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| 00:42.57 | lvlinux | can i have asterisk mark BOTH CoS and ToS values? Or is it one or the other? |
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| 00:51.05 | fornax | WIMPy: When I do not load the module from the kernel, dahdi does not stop since it does not find the device. So I assume it uses the driver from the kernel |
| 00:51.52 | WIMPy | No. You need to disable the linux driver for dahdi to work. |
| 00:52.19 | WIMPy | DAHDI has its own driver. |
| 00:53.33 | lvlinux | hey WIMPy by "Linux driver" do you mean like what is part of libpri? or is that something else? |
| 00:54.34 | WIMPy | No. What is part of the linux kernel. |
| 00:54.43 | WIMPy | And that doesn't use libpri. |
| 00:56.28 | lvlinux | oh ok I see. |
| 00:58.13 | fornax | WIMPy: Before I installed kernel 3.12 I did not have to install a kernel driver. But when I rebooted dahdi did not start because it did not find the driver for the bri card anymore. I had a look into the kernel and there was an entry marked as new in the kernel for the isdn card. I enabled the driver, compiled it as module, made a modprobe and after it was listed with lsmod dahdi was able to start. So is dahdi not using the kernel driver? |
| 00:59.36 | WIMPy | Maybe it's still not clear: DAHDI are kernel modules and incude driver modules, but they are not related to the drivers supplied in the Linux Kernel. |
| 00:59.52 | WIMPy | And they don't use them, either. |
| 01:00.05 | WIMPy | If you have one of them loaded, you can't load the other. |
| 01:01.22 | fornax | WIMPy: But why does dahdi say /dev/dahdi not found when I do not load the driver from the kernel via modprobe? the node is not available when I do not select the module |
| 01:01.56 | WIMPy | You have to load the dahdi modules to get that device. |
| 01:03.08 | WIMPy | The dahdi driver is wcb4xxp, the linux driver is hfcmulti. |
| 01:04.12 | fornax | WIMPy: Ah, okay, but why does it not work when I set n for misdn_hfcmulti? |
| 01:04.36 | fornax | WIMPy: even when wcb4xxp is in my dahdi modulesß |
| 01:05.02 | WIMPy | I dont know. But I know that dahdi will NOT work if you have hfcmulti loaded. So you don;t need to enable it. |
| 01:05.06 | WIMPy | Or shouldn;t have to. |
| 01:06.15 | fornax | WIMPy: Strange, very strange, because before dahdi worked without recompiling the kernel, 3.12 was the first kernel that did only work with enabled kernel driver |
| 01:06.17 | WIMPy | I had the issue with an older version of dahdi that the card I used didn't work, unless I loaded hfcmult _and_unloaded_it_again_ before loading wcb4xxp. |
| 01:06.32 | WIMPy | But that issue should be long gone. |
| 01:06.52 | fornax | WIMPy: I will investigate that tomorrow, you have the same timezone and know how late it is ;-) |
| 01:07.22 | WIMPy | Work gourts ar til 4 :-) |
| 01:07.27 | WIMPy | hours |
| 01:10.09 | fornax | WIMPy: Yes, you are right, it’s almost the same for me, but I have done my work today and can have a good night |
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| 01:33.38 | raspberrypifan | hi |
| 01:34.05 | WIMPy | lo |
| 01:34.27 | raspberrypifan | hows it going wimp |
| 01:34.54 | WIMPy | No news |
| 01:36.34 | raspberrypifan | sad |
| 01:38.19 | WIMPy | Sometimes it's good, sometimes it's bad. |
| 01:38.25 | raspberrypifan | yea true |
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| 03:16.38 | linocisco | from someone's extension on another desk, how can we call using own extension? for example, I am john and I have to use Tom's phone on his desk, I would like others to see I am calling using my extension, not Tom's extension |
| 03:22.18 | ChannelZ | Just fudge the CID |
| 03:22.46 | ChannelZ | make some extension code |
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| 04:56.32 | linocisco | from someone's extension on another desk, how can we call using own extension? for example, I am john and I have to use Tom's phone on his desk, I would like others to see I am calling using my extension, not Tom's extension |
| 05:00.18 | r00f | you were answered. just make some extension prefix for that |
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| 05:02.07 | dddh | yo |
| 05:02.20 | r00f | sup |
| 05:02.24 | dddh | r00f: hi |
| 05:02.48 | r00f | dddh: hi |
| 05:02.53 | dddh | I have a strange dialplan with MixMonitor and StopMonitor |
| 05:03.31 | dddh | it seems to me that StopMonitor doesn't stop MixMonitor, but StopMixMonitor should be used instead |
| 05:04.34 | r00f | so why don't you try? |
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| 05:13.21 | dddh | r00f: I'll just change this one line .. |
| 05:13.22 | dddh | http://devopsreactions.tumblr.com/post/84505783088/ill-just-change-this-one-line |
| 05:13.46 | r00f | 8) |
| 05:14.35 | r00f | well, it's been a while since i used mixmonitor, and besides i didnt use stop, channel hangup was ok for me. but wiki says that you are correct about stopmixmonitor |
| 05:15.40 | dddh | r00f: there are endless loops in this dialplan with MixMonitor and StopMonitor |
| 05:16.21 | dddh | and every time it starts new MixMonitor a new file is created |
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| 05:42.46 | dddh | r00f: and when call is finished I see 20-30 files there :) |
| 05:43.53 | r00f | dddh: so may be the source of problem are loops, not the wrong stopmonitor application? ) |
| 05:45.06 | dddh | r00f: yes, it should stopmixmonitor before next cycle(where new mixmonitor will start) |
| 05:47.43 | dddh | seems like the problem was that it started new mixmonitor(with different filename), but didn't stop previous one |
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| 07:48.49 | Zogot | ahoyhoy |
| 07:49.02 | r00f | sup |
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| 07:54.46 | Zogot | Could anyone know of any reasons why all phones wont ring when called to an IVR. Out of the 9, 2 dont ring. |
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| 08:01.32 | Zogot | is there a limit to the number of FastAGI calls? The 2 phones that dont ring get a FastAGI errror: Couldn't connect to any host. FastAGI failed |
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| 09:18.22 | linocisco | <PROTECTED> |
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| 10:21.36 | *** join/#asterisk DeviantPeer (~kvirc@s529d8584.adsl.online.nl) |
| 10:22.30 | DeviantPeer | hi all. Can anyone tell me how reliable the res_fax is? (faxes by normal are unreliable anyway...) |
| 10:22.48 | DeviantPeer | in the issue list there are only 4 minor bug open |
| 10:23.06 | DeviantPeer | but is that as the re_fax is a good module? or noone is using it? |
| 10:23.09 | vedic | Is there any way to read a live audio ? I am using IVR to record user's message but I would like to access it as soon as I can (before the complete file gets saved to the system). What is the suggested way? |
| 10:24.59 | DeviantPeer | vedic: I think you have to look at EAGI |
| 10:25.37 | vedic | DeviantPeer: Could you point me to such an example? |
| 10:25.39 | DeviantPeer | vedic: EAGI is like AGI but provides the script with file descriptors of the audio channels. |
| 10:25.47 | DeviantPeer | http://www.voip-info.org/wiki/view/Asterisk+EAGI |
| 10:25.55 | DeviantPeer | I never tried it... |
| 10:26.31 | DeviantPeer | but looks simple enough |
| 10:27.54 | vedic | DeviantPeer: What if multiple calls are coming at the same time? |
| 10:28.04 | vedic | File descriptor 3 will get shared? |
| 10:28.23 | DeviantPeer | I don't think it's shared |
| 10:28.44 | DeviantPeer | I think asterisk will write PCM data to that pipe |
| 10:28.47 | DeviantPeer | and you can read it |
| 10:29.11 | vedic | DeviantPeer: But will it differentiate from one call to another call? |
| 10:29.16 | DeviantPeer | yes |
| 10:29.31 | DeviantPeer | the pipe fs will be diferent |
| 10:29.36 | DeviantPeer | s/fs/fd/ |
| 10:29.43 | DeviantPeer | yes |
| 10:29.48 | vedic | ok |
| 10:29.54 | DeviantPeer | good bot. |
| 10:31.11 | r00f | DeviantPeer: by reliablility you mean fax delivery or module itself? |
| 10:31.44 | r00f | reliability of fax-over-voip is very vast topic to discuss. alot of articles over internet |
| 10:31.46 | DeviantPeer | fax delivery |
| 10:32.10 | DeviantPeer | r00f: yes... it's not a simple subject at all. |
| 10:32.20 | r00f | for me, via 711 over PSTN it works almost 99% times |
| 10:32.35 | r00f | over via 711 over SIP - at least 80% |
| 10:32.44 | r00f | T38 hates me completely |
| 10:33.59 | DeviantPeer | hehe |
| 10:34.44 | r00f | however, recently Samsung guys were trying to connect MGI card with my asterisk. we achieved like 100% over T38 (in tests), but client needed 711 |
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| 10:35.07 | r00f | and they just could not configure it for 711. some bugged card, i guess |
| 10:35.43 | DeviantPeer | interoperability is always a darn thing... :) |
| 10:36.15 | DeviantPeer | but if g711 works well.. :) that should be enough for the old people to send faxes around. ;) |
| 10:37.46 | r00f | yeah, for me 711 is good. we switched to it everywhere, thanks god nowadays internets are fast enough |
| 10:38.00 | DeviantPeer | I asked because it seems to be a much simpler solution than using iaxmodem and hylafax (or something like that). |
| 10:38.11 | r00f | maybe it is because i dont understand T38, it's not for me to decide |
| 10:38.17 | r00f | but still |
| 10:39.30 | DeviantPeer | the problem might not be on your side.. |
| 10:39.34 | r00f | just give it a try, i havent used any of those |
| 10:39.55 | DeviantPeer | I think I will and then I'll come back here and share the experience. |
| 10:40.29 | r00f | however, via 711 i can send a fax to my desktop even over voippro. who make no promises about faxes, and god knows how they are routing call to my pstn operator |
| 10:41.11 | r00f | it simply works |
| 10:41.24 | DeviantPeer | that's not bad at all... |
| 10:41.59 | DeviantPeer | voippro must be taking some care with faxes nevertheless |
| 10:42.01 | r00f | yep. i was surprised tbh |
| 10:42.23 | DeviantPeer | they just don't promiss anything.. because it's really a very "jaggering" solution :) |
| 10:45.26 | DeviantPeer | r00f: thx for sharing your experience. I'll go and play asterisk config and sending some faxes to see how well I can get it to work. |
| 10:46.50 | r00f | np |
| 10:47.04 | r00f | have fun |
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| 10:50.23 | CommaCrazy | hi all, question, is there a way to patch OpenVox A1200 card into already running asterisk or does it really have to be recompiled with it for it to work? |
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| 10:53.17 | Rovanion | Must every user have their own entry in extensions.conf? |
| 11:01.18 | CommaCrazy | anyone |
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| 14:14.14 | neeby_goosey | Hello. |
| 14:14.46 | YAVoiper | hi |
| 14:16.18 | YAVoiper | Is this the proper place to ask questions about yum installs for asterisk? |
| 14:16.57 | Qwell | probably, but it depends |
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| 14:19.00 | YAVoiper | well, I'm new here so pls let me know if I err. We have installed asterisk 11 via yum for easier updated (always compiled in the past). Everything was great EXCEPT we are unable to execute CURL or SYSTEM dialplan commands. I was wondering if there was an additional yum install for those or if it is possible to use the those commands with the yum install packaging. |
| 14:19.57 | Qwell | That's where the "it depends" comes in. |
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| 14:21.50 | neeby_goosey | There was a problem with my asterisk system when (VoIP provider => asterisk => VoIP provider) calls were all silent and no RTP was sent either way. I asked about this here and user 'scouture' suggested adding 'Playback(silence/1);' before the outbound call and - voila - everything works. The question is - how does this workaround works? |
| 14:22.22 | Qwell | neeby_goosey: It takes time for those audio paths to be setup. |
| 14:22.46 | neeby_goosey | Wow. That is really all? |
| 14:23.38 | Qwell | Playback also causes an answer to happen. Were you answering the call? |
| 14:24.23 | YAVoiper | You can also take a tcpdump with/without & observe the differences |
| 14:25.14 | neeby_goosey | Qwell, yes, and that was the first #asterisk's suggestion. |
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| 14:28.10 | YAVoiper | Qwell, can you elaborate on the dependency regarding yum install * 11 & SYSTEM commands? |
| 14:28.56 | Qwell | YAVoiper: not without knowing anything about the packages you're using |
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| 14:33.41 | YAVoiper | Qwell, here is the yum list of installed packages for asterisk: |
| 14:33.53 | YAVoiper | asterisk.x86_64 11.7.0-1_centos6 @asterisk-11 |
| 14:33.55 | YAVoiper | asterisk-core.x86_64 11.7.0-1_centos6 @asterisk-11 |
| 14:33.56 | YAVoiper | asterisk-dahdi.x86_64 11.7.0-1_centos6 @asterisk-11 |
| 14:33.58 | YAVoiper | asterisk-doc.x86_64 11.7.0-1_centos6 @asterisk-11 |
| 14:33.59 | YAVoiper | asterisk-sounds-core-en-gsm.noarch |
| 14:34.01 | YAVoiper | <PROTECTED> |
| 14:34.02 | YAVoiper | asterisk-voicemail.x86_64 |
| 14:34.04 | YAVoiper | <PROTECTED> |
| 14:34.05 | YAVoiper | asterisknow-version.noarch |
| 14:34.07 | YAVoiper | dahdi-linux.x86_64 2.9.0-1_centos6 @asterisk-current |
| 14:34.08 | *** kick/#asterisk [YAVoiper!~north@pdpc/sponsor/digium/Qwell] by Qwell (pastebin) |
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| 14:40.33 | YAVoiper | Qwell, I'll say again, I am new here. Is it not proper etiquette here to list *short* lists? |
| 14:40.55 | Qwell | ~pastebin |
| 14:40.55 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:41.25 | YAVoiper | ok. Thanks Qwell |
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| 14:42.01 | Qwell | Yes, there is a curl package, no, system does not require anything special. |
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| 14:47.08 | YAVoiper | Ok, thanks. We'll dig a bit more then. Just wanted to confirm that it was included with yum. It looked like it was. Thanks again... |
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| 15:51.26 | [diecast] | may not be relevant to this channel, but anyone know if you can pass SS7 via audio cable on a phone device like iPhone |
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| 16:01.10 | jwwwww_ | Hello. |
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| 16:08.57 | navaismo | o/ |
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| 16:48.21 | igcewieling | I'm looking for some guidance with hangup handlers. Previously (when using "h" exten and "g" option to dial) when one side hangs up first, then the OTHER call leg either continues in the dialplan or jumps to the "h" extension, depending on which leg hung up first, and an audio message is played to the non-hungup leg. I am NOT having a problem with that. My issue is when using hangup handlers it seems the leg which hangs up gets |
| 16:49.10 | [TK]D-Fender | igcewieling: https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers |
| 16:49.18 | [TK]D-Fender | igcewieling: If you want to do it the "good" way |
| 16:49.42 | igcewieling | [TK]D-Fender: Um, that is what I'm using. |
| 16:49.54 | igcewieling | I'll pastebin some CLI output showing the issue. |
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| 16:54.12 | igcewieling | [TK]D-Fender: In fact it looks like that page is the only documentation in the universe for hangup handlers |
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| 16:56.46 | paulc | I never knew about hangup handlers! Much easier than having an h extension in every context that goto's our "hangup handling" post call processing.. interesting! |
| 16:58.47 | igcewieling | paulc: That's what I thought until I ran into this issue. |
| 16:59.45 | paulc | That said, the "h in every context" bit worked well - ours is a big IVR app, and it mirrors our legacy system a bit.. |
| 17:00.09 | paulc | I remember AGES ago, the h extension would fire if you used AMI to transfer a call to a different context/extension.. in the previous context I think.. |
| 17:00.35 | paulc | we kludged around it by transferring to a context with no h, that then did a goto to where we wanted to be. Not pretty, but it worked. And the channels no longer got hung up on. |
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| 18:11.39 | cmendes0101 | In an attended transfer I can't seem to get the callerid to change after the transfer. I have sendrpid=yes but not happening on the device. Its with asterisk 1.6 but I thought this was possible in that version |
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| 18:27.13 | *** join/#asterisk eduardonunesp (~eduardonu@187.103.104.42) |
| 18:27.22 | eduardonunesp | hello everybody |
| 18:28.51 | eduardonunesp | i have a asterisk 1.4.44 installed, ok i know that needs upgrade, but some very strange is happening, the number of calls for channels is very wrong |
| 18:29.14 | eduardonunesp | sometimes the asterisk get 100 channels and 400 calls |
| 18:29.42 | eduardonunesp | i think that calls don't hung up properly |
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| 18:33.45 | Zogot | is there any limit on the number of AGI calls? |
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| 18:36.27 | [TK]D-Fender | eduardonunesp: The counter is simply wrong. |
| 18:36.35 | [TK]D-Fender | eduardonunesp: And that version no longer supported at all |
| 18:37.17 | [TK]D-Fender | cmendes0101: 1.6 isn't a branch, nor is 1.6.x supported any more either... |
| 18:37.20 | eduardonunesp | [TK]D-Fender, yeah, i know it, but the server is in production |
| 18:37.40 | [TK]D-Fender | eduardonunesp: You'll have to live with it until you're off of it. |
| 18:37.54 | eduardonunesp | [TK]D-Fender, agree |
| 18:38.09 | eduardonunesp | Zogot, i will check |
| 18:38.29 | [TK]D-Fender | Zogot: Your system resources. |
| 18:38.36 | Zogot | eduardonunesp: thanks |
| 18:38.53 | Zogot | [TK]D-Fender: some of my phones are not all calling, seems to be random |
| 18:39.05 | Zogot | and i get a timeout on agi on those phones |
| 18:39.08 | Zogot | that dont call |
| 18:39.22 | [TK]D-Fender | Zogot: What does "some of my phones are not all calling" mean? |
| 18:39.32 | Zogot | [TK]D-Fender: they dont ring |
| 18:39.43 | [TK]D-Fender | Zogot: We know nothing about what your AGI is doing. |
| 18:40.09 | [TK]D-Fender | Zogot: You should be showing us the call and the code and better describing where this "call" is supposed to be happening. |
| 18:40.10 | eduardonunesp | Zogot, my agi is limited at 120 |
| 18:40.13 | [TK]D-Fender | ~pb |
| 18:40.14 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 18:40.15 | [TK]D-Fender | ^^^ |
| 18:42.51 | Zogot | [TK]D-Fender: they are communicating with a mysql, perhaps that is being the problem with mysql blocking the usage, causing the agi to time out |
| 18:43.05 | Zogot | il paste in a sec, on phone with manager |
| 18:47.31 | eduardonunesp | i'll continue with the monitoring, thanks guys |
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| 19:38.12 | tom_w | Strange issues. I just upgraded to trunk current. When a call comes in now it goes to privacy manager and prompts for a name. I do not have P or p set. What would cause this |
| 19:38.28 | Qwell | running trunk |
| 19:39.41 | tom_w | asterisk-11-current.tar.gz would you recommend another version/ |
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| 19:56.26 | twanny796 | could someone give me a dialing rule to dial 9xxxxxxxx and asterisk dials 00356xxxxxxxx ? |
| 19:59.54 | Penguin | twanny796: exten => _9XXXXXXXX,1,Dial(DAHDI/00356${EXTEN:1},60) |
| 20:01.26 | twanny796 | Penguin, ok, very good 10x |
| 20:01.39 | navaismo | tranny just add the X |
| 20:01.50 | navaismo | or that was like the +10? |
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| 20:30.03 | greenlava | hi there - anyone have any good resources for understanding DTMF? my phone system for a small business works in most cases, but I have 2 customers whose IVR's won't accept my keypresses for some reason |
| 20:39.28 | *** join/#asterisk Curs0r (46b85355@gateway/web/freenode/ip.70.184.83.85) |
| 20:39.33 | Kobaz | greenlava: use sip info or rfc2833 for dtmp |
| 20:39.34 | Kobaz | dtmf |
| 20:40.23 | Curs0r | Hi all. Is there some magical permission setting I need to set on /var/spool/asterisk/outgoing to have a php script move a .call file to that folder from /tmp? |
| 20:41.01 | Kobaz | magical? no |
| 20:41.19 | Curs0r | Ok, arcane (to me) setting then? :) |
| 20:41.37 | Kobaz | chmod/chown to proper ownership/permissions so your php script can write to the directory *and* asterisk can read/write it as well |
| 20:43.57 | Curs0r | So in this case I'd want to like, make a new group, add apache and asterisk to said group, then make that group the owner? |
| 20:44.20 | Kobaz | that's one way, sure |
| 20:44.32 | Kobaz | you'll need to restart apache and asterisk so that they become members of the new group |
| 20:45.04 | Curs0r | Obie kabie. I will give that a shot. Thanks |
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| 20:48.22 | Nugget | tree times a mabey. |
| 20:49.16 | file | Nugget, shhhhhh |
| 20:49.24 | file | Nugget, or I'll throw butter at you |
| 20:50.02 | Nugget | poutine |
| 20:50.13 | Nivex | poutine is nasty stuff |
| 20:51.17 | Curs0r | Hrmm, either I didn't do that right or it didn't work. Probably didn't do it right |
| 20:51.33 | Kobaz | ls -al /var/spool/asterisk/outgoing |
| 20:51.58 | Curs0r | drwxrwxrwx 2 asterisk callinit 4096 May 5 13:31 . drwxr-x--- 9 asterisk asterisk 4096 Jun 18 2012 .. |
| 20:52.10 | Curs0r | callinit is the group I added asterisk and www-data to |
| 20:53.55 | Kobaz | so your group ownership is read and execute |
| 20:54.04 | Kobaz | er, group permissions |
| 20:55.32 | Curs0r | Oh, it read to me as if it had full read/write for the callinit group |
| 20:56.11 | Kobaz | it's user/group/world |
| 20:56.42 | Kobaz | well i was looking at the end actually |
| 20:56.43 | Kobaz | .. |
| 20:56.45 | Kobaz | which is asterisk |
| 20:57.02 | Kobaz | this is what happens when doing 5 things at once |
| 20:57.22 | Kobaz | okay your . directory which is /var/spool/asterisk/outgoing, is 777 all access to everyone |
| 20:57.31 | Penguin | Sounds dangerous. |
| 20:57.39 | Kobaz | that works too, but it's not at all secure if you have multiple users on the machine |
| 20:58.05 | Penguin | chmod o-w /var/spool/asterisk/outgoing |
| 20:58.39 | Curs0r | drwxrwxr-x 2 asterisk callinit 4096 May 5 13:31 is what it says now |
| 20:59.26 | Penguin | asterisk can read, write and search, any user in the callinit group can do the same. Anyone else can only read and search in the directory. |
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| 20:59.46 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 21:01.46 | Curs0r | www-data still seems unable to move the file though |
| 21:03.29 | Penguin | Is that a uid or gid? |
| 21:03.37 | Curs0r | uid |
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| 21:03.56 | Penguin | Is www-data in the callinit group? |
| 21:04.05 | Curs0r | Yes |
| 21:04.19 | Curs0r | ls -lg /tmp/1.call returns: --rw-r--r-- 1 www-data 116 May 5 14:01 /tmp/1.call |
| 21:06.18 | *** join/#asterisk Zogot (~Adium@90-145-116-55.bbserv.nl) |
| 21:06.35 | Curs0r | woops, using -al I get: -rw-r--r-- 1 www-data www-data 116 May 5 14:01 /tmp/1.call |
| 21:07.31 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 21:14.34 | Curs0r | I definitely restarted both services... it's a puzzler haha |
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| 21:20.56 | *** join/#asterisk Geriatrix (~kvirc@mail.vancouveronthenet.com) |
| 21:21.48 | Geriatrix | can someone point me in the right direction with coding a php interface that interacts with Manager - to display line status - ringing status etc |
| 21:23.21 | dfighter | how would such an application work? I mean if you want it to show stuff real time |
| 21:24.10 | [TK]D-Fender | ~book |
| 21:24.10 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:24.14 | [TK]D-Fender | ~asteriskwiki |
| 21:24.15 | infobot | somebody said asteriskwiki was http://wiki.asterisk.org |
| 21:24.17 | [TK]D-Fender | ^^^^^^ |
| 21:24.27 | [TK]D-Fender | Thre |
| 21:24.56 | Geriatrix | my developer is having issue receiving the events - i need to point ihim in the right direction for help |
| 21:25.16 | [TK]D-Fender | The docs are there. |
| 21:26.21 | Curs0r | sudo chown -R :callinit /var/spool/asterisk/ <-- that worked |
| 21:26.55 | dfighter | Geriatrix, I'm still wondering how and what application you are building exactly |
| 21:27.21 | *** join/#asterisk supersoaker (45a958c4@gateway/web/freenode/ip.69.169.88.196) |
| 21:27.37 | dfighter | because ofc you can use PHP to connect to the manager, but while you are connected and are receiving the events, you can't render the webpage afaik |
| 21:28.00 | dfighter | that's why I myself built a java applet instead for our client |
| 21:28.37 | *** join/#asterisk petris (~petris@2607:5300:60:5475:64c2:903:2ab1:7d43) |
| 21:28.45 | supersoaker | Yo! I have a very interesting issue, when I dial out to most numbers it works great, if i dial out to a certain number it decides to setup a bridge and there is no audio |
| 21:28.49 | supersoaker | anyone seen this? |
| 21:30.51 | *** join/#asterisk petris (~petris@2607:5300:60:5475:64c2:903:2ab1:7d43) |
| 21:31.28 | Geriatrix | it woudl be similar to fop |
| 21:31.33 | Geriatrix | but not to that extent |
| 21:31.47 | [TK]D-Fender | Geriatrix: You wouldn't do this on code that hits a web-page direct |
| 21:31.50 | Geriatrix | i jsut need a small windows that shows stauts of the phone |
| 21:31.58 | [TK]D-Fender | Geriatrix: This isn't a "background" thing |
| 21:32.00 | Geriatrix | kind of like a side cart for the receptionist |
| 21:32.09 | *** part/#asterisk tom_w (~telcomind@ip68-104-29-79.lv.lv.cox.net) |
| 21:32.14 | [TK]D-Fender | Geriatrix: If you expect live, then it's the wrong approach |
| 21:32.23 | [TK]D-Fender | Geriatrix: There's a reason FOP was flash... |
| 21:32.24 | dfighter | Geriatrix, I think you didn't understand what I was telling you :) |
| 21:32.45 | Geriatrix | dfighter: i guess not |
| 21:32.59 | Qwell | It's no longer 1995. PHP for something live like that is fine. |
| 21:33.01 | [TK]D-Fender | PHP on a flat page is BLOCKING. this is not for an interactive page |
| 21:33.26 | dfighter | Qwell, ok but how do you render and communicate at the same time? |
| 21:33.28 | [TK]D-Fender | You could have a back-end part that uses it, but it can't be in the live display page |
| 21:33.43 | [TK]D-Fender | dfighter: You don't within a single process like that |
| 21:33.45 | Qwell | dfighter: Very basic webpage design. |
| 21:34.01 | Geriatrix | how can you do something like that ? |
| 21:34.05 | [TK]D-Fender | Qwell: Point was that he can't get live updates pushed like that normally.... |
| 21:34.13 | Qwell | [TK]D-Fender: *shrug*, not hard |
| 21:34.15 | [TK]D-Fender | Geriatrix: Your programmer should aready know better |
| 21:34.19 | dfighter | Qwell, I am not a web developer :) |
| 21:34.24 | dfighter | Qwell, so please elaborate! |
| 21:34.29 | [TK]D-Fender | Geriatrix: AJAX, etc |
| 21:34.34 | Qwell | ^ |
| 21:34.38 | dfighter | oh |
| 21:34.49 | Geriatrix | ajax can subscribe to ami ? |
| 21:35.01 | Qwell | Oh look, my bus is here. |
| 21:35.09 | dfighter | well I sure wouldn't like that, that would basically mean you need to log in every time, and poll the statuses, wouldn't it? |
| 21:35.14 | Geriatrix | i need to tell him where to look to figure it out :) |
| 21:35.29 | Qwell | Geriatrix: If he's a web developer, he'll know all about it. |
| 21:35.32 | dfighter | instead of just getting the events to update the status |
| 21:35.42 | [TK]D-Fender | Dev already seems less than competant... |
| 21:35.42 | Geriatrix | well he is and he doesn't :) |
| 21:35.53 | Geriatrix | i know asterisk - but i am not good at web stuff |
| 21:35.59 | Geriatrix | lol |
| 21:36.12 | [TK]D-Fender | Him neither I'm suspecting. |
| 21:36.49 | Geriatrix | cdr stats logs in all the tiem for "real time" graph |
| 21:37.08 | Geriatrix | i dont' liek that abotu cdr stats - but it doesn't seems to be much of a performance hit to manager |
| 21:38.54 | hecatae | something that reads the asterisk cli and outputs meaningful information would be useful |
| 21:39.42 | hecatae | xima chronicall does this for avaya ip office, reads the cli and outputs to a webpage or agent widget |
| 21:40.26 | hecatae | of course I comparing proprietary to opensource but still |
| 21:41.47 | [TK]D-Fender | checkout time, BBL |
| 21:42.02 | Geriatrix | looking in the book - ami over http seems to be the way to go |
| 21:42.12 | Geriatrix | but they do mention flash use in fop2 |
| 21:43.19 | dfighter | Geriatrix, I don't know cdr stats, but I suppose that has no interface to work with like AMI :) |
| 21:43.22 | dfighter | so no other choice |
| 21:44.16 | Geriatrix | seems liek ami over https is the way to go for outbound calls etc |
| 21:44.21 | Geriatrix | we already have call files for that |
| 21:44.37 | Geriatrix | i need to subscripbe to hints like ringing - status - etcccc. |
| 21:44.56 | Geriatrix | that is dynamic contents and i am not sure how that can be delivered to web |
| 21:45.03 | Geriatrix | ajax may be the way to go ? |
| 21:46.12 | dfighter | as I said I used an applet for my monitor application |
| 21:46.19 | dfighter | you could probably go with websockets too |
| 21:46.23 | Geriatrix | java applet ? |
| 21:46.26 | dfighter | yep |
| 21:46.51 | Geriatrix | going to google webcsockets real quick :) |
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| 21:47.07 | *** join/#asterisk tdonahue (~tdonahue@vonmail.vonworldwide.com) |
| 21:47.15 | tdonahue | hi all, any digium guys around? |
| 21:47.16 | navaismo | only for 12 |
| 21:48.27 | supersoaker | Okay, Here is the logs of my issue: http://pastebin.com/9LKq7nsq |
| 21:52.41 | supersoaker | When i dial one outbound number is get: Packet2Packet bridging SIP/cisco7940-00000641 and SIP/outgoingphone#-00000643 |
| 21:52.41 | *** join/#asterisk gavit (~gavit@190.98.71.89) |
| 21:52.41 | supersoaker | verses all other outbound dials i get : SIP/outgoingphone#-00000640 is making progress passing it to SIP/cisco7940-0000063e |
| 21:52.41 | supersoaker | Asterisk 1.4.30 |
| 21:52.41 | supersoaker | When the Packet2Packet bridging happens I have no audio on the channel |
| 21:53.39 | gavit | is there a iscsi/diskless version of asterisk/freepbx? |
| 21:57.22 | *** part/#asterisk hecatae (~Philip@host-92-27-124-62.static.as13285.net) |
| 21:59.30 | *** join/#asterisk CoreIT84 (~Corey84@wsip-98-191-213-227.dc.dc.cox.net) |
| 21:59.53 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:00.28 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 22:00.42 | Qwell | tdonahue: Now that's a domain I haven't seen for a very long time. |
| 22:06.18 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 22:14.09 | *** join/#asterisk mattcen (~mattcen@c110-22-201-130.sunsh4.vic.optusnet.com.au) |
| 22:19.30 | tdonahue | Qwell, was just wondering since i couldn't find an email address for them and part of their website was down at the time |
| 22:23.36 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 22:46.16 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 22:47.02 | newtonr | tdonahue, what website was down? who are you trying to E-mail? |
| 22:47.06 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-126-182.dynamic.qsc.de) |
| 22:47.45 | newtonr | tdonahue, I'm about to step out, but I'll PM you my E-mail if you need help trying to reach a specific Digium address. |
| 22:52.47 | *** join/#asterisk mushroomed (~mushroome@li173-111.members.linode.com) |
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| 23:39.37 | mushroomed | Hi, is there anyway to call an internal extension from my extension and then Playback(telephone-number), then Read(phone_number) and make asterisk call that number and when it answers asterisk Playback(one-moment-please) to the callee and then transfer that call to my extension? |
| 23:40.10 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 23:40.54 | *** join/#asterisk petris (~petris@2607:5300:60:5475:64c2:903:2ab1:7d43) |
| 23:41.07 | [TK]D-Fender | mushroomed: First, don't call the device you're talking on an "extension". Next, for teh call-bnnack : "core show application originate" |
| 23:41.19 | mushroomed | For example, I call my internal *99 and it asks for a number so I enter 12345678 and then it says 'one-moment-please', hangs up and call 12345678. When it answers, asterisk plays 'one moment please' and transfer that call to my extension |
| 23:43.30 | [TK]D-Fender | Do you read. Push that number somewhere. Then originate the call-out. |
| 23:43.40 | [TK]D-Fender | your* |
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| 23:56.14 | navaismo | or disa maybe? |
| 23:56.46 | [TK]D-Fender | no... |
| 23:56.54 | navaismo | :( |