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| 00:06.12 | Max_E | Jinxed-: your dialplan |
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| 01:02.24 | WIMPy | <oquidave> what's the right procedure of upgraading!! |
| 01:02.31 | WIMPy | <DynamicFail> How out of date is asterisk 1.8.5 in terms of either MeetMe or |
| 01:02.36 | WIMPy | oops |
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| 01:26.19 | ChannelZ | apb1963: there might not be anything you can do about it, besides have a router in front of your modem that does bandwidth shaping so as not to fill up the modem's buffer. The cable company might be able to dick around in some advanced settings they lock off from the user.. |
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| 01:46.36 | Jinxed- | how would I place a call and put that call into a conference |
| 01:46.47 | Jinxed- | so a user dials 1234 which goes to my asterisk box |
| 01:47.07 | Jinxed- | I want my box to call 7777 which is over a trunk |
| 01:47.14 | Jinxed- | and place that call in a specified conference |
| 02:02.00 | Max_E | hmm sounds like a combination of execif plus originate |
| 02:03.43 | Jinxed- | I tried to do |
| 02:04.05 | Jinxed- | exten => 88,1,Orginate(SIP/1001,exten,conferences,555) |
| 02:04.13 | Jinxed- | so when a user dials 88 |
| 02:04.26 | Jinxed- | it would call user 1001 and try place them in the conference |
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| 03:11.58 | StanAccy | Does Asterisk 11.6 do anything with the MWI NOTIFY packet? |
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| 03:13.51 | file | I don't understand the question. |
| 03:14.29 | StanAccy | My voip provider has a message waiting . Its sent a UDP SIP NOTIFY packet to me, and its hitting the Asterisk server. |
| 03:14.41 | StanAccy | Im trying to get that message notification through to the IP phones on the other side of asterisk |
| 03:14.48 | StanAccy | so their red lights can light up |
| 03:15.13 | [TK]D-Fender | file: Answering in #freepbx |
| 03:15.23 | [TK]D-Fender | (since that's what he's using and where he first asked) |
| 03:15.48 | [TK]D-Fender | If you want to chime in, pop over there so it's all in one place |
| 03:17.27 | StanAccy | Thanks - didnt realize you were on here as well. |
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| 04:57.46 | StanAccy | how does asterisk process incoming NOTIFY packets for the MWI on a trunk SIP line? Im not subscribing to these notifications - my provider sends them. If I use the "mwi => " construct my provider respondes with "Met5hod not allowed" |
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| 05:09.06 | dar123 | how can i control RTP in asterisk |
| 05:09.25 | dar123 | if i want it to flow between endpoints or via asterisk |
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| 06:05.53 | leibert | ) |
| 06:05.54 | leibert | 22:17 < StanAccy> Thanks - didnt realize you were on here as well. |
| 06:05.54 | leibert | 23:57 < StanAccy> how does asterisk process incoming NOTIFY packets for the MWI on a trunk SIP line? Im not subscribing to these notifications - my provider sends them. If I use the "mwi => " construct my provider respondes with "Met5hod not allowed" |
| 06:05.57 | leibert | Day changed to 13 Dec 2013 |
| 06:06.00 | leibert | 00:09 < dar123> how can i control RTP in asterisk |
| 06:06.02 | leibert | 22:17 < StanAccy> Thanks - didnt realize you were on here as well. |
| 06:06.05 | leibert | 23:57 < StanAccy> how does asterisk process incoming NOTIFY packets for the MWI on a trunk SIP line? Im not subscribing to these notifications - my provider sends them. If I use the "mwi => " construct my provider respondes with "Met5hod not allowed" |
| 06:06.09 | leibert | Day changed to 13 Dec 2013 |
| 06:06.11 | leibert | 00:09 < dar123> how can i control RTP in asterisk |
| 06:12.13 | ChannelZ | thanks for that |
| 06:13.50 | fling | How to specify which contexts to use cdr? |
| 06:14.00 | fling | I want to log only few contexts! |
| 06:16.09 | Penguin | I don't even understand what you're saying. |
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| 06:18.36 | keanne | anyone using asterisk client over openvpn? client can ping the asterisk over openvpn but cannot register. any idea on what to check? |
| 06:22.55 | kaldemar | keanne: that the client packets go over the vpn, and whether a firewall is blocking sip. |
| 06:23.21 | fling | keanne: I'm using iax2 over openvpn |
| 06:24.27 | kaldemar | fling: CRD is not related to contexts like that, it's related to channels. you could put app NoCDR to all extensions that you don't want to log CDR but that's not really what you want is it? |
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| 06:25.03 | fling | kaldemar: yes, I want to put NoNoCDR in three contexts :D |
| 06:25.14 | kaldemar | keanne: also, what are the symptoms of "cannot register"? |
| 06:25.55 | kaldemar | fling: are you still trying to log a subset of calls with postgresql? |
| 06:26.13 | fling | kaldemar: just started doing this, created a db |
| 06:26.33 | fling | rebuildign wish postgresql support |
| 06:26.52 | kaldemar | you could just write to the db from dialplan. |
| 06:27.32 | kaldemar | or just let CDR log everything and select what you want to see. |
| 06:28.18 | keanne | kaldemar: thanks, right now it starting to look like firewall issue because zoiper softphone is just stuck in "registering" |
| 06:29.24 | kaldemar | keanne: look at sip debug on asterisk side when you try to register. you'll see if asterisk gets any messages. "asterisk -vvvr" to attach to CLI and "sip set debug on" to enable sip debug. |
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| 06:46.51 | keanne | kaldemar: thanks for the tip. it was both a firewall issue and a NAT issue. have to open asterisk IP from LAN side and NAT for the extension. thanks for the tip |
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| 07:00.31 | nunne | Does anyone have experience hooking up asterisk with CUCM? I have a client that needs to see it "working" before they can choose asterisk as a new platform. Basically I know nothing about CUCM and I have asked the guy administrating the CUCM (not to good at it...) to setup a trunk following a freepbx+cucm how-to. He can now call me - but if I try to call the CUCM i get BUSY/CONGESTION.. Which |
| 07:00.31 | nunne | probably means he doesn't have any dialplan rules for the trunk or something. Does anyone know how to, simplest possible way, route all trafic from 1 sip trunk in CUCM to 1 phone (so he can see that it works). Any help would be MUCH appreciated! |
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| 07:30.44 | fling | kaldemar: ok it is now logging all the calls. |
| 07:31.00 | fling | kaldemar: I will now try adding nocdr |
| 07:31.15 | fling | kaldemar: and will change it later with my next extensions rewrite ;> |
| 07:31.25 | fling | If it will be needed… |
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| 07:35.45 | fling | hmm hmm btw gosub stopped working after an upgrade |
| 07:35.47 | fling | WARNING[21338][C-00000010]: pbx.c:4621 in pbx_extension_helper: No application 'GoSub' for extension |
| 07:35.50 | fling | how to fix this? |
| 07:36.29 | fling | ah right not all the modules autoloaded |
| 07:38.45 | fling | copied this fix from another server with the same problem after upgrade > http://dpaste.com/1502783/ |
| 07:38.55 | fling | it was a while ago… |
| 07:46.53 | fling | cdr shows 'dst' as 's'; I need to rewrite it somehow… |
| 08:00.20 | fling | same => n,Set(CDR(dst)=${ARG1}) |
| 08:00.28 | fling | ^ this is not working, still getting 's' |
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| 08:01.59 | fling | -- Executing [s@u-kit-out:2] Set("SIP/306-0000001e", "CDR(dst)=+74952280412") in new stack |
| 08:02.09 | fling | do I need to rewrite it earlier? |
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| 08:05.57 | fling | ok tried before the first gosub, still nothing |
| 08:06.04 | fling | What am I doing wrong? |
| 08:07.35 | fling | Looks like I need to tune cdr….conf |
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| 08:09.16 | fling | something like this? > http://lists.digium.com/pipermail/asterisk-users/2009-January/225064.html |
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| 08:12.36 | fling | no, it does not help |
| 08:12.52 | fling | Am I not supposed to set this? |
| 08:15.41 | fling | Ok, I will try to fix this later. |
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| 08:39.51 | Faustov | if you ever have to do business with TATA - DON'T. You've been warned. |
| 08:46.31 | wdoekes | I would expect at least a blog post to back that statement up, otherwise you're just spamming the channel |
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| 08:57.42 | nunne | Anyone done any CUCM? http://picpaste.com/648152605fe54c39778ffcd577bfb200.JPG <- I get SIP 404 Not Found (UNALLOCATED NUMBER) trying to dial in to this. Should "Calling search space" really be empty? or is this the problem. |
| 08:58.13 | fling | kaldemar: how to >> select what I want to see? |
| 08:58.23 | fling | kaldemar: you mean when I access the db afterwards? |
| 08:59.51 | nunne | fling: can you really use Set(CDR(dst)=)?? isn't dst read only? |
| 08:59.58 | kaldemar | fling: yes |
| 09:00.30 | nunne | isn't only userfield, accountcode and amaflags settable? |
| 09:01.09 | kaldemar | yes, you can't write to dst. |
| 09:02.49 | nunne | fling: do you write CDR to SQL? because I have found it easier to just do "cdr-logging" using custom ODBC-functions than to use CDR-function due to all the read-only flags |
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| 09:10.52 | fling | nunne: oh! tell me more |
| 09:11.22 | Faustov | wdoekes: sorry for the amsg, I think I'm doing everyone a favor with this information ;) |
| 09:11.23 | fling | I'm putting it in postgresql |
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| 09:25.58 | skrusty | morning |
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| 11:07.02 | danjenkins | hey |
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| 12:06.11 | nunne | fling: read on http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/getting_funky.html |
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| 12:07.06 | nunne | fling: basically what you do is setting by custom ODBC-function on incoming call the date and all other info you need. Then using a gosub on Dial-answer.. and then h-extension to update the ODBC when call ends |
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| 12:10.58 | James87 | Hi, can anyone tell me if it is a security risk to enable auth_message_requests on a public asterisk 11 server? |
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| 12:33.00 | oquidave | hello, am having trouble with configuring asterisk with gtalk. It says Unable to create Jingle session on endpoint 'google' as no capable resource for target 'oloborama@gmail.com' was found here's my configs http://pastebin.com/wgN605Pq |
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| 13:10.20 | StanAccy | Can I take an external SIP providers MWI NOTIFY packet and signal asterisk on an existing mailbox for an extension (or group of extensions) ? |
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| 13:19.54 | WIMPy | StanAccy: AFAIK that's possible, yes. Did you read the sample sip.conf? |
| 13:20.15 | StanAccy | WIMPy: With the mwi lines? |
| 13:20.43 | StanAccy | Yeh, but that looks like Asterisk is trying to subscribe to notifications. My provider reeturns a "Method not allowed" on the subscription attempt. It sends the notifications regardless. |
| 13:21.14 | [TK]D-Fender | So you clearly don't need to subscribe |
| 13:21.27 | [TK]D-Fender | I gave you the 3 different places you could do this last night.... |
| 13:21.30 | StanAccy | so when the NOTIFY packet hits the asterisk server I dont know how to link those two events |
| 13:21.30 | Jinxed- | So I figured out two methods of inviting people to join a conference. One via orginate in the dial plan and one via .call files, but I haven't figured out a method other than CLI for kicking individuals or all members from a conference |
| 13:21.40 | Jinxed- | I'm using Asterisk 11 with confBridge |
| 13:21.49 | [TK]D-Fender | Jinxed-: Or CLI originate, or AMI originate..... |
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| 13:21.56 | Jinxed- | I saw something about AMI events, but I didn't understand how to use them |
| 13:22.06 | StanAccy | Ive got 2 extensions set up, 100 and 200. I want to signal extension 100 |
| 13:23.28 | WIMPy | Jinxed-: Events are sent from Asterisk. If yout want to tell Asterisk to do somethign you need an Action. |
| 13:23.34 | [TK]D-Fender | StanAccy: I have you the exact lines for this already.... |
| 13:23.47 | [TK]D-Fender | gave* |
| 13:23.51 | StanAccy | [TK]D-Fender: Those mwi => lines from the sample conf dont work |
| 13:24.00 | StanAccy | or there |
| 13:24.07 | StanAccy | or there's something else Im missing |
| 13:24.12 | [TK]D-Fender | StanAccy: Amongst which were ones that let you specify what box to attribute the messages to |
| 13:24.28 | [TK]D-Fender | StanAccy: that was ONE of the 3 different places I gave you |
| 13:25.24 | StanAccy | [TK]D-Fender: Just keep repeating the same thing over and over - that will help a lot |
| 13:26.15 | [TK]D-Fender | StanAccy: I listeds the line numbers specifically, what more do you want? One of them was the "MWI =>", the 2 others clearly weren't. |
| 13:26.53 | Jinxed- | WIMPy, do I use AMI events in the dialplan somehow? |
| 13:27.14 | [TK]D-Fender | StanAccy: if you look at the sample config for "mwi" yourself you'd seem then just the same |
| 13:27.25 | WIMPy | Jinxed-: No. You use them in external applications. |
| 13:27.34 | [TK]D-Fender | Jinxed-: Not via *. AMI is an external thing. You'd have to call a script. |
| 13:27.48 | [TK]D-Fender | Jinxed-: "call files" are probably your easiest option |
| 13:28.10 | Jinxed- | call files run ami events? |
| 13:28.18 | [TK]D-Fender | no |
| 13:28.28 | WIMPy | No. Completely different thing. |
| 13:28.56 | [TK]D-Fender | Jinxed-: just use call files for your origination needs |
| 13:30.40 | Jinxed- | Are there any suggested locations on how I might learn how to use AMI actions such as ConfbridgeKick even if it's via scripting |
| 13:30.46 | Jinxed- | is it like bash/python scripting? |
| 13:31.02 | StanAccy | I see a group of mwi => lines in tghe sample conf - i understand the syntax, but they dont work. Next is a mailbox=1234@SIP_remote line. That also doesnt work. It seems that mailbox= is set in the extension specific sections, so Im not sure why Id be setting this in the sip.conf (or equiv file for Freepbx) |
| 13:31.37 | WIMPy | Jinxed-: It would be pretty hard as a shell script. You need to do some socket programming. |
| 13:31.42 | StanAccy | and if you can find a google hit that talks about this that works Ill be happy to read it if I havent already read it |
| 13:32.02 | [TK]D-Fender | StanAccy: Show us the other attempts |
| 13:32.23 | Jinxed- | WIMPy, well I'm not opposed to coding, I just haven't had much luck on google finding any examples of this type of stuff |
| 13:32.29 | StanAccy | those are the two ion the sip conf file. I dont recall a 3rd |
| 13:32.30 | Jinxed- | call files will prob be a short term solution |
| 13:32.46 | [TK]D-Fender | StanAccy: "mailbox= is set in the extension specific sections" <- this is for * sending MWI |
| 13:32.47 | Jinxed- | I assume they can just run normal asterisk cli commands |
| 13:32.47 | StanAccy | I added the mwi lins in sip_general_custom.conf |
| 13:32.57 | [TK]D-Fender | StanAccy: Go do it and show us |
| 13:33.09 | StanAccy | what do you want to see> |
| 13:33.16 | file | the mwi capability requires the remote server allow subscribing to the mailbox which may or may not be supported by the remote SIP server, it's not a frequently used feature and it wouldn't surprise me if it's finicky - otherwise there is no capability to pass through MWI information |
| 13:33.20 | [TK]D-Fender | StanAccy: Exactly how you added it and where |
| 13:33.53 | StanAccy | file: When Asterisk triees to subscribe to my bvoip provider, the server replies with a 405 Method Not allowed packet. It sends the notifiy packets regardless |
| 13:34.12 | file | StanAccy, unsolicited NOTIFY messages are not supported with the mwi capability |
| 13:34.30 | WIMPy | Jinxed-: AMI is a very powerfull, elegant and fast way to control Asterisk. So if you don't mid some coding, move in that direction. |
| 13:34.44 | file | StanAccy, therefore... that won't work |
| 13:34.54 | [TK]D-Fender | StanAccy: That's fine,the 2nd method that * lists should work |
| 13:34.54 | WIMPy | Jinxed-: It's a simple text interface. You can try it manually by just telnetting in to it. |
| 13:34.56 | Jinxed- | WIMPy, I found a list of libraries... there are tons of them in many different languages |
| 13:35.19 | Jinxed- | are any more suggested than others (maintained, easier to use, etc) |
| 13:35.24 | [TK]D-Fender | StanAccy: So let's see exactly how you have done it (assuming it turns out the same as your first attempt) |
| 13:35.25 | Jinxed- | http://www.voip-info.org/wiki/view/Asterisk+manager+Examples |
| 13:35.29 | Jinxed- | what I was refering to |
| 13:35.38 | file | well, it may depending on what version of Asterisk |
| 13:35.47 | [TK]D-Fender | Jinxed-: How are you looking to trigger these call-outs? |
| 13:35.59 | file | unsolicited_mailbox option is what you want |
| 13:36.01 | WIMPy | Jinxed-: I haven;t tried any of them. Too many bad experiences with stuff that's supposed to make things easier, but in the end makes it harder or fial. |
| 13:36.08 | WIMPy | fail. |
| 13:36.11 | Jinxed- | Step 1: Via dialing. Step 2: Build a Web interface |
| 13:36.25 | file | wonders when that got added |
| 13:37.11 | Jinxed- | Step 1: Have a user dial a number, and it invites a predefined group of people to a conference. Then if a user dials a second number it kicks everyone from the conference. Step 1.5 Allow for custom invites via some sort of menu system |
| 13:37.15 | file | apparently I added it 4 years ago |
| 13:37.27 | [TK]D-Fender | Jinxed-: If your web interface is on the * box itself you could just use call-files. Beyond that, AMI |
| 13:37.29 | Jinxed- | Step 2: Build a Jquery Web Interface to allow a user to dynamically setup conferences |
| 13:37.33 | Jinxed- | and control them |
| 13:37.34 | file | or rather, committed the change |
| 13:37.46 | file | talks to self |
| 13:38.09 | [TK]D-Fender | file: I think of it as "a conversations with a very interesting person" ;) |
| 13:38.26 | [TK]D-Fender | And I'm just slapping "s" on everything this morning.... |
| 13:38.31 | Jinxed- | [TK]D-Fender, I guess call files are the way to start for now, with dialing |
| 13:39.10 | [TK]D-Fender | Jinxed-: or System + CLI originate if you don't need to pass anything interesting to it |
| 13:40.37 | WIMPy | Wait. Didn't I mention telnet? |
| 13:41.13 | WIMPy | Noone there to tell us it's evil? |
| 13:42.14 | *** join/#asterisk mjordan (~matt@75.76.55.191) |
| 13:42.15 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:42.16 | [TK]D-Fender | WIMPy: His script has a 24H reset timer... |
| 13:42.42 | WIMPy | So it's only evil once a day? |
| 13:42.46 | [TK]D-Fender | WIMPy: Which is what stops it from being obnoxious |
| 13:43.00 | [TK]D-Fender | WIMPy: So you're warned only once a day :) |
| 13:43.09 | [TK]D-Fender | WIMPy: It's evil 24/7 :p |
| 13:45.05 | StanAccy | file: Thanks |
| 13:45.22 | Jinxed- | WIMPy, I tried telnet |
| 13:45.25 | Jinxed- | It keeps crashing |
| 13:45.34 | Jinxed- | enabled it in manager.conf |
| 13:45.39 | Jinxed- | telnet to port 5038 |
| 13:45.40 | WIMPy | What crashes??? |
| 13:45.46 | Jinxed- | the telnet session |
| 13:45.48 | Jinxed- | or putty |
| 13:46.19 | StanAccy | and Im using Asterisk 11.6 |
| 13:46.20 | WIMPy | Well, so I guess putty is much more evil than telnet. |
| 13:46.37 | WIMPy | wonders how it's even ppossible for a telnet to crash. |
| 13:47.20 | Jinxed- | Do I just type commands like |
| 13:47.30 | Jinxed- | ConfbridgeListRooms |
| 13:47.36 | Jinxed- | once I'm telnet'd in |
| 13:47.47 | WIMPy | First you need the login action. |
| 13:47.59 | StanAccy | file: For each extension do I set their mailbox setting to myphonenumber@SIP_Remote (i.e. SIP_Remote, not the IP address of my voip provider)? |
| 13:48.21 | file | StanAccy, if you put unsolicited_mailbox=myphonenumber then yes |
| 13:48.24 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 13:48.53 | *** join/#asterisk danjenkins (~danjenkin@213.219.7.163) |
| 13:49.46 | StanAccy | so this is a trunk setting? |
| 13:50.14 | StanAccy | the sample sip.conf lists this setting under "sample definition for provider" - Im assuming thats a trunk defn |
| 13:50.37 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 13:51.08 | *** join/#asterisk imox (~imox@p57A96F54.dip0.t-ipconnect.de) |
| 13:51.38 | file | unsolicited_mailbox is only applicable to peer types. |
| 13:52.35 | StanAccy | I just removed the mwi => line I added in sip_general_custom.conf, reloaded sip and core, and I see a whole bunch of warnings about my SIP provider not implemeting SUBSCRIBE *(these are the 405 errors reported back). I didnt expect to see any subcsription attempts. What controls Asterisk trying to subscribe to remote Vmail? |
| 13:53.47 | [TK]D-Fender | Might require a full restart of * to flush |
| 13:53.55 | file | 'tis possible |
| 13:55.08 | StanAccy | ok, restarted. file: So my extensions are peers, correct? Are my SIP Trunks not also peers? |
| 13:55.14 | StanAccy | (they are listed in 'sip show peers') |
| 13:55.26 | file | then they are peers |
| 13:58.52 | StanAccy | should I have voicemail enabled on the extensions (in order for mwi to work), or is mwi independent of the voicemail? |
| 13:59.04 | file | I don't know FreePBx |
| 14:00.22 | [TK]D-Fender | StanAccy: If you want your phones to see that MWI you have to define it in the extension. |
| 14:00.42 | StanAccy | [TK]D-Fender: Using the mailbox= line ? |
| 14:01.21 | StanAccy | I added 555111222@SIP_Remote to the extension, and added the unsolicted_mailbox=555111222 to the trunk's section in the config |
| 14:01.37 | StanAccy | just waiting for a NOTIFY |
| 14:02.04 | Penguin | stanaccy: In asterisk, you set the mailbox in the sip peer entry in sip.conf; the extension isn't related to MWI. In anything other than Asterisk, this is the wrong place. |
| 14:02.27 | [TK]D-Fender | Penguin: Using FreePBX terminology on this one |
| 14:02.37 | [TK]D-Fender | Penguin: As that's what he's running |
| 14:03.17 | StanAccy | ok NOTFIY came in with a 1/0 count, but nothign being picked up here. |
| 14:05.13 | [TK]D-Fender | StanAccy: "help mailbox", from * CLI .... and show us the config generated for your extension, your trunk, and the debug of the NOTIFY attempt |
| 14:05.19 | StanAccy | for my unsolcited mailbox, do I need to add the remote server somehow? |
| 14:05.50 | StanAccy | no such command |
| 14:06.30 | file | the code matches to the peer entry to associate the NOTIFY with it, and to place the info in the unsolicited_mailbox provided - if the NOTIFY is coming from a different place it won't match and won't work |
| 14:06.47 | *** join/#asterisk dxd828 (~dxd828@host217-43-217-142.range217-43.btcentralplus.com) |
| 14:07.14 | StanAccy | file: The notify is coming from the same server Im using for normal communications |
| 14:07.21 | [TK]D-Fender | StanAccy: "mailbox show users" should do it |
| 14:07.31 | [TK]D-Fender | StanAccy: Show us |
| 14:07.50 | StanAccy | there's a "voicemail show users" |
| 14:07.57 | StanAccy | no mailbox command on this systemn\ |
| 14:08.06 | StanAccy | just the 1 entry: default 100 100 0 |
| 14:08.54 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.98) |
| 14:09.21 | Penguin | voicemail show users is what you're after. |
| 14:09.46 | file | that won't show you what you want, I don't think there's a CLI command for it |
| 14:10.39 | file | as [TK]D-Fender mentioned seeing the generated configs and a "sip set debug on" with the NOTIFY and such would be useful |
| 14:10.57 | StanAccy | working on it |
| 14:11.50 | *** join/#asterisk ehsjoar (~jonas@24.9.87.13) |
| 14:11.51 | [TK]D-Fender | file: The tally it's supposed to accept from the remote peer won't get listed as a "user"? |
| 14:12.14 | file | [TK]D-Fender, nah |
| 14:12.47 | file | the output there is configured mailboxes from the voicemail application itself |
| 14:13.18 | file | what the "virtual mailbox" stuff is is an event that gets published by chan_sip, and cached so other stuff can retrieve it |
| 14:13.20 | [TK]D-Fender | file: Shame it doesn't pool from the various places.... |
| 14:13.51 | file | it doesn't exist as a real entity, it's merely a snapshot of the state of some arbitrarily named thing |
| 14:14.32 | file | given the tools we have today it could be done better |
| 14:15.09 | [TK]D-Fender | file: Silly though.. would the mailbox dialplan apps work on it properly? |
| 14:15.12 | [TK]D-Fender | thought* |
| 14:15.19 | file | what mailbox dialplan app? |
| 14:15.32 | [TK]D-Fender | functions rather |
| 14:15.55 | [TK]D-Fender | Doubt "MAILBOX_EXISTS" would, here was a "count" one, no? |
| 14:16.43 | file | those are all exposed by app_voicemail and work against it |
| 14:16.55 | [TK]D-Fender | That would be a "no"... hrm |
| 14:16.58 | StanAccy | just waitinfg for a notify capture. |
| 14:17.13 | [TK]D-Fender | file: So no way to actually see it anyway. |
| 14:17.17 | [TK]D-Fender | anywhere* |
| 14:17.39 | file | you can't query it in the dialplan presently, it's merely for passing through the MWI indication |
| 14:17.44 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 14:19.16 | [TK]D-Fender | file: the fact it's using a "virtual context" would make it appear as being stored in memory, or is that something that check immediately against loaded SIP peers and forwards raw without storage? |
| 14:19.33 | file | the event itself is cached |
| 14:19.53 | *** join/#asterisk serafie (~erin@nat/digium/x-fsdzcqqojggpgrrw) |
| 14:19.54 | file | it gets looked up based on the mailbox and context |
| 14:19.54 | [TK]D-Fender | (not just SIP, but anything with a "mailbox=" I guess technically) |
| 14:20.28 | file | and then users of MWI either subscribe for events, query the cache, or failing that ask app_voicemail directly |
| 14:20.32 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:22.02 | *** join/#asterisk dar123 (~IceChat77@2.88.3.195) |
| 14:23.26 | StanAccy | what SIP packets do you want to see - just one of the NOTIFY ? |
| 14:23.33 | StanAccy | with the mwi indicator set on it? |
| 14:23.48 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 14:24.01 | [TK]D-Fender | that whole conversation, and a check for a while following to see if * forwards it on |
| 14:24.29 | StanAccy | I dont see anything in the logs related to a forward. |
| 14:24.35 | StanAccy | and I jsut tiurned the blasted thing off. |
| 14:24.46 | [TK]D-Fender | :| |
| 14:24.47 | StanAccy | I have a wireshark captur |
| 14:24.58 | [TK]D-Fender | * only... |
| 14:25.07 | [TK]D-Fender | Don' use external tools for this |
| 14:25.13 | file | a wireshark capture shows the traffic, it doesn't show some Asterisk stuff (such as chan_sip matching the peer) |
| 14:25.33 | StanAccy | understood - I have the SPAN setup and wired up so Ive been using it in parallel |
| 14:25.54 | StanAccy | ill get what I can from the console buffer |
| 14:26.25 | *** join/#asterisk BQffen (~BQffen@93-162-76-78-static.dk.customer.tdc.net) |
| 14:26.28 | [TK]D-Fender | You can get it all |
| 14:27.08 | BQffen | I am look for some help in outgoing SIP calls. Anybody up for that? |
| 14:27.09 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 14:28.24 | [TK]D-Fender | BQffen: Dive us details.... |
| 14:28.28 | [TK]D-Fender | Give |
| 14:28.34 | [TK]D-Fender | DAMN I can't type for beans today... |
| 14:29.11 | BQffen | I have a SIP trunk setup, and ingoing calls work fine |
| 14:29.13 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 14:29.27 | BQffen | But on outgoing call I gat: |
| 14:29.29 | BQffen | get* |
| 14:29.34 | BQffen | Failed to authenticate on INVITE to ... |
| 14:29.40 | BQffen | Call failed to go through, reason (8) Congestion (circuits busy) |
| 14:30.13 | [TK]D-Fender | ~pb |
| 14:30.14 | 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. |
| 14:30.15 | [TK]D-Fender | ^^^ |
| 14:30.38 | [TK]D-Fender | show us the complete call attempt with "sip set debug on" and "core set verbose 10" |
| 14:31.22 | BQffen | Roger, just a second |
| 14:32.00 | StanAccy | Ok, info here: http://pastebin.com/1eZQcPLM |
| 14:32.12 | StanAccy | its the XXXXXXXXXX line Im working on |
| 14:32.28 | StanAccy | ignore the "ignoreme" line. thats another account at the same provider |
| 14:34.06 | BQffen | D-Fender: Here you go: http://pastebin.com/vn8UFfy3 |
| 14:35.00 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 14:37.19 | [TK]D-Fender | StanAccy: I don't see you telling * to accept the MWI in the first place |
| 14:37.49 | StanAccy | unsolicited_mailbox 0- last line of file |
| 14:38.18 | StanAccy | I trunc'd it off the pastebin, my bad |
| 14:38.35 | StanAccy | unsolicited_mailbox=XXXXXXXXXX |
| 14:38.39 | [TK]D-Fender | BQffen: Reliably Transmitting (no NAT) to 193.105.190.241:5060: Contact: <sip:38403600@10.0.1.13:5060> <- your * is sending it's PRIVATE IP to the remote peer. You have not set your server up properly to work from behind NAT |
| 14:38.52 | Penguin | its |
| 14:39.07 | [TK]D-Fender | StanAccy: You're masking the first parts, excluding the other ciritical values and we've got little have faith in here... |
| 14:39.20 | StanAccy | Im masking the first parts of what? |
| 14:39.35 | StanAccy | all I did was search and replace the phone numbers for XXXXXXXXXX and ignoreme |
| 14:39.36 | [TK]D-Fender | XXXXXXXXXX |
| 14:39.58 | BQffen | D-Fender: I tried various combinations. With nat settings incoming calls seems to fail on hangup. Any clue on that? |
| 14:40.00 | StanAccy | I pasted the contents in to a text editor (SIP debug and sip conf) and did a search/replace and masked the passwords |
| 14:40.01 | Penguin | You mean that hiding the important part doesn't help? |
| 14:40.22 | StanAccy | wherever theres a XXXXXXXXXX insert phone number. |
| 14:40.29 | StanAccy | it all ties together |
| 14:40.30 | [TK]D-Fender | BQffen: You've done them wrong. * is advertising the wrong IP |
| 14:40.35 | Penguin | Some random phone number? |
| 14:41.05 | StanAccy | no, my phone number,. the one I use to register and send/receive calls with |
| 14:41.23 | Penguin | If you wouldn't have tried to make it so secretive, we could have avoided this entire part of the conversation. |
| 14:41.33 | StanAccy | I dnot understand why this is part of the conversation |
| 14:41.49 | StanAccy | the SIP debug trace was updated accordingly |
| 14:41.58 | file | config and signaling seems fine, without labbing it up myself and isolating things I can help no more |
| 14:42.01 | StanAccy | so everywhere the phone number was is now tghe XXX string |
| 14:42.19 | Kobaz | okay |
| 14:42.21 | Kobaz | question |
| 14:42.23 | StanAccy | file: That directed at me? |
| 14:42.25 | *** part/#asterisk mjordan (~matt@75.76.55.191) |
| 14:42.27 | file | yes |
| 14:42.31 | StanAccy | file: Thanks |
| 14:42.54 | StanAccy | should there be some debug out when * processes the NOTIFY? |
| 14:43.09 | Kobaz | if I do MusicOnHold(otherMusicClass) |
| 14:43.13 | StanAccy | assuming it had read the unsolicited_mailbox line correctly? |
| 14:43.28 | Kobaz | that's not supposed to change the default music class on the channel... is it? |
| 14:44.37 | Penguin | No, you set the channel's default class with CHANNEL(musicclass). |
| 14:45.09 | file | StanAccy, doubt it... but it should be sending back a 200 OK which is strange... |
| 14:45.27 | file | StanAccy, (since it's not upon a second look) |
| 14:45.43 | StanAccy | I stopped the sip debug trace shortluy after the notify was received |
| 14:45.51 | StanAccy | Ill regrab and check |
| 14:46.35 | Penguin | Set(CHANNEL(musicclass)=oneClass); MusicOnHold() would then play oneClass and MusicOnHold(otherMusicClass) would play otherMusicClass. |
| 14:47.14 | *** join/#asterisk zamba (marius@flage.org) |
| 14:47.36 | *** join/#asterisk davlefouAMD (~david@197.15.254.31) |
| 14:47.50 | StanAccy | Ive disabled the second trunk to reduce the debug noise |
| 14:51.05 | Kobaz | Penguin: yeah i know how that works |
| 14:51.10 | Kobaz | Penguin: i think i found a bug in MusicOnHold |
| 14:51.26 | BQffen | D-Fender: Got my PM? |
| 14:51.45 | Kobaz | like, if you Set(CHANNEL(musicclass)=default); MusicOnHold(otherMusicClass, 30); |
| 14:52.02 | Kobaz | and then you do a Bridge() to that music-ing channel |
| 14:52.17 | Kobaz | the CHANNEL(musicclass) is otherMusicClass and not default |
| 14:52.37 | Kobaz | so the original music class doesn't get reset back to 'default' |
| 14:53.02 | Kobaz | MusicOnHold(otherMusicClass, 30); should *only* change the music class for 30 seconds, *or* however long the application is running |
| 14:53.23 | Penguin | What if you wait until after the 30 seconds of otherMusicClass, when the dialplan moves on to the next step, and then Bridge the channel? |
| 14:53.35 | BQffen | D-Fender: nat seems to be configured now, but same symptoms. |
| 14:54.39 | Kobaz | Penguin: probably works as expected... i'll do some testing |
| 14:55.03 | Kobaz | pssing in otherMusicClass is a temporary setting (at least that's what the docs imply) |
| 14:56.00 | Penguin | I'd be interested in checking the CHANNEL(musicclass) in the next step after MusicOnHold(otherMusicClass,30) has finished. |
| 14:56.56 | Kobaz | yeah |
| 14:57.00 | Kobaz | it's probably 'default' |
| 14:57.04 | Kobaz | it should be |
| 14:57.53 | Penguin | But if it isn't, then it's definitely a problem. |
| 14:57.58 | StanAccy | ok, got a larger trace |
| 14:58.42 | StanAccy | I dont see any 200 responses from * to any of the NOTIFY packets. Only a 200 response to an OPTIONS packet |
| 15:02.50 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:02.50 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:03.06 | StanAccy | long SIP trace: http://pastebin.com/u2HTvHke |
| 15:03.17 | StanAccy | And yes, I masked the phone number with 2125551111 |
| 15:03.44 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.113) |
| 15:03.57 | StanAccy | since a sequence of X's was bad :-) |
| 15:04.15 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 15:06.43 | StanAccy | just reading the RFC 3842, is a SIP server required to ack with a 200 status to a NOTIFY? |
| 15:07.44 | StanAccy | or is that a standard SIP protocol thing? (that all messages are ACKd) |
| 15:09.35 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:09.36 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:09.58 | *** join/#asterisk Marverick (~angelslie@unaffiliated/marverick) |
| 15:10.06 | *** part/#asterisk Marverick (~angelslie@unaffiliated/marverick) |
| 15:10.09 | wdoekes | don't say ACK, because that creates confusion. but yes, all requests (e.g. notify) should get one or more responses (e.g. 200) |
| 15:11.13 | StanAccy | ok, so what might stop Asterisk form sending a Status: 200 OK response |
| 15:11.15 | wdoekes | but it depends on the contents of the notify request whether that should be a 200 or 481 or something else |
| 15:11.26 | StanAccy | its a MWI NOTIFY |
| 15:12.08 | StanAccy | SIP debug here: http://pastebin.com/u2HTvHke |
| 15:12.26 | StanAccy | Im only seeing * send 200 responses to the Request: OPTIONS packets |
| 15:12.41 | wdoekes | From: "VoiceMail Notifier" <sip:vmnotify@50.97.129.82> <-- no from-tag.. this request is dropped |
| 15:13.25 | StanAccy | from-tag? |
| 15:13.48 | wdoekes | look at the requests that *do* get a response |
| 15:14.54 | StanAccy | Ah, the OPTIONS has a from tag line under the From: section |
| 15:15.03 | StanAccy | the NOTIFY has a 'from address' instead |
| 15:16.05 | wdoekes | no |
| 15:16.21 | StanAccy | well, they both have that |
| 15:16.26 | wdoekes | the ;tag= bit |
| 15:16.54 | StanAccy | the penultimate OPTIONS poacket has a SIP from tag: 52ab20 |
| 15:17.10 | StanAccy | ok a couple of questions: |
| 15:17.26 | StanAccy | 1. is there a way I can get asterisk to tell me its ignoring packets (and for what reasons) ? |
| 15:17.38 | wdoekes | core set debug 20 chan_sip |
| 15:17.45 | StanAccy | 2. is there a way to work around this (or make asterisk assume a certain value for that tag) |
| 15:17.49 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:17.49 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:17.55 | wdoekes | pedantic=no |
| 15:19.00 | StanAccy | is there a non pedantic answer? :-) |
| 15:20.07 | wdoekes | grep pedantic /etc/asterisk/sip.conf |
| 15:20.32 | StanAccy | haha, its a setting? |
| 15:20.40 | cusco | yes |
| 15:21.12 | StanAccy | ill give that a try once Ive seen this debug log on the NOTIFY receipt |
| 15:22.06 | StanAccy | the colored debug logs are awesome |
| 15:22.23 | Katty | good morning cupcakes! |
| 15:23.55 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 15:25.02 | StanAccy | is pedantic a global setting or done on a per peer/trunk basis? |
| 15:28.24 | *** join/#asterisk newtonr (~newtonr@173-17-135-67.client.mchsi.com) |
| 15:28.27 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:28.33 | wdoekes | StanAccy: global |
| 15:28.38 | Penguin | The sip.conf sample shows what is valid where. |
| 15:28.58 | StanAccy | just looked it up now that ive seen the debug notice of the packet being dropped |
| 15:29.57 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 15:30.08 | *** join/#asterisk adeel (~adeeln@199.48.103.26) |
| 15:32.00 | adeel | after I've called the Transfer(SIP/foo@blah.com) in my dialplan, is there a way to cancel it, say if a timer hits? |
| 15:32.37 | [TK]D-Fender | [10:17]wdoekespedantic=no <- that's just for URI re-writes really... doesn't show you anything more. |
| 15:32.55 | [TK]D-Fender | adeel: There is no timer. its a redirect right off your server |
| 15:33.54 | StanAccy | huh, now * is replying to the NOTIFY with a "487 Bad Event" |
| 15:34.16 | adeel | [TK]D-Fender: ok, that's what i figured...as soon as the other side sends a 202 Accepted, the call control on * is gone |
| 15:35.21 | [TK]D-Fender | adeel: call = bai bai |
| 15:37.24 | StanAccy | is it normal that a Voip provider would have such a badly formed MWI notiftication? |
| 15:37.56 | StanAccy | I set pedantic=no, and restarted Asterisk. its still rejecting the MWI packets |
| 15:40.09 | *** join/#asterisk mjordan (~matt@nat/digium/x-wvnknknifyofdewz) |
| 15:40.09 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:41.49 | *** join/#asterisk kchehab (~kchehab@37.220.108.2) |
| 15:41.50 | kchehab | please anyone know in which part of RFC i can find the paragraph that says //billing will start when 200 ok is set from trunk or something like that |
| 15:42.07 | StanAccy | wdoekes: Can upi see another reason why that packet would be rejected? The chan_sip debug log doesnt say why its considered bad. |
| 15:42.52 | *** join/#asterisk Pullphinger (~Pullphing@c-24-13-69-42.hsd1.il.comcast.net) |
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| 15:48.45 | Katty | TOAD IN A HOLE. |
| 15:49.01 | WIMPy | There is a RFC that covers billing? |
| 15:49.38 | [TK]D-Fender | WIMPy: You'll find on the shelf next to the books on unicorns and Biggfoot :) |
| 15:49.56 | WIMPy | That makes sense. |
| 15:50.00 | Katty | and the other book is Aliens, from the historyc hannel |
| 15:50.07 | Katty | <PROTECTED> |
| 15:51.01 | wdoekes | StanAccy: what would you want asterisk to do with the notify? |
| 15:51.07 | wdoekes | it is an out-of-dialog request |
| 15:51.42 | wdoekes | [TK]D-Fender: ? did you see it change the behaviour of ignoring the request to replying with a 487 just now? |
| 15:52.23 | [TK]D-Fender | I've seen nothing of the sort |
| 15:53.15 | StanAccy | wdoekes: Im trying to get it to trigger the MWI on a phone ( based on this description: https://issues.asterisk.org/jira/browse/ASTERISK-17362 ) |
| 15:53.31 | [TK]D-Fender | And this is no longer worth my time... |
| 15:53.40 | wdoekes | shouldn't you send it to the phone then? |
| 15:53.53 | StanAccy | wdoekes: Im trying to get it to the phone |
| 15:54.19 | StanAccy | phone talks to Asterisk, Asterisk trunks to Voip provider (over NAT) |
| 15:55.05 | StanAccy | Voip sends unsolicited MWI notification, I now need Asterisk to receive that and set a flag on the mailbox that the phone is setup to monitor |
| 15:55.10 | *** join/#asterisk adeel (~adeeln@199.48.103.26) |
| 15:55.43 | mjordan | You want the external system to change the mailbox counts via a receive SIP NOTIFY in chan_sip? |
| 15:55.55 | StanAccy | im surprised (maybe I shouldnt be) that a VOIP provider doesnt send a MWI NOTIFY packet that Asterisk can handle |
| 15:56.09 | StanAccy | mjordan: I think so, yes. |
| 15:56.39 | StanAccy | Isnt that what ASTERISK-17362 patched? |
| 15:57.12 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 15:57.28 | mjordan | That patch is not in Asterisk |
| 15:57.30 | StanAccy | duh, I just read the end comments |
| 15:57.47 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:57.53 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 15:58.17 | StanAccy | so is there any way to get such unsolcited MWI notifications to light up the MWI indicator on a phone attached to asterisk? |
| 15:58.37 | Katty | chuckf: ping |
| 16:00.25 | pabelanger | StanAccy, define unsolicited |
| 16:00.39 | StanAccy | pabelanger: The voip provider is sending them without being asked |
| 16:00.56 | StanAccy | pabelanger: From what I understand there's also a scenario in which ASterisk tries to subscribe to such notices |
| 16:00.57 | pabelanger | StanAccy, it is possble |
| 16:01.04 | pabelanger | is the phone registered to the ITSP? |
| 16:01.07 | mjordan | not without patching Asterisk |
| 16:01.14 | StanAccy | pabelanger: ITSP? |
| 16:01.18 | mjordan | hum. I might stand corrected. |
| 16:01.19 | pabelanger | voip provider |
| 16:01.23 | file | mjordan, you do |
| 16:01.28 | mjordan | StanAccy: do you have unsolicited_mailbox set on the peer? |
| 16:01.36 | StanAccy | pabelanger: No, its registered to Asterisk. The stupid Cisco SIP firmware wont traverse NAT properly |
| 16:01.41 | file | ASTERISK-17362 is for supporting multiple mailboxes at an ITSP |
| 16:01.48 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 16:01.51 | StanAccy | mjordan: Yes |
| 16:01.56 | mjordan | hm. |
| 16:02.02 | StanAccy | but Asterisk is rejecting the packet as being bad |
| 16:02.14 | mjordan | 489 Bad Event? |
| 16:02.16 | StanAccy | chan_sip debug doesnt say why |
| 16:02.19 | StanAccy | mjordan: Yes |
| 16:02.50 | StanAccy | mjordan: I had to set pedantic=no in order for that to occur. Prior to that * was simply dropping the packet since there was no from-tag |
| 16:03.01 | mjordan | well, that's not a great sign, but okay |
| 16:03.11 | mjordan | Does the SIP peer have the "mwi" entry set? |
| 16:03.50 | file | no. |
| 16:04.14 | StanAccy | mjordan: No I dont see it |
| 16:04.27 | file | mjordan, that's not a peer option and is for subscribing for MWI |
| 16:04.35 | StanAccy | does bad request mean thart the packet is bad ior that Asterisk has no handling loop for that event? |
| 16:04.56 | StanAccy | and by bad, I mean badly formed or missing required data |
| 16:06.42 | file | if it's dropping with pedantic on it means it is badly formed, if a 489 Bad Event is being sent it means that 1. It's not matching the right peer 2. It's parsing it incorrectly |
| 16:07.01 | StanAccy | a 489 is being sent |
| 16:10.44 | ipengineer | Is it possible to have multiple parking lot contexts? Basically I don't want my parking lots to overlap. Customer A shouldn't be able to retrieve calls from Customer B so something like customer_a_park and customer_b_park |
| 16:11.11 | StanAccy | This is the SIP debug and chan_sip debug trace frmo the asterisk console: http://pastebin.com/tY1xw80P |
| 16:12.09 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 16:16.35 | Penguin | ipengineer: I seem to remember seeing that before. Check features.conf. |
| 16:16.38 | [TK]D-Fender | ipengineer: Yes, it's all in the sample config |
| 16:18.07 | ipengineer | Ok. I am seeing that in there.. I wasn't sure if they all used the default parking context on the backside.. I will try to add a couple. |
| 16:19.04 | Penguin | ipengineer: You can specify the parking context in the dialplan. |
| 16:19.30 | ipengineer | Alright. Thanks! |
| 16:22.32 | Penguin | I guess "parking context" isn't the right term for it. Parking lot name seems to be what they call it. |
| 16:23.18 | [TK]D-Fender | Well there is a dialplan context used by the lot... |
| 16:23.28 | [TK]D-Fender | So it is half of the story... |
| 16:23.40 | *** join/#asterisk generalhan_ (~generalha@about/windows/staff/generalhan) |
| 16:23.59 | Penguin | You can set the parking lot in the Park() app, set it by the PARKINGLOT variable, or with CHANNEL(parkinglot)... so you've got some options. |
| 16:26.37 | *** join/#asterisk bitfury (~bitfury@unaffiliated/bitfury) |
| 16:27.48 | ipengineer | Penguin: I was able to get this working with this: http://paste.laravel.com/1dZj my only other question, and I don't know if it is possible is how can I retrieve this call at the same extension? I basically want to set a BLF button to park a call, if that call is parked the BLF key will light up, they can hit that key again to retrieve it... Basically a replacement for SLA |
| 16:31.51 | Penguin | I think you're going to have to write some logic in the dialplan for this. I haven't created it myself, but I did an install for a company who had their line keys on the Polycom phones set up like that. |
| 16:34.01 | Penguin | My preliminary thinking is of something like the exten retrieves the call from the lot if there's call parked in it; or if there is no call parked there, park it there. |
| 16:34.28 | Penguin | I'd have to actually create dialplan for it and see if it makes sense and works. |
| 16:36.55 | *** join/#asterisk ipengineer_ (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 16:37.34 | *** join/#asterisk smirker (~x@CPE-123-211-202-192.lnse4.woo.bigpond.net.au) |
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| 16:45.19 | *** join/#asterisk JonMR (JonMR@2600:3c03::f03c:91ff:feae:9b1d) |
| 16:47.58 | *** join/#asterisk zamba (marius@flage.org) |
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| 16:54.05 | StanAccy | can I have more than 1 unsolicited mailbox (1 per configured trunk) ? |
| 16:54.06 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 16:54.57 | StanAccy | Ive got asterisk to process the NOTIFY packet, but when I enable the second trunk (with a second extension and 2nd mailbox), asterisk is sending the MWI to the wrong extension. |
| 16:55.41 | StanAccy | ive got: phone1@SIP_REMOTE and phone2@SIP_REMOTE configured for the two extensions. |
| 16:55.57 | StanAccy | and the unsolivited_mailbox=phone1 and =phone2 for the two VOIP trunks respectively |
| 16:56.47 | StanAccy | Asterisk receives the NOTIFY from the provider, sends back a status 200, then issues a NOTIFY to the phone, but it issues it on the 200 extension and not the 100. Voicemail was left on phone1 (which extension 100 is connected to via the routing) |
| 16:56.56 | StanAccy | and mailbox=phone1=SIP_REMOTE |
| 16:57.03 | StanAccy | have I missed something here? |
| 16:58.51 | file | that is what the patch previously mentioned adds the ability to do |
| 16:59.18 | StanAccy | ah, so I cant run two separate VOIP links with unsolicited_mailbox (as things stand right now) |
| 16:59.25 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 16:59.33 | file | not to the same provider |
| 16:59.41 | StanAccy | even though they are different accounts? |
| 16:59.44 | Penguin | You get one per peer, don't you? |
| 17:00.13 | file | it matches based on the source IP address and port, which would be the same despite having different accounts |
| 17:00.47 | StanAccy | I wonder if I can shift the SIP port for account 2 |
| 17:01.28 | *** join/#asterisk cancerstik (~Lingo@96.40.201.189) |
| 17:01.49 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 17:02.09 | StanAccy | file: That would work, right? |
| 17:02.20 | StanAccy | since its matching on IP + PORT |
| 17:02.32 | file | you can't force your ITSP to do that. |
| 17:02.37 | Penguin | Getting your ITSP to change the port is going to be tough. |
| 17:03.04 | Penguin | And by tough, I mean it's not going to happen. |
| 17:03.10 | StanAccy | I know I cant, but they might accept SIP traffic on multiple ports. I know they have port 5079 open |
| 17:05.22 | StanAccy | or better, whats involved in getting that patch referemced earlier in to the source tree? |
| 17:05.58 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:c443:540f:ed95:1602) |
| 17:06.16 | Penguin | You can add any patches you want to your own source tree before you compile. |
| 17:06.32 | StanAccy | Im referring to the mainline |
| 17:06.54 | StanAccy | seems like someone already did a bunch of working in analyzing and creating the patch files |
| 17:10.08 | StanAccy | butu they do have multiple SIP servers :-) |
| 17:14.08 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 17:14.39 | *** join/#asterisk hfp (~hfp@MTRLPQ0736W-LP130-01-2925268998.dsl.bell.ca) |
| 17:16.02 | hfp | Hi! I migrated to Asterisk 1.8 from ASterisk 11. One of my extensions is using System(echo -e "what to put\\nin .call file"). This was working fine before but now asterisk includes in the .call file the "-e" argument... What am I missing? |
| 17:21.18 | *** join/#asterisk moy_ (~moy@173.239.155.74) |
| 17:25.05 | hfp | This is the extension: http://pastie.org/private/wqunyskcvnnstj8orloow |
| 17:27.30 | *** join/#asterisk jonlewisorg (~Adium@gateway.wp.highwinds.com) |
| 17:30.23 | ipengineer | Penguin: Does this look right to you? I am trying to get the call to park in abcdefg parking lot but it looks like it keeps going to default for some reason.. http://paste.laravel.com/1e03 |
| 17:31.14 | ipengineer | This is the output from "parking show" http://paste.laravel.com/1e04 |
| 17:35.07 | jonlewisorg | what are people who were using meetme / web-meetme doing for an easy to use user interface for confbridge provisioning? |
| 17:38.40 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.127.113) |
| 17:39.31 | *** join/#asterisk vlad_sta_ (~vlad_star@91.206.59.134) |
| 17:40.19 | newtonr | hfp, that looks like a bug to me. File an issue on issues.asterisk.org/jira |
| 17:40.25 | [TK]D-Fender | ipengineer: Did you restart * after making your changes? |
| 17:40.32 | ipengineer | Yes |
| 17:40.56 | *** join/#asterisk zamba (marius@flage.org) |
| 17:41.10 | hfp | newtonr: Meh, I'll try and compile 11.6 from source. Thanks |
| 17:41.38 | newtonr | newtonr, I just tried latest SVN of 11 and got the same result |
| 17:42.20 | Penguin | ipengineer: Try parking_lot_name option in the Park(). |
| 17:44.31 | hfp | newtonr: Weird, it works on the 11.2.1 I was using |
| 17:44.46 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 17:45.23 | oquidave | anyone had problems with gtalk and asterisk...am getting an error "ERROR[7571]: chan_motif.c:2035 jingle_interpret_google_transport: Received Google transport information on session '2361049726' but ICE support not available" and yet my rtp.conf has icesupport=true |
| 17:46.16 | newtonr | hfp, perhaps a regression then |
| 17:46.23 | ipengineer | Penguin: Looks like either that or me completely wiping the file out and redoing it cleaned it up.. http://paste.laravel.com/1e0l |
| 17:47.07 | [TK]D-Fender | oquidave: Perhaps you didn't actually have compiled-in support... |
| 17:47.15 | ipengineer | Could I retrieve that call with: exten => 701,1,ParkedCall(701) |
| 17:47.49 | oquidave | [TK]D-Fender: i've chan_motif properly compiled. the module is loaded |
| 17:48.18 | [TK]D-Fender | oquidave: That is not implicitly ICE IIRC... go check menuconfig |
| 17:48.25 | Penguin | Normally you just send the call to extension 701 at the parking context and it retrieves without having to run ParkedCall(). |
| 17:48.58 | oquidave | [TK]D-Fender: you mean "make menuselect"? |
| 17:49.15 | [TK]D-Fender | oquidave: Yes... |
| 17:49.17 | dar123 | just to confirm, DAHDI is required only if i have analog interfaces? |
| 17:50.06 | WIMPy | DAHDI is required if you have hardware that needs it or to use MeetMe. |
| 17:50.19 | oquidave | [TK]D-Fender: okay, what's it under? what category? modules, applications |
| 17:50.22 | file | libuuid-dev (or equivalent on the respective Linux distribution) has to be installed or else ICE/STUN support will not be built |
| 17:50.50 | dar123 | for sip to sip calls i only need asterisk right? |
| 17:51.43 | ipengineer | Penguin: Yea, the only thing I get trying the ParkedCall is there is no extension parked there message |
| 17:52.20 | Penguin | ipengineer: I'd guess you're doing it wrong. You don't explicitly run ParkedCall() at all. |
| 17:52.58 | Penguin | The extensions to pick up are created automatically, so you just call the extension that has the call on it. |
| 17:53.29 | ipengineer | That is if parkext is set though correct? |
| 17:53.46 | ipengineer | Is there no way to explicitly declare, in the dial plan an extension and have it pickup that parked call? |
| 17:54.12 | [TK]D-Fender | [12:50]dar123for sip to sip calls i only need asterisk right? <- what does that technically mean for your processing of those calls? |
| 17:54.29 | [TK]D-Fender | ipengineer: "core show function parkedcall" |
| 17:54.57 | oquidave | [TK]D-Fender: i've [*] res_xmpp and [*] chan_motif compiled which gtalk depends on to work according to this doc https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 17:55.12 | Penguin | That command is probably going to fail. |
| 17:55.14 | file | oquidave, my comment was to you |
| 17:55.25 | [TK]D-Fender | ipengineer: "core show application parkedcall" |
| 17:55.39 | ipengineer | Command 'core show function parkedcall' failed. |
| 17:55.42 | [TK]D-Fender | Penguin: Yup, I am full of "oops" today... |
| 17:55.54 | oquidave | file: oh, checking it out |
| 17:55.59 | [TK]D-Fender | ipengineer: see above |
| 17:57.16 | oquidave | file: i've libuuid-devel - Development files for libuuid1 installed already |
| 17:58.10 | ipengineer | Penguin: That did the trick! Thanks.. The documentation online was old. I will have to remember I can access that from the CLI |
| 18:00.27 | file | oquidave, did you just build Asterisk? |
| 18:01.17 | nuken | does anybody already use Cisco SPA8800 as FXO gateway ? |
| 18:01.26 | nuken | i'm having some problems with outbound calls |
| 18:01.30 | nuken | they don't work |
| 18:01.40 | nuken | inbound works fine |
| 18:02.06 | oquidave | file: sure |
| 18:02.22 | nuken | for outbound i receive a message of circuits busy |
| 18:02.28 | file | is that a yes or no... |
| 18:03.13 | oquidave | file: i did built it |
| 18:03.25 | file | oquidave, I mean when... |
| 18:03.37 | oquidave | file: i built it yesterday |
| 18:03.50 | oquidave | file: am running asterisk 11.6 btw |
| 18:03.52 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 18:05.51 | file | when you run ./configure does it say: checking for uuid_generate_random in -luuid... yes |
| 18:06.21 | *** join/#asterisk apb1963 (~quassel@174.134.232.228) |
| 18:07.45 | oquidave | file: yes it does "checking for uuid_generate_random in -luuid... yes" |
| 18:07.52 | file | k |
| 18:08.36 | file | do... nm -a res/res_rtp_asterisk.so | grep pjnath |
| 18:08.41 | file | does it output anything? |
| 18:09.30 | oquidave | file: that's in the source dir, right? if so, it doesn't output anything |
| 18:09.49 | file | then it wasn't built with ICE support, interestingggggg |
| 18:10.05 | file | do make distclean, ./configure, make |
| 18:10.22 | file | and then try the above nm -a thing again and see if it outputs anything again |
| 18:10.55 | oquidave | file: okay |
| 18:11.01 | [TK]D-Fender | [13:02]nukenfor outbound i receive a message of circuits busy <- show us an actual complete call with SIP debug enabled |
| 18:11.03 | [TK]D-Fender | ~pb |
| 18:11.03 | 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. |
| 18:11.08 | [TK]D-Fender | ^^^ |
| 18:11.44 | [TK]D-Fender | nuken: that "message" alone doesn't tell us anything, and provide your peer setup for those ports & screenshots of the SPA side |
| 18:12.00 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 18:12.31 | oquidave | file: i have to step out a bit, but let do that i update on the results. thanks |
| 18:12.36 | Penguin | ipengineer: I'm interested in how that "Park" button was set up on those phones. On a 6-line phone, the 5th key said Park 1. If you had a call on the phone, you just hit the Park 1 button and it put the call into the parkinglot. All the phones shared that, so when you parked the call on one phone, the button lit up on the other phones. Press the button again to retrieve the call. |
| 18:14.26 | ipengineer | Yea that is what I am trying to do.. the only issue is v12 does not support event dialog yet. |
| 18:15.24 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 18:15.48 | Penguin | As soon as I can get something to eat, I'm going to think about making it work. |
| 18:16.25 | ipengineer | Also, instead of having one button they hit and it picks the "next" parking lot I want them to hit button 1 that is BLF for 701. If that lot is empty then the call will be parked there. If it is inUse then it will pickup the call. Going back to our scripting discussion earlier I think I can setup hints on the parking extension (which I already have done) and when the extension is dialed check to see if it is inUse, if so |
| 18:16.26 | ipengineer | the above. |
| 18:17.30 | ipengineer | The hints have to be setup anyways for BLF to work.. Does that make sense and am I answering your question? I think I have this going except for the scripting part.. Trying to figure out what cmd to use to check a hint status |
| 18:17.39 | nuken | [TK]D-Fender, this is the sip debug of a call -> http://pastebin.com/zhrWfrQG |
| 18:18.11 | Penguin | ipengineer: To check the hints, you have to subscribe to the extension. I don't know what settings are needed for your phone to do that. |
| 18:18.35 | ipengineer | Penguin: No I get that.. That is where I get this error: No registered subscribe handler for event dialog |
| 18:18.56 | ipengineer | Penguin: What I am saying is I don't know how to, from the dial plan check the status of a parking spot |
| 18:19.30 | [TK]D-Fender | nuken: SIP/2.0 404 Not Found <- your SPA doesn't know what to do with that number. You have misconfigured its dialplan from the look of things |
| 18:19.58 | Penguin | ipengineer: First thought, DIALPLAN_EXISTS() |
| 18:20.07 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 18:20.25 | [TK]D-Fender | nuken: Reliably Transmitting (no NAT) to 192.168.73.2:5060: <- also go verify what PORT you should be actually sending calls to to get to the FXO ports on it. Traditionally they reserve the lower ports for the FXS ports |
| 18:20.27 | Penguin | ipengineer: If the parkinglot hasn't parked anything there, the extension shouldn't exist. I believe those are created and destroyed dynamically. |
| 18:20.50 | ipengineer | Ok, perfect.. Let me try that. I am mocking a macro up and will share it with you momentarily |
| 18:20.54 | ipengineer | Are you fixing to go eat? |
| 18:21.47 | nuken | [TK]D-Fender, but the dial plan of SPA is a bit confuse |
| 18:21.57 | Penguin | I'll be here. I can eat and work. It just takes a few extra seconds to do things at the keyboard. |
| 18:22.04 | nuken | [TK]D-Fender, this is the print screen of spa http://postimg.org/gallery/tfi3d04/3ec9b0b4/ |
| 18:22.56 | ipengineer | Penguin: Ok.. |
| 18:27.29 | *** part/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 18:28.42 | StanAccy | on a SIP "Contact" header, what does the "user=phone" signify? |
| 18:29.16 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 18:30.58 | ipengineer | Penguin: Doesn't look like DIALPLAN_EXISTS unless I am doing it wrong: http://paste.laravel.com/1e12 |
| 18:31.11 | ipengineer | I also tried abcdefg_parked as the context |
| 18:31.31 | [TK]D-Fender | nuken: For the "VoIP-to-PSTN" section you didn't bother specifying any dialplan for it to use. |
| 18:31.35 | *** part/#asterisk jonlewisorg (~Adium@gateway.wp.highwinds.com) |
| 18:34.14 | Katty | chuckf: y u no stop by?! |
| 18:34.26 | Katty | chuckf: i haz srs business to discuss! |
| 18:34.52 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
| 18:36.54 | nuken | [TK]D-Fender, i've already try to specify one with ".xxx" but didn't work too... |
| 18:37.19 | nuken | [TK]D-Fender, do you have any idea how ths dial plan have to be ? |
| 18:37.38 | [TK]D-Fender | nuken: definitely no "." in front...and as I said you didn't set ANY |
| 18:38.50 | Penguin | ipengineer: Verbose(0,${IF($[${DIALPLAN_EXISTS(ParkingLotName,someExtension)}]?a call is parked:a call is NOT parked)}) |
| 18:39.25 | nuken | [TK]D-Fender, sorry, i did, Dial Plan 4: (xx.) and I specified in Voip-to-PSTN the number 4, but didn't work too |
| 18:39.27 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.113) |
| 18:39.41 | Penguin | ipengineer: crap, forgot the priority... |
| 18:39.49 | Penguin | ipengineer: Verbose(0,${IF($[${DIALPLAN_EXISTS(ParkingLotName,someExtension,1)}]?a call is parked:a call is NOT parked)}) |
| 18:39.55 | [TK]D-Fender | nuken: try it again. |
| 18:41.25 | *** join/#asterisk navaismo (~navaismo@189.241.100.102) |
| 18:42.22 | Penguin | ipengineer: I don't know if I'm thinking of going about this in the wrong way. I can probably build really good dial plan logic to make the determinations about if things are parked or not, but that doesn't mean it will be the best or the most elegant way to do it. |
| 18:42.58 | [TK]D-Fender | There is a standard option for a hint targeting a parked call already... |
| 18:43.04 | [TK]D-Fender | it's in the docs.. |
| 18:43.25 | nuken | [TK]D-Fender, same error again... SIP/2.0 404 Not Found |
| 18:43.40 | ipengineer | Penguin: So if I set the context to abcdefg_park then I get NOT parked, even if it is. If i change the context to abcdefg then I get it is parked even when it isn't. I think that is because of the hints setup: http://paste.laravel.com/1e15 |
| 18:44.04 | [TK]D-Fender | nuken: Show the new scrrens |
| 18:44.53 | [TK]D-Fender | ipengineer: clearly wrong context on the hint |
| 18:45.24 | nuken | [TK]D-Fender, that's just the part that i've changed http://postimg.org/image/fuepntqgd/full/ |
| 18:46.16 | ipengineer | [TK]D-Fender: IDK what it should be then because that is the only way it works. The state changes if the call is parked or not. Here is an example. I was parked then I hung up: http://paste.laravel.com/1e16 |
| 18:47.14 | Penguin | From what I can see, it's working properly. |
| 18:47.28 | Penguin | It shows InUse first, then Idle next time. |
| 18:47.38 | ipengineer | Penguin: So it looks like 701 is not adding to the dialplan when on a call: http://paste.laravel.com/1e17 |
| 18:49.09 | [TK]D-Fender | nuken: Why did you set FXS on that? |
| 18:49.18 | [TK]D-Fender | nuken: This has nothing to do with the FXS ports |
| 18:49.52 | nuken | [TK]D-Fender, ok i will change the FXS options and try again |
| 18:50.06 | nuken | i will put it in "none" |
| 18:51.00 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-qgvdgbpthgwfhstp) |
| 18:59.22 | wdoekes | hfp: did your OS upgrade at the same time, by any chance? |
| 18:59.28 | wdoekes | echo -e is not always portable |
| 18:59.53 | wdoekes | use printf instead (don't forget to add the trailing \n that echo adds put printf doesn't) |
| 19:01.05 | hfp | wdoekes: I changed system and distribution altogether in the move. Apparently using /bin/echo -e instead of echo -e fixes it |
| 19:01.23 | wdoekes | use printf in the future |
| 19:01.45 | hfp | Ok so I can use printf and add a trailing \n in whatever it is I want written to the file? |
| 19:02.23 | wdoekes | yes, instead of "echo -e 'abc'" you do "printf 'abc\n'" |
| 19:03.03 | *** join/#asterisk jblack (~jblack@107.1.75.186) |
| 19:03.14 | hfp | Understood, thanks |
| 19:03.42 | hfp | The reason being that printf is consistent across all ditributions but echo isn't, correct? |
| 19:03.47 | wdoekes | when you did echo (without bin) you got the builtin shell command |
| 19:04.02 | wdoekes | and you apparently got a new default shell (dash instead of bash, perhaps) |
| 19:04.31 | wdoekes | and the builtins differ in behaviour, as does the /bin/echo implementation accross distribution |
| 19:04.34 | wdoekes | +s |
| 19:04.53 | wdoekes | -c |
| 19:05.08 | hfp | I understand, thanks |
| 19:10.28 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:8d4:9259:afff:fe5b:1b90) |
| 19:10.30 | gusto | hey |
| 19:10.37 | gusto | americans fixed television! |
| 19:10.45 | gusto | Penguin: are you here? |
| 19:12.23 | nuken | [TK]D-Fender, thank you man ! it works on Line 1 perfectly ! |
| 19:13.15 | [TK]D-Fender | nuken: You're welcome. |
| 19:13.28 | nuken | [TK]D-Fender, now i'm having problems with Line 2, please, have a look in this log when I active the sip debug http://pastebin.com/UbrHNuAq |
| 19:13.55 | nuken | [TK]D-Fender, the debug is very big there is a lot of same messages |
| 19:14.22 | [TK]D-Fender | nuken: Three is no call in there at all |
| 19:14.24 | [TK]D-Fender | there* |
| 19:14.40 | nuken | [TK]D-Fender, there is no call active |
| 19:14.57 | nuken | [TK]D-Fender, when i try to make one call, circuits busy appears again |
| 19:15.01 | [TK]D-Fender | nuken: You are not showing me a failed call that ever touched your server |
| 19:16.38 | war9407 | [TK]D-Fender: curious btw, are you an asterisk developer? |
| 19:16.40 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 19:16.53 | [TK]D-Fender | war9407: Nope, just a user like everyone else. |
| 19:17.09 | *** join/#asterisk cancerstik (~Lingo@96.40.201.189) |
| 19:17.32 | war9407 | [TK]D-Fender: kk, cool :) |
| 19:18.25 | nuken | [TK]D-Fender, i have the circuits busy error when i try a call... but i can't show you the debug, because when i active it, a LOT of messages (like that in pastebin) start to appear in the console, |
| 19:18.45 | nuken | it's a lot of messages... |
| 19:19.10 | dumby | it often is, the folks here are good at parsing through it |
| 19:19.20 | war9407 | [TK]D-Fender: one other note, have you done integration before with SIP providers or hardware PBXs? Ive done some with SPA3102s and Google Voice via Obi110s - things like that and an IAX2 trunk for VoIP through the 3102 but im not sure Id say that's super advanced with asterisk? |
| 19:19.31 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.136) |
| 19:21.49 | war9407 | what would be considered an expert asterisk admin? |
| 19:27.43 | Penguin | The criteria are many. |
| 19:27.46 | navaismo | whisper: a dCAP |
| 19:28.39 | gusto | ha? |
| 19:28.50 | gusto | Penguin: how are your mirrors? |
| 19:35.37 | [TK]D-Fender | war9407: I've used analog cards, T1/PRI, SIP, IAX2 providers, analog gateways, etc, including the SPA-3000/3102 |
| 19:36.18 | [TK]D-Fender | war9407: the OBI's look too "cheap" and I wouldn't be on them being quality, espcially for FXO purposes |
| 19:37.56 | [TK]D-Fender | [14:21]war9407what would be considered an expert asterisk admin? [14:27]PenguinThe criteria are many. <- Generally should be able to get most answers from the docs & tarballs without having to ask. strong dilplan understanding, good networking background, and a moderate amount of SIP transaction-flow is a start |
| 19:38.34 | [TK]D-Fender | war9407: the concept of certs means little to me and many. Most are multiple-choice tests and who knows how useful a person is going to be when they aren't jsut tasked with one little thing |
| 19:54.10 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 19:55.54 | war9407 | [TK]D-Fender: thank you for this information |
| 19:56.05 | nuken | [TK]D-Fender, thank you man for the help ! all the problems that i had was related with ports... |
| 19:56.13 | nuken | [TK]D-Fender, thank you very much |
| 19:57.06 | [TK]D-Fender | nuken: You're welcome |
| 19:58.07 | [TK]D-Fender | war9407: As for SPA's being "super-advanced with Asterisk", no. They have a few tiny quirks in the setup of the device and your peer on *. Little more than setting up any other ATA |
| 20:01.13 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.125) |
| 20:01.25 | ipengineer | Penguin: Here is my final, working solution for parking. http://paste.laravel.com/1e27 |
| 20:21.43 | *** join/#asterisk jblack (~jblack@107.1.75.186) |
| 20:22.48 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 20:25.09 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 20:29.15 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 20:40.01 | *** join/#asterisk calum_ (~calum_@cpc4-harg5-2-0-cust371.7-1.cable.virginm.net) |
| 20:42.20 | StanAccy | Thanks for the assistance - Ive got it working now - using different SIP ports for the two SIP trunks (to the provider) |
| 20:43.22 | StanAccy | file: [TK]D-Fender: ^^ (and someone else that's gone offline) |
| 20:43.35 | file | ack |
| 20:52.40 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 20:57.40 | *** join/#asterisk bitfury (~bitfury@unaffiliated/bitfury) |
| 21:06.42 | Penguin | ipengineer: Line 4 has a discrepancy with lines 6/9. |
| 21:08.01 | ipengineer | Penguin: Yes, but for whatever reason that is the only way the hints work. The hints are bound to the Parking Context whereas the rest is bound to the Parking Lot |
| 21:08.16 | Penguin | ipengineer: And remember what "core show application ParkedCall", as well as I, said about using ParkedCall(). |
| 21:08.38 | Penguin | "You do not need to explicitly use this application" |
| 21:08.38 | Penguin | in your dialplan. Instead, all you should do is include the parking lot |
| 21:08.38 | Penguin | context in your dialplan. |
| 21:08.48 | Penguin | well, quote failure... |
| 21:08.59 | Penguin | But hopefully you got the message anyway. |
| 21:09.45 | ipengineer | Right, But I do not believe that will work like I am wanting.. I don't think it works the same way. Or are you saying include it, park a call via this method but retrieve them by letting the parking lot do its own thing |
| 21:09.53 | Penguin | I'm still going to try to build what I envisioned earlier. I got called away and didn't get to work on it like I wanted. |
| 21:11.41 | ipengineer | Ok, sounds good. Please share if you do get it going |
| 21:31.33 | gusto | Penguin: you still did not answer my question how your mirrors are, ha? |
| 21:40.19 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 21:40.56 | Penguin | ipengineer: I think I found a bug or two in Park(). Did you read the notes in core show application Park? |
| 21:41.10 | ipengineer | No. I was online looking.. |
| 21:41.41 | Penguin | They say Park() needs to be the first extension priority. |
| 21:42.02 | Penguin | That part makes it behave like a hint. |
| 21:42.12 | Penguin | I don't know if it actually does that. |
| 21:42.29 | Penguin | And the second note does not seem to pertain to Park() at all! |
| 21:42.37 | Penguin | More like ParkedCall() |
| 21:44.38 | *** join/#asterisk aarobc (~ac@asa-2.fbp.ore.fiber.net) |
| 21:46.33 | mjordan | Penguin: which documentation? |
| 21:46.51 | Penguin | core show application Park in 1.8 |
| 21:47.02 | Penguin | The notes. |
| 21:47.25 | Penguin | It says you do not need to use the application. |
| 21:47.30 | Penguin | Well, ya kinda do. |
| 21:47.39 | mjordan | well. Kind of. |
| 21:47.44 | mjordan | What are you thinking of doing? |
| 21:47.57 | Penguin | But the ParkedCall() app does not need to be called explicitly. |
| 21:48.23 | mjordan | Parking is some strange mumbo-jumbo. If you try to do things outside of what gets created in extensions.conf via features.conf, you may find gremlins |
| 21:48.50 | aarobc | Hi guys, I just installed asterisk 11, made sure I was including chan_motif when compiling, but when I run "module show like chan_motif" nothing comes up. I created the motif.conf file, restarted, still nothing. |
| 21:49.05 | aarobc | any ideas |
| 21:49.51 | mjordan | what does your log file say? |
| 21:50.01 | mjordan | and what happens when you load the module? |
| 21:50.36 | aarobc | mjordan: still new to this, what's the syntax for manually loading the module? |
| 21:51.24 | mjordan | module load chan_motif.so |
| 21:51.43 | ipengineer | Penguin: I am on v12 and not seeing that in the documentation unless I am missing something. |
| 21:52.09 | mjordan | ipengineer: parking was completely rewritten in 12 |
| 21:52.22 | mjordan | and it is now usable pretty much anywhere (with a few very minor caveats) |
| 21:52.45 | ipengineer | Right.. Me and Penguin have been discussing it off and on today. It is working very well for me in 12 |
| 21:52.56 | mjordan | yay! |
| 21:53.09 | mjordan | jrose_atDigium: ^^^ |
| 21:54.21 | ipengineer | Now if I can just get that event dialog going Ill be set ! [:) |
| 21:54.24 | aarobc | mjordan: https://gist.github.com/aarobc/940edca47a0045618bce |
| 21:54.42 | aarobc | I checked the directory, it does exist |
| 21:54.49 | file | aarobc, res_xmpp.so has to be loaded before chan_motif.so |
| 21:54.56 | jrose_atDigium | Always nice to hear stuff in 12 is working. *catching up* |
| 21:55.09 | mjordan | and if res_xmpp isn't loaded, that's a good sign that iksemel isn't installed |
| 21:55.23 | ipengineer | g2g.. Anniversary today.. Thanks Penguin file mjordan for the help today |
| 21:55.26 | file | chan_motif wouldn't build without it installed |
| 21:56.16 | aarobc | file: Okay, very good. Manually loading res_xmpp then chan_motif seems to work |
| 21:56.43 | aarobc | So now the question is how to make that autorun when asterisk starts |
| 21:57.44 | file | modules.conf controls module load order and explicit module loading |
| 21:58.43 | mjordan | If you use autoload in modules.conf, it should just work |
| 21:59.14 | aarobc | Good to know |
| 21:59.39 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:00.29 | *** join/#asterisk Katty (~angela@96-35-230-10.dhcp.stls.mo.charter.com) |
| 22:01.08 | Penguin | pokes katty a little |
| 22:02.02 | Katty | squeaks. a bit. |
| 22:02.14 | Penguin | *giggle* |
| 22:02.20 | Katty | how're you dear? |
| 22:02.35 | Penguin | The snow is falling on my head! |
| 22:02.43 | Katty | boo!! it's sleeting here. :< |
| 22:03.33 | Penguin | It was going sideways a minute ago. Lots of wind from the south. |
| 22:05.23 | Katty | i wanna go south :< |
| 22:05.35 | carrar | Hawaii! |
| 22:05.48 | Katty | YES |
| 22:05.51 | Katty | also, carrar |
| 22:05.57 | Penguin | What will you do in the south? Sun tan? |
| 22:06.00 | Katty | do you want xmas card this year? sorry if i already asked you...i just can't remember |
| 22:06.09 | carrar | There is a nice Cafe on Kona that is about as south as you can get |
| 22:06.12 | Katty | Penguin: defrost. |
| 22:06.16 | carrar | and stil be in the US |
| 22:07.48 | Katty | chuckf: and i wish you'd show up |
| 22:07.51 | Katty | chuckf: srsly. |
| 22:08.08 | Katty | chuckf: my luck you will show up 5 minutes after i mail all the cards out :< |
| 22:08.12 | carrar | Punalu'u Bake SHop |
| 22:08.24 | Katty | carrar: answer meee |
| 22:08.25 | carrar | "The Southern Most Bkery in the USA" |
| 22:08.29 | Katty | carrar: do you want an xmas card this year? |
| 22:08.31 | carrar | what? |
| 22:08.35 | carrar | YES |
| 22:08.39 | Katty | send me your address, yo. |
| 22:08.47 | Katty | or i'll ship your card to Penguin |
| 22:09.14 | *** join/#asterisk Weezey (~ohno@i.am.weezey.com) |
| 22:09.21 | Penguin | Then I'll get two! |
| 22:09.35 | Penguin | I sure hope they aren't the same. |
| 22:11.04 | Katty | https://scontent-a-ord.xx.fbcdn.net/hphotos-prn2/1454904_10100835625322327_1439881066_n.jpg <- here's some from last year |
| 22:11.10 | Katty | carrar might recognize his |
| 22:11.43 | carrar | heh |
| 22:13.29 | Katty | yes. that is a cactus in the window. |
| 22:13.35 | carrar | wife is just in the process of getting cards made |
| 22:13.37 | Katty | no, i do not live in the desert. |
| 22:13.43 | Katty | horay! |
| 22:15.52 | *** join/#asterisk jblack (~jblack@107.1.75.186) |
| 22:17.16 | Katty | chuckf: when you come back, go check your fb |
| 22:20.06 | Katty | hi james |
| 22:21.06 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 22:30.54 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 22:33.45 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:41.22 | *** join/#asterisk polysics (~Adium@50-192-47-77-static.hfc.comcastbusiness.net) |
| 22:41.27 | polysics | hello |
| 22:41.55 | polysics | for my test rig extension set, how would you recommend making an extension that never answers, but just rings for a while? |
| 22:42.01 | polysics | just Wait 30, hangup? |
| 22:42.21 | Penguin | You could do that. |
| 22:42.44 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 22:48.16 | outtolunc | Please hold while we never attempt to find an available agent. ;) |
| 22:49.13 | polysics | that's the idea |
| 22:49.54 | *** part/#asterisk aarobc (~ac@asa-2.fbp.ore.fiber.net) |
| 22:51.34 | jrose_atDigium | polysics: Always make sure to conclude the call with Playback(tt-weasels). |
| 22:51.34 | outtolunc | haha ok, goodluck with that |
| 22:52.36 | polysics | I am using that as "dialogue" |
| 22:52.51 | polysics | can I Goto with context only? |
| 22:53.04 | polysics | exten => _4XXX,1,Goto(default) or something |
| 22:53.13 | jrose_atDigium | No, if you only include 1 argument it has to be extension... but... |
| 22:53.14 | Katty | hello my asterisk does not work at all how to fix plz?? is urgent thx. |
| 22:53.44 | jrose_atDigium | what you can do is Goto(default,${CHANNEL(extension)},${CHANNEL(priority)}) or something like that |
| 22:53.45 | Penguin | polysics: No. |
| 22:54.08 | jrose_atDigium | not sure if that's what you would intend. |
| 22:54.08 | outtolunc | (it used to fallthru to s) |
| 22:54.09 | Penguin | polysics: Goto priority only, extension and priority, or context and extension and priority. |
| 22:55.20 | polysics | that should work in some form |
| 22:55.52 | polysics | Katty: you would need to provide at least a faint try at some actual info there. |
| 22:55.58 | Katty | orly |
| 22:56.17 | Katty | do i want to make bbq chicken pizza for dinner? or chicken parmesan? |
| 22:56.25 | Katty | ANSWER ME THAT, polysics |
| 22:56.26 | Qwell | yes to both |
| 22:56.33 | Katty | mister Qwell! |
| 22:56.35 | Katty | hugs Qwell |
| 22:56.38 | polysics | if you make either you have bad taste. |
| 22:56.51 | Katty | polysics: i hereby disown you. |
| 22:56.54 | polysics | none of those are actual Italian dishes :D |
| 22:57.12 | Penguin | Does it have to be Italian to be able to have it for supper? |
| 22:57.28 | Katty | newp |
| 22:57.31 | Katty | just tasty! |
| 22:57.45 | polysics | I guess not. Also, sorry at not catching the humor above. I am used to people that say those things for real. |
| 22:57.58 | navaismo | the echo application create the "echo" for the video too? |
| 22:58.05 | Katty | polysics: ^_^ |
| 22:58.13 | Penguin | VideoEcho() |
| 22:58.42 | Penguin | I think that would be called a mirror. |
| 22:58.58 | Katty | mirror mirror, on the wall. isn't navaismo the fairest of them all? |
| 23:00.09 | navaismo | "Echos back any media " |
| 23:00.19 | Penguin | espejo |
| 23:00.20 | jrose_atDigium | polysics: Sorry, a small correction to the above fake... it would be Goto(<context>,${EXTEN},${PRIORITY}) |
| 23:00.42 | Katty | Goto(kitchen,fridge,topshelf) |
| 23:00.52 | navaismo | yep i want to see me, to test the delay of the video |
| 23:01.41 | jrose_atDigium | if you try to go to the extension and it doesn't exist, you'll get dropped into i. |
| 23:01.48 | jrose_atDigium | If that doesn't exist, you'll get hung up. |
| 23:02.14 | jrose_atDigium | or... I guess it actually tries to run the h extension first and then hangs up. |
| 23:02.29 | jrose_atDigium | but that's all in the Goto documentation anyway. |
| 23:02.30 | Penguin | hangup, then h. |
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