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| 01:23.44 | rockydoggy | Hi everyone, i'm looking for help to build a daily script. The script gonna ping a specific hostname and update my sip.conf "permit=" ip address. I has been looking many website last week and cannot find any example. thanks for helping |
| 01:24.57 | MaliutaLap | That's not really a * thing, more of a bash/perl scipting thing |
| 01:25.58 | rockydoggy | i was thinking about a bash script that I can add to my cron service |
| 01:27.19 | rockydoggy | I know : dig +short google.ca works to get the ip. but cannot make it write the file. |
| 01:28.55 | MaliutaLap | create a template file and the use a sed command to do what you want into a temp file, cp the temp file into place, restart/reload * |
| 01:29.03 | MaliutaLap | problem solvered |
| 01:30.24 | [TK]D-Fender | Or forget "permit" altogether and just set that hostname as the "host" |
| 01:30.53 | [TK]D-Fender | because a fixed host would only permit them anyway |
| 01:32.18 | MaliutaLap | [TK]D-Fender: doesn't the hostname thing only do a dns lookup on load? which is bad for dynamic ip's |
| 01:32.47 | MaliutaLap | [TK]D-Fender: but still your method would only need a sip reload to refresh |
| 01:39.18 | rockydoggy | MaliutaLap Does SIP reload hang up current call? |
| 01:43.01 | MaliutaLap | not AFAIK |
| 01:43.23 | MaliutaLap | [TK]D-Fender might have a more definitive answer |
| 01:43.34 | [TK]D-Fender | rockydoggy> MaliutaLap Does SIP reload hang up current call? <- no |
| 01:44.16 | [TK]D-Fender | <MaliutaLap> [TK]D-Fender: doesn't the hostname thing only do a dns lookup on load? which is bad for dynamic ip's <- I suspect it does a DNS lookup on demand as it goes.... |
| 01:44.20 | [TK]D-Fender | Could be wrong. |
| 01:44.39 | [TK]D-Fender | easy enough to script of course if you wanted to go the other way |
| 01:45.15 | rockydoggy | [TK]D-Fender : Thank you. Just to confirm, I do replace my host=dynamic by host=myhostname.com and it should only allow that host. |
| 01:45.17 | rockydoggy | correct? |
| 01:45.27 | [TK]D-Fender | it will. |
| 01:45.36 | rockydoggy | Very appreciate I will test it. |
| 01:45.37 | MaliutaLap | [TK]D-Fender: now you're destroying my illusions of the world! I thought you were all knowing, all seeing! :( |
| 01:45.39 | rockydoggy | Thanks again |
| 01:45.44 | [TK]D-Fender | the question MaliutaLap brought up was if it only looked on load. |
| 01:46.04 | [TK]D-Fender | then again.... you could reload on interval as well which is easier to script |
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| 01:48.35 | MaliutaLap | [TK]D-Fender: my understanding was that it did the hostname lookup once, on load (unless "host=" was set to dynamic). I also understood that using a hostname was not compatible with dynamic |
| 01:50.13 | MaliutaLap | which is part of my understanding of the hostname lookup method |
| 01:53.42 | rockydoggy | It does resolve the correct hostname IP, but receiving error : chan_sip.c:28235 handle_request_register: Registration from '<sip:MYEXTENSION@MYSERVERIP>' failed for 'MYREMOTEPHONEEXTIP:1024' - Peer is not supposed to register |
| 01:55.38 | [TK]D-Fender | indeed |
| 01:55.50 | [TK]D-Fender | because if the host is fixed then they are not ALLOWED to register |
| 01:55.57 | [TK]D-Fender | registering should not be a requirement |
| 01:56.02 | [TK]D-Fender | because you KNOW where they are |
| 01:57.53 | MaliutaLap | I think I leave my cisco on dynamic so I can have it register each line |
| 02:02.17 | [TK]D-Fender | You think you know why you did things? |
| 02:02.19 | [TK]D-Fender | ;) |
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| 02:05.14 | MaliutaLap | 'twas a while ago |
| 02:05.29 | MaliutaLap | didn't document my reasoning |
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| 05:23.11 | linocisco | hi all from west of the globe, if you are not sleeping , let me ask you one |
| 05:24.04 | linocisco | is there any aseterisk based free hotel solution? |
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| 05:45.18 | [TK]D-Fender | linocisco, Who's going to make an entire solution for that kind of business and give it away for FREE? |
| 05:47.14 | linocisco | the question is Yes Or No only? CentOS is free, Ubuntu is free claimed to be free forver so far. It is not easy to compare but I just wonder it exists or not. |
| 05:53.11 | MaliutaLap | Debian is free, Asterisk is free |
| 05:53.19 | MaliutaLap | build it yourself is free :) |
| 05:53.26 | [TK]D-Fender | linocisco, Because all of those are GENERIC |
| 05:54.30 | [TK]D-Fender | CoentOS is just one distro. |
| 05:54.33 | [TK]D-Fender | And a BAD example |
| 05:54.36 | linocisco | because some call center solutions and CRM to be used with asterisk/Elastix is free |
| 05:54.45 | [TK]D-Fender | it is a REBRANDING of RHEL which is a commercial distro |
| 05:54.48 | linocisco | I am just wondering |
| 05:56.16 | [TK]D-Fender | linocisco, that is a VERY specific industry. People who'd make a system to run that entire specific business would charge for the work it takes to implement. |
| 05:56.45 | MaliutaLap | I doubt there is anything pre-built for a specific industry that is going to be free, or fit the business model as every business in that industry |
| 05:56.47 | [TK]D-Fender | How many free billing solutions are there for *? |
| 05:56.58 | [TK]D-Fender | Not much. And most are crap |
| 05:57.11 | linocisco | [TK]D-Fender, yes. I agree but call centers solutions and CRMs are found commerciially and usable for SMBs somehow with basic features |
| 05:57.12 | MaliutaLap | [TK]D-Fender: including the roll your own? Infinity ;) |
| 05:57.13 | [TK]D-Fender | Actually, that'd be "justabout all" |
| 05:58.10 | linocisco | I dont mean complete perfect all in one solution. Just for basic most common features that would benefit for small business |
| 05:59.45 | [TK]D-Fender | There is no "'basic" for this. |
| 05:59.50 | [TK]D-Fender | Those pieces need to fit. |
| 06:00.01 | [TK]D-Fender | Call limiting, billing, checkin/out scheduling, etc |
| 06:00.12 | [TK]D-Fender | "hotel business" != "generic business. |
| 06:00.31 | [TK]D-Fender | Generic Business = FreePBX <- |
| 06:03.30 | MaliutaLap | eww |
| 06:04.59 | MaliutaLap | I would figure the profit margin on a hotel providing that kind of service would justify the cost to get someone to come in and do something up to fit with their model |
| 06:07.13 | [TK]D-Fender | Hence why I can't imagine anyone going through that much specific effort to give away the product for free |
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| 07:03.37 | coppice | I was in a hotel recently that seemed to have shiny new Nortel phones. I wonder where those were dug up from? |
| 07:04.07 | MaliutaLap | coppice: did you plug into the phone port and see what you could find? |
| 07:05.01 | MaliutaLap | coppice: I know this guy who travels alot for work, he tends to do things like get root on every IPTV setop box in the hotel |
| 07:05.25 | MaliutaLap | actually this is one of his other "things" - http://blog.docbert.org/spoofing-public-wifi-networks/ |
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| 07:24.06 | TriJetScud | coppice: probably old stock "avaya" phones that they never bothered to rebadge as nortel |
| 07:24.15 | TriJetScud | it could very well be refurbed phones |
| 07:25.16 | coppice | refurbed phones always look what they are |
| 07:26.13 | TriJetScud | true |
| 07:26.40 | TriJetScud | coppice: what models did you run into? |
| 07:26.50 | TriJetScud | were they the norstar's or the newer SIP phones |
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| 07:52.59 | seik0 | continued investigation of asterisk sudden low performance (before it worked well, I even tested it under load). Is it normal, that system generates a lot of "TLB shutdown" and "Resheduling interrupts" while running asterisk? |
| 07:53.20 | seik0 | For a moment I can't test system with no asterisk running |
| 07:53.37 | seik0 | it's a bit productive |
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| 08:26.55 | WIMPy | Doesn't sound normal to me. |
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| 08:29.31 | seik0 | WIMPy, again stopped on that memory leak causing that |
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| 08:30.42 | seik0 | have two asterisk machines with memory leak problem and 3 without. Former has similar symptoms |
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| 08:32.59 | WIMPy | Memory leaks are evil. Do you know whereabout they are? |
| 08:33.50 | seik0 | they are in oracle odbc driver |
| 08:34.02 | seik0 | oracle odbc driver for x86 |
| 08:34.53 | seik0 | however, we do not have leak in them in asterisk 1.4, but have in 1.8 |
| 08:35.07 | seik0 | current workaround is to use x64 |
| 08:35.20 | seik0 | x64 oracle odbc driver does not show memory leak |
| 08:35.26 | WIMPy | Is the Oracle part as old as the Asterisks? |
| 08:36.04 | seik0 | =), no, tried most recent drivers |
| 08:37.11 | seik0 | btw, asterisk 1.4 quite stable. missing some sweet features maybe |
| 08:38.02 | WIMPy | Missing a lot. |
| 08:38.24 | seik0 | missing supported dahdi, though |
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| 08:42.05 | seik0 | the only thing is that leak was all the way from the beginning and asterisk became slow yesterday |
| 08:42.23 | WIMPy | I can remember some random features of 1.4 with zaptel. |
| 08:42.36 | seik0 | zaptel in 1.4 quite bad |
| 08:42.45 | WIMPy | IHMO 1.8 was the first usable version. Even if 1.4 was indeed stable. |
| 08:43.18 | seik0 | especially when it lacks resources - it hangs and very hard to restore drivers without reboot |
| 08:43.50 | WIMPy | That's not somethign I remember. |
| 08:46.07 | seik0 | that's the only issue with zaptel I know =), but I experienced only a few installations |
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| 09:10.17 | MaliutaLap | zaptel??? haven't moved to dahdi? |
| 09:12.00 | WIMPy | You'd need a reasonably recent Asterisk version to do so. |
| 09:14.40 | MaliutaLap | god 1.4 is ancient |
| 09:20.37 | seik0 | once we will surely migrate =) |
| 09:22.36 | WIMPy | Unless your dialplan is even older, that should be easy. |
| 09:25.10 | seik0 | dialplan is in db, migrating is easy. moreover, we have 1.4 and 1.8 processing same dialplan using db views |
| 09:25.18 | MaliutaLap | WIMPy: you are making me want to do the IRC equivalent of stick my fingers in my ears and going "la la la la" |
| 09:25.21 | seik0 | harder to migrate hardware |
| 09:26.02 | MaliutaLap | seik0: what hardware are you migrating? |
| 09:26.47 | seik0 | not migrating, but rather would be good to migrate ) |
| 09:26.56 | seik0 | new machine, maybe virtual |
| 09:27.08 | seik0 | mainly e1 |
| 09:27.31 | WIMPy | Does that work now? |
| 09:27.49 | seik0 | work |
| 09:28.09 | MaliutaLap | zaptel/dahdi devices in a VM? are you using pci pass through or soemething |
| 09:28.12 | seik0 | just keep cpu a bit idle for zaptel not to hang entire system |
| 09:28.14 | WIMPy | I mean virtualization and dahdi? |
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| 09:28.33 | MaliutaLap | ^^ my point |
| 09:28.49 | seik0 | we have working e1-eth in production and it's ok |
| 09:29.13 | seik0 | virtualisation sometimes have issues with old kernels |
| 09:29.17 | WIMPy | Ugh |
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| 11:39.40 | BigKristoff | Hi :) Should Asterisk detect fax through faxdetect option if CNG signal is comming in RTFC 2833 RTP Event? |
| 11:40.36 | eirirs | ppl still using fax in 2015? :) |
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| 11:41.36 | BigKristoff | yeah... unfortunately ;) |
| 11:42.28 | eirirs | you better laugh at them |
| 11:42.39 | eirirs | we should make fax-user ashamed of it, so they start moving on |
| 11:42.48 | eirirs | users* |
| 11:44.26 | BigKristoff | i Will, but now they are laughing ant me ;) Fax detecrtion works perfeclty fine when CNG signal comes in RTP stream. But in my case I have CNG in seperate event (RTP Event) and... I'm not shure if this is a Asterisk Bug or Asterisk supports only CNG in Audio |
| 11:46.41 | coppice | its quite unusual to see it sent as RFC4733 event. asterisk didn't support it the last time I looked, but that might have changed. |
| 11:49.07 | BigKristoff | For now I have one machine that is using RFC. |
| 11:49.32 | BigKristoff | its WorkCentre 3220 |
| 11:51.29 | coppice | that's a good point. quite a few of those types of machines are probably capable of sending those events, yet the problem seldom arises. I wonder why |
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| 11:55.25 | BigKristoff | i should mentioned that it is Asterisk 11.16.0 |
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| 12:32.28 | pbxman | hello |
| 12:36.21 | BigKristoff | hello |
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| 14:05.23 | adeel | on * 1.8, if fax detection is enabled on a sip peer, is it possible to NOT have * jump to the fax extension? |
| 14:06.01 | [TK]D-Fender | No, that is the entire point of detection. |
| 14:06.08 | [TK]D-Fender | So that it can jump to where it needs to jump to |
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| 14:08.37 | adeel | thanks |
| 14:09.15 | WIMPy | Why do you enable fax detection if you don't want it? |
| 14:09.23 | adeel | WIMPy: i do want it...just not always |
| 14:10.18 | adeel | but i see i can override it on a per peer definition...so let's see what happens when i set the endpoint to have faxdetect=no |
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| 15:02.49 | russlevy | hi, i'm having an issue with the opus codec. a user seems to have figured out how to crash the server when connecting through websockets |
| 15:02.57 | russlevy | this is the last message before the crash: http://pastebin.com/SaiUNGxX |
| 15:04.04 | WIMPy | ~collectdebug |
| 15:04.04 | infobot | collectdebug is, like, a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 15:04.30 | WIMPy | If it crashes, make sure you have core dumps enabled. |
| 15:05.10 | russlevy | don't have coredump. how do i enable? |
| 15:05.30 | *** join/#asterisk overyander (~Jeff@12.49.160.131) |
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| 15:06.02 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 15:06.58 | russlevy | going to wait for us to reproduce, debug logging enabled to file now |
| 15:10.45 | WIMPy | The wiki page should explain it all. Analyzing the core dump is the key when it crashes. |
| 15:12.51 | russlevy | thx -- core dumps enabled, debugging on, so will try to crash again |
| 15:16.18 | newtonr | That page doesn't explain core dumps or getting a backtrace |
| 15:16.20 | newtonr | here you go https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 15:17.41 | *** part/#asterisk monsterco (~monsterco@TOROON474AW-LP130-01-1177674788.dsl.bell.ca) |
| 15:21.09 | WIMPy | Oh. Do we have a short cut for that in the bot? |
| 15:21.30 | russlevy | I have the core dump, but i guess i need to recompile with better backtraces? |
| 15:24.50 | WIMPy | Probably, but look at what you've got. |
| 15:26.17 | russlevy | how would i find the issue? |
| 15:26.29 | russlevy | here's the only extra error message that seems to mean anything: [2015-06-23 08:19:24] WARNING[17108][C-0000002a] translate.c: Out of buffer space |
| 15:26.47 | russlevy | (in addition to the same ones i have in the pastebin) |
| 15:26.47 | WIMPy | That looks usefull. |
| 15:27.11 | russlevy | as well as "[2015-06-23 08:19:24] DEBUG[17108][C-0000002a] translate.c: Sample size different 160 vs 960" |
| 15:27.43 | russlevy | but this is all happening while i'm getting the opus corrupted stream messages |
| 15:28.30 | *** join/#asterisk aquaguy (~Arkaitz@167.83-213-48.dynamic.clientes.euskaltel.es) |
| 15:28.56 | russlevy | which compiler flags should i enable i menuconfig? dont_optimize and debug_threads? |
| 15:35.56 | *** join/#asterisk probonic (~philevans@2a02:40:2:0:4a5b:39ff:feaa:3b6f) |
| 15:42.06 | mastahh | I do not know how many of you guys sell Asterisk to your customers as a VoIP solution, but what do you normally charge with Asterisk being open-source? Do you just literally charge for professional services? |
| 15:45.19 | [TK]D-Fender | 100 flavours of "depends" |
| 15:45.23 | [TK]D-Fender | To ogeneric |
| 15:45.40 | [TK]D-Fender | Some people use * as small hooking for other systems jsut as an "announcement" tool |
| 15:45.45 | [TK]D-Fender | Others setup full call-centers |
| 15:45.50 | [TK]D-Fender | Other setup as SMB PBX's. |
| 15:45.55 | [TK]D-Fender | Others as fax gateways |
| 15:46.03 | [TK]D-Fender | EVERY kind of project is 100% different than another |
| 15:46.14 | [TK]D-Fender | I use Asterisk as a JUKEBOX, and a COFFEE MAKER |
| 15:46.17 | [TK]D-Fender | YMMV |
| 15:46.42 | [TK]D-Fender | The question is at best a crappy poll you shouldn't expect too much actionable info from. |
| 15:48.32 | mastahh | I am looking to use Asterisk as a complete alternative to Cisco Unified Communication. |
| 15:50.04 | [TK]D-Fender | Depends if you're offering it as in-house. Maybe "on-site" as a finished "good" |
| 15:50.18 | [TK]D-Fender | Go look what other companies are making similar offerings |
| 15:50.28 | [TK]D-Fender | doesn't matter if it"s * behind the scenes or not |
| 15:51.01 | mastahh | With Asterisk, you are not having to pay through your teeth for licensing. That is where vendors such as Cisco really get you. |
| 15:51.30 | *** join/#asterisk gusto (~gusto@2001:4c50:62f:f213:8219:34ff:fecf:17f0) |
| 15:51.37 | mastahh | You end up paying half of the total cost of the solution mainly on licensing features where Asterisk can be built to any specification on pretty much any budget and still get the same functionality of Cisco. |
| 15:52.17 | [TK]D-Fender | Not quite |
| 15:52.25 | [TK]D-Fender | Because * is only one little part of the equation |
| 15:52.33 | [TK]D-Fender | [11:42]mastahhI do not know how many of you guys sell Asterisk to your customers as a VoIP solution, but what do you normally charge with Asterisk being open-source? Do you just literally charge for professional services? |
| 15:52.36 | [TK]D-Fender | And look at your question. |
| 15:52.45 | [TK]D-Fender | There are lots of other UC solutions out there. |
| 15:52.57 | [TK]D-Fender | They ALSO cost a fair amount and are almost all licensed per-user, etc |
| 15:53.01 | [TK]D-Fender | this is a BUSINESS QUESTION |
| 15:53.13 | [TK]D-Fender | This has nothing to do with the fact 1 piece of the engine is "free" |
| 15:53.22 | [TK]D-Fender | you are paying for a finished product |
| 15:53.35 | mastahh | I see your point. |
| 15:53.37 | [TK]D-Fender | How cheap do you think you are going to offer your solution vs the rest of the market? |
| 15:53.41 | [TK]D-Fender | That's entirely up to you. |
| 15:53.47 | [TK]D-Fender | But there is no relationship |
| 15:54.05 | mastahh | That is true. |
| 15:54.12 | [TK]D-Fender | You should go do some market research on UC in general. |
| 15:54.22 | [TK]D-Fender | The client generally rarely knows or cares what's on the backend |
| 15:54.47 | mastahh | Definitely, I was trying to artituclate that and you have hit the nail on the head [TK]D-Fender |
| 15:55.26 | mastahh | I am very Cisco-centric and have come from a Cisco partner and now I am coming in to a world where as long as the solution does the job, it does not matter whose or what badge is on the kit. |
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| 16:06.12 | *** part/#asterisk aandrew (~tzanger@gromit.mixdown.ca) |
| 16:20.01 | carrar | mastahh: Is free, so we don't charge our customer anything!! :) |
| 16:20.05 | carrar | it's |
| 16:22.02 | mastahh | That is what I thought. You would simply charge for the time required to provision servers and configure the system to meet the needs of the customer, carrar |
| 16:22.56 | [TK]D-Fender | mastahh: nope |
| 16:23.08 | [TK]D-Fender | mastahh: Have you looked at all those other UC service vendors? |
| 16:23.23 | [TK]D-Fender | mastahh: those are ways it gets sold |
| 16:23.40 | [TK]D-Fender | mastahh: REmove Asterisk from you mind completely concerning this |
| 16:24.01 | mastahh | [TK]D-Fender: are you referencing vendors such as Mitel, ShoreTel etc? |
| 16:24.03 | [TK]D-Fender | mastahh: You want to know how the market sells UC? Look at how those companies DO. |
| 16:24.05 | *** part/#asterisk aquaguy (~Arkaitz@167.83-213-48.dynamic.clientes.euskaltel.es) |
| 16:24.35 | [TK]D-Fender | mastahh: I'm referring to any UC vendor. There are HUNDREDS |
| 16:25.03 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 16:25.18 | [TK]D-Fender | mastahh: Permanently ignore the fact that Asterisk could be one piece of YOUR offering, and that it happens to be free. |
| 16:25.54 | [TK]D-Fender | mastahh: Tons oc ompanies use BSD as a backing OS in their otherwise closed solutions. Has NO impact on how they sell the finished complete solution |
| 16:26.08 | mastahh | That is true. I never thought of it that way |
| 16:26.52 | mastahh | Another way to look at it is high-end storage platforms such as NetApp utilise FreeBSD -- free being the key word |
| 16:26.57 | mastahh | and they charge a premium for their product. |
| 16:27.16 | mastahh | [TK]D-Fender I see your point, completely now. |
| 16:27.22 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-spsudaidcfczicnk) |
| 16:27.23 | [TK]D-Fender | So forget about this aspect entirely. |
| 16:28.04 | [TK]D-Fender | "What kind of packagaing and licensing terms do other UC vendors sell their products via?" |
| 16:31.42 | *** join/#asterisk t4nk093 (57ed466d@gateway/web/freenode/ip.87.237.70.109) |
| 16:34.57 | t4nk093 | hi, I am scratching my head over something and just wanted to get the thoughts of an expert, I have one side audio on SIP when dialing between an asterisk phone system and a Unify X8 - OSB V1, I captured a dump of the nic and can see that the rtp returns are attempting to go to an ip address outside of my local range |
| 16:36.07 | t4nk093 | I dont appear to be able to change the STUN details on the Unify (the PBX with out audio) so was wondering if there was any way to repoint the rtp traffic to the Unify local address |
| 16:36.21 | [TK]D-Fender | Fix your peer |
| 16:36.31 | [TK]D-Fender | You tell it whether to trust what the other side says or not |
| 16:36.39 | t4nk093 | ok |
| 16:36.44 | [TK]D-Fender | nat= <------------------- |
| 16:36.54 | t4nk093 | yeh nat=no |
| 16:36.54 | [TK]D-Fender | canreinvite= <------------ |
| 16:37.05 | t4nk093 | canreinvite=yes |
| 16:37.10 | [TK]D-Fender | nat=no = trust whatever they say |
| 16:37.16 | t4nk093 | ok |
| 16:37.38 | [TK]D-Fender | Which doesn't seem to be working out too well for you so far |
| 16:37.51 | t4nk093 | no isn't |
| 16:38.15 | t4nk093 | i will try changing it |
| 16:38.19 | t4nk093 | thanks for your help |
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| 17:24.22 | aquaguy | Hello. I'm receiving calls from by SIP provider, in extensions.conf I've added the extension to answer those calls, I've tried answering the call and playing the music on hold and it works. |
| 17:24.44 | aquaguy | The problem is that instead of playing music on hold I want to pass the call to an IVR and that does not work |
| 17:24.45 | aquaguy | http://pastebin.com/aBq4se0s |
| 17:26.14 | WIMPy | The Goto is wrong and your use of Background and MusicOnHold doesn't make sense. |
| 17:27.41 | linocisco | how to provision grandstream GXP1405 phone? |
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| 17:28.33 | aquaguy | WIMPy: there is a WatExten and some extensions bellow that, I havent posted it cause I though it was irrelevant for the question I was asking |
| 17:31.09 | hexanol | on asterisk 13.4.0, we were doing some tests with app_queue, and we found a segfault on the following scenario |
| 17:31.32 | hexanol | a caller calls a queue, then a queue member answer the calls; when the caller does a blind transfer to someone else, asterisk segaults |
| 17:31.59 | hexanol | someone is aware of this bug ? we've look at issues.asterisk.org, but did not found any related ticket |
| 17:32.12 | *** join/#asterisk airjump (~Thunderbi@p4FDF09DC.dip0.t-ipconnect.de) |
| 17:32.52 | [TK]D-Fender | hexanol: Congratulations, you win the earlybird prize! |
| 17:33.02 | WIMPy | aquaguy: The two toints are still valid. |
| 17:33.19 | hexanol | alright, it's just that sometimes I try looking for issues on the tracker and don't find them |
| 17:33.30 | hexanol | so I'm asking before opening a new issue |
| 17:34.31 | hexanol | because I'm a bit surprised this hasn't been found, since this happens in a systematic way, but oh well |
| 17:36.32 | [TK]D-Fender | aquaguy: MoH breaks your Background and is "blocking". This is not a proper IVR. |
| 17:36.46 | [TK]D-Fender | aquaguy: Also note that they cannot interrupt your firs Playback |
| 17:46.45 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 18:00.14 | hexanol | I've opened a new issue: https://issues.asterisk.org/jira/browse/ASTERISK-25185 |
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| 18:03.18 | *** join/#asterisk pjensen00 (~per@ip-69-178-218-66.far.ideaone.net) |
| 18:05.14 | pjensen00 | Quick question, is there a way to listen for SIP responses in ARI? For instance, if i'm doing a dial I want to be able to see if I get a SIP 404 or a 502 error |
| 18:05.28 | *** join/#asterisk vader- (~Adium@204.183.88.4) |
| 18:05.30 | mjordan | hexanol: please generate a full backtrace for the issues. |
| 18:05.42 | mjordan | hexanol: instructions for generating backtraces can be found on the wiki: |
| 18:06.07 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
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| 18:10.46 | hexanol | mjordan: I've removed the previous files and replaced them with the gdb command from the wiki |
| 18:11.37 | mjordan | apparently the caller is NULL in those cases. That's odd. |
| 18:17.54 | DivideBy0 | pjensen00: I don't believe so, but others may know more |
| 18:18.12 | DivideBy0 | the call doesn't get passed into your stasis app until it connects, so you wont see those sorta things either |
| 18:18.21 | pjensen00 | I did'nt think so either, since it's kind of wrapped up in the asterisk hangup codes |
| 18:18.35 | pjensen00 | as far as "BYE" goes |
| 18:18.38 | DivideBy0 | you can get a hangup code |
| 18:18.44 | pjensen00 | yep, I have those |
| 18:18.46 | pjensen00 | :) |
| 18:18.53 | DivideBy0 | oh, ok |
| 18:19.15 | pjensen00 | They map some of the SIP error codes into asterisk hangup codes |
| 18:20.08 | DivideBy0 | which one maps to "slammed down handset"? I miss slamming the phone down after a phone call |
| 18:20.54 | WIMPy | If you slam it down hard enough to destry it, that would be a timeout. |
| 18:21.13 | DivideBy0 | I push the touch-screen really hard. It's just not the same |
| 18:21.46 | DivideBy0 | can I borrow your cell? |
| 18:21.53 | DivideBy0 | I want to test the timeout |
| 18:22.06 | WIMPy | Just hold your finger the hangup icon until the "really hard" option appears :-) |
| 18:22.39 | DivideBy0 | oh, lemme try that! |
| 18:22.39 | WIMPy | On a cellphone it's easy. Just drop the battery. |
| 18:23.52 | WIMPy | That just makes me wonder what happened to that issue. Did someone decide on using the Cisco code or the correct one, yet? |
| 18:23.57 | pjensen00 | I have a deskphone that allows me to slam some serious plastic. |
| 18:24.52 | WIMPy | Looks like that one is still open. |
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| 19:08.20 | *** join/#asterisk BigKristoff (~Kristoff@89-69-230-70.dynamic.chello.pl) |
| 19:08.49 | BigKristoff | Hi :) |
| 19:08.49 | BigKristoff | Does Asterisk supports fax detection through faxdetect option if CNG signal is comming in RFC 2833 RTP Event? |
| 19:08.49 | BigKristoff | I have WorkCentre 3220 traing to send FAX to may Asterisk 11.16.0 (faxdetect = yes). The CNG tone is comming to Asterisk through RFC 2833 RTP Event, and Asterisk seems to ignore it, event when i sent DTMF mode to RFC. |
| 19:08.49 | BigKristoff | When CNG comes in RTP stream everything works fine. |
| 19:11.25 | hexanol | another question about app_queue, about the AgentComplete event that is sent when the caller or the member hangups a call |
| 19:11.41 | hexanol | in the AgentComplete event, there's never both the information about the caller and the agent |
| 19:11.56 | hexanol | if the agent hangs up, only the caller information is in the event |
| 19:12.08 | hexanol | and if the caller hangs up, only the agent information is in the event |
| 19:12.18 | hexanol | someone is aware of this behaviour ? |
| 19:19.49 | TazzNZ | BigKristoff, have you enabled DTMF debugging to confirm you are getting the signal ? |
| 19:20.01 | *** join/#asterisk tompaw (U2FsdGVkX1@tompaw.xxx) |
| 19:20.05 | tompaw | Hello |
| 19:21.14 | TazzNZ | hexanol, I havn't looked into that tbh, but you know that you can declare varibles that will remain with the call until both parties have hung up - like Set(__VAR,bla) |
| 19:21.43 | TazzNZ | no matter where you are, you will then have access to that - within the same call ofcourse :) |
| 19:22.41 | hexanol | I understand that, but in this case that does not help |
| 19:26.11 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 19:26.23 | TazzNZ | why not save both the details, then recall it when you need ? |
| 19:29.59 | hexanol | the problem in this case seems to be that the hangup callback in app_queue is called after the channel's destructor has run, so the channel snapshot of the channel which hanged up is not in the ast_channel_cache anymore and is thus never available in the AgentComplete AMI message |
| 19:31.33 | hexanol | well I have an external application (using the AMI) which we are updating for asterisk 13 (from asterisk 11), and in asterisk 11 that information was there |
| 19:32.08 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 19:32.52 | TazzNZ | log a bug then imho |
| 19:33.33 | TazzNZ | have you checked to see if the info wasn't deprecated in 12 ? |
| 19:34.28 | hexanol | I've looked on the wiki and the CHANGES and UPGRADE files and no, there's no word about it |
| 19:34.44 | vader- | Any of you guys deal with 8x8? |
| 19:34.49 | hexanol | but indeed there would be a different way to retrieve the information I'm looking for but it would ask for extra bookeeping in the application |
| 19:35.06 | hexanol | *the external application that is |
| 19:35.42 | TazzNZ | vader-, that is a vary opened ended question |
| 19:36.06 | vader- | im just trying to find out any pricing type info |
| 19:36.11 | vader- | their website has nothing unless you contact them |
| 19:36.23 | vader- | so i contacted them and didn't get any info there without reaching out to another person, etc |
| 19:36.30 | vader- | so i was looking at reselling cloud pbx services |
| 19:36.40 | TazzNZ | hexanol, perhaps one of the dev's can help you out - it might pay to try and get them to respond here :) |
| 19:37.03 | TazzNZ | vader-, you in the wrong channel for that :) |
| 19:37.21 | TazzNZ | unless one of their employees hang out here |
| 19:37.36 | vader- | there are plenty of Telephony guys in here that deal with things other than Aserisk |
| 19:37.41 | vader- | was just hopign to get lucky and find one |
| 19:37.56 | TazzNZ | totally agree :) |
| 19:37.57 | BigKristoff | TazzNZ th for reply. I did not. I will try that tomorow when i get bac to the office. But i think that asterisk can see them becacuse they still works in DTMM menu |
| 19:39.36 | mjordan | TazzNZ: We watch this channel. |
| 19:39.39 | mjordan | shrugs |
| 19:40.06 | mjordan | Frankly, I can't recall why that information isn't always available. I suspect that with the more strict model of channel lifetime that occurs in 13, it's possible that the ordering changed and we didn't realize it. |
| 19:40.08 | pjensen00 | I wonder if the NSA watches this channel..... |
| 19:40.18 | mjordan | Or, we changed the ordering and we forgot to update UPGRADE, and it can't be fixed. |
| 19:40.24 | mjordan | Without staring at app_queue for awhile, I'm not sure. |
| 19:40.31 | TazzNZ | pjensen00, you know they do ! :) |
| 19:40.34 | mjordan | A bug report is fine, as that will make sure it gets tracked in some form or another. |
| 19:41.06 | TazzNZ | there you go hexanol :) |
| 19:41.19 | pjensen00 | "Tell J and K to go out there and hunt some bugs instead of watching us" |
| 19:43.31 | TazzNZ | yeah mjordan - I know you watch it, but you also have a 100 other things on :) |
| 19:43.44 | hexanol | mjordan: thanks you for your answer as always, I'll open a issue |
| 19:43.52 | hexanol | *thank |
| 19:44.02 | mjordan | hexanol: np |
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| 19:59.36 | hexanol | there it is: https://issues.asterisk.org/jira/browse/ASTERISK-25187 |
| 19:59.47 | edwin_quijada | Hi! |
| 20:00.40 | *** part/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 20:01.40 | edwin_quijada | http://pastebin.com/dmtbz4aK |
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| 22:08.54 | rsdvd | hello All - back with another call for help - our ISDN config has broken again. We had outbound calls working fine - then we installed a config module in completePBX and they stopped working |
| 22:09.13 | rsdvd | I need a hand checking what it broke because it all looks fine to me |
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| 22:21.17 | *** join/#asterisk busymind (~textual@li742-188.members.linode.com) |
| 22:21.48 | busymind | Could someone please help me understand priority orders in Asterisk? |
| 22:22.48 | [TK]D-Fender | ~ask |
| 22:22.48 | 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. |
| 22:25.42 | busymind | so inside of context priorities, you can either use 1,2,3,4 etc, or 1,n,n,n etc. I get that, but how can you insert a new step inside of a context without it overwriting the "n". |
| 22:26.30 | busymind | for example, if I have 1,n,n,n in my context, and I want to specify an include like FreePBX does, and then inside of that include I put, say, 2, right now it overwrites the second N |
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| 22:32.42 | [TK]D-Fender | n = next |
| 22:32.57 | busymind | i know that |
| 22:32.59 | [TK]D-Fender | It follows whichever was immediately BEFORE it |
| 22:33.10 | [TK]D-Fender | So if there is hard numbering ... then there is hard numbering |
| 22:33.26 | busymind | but Asterisk replaces the n with the number in RAM, right? |
| 22:33.34 | [TK]D-Fender | No. |
| 22:33.40 | [TK]D-Fender | It loads from the file in order |
| 22:33.51 | [TK]D-Fender | not just "from ram" |
| 22:33.53 | [TK]D-Fender | from LOAD |
| 22:34.06 | [TK]D-Fender | Because once loaded into ram everything has already been converted to "hard numbered' |
| 22:34.06 | busymind | okay, well, how can I insert a string inside of a context from a different file? |
| 22:34.13 | busymind | i see |
| 22:34.29 | [TK]D-Fender | You don't "insert" |
| 22:34.36 | [TK]D-Fender | There is no such thing as "insert" |
| 22:35.00 | [TK]D-Fender | All you have is what's merged in via include's, and duplicate context entries |
| 22:35.27 | busymind | ah, okay, that makes sense |
| 22:35.33 | [TK]D-Fender | Which on Freepbx means using that overrides config, not just the "custom" |
| 22:35.36 | [TK]D-Fender | based on load order |
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| 22:35.57 | [TK]D-Fender | because repeats tend to get ignored |
| 22:36.43 | busymind | does the override config in freepbx merge with the custom contexts? or does it get prepended to them? |
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| 22:39.11 | [TK]D-Fender | it get's FORCED where you put it and the generated stuff becomes the ignored duplicates |
| 22:39.51 | busymind | ah, so there really is no way to gently insert a few lines? I have to write the entire context as I want it in the extensions_custom? |
| 22:40.01 | [TK]D-Fender | <[TK]D-Fender> There is no such thing as "insert" |
| 22:40.26 | [TK]D-Fender | You have to override explicit priorities |
| 22:42.43 | [TK]D-Fender | it loads the override then the original |
| 22:42.47 | [TK]D-Fender | duplicates get dropped |
| 22:43.11 | [TK]D-Fender | So if you replace prioirty 1, then the rest should resume |
| 22:44.38 | busymind | okay, but how can you replace priority 1 and 2, but still keep the originals? |
| 22:44.59 | busymind | maybe re-prioritize them to 3,4? |
| 22:45.03 | [TK]D-Fender | the originals will parse their sequence and one by one get dropped |
| 22:45.20 | [TK]D-Fender | If you make #2 then ONLY #2 gets replaced |
| 22:45.47 | [TK]D-Fender | There is no reprioritizing. |
| 22:46.28 | [TK]D-Fender | heads out for the evening.... |
| 22:46.59 | busymind | okay |
| 22:47.03 | busymind | thank you :) |
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