in the bathroom |
| 02:15.21 | Slugs_ | oh jaytee, absolutly, thank god for astrisk docs |
| 02:15.26 | Katty | and tinker a lot (= |
| 02:15.30 | Slugs_ | lol |
| 02:15.32 | Slugs_ | yes |
| 02:15.48 | Slugs_ | im so happy i started doing this |
| 02:16.12 | jaytee | build it, then deliberately break it to see what happens, gives you a good handle on cause and effect for troubleshooting. |
| 02:16.25 | Slugs_ | definitly |
| 02:16.39 | Slugs_ | jaytee, how long have you been doing this? |
| 02:16.45 | Katty | i still break stuff all the time |
| 02:16.53 | jaytee | since 2006 |
| 02:17.41 | Slugs_ | non of you guys use freepbx? |
| 02:17.45 | Slugs_ | none* |
| 02:17.49 | Katty | i tried it once. |
| 02:17.59 | Katty | i had a difficult time understanding how it worked. |
| 02:18.11 | Slugs_ | it seems like you have less control |
| 02:18.14 | jaytee | it tends to restrict the flexibility and it's a very complex system |
| 02:18.15 | Slugs_ | i got rid of it |
| 02:18.18 | Katty | then i realized it made its own config files...and the macros were such a...nightmare to sift through |
| 02:18.34 | Slugs_ | yes |
| 02:18.39 | Katty | i ended up taking a few key items from it, and then dumped it |
| 02:18.43 | Katty | asterisk-stat was one of them |
| 02:19.08 | Katty | FoP was also on trixbox |
| 02:19.10 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 02:19.13 | Katty | but i twas broken. |
| 02:19.19 | VoIP-Penguin | You get lots less control using a GUI. |
| 02:19.26 | Katty | luckily i'd setup FoP before, so i dug through it and figured out what was broken... |
| 02:19.46 | Katty | overall, i didn't much care for it |
| 02:19.51 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 02:19.55 | jaytee | I'm designing a management interface that runs on Windows and uses SSH to Asterisk, still uses the standard config files and still lets the admin have full control over the config files |
| 02:20.06 | Katty | it has certain advantages for an office who wants to manage their own system...and not really know a clue about it |
| 02:20.10 | Katty | rather than managed services. |
| 02:20.35 | Slugs_ | that's pretty cool |
| 02:20.37 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 02:22.57 | Slugs_ | im getting closer to connecting avaya to asterisk via h323, so im getting more excited |
| 02:24.35 | *** join/#asterisk elvisthedj (~kris@68.119.14.243) |
| 02:24.58 | Katty | i have a sleepy sammy ferret |
| 02:25.31 | elvisthedj | anyone here running gentoo? this might not be distro specific, but i can't figure out what rtc kernel option to enable to make dahdi happy |
| 02:25.45 | simcop2387-lap | i am |
| 02:26.08 | simcop2387-lap | lemme go look at what i've got (though i think i've got every one of them compiled as modules) |
| 02:26.23 | elvisthedj | simcop2387-lap: thanks! :) |
| 02:27.24 | simcop2387-lap | yea everything as modules |
| 02:27.30 | simcop2387-lap | though let me check something |
| 02:27.59 | simcop2387-lap | hmm |
| 02:28.14 | simcop2387-lap | the only depends line that i get is depends: crc-ccitt |
| 02:28.18 | simcop2387-lap | from modinfo |
| 02:29.13 | elvisthedj | simcop2387-lap: thanks! lemme take a look over here |
| 02:31.30 | elvisthedj | simcop2387-lap: i have crc-ccitt compiled in .. trying to emerge dahdi says i don't have rtc support .. driving me crazy |
| 02:33.09 | elvisthedj | simcop2387-lap: if you have a moment, maybe you can see if you see anything obvious .. i'm pretty close to going the non-portage route on this one |
| 02:33.15 | elvisthedj | http://pastebin.com/f0mM1eKa |
| 02:34.26 | simcop2387-lap | yea let me look |
| 02:35.17 | simcop2387-lap | what kernel version? out of curiosity |
| 02:35.40 | elvisthedj | 2.6.29 |
| 02:35.49 | simcop2387-lap | oh its at the bottom of the paste heh |
| 02:35.58 | elvisthedj | :D |
| 02:39.08 | simcop2387-lap | the only differences i'm seeing is i've got RTC_DEBUG on (shouldn't matter) and all the drivers as modules, except the test driver (o' |
| 02:39.17 | simcop2387-lap | i've got the test driver built in |
| 02:39.31 | simcop2387-lap | you've got cmos built in |
| 02:39.39 | simcop2387-lap | oh and the test |
| 02:40.13 | simcop2387-lap | oh doh i'm backwards |
| 02:40.46 | simcop2387-lap | i've got everything BUT the test driver installed as modules |
| 02:40.50 | elvisthedj | well.. i guess i'm just going to make my own ebuild for it. maybe it will either work or give me a better idea as to why it is not |
| 02:40.51 | simcop2387-lap | give that a shot |
| 02:40.57 | simcop2387-lap | yea |
| 02:40.59 | elvisthedj | oh okay |
| 02:42.26 | simcop2387-lap | i've only got rtc-cmos loaded though |
| 02:45.33 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 02:47.14 | simcop2387-lap | i just wish google would fix gizmo, its gone way down hill since the take over, i used to have google voice+gizmo working great to let me make free outgoing (and incoming) calls |
| 02:49.39 | elvisthedj | i just started playing with google voice a couple weeks ago.. haven't tried to integrate it (but free in/out sounds nice ;) |
| 02:50.14 | elvisthedj | i got dahdi to build :D .. now dahdi tools is screwing up .. more googling |
| 02:50.34 | *** join/#asterisk pentanol (~pentanol@77-35-6-037.pppoe.primorye.net.ru) |
| 02:51.01 | *** join/#asterisk Slugs_ (Slugs_@c-76-97-205-31.hsd1.ga.comcast.net) |
| 02:51.04 | ManxPower-work | elvisthedj, did you read the dahdi README? |
| 02:53.00 | elvisthedj | ManxPower-work: I haven't. I'll take that as a suggestion :) |
| 02:53.16 | ManxPower-work | should have been the first thing done |
| 02:53.34 | ManxPower-work | chances are it will list what software is required to build it |
| 02:56.38 | xpot | anyone know how to solve voicemail issues where extension digits are repeated and password digits are repeated? IE: extension 104, shows entered in cli as 110044, and pass 23456 shows up as 2233445566? I asume jitter any suggestions? |
| 02:56.58 | elvisthedj | ManxPower-work: I was able to build/install dahdi. I always look at the readme when i'm downloading src, but you kinda hope that the deps have been addressed in the ebuilds |
| 02:57.25 | ManxPower-work | elvisthedj, Generally Packaged versions of Asterisk suck and are not supported here. |
| 02:57.25 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 03:00.20 | *** join/#asterisk siera04 (~chatzilla@113.18.8.82) |
| 03:00.43 | *** join/#asterisk sjobeck (~Adium@67.136.135.134) |
| 03:04.27 | siera04 | I made the outbound call with Zaptel FXS phone, but the destination plays ivr and requires something(like language, password) before answer. |
| 03:04.29 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 03:04.46 | siera04 | I can't input password on Zaptel FXS phone because the destination didn't answer to me. |
| 03:04.48 | siera04 | How can i input password? |
| 03:04.54 | siera04 | SIP phone has no problems to input the password. |
| 03:05.08 | Slugs_ | . |
| 03:05.09 | ManxPower-work | siera04, How are you connecting to the PSTN? |
| 03:05.30 | siera04 | using zaptel FXO. |
| 03:06.10 | ManxPower-work | siera04, All FXO ports are considered ANSWERED as soon as dialing is finished. |
| 03:06.46 | ManxPower-work | The only exception to this is if you are using the (rather problematic) busydetect=yes or (and only insane people use this option) callprogress=yes |
| 03:06.49 | siera04 | ... |
| 03:07.30 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 03:07.33 | ManxPower-work | siera04, What specific CARD do you have? |
| 03:07.54 | siera04 | 4FXS4FXO digium card |
| 03:09.38 | siera04 | NOW in zaptel.con busydetect=yes,and callprogress=no |
| 03:14.32 | ManxPower-work | siera04, great1 set them both to no |
| 03:17.50 | siera04 | i set "callprogress=yes" and test it just now. |
| 03:18.12 | siera04 | it looks like no problems now... |
| 03:19.47 | ManxPower-work | you should set them both to no or you'll have calls randomly disconnect as the systems incorrectly detects a BUSY condition. |
| 03:20.12 | ChannelZ | Yeah when you get that sound effects guy from Police Academy on the phone, it's a nightmare |
| 03:21.05 | ManxPower-work | ChannelZ, usually people with "loud" voices or women with higher pitched voices |
| 03:21.27 | siera04 | ManxPower-work: as you said, if "callprogress=yes", the FXO has answered |
| 03:21.45 | siera04 | and after a few minutes, call is hangup automatically. |
| 03:21.47 | ChannelZ | Busy detect I get buy what is the call progress actually looking for? |
| 03:21.52 | ManxPower-work | English is not your native language? |
| 03:21.58 | siera04 | sorry |
| 03:22.02 | siera04 | me is poor enblish. |
| 03:22.22 | ManxPower-work | ChannelZ, a pathetic attempt at trying to determine busy, answer, etc on FXO ports. |
| 03:22.42 | ManxPower-work | siera04, use whatever option works for you. |
| 03:22.54 | siera04 | yes. |
| 03:23.11 | ManxPower-work | I wish they would either change callprogress=yes to randomlydisconnectmycalls=yes (it's much more truthful) or remove the feature all togather. |
| 03:24.28 | siera04 | yes, as you said. call is hanged up randomly... |
| 03:25.33 | siera04 | but i saw "--Called g2/17*** -- Zap/35-1 answered Zap/37-1", if i use "callprogress=yes". |
| 03:25.54 | siera04 | i saw only "--Called g2/17*** ", if i use "callprogress=no". |
| 03:28.16 | siera04 | any other help?? i saw only "--Called g2/17*** ", if i use "busydetect = no" |
| 03:29.18 | ManxPower-work | siera04, set it to whatever works for you. |
| 03:30.37 | siera04 | i can't see "-- Zap/35-1 answered Zap/37-1" log. |
| 03:30.49 | siera04 | so i can't input the password after ivr menu. |
| 03:36.32 | *** join/#asterisk chendy_ (~chatzilla@204.152.211.137) |
| 03:42.15 | *** join/#asterisk jhirley (~jhirley@adsl-3-129-18.mia.bellsouth.net) |
| 03:43.34 | jhirley | anyone know the default mysql root password for the AsteriskNow distro ? |
| 03:44.08 | *** join/#asterisk Slugs_ (Slugs_@c-76-97-205-31.hsd1.ga.comcast.net) |
| 03:44.29 | *** join/#asterisk chendy_ (~chatzilla@119.139.18.189) |
| 03:44.30 | jhirley | anyone know the default mysql root password for the AsteriskNow distro ? |
| 03:47.13 | TJNII | No, but I bet #asterisknow does. |
| 03:48.05 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 03:49.38 | *** join/#asterisk jhirley_ (~jhirley@adsl-145-32-23.mia.bellsouth.net) |
| 03:53.46 | hluesea | hello |
| 03:55.19 | *** join/#asterisk Brack10 (~travis@97.90.64.53) |
| 03:55.23 | Brack10 | Hey |
| 03:55.58 | Brack10 | Does Asterisk support SIMPLE messaging and presence |
| 04:00.20 | beek | infobot: seen jaytee |
| 04:00.23 | infobot | jaytee is currently on #asterisk (3h 23m 23s). Has said a total of 12 messages. Is idling for 1h 40m 28s, last said: 'I'm designing a management interface that runs on Windows and uses SSH to Asterisk, still uses the standard config files and still lets the admin have full control over the config files'. |
| 04:03.24 | *** join/#asterisk Slugs_ (Slugs_@c-76-97-205-31.hsd1.ga.comcast.net) |
| 04:03.53 | *** join/#asterisk chendy_ (~chatzilla@119.139.18.189) |
| 04:06.16 | Slugs_ | . |
| 04:09.25 | ManxPower-work | phpagi is crap compared to asterisk-perl |
| 04:10.03 | ManxPower-work | Brack10, No. It does not, unless it's new to 1.6, in which case it should be mentioned in the UPGRADE*.txt files that come with the Asterisk source code. |
| 04:10.20 | Brack10 | well damnit |
| 04:10.48 | ManxPower-work | It supposed SUBSCRIBE/NOTIFY based presence. |
| 04:10.55 | ManxPower-work | supports, that is. |
| 04:10.56 | Brack10 | there's no good way to integrate xmpp since presence on xmpp enabled clients wouldn't be visible to hard phones and visa versa |
| 04:11.24 | ManxPower-work | that was not your question. |
| 04:12.05 | ManxPower-work | if the xmpp (jabber?) channel driver supports presence, then it should be integrated into the Asterisk DEVSTATE stuff, which works via SUBSCRIBE/NOTIFY. |
| 04:12.57 | ManxPower-work | So, yes, there's a good chance you can get presence working between an xmpp channel driver and SIP phones. No, you woud not be using SIMPLE. |
| 04:13.24 | hluesea | i have take like that notice and i can't see callers cid chan_sip.c:18044 handle_request_invite: Call from '' to extension 'XXXXXXXXX' rejected because extension not found |
| 04:13.42 | ManxPower-work | hluesea, then you are not receiving that information. |
| 04:14.02 | VoIP-Penguin | to extension 'XXXXXXXXX' rejected because extension not found |
| 04:14.03 | hluesea | said that not found extension but this coming calls come default scobe and |
| 04:14.17 | VoIP-Penguin | plain |
| 04:14.20 | VoIP-Penguin | text |
| 04:14.21 | ManxPower-work | hluesea, but that wasn not your question. Your question was about callerid. |
| 04:14.32 | *** join/#asterisk chendy (~chatzilla@204.152.211.137) |
| 04:14.51 | ManxPower-work | hluesea, do you have allowguest=no in your sip.conf [general]? |
| 04:15.02 | hluesea | ok my question is system was working properly yesterday and i can see the callerid etc. and system accept the callers |
| 04:15.06 | hluesea | i am looking in |
| 04:15.29 | ManxPower-work | hluesea, I suspect the call is not matching the peer you think it's matching |
| 04:15.33 | hluesea | it is unselected |
| 04:15.34 | VoIP-Penguin | Another case of magical changes. |
| 04:15.38 | hluesea | actually commented |
| 04:16.11 | ManxPower-work | hluesea, Add allowguest=no to the [general] section of sip.conf and do a "sip reload" in the CLI. |
| 04:16.28 | ManxPower-work | then try your call again. You should get a better error message on the CLI now. |
| 04:18.48 | hluesea | ok i am trying |
| 04:19.51 | *** join/#asterisk fofware (~chatzilla@190.225.219.125) |
| 04:20.50 | *** join/#asterisk wdbl (~not@ool-44c0668f.dyn.optonline.net) |
| 04:21.07 | *** join/#asterisk tecnico (~tecnico@75.76.169.148) |
| 04:22.33 | hluesea | <PROTECTED> |
| 04:22.45 | hluesea | i guess it is a2billing issue |
| 04:23.46 | *** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68) |
| 04:25.21 | *** part/#asterisk ruben23 (~ITadmin@122.55.48.243) |
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| 04:44.36 | siera04 | ManxPower-work: I resolved that problem. i changed "answeronpolarityswitch=yes" to "answeronpolarityswitch=no" |
| 04:45.10 | siera04 | thank your for your help. |
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| 06:54.22 | essiene | hi all |
| 06:55.08 | essiene | what's the preferred way of doing SS7 in asterisk? libss7 or chan_ss7? |
| 06:55.22 | essiene | that is if any is preferred... or are they both mature and it doesn't matter which i use? |
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| 07:09.34 | ChannelZ | simcop2387-lap: hey I tried your system again last night and still got no audio. Is your * partially behind a firewall? |
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| 08:05.07 | simcop2387-lap | ChannelZ: yea it is, it is set to always believe its natted though |
| 08:06.03 | ChannelZ | My only guess was that the RTP range my side is using is being stopped on yours (outbound) |
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| 08:07.57 | *** join/#asterisk Znuff (~ibm86@2001:0:53aa:64c:c97:55aa:a6d0:dcfd) |
| 08:08.00 | Znuff | Hi. |
| 08:08.18 | Znuff | Anyone knows of a softphone that can open an unlimited lines? |
| 08:08.25 | Znuff | Something like Zoiper |
| 08:08.40 | Znuff | For Windows |
| 08:09.42 | ChannelZ | not really since it's pretty hard to have an unlimited number of people using a single app |
| 08:10.44 | Znuff | well, Zoiper can open up unlimited lines |
| 08:10.53 | *** join/#asterisk Gary_B (~gary@85.211.228.89) |
| 08:10.59 | Znuff | but I'm faced with a weird issue, the call timeouts after ~30 seconds |
| 08:19.11 | Gary_B | do you hear anything during the 30 secs? |
| 08:23.48 | Znuff | Gary_B, absolutely nothing |
| 08:25.02 | Znuff | only the initial ring |
| 08:25.20 | Gary_B | has it ever worked? Do calls to other exyensions work - ie is it just calls going via sip/iax provider that dont work? |
| 08:26.47 | Znuff | dialing to another extension to my computer, same thing... |
| 08:28.05 | Znuff | But unfortunatelly I'm not the asterisk's admin :-/ |
| 08:28.23 | Znuff | I would have hoped it's something I could fix client-side |
| 08:30.10 | ChannelZ | I guess I'm missing the part where a client with unlimited lines is significant |
| 08:30.32 | Gary_B | so you are trying to use a softphone on your pc and you can not dial out at all, in this case id check your firewall settings on that computer first |
| 08:31.31 | Znuff | ChannelZ, that was just a bonus |
| 08:31.40 | Znuff | Gary_B, I can do calls using other sip accounts just fine |
| 08:31.43 | Znuff | It's just this one that fails |
| 08:32.12 | Gary_B | are the other sip accounts outside your loacal LAN while the offending one is inside by any chance? |
| 08:32.39 | Znuff | nope, all external |
| 08:34.31 | Gary_B | have you looked into codecs at all, is your softphone setup to use a codec with the other accounts that perhaps this one doesnt support? |
| 08:35.00 | Znuff | this is trying u-law by default |
| 08:35.11 | Znuff | pretty sure I've heard u-law sounds coming out of these speakers |
| 08:35.27 | Gary_B | btw - has it ever worked? |
| 08:35.50 | Znuff | On this sip account, nope |
| 08:37.19 | Gary_B | and the other sip accounts, you have confirmed that at least 2 completely different accounts worked recently, in this case, i think you need to contact the asterisk admin |
| 08:37.40 | Znuff | Tought so =) |
| 08:37.45 | Znuff | just my luck, I guess |
| 08:39.39 | Znuff | Now, another thing that it's worth mentioning |
| 08:40.06 | Znuff | if I open enough lines, fast enough, the calls drop after 40-50-60 even 70 seconds |
| 08:40.48 | Gary_B | but you still only hear one ring yea? |
| 08:41.20 | Znuff | yeah |
| 08:41.46 | Znuff | erm, keep forgetting that the client crashes if I do this too fast :P |
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| 11:17.11 | whtsup | hello |
| 11:17.35 | whtsup | how can i install speex into asterisk |
| 11:24.06 | whtsup | hello |
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| 11:52.20 | Ad-Hoc | hi ppl |
| 11:52.25 | jaytee | hi |
| 11:56.13 | whtsup | how to install speex in asterisk |
| 11:56.20 | whtsup | need help plz |
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| 12:50.16 | whtsup | WARNING[28981]: loader.c:381 load_dynamic_module: Error loading module 'codec_speex.so': libspeex.so.1: cannot open shared object file: No such file or directory |
| 12:50.27 | whtsup | error comming when i m loading speex module |
| 12:52.11 | gladier | a) has it been compiled and b) does it libspeex.so.1 exist in /usr/lib/asterisk/modules/ |
| 12:53.35 | whtsup | nops |
| 12:53.42 | whtsup | codec_speex.so |
| 12:54.04 | whtsup | is exist |
| 12:55.26 | whtsup | when i try to load this file this error is comming |
| 12:55.34 | whtsup | i had install speex |
| 12:55.40 | whtsup | and recompile asterisk |
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| 12:57.40 | gladier | yep ... because your shared library isn't located anywhere that asterisk knows to look |
| 12:58.08 | gladier | do a 'updatedb && locate libspeex.so.1' and tell me where the file is |
| 12:58.09 | whtsup | so wht shud i do now |
| 12:59.48 | whtsup | updatedb command not found |
| 13:00.34 | gladier | sigh ... lets do this the old way then ... find / -name libspeex.so.1 -print |
| 13:00.48 | russellb | the error is from the dynamic linker. normally the library would be in /usr/lib |
| 13:01.08 | jaytee | waves at russellb |
| 13:01.14 | russellb | waves back to jaytee |
| 13:01.40 | jaytee | russellb, how've ya been? you've been travelling alot lately |
| 13:02.16 | russellb | doing well! |
| 13:02.24 | whtsup | usr/src/speex-1.2rc1/libspeex/.libs/libspeex.so.1 |
| 13:02.26 | russellb | but for now ... i need to go back to sleep, i got up crazy early |
| 13:02.45 | whtsup | usr/local/lib/libspeex.so.1 |
| 13:03.04 | gladier | ok is /usr/local/lib/ in /etc/ld.so.conf ? |
| 13:03.08 | jaytee | rest well, russellb |
| 13:03.17 | gladier | if it isn't - add it then run a ldconfig |
| 13:03.42 | gladier | yay for distros that dont expect you to compile anything locally |
| 13:03.57 | whtsup | ok cheking |
| 13:04.19 | jaytee | gladier, what distro do you use? |
| 13:04.41 | gladier | depends what im doing - for asterisk i generally use centos |
| 13:04.45 | whtsup | no there is nothing in /etc/ld.so.conf |
| 13:04.53 | jaytee | gladier, same here |
| 13:05.04 | jaytee | for desktop I use Ubuntu |
| 13:05.20 | gladier | echo "/usr/local/lib/" >> /etc/ld.so.conf |
| 13:05.22 | gladier | then ldconfig |
| 13:05.27 | jaytee | loves RHEL 5 64 bit |
| 13:05.49 | whtsup | done |
| 13:05.58 | gladier | desktop i have a mac :P - firewalls i normally build a base gentoo box with iptables |
| 13:06.04 | gladier | whtsup: try and load the module again |
| 13:06.09 | whtsup | ok |
| 13:06.46 | whtsup | thanks alot dude |
| 13:06.50 | gladier | worked? |
| 13:06.53 | whtsup | yah |
| 13:06.57 | whtsup | core show translation |
| 13:07.01 | whtsup | now it is showing |
| 13:07.02 | gladier | what distro out of interest? |
| 13:07.09 | whtsup | Centos |
| 13:07.44 | gladier | strange... i've never had that issue |
| 13:08.01 | gladier | and i have funky modules like chan-sccp_b |
| 13:08.15 | gladier | goes back to studying for his win7 exam |
| 13:08.30 | whtsup | take care gladier n have a nice time |
| 13:08.35 | whtsup | n thanks again |
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| 13:41.33 | DelphiWorld | hi |
| 13:41.35 | DelphiWorld | anyone using fring? |
| 13:45.33 | DelphiWorld | looking for adding sip account to fring after adding sip |
| 13:46.11 | DelphiWorld | looking for adding sip account to fring after adding skype... |
| 13:46.15 | DelphiWorld | s/sip/skype/ |
| 13:51.18 | *** part/#asterisk DelphiWorld (~Miranda@41.104.31.92) |
| 13:58.41 | zamba | doesn't fring suck? |
| 13:58.47 | zamba | the sip support, i mean |
| 13:59.00 | zamba | there's no native sip support.. everything is routed over their own servers |
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| 14:32.24 | jkroon | tzafrir, did you ever manage to sort out http://lists-archives.org/git/711956-git-svn-cannot-lock-the-ref-refs-remotes-tags-autotag_for_.html ? |
| 14:33.04 | tzafrir | jkroon, basically by manually removing that tag |
| 14:33.27 | jkroon | ok. how do i do that? |
| 14:33.27 | tzafrir | hmm.. not tag. ref |
| 14:34.24 | tzafrir | the line in .git/info/refs , IIRC |
| 14:35.42 | tzafrir | but maybe also removing the file under .git/logs/refs/remotes/tags |
| 14:35.53 | jkroon | the only reference to autotag in .git for me is in .git/configs (ignore-paths) |
| 14:38.02 | jkroon | find locates the name in .git/svn/refs/remotes/tags, rm -rf'ing it there doesn't solve it for me. |
| 14:38.27 | jkroon | it just recreates it :( |
| 14:39.36 | jkroon | going to try #git perhaps ... |
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| 15:15.49 | tzafrir | jkroon, .git/info/refs is a single file |
| 15:16.26 | jkroon | yes, but it's not in there. i'm doing a clean git svn clone, it's getting stuck at that ref, not going past it and not recording it to remove it either. |
| 15:17.02 | jkroon | doing weird svn list things it actually looks like someone removed that actual path using git dump/load without also removing the commit referencing the path. |
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| 15:19.18 | jkroon | http://pastebin.co.za/96347 |
| 15:21.58 | tzafrir | jkroon, but is there any actual problem? e.g.: does 'git gc' work? |
| 15:22.25 | jkroon | checks. |
| 15:24.00 | jkroon | re problem, yes, there is. the svn repo is broken from the looks of it. however, my concern is more that git won't check it out (even though it is broken). git svn clone http://svn.digium.com/svn/asterisk -s --ignore-paths='.*autotag_for.*' asterisk fails at svn revision 47394. |
| 15:24.28 | jkroon | git gc works fine even though the repo is only checked out to revision 47393 ... can't get past that revision. |
| 15:24.55 | *** join/#asterisk r0fl (~r0fl@unaffiliated/r0fl) |
| 15:25.12 | jkroon | and digium can't go back and edit the svn history as it'll cause major re-numbering and major other crap (everybody will probably have to re-checkout their repos type of crap) |
| 15:28.27 | dlynes | asterisk is switching to git? |
| 15:29.08 | jkroon | no. i'm just using git to checkout the svn repo because it's simpler to keep my own repo on top of what's happening upstream with it. |
| 15:29.37 | jkroon | my other (non-ideal) option is to explicitly checkout just trunk/ and not have the tags available at all. |
| 15:29.52 | Katty | hi |
| 15:30.04 | dlynes | herro katty |
| 15:30.09 | jkroon | hi |
| 15:30.27 | Katty | is it time for breakfast |
| 15:30.34 | dlynes | long since past |
| 15:30.38 | dlynes | it's almost noon |
| 15:30.43 | Katty | :< |
| 15:30.58 | Katty | time for lunch? :> |
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| 15:40.37 | VoIP-Penguin | Lunch, and 10:30? |
| 15:42.02 | *** join/#asterisk Whtsup (~sssi@203.81.226.170) |
| 15:42.06 | Whtsup | hello |
| 15:42.33 | Whtsup | if i recompile asterisk can i use my old config or it will be change |
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| 15:51.14 | VoIP-Penguin | Depends if you clean out your old config or change it before you compile again. |
| 15:54.43 | Whtsup | i want to use my old config |
| 15:54.55 | Whtsup | soo wht shud i do in recompile |
| 15:55.03 | Whtsup | ./configure make make install right ? |
| 15:56.04 | VoIP-Penguin | What is your reason to recompile? |
| 15:56.18 | Whtsup | installing speex |
| 15:56.26 | Whtsup | codec |
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| 15:58.36 | dlynes | Whtsup, it also depends on what you're upgrading to, and what you're upgrading from, whether you'll need to change your config, or not |
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| 16:03.10 | VoIP-Penguin | On that note, Linux has an "oldconfig" make target to incorporate an old config with a new version that has new/more options... does asterisk have such a thing? |
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| 17:02.38 | otavio | Can someone help me to identify why my asterisk doesn't load? |
| 17:02.45 | otavio | How I can identify it? |
| 17:03.44 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 17:05.16 | russellb | otavio: run it from the console... # asterisk -vvvvvvvvgc |
| 17:05.22 | russellb | and see what the last error is |
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| 17:09.00 | Aven | Somebody uses SFA? |
| 17:09.29 | russellb | Yes, somebody does :-) |
| 17:10.31 | dlynes | Aven, lots of strange people do |
| 17:10.49 | Aven | I can not understand then, but after the taken place conversation there are errors |
| 17:10.51 | dlynes | Aven, apparently it's one of digium's best selling products for whatever reason |
| 17:11.32 | dlynes | Aven, ah...I guess you're one of them :) |
| 17:11.35 | Aven | flood in console ^( |
| 17:11.42 | Aven | [Mar 20 20:09:50] WARNING[9045]: tcptls.c:254 ast_tcptls_server_root: Accept failed: Bad file descriptor |
| 17:12.27 | dlynes | Aven, Just a quick question...has one end of the call already hung up? |
| 17:12.43 | russellb | well, make sure you're using the latest version |
| 17:12.49 | Aven | Asterisk 1.6.2.6, skypeforasterisk-1.6.2.0_1.0.9.2-x86_32.tar.gz |
| 17:12.59 | russellb | aside from that, contact http://www.digium.com/en/supportcenter/ |
| 17:13.52 | *** join/#asterisk Shazaum (~Shazaum@189.73.100.45) |
| 17:14.24 | Aven | Simple call. Skype the client calls on my account, I do not lift a tube. Skype the client ceases to call, but at me are sonorous continue to go. If to lift a tube and to put, all dies... |
| 17:15.23 | Aven | Similar it has begun after updating till 1.6.2.6 |
| 17:15.31 | dlynes | Aven, sonorous? |
| 17:16.00 | ManxPower-work | Aven, very few people on this channel use SFA. I think that's why russellb pointed you to the Digium Support Center |
| 17:16.25 | Aven | dlynes, ipphone continues to call |
| 17:16.26 | russellb | well even if people use it, it's a commercial product, and our support team will be able to help the best |
| 17:16.52 | Aven | ok, thx) |
| 17:18.45 | Aven | excuse for my English, Russian I know much better :) |
| 17:19.21 | russellb | it's fine. your English is _much_ better than my Russian. |
| 17:20.05 | Aven | =)) |
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| 17:39.32 | jkroon | tzafrir, the trick is to use git svn fetch -r ${bad_rev}+1:HEAD |
| 17:40.17 | antiwire | haha |
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| 17:46.49 | doolittlework | hi there people anyone active? |
| 17:47.18 | Dovid | nope |
| 17:47.22 | Dovid | whats up / |
| 17:47.22 | doolittlework | lol |
| 17:47.23 | Dovid | ?* |
| 17:47.47 | doolittlework | nothing mush i need someone to help me understand some code, can i paste it? |
| 17:48.13 | russellb | ~pb |
| 17:48.14 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 17:48.16 | Dovid | use pastebin |
| 17:48.17 | Dovid | lol |
| 17:48.59 | Shazaum | combo break |
| 17:50.37 | doolittlework | http://pastebin.com/03uDWyAV |
| 17:53.43 | Shazaum | ... |
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| 17:57.03 | [TK]D-Fender | doolittlework: You're going to point out the line in question... RIGHT? |
| 17:57.18 | Dovid | lol |
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| 17:58.04 | patrb | If I make asterisk run as user 'root' and group 'apache', will that write my voicemail files with that user and group? (im trying to change my voicemail files to be owned by group apache) |
| 17:58.40 | patrb | looking at the following init script: http://www.voip-info.org/wiki/index.php?page=Asterisk+non-root |
| 18:00.23 | doolittlework | ok here goes the fist line of the mainout contaxt points to recording-check context |
| 18:01.35 | doolittlework | _[*0-9].,1,Gosub(recording-check,${EXTEN},1(${recordoutbound})) what does the 1(${recordoutbound}) do? |
| 18:02.35 | *** join/#asterisk SeriousMatters (~Sirius@87.114.35.31.plusnet.thn-ag3.dyn.plus.net) |
| 18:02.49 | SeriousMatters | Hi, |
| 18:02.57 | doolittlework | hi |
| 18:03.16 | SeriousMatters | Any recommendation on how to partition for an Asterisk-Thirdlane box? |
| 18:04.16 | [TK]D-Fender | SeriousMatters: Very carefully |
| 18:04.38 | [TK]D-Fender | doolittlework: how should we know? It isn't in the pastebin |
| 18:04.45 | doolittlework | yes |
| 18:05.08 | ManxPower-work | doolittlework, what is the value of the ${recordoutbound} variable. |
| 18:05.21 | doolittlework | [TK]D-Fender: its on the first line |
| 18:05.43 | [TK]D-Fender | doolittlework: whatever sets it. Perhaps its set against a PEER |
| 18:05.51 | [TK]D-Fender | doolittlework: Considered looking at the peer? |
| 18:06.08 | jblack | has anyone fiddled with xtreemfs? |
| 18:06.39 | SeriousMatters | Is everything in one ext3 partition acceptable? |
| 18:06.43 | doolittlework | i take it points to context recording-check is the ${recordoutbound} variable $[ARG1} when passed on with the Gosub command? |
| 18:06.49 | jblack | SeriousMatters: yes |
| 18:07.00 | ManxPower-work | SeriousMatters, Whatever your DISTRO recommends. |
| 18:07.21 | [TK]D-Fender | SeriousMatters: If you feel like it |
| 18:08.11 | ManxPower-work | doolittlework, no. ARG1 would be the ehe VALUE of ${recordoutbound}. There's a big difference, which I would have explained if you had answered my question. |
| 18:08.45 | doolittlework | ManxPower-work: whwere do i find this |
| 18:09.18 | [TK]D-Fender | doolittlework: Either somewhere else in the dialplan being executed, or in a peer entry that sets it |
| 18:09.22 | ManxPower-work | It would show on the CLI. But I would have hoped you would know. |
| 18:09.40 | ManxPower-work | doolittlework, Didn't you write the dialplan? |
| 18:10.03 | [TK]D-Fender | ManxPower-work: Don't recognize it? :) |
| 18:10.26 | ManxPower-work | [TK]D-Fender, the name looks GUIish, but no it doesn't |
| 18:10.41 | [TK]D-Fender | ManxPower-work: Smells familiar... |
| 18:10.42 | ManxPower-work | Maybe I really should stop coming here. All we seem to get these days is GUI questions. |
| 18:10.58 | patrb | i had a good questions |
| 18:11.02 | patrb | If I make asterisk run as user 'root' and group 'apache', will that write my voicemail files with that user and group? (im trying to change my voicemail files to be owned by group apache) |
| 18:11.18 | ManxPower-work | patrb, I don't know. It is trivial to try and see. |
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| 18:11.47 | patrb | dont have a test box at the moment..only production, trying to research so I can test later |
| 18:11.51 | ManxPower-work | You should also check the sip.conf.sample to see if there are any obvious options to try. |
| 18:12.22 | ManxPower-work | why don't you just run Astrisk as user and group apache? |
| 18:12.30 | jblack | my lord, please don't do that. |
| 18:12.42 | patrb | bad plan |
| 18:12.48 | ManxPower-work | Asterisk will do the rootish stuff, then change it's userid to apache. |
| 18:12.58 | jblack | There's got to be something better than running two daemons under teh same uid. |
| 18:12.59 | *** join/#asterisk seanjohn (~john@173.50.101.10) |
| 18:13.13 | ManxPower-work | jblack, when dealing with things like web servers, I've never found one. |
| 18:13.14 | seanjohn | where is the sql file to setup the database for asteriskcdrdb? |
| 18:13.37 | jblack | For the _very_ least, he can look at suexec. |
| 18:13.39 | [TK]D-Fender | seanLook in your tarball |
| 18:13.40 | seanjohn | i can't find it in asterisk-addons source |
| 18:13.41 | ManxPower-work | seanjohn, I hope it would be included with Asterisk. |
| 18:13.48 | jblack | and for one virtual host, have the webserver become asterisk. |
| 18:14.09 | ManxPower-work | jblack, that works as well. |
| 18:14.24 | ManxPower-work | I think that is how most guis do it. |
| 18:14.26 | jblack | hell, apache2 makes it easy with assignuserid |
| 18:14.39 | seanjohn | can someone point me to it? |
| 18:15.21 | ManxPower-work | looks like the FreePBX/Trixbox and their ilk run the web server as "asterisk" |
| 18:15.50 | jblack | It kinda makes sense on single-purpose boses. |
| 18:15.54 | jblack | boxes, that is, |
| 18:16.38 | jblack | but on a multipurposed server.. do you really want * running as the same uid as the unrelated drupal installation 2 dirs over? Just.. ewww. |
| 18:16.59 | ManxPower-work | jblack, of course not. But your asterisk box should never be a multipurposed server |
| 18:17.35 | [TK]D-Fender | seanjohn: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-addons-1.6.2.0.tar.gz |
| 18:17.44 | jblack | That's the difference between ideals and the reality of funding. |
| 18:18.09 | ManxPower-work | jblack, It's the difference between being smart and being stupid. |
| 18:18.22 | jblack | You should never have two discrete tasks on the same server, but that's cost prohibitive. |
| 18:18.34 | doolittlework | ManxPower-work: found it in sip.conf the variale for ${recordoutbound} setvar=recordoutbound=yes |
| 18:18.45 | seanjohn | [TK]D-Fender: i have the source, im looking through all the directories for the sql file |
| 18:18.49 | ManxPower-work | jblack, then use a VM or spend the massive extra time and effort to make it work all on one server. |
| 18:19.05 | jblack | Whoah, backup partner. You're conflating poor with intelligence, and I know you insulted several people here that are not stupid. |
| 18:19.21 | ManxPower-work | The amount of time to make the same web server handle GUIs and management for multiple applications would be just staggering. |
| 18:20.15 | seanjohn | i found it; its in the freepbx source |
| 18:20.28 | [TK]D-Fender | seanjohn: It isn't in a ".sql" file. Try looking for one with an obvious name |
| 18:21.10 | [TK]D-Fender | seanjohn: Then go ask in their channel. Their requirements are probabably more than the minimum. |
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| 18:44.44 | ChannelZ | yawns |
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| 19:08.54 | redax | hi |
| 19:11.07 | redax | how to debug AGI scripts? 'agi set debug on' wont help |
| 19:11.38 | Shazaum | agi set debug? |
| 19:12.03 | redax | if I start the script manually It returns SET VARIABLE TOCALL "335" |
| 19:12.25 | redax | Shazaum: agi set debug on does nothing |
| 19:12.43 | Shazaum | 'agi set debug' |
| 19:13.02 | Qwell | does nothing or gives an error? very big difference |
| 19:13.28 | redax | Shazaum: 'agi set debug' gives usage. |
| 19:14.02 | VoIP-Penguin | Then agi set debug probably turns it on. |
| 19:14.05 | redax | Qwell: after 'agi set debug on' I call the given extension and I don't see any debug at all |
| 19:14.13 | VoIP-Penguin | And it will be waiting for agi to debug. |
| 19:14.21 | Qwell | do you have debug messages going to console? |
| 19:14.28 | redax | not at all |
| 19:14.35 | redax | just the dialplan |
| 19:14.36 | Qwell | check logger.conf |
| 19:14.53 | redax | nah. so the dialplan messages. |
| 19:14.59 | redax | but nothing from the agi debugging |
| 19:15.11 | Qwell | dialplan isn't debug |
| 19:16.19 | redax | so it should be there... if I turn on sip debug, I see the sip debug messages in the console, so if I turn on agi debug messages it should be there too_ |
| 19:17.08 | redax | asterisk version is 1.6.0.25 bw |
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| 19:23.00 | redax | got it :) pbx.c: The application delimiter is now the comma, not the pipe. Did you forget to convert your dial |
| 19:23.02 | redax | plan? (AGI(autocallerid.pike|06706236818) |
| 19:23.04 | redax | ... |
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| 21:30.58 | LemensTS | anyone got a link on how to setup checking of voicemail remotely by hitting * during the voicemail message? |
| 21:31.32 | ChannelZ | eh? |
| 21:33.05 | ChannelZ | I don't understand the question |
| 21:33.57 | LemensTS | if you call a sip phone and get their voicemail, you can hit * to reach VoiceMailMain to check the voicemail remotely. |
| 21:34.00 | LemensTS | On freepbx |
| 21:36.56 | ChannelZ | hmm.. I just create an extension to call VoiceMailMain with no mailbox so they get asked |
| 21:37.51 | ChannelZ | but I imagine they are using 'exitcontext' in voicemail.conf to send it to a context that does a VoiceMailMain |
| 21:38.01 | LemensTS | huh lol --> http://www.mail-archive.com/asterisk-users@lists.digium.com/msg238172.html |
| 21:38.29 | LemensTS | see they have an 'a' on that last line. I had * there, when i switch it to 'a' it works. What does 'a' mean? |
| 21:38.53 | ChannelZ | asterisk perhaps |
| 21:39.08 | LemensTS | well long as it works i dont care then ha. thanks |
| 21:39.22 | ChannelZ | http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 21:40.35 | LemensTS | ChannelZ: ahh cool well i learned something new, never kinew about that before :) |
| 21:41.49 | ChannelZ | although you can use * in extension names so 'a' is somewhat special for this task it seems |
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| 21:54.24 | ruben23 | hi |
| 21:54.26 | ruben23 | there |
| 21:55.41 | Dovid | hi |
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| 22:10.30 | ruben23 | hi can i used zaptel instead of dahdi for asterisk ver. 1.4.27 - 28...? |
| 22:15.16 | voipmonk | read the changelog |
| 22:15.44 | voipmonk | why not use dahdi? |
| 22:16.47 | ruben23 | voipmonk:its just there are strange happening when i install asterisk 1.4.27-----asterisl CLI is not displaying colors for Notice, error, Agi. |
| 22:16.59 | ruben23 | i want ot isolte if dahdi is causing it. |
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| 22:28.12 | voipmonk | what happened to the person who was supposed to come in and help you with that? |
| 22:30.59 | ruben23 | voipmonk: he is still put on hold..higher management problem.. |
| 22:36.10 | voipmonk | meanwhile the world is passing him by |
| 22:36.11 | voipmonk | :) |
| 22:38.36 | voipmonk | WAIT A MINUTE |
| 22:38.39 | voipmonk | wrong window |
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| 22:44.37 | ruben23 | <PROTECTED> |
| 22:50.58 | Katty | i bought an ipod nano |
| 22:51.06 | ChannelZ | ruben23: 1.4.27 isn't the cause of no colors in the console |
| 22:51.21 | ChannelZ | err dahdi I mean |
| 22:51.40 | Katty | what does cause the colors, anyway |
| 22:52.04 | Katty | CAN I MAKE MINE PINK |
| 22:52.17 | ChannelZ | Well it looks to see if the console supports colors and uses them if it does |
| 22:52.23 | Katty | oh :< |
| 22:52.48 | Katty | anyone here have an ipod nano |
| 22:52.55 | ChannelZ | The problem is if you launch Asterisk at startup it's under a terminal that doesn't support colors |
| 22:53.19 | ruben23 | ChannelZ: extraputty..? |
| 22:53.32 | ChannelZ | so when you asterisk -r you don't get any because you're really just getting a dump of what the main asterisk process is outputting.. even though YOU are in a color-capable terminal when you asterisk -r |
| 22:54.02 | ruben23 | ChannelZ: what are the corrections for this.. |
| 22:54.53 | ChannelZ | well one way which may or may not be ideal for you is to run safe_asterisk instead of asterisk |
| 22:55.27 | ChannelZ | safe_asterisk doesn't launch asterisk in detach mode |
| 22:55.58 | ChannelZ | so it maintains the terminal from safe_asterisk and the colors should continue to work |
| 22:56.38 | ruben23 | ChannelZ:ow ok i have tried asterisk 1.4.21 ---colors are ok, just exprience it with 1.4.27 - 28 |
| 22:56.57 | ChannelZ | are you building from source or using packages? |
| 22:57.14 | ruben23 | buidling source |
| 22:57.24 | ChannelZ | are you running 'make config'? |
| 22:57.28 | ruben23 | yes |
| 22:57.32 | ChannelZ | that's probably why |
| 22:57.42 | ChannelZ | the init scripts changed at one point or another to run asterisk instead of safe_asterisk |
| 22:58.29 | ruben23 | ow ok, so i shouldnt run it..? but how do i start my asterisk automatically on startup..? |
| 22:59.23 | ChannelZ | No I just mean that when you run 'make config' it installs init scripts to start up asterisk.. so the reason you see a change is probably because the old version init script was running safe_asterisk but newer versions run the main asterisk process |
| 22:59.51 | ruben23 | ok got it. |
| 23:00.05 | ChannelZ | you can just contiue to use the init script from an old version and not 'make config' which overwrites it |
| 23:00.59 | ruben23 | old versiosn init script, with new version os asterisk right.. |
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| 23:04.34 | ChannelZ | yeah or just hack the 'new' init script to run safe_asterisk instead |
| 23:08.16 | ruben23 | <PROTECTED> |
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