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| 00:06.15 | elcontrastador | Hi guys...been out of Asterisk for awhile. Using 14.6.0 and looking for the diaplan app 'System'. It doesn't appear to be there anymore. What is the current way to execute as system script? |
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| 00:22.09 | drmessano | elcontrastador: Did you install from source? |
| 00:22.16 | elcontrastador | y |
| 00:23.04 | elcontrastador | i'm looking at the asterisk wiki and it shows it's still current...tons of dialplan apps missing from my core show applications ? query |
| 00:24.50 | drmessano | Go back and look at the menuselect |
| 00:25.03 | elcontrastador | ok... |
| 00:25.34 | elcontrastador | that answers that...i was starting to figure i missed some configuration items after seeing the list of avail apps i was missing |
| 00:25.49 | elcontrastador | thank u |
| 00:33.14 | elcontrastador | drmessano: looks like i had all the apps built in menuselect |
| 00:33.21 | elcontrastador | app_system is checked |
| 00:33.39 | drmessano | Pastebin your dialplan then |
| 00:34.38 | elcontrastador | [Aug 6 17:34:22] WARNING[14716][C-00000001]: pbx.c:2903 pbx_extension_helper: No application 'System' for extension (do_pbxin, 400, 2) |
| 00:34.57 | drmessano | Pastebin your dialplan then |
| 00:34.58 | elcontrastador | that's the error...not even calling the script |
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| 00:35.44 | drmessano | ok |
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| 00:37.18 | elcontrastador | https://pastebin.com/uQX95YYE |
| 00:38.44 | elcontrastador | https://pastebin.com/Ccx3qVcU |
| 00:39.29 | elcontrastador | system application is not shown in list of available apps...like it's not enabled |
| 00:41.50 | drmessano | do you have app_system.so in /usr/lib/asterisk/modules ? |
| 00:43.09 | elcontrastador | y...just loaded it with module load app_system.so.... i'll need to add this to modules.conf...wasn't there |
| 00:43.44 | elcontrastador | i didnt know thery werent all loaded by default...i've been out of Asterisk for years... |
| 00:43.50 | elcontrastador | i'm fixing this now |
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| 00:46.20 | elcontrastador | drmessano: thank you... |
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| 00:53.57 | drmessano | Go find the sample modules.conf |
| 00:54.21 | drmessano | Should give you an idea as to what to preload and what to load |
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| 01:02.32 | elcontrastador | ok...will do |
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| 01:47.53 | elcontrastador | drmessano: https://pastebin.com/EFTR4XLe |
| 01:48.44 | elcontrastador | i've got this mostly working...put intent in the top of pastie. Just need some holistic idea of best approach here as I'm struggling where to start reading |
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| 01:59.50 | elcontrastador | clearly, i have to inform asterisk that to hold until it receives notification to proceed as this is not a channel thru asterisk... just trying to determine best way. The phones are skinny...it will know the phone's off hook...trying to figure out to tell it |
| 02:32.37 | elcontrastador | How do I register a channel type: [Aug 6 19:30:45] WARNING[17324][C-00000001]: channel.c:6261 ast_request: No channel type registered for 'MulticastRTP'. The module is loaded |
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| 05:20.13 | elcontrastador | How do I register a channel type: [Aug 6 19:30:45] WARNING[17324][C-00000001]: channel.c:6261 ast_request: No channel type registered for 'MulticastRTP'. The module is loaded |
| 05:21.04 | elcontrastador | hjuhsd-voip1*CLI> core show channeltype |
| 05:21.04 | elcontrastador | Announcer CBAnn CBRec Local Recorder SCCP Surrogate |
| 05:21.53 | drmessano | You don't have the needed modules loaded |
| 05:22.06 | drmessano | are you autoloading or trying to load these with specific load directives? |
| 05:24.57 | drmessano | ??? |
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| 06:00.08 | drmessano | elcontrastador: ? |
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| 09:25.04 | Haris | hello all |
| 09:25.42 | Haris | I have 30 channels from a sip trunk/exchange. how do I limit extensions behind my box to be able to do only one call at one time, no matter what the direction (i.e., inbound/outbound) ? |
| 09:26.19 | Haris | I'm watching a *perhaps* overlap in call making |
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| 09:43.02 | elcontrastador | drmessano: so, sorry... i'm not autoloading... i did add this module to modules.conf. So,, it's dependency modules it's after? Should i turn on autoload and just 'noload' what i dont want? |
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| 11:01.57 | Haris | guys, anyone around ? |
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| 12:11.33 | elcontrastador | I've wondering why my dialplan stanza doesn't continue after the Page app is called. More detailed question, dialplan, and console output here: https://pastebin.com/9YftpCAH |
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| 13:13.27 | Samot | elcontrastador: What do you mean it "doesn't continue"? How doesn't it continue? |
| 13:14.26 | elcontrastador | samot: did u see the pastebin? It never executes anything below it...the first thing after the Page is a NoOp and then a script to stop the handsets from listening...it just stops processaing after page |
| 13:16.26 | elcontrastador | so, after line 70 (of pastebin)...no other dialplan entries are executed in that stanza...i thought it would fall thru like Dial would |
| 13:19.02 | Samot | elcontrastador: Page is like a Dial(). |
| 13:19.15 | Samot | It's dialing the device and once it answers, that's it. It is in a call. |
| 13:19.24 | Samot | Then next step after that is basically hanging up. |
| 13:19.32 | elcontrastador | yeah...then hang up and should do next step right? |
| 13:19.45 | Samot | If you want to execute further dialplan you need to look at the Page() options to do so. |
| 13:19.48 | Samot | No |
| 13:19.52 | Samot | Hang up = No more call |
| 13:20.14 | elcontrastador | no Page() options to execute further dialplan that i could tell |
| 13:20.22 | elcontrastador | ahh... |
| 13:22.11 | Samot | Yeah you're probably right. There's not. |
| 13:22.26 | Samot | Because Page is a "call these devices and send message" done. |
| 13:22.33 | elcontrastador | i need to execute the script following the Page. Just holistically, at least, can you give me an idea of the best way to process that script when Page exits? |
| 13:23.00 | Samot | What is supposed to happen after the Page? |
| 13:23.01 | elcontrastador | i'll do the reading...just hard to know what to look into |
| 13:23.14 | elcontrastador | another System() |
| 13:23.26 | Samot | Not happening. |
| 13:23.34 | Samot | Page sends an outbound call to the device. |
| 13:23.37 | Samot | Or devices |
| 13:23.43 | Samot | Put them in a conf room |
| 13:24.18 | Samot | And when the original caller/pager ends the call that's it |
| 13:24.19 | Samot | Done. |
| 13:24.32 | Samot | Everyone is dumped from the conf room and the conf room is destroyed. |
| 13:24.49 | elcontrastador | i understand that... |
| 13:24.53 | elcontrastador | now... |
| 13:25.09 | Samot | Let me try again with this |
| 13:25.20 | Samot | You cannot continue on in the dialplan after the Page is comepleted. |
| 13:25.22 | elcontrastador | i meant, knowing that, what would be an option to tear down the mcast listeners i started before page |
| 13:25.38 | Samot | You cannot carry on after an answered/connected Page. |
| 13:25.43 | elcontrastador | ok |
| 13:25.44 | Samot | I just told you why. |
| 13:26.11 | Samot | The Pager hangsup. So they are out of the call. |
| 13:26.30 | Samot | When the Pager hangsup, all the other devices that were Paged and connected are dropped. |
| 13:26.36 | Samot | So there are no more active channels |
| 13:26.40 | Samot | No more active calls. |
| 13:26.46 | Samot | No more dialplan to be run. |
| 13:27.18 | elcontrastador | its a multicast page... |
| 13:27.29 | Samot | OK. |
| 13:27.40 | Samot | You get that ALL the channels are DISCONNECTED |
| 13:27.41 | Samot | Right? |
| 13:27.48 | Samot | There isn't a single channel left opened. |
| 13:27.59 | elcontrastador | i system call a script that pushes the xml to the phones so they listen, Page successfully, then i need to push xml for them to stop listening... |
| 13:28.01 | Samot | How do you plan to continue on when there are no calls? |
| 13:28.01 | elcontrastador | i get it |
| 13:28.16 | Samot | That's not how Page works. |
| 13:28.20 | Samot | It doesn't push XML files. |
| 13:28.34 | elcontrastador | ugh...did u look at the pastebin? |
| 13:28.40 | Samot | Yes. |
| 13:28.42 | elcontrastador | I pushed the xml |
| 13:28.48 | elcontrastador | made the phones listen |
| 13:29.06 | elcontrastador | after the page i need to make them not listen or they stay listening |
| 13:29.12 | elcontrastador | so i'll just figure that out |
| 13:29.50 | Samot | Then the caller/pager CANNOT hang up |
| 13:29.53 | Samot | Do you get that?! |
| 13:31.13 | elcontrastador | y...thx |
| 13:31.35 | Samot | So I'm not sure how you would do what you want. |
| 13:33.31 | elcontrastador | Dial(local/extension@context/n, then G(context^exten^pri)) maybe...in the other extension i put the code to stop the phones from receiving mcast. What i'm going to try |
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| 14:35.24 | jfcl | Hello, is it possible to play a sound with asterisk when dialing an intercom number? |
| 14:35.47 | jfcl | We got some Yealink phones here and we want to be able to play a sound when we dial the extension 255 |
| 14:35.52 | jfcl | kinda like an alarm |
| 14:36.03 | voipmonk | do you want the sound played as an alert that the intercom was dialed or a sound file played through the intercom? |
| 14:36.13 | voipmonk | or am I missing something... |
| 14:36.36 | voipmonk | gets coffee |
| 14:36.39 | eric_hill | I use PlayTones for that in both cases. |
| 14:36.39 | jfcl | okay |
| 14:36.46 | jfcl | well its more like a sound played through |
| 14:37.17 | jfcl | PlayTones? do you have a custom script you wrote for that? |
| 14:37.39 | eric_hill | No, it's a built in application. |
| 14:37.45 | jfcl | Oh okay ;) |
| 14:40.39 | voipmonk | check the Dial command |
| 14:40.48 | voipmonk | look at A |
| 14:40.57 | voipmonk | for Play announcement to the called party |
| 14:43.02 | jfcl | thank you voipmonk, you are as helpful as always. you helped me in the past using cisco phones for asterisk. you have been a great help |
| 14:43.03 | jfcl | :P |
| 14:43.21 | DanQuinney | Monks are normally very helpful |
| 14:43.26 | jfcl | lol |
| 14:44.03 | voipmonk | you're welcome - and may opus be with you |
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| 15:48.53 | avb | hey guys. Is there any modern hook for a queue, which will allow me to add some predial actions? Say I want to implement a blacklist |
| 15:49.02 | avb | and skip agentA for calls X |
| 15:49.07 | avb | and skip agentA for a caller X |
| 15:49.16 | avb | and I dont want to use Local |
| 15:56.21 | [TK]D-Fender | no |
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| 16:25.58 | Bhakimi | hi guys, is jitterbuffer enabled by default in the iax driver? |
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| 16:32.22 | Dovid | What would cause this issue? TLS transport or SSL error reading data |
| 16:35.40 | Samot | Dovid: Is there a real question with some real context coming? |
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| 16:44.42 | avb | [TK]D-Fender: thats what i thougt :) |
| 16:45.13 | [TK]D-Fender | avb, App instructions are built in and the sample configs tell you what you can do with it |
| 16:45.20 | [TK]D-Fender | If it's not there it doesn't exist |
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| 16:54.14 | dovid | I am trying to connect via WebRTC and I keep getting that error |
| 17:01.54 | overyander | anyone use * on freebsd? |
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| 17:27.41 | [TK]D-Fender | ~polls |
| 17:27.41 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
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| 18:00.16 | Janos | hey there, got a peculiar problem after upgrading from 1.8 to 13, some phones have one way audio problems, cisco SPA work like always, but polycom is presenting the issue, normally this means NAT problems but these two phones are on the same network etc, and checking with tcpdump I do see the two way UDP stream to and from the phone when the call is established |
| 18:01.38 | Janos | I was thinking this could be related to some codec change or maybe some default value changed for the sip protocol itself, still using the sip channel, not pjsip |
| 18:01.52 | Janos | any advice on how to diagnose this is welcome |
| 18:03.39 | salviadud | are you using sip or pjsip? |
| 18:04.03 | salviadud | Sorry, I read that now.. |
| 18:04.14 | salviadud | regular sip |
| 18:04.56 | salviadud | Ummm, you might want to debug at least one peer |
| 18:05.09 | salviadud | sip set debug [peer] |
| 18:05.23 | salviadud | Start from there, perhaps. |
| 18:06.15 | Janos | salviadud, thanks, will do |
| 18:08.06 | Janos | also, when calling phone to phone, it does work, but when the call pass through a queue for example the problem also presents itself, and calls to sip trunks always present the problem |
| 18:10.56 | salviadud | Janos, on your sip.conf file, referring to a peer that presents the problem, is directmedia set to =yes ? |
| 18:12.53 | Janos | `sip show peer <peer>` shows DirectMedia: No |
| 18:14.14 | salviadud | try adding directmedia=yes to a peer that has problems |
| 18:14.17 | salviadud | then do a sip reload |
| 18:15.25 | Janos | thanks let me check |
| 18:17.30 | drmessano | Wait, what? |
| 18:17.48 | [TK]D-Fender | completely backwards |
| 18:17.59 | [TK]D-Fender | Direct media is what SCREWS you most of the time |
| 18:18.03 | salviadud | So, it has nothing to do with directmedia |
| 18:18.12 | salviadud | I got my phones with directmedia=yes |
| 18:18.14 | salviadud | no probs here |
| 18:18.18 | drmessano | Thats dumb |
| 18:18.24 | [TK]D-Fender | Who says they even get the chance to use it? |
| 18:18.33 | [TK]D-Fender | Or that your networking scenario is the same as his? |
| 18:18.37 | [TK]D-Fender | and that's just PHONES |
| 18:18.50 | [TK]D-Fender | Maybe ... just MAYBE we should be looking at a *call* |
| 18:19.01 | salviadud | Yeah, I told him to debug it |
| 18:19.07 | salviadud | I'm dancing in the dark here. |
| 18:19.13 | drmessano | Yeah, then told him how to break it |
| 18:19.23 | [TK]D-Fender | I believe the word is "fumbling" |
| 18:19.36 | salviadud | Janos, come back |
| 18:19.37 | salviadud | haha |
| 18:19.46 | salviadud | YOu might break all your stuff |
| 18:19.54 | Janos | back |
| 18:20.03 | salviadud | Yeah, about directmedia |
| 18:20.10 | salviadud | That might not be the solution you are looking for. |
| 18:20.34 | salviadud | You should debug a call, take a look at what's going on. |
| 18:20.38 | drmessano | salviadud: Probably best not to recommend settings if you don't know what they do |
| 18:20.59 | salviadud | Like I said... mine works, but that's my setup. |
| 18:21.14 | drmessano | I could add in a lot of settings |
| 18:21.22 | drmessano | that "work" but have consequences |
| 18:21.54 | drmessano | directmedia does nothing until you try to re-invite a call across a NAT |
| 18:22.17 | drmessano | Which doesn't work, at all |
| 18:22.30 | drmessano | That's literally 99% of all installs |
| 18:24.22 | salviadud | So, I could probably set directmedia=no and I'd be saving myself the pain |
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| 18:25.35 | drmessano | s/could probably/should/ |
| 18:26.26 | *** join/#asterisk tzafrir (~tzafrir@206.167.44.205) |
| 18:31.01 | Janos | I going to do some tcpdump and sip debugs and report back on the findings |
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| 18:39.40 | drmessano | Janos: Nobody really wants to see tcpdump |
| 18:39.48 | drmessano | Get some SIP debug and pastebin it |
| 18:40.13 | drmessano | Enable SIP debug, make a call, pastebin the whole thing |
| 18:45.30 | Janos | will do |
| 18:45.45 | Janos | is there any way to redirect the debug to a file ? |
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| 18:49.22 | Janos | found it on the wiki, sorry :P |
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| 18:50.22 | Janos | sweet, 13 allows to do `logger add channel` |
| 18:52.41 | RovingWriter | i just wish there was a way for * to log data about specific sip extensions. like, in sip.conf, turn "logging=on" on a sip extension, and everything that has to do with calls to/from that, get all their debug info logged |
| 18:58.50 | RovingWriter | for when there are a lot of calls going through the pbx |
| 19:07.09 | drmessano | You can already do that |
| 19:07.37 | drmessano | sip set debug peer (peer) |
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| 19:25.06 | Bhakimi | hi guys, is jitterbuffer enabled by default in IAX ? |
| 19:26.02 | Samot | https://github.com/asterisk/asterisk/blob/master/configs/samples/iax.conf.sample |
| 19:26.59 | Bhakimi | so it uses global if not set in iax, where is global defined? |
| 19:27.45 | Samot | Huh? |
| 19:27.48 | Samot | Right there. |
| 19:28.03 | Bhakimi | so if its not set in my iax config its disabled by default right? |
| 19:28.11 | Samot | Does it say that? |
| 19:28.18 | Bhakimi | no |
| 19:28.25 | Bhakimi | it deosnt say what its set to by default |
| 19:28.38 | Bhakimi | its not very clear thats why im asking |
| 19:28.48 | drmessano | iax2 show settings |
| 19:28.49 | Samot | OK |
| 19:28.58 | Samot | Let's try this. |
| 19:29.12 | Samot | Do you want jitterbuffer set globally? |
| 19:29.24 | Samot | Whether it be "yes" or "no"? |
| 19:29.33 | Bhakimi | no, i want to make sure its disabled by default |
| 19:29.38 | Samot | OK |
| 19:29.53 | Bhakimi | i got 100 servers and i wanted toi see if its disabled by default so i dont have to update all of them to explicitly disable them |
| 19:29.56 | Samot | So then that would be "Yes, I do." |
| 19:30.04 | Samot | Dude. |
| 19:30.12 | Samot | You have to have an iax.conf in all the servers |
| 19:30.14 | Samot | It's a setting |
| 19:30.19 | Samot | Just set it to NO |
| 19:30.19 | Bhakimi | i do\ |
| 19:30.28 | Samot | [general] |
| 19:30.38 | Samot | jitterbuffer=no |
| 19:30.41 | Bhakimi | i get that i can set it to no, im wondering if its disabled by defaulr ot not, i get these error messages and i read its related to jitter buffer |
| 19:30.46 | Samot | Now put that on all the servers. |
| 19:30.47 | Samot | Done. |
| 19:30.57 | Bhakimi | before i go make changes everywhere i like to understand what its set by defauklt if its not set at all |
| 19:31.16 | Samot | Well |
| 19:31.24 | Samot | You could take one server you are getting these messages on |
| 19:31.29 | Samot | And set it there to no |
| 19:31.33 | Samot | If they go away.. |
| 19:31.36 | Samot | That's your answer. |
| 19:31.36 | Bhakimi | k |
| 19:31.40 | Bhakimi | thanks |
| 19:31.50 | Samot | IAX2 is kinda not even developed anymore |
| 19:32.05 | Bhakimi | i use it for trunks between asterisk servers |
| 19:32.16 | Bhakimi | do you suggest i use sip ? |
| 19:32.50 | Samot | I, personally, wouldn't recommend IAX. |
| 19:33.02 | Samot | But that's me. |
| 19:33.12 | Bhakimi | i get these messages that im trying to figure out how to solve "chan_iax2.c:9990 socket_process_meta: Received trunked frame before first full voice frame" |
| 19:33.22 | Samot | Don't know. |
| 19:33.25 | Samot | I don't use IAX. |
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| 20:04.29 | Janos | is there a specific pastebin for this channel or paste.debian.net is good ? |
| 20:06.17 | *** join/#asterisk jkroon (~jkroon@165.16.204.162) |
| 20:12.31 | Janos | ok here it is |
| 20:12.42 | Janos | working call from cisco phone: https://paste.debian.net/hidden/7d2bd917/ |
| 20:12.54 | Janos | broken call from the polycom phone: https://paste.debian.net/hidden/6a497b12/ |
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| 20:25.06 | voipmonk | Janos, audio issues? |
| 20:26.38 | *** join/#asterisk tzafrir (~tzafrir@206.167.44.205) |
| 20:27.14 | drmessano | Janos: You need to remove G729 from your Polycom codec list |
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| 20:39.27 | Janos | drmessano, thanks, will try that, got a min to explain ? from what I see on the debug "combined - (ulaw|alaw)" is the same for both, shouldn't asterisk prefer one of those ? also the cisco reports g729 too |
| 20:41.07 | Janos | voipmonk, yeah, Polycom phones not working for some calls after upgrading from 1.8 to 13, while cisco phones work |
| 20:50.08 | Janos | drmessano, by remove you mean in the asterisk con en sip.conf or in the actual phone config ? |
| 21:03.39 | *** join/#asterisk u0m3 (~u0m3@5-12-198-162.residential.rdsnet.ro) |
| 21:03.46 | Janos | disabled g729 on the phone but call still does not work, here is the debug https://paste.debian.net/hidden/7a279679/ |
| 21:05.39 | eric_hill | Janos, what does "not work" mean? No audio? Audio jitter? One-way audio? Sounds of seagulls? |
| 21:05.52 | Janos | one way audio |
| 21:06.13 | eric_hill | which way? |
| 21:06.30 | *** join/#asterisk salviadud (~ralfalfa@189-211-190-134.static.axtel.net) |
| 21:06.42 | Janos | caller can hear the callee but not the other way around |
| 21:07.05 | eric_hill | Aaaand is the caller the Cisco or the Polycom? |
| 21:07.33 | Janos | sorry, got that worong, callee can hear the caller but not the other way around |
| 21:07.58 | Janos | when the call is done with a cisco phone it works |
| 21:08.06 | Janos | but the same call with the polycom does not work |
| 21:08.13 | Janos | you get one way audio |
| 21:08.39 | eric_hill | So when the Polycom calls a Cisco, the Cisco can hear the Polycom, but the Polycom CANNOT hear the Cisco. Right? |
| 21:08.55 | eric_hill | Can a Polycom call a Polycom? |
| 21:09.19 | *** join/#asterisk u0m3 (~u0m3@5-12-198-162.residential.rdsnet.ro) |
| 21:09.39 | Janos | sorry, let me explain again, the callee is some remote phone behind another sip pbx, the caller can be a cisco or a polycom |
| 21:10.29 | salviadud | Janos, so the ciscos that originate the call have no probs, just the polycoms. |
| 21:10.34 | Janos | when the call is placed by a cisco phone say ext 6200 it works, but when placed on a polycom phone say ext 6121 it gets one way audio |
| 21:10.36 | Janos | correct |
| 21:10.44 | *** join/#asterisk u0m3 (~u0m3@5-12-198-162.residential.rdsnet.ro) |
| 21:11.01 | salviadud | debug a polycom call |
| 21:11.09 | salviadud | and pastebin, please |
| 21:11.12 | eric_hill | I'm guessing the call is going out via one codec on the Cisco and another on the Polycom. |
| 21:11.27 | Janos | call from cisco: https://paste.debian.net/hidden/7d2bd917/ |
| 21:11.41 | Janos | call from polycom: https://paste.debian.net/hidden/6a497b12/ |
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| 21:11.55 | Janos | call from polycom with g729 disabled: https://paste.debian.net/hidden/7a279679/ |
| 21:12.09 | drmessano | Your SDP still has G729 in it |
| 21:12.13 | drmessano | Remove it |
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| 21:13.35 | eric_hill | Doesn't look like 729 is enabled in the last example. |
| 21:13.48 | drmessano | Like hell it doesnt |
| 21:14.01 | eric_hill | Where is it? What line? I'm missing it then. |
| 21:14.11 | drmessano | s=Polycom IP Phone |
| 21:14.11 | drmessano | c=IN IP4 192.168.5.220 |
| 21:14.11 | drmessano | t=0 0 |
| 21:14.11 | drmessano | m=audio 2234 RTP/AVP 0 8 18 101 |
| 21:14.11 | drmessano | a=rtpmap:0 PCMU/8000 |
| 21:14.11 | drmessano | a=rtpmap:8 PCMA/8000 |
| 21:14.11 | drmessano | a=rtpmap:18 G729/8000 |
| 21:14.29 | eric_hill | Not this one: https://paste.debian.net/hidden/7a279679/ |
| 21:14.41 | drmessano | Thats the Cisco |
| 21:14.43 | drmessano | Which works fine |
| 21:14.44 | Janos | no |
| 21:14.45 | drmessano | Not the Polycom |
| 21:14.47 | drmessano | Which doesnt |
| 21:14.53 | eric_hill | AH, thought that was the Polycom. |
| 21:15.00 | Janos | no |
| 21:15.09 | Janos | that's the Polycom without g729 |
| 21:15.27 | Janos | check line 21 of https://paste.debian.net/hidden/7a279679/ |
| 21:15.32 | eric_hill | Janos, it /does/ say s=Polycom IP Phone |
| 21:15.46 | Janos | User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.7.0134 |
| 21:16.36 | Janos | Again, Cisco phone: https://paste.debian.net/hidden/7d2bd917/ it works and has g729 enabled |
| 21:16.56 | Janos | Polycom with g729: https://paste.debian.net/hidden/6a497b12/ one way audio problem |
| 21:17.16 | Janos | Polycom without g729: https://paste.debian.net/hidden/7a279679/ still one way audio problem |
| 21:18.32 | eric_hill | Same codecs are accepted on both examples. This may not be a problem on your side. |
| 21:21.40 | Janos | only diff I see between both phones is that the cisco reports more codecs g726 and g722 to be precise, what I can't explain is how this worked in 1.8 |
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| 21:22.22 | salviadud | how are your allow and disallow parameters? |
| 21:22.34 | salviadud | are they global for all sip devices? |
| 21:27.38 | *** join/#asterisk Bhakimi (~textual@rrcs-69-75-121-202.west.biz.rr.com) |
| 21:27.48 | Bhakimi | anyone knows where i can get a ringing audio file? |
| 21:27.59 | Bhakimi | i need to play back a ringing inside a agi script and i dont have a ringing audio file to use |
| 21:28.14 | Bhakimi | unless there is a way to create a ringing effect without playing back a audio file? |
| 21:28.37 | eric_hill | Bhakimi, PlayTones? |
| 21:29.09 | Bhakimi | and use agi->exec to execute that command? |
| 21:29.33 | eric_hill | I *think* you can, but I don't use AGI so YMMV. |
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| 21:39.40 | Bhakimi | that worked eric_hill: thx |
| 21:40.05 | eric_hill | np |
| 21:50.28 | eric_hill | Does anyone have a good chan_sip to res_pjsip migration guide? |
| 21:51.12 | file | have you checked the wiki? |
| 21:51.30 | file | https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip |
| 21:51.46 | eric_hill | reading that right now. |
| 21:52.38 | eric_hill | I'm interested in the realtime over http if it's at all possible. |
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| 21:54.11 | Samot | So you want WebRTC |
| 21:55.33 | eric_hill | Samot, no, realtime configuration of sip peers and dialplans from an HTTP back-end |
| 21:56.18 | eric_hill | It looks like pjsip plugs into sorcery, which can expose that functionality for me. I may be making this more difficult than I need to. |
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| 22:07.14 | salviadud | sorcery, haha |
| 22:08.45 | drmessano | salviadud: Whats wrong with sorcery? |
| 22:09.25 | eric_hill | yea.... pjsip -> sorcery -> realtime wizard -> res_config_curl ... If it works, great. When it breaks, whoa momma. |
| 22:09.40 | eric_hill | Plus to debug it I'd need a +3 battleaxe. |
| 22:10.35 | drmessano | I found salviadud on YouTube |
| 22:10.40 | drmessano | https://www.youtube.com/watch?v=_wKN3OF7oiI |
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| 22:11.10 | eric_hill | Ah! You can get yourself pregnant!? |
| 22:15.34 | salviadud | yeah, very funny drmessano |
| 22:15.51 | salviadud | didn't know what type of sorcery was being mentioned. |
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| 22:32.54 | file | sorcery is the name of the data access layer which newer generation things use |
| 22:34.45 | *** join/#asterisk ttaylor (~ttaylor@vpn.duh.net) |
| 22:35.31 | RovingWriter | have 57 BTC. bought them long time ago for $91 ... have cashed out 3 times small amounts in the tune of $300.... so now everything is complete profit. it's soooooo freaking hard to not just cash out and turn a blind eye |
| 22:36.10 | salviadud | RovingWriter, ur a player. |
| 22:36.25 | RovingWriter | negative |
| 22:36.27 | salviadud | Better than stock eh |
| 22:36.31 | RovingWriter | just got lucky i suppose |
| 22:36.40 | RovingWriter | but its hard to decide if I should just get out now, take the profit and run, or hang on and see where it goes |
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| 22:38.21 | drmessano | You have $200k in BTC? |
| 22:39.40 | RovingWriter | yeah, something around there, drmessano |
| 22:39.53 | salviadud | That or buy some goods. |
| 22:39.56 | RovingWriter | but i bought it for $91, many years ago |
| 22:40.16 | RovingWriter | so i kinda wanna just see where it goes, but.... man, is that stupid? |
| 22:40.35 | RovingWriter | i own my house outright, no debt, 60k in bank |
| 22:42.56 | salviadud | RovingWriter, sounds like you need to have this discussion with your wife and kids maybe |
| 22:43.39 | RovingWriter | that's how i own my house outright and no debt, i dont have wife or kids |
| 22:43.41 | RovingWriter | haha |
| 22:44.09 | salviadud | You could get married with that coin. |
| 22:44.19 | salviadud | Ladies love economic stability! |
| 22:44.25 | RovingWriter | lol |
| 22:44.42 | RovingWriter | i could buy like 200k lottery tickets |
| 22:45.24 | salviadud | Well, it's up to you. |
| 22:45.38 | salviadud | I'd start doing all bitcoin transaction.s |
| 22:45.43 | salviadud | Not even cashing out |
| 22:45.47 | RovingWriter | i dont really buy things |
| 22:45.51 | RovingWriter | i am very minimalistic |
| 22:45.56 | RovingWriter | i just save $$ |
| 22:46.12 | salviadud | You could buy a Phillip Glass album |
| 22:46.27 | salviadud | He's a minimalist you might enjoy. |
| 22:46.31 | salviadud | Jokes aside. |
| 22:46.46 | salviadud | You could just keep saving it, it'll probably go up. |
| 22:47.04 | RovingWriter | yeah, lots of people seem to think so |
| 22:47.17 | salviadud | Are you a buddhist as well? |
| 22:47.22 | RovingWriter | nope |
| 22:47.25 | salviadud | Maybe you believe that less is more. |
| 22:47.34 | salviadud | Which is a humble way to live, I guess. |
| 22:48.03 | RovingWriter | i think my minimalistic ways are from moving around a lot in military days |
| 22:48.11 | RovingWriter | big tv? no, i don't wanna move it later. |
| 22:48.14 | salviadud | Get an MK23 |
| 22:48.32 | salviadud | I heard it's very accurate... |
| 22:48.35 | salviadud | not cheap |
| 22:48.37 | RovingWriter | i do have those |
| 22:48.41 | RovingWriter | i do have a lot of guns already |
| 22:48.42 | RovingWriter | lol |
| 22:49.12 | salviadud | That is the stuff you buy then. |
| 22:49.26 | RovingWriter | 200k worth of guns? |
| 22:49.36 | salviadud | Are you also a doomsday prepper? Perhaps you want to invest in a fallout shelter. |
| 22:49.45 | RovingWriter | go into gun store. "yes, do you accept bitcoins?" |
| 22:49.51 | RovingWriter | no i don't do doomsday prep stuff |
| 22:50.05 | salviadud | Lets just hope they are all wrong. |
| 22:50.19 | salviadud | And they prepped for another katrina, at most. |
| 22:50.45 | salviadud | It be kind of dissapointing to prep for doomsday, and not even get zombies. |
| 22:50.48 | salviadud | Or a purge. |
| 22:51.11 | RovingWriter | yeah, that would be lame |
| 22:51.13 | salviadud | "What am I going to do with all those tuna cans and .22 stockpiled ammo?" |
| 22:51.16 | RovingWriter | i just have lots of guns haha |
| 22:51.25 | RovingWriter | and by lots, i only mean like 25 |
| 22:51.27 | salviadud | What's the highest caliber you got? |
| 22:51.39 | salviadud | You own a barret by any chance? |
| 22:51.40 | RovingWriter | .50 BMG |
| 22:51.50 | RovingWriter | mine is not barret, but shoots same round |
| 22:52.07 | salviadud | Do you own an mp7? |
| 22:52.10 | RovingWriter | mine is an armalite AR-50. bolt action |
| 22:52.13 | salviadud | man, I love that lil thing |
| 22:52.22 | RovingWriter | nope, you mean MP5? |
| 22:52.32 | salviadud | MP5 shoots 9mm |
| 22:52.38 | salviadud | MP7 shoots a smaller round |
| 22:52.51 | RovingWriter | the most exotic things I have are... Kriss Vector .45, Kriss Vector .9mm, FN FS2000 |
| 22:52.53 | salviadud | They are both from HK |
| 22:53.09 | salviadud | FN has a nice thumb guard |
| 22:53.24 | RovingWriter | yeah, but the gas blowback while suppressed sucks |
| 22:53.28 | salviadud | Kriss vector is an interesting recoil system. |
| 22:53.31 | RovingWriter | right back into your face |
| 22:53.45 | RovingWriter | my .45 Vector is a piece of shit. its one of the Gen1 ones.... then 9mm is gen2, and it shoots like a dream |
| 22:53.59 | salviadud | Well, it is a smaller round |
| 22:54.10 | salviadud | And you got full-auto or just civilian model? |
| 22:55.24 | salviadud | Well, I think you might need to get a portable missile system, for 57 btc, you could afford it. |
| 22:55.41 | salviadud | Your own SAM |
| 22:55.56 | RovingWriter | all of mine are just cililian |
| 22:55.59 | RovingWriter | with suppressors |
| 22:56.03 | RovingWriter | so tax stamps there |
| 22:56.31 | salviadud | RovingWriter, you are even discreet with the guns, sub-sonic and such. |
| 22:56.48 | salviadud | Go get a girlfriend already |
| 22:57.06 | salviadud | Or maybe, you'd like to buy 15 7942 cisco phones |
| 22:57.22 | salviadud | Another man's junk is another man's treasure, yarrrrr |
| 22:58.50 | salviadud | RovingWriter, I'm out of ideas, I guess you are in fact blessed. I hope you make the right choice. |
| 22:58.57 | RovingWriter | i have a girlfriend, haha |
| 22:59.14 | RovingWriter | but she doesn't control any of my $$ decisions |
| 22:59.20 | RovingWriter | she's horrible with $$ hehe |
| 22:59.58 | salviadud | If you could use that for beer money, for how long would it last? |
| 23:00.11 | salviadud | Or just doing the groceries |
| 23:00.15 | RovingWriter | haha, i don't drink |
| 23:00.20 | RovingWriter | but for groceries, a long time |
| 23:00.27 | salviadud | You don't drink, what a trooper |
| 23:00.55 | RovingWriter | its really actualliy quite lame. i just hate the taste of alcohol :(" |
| 23:01.50 | salviadud | Really? ALL alcohol? |
| 23:02.04 | salviadud | You might need to try one that fits your style. |
| 23:02.14 | RovingWriter | i usually try new ones every time i go somewhere |
| 23:02.18 | RovingWriter | to see if there is something I like |
| 23:03.33 | salviadud | Are you close to the the state of Nevada, Portland, Colorado or Washington ? |
| 23:03.44 | salviadud | Maybe you are a puff kinda guy. |
| 23:03.58 | salviadud | Like some of the linux/unix gurus I've met. |
| 23:04.31 | salviadud | 57 BTC worth of indica, it be a couple of tons. |
| 23:04.38 | salviadud | :O |
| 23:05.11 | salviadud | What I'm getting at is this: you might want to use it for travelling |
| 23:05.27 | salviadud | If you don't want tangible stuff, you could pay for an experience. |
| 23:05.35 | salviadud | Some trip in an island paradise. |
| 23:05.46 | RovingWriter | haha, yes, WA |
| 23:05.59 | salviadud | Go get baked! |
| 23:06.05 | salviadud | If you don't like puffing |
| 23:06.08 | salviadud | Buy an edible. |
| 23:06.16 | salviadud | You'll feel fiiiiine. |
| 23:06.29 | salviadud | You're gonna want to listen to musiconhold forever! |
| 23:06.39 | RovingWriter | lol, i get all high, and then wake up next morning, and have donated 56btc to hillary clinton |
| 23:06.40 | RovingWriter | :( |
| 23:06.48 | salviadud | Yuck |
| 23:06.59 | salviadud | No, don't do that |
| 23:07.02 | RovingWriter | hence the :( :( |
| 23:07.22 | salviadud | Don't even mention her scummy name! haha |
| 23:07.33 | salviadud | Man, she was the worst candidate... |
| 23:07.41 | salviadud | feels for Bernie... |
| 23:08.03 | salviadud | You could start your own version of the DNC |
| 23:08.09 | salviadud | one that actually listens to ppl |
| 23:09.18 | RovingWriter | yeah i feel for bernie |
| 23:09.21 | RovingWriter | but he was a pushover too |
| 23:09.27 | RovingWriter | he had so much support |
| 23:09.51 | salviadud | He's a bit of a pacifist, not much of a fighter. |
| 23:10.30 | salviadud | I guess he ended up showing how corrupt and bought the system really is. |
| 23:10.40 | salviadud | Without getting shot! |
| 23:10.51 | salviadud | Such a smart one, that Bernie. |
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| 23:32.27 | Lovepump | hello - I'm currently trying to build asterisk 14.6.0 from source. I've made it past the dahdi and libpri builds and trying to figure out what to do with pjproject, as outlined here: https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-bundled |
| 23:33.01 | Lovepump | The wiki suggests I could try using the "bundled" pjproject that comes with asterisk >= 13.8 |
| 23:33.40 | Lovepump | when I follow that link, it takes me to the middle of the same page and it has no further information on the fabled bundled pjproject |
| 23:34.16 | Lovepump | The sentence I am referring to is this one: If you're using Asterisk 13.8.0 or greater, consider using the Bundled Version of pjproject. |
| 23:34.55 | Lovepump | the link is: https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-BuildingandInstallingpjprojectfromSource |
| 23:35.09 | Lovepump | and it takes me to a "solution" that says |
| 23:35.22 | Lovepump | <PROTECTED> |
| 23:36.05 | Lovepump | so the link is obviously going to the wrong part of something - can someone point me at the documentation for the bundled pjproject? |
| 23:36.22 | Samot | First, run ./contrib/scripts/install_prereq. Building the bundled pjproject requires the python development libraries which install_prereq has already installed. All you have to do now is add the --with-pjproject-bundled option to your Asterisk ./configure command line and remove any other --with-pjproject option you may have specified. |
| 23:36.41 | Lovepump | ok cool - thanks |
| 23:36.47 | Samot | Right there |
| 23:36.51 | Samot | At the bottom of the page |
| 23:37.29 | Lovepump | ahh ok - so the link is just anchored incorrectly - I'll trudge on |
| 23:37.48 | Samot | I don't know. I clicked the link in the side menu of that page and it took me there. |
| 23:37.56 | Samot | But.. |
| 23:38.12 | Lovepump | this was the link contained in this sentence: |
| 23:38.15 | Samot | Even if the link isn't anchored correctly..... |
| 23:38.20 | Lovepump | The sentence I am referring to is this one: If you're using Asterisk 13.8.0 or greater, consider using the Bundled Version of pjproject. |
| 23:38.25 | Samot | Reading the whole page helps |
| 23:38.30 | Samot | Right |
| 23:38.39 | Samot | And all of what I pasted is right _below_ it |
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| 23:41.38 | Lovepump | And since I don't make the documentation and can't fix the link, I figured letting someone know it is confusing would be the right thing to do. |
| 23:42.02 | Samot | https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-bundled <-- That link took me right to that section |
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