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| 02:07.09 | skyroveRR | Is it possible to test asterisk's SIP functionality with only a single SIP account? I made an account in Ekiga SIP client, but I'm not sure if I will get to test asterisk using only 1 SIP account. |
| 02:19.02 | chandoo | hi |
| 02:19.16 | chandoo | how t o install astgerisk on freebsd jail |
| 02:21.31 | chandoo | i foudn this note looks like it is not good for 9..3 |
| 02:21.32 | chandoo | https://forums.freenas.org/index.php?threads/how-to-install-asterisk-1-8-pbx-and-asterisk-gui-in-a-portjail.19829/ |
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| 02:22.26 | WIMPy | The Asterisk GUI is dead. |
| 02:23.56 | chandoo | WIMPy, what do you recommed for ip phone |
| 02:24.14 | WIMPy | none |
| 02:24.24 | chandoo | i build freenas and want to install asterisk |
| 02:25.12 | Synthase_ | skyroveRR: Yes. The sample files contain dialplan steps that will playback messages to a single endpoint. |
| 02:26.21 | skyroveRR | Synthase_: there's one more thing: I can't seem to find asterisk binding to port 5060... when I register an account with ekiga on localhost, it says that it cannot register the account, remote host is offline.. |
| 02:26.55 | WIMPy | You can't have multiple applications on the same port. |
| 02:27.25 | skyroveRR | WIMPy: so what shall I do instead? |
| 02:27.47 | WIMPy | Change it on either one. |
| 02:27.57 | WIMPy | Or use a 2nd machine. |
| 02:28.01 | skyroveRR | Ekiga is closed now, but still I don't see asterisk listening on port 5060. |
| 02:28.23 | WIMPy | Did you restart it? |
| 02:28.30 | skyroveRR | Yes. |
| 02:28.50 | WIMPy | The it's not or badly configured. |
| 02:29.43 | WIMPy | Try 'module load chan_sip' with some verbose and possibly debug enabled and see what it has top tell you. |
| 02:30.07 | skyroveRR | *CLI> module load chan_sip |
| 02:30.09 | skyroveRR | Module 'chan_sip' already exists. |
| 02:30.11 | skyroveRR | Unable to load module chan_sip |
| 02:30.13 | skyroveRR | Command 'module load chan_sip' failed. |
| 02:30.31 | skyroveRR | Do I need to open any firewall or something just for listening? |
| 02:30.55 | WIMPy | Nom but to make it work. |
| 02:31.16 | WIMPy | 'sip show settings' |
| 02:32.37 | skyroveRR | WIMPy: http://tracefl.in/pastes/settings.txt |
| 02:33.55 | WIMPy | Looks like it should losten. |
| 02:33.59 | WIMPy | listen |
| 02:34.45 | WIMPy | That looks like a somewhat old version, BTW. |
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| 02:35.50 | skyroveRR | Oh. Now it is listening. |
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| 06:32.16 | asteriskmo | I've installed 3 AsteriskNOW versions. One with Asterisk 11 another with 12 last with 13. Only Asterisk 12 has res_calendar* modules natively in it. Any reason why those modules are not compiled in Asterisk 11 and 13? |
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| 06:35.25 | asteriskmo | Also, I saw that I can compile them myself but I'd like to avoid that complication if possible.. (that's why I've chosen a distro) |
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| 08:15.38 | Moggie | Hi, please can I check something. If Asterisk fetches SIP users from a database, it cannot support subscriptions, voicemail waiting notifications, or NAT keep alive to maintain registrations? Thanks in advance! |
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| 08:59.28 | Rico | hi |
| 08:59.46 | Rico | is this the right place to ask about fax for asterisk ? |
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| 09:02.24 | jzu_ | Rico: ...yes? |
| 09:02.44 | Rico | jzu_: I have problems with sending some faxes |
| 09:03.03 | Rico | I don't know if it's related to fax fort asterisk or related to PDF -> tiff conversion |
| 09:03.12 | Rico | it works well with 99.99% of faxes |
| 09:03.24 | jzu_ | I don't even know how you send fax using Asterisk :) |
| 09:03.25 | Rico | but one of my client is unable to send faxes |
| 09:03.34 | jzu_ | is there some web-interface for sending & receiving? |
| 09:03.45 | Rico | jzu_: http://www.digium.com/en/products/software/fax-for-asterisk |
| 09:03.47 | Rico | nop |
| 09:03.57 | Rico | core show application sendfax |
| 09:05.04 | Moggie | strange, I got the welcome email from digium but no password when I registered on the forum :o |
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| 11:29.51 | h8 | hi everyone |
| 11:30.17 | h8 | is there a solution to stop Call from '' (74.91.31.172:5071) to extension '900441226693583' rejected because extension not found in context 'central'. kind of messages? |
| 11:30.31 | h8 | or even others trying to call through the voip serveR? |
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| 11:33.45 | asteriskmo | Hello, anyone here? |
| 11:35.03 | asteriskmo | Any AsteriskNOW expert pl? |
| 11:37.49 | asteriskmo | I thought this was #asterisk ..? |
| 11:38.02 | Moggie | it is |
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| 11:39.46 | asteriskmo | Ok cool. Actually I'm not sure of my question is for #asterisk or #asterisk-dev: I wonder why res_calendar* modules are not in Asterisk 11 or 13 but are in Asterisk 12 ? |
| 11:40.07 | Moggie | yeah i dunno about that one i'm afraid |
| 11:41.08 | Moggie | I myself have been trying to confirm whether subscriptions / NAT keep-alive will work when Asterisk runs in realtime mode. The docs suggest those features are unavailable, but I'm not sure if it's just me reading it wrongly or the page is out of date. |
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| 11:42.11 | asteriskmo | Oki. Thanks Moggie. Would you know how to add those modules to 11 or 13? I've installed AsteriskNOW distro and they come without source code. Can I still compile those packages' source? |
| 11:47.02 | Moggie | I would think so. I think it mostly functions like a regular linux box |
| 11:47.37 | Moggie | have to be a bit careful if/when you want to perform upgrades though |
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| 12:15.27 | woopstar | Hi. Can anybody help me with this error using pjsip and realtime: There was no wizard for object type 'registration' - What do i need to add to the sorcery.conf file? I cannot find it, in the official documentation |
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| 12:28.23 | babak | Hi, using func_odbc vs agi for database access has any advantages like performance or ...? |
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| 12:35.58 | [TK]D-Fender | babak: Lower resources required, get to stay in the dialplan |
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| 12:51.46 | Moggie | with asterisk running in realtime mode, does anyone use NAT keep-alive or voicemail message waiting indicators? |
| 12:52.56 | [TK]D-Fender | What;s your issue? |
| 12:53.39 | mjordan | woopstar: generally, your registration objects would not be stored in dynamic realtime. Outbound registrations are only read when the module is loaded; there is no way for Asterisk to know that an outbound registration has been updated in the database. You can use standard static realtime if you want to store an outbound registration in a database as opposed to a conf file. |
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| 12:54.14 | woopstar | mjordan: Then what is the ps_registrations table in mysql for, which is provided by alembic ? |
| 12:54.53 | woopstar | because the ps_registrations table from mysql, which alembic creates is not documented anywhere it seems? |
| 12:55.13 | mjordan | once upon a time, we had a table for those objects. That was when we first created the scripts, and a number of people used it. |
| 12:55.23 | mjordan | Rather than destroy their db schemas by dropping the table, we just left it. |
| 12:56.13 | mjordan | Really, Asterisk will probably read objects out of that table on startup and create the outbound registrations, if you tell it to do so via sorcery.conf (I vaguely recall that it will still work, but it has been awhile since someone tried it). However, it will still only do that on startup. |
| 12:57.34 | woopstar | Oh ok. Then it all makes sense, but It's a bit unclear when it's not present in the documentation :) |
| 12:57.45 | mjordan | woopstar: I'd follow the wiki page - https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime |
| 12:57.58 | mjordan | you'll note it doesn't mention outbound registrations :-) |
| 12:58.10 | woopstar | yup, i did that. And after that i was wondering why there were tables from alembic |
| 12:58.13 | woopstar | that was not in the wiki |
| 12:58.14 | mjordan | This, however, does: https://wiki.asterisk.org/wiki/display/AST/Configuring+Outbound+Registrations |
| 12:58.48 | mjordan | specifically: https://wiki.asterisk.org/wiki/display/AST/Configuring+Outbound+Registrations#ConfiguringOutboundRegistrations-Realtime |
| 13:00.34 | woopstar | Well. Seems i've done things right then :) Next step would now be to add a sip trunk to the pjsip tables in the db |
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| 13:53.18 | Moggie | [TK]D-Fender: I don't currently have an issue, but I'm trying to confirm what it says here: https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration |
| 13:53.47 | Moggie | It seems to suggest that in realtime mode, Asterisk cannot support things like subscriptions, voicemail notifications, and NAT keep alives. |
| 13:54.05 | Moggie | But maybe I miss-understood. |
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| 14:06.25 | Rico | I can't understand how I can have more than one incoming call on my sip phone, having limitonpeer=yes, call-limit=4 and busylevel=1 |
| 14:06.32 | Rico | can anyone explain me ? using asterisk 1.8 |
| 14:06.43 | Rico | asterisk 1.8.26.0 |
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| 15:11.04 | Rico | any idea about my call-limit / busy-level problem ? |
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| 15:21.25 | ChannelZ | I can't remember.. you need to have callcounter turned on as well IIRC |
| 15:21.55 | Rico | ok |
| 15:22.15 | Rico | callcounter=yes in my sip.conf |
| 15:22.29 | Rico | and limitonpeer=yes too |
| 15:23.08 | Rico | ChannelZ: http://pastebin.com/g8VjQhCp |
| 15:23.18 | Rico | but I can't find anything about limit on peer |
| 15:24.13 | [TK]D-Fender | Rico: You aren't clear on what the failure is |
| 15:24.24 | [TK]D-Fender | Rico: Please rephrase your original question |
| 15:24.45 | Rico | [TK]D-Fender: sorry. I want my SIP peer to receive only one call at the same time |
| 15:25.00 | Rico | I wan it to reply with "busy" state when already in use |
| 15:25.15 | Rico | sorry |
| 15:25.19 | Rico | forget my last sentence |
| 15:25.28 | [TK]D-Fender | Rico: Then actually put a limit on it, or check first |
| 15:25.40 | Rico | and replace it with "I want asterisk to don't send the call to my phone if it is already in use" |
| 15:25.42 | [TK]D-Fender | RicoI can't understand how I can have more than one incoming call on my sip phone, having limitonpeer=yes, call-limit=4 and busylevel=1 <- This allows more than 1 |
| 15:25.55 | [TK]D-Fender | Rico: You allw 4 there |
| 15:26.18 | Rico | yes but shouldn't the phone reply with BUSY when having already one active call ? |
| 15:26.25 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:26.38 | [TK]D-Fender | Rico: Your phone SUPPORTS multiple simultaneous calls |
| 15:26.56 | [TK]D-Fender | Rico: that is the PHONE choosing how it replies. That has nothing to do with ASTERISK limiting anything |
| 15:26.58 | Rico | [TK]D-Fender: as I understood, asterisk would not send the call to the phone if its state is BUSY |
| 15:27.02 | [TK]D-Fender | Rico: You are mixing up directions. |
| 15:27.08 | Rico | [TK]D-Fender: I know |
| 15:27.11 | Rico | let me be more clear |
| 15:27.17 | Rico | forget about the phone |
| 15:27.20 | [TK]D-Fender | Rico: You misunderstood.. because you set the limit to 4... not 1 |
| 15:27.30 | *** join/#asterisk sgo11 (~song@106.117.89.95) |
| 15:27.30 | Rico | I want asterisk to don't send the call to my phone if it is busy |
| 15:27.45 | [TK]D-Fender | Rico: So either limit your peer properly or check if it's on a call first |
| 15:28.05 | Rico | when setting busy-level to 1, should'nt asterisk detect that the user has one active call and don't send hilm another call ? |
| 15:28.19 | [TK]D-Fender | Rico: You clearly didn't read what that parameter is even FOR |
| 15:28.32 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/1.8/configs/sip.conf.sample?revision=423066&view=markup |
| 15:28.43 | [TK]D-Fender | that is for PRESENCE notifications. |
| 15:28.44 | Rico | if I set call-limit to 1, my phone will not be able to do an outgoing call if he has an active comm |
| 15:28.59 | [TK]D-Fender | Rico: Exactly. That's what the word LIMIT means. |
| 15:29.11 | [TK]D-Fender | Rico: There is no "inbound limit" vs "outbound limit" |
| 15:29.21 | [TK]D-Fender | Rico: There is only "hard limit" as a SIP option. |
| 15:29.31 | [TK]D-Fender | Rico: Anything more is up to you and your dialplan to check and control. |
| 15:29.44 | Rico | so what is the busy-level for ? Just marking the phone as BUSY (hint) ? |
| 15:30.13 | [TK]D-Fender | Rico: Read the sample config |
| 15:30.18 | [TK]D-Fender | Rico: Presence only |
| 15:31.47 | sgo11 | Hi, I am looking for a way to setup a video/voice chat server on ubuntu. People can use different clients (mobile app/PC/Browsers) to do video/voice chat on that server. When I google such task, google shows me asterisk. but when I read some asterisk tutorials, it says it has to connect to a trunk which is a third party service like google voice. this is not what I want. I just want a private server to handle such thing. can I use asterisk without a third-pa |
| 15:31.48 | sgo11 | rty trunk service? I don't need to call mobile numbers directly. thanks. |
| 15:32.32 | WIMPy | Asterisk doesn't need any 3rd party services. |
| 15:32.56 | WIMPy | You only need them if you want to call PSTN numbers for obvious reasons. |
| 15:34.08 | sgo11 | WIMPy, thanks a lot for your reply. can you show me some tutorials on how to setup and use asterisk without 3rd party services? I am confused when I read all the tutorials and docs talking about google voice. |
| 15:34.17 | [TK]D-Fender | sgo11: You connect whatever you want (who protocol is supported) to * and handle those calls however you want. |
| 15:34.28 | [TK]D-Fender | sgo11: Just don't do that part <- |
| 15:34.41 | [TK]D-Fender | sgo11: leave all that off entirely |
| 15:34.53 | WIMPy | Jepp, just skip the parts you don't need. |
| 15:35.24 | [TK]D-Fender | sgo11: Asterisk is a collection of pieces. You say "here's a SIP phone definition". "Here's an IAX definition for my softphone", etc and do what you want |
| 15:35.51 | [TK]D-Fender | sgo11: Sgo Don't want any 3rd partyy service? Don't go making entires for them. Don't sign up for service |
| 15:35.51 | sgo11 | thanks. what clients and mobile apps can I use to do video/audio talk via my asterisk server? |
| 15:36.08 | [TK]D-Fender | sgo11: Look at the PROTOCOLS that they support |
| 15:36.27 | [TK]D-Fender | sgo11: Never think "brands/models". It's a question of the language they speak |
| 15:36.29 | [TK]D-Fender | ~book |
| 15:36.30 | 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 |
| 15:36.31 | [TK]D-Fender | ^^^ |
| 15:36.48 | [TK]D-Fender | sgo11: Go learn how * works, and what protocol it can speak. Then you can start making your choices |
| 15:37.37 | sgo11 | ok. Thanks a lot. at least, I know Asterisk is what I want now. I will go and read more docs. just to make sure I am on the right direction. thanks. |
| 15:45.10 | *** join/#asterisk deranged (Bit@lolnerd.net) |
| 15:54.45 | *** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 15:57.58 | sgo11 | what is the difference btw incredible pbx and elastix ? |
| 16:00.32 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 16:08.44 | *** join/#asterisk airjump (~Thunderbi@p4FDF06C8.dip0.t-ipconnect.de) |
| 16:09.00 | *** join/#asterisk Greenlight (wluke@cpc18-dund12-2-0-cust490.16-4.cable.virginm.net) |
| 16:09.14 | Greenlight | Afternoon |
| 16:12.40 | Greenlight | Had 11.16.0 deadlock on me a few times over the last couple of weeks. Tis a rather heavily loaded system. I've managed to capture a backtrace the last time it occured. Would greatly appriciate any pointers on how to track down where it's occuring. http://pastebin.com/UaUk29nR if anyone's got a few moments to take a look |
| 16:12.47 | *** join/#asterisk mirton (~jazz@mirtoff.ru) |
| 16:13.44 | Synthase_ | Greenlight: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines |
| 16:14.05 | Rico | is $DIALSTATUS r/w or just read ? |
| 16:14.46 | WIMPy | RO |
| 16:15.30 | Rico | sh*t |
| 16:16.00 | WIMPy | That is, you can probably write to it as to any other variable, but it doesn't have any special meaning or so. What are you trying to do? |
| 16:16.02 | Greenlight | Yup will get a issue raised, as I have in the past when tracking down deadlocks, but am happy to have a dig into the code just was hoping for some pointers on where to start looking |
| 16:17.24 | Rico | WIMPy: trying to set dialstatus=BUSY if DEVSTATE is BUSY |
| 16:17.42 | Rico | it would make my agi more simple |
| 16:18.36 | WIMPy | Just don't Dial if you don't want to. |
| 16:18.56 | Rico | yes, that's right |
| 16:19.24 | WIMPy | And for whatever happens next in your dialplan see above. You can probably use it like a normal variable. |
| 16:21.10 | WIMPy | Greenlight: If yu can run your installation with thread debugging enabled, that would probably help a lot so you can add 'core show locks|threads|taskprocessors'. |
| 16:21.36 | Greenlight | Yea, i tried that earlier in the week... Box died in 5 mins |
| 16:21.56 | Greenlight | This is a Dell 720, dual Xeon 192GB mem... so hefty box too |
| 16:22.05 | WIMPy | Did you gather the informatin from that? |
| 16:22.09 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-josrwycidqvfbuxt) |
| 16:22.26 | Greenlight | Naa it didn't die to the deadlock situation... the overhead from DEBUG_THREADS was too much |
| 16:22.29 | Rico | [TK]D-Fender: it works, thanks |
| 16:22.32 | Rico | (busylevel) |
| 16:22.38 | WIMPy | What do you put in all that RAM? |
| 16:22.58 | WIMPy | Greenlight: Well, that's bad. |
| 16:23.40 | Greenlight | BUt I've at least got a backtrace with it deadlocked |
| 16:23.41 | WIMPy | had the ipression that thread debugging has become a lot more lightweight again. |
| 16:23.53 | Greenlight | I thouught so too... which is why i was happy to enable it. |
| 16:24.05 | Greenlight | It didn't end well :S |
| 16:24.57 | Greenlight | The box is busy thought... since it was restarted after last deadlock, umm 30 mins ago, 50k calls processed |
| 16:25.14 | Synthase_ | There was a condition that would cause a lock in chan_sip if you ran in production with DEBUG_THREADS enabled, but that's been resolved. The performance hit is still going to be there, regardless. |
| 16:25.17 | WIMPy | It might also help to find out whate exactely doesn't work any more and what still works. |
| 16:26.15 | *** join/#asterisk airjump (~Thunderbi@p4FDF06C8.dip0.t-ipconnect.de) |
| 16:26.20 | [TK]D-Fender | sgo11what is the difference btw incredible pbx and elastix ? <- Shit vs crap. |
| 16:27.38 | sgo11 | [TK]D-Fender, They seem providing an easy way to setup asterisk for a newbie like me. I may try incredible pbx on ubuntu. :) |
| 16:27.40 | *** join/#asterisk mirton (~jazz@mirtoff.ru) |
| 16:27.42 | Greenlight | Seems channels could still be created. But the Manager interface stopped passing any data whatsoever. A "core restart" hung the CLI completely after |
| 16:28.35 | Greenlight | I can see a number of threads inside a poll() after ast_waitfor_nandfds |
| 16:28.36 | WIMPy | Greenlight: Yes the locked up consoles are annoying. But you can still use 'asterisk -x'. |
| 16:30.32 | Greenlight | And another bunch stuck around the audiohook stuff |
| 16:31.31 | [TK]D-Fender | sgo11: Except for all of the extra structures they creat that you have no need of. No control over who can dial who, etc. |
| 16:35.56 | sgo11 | [TK]D-Fender, no control over who can dial who? I thought they are created on the top of asterisk. less feature than asterisk? I am just afraid it will take too much time on learning how to configure asterisk. Some people comment that asterisk is like a linux kernel and normal user should not learn and should go and use incredible pbx or elastix. |
| 16:36.16 | [TK]D-Fender | sgo11: You talk about "on top of asterisk" in a completely vague capacity. |
| 16:37.22 | [TK]D-Fender | sgo11: Your dialplan who is allowed to call what, and what happens when anything is dialed. GUI's have their own rules for this and those 2 are both based on FreePBX where everyone is always allowed to call everyone, etc |
| 16:40.47 | [TK]D-Fender | Elastix is worse still for using a hacked up and embedded form of FreePBX for this. |
| 16:41.24 | sgo11 | [TK]D-Fender, ok. got it. thanks. I will read more doc and decide which way I should go. I basically read this thread: http://community.spiceworks.com/topic/270839-i-want-to-try-asterisk-where-do-i-start and most people suggest to use a packaged version, especial the guy called 'Scott' |
| 16:42.22 | [TK]D-Fender | sgo11: "Some guy called 'Scott'" must be the WORST for of "guess this must be authoritative!" I've heard in a long time... |
| 16:42.48 | [TK]D-Fender | On some 3rd party misc blog site |
| 16:43.23 | sgo11 | [TK]D-Fender, ^_^ |
| 16:43.24 | WIMPy | Hello. My name is Scott. |
| 16:44.09 | sgo11 | lol. |
| 16:44.58 | mirton | c/lear |
| 16:45.04 | mirton | oops |
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| 16:50.35 | sgo11 | [TK]D-Fender, thanks a lot for your help. I gotta go. I will try to setup asterisk alone. if that is too difficult, I may go with incredible pbx. |
| 16:51.02 | [TK]D-Fender | sgo11: I'd recommend getting FreePBX's bundled ISO over that |
| 16:51.09 | [TK]D-Fender | sgo11: Far better support |
| 16:51.47 | sgo11 | [TK]D-Fender, got it. thanks. I will look for it. see you. |
| 16:51.59 | [TK]D-Fender | sgo11: And before you get ANYTHING... go read the book to understand the protocols * speaks first and plan what your CLIENTS will be base don that |
| 16:52.36 | sgo11 | [TK]D-Fender, yeah, I will do that and did some search on clients. |
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| 17:34.36 | *** join/#asterisk karelk (~karel@84-72-164-65.dclient.hispeed.ch) |
| 17:36.24 | Moggie | Hi, I'm trying to confirm what it says here please: https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration |
| 17:36.29 | Moggie | It seems to suggest that in realtime mode, Asterisk cannot support things like subscriptions, voicemail notifications, and NAT keep alives. |
| 17:36.36 | Moggie | But maybe I miss-understood. |
| 17:37.58 | [TK]D-Fender | Where does it say that on that page? |
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| 17:57.34 | Moggie | [TK]D-Fender: in the realtime SIP friends section: "The SIP realtime objects are users and peers that are loaded in memory when needed, then deleted. This means that Asterisk currently can't handle voicemail notification and NAT keepalives for these peers." |
| 18:04.23 | [TK]D-Fender | Moggie: "With caching, the device stays in memory for a specified time. More information about this is to be found in the sip.conf sample file." <- I believe this may be related to getting it "functional" |
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| 18:31.55 | Moggie | [TK]D-Fender: ah, right that's interesting. |
| 18:32.57 | Moggie | [TK]D-Fender: I'm still a little confused though. If a device is sending NAT keep-alive messages, for example, is that device not still needed and therefore should not be deleted from memory? |
| 18:36.57 | Moggie | However, if asterisk always keeps the device in memory, it would never reload it fron the database and thus would never pick up DB changes, hrm. |
| 18:45.56 | Moggie | If I understand correctly, setting the cache expire time to 1 hour (the same as the registration expire time) could be a workaround. |
| 18:46.30 | Moggie | For up to 1 hour, Asterisk should still process subscription / NAT keep-alive messages for the device? |
| 18:46.53 | [TK]D-Fender | I don't know the fine point myself as I haven't worked with this before. Just passing on what I see in case it may come in handy |
| 18:48.17 | Moggie | Understood, I appreciate your perspective, thanks. It's tricky to properly interpret the meaning and side effects of settings sometimes :\ |
| 18:52.03 | Moggie | I wanted to get a second opinion in case I'd gotten it wrong, that's why I came here :) |
| 18:54.44 | [TK]D-Fender | Look for a third & fourth ;) |
| 18:56.37 | Moggie | I would love to, but I can't seem to register on the forum (it wont send me a registration confirmation and password), and you're the only person who's talked back to me on here lol. |
| 18:57.14 | [TK]D-Fender | Ask again in a few hours |
| 19:00.21 | *** join/#asterisk Visage (~visage@pdpc/supporter/active/visage) |
| 19:00.46 | TazzNZ | Moggie [TK]D-Fender - I think what is meant there is that the voicemail/etc modules can't be moved to realtime |
| 19:01.14 | TazzNZ | I think that you move "sip.conf" into realtime, but leave the rest as per normal |
| 19:04.51 | *** join/#asterisk areski (~areski@80.174.128.88.dyn.user.ono.com) |
| 19:06.35 | Moggie | TazzNZ: Interesting, you understand it to mean voicemail, subscriptions, NAT keep-alive is still supported but only for sip.conf users? |
| 19:08.33 | Moggie | I read the comment for the rtcachefriends configuration option saying: "Cache realtime friends by adding them to the internal list just like friends added from the config file only on a as-needed basis? (yes|no)" |
| 19:09.48 | Moggie | Perhaps suggesting that when realtime devices from the database are cached, they behave just like devices / peers from sip.conf; them I presume that, while cached - voicemail, subscriptions, and NAT keep-alive should work as normal for regular sip.conf devices? |
| 19:19.40 | TazzNZ | Moggie: no, I understand it to be that "sip.conf" is in realtime config, but the rest of the "stuff" is in the files/where they are now |
| 19:20.20 | TazzNZ | it makes no sense that something like subscriptions would stop working, purely because the device is now loaded from a DB rather than a file, imho |
| 19:20.50 | TazzNZ | keep in mind that you still need to issue a reload command to see the changes |
| 19:21.18 | TazzNZ | if you insert a row into "your sip.conf db", it won't magically appear in asterisk |
| 19:21.33 | TazzNZ | from memory, the only thing that will do that is the dialplan |
| 19:21.50 | TazzNZ | which is also they only thing, imho, that makes sense to do in realtime |
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| 19:54.48 | *** join/#asterisk ddegendt (~smuxi@78-20-43-73.access.telenet.be) |
| 19:55.34 | ddegendt | Hi, I was wondering if someone was able to setup PJSIP with LDAP integration? Can't seem to find a good res_ldap.conf. Auth keeps going wrong :-) |
| 20:04.47 | *** join/#asterisk karelk (~karel@84-72-164-65.dclient.hispeed.ch) |
| 20:11.10 | avb | Moggie: rtpcachefriend works somehow |
| 20:11.27 | avb | if you want to refresh you can do a cronjob with asterisk -rx'sip reload' |
| 20:11.35 | avb | say every minute |
| 20:11.48 | avb | works pretty good |
| 20:12.09 | avb | i would love to get rid of the mess but seems there is no straight solution here |
| 20:12.29 | WIMPy | that might produce quite som load. |
| 20:12.40 | avb | like what? |
| 20:12.49 | WIMPy | And you would re-register every time. |
| 20:12.57 | avb | indeed |
| 20:13.04 | avb | thats a con |
| 20:13.27 | PHunter | why not check for changes in files, and reload when nessecary? |
| 20:13.39 | PHunter | necessary even.. |
| 20:13.52 | avb | im not sure why Moggie wants to have up 2 date changes |
| 20:13.56 | WIMPy | How do the changes get there? |
| 20:14.30 | avb | if not in cache device will try to register, it will be queried from the database |
| 20:15.06 | avb | the only issue i see with sip realtime is that sip show peers do not show everything, but just online users |
| 20:15.13 | PHunter | like regen of sip files and reload? |
| 20:15.37 | WIMPy | PHunter: Sounds like a plan. |
| 20:15.51 | avb | that will work |
| 20:15.57 | avb | but id take realtime |
| 20:16.23 | PHunter | well realtime is relative tbh |
| 20:16.36 | PHunter | if reloading constantly would cause issues, setup a slave script. |
| 20:16.51 | PHunter | Watch the files housing the config, when it changes, issue the command. |
| 20:17.02 | avb | as I told, there is no need to reload usually |
| 20:17.38 | PHunter | Perhaps im missing something. |
| 20:17.45 | avb | you need to reload if changes were made to a registered peer |
| 20:17.54 | PHunter | Are we talking sip endpoint configuration anre reloading for them to take effect? |
| 20:17.54 | avb | but there is a workaround |
| 20:18.14 | avb | endpoint and peers |
| 20:18.17 | PHunter | ok |
| 20:18.49 | avb | from voip-info |
| 20:18.51 | avb | Update : use "sip prune realtime PEERNAME" then "sip show peer PEERNAME load" to flush the peer and reload from db - (Voicemeup) |
| 20:18.52 | PHunter | As far as the peers and endpoints grabbing a change, i am not sure how that would work, but reloading should only occur when there is a change. |
| 20:18.55 | avb | http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip |
| 20:19.14 | avb | im using this in production for couple years |
| 20:19.18 | avb | works perfect |
| 20:19.51 | PHunter | Oh yeah |
| 20:20.01 | PHunter | I use conf files still because, well, it works. |
| 20:20.21 | avb | i used to use them long time ago :) |
| 20:20.28 | PHunter | I have had dreams of making all of our config web driven.. |
| 20:20.30 | avb | try realtime, i bet you will like it |
| 20:20.59 | PHunter | Its not about me liking it, its about someone dicking with the config who isn't an asterisk tech. |
| 20:21.16 | avb | as a plus you will be able to query endpoins UAs and avialability straight from DB without ami |
| 20:21.28 | PHunter | hehe |
| 20:21.38 | PHunter | you don't want to know what i do with AMI |
| 20:21.46 | PHunter | lets just say node.js is involved. |
| 20:21.47 | avb | well, such guys will ruin anything even if you will give them user rights |
| 20:21.48 | avb | lol |
| 20:21.56 | WIMPy | How is being able to query from a DB an advantage over being actively informed via AMI? |
| 20:22.15 | PHunter | its just delivery method |
| 20:22.24 | PHunter | my boss is old school. Loves DB |
| 20:22.33 | avb | WIMPy: it is if you doing small project and not using ami at all :) |
| 20:22.34 | PHunter | he will poke around in it and do stupid things with the data |
| 20:23.13 | avb | if you want just display list of extensions with its status, its a good part :) |
| 20:23.34 | WIMPy | How does samll fit in with using a DB? |
| 20:23.44 | avb | easy :) |
| 20:24.20 | WIMPy | That's one thing I do and I can't see an easier way than AMI. Plus it's realtime. |
| 20:25.08 | avb | well, if you do CRM which already have sql |
| 20:25.23 | avb | and there is no websockets etc |
| 20:25.56 | WIMPy | Where do websockets come in to the equation? |
| 20:25.56 | avb | i do not want to deal with ami without a need to actually use it |
| 20:26.12 | avb | i thought you are talking realtime |
| 20:26.22 | avb | in the gui |
| 20:26.24 | *** join/#asterisk Garibaldo (~smuxi@177.133.126.166) |
| 20:26.37 | WIMPy | Yes. |
| 20:27.01 | avb | or you want to run background $.post() on timeout? :) |
| 20:27.17 | WIMPy | What? |
| 20:27.21 | avb | nevermind |
| 20:27.41 | WIMPy | I don't need any facy shit to do it. |
| 20:27.59 | ddegendt | Is there any documentation out there to combine PJSIP with res_ldap? Can't seem to figure out if res_ldap is taking any variables at all... |
| 20:28.11 | avb | oh, you is a fan of a 100 xmlhttprequest posts? :-D |
| 20:28.20 | avb | or of a constant page refreshes? |
| 20:28.34 | PHunter | jQuery ftw. |
| 20:28.37 | Moggie | I thought the idea was for Asterisk to check the DB when processing calls, meaning SIP devices and extensions can be added/modified dynamically without the need for a reload each time something changes. |
| 20:29.01 | avb | hides with his databases, jquery, websockets and other fancy shit |
| 20:29.08 | WIMPy | Just a good old delayed refresh. Super simple and effective. |
| 20:29.22 | avb | lol :) |
| 20:29.56 | Moggie | If a reload was always required to refresh changes, there would be no need for deleting objects from memory or implementing a SIP friend caching system? |
| 20:29.58 | WIMPy | It might even work in lynx. Maybe I should try that. |
| 20:30.09 | avb | is too nervious to see on this blinking constantly reloading pages |
| 20:30.39 | avb | Moggie: use "sip prune realtime PEERNAME" then "sip show peer PEERNAME load" to flush the peer and reload from db |
| 20:30.41 | Moggie | There has to be more to it than just loading friends from a database and having to perform a reload every time something changes. |
| 20:30.41 | WIMPy | I don't see anything the like. |
| 20:31.00 | avb | Moggie: that would reload needed peer |
| 20:31.52 | avb | WIMPy: :) well, i will owe u if you will bring me customers with lynx |
| 20:31.56 | WIMPy | Nah, neither lynx nor links wil refresh. :-( |
| 20:31.59 | avb | as a requirement |
| 20:32.01 | Moggie | yeah, but it's a manual process. It doesn't make sense to me that I have to run asterisk commands for each peer every time I update one of them. |
| 20:32.21 | avb | Moggie: why manual? |
| 20:32.25 | avb | cronjob |
| 20:32.50 | Moggie | a cronjob to reload every registered user? that sounds less efficient :( |
| 20:33.09 | avb | echo 'select ext from users where lastupdated > $lastCheckTime'|while read ext;do asterisk -rx'...';done |
| 20:33.29 | avb | echo 'select ext from users where lastupdated > $lastCheckTime'|mysql ... |while read ext;do asterisk -rx'...';done |
| 20:33.48 | avb | Moggie: in reality its not that many edits :) |
| 20:35.02 | avb | WIMPy: im not sure why peoples do not like databases. they are very handy. you do not need to parse |
| 20:35.11 | avb | and you can do guis in no time |
| 20:35.24 | WIMPy | They eat lots of resources. |
| 20:35.25 | avb | and api |
| 20:35.30 | avb | cmon :) |
| 20:35.35 | avb | they do not |
| 20:35.49 | WIMPy | They do. |
| 20:35.58 | avb | used to have a server with mysql and apache on a server with 128mb ram |
| 20:36.01 | avb | they dont |
| 20:36.09 | Moggie | Also, it still doesn't explain why the friend caching / deleting exists. If Asterisk requires commands to be run either manually or via cronjob to pick up changes, why where these settings like rtcachefriends to control dynamic behaviour |
| 20:36.10 | avb | if you know how to cook them |
| 20:36.12 | WIMPy | And you are adding additional possibilities for failure. |
| 20:36.39 | Moggie | Also, it still doesn't explain why the friend caching / deleting exists. If Asterisk requires commands to be run either manually or via cronjob to pick up changes, why were these settings like rtcachefriends to control dynamic behaviour added |
| 20:36.48 | avb | WIMPy: if you are not doing anything, it will be even less possibilities for failure :-D |
| 20:36.48 | Moggie | hopefully that makes more sense, heh |
| 20:37.13 | avb | Moggie: well, thats a great question to realtime developers |
| 20:37.20 | WIMPy | True. So always keep things as simple as possible. |
| 20:38.11 | avb | WIMPy: mysql is a mature product. its 100% reliable. and if we are talking about possibilities of failure, then there is clustering |
| 20:38.38 | avb | if you have no clustering, then for sure conversation about points of failure is pointless |
| 20:39.11 | avb | before mysql guys were using bdb |
| 20:39.24 | avb | berkley db |
| 20:39.44 | avb | its the same mysql with way less possibilities |
| 20:40.01 | avb | look at your sendmail/postfix |
| 20:40.04 | avb | they still use it |
| 20:40.41 | avb | if you trying to act as an old school unix guru, sorry to say, but databases were heavy in use that time as well |
| 20:41.16 | avb | alright, im off to bed. :) have a great day/evening guys |
| 20:41.18 | MaliutaLap | mysql is a piece of shite - go for postgresql |
| 20:41.27 | WIMPy | Databases (as in RDBMS) are great if you're not sure what you're doing. |
| 20:41.39 | avb | lol :) |
| 20:41.55 | avb | tell this to a telco and banks and insurance companies |
| 20:42.02 | MaliutaLap | WIMPy: perfect for DBA's then :P |
| 20:42.25 | avb | MaliutaLap: show me painless master-master cluster on pg and we will talk :) |
| 20:42.51 | WIMPy | Well, if the programmer knew what they were doing they wouldn't need DBAs, would they? |
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| 20:43.37 | avb | funny |
| 20:43.38 | MaliutaLap | WIMPy: most DB systems are so crap they need a DBA to keep them running, regardless of the quality of the programmer |
| 20:43.48 | avb | i were like you like 5 years ago |
| 20:43.56 | MaliutaLap | WIMPy: not that the quality of programmers is getting better |
| 20:44.15 | WIMPy | MaliutaLap: Definitely not at all :-( |
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| 22:08.49 | mrc3 | hey! i keep getting "acf_jabberstatus_read: Resource (null) of buddy [...]" when trying to read status via JABBER_STATUS. i found ASTERISK-23456 but seems to be closed a non-bug. would anyone happen to have any leads? |
| 22:09.41 | mrc3 | that with both asterisk 11 and asterisk 13, running with openfire 3.8.2 as the jabber server |
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| 23:21.31 | smkelly | file: hi |
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