| 00:06.23 | *** part/#asterisk nny (~Scott@cpe-071-076-058-253.sc.res.rr.com) |
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| 00:13.01 | telnettech | wanting to learn more indepth on how to use the AGI and AMI part of asterisk. anybody know where besides the book? |
| 00:14.42 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 00:22.38 | paulc | I'm building something in the dialplan, and thinking "this might be a good idea to give AEL a whirl" - it will jive better with the target audience I'll be showing this to. In my AEL I have "SomeVariable=1" and I see it gets translated to MSet(SomeVariable=1)... leading me to stumble upon MSet vs Set. Do I need to care about this (using 1.6.2.6) or should I be explicitly saying "Set(SomeVar=1)" in my AEL code? |
| 00:22.44 | paulc | is an AEL noob |
| 00:24.39 | *** join/#asterisk geoffmcc123 (~Geoff@cpe-72-231-200-14.buffalo.res.rr.com) |
| 00:25.18 | hardwire | any way to see the permit/deny options for a peer? |
| 00:25.21 | hardwire | sip |
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| 00:25.32 | hardwire | I can't seem to find it |
| 00:25.58 | geoffmcc123 | having trouble clarifiying this - net gives alot of same examples so not getting it. - setting up nat : localnet= gateway or trixbox address |
| 00:28.03 | geoffmcc123 | no one? |
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| 00:52.20 | kerx | hi all, anyone know how i can use AMI to get call duration for an agent? |
| 00:54.50 | encinoman | Anyone have any advice on implementing call conferencing? |
| 00:54.58 | ChannelZ | MeetMe() |
| 00:55.48 | encinoman | ChannelZ: OK, thanks. I'll take a look. |
| 00:56.11 | kerx | ChannelZ, any idea on how to get call duration of an agent while they are on phone call? |
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| 01:22.14 | torrio | Hi guys, when using "Playback(...)" or "Background(...)" and pressing a digit before the end of the sound-file, the i-extension catches and plays another file, but that file sounds distrubed ! What could fix that ? |
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| 01:41.16 | paulc | Is the problem that the i extension plays something, and it sounds wrong? Or that the i extension shouldn't fire when someone's pressing a key? |
| 01:41.28 | paulc | torrio: delayed reply there ;) |
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| 01:49.42 | encinoman | ChannelZ, you still there? |
| 01:50.51 | encinoman | What I REALLY would like to implement for call conferencing is: 1) You call into the PBX 2) press *number 3) Dial another number, 4) add them to the conference. |
| 01:51.13 | encinoman | meetme() doesnt seem to do that. |
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| 01:57.44 | Andrew_M | I plugged into the network a new polycom ip600, and it's stuck at "Running App = sip.ld" I was hoping to get a DHCP IP address, then use the phone's web interface for configuration, but the phone gets stuck at that screen. Any ideas? |
| 02:08.20 | paulc | encinoman: You could build it with Local channels and a bit of dial plan logic |
| 02:08.45 | paulc | Andrew_M: How long's it sit there - does it ever boot up or does it just sit there forever? |
| 02:11.04 | Andrew_M | paulc: forever... over 5-10 mins. |
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| 02:14.20 | Andrew_M | paulc: It's the first Polycom set on the network, and I don't have any conf servers, neither ftp, nor tftp. There is DHCP, and an Aastra phone booted fine with it. |
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| 02:16.16 | paulc | Andrew_M.. weird.. I'd set the boot up server thing to Static (DHCP menu on the phone, as it's powering up) and point it to an FTP server where it can pull some config from |
| 02:16.28 | paulc | but that's me - I like my Polycoms to provision from config files, not web |
| 02:16.41 | paulc | but.. doing that (even with no config) will give you the latest boot ROM and firmware |
| 02:17.07 | paulc | You can get the latest boot + firmware here: http://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip600.html |
| 02:17.32 | paulc | I gotta head home, 7:15pm and I'm still at the office.. but I'll be back in 45 to 60ish if you're still stuck and we can sort it out. |
| 02:17.48 | Andrew_M | paulc: I want to do that eventually, but it's a small test system, and I would like to use manual config. |
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| 02:19.09 | Andrew_M | paulc: Thank you... I am on Eastern time, I will be back from the office tommorrow where the phone is. |
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| 02:27.53 | philipp10 | Hello |
| 02:28.18 | ChannelZ | ohell |
| 02:28.53 | philipp10 | Does anybody know how define outbound routes to route traffic for different trunks via different IPs on a Virtual Server? |
| 02:29.13 | philipp10 | 'till now I haven't found any option to do this |
| 02:33.07 | philipp10 | I think I'll try it with this question in the official board |
| 02:33.10 | p3nguin | "man route" |
| 02:33.32 | philipp10 | I mean something different |
| 02:34.30 | philipp10 | If you want to connect to mysip-provider.com for customer 1 via ip xxx.xxx.xxx.xxa |
| 02:34.51 | philipp10 | and then you want to connect to mysip-provider.com for customer 2 via ip xxx.xxx.xxx.xxb |
| 02:34.55 | philipp10 | how to solve this problem? |
| 02:35.12 | philipp10 | a normal route doesn't match the problem because the provider adress is the same |
| 02:35.28 | philipp10 | but the trunk lines aren't the same because billing at the providers side is identified by IPs |
| 02:35.41 | philipp10 | so the different customers on asterisk have to use different IPs for outgoing calls |
| 02:35.58 | philipp10 | that's the probelm |
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| 02:45.51 | philipp10 | (mysip-provider.com is only an example, this adress doesn't exist really) |
| 02:50.05 | carrar | philipp10, run multiple instances of asterisk each bound to a different IP |
| 02:50.44 | carrar | is about the only way you are gonna control the source IP unless you use iptabes to re-write your source IP |
| 02:51.02 | carrar | that would be messy and probably not work |
| 02:57.57 | *** join/#asterisk TJNII (~TJNII@207.189.199.62) |
| 02:59.22 | Naikrovek | or *gulp* two asterisk servers. |
| 02:59.41 | Naikrovek | all solutions to that problem sound hackish and problematic |
| 03:00.49 | Naikrovek | a better way may be to bill the customers yourself, via asterisk call logging. you get one bill from your provider, you sort out calls via the CDR, then bill your customers appropriately |
| 03:01.12 | Naikrovek | room for markup this way, and it's expandable to a lot of customers |
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| 03:10.59 | Micc_ | has anyone here succesfully used a handytone to send a fax with t.38 to asterisk? |
| 03:11.25 | Micc_ | I have no problem sending with an spa8000, but the handytone negotiates t.38 then hangs up. |
| 03:12.21 | Micc_ | digium said asterisk sends the udptl packet and gets no response from the handytone. But I don't think its a firewall issue or the spa8000 wouldn't work either, unless the spa8000 is doing nat traversal better. |
| 03:13.57 | Naikrovek | the handytone not responding is probably not a firewall problem. it's a handytone problem |
| 03:14.21 | Naikrovek | but i'm guessing |
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| 03:42.56 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.7, 1.6.1.19, 1.6.0.27 (2010/05/04), 1.4.31 (2010/05/04), *-Addons 1.6.2.1, 1.6.1.3, 1.6.0.5, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 03:42.59 | Micc_ | maybe that will help. |
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| 04:45.54 | Micc | It looks like the problem may have been fixed in a firmware update from grandstream. I'm testing it now. |
| 04:46.13 | Micc | release notes said something about failing after invite with t38faxrate |
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| 04:54.37 | ruben23 | hi anyone familiar with this warning--> on asterisk 1.4 ---> http://pastebin.com/0up3j6xL |
| 04:54.42 | ruben23 | any idea on this..? |
| 04:58.10 | Naikrovek | ruben23: looks like you have some AMI client called "sendcron" and it didn't behave |
| 04:58.40 | ruben23 | Naikrovek: is this an issue..? can this be corrected..? |
| 04:59.23 | Naikrovek | if i'm right (let's not assume that i am) then it's a problem with whatever program logs into the AMI with the user "sendcron" |
| 04:59.40 | Naikrovek | find out what app that is and see if you can update it maybe |
| 05:00.44 | Naikrovek | if you don't know what the sendcron AMI user is then I would remove that entry from /etc/asterisk/manager.conf and forget about it |
| 05:00.51 | Naikrovek | but listen |
| 05:00.56 | Naikrovek | im super tired and not thinking clearly |
| 05:00.59 | Naikrovek | i could be way off |
| 05:01.09 | ruben23 | its ok |
| 05:01.14 | ruben23 | tahnks for the suggestion |
| 05:02.56 | *** part/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 05:04.57 | gpled | anyone have an @ekiga.net sip number? |
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| 05:14.42 | Jumpie | hey guys |
| 05:15.02 | Jumpie | is there a recommended site/resource that helps a from scratch build step by step? including centos tweaks? |
| 05:15.07 | Jumpie | is that listed in ~book? :P |
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| 05:46.08 | checkerboard123 | hi all |
| 05:46.33 | checkerboard123 | It's been a while that I have been trying to figure out to use asterisk as a phone confirmation system |
| 05:46.48 | checkerboard123 | So is there anyone in here with any experience with it |
| 05:46.50 | checkerboard123 | ? |
| 05:46.55 | checkerboard123 | Any help will be appreciated |
| 05:47.39 | *** join/#asterisk Gopal (~Miranda@61.12.17.170) |
| 05:48.21 | ChannelZ | A confirmation for what? |
| 05:49.24 | checkerboard123 | like say a user has a pin number and a verification is required through phone. So an automatic call is sent to that user and he could dial in his pin number for the verification |
| 05:49.46 | checkerboard123 | Is there any add on or something for that?? |
| 05:49.53 | checkerboard123 | or is it a feature to be added |
| 05:50.12 | checkerboard123 | I haven't found any implementation like this using asterisk |
| 05:50.47 | ChannelZ | Probably have to build it yourself |
| 05:50.50 | Gopal | checkerboard123: if the pin number is wrong what happen to the automatic dialed call? |
| 05:53.40 | Jumpie | hey ChannelZ ever used astlinux ? |
| 05:54.07 | checkerboard123 | hey gopal the user will be informed of invalid pin number |
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| 05:54.14 | checkerboard123 | anyways thanks for the help ChannelZ |
| 05:54.51 | Gopal | checkerboard123: you have to use AGI and need to build yourself for this |
| 05:55.20 | ChannelZ | Jumpie: nope |
| 05:55.40 | checkerboard123 | ok thanks for the info Gopal |
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| 06:09.07 | Micc | I think I figured out my handytone problem. It seems to require a different type of error correction. fec didn't work. changing to redundancy I think fixed it. No more hangup right after t38 negotiation. |
| 06:10.12 | checkerboard123 | hey gopal is there any other open source software that provides the phone confirmation feature |
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| 06:10.57 | Gopal | checkerboard123: no |
| 06:11.22 | checkerboard123 | any suggestions or paperworks that will help to get started with to add this featue |
| 06:11.38 | checkerboard123 | I can get nothing more than a demo of this feature |
| 06:11.44 | checkerboard123 | I could see one in phoneconfirm.com |
| 06:11.59 | checkerboard123 | But I need the technical details and all that will help |
| 06:12.05 | checkerboard123 | Gopal any suggestions?? |
| 06:12.12 | Gopal | checkerboard123: you need to write AGI script using java or PHP |
| 06:13.08 | Gopal | checkerboard123: JAVA with asterisk manager connection |
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| 06:14.10 | checkerboard123 | ok thanks gopal |
| 06:16.19 | kaldemar | checkerboard123: originate a call to the user with either AMI or a call file and connect him to an extension that asks for the PIN number and checks it. this is all doable in extensions.conf. |
| 06:18.20 | kaldemar | checkerboard123: use app Playback to give the user audio feedback and for example app Read to get the PIN. checking the PIN depends on where you want to store them. |
| 06:20.41 | checkerboard123 | thanks kaldemar for the info |
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| 07:02.00 | decaffeine | good morning everyone |
| 07:02.46 | MAbbas | Hi, in a call center setup. I am using AgentLogin() for agent login and for incoming calls I am using Answer() and Queue(). I am testing this setup using Winsip for agents as well as calls. However after sometime all my agent are logged out and asterisk become un-responsive. I am using asterisk version 1.6.2.7. |
| 07:16.10 | Jumpie | ChannelZ im kinda looking for a solution for rapid deployment of servers |
| 07:16.12 | Jumpie | preferably in a VM |
| 07:16.17 | Jumpie | thats why i wal ooking at astlinux |
| 07:16.20 | Jumpie | any suggestions? |
| 07:16.33 | Jumpie | inflate my image |
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| 07:18.18 | ChannelZ | sorry no idea |
| 07:20.34 | Jumpie | k |
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| 07:36.41 | jkroon | hi guys, with whom do I take up bugs in the asthostid code? digium direct I take it? |
| 07:37.27 | jkroon | I've now gone as far as stracing the asthostid program and there is definitely a bug in the way they deal with findind an ethernet mac to base the asthostid on. |
| 07:37.45 | ChannelZ | Yeah.. it's their code |
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| 07:39.00 | smooth_penguin | HEY |
| 07:39.19 | ChannelZ | HEY! Do you like hot dogs? |
| 07:39.45 | ChannelZ | If you were a hot dog, and you were starving, would you eat yourself? |
| 07:41.11 | smooth_penguin | no |
| 07:41.28 | smooth_penguin | ChannelZ, I like pizzas |
| 07:45.41 | ChannelZ | Me too. |
| 07:46.33 | smooth_penguin | wheres Katty |
| 07:51.15 | carrar | sleeping |
| 08:00.39 | smooth_penguin | ahk |
| 08:00.56 | MAbbas | hay, all. My asterisk just crashed following error was printed. ERROR[26774]: astobj2.c:321 __ao2_ref: refcount -1 on object 0xb7659ff0 |
| 08:01.07 | MAbbas | any ideas? |
| 08:01.08 | smooth_penguin | man I think Im going to climb that telephone pole myself and fix my phoneline |
| 08:01.40 | carrar | yeah |
| 08:01.45 | carrar | video tape it |
| 08:02.18 | carrar | MAbbas, try restarting asterisk |
| 08:04.18 | MAbbas | <carrar> It restarted fine. But any idea what might have caused it? |
| 08:04.44 | carrar | Do a backtrace on your core file against the source |
| 08:04.53 | kaldemar | MAbbas: if you can repeat the behavior, see doc/backtrace.txt in the source package for backtrace instrunctions. |
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| 08:09.25 | MAbbas | <kaldemar> I did not started asterisk with -g option :( |
| 08:11.29 | kaldemar | as said, run asterisk so that it can produce a useful core dump and try to repeat the behavior. you could also search mantis (issues.asterisk.org) for a similar issue if you know what was happening during the crash. |
| 08:11.48 | kaldemar | that error message is pretty useless by itself. |
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| 08:44.37 | sun28 | moin |
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| 09:13.54 | Jumpie | ~book |
| 09:13.55 | 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 |
| 09:13.59 | Jumpie | lol ty |
| 09:15.31 | *** join/#asterisk joobie (~joobie@CPE-121-220-3-162.lnse1.win.bigpond.net.au) |
| 09:16.04 | *** join/#asterisk albasheers (~basheer@89.148.43.51) |
| 09:16.29 | *** join/#asterisk albasheers (~basheer@89.148.43.51) |
| 09:16.53 | *** part/#asterisk albasheers (~basheer@89.148.43.51) |
| 09:19.34 | *** join/#asterisk albasheers (~basheer@89.148.43.51) |
| 09:22.38 | joobie | guys im looking at a GSM card to receive SMS's .. the OpenVox G400P is what ive been looking at, though it seems to be a patch / hack job to get SMS receive wokring.. is there a card that's better suited for this function? |
| 09:25.58 | *** join/#asterisk jrz (~jrz@a190165.upc-a.chello.nl) |
| 09:28.25 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
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| 09:45.47 | decaffeine | I'm completly lost |
| 09:46.07 | decaffeine | don't know what to do :( |
| 09:47.13 | decaffeine | Can somebody help with incoming and outgoing sip providers? |
| 09:49.40 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
| 09:58.09 | *** join/#asterisk aceio (~5188dc77@gateway/web/freenode/x-ulzisnlrnyakhtof) |
| 09:58.20 | torrio | Hi guys, i wanted to use "Playback(...)" or "Background(...)" and the i-extensition to catch non-used digits. That's working, but when playing a file in i,2,Playback("Sorry, you pressed the wrong key") that sounds distrubed or blurred ! How to fix that ? |
| 09:58.38 | *** join/#asterisk pif (~ldm@LSt-Amand-152-33-2-89.w193-251.abo.wanadoo.fr) |
| 10:10.05 | aceio | hi all |
| 10:10.56 | aceio | iam unable to make calls over iSDN |
| 10:11.02 | aceio | please help |
| 10:11.40 | *** join/#asterisk doolittlework (~d@196.211.34.2) |
| 10:12.30 | WIMPy | details? |
| 10:12.39 | *** join/#asterisk guyvdb_ (~guy@dsl-240-186-23.telkomadsl.co.za) |
| 10:14.37 | doolittlework | stange request, i have been pumping cdr to mysql and made a php program to view the call records, i now have a customer that has a small 10 extension system, and he does not wan the mysql --php page, he has his own management system, and whats me to send him the cdr in a txt file but he wants it a specific format, how does one edit the way the data is writen to the master.csv file. |
| 10:15.01 | guyvdb_ | what is an easy way to call and external script and get back a bool value from within the dial plan? Is it AIG? |
| 10:15.37 | kaldemar | torrio: what version and codec are you using? |
| 10:16.15 | guyvdb_ | doolittlework: how are you sending the data? Via a cron job? Why not just "transcode" the file via a script? |
| 10:18.21 | kaldemar | doolittlework: use cdr_custom and define the format in cdr_custom.conf |
| 10:18.34 | aceio | WIMPy: http://pastebin.com/Ri1tHb7Z this my output from cli |
| 10:19.09 | torrio | kaldemar, Asterisk 1.4.21.2~dfsg-3+lenny1 |
| 10:19.17 | kaldemar | aceio: have you installed dahdi? |
| 10:19.36 | doolittlework | guyvdb_ i still need to figure all this out, he wants it a specific format like date => extension numberdialed duration channel |
| 10:19.43 | aceio | kaldemar: yes i have |
| 10:20.01 | kaldemar | torrio: and codec of the sound files? gsm? |
| 10:20.11 | torrio | wav |
| 10:20.40 | WIMPy | aceio: You try to use a channeldriver that does not exist on your system. What hardware? What connection? What have you configured so far? |
| 10:21.01 | doolittlework | guyvdb_ the => and <= out is how they deside what is inbound and outbound |
| 10:21.11 | kaldemar | aceio: chan_dahdi.so is not loaded. what does 'module load chan_dahdi.so' in CLI tell you? |
| 10:22.21 | aceio | kaldemar: Module 'chan_dahdi.so' already exists. |
| 10:22.43 | aceio | WIMPy: i have the b410p isdn card |
| 10:23.17 | guyvdb_ | doolittlework: you could have a simple ruby or perl or python or whatever lang script that reads the CSV file daily, hourly, etc and rewrites it to the field order you require |
| 10:23.20 | kaldemar | aceio: what do your dahdi configurations look like? |
| 10:23.43 | aceio | kaldemar: okay 1 sec |
| 10:23.49 | torrio | kaldemar ... when i wait for a beep to insert a digit, its working right. The sound is only choppy when a digit is pressed while another one is played |
| 10:24.19 | torrio | or another sound file whatever |
| 10:28.19 | aceio | kaldemar: http://pastebin.com/E2Dz0uUh |
| 10:28.22 | aceio | WIMPy: http://pastebin.com/E2Dz0uUh |
| 10:29.27 | doolittlework | guyvdb_ can you write a script like that? |
| 10:29.42 | *** join/#asterisk MariusKarthaus (~quassel@s5593c029.adsl.wanadoo.nl) |
| 10:30.17 | kaldemar | aceio: this line is broken: signalling = bri_cpe point-to-point |
| 10:31.40 | aceio | kaldemar: so i to not need it |
| 10:32.37 | WIMPy | aceio: Yo do need it, but the correct name is bri_cpe_ptp IIRC. Certainly no spaces. |
| 10:32.51 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 10:32.58 | kaldemar | aceio: yes you need a signalling line to define the signalling of the channels. but it needs to at least conform to the syntax. |
| 10:33.32 | aceio | kaldemar: WIMPy okay |
| 10:34.07 | guyvdb_ | doolittlework: All you have to do is parse the file and change the field order. If you do not know how to do that, you could open it in a spreadsheet program such as open office calc and change the column order in there and save it. |
| 10:34.50 | *** part/#asterisk jkroon (~jkroon@dsl-244-55-145.telkomadsl.co.za) |
| 10:34.58 | doolittlework | thanks guyvdb_ i will look into that thanks for the steer in the right direction |
| 10:35.01 | kaldemar | doolittlework: or just use the tool that already comes with asterisk and spare yourself from custom scripting... |
| 10:36.01 | aceio | WIMPy: kaldemar signalling = bri_cpe_ptp |
| 10:36.45 | doolittlework | what tool is that kaldemar? |
| 10:37.12 | MariusKarthaus | Hi, I'd like to transfer a call to an extension from a AGI script after AGI answered the call. It should be possible as a human can do it from any sip phone, but how could I do that using AGI?. What I want is: (1) A calls our number (2) asterisk/agi answers with a voice menu "select x for a live person" etc (3) asterisk makes a call to several phones (depending on selection, time of day/night etc) (4) employee X answers at a line, may |
| 10:37.12 | MariusKarthaus | <PROTECTED> |
| 10:37.18 | kaldemar | guyvdb_: about external scripts, you can execute them with app System and examine the SYSTEMSTATUS variable, or use func SHELL to store stdout to a variable. |
| 10:37.28 | MariusKarthaus | In fact I'd like to write the logic of a secretary in AGI |
| 10:38.05 | doolittlework | thx guys |
| 10:38.07 | kaldemar | doolittlework: as i said earlier, the custom cdr module (cdr_custom) and define the format in cdr_custom.conf. see the sample config for an example. |
| 10:38.09 | Chainsaw | "I'm going to replace you with a small shel.. eh, AGI script." |
| 10:38.37 | WIMPy | MariusKarthaus: Doesn;t look like you need AGI for that. |
| 10:39.54 | MariusKarthaus | WIMPy: what whould you suggest ? |
| 10:40.19 | doolittlework | kaldemar: i know how to do that but how do i get rid of the , delimiter |
| 10:40.31 | MariusKarthaus | I'd like AGI because it will allow me to program complex logic into the 'secretary' |
| 10:40.37 | WIMPy | MariusKarthaus: Dialplan, maybe with a queue. |
| 10:41.04 | doolittlework | join #linux |
| 10:41.41 | guyvdb_ | kaldemar: thx |
| 10:43.02 | *** join/#asterisk Z_God (~julius@wlan225043.mobiel.utwente.nl) |
| 10:43.03 | kaldemar | doolittlework: sed < file.csv 's/\,/ /g' for example |
| 10:43.05 | MariusKarthaus | WIMPy: can I do Answer(), play() , Dial() in that order from the dialplan? |
| 10:43.12 | aceio | WIMPy: kaldemar still no joy |
| 10:43.53 | kaldemar | aceio: reload the module and see what it says. also see what dahdi show channels tells you when it is supposed to be loaded. |
| 10:43.58 | WIMPy | MariusKarthaus: That's what it's there for. |
| 10:44.27 | aceio | kaldemar: |
| 10:44.35 | aceio | kaldemar: okay |
| 10:47.10 | *** join/#asterisk Tim_Toady (~moi@193.92.247.185.dsl.dyn.forthnet.gr) |
| 10:47.33 | MariusKarthaus | WIMPy: hmm ok I was under the impression that something extra would need to be done in order to make that work after asterisk already 'talked' to the caller. I'm going to experiment |
| 10:48.45 | aceio | kaldemar: http://pastebin.com/D72btUD7 |
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| 10:58.58 | *** join/#asterisk iscario (~quassel@laureades.davout.pck.nerim.net) |
| 10:59.06 | aceio | WIMPy: i am still having this problem |
| 10:59.33 | aceio | looks like the problem is with my channels |
| 10:59.44 | aceio | please help |
| 10:59.52 | *** join/#asterisk iscario (~quassel@laureades.davout.pck.nerim.net) |
| 11:00.10 | *** join/#asterisk kazaa_lite (~eddie@212.183.140.48) |
| 11:00.30 | WIMPy | aceio: I don't understand, what you chan_dahdi is trying to do there. |
| 11:03.34 | aceio | WIMPy: i need to creat a channel that will allow exten => _X.,1,Dial(DAHDI/g1/${EXTEN},${DIAL_TOUT},Hrs(600) TWK) |
| 11:06.58 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 11:07.20 | aceio | http://pastebin.com/QsBBx8KX |
| 11:07.24 | aceio | WIMPy: http://pastebin.com/QsBBx8KX |
| 11:10.15 | WIMPy | aceio: That shouldn't be related to the error you're getting, but are you sure, you want hardhdlc? |
| 11:11.07 | aceio | WIMPy: no sure |
| 11:11.33 | aceio | WIMPy: what are my options |
| 11:12.41 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 11:16.59 | WIMPy | aceio: 'dchan' instead of 'hardhdlc' |
| 11:17.48 | aceio | okay |
| 11:18.21 | aceio | WIMPy: but what can i not see any channels |
| 11:19.20 | aceio | WIMPy: sorry |
| 11:20.34 | aceio | WIMPy: why i can't see any channel info when i execute dahdi show channels |
| 11:20.40 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 11:21.34 | *** join/#asterisk guyvdb_ (~guy@dsl-240-186-23.telkomadsl.co.za) |
| 11:21.56 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:21.56 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:22.20 | guyvdb_ | Is it still recommended to use a Proxy (such as AstManProxy) when using AMI with say max 20 sessions? |
| 11:23.27 | WIMPy | aceio: I have no clue. Looks pretty strange when dahdi moans about old zaptel config files. |
| 11:23.58 | aceio | WIMPy: great |
| 11:24.17 | aceio | this is looking like a rebuild |
| 11:36.38 | *** join/#asterisk coppice (~chatzilla@m121-203-209-202.smartone-vodafone.com) |
| 11:39.50 | *** join/#asterisk eliel (~eliels@200.43.6.244) |
| 11:43.54 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 11:48.18 | aceio | WIMPy: is the a way to help me configure the /etc/asterisk/chan_dahdi.conf to get some channels up |
| 11:48.42 | aceio | kaldemar: also kaldemar can you help |
| 11:49.20 | aceio | i really need some help |
| 11:54.14 | *** join/#asterisk kazaa_lite (~eddie@78-86-126-14.zone2.bethere.co.uk) |
| 11:57.48 | *** join/#asterisk supa_disko (~bleh@secure27.lnk.telstra.net) |
| 11:58.15 | [TK]D-Fender | bai bai |
| 12:04.32 | tzafrir_laptop | WIMPy, for the record, there's no chan_dahdi signalling called 'bri_cpe_ptp' |
| 12:04.44 | tzafrir_laptop | It's simply bri_cpe |
| 12:09.59 | *** join/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
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| 12:30.28 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-qawhiifelkoeruhf) |
| 12:32.14 | *** join/#asterisk Krillec (~sanxioned@fw2-ad-gw.orbital.net) |
| 12:37.46 | Krillec | does anyone have experience with the custom-context module for altering the from-did-direct context using asterisknow or ref to a usefull dump on tinternet with supporting information |
| 12:40.23 | devmod | Can I somehow whenever a SIP device isnt available, set a hint on a local extension saying the local extension isnt available as well ? |
| 12:42.59 | Gido-E | Krillec heu, i think everybody in this channel with some experience |
| 12:43.36 | Gido-E | Krillec if i decoded it good what you did type |
| 12:43.41 | Gido-E | asterisknow is? |
| 12:44.22 | [TK]D-Fender | ~freepbx |
| 12:44.23 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 12:44.25 | [TK]D-Fender | ^^^^^^^ |
| 12:45.29 | Krillec | cheers for the advice I will head over and try my luck there o/ thanks all |
| 12:51.18 | *** part/#asterisk Krillec (~sanxioned@fw2-ad-gw.orbital.net) |
| 12:52.48 | *** join/#asterisk aceio (~5188dc77@gateway/web/freenode/x-eghmxtychtnvpqeo) |
| 12:53.55 | *** join/#asterisk superbeef (~lanej@74.84.194.4) |
| 12:54.45 | tzafrir_laptop | aceio, 'signalling = bri_cpe_ptp' is invalid. Use 'signalling = bri_cpe' instead |
| 12:55.27 | aceio | tzafrir_laptop: okay i will try that now |
| 12:57.20 | aceio | tzafrir_laptop: still not working |
| 12:57.44 | tzafrir_laptop | "not working" == ? What does happen? |
| 12:58.23 | aceio | tzafrir_laptop: No channel type registered for 'DAHDI' at cli |
| 12:58.36 | tzafrir_laptop | module unload chan_dahdi.so |
| 12:58.42 | tzafrir_laptop | module load chan_dahdi.so |
| 12:58.53 | tzafrir_laptop | What version of asterisk is it? |
| 12:59.46 | aceio | 1.4.30 |
| 13:00.52 | aceio | tzafrir_laptop: sending you an output |
| 13:01.41 | aceio | tzafrir_laptop: http://pastebin.com/KFLFKX1g |
| 13:03.05 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:03.11 | *** join/#asterisk vader-- (~me@c-71-225-201-226.hsd1.nj.comcast.net) |
| 13:03.25 | tzafrir_laptop | hmm... "Unable to load zapata.conf" . Does that file exist? |
| 13:03.27 | *** join/#asterisk jblack (~jblack@pool-96-243-97-64.sctnpa.east.verizon.net) |
| 13:04.29 | aceio | tzafrir_laptop: so what do you think |
| 13:05.21 | tzafrir_laptop | aceio, see my question above |
| 13:06.38 | aceio | tzafrir_laptop: Asterisk 1.4.30 |
| 13:06.55 | [TK]D-Fender | [09:03]<tzafrir_laptop>hmm... "Unable to load zapata.conf" . Does that file exist? |
| 13:06.56 | *** join/#asterisk pabelanger (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 13:07.09 | aceio | yes |
| 13:07.19 | tzafrir_laptop | Can you pastebin it? |
| 13:07.30 | aceio | tzafrir_laptop: sure |
| 13:07.39 | [TK]D-Fender | aceio: "ls -la /etc/asterisk" <------------ |
| 13:08.15 | *** join/#asterisk coppice (~chatzilla@153.166.232.220.dyn.pacific.net.hk) |
| 13:09.37 | *** join/#asterisk Andrew_M_ (~c6537863@gateway/web/freenode/x-baaivjjzweqblabq) |
| 13:10.29 | aceio | http://pastebin.com/vcaE543a |
| 13:11.42 | aceio | [TK]D-Fender: i have included in the above link |
| 13:12.56 | [TK]D-Fender | cs:~# cat /etc/zapata.conf <--- should be /etc/asterisk |
| 13:13.31 | aceio | [TK]D-Fender: http://pastebin.com/vcaE543a |
| 13:13.48 | [TK]D-Fender | cs:~# cat /etc/zapata.conf <--- should be /etc/asterisk |
| 13:15.17 | aceio | okay |
| 13:18.05 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 13:18.25 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-147-211.dsl.stlsmo.sbcglobal.net) |
| 13:18.49 | aceio | [TK]D-Fender: are you saying you want to move /etc/zapata.conf to /etc/asterisk |
| 13:18.55 | LemensTS | is there any way to make cli tell what codec a call is using when it is placed? |
| 13:19.07 | [TK]D-Fender | aceio: yes |
| 13:19.25 | [TK]D-Fender | LemensTS: sip show channels |
| 13:19.51 | aceio | [TK]D-Fender: okay that is done now,should i reload |
| 13:19.59 | [TK]D-Fender | aceio: Restart * |
| 13:20.45 | aceio | okay |
| 13:21.15 | LemensTS | thx |
| 13:22.23 | aceio | okay done |
| 13:22.59 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:22.59 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:24.55 | aceio | [TK]D-Fender: stil no joy |
| 13:25.12 | *** part/#asterisk LemensTS (~LemensTS@adsl-70-238-147-211.dsl.stlsmo.sbcglobal.net) |
| 13:25.15 | [TK]D-Fender | aceand I don't see any backup <- |
| 13:25.16 | Jumpie | so finally converting back to vanilla asterisk |
| 13:25.19 | [TK]D-Fender | aceio: and I don't see any backup <- |
| 13:25.29 | Jumpie | bit more of a learning curve but tired of the bugs and nightmares with freepbx |
| 13:27.06 | aceio | [TK]D-Fender: my problem to being with i was, i was unable to make outside calls over isdn b410p card using the dahai drives |
| 13:27.42 | *** join/#asterisk rickead (~richard@hq.ipbase.co.uk) |
| 13:27.59 | [TK]D-Fender | aceio: where is the new CLI output from trying to load the modules? Where is the new LS dump? Where do I see your other zatel conf? |
| 13:28.27 | rickead | is it possible to load a codec driver into asterisk without a restart? |
| 13:29.23 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:29.43 | [TK]D-Fender | rickead: yes |
| 13:29.57 | rickead | whats the command? |
| 13:30.06 | rickead | my g729 codec keeps dropping out of a running system for some reason |
| 13:31.28 | [TK]D-Fender | rickead: Shouldn't |
| 13:31.47 | [TK]D-Fender | rickead: go call Digium support, and try to get some CLI logging from when it happens |
| 13:35.02 | aceio | [TK]D-Fender: http://pastebin.com/2wS1tgvb |
| 13:35.18 | [TK]D-Fender | aceio: [May 13 14:28:56] WARNING[3074]: chan_dahdi.c:1231 dahdi_open: Unable to open '/dev/zap/channel': No such file or directory |
| 13:35.25 | [TK]D-Fender | aceio: not initialized first... |
| 13:35.40 | [TK]D-Fender | aceio: stop *. "dahdi_cfg -vvvv". If that works, restart * |
| 13:35.56 | Jumpie | good ol dahdi issues |
| 13:36.09 | aceio | [TK]D-Fender: okay |
| 13:36.29 | tzafrir_laptop | aceio, asterisk 1.4.30 does not have support for BRI signallings |
| 13:37.07 | tzafrir_laptop | You also seem to have a version of asterisk built vs. zaptel, but try to use DAHDI drivers |
| 13:37.25 | tzafrir_laptop | there's a patch somewhere to add that BRI support to later 1.4-s |
| 13:39.40 | *** join/#asterisk Polysics (~Luca@95.237.66.228) |
| 13:39.42 | Polysics | hello |
| 13:40.00 | Polysics | <PROTECTED> |
| 13:40.08 | Polysics | something like "your credit is running out" |
| 13:40.17 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:40.18 | aceio | [TK]D-Fender: http://pastebin.com/sui36bac |
| 13:40.42 | aceio | tzafrir_laptop: where can i find this patch |
| 13:40.51 | [TK]D-Fender | aceio: Go start * |
| 13:41.15 | aceio | [TK]D-Fender: done |
| 13:42.26 | [TK]D-Fender | aceio: ...............AND?!?! |
| 13:42.49 | Polysics | i was thinking of using JACK, but it looks a little of a kludge |
| 13:43.03 | *** join/#asterisk coppice (~chatzilla@153.166.232.220.dyn.pacific.net.hk) |
| 13:43.11 | aceio | [TK]D-Fender: am still getting Unable to open '/dev/zap/channel': No such file or dir |
| 13:43.24 | [TK]D-Fender | Polysics: When is the decision for this playback made? |
| 13:43.35 | leifmadsen | Polysics: use whisper mode in chanspy() along with some scripts to trigger it via AMI or a callfile |
| 13:43.42 | [TK]D-Fender | aceio: I suggest convering over to full DAHDI config files... |
| 13:43.49 | Polysics | [TK]D-Fender, AMI and a timer |
| 13:44.19 | Polysics | i poll the call from time to time and deduct credits |
| 13:44.28 | Polysics | so far i can disconnect the call brutally |
| 13:44.45 | aceio | TK |
| 13:45.28 | [TK]D-Fender | Polysics: leifmadsen covered what I'd suggest in this case |
| 13:45.32 | aceio | [TK]D-Fender: okay maybe i will upgrade to 1.6 |
| 13:45.44 | Polysics | i wil lread up a little |
| 13:45.46 | leifmadsen | 1.6.what? |
| 13:45.49 | leifmadsen | 1.6 is not a version |
| 13:45.56 | leifmadsen | 1.6.0, 1.6.1, and 1.6.2 are all branches |
| 13:46.28 | Polysics | and btw, about branches, is it worth it to upgrade from 16.1. to 16.2, or since the system works, you would leave it alone? |
| 13:46.29 | leifmadsen | where I'm going with this, is that 1.6.0 and 1.6.1 are now switching to security fix only mode -- so if you're going to move to a 1.6 branch, then move to 1.6.2 |
| 13:46.39 | leifmadsen | don't fix what ain't broke |
| 13:46.53 | leifmadsen | ~asterisk-versions |
| 13:46.54 | infobot | [asterisk-versions] http://www.asterisk.org/asterisk-versions |
| 13:47.30 | Polysics | hmm, chanspy looks like something that should be used interactively |
| 13:47.33 | leifmadsen | infobot: no, asterisk-versions is Information about the maintenance support and when branches will move into security fix only mode, and eventually end-of-life is available at http://www.asterisk.org/asterisk-versions |
| 13:47.34 | infobot | okay, leifmadsen |
| 13:48.18 | *** join/#asterisk analogkid (~xchat@dialin-212-144-188-157.pools.arcor-ip.net) |
| 13:48.35 | *** join/#asterisk sasargen (~chatzilla@98.158.121.6) |
| 13:52.28 | coppice | "and eventually end-of-life is available at" - is this a euthenasia thing? |
| 13:53.52 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 13:55.33 | *** join/#asterisk grandpapadot (~nonya@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 13:56.15 | grandpapadot | Hey guys, what's the dealio on using the asterisk logo and name in a companion product? |
| 13:56.29 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 13:59.29 | *** part/#asterisk grandpapadot (~nonya@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 14:00.33 | leifmadsen | wow... 3 minutes! |
| 14:00.43 | leifmadsen | the answer is... "contact Digium for permission" |
| 14:00.51 | Polysics | too late :-) |
| 14:00.56 | leifmadsen | I know |
| 14:01.02 | leifmadsen | hence my "wow... 3 minutes!" remark |
| 14:01.40 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 14:01.47 | leifmadsen | coppice: yes, we send it out to pasture with the promise of a better life, and that all it's work and sacrifice for the greater good will be rewarded upon retirement. |
| 14:02.18 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:02.26 | coppice | leifmadsen: and then you club it to death while its back is turned? |
| 14:02.31 | Polysics | in reality, the old version ends up living off welfare in a cardboard box, below a bridge in Tacoma |
| 14:02.33 | leifmadsen | coppice: just like a baby seal! |
| 14:02.47 | Polysics | remember that warranty is void if seal is broken |
| 14:03.10 | coppice | good one |
| 14:03.11 | Polysics | ok, i got how ChanSpy is supposed to work, but not how to play audio through it |
| 14:04.55 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 14:04.55 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:05.04 | Polysics | so far i have managed to whisper, but by talking |
| 14:05.13 | Polysics | how would i play a file in there? |
| 14:06.13 | [TK]D-Fender | Polysics: AMI originate w/ local channel; |
| 14:09.20 | Polysics | so i would dial a local channel having the proper dialplan? |
| 14:09.35 | Polysics | which would be ChanSpy followed by Playback? |
| 14:10.47 | *** join/#asterisk gelpg (~chatzilla@94-21-163-153.pool.digikabel.hu) |
| 14:12.19 | gelpg | hi, I have 2 HFP-PCI ISDN card |
| 14:12.34 | gelpg | I compiled the mISDN v2 driver |
| 14:12.48 | gelpg | misdn_info shows the two cards |
| 14:12.58 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 14:13.00 | wcselby | o/ |
| 14:13.02 | gelpg | but when I try to compile asterisk |
| 14:13.16 | gelpg | it doesn't allow me the misdn channel |
| 14:15.01 | gelpg | what do I miss? |
| 14:15.25 | Polysics | or do i need two extensions, one for the playback and one for the chanspy? |
| 14:15.48 | *** part/#asterisk sasargen (~chatzilla@98.158.121.6) |
| 14:15.58 | wcselby | gelpg - there should be a requirement listed at the bottom of make menuselect for mISDN support, that should tell you what you're missing... |
| 14:17.35 | gelpg | wcselby: yes, isdnnet(E), misdn(E), suppserv(E) |
| 14:17.44 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:17.45 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:17.51 | gelpg | wcselby: but what are they? |
| 14:18.33 | wcselby | gelpg - you need to figure out how to install those dependancies |
| 14:18.40 | wcselby | i've never worked with misdn before |
| 14:18.46 | wcselby | but i'm sure google can help you |
| 14:21.42 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 14:22.40 | [TK]D-Fender | Polysics: Yes |
| 14:26.31 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 14:28.24 | devmod | can I log in agents using AMI? I only see AgentLogOff |
| 14:29.44 | leifmadsen | devmod: you can run any asterisk dialplan application from AMI so you can use that if necessary |
| 14:30.57 | *** join/#asterisk netman86 (~como@cpe-74-65-131-73.maine.res.rr.com) |
| 14:31.04 | [TK]D-Fender | leifmadsen: Really? What command? |
| 14:31.12 | devmod | leifmadsen, I was trying to create a web based login for the agents. wouldnt that require me to dial an extension? |
| 14:31.21 | [TK]D-Fender | devmod: QueueAdd agent,all Add interface to queue. |
| 14:31.36 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
| 14:31.48 | leifmadsen | I think it's just Command or something |
| 14:32.25 | [TK]D-Fender | leifmadsen: Yeah, thats to run any CLI command, but not a diallpan line. |
| 14:32.50 | leifmadsen | ah, then I was mistaken. Still applies. |
| 14:34.05 | devmod | [TK]D-Fender, that would add an Agent to the queue but how to tell the queue where the agent is reachable? |
| 14:35.00 | tzafrir_laptop | any idea where I may find the "Karl will be with you shortly" recording of Allison? |
| 14:35.17 | netman86 | I have a system with a single trunk that has two channels. Our internal IP phones have L1-L4 buttons on them, but they all map to trunk 0 channel 1. I'm trying to find a way to specify on the phone that I'd like to dial out on channel 1 or 2 on that trunk, but I cant seem to find anything on google. Are there certain key words I should know? |
| 14:40.00 | [TK]D-Fender | devmod: it adds an INTERFACE. that is whatever you tell it. |
| 14:40.55 | [TK]D-Fender | netman86: Set each as a separate registration and set your dialplan accordingly |
| 14:42.26 | devmod | [TK]D-Fender, right, so I am guessing I can add the interface directly without using "Agent/XXXX". I was wondering if there were any benefits when using "Agent/XXX" vs actual devices in the queue. |
| 14:42.51 | [TK]D-Fender | devmod: Agent/XXX *is* an interface |
| 14:45.01 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca) |
| 14:45.37 | wcselby | can you use Read() inside a Macro? |
| 14:45.59 | netman86 | [TK]D-Fender ; Sorry if this seems a bit stupid, but What is a registration? Is it anything like a trunk? |
| 14:46.24 | devmod | [TK]D-Fender, right I understand that. So, internally what happens when u do an Agent login through an extension. Somewhere it gets stored (Agent/XXX => OtherInterface) ? |
| 14:47.06 | [TK]D-Fender | netman86: ... Your phone registers to *. this is an "identity". Give that phone MULTIPLE ones, each pointing to a different context or make your common dialplan logic look at which reg the call is from to determine what to do. |
| 14:47.15 | [TK]D-Fender | wcselby: Yes |
| 14:48.43 | netman86 | [TK]D-Fender Thanks for the help, but I'm afraid you're just too far over my head! I just found out yesterday that a trunk can have more than one channel. |
| 14:48.48 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:48.51 | [TK]D-Fender | devmod: That is adding an interface to a queue, not efecting a login to chan_agent. This is a separate action you'll need to Originate a local channel to "log in" |
| 14:49.09 | [TK]D-Fender | netman86: This has nothign to do with"trunks". This has to do with your PHONE |
| 14:49.49 | netman86 | Okay. So what is the term for teaching the phone how to use a different channel? |
| 14:50.06 | wcselby | netman86 - what kind of phone? |
| 14:50.46 | netman86 | Aastra phones. This happens to be an 57i |
| 14:51.22 | [TK]D-Fender | netman86: Seat each line to register as a different user |
| 14:51.40 | netman86 | But wouldnt a different user require a different extension? |
| 14:51.42 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:52.53 | [TK]D-Fender | netman86: Do NOT over-associate an EXTENSION with a DEVICE. |
| 14:53.00 | [TK]D-Fender | netman86: Your dialplan does whatever you tell it to. |
| 14:53.37 | [TK]D-Fender | netman86: If you want it to cascade through the multiple devices a single phone may register as then this is just a little extra dialplan logic |
| 14:55.31 | netman86 | I suppose that's fair. As a test, is there a quick and dirty way I can make a call to specify that I'd like to dial out on the trunk0 channel 2 line? (Just to know that it works) |
| 14:56.06 | [TK]D-Fender | netman86: How does a "trunk" have specific channel #'s? |
| 14:56.34 | netman86 | Well it has a "maximum" of 2 channels, which are plugged into 2 POTS lines that go to different phone companies |
| 14:56.42 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:56.56 | netman86 | we need to be able to choose which line we want to use easily, otherwise my boss will throw a fit |
| 14:57.00 | [TK]D-Fender | entAnd how do you specify which to use? |
| 14:57.08 | [TK]D-Fender | netman86: And how do you specify which to use? |
| 14:57.24 | netman86 | We don't right now- that's the problem. But if the first one is in use and someone tries to make a call, it seems to use the second one just fine. |
| 14:57.25 | [TK]D-Fender | netman86: You are being very vague about what hardware & services youa re dealing with |
| 14:57.34 | netman86 | I'm sorry, this is all very new to me. |
| 14:58.01 | [TK]D-Fender | [10:56]<netman86>Well it has a "maximum" of 2 channels, <-- what piece of HARDWARE is "it"? |
| 14:58.22 | netman86 | OH. Sorry. It's a sangoma a200 with a 2 channel FXS card and a 2 channel FXO card. |
| 14:58.35 | [TK]D-Fender | entthen YOu aer in control over how you dial out. |
| 14:58.46 | [TK]D-Fender | netman86: Then YOU are in control over how you dial out. |
| 14:58.52 | netman86 | Well I certainly should be ;) I just don't know what I'm doing. |
| 14:59.09 | [TK]D-Fender | netman86: If you want to prioritize your usage, tehn just pick the specific chanenl you want instead of grouping them |
| 15:00.08 | netman86 | I don't know how, though. I just found out last night that the FXS card was 2 channel as opposed to one, I thought a port was a trunk- clearly I'm just bumbling around. |
| 15:00.22 | [TK]D-Fender | ~book |
| 15:00.23 | 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 |
| 15:00.25 | [TK]D-Fender | netman86: ^^^^^ |
| 15:00.28 | netman86 | Oh, score. |
| 15:00.50 | [TK]D-Fender | netman86: Seems you are doing so with no sense of how * works at all really... time to hit the books ro hire a consultant |
| 15:00.55 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 15:01.17 | netman86 | Yeah, this is the learning project |
| 15:02.03 | beefpastry | kudos to the bookwriters, btw...easy reading so far |
| 15:03.14 | devmod | are there any reserved characters on a extension? like _ at the beginning of a extension etc. |
| 15:03.32 | wcselby | devmod - s, i, n, t, h are all reserved |
| 15:03.49 | wcselby | well, s may not be technically reserved |
| 15:03.57 | wcselby | but I think it is |
| 15:04.15 | [TK]D-Fender | devmod: only when specifying a pattern |
| 15:04.49 | devmod | cool |
| 15:06.17 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 15:07.19 | *** join/#asterisk JasonL (~jason@216.223.114.3) |
| 15:07.50 | *** join/#asterisk Belgarath (~Belgarath@banda.pl) |
| 15:07.57 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 15:12.47 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 15:13.44 | JasonL | I'm reading on voip-info about penalties for queue members.. it's unclear if they works with all ring strategies or only specific ones? |
| 15:13.45 | *** join/#asterisk jart (~jart@c-76-23-206-246.hsd1.ct.comcast.net) |
| 15:13.51 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 15:14.26 | [TK]D-Fender | JasonL: Ringall will pick all those of a give penalty, etc |
| 15:14.38 | [TK]D-Fender | JasonL: Should scale lgoically |
| 15:15.52 | *** join/#asterisk kartik (~koolkarti@117.199.113.116) |
| 15:15.55 | *** join/#asterisk ethel29 (~ethel29@58.69.37.108) |
| 15:16.39 | JasonL | ok, thank you.. i will test it some more.. we need to use it with rrmemory and leastrecent, but we're seeing funny things happening |
| 15:18.09 | ethel29 | good day guys, is there a manager command to identify the assosiated sip channel on specific extension?, what is the best way to disconnect a call using a manager command ? thanks i'm fairly new to asterisk i'm really sorry if this was asked for so many times |
| 15:18.22 | *** join/#asterisk dsfr (~dsfr@pdpc/sponsor/digium/dsfr) |
| 15:18.23 | *** join/#asterisk knot (yiffstar66@unaffiliated/devemo) |
| 15:18.27 | ethel29 | thanks |
| 15:20.30 | [TK]D-Fender | ethel29: "soft hangup [channel] |
| 15:21.15 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 15:21.23 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 15:21.38 | *** join/#asterisk aceio (~5188dc77@gateway/web/freenode/x-yhhkalnjhqpfelwr) |
| 15:21.46 | ethel29 | [TK]D-Fender: will check it right away thanks |
| 15:22.04 | aceio | [TK]D-Fender: hi |
| 15:23.05 | TSM | i am looking to get the Skype plugin for asterisk, is there a way to limit which eth interfaces it binds the key against, i have a onboard card and an addin card, i want it to only bind against that one |
| 15:23.21 | aceio | [TK]D-Fender: i am now using full dahdi noo more zap |
| 15:23.52 | aceio | [TK]D-Fender: i am no getting channels |
| 15:24.01 | TSM | also does the SFA plug look at active eth interfaces ie from ifconfig or does it look at the hardware installed |
| 15:24.14 | aceio | [TK]D-Fender: but still can't pick an outside line |
| 15:25.08 | [TK]D-Fender | aceio: And you're not showing anything sueful |
| 15:25.12 | [TK]D-Fender | useful* |
| 15:25.28 | aceio | http://pastebin.com/4p3d6biu |
| 15:26.38 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:26.38 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 15:26.53 | [TK]D-Fender | aceio: Where are the configs? Where is the dump of dahdi channels? |
| 15:27.32 | aceio | [TK]D-Fender: coming up |
| 15:28.50 | *** join/#asterisk Netgeeks (~chris@173.11.68.155) |
| 15:30.37 | ethel29 | [TK]D-Fender: it was channel request hangup for 1.6 thank you so much, sorry failed to mention my asterisk installation |
| 15:31.28 | aceio | [TK]D-Fender: http://pastebin.com/JhUAVYmr |
| 15:31.30 | *** join/#asterisk lep (~lep@93-62-167-170.ip23.fastwebnet.it) |
| 15:33.26 | drmessano | TSM: The /var/lib/asterisk/licenses directory contains the Host-ID specific license files for your system. These license files are tied to the MAC address of all the ethernet devices installed in your system. |
| 15:33.40 | *** join/#asterisk dsfr (~dsfr@pdpc/sponsor/digium/dsfr) |
| 15:34.15 | TSM | so i need to disable the eth in the bios of the card i dont use, i want to tie it to the NIC incase i need to move asterisk to another machine |
| 15:34.27 | aceio | [TK]D-Fender: what do you think |
| 15:34.32 | drmessano | You can move it once. How often do you rebuild? |
| 15:34.42 | TSM | drmessano: would have been better if they made it so it tied to one nic of choice |
| 15:34.53 | ethel29 | [TK]D-Fender: last question is there a command to list all available channels? |
| 15:35.05 | [TK]D-Fender | aceio: Restart & retest |
| 15:35.08 | TSM | drmessano: not often, just covering my ass i dont want it bound to the onboard nic that i dont use |
| 15:35.13 | TSM | hardly ever |
| 15:35.17 | TSM | never actualy |
| 15:35.17 | [TK]D-Fender | ethel29: "core show channels" |
| 15:35.53 | ethel29 | [TK]D-Fender: hehehe many thanks |
| 15:38.20 | aceio | [TK]D-Fender: http://pastebin.com/SnkRELg2 |
| 15:38.35 | *** join/#asterisk outtolunc (~me@c-67-160-192-210.hsd1.ca.comcast.net) |
| 15:38.46 | wcselby | any idea what "fram type 64" is in the following error message ---> chan_sip.c:3806 sip_write: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) |
| 15:39.45 | *** join/#asterisk puzzled_ (~patrick@puzzled.xs4all.nl) |
| 15:40.02 | [TK]D-Fender | aceio: all sorts of warnings... looks like you're configuring unconencted channels as well |
| 15:40.26 | *** join/#asterisk iscario (~quassel@laureades.davout.pck.nerim.net) |
| 15:40.45 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 15:40.50 | aceio | [TK]D-Fender:help me |
| 15:41.13 | [TK]D-Fender | aceio: Do you have unconnected ports? |
| 15:41.29 | *** join/#asterisk iscario (~quassel@LMontsouris-156-24-36-77.w217-128.abo.wanadoo.fr) |
| 15:41.30 | aceio | [TK]D-Fender: yes |
| 15:41.49 | [TK]D-Fender | aceio: then do not configure those channels |
| 15:42.05 | aceio | [TK]D-Fender: i have cables in port1 and port2 |
| 15:42.17 | aceio | [TK]D-Fender: okay |
| 15:43.53 | *** part/#asterisk gelo (~gelo@209.138.60.213.dynamic.mundo-r.com) |
| 15:44.37 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:46.58 | aceio | [TK]D-Fender: YES |
| 15:47.11 | aceio | [TK]D-Fender: YES IT'S WORKING |
| 15:47.30 | [TK]D-Fender | aceio: lesson : Don't try to use channels you didn't plug in in the first place |
| 15:48.02 | [TK]D-Fender | aceio: And you're welcome |
| 15:48.28 | aceio | [TK]D-Fender: lesson well leant |
| 15:48.35 | aceio | [TK]D-Fender: wow |
| 15:53.47 | jblack | That can be generalized to a life lesson -- don't use things that don't exist. |
| 15:53.52 | jblack | 'specially useful on sundays |
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| 15:54.41 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:56.19 | *** join/#asterisk Knightfal (~j@mailer.1callres.com) |
| 15:58.35 | aceio | [TK]D-Fender: one more |
| 15:59.09 | aceio | [TK]D-Fender: incomming are being rejected |
| 15:59.21 | [TK]D-Fender | ... |
| 15:59.24 | aceio | [TK]D-Fender: sending you an output |
| 15:59.35 | *** join/#asterisk RutgerKirkels (~rutgerkir@82-170-132-31.ip.telfort.nl) |
| 16:02.07 | *** join/#asterisk MatBoy (~MatBoy@wiljewelwetenhe.xs4all.nl) |
| 16:02.38 | aceio | http://pastebin.com/if2ZgEHF |
| 16:03.55 | [TK]D-Fender | aceio: [dial-in] does not have an exten to match that number, just like it says |
| 16:07.52 | aceio | [TK]D-Fender: how do i correct this |
| 16:09.39 | aceio | [TK]D-Fender: sorry too ask |
| 16:16.07 | aceio | [TK]D-Fender: |
| 16:17.37 | *** part/#asterisk RutgerKirkels (~rutgerkir@82-170-132-31.ip.telfort.nl) |
| 16:17.55 | *** join/#asterisk RutgerKirkels (~rutgerkir@82-170-132-31.ip.telfort.nl) |
| 16:19.47 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 16:20.03 | ujjain | Is there a fast ISO mirror for AsteriskNow? |
| 16:20.17 | *** join/#asterisk lmns972 (~lmns972@cac94-8-82-245-26-103.fbx.proxad.net) |
| 16:24.19 | Qwell | ujjain: asterisknow.org |
| 16:24.30 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 16:25.21 | *** join/#asterisk MiserySoft (~lnd@89.193.94.154) |
| 16:27.10 | *** join/#asterisk QaDeS (~mklaus@p4FC7032E.dip0.t-ipconnect.de) |
| 16:28.02 | korcan | I am having trouble creating a socket connection from a php script... my debug within asterisk is giving me "NOTICE[23347]: manager.c:1087 authenticate: xx.xx.xx.xx failed to pass IP ACL as 'ClickToDial'" |
| 16:28.36 | korcan | my manager.conf has permit=0.0.0.0/0.0.0.0 set |
| 16:28.42 | wcselby | i have one user that when a call is manually blind transfered to them (person presses transfer button on their phone before the user answers), the user cannot hear anything, although the original caller can hear them. And my asterisk cli is flooded with this --> chan_sip.c:3806 sip_write: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4). I haven't been able to reproduce this with other user |
| 16:29.32 | wcselby | actually, nevermind, i'll post more when I get back from lunch |
| 16:30.42 | *** join/#asterisk MatBoy (~MatBoy@wiljewelwetenhe.xs4all.nl) |
| 16:31.56 | ujjain | Qwell: Asterisknow.org mirror is 40kB/sec |
| 16:31.57 | ujjain | it's awful. |
| 16:32.56 | Naikrovek | asterisknow.org is not a mirror |
| 16:33.57 | Naikrovek | and fyi: i'm downloading it a lot faster than that, right now. |
| 16:33.57 | *** join/#asterisk aruntomar (~aruntomar@61.17.193.163) |
| 16:34.09 | Naikrovek | so it's not that site that's slow, it's something between that site andyou |
| 16:34.10 | ujjain | I asked for a mirror. |
| 16:34.27 | Naikrovek | there are no distribution mirrors for asterisk or asterisknow, that i know of. it's all digium |
| 16:34.29 | ujjain | Yeah obviously could be because you use different transit |
| 16:34.44 | ujjain | Hmm,, I see. |
| 16:35.06 | ujjain | I would love the extra fax, voicemail features easily without the hassle of spending many days to get to know the basics of Asterisk text-line. |
| 16:35.44 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 16:35.58 | Naikrovek | well learning that config file stuff will reward you in the future |
| 16:36.06 | Naikrovek | but if you don't think so, just download it at 40kbsp |
| 16:36.08 | Naikrovek | kbps |
| 16:37.00 | ujjain | Yeah, I'll find a faster way :) |
| 16:37.15 | ujjain | It would rewarwd me, but all I care for is a working solution quickly. |
| 16:37.23 | ujjain | I have already configured voip-in/voip-out, already works fine. |
| 16:37.42 | ujjain | Most time was spent because of my bad SIP provider. |
| 16:37.58 | ujjain | anyway, there's a pengiun here that helped me out a lot. |
| 16:42.11 | *** join/#asterisk afink (~chatzilla@204.26.87.226) |
| 16:43.39 | afink | Is there a way to get automon to automatically combine the in and out files of a recording? |
| 16:45.10 | *** join/#asterisk scalex000 (~chatzilla@190.80.195.104) |
| 16:45.19 | scalex000 | hello |
| 16:45.32 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 16:48.32 | aruntomar | what would be the best strategy for incoming calls to be automatically distributed by asterisk based on extensions that are free |
| 16:49.29 | edwin_quijada | Hi! |
| 16:49.52 | edwin_quijada | I need to show an info from my client suing callerid when a call come in |
| 16:49.53 | Knightfal | aruntomar: app_queue or dev_state |
| 16:49.54 | ChannelZ | aruntomar: a queue, if I understand your question |
| 16:49.57 | afink | aruntomar: http://www.voip-info.org/wiki/view/Asterisk+call+queues |
| 16:50.01 | Naikrovek | aruntomar: queue |
| 16:50.10 | Naikrovek | ooh hit the enter key too late. now it looks like i copied |
| 16:50.36 | Naikrovek | edwin_quijada: run windows? |
| 16:50.42 | edwin_quijada | no |
| 16:50.57 | edwin_quijada | is for agents from a pizzeria |
| 16:51.19 | edwin_quijada | when the call come in they want to see all information from that client |
| 16:51.24 | edwin_quijada | any clues? |
| 16:51.26 | aruntomar | we have a pri line, with 10 extensions, customers will call on the given numbers, those calls needs to be routed to telecallers based on, which telecaller is free at the moment |
| 16:51.38 | aruntomar | i'm reading queues |
| 16:51.42 | Knightfal | Ya Queues |
| 16:51.50 | edwin_quijada | Just I need rigth direction to do this |
| 16:52.16 | Naikrovek | edwin_quijada: just add the logic to the dialplan. call a program that queries a database and instant messages the person whose phone is ringing or something |
| 16:52.38 | aruntomar | we already have an asterisk setup but that's for outbound calls, this is a new setup that will mostly be used for incoming calls |
| 16:53.03 | Naikrovek | aruntomar: asterisk can handle incoming calls too |
| 16:53.05 | edwin_quijada | Naikrovek: I need an app running in my client always |
| 16:53.23 | Naikrovek | edwin_quijada: yes but the asterisk server has to tell you who is calling, right? |
| 16:53.37 | korcan | anyone have any ideas for my authentication issue? |
| 16:53.39 | Naikrovek | or are you using softphones |
| 16:53.40 | aruntomar | another query was, that we want to integrate asterisk to our web application, for that i guess, we'll need webphone, any suggestions for webphones? |
| 16:53.45 | korcan | I am having trouble creating a socket connection from a php script... my debug within asterisk is giving me "NOTICE[23347]: manager.c:1087 authenticate: xx.xx.xx.xx failed to pass IP ACL as 'ClickToDial'" |
| 16:53.47 | korcan | my manager.conf has permit=0.0.0.0/0.0.0.0 set |
| 16:53.51 | aruntomar | Naikrovek: this is a new client |
| 16:55.10 | aruntomar | thanx guys, i'll implement queues |
| 16:55.17 | aruntomar | any suggestions for webphones |
| 16:55.48 | afink | aruntomar: xlite |
| 16:56.49 | [TK]D-Fender | not a "webphone" |
| 16:57.21 | aruntomar | afink: as far as i know, xlite is a softphone |
| 16:57.58 | aruntomar | plus we r on ubuntu os, so anything that works on that would be gr8 |
| 16:58.24 | aruntomar | in softphones, we find zoiper to be very good on ubuntu |
| 16:59.22 | aruntomar | but zoiper webphones are only for windows and IE |
| 17:00.15 | [TK]D-Fender | aruntomar: Why would you be using a "webphone" instead of a regular soft-phone? |
| 17:00.54 | aruntomar | [TK]D-Fender: we want to integrate our webapp with asterisk, and calls to be made directly from the browser |
| 17:01.12 | aruntomar | our vendors are using adhersion framework based on rails |
| 17:01.59 | aruntomar | they r able to make calls, but logically it's the web server that's making the call, so the call doesn't comes to the agent |
| 17:02.02 | [TK]D-Fender | GAH |
| 17:02.04 | [TK]D-Fender | NVM |
| 17:03.07 | aruntomar | [TK]D-Fender: so in order to make a call the agent will have to accept the call, that's a bit weird for them ;) |
| 17:03.31 | scalex000 | hi, how to create secure connection with sip? |
| 17:03.38 | Corydon76-dig | aruntomar: why not run a Java softphone, instead, which can be launched from the browser? |
| 17:04.10 | Corydon76-dig | scalex000: tunnel it over a VPN |
| 17:04.39 | [TK]D-Fender | Corydon76-dig: Still no SRTP in 1.6.2.x? |
| 17:04.52 | Corydon76-dig | won't ever be in 1.6.2 |
| 17:05.06 | Corydon76-dig | Once the release is out, it's feature-frozen |
| 17:05.16 | [TK]D-Fender | Corydon76-dig: Wasn't sure if it was prepped for 1.6.1 and avail in 1.6.2 |
| 17:05.32 | Corydon76-dig | Nope, 1.8 would be the earliest version it's in |
| 17:05.33 | [TK]D-Fender | Corydon76-dig: Guess it's "neither |
| 17:05.38 | scalex000 | corydon76, any documents I can see it |
| 17:06.02 | Corydon76-dig | scalex000: documents? |
| 17:06.18 | aruntomar | Corydon76-dig: that's the idea i was looking for ;) |
| 17:08.04 | scalex000 | corydon76, you say tunnel |
| 17:08.13 | scalex000 | corydon76, a vpn connection |
| 17:08.42 | *** join/#asterisk BSE-R (~BSE-R@190.34.191.8) |
| 17:09.25 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:10.12 | BSE-R | A 503 Service Unavailable results in the phone being unable to dial out until a new registration takes place. If you manually re-register the phone, it will work again until the 503 occurs again. |
| 17:10.20 | BSE-R | The worst part about this behavior is that the phone will not even dial out to other known working extensions during this period. |
| 17:10.27 | BSE-R | im using xlite |
| 17:10.37 | Corydon76-dig | aruntomar: phonefromhere.com, perhaps? |
| 17:11.09 | aruntomar | Corydon76-dig: http://www.mizu-voip.com/Products/WebPhone.aspx |
| 17:11.17 | aruntomar | Corydon76-dig: i found this also |
| 17:20.56 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 17:21.51 | *** join/#asterisk Znuff (~ibm86@2001:0:53aa:64c:2832:3add:a6d0:dcf6) |
| 17:27.55 | *** join/#asterisk dsfr (~dsfr@pdpc/sponsor/digium/dsfr) |
| 17:31.39 | *** join/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
| 17:32.10 | mbrevda | using anager API Action QueueStatus how do you filter for a specific memeber? |
| 17:32.59 | mbrevda | *Manager |
| 17:37.33 | iscario | hi, i downloaded a branch of asterisk to enable srtp (srtp_reboot), but i'm having trouble configuring it because i cannot find any documentation .... could you help ? |
| 17:39.35 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 17:47.31 | *** join/#asterisk vally (~vally@ip-109-90-253-92.unitymediagroup.de) |
| 17:48.56 | *** join/#asterisk Znuff (~ibm86@2001:0:53aa:64c:2832:3add:a6d0:dcf6) |
| 17:51.13 | *** join/#asterisk orioni (~chatzilla@95.107.226.183) |
| 17:51.54 | orioni | hi there , where can i get some infos how wholesale telephony works , and what tools / services are needed ? |
| 17:52.11 | orioni | just general ideas so i can search myself later |
| 17:53.46 | *** part/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
| 18:01.33 | pabelanger_ | m5413 |
| 18:01.47 | MuffinMan | [assigned] [Asterisk] Channels/chan_sip/NewFeature 0005413: [patch] [branch] Secure RTP (SRTP) reported by mikma https://issues.asterisk.org/view.php?id=5413 |
| 18:01.49 | pabelanger_ | iscario: ^^^ |
| 18:02.02 | pabelanger_ | iscario: any documentation would be located in that thread |
| 18:04.26 | iscario | thx MuffinMan, pabelanger_ , i 'll have a look there then |
| 18:05.01 | *** join/#asterisk UberDuper (~jsatter@elrond.uberduper.com) |
| 18:05.27 | UberDuper | What's the recommended linksys gateway for a home wrt asterisk setup? |
| 18:05.45 | UberDuper | Was it the SPA2100? |
| 18:06.41 | pabelanger_ | UberDuper: http://lmgtfy.com/?q=SPA2100 |
| 18:07.27 | wcselby | i have one user that when a call is manually blind transfered to them (person presses transfer button on their phone before the user answers), the user cannot hear anything, although the original caller can hear them. And my asterisk cli is flooded with this --> chan_sip.c:3806 sip_write: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4). I haven't been able to reproduce this with other user |
| 18:07.40 | UberDuper | So uhh. If you're gonna lmgtfy someone... at least give them a useful search. |
| 18:07.59 | wcselby | i've also recreated the same user's extension on another phone, and the same issue persists |
| 18:08.30 | UberDuper | There was one in particular that was highly recommended last time I was looking. I just don't remember which model it was. |
| 18:08.41 | wcselby | the flood of messages on the CLI happen when the user answers the phone |
| 18:09.56 | [TK]D-Fender | UberDuper: What KIND of gateway? |
| 18:10.01 | wcselby | let me grab a trace of the cli and pastebin it |
| 18:11.01 | pabelanger_ | m16287 |
| 18:11.03 | MuffinMan | [acknowledged] [Asterisk] Channels/chan_sip/CodecHandling 0016287: One way audio after attended transfer reported by cbkm https://issues.asterisk.org/view.php?id=16287 |
| 18:11.07 | pabelanger_ | wcselby: ^^^ |
| 18:11.34 | wcselby | thanks pabelanger, let me look into it |
| 18:11.41 | jjjj-away | To connect asterisk with a Digium t1 card to an avaya PBX with a CSU do you need a T1 cable? |
| 18:12.08 | pabelanger_ | jjjj-away: likely a T1 crossover cable |
| 18:12.15 | jjjj-away | thanks |
| 18:12.44 | pabelanger_ | ~crossover |
| 18:12.45 | infobot | hmm... crossover is a proprietary piece of software that allows x86 linux users to use some Windows Netscape plugins with Netscape/Mozilla/Konq. I'd give you a link, but I don't like spam. :-P, or a different wiring for CAT5 cabling, that allows you to hook 2 ethernet cards directly together with no hub in between. For more information, see http://www.gcctech.com/support/doc/crossover.html |
| 18:12.57 | pabelanger_ | ~t1crossover |
| 18:13.05 | *** join/#asterisk MiserySoft (~lnd@109.180.222.4) |
| 18:14.10 | lmns972 | hello everybody |
| 18:14.20 | lmns972 | I did some tests in the asterisk until everything is ok |
| 18:14.53 | lmns972 | Then I wanted to test french sound files |
| 18:15.19 | lmns972 | So I installed the following package: asterisk-prompt-fr-proformatique and since asterisk does not work |
| 18:15.33 | lmns972 | If someone has an idea? |
| 18:15.47 | lmns972 | http://pastebin.com/HJVpjpkP |
| 18:16.20 | Qwell | lmns972: don't load multiple copies of app_voicemail. see /etc/asterisk/modules.conf, and go yell at your distribution |
| 18:17.56 | *** join/#asterisk |amadeus| (~amadeus@75.130.147.152) |
| 18:18.03 | *** join/#asterisk MiserySoft (~lnd@109.180.222.4) |
| 18:20.03 | lmns972 | Qwell, I do not see the problem in the modules.conf file. Can you tell me more? |
| 18:20.18 | UberDuper | Okay I think it was the SPA3000 I needed. |
| 18:20.22 | lmns972 | and i use debian |
| 18:20.23 | Qwell | noload the files that it's complaining about in your log |
| 18:20.23 | lmns972 | http://pastebin.com/8QHm2dTv |
| 18:20.24 | UberDuper | 1 FSX and 1 FXO |
| 18:20.55 | Qwell | tzafrir_laptop: what's up with that? I've seen it a few times now. |
| 18:21.17 | [TK]D-Fender | UberDuper: SPA-3102 |
| 18:21.51 | UberDuper | [TK]D-Fender: I think you answered some questions for me a few months back on this. |
| 18:22.19 | UberDuper | All I want to do is put an asterisk box between my analog phone service and my analog phones. |
| 18:22.42 | korcan | anyone have any ideas for my authentication issue? |
| 18:22.46 | korcan | I am having trouble creating a socket connection from a php script... my debug within asterisk is giving me "NOTICE[23347]: manager.c:1087 authenticate: xx.xx.xx.xx failed to pass IP ACL as 'ClickToDial'" |
| 18:22.49 | korcan | my manager.conf has permit=0.0.0.0/0.0.0.0 set |
| 18:23.27 | tzafrir_laptop | Qwell, http://patch-tracker.debian.org/patch/series/view/asterisk/1:1.6.2.7-1/hack-multiple-app-voicemail |
| 18:23.41 | UberDuper | What's the difference between the SPA3102 and SPA3102-NA? |
| 18:23.49 | UberDuper | NA is unlocked? |
| 18:23.58 | tzafrir_laptop | I can't think of a saner way to provide all three voicemail storage options |
| 18:24.25 | Qwell | tzafrir_laptop: can't put them in separate packages, or put them noloaded in the default modules.conf? |
| 18:24.52 | Qwell | the multiple files thing is exactly what AN does. it just has separate packages for them |
| 18:26.32 | [TK]D-Fender | UberDuper: Correct. Though just about any 3102 yuo ffind should eb unlocked regardless |
| 18:27.00 | [TK]D-Fender | korcan: And that singular line offers us no proof that yoru configs are right overall |
| 18:27.34 | korcan | [TK]D-Fender, what else should I be looking at? |
| 18:28.07 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.225.159.dsl.dyn.forthnet.gr) |
| 18:28.11 | korcan | read = All write = All |
| 18:29.28 | [TK]D-Fender | korcan: PASTEBIN yrou configs and look at the TCPDUMP for the attempt |
| 18:29.56 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 18:30.10 | UberDuper | [TK]D-Fender: Thanks. |
| 18:30.40 | raden_work | msg Naikrovek hey honey im home LOL |
| 18:30.53 | Naikrovek | wtf |
| 18:31.01 | raden_work | lol i didnt shrink the boss yet |
| 18:31.19 | Naikrovek | your boss needs to get duct taped to the wall or something |
| 18:31.41 | raden_work | no shit |
| 18:35.16 | korcan | [TK]D-Fender, I have it working on my LAN, however the webserver my script sits on is outside the firewall... where else should I set the external IP for the webserver for proper authentication? |
| 18:35.29 | korcan | just confirmed that... |
| 18:35.31 | *** join/#asterisk knot (yiffstar66@unaffiliated/devemo) |
| 18:35.55 | korcan | [TK]D-Fender, so it does work, however I don't know where else asterisk would check for an IP auth |
| 18:36.14 | [TK]D-Fender | korcan: I don't see a complete entry anywhere |
| 18:36.37 | orioni | hi there , where can i get some infos how wholesale telephony works , and what tools / services are needed ?just general ideas so i can search myself later |
| 18:37.03 | korcan | [TK]D-Fender, which config should I send you? |
| 18:37.07 | *** part/#asterisk aruntomar (~aruntomar@61.17.193.163) |
| 18:37.34 | [TK]D-Fender | manager.conf |
| 18:37.45 | [TK]D-Fender | orioni: It works like anything else |
| 18:38.21 | orioni | the network idea is ok , qos and the ethernet-e1/t1 is ok |
| 18:38.22 | korcan | !bin |
| 18:38.26 | korcan | ~bin |
| 18:38.28 | infobot | extra, extra, read all about it, bin is where you put rubbish, or your little brother/sister, or your copy of Windows 98 |
| 18:38.38 | [TK]D-Fender | ~pb |
| 18:38.39 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 18:38.43 | orioni | just still dont get the idea of lcr and the billing part |
| 18:38.48 | *** join/#asterisk hfb (~hfb@pool-98-112-146-69.lsanca.dsl-w.verizon.net) |
| 18:39.12 | orioni | or better how are this servers serving those services connected |
| 18:42.15 | [TK]D-Fender | If you call France and you ahve 2 possible providers for that area and you choose the cheaper one of the two.... that's LCR |
| 18:42.36 | [TK]D-Fender | orioni: And billing... if I give you service I BILL YOU for it. Also not complicated |
| 18:43.07 | *** join/#asterisk chazzam (~chazz@173-24-238-25.client.mchsi.com) |
| 18:43.13 | orioni | the software needed , how the are conencted etc sth like this but more organized http://www.kamailio.org/wp-images/kamailio-sip-ucp.png i.e internet - gateway - openser - billing - E1 - telecom |
| 18:45.18 | [TK]D-Fender | orioni: Who said anything was "needed |
| 18:45.20 | [TK]D-Fender | "? |
| 18:45.35 | [TK]D-Fender | orioni: Who says you need an E1? |
| 18:45.53 | [TK]D-Fender | orioni: What is "telecom" in that list.... EVERYTHING before it counts... so why is this term on its own |
| 18:46.25 | orioni | that i wanted to specify the telecom part where i "inject" the traffic i got from the other provider |
| 18:46.56 | orioni | suposse italy telekom - albania (me) - albanian telecom |
| 18:47.25 | orioni | so the (me) need to know how that works and what tools do i need in order to make some profits , in general terms |
| 18:47.28 | *** join/#asterisk torrio (~f3k@p54A2D41A.dip.t-dialin.net) |
| 18:48.34 | *** join/#asterisk rare1980_ (~rare1980@119.158.24.200) |
| 18:50.26 | torrio | hi, when using Background() ... is it possible to "catch" the digit, that was pressed ? |
| 18:51.09 | [TK]D-Fender | orioni: This is looking to be very much out of your league... |
| 18:51.25 | [TK]D-Fender | torrio: It jumps to the appropriate exten. |
| 18:51.33 | wcselby | torrio - outside of macro's, yeah, pretty much. or you could use a Read(var,filename) |
| 18:51.58 | wcselby | torrio - where var is the variable where you catch the digit, and filename is the sound file you would normally Background() |
| 18:52.23 | torrio | and read will stop playing when a digit was pressed ? |
| 18:52.27 | wcselby | torrio - if you're outside of a macro, you can just create new exten => commands for the digits you want to catch |
| 18:52.57 | torrio | i want to catch all possible digits and send them to a agi-script |
| 18:53.04 | *** join/#asterisk dymaxion (~dymaxion@203080238194.static.ctinets.com) |
| 18:53.22 | [TK]D-Fender | [14:52]<torrio>and read will stop playing when a digit was pressed ? <- yes |
| 18:53.32 | wcselby | torrio - then why not catch them in the AGI script? |
| 18:53.46 | wcselby | or just catch them in a Read() variable and pass that to the AGI |
| 18:54.42 | torrio | trying, thanks. |
| 18:58.00 | *** join/#asterisk tjwallace (~jeff@76-10-176-148.dsl.teksavvy.com) |
| 19:00.08 | tjwallace | has anyone had any luck with ubuntu 10.04, asterisk and freepbx? |
| 19:00.26 | tjwallace | i am trying right now and have run into a problem |
| 19:00.41 | orioni | <[TK]D-Fender : yep , im trying to get the idea |
| 19:00.59 | tjwallace | asterisk kills itself when its starting up, with little information in the logs |
| 19:01.04 | tjwallace | here is the startup: http://pastebin.com/mN4Z97ZU |
| 19:01.11 | Qwell | wild guess |
| 19:01.15 | Qwell | <PROTECTED> |
| 19:01.34 | *** join/#asterisk ariel_ (~chatzilla@252-68-174-206.gci.net) |
| 19:01.36 | tjwallace | cd /var/run/ |
| 19:01.36 | tjwallace | ll |
| 19:01.40 | Qwell | or not |
| 19:02.17 | *** join/#asterisk The-Bat (~The-Bat@59.162.86.164) |
| 19:02.43 | Qwell | voicemail strikes again |
| 19:02.43 | tjwallace | Qwell: ? |
| 19:02.59 | ariel_ | Hello folks |
| 19:03.06 | Qwell | <Qwell> don't load multiple copies of app_voicemail. see /etc/asterisk/modules.conf, and go yell at your distribution |
| 19:04.25 | Qwell | actually, no |
| 19:04.31 | Qwell | freepbx overwrote your modules.conf |
| 19:04.38 | Qwell | yell at both |
| 19:05.18 | tjwallace | should i use what my distro used? it boost up fin |
| 19:05.21 | tjwallace | fine* |
| 19:05.56 | Qwell | no. FreePBX doesn't care what distro you're running, or what it's doing differently than stock Asterisk |
| 19:06.02 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 19:06.19 | Qwell | add the noloads to modules.conf, and you'll be fine. but really, go yell at people |
| 19:06.35 | tjwallace | the noloads from my distro? |
| 19:06.40 | tjwallace | distro's config |
| 19:06.43 | tjwallace | i have it backed up |
| 19:06.45 | Qwell | as more and more people upgrade to Ubuntu 10.04, people will keep seeing this |
| 19:06.52 | Qwell | tjwallace: no, just add the noload lines for voicemail |
| 19:06.58 | *** part/#asterisk torrio (~f3k@p54A2D41A.dip.t-dialin.net) |
| 19:06.58 | tjwallace | gotcha |
| 19:07.08 | Qwell | and app_directory or whatever else it's doing |
| 19:07.22 | Qwell | mmm, I think that's fine in 1.6 now. just voicemail |
| 19:07.34 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
| 19:10.32 | tjwallace | Qwell: excellent diagnosis! worked like a charm! |
| 19:10.40 | tjwallace | Qwell: thanks a lot |
| 19:10.45 | tjwallace | Qwell: now i will start yelling |
| 19:11.01 | Qwell | neither and both are at fault. |
| 19:11.05 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
| 19:11.08 | Qwell | neither will accept fault. |
| 19:11.10 | tjwallace | Qwell: where in the startup log did you notice the problem |
| 19:11.20 | Qwell | tjwallace: I just knew |
| 19:11.31 | tjwallace | Qwell: really? impressive |
| 19:11.47 | Qwell | I've seen it a bunch of times now.. |
| 19:11.50 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
| 19:13.52 | ChannelZ | Clairvoyance is an asset |
| 19:20.01 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 19:25.02 | *** join/#asterisk MiserySoft (~lnd@94-116-26-54.dynamic.thecloud.net) |
| 19:26.29 | ujjain | I have that too, did you ever happen to predict that you were going to sneeze? |
| 19:26.38 | ujjain | I think I have clairvoyance. |
| 19:26.47 | ujjain | how do I correctly apply the noun? :D |
| 19:26.58 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 19:27.24 | ujjain | dmz: Howdy. |
| 19:28.41 | dmz | howdy |
| 19:29.24 | ChannelZ | If you were clairvoyant, you'd already know... |
| 19:29.42 | ChannelZ | "ERROR, ERROR" *head explodes* |
| 19:30.54 | *** join/#asterisk mcateer (~mcateer@75.76.254.162) |
| 19:33.38 | *** join/#asterisk RutgerKirkels (~rutgerkir@82-170-132-31.ip.telfort.nl) |
| 19:34.35 | RutgerKirkels | hi |
| 19:35.23 | RutgerKirkels | does anyone have an idea what would be the best way to make an ISDN fallback for outgoing calls? |
| 19:36.35 | *** part/#asterisk lmns972 (~lmns972@cac94-8-82-245-26-103.fbx.proxad.net) |
| 19:37.51 | mcateer | @Rutger: Put 'Dial' in a loop and check the dial status to determine whether to try again. |
| 19:38.27 | mcateer | Does anyone here have any particular sentiments (good or bad) about running Asterisk with the '-p' option? |
| 19:39.55 | RutgerKirkels | i've made this for cell numbers, it gives me 2 beeps when the voip dialout doesn't work: |
| 19:39.56 | RutgerKirkels | exten => _0[6].,1,Dial(SIP/voipcheap/${EXTEN},60,Trt) |
| 19:39.56 | RutgerKirkels | exten => _0[6].,2,Playback(beep) |
| 19:39.57 | RutgerKirkels | exten => _0[6].,3,Playback(beep) |
| 19:39.57 | RutgerKirkels | exten => _0[6].,4,Dial(Zap/g1/$(EXTEN),60,Trt) |
| 19:40.48 | RutgerKirkels | but i guess this also tries to dial again through ISDN when the other side is busy... |
| 19:42.33 | *** join/#asterisk imcdona (imcdona@173.160.189.77) |
| 19:45.45 | tjwallace | where does asterisk 1.6 store sounds? |
| 19:47.02 | tjwallace | /usr/share/asterisk/sounds ? |
| 19:47.03 | [TK]D-Fender | tjwallace: in the lib folder specified in asterisk.conf |
| 19:47.05 | *** join/#asterisk djMax (~chatzilla@66.92.65.4) |
| 19:47.34 | djMax | so I can't update an asterisk install because packages.digium.com seems to have issues. Known problem? |
| 19:47.49 | tjwallace | [TK]D-Fender: astvarlibdir => /var/lib/asterisk ? |
| 19:48.13 | [TK]D-Fender | tjwallace: under there... |
| 19:49.22 | pabelanger_ | djMax: What issues |
| 19:49.38 | tjwallace | [TK]D-Fender: thats strange because im getting this error: file.c: File custom/ew-greeting-with-ext does not exist in any format |
| 19:49.50 | tjwallace | Unable to open custom/ew-greeting-with-ext (format 0x4 (ulaw)): No such file or directory |
| 19:49.54 | djMax | trying a yum clean now to see if that helps, but "repdata/filelists.xml.gz metadata file does not match checksum" |
| 19:50.01 | djMax | repodata I meant. |
| 19:50.33 | *** join/#asterisk brezular (~brezular@adsl-dyn135.78-99-111.t-com.sk) |
| 19:50.58 | wcselby | queue show |
| 19:51.48 | [TK]D-Fender | tjwallace: what tells me you have a proper file in a proper place? |
| 19:52.26 | jasonjjohnsonjr | <PROTECTED> |
| 19:53.10 | tjwallace | ew-greeting-with-ext.wav is in /var/lib/asterisk/sounds/custom/ |
| 19:53.20 | tjwallace | [TK]D-Fender: ew-greeting-with-ext.wav is in /var/lib/asterisk/sounds/custom/ |
| 19:53.47 | wcselby | doesn't 1.6.x store in /var/lib/asterisk/sounds/en/ |
| 19:53.48 | wcselby | ? |
| 19:54.19 | pabelanger_ | wcselby: usually yes, but some packages change the locate of files |
| 19:54.51 | tjwallace | im using ubuntu 10.04 with freepbx |
| 19:55.35 | tjwallace | hmm ok i got it going, had to move the file to where ubuntu keeps sound files |
| 19:55.55 | pabelanger_ | tjwallace: yes, the debian packaging changes the locate of sound files |
| 19:56.14 | tjwallace | pabelanger_: can that directory be configured in asterisk? |
| 19:56.35 | pabelanger_ | tjwallace: Aye, asterisk.conf file |
| 19:57.03 | *** join/#asterisk ddickenson (~chatzilla@67-198-0-5.static.grandenetworks.net) |
| 19:57.54 | tjwallace | pabelanger_: ah yes i see i have to remove the '(!)' to enable directory overrides |
| 19:58.03 | tjwallace | is there a way to override freepbx? |
| 19:58.51 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:58.55 | djMax | yum clean all seemed to resolve the digium package issues, so I guess whatever. |
| 19:59.07 | wcselby | so if I'm having this same issue - https://issues.asterisk.org/view.php?id=14249, even though it's been suspended, how can I get some followup on this? |
| 20:02.12 | pabelanger_ | m16287 |
| 20:02.14 | MuffinMan | [acknowledged] [Asterisk] Channels/chan_sip/CodecHandling 0016287: One way audio after attended transfer reported by cbkm https://issues.asterisk.org/view.php?id=16287 |
| 20:02.27 | pabelanger_ | wcselby: ^^^ just subscribe to this issue |
| 20:02.33 | pabelanger_ | since they are related |
| 20:03.40 | wcselby | pabelanger - the work around mentioned in the comments doesn't work for me |
| 20:03.49 | [TK]D-Fender | [15:57]<tjwallace>is there a way to override freepbx? <- pardon? |
| 20:04.16 | ddickenson | anyone tackled the change from macro to gosub? I'm building a new system and I figure if macro's depreciated I ought to learn the replacement and be a tad more futureproof. |
| 20:05.00 | pabelanger_ | wcselby: pastebin your output |
| 20:05.00 | *** part/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 20:12.54 | wcselby | http://pastebin.com/hLKzQ1gh |
| 20:18.07 | pabelanger_ | wcselby: what version are you using? |
| 20:18.12 | wcselby | 1.4.26 |
| 20:27.19 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 20:28.05 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 20:29.16 | devmod | any suggestions on a free queue monitoring n reporting tool |
| 20:34.53 | *** join/#asterisk jaxxan (~jaxxan@216.130.138.34) |
| 20:35.03 | *** part/#asterisk jaxxan (~jaxxan@216.130.138.34) |
| 20:36.29 | *** join/#asterisk QaDeS_ (~mklaus@p4FC70665.dip0.t-ipconnect.de) |
| 20:37.36 | leifmadsen | devmod: not sure of any free ones -- they likely don't exist, but queue metrics has been pretty decent in an installation I installed it on |
| 20:38.59 | wcselby | i think i may have resolved my issue |
| 20:39.12 | wcselby | by removing the Answer() statement on the extension i'm trying to transfer TO |
| 20:43.02 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 20:46.12 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:04.59 | *** join/#asterisk smooth_penguin (~smoove@59.95.22.75) |
| 21:10.45 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 21:16.16 | *** join/#asterisk decaffeine (~decaffein@109.167.129.114) |
| 21:26.44 | *** join/#asterisk smooth_penguin (~smoove@59.95.6.110) |
| 21:28.39 | tjwallace | is there any way to get around these errors: Module 'cdr_addon_mysql.so' was not compiled with the same compile-time options as this version of Asterisk. |
| 21:28.48 | tjwallace | and then it won't load it |
| 21:30.48 | *** join/#asterisk Znuff (~ibm86@2001:0:53aa:64c:2832:3add:a6d0:dcf6) |
| 21:31.09 | *** join/#asterisk ecrane (~ecrane@o1-69-19-166-10.static.o1.com) |
| 21:31.14 | orioni | what is the difference of opensips and kamailio ? |
| 21:34.53 | ChannelZ | tjwallace: recompile both? |
| 21:37.02 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 21:38.50 | tjwallace | ChannelZ: they are packaged from ubuntu 10.04 |
| 21:39.43 | paulc | I'm playing around with MeetMe on a box that doesn't have any DAHDI hardware. I'm using a Local channel to connect an extension into the conference where that extension plays an announcement and hangs up. Sometimes, the Playback seems to stop - the channel is still up/active, but "frozen".. any ideas on what's going on? |
| 21:47.46 | *** join/#asterisk ryanlin (~ryanlin@you.better.not.nulroute.us) |
| 21:48.13 | ryanlin | can i use asterisk for auto attendent with cisco callmanager express? |
| 21:50.06 | p3nguin | I don't see why it wouldn't be possible. Whether or not you are capable... only you know. |
| 21:50.11 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 21:51.40 | *** join/#asterisk HorizonXP (~xitij@206-248-131-202.dsl.teksavvy.com) |
| 21:52.56 | paulc | ryanlin: send calls into Asterisk via CME then transfer calls back to CME endpoints? Sounds doable |
| 21:53.10 | ryanlin | yeah |
| 21:53.22 | ryanlin | but i can not get it to work for some reason..let me show you guys my sip.conf |
| 21:54.00 | ryanlin | http://pastebin.com/i53Qs7iW |
| 21:54.09 | ryanlin | does this look right to you guys? |
| 21:54.24 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-155-51.cinci.res.rr.com) |
| 21:55.32 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 21:57.35 | p3nguin | ryanlin: http://pastebin.com/1LLwBBFK |
| 21:57.43 | p3nguin | ryanlin: I like this better than yours. |
| 21:57.59 | ryanlin | ahh |
| 21:58.02 | *** join/#asterisk Chilling_Silence (~Chilling_@smtp.fcsltd.co.nz) |
| 21:58.07 | ryanlin | maybe this will work |
| 21:58.12 | ryanlin | let me try it |
| 21:58.27 | Chilling_Silence | Is there somebody who knows about how Asterisk behaves when it loses internet connectivity, and SIP Peers (Extensions) then cannot register? |
| 21:58.28 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:58.50 | Chilling_Silence | I'm hoping somebody can shed some light on the subject of using the PBX while the internet is "offline" |
| 21:58.54 | p3nguin | chilling_silence: First of all, SIP peers are not extensions. |
| 21:59.05 | Chilling_Silence | p3nguin: OK, I stand corrected :D |
| 21:59.40 | Chilling_Silence | I run "sip show peers" and it shows me both my SIP <-> ISDN trunk (Patton SN4554), and it also shows me all my Exts, as well as the SIP trunks |
| 21:59.51 | p3nguin | chilling_silence: Second, if the peers are not registered, if you try to make calls to them, the dialplan should progress to the next step after the Dial(). |
| 22:00.13 | Chilling_Silence | But not if it's not allowing the Exts to register |
| 22:00.24 | p3nguin | Extensions don't register, peers do. |
| 22:00.28 | p3nguin | already been over this part. |
| 22:00.33 | Chilling_Silence | So what do Extensions do ? |
| 22:00.46 | p3nguin | whatever you tell them to do. See extensions.conf. |
| 22:00.47 | *** join/#asterisk voxter (~voxter@m285636d0.tmodns.net) |
| 22:01.41 | Chilling_Silence | OK so what you're saying is: In the event of connectivity loss to the internet, there should be no functionality impairment for asterisk? And that my issue is something FreePBX specific? |
| 22:01.46 | ryanlin | hmm |
| 22:01.57 | ryanlin | when i make a call, it goes to a busy signal |
| 22:01.58 | tjwallace | \quit |
| 22:02.01 | p3nguin | FreePBX is an issue all by itself. |
| 22:02.13 | Chilling_Silence | p3nguin: Point taken ;) |
| 22:03.06 | p3nguin | Asterisk doesn't care if the devices are registered or not. It will want to dial them if your dialplan says to dial them, but if they are not available for calls, whatever is after the Dial() command is what happens next. |
| 22:03.27 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 22:03.28 | p3nguin | I believe changing the fallthrough setting might change that behavior, too. |
| 22:03.39 | korcan | debug within asterisk is giving me "NOTICE[23347]: manager.c:1087 authenticate: xx.xx.xx.xx failed to pass IP ACL as 'test'" |
| 22:03.43 | korcan | my manager.conf has permit=0.0.0.0/0.0.0.0 set |
| 22:04.09 | Chilling_Silence | You seem like you may be able to offer insight on my next part of the question then. If I'm losing calls, mid-call, from a variety of sources (ISDN or SIP to ITSP), to / from a variety of 7 different handsets, what's likely to be causing that? It's not internet, (As it affects ISDN), it's not codecs, and we've even tried replacing all the hardware |
| 22:04.18 | Chilling_Silence | p3nguin: Fallthrough setting? |
| 22:04.20 | p3nguin | korcan: I would remove the deny and permit lines if you're just going to permit everything anyway. |
| 22:04.20 | Chilling_Silence | goes to google it |
| 22:04.39 | korcan | p3nguin, I will try that now.. |
| 22:04.50 | ryanlin | p3nguin: any ideas? |
| 22:04.51 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 22:04.57 | p3nguin | chilling_silence: autofallthrough=yes or autofallthrough=no |
| 22:05.35 | p3nguin | ryanlin: The dialplan isn't right. |
| 22:05.47 | korcan | [May 13 18:05:11] NOTICE[31442]: manager.c:1087 authenticate: xx.xx.xx.xx failed to pass IP ACL as 'test' |
| 22:05.47 | korcan | [May 13 18:05:11] NOTICE[31442]: manager.c:1113 authenticate: xx.xx.xx.xx failed to authenticate as 'test' |
| 22:05.47 | korcan | [May 13 18:05:12] == Connect attempt from 'xx.xx.xx.xx' unable to authenticate |
| 22:06.26 | korcan | p3nguin, any other ideas? |
| 22:06.48 | korcan | p3nguin, no need to reload anything? correct? |
| 22:07.01 | Chilling_Silence | p3nguin: autofallthrough=yes is default, so that's never been changed... |
| 22:07.32 | Chilling_Silence | korcan: You've removed both the allow and the deny lines? It says it's failing on the IP ACL |
| 22:07.47 | korcan | I put a ; in front of them |
| 22:08.18 | Chilling_Silence | Try a #, but even that will be interpereted eventually (So the errors in the logs tell me at least), so just remove them altogether |
| 22:08.40 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 22:08.42 | korcan | [test] |
| 22:08.42 | korcan | secret = test |
| 22:08.42 | korcan | read = All |
| 22:08.42 | korcan | write = All |
| 22:09.05 | korcan | same error |
| 22:09.25 | korcan | I have been banging my head on my desk all day... |
| 22:09.59 | korcan | any other ideas? anyone? |
| 22:10.01 | Chilling_Silence | korcan: I have mine as: http://pastie.org/959454 |
| 22:10.20 | Chilling_Silence | For the TAPI interfaces this particular client uses from their LAN |
| 22:10.51 | *** join/#asterisk Znuff (~ibm86@2001:0:53aa:64c:2832:3add:a6d0:dcf6) |
| 22:11.09 | Chilling_Silence | p3nguin: So you wouldn't happen to have any ideas on what would cause a call to randomly drop, mid-call, and then have the SIP debug show it doing a normal hangup? |
| 22:11.20 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 22:11.28 | korcan | Chilling_Silence, mine is above... |
| 22:11.37 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 22:11.51 | Chilling_Silence | p3nguin: It happens semi-regularly, but irregularly. Sometimes there will be multiple calls and one will hang up, sometimes they all hang up within 10 seconds of each other |
| 22:12.09 | Chilling_Silence | korcan: What are you trying to connect through the AMI? |
| 22:13.25 | korcan | http://pastie.org/959462 |
| 22:13.31 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 22:13.32 | korcan | thats the php code... |
| 22:13.43 | korcan | the variables are set before that part... |
| 22:13.58 | *** join/#asterisk smooth_penguin (~smoove@59.95.6.110) |
| 22:14.20 | Chilling_Silence | korcan: I've seen that code floating around the internet before :D |
| 22:14.37 | korcan | The crazy thing is it works from the DMZ |
| 22:14.40 | korcan | no errors |
| 22:14.53 | korcan | when I put it outside my walls it breaks... |
| 22:15.06 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 22:15.43 | korcan | pbx is on the LAN |
| 22:15.47 | Chilling_Silence | What host are you trying to use for the PBX IP? |
| 22:15.54 | Chilling_Silence | $strHost? |
| 22:16.12 | korcan | public IP that points to the pbx |
| 22:16.20 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 22:16.33 | Chilling_Silence | Just making sure |
| 22:16.40 | korcan | the pbx is 10.1.1.50 |
| 22:16.44 | Chilling_Silence | And 5038 TCP must be being forwarded, as it's complaining about the authentication |
| 22:16.45 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 22:16.54 | korcan | yes |
| 22:17.13 | Chilling_Silence | Do you know the IP Address of your webserver that it's coming from? It's not going to be changing? You could setup deny 0.0.0.0/0.0.0.0 and then allow= the webserver IP? |
| 22:17.21 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 22:17.30 | Chilling_Silence | So if you run that same PHP code in your LAN, it works OK ? |
| 22:17.33 | korcan | I also have this in the manager.conf bindaddr = public_ip |
| 22:17.38 | korcan | yes |
| 22:17.50 | korcan | DMZ, but it points the LAN ip |
| 22:17.56 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 22:18.41 | korcan | DMZ webserver is 192.... PBX ip is 10.... |
| 22:18.59 | korcan | Publis Webserver has public IP |
| 22:19.08 | korcan | DMZ server connects to 10.... |
| 22:19.12 | korcan | and it works |
| 22:19.19 | korcan | Chilling_Silence, does that make sense? |
| 22:19.25 | p3nguin | korcan: If you edit config files, you have to reload SOMETHING. Either the module that the conf is for or the component or the entire asterisk. |
| 22:19.48 | korcan | manager gets read every time it is accessed |
| 22:19.58 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 22:19.58 | korcan | I think :) |
| 22:20.07 | korcan | er, I read |
| 22:20.16 | Chilling_Silence | That's what I thought too, I didn't believe you had to reload if you've edited manager.conf ? I certainly don't ? |
| 22:20.30 | *** join/#asterisk Znuff (~ibm86@2001:0:53aa:64c:2832:3add:a6d0:dcf6) |
| 22:20.51 | p3nguin | chilling_silence: I'd want to see a debug of a call when it is being dropped to try to figure out it. |
| 22:20.58 | korcan | so how do I get manager to bind to this public ip? I specified the public ip with bindaddr = |
| 22:21.07 | Chilling_Silence | p3nguin: That can be arranged ;) 1 sec I've got several |
| 22:21.28 | korcan | I have 2 bindaddr lines, one for 10.1.1.50 and the second with 65.x.x.x |
| 22:21.36 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 22:21.48 | korcan | p3nguin, any ideas on my dilemma? |
| 22:22.08 | p3nguin | korcan: I don't think you can bind to two addresses. You get to choose one or all. |
| 22:22.18 | Chilling_Silence | p3nguin: this has the entire call from the phone number ending in 0499: |
| 22:22.18 | Chilling_Silence | http://pastie.org/956126 |
| 22:22.24 | korcan | ok, so 0.0.0.0? |
| 22:22.36 | Chilling_Silence | Yup, no specific need to bindaddr |
| 22:22.37 | p3nguin | korcan: right, that would be all addresses on all interfaces. |
| 22:23.05 | korcan | maybe thats it... the pbx doesn't have this ip set anywhere |
| 22:23.19 | korcan | the public ip is not on an interface |
| 22:24.08 | p3nguin | You can't bind to addresses that are not on directly connected interfaces. |
| 22:24.22 | p3nguin | If the box is on a LAN, you've probably only got a single private address on it. |
| 22:24.36 | korcan | p3nguin, Chilling_Silence, gotta run, thanks for the help.. I will try to bind it later :) |
| 22:24.39 | Chilling_Silence | p3nguin: You'll see multiple calls failing to / from that number if you search for 0499 in the above pastie. |
| 22:24.42 | korcan | :) |
| 22:25.56 | Chilling_Silence | korcan: good luck ;) |
| 22:26.09 | Chilling_Silence | p3nguin: I've probably got tcpdump logs for those calls also |
| 22:26.16 | Chilling_Silence | Around 2844 you see one of the hangups |
| 22:26.30 | Chilling_Silence | *line 2844 on that pastie I mean |
| 22:28.20 | *** part/#asterisk Brooklyn (miracle@the.city.that.never-sleeps.us) |
| 22:28.37 | Chilling_Silence | p3nguin: You would be the worlds biggest legend if you could figure that one out. I've literally had dozens of "VoIP Experts" look at it and shrug, nobody can seem to figure out why it's happening |
| 22:30.38 | p3nguin | I really don't want to sift through 3180 lines of what is partially useless. I'd rather have a debug on a single call which at some point drops or otherwise fails. |
| 22:34.39 | paulc | Anyone in here use app_konference? |
| 22:36.08 | *** join/#asterisk farkus_ (chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 22:37.35 | Chilling_Silence | p3nguin: Yeah so if you look at like 2844 is where it "drops" |
| 22:37.43 | Chilling_Silence | The few lines prior show the call active |
| 22:37.48 | Chilling_Silence | Basically I rang the guy back a couple of times |
| 22:39.36 | Chilling_Silence | p3nguin: Sorry, 2884. Ext is 708 (Cordless), and 192.168.0.250 as you can probably see is the PBX |
| 22:46.54 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 22:46.59 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 22:49.56 | Chilling_Silence | p3nguin: Any ideas? :D |
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| 23:08.14 | sawgood | hi group! |
| 23:08.21 | sawgood | a few random thoughts for the day ... |
| 23:08.42 | sawgood | is there any 'advantage' to using a setting above 9 for core set verbose 9 |
| 23:15.27 | sawgood | If I wanted Asterisk to always start with both core and verbose logging set a 9, which file would I edit with vi to do this? |
| 23:20.03 | leifmadsen | sawgood: how goes your asterisk start? |
| 23:20.25 | [TK]D-Fender | [19:20]===sawgood: member of #freepbx, #asterisknow, and #asterisk |
| 23:20.30 | [TK]D-Fender | leifmadsen: Guess :) |
| 23:20.37 | leifmadsen | sawgood: there isn't an advantage to using anything above 5 |
| 23:20.43 | leifmadsen | most are 4 or lower |
| 23:21.09 | sawgood | thak you, sir! |
| 23:21.21 | sawgood | My Asterisk start is going good ... thanks for askin |
| 23:21.32 | ChannelZ | sawgood: cli.conf |
| 23:22.14 | sawgood | ChannelZ: thank you |
| 23:22.28 | ChannelZ | though if you want it as such in you logfiles as well, you'd have to run the parent process with whatever options.. so edit your init script, or whatever it is that runs * for you on boot |
| 23:24.35 | ChannelZ | oh.. no actually cli.conf I think is read on * startup period, for some reason I was thinking it was only for connected consoles. Snap. |
| 23:25.12 | sawgood | no problem ... |
| 23:25.31 | sawgood | I think I am going to edit the file to start both verbose and debug at 5 |
| 23:25.43 | sawgood | so, I do not have to start that which each Asterisk restart |
| 23:25.55 | sawgood | I do not mind the output on the screen |
| 23:26.02 | sawgood | I kind of like it myself |
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| 23:34.13 | sawgood | say, I just wanted to 'know' what level of verbose logging is currently happening ... is there a command to confirm it (rather than having to exit out and back in again)? |
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| 23:36.51 | sawgood | got it ... never mind ... sorry! |
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