| 00:38.28 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:38.28 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.8.1 (2014/03/10), 1.8.26.1 (2014/03/10); Standard: Asterisk 12.1.1 (2014/03/10); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 00:38.47 | DevWork_ | I see it is a sample. I don't understand the sample. I get that if I dial 7015, it does some stuff based on the DND script. I get that the BLF key is linked to DND_7015 and toggles the state variable. I don't know why I have that first line, or where it goes. |
| 00:38.56 | DevWork_ | exten => DND_7015,hint,Custom:DND_7015 what is this doing. |
| 00:39.10 | [TK]D-Fender | that IS the difinition of the BLF |
| 00:39.19 | [TK]D-Fender | hint = BLF |
| 00:39.49 | DevWork_ | Ok. |
| 00:42.10 | DevWork_ | Do I need anything in the sip.conf to allow BLF's or is it all device configuration |
| 00:42.57 | digitalirony | well, im making progress |
| 00:43.16 | digitalirony | SV now accepts the call, but I can't tell that it actually does anything lol |
| 00:44.48 | DevWork_ | hmm it looks like it works actually without any further config |
| 00:45.02 | DevWork_ | just add the entry in the phones config. |
| 00:48.15 | [TK]D-Fender | dip.conf entry specifies what context to lok for hints when the phone tries to subscribe. From there it's up to the phone to ask for it |
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| 05:51.29 | zpotoloom | hi, anyone using Sangoma W400 GSM card with asterisk ( libwat ) ? |
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| 07:03.21 | michoelc | Hi all, wondering if someone can help me with getting call pickup working for queues? Basically want to be able to dial an extension to pickup the next call from a queue.. |
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| 07:06.24 | michoelc | This is what I have got so far but not working yet: http://pastebin.com/6biNdz8T |
| 07:07.06 | michoelc | I get app_directed_pickup.c:302 pickup_exec: No target channel found for s@incoming |
| 07:11.54 | kaldemar | michoelc: what does core show channel <channel> tell you about the channel's extension and context? |
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| 07:15.32 | michoelc | kaldemar: it says context: "incoming", extension: "s" |
| 07:17.22 | ChannelZ | Interesting. I don't know that you can't, but I'm not sure if you can snatch a call out of a queue like that. Pickup generally only works on a channel currently ringing AFAIK |
| 07:19.09 | michoelc | ChannelZ: from what I've found here someone's gotten it to work https://reviewboard.asterisk.org/r/1619/ |
| 07:20.16 | michoelc | ChannelZ: I'm assuming that this was eventually merged in as I the hint part of it is working fine on 11.7 |
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| 07:25.23 | frederik_jensen | Hello! I am new to Asterisk |
| 07:25.23 | frederik_jensen | I am supposed to use a headset and just make calls by running AMI actions - this will be the easiest for me i think - and i dont want to touch my Cisco 502g at all |
| 07:25.23 | frederik_jensen | Is it possible to make a call by sending a series of AMI actions to a Cisco 502g phone - without pressing the headset button in order ot initiate the call? |
| 07:27.15 | ChannelZ | michoelc: without screwing around with it myself which I don't have time to do, I have to wonder if this is somewhat of a hack and that using Goto into the separate context that just does the Queue is intentional. The example is either bad or is tricky on purpose. |
| 07:27.44 | michoelc | frederik_jensen: You can use the Originate action from ami to initiate the call |
| 07:28.10 | ChannelZ | frederik_jensen: not sure on AMI specifically but you can use SipAddHeader and make the 502g auto-answer |
| 07:28.33 | ChannelZ | SIPAddHeader(Call-Info: \;answer-after=0) |
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| 07:28.48 | ChannelZ | at least I think that still works on the 5xx's |
| 07:29.09 | frederik_jensen | ok! |
| 07:29.29 | frederik_jensen | thank you for replying |
| 07:29.30 | kaldemar | michoelc: try changing the answer in your queue extension to Ringing. |
| 07:29.54 | ChannelZ | good luck everyone, off to bed. |
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| 07:36.02 | michoelc | ChannelZ: thanks, but tried that and no luck :-( |
| 07:36.17 | michoelc | sorry, I meant kaldemar |
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| 07:54.15 | michoelc | Anyhow need to go now, will try again tomorrow |
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| 10:59.10 | davlefou | hi, where i can find the iax sql file for postgresql? |
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| 11:01.38 | kaldemar | davlefou: contrib/realtime/postgresql/postgresql_config.sql |
| 11:05.10 | davlefou | kaldemar, i get that file but there no iax_conf sql file |
| 11:05.49 | davlefou | in my asterisk 11.8.1 |
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| 11:24.09 | kaldemar | davlefou: http://svn.asterisk.org/svn/asterisk/tags/12.1.1/contrib/realtime/postgresql/postgresql_config.sql |
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| 12:27.23 | davlefou | kaldemar, it is an very intereting table, |
| 12:30.15 | davlefou | kaldemar, can we use materialized view for sipregs? |
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| 12:30.34 | Zogot | Ahoyhoy |
| 12:31.37 | Zogot | Quick question: sip context names, email is a valid one eh? [test@example.com]context=default type=friend secret=exampleSecret |
| 12:31.48 | Zogot | for example |
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| 12:44.20 | [TK]D-Fender | That'll probably pose parsing issues somewhere. |
| 12:45.06 | [TK]D-Fender | I'd recommend just giving it a regular name... and remember that that doesn't have to imply the username it auths |
| 12:46.40 | Zogot | [TK]D-Fender: oh? how could you define a different username? |
| 12:46.58 | Zogot | defaultuser=? |
| 12:47.09 | [TK]D-Fender | Zogot: you should really read the sample config... |
| 12:47.19 | Zogot | [TK]D-Fender: thanks, il take a look |
| 12:47.30 | [TK]D-Fender | Zogot: that's the normal one. username / defaultuser |
| 12:49.21 | Zogot | [TK]D-Fender: and the values in under the [] can just be directly related to the database columns if i were to use func_odbc |
| 12:50.21 | [TK]D-Fender | I don't see where you are drawing an automatic association of FUNC_ODBC ... and sip.conf |
| 12:51.53 | Zogot | [TK]D-Fender: Well they go into the sip table in mysql for example, i would preferably be using that over the conf files |
| 12:52.48 | Zogot | ah it is :) |
| 12:52.50 | [TK]D-Fender | Lots of things become more painful like getting info at CLI. I'd reserve that method for when there's a real payoff for doing it |
| 12:52.52 | Zogot | http://svn.asterisk.org/svn/asterisk/tags/11.8.1/contrib/realtime/mysql/sippeers.sql |
| 12:53.35 | Zogot | [TK]D-Fender: aye, this is for a multi tennant system im working on. Im quite fresh to asterisk so you must excuse any of my stupid questions :p |
| 12:54.07 | Zogot | theres someone else here who has significantly more knowledge than I on asterisk |
| 12:54.28 | Zogot | but i just wanted to verify something |
| 12:55.55 | [TK]D-Fender | Zogot: Most platforms I see still run on the concept of using DB's for conceptual PBX constructs and just generate * configs to match |
| 12:56.25 | [TK]D-Fender | Zogot: Reduces latency, load, and dependency on other services to keep it running. |
| 12:56.40 | Zogot | [TK]D-Fender: Requires doing a sip reload though for every change right? |
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| 12:59.30 | Zogot | [TK]D-Fender: I will look into the latency/load though. That could be something. I'll have to check with my colleague |
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| 13:00.38 | [TK]D-Fender | Zogot: Or at a time when you're ready to "batch" them. |
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| 13:03.34 | Zogot | [TK]D-Fender: Thanks for your responses |
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| 13:08.16 | [TK]D-Fender | You're welcome |
| 13:09.12 | Zogot | [TK]D-Fender: You work at Digium? |
| 13:10.10 | [TK]D-Fender | Zogot: Nope, just been around the field a long while and have seen many * platofrms, multi-tennent and all |
| 13:11.18 | Zogot | [TK]D-Fender: ah fair enough. Not that Im challenging what you are saying or so, just making conversation |
| 13:13.21 | Zogot | [TK]D-Fender: Well, I'll be around more often, I may sometimes have additional queries. Plus IRC is a great resource to learn from, see what other people run into and then seeing how/if they solved it or so |
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| 13:14.31 | file | I have a blog post about why I personally dislike realtime :D |
| 13:14.39 | [TK]D-Fender | Zogot: DB losing comms takes out your PBX, is one more thing to secure, etc |
| 13:14.50 | Zogot | file: oh? would love a read |
| 13:14.58 | file | http://www.joshua-colp.com/realtime-i-love-to-hate-you/ |
| 13:15.28 | Zogot | [TK]D-Fender: supervisord should help with that eh? |
| 13:17.44 | [TK]D-Fender | Zogot: Not something I have experience with |
| 13:18.44 | Zogot | file: How do you generate the configs? Some seperate script that runs every so often? |
| 13:18.55 | file | sure |
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| 13:20.12 | Zogot | file: Thanks for the article. |
| 13:20.31 | file | everything has pros and cons but if you understand the pitfalls then you can make an informed decision |
| 13:20.58 | Zogot | file: aye indeed. Nah its some good points from you and [TK]D-Fender. I will bring them up |
| 13:22.28 | ghost75 | is pjsip having a features for matching multiple hosts instead a single peer? |
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| 13:27.06 | dar123 | i want to connect asterisk with mysql, for user database and CDR's. |
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| 13:30.33 | [TK]D-Fender | darGo for it |
| 13:30.37 | [TK]D-Fender | dar123: Go for it |
| 13:32.20 | ghost75 | *deep voice* do it |
| 13:34.19 | dar123 | unable to find any good tutorial or doc so far |
| 13:34.34 | [TK]D-Fender | ~wiki |
| 13:34.37 | [TK]D-Fender | ~book |
| 13:34.37 | 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 |
| 13:34.40 | Zogot | [TK]D-Fender, file: http://pastebin.com/NWkFrM1G what happened to this guy to explode like that :p |
| 13:34.49 | Zogot | pretty funny stuff |
| 13:34.50 | [TK]D-Fender | ~asteriskwiki |
| 13:34.51 | infobot | hmm... asteriskwiki is http://wiki.asterisk.org |
| 13:34.53 | [TK]D-Fender | ^^^^^^ |
| 13:35.24 | Zogot | file: did you go on skype with him? please tell me you recorded it :p |
| 13:35.28 | [TK]D-Fender | Zogot: * comes with sample configs for this, SQL setup scripts and is well documented in the book & wiki |
| 13:35.36 | [TK]D-Fender | dar1 : * comes with sample configs for this, SQL setup scripts and is well documented in the book & wiki |
| 13:35.44 | [TK]D-Fender | has targeting issues this morning... |
| 13:36.45 | dar123 | thanks |
| 13:37.23 | [TK]D-Fender | Zogot: Yeah... he was a "special" one... |
| 13:37.49 | ghost75 | i hope now banned |
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| 13:40.12 | [TK]D-Fender | ghost75: You could see what he flung at file there, and he did as much and worse to other ops as well, made threats to pirate and distribute Digium products, etc. |
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| 13:41.39 | jameswf | Digium grows produce now? |
| 13:42.34 | file | hrm? |
| 13:42.36 | file | oh that |
| 13:42.38 | file | no, I did not go on Skype or anything with him |
| 13:43.23 | ghost75 | those are just trolls |
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| 13:44.37 | Zogot | did you know, that the word troll(when refering to an internet troll or so) is in reference to the fishing technique |
| 13:44.41 | Zogot | and not the thing under the bridge |
| 13:44.52 | ghost75 | nope |
| 13:45.23 | Zogot | aye :p, trolling |
| 13:45.37 | ghost75 | see me trollin they hatin |
| 13:45.42 | Zogot | to leave many lines out and hope to catch fish, not to be confused with trawling |
| 13:45.42 | Zogot | which is a giant net |
| 13:46.22 | Zogot | http://en.wikipedia.org/wiki/Trolling_(fishing) |
| 13:46.22 | Zogot | so there go you, TYL |
| 13:46.32 | whizzi | only in England >:) |
| 13:46.37 | ghost75 | trolling the bait |
| 13:47.26 | ghost75 | i want this http://upload.wikimedia.org/wikipedia/commons/6/66/Troll_Warning.jpg |
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| 14:00.28 | [TK]D-Fender | Actually he wasn't technically a troll, just a rude, ignorant, delusional asshole with a tragically disproportionate sense of self-entitlement. |
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| 14:27.09 | tomodachi | hi is there a way to see what part of the dialplan im currently in , line by line? im trying to debugg som trange message i get when pressing # in a conference |
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| 14:28.05 | WIMPy | core set verbose 3 |
| 14:28.33 | tomodachi | WIMPy: thnxk ill give it a shot :) |
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| 14:34.16 | tomodachi | exten => _0046X.#,1,Macro(dial_iax_trunk,0${EXTEN:4}) |
| 14:34.33 | tomodachi | what does # do here? i cant find it in the pattern matching documentation |
| 14:34.37 | tomodachi | or does it just mean # |
| 14:34.49 | WIMPy | That is not a valid extension pattern. |
| 14:35.15 | tomodachi | yeah thougt it could be what was causying my problems :) a typo perhaps |
| 14:35.27 | WIMPy | You can't have anything after . or ! . |
| 14:36.28 | [TK]D-Fender | Well... you can... but it won't do anything... |
| 14:36.51 | [TK]D-Fender | tomodachi: and "#" means .... "#" |
| 14:36.58 | tomodachi | [TK]D-Fender: i recongnice your nick , you have been here for years! , when i first started looking into asterisk >:) |
| 14:37.01 | WIMPy | Yeah, it's probably just ignored. |
| 14:38.19 | [TK]D-Fender | indeed |
| 14:38.21 | tomodachi | i migrated this dialplan from a an old 1.2 asterisk god knows what elese there is in there |
| 14:41.20 | tomodachi | hmm my problems seems fairly global, whenver i call into an application and press # a voice says transfer |
| 14:41.42 | WIMPy | features.conf |
| 14:42.07 | [TK]D-Fender | DTMF transfers is for chumps :) |
| 14:42.07 | WIMPy | And you have t and/or T as option to Dial(). |
| 14:43.32 | tomodachi | that was it, |
| 14:44.27 | tomodachi | so that flag just ads a transfer message to wherever i call? |
| 14:44.28 | *** join/#asterisk lorsungcu_ (~anonymous@67.138.198.66) |
| 14:44.37 | tomodachi | why would anyone want that! |
| 14:45.08 | WIMPy | Because they use a phone that can't do a transfer via Asterisk. |
| 14:45.32 | *** join/#asterisk yoavz (yoavz@yoavz.net) |
| 14:45.51 | tomodachi | must have been something legacy, since it seems to work at least :=) thanx a bunch guys! |
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| 15:11.50 | whizzi | can somebody explain where the defaultuser is needed for on a trunk ? |
| 15:12.36 | [TK]D-Fender | whizzi: If your inbound caller's username is not [whatyouputinhere] |
| 15:13.55 | whizzi | ok, so it’s only needed for incoming |
| 15:14.51 | [TK]D-Fender | May apply to outbound as well |
| 15:15.11 | [TK]D-Fender | username / fromuser do (in different ways) |
| 15:16.02 | whizzi | aye, that I know. |
| 15:20.29 | whizzi | tnx :) |
| 15:27.24 | Penguin | I doubt it is used for incoming at all. |
| 15:31.36 | ghost75 | somebody know if its possible to use multiple hosts on sip peer section |
| 15:31.56 | [TK]D-Fender | it is not |
| 15:32.17 | ghost75 | wants |
| 15:33.44 | file | PJSIP support in 12 has that... I purposely did it as many people have said so in the past |
| 15:34.09 | ghost75 | thats great news |
| 15:34.42 | ghost75 | how they are added? just multiple host lines? |
| 15:35.15 | file | right now they primarily exist for matching incoming calls, and you can specify multiple hosts or ranges and map them back to an endpoint |
| 15:35.34 | file | ie: everything from 172.16.1.0/24 should be treated as coming from "dave" |
| 15:35.45 | ghost75 | matching incoming calls what exactly what i was looking |
| 15:36.38 | ghost75 | because many isp are using load balancer |
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| 15:49.48 | ghost75 | file: how is it to be entered in sip.conf? |
| 15:49.58 | file | it's not, that is chan_sip |
| 15:50.13 | ghost75 | wooza :> |
| 15:50.32 | file | PJSIP is a completely new channel driver and architecture |
| 15:52.16 | ghost75 | but pjsip.conf is similar to sip.conf ? |
| 15:55.23 | file | no, it's vastly different |
| 15:55.30 | file | conceptually at least |
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| 16:18.52 | file | wobbles |
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| 16:46.28 | willwh | <PROTECTED> |
| 16:46.50 | willwh | so who is trying to get working secure websockets going with asterisk 12? ;) |
| 16:46.56 | willwh | if you are, I want to talk to you |
| 16:47.24 | willwh | I've read some people have had success with 12.2.0-rc1 (moy, for example, whose /team/moy/webrtc-12 branch is in rc-1 |
| 16:47.32 | willwh | I have not had any luck though :( |
| 16:47.35 | Katty | clears throat |
| 16:51.46 | willwh | squints at Katty |
| 16:51.47 | willwh | hmm |
| 16:51.47 | willwh | ? |
| 16:53.17 | Katty | oh, nothing. |
| 16:53.21 | Katty | sinus drainage. |
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| 16:56.47 | willwh | ah, 'tis the season |
| 16:56.53 | willwh | sneezes for good measure |
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| 17:46.48 | ghost75 | file: is external_media_address and external_signal_address working also with ddns ? |
| 17:46.53 | ghost75 | like externhost |
| 17:47.12 | file | yes |
| 17:47.37 | ghost75 | so there is whatever refresh inteval |
| 17:48.55 | ghost75 | and here in the example: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip |
| 17:49.04 | file | I think we may actually treat it strictly as a string right now |
| 17:49.06 | ghost75 | why there are 2 entries for 6001 device |
| 17:49.14 | *** join/#asterisk TechSmurf (~jdaniel@unaffiliated/techsmurf) |
| 17:49.43 | file | because they are of different types and represent two conceptually different SIP things |
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| 17:51.41 | ghost75 | so endpoint is like a phone and aor is like user |
| 17:52.14 | file | an endpoint is a physical device, an aor is an address of record in SIP land and is a mapping of an identity to a way of contacting the device or devices |
| 17:52.24 | file | well - I say physical device... could be a softphone |
| 17:52.26 | file | or a trunk |
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| 20:11.48 | *** join/#asterisk FlipZTechGuy (~FlipZTech@69.197.92.114) |
| 20:13.12 | FlipZTechGuy | Quick question. What flavor of asterisk is most recommended? |
| 20:14.58 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions will tell you how long the various versions will be supported for |
| 20:15.23 | file | Asterisk 1.8 goes security fix only this year, 11 in 2016 |
| 20:16.03 | file | so 11 is a pretty safe bet, and you can then migrate to 13 if you desire |
| 20:19.23 | FlipZTechGuy | cool |
| 20:19.25 | FlipZTechGuy | thanks! |
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| 20:29.40 | *** join/#asterisk SuperNull (~DaBombdig@24-148-101-238.ip.mhcable.com) |
| 20:29.55 | SuperNull | What takes priority in an asterisk realtime configuration, sip.conf or realtime database ? |
| 20:30.56 | file | sip.conf |
| 20:31.06 | SuperNull | alright. ty. |
| 20:31.11 | SuperNull | fclose(file); |
| 20:31.22 | file | Permission Denied. |
| 20:31.35 | SuperNull | you must be female, that was quick. |
| 20:31.42 | navaismo | sudo |
| 20:31.43 | SuperNull | DAYUM DAYUM DAYYUMMM |
| 20:31.50 | SuperNull | sudo fclose(file); |
| 20:31.52 | SuperNull | HAR |
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| 20:39.13 | SuperNull | is there no way to just reload the peers without dropping all sip sessions ? (real time involved) |
| 20:45.15 | SuperNull | looks like leif answered this on the forums. 'no' seems to be the answer. |
| 20:46.22 | file | define sip sessions... |
| 20:46.34 | file | doing a sip reload won't kill calls |
| 20:46.50 | SuperNull | kills registration for all realtime peers. (possibly all peers) |
| 20:47.05 | file | any cached realtime peers are expunged |
| 20:47.29 | SuperNull | i forget why i have caching on but i know i needed it. |
| 20:47.34 | SuperNull | perhaps it was MWI |
| 20:48.25 | SuperNull | ultimately asterisk is the wrong solution for our 'problem'. |
| 20:49.10 | navaismo | lmao |
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| 21:23.16 | SuperNull | if you guys use 'info' dtmfmode do you expect to hear audio tones on the call also if its an ATA ? |
| 21:23.25 | *** part/#asterisk fullstop (~fullstop@64-121-16-14.c3-0.tlg-ubr1.atw-tlg.pa.cable.rcn.com) |
| 21:26.07 | file | generally no |
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| 21:31.13 | SuperNull | well oddly.. we did a firmware upgrade on our GPON gear which has integrated ATA, and there is quote 'a bug' keeping them working.. which allows inband+info at the same time. problem is .. they fixed 'the bug' in a different model and now in-call DTMF doesnt work at all. |
| 21:31.29 | SuperNull | they expect info, asterisk is explicitly set to info on the peer.. no workie. |
| 21:31.37 | dan_j | Hi. Has anyone got any experience with auto-diallers and asterisk? Are there pre-made modules which can be added to asterisk to make the process simpler? Its for a call centre which calls new clients, but they are finding that some data is bogus. |
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| 22:00.24 | JeffC_NN | Does anyone have experience with dialplan application DAHDISendCallreroutingFacility()? |
| 22:00.43 | JeffC_NN | No matter what I try I get "PRI Span: 1 Could not schedule facility message for CallRerouting/CallDeflection message." |
| 22:01.53 | JeffC_NN | Using Framing/Coding ESF/B8ZS |
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| 22:08.48 | JeffC_NN | The problem I'm trying to fix is that my PRI provider doesn't have a way to forward DID's to different numbers when I run out of channels, so we're going to check if we're almost out of them, and use the last channel to redirect away to other nubmers. DAHDISendCallreroutingFacility seemed like the best/only solution, but either I'm not using it right, or my provider doesn't support it....? |
| 22:09.26 | JeffC_NN | My dialplan looks like exten = 7598,1,DAHDISendCallreroutingFacility(19710000000,19711111111,1) |
| 22:09.38 | JeffC_NN | (I changed the phone numbers, but the # of digits is the same) |
| 22:10.13 | JeffC_NN | I've tried with only the deflecting number, aka exten = 7598,1,DAHDISendCallreroutingFacility(19710000000) |
| 22:10.14 | JeffC_NN | and it still gives me the same message |
| 22:27.24 | rmudgett | JeffC_NN: DAHDISendCallreroutingFacility is only valid on EuroISDN(ETSI) and Q.SIG. |
| 22:28.52 | JeffC_NN | ok, thanks :/ |
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