| 00:05.03 | wyoung | Kobaz: Violence is bad mm'kay |
| 00:17.41 | *** join/#asterisk Frod (~Frod@200.9.182.82) |
| 00:17.47 | Frod | hello all |
| 00:18.09 | Frod | what good queue monitors are available for asterisk |
| 00:18.20 | Frod | like a wallboard for a call center |
| 00:18.40 | Frod | i have seen projects like qpanel and monast |
| 00:19.05 | Frod | that do the job but i want to know what people that use asterisk use |
| 00:19.28 | [TK]D-Fender | Do people actually use wallboards anymore? |
| 00:20.20 | Kobaz | yeap |
| 00:20.29 | Kobaz | i sell one wallboard with every call center we build |
| 00:20.34 | Kobaz | everyone loves it |
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| 00:21.53 | Kobaz | qpanel looks nifty, i may have to borrow some ideas from that |
| 00:23.25 | Kobaz | Frod: i built a wallboard in c# that we sell, but it only works with our call center platform, not vanilla asterisk |
| 00:24.07 | wyoung | hai Frod |
| 00:25.17 | Frod | Kobaz: what is your platform name |
| 00:25.24 | Kobaz | Intellasoft |
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| 03:17.31 | life_of_e | I've got my FXO ATA set to automatically forward any calls to Asterisk. It appears to autoanswer the call when it performs the forward so the other side hears silence. I used Ringing() in the dialplan to pass back ringing and then it's followed by Answer() before Dial(). |
| 03:17.59 | life_of_e | There's a blacklist check right after answering which will perform a PlayTones() then hang up. Do I really need the Answer() in this? |
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| 04:31.10 | [TK]D-Fender | the SIP device answered the line so the telco doesn't continue ringing. As it calls in to * there is no audio from * until * answers |
| 04:32.24 | [TK]D-Fender | Good odds the ATA doesn't have any logic in it to take ringing back from * and generate tones |
| 04:39.05 | life_of_e | Yeah, I just started experimenting and I half needed Answer(). I didn't need it prior to the Dial() but I did need it for PlayTones() |
| 04:41.08 | life_of_e | I have my blacklist playing the intercept tone to dump bogus calls. Moving Answer() into that branch and leaving it out of the Dial() branch leaves the telco's ring audio in place. It sounded jarring when the Answer() picked up and then Asterisk's Ringing() audio came through. |
| 04:41.56 | [TK]D-Fender | Actually "playtones" is inherently audio... if you used Ringing() instead maybe it would have worked |
| 04:42.38 | [TK]D-Fender | holdon... scratch that.. you already tried that |
| 04:42.44 | [TK]D-Fender | I'm a little tired here. |
| 04:42.49 | life_of_e | No worries |
| 04:42.53 | [TK]D-Fender | so yeah I guess myt first idea stands |
| 04:42.56 | life_of_e | It was worth the experiment |
| 04:43.49 | life_of_e | I got it working so it sounds right to the caller on the telco side (stays on the telco's ring signal without switching) and answers to do PlayTones if it's blacklisted |
| 04:44.33 | life_of_e | It was pretty cool, my cell phone hung up as soon as it sensed the intercept tones. :) |
| 04:45.34 | [TK]D-Fender | that sounds unusual considering its call to the telco should be considered answered and those would be inband |
| 04:46.19 | life_of_e | Yeah, I guess there's a process that siphons the audio out and runs it through a DSP |
| 04:46.38 | [TK]D-Fender | Maybe it was the ATA |
| 04:46.51 | life_of_e | That's always possible |
| 04:46.53 | [TK]D-Fender | actually it would ahve to be... |
| 04:47.10 | [TK]D-Fender | cell phones don't process audio to make decisions |
| 04:47.17 | *** part/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 04:47.20 | life_of_e | Even funnier then considering the ATA was taking the call |
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| 04:48.40 | life_of_e | I'm fine with it either way, the tones make it through to the other end which is enough for a robodialer to drop the number |
| 04:48.55 | [TK]D-Fender | "it works"... yeah that's usually enough for me |
| 04:49.18 | life_of_e | Precisely :) This is just a personal project so I'm in it for the Stupid Asterisk Tricks |
| 04:50.58 | life_of_e | But having this feature will help with our robodialer problem. I currently have a serial modem and a raspberry pi watching the caller ID and doing an answer/hangup on blacklists. |
| 04:51.36 | life_of_e | It was just annoying to edit the list to add new callers. Now with Asterisk I was able to record the last caller ID then have an extension I can call to put the number into the blacklist |
| 05:19.17 | wyoung | o/ |
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| 11:09.31 | DanielYK | I have a RTP codec problem with a specific endpoint. The RTP payload is switching every 3-5 seconds. If g722 is the only allowed codec, it is switching between Payload Type 9 (g722) and Payload Type 6 (DVI4 16k). If alaw is the only allowed codec, it is switching between Payload Type 8 (alaw) and 1 (fs-1016). I have set preferred_codec_only=yes and asymmetric_rtp_codec=no without success. Only the RTP stream from asterisk to the |
| 11:09.32 | DanielYK | <PROTECTED> |
| 11:15.23 | DanielYK | Does anyone have an idea? |
| 11:44.17 | DanielYK | @file Maybe you? |
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| 15:45.18 | pts0 | How do you setup pjsip for ip auth like you do with chan_sip? |
| 15:45.30 | pts0 | Is this part still the same? same => n,Dial(SIP/25501293*12125678974@flowroute,,r) |
| 15:46.23 | pts0 | What is different in pjsip.conf compared to the wiki tutorial example? |
| 15:46.40 | [TK]D-Fender | The auth in your endpoint is just like in the chan_sip peer |
| 15:46.53 | [TK]D-Fender | that dialplan line doesn't actually contain "auth" |
| 15:47.31 | sibiria | PJSIP/123@endpoint for pjsip |
| 15:49.44 | pts0 | So for the trunk example here, I can leave out the auth section? https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples |
| 15:50.16 | pts0 | type=auth |
| 15:50.16 | pts0 | auth_type=userpass |
| 15:50.16 | pts0 | password=1234567890 |
| 15:50.16 | pts0 | username=1234567890 |
| 15:50.21 | [TK]D-Fender | PASTEBIN <- |
| 15:50.23 | pts0 | leave that out? |
| 15:50.29 | pts0 | oops sorry |
| 15:50.53 | [TK]D-Fender | What do you even mean by "auth" here exactly? |
| 15:51.04 | [TK]D-Fender | what action are you trying to take that is specifically failing? |
| 15:51.56 | pts0 | I'm just going to transition to pjsip and wanted to know |
| 15:52.18 | pts0 | I suppose I don't have to login to flowroute since I am using ip auth |
| 15:52.30 | file | if you are referring to the remote party, flowroute, doing ip auth on your requests - then you don't configure an auth section and don't configure the outbound auth in the endpoint |
| 15:52.33 | pts0 | so I'm guessing that in pjsip I can just leave that out as I do in sip.conf |
| 15:52.53 | file | "ip auth" can refer to incoming IP based authentication, so in the future it's best to be explicit as most people would think of that first |
| 15:52.55 | [TK]D-Fender | The term you were using is rather vague and when it's even used by others typically the direction of the call is what's important in determining the actual usage |
| 15:53.36 | pts0 | ok sorry...thanks I think I got it |
| 15:53.55 | pts0 | by the way, what is the ,,r in that line above--the dialplan line/ |
| 15:54.36 | sibiria | no timeout + dial tone |
| 15:54.52 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial |
| 15:57.07 | pts0 | thanks, looks like it can be left out since it says default |
| 15:57.10 | pts0 | Default: Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling party until the called channel has answered. |
| 15:57.44 | pts0 | thanks all |
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| 15:58.56 | [TK]D-Fender | That isn't technically all that it means. |
| 15:59.38 | [TK]D-Fender | The other side can be ringing, it's just whether * cares to count it as such in the absence of them telling you it's ringing. |
| 16:00.36 | [TK]D-Fender | I've seen a few odd carriers that failed to pass call progress between networks and you had to take on faith that the destination was actually successfully reached to be able to start ringing. |
| 16:01.02 | [TK]D-Fender | I think is was some oddball African country a guy came in trying to connect with or some such. Was a LONG time ago.... |
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| 18:16.29 | life_of_e | If a Dial() command is issued to dial with options an extension in another context and that extension issues its own Dial() with options, which set of options is in effect for the channel? |
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| 18:18.48 | file | if you mean using a Local channel, then both |
| 18:19.24 | file | the particular interaction depends on the options in use. |
| 18:20.33 | life_of_e | Ok, so if I have Dial(local/exten@context,,options_set_A) and within context I have exten issue Dial(technology/resource,,options_set_B) then in most cases the total set of options will be the union of options_set_A and options_set_B? |
| 18:21.36 | file | in a way - there is no special logic or anything to combine them |
| 18:22.12 | file | it's just by virtue of Local channels internally exchanging media/events that there is a relation |
| 18:22.17 | life_of_e | Ok, so as long as there's no conflicting values. Say set_A was just X and set_B was k, the end result would be a channel with Xk |
| 18:23.31 | file | yes |
| 18:23.36 | life_of_e | Cool, thank you |
| 18:23.40 | file | although what exactly would happen depends - because it's not 1 channel |
| 18:23.45 | file | there's 4. |
| 18:23.51 | life_of_e | Oh I see |
| 18:24.00 | file | Caller -> Local,1 -> Local,2 -> Called |
| 18:27.45 | life_of_e | I asked because I have one context handling my incoming FXO ATA. I used Dial() to jump to my internal-phones context which has lots of included contexts. There the extension performs a Dial() on all the phones. I wanted to enable call recording by the callee for incoming calls but I had the chained Dial()s so I wasn't sure. |
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