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| 00:47.34 | navaismo | is anybody here? If so, does anyone know how to use websocket to pull info from asterisk 11, i know that 12 has ARI |
| 00:48.33 | WIMPy | The wiki? |
| 00:51.46 | navaismo | ah that site is banned from my mind, time to looking at |
| 00:51.51 | file | that functionality does not exist in 11 |
| 00:51.57 | navaismo | oh :( |
| 00:52.28 | navaismo | well then, time to struggle with wss again in 12 |
| 01:13.35 | epaphus | Could anybody can please provide me an example or link of the screening function? much appreciated/ |
| 01:16.08 | navaismo | http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial option p |
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| 01:27.53 | Dovid | hello ev1 |
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| 01:37.59 | MaliutaLap | Dovid: nobody is here, we all died of heartbleed ;P |
| 01:38.11 | Dovid | lol |
| 01:39.53 | lvlinux | lol |
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| 01:44.44 | navaismo | I know im missing something and it must be very stupid but my cli have a flood with " res_pjsip_mwi.c:589 mwi_validate_for_aor: Endpoint '5000' already configured for unsolicited MWI for mailbox '5000@default'. Denying MWI subscription to 5000" |
| 01:59.40 | lvlinux | i don't know but it sounds to me like your phone is trying to subscribe and you also have "mailbox=5000@default" on the peer in sip.conf?? Does that make sense? |
| 02:00.25 | lvlinux | i would say try changing your phone settings to not check a mailbox with the LED or whatever, or delete that line in sip.conf |
| 02:01.12 | navaismo | yep mailbos is set in the table, and mailbox exist |
| 02:01.16 | navaismo | mailbox* |
| 02:02.09 | lvlinux | what kind of phone is it? |
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| 02:10.28 | navaismo | yealink |
| 02:12.01 | navaismo | anyway im going to shutdown my machine this is my asterisk test system so no problem |
| 02:12.06 | navaismo | thanks! |
| 02:12.22 | navaismo | and see you all |
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| 09:42.38 | Ibrahim22 | Hi, everyone! Is it possible to stop MOH on a certain channel, if you have the name of the channel available? like StopMusicOnHold(<channel name>) |
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| 12:17.32 | WIMPy | Ibrahim22: No. You could only redirect it to somewhere else in the dialplan. |
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| 12:20.33 | Ibrahim22 | Ah, okay |
| 12:22.23 | Ibrahim22 | I want to be able to have AgentRequest fire its AgentConnect event, have the caller stay on hold for another x seconds before actually connecting the two. Can this be done? |
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| 12:23.30 | WIMPy | I'm not sure what exactely you want to do. |
| 12:25.30 | Ibrahim22 | I want a delay of X seconds before connecting the caller and agent when AgentRequest in the dialplan is called. |
| 12:25.53 | Ibrahim22 | so my script (AMI client) is loaded before the caller is connected with my agent |
| 12:26.31 | WIMPy | Why isn't it connected? |
| 12:26.59 | WIMPy | AMI is usually best used with a permanent connection. |
| 12:27.23 | Ibrahim22 | Sorry, I didn't explain correctly |
| 12:27.26 | Ibrahim22 | One second |
| 12:31.10 | Ibrahim22 | I have an AMI client (daemon) that listens for the AgentConnect event, and shows a script (actual text script) to the agent. My manager wants a delay of x seconds before the agent actually connects to the caller so they can read it beforehand. I tried using BridgeWait which actually did almost exactly what I wanted, except it didn't play the announ |
| 12:31.10 | Ibrahim22 | cement file to the agent. So now I'm looking for something else. |
| 12:31.25 | file | oh, announcement... not cement |
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| 12:36.11 | WIMPy | Ok, I see. Nut I don't use AgentConnect so I have no idea, where to sensibly hook in to that one. |
| 12:36.59 | WIMPy | I don't even have that anywhere. |
| 12:37.22 | Ibrahim22 | I thought of using a UserEvent before calling AgentRequest, but some data is missing for me to form an actual script |
| 12:39.46 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 12:39.46 | *** 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 |
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| 13:45.00 | syshape | is it possible to simulate activity on asterisk ? I'd like to test a monitoring program. |
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| 13:58.50 | file | syshape, I don't understand the question |
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| 14:04.08 | napnap | Hi all, how can I get the dahdi channel number of a fxs port (for example) ? |
| 14:10.09 | jameswf | napnap: answered in #FreePBX |
| 14:20.19 | syshape | file: I'd like to test my monitoring program, but there's no activity on my asterisk, I'd want fake data |
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| 14:25.05 | file | sipp can generate SIP calls, or you can originate arbitrary calls from within Asterisk |
| 14:28.50 | syshape | to generate calls ? but how do we answer ? |
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| 14:34.29 | file | you send them to an extension that does something, or to somewhere else through Asterisk |
| 14:39.43 | jameswf | syshape: I use to run them to an extension with MOH then tear em down |
| 14:40.09 | jameswf | puts a load on the system |
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| 14:45.56 | *** join/#asterisk DruidZ (~darcy@dilbert.druid.net) |
| 14:47.42 | DruidZ | Good morning |
| 15:06.57 | syshape | thanks |
| 15:07.03 | syshape | cyou last week |
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| 15:13.18 | DruidZ | Anyone have any pointers? Some callers to me do not hear ringing but some do. |
| 15:13.50 | DruidZ | Every page I find talks about how to fix that it never works. |
| 15:14.01 | [TK]D-Fender | Your description is very weak.... |
| 15:14.07 | DruidZ | I know. |
| 15:14.38 | DruidZ | Basically people call me and when I answer they tell me that they never hrard a ringing on their end. |
| 15:14.55 | DruidZ | But not everyone. |
| 15:15.16 | DruidZ | For example, I cannot reproduce by calling in on my cell phone. |
| 15:15.41 | [TK]D-Fender | Still no useful details... |
| 15:15.50 | DruidZ | Asterisk 11.7.0. |
| 15:16.04 | [TK]D-Fender | Asterisk version has nothing to do with that.... |
| 15:16.20 | DruidZ | I know but people always ask. |
| 15:16.49 | [TK]D-Fender | Generally only when it actually matters... or the requester has no clue |
| 15:23.35 | DruidZ | One page suggested Playtones(ring) but that just gives me an error. Probably an old page. |
| 15:27.55 | DruidZ | The dialplan couldn't be much simpler: http://www.vex.net/~darcy/NoRing.txt |
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| 17:26.44 | Vendigroth | Does any one know if there Is there a way to get a Record-Route header somehow? it doesn't seem to be saved in any of the Chan variables |
| 17:28.57 | [TK]D-Fender | Vendigroth: "core show function SIP_HEADER" |
| 17:35.27 | Vendigroth | [TK]D-Fender aha! that should work. thank you |
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| 17:42.10 | DruidZ | I seem to have been disconnected from IRC all day but with no notification. Did anyone have any suggestions for my no-ringing issue earlier? |
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| 17:43.19 | [TK]D-Fender | You gave no useful details |
| 17:44.12 | [TK]D-Fender | And you asked 2 and a half hours ago.... not "disconnected all day" |
| 17:44.27 | [TK]D-Fender | And only dropped off a little over an hour ago |
| 17:57.49 | newtonr | DruidZ, you need to show debug from a call where the far end reports not hearing ringing, and debug from a call where they hear ringing. Then people here could at least help you tell if Asterisk is doing anything different in the two cases.. |
| 17:58.28 | newtonr | as for me, i'll be back later |
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| 23:38.25 | gusto | hi WIMPy |
| 23:38.31 | gusto | hi folks |
| 23:39.00 | gusto | what happend to this asterisk12? i have troubles installing it on every platform because of res_rtp_asterisk lacking some dependencies |
| 23:40.20 | gusto | now, under arch i have following problem: http://ur1.ca/h43kx |
| 23:40.25 | [TK]D-Fender | Have you considered installing the dependencies? |
| 23:40.32 | gusto | yes |
| 23:40.38 | gusto | but i can not find that libpj |
| 23:40.55 | gusto | and i have no idea what it is |
| 23:48.13 | lvlinux | what distribution |
| 23:48.15 | lvlinux | ? |
| 23:48.17 | gusto | arch |
| 23:49.10 | lvlinux | i think libpj is from pjsip, which you can download the source. |
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