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| 00:17.42 | Shaan | hey does vicidial havea channel on freenode? |
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| 00:29.11 | AnonGirl | Shaan: no :/ |
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| 02:55.28 | Milos|Work | using host and outboundproxy, I can register successfully, but seconds later it says registration timeout and retransmission timeouts |
| 02:55.36 | Milos|Work | any idea? |
| 02:56.13 | [TK]D-Fender | Look at the actual attempt |
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| 03:22.57 | ganbold_ | I'm trying to add authentication support to ooh323 and I may have some questions, whom should I talk to here? Does Alexander Anikin irc here? |
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| 03:52.18 | Milos|Work | [TK]D-Fender, ok, I resolved it by not using the outboundproxy given, but now I get 403 forbidden trying to make a call. Ideas? |
| 03:53.11 | [TK]D-Fender | Sounds like auth is wrong |
| 03:53.21 | Milos|Work | you mean my user/pass? |
| 03:53.24 | [TK]D-Fender | Of course I still see NOTHING... so that's just another guess |
| 03:53.26 | Milos|Work | but `sip show peers` says OK |
| 03:54.19 | [TK]D-Fender | That means nothing |
| 03:54.39 | Milos|Work | aweseoe |
| 03:54.49 | [TK]D-Fender | Other than they respond to SIP OPTIONS requests |
| 03:54.53 | Milos|Work | awesome |
| 03:55.00 | Milos|Work | so OK doesn't mean it authed? |
| 03:56.52 | [TK]D-Fender | <[TK]D-Fender> That means nothing |
| 03:56.58 | [TK]D-Fender | <[TK]D-Fender> Other than they respond to SIP OPTIONS requests |
| 03:57.48 | Milos|Work | awesome |
| 03:57.58 | Milos|Work | so `sip show registry` will tell me if it authed |
| 03:58.39 | [TK]D-Fender | .... |
| 03:59.12 | [TK]D-Fender | Registration is one thing it does NOT mean that a CALL going out your peer is going to work. |
| 03:59.29 | Milos|Work | great, well I can't possibly care about that at this stage because it says State: Rejected. |
| 03:59.29 | [TK]D-Fender | FORBIDDEN <----------- |
| 03:59.34 | Milos|Work | REJECTED <--- |
| 03:59.41 | [TK]D-Fender | Just because X works doesn't mean Y will |
| 03:59.46 | Milos|Work | indeed |
| 03:59.53 | Milos|Work | and I sure can't place a phone call without registering ;) |
| 04:00.00 | [TK]D-Fender | Depends |
| 04:00.07 | Milos|Work | right. |
| 04:00.16 | [TK]D-Fender | Registration isn't always a requirement |
| 04:00.34 | Milos|Work | in the 1% of cases |
| 04:00.39 | [TK]D-Fender | far from |
| 04:00.41 | Milos|Work | ic |
| 04:01.52 | [TK]D-Fender | Do you forsee a point in this process where you are actually going to show us the problem? |
| 04:02.26 | Milos|Work | the problem is pretty clear isn't it? |
| 04:02.33 | Milos|Work | state: rejected |
| 04:02.36 | Milos|Work | I have the wrong password or something |
| 04:02.38 | Milos|Work | can't read my handwriting |
| 04:02.44 | Milos|Work | need to verify password with ISP |
| 04:03.01 | Milos|Work | not sure if l or 1 or L etc. I did try all of those though |
| 04:03.18 | [TK]D-Fender | it may also be something else |
| 04:03.41 | Milos|Work | what do you have in mind |
| 04:03.46 | Milos|Work | they gave me an outbound proxy which I'm not using |
| 04:04.02 | Milos|Work | sorry, rather, they gave me a proxy which I'm not using |
| 04:04.05 | Milos|Work | but I am using ountbound |
| 04:04.36 | [TK]D-Fender | Sorry but I'm not up for guessing games tonight. |
| 04:04.50 | Milos|Work | indeed, I don't actually need anything from you at this point |
| 04:04.58 | Milos|Work | because I realised sip show registry said invalid registration |
| 04:05.05 | Milos|Work | so I'll confirm my details first, and ask if I have issues later |
| 04:06.11 | jeev | i'm so bored, i'm calling my hylafax server, then calling it again, then merging the calls and watching the faxes confuse eachother |
| 04:06.21 | Milos|Work | neat |
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| 06:18.49 | Milos|Work | [TK]D-Fender, with the correct user/pass, the response I get back is SIP 604 |
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| 06:34.30 | sofltech | Hi All - I'm using AMP with PHP to trigger an action via extension.conf. I'm using "Action: Originate" with proper settings and in fact it works (sort of). At the moment, for the action to be complete, the originating SIP account has to be logged in and pick up the "call" which than triggers the call to the extension. All I really want is to call the extension (virtually) without needing |
| 06:34.30 | sofltech | the SIP account to be logged in and pick up the call. |
| 06:34.42 | sofltech | I meant AMI not AMP... |
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| 06:59.05 | AnonGirl | win 150 |
| 06:59.06 | AnonGirl | whups |
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| 08:34.03 | tparcina | Tuju: I don't thnik that's the reason. :) |
| 08:34.14 | tparcina | We are having small periods of time (second, maybe two) when they don't hear the other person (or the other person doesn't hear them). |
| 08:34.20 | tparcina | We record all phone calls, and when I listen for the phone call, I can hear both speakers fine. |
| 08:34.29 | tparcina | I'm checking the network usage, and it's under 100 Mbit (tipicly 20-30 Mbit) on 1 Gbit network. |
| 08:34.57 | *** part/#asterisk apten (~apten@b2b-46-252-133-250.unitymedia.biz) |
| 08:35.26 | tparcina | Asterisk is on VM on XenServer, and that physical server has low processor usage (around 20%). |
| 08:36.27 | tparcina | VM server which runs Asterisk is on all 8 processors, and has the highest processor priority (all other VM's have normal priority). |
| 08:38.00 | tparcina | If it was processor issue, would I hear the recording fine? |
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| 08:42.03 | pchero_work | Hello, I just released asterisk-zmq module. It asterisk module working with zmq and json. https://github.com/pchero/asterisk-zmq |
| 08:42.58 | pchero_work | It still has a lots of bugs in there, but it would be useful if you want use zmq and json like an AMI. :) |
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| 09:11.24 | LoadDefaults | Is there a way to load the default settings for AsteriskNow? |
| 09:13.31 | LoadDefaults | I'm getting a mysql syntax error so I would like to start over but I no longer have physical access to the server |
| 09:14.09 | doop | why not just ssh in then |
| 09:14.54 | skrusty | pchero_work: interesting, will you support other queues? like rmq etc? |
| 09:16.07 | pchero_work | skrusty: Thank you for having interesting! :) But nope, I don't have a plan for that.. |
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| 09:16.41 | pchero_work | I'll move for next step. :) |
| 09:16.43 | skrusty | have you submitted it to the review board? |
| 09:17.01 | pchero_work | review board? |
| 09:17.13 | pchero_work | skrusty: I never heard about that, what is that? |
| 09:17.17 | skrusty | are you going to submit this as an offical patch? |
| 09:17.27 | pchero_work | I hope so. :) |
| 09:17.56 | pchero_work | skrusty: Could you tell me about that more? I'm really interesting about. |
| 09:18.01 | skrusty | maybe have a read of this then: http://www.asterisk.org/community/developers |
| 09:18.38 | pchero_work | oh, thanks! :) |
| 09:18.45 | skrusty | using review board, asterisk community can peer review your code |
| 09:26.21 | pchero_work | I tried to login at https://reviewboard.asterisk.org/account/login/?next_page=/dashboard/ . Btw, how can I create account for access here? It says no need to register, but, doesn't work. :'( |
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| 09:35.48 | pchero_work | I found it. here. https://signup.asterisk.org |
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| 12:08.15 | wasanzy | I have I have this in my sudo file nagios ALL=(ALL) NOPASSWD: /usr/local/nagios/libexec/ when anytime I run a plugin like this: sudo su - nagios /usr/local/nagios/libexec/check_asterisk_channels |
| 12:08.15 | wasanzy | <PROTECTED> |
| 12:08.22 | davlefou | Hi, how can i have opus codec under asterisk? |
| 12:08.27 | wasanzy | this is to monitor asterisk chanels |
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| 13:14.30 | alexises | Hi |
| 13:15.58 | alexises | I'm using with dahdi module, all works well except the last update that cause dahdi fail to work multiple time a day |
| 13:16.31 | alexises | I need to restart dahdi service and asterisk to solve the issue |
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| 14:05.32 | sofltech | Hi - getting this error: [Feb 20 09:00:55] ERROR[21848] utils.c: fwrite() returned error: Broken pipe |
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| 14:06.12 | sofltech | I have a U action (gosub) which on some conditions, terminates with an abort, this is when I get this error. |
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| 14:13.05 | davlefou | What are the good version of asterisk actually? |
| 14:13.48 | davlefou | i have the 11.8, is the 13.2 as good? |
| 14:14.17 | RPerre | davlefou, I'm using 13.2 and I have no problems |
| 14:14.45 | RPerre | been using 12 on other servers, also zero problems so far |
| 14:15.54 | davlefou | i ll try it |
| 14:16.16 | RPerre | davlefou, are you looking for features that asterisk 11 does not have? |
| 14:16.33 | RPerre | its allways a question of what you get by updating |
| 14:16.50 | [TK]D-Fender | sofltech: Show us an actual call doing this |
| 14:16.52 | [TK]D-Fender | ~pb |
| 14:16.53 | 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. |
| 14:16.54 | [TK]D-Fender | ^^^ |
| 14:20.28 | davlefou | Certified or not? |
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| 14:29.59 | sofltech | TK: will do so in a minute, thank you |
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| 14:33.27 | ldc | hello! is there some best practice for extension numbering? |
| 14:33.51 | ldc | e.g. I have a few users who have a phone both on their office desk, both in their house (via vpn), both as a softphone on their mobile |
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| 14:35.32 | sofltech | TK: I'm using AMI with PHP to originate the call, this is when I'm seeing this error, meaning, I don't think the issue is with the dialplan in the conf file. I don't see the error when I initiate the call directly from another extension. |
| 14:35.48 | mjordan | ldc: following the practices in A:TDG isn't a bad idea. |
| 14:35.56 | [TK]D-Fender | ldc: Only use odd numbers between 2000-4500. Everywhere is is ok to use even numbers too unless the value of the digits adds up to 8. That's never cool... |
| 14:36.17 | mjordan | [TK]D-Fender: how about only palindromes? |
| 14:36.22 | robmal | Also, avoid prime numbers. |
| 14:36.34 | robmal | Especially if they have 3 in them. |
| 14:36.39 | [TK]D-Fender | mjordan: Except on Tuesdays. |
| 14:36.52 | mjordan | that would be an interesting dialplan |
| 14:37.18 | ldc | haha |
| 14:37.38 | ldc | no, but I see this quickly turning into a mess. can't they do a SIP with multiple registries? :p |
| 14:37.59 | [TK]D-Fender | ldc: chan_pjsip supports this. |
| 14:38.05 | ldc | like different buttons on cme |
| 14:38.22 | [TK]D-Fender | ldc: And you've just made the first cardinal error of calling DEVICES "extensions" |
| 14:38.56 | [TK]D-Fender | ldc: and "extension" is a line of dialplan in extensions.conf |
| 14:39.04 | *** part/#asterisk mjordan (~mjordan@75.76.55.191) |
| 14:39.43 | ldc | [TK]D-Fender: my dialplan is pretty simple at the moment, it's a lookup table between the registered sip user and the extension associated to it |
| 14:41.35 | ldc | I was thinking of ring groups, maybe, and giving out usernames for the devices instead of numbers |
| 14:42.02 | [TK]D-Fender | ldc: mixed alpha-numeric is best |
| 14:45.35 | ldc | I'll also look into pjsip |
| 14:45.36 | ldc | thanks :) |
| 14:46.05 | sofltech | looks like adding async:yes solved the problem... would be good to know why |
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| 15:19.17 | crised | [TK]D-Fender: Hi there, all my problems were fixed, when I chose to use an outbound proxy and expire the connection every 60s |
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| 15:30.21 | [TK]D-Fender | sofltech: Keep it in-channel |
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| 15:56.46 | *** join/#asterisk War_Bear (~War_Bear@shiftingshadow.warbear.co.uk) |
| 15:58.01 | War_Bear | Hi guys, I have a weird situation. I have a very basic asterisk install that is just accepting all sip calls and sending them to another server. This works fine. However when the calls get over 200 concurrent the box stops accepting new calls and the CLI starts to act weird. Like "core show channels" never completes and I cant tab complete?! any ideas? |
| 15:58.16 | War_Bear | also asterisk wont stop with core stop now |
| 16:02.07 | [TK]D-Fender | kill the process |
| 16:02.17 | sofltech | TK: thanks, that was a mis-click :-) |
| 16:02.21 | [TK]D-Fender | You don't have to go through * CLI to kill it.... |
| 16:02.48 | War_Bear | I have killed it. When I restart its the same and wont die. I assume maybe a permission issue? |
| 16:03.39 | sofltech | TK: I think I figured it out :-). In my php code I wasn't waiting for all of the responses from Asterisk and so the connection was closing permaturely. Surprised that I didn't see any examples talking about this or showing how to do it in PHP - do you know of any? |
| 16:04.19 | sofltech | TK: any URLs with examples. At the moment I'm just looping through "fgets" to the socket until its done. |
| 16:05.08 | pabelanger | War_Bear, out of file descriptors? |
| 16:05.44 | pabelanger | ulimit -n |
| 16:05.48 | *** join/#asterisk mirela666 (~mirko@212.200.146.242) |
| 16:06.10 | War_Bear | pabelanger that returns 65535 |
| 16:06.38 | pabelanger | make sure you run the command as the same user you are starting asterisk as |
| 16:06.48 | *** part/#asterisk mirela666 (~mirko@212.200.146.242) |
| 16:07.02 | War_Bear | yeah I did it as the user asterisk |
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| 16:25.08 | t4nk722 | hello |
| 16:25.32 | t4nk722 | I need help please |
| 16:26.18 | pabelanger | so say we all |
| 16:26.23 | pabelanger | ~ask |
| 16:26.23 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:26.26 | *** join/#asterisk RPerre (~RPerre@alpes.nortenet.pt) |
| 16:27.03 | [TK]D-Fender | Weasels hate meta-questions..... |
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| 16:29.13 | t4nk722 | i've 3 sip account in a group like this dial(sip1&sip2&sip3,120,rtT) but if 1 sip account is not registred then the incoming call go to nobodyavailable |
| 16:29.51 | [TK]D-Fender | t4nk722: Shouldn't. Show us a complete call. |
| 16:29.59 | [TK]D-Fender | t4nk722: with SIP DEBUG enabled |
| 16:30.01 | [TK]D-Fender | ~pb |
| 16:30.01 | 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. |
| 16:30.03 | [TK]D-Fender | ^^^ |
| 16:30.45 | t4nk722 | ok :) |
| 16:31.56 | War_Bear | What file permissions should astdb have? |
| 16:32.44 | t4nk722 | what is astdb ? |
| 16:32.51 | [TK]D-Fender | War_Bear: -rw-rw-r-- 1 asterisk asterisk 110592 Feb 20 16:32 astdb |
| 16:32.57 | [TK]D-Fender | War_Bear: on mine. |
| 16:33.00 | War_Bear | Thanks |
| 16:33.09 | [TK]D-Fender | t4nk722: ... |
| 16:33.13 | [TK]D-Fender | ~book |
| 16:33.13 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:33.15 | [TK]D-Fender | ^^^ |
| 16:33.21 | [TK]D-Fender | Asterisk Database |
| 16:33.30 | *** join/#asterisk alexscott (~alexscott@2a01:6380:fffd:1:5553:48c:4fa6:e6b2) |
| 16:33.40 | [TK]D-Fender | read up after providing what was requested.... |
| 16:35.59 | newtonr | t4nk722, https://wiki.asterisk.org/wiki/display/AST/Asterisk+Internal+Database |
| 16:36.23 | t4nk722 | Thanks |
| 16:51.17 | t4nk722 | my database persissions is -rw-r--r-- |
| 16:58.19 | t4nk722 | quit |
| 17:01.12 | War_Bear | Thanks all |
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| 17:26.42 | liquidamber | greetings - on https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages - will this not work with centos 7? |
| 17:28.13 | [TK]D-Fender | liquidamber: http://packages.asterisk.org/centos/ |
| 17:28.19 | [TK]D-Fender | liquidamber: No C7 yet |
| 17:31.20 | liquidamber | [TK]D-Fender, gracias |
| 17:34.36 | saint_ | what is going on people ? |
| 17:34.40 | saint_ | long time no talk |
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| 18:12.31 | *** join/#asterisk sofltech (~sofltech@c-76-108-216-19.hsd1.fl.comcast.net) |
| 18:14.10 | sofltech | I have a php script which retrieves a list of phone numbers, between 1 and 6 numbers. I've configured a dialplan which I can activate for a single number (passed via a parameter). What is the best way to pass the list of numbers to asterisk and loop through them? |
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| 18:22.33 | [TK]D-Fender | sofltech: please elaborate on your description. |
| 18:23.10 | [TK]D-Fender | sofltech: "activate" means nothing specific here, nor does "passing" Asterisk a list. You aren't being clear as to when things actually take place and how. |
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| 18:34.53 | sofltech | td: let me try more clearly. I need to iterate thruogh a set of phone numbers and execute a dialplan on each one in sequence, only if the previous dialplan failed (no answer or no acknloedgement by the user). I already have the dial plan set up and I can get it to run for a single number. |
| 18:35.38 | sofltech | td: I have a php script set up which can loop through the numbers and can initiate the dial plan on each number via ami. |
| 18:36.34 | sofltech | td: the problem is that I don't know how to determine if the dialplan succeeded or failed (return a value to the php script) and based on that value, do the same for the next number or quit. |
| 18:37.19 | sofltech | td: alternatively, I could send the numbers via a comma delimited parameter to the script and have it iterate through all of the numbers where I have more control over the status / flow. |
| 18:37.57 | sofltech | td: I also read that I could execute a odbc readsql call within asterisk and retrieve multiple rows and fetch the data, so that may be an option as well. |
| 18:39.05 | sofltech | td: I'm trying to understand which of these options is considered "best practice" or best for this case. |
| 18:40.12 | sofltech | td: one last thing, if all the numbers fail, the whole process should start again after 10 minutes - for the same group of numbers. I'd either have to "cron" that somehow via php or maybe the dialplan can handle seeting idle via a function for 10 minutes and restarting the process. |
| 18:43.04 | [TK]D-Fender | If your PHP is already listening on AMI then have your dialplan send out a USEREVENT to AMI to tell your script the result. |
| 18:43.28 | sofltech | td: thanks, I'm going to look into that right now. |
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| 19:05.48 | sofltech | tk: I'm getting a gosub "popping routine return locations" error on my script. I don't think I've got the gosub returns set up correctly, could you have a quick look? see: http://pastebin.com/i6HJFv5V thanks. |
| 19:06.08 | sofltech | tk: i was able to add the uservalue and I think that will work. |
| 19:06.28 | [TK]D-Fender | You didn't show the error.... |
| 19:06.42 | [TK]D-Fender | For all I know your problem has nothng to do with the dialplan you're showing me. |
| 19:07.38 | [TK]D-Fender | sofltech: same => n,WaitExten(3) <-NEVER use waitexten to make IVR's in a Dial gosub like that. |
| 19:07.53 | [TK]D-Fender | sofltech: use Read() |
| 19:08.25 | sofltech | tk, thanks, this is the actual error: [Feb 20 13:58:13] NOTICE[26016][C-000000c1] app_stack.c: SIP/ilan-000000ed Abnormal 'Gosub(subAlert,s,1(s,1))' exit. Popping routine return locations. |
| 19:09.10 | [TK]D-Fender | And showing the error doesn't show WHERE it happened. What occurred before, and what occurred after |
| 19:11.10 | sofltech | ok, how can I tell where it happened? |
| 19:11.34 | sofltech | also, I'm not clear on where you would use WaitExten vs. Read? |
| 19:11.56 | sofltech | I get it that read is more generic, but what is the issue with WaitExten? |
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| 19:12.19 | sofltech | do I still use background as well? |
| 19:12.29 | sofltech | all I want is for them to press 1 to confirm. |
| 19:27.07 | [TK]D-Fender | [14:12]sofltechdo I still use background as well? <- no. |
| 19:27.19 | [TK]D-Fender | do not use anything that will jump out of that exten. |
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| 19:48.37 | sofltech | tk: that extension or that context or both? |
| 19:49.32 | [TK]D-Fender | sofltech: For the context you are Gosub-ing for your dial |
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| 20:48.21 | Jacoby6000 | does AddQueueMember support negative numbers for penalty, so that they function more as priority than penalty? |
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| 21:14.26 | pabelanger | Jacoby6000, I _think_ so.... I'd have to check some of my old dialplan |
| 21:26.32 | *** join/#asterisk tristan-mei (tristan@unaffiliated/tristan-mei) |
| 21:28.25 | tristan-mei | I'm having an issue where once a caller is placed on hold and picked back up, outbound audio stops working - we can hear them but not vice versa. This is a new issue, and we haven't made any networking/firewall changes since it was working properly - does anybody have a suggestion as to where I should start looking? |
| 21:33.03 | Geriatrix | tristan-mei: i woudl say firewall - start tracing packets - is your direct media on ? |
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| 23:28.07 | *** join/#asterisk Sudravirodhin (~sudraviro@162.218.22.124) |
| 23:29.21 | Sudravirodhin | Hi all. Quick question, answer when convenient: is it possible to have automixmon record on a call that's been bridged with another using the atxferthreeway option? I tried and get an error saying it can't identify peer. Any help is appreciated, thanks. |
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| 23:30.54 | Sudravirodhin | Latest Asterisk 13, btw. Sorry for double messages. |
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| 23:33.47 | jmetro | ~book |
| 23:33.47 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 23:34.47 | mjordan | Sudravirodhin: nope. The features stuff that wants to be invoked on a peer only works in a two party bridge. |
| 23:34.51 | jmetro | are there any up-to-date resources for using asterisk realtime? i wish voip-info would diaf. |
| 23:34.54 | mjordan | It has no way to know who you think the "peer" is |
| 23:35.08 | mjordan | jmetro: from what perspective? |
| 23:35.34 | jmetro | configuring the tables within realtime, for post-1.6 |
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| 23:35.47 | jmetro | have all the connections established, getting peers across, just trying to do the VM setup now |
| 23:35.56 | Sudravirodhin | mjordan: Thanks, that's what I was afraid of. Do you have any suggestions on how to remedy this? I have a client who needs to record a bridged call and it looks like ConfBridge is my only solution. |
| 23:35.58 | mjordan | jmetro: what version of Asterisk? |
| 23:36.01 | jmetro | 11 |
| 23:36.08 | mjordan | Sudravirodhin: use ARI! |
| 23:36.12 | mjordan | you can record the bridge natively |
| 23:36.51 | mjordan | jmetro: you can probably use the Alembic scripts from 13 to get the schemas, although the schemas should be in 11 as well (just not in as nice of a management form) |
| 23:37.28 | jmetro | If i just update to 12/13 would this be easier? |
| 23:37.36 | jmetro | Im still developing the system so i am open to it |
| 23:38.07 | mjordan | well, you can use Alembic to manage the databases in 13 |
| 23:38.16 | mjordan | but that's just DB schema management |
| 23:38.23 | mjordan | the act of configuring Asterisk for ARA hasn't changed |
| 23:38.35 | Sudravirodhin | mjordan: That would take control of it away from the user, though. The agent making the atxferthreeway call needs to have control of recording, see. |
| 23:38.56 | mjordan | they still can. You would have to implement the DTMF handling to invoke it |
| 23:39.08 | mjordan | but the user can still trigger the bridge recording |
| 23:39.22 | Sudravirodhin | So a DTMF function to invoke the ARI to record the current bridge? |
| 23:39.39 | Sudravirodhin | Just making sure I understand. |
| 23:39.41 | mjordan | er |
| 23:39.58 | mjordan | not exactly. If you go down this path, you'd really be replacing Dial. |
| 23:40.02 | jmetro | I started this project about a year ago, so maybe i should just update to 13 now |
| 23:40.17 | mjordan | you can't really merge the features/dial code in Asterisk with ARI. |
| 23:40.31 | mjordan | but if you need custom features that go beyond what Asterisk can provide, ARI is the way to go. |
| 23:41.18 | Sudravirodhin | mjordan: Yikes. Maybe I can have a feature mapping for a custom confbridge that records until it's empty. Would that be viable or not recommended? |
| 23:41.33 | mjordan | you could do that |
| 23:41.59 | mjordan | The only problem with using ConfBridge is that it makes your two party bridges more "expensive" |
| 23:42.16 | Sudravirodhin | mjordan: CPU-wise? |
| 23:42.19 | mjordan | yes |
| 23:42.23 | jmetro | any reason people prefer PJSIP over SIP? |
| 23:42.49 | mjordan | it doesn't switch to a more optimal bridging model, it assumes that it must use bridge_softmix at all times |
| 23:43.11 | Sudravirodhin | jmetro: I prefer it because you have fine-grain control over the peer versus SIP, plus you can have multiple AORs so no need for "cellphone" peers. That's my experience, though. |
| 23:43.47 | Sudravirodhin | mjordan: I'll have to see how frequent these kinds of calls would be made, then I'll make a judgment as to whether or not I'll pursue that implementation. Thanks much. |
| 23:43.51 | mjordan | jmetro: it supports multi-threading; realtime integration will generally be better due to the multi-threading; has a few nice features that chan_sip doesn't (although it doesn't have every chan_sip feature either)) |
| 23:46.07 | mjordan | Sudravirodhin: np. ARI is a big jump for that use case, as you have to implement a fair amount of the dial logic yourself. If you find it worthwhile, you may want to take a look at the app-dev list postings. Anecdotally, some have found it a smaller task than they initially thought: http://lists.digium.com/pipermail/asterisk-app-dev/2014-December/000629.html |
| 23:48.29 | Sudravirodhin | mjordan: I'll bookmark this, thanks. Yes, ARI is a bit daunting for me but I do need to get into it for when I develop the panel for a company product. Maybe this is a good intro. Thanks again, and have a good one. |
| 23:52.30 | Sudravirodhin | MixMon() only operates on a bridge between two channels, correct? |
| 23:52.38 | Sudravirodhin | MixMonitor* |
| 23:52.45 | Sudravirodhin | The function itself. |
| 23:53.50 | AnonGirl | hiiiiiiiiiiiı |
| 23:54.38 | [TK]D-Fender | Sudravirodhin> MixMon() only operates on a bridge between two channels, correct? <- no. |
| 23:54.55 | [TK]D-Fender | Sudravirodhin, "core show application mixmonitor" <- read the instructions |
| 23:56.38 | Sudravirodhin | [TK]D-Fender, thanks for pointing that out. I should have checked there first. |
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