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| 01:08.20 | scv | jeffg: no prob, glad you got it. |
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| 03:34.56 | nickgaw | Hi, Is there such a thing as a hosted asterisk system that a company or someone can help me setup should I require such a service for use with sip phones or is it best to run my own virtual private server and setup my own system? |
| 03:40.33 | scv | either choice is available |
| 03:41.11 | nickgaw | What do you do on your setup? |
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| 03:54.34 | scv | i'm a hosted pbx provider |
| 03:55.21 | nickgaw | How much do people generally charge to setup a small system with three extentions and a prompt that says to dial the extention and to have each extention to have their own voicemail and for the calls to be recorded and stereo mixed where the caller is on one channel and the other person is on the other channel? |
| 03:57.30 | nickgaw | What is your companies web site so I can take a look and do you personally work on people's systems? |
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| 07:08.54 | Vamp898 | Hi there, starting with today we get a lot of these when i look at asterisk with asterisk -rvvv "failed to extend from 256 to 451" these seem to happen when the system is idle (no call going on). But it seems like it have no real effect... telephoning (at least i think so) works as expected... |
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| 07:09.54 | Vamp898 | Ah ok, i just received a call from one user, hes telephone is not able to connect... |
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| 07:15.36 | Vamp898 | Ok, he had a different problem :D so it still seems like this error have no effect, but spams our logs |
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| 08:54.30 | *** join/#asterisk Haris (~haris@unaffiliated/haris) |
| 08:59.02 | Haris | ok, with sipp tool, I'v simulated basic udp sip registration |
| 08:59.18 | Haris | on my fresh, manual asterisk 13 install |
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| 09:31.36 | Samot | And have you been able to do an actual and real registration? |
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| 10:51.26 | Kunsi | http://paste.debian.net/788601/ do you see any errors in here? call is just ended, like there isn't any extension1 |
| 10:51.48 | Kunsi | oh, gotoin |
| 10:53.23 | Kunsi | putting a "hello world"-playback in front of it does work, but then it exits without an error |
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| 10:58.27 | Kunsi | "dialplan show" shows http://paste.debian.net/788603/, /tmp/raumstatus is 0644, and currently contains "1" |
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| 11:00.37 | fbnts | Hi, does anyone know what 'SIP/2.0 503 Service Unavailable (can't handle one-legged xfers)' means? I have tried googling but can't find any actual details of why transfer's are failing |
| 11:02.05 | file | what are you trying to transfer and to where? |
| 11:03.59 | fbnts | its from grandstream handsets. If they initiate a call out they are unable to blind transfer the call. (The screen says "Transfer Failed") but if they are the recipient of a call they can blind transfer |
| 11:04.26 | fbnts | the Asterisk console logs: "NOTICE[17930][C-000002b6]: chan_sip.c:24051 handle_response_notify: Got OK on REFER Notify message" |
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| 11:44.37 | Haris | anyone used sipp tool to troubleshoot or test sip calls using different transport methods when using asterisk as sip server ? |
| 11:45.01 | Samot | Haris: Have you tried to actually register an endpoint? |
| 11:45.24 | Samot | Again, that's how most people do it. Configure, connect, test. |
| 11:45.27 | Haris | yes. sipp's output suggests it was successful. asterisk cli shows header exchange for it too |
| 11:45.33 | Samot | OK |
| 11:45.36 | Haris | but! |
| 11:45.39 | Samot | Did you register a REAL endpoint? |
| 11:45.42 | Haris | this was through udp |
| 11:45.45 | Haris | not ws |
| 11:45.48 | Samot | Of course |
| 11:46.00 | Samot | And that tool is three years old since it's last update and doesn't support WSS. |
| 11:46.01 | Haris | sipp uses xml files for scenarios'. I have yet to devise one for ws/wss |
| 11:46.11 | Samot | Because that tool doesn't support it. |
| 11:46.16 | Haris | yep. it doesn't. but its the only tool I could find |
| 11:46.29 | Samot | What have you ACTUALLY done? |
| 11:46.45 | Haris | fresh VM. fresh asterisk install - manual build |
| 11:47.02 | Samot | Did you create your peers for the endpoints? |
| 11:47.02 | Haris | manually configure http.conf, sip.conf, rtp.conf |
| 11:47.10 | Samot | Did you setup any dialplan? |
| 11:47.16 | Haris | just added 2 extensions in sip.conf. nothing else |
| 11:47.24 | Samot | Do they register? |
| 11:47.29 | Haris | sip show peers does show I have 2 peers configured |
| 11:47.36 | Samot | Do they register? |
| 11:47.39 | Haris | registration works via udp |
| 11:47.43 | Samot | OK |
| 11:47.46 | Samot | So now... |
| 11:48.08 | Samot | Have you setup any dialplan to actually handle calls? |
| 11:48.16 | Haris | I checked WS connection via npm tool wscat. it connects to port 8088. don't know yet how to simulate something further till register or options headers from that point |
| 11:48.16 | Samot | Because you realize you need to do that right? |
| 11:48.22 | Haris | yes |
| 11:48.27 | Haris | that part - not done yet |
| 11:48.34 | Samot | You're not even anywhere close to testing WS. |
| 11:48.34 | Haris | I have configured the following though |
| 11:48.41 | Samot | Because you haven't configured anything for it. |
| 11:48.59 | Haris | dial=SIP/1060 under the extension config in sip.conf |
| 11:49.17 | Haris | I have yet to configure the 3-4 liner dialplan in extensions.conf |
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| 11:55.30 | Haris | this http://pastebin.ca/3686189 is my current config |
| 11:55.33 | Haris | on this manual install |
| 11:56.47 | Haris | checking up on this ( http://tomeko.net/other/sipp/sipp_cheatsheet.php?lang=en ) for tools |
| 11:57.25 | Haris | I have followed this ---> https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 |
| 11:58.16 | Haris | going to configure extensions.conf part now |
| 11:58.43 | Samot | So you've created the cert? |
| 11:58.46 | Samot | You're installed it? |
| 11:58.56 | Samot | You have the TLS part working? |
| 11:59.23 | Haris | I have the ssl cert .. that free trial one. I'v not yet installed it. going to do that too |
| 11:59.35 | Haris | tls is not that much of a change. first I want to get ws based calls working |
| 11:59.39 | Haris | then I can switch from ws to wss |
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| 12:03.45 | Haris | I put the same dialplan as mentioned here ( https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 ) |
| 12:03.53 | Haris | the 4 liner |
| 12:04.38 | Haris | replaced 1000 with my two extensions ( i.e., 1060 and 1061 ) |
| 12:05.25 | Samot | And can they call each other? |
| 12:06.35 | Haris | http://pastebin.ca/3686201 |
| 12:06.54 | Haris | added tiny dialplan mentioned at the end |
| 12:07.06 | Haris | will have to check |
| 12:07.06 | Haris | yet |
| 12:07.40 | Haris | is this config ok for basic call ? |
| 12:07.56 | Haris | asterisk is listening on udp/5060 and http/8088 |
| 12:09.56 | Haris | http://pastebin.ca/3686208 |
| 12:10.06 | Haris | hold please. need 20 mins break |
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| 12:37.01 | Haris | back |
| 12:42.49 | Haris | I have to enable audio/video codecs |
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| 12:46.03 | [TK]D-Fender | So do it |
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| 13:12.24 | Haris | how to see loadded codec ? |
| 13:12.32 | Haris | loaded+ codecs+ |
| 13:12.48 | Haris | show codecs command doesn't exist |
| 13:12.53 | Haris | http://www.voip-info.org/wiki/view/Asterisk+codecs |
| 13:13.12 | crisderock | core show codecs |
| 13:13.46 | Haris | long list |
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| 13:39.46 | [TK]D-Fender | Created by: flavour, Last modification: Fri 01 of Jul, 2011 (14:57 UTC) by hindmasj |
| 13:39.52 | [TK]D-Fender | 5 YEARS old. |
| 13:39.56 | [TK]D-Fender | Nice & current.... |
| 13:41.46 | Samot | Haris is a master of following current and relative documentation. |
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| 14:04.19 | Kunsi | voip-info.org is outdated, yes |
| 14:04.38 | Kunsi | or atleast not reflecting asterisk 11 and 13 commands and stuff |
| 14:09.59 | [TK]D-Fender | Dated... yes. The carbon kind... |
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| 14:58.08 | Samot | Just think, even if he gets WSS working. He'll have to write dialplan next. |
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| 16:11.48 | UnixGuru | is there a way to do MWI externally with asterisk 11? |
| 16:14.10 | Samot | You mean you want a remote (different PBX) endpoint to be notified? |
| 16:14.36 | UnixGuru | i want to do voicemail with out the voicemail app, but the MWI is tied to the voicemail app |
| 16:14.52 | UnixGuru | i have the agis, etc, for the voicemail |
| 16:15.08 | UnixGuru | the problem is setting the MWI and letting asterisk know the message count |
| 16:15.09 | Samot | So you're sending the voicemail to another server? |
| 16:15.12 | UnixGuru | yes |
| 16:15.18 | UnixGuru | well, no, but tyou could say, yes |
| 16:15.26 | Samot | It still will tell the endpoint that it has messages. |
| 16:15.32 | Samot | That's the point of MWI. |
| 16:15.40 | UnixGuru | endpoint registers to asterisk |
| 16:15.44 | Samot | Right. |
| 16:15.49 | UnixGuru | but asterisk gets the MWI info from app_voicemail |
| 16:15.59 | UnixGuru | app_voicemail does not know how many voicemails the user has |
| 16:16.03 | UnixGuru | because i am not using that |
| 16:16.13 | Samot | So turn off MWI |
| 16:16.30 | UnixGuru | ok, but i want MWI to work on the endpoint |
| 16:16.44 | UnixGuru | with asterisk 13, there is res_mwi_external, but i cant upgrade at this time |
| 16:16.54 | UnixGuru | i need something or a workaround for 11 |
| 16:16.57 | Samot | Then you need to tell the endpoint where to look for it's voicemail. |
| 16:17.28 | UnixGuru | what do you mean? |
| 16:17.37 | UnixGuru | normally the subscrube goes to where the registration goes |
| 16:18.29 | Samot | But it doesn't have to. |
| 16:18.44 | Samot | And res_mwi_external is tied to app_voicemail. |
| 16:19.06 | Samot | https://wiki.asterisk.org/wiki/display/AST/Message+Waiting+Indication |
| 16:20.31 | UnixGuru | in 13? that doc is old, you cant load app_voicemail and external mwi in 13, its one or the other |
| 16:20.45 | UnixGuru | but anyhow, i know thats the long term fix |
| 16:20.46 | Samot | That also show how to do it with Chan_SIP |
| 16:20.50 | UnixGuru | im trying to get a workaround |
| 16:20.59 | Samot | External MWI is another option. |
| 16:21.02 | shido6 | http://www.voip-info.org/wiki/view/Asterisk+at+large |
| 16:21.08 | shido6 | page full of workarounds |
| 16:21.13 | Samot | Which was introduced in Asterisk 12. |
| 16:21.14 | shido6 | enjoy - Good Luck, next |
| 16:21.15 | UnixGuru | thank you, im reading |
| 16:22.45 | Samot | Except on that page it shows that you need to patch Chan_SIP to do remote MWI. |
| 16:22.53 | Samot | The Wiki page I posted isn't from 2005 and Chan_SIP has that functionality in it. |
| 16:22.58 | Samot | So you just need to do what the page says. |
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| 17:06.42 | UnixGuru | jason@dynamicpacket.com |
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| 19:18.03 | scv | swet |
| 19:35.59 | scv | three cheers for gtjoseph for fixing #26241 |
| 19:36.10 | scv | and finding an additional case too |
| 19:36.32 | gtjoseph | which one was that? :) |
| 19:36.37 | scv | short rpai headers |
| 19:36.57 | gtjoseph | :) |
| 19:37.13 | gtjoseph | rmudgett found the additional case. :) |
| 19:37.27 | scv | additional cheers for rmudgett! |
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| 21:22.31 | raspberrypifan | could someone please help me out |
| 21:22.33 | raspberrypifan | i keep getting busy here |
| 21:22.40 | raspberrypifan | from my gsm gateway via asterisk |
| 21:22.45 | raspberrypifan | it works fine with my other softphones |
| 21:26.50 | Samot | Show a call and what is happening. |
| 21:26.59 | Samot | Is it an inbound or outbound call? |
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| 21:34.05 | raspberrypifan | inbound |
| 21:34.08 | raspberrypifan | i got logs |
| 21:34.38 | raspberrypifan | http://pastebin.com/W25AqD77 |
| 21:36.28 | raspberrypifan | thats from the gateway side |
| 21:38.33 | [TK]D-Fender | Show what * gets |
| 21:39.02 | raspberrypifan | alright one sec |
| 21:47.40 | raspberrypifan | http://pastebin.com/DL2nRXQY |
| 21:47.46 | raspberrypifan | i hope i cut out the relevant info |
| 21:48.56 | raspberrypifan | 209.208.212.148 |
| 21:49.01 | raspberrypifan | is the ip of origin |
| 22:12.22 | raspberrypifan | [TK]D-Fender: |
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| 23:48.57 | Samot | raspberrypifan: So you get busies on the inbound calls? |
| 23:49.09 | raspberrypifan | yes |
| 23:49.13 | raspberrypifan | well |
| 23:49.21 | raspberrypifan | the call comes in goes through asterisk and then goes out |
| 23:49.28 | raspberrypifan | it is when the call leaves asterisk that the issue starts |
| 23:50.25 | Samot | 186.4.236.29 <-- What IP is this for? |
| 23:50.35 | Samot | PBX, endpoint? |
| 23:50.52 | raspberrypifan | gsm gateway endpoint |
| 23:51.15 | Samot | 104.238.129.79? |
| 23:51.29 | raspberrypifan | asterisk box |
| 23:52.15 | Samot | https://www.irccloud.com/pastebin/dmi5OwHB/ |
| 23:52.31 | Samot | Looks like the GSM Gateway is returning the busies. |
| 23:56.17 | raspberrypifan | yea |
| 23:56.28 | raspberrypifan | thats the issue why is it doing that only for that user |
| 23:56.32 | raspberrypifan | for the other users its fine |
| 23:57.49 | raspberrypifan | would a succesful log be useful to look at? |