| 00:01.15 | *** join/#asterisk jblack (~jblack@pool-71-181-222-126.sctnpa.east.verizon.net) |
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| 00:18.05 | p3nguin | I'm having a hard time believing that no one is commenting in that book. |
| 00:19.30 | p3nguin | I, of course, don't get paid enough to read over the entire thing and fix all the mistakes, so someone else needs to contribute. |
| 00:21.18 | sawgood | I think is rather new ... |
| 00:23.09 | p3nguin | I've never used this system before... if I change my mind after I submit a comment, can I edit/delete it afterward? |
| 00:23.43 | sawgood | I have never left comments before |
| 00:24.05 | sawgood | enter a test comment (tell me the URL) ... |
| 00:24.07 | p3nguin | I'm a bit impulsive and passive-aggressive... so I sometimes say things that need to be changed. |
| 00:24.10 | sawgood | then we will see if you can edit |
| 00:24.28 | sawgood | I've never seen that in you :> |
| 00:24.33 | p3nguin | lol |
| 00:24.34 | p3nguin | right |
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| 01:28.03 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 01:28.16 | raden | how can i make asterisk dial from a number on a webpage like click to dial |
| 01:29.54 | p3nguin | What do you mean by "dial FROM a number" on a web page? |
| 01:30.12 | p3nguin | Do you actually mean dial TO your phone FROM a web page? |
| 01:30.18 | raden | like our customer management software is web based |
| 01:30.42 | raden | how can i make it so users can just click on the number for the user and it will dial that person and also there extension |
| 01:32.39 | p3nguin | It can be done with a very simple originate command. |
| 01:33.10 | p3nguin | I don't know if there is a more elegant way to do it. |
| 01:37.33 | atan | raden, by chance are you talking about like how RingCentral does it? =) |
| 02:01.23 | raden | atan, dunno but skype has it where u just click on a number on a webpage and it works |
| 02:02.11 | atan | Err. They are slight different in how they do it though since they have control over the computer browsing, more or less. |
| 02:03.02 | atan | Why on earth does this work http://pastebin.com/8Yks4Shg but if I change 16 to 2 it stops working. It doesn't just "stop" at 2 like it should. |
| 02:03.13 | atan | Do you need to reload something in asterisk when you change part of an AGI? |
| 02:07.39 | atan | My own fault. Added escape digits and we are good again. Oddly it worked by default with 16, just not 4. Oh well. |
| 02:23.47 | tyman | p3nguin: I flippin' love asterisk! Man...losing sleep with I'll the ideas going thru my head! :-) |
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| 02:31.54 | p3nguin | Yep, it certainly has its usefulness. |
| 02:34.18 | *** join/#asterisk thehar (~thehar@diddlebox.thehar.com) |
| 02:40.31 | raden | fuck I cant even use skype for calling out limit 50 calls a day :( |
| 02:40.34 | raden | Im soooo screwed |
| 02:40.52 | raden | is there a softphone for asterisk that has click to call on web ? |
| 02:41.38 | raden | OMFG i need another monitor :( |
| 02:43.28 | p3nguin | It's not a softphone FOR Asterisk, but it's a softphone and it works WITH Asterisk... Zoiper Web. http://www.zoiper.com/webphone/ |
| 02:46.49 | atan | Okay. I moved my CGI from my outbound context where it had a 123 extension, worked fine. Played my audio back just fine. Now that I moved it to my inbound it does not play audio. My phone connects to it, that's fine, and the console shows it playing the sound files, but there is no sound... pfft |
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| 02:52.29 | atan | I see. I didn't send 'ANSWER' my bad ^_^ good now |
| 02:56.37 | raden | p3nguin, what I need is a way to just click on the number in a browser and have it put the call through be it rining the extension at that desk or a softphone |
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| 03:55.23 | tyman | raden: Are you use the skype channel for asterisk? Is it limitted to 50 calls a day? |
| 03:56.10 | atan | tyman, I don't believe he is no |
| 03:56.45 | tyman | atan: ok, not understanding the 50 calls a day statement |
| 03:57.05 | atan | tyman, he was using skype but hit a call limit, now he wants to use asterisk and his provider.. and blah blah blah. |
| 03:57.25 | atan | http://www.voip-info.org/wiki/view/Asterisk+click+to+call might be one option perhaps |
| 03:58.14 | atan | many options even |
| 04:02.11 | WIMPy | Yes, it's really bad how bodern bowsers or rather operating systems are completely unable to provide a connom way to handle url methods. |
| 04:02.38 | atan | Well there is a tel: RFC I believe but nobody uses it |
| 04:03.03 | WIMPy | That's what I'm talking about. It is actually really easy. |
| 04:09.30 | WIMPy | Maybe that would be a nice little task to wake up to. |
| 04:10.37 | atan | ^_^ |
| 04:10.55 | atan | I suppose we could write a nice little firefox plugin for it ;) |
| 04:11.16 | WIMPy | I would guess there are soem already. |
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| 04:33.42 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 04:33.46 | AliRezaTaleghani | L-) |
| 04:33.59 | AliRezaTaleghani | is there any body help me? |
| 04:34.02 | AliRezaTaleghani | how can i get the Queue Max Length (maxlen variable) with in a Dialplan? |
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| 04:40.42 | WIMPy | Hmm. Where do I get the URI? |
| 04:41.27 | AliRezaTaleghani | me :-/ |
| 04:41.37 | AliRezaTaleghani | ? |
| 04:42.19 | WIMPy | No, me. |
| 04:44.41 | AliRezaTaleghani | :D |
| 04:44.49 | AliRezaTaleghani | WIMPy: look... |
| 04:45.07 | AliRezaTaleghani | i had been defined a Queue manlen=10 variable in my queue definition |
| 04:45.31 | WIMPy | I have never used queues. |
| 04:45.40 | AliRezaTaleghani | know in my dialplan.... i want to check that if this limit is reach... :) |
| 04:45.55 | AliRezaTaleghani | woops, so just thanks.... WIMPy ;) |
| 04:46.47 | ChannelZ | look at the function QUEUE_VARIABLES |
| 04:47.04 | AliRezaTaleghani | i check... but it don't work at all |
| 04:47.07 | AliRezaTaleghani | :( |
| 04:47.25 | AliRezaTaleghani | https://wiki.asterisk.org/wiki/display/AST/Function_QUEUE_VARIABLES?focusedCommentId=10649975#comment-10649975 |
| 04:47.52 | AliRezaTaleghani | i search more, i don't know if i use this function in a wrong manner or not |
| 04:52.03 | ChannelZ | do you have setqueuevar turned on in queues.conf >? |
| 04:54.25 | AliRezaTaleghani | yes... |
| 04:54.34 | AliRezaTaleghani | i did it |
| 04:55.46 | AliRezaTaleghani | grep -i "^setqueuevar" queue* |
| 04:55.48 | AliRezaTaleghani | queues.conf:setqueuevar=yes |
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| 05:08.53 | ChannelZ | is it turned on specifically for the queue in question? |
| 05:09.24 | ChannelZ | (it's not a global option) |
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| 05:12.44 | epaphus | hello all. Anybody know of a mirror i can download asterisknow32.iso asterisk.org is still down |
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| 05:15.35 | engaged | Is it possible if you do an outbound call, then conference in someone else, the original person making the call out can disconnect to make another call out and have the second two people talk? |
| 05:21.27 | AliRezaTaleghani | ChannelZ: do you mean in my Queue config? it's mine |
| 05:21.29 | AliRezaTaleghani | http://paste.ubuntu.com/547641/ |
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| 05:24.41 | engaged | anyone? |
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| 05:45.17 | WIMPy | That was a lot harder than I thought. Looks like I'm getting rusty :-( |
| 05:48.35 | WIMPy | But I'm awake now :-) |
| 05:51.12 | engaged | Is it possible if you do an outbound call, then conference in someone else, the original person making the call out can disconnect to make another call out and have the second two people talk? |
| 05:51.43 | WIMPy | How do you do the conference? If you use MeetMe that's what will happen. |
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| 06:02.20 | engaged | WIMPy: I use five9, i just want to know if it is technically possible... |
| 06:02.41 | WIMPy | What is that? |
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| 06:07.36 | engaged | virtual call center |
| 06:07.44 | engaged | i jsut need to know if its possible... |
| 06:08.12 | WIMPy | Is that somehow related to Asterisk? |
| 06:08.37 | WIMPy | And I told you that it is what will happen if you use MeetMe. So: yes. |
| 06:20.05 | *** part/#asterisk epaphus (epaphus@201.199.62.74) |
| 06:20.07 | WIMPy | Damned security features. My phone won't accept my request to dial. |
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| 06:33.08 | ChannelZ | AliRezaTaleghani: Yes setqueuevar must be set in the queue, not in [general] |
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| 06:49.39 | AliRezaTaleghani | i think that is enough to define in queue template, ( instead of per queue ) i will check it and test again.... :) tnx |
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| 07:44.28 | p3nguin | wimpy: There's a setting in my Cisco SIP config to allow or not allow the two external parties to stay connected, when I hang up, after being joined by the conf key on the phone. The default is to leave them connected. |
| 07:50.59 | WIMPy | That was someone elses question. |
| 07:51.41 | WIMPy | I'd just like to know what securty feature I shouldn't have turned on in order to dial using an URL request to the phone. |
| 07:53.48 | sol326 | is asterisk.org down tonight? |
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| 07:54.20 | WIMPy | Only www. Issues and downloads are ok. |
| 07:55.20 | sol326 | what's issues sight? |
| 07:55.22 | sol326 | site |
| 07:55.36 | WIMPy | issues.asterisk.org |
| 07:58.52 | fofware | helo, I have a spa941 that don't boot, start to boot and stop after show Linksys A division .... on display any one know what can i do to debrick it? |
| 08:15.21 | drmessano | It never gets to "Initializing network"? |
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| 09:12.38 | tyman | Is it possible "bridge" music over the top of my ivr? |
| 09:24.15 | ChannelZ | AliRezaTaleghani: Probably, but you didn't do that. |
| 09:26.26 | AliRezaTaleghani | ChannelZ: yes, your right ;) i check it and all is working nice |
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| 12:17.44 | twanny796 | how can I get more info on notifymimetype? |
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| 14:00.31 | AliRezaTaleghani | L-) |
| 14:01.02 | AliRezaTaleghani | how can i execute a Function manually on asterisk console? |
| 14:01.06 | AliRezaTaleghani | is there anyway... |
| 14:01.09 | AliRezaTaleghani | ? |
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| 16:22.12 | *** join/#asterisk DrDamnit (~michael@highpoweredhelp.com) |
| 16:23.02 | DrDamnit | In asterisk 1.8.1.1, Getting: No SRTP module loaded, can't setup SRTP session. How do I load this module? Where is it? (I used to run the SRTP patch in 1.6, but this has been merged into 1.8.x.x). |
| 16:23.31 | WIMPy | module load res_srtp |
| 16:23.54 | WIMPy | But with the wrong client it can make your Asterisk crash. |
| 16:24.20 | DrDamnit | Not there. Do I need to select it during make menuselect? |
| 16:24.36 | DrDamnit | what do you mean wrong client. Wrong endpoints? I am 100% polycom. |
| 16:25.17 | WIMPy | I can't remember if it's selected by default. |
| 16:25.46 | DrDamnit | Running make menuselect now, it's not there. :-( |
| 16:26.30 | DrDamnit | Is it in branch and not release? |
| 16:26.37 | DrDamnit | I am running 1.8.1.1 release. |
| 16:26.41 | WIMPy | Then something else is missing. |
| 16:26.57 | WIMPy | Maybe you need a newer version of srtp? |
| 16:27.06 | DrDamnit | download from sourceforge? |
| 16:31.30 | DrDamnit | Got the package from sourceforge, and installed current. Still don't see it. I have obviously missed a step. Thoughts? |
| 16:31.56 | atan | http://pastebin.com/mV7EXqeb Am I diaialing wrong for Macedonia Skopje? |
| 16:32.03 | WIMPy | Doesn't menuselect mention anything? |
| 16:32.13 | atan | I don't see why that international call doesn't correct and says declined. |
| 16:33.15 | DrDamnit | WIMPy: No. Under Resource Modules, the last entry is res_rtp_asterisk. however, under /usr/src/asterisk1.8/res/res_srtp.c is present. |
| 16:35.31 | DrDamnit | atan: You have successfully contacted the remote server; however, you were expressly declined by their box. Not an asterisk problem unless it is a configuration problem with a user/pass, IP filter (on their side) or your account is not valid with them |
| 16:38.05 | atan | I thought it might have had to do with my number format. =( |
| 16:39.17 | DrDamnit | atan: It does not appear so. But you won't know unless you can confirm with the remote side. That box is specifically telling you "no" before it even tries to dial the number. Confirm you can (are allowed) to connect to them first, then worry about the number format. |
| 16:41.47 | DrDamnit | WIMPy: This is essentially what I did to get it working in 1.6. http://www.e164.org/wiki/AsteriskSRTP What step am I missing to get it working in 1.8? |
| 16:42.27 | WIMPy | The only thin I needed was the srtp package. |
| 16:42.54 | DrDamnit | hmmmm.... I am trying make clean && ./configure && make menuselect. We'll see what I get... |
| 16:43.24 | WIMPy | Oh yes, you need to ./copnfigure after changing the prerequisites. |
| 16:43.53 | DrDamnit | WIMPy: Fail. That didn't work. Are you using release (from asterisk.org) or trunk or branch? |
| 16:44.29 | DrDamnit | I AM SO RETARDED. |
| 16:44.33 | DrDamnit | I didn't scroll down. |
| 16:44.40 | DrDamnit | It's there. |
| 16:45.51 | DrDamnit | WIMPy: Got it working. Thanks for your help. |
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| 17:41.30 | epaphus | Anybody know of a mirror for asterisknow 32bit version ISO? Website has been down too much... or anybody care to send it to me... no torrents available online |
| 17:56.44 | ChannelZ | I don't think there are any |
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| 18:36.12 | thehar | dances |
| 19:04.05 | ChannelZ | takes pictures |
| 19:07.47 | thehar | sells them for money |
| 19:07.52 | thehar | gotta make christmas money back somehow |
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| 22:55.08 | dwery | tzafrir_laptop: hi. any news on asterisk 1.8 in debian? |
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| 23:39.06 | ChannelZ | just build it |
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