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| 01:55.33 | *** join/#asterisk lkthomas (lkthomas@n119236143112.netvigator.com) |
| 01:55.35 | lkthomas | hey guys |
| 01:55.45 | lkthomas | any free softphone you guys using to test asterisk ? |
| 01:55.51 | lkthomas | x-lite web site can't access now |
| 01:56.22 | WIMPy | zoiper |
| 01:56.39 | WIMPy | Or any of the hundreds of others. |
| 01:57.06 | lkthomas | thanks, let me download it |
| 02:07.16 | lkthomas | WIMPy: first problem, when I setup softphone, it ask for domain username and password |
| 02:07.23 | lkthomas | where to set username and password on asterisk ? |
| 02:08.22 | lkthomas | OH, username = extension |
| 02:17.12 | WIMPy | No |
| 02:17.18 | lkthomas | elastix seems good |
| 02:17.27 | WIMPy | Create a section in sip.conf or iax.conf |
| 02:17.29 | lkthomas | WIMPy: I could register if use username = extension number |
| 02:18.27 | WIMPy | So the evil cracker already knows your usernames and only needs to guess your passwords. |
| 02:18.51 | lkthomas | you are right, so what should I create on sip.conf ? |
| 02:19.15 | WIMPy | User some less obvious usernames. |
| 02:19.27 | WIMPy | -r |
| 02:40.03 | lkthomas | WIMPy: do you suggest start to learn asterisk using freepbx interface or build from scratch ? |
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| 02:43.52 | WIMPy | If you want to learn Asterisk, forget about the GUIs. |
| 02:44.16 | WIMPy | If you just want something to do some basic stuff, you can try them. |
| 02:46.18 | lkthomas | I think GUI could give me basic idea on what asterisk could be done, and I have to drill from that |
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| 02:47.34 | WIMPy | Maybe it can give you some hints on what is possible, but maybe not on how to do it. And beware of the diffeering terminology. |
| 02:48.03 | lkthomas | right |
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| 03:23.41 | lkthomas | WIMPy: I see this on applications.conf: Macro(user-callerid) |
| 03:23.47 | lkthomas | where is the Macro defined ? |
| 03:24.21 | WIMPy | extensions.conf |
| 03:24.31 | WIMPy | Or anything included there. |
| 03:26.39 | lkthomas | I did grep and found out extension is calling macro |
| 03:26.43 | lkthomas | but nothing define that macro |
| 03:30.29 | lkthomas | no wait |
| 03:30.35 | lkthomas | I seems found something |
| 03:33.38 | DrDigital | is there a way for ip-pbx systems to work with cell phones? like in actually providing them service? |
| 03:33.59 | DrDigital | like if i bought a radio and an antenna and put it on a building |
| 03:34.07 | DrDigital | to service the building |
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| 03:41.11 | *** join/#asterisk emsLinux (~emsLinux@adsl190-28-186-91.epm.net.co) |
| 03:42.25 | emsLinux | Good night people, any of you guys know if there is any issue with Grandstream HandyTone 502 ATAs connecting to the last version of Asterisk, no matter what i do, can't make it register to my server. |
| 03:45.16 | lkthomas | emsLinux: I am sorry that I am a newbie on VOIP, but did you check asterisk log and ATA log as well ? |
| 03:52.10 | lkthomas | I am confused with Macro() |
| 03:52.29 | lkthomas | is it a subfunction to do decision making point if I send the user to a Macro() ? |
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| 04:02.05 | kaldemar | lkthomas: Macro is a dialplan application that executes a block of dialplan code. |
| 04:02.31 | kaldemar | emsLinux: what does CLI say with verbosity and sip debug enabled? |
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| 04:03.37 | lkthomas | isn't include() already does what it suppose to do ? |
| 04:04.09 | kaldemar | lkthomas: Macro(foo) will execute "exten => s" in [macro-foo]. |
| 04:04.32 | lkthomas | sorry, what extern => s do ? |
| 04:04.46 | kaldemar | lkthomas: no. include, not include(). include doesn't execute anything, it just includes a context to another one. |
| 04:05.04 | lkthomas | I see |
| 04:05.10 | lkthomas | so it's like a subfunction |
| 04:05.14 | kaldemar | exten, not extern. it doesn't do anything, it is a part of an extension definition. |
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| 04:23.02 | samandiriel | Hullo there... quick question about variable scope. If I use GOSUB to call a routine in a completely different context, and both contexts have a variable with the same name... can the GOSUB overwrite the value of the variable with the same name in the calling context? |
| 04:23.24 | samandiriel | I thought the context variable scope stayed separate, but it looks like it doesn't... |
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| 04:32.41 | kaldemar | samandiriel: yes, it can overwrite it. contexts do not separate variable spaces, channels do. |
| 04:33.13 | samandiriel | ooo, poop. now I know where some subtle bugs have been coming from. thanks kaldemar |
| 04:34.38 | samandiriel | I think I will go back and namespace all my variables by tossing the context name on the end of them... |
| 04:36.09 | lkthomas | guys, when I dial *97, it goes to my voice mail, then ask for password |
| 04:36.23 | lkthomas | but I can't find where in extension config asking for password |
| 04:37.08 | lkthomas | http://pastebin.com/DwCe7e8L |
| 04:38.00 | lkthomas | if I am not wrong, this should be core function when I call *97 |
| 04:38.23 | samandiriel | possibly happening in one of the macros? |
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| 04:42.19 | lkthomas | would it be in VoiceMailMain() ? |
| 04:42.23 | lkthomas | it seems a predefiend function |
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| 04:43.15 | samandiriel | ah, yes, that's likely it |
| 04:43.31 | lkthomas | I didn't know there is a voicemail function |
| 04:43.31 | samandiriel | you can pass an option to voicemail to suppress password I think |
| 04:43.59 | lkthomas | asterisk book tell me that need to be DIY instead |
| 04:44.06 | samandiriel | http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMailMain |
| 04:44.17 | lkthomas | yes |
| 04:44.22 | samandiriel | if you put an 's' in front it will skipp asking for the password |
| 04:44.44 | lkthomas | you mean like voicemailmain(s) ? |
| 04:44.49 | samandiriel | VoiceMailMain(s5000) |
| 04:44.56 | samandiriel | you have to have the mailbox number in front |
| 04:45.01 | samandiriel | I mean after |
| 04:45.02 | samandiriel | the s |
| 04:45.08 | lkthomas | I am confused |
| 04:45.14 | lkthomas | current config is like this: |
| 04:45.18 | lkthomas | exten => *97,n,VoiceMailMain() |
| 04:45.25 | lkthomas | nothing in the bracket |
| 04:45.33 | lkthomas | how should I modify it ? |
| 04:46.04 | samandiriel | try exten => *97,n,VoiceMailMain(s${EXTEN}) |
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| 04:52.08 | kaldemar | more like VoiceMailMain(5000,s) |
| 04:52.59 | kaldemar | *97,n,VoiceMailMain(s${EXTEN}) would expect the mailbox to be "s*97", and ask for a password. |
| 04:53.10 | samandiriel | I think both would work... just was scrolling down, the first is the older version pre 1.2 |
| 04:53.20 | kaldemar | core show application VoiceMailMain |
| 04:53.27 | lkthomas | god damn |
| 04:53.34 | samandiriel | fie... well, serves me right for trying to helpful :P |
| 04:53.41 | lkthomas | I am not sure what I have done, but after I key in password, it disconnect |
| 04:53.57 | kaldemar | that's why you shouldn't use voip-info as a command reference. it's the last place you should look for syntax. |
| 04:54.33 | samandiriel | what's a better resource then? |
| 04:54.53 | kaldemar | quite a bit of the information there is outdated and written for some other version. |
| 04:55.23 | kaldemar | application documentation in asterisk itself is the best resource. you'll always get the syntax for the version you're currently using. |
| 04:55.48 | kaldemar | in CLI, "core show applications" and "core show application <application>" |
| 04:56.15 | samandiriel | I find https://wiki.asterisk.org/wiki/display/AST/Asterisk+Command+Reference helpful, but lacking in examples. some of the stuff is obtuse without a working example |
| 04:56.37 | samandiriel | I think that the wiki and core show have very similar / the same info? |
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| 05:00.51 | kaldemar | samandiriel: yes, but the wiki has the information for the newest version, which you might not use. |
| 05:01.01 | lkthomas | damn |
| 05:01.05 | lkthomas | it still ask for password |
| 05:01.05 | kaldemar | samandiriel: and the wiki also has mistakes. |
| 05:01.33 | kaldemar | lkthomas: what do you see in CLI when making a call? |
| 05:01.41 | c0dyhi11 | Hello, I have an AEX808E equiped with the VPMADT032 echo cancellation module. I just installed AsteriskNow and when the system boots it throws quite a few errors when it comes to DAHDI |
| 05:01.57 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 05:02.09 | lkthomas | no, do I need to enable debug mode on cli or something ? |
| 05:02.09 | c0dyhi11 | I did a yum update to get the lates versoin of everything but it is still throwing errors. |
| 05:02.38 | kaldemar | lkthomas: "core set verbose 10" |
| 05:02.40 | samandiriel | kaldemar: thanks. I'm using 1.8, so it should be pretty congruent. as for mistakes, I can probably live with them... what's nice about the wiki is that it also shows them all in a nice list, which makes it easy to read and discover new functions. I have definitely been taking EVERYTHING with a grain of salt, tho!!! |
| 05:02.42 | c0dyhi11 | I looked at the card and it has 3 red LEDs light up on the echo canceller |
| 05:03.10 | c0dyhi11 | does anyone know what those red LEDs mean? |
| 05:05.25 | lkthomas | kaldemar: it does not pass s into function |
| 05:08.16 | kaldemar | lkthomas: pastebin what you see in CLI. |
| 05:08.18 | kaldemar | ~pb |
| 05:08.18 | 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.us , 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. |
| 05:09.31 | ChannelZ | Does Android support the 'gsm in wav' codec for voicemails? |
| 05:09.52 | lkthomas | this is stupid |
| 05:10.05 | lkthomas | I comment out voicemailmain() and voicemail still working |
| 05:10.15 | lkthomas | so it does not seems to be that function I am modifing |
| 05:10.22 | lkthomas | kaldemar: let me pastebin it |
| 05:11.48 | lkthomas | http://pastebin.com/YQHAAvZS |
| 05:13.08 | lkthomas | from-internal:106 |
| 05:13.11 | lkthomas | what is 106 means |
| 05:13.29 | kaldemar | lkthomas: either you commented it out in the wrong place or did not reload dialplan after you did it. |
| 05:13.38 | kaldemar | 106 is a priority. |
| 05:13.42 | kaldemar | ~book |
| 05:13.42 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 05:13.56 | kaldemar | lkthomas: ^ that will tell you more. |
| 05:14.15 | samandiriel | I'll second that - that book has been VERY helpful to me |
| 05:15.12 | lkthomas | [from-internal] |
| 05:15.14 | lkthomas | include => from-internal-xfer |
| 05:15.14 | lkthomas | include => bad-number |
| 05:15.31 | lkthomas | so I have to track down on from-internal-xfer ? |
| 05:15.52 | lkthomas | it doesn't seems to be there |
| 05:16.58 | kaldemar | lkthomas: what are you trying to do? |
| 05:17.26 | kaldemar | lkthomas: are you trying to learn asterisk or torture yourself with freepbx configs for the fun of it? |
| 05:18.01 | lkthomas | my company said we are going to use freepbx and do modify on it |
| 05:18.11 | lkthomas | before I could modify, I want to understand logic of the core piece |
| 05:18.59 | samandiriel | lkthomas: I feel your pain there... I'm doing much the same thing. While some of the stuff has been good examples, I've quickly come to the conclusion that if you need to mod stuff freepbx is more a hindrance than anything |
| 05:19.12 | lkthomas | I know |
| 05:19.16 | lkthomas | but I don't make this decision |
| 05:19.19 | lkthomas | anyway |
| 05:19.24 | lkthomas | how could I do detail trace here |
| 05:19.45 | lkthomas | pirority is hard to chase as all using "n" |
| 05:20.28 | samandiriel | when you're in the asterisk CLI, turn up the verbosity just like kaldemar said to |
| 05:20.37 | lkthomas | it's on 10 now |
| 05:21.13 | samandiriel | I'm using putty and I capture it all to a file; I find it easier to go thru the debug in the file than in my terminal window |
| 05:21.27 | lkthomas | samandiriel: it's on pastebin now |
| 05:22.54 | kaldemar | lkthomas: you better hang around in #freepbx then. freepbx is not supported on this channel and people here tend to avoid tinkering with it. |
| 05:23.34 | lkthomas | I will ask on freepbx channel, but how do you do trace on such situation ? |
| 05:23.39 | lkthomas | it's generate asterisk question |
| 05:24.31 | samandiriel | you're already tracing if you're watching commands fly by in the CLI lkthomas |
| 05:24.46 | kaldemar | lkthomas: you already did. there is an extension *97 in the context from-internal with priority 106 that has the VoiceMailMain app. |
| 05:25.10 | lkthomas | now if priority is using "n", how could you trace down what is 106 ? |
| 05:26.34 | kaldemar | the 106 is most likely 106 and not "n". otherwise there would have to be at least 106 priorities in the extension, which would be idiotic. but, you would count up from the previous numbered priority, be it 1 or something else. |
| 05:27.22 | lkthomas | so if it call multiple macros, I have to follow the code as well ?! |
| 05:28.16 | samandiriel | yup. thru oodles and oodles of files and loops |
| 05:28.31 | lkthomas | I found the core function |
| 05:28.35 | lkthomas | it's 3lines away |
| 05:28.52 | lkthomas | exten => *97,n,VoiceMailMain() <--- but I still don't understand what is it use for |
| 05:28.58 | lkthomas | it just call without variable |
| 05:29.00 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 05:29.45 | samandiriel | have you read the book? it really does help a lot: ~book |
| 05:29.57 | samandiriel | ~book |
| 05:29.57 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 05:30.33 | lkthomas | I am reading book and playing with asterisk |
| 05:30.41 | lkthomas | throwing books to me doesn't help my friend |
| 05:31.51 | samandiriel | you can call a function without a variable, btw. depends on the function |
| 05:32.17 | lkthomas | I know, my question is that when you call voicemailmain() without variable, what would happen |
| 05:32.25 | lkthomas | it seems hangup to me now because I just tested |
| 05:32.48 | kaldemar | lkthomas: you would know what VoiceMailMain() does if you read the application documentation. "core show application VoiceMailMain" |
| 05:32.53 | samandiriel | when I do it, it asks me for a password. providing of course that you have voicemail set up for that extension in freepbx |
| 05:33.42 | samandiriel | kaldemar: I think freepbx is probably getting in the way, just like it does for me |
| 05:33.52 | kaldemar | samandiriel: it does not ask for a password, it asks for a mailbox. |
| 05:34.05 | samandiriel | all that crap it sets for you in the background, trying to be 'helpful'... much likes windows :P |
| 05:34.07 | lkthomas | funny thing is that it doesn't |
| 05:34.14 | lkthomas | wait |
| 05:34.38 | samandiriel | kaldemar: I have mine set up to put the extension number in based on callerid :) |
| 05:35.39 | samandiriel | hard to keep track of what I have fiddled and what I haven't.... |
| 05:35.41 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 05:35.46 | samandiriel | wheee! |
| 05:36.35 | lkthomas | exten => *97,check+101(mbexist),VoiceMailMain(${AMPUSER}@${VMCONTEXT},s) <--- on this function, priority is defined by current priority + 101 ? |
| 05:36.58 | lkthomas | check+101(mbexist) <--- what is this means ? |
| 05:37.14 | samandiriel | "does the mailbox exist" |
| 05:37.26 | samandiriel | would be my guess |
| 05:37.46 | lkthomas | let me think |
| 05:37.53 | kaldemar | mbexist is a label |
| 05:38.15 | lkthomas | what does it do this this line ? |
| 05:38.26 | lkthomas | add more numbers to priority ? |
| 05:41.40 | lkthomas | kaldemar: is it possible to "Echo" mbexist value on CLI during execution ? |
| 05:43.06 | kaldemar | lkthomas: it is invalid syntax. |
| 05:43.27 | lkthomas | kaldemar: why invalid ? |
| 05:43.32 | kaldemar | lkthomas: mbexist is not a variable, it is just a label. it doesn't have a value. |
| 05:43.53 | lkthomas | ok, so it does not have any useful meaning in the program right ? |
| 05:43.59 | kaldemar | lkthomas: a priority is a number or "n", with or without a label. other crap like that is invalid. |
| 05:45.21 | kaldemar | lkthomas: no, unless the version of asterisk is modified to interpret it in some way. |
| 05:45.57 | lkthomas | ok, because I don't get what you mean as a "label" |
| 05:49.11 | *** join/#asterisk dadad (~Miranda@commsrv.spb.ec-energo.ru) |
| 05:52.33 | wdoekes2 | lkthomas: you can jump to a number or to a label -- which is just an alias for that number |
| 05:52.41 | kaldemar | exten => <extension>,<priority[(<label>)]>,<application> |
| 05:53.21 | lkthomas | I see |
| 05:54.37 | wdoekes2 | having said that.. normally, you don't use numbers, except for '1'.. you use 'n'.. and labels if you want to jump to anywhere else than 1 |
| 05:55.16 | lkthomas | so it I have 1 -> n -> n(three), next time I could call three and will jump to number 3 of n ? |
| 05:56.04 | wdoekes2 | 3 of n? if you jump to three or 3 you go to the same place |
| 05:56.10 | lkthomas | exten => *97,n,GotoIf($["${VMBOXEXISTSSTATUS}" = "SUCCESS"]?mbexist) |
| 05:56.13 | lkthomas | right on |
| 05:56.30 | lkthomas | so if vmboxexist = success, jump to mbexist line |
| 05:56.41 | lkthomas | which is exten => *97,check+101(mbexist),VoiceMailMain(${AMPUSER}@${VMCONTEXT},s) |
| 05:57.07 | lkthomas | start to understand the structure now :P |
| 05:57.12 | wdoekes2 | if you remove the garbage, like kaldemar said |
| 05:58.25 | lkthomas | hey, my boss also ask me to look into voice recording, what is this ? |
| 05:58.57 | wdoekes2 | the Monitor app? |
| 05:59.31 | wdoekes2 | or Record, depending on what your boss means ;) |
| 05:59.38 | lkthomas | OH, it record the conversation right ? |
| 05:59.49 | wdoekes2 | CLI> core show applications |
| 05:59.54 | wdoekes2 | CLI> core show application SomeApplication |
| 06:00.25 | lkthomas | <PROTECTED> |
| 06:02.42 | lkthomas | I see |
| 06:02.44 | lkthomas | nice |
| 06:02.46 | lkthomas | Monitor apps :P |
| 06:02.57 | lkthomas | so powerful my friend |
| 06:14.53 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 06:17.34 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 06:21.06 | irroot | morning trying something intresting with G+ posting my * changes to the company and digium circles lets see how it works |
| 06:22.28 | *** join/#asterisk dmz (~dmz@64.203.235.49) |
| 06:27.05 | *** join/#asterisk binilivi (~jorixFA37@banjo12.orchestra.cse.unsw.EDU.AU) |
| 06:30.51 | *** join/#asterisk wasanzy (~emmanuel@196.201.43.55) |
| 06:36.46 | *** join/#asterisk jmls (~Julian@host217-36-208-155.in-addr.btopenworld.com) |
| 06:39.13 | *** join/#asterisk galaxywatcher (~galaxywat@pdpc/supporter/active/galaxywatcher) |
| 06:40.09 | dadad | Hi. May I ask? When closed issue will be accepted to release? |
| 06:40.49 | ChannelZ | the next time usually |
| 06:45.26 | wasanzy | Kaldemar: hello good morning |
| 06:48.50 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 06:49.36 | kaldemar | wasanzy: morning. i took a look at your interface dumps, and it did not look normal. there was only packets from .94 to .11. so there was no audio going from .11 to .94. see that the phone is not on mute and the microphone works and there is no silence suppression or something similar enabled. |
| 06:50.13 | Cobadol | Its closed 8 July but not appear in 1.8.5. I need to wait 1.8.6 version, i guess? What date of 1.8.6 release (aproximately)? |
| 06:51.05 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 06:53.00 | wasanzy | kaldemar: thank you very much. but I will pastebin my sip.conf for you again while I wait for the other guys to come in so I can use their machine for the test |
| 06:55.02 | wasanzy | here is my sip.conf: http://pastebin.com/wrS4GD8x |
| 06:55.34 | wasanzy | but one thing is that, the two machines are able to talk on skype voip correctly |
| 06:57.17 | kaldemar | wasanzy: well, on skype you're using a different software and a totally different network scenario and protocols. so that really only proves that the machines have working audio input/output, nothing more. |
| 06:58.27 | wasanzy | ok, can you look at my conf, maybe some thing might be wrong with it |
| 06:58.29 | kaldemar | wasanzy: no need to have both directmedia and canreinvite. canreinvite is an old name for the same option. |
| 06:58.52 | wasanzy | oh ok |
| 06:58.59 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 06:58.59 | wasanzy | I will change that then |
| 06:59.56 | wasanzy | could that be the problem? |
| 07:00.06 | *** join/#asterisk MariusAgon (~MariusAgo@89.249.83.26) |
| 07:00.23 | kaldemar | there is nothing wrong with config. you could try to set directmedia=no for the peers to have the audio go through asterisk. |
| 07:00.35 | kaldemar | having both directmedia and canreinvite is not the problem. |
| 07:01.17 | kaldemar | the problem is not asterisk, it is twinkle, firewall on the machines or your network. |
| 07:01.42 | wasanzy | oh ok good |
| 07:02.22 | wasanzy | I will check the twinkle's codec or the machines to disable firewall then |
| 07:05.26 | kaldemar | twinkle's codec is not likely to be a problem since they both know the same codecs. but you should enable disallow and allow lines in sip.conf. |
| 07:06.53 | wasanzy | disallow and allow you mean the codecs part in the conf? |
| 07:08.25 | lkthomas | anyone using voicwmailmain() ? |
| 07:10.43 | wasanzy | kaldemar: which of the codecs is best to use? |
| 07:13.09 | kaldemar | wasanzy: yes, the codecs part. what is the best one depends on what you want. if you want low bandwidth usage, continue with gsm, if you want better audio quality, use alaw or ulaw. |
| 07:13.30 | kaldemar | lkthomas: what do you want to know? |
| 07:16.08 | wasanzy | Kaldemar: thank you |
| 07:16.26 | wasanzy | I will try all this options can get back to you |
| 07:19.16 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:24.25 | lkthomas | kaldemar: what if I want to change the menu structure of voicemailmain(), how could I do that |
| 07:25.05 | lkthomas | brb |
| 07:25.08 | lkthomas | reboot computer |
| 07:26.09 | *** join/#asterisk lkthomas (~lkthomas-@n119236143112.netvigator.com) |
| 07:26.13 | lkthomas | back |
| 07:26.16 | Cobadol | lkthomas: may be wrong, but change structure of menu by changing source code only. |
| 07:26.21 | kaldemar | lkthomas: by changing the application source code (app_voicemail.c) and recompiling the module. |
| 07:27.35 | lkthomas | woo |
| 07:27.39 | lkthomas | that's complicated, LOL |
| 07:28.18 | lkthomas | let's try to change apache source code if I want to add a word on web :P kidding :P |
| 07:29.24 | kaldemar | lkthomas: maybe minivm suits your needs better. http://www.asterisk.org/node/48326 |
| 07:30.04 | kaldemar | core show applications like Minivm, core show functions like MINIVM |
| 07:31.20 | kaldemar | lkthomas: if you only want to change the voice prompts, you can record your own. :) |
| 07:34.15 | Cobadol | I have to compile module pbx.c on virtual machine and copy compiled module to real similar (mean from same distrib - AsteriskNOW) computer. But it havn't work well. Asterisk starts, but dialplan is not loaded. Can you idvise me where search a problem? |
| 07:41.07 | kaldemar | Cobadol: do you see any warnings or errors when starting asterisk with "asterisk -vvvc"? |
| 07:46.23 | Cobadol | kaldemar: Maybe, but not really matters. I mean where's no restrictions to change right compiled (with take in mind OS version, hardware) module with old one, right? No CRC, other checks for modules? |
| 07:48.07 | kaldemar | if it's built in the same architecture and against compatible headers, it should work. |
| 07:48.39 | kaldemar | asterisk will tell you if it doesn't like a module. |
| 07:49.30 | lkthomas | most of the shit already in elastix |
| 07:49.31 | Cobadol | kaldemar: Ok. Thanks. I can't take any information now. Asterisk in work. |
| 08:10.47 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 08:11.28 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 08:11.43 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 08:12.43 | lkthomas | I have a question on DID |
| 08:13.08 | lkthomas | if I get DID number from third party provider |
| 08:13.22 | lkthomas | so when user dial from worldwide to my number, they have to pay long distance ? |
| 08:15.03 | kaldemar | lkthomas: depends on where they dial from and where the provider has their PSTN termination. |
| 08:15.18 | lkthomas | so there is no catch all DID provider right ? |
| 08:15.40 | lkthomas | I mean, is it possible to get a DID which call everywhere with same price as local phone call ? |
| 08:15.59 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:16.33 | kaldemar | not possible. that's pretty much why you'd want to use VoIP over the internet in the first place. |
| 08:16.34 | *** join/#asterisk Eye_Trust_No1 (~Eye_Trust@84.203.229.194) |
| 08:17.48 | lkthomas | kaldemar: ok, and also, is it the best thing to do is to put asterisk box which is same location as DID termination to lower down latency ? |
| 08:18.38 | kaldemar | depends on what you do with the box. |
| 08:19.23 | lkthomas | but if I get DID with USA number then I put the asterisk box in china, user dial DID will route from USA to china and back to USA, am I correct ? |
| 08:20.17 | lkthomas | http://www.phone2net.com/charges/ <--- what is extra channel means? same DID number with 2 channel which means two person could be on the same line? |
| 08:20.35 | kaldemar | it will go from USA to china, the rest is up to you. |
| 08:21.23 | kaldemar | lkthomas: read the lower part of the page. |
| 08:21.56 | lkthomas | so 2 simultanious call means two person could call and answer at the same time |
| 08:22.05 | lkthomas | extra channel cost a lot |
| 08:22.31 | kaldemar | two simultaneous calls between the provider and what you use to connect to them. |
| 08:24.10 | *** join/#asterisk bratner (~bratner@95.211.21.37) |
| 08:24.30 | lkthomas | kaldemar: are we talking about same number ? |
| 08:25.55 | bratner | hi all! i have an incoming sip call and i want to dial back through the same SIP/channel<num> it came from. is there a dialplan variable that might help me? |
| 08:26.03 | lkthomas | kaldemar: how could I test my pbx if I got a US DID ? |
| 08:26.08 | lkthomas | I am not in USA :P |
| 08:30.18 | kaldemar | it doesn't matter where you are. |
| 08:31.33 | lkthomas | I just want to test pbx |
| 08:31.55 | lkthomas | I call will charge long distance fee |
| 08:31.56 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 08:32.13 | kaldemar | bratner: func CHANNEL has a name field that will output the current channel. either parse ${CHANNEL(name)} or use SIP/${CHANNEL(peername)} or what suits you best. |
| 08:33.13 | bratner | kaldemar, thanks! |
| 08:33.16 | kaldemar | lkthomas: you don't need a DID to test your PBX. use a voip phone to dial in it. if you want to test it with a DID, just connect to an ITSP with your asterisk. |
| 08:33.39 | lkthomas | ITSP ? |
| 08:34.15 | kaldemar | internet telephony service provider |
| 08:34.28 | lkthomas | such as ? |
| 08:34.29 | kaldemar | see http://ofps.oreilly.com/titles/9780596517342/asterisk-OutsideConn.html#OutsideConnectivity_id291235 and especially "Connecting an Asterisk system to a SIP provider" |
| 08:34.38 | kaldemar | ~itsp-list |
| 08:34.38 | infobot | itsp-list is, like, Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 08:36.19 | lkthomas | all are not free , haha |
| 08:37.42 | tamiel | lkthomas: maybe try with IPKALL |
| 08:38.09 | kaldemar | lkthomas: you can't expect everything to be free. |
| 08:38.15 | lkthomas | LOL |
| 08:38.46 | lkthomas | kaldemar: one question, does anyone put their mobile phone sim act like a trunk ? |
| 08:39.33 | lkthomas | http://www.voip-info.org/wiki/view/Asterisk+Connecting+to+the+Cellular+Network |
| 08:40.14 | lkthomas | http://fonality.com/trixbox/node/24977 |
| 08:40.15 | lkthomas | LOL |
| 08:40.18 | lkthomas | it's possible! |
| 08:41.17 | kaldemar | yes, by chan_mobile with a phone through bluetooth or chan_datacard with a USB-connected modem. |
| 08:41.33 | lkthomas | is it mature now? that post was couple years ago |
| 08:44.49 | kaldemar | don't know about maturity. i've tried chan_mobile briefly and it seemed to work. |
| 08:45.18 | lkthomas | cool! did you ask cellular phone provider to do special configuration on this or just normal sim card would work ? |
| 08:48.57 | kaldemar | nothing special was needed, just used my cell phone. |
| 08:49.04 | lkthomas | heh, ok |
| 08:50.49 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:52.17 | lkthomas | any max participants on conference limit on asterisk ? |
| 08:53.36 | *** join/#asterisk Nasga (~Nasga@85.212.10.93.rev.sfr.net) |
| 08:56.34 | kaldemar | lkthomas: nothing programmed in. resources determine that. |
| 08:56.48 | lkthomas | ok |
| 08:57.15 | lkthomas | I start to love asterisk |
| 08:57.27 | lkthomas | first time I am able to setup my phone system in my life, haha |
| 08:59.35 | lkthomas | what is a call routing means ?! |
| 09:00.27 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 09:07.14 | Nasga | lkthomas: like networking routing, you receive a call and you can forward it in différents dialplan/gateways... |
| 09:07.47 | *** join/#asterisk banditti (~banditti@unaffiliated/banditti) |
| 09:08.24 | banditti | are there any tricks to asterisk and a polycom IP4000? I have been pissing with this turn for a day and a half |
| 09:11.52 | banditti | nevermind, just got it! |
| 09:18.10 | *** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45) |
| 09:18.31 | *** join/#asterisk CoderForLife (~Miranda@unaffiliated/coderforlife) |
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| 09:48.01 | *** join/#asterisk Cadey (~x@62.84.178.106) |
| 09:48.26 | Cadey | Hi guys, anyone in the UK intrested in buying some used Aastra 57i's ? we have 65 potentialy being sold soon |
| 10:01.57 | *** join/#asterisk bjhaid (~abejide@41-147.rv.ipnxtelecoms.com) |
| 10:13.29 | Cobadol | Do anyone know when could be next asterisk release? |
| 10:13.50 | irroot | Cobadol 1.8 or 1.10 ?? |
| 10:14.01 | Cobadol | 1.8.6 |
| 10:14.02 | irroot | 1.8.6 is in "sprint" |
| 10:14.38 | Cobadol | What does it mean "sprint"? |
| 10:14.59 | kaldemar | RC of 1.8.6 comes next week, maybe. |
| 10:15.01 | irroot | merging of outstanding patches and fixes |
| 10:15.09 | irroot | maybe not :P |
| 10:15.25 | jacc0 | irroot: is my patch going to be in there (the app_originate one) |
| 10:15.26 | irroot | Cobadol why you ask |
| 10:15.56 | irroot | jacc0 possible what is the ref on it again |
| 10:16.34 | Cobadol | )) ok. thanks. Wait my patch (closed issue) |
| 10:16.51 | jacc0 | https://issues.asterisk.org/jira/browse/ASTERISK-17015 |
| 10:19.23 | irroot | jacc0 yeah needs a "fixed" patch |
| 10:19.36 | irroot | once the change to sscanf is made should be good to go |
| 10:19.59 | irroot | but not for 1.8 as it introduces a new "feature" |
| 10:20.16 | irroot | with 1.10 close 1.8 is getting more frozen |
| 10:24.51 | irroot | Cobadol can use SVN ?? checkout /branches/1.8 |
| 10:25.23 | *** join/#asterisk Polysics (~luca@host210-142-static.228-95-b.business.telecomitalia.it) |
| 10:25.25 | Polysics | hello |
| 10:25.52 | Polysics | i have a freshly installed machine where when i connect using asterisk -rvvvvvv i basically do not see anything in the console |
| 10:25.55 | Polysics | asterisk 1.8 |
| 10:26.05 | Polysics | my first 1.8 install, does it act different? |
| 10:26.24 | Polysics | i have no links and need to start it through /usr/sbin/asterisk but that is probably correct |
| 10:28.10 | Polysics | might it be that this asterisk is running as root? |
| 10:28.54 | jacc0 | type: core show uptime |
| 10:28.56 | *** join/#asterisk gravin (~gravin@175.139.236.219) |
| 10:29.07 | jacc0 | to see how long it has been running |
| 10:29.41 | jacc0 | if there are no incomming calls you basicly see nothing in the CLI |
| 10:30.31 | jacc0 | what does "basicly nothing" meen? what do you see if you do : asterisk -rvvvvvv |
| 10:30.36 | jacc0 | (pastebin it pls) |
| 10:30.56 | Polysics | jacc0, nothing at all, nothing to pastebin |
| 10:31.04 | Polysics | if i write commands they do show though |
| 10:31.11 | Polysics | such as sip show peers, it does show them |
| 10:31.41 | jacc0 | okay, then it is running |
| 10:31.41 | *** join/#asterisk bjhaid (~abejide@41-147.rv.ipnxtelecoms.com) |
| 10:32.12 | jacc0 | afet running asterisk -rvvvvvvvvvvvvvvvvvvv you should see something like : Asterisk 1.8.5.0, Copyright (C) 1999 - 2011 Digium, Inc. and others. |
| 10:32.19 | jacc0 | at the first line |
| 10:32.21 | Polysics | i do calls, echo tests, everything |
| 10:32.55 | Polysics | at verbosity 46 i still do not see anything |
| 10:32.57 | jacc0 | you don't even see a prompt? |
| 10:33.07 | Polysics | just the prompt |
| 10:33.19 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:33.21 | Polysics | what's fun is that is i enable sip debug , which normally swamps me, i get no output |
| 10:33.41 | jacc0 | use "tshark -R sip" if there is any incomming sip trafic at alll |
| 10:33.59 | Polysics | calls work |
| 10:34.21 | jacc0 | maybe the calls you make are going to some other server? you could be mistaking ; try tshark |
| 10:34.24 | Polysics | so there has to be SIP traffic |
| 10:34.51 | jacc0 | assumtion is the mother of.... |
| 10:36.08 | Cobadol | irroot: yes. changes are there. I not yet familar with opensource code approving. :-[ |
| 10:36.48 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 10:37.40 | irroot | Cobadol ah ok you using source build or package ?? it takes a while after the final release for packages to be put up by maintainers |
| 10:39.25 | *** join/#asterisk aiksa[LV] (~aiksaLV]@212.70.182.16) |
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| 11:20.38 | *** join/#asterisk Dovid (~Dovid@ool-4355c88f.dyn.optonline.net) |
| 11:21.09 | Dovid | hi. is there any way to have asterisk liste on two IP's ? i have 4 IP's on the box but want asterisk to only listen to 2 |
| 11:21.32 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 11:21.56 | Dovid | hello irroot |
| 11:22.11 | irroot | yo there connex problems |
| 11:22.25 | leifmadsen | Dovid: no |
| 11:22.38 | leifmadsen | Dovid: it's either listen on all 4 interfaces, or 1 interface |
| 11:22.53 | leifmadsen | although you could listen with UDP on one interface and TCP on another |
| 11:23.33 | *** join/#asterisk gravin (~gravin@175.139.236.219) |
| 11:23.38 | Dovid | :(. |
| 11:23.54 | Dovid | I have OpenSIPS on the same box. i need OpenSipS to take 2 and then Asterisk to take another 2 |
| 11:23.58 | *** part/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 11:25.22 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 11:25.46 | leifmadsen | Dovid: then you will need to modify the source of Asterisk |
| 11:27.31 | Dovid | leifmadsen: Any pointers on how to do it? |
| 11:27.40 | leifmadsen | no, because it's not possible without changing the source |
| 11:27.50 | leifmadsen | (and I have no idea what to change in the source) |
| 11:28.17 | Dovid | ok. thanks a lot |
| 11:29.11 | kaldemar | try redirecting incoming traffic to ports 5060 to different interfaces to some other ports that asterisk and opensips listen to. |
| 11:31.18 | Cobadol | irroot: package. with 'yum' updating (asteriskNOW) |
| 11:32.17 | wasanzy | kaldemar: I did the test once again and this time too. it didn't work. the tcpdump: http://pastebin.com/s8RBUZqJ |
| 11:47.50 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 11:51.22 | *** join/#asterisk Eye_Trust_No1 (~Eye_Trust@84.203.229.194) |
| 11:51.34 | aberrios | is "core show locks" removed in 1.8? |
| 11:51.45 | *** join/#asterisk Eye_Trust_No1 (~Eye_Trust@84.203.229.194) |
| 11:51.46 | aberrios | trying to find a reference |
| 11:52.53 | aberrios | ah DEBUG_THREADS.... |
| 11:56.10 | Cobadol | irroot: Now I think that if I had a free time, its better for me to solve problem with substituting module from virtual machine to real, than wait for package release ) thanks |
| 11:59.10 | *** join/#asterisk kilgorex (~rob@82-69-63-23.dsl.in-addr.zen.co.uk) |
| 12:11.35 | *** join/#asterisk jstapleton (~jstapleto@c-24-125-171-223.hsd1.va.comcast.net) |
| 12:14.32 | Cadey | anyone interested in some Aastra 57i hand sets? |
| 12:22.47 | *** join/#asterisk euphor][a (~mallchin@mail.dataproservices.co.uk) |
| 12:22.56 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:23.27 | chuckf | for free? |
| 12:23.30 | euphor][a | hi guys, having trouble starting zaptel, I get "ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25)" |
| 12:23.36 | euphor][a | tried googling but not found anything useful |
| 12:23.47 | euphor][a | any ideas please? :) |
| 12:35.33 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 12:37.05 | *** join/#asterisk wasanzy (~emmanuel@196.201.43.55) |
| 12:37.45 | *** join/#asterisk TimeRider (~steve@host-92-27-131-175.static.as13285.net) |
| 12:38.02 | Cadey | chuckf : no lol not for free :) |
| 12:38.18 | Cadey | chuckf : we are replacing 65 of them potentialy |
| 12:38.46 | kaii | euphor][a: your config is wrong ;-) |
| 12:42.05 | Cadey | Humm quick question. I have set the language in general to en_gb so we heard the sounds with an english accent, however we receive all of our incomming calls over an open sip tunnel which isnt defined in sip.conf which means our custeomrs are hearing the default american accent |
| 12:42.20 | Cadey | how can we change what sounds are the default with out over ridding the actual sound files? |
| 12:42.57 | Cadey | voicemail sounds |
| 12:44.01 | wasanzy | Kaldemar: are you there? |
| 12:45.42 | kaldemar | wasanzy: yes |
| 12:46.10 | wasanzy | have you looked at my latest tcpdump? |
| 12:46.22 | kaldemar | nothing new there. |
| 12:46.28 | wasanzy | is the same thing though. I am right now confuse |
| 12:47.20 | wasanzy | am going to direct the audios through the asterisk when am done installing my virtualbox |
| 12:47.27 | Cobadol | Cadey: and where this incoming sip tunnel defined? |
| 12:48.50 | Cadey | its not defined Coladol |
| 12:49.01 | Cadey | they simply point calls to our IP and we accept them |
| 12:49.21 | Cadey | its not a peer you see |
| 12:50.57 | kaldemar | is something preventing you from making a peer that matches those calls? |
| 12:52.35 | Cadey | yeah, its how they do it as its a wholesale agreement |
| 12:54.57 | Cadey | may be its a flag we can send to the Voicemail app in the dial plan |
| 12:57.06 | leifmadsen | Set(CHANNEL(language)=en_gb) |
| 12:57.13 | leifmadsen | Voicemail(1234@foo,u) |
| 12:57.55 | Cadey | arr harrr |
| 12:57.57 | Cadey | thanks leig |
| 12:58.07 | wasanzy | Kaldemar: Am trying to configure one of the account on a machine which is not in the same network with asterisk and for matter, a different ISP is providing Internet for that machine, and now, I am getting service unavailable or if I use the public IP, I get request timed out. How can I deal with this? |
| 12:58.35 | wasanzy | I already done port forwarding in the router pointing to the asterisk server |
| 12:58.50 | kaldemar | ~sipnat |
| 12:58.50 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 13:06.16 | aberrios | when changing iaxmaxthreadcount and iaxthreadcount, after a reload cli says its ingoring the new settings. Is the change only applied after a restart? |
| 13:06.22 | *** join/#asterisk jstapleton (~jstapleto@173-15-197-75-BusName-Richmond.hfc.comcastbusiness.net) |
| 13:07.39 | leifmadsen | aberrios: sounds like it -- module unload chan_iax2.so then module load chan_iax2.so |
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| 13:22.48 | puzzled | hi |
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| 13:39.06 | euphor][a | kaii: config looks sane, have auto-generated one too and that didn't work -- could I post config to channel? :) |
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| 14:15.22 | asteriskATmarmuD | what would be the easiest way to connect a inhouse sip phone to an external number from "outside" (via some kind of script)? |
| 14:16.38 | *** join/#asterisk ihor (~Miranda@194.44.15.90) |
| 14:24.56 | JonathanRose | I take it a call file doesn't meet your fancy. |
| 14:25.40 | asteriskATmarmuD | no, moving a call file isn't what I am looking for ;) |
| 14:26.23 | WIMPy | AMIT |
| 14:26.26 | WIMPy | oops |
| 14:26.31 | WIMPy | AMI |
| 14:26.33 | asteriskATmarmuD | more an AGI script with parameter, e.g. 1. parameter = internal sip-phone to call, 2. parameter external phone to call... |
| 14:26.35 | WIMPy | asterisk -rx |
| 14:26.44 | JonathanRose | I would have thought AMI as well, but I'm not sure what the problem is exactly. |
| 14:26.57 | asteriskATmarmuD | ok, gotta look for the difference of AGI and AMI (Asterisk manager interface) as far as I remember |
| 14:26.58 | file | an AGI script requires a call to be in progress |
| 14:27.02 | WIMPy | AGIs are calld from the dialplan i.e. on an already existing call. |
| 14:27.19 | asteriskATmarmuD | a of course, forgot that |
| 14:27.31 | asteriskATmarmuD | using a lot of AGI in the dialplan.. ok, so AMI |
| 14:28.39 | asteriskATmarmuD | ok, how to call an internal sip phone and connect that to a given number... we want to trigger this process from another server |
| 14:29.22 | JonathanRose | Well, you could originate the channel to an extension that calls the number. |
| 14:29.30 | irroot | file AGI can be run from h exten after the call has ended ... |
| 14:29.34 | JonathanRose | I'm not sure if that would do it at the same time or not though. |
| 14:29.42 | file | irroot, the call is still active within Asterisk |
| 14:29.48 | file | a channel still exists |
| 14:30.52 | JonathanRose | But yeah, you could combine originate with the dialplan to make the call to the first phone, then once it picks up, you could dial out to the number you want. |
| 14:31.55 | JonathanRose | And AMI works remotely as long as you can connect to the server you set it up on at the port you set it to use. |
| 14:32.04 | WIMPy | is not sure, what the exact scenatio is like. |
| 14:32.29 | JonathanRose | I'm guessing he wants to call an inside line from Asterisk and then once inside line picks up, he wants Asterisk to connect it to something outside. |
| 14:32.41 | asteriskATmarmuD | ok, I think it would be great not to use the dial plan. AMI would be great. any examples (online) on how to do what I mentioned |
| 14:32.48 | asteriskATmarmuD | JonathanRose: exactly |
| 14:33.34 | JonathanRose | Why don't you want to use the dialplan at all? You can make it totally distinct from the rest of the dialplan just by throwing it in its own context and using that context as one of your arguments for originate. |
| 14:33.45 | WIMPy | Once it picks up? I.e. put the caller, the callee and an external to a conference? |
| 14:34.08 | JonathanRose | I don't think he mentioned anything about conferences. |
| 14:34.26 | asteriskATmarmuD | WIMPy: yes, meetme room would be nice |
| 14:34.38 | WIMPy | I know, but The "picks up" patr doesn't make sense to me. |
| 14:34.43 | WIMPy | Ok |
| 14:34.59 | WIMPy | You can do that entirely from the dialplan, I think. |
| 14:35.33 | WIMPy | But AMI might be easier, if you can do some socket communication. |
| 14:35.39 | asteriskATmarmuD | WIMPy: meetme is not needed, can't use the dialplan, since this should be triggered from the outside (another server in our network) |
| 14:35.52 | asteriskATmarmuD | WIMPy: sockets are great :) |
| 14:36.10 | JonathanRose | The fact that it's triggered from outside the network doesn't mean you can't use the dialplan at all. |
| 14:36.18 | WIMPy | Ok, back to the original question: What's the exact scenatio? |
| 14:36.31 | WIMPy | Now I thought the internal phone being picked up is the trigger. |
| 14:36.55 | WIMPy | is being picked up in a few minutes as well, BTW. |
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| 14:39.04 | c4rg | yo, does anyone know what REASON channel variable = 0 means, when an outgoing call is generated by placing a file in /var/spool/asterisk/outgoing? |
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| 15:04.38 | ssureshot | clarify this statement.... .Incoming Telephone Line or PABX Station is an FXO device ? |
| 15:05.01 | ssureshot | err that sounds wrong... lol let me fix it |
| 15:05.18 | Chainsaw | ssureshot: I was about to say, that sets off the ambiguous input alarms everywhere. |
| 15:06.49 | ssureshot | the manual for my pa system States this for the phone port "Incoming Telephone Line or PABX Station" |
| 15:07.06 | ssureshot | so I believe that is an FXO |
| 15:09.50 | ssureshot | Basically I need to know what typ of device this is,, I believe its an fxo device... http://www.google.com/url?sa=t&source=web&cd=1&ved=0CDQQFjAA&url=http%3A%2F%2Fwww.valcom.com%2Fpdf%2Fv-9970.pdf&rct=j&q=valcom%20v-9970&ei=iu8mTseZK8SBgAfioMxc&usg=AFQjCNHlYDerHv6etS7-KnYHQvYw_kPOcQ&sig2=hUKpaXDztzqveoYQlFWTnQ&cad=rja ... |
| 15:10.14 | ssureshot | last two pages show the phone port |
| 15:12.23 | asteriskATmarmuD | thanks for your help guys, had to leave the desk... will try my best and come back for help if needed :) |
| 15:13.12 | _Corey_ | ssureshot: Looks like that's designed to be plugged into an FXS port |
| 15:23.04 | *** join/#asterisk moy_ (~moy@173.239.130.12) |
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| 15:30.33 | malcolmd | agree. that's an fxo device, it accepts ring voltage. |
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| 15:35.28 | spck | Does anyone know of any tools/solutions to make provisioning phones easier? |
| 15:35.48 | valera | spck: dhcp :) |
| 15:36.43 | Chainsaw | spck: I use a PHP script (my Polycoms provision over HTTP). |
| 15:37.00 | valera | or just curl |
| 15:37.05 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 15:37.30 | raden | what would be the difference between a IP phone and SIP phone ? aastra has them classified seperately on there website |
| 15:38.03 | malcolmd | maybe one doesn't support SIP (MGCP, H.323?) while the other one does? |
| 15:39.48 | valera | raden: whats in the specs ? short diff |
| 15:40.04 | ssureshot | _Corey_: got it,, then guess Ill make my purchase and cross my fingers lol |
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| 15:40.36 | _Corey_ | ssureshot: Valcom and Bogen usually have pre-sales engineering support, so if you're still unsure you can probably call and ask |
| 15:42.00 | ssureshot | ah good call, thank you |
| 15:42.25 | spck | i get the whole provisioning thing myself but i want to make easier for those working the help desk to use |
| 15:48.31 | *** join/#asterisk TheJA (~johnnyast@89.204.168.10) |
| 15:49.10 | MrTelephone | Some phones use IP but have a proprietary communication protocol or use MGCP/H323 only |
| 15:49.22 | MrTelephone | If cisco had their way SIP wouldn't even exist anymore :( |
| 15:53.39 | *** join/#asterisk b0ot (~Jinxed---@147.177.56.63) |
| 15:53.53 | b0ot | in the voicemail.conf do you specify the number or username |
| 15:54.12 | b0ot | for the format |
| 15:55.12 | wasanzy | Kaldemar: are you there please? |
| 15:57.16 | wasanzy | I set directmedia to no and got this error when calling from my vitualbox: http://pastebin.com/7KtXuWW6 |
| 15:57.40 | wasanzy | that is actually my rtp debug on the asterisk server |
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| 16:12.32 | wasanzy | Kaldemar: are you there please? |
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| 16:41.12 | wasanzy | hello |
| 16:41.45 | ChannelZ | OH HAI! |
| 16:41.54 | wasanzy | what is this error about in rtp debug? http://pastebin.com/7KtXuWW6 |
| 16:42.09 | wasanzy | because my sound is still not working |
| 16:42.38 | ChannelZ | well your SIP probably isn't either |
| 16:43.13 | ChannelZ | It basically means it got no response to a message.. either because the response didn't make it back, or the original request never got to the other end in the first place |
| 16:43.54 | MrTelephone | When you use asterisk realtime do the peers only show up 'sip show users/peers' when they register? |
| 16:44.40 | *** join/#asterisk johnnyasterisk (~johnnyast@smtp.ardmore-hotel.com) |
| 16:44.41 | billmania | In asterisk 1.8.4.2, is it a separate module which parses logger.conf and writes to the queue_log file or is it part of the "main" functionality? |
| 16:46.08 | wasanzy | please some body help me |
| 16:47.30 | ChannelZ | wasanzy: you have networking problems |
| 16:48.14 | ChannelZ | billmania: logging in general I believe is in core |
| 16:48.24 | ChannelZ | billmania: what is your actual problem? |
| 16:48.53 | billmania | The file /var/log/asterisk/queue_log never gets created and there is no entry in /var/log/asterisk/verbose stating "parsing logger.conf". |
| 16:49.09 | billmania | When the dialplan calls QueueLog(), there is no complaint however. |
| 16:50.27 | *** join/#asterisk Ehsanfzali (5b62a1ad@gateway/web/freenode/ip.91.98.161.173) |
| 16:50.45 | Ehsanfzali | Hi there, |
| 16:51.00 | Ehsanfzali | Anyone used any voicexml browser with asterisk? |
| 16:51.12 | *** join/#asterisk Nasga (~Nasga@85.212.10.93.rev.sfr.net) |
| 16:51.23 | Ehsanfzali | I want to know which one should I use? Voxy, Voiceglue or VXI* |
| 16:52.26 | irroot | billmania queue_log is in app_queue and can be realtime ... |
| 16:53.14 | billmania | irroot: Is app_queue module still able to write to the file queue_log? |
| 16:53.41 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053179.ics.uci.edu) |
| 16:53.43 | ChannelZ | billmania: make sure /var/log/asterisk is accessable by whatever user your asterisk runs as |
| 16:53.55 | irroot | ast_queue_log <- the heart is in main/logger |
| 16:54.13 | billmania | ChannelZ: Way ahead of you. Permissions are wide open on that directory. |
| 16:54.46 | ChannelZ | what does 'logger show channels' tell you? |
| 16:55.40 | billmania | Looking … |
| 16:56.39 | Ehsanfzali | Hello!!! no one knows about VoiceXML support in asterisk? |
| 16:56.59 | ChannelZ | Ehsanfzali: Apparently no one whose currently listening, no |
| 16:57.47 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 16:58.31 | irroot | billmania in the general section of logger.conf set queue_log and queue_log_to_file and queue_log_name |
| 16:58.37 | billmania | ChannelZ: The five files in /var/log/asterisk: debug, fax, dtmf, verbose and full. |
| 16:58.41 | ChannelZ | and wasanzy's network problems grow |
| 16:58.59 | irroot | wasanzy has a notwork |
| 16:59.05 | ChannelZ | heh indeed |
| 16:59.58 | ChannelZ | billmania: assuming queue_log and queue_log_name is not commented out and/or set incorrectly such that they're not getting parsed, not sure. |
| 17:00.03 | *** join/#asterisk wasanzy (~emmanuel@196.201.43.55) |
| 17:00.13 | irroot | ChannelZ got contracted to fix a botched job about 100km away the installer created a Vlan with a duplicate ip/range of the default network .... |
| 17:00.25 | ChannelZ | oops |
| 17:00.58 | ChannelZ | Where do I route these packets? Wherever! |
| 17:01.01 | billmania | irroot: My /etc/asterisk/logger.conf has, in the [general] section: |
| 17:01.01 | billmania | queue_log = yes |
| 17:01.02 | billmania | queue_log_to_file = yes |
| 17:01.02 | billmania | queue_log_name = queue_log |
| 17:01.26 | billmania | But I have no indication that file (logger.conf) is even being parsed by asterisk. There's no mention of it in /var/log/asterisk/verbose. |
| 17:01.32 | irroot | and /etc/asterisk/extconfig.conf |
| 17:01.47 | ChannelZ | well then I dunno where the rest of your logs are coming from :) |
| 17:02.08 | ChannelZ | But yes as irroot says make sure that's not realtime for some reason in extconfig |
| 17:02.09 | billmania | ChannelZ: That makes two of us. |
| 17:02.19 | wasanzy | ChannelZ: the three machines are in the same network, but one of them is virtualbox as you can see in the log. I did port forwarding and even set the nat parameters in the sip.conf, so what could be the usual problem with this kind of error? |
| 17:02.27 | billmania | First things first: should there be mention in /var/log/asterisk/verbose about logger.conf being parsed? |
| 17:02.43 | ChannelZ | there is on the console, not sure in the disk log |
| 17:02.54 | ChannelZ | (IE if you just do 'reload', it's pretty much the first thing said for me..) |
| 17:03.17 | *** join/#asterisk Stormcrow (~Phydeaux@96.57.40.218) |
| 17:03.33 | Stormcrow | Is there a way to cancel a "restart when convenient" or check to see if one is in place? |
| 17:03.43 | billmania | ChannelZ: I'll give that a try and see what I see. |
| 17:04.20 | irroot | Stormcrow core show threads should show a thread |
| 17:04.49 | ChannelZ | billmania: I just turned on my verbose log and no I don't see it. BUt it depends on what your core asterisk process was run with |
| 17:05.21 | ChannelZ | oh wait nevermind |
| 17:05.30 | ChannelZ | I do see it in there for me |
| 17:05.33 | irroot | ChannelZ i suspect with proper crafting with iptables and mark based routing on multiple route tables one could run a vlan and lan on same ip/net but why |
| 17:05.34 | Stormcrow | irroot: I see (Sorry. I had a live asterisk server dumped on me 10 minutes ago when I don't know the first thing). Is the thread I'm looking for obviously named? |
| 17:06.11 | irroot | stormcrow may be cant think of it now |
| 17:06.29 | Stormcrow | It's not sched_thread, is it? |
| 17:07.02 | wasanzy | guys please help me, I know am disturbing but I need your help |
| 17:07.05 | ChannelZ | though I'm not sure what verbose level those come out as, but I typically run vvv |
| 17:07.18 | ChannelZ | wasanzy: your network is screwed up |
| 17:07.54 | irroot | Stormcrow just tried it does not look like it |
| 17:08.23 | irroot | 0xac661b70 cleanup started at [ 417] pbx_realtime.c load_module() |
| 17:08.29 | irroot | might be it |
| 17:08.49 | wasanzy | now when I made the call with a different machine not the virtualbox, the error didn't appear again and the sound went through a little, but I can't hear the other partner whiles he could heard me |
| 17:09.02 | Stormcrow | That's not on my list. -nod- Odd, really. I was pretty sure I gave the command. |
| 17:09.26 | irroot | nope its there after restart too |
| 17:09.36 | ChannelZ | wasanzy: you have network or firewall/nat problems |
| 17:10.08 | wasanzy | let me narrate the network a little |
| 17:10.19 | Stormcrow | Thanks, though. The situation is that I gave it that command to try to fix a problem, and now I'm hearing our best practices is to schedule any restart overnight, just in case. Makes sense. Of course, I already gave it that command, so... :D Not a big deal, but still. |
| 17:10.23 | irroot | Stormcrow sorry it does not look like seperate thread thougt it was |
| 17:11.01 | ChannelZ | with directmedia=no, you have 4 different network connections that all need to work, and that's just from Asterisk's point of view |
| 17:12.16 | wasanzy | we have one public IP from the ISP, and that IP is configured in a router. this router is also connected to another wireless router which assign lan IP automatically. now the three machine, asterisk and the two phones are in the same network, |
| 17:12.33 | Stormcrow | There's no chance that 'abort halt' would kill it, I'm guessing. I'd rather not enter random commands. Sorrow. I thought I was doing the right thing, here. :) |
| 17:12.50 | Stormcrow | (And I probably DID do the right thing, but policy is policy) |
| 17:12.52 | ChannelZ | wasanzy: the same LAN network? |
| 17:12.57 | irroot | <PROTECTED> |
| 17:12.59 | wasanzy | yes |
| 17:13.21 | ChannelZ | And what are the 2 phones running? |
| 17:13.31 | wasanzy | 192.168.1.0/24 |
| 17:13.45 | ChannelZ | Are they actual devices or softphones? |
| 17:13.47 | irroot | Stormcrow you do know murphy will do it when the CEO wants to call the CTO |
| 17:13.58 | wasanzy | they are running twinkle |
| 17:14.10 | Stormcrow | Yes, I do. :) |
| 17:14.19 | Stormcrow | And nevermind... I just saw in the console that it restarted. :D |
| 17:14.27 | *** join/#asterisk johnnyasterisk (~johnnyast@smtp.ardmore-hotel.com) |
| 17:14.31 | ChannelZ | and no firewalls on them? |
| 17:14.34 | ChannelZ | or Asterisk? |
| 17:14.44 | wasanzy | no |
| 17:15.16 | ChannelZ | And have you gotten one of them to work just between it and Asterisk? IE make a test extension that plays a sound, and does an Echo test |
| 17:15.34 | billmania | ChannelZ: irroot: I executed "logger reload" via the CLI and I DID get a message about '/etc/asterisk/logger.conf' being parsed. It also stated that the "Asterisk Queue Logger restarted". |
| 17:15.39 | wasanzy | how do I do that? |
| 17:15.44 | *** join/#asterisk elfelvin (~elfelvin@87-194-69-88.bethere.co.uk) |
| 17:16.03 | billmania | However, there is no Channel for queue_log in the output of "logger show channels". |
| 17:16.04 | *** join/#asterisk johnnyasterisk (~johnnyast@smtp.ardmore-hotel.com) |
| 17:16.10 | ChannelZ | wasanzy: uhm.. make an extension in extensions.conf that does a Playback and an Echo..... |
| 17:16.35 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 17:17.01 | ChannelZ | Answer, Playback(hello-world), Echo |
| 17:17.02 | wasanzy | when we made the call, one person could hear his partner talk he doesn't hear a respond back |
| 17:17.08 | wasanzy | oh ok |
| 17:17.17 | wasanzy | let me do that and see |
| 17:17.41 | wasanzy | but here is my sip.conf: http://pastebin.com/9g2wSeLx |
| 17:17.54 | irroot | wasanzy also use monitor b4 any without b option |
| 17:18.10 | irroot | so you can listen to the servers perspective |
| 17:19.01 | wasanzy | monitor in sip.conf? |
| 17:19.25 | irroot | yeah as a line before playing hello-world |
| 17:20.16 | ChannelZ | Echo will tell you what you need to know |
| 17:20.25 | wasanzy | ok |
| 17:20.53 | ChannelZ | If you hear hello world, we know Asterisk can get sound to you. If you can then hear yourself in the echo, we know your audio can get to Asterisk. |
| 17:21.54 | wasanzy | is it some thing like this? |
| 17:22.04 | wasanzy | http://pastebin.com/dQtEXK9X |
| 17:22.35 | ChannelZ | well that's the general syntax if that's what you're asking |
| 17:22.53 | ChannelZ | but not really what I said |
| 17:23.12 | irroot | ~beer channelz |
| 17:23.13 | infobot | ACTION pulls out a excellent Piraat for channelz |
| 17:23.51 | wasanzy | am really new, so if you can some thing sample in pastebin for me, I will appreciate it |
| 17:24.02 | ChannelZ | http://pastebin.com/WB8bcCP9 |
| 17:24.54 | ChannelZ | assuming it's in the right context you should then be able to dial 555 (after reloading the dialplan) |
| 17:25.35 | wasanzy | don't I have to add it to the sip.conf as well? |
| 17:25.37 | ChannelZ | which according to your last pastes, 'emma' and 'elarty' are in the 'phones' context so that's where they aught go |
| 17:25.42 | ChannelZ | no |
| 17:25.47 | Stormcrow | irroot: Regardless of everything else, thanks for the assistance. :) |
| 17:26.07 | wasanzy | ok |
| 17:26.27 | ChannelZ | hmm does voicemail just call the 'sox' binary or is it compiled in? |
| 17:26.35 | wasanzy | so emma can dial 155 right? |
| 17:26.38 | irroot | Stormcrow all good wondered for myself there ... never thought about it |
| 17:26.48 | ChannelZ | 555 |
| 17:27.11 | wasanzy | oh sorry, wrong typing, ok |
| 17:29.25 | ChannelZ | oh, barf. I had libsox installed but not the actual CLI sox. Oops. |
| 17:30.36 | wasanzy | now my extension.conf: http://pastebin.com/7uRZeN7W |
| 17:31.09 | *** join/#asterisk johnnyasterisk (~johnnyast@89.204.225.16) |
| 17:31.19 | ChannelZ | ok well you have a problem but not with this test so we'll ignore it for now |
| 17:31.43 | *** join/#asterisk moy_ (~moy@209.250.158.226.tor.pathcom.com) |
| 17:32.01 | wasanzy | ok let me make the call and see |
| 17:32.21 | billmania | ChannelZ: irroot: OK, I don't understand. After executing "logger reload" I now have the queue_log file and all appears to be working. It's still not listed in "logger show channels" however. |
| 17:32.26 | irroot | dialplan reload first if you have not |
| 17:32.35 | billmania | I'm going to stop and restart asterisk to see if this is just a fluke. |
| 17:32.54 | wasanzy | oh sure thanks |
| 17:33.29 | ChannelZ | billmania: did you ever check extconfig.conf and make sure you don't have the queue log in a database? (I don't use realtime OR queues so I don't know if that would or wouldn't show up in the logger show config...stabbing in the dark here) |
| 17:34.25 | ChannelZ | oh nevermind |
| 17:34.37 | billmania | ChannelZ: I did. My entire extconfig.conf is commented out. Only the [settings] line isn't commented. |
| 17:34.40 | ChannelZ | you said you DO have the queue log file, my bad |
| 17:35.11 | wasanzy | wow the test worked |
| 17:35.26 | billmania | The only thing I'm aware of having done differently this time is the "logger reload". I had never done that before. |
| 17:35.31 | ChannelZ | wasanzy: ok so you heard hello world and could year yourself echoed back? |
| 17:35.48 | wasanzy | I heard some sweet voice saying hello world |
| 17:36.03 | ChannelZ | then did you talk? |
| 17:36.12 | wasanzy | no I didn't talk |
| 17:36.25 | ChannelZ | well do talk. You should hear your audio come back slightly delayed |
| 17:36.35 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 17:36.42 | wasanzy | the machine is far from the asterisk so am going for it |
| 17:37.35 | kuku | I would like to play an announcement ( only to the caller ) after 15 and 30 seconds from the start of the call. Like a whisper. Any suggestions which command would make that happen ? |
| 17:38.04 | irroot | kuku possibly use app_queue it has that ability |
| 17:38.20 | irroot | mmm maybe not |
| 17:38.42 | irroot | kuku before its answerd ?? |
| 17:39.39 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 17:40.33 | wasanzy | yes I could hear my self talk |
| 17:40.51 | ChannelZ | ok so now do the same thing on the other phone and make sure it works too |
| 17:41.04 | wasanzy | oh ok |
| 17:42.07 | ChannelZ | potty break |
| 17:55.22 | wasanzy | am still waiting for the guy so I can use the machine for the test, will get back when done |
| 17:56.56 | wasanzy | wow, it also worked |
| 17:57.28 | wasanzy | the echo came and I talked back and it worked so what next? |
| 17:58.44 | ChannelZ | but if you use your 1000 or 1002 extensions (depending on which side is which, I don't know) you only get one-way audio? |
| 17:59.12 | wasanzy | that is how it is looking like and am confused |
| 17:59.17 | *** join/#asterisk [netman] (netman@152.252.22.95.dynamic.jazztel.es) |
| 17:59.36 | ChannelZ | Which way? |
| 18:02.56 | wasanzy | I mean one person could only heard the other person but he can no hear him |
| 18:03.45 | wasanzy | is like if I call you, you can hear me talk slowly but I can't hear you at all |
| 18:04.45 | wasanzy | is there any thing wrong in my extension.conf? |
| 18:04.47 | ChannelZ | talk slowly? |
| 18:05.06 | wasanzy | the sound delays I mean |
| 18:05.49 | ChannelZ | with everything setup the way it is now (the echo test was successful on both phones) is it still behaving this way if you call one another? |
| 18:06.16 | wasanzy | let me test it again |
| 18:06.35 | ChannelZ | because what you were posting before does not match this, with the SIP packet timeouts and such |
| 18:07.11 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 18:13.36 | wasanzy | is unfortunate that, the guy is leaving the office so I can't do the test, thank you very much for your great help, and hope to talk to you tomorrow. |
| 18:14.32 | ChannelZ | Hmm. I thought he said they were both on the same LAN... |
| 18:14.55 | ChannelZ | Guess the guy must lock his office up. <shrugs> |
| 18:15.30 | *** join/#asterisk Jcook_5xData (~Jcook_5xD@173.162.32.1) |
| 18:16.12 | *** join/#asterisk johnnyasterisk (~johnnyast@89.101.130.55) |
| 18:16.34 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 18:16.37 | Jcook_5xData | stupid ? when watching the asterisk console. this message forever fill it up meetme.conf': == Found is there a way to stop this |
| 18:19.55 | pigpen | Hi all, I know this isn't relevant, but I have a new asterisk box in prep and after every command, I get a 1- - 15 second pause. /etc/hosts and /etc/resolv.conf are setup correctly. ideas? |
| 18:20.14 | billmania | ChannelZ: irroot: If I restart asterisk, queue_log doesn't work. If I then "logger reload", all is well. Sound like a defect? |
| 18:20.56 | irroot | billmania try look at loading order in modules.conf |
| 18:20.56 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:20.57 | ChannelZ | billmania: it sounds like a module is loading in the wrong order or something but I'm not sure what that would be |
| 18:21.46 | irroot | channelz billmania perhaps preload app_queue |
| 18:22.25 | Jcook_5xData | pigpen, what you commands like ls and ifconfig stuff like that |
| 18:22.29 | WIMPy | pigpen: Is your hostname resolvable? |
| 18:22.29 | billmania | I'll give that a try. |
| 18:22.35 | pigpen | yeah. |
| 18:23.08 | Jcook_5xData | pigpen, I am with WIMPy check you host name |
| 18:23.14 | pigpen | yeah, any command... |
| 18:23.58 | WIMPy | Any is too much, really. |
| 18:25.12 | *** join/#asterisk [ctrl][alt][del] (~Jabber@pdpc/supporter/active/ctrl-alt-del) |
| 18:25.42 | Jcook_5xData | pigpen, check this link seem good http://www.ducea.com/2006/08/07/how-to-change-the-hostname-of-a-linux-system/ |
| 18:27.22 | pigpen | This image is a cookie cutter system, we have done about 20 now, about 400 to go. |
| 18:27.40 | pigpen | I am needing to experiment with some scripts, and naturally the guy (a kern dev) is on vacation. |
| 18:27.46 | Bipul | p3nguin, ping |
| 18:27.50 | pigpen | so I have figured out the image processing, just having this delay. |
| 18:28.06 | Qwell | pigpen: What commands pause? |
| 18:28.29 | pigpen | ls, ps, echo, sed, all. |
| 18:28.37 | pigpen | seems like hostname/reslove. |
| 18:28.54 | Qwell | Do you have the hostname in your prompt or something weird? |
| 18:29.01 | Qwell | echo $PS1 |
| 18:29.04 | pigpen | heh....yeah |
| 18:29.47 | Qwell | w;w |
| 18:29.48 | Qwell | run that |
| 18:29.56 | Qwell | Does it output both at the same time, or is there a delay between them? |
| 18:29.58 | billmania | ChannelZ: irroot: I'm pre-loading app_queue.so in /etc/asterisk/modules.conf. Still have the same issue with queue_log. |
| 18:31.56 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 18:32.00 | ChannelZ | billmania: yeah but logging is in the core and should be started before preloads |
| 18:33.05 | billmania | ChannelZ: Makes sense to me. The only way I can get queue_log to work, at present, is by manually executing "logger reload" after asterisk has started. |
| 18:33.17 | pigpen | Qwell, ok, figured it out. My business partner called. He is setting a custom hostname. |
| 18:33.22 | pigpen | you were right. |
| 18:33.41 | ChannelZ | which is why it doesn't make sense. You might be running into this: https://issues.asterisk.org/jira/browse/ASTERISK-17036 |
| 18:35.58 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 18:37.44 | tuxxie | is it connect to astersk servers via iax2 if one server is running asterisk 1.4 and the other is running asterisk 1.6? |
| 18:37.59 | Qwell | tuxxie: I think you a word there. |
| 18:39.56 | ChannelZ | but if he asked what I think he asked, yes it should be OK |
| 18:40.24 | billmania | ChannelZ: Yes, that sort of looks like my issue with queue_log, except queue_log doesn't work from startup until I "logger reload". I'll try that patch in my copy of the source. |
| 18:41.07 | ChannelZ | Read down all the comments |
| 18:41.39 | ChannelZ | there was someone who said his didn't work until reload as well (though I don't think he ever replied back if the patch solved it for him specifically) |
| 18:42.57 | tuxxie | Qwell: :( i so. |
| 18:43.29 | tuxxie | can i link to asterisk servers via iax2 running different asterisk verisons? |
| 18:43.53 | tuxxie | i.e. asterisk 1.4 and asterisk 1.6 |
| 18:44.23 | Qwell | sure |
| 18:45.06 | Jcook_5xData | leifmadsen, I may have fix the ringing when on a call. under my sip user I had 'busyallow=1' I remove it. So far today it seems good no ring when on call |
| 18:48.17 | sunfone | Looking for a good method of collecting jitter information per peer, for purposes of graphing... anyone got any pointers? |
| 18:49.54 | tuxxie | Qwell: would if you expected high call volume between the servers? |
| 18:50.05 | Qwell | Would I what? |
| 18:50.40 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 18:50.43 | tuxxie | link 2 servers via iax on different asterisk versions |
| 18:51.07 | Qwell | sure, why not? |
| 18:51.27 | ChannelZ | it's designed for it! |
| 18:52.18 | tuxxie | I was just making sure prior to rollowing out my network |
| 18:52.19 | tuxxie | thanks. |
| 18:55.02 | jaytee | there's a parameter you have to set in iax.conf though on the 1.6 server IIRC to work with 1.4 |
| 18:59.36 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 18:59.51 | ChannelZ | requirecalltoken you might be referring to |
| 19:10.18 | jaytee | <PROTECTED> |
| 19:11.49 | jaytee | at least it had to be set on 1.6 to work with 1.4 awhile but there may have been a backport to a later 1.4.x version that made that unneccessary. |
| 19:19.09 | *** join/#asterisk wonderworld (~ww@port-92-201-117-40.dynamic.qsc.de) |
| 19:19.15 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 19:24.04 | *** join/#asterisk jsman (~jsman@unaffiliated/jsman) |
| 19:30.30 | *** join/#asterisk f2Knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 19:30.38 | *** join/#asterisk fullstop (~fullstop@173.210.91.4) |
| 19:30.58 | fullstop | Anyone here built asterisk with clang? How did it go for you? |
| 19:32.00 | f2Knight | Q: Asterisk Queues, I need a way to write information to a database AFTER a person in a queue has been picked up by an agent. Do Channel Variables pass through ? |
| 19:34.28 | raden | anyone have any input on polycom 321's ? |
| 19:35.28 | leifmadsen | it's too bad "busyallow" does't exist |
| 19:35.57 | raden | I need to get away from aastra phones :( |
| 19:36.06 | leifmadsen | I like the Polycom 335's |
| 19:36.19 | leifmadsen | (other than a ringing issue I have with a couple phones at one particular location) |
| 19:40.23 | raden | leifmadsen, none of my aastra phones ( over 300 in offices ) will work with asterisk 1.8.x properly |
| 19:40.28 | raden | no music on hold |
| 19:40.31 | raden | call transfer issues |
| 19:40.35 | leifmadsen | odd |
| 19:40.55 | raden | 9133's 9143's 480i's 57i's |
| 19:41.06 | raden | and aastra just is like we dont know |
| 19:41.30 | raden | all newest firmware , backdated firmware , setup testing server... changed a bazzilion things have over 200 hours on issue |
| 19:41.44 | raden | ended up rolling back our 4 main asterisk servers to 1.6.x |
| 19:42.41 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 19:44.05 | raden | Just wish I understood what was happening :( |
| 19:44.12 | raden | had 3 other people work on the issue as well :( |
| 19:44.51 | raden | Anyone here use Asterisk2Billing ??? |
| 19:47.51 | *** join/#asterisk Nasga (~Nasga@AAmiens-157-1-56-250.w86-196.abo.wanadoo.fr) |
| 19:49.00 | f2Knight | raden, I am glad you said something about the aastra and 1.8... i was just thinking of recommending them for a new server that is 1.8. (no option to go 1.6 as they want to use google voice trunks) |
| 19:49.39 | raden | f2Knight, have hundreds of hours on the issue with no resolve :( |
| 19:49.43 | leifmadsen | I suggest testing google voice trunks, and not relying on it, as Google keeps changing their interface. |
| 19:49.59 | raden | f2Knight, polycom 3xx series works nice\ly on it no issues |
| 19:50.07 | raden | and supports 722 codec |
| 19:50.22 | f2Knight | leifmadsen, I agree and have noticed that. They just want to use it as a cheap longdistance trunk when applicable. |
| 19:51.01 | f2Knight | raden, I just don't like Polycom / Cisco configs... |
| 19:51.40 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 19:51.41 | f2Knight | raden, I will get flamed here probably but I actually use a lot of Grandstream 2110's |
| 19:55.13 | *** join/#asterisk bn-7bc (~bjarne-im@pdpc/supporter/active/bn-7bc) |
| 19:56.08 | Qwell | How could anyone not like Polycoms? |
| 19:56.48 | WIMPy | Wait a few years, and maybe voip phones will become usable. |
| 19:57.37 | _Corey_ | Qwell: I have learned in the last 10 years doing this that some people are psychotic when it comes to phones |
| 19:58.19 | _Corey_ | no knock at you f2Knight :-) |
| 19:58.27 | raden | how are the cisco phones compared to polycom |
| 19:58.50 | _Corey_ | raden: They're all pretty bad, in varying degrees |
| 19:59.53 | f2Knight | _Corey_, no offense taken. |
| 20:01.01 | *** join/#asterisk bn-7bc (~bjarne-im@pdpc/supporter/active/bn-7bc) |
| 20:01.55 | leifmadsen | f2Knight: I find polycom configs fine once you automate building them |
| 20:02.07 | _Corey_ | f2Knight: I was referring more to customers' preferences and strange comments on phone sets... |
| 20:03.02 | f2Knight | raden, Qwell, My only real issue with Polycom/Cisco (which is made by Polycom) is the config process. Why should I have to have a TFTP server to configure the phones. No Phone UI to configure (polycoms I think do, but is ehh) I have used Snom, Grandstreams, and Aastra I like all of them each has some issue or another, but Feature per dollar, Grandstream packs a lot in. and has good sound clearity. G722 , and Ulaw are all i use. |
| 20:03.12 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 20:03.32 | f2Knight | leifmadsen, I agree if your deploying lots of them then automating it is not a big deal, but if your only deploying 5 or 6 phones... ugh |
| 20:03.45 | leifmadsen | I do that for 3 phones |
| 20:03.55 | _Corey_ | Seriously, I'd do it for 1 |
| 20:03.55 | leifmadsen | if I have ot do it more than 1-2 times, then automation is how to do it |
| 20:04.02 | leifmadsen | plus you only have to build the tools once |
| 20:04.07 | *** join/#asterisk tamiel (~tamiel@ip-38.net-81-220-92.rev.numericable.fr) |
| 20:04.09 | f2Knight | leifmadsen, then I want a copy of your deploy scripts lol |
| 20:04.14 | _Corey_ | The whole idea of managing phones individually via a web interface is kind of repugnant |
| 20:04.30 | leifmadsen | I now use my Android phone to scan barcodes which writes to a text file, then the script builds the configus |
| 20:04.47 | leifmadsen | f2Knight: they aren't complicated... just something hacked together in php |
| 20:05.10 | f2Knight | I got 2 Polycoms 301's and a 2 7960, and 3 7940's (I think all reflashed sip) that I just don't care/want to config because its a night mere lol |
| 20:05.18 | _Corey_ | leifmadsen: What's the andriod app? I've got a couple guys here that would love to ditch our USB scanner... :) |
| 20:05.57 | f2Knight | leifmadsen, the android app sounds nice.. something you wrote yourself? |
| 20:06.07 | leifmadsen | it's just called...... barcode2file I think |
| 20:06.12 | _Corey_ | cool |
| 20:06.22 | leifmadsen | it just takes the data and saves it to a text file |
| 20:06.36 | leifmadsen | then I send that file (email, scp, whatever) to the server |
| 20:06.39 | f2Knight | I see then your script runs and uses the file for mac addresses |
| 20:06.44 | leifmadsen | just loop through mac addresses, build configs, done |
| 20:06.51 | leifmadsen | yes |
| 20:07.12 | f2Knight | so the magic is in leifmadsen, php script |
| 20:07.31 | leifmadsen | it's mostly just a php whle loop |
| 20:08.10 | fullstop | If you had to take a 7 year old girl to the movies, would you take her to see Cars 2 or Winnie the Pooh? |
| 20:08.13 | raden | what operating system you use leifmadsen ? |
| 20:08.18 | sunfone | I do something very similar with a bash script for Polycom builds |
| 20:08.42 | sunfone | I have a text file that maps MAC to extensions numbers, and names |
| 20:08.54 | sunfone | I update the text file and run the bash script which builds all the Polycom XML files |
| 20:09.09 | leifmadsen | raden: linux of course |
| 20:09.18 | leifmadsen | sunfone: yep exactly -- same idea |
| 20:09.25 | leifmadsen | btw: app is barcode2file |
| 20:09.38 | leifmadsen | fullstop: I'm not sure that is legal in my country |
| 20:09.44 | _Corey_ | yeah, we found it :) |
| 20:09.49 | raden | leifmadsen, Sorry what distro is what I meant ? |
| 20:09.56 | leifmadsen | raden: Ubuntu and CentOS |
| 20:10.10 | fullstop | leifmadsen: I'm sure that you can do that in Canada.. ;-) |
| 20:11.11 | raden | I started using Ubuntu other night and only thing driving me nuts is the unity thingy staying on the left of screen :( Been using suse for 9 years now and just starting to feel broken in ways ) maybe its more KDE is feeling broken not sure .... |
| 20:11.32 | leifmadsen | raden: well it depends if you're talking about desktop or server -- those are totally different |
| 20:11.59 | McBoing | Trying to figure out why my Polycom phone is showing the time as EPOCH and the display is blinking, I know it has to be my sip_325.cfg file I created for it because when I go back to the legacy cfg file and reboot, the time is ok, isnt it only the SNTP paramaters that I need to setup, because they are the same for both cfg files, any ideas? |
| 20:12.06 | leifmadsen | I'm currently using Ubuntu 10.10 Desktop and Fedora 15 (F15 primary), and then I use Ubuntu Server 10.04 for servers, or CentOS 5.x sometimes |
| 20:12.19 | leifmadsen | McBoing: because it didnt' get connected to an NTP server |
| 20:12.50 | raden | leifmadsen, as a server SUSE has been solid , desktop ( lots of quirks ) whats your experiance ? |
| 20:12.52 | McBoing | leifmadsen: but the SNTP server IP is the same in both cfg files, and I dont see any errors in the boot log |
| 20:13.11 | leifmadsen | ok |
| 20:13.19 | leifmadsen | raden: with SuSE? Zero. |
| 20:13.27 | leifmadsen | and I don't plan on trying it ever |
| 20:13.38 | leifmadsen | I don't believe in learning 5 tools that do the same thing |
| 20:15.18 | raden | leifmadsen, I do agree , What do you use ? I'm trying to simplify things when it comes to technology .... |
| 20:15.28 | leifmadsen | raden: asked and answered |
| 20:15.49 | raden | Ubuntu pretty good all around ? Server and Desktop ? |
| 20:15.58 | McBoing | CentOS! |
| 20:16.57 | raden | I'm getting sick of having to run 5 different operating systems and Ubuntu does seem to be the future |
| 20:17.06 | raden | I could totally be wrong though . |
| 20:17.13 | leifmadsen | so stop using many distros |
| 20:17.23 | raden | lol |
| 20:17.30 | leifmadsen | if you want to stop losing money in stocks, stop losing money in stocks |
| 20:17.30 | raden | getting everything to work on one is a PITA |
| 20:17.41 | raden | I agree |
| 20:17.48 | xbp | hello beautifuls |
| 20:17.51 | xbp | humpday |
| 20:18.02 | leifmadsen | humps the air |
| 20:18.26 | jaytee | CentOS is great. Once Russell starts kickin ass at Red Hat the code in CentOS can only get better. |
| 20:18.46 | leifmadsen | I'm not sure he'll be working in that area :) |
| 20:18.52 | leifmadsen | he's doing clustering stuffz |
| 20:19.16 | jaytee | his influence will rub off, it will be a cascading effect of code awesomeness :-) |
| 20:19.18 | _Corey_ | I hear Fedora has some good people... ;) |
| 20:19.30 | leifmadsen | I hear Linux has good people |
| 20:19.50 | fullstop | Russell is with redhat now? |
| 20:19.57 | jaytee | will be soon |
| 20:20.00 | leifmadsen | www.russellbryant.net |
| 20:20.05 | fullstop | No longer with digium? |
| 20:20.12 | leifmadsen | see link above |
| 20:20.14 | jaytee | his last day is the 29th |
| 20:20.15 | leifmadsen | that's why I typed it |
| 20:20.53 | fullstop | Well, that is bittersweet, I suppose. |
| 20:21.14 | McBoing | make time work on my phone |
| 20:21.23 | jaytee | huh? |
| 20:22.50 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 20:22.54 | McBoing | lol just being a PITA, my polycom soundpoint IP 6000 display is flashing with Jan 1 2007 or some nonsense, if I point the device to the old cfg file and reboot the time is ok, checking the SNTP section they are the same, must be some other parameters to ensure the phone gets an ntp server? |
| 20:23.14 | jaytee | GMTOffset |
| 20:23.20 | xbp | it gets it from the pbx i think |
| 20:23.23 | xbp | is it in the cfg? |
| 20:23.29 | McBoing | yeah |
| 20:23.31 | xbp | what is different between the stock cfg and that one |
| 20:23.41 | fullstop | dhcp option type differences. |
| 20:23.47 | _Corey_ | Check your DHCP override setting, in the SNTP area... maybe getting something bogus from elsewhere |
| 20:23.48 | xbp | i used to know the unix command for checking differences between files |
| 20:23.49 | xbp | :( |
| 20:23.55 | McBoing | well diff shows tons of crap |
| 20:23.58 | xbp | was it ds for delete spaces |
| 20:24.04 | xbp | lol lost |
| 20:24.06 | McBoing | thats why its easier to ask, if SNTP is the only section I need to worry about |
| 20:24.13 | xbp | ya |
| 20:24.23 | xbp | diff |
| 20:24.25 | xbp | right! |
| 20:24.34 | xbp | <3! |
| 20:27.12 | McBoing | gdi "Uploading boot log, time is Mon Jan 1 00:00:16 2007" lol |
| 20:28.27 | *** part/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:28.56 | WIMPy | . |
| 20:29.09 | McBoing | .. |
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| 20:50.32 | f2Knight | my 2cents on CentOS... why use any distro that is based off another , that is that has to wait for the primary distro to release before it can update. That just seems stupid to me. Same feeling about using LinuxMint for a desktop as opposed to just using Ubuntu which its built from. |
| 20:51.17 | *** join/#asterisk galaxywatcher (~galaxywat@pdpc/supporter/active/galaxywatcher) |
| 20:51.23 | f2Knight | that being said.. production servers, Ubuntu LTS, RedHAT Enterprise (if you can afford it), SuSE Server (If you can afford it) that really leaves only one option I know |
| 20:52.31 | f2Knight | Desktops, play explore use what you like but stability... security updates etc... above rule applies. Why wait for someone else to update so someone else can repackage the same stuff. |
| 20:53.50 | f2Knight | That being said I think its fun to play with desktops, I was a long time Suse user, and swtitch to Ubuntu only about 5 years ago. I have only 2 CentOS boxes left in the wild.. one is a Trixbox server the other runs asterisk and is the T1 interface. |
| 21:03.47 | *** join/#asterisk SpiderMon (~SpiderMon@68.152.22.33) |
| 21:04.07 | SpiderMon | can i get some help with a multi location setup |
| 21:04.39 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:04.39 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:05.23 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 21:05.36 | timeshell | Sidetone: http://forums.asterisk.org/viewtopic.php?f=1&t=18321 |
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| 21:06.05 | timeshell | I have started to have issues as per ^^^^ with some of my users on occasional/regular daily basis. |
| 21:06.26 | timeshell | Seems to have come up since I upgraded away from 1.6.0.x to 1.6.2.x or 1.8.4.4 |
| 21:06.39 | timeshell | Phones are all Polycom IP phones. |
| 21:06.42 | *** join/#asterisk Wiretap_Work (~wiretap@unaffiliated/wiretap) |
| 21:06.49 | timeshell | Only happens on PSTN incoming calls. |
| 21:06.59 | timeshell | Occasionally, not every call. |
| 21:07.10 | timeshell | Any idea on how to deal with it? |
| 21:07.19 | SpiderMon | can i get some help with a multi server/location setup |
| 21:07.20 | malcolmd | what's an aak? |
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| 21:24.23 | jeffspeff | on a fresh install on centos, i ran "make config" after all was compiled... i then ran "service asterisk start" it started fine and i was able to console into it. i rebooted the machine without stopping asterisk first... now when i run "service asterisk start" it only starts "safe_asterisk" as evident with a "ps aux | grep asterisk"... how do i fix this? |
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| 21:46.31 | Bipul | p3nguin, ping |
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| 22:10.36 | f2Knight | guess spiderman doesn't know to how to keep the window open in the back ground |
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| 22:42.41 | prologic | Anyone know of a good resource of ringtones suitable for use by asterisk/freepbx ? |
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| 22:54.07 | ChannelZ | Ringtones? |
| 22:54.13 | ChannelZ | That has more to do with your phones than Asterisk |
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| 23:06.01 | datarecall | i am trying to figure out why asterisk wont make outgoing calls i have the follow me settings setup properly which log would it be in |
| 23:08.35 | datarecall | It's not our turn (SIP/1962941342-0000002a). is what shows up in asterisk -vvvvvrd but the follow me settings are setup to make outgoing calls |
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| 23:41.18 | f2Knight | Q: Okay so Channel variables do not pass through the Queue command.. Does anyone have any idea how to extract exactly how long a caller spends talking to an Agent? |
| 23:41.33 | f2Knight | Perferably access able from the dialplan |
| 23:43.13 | p3nguin | If your variables are not inheriting as new channels get created, you could try setting the variable using an underscore or two underscores. Set(_myVar=foo) or Set(__myVar=foo) |
| 23:51.18 | *** join/#asterisk kaushal (~kaushal@115.118.156.163) |
| 23:51.21 | kaushal | Hi |
| 23:51.27 | *** join/#asterisk titter (~Justin@c-76-101-15-40.hsd1.fl.comcast.net) |
| 23:51.44 | kaushal | can i have detail explanation of error codes reported by Asterisk are 101 (for 448 calls) and 16 (for 239 calls) |
| 23:52.18 | titter | I have a dynamic app in features.conf use a goto it seems not to do anything. It just says goto the context, and doesn't go any further. Nothing is executed. |
| 23:54.09 | johnnyasterisk | titter: what does the feature dial? have you tried using extension 's' in that context? |
| 23:54.23 | f2Knight | p3nguin, what does the _ and __ do? |
| 23:55.11 | f2Knight | kaushal, I think what you are looking for is SIP response codes. http://en.wikipedia.org/wiki/List_of_SIP_response_codes |
| 23:55.13 | titter | johnnyasterisk: It's for recording a call. If I use a macro it works, however I need to be able to play a beep every 15 seconds. |
| 23:56.07 | f2Knight | titter, if you find a solution to this let me know please, it sounds like one that will pop up now and then and would be good to add to the ticks book |
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| 23:57.12 | titter | Ya I thought I had it, did a Macro and tried to do a goto to jump out of the Macro ... same thing, just stalls. |
| 23:57.37 | kaushal | f2Knight: it does not detail |
| 23:58.19 | kaushal | Error code 101 actually |
| 23:58.28 | f2Knight | kaushal, sure what you mean I am not, |
| 23:58.46 | kaushal | where can i seek info ? |
| 23:59.08 | kaushal | Can someone please guide me |
| 23:59.29 | f2Knight | kaushal, Not sure what your referreing too . Are you getting this error on the asterisk CLI? or on your SIP phone? |
| 23:59.36 | kaushal | yes |
| 23:59.42 | kaushal | asterisk CLI |
| 23:59.48 | f2Knight | pastebin it |