| 16:32.49 | *** join/#asterisk infobot (ibot@69-58-80-29.ut.vivintwireless.net) |
| 16:32.49 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.6.0 (2015/10/09), 11.20.0 (2015/10/09); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 16:42.59 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 16:43.45 | Phil-Work | I'm having trouble getting BackgroundDetect() to detect talking and hand off to the talk extension |
| 16:43.54 | Phil-Work | does anything special need enabling (modules, configs, etc.)? |
| 16:45.56 | Phil-Work | it doesn't appear to respond to DTMF either |
| 16:46.08 | igcewieling1 | Phil-Work: have you looked at the AMD app?: |
| 16:46.28 | *** join/#asterisk saratogga (~saratogga@107.85.105.213) |
| 16:46.48 | igcewieling1 | Phil-Work: there is a good chance you will have to answer the call before Asterisk can detect |
| 16:46.55 | *** join/#asterisk tompaw (U2FsdGVkX1@tompaw.xxx) |
| 16:46.59 | Phil-Work | igcewieling1, yeh but I'm really just trying to detect the presence of audio here |
| 16:47.28 | Phil-Work | igcewieling1, it's an inbound call into Asterisk - it does an Answer() then a BackgroundDetect() |
| 16:47.35 | igcewieling1 | Phil-Work: not trying to detect the presence of a fax tone? |
| 16:47.53 | Phil-Work | the audio file given to BackgroundDetect() plays but it doesn't detect talking or DTMF |
| 16:47.57 | Phil-Work | not trying to detect fax, no |
| 16:48.05 | Phil-Work | trying to set up some automated telephony testing |
| 16:48.12 | Phil-Work | just need to make sure there's audio heard |
| 16:48.21 | tompaw | Guys, anyone dealing with this sort of segfaults? |
| 16:48.22 | tompaw | http://hastebin.com/kafagetugo.vala |
| 16:48.37 | igcewieling1 | Phil-Work: good luck. I've not seen many people trying that, usually they are trying to detect a fax tone. |
| 16:48.39 | tompaw | It looks like listing channels via ari-py overflows the memory |
| 16:48.51 | tompaw | #44 0x00000000005218ec in handle_uri (headers=0x7f87dc379250, method=AST_HTTP_GET, uri=0x7f87be2c6be9 "api-docs/channels.json", ser=0x7f8870004e08) at http.c:1480 |
| 16:50.25 | igcewieling1 | tompaw: you might check #asterisk-dev |
| 16:50.59 | tompaw | ok |
| 16:51.27 | Phil-Work | playing a longer message, I can get it to respond to DTMF now |
| 16:51.29 | Phil-Work | just not talking |
| 16:51.58 | *** join/#asterisk theron (~theron@2620:10d:c091:200::9:3ce2) |
| 16:52.49 | Phil-Work | oh, that got it |
| 16:52.50 | Phil-Work | same => n,BackgroundDetect(bombsquad, 1, 1, 5000) |
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| 18:47.19 | mcargile | I am looking for a very simple WebRTC phone that I can integrate into my website and connect into my Asterisk server. Any recommendations? |
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| 19:05.08 | solmsted | Hello, I'm using Asterisk with a custom FastAGI application. Is there any way to cancel, interrupt, or stop an AGI command that is in progress? |
| 19:05.40 | WIMPy | hangup, redirect |
| 19:05.52 | WIMPy | Or kill that AGI |
| 19:06.29 | solmsted | I tried dropping the TCP socket connection, but Asterisk continues executing the command |
| 19:08.15 | solmsted | If I send another AGI command while the previous one is still running it seems to just queue them up |
| 19:08.37 | WIMPy | Yes, That's the way it works. |
| 19:08.56 | solmsted | is there no way to stop one early? |
| 19:09.44 | WIMPy | Not from the AGI. |
| 19:10.17 | solmsted | is there another interface, outside of AGI, that can stop a running AGI command? |
| 19:10.32 | WIMPy | *CLI and AMI. |
| 19:10.59 | WIMPy | Or you could change over and use ARI. That might have more possibiliteis, but I'm not in to that. |
| 19:11.08 | igcewieling1 | you can use non-fast agi |
| 19:11.18 | *** part/#asterisk mcargile (~mikec@office1.vicidial.com) |
| 19:11.32 | WIMPy | What difference does that make? |
| 19:11.34 | igcewieling1 | that, along with catching SIGHUP might do what you need to do. |
| 19:11.37 | solmsted | Yes, ARI would solve the issue, but last I checked ARI doesn't support speech recognition which I need |
| 19:12.13 | WIMPy | How is that related to AGI? |
| 19:12.22 | solmsted | what AMI action could do it? I'm not very familiar with AMI. I was reading through the docs looking for something to do this. |
| 19:12.24 | igcewieling1 | WIMPy: might make asterisk see the disconnect. I *never* have problems with Asterisk continuing to send data to my AGIs |
| 19:12.36 | igcewieling1 | WIMPy: oddly, AGI and FastAGI are similar. He said he is using FastAGI. |
| 19:12.42 | WIMPy | redirect |
| 19:12.56 | igcewieling1 | WIMPy: he's not using Manager. |
| 19:13.11 | WIMPy | AGI and FastAGi should do the same, just on another connection type. |
| 19:13.18 | igcewieling1 | solmsted: often using both AGI and Manager are done. |
| 19:13.43 | igcewieling1 | WIMPy: I doubt it sends HUP accross the tcp socket. |
| 19:14.16 | WIMPy | But it would reset the connection, which might do the same. |
| 19:14.47 | igcewieling1 | "Asterisk will send a SIGHUP signal when the user hangups the channel." the AGI needs to catch that to see the hangup. |
| 19:15.19 | solmsted | so an AMI redirect action would immediately send the channel to another extension in the dial plan? If that works, I could probably build something with that and have it just reconnect to FastAGI |
| 19:15.42 | WIMPy | yes |
| 19:15.51 | igcewieling1 | solmsted: it should. Any NEED for fastagi .vs. regular AGI? |
| 19:16.35 | solmsted | we don't want to be running 1 process per call, we can have a small pool of processes handle all the tcp connections |
| 19:17.13 | solmsted | or we have the option of moving that service to another machine if we want to |
| 19:17.14 | igcewieling1 | solmsted: unless you have MANY calls I strongly doubt it will matter. |
| 19:18.54 | solmsted | averaging 800-1000 calls per hour, so the process count is somewhat significant |
| 19:20.01 | WIMPy | Doesn't sound too interesting. |
| 19:21.31 | solmsted | but thanks! I'll look at redirect. |
| 19:21.50 | igcewieling1 | solmsted: 8000-ish calls per day. could be considered "many" 8-) |
| 19:23.09 | solmsted | igcewieling1, yes we hit that regularly |
| 19:24.01 | WIMPy | A fork every 3.6 seconds may not be nice, but I see no cause for concern. |
| 19:24.02 | igcewieling1 | one of my main servers: 890432 calls processed in 10 weeks, 4 days |
| 19:24.17 | igcewieling1 | I run at a min 2 agi per call |
| 19:27.08 | solmsted | idk all the details, we were hitting performance limits with AGI at about 100 simultaneous calls. We switched to FastAGI and it's been great. |
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| 20:32.03 | *** join/#asterisk jpastore (~jpastore@50-200-51-210-static.hfc.comcastbusiness.net) |
| 20:32.54 | jpastore | hi I was wondering how I can modify the sip headers for the invite request. I have a call failing. The capture shows a difference in the Allow header, but I don't see where I can modify that |
| 20:34.37 | *** join/#asterisk jrrose (jon@nat/digium/x-mdoroeziebeqbmzj) |
| 20:35.06 | igcewieling1 | jpastore: For the most part you don't mess with SIP headers in Asterisk. Is there a specific header you want to modify? |
| 20:36.40 | *** part/#asterisk DragonAzul (~DragonAzu@187.208.73.113) |
| 20:37.39 | jpastore | igcewieling1, yes, "allow" under the message header. as well as "supported" the call that fails shows: replaces, timer...the successful call shows timer, 100rel, precondition. |
| 20:37.55 | igcewieling1 | jpastore: not going to happen in Asterisk. |
| 20:38.33 | igcewieling1 | For direct modification of SIP you'll need a SIP proxy like Kamailio |
| 20:38.57 | igcewieling1 | I suggest fixing the cause rather than the symptoms |
| 20:39.13 | jpastore | what would the cause be? |
| 20:39.34 | igcewieling1 | no idea, but messing with allow and supported is not the solution |
| 20:40.07 | jpastore | Is it possible to enable a configuration option to allow 100rel and precondition? |
| 20:40.10 | igcewieling1 | if you pastebin the invite for a working and non-working call someone (not me) might be interested in looking at it. |
| 20:41.51 | jpastore | igcewieling1, why would I want to setup Kamailio? what benefit does it serve? |
| 20:43.06 | igcewieling1 | jpastore: it allows you to mess with the sip packets. |
| 20:43.21 | igcewieling1 | For direct modification of SIP you'll need a SIP proxy like Kamailio |
| 20:43.41 | igcewieling1 | if you don't need to mess with the SIP packets there is little reason to set up Kamailio |
| 20:43.43 | solmsted | igcewieling1, WIMPy: thanks I just did a proof-of-concept test with AMI redirect and it's working! |
| 20:44.24 | jpastore | igcewieling1, well SIP packet modification seems to be the goal. I'll download this and see if I can find a primer on it thanks for the lead |
| 20:44.53 | igcewieling1 | jpastore: modifying the SIP packets is a terribly dumb idea. If you want to do it though..... |
| 20:45.21 | igcewieling1 | If this was a general problem there should be zillions of hits in a web search. |
| 20:46.37 | jpastore | thanks I apprecaite the input |
| 20:46.51 | igcewieling1 | you're going to waste a lot of time on this. |
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| 20:51.31 | jpastore | I appreciate your position. I imagine with my experience where it's at will result in that regardless |
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| 22:59.13 | xpheres | hello, it is possible to use two trunks at the same time, one for inboun and the other for outbound? |
| 22:59.27 | infinity1 | rrittgarn: i have access to a block of about DIDs (company going out of business). i'm hoping to get some of them (maybe 10, or more depending on cost.). I don't plan on using the DIDs right now and i need a place to hold them. any thoughts? You mentioned e4, NexVortex, and Bandwidth before, but all of those seem like i need a large account to get a descent price. |
| 22:59.42 | infinity1 | s/about DIDs/about 150 DIDs/ |
| 23:00.10 | infinity1 | xpheres: yea |
| 23:00.17 | xpheres | ok |
| 23:00.47 | infinity1 | xpheres: you can have different trunks for different area codes, dial patterns, whatever. |
| 23:00.57 | xpheres | right |
| 23:01.49 | rrittgarn | PM'd you infinity1 |
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| 23:32.22 | gruetzkopf | DID is so much cleaner with variable-length numbering plans |
| 23:33.24 | gruetzkopf | we are supposed to use -00 to -99 at our location, but up to 11 digits work.. |
| 23:34.03 | WIMPy | That would off course depend on where you're calling from. |
| 23:35.19 | gruetzkopf | yeah |
| 23:35.33 | gruetzkopf | it doesnt even work from all germany mobile carriers |
| 23:36.02 | gruetzkopf | but we haven't found a POTS or ISDN provider where it doesn't |
| 23:36.42 | gruetzkopf | everything that does step-by-step dialing works. |
| 23:36.58 | WIMPy | overlap |
| 23:38.36 | gruetzkopf | we do weird stuff with that much numbering space |
| 23:38.59 | WIMPy | Has that become legal, BTW? |
| 23:39.06 | gruetzkopf | ? |
| 23:39.21 | WIMPy | I just saw that sipgate offers to add a single digit to any registered number. |
| 23:39.37 | gruetzkopf | why shouldn't they |
| 23:40.22 | WIMPy | Because it used to be illegal to extend numbers. |
| 23:40.46 | WIMPy | Not that it has always been done. I only found out by pure chance. |
| 23:41.35 | gruetzkopf | well, the BNetzA did complain about us wanting 2 digits for a single ISDN BRI |
| 23:45.07 | *** join/#asterisk cyford (~support@c-73-137-1-6.hsd1.ga.comcast.net) |
| 23:45.20 | gruetzkopf | but yeah, we run 5 PBXes, we need some phone numbers |
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| 23:49.12 | pjensen00 | Hey, I'm going nuts ovah hyah. I'm trying to handle a SIP 302. Does anything look wrong in this debug output of processing of the 302? |
| 23:49.13 | pjensen00 | http://pastebin.com/29VWWgZM |