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| 02:13.36 | raspberrypifan | anyone around im getting an issue [Jun 30 22:09:51] NOTICE[26238][C-00000106]: chan_sip.c:25859 handle_request_invite: Call from 'tech1' (xxx.xxx.xxx.xxx:5060) to extension '09xx463592' rejected because extension not found in context 'stations'. |
| 02:13.43 | raspberrypifan | im trying to dial out of my gateway |
| 02:13.47 | raspberrypifan | not directly to an extension |
| 02:13.51 | raspberrypifan | inside asterisk |
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| 03:21.01 | wyoung | <PROTECTED> |
| 03:23.46 | raspberrypifan | wut |
| 03:24.18 | wyoung | I am here |
| 03:24.41 | raspberrypifan | do i know u |
| 03:24.57 | wyoung | probably not, but you did ask if any one was around |
| 03:25.16 | wyoung | I am assuming you want assistance to an issue? |
| 03:25.17 | raspberrypifan | oh |
| 03:25.20 | raspberrypifan | yup |
| 03:25.23 | wyoung | ok I have read it |
| 03:25.33 | wyoung | how are you placing the call? |
| 03:25.40 | wyoung | from a SIP phone? |
| 03:25.58 | raspberrypifan | yea |
| 03:28.40 | wyoung | ok so what does your stations context look like? can you paste it in bpaste or hastebin? |
| 03:29.06 | wyoung | (extensions.conf, unless you are using .lua or .ael) |
| 03:30.13 | raspberrypifan | sure |
| 03:30.20 | raspberrypifan | one moment stvp |
| 03:35.02 | wyoung | ok |
| 03:35.14 | raspberrypifan | http://pastebin.com/FErmxVSr |
| 03:37.08 | wyoung | I want to look at your extensions.conf only, not advertisements too |
| 03:38.52 | raspberrypifan | dont see no adverts |
| 03:38.54 | raspberrypifan | but sorry if i did |
| 03:41.48 | wyoung | what extension is the top extensions in/ |
| 03:41.54 | wyoung | oops, what context even |
| 03:42.14 | wyoung | and in your sip.conf, what context are your SIP phones in? |
| 03:43.08 | raspberrypifan | [outbound] |
| 03:43.15 | raspberrypifan | sip.conf they are in stations |
| 03:54.00 | raspberrypifan | apparently the solution was to add a . |
| 04:14.20 | wyoung | ? |
| 04:16.04 | snadge | this is going to be a super dumb question.. but if you have [SomeTrunk] host=dynamic username=test secret=password |
| 04:16.18 | snadge | why do i have to register with username SomeTrunk instead of test |
| 04:17.46 | snadge | im scratching my head wondering why im getting auth failed.. going.. no.. thats the right username.. thats the right password.. wtf.. in this case the username and the trunk name were the same, but the trunk had a capital at the beginning |
| 04:18.01 | wyoung | snadge: you register with the username, the [SnomTrunk] is what you use to reference it in extensions.conf |
| 04:18.14 | snadge | thats what i thought too |
| 04:18.33 | snadge | this is sip.conf rather |
| 04:18.39 | wyoung | snadge: you can also have situtations where you register with one username and password and have a peer entry ([SomeTrunk]) that uses different credentials |
| 04:19.04 | drmessano | username is used in outbound authentication to another peer |
| 04:19.07 | wyoung | I mean I don't know why you would want to do it but it is possible |
| 04:19.23 | drmessano | If you're registering TO Asterisk you use the [peername] |
| 04:20.07 | snadge | right.. of course that makes sense now.. i clearly need more sleep or something |
| 04:22.45 | snadge | normally i create separate inbound and outbound trunks.. i knew there was a reason why :P |
| 04:22.53 | snadge | this one just happens to be used for both.. because lazy |
| 04:23.26 | snadge | and apparently stupid too |
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| 05:37.32 | wyoung | snadge: I just setup a friend and stick it in a pretty limited context |
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| 08:46.54 | UncleKiwi | thanks for asterisks |
| 08:46.57 | UncleKiwi | :) |
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| 11:12.12 | cervajs2 | hi, i have asterisk pjsip trunk - asterisk chan_sip trunk . trunk works but sip options from chan_sip to pjsip are not authorized. any hint? |
| 11:12.32 | cervajs2 | res_pjsip/pjsip_distributor.c:368 log_unidentified_request: Request from |
| 11:18.07 | cervajs2 | solved: mistake in identify section |
| 11:19.11 | cervajs2 | partially solved. log_unidentified_request are gone. but in sngrep i see that response is still 401 unautorized |
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| 11:23.38 | [[thufir]] | does this article describe just inbound calling, receiving calls from the trunk? http://support.siptrunk.com/hc/en-us/articles/204848079-SIPTRUNK-com-CONFIGURATION-GUIDE-FOR-ASTERISK |
| 11:36.59 | [[thufir]] | I can't find the syntax, but I recall reading that it's possible to break down extensions.conf into seperate files. What is that called? |
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| 11:37.52 | [[thufir]] | #include I think |
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| 11:56.30 | [TK]D-Fender | [[thufir]], that just should the SIP pper & register strings. that covers both directions |
| 11:56.34 | [TK]D-Fender | peer* |
| 11:59.11 | [[thufir]] | [TK]D-Fender: but the context is from-trunk. Shouldn't there be a corresponding to-trunk? |
| 11:59.36 | [TK]D-Fender | Peers only have one context |
| 11:59.47 | [TK]D-Fender | When you use a peer to call out SIP knows NOTHING about "context" |
| 11:59.52 | [TK]D-Fender | that is an * dialplan concept |
| 12:00.10 | [TK]D-Fender | context= is where calls FROM the trunk will land |
| 12:00.16 | [TK]D-Fender | and that name is whatever you want it to be |
| 12:00.45 | [TK]D-Fender | They are very clearly showing this set up for a FreePBX system |
| 12:01.19 | [TK]D-Fender | And that is the context FreePBX uses to handle calls from what they'd consider a "trunk" so that it gets matched against Inbound Routes |
| 12:01.38 | [[thufir]] | ok |
| 12:01.43 | [TK]D-Fender | ~book |
| 12:01.43 | 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 |
| 12:01.45 | [TK]D-Fender | ^^^ |
| 12:02.28 | [[thufir]] | why do you say freepbx, though? the title has "asterisk". |
| 12:05.29 | [TK]D-Fender | A. because of the specific context they chose. B. Becuase read the LINKS on the right hand side. FREEPBX MODULE. ELASTIX. FREEPBX. |
| 12:06.17 | [TK]D-Fender | And they know anyone who knows anything about Doing it in Asterisk for their own manual dialplan will know to set the context to wherever they wanted anyway |
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| 12:09.34 | [[thufir]] | ok |
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| 14:11.10 | doppler | good morning |
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| 14:16.01 | doppler | if anyone here has experience with polycom equipment, give me a holler :) |
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| 15:04.43 | ||cw | doppler: just ask the question |
| 15:05.28 | doppler | well⦠|
| 15:05.42 | doppler | how well do soundstations typically play with asterisk? |
| 15:06.05 | doppler | i'm thinking of picking up a few 550's or 650's, and from what i can tell, they will *work* |
| 15:06.11 | doppler | but the question is, how well? :) |
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| 15:06.49 | doppler | primarily interested in polycom since their speakerphones are supposedly excellent |
| 15:07.02 | doppler | (and they don't look like toys) |
| 15:08.16 | doppler | they're cheap enough on the used market that i could just buy a few and find out firsthand, but i thought i'd see if some other users had an opinion |
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| 15:12.53 | dwigton_ | Does it matter where your asterisk box is in your network topology as long as it has access to the web? Or does it need to be outside of any firewalls etc? Or at least have a 1-1 NAT? |
| 15:14.10 | dwigton_ | I am just a bit confused how it can sit on your LAN without opening up a hole. |
| 15:14.56 | dwigton_ | Unless asterisk intiates the connection to your SIP provider. |
| 15:15.47 | shido6 | eliminate nat if possible - but it can be behind a firewall - test at scale before certifying the service as production ready. |
| 15:16.12 | shido6 | you may find your firewall has hidden functionality |
| 15:16.36 | dwigton_ | hidden functionality? |
| 15:16.48 | shido6 | SIP ALG, SIP DDoS detection, etc. |
| 15:17.49 | dwigton_ | we are using pfsense |
| 15:18.09 | shido6 | in that case, hidden sip proxy |
| 15:18.43 | shido6 | If someone went out of their way to configure the proxy - you may need to disable it - |
| 15:19.42 | dwigton_ | I have no idea any of that means. Any SIP traffic on the lan gets handled by pfsense instead of making it out to the isp? |
| 15:20.11 | shido6 | its possible but we can discern those things when make test calls and post the debug in pastebin.com and send us the link to review the data |
| 15:21.01 | dwigton_ | I am not at that point yet. I am in the reseach phase to make a proposal for replacing our ancient pbx. |
| 15:22.53 | dwigton_ | I will try to scrounge up some hardware or a vm for a freepbx install. Can asterisk make connections to the SIP provider that by passes the current appliance the isp provided for our legacy system? |
| 15:23.12 | shido6 | yes |
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| 15:23.32 | shido6 | do you have a link to the public internet that doesn't use that appliance? |
| 15:23.55 | dwigton_ | No, I don't think so. |
| 15:25.04 | dwigton_ | Does the test need to be done on their fiber or can we connect from another location? |
| 15:26.03 | dwigton_ | Also I need to verify that they actually are providing sip. Do we just ask them for credentials for our 5 lines? |
| 15:26.54 | dwigton_ | ^probably an idiot question but no idea how this really works. |
| 15:27.00 | shido6 | you can ask them , sure |
| 15:27.11 | shido6 | find out what you are paying |
| 15:27.30 | shido6 | look at your bills for the last 6 months - |
| 15:27.59 | dwigton_ | Ah looking for SIP charge? |
| 15:28.26 | dwigton_ | I will have to hunt the real admin down and see if he will let me look at those. |
| 15:28.45 | shido6 | phone bill - telecom charges, lines, phone numbers, you can get on the phone your provider if you have questions - |
| 15:29.02 | shido6 | with with with with - whats wrong with my fingers today with with with |
| 15:29.24 | shido6 | let you? |
| 15:30.12 | shido6 | Speak with an intelligent tone and inquire |
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| 15:54.22 | albatrosss | hi all |
| 15:56.15 | albatrosss | is it possible to call a global variable by adding channel variable |
| 16:01.14 | albatrosss | i have a global variable ----> system_day_on=0900 and system_day_off=1500, i can call this in the dialplan no problem, i however want to call it changing the day why won't this work $[${system_(${day})_open}] |
| 16:02.21 | albatrosss | forgot to mention i have multible global variable with a different day |
| 16:03.51 | WIMPy | So you have a system_monday_on etc? |
| 16:04.22 | albatrosss | yes |
| 16:04.34 | WIMPy | But you don't have variables called system_(monday)_on? |
| 16:04.54 | albatrosss | i do |
| 16:05.05 | albatrosss | for every day of the week |
| 16:05.25 | WIMPy | You have both names? |
| 16:05.51 | albatrosss | ? |
| 16:06.27 | WIMPy | Hint: With or without brackets. |
| 16:07.33 | albatrosss | global var looks like this system_Friday_open=0900 |
| 16:07.37 | albatrosss | no brackets |
| 16:08.17 | WIMPy | So you won't find a value if you're looking for a varaibale wit brackets in it's name. |
| 16:08.45 | albatrosss | i get the day using STRFTIME and want to pass that too the global variable |
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| 16:12.02 | albatrosss | set(foo="Friday) then call global NoOp(${system_${foo}_open}) |
| 16:12.15 | albatrosss | i think i lost the plot this also does not work |
| 16:12.33 | TandyUK2 | anyone aware of an ATA with 4 analog ports (for 4 seperate accounts) |
| 16:12.49 | WIMPy | "Friday" or "friday"? |
| 16:12.51 | albatrosss | grandstream |
| 16:13.13 | albatrosss | Friday |
| 16:14.11 | TandyUK2 | HT704? you actually used one? |
| 16:14.18 | WIMPy | And die you set foo=Friday or foo="Friday"? |
| 16:14.35 | Penguin | set(foo="Friday) |
| 16:14.38 | Penguin | "Friday |
| 16:14.42 | WIMPy | Building a similar name doesn't do anything. It has to be exact. |
| 16:15.10 | albatrosss | lol sorry gents been staring at this screen the whole day missed that one |
| 16:15.27 | albatrosss | my code does have both "" |
| 16:15.56 | WIMPy | Remove them. |
| 16:15.59 | Penguin | I suggest that you don't use quotation marks in your set. |
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| 16:16.38 | albatrosss | thx will give it a go |
| 16:20.38 | albatrosss | you guys rock, working thank you so much |
| 16:25.49 | TandyUK2 | anyoen know of a good voip handset for use in very dirty (dusty) environments |
| 16:26.12 | TandyUK2 | it would be wall mounted, where they cut stone blocks |
| 16:26.17 | TandyUK2 | so insanely dusty in there :P |
| 16:44.16 | albatrosss | TandyUK2:why don't you slap down a ata, analog handsets are more robust the voip handsets, also much cheaper to replace |
| 16:45.57 | Samot | ^ Probably the best option. |
| 16:46.35 | Samot | Don't want to spend $100 on a phone to have it destroyed by brick dust getting inside it or shrapnel hitting it. |
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| 16:48.03 | Penguin | I would suggest a $10-$15 basic bell telephone installed on the wall and an ATA placed in a cleaner location with a regular phone cord connecting the two. |
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| 17:00.49 | doppler | i second this |
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| 18:53.46 | albatrosss | is there a way that you can change a quember possition on the fly? |
| 18:54.01 | albatrosss | that is queuemember sorry |
| 18:57.53 | file | no. |
| 19:10.49 | albatrosss | thx |
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| 20:05.55 | albatrosss | if you force a queue member to pause whiles agent is on a call will you disconnect the call the user is busy with? |
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| 21:06.07 | WIMPy | Hmm. Does Asterisk accept both = and == in comparisons? |
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| 21:30.54 | TandyUK | WIMPy: if its like other lanbuages, = sets something, == compares |
| 21:35.14 | WIMPy | It's not a programming language :-) |
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| 21:47.53 | brettnem | hey all, I'm calling an AGI script which drops a user into a conference room. But what I really want to do is have the user basically on hold while my AGI script loops, making REST calls until a condition is met, then redirect the call. Because the callout to ConfBridge is blocking, I can't really check my REST call. What's a good way to do this? |
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| 21:49.55 | brettnem | I tried using AMI from my agi to redirect the call into a conference room and continue in my AGI, but when I redirect, the AGI is terminated. I've tried various combinations of signal trapping, and AGIEXITONHANGUP and AGISIGHUP but I can't seem to figure it out |
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| 21:51.50 | WIMPy | Because thee redirect action redirects the call away from the AGI application. |
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| 21:56.18 | robmal | Hey, anyone familiar with cisco 79xx? I've almost got them to do their work but DnD still works not the way i'd want. When enabled all it does it mutes the ringing, but i'd like it to i.e. go to voicemail. Cisco says it can be done when call forwart no answer is set but i can't find any way to set it via xml configs. Any clues? |
| 22:04.42 | brettnem | WIMPy: yeah gathered that. I'm looking for a better way to do this. Any ideas? |
| 22:05.38 | WIMPy | Dunno what you're up to. |
| 22:07.01 | brettnem | WIMPy: well just like I said above. I want to be able to do blocking things, like have the call wait in a conference room until an event happens then redirect it away. So I want to be able to park the call for some time, then later come back and redirect it.. Problem is with straight agi, once I drop into a conference I can't do anything else in the channel because it's blocking |
| 22:08.38 | WIMPy | Yes, that's the way the dialplan works. |
| 22:09.25 | brettnem | WIMPy: I'm aware that's how it works. I'm asking if anyone has any ideas on how to do what I'm trying to do here |
| 22:09.26 | WIMPy | But I still don't konw *when* you want to do something. But maybe you want to use more AMI? Or even ARI? |
| 22:10.09 | brettnem | WIMPy: I want to drop the call into a conference room. Then every 10 seconds I'm going to call to a REST endpoint. it will tell me "wait" or "go" when it says "go" I want to redirect the channel |
| 22:11.02 | WIMPy | Then don't do it inside your AGI. Fork another process. |
| 22:11.40 | WIMPy | Or write an application that stays connected to AMI. |
| 22:12.30 | brettnem | WIMPy: yeah my next attempt was a fork. Just wondering if there was a cleaner asterisk way to do it while maintaining the local channel connection. |
| 22:12.57 | WIMPy | AMI/ARI |
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| 22:13.28 | brettnem | I see some reference to async AGI whichj looks appealing but I'm failing to understand how using it is even possible |
| 22:13.46 | brettnem | once you call async:agi you are blocked and can't run a background app that performs the AMI actions on it. |
| 22:14.31 | WIMPy | Either fork or don't use AGI. |
| 22:15.45 | brettnem | ok thanks |
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