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| 01:13.19 | almostworking | i dont know a ton about , VOIP, however should be able to run a SIP from a unmanaged VPS, connecting , via VPN ? |
| 01:17.09 | WIMPy | sure |
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| 01:18.12 | almostworking | cool, i might do , that, and use old cell as softphone, any good android, paid or free VOIP apps? |
| 01:18.34 | almostworking | ( dedicated router handles the VPN part) |
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| 03:43.25 | ChannelZ | almostworking, CSipSimple is pretty good on Android |
| 03:44.51 | almostworking | ahh, ok cool, thanx, ChannelZ .. .. il add to wishlist now, ( for addding to old nexus later) |
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| 07:57.21 | asteriskbizua | anybody home :) WHO IS FAMILIAR WITH TLS ? |
| 07:58.54 | wdoekes | ~polls |
| 07:58.54 | 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> |
| 07:59.08 | wdoekes | ~ask |
| 07:59.08 | 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. |
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| 08:17.10 | asteriskbizua | OK THE QUESTION IS -- according this article https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial they propose to generate client sert ----I dont understand why i need him ----i have only 1 server and 2 softphones ----and they areworking |
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| 08:29.08 | R4v3n | hello world :) |
| 08:29.57 | R4v3n | I'm here to ask some questions and chew bubblegum, and I'm all outta gum ! |
| 08:30.07 | ChannelZ | exit code -12: syntax error |
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| 08:32.57 | R4v3n | I'm using asterisk 1.4.22 with 2 T0 (ISDN) lines (so a total of 4 lines) since 2008, with thomson SIP phones. Everything is running fine. But now I need to add some videoconference via a polycom realpresence group system, with support for SIP (it works) and H323 (it doesn't work very well) |
| 08:33.35 | WIMPy | Wow. That's old. |
| 08:34.08 | R4v3n | when I try to contact the polycom system using asterisk as a gateway to translate SIP > H323, I join the polycom system via ooh323c but it hang up immedialty |
| 08:34.17 | R4v3n | WIMPy, yep but it works, not connected to the internet |
| 08:34.51 | R4v3n | and when I try to contact do polycom (h323) > SIP phone, it makes asterisk crash immediatly |
| 08:35.25 | R4v3n | I read a lot of topics about ooh323, and I have 2 questions about h323 in general in asterisk |
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| 08:36.25 | ChannelZ | Well apparently it _doesn't_ work |
| 08:36.31 | R4v3n | 1 - Will asterisk (last versions ?) can be used with this kind of system (polycom videoconf) and will let pass the video ? |
| 08:36.42 | R4v3n | 2 - is ooh323 the problem, or asterisk ? |
| 08:36.45 | WIMPy | Well, H.323 support is probably limited to if it works for you, be happy. |
| 08:37.09 | R4v3n | ChannelZ, for the SIP part only, it works well since 2008 ! :) |
| 08:37.23 | WIMPy | And as per usual for limit values of "works". |
| 08:38.21 | R4v3n | WIMPy, so do you think with the last asterisk, I can make the videoconf through h323 work ? Or must I found an alternative like freeswitch or something ? |
| 08:38.50 | R4v3n | last asterisk/last ooh323 i mean |
| 08:39.15 | R4v3n | maybe somebody here already used a h323 videoconference system with asterisk ? |
| 08:39.26 | WIMPy | I have no idea. But talking about 1.4 certainly doesn't make sense. |
| 08:40.12 | R4v3n | yep, sure. I'll install the last in a VM, hoping it will be fast to configure, and it has a "great" h323 support |
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| 08:44.34 | ChannelZ | Call from '' (199.168.102.90:5071) to extension '.0027218139313' rejected -- . ? That's new |
| 08:45.11 | WIMPy | A PHP fraggle having made a mistake? |
| 08:46.52 | ChannelZ | probably |
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| 12:20.06 | stefan27 | if i launch asterisk with vvvg replaced by vvv it will not create a core-dump file upon segmentation fault? |
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| 12:22.16 | asteriskbizua | ANYBODY CAN TELL ME THE TRUTH? why i need to generate second cert while using TLS &&&&&&&&&&&&&&& |
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| 12:29.35 | WIMPy | stefan27: Depends on the contents of your asterisk.conf. |
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| 13:56.10 | RadJackson | Hello , we have asterisk PBX running on a dedicated Server , our server is starting to get slower, or instance when we a context calls an AGI Script that runs mysql queries , once a huge query lags or get stuck, the calls gets blocked, this was an example , sometimes asterisk process stops due to the huge amount of calls being treated simultenously , we would like to upgrade to a higher |
| 13:56.10 | RadJackson | system, something extensible that can be upgraded easily as our IPBX keeps receiving more and more calls , we were about to get a better server , but we may have the same problem in the future, and it takes a lot to migrate the whole system (trunks , sips configs to a new server). What would you suggest for us ? thanks |
| 13:59.51 | [TK]D-Fender | get another server at the same host with direct connectivity for your DB's to run off of. |
| 14:00.39 | RadJackson | The problem is not only related to my mysqldatabase , sometimes asterisk process is running 90% of CPU , its the whole system. I would like to upgrade to a better server in every way not only db |
| 14:01.42 | [TK]D-Fender | And what are you running specifically? |
| 14:03.13 | RadJackson | sometimes we receive 2000 simultenous calls , every call runs an AGI script (thru PHP) so sometimes even apache consumes a lot of CPU , and every PHP call has a mysql connections with a couple of query being runned , so its the whole server getting slower ... |
| 14:03.27 | RadJackson | the idea is to have a better server that could be upgraded easily without migrating |
| 14:04.19 | [TK]D-Fender | Again, what are you running exactly? |
| 14:04.35 | RadJackson | Sorry ,what do you mean by running ? |
| 14:04.54 | [TK]D-Fender | ERSIONS |
| 14:04.56 | [TK]D-Fender | +V |
| 14:05.19 | RadJackson | Asterisk 1.6.2.9-2+squeeze10 |
| 14:06.04 | RadJackson | PHP 5.3.3-7+squeeze19 with Suhosin-Patch |
| 14:06.48 | RadJackson | MYSQL 5.1.49-3 |
| 14:06.52 | [TK]D-Fender | Your are tragically outdated. 5 entire branches behind current. Do you know how many stability and perfomance improvements there have been since then? |
| 14:07.33 | [TK]D-Fender | That system has a number of staggering vulnerabilities as it is and that branch is not supported |
| 14:07.51 | [TK]D-Fender | This is an automatic strike to your running platform |
| 14:07.52 | WIMPy | If you have issues with calling AGIs, optimize your AGIs or try to move some functionality to something else. |
| 14:08.15 | RadJackson | In case i upgrade to the last Asterisk version , do i risk some DialPlan Problems? |
| 14:09.12 | WIMPy | unlikely, yet possible. |
| 14:09.23 | [TK]D-Fender | And if you still have too many calls, distribute them between servers |
| 14:11.06 | RadJackson | Ok , so upgrading to the last Asterisk version , optimising AGI scripts , may be migrating DB to another server at the same host , and in the end dispatching calls between multiple servers. |
| 14:11.19 | RadJackson | I will try this , Thanks you [TK]D-Fender & WIMPy |
| 14:12.00 | RadJackson | One last question, is there a CLI command to upgrade asterisk ,or should i install the new version |
| 14:12.15 | [TK]D-Fender | Install |
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| 14:18.20 | RadJackson | What about VPS ? cloud virtual private servers, they can be upgraded easily , would it be a good idea to run asteirsk on such servers ? |
| 14:19.52 | WIMPy | A pretty common thing. |
| 14:21.10 | RadJackson | do you have any idea about virtualisation softwares? Proxmox VE3 , Windows Hyper V Server or Microsoft Azure ? any idea about the best between these? |
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| 14:25.42 | [TK]D-Fender | RadJackson: First 2 both run very close to bare-metal speeds, but it still depends how they are provisioned |
| 14:25.57 | [TK]D-Fender | RadJackson: I don't know of Azure at all. |
| 14:26.21 | [TK]D-Fender | (aside from being sure I've heard the name a few times with no idea what it was) |
| 14:28.06 | file | it's the name of Microsoft |
| 14:28.07 | file | er |
| 14:28.18 | file | Microsoft's cloud services stuff - virtual machines and other stuff |
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| 14:29.33 | RadJackson | What about VMWare ? |
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| 14:37.42 | [TK]D-Fender | ESXi is good |
| 14:38.43 | [TK]D-Fender | RadJackson: However you keep talking about performance issues.... and now you're talking about sharing that physical hardware with OTHER THINGS. Are you actively trying to fail? |
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| 14:41.09 | marceloamorim | guys, there is some way to put "or" on pattern like _[9|99]XXX to accept 9XXX or 99XXX? |
| 14:42.07 | WIMPy | Good point. |
| 14:42.28 | WIMPy | marceloamorim: No, only for single digits. |
| 14:42.43 | marceloamorim | =( thx WIMPy |
| 14:43.31 | sekil | is SLA supported nowadays on *? |
| 14:45.24 | [TK]D-Fender | marceloamorim: No |
| 14:45.34 | [TK]D-Fender | sekil: No |
| 14:48.16 | RadJackson | [TK]D-Fender the reason i asked about Virtualisation is not for the current server , but may be a better extensible server , i'm speaking precisely about this : |
| 14:48.17 | RadJackson | http://www.voip-info.org/wiki/view/Asterisk+and+Virtual+Private+Servers |
| 14:48.20 | Katty | hello my asterisk does not work at all how to fix pls is urgent thx. |
| 14:50.30 | [TK]D-Fender | RadJackson: Yes, and that implies your host could be over-provisioned and you're screwed. If you're willing to risk that then I don't think you're taking your solution too seriously |
| 14:52.40 | RadJackson | Ok , There are too many solutions on the internet , reason why i am asking... |
| 14:53.26 | file | there are options, they may or may not be solutions to the problem |
| 14:53.27 | [TK]D-Fender | RadJackson: And you are looking at things that cut into performance while mentioning issues you have. |
| 14:53.59 | RadJackson | I dont recognize the best solution, yet the most extensible, i will stick to the solution you have suggested, two servers , one frontal for asterisk and another for DB being hosted at the same host |
| 14:54.04 | RadJackson | Thanks a lot :) |
| 14:54.33 | [TK]D-Fender | RadJackson: Not just that.... mulktiple * servers and split the calls up |
| 14:55.53 | RadJackson | by multiple servers you mean asterisk being installed on each one ? |
| 14:56.16 | [TK]D-Fender | "Multiple Asterisk Servers" is pretty clear... |
| 14:56.33 | [TK]D-Fender | More than 8 server running Asterisk. Split your calls between them |
| 14:56.36 | RadJackson | hands [TK]D-Fender a beer |
| 14:56.36 | [TK]D-Fender | * |
| 14:56.52 | RadJackson | Ok i will think about that. sounds a good idea |
| 14:56.56 | [TK]D-Fender | "Too much for 1 machine to handle? GET ANOTHER" |
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| 15:09.10 | RadJackson | [TK]D-Fender sorry one more question :-) , Suppose i have two asterisk servers, and i split the calls everytime to a server , for such example both servers have to be synchronised , especially extensions.conf DialPlan code , no ? is there a way to synchronize both servers ? I do not care about DB as it will be hosted on a 3rd serveur same host |
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| 15:16.42 | RadJackson | slaps [TK]D-Fender around a bit with a large trout |
| 15:21.31 | [TK]D-Fender | RadJackson: You would think they would process calls the same way, wouldn't you? Why would this even be a question? |
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| 15:24.46 | RadJackson | May be there is some technology that would let me sunchronize dialplan automatically , instead of editing the two servers and reloading dialplan for every modification |
| 15:24.52 | gwenloh | Hello |
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| 15:30.12 | [TK]D-Fender | RadJackson: Are you a server admin at all? |
| 15:31.08 | [TK]D-Fender | RadJackson: I'm getting a strong impression of "no" from every point in this conversation so far... |
| 15:33.56 | RadJackson | well , i am, and i am able to synchronise files as a "linux server admin" , but my question is from VOIP and ASTERISK point of view, instead of synchronizing every folder every agi script i thought MAY BE you would suggest me a technology , as there are several ones for instance "kamailio" that would do it automatically without linux server knowledge |
| 15:34.28 | tuxx- | wow. :D |
| 15:34.32 | RadJackson | anyway thanks for the help |
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| 15:38.58 | Ibrahim22 | Hi, how would one scale an ARI application (running on node). Node has the ability to create workers (node-cluster), but creating websocket connection for each worker would not work, as they would all connect to the same events and just process every event n times with n being the number of workers. |
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| 15:40.03 | mjordan | Ibrahim22: there's a number of ways you could approach it. Are you looking to scale out Asterisk, or scale out the node.js app? |
| 15:42.37 | pabelanger | SCALE ALL THE THINGS |
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| 15:44.10 | mjordan | pabelanger: yup, but different solutions for each side of that equation :-) |
| 15:44.21 | pabelanger | mjordan, indeed |
| 15:44.35 | mjordan | so, speaking very generally, if you want to scale out your node.js apps, you'd want to proxy/farm out the work from each websocket connection |
| 15:44.59 | mjordan | that is, have a node.js app that pulls information off the websocket, figures out where it's supposed to go, and then dispatches that to another node.js process/app for handling |
| 15:45.05 | pabelanger | Ibrahim22, we've had some good success running an ARI proxy on the local asterisk box, which converts events over a messagebus to our ARI cluster (apps). Seems to be working well |
| 15:45.15 | pabelanger | but ya, some sort of ws proxy could work too |
| 15:45.36 | pabelanger | nginx has support for a ws proxy |
| 15:45.39 | mjordan | yup |
| 15:45.44 | pabelanger | using it with kamailio right now and webrtc |
| 15:45.50 | mjordan | yay HTTP and HTTP-related friends |
| 15:45.56 | pabelanger | since I need wss |
| 15:46.47 | *** part/#asterisk jwr___ (~quassel@205.196.167.54) |
| 15:47.11 | [TK]D-Fender | RadJackson: Asterisk is just a bunch of configuration files, and wherever you do your AGI's from. You should already know how to move those around. Asterisk doesn't know anythiing about anything unless yous et it up to., and syncing files isn't part of what it does. |
| 15:47.33 | mjordan | pabelanger: just to clarify - Asterisk supports WSS. Do you need nginx for WSS with Kamailio? |
| 15:48.01 | pabelanger | mjordan, ya. I don't use websockets at asterisk, it just processes SIP |
| 15:48.07 | mjordan | gotchya |
| 15:48.22 | mjordan | pabelanger: 1.8 FL? ;-) |
| 15:50.22 | [TK]D-Fender | 1.8 EOL :D |
| 15:50.55 | pabelanger | mjordan, indeed |
| 15:51.03 | pabelanger | working on 13 testing now |
| 15:51.13 | pabelanger | but, likely be a few more months before I migrate |
| 15:51.16 | Ibrahim22 | @mjordan I'm looking to scale ARI |
| 15:51.47 | pabelanger | Ibrahim22, what are you building? |
| 15:51.53 | pabelanger | mostly curious |
| 15:52.12 | Ibrahim22 | I'm creating a replacement for app_queue and app_agent_pool |
| 15:53.00 | pabelanger | :) |
| 15:53.06 | pabelanger | welcome to the future |
| 15:53.23 | [TK]D-Fender | No, this is the present. |
| 15:53.28 | [TK]D-Fender | When will then be now? |
| 15:53.29 | [TK]D-Fender | SOON! |
| 15:53.42 | [TK]D-Fender | </spaceballs> |
| 15:54.02 | mjordan | [TK]D-Fender: +1 |
| 15:54.53 | Ibrahim22 | Haha, I'm frustrated with the secrecy that app_queue and app_agent_pool runs on, no customization, so ARI is perfect to build my own implementation :) if it works well in production, i'll publish it on github for scrutiny and severe criticism :) |
| 15:55.38 | Ibrahim22 | @mjordan what are the ways to scale ARI then? |
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| 15:56.29 | Ibrahim22 | @pabelanger I already thought of a proxy that retransmits events, but the bottleneck still exists at the proxy... |
| 15:57.00 | pabelanger | Ibrahim22, we're doing a queue app too, for python. For scaling out ARI, we decided not to worry bout scaling that part. Basically, we do a think ARI proxy on running on each asterisk box. Which, we exposed over a messagebus to our backend. Everything in our apps is already externalized into redis / database so if one app goes down, another is there to fetch the data. |
| 15:57.06 | pabelanger | I think I have a diagram around some place |
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| 15:58.47 | Ibrahim22 | But you have several (of the same?) apps running, each connected to websockets? How are you then preventing duplicate tasks being run? |
| 15:59.47 | rrittgarn | use filters on what you're processing in node |
| 15:59.52 | rrittgarn | have each process do something different event wise |
| 15:59.56 | rrittgarn | and then handle it all together |
| 16:00.00 | rrittgarn | (or bring it back together) |
| 16:00.15 | rrittgarn | (how we handle stuff listening to AMI events with multiple Node servers) |
| 16:00.31 | pabelanger | Ibrahim22, no |
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| 16:00.56 | pabelanger | we have multiple apps running |
| 16:01.06 | pabelanger | then, using the messagebus to round robin messages to each app |
| 16:01.33 | pabelanger | then use that to transmit / receive info |
| 16:02.02 | pabelanger | Ibrahim22, from the WS point of view |
| 16:02.13 | pabelanger | we only have 1 connection per asterisk box we have |
| 16:02.20 | Ibrahim22 | Ah, okay, I see |
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| 16:03.15 | Ibrahim22 | Your proxy receives all the events, and sends the messages down to the apps (round robin wise) |
| 16:03.31 | Ibrahim22 | But if your proxy fails, your entire infrastructure fails, am I right? |
| 16:04.14 | pabelanger | well, there is always points of failure |
| 16:04.16 | pabelanger | but yes, |
| 16:04.25 | pabelanger | we cluster the proxy |
| 16:04.30 | pabelanger | err |
| 16:04.32 | pabelanger | messagebus |
| 16:05.15 | Ibrahim22 | How can you cluster your proxy/messagebus, wouldn't every worker receive the same event that it needs to send out to the apps ? |
| 16:05.31 | pabelanger | personally, I think it is easier to make services redundant (redis, rabbitmq, mysql) then dealing with it in your application |
| 16:06.17 | pabelanger | Ibrahim22, here is the diagram I was talking about: http://wiki.kickstand-project.org/pabelanger?action=AttachFile&do=view&target=payload.png |
| 16:06.30 | pabelanger | little old, but the basic concept of our queue |
| 16:08.58 | Ibrahim22 | Okay, so it is only 1 proxy per box |
| 16:09.33 | pabelanger | yes |
| 16:09.40 | pabelanger | we bind to localhost for our WS |
| 16:09.48 | pabelanger | and if our WS is down, or not connected |
| 16:09.59 | pabelanger | we 302 redirect traffic to another asterisk gox |
| 16:10.01 | Ibrahim22 | It is easier, but is it production ready. What if your proxy has a segmentation fault or a memory leak or god forbid unhandled exception |
| 16:10.01 | pabelanger | box* |
| 16:10.15 | pabelanger | then, remove said box from the incoming route |
| 16:10.43 | Ibrahim22 | Hm, okay, so you scale by box and not by app |
| 16:10.50 | pabelanger | basically |
| 16:11.05 | pabelanger | If something happens to our box, for what ever reason |
| 16:11.13 | pabelanger | we destroy it and create a new one |
| 16:11.20 | Ibrahim22 | Is that the only way to scale ARI ? |
| 16:11.26 | pabelanger | deal with troubleshooting later |
| 16:11.29 | pabelanger | no |
| 16:11.40 | pabelanger | this is the way we choose to do it. there is many ways to scale |
| 16:12.07 | pabelanger | ARI is still young, so you'll likely have to do alot of work yourself and see what works |
| 16:12.11 | Ibrahim22 | Can you tell me about the others ? :) I would like to learn them, I can't seem to find info on the wiki or even Google |
| 16:13.08 | pabelanger | you are better to ask on the mailing list: http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev |
| 16:13.28 | pabelanger | last time I talked to people, there are only a handful of people doing ARI |
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| 16:14.05 | pabelanger | I know digium is doing a ARI voicemail application, mjordan likely has more information on it. But not sure what their plans are for scaling it up is. |
| 16:14.45 | Ibrahim22 | I'm glad that is the case, it means we are the early adopters generations from now will talk about in awe and envy |
| 16:14.50 | Ibrahim22 | :) |
| 16:15.44 | [TK]D-Fender | Or will simply serve as a lesson to others! |
| 16:15.48 | [TK]D-Fender | :D |
| 16:16.13 | [TK]D-Fender | s/lesson/warning |
| 16:16.17 | [TK]D-Fender | (for clarity) |
| 16:18.22 | pabelanger | Ibrahim22, There are a handful of people interested in ARI queue apps. I'd love to get everybody together and agree on some common format between apps.; like events / rest API. But, everybody has their own purpose for there apps, so going to be hard to coordinate. |
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| 16:19.32 | pabelanger | ~collectdebug |
| 16:19.32 | infobot | it has been said that collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 16:20.39 | Ibrahim22 | Maybe start a thread on the asterisk-app-dev mailing list to coördinate? |
| 16:21.08 | Ibrahim22 | @pabelanger |
| 16:21.46 | sekil | [TK]D-Fender: sorry you said that shared line appearance is not supported? the sla.conf stuff is then just a little hack I guess |
| 16:22.18 | pabelanger | Ibrahim22, I won't paste here (don't want to flood channel with info [not cool]), but PM if you want some more info about what we are trying to do. |
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| 16:24.39 | [TK]D-Fender | sekil: "Little" is not a word I've used historically, but might serve for now. |
| 16:25.34 | [TK]D-Fender | sekil: If you think you're setting "SLA" as an option on your phone you will be greatly disappointed |
| 16:27.57 | klonstein | hi, what is the best way to enforce transcoding without negotiate a large pool defined into sip.conf |
| 16:28.35 | [TK]D-Fender | klonstein: Your question is unclear, please rephrase. |
| 16:32.39 | sekil | [TK]D-Fender: I hear you...no support for that presence enhancements then..like Call-ID etc. |
| 16:32.52 | sekil | [TK]D-Fender: thanks |
| 16:38.45 | R4v3n | WIMPy, hey, with asterisk 12 and h323plus, I got H323 voice calls, but no video, do you know it a little or not at all ? |
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| 17:00.19 | WIMPy | R4v3n: I have never tried video. |
| 17:00.55 | R4v3n | ok. I've the feeling i'm the only one who tried to do things with videoconferencing systems :x |
| 17:01.28 | WIMPy | Definitely not. |
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| 17:09.18 | f0ner00t | Hello.. I am getting the following error when loading chan_sip.so can someone help me? Module 'chan_sip.so' was not compiled with the same compile-time options as this version of Asterisk. |
| 17:10.21 | [TK]D-Fender | f0ner00t: recompile with the proper oiptions then. You did a broken combo install |
| 17:10.40 | WIMPy | Your asteris and the chan_sip module are not from the same install. |
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| 17:16.09 | f0ner00t | [TK]D-Fender: or WIMPy How do I fix ? |
| 17:16.33 | WIMPy | Correctly install Asterisk. |
| 17:16.59 | WIMPy | Maybe you have multiple versions installed. If so, remove them. |
| 17:17.08 | f0ner00t | I am thinking thats it |
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| 17:28.56 | f0ner00t | bah |
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| 18:13.33 | klonstein | [TK]D-Fender: I'm trying to force setup codecs between 2 legs, is only to get a clear dialog without a large pool of codecs |
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| 18:18.26 | [TK]D-Fender | klonstein: So set your codecs properly in each peer |
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| 18:24.58 | klonstein | [TK]D-Fender: currently I have a different setup for the same ip / I mean: [xx] ip: 1,2,3,4 allow: g729^ulaw ... [yy] ip: 1,2,3,4 allow: g723 ..... this setup is for outboubd purposes , so , I'm trying to avoid this setup |
| 18:26.01 | [TK]D-Fender | klonstein: For outbound you can force the codec to be used via SIP_CODEC |
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| 19:36.53 | ChannelZ-Wk | Hmm. I'm pretty sure ChanSpy screws up RTCP stats. |
| 19:37.46 | file | I don't see how it... would... |
| 19:39.34 | ChannelZ-Wk | Well I need to do some more tests to be sure that's what is causing it, but: |
| 19:39.40 | ChannelZ-Wk | Channel Stats: ssrc=141577309;themssrc=1888495861;lp=0;rxjitter=0.000000;rxcount=1761;txjitter=0.000193;txcount=22106;rlp=126358;rtt=0.156046 |
| 19:41.06 | ChannelZ-Wk | rlp=12635 ? I don't think that could possibly be. |
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| 19:42.39 | ChannelZ-Wk | Actually several things about that don't make sense to me, rxcount seems wrong (that's received voice packets right?) |
| 19:43.00 | file | points at mjordan |
| 19:43.43 | ChannelZ-Wk | looks at the call log... |
| 19:47.55 | ChannelZ-Wk | Maybe I don't know what those counters mean. A call that just ended had an rxcount of 212 and a txcount of 0 yet it was a ~3 minute call |
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| 19:48.49 | ChannelZ-Wk | But if those were RTP packets, 212 would only be ~4 seconds (20ms packets for ulaw aren't they?) |
| 19:49.21 | file | um |
| 19:49.26 | Cuzner | Who knows Polycom Digitmaps well? I think I may have stumbled across a bug, and I want to ensure it's not just a misunderstanding I have with how digitmaps work... |
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| 19:50.43 | Cuzner | Why would dialplan.digitmap="" allow me to off-hook press as many keys as I want without sending, but dialplan.digitmap="*8T" send immediately on a single keypress unless it's *8 (then it waits the default timeOut of 3 seconds). |
| 19:51.40 | file | ChannelZ-Wk, this is stuff I have put into long term storage |
| 19:51.47 | file | (the knowledge, that is) |
| 19:52.28 | [TK]D-Fender | Cuzner: because *8 is explicit and when you fail the explicit... |
| 19:52.31 | ChannelZ-Wk | Cuzner, I imagine because 'nothing' means 'anything', whereas 'something' means whatever it is. |
| 19:52.41 | Cuzner | interesting |
| 19:52.42 | [TK]D-Fender | Cuzner: when it's blank... well.. no rule = no wrong. |
| 19:52.51 | ChannelZ-Wk | Does it actually send the call through or fail with the *8T when you don't dial *? |
| 19:53.00 | [TK]D-Fender | ChWhen you do anythign ELSE |
| 19:53.21 | Cuzner | so if I specify at least 1 pattern or more, I have to create a map for everything. |
| 19:53.31 | ChannelZ-Wk | (IE when you press a key you probably get a busy, but did it actually place a call or fail locally on the phone?) |
| 19:53.52 | Cuzner | ChannelZ-Wk: If I press *# other than 8, it will send immediately |
| 19:54.00 | Cuzner | instead of waiting for the 3 second timeout |
| 19:54.18 | ChannelZ-Wk | Oh OK that's different. I thought you meant you were dialing like 4 or something random |
| 19:54.31 | Cuzner | ChannelZ-Wk: that too |
| 19:54.38 | Cuzner | any other single keypress will send immediately |
| 19:55.02 | ChannelZ-Wk | Well that one I don't understand then |
| 19:56.13 | Cuzner | no, i honestly think you've unknowingly pointed me in the right direction |
| 19:56.14 | ChannelZ-Wk | Sounds like a strange default fallthrough behavior.. but I don't know anything specifically about how Polycoms behave specifically |
| 19:56.55 | Cuzner | these are explicit rules, and I don't have a rule that covers the pattern I'm trying to dial |
| 19:57.08 | [TK]D-Fender | Cuznerinstead of waiting for the 3 second timeout <- because the timeout is if you DO match. You do not. It is therefor and INVALID dial, and has its own handler |
| 19:57.17 | [TK]D-Fender | Cuzner: Read their admin guide |
| 19:57.18 | Cuzner | if I create 1 rule, i have to create a rule for every type of dial pattern. |
| 19:57.41 | Cuzner | [TK]D-Fender: yeah, been reading through this http://support.polycom.com/global/documents/support/technical/products/voice/Understanding_Digit_Maps_Tech_Tip.pdf |
| 19:59.02 | Cuzner | My original problem was that a user opened a support ticket stating that they couldn't off-hook dial more than 5 digits, unless 0 was the first keypress. They were using this digitmap: <dialplan dialplan.digitmap="*8T|*xxxT|*xxxx|**xxT|**xxxx|#xxxT|#xxxx|112T|[2-9]11T|0T|00xxx.T|0[1-9]xxxxx.T|xxxxT"> |
| 19:59.35 | [TK]D-Fender | Correct |
| 19:59.55 | [TK]D-Fender | indeed they can't. So don't complain when it's following the rules |
| 19:59.58 | Cuzner | There is no rule to send after 5 keypresses, but I think because xxxxT should be xxxx.T, this is what's causing it. |
| 20:00.12 | file | phone digit maps: black magic |
| 20:00.43 | Cuzner | [TK]D-Fender: yes, in the end i believe it's because i misunderstood the default behaviour when there is no match. |
| 20:00.48 | ChannelZ-Wk | phone numbers/extension crossovers are hard |
| 20:00.51 | [TK]D-Fender | file: Or in the case of Polycom : clearly documented and does what it says. |
| 20:01.08 | file | still black magic |
| 20:01.16 | Cuzner | omg it's SO CLEAR |
| 20:01.31 | Cuzner | so clear i felt the need to come in here and spam your channel even. |
| 20:01.36 | [TK]D-Fender | yup! |
| 20:02.26 | [TK]D-Fender | One pattern has a "." and allows whatever length... the other doesn't and fails. #empiricalevidence |
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| 20:05.14 | Cuzner | [TK]D-Fender: thanks man... you might be a condescending prick 99% of the time, but you always point me in the right direction in a timeley fashion. |
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| 20:06.47 | c0ldg0ld | I haven't yet been able to google any information about this but perhaps I'm using the wrong keywords. Are templates allowed in res_digium_phone.conf? If so what's the syntax? Same as sip.conf? |
| 20:07.24 | c0ldg0ld | After just a few phones you repeat a lot of stuff that apparently can't be global (like timezone) and I'd like to be able to clean up my conf files |
| 20:08.19 | [TK]D-Fender | Cuzner: Not quite so bad! I have made no real swipe at you here. Chill... |
| 20:08.32 | file | c0ldg0ld, it uses the same config framework as everything else - so templates will work |
| 20:08.47 | c0ldg0ld | file: excellent. Thanks |
| 20:09.05 | Cuzner | [TK]D-Fender: oh, no offense intended my friend :) |
| 20:14.40 | Cuzner | impossibleMatchHandling |
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| 20:15.09 | Cuzner | That's what's doing it, set it to "2" and on these "invalid" strings people dial, it will wait for the user to hit Send |
| 20:20.09 | [TK]D-Fender | yup |
| 20:20.33 | [TK]D-Fender | I allow absolutely everything + wait on all of my phones and let my PBX do it's own thing |
| 20:20.33 | Penguin | [TK]D-Fender: its |
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| 20:42.05 | Cuzner | [TK]D-Fender: so I proposed both solutions to the powers that be, they prefer xxx.T over impossiblematchhandling=2 :P |
| 20:43.22 | [TK]D-Fender | Cuzner: I also do both. It's the "STFU And Do It Dialplan" |
| 20:45.47 | Cuzner | our dialplan is nightmare inducing |
| 20:46.34 | Cuzner | i'm 99% certain we have the largest asterisk dialplan in production, globally. |
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| 20:50.40 | jvwjgames | Hi |
| 20:53.15 | jvwjgames | how do i how do i add speech recognition to asterisk |
| 20:53.22 | jvwjgames | and elastix |
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| 21:04.17 | jvwjgames | anyone here to help |
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| 21:16.31 | [TK]D-Fender | Go pick an engine and install it. |
| 21:17.08 | [TK]D-Fender | jvwjgames: Elastix is its own mess. Getting that GUI to accept and give you GUI options is something to take up in their support channel. |
| 21:27.38 | *** join/#asterisk gp (~IceChat77@c-24-30-17-154.hsd1.ga.comcast.net) |
| 21:29.15 | gp | Any recommendations for figuring out why srtp encryption causes 1-way audio for the encrypted endpoint? 2-way audio for both endpoints without srtp |
| 21:32.02 | [TK]D-Fender | checkout time, heading home... |
| 21:33.49 | jvwjgames | GP how did you setup srtp |
| 21:33.59 | jvwjgames | i want to know how to set that up |
| 21:35.40 | gp | Well it isn't working 100% atm for me. It also depends on your linux distro and environment |
| 21:36.08 | gp | Here is a quick summary that I posted to a message board: http://pbxinaflash.com/community/index.php?threads/install-and-build-asterisk-with-srtp-support.16224/ |
| 21:36.13 | jvwjgames | i have asterisk 1.8 centos |
| 21:36.22 | jvwjgames | using elastix |
| 21:36.29 | gp | This is helpful: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
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| 21:37.25 | jvwjgames | also what about speech recognition |
| 21:37.44 | gp | Unfortunately I am still in the learning phase =( |
| 21:37.51 | jvwjgames | ok |
| 21:44.26 | jvwjgames | do you use a gui |
| 21:44.45 | jvwjgames | or what do you use for your asterisk administration |
| 21:44.50 | jvwjgames | gp |
| 21:47.27 | Cuzner | vi and sublime :) |
| 21:47.42 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 21:47.57 | Cuzner | though I really wish sublime had proper asterisk dialplan syntax highlighting, but that's asking way too much. |
| 21:48.47 | jvwjgames | gp: what do you use for asterisk administration |
| 21:49.17 | gp | Currently IncrediblePBX11.4 on CentOS6.5. http://nerdvittles.com/?p=10079 It is very quick to get something up and running. I would consider myself a beginner at this though so I don't know if you should take advice from me. But I don't think this channel is about that though. |
| 21:50.03 | jvwjgames | ok |
| 21:50.29 | gp | This forum has helpful people for getting started with it: http://pbxinaflash.com/community/index.php |
| 21:50.30 | Cuzner | jvwjgames: you don't like elastix? it has a gui front end, no? |
| 21:51.22 | Cuzner | FreePBX is another one... It used to be just a front-end to asterisk, but i think they pitch it as an integrated "distro" now. |
| 21:51.50 | gp | IncrediblePBX11.4 uses the FreePBX GUI. It just adds a lot of helpful apps out of the box |
| 21:51.58 | Cuzner | ahh |
| 21:52.07 | jvwjgames | i like elastix |
| 21:52.18 | gp | I think in Version 12 they swap GUIs though |
| 21:52.27 | Cuzner | the only thing I know about elastix is that it's german |
| 21:52.31 | jvwjgames | but it doesn't have the awesome freature codes or voice recognition i want |
| 21:53.26 | gp | jvwjgames: IncrediblePBX is very easy to work with. Maybe you would like it better. The tutorials for getting up and running are fairly comprehensive |
| 21:54.20 | jvwjgames | ok |
| 21:54.35 | jvwjgames | where is the direct link you donloaded it from |
| 21:55.18 | gp | The link I posted to nerdvittles is the blog of sorts that the developers post releases and tutorials on. I linked you directly to the one I used |
| 21:55.44 | gp | I must afk for a bit |
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| 22:15.29 | Baylink | QUERY: Does Dahdi stuff have its own channel? 2.10.1 (in PIAF-Green/FPBX11/Asterisk11) is throwing me "FXO PCI Master abort" errors after boot with 1 or 3 cards installed; ABit mobo. wcfxo module only; Ambient clone cards which were working in an older release. |
| 22:16.06 | f0ner00t | Any ideas why when I set up streaming MOH I get this error in my CLLI monmp3thread: poll() failed: Interrupted system call |
| 22:16.26 | f0ner00t | Sorry Full Error: res_musiconhold.c:695 monmp3thread: poll() failed: Interrupted system call |
| 22:30.53 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:33.16 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 22:37.36 | newtonr | Baylink, if you are using a GUI to administrate Asterisk, then it is best to ask about your DAHDI issues in that forum/channel as the troubleshooting steps will likely involved use of your management interface. |
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| 22:45.49 | Baylink | In fact, while PIAF is supposed to have a DAHDI config module, it doesn't appear to be there, so I was expecting to have to fall back on manual config, newtonr. Driver console/dmesg errors don't seem to fall in that scope anyway; do they? |
| 22:46.17 | Baylink | It's been back to just when dahdi got that name I had to touch it last, and that cluster was all PRI. |
| 22:47.41 | newtonr | Baylink, FreePBX does have a DAHDI module. No idea if it supports your clone cards : http://wiki.freepbx.org/display/F2/DAHDI+Configs |
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| 22:52.56 | newtonr | Baylink, If that is the old school X100P design I don't think that is supported at all in DAHDI. Perhaps back when it was Zaptel. However I'm not 100% sure. |
| 22:54.15 | Baylink | newtonr: There's still a wcfxo module, and it is active; has it gotten pickier about what clones it will talk to now? |
| 22:56.23 | Baylink | These cards were in service in two older releases; one was PIAF 1.7, with whatever that ran; Asterisk 1.6, I think. The drive, alas, died. Hmmm. It *was* zaptel, and I have the zapata.conf. Can I transplant that by a rename, or do I have to modify the contents? Can pastebin. |
| 22:56.56 | Baylink | http://pastebin.com/nNB4MfTX |
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| 22:58.35 | Baylink | I'm dropping that in dahdi-channels.conf, and I'll see what happens. |
| 22:59.25 | newtonr | I'm unsure about the state of the wcfxo module. Configuration will likely have to be reworked, but not much. If you just google around for some modern DAHDI configurations it'll help. I'd hope the manufacturer has updated docs for DAHDI |
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| 23:00.30 | newtonr | /etc/dahdi/system.conf and your /etc/asterisk/* dahdi files will both have to be configured. I haven't touched any cards in a long time and especially not with clones, so here is where my help ends. :D |
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| 23:01.35 | Baylink | Understood; I often provide "as much help as I have available" as well. :-) |
| 23:02.24 | Baylink | And as it happens, that conf file has at least killed off my Abort errors. |
| 23:02.54 | Baylink | Well, I got *one*, but not one every 3 seconds or so. |
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| 23:08.43 | f0ner00t | Any ideas why when I set up streaming MOH I get this error in my CLLI monmp3thread: poll() failed: Interrupted system call |
| 23:08.49 | f0ner00t | Sorry Full Error: res_musiconhold.c:695 monmp3thread: poll() failed: Interrupted system call |
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| 23:11.45 | newtonr | f0ner00t, probably better off just posting to the users mail list, rather than reposting here. |
| 23:12.55 | newtonr | If you do. You'll probably want to pastebin a larger section of log than just the one line. |
| 23:13.36 | newtonr | Probably want 'DEBUG' type logger channels on as well: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 23:18.41 | f0ner00t | Newtonr ; I was hoping something it was something easy |
| 23:19.14 | newtonr | f0ner00t, Maybe it is! I've never setup streaming music on hold. |
| 23:19.39 | newtonr | Maybe the right person just isn't online right now. |
| 23:21.56 | f0ner00t | newtonr :) I'll ask another time thank you |
| 23:22.05 | f0ner00t | or go to the link |
| 23:22.32 | newtonr | go to the link and post on the users list. :D only takes a minute |
| 23:22.55 | newtonr | If you have never used the mail lists before: http://www.asterisk.org/community/discuss |
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| 23:34.48 | f0ner00t | newton : I have I'm just looking where the mailing lists are |
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