| 00:23.30 | *** join/#asterisk sinaowolabi (~Sina@105.112.38.3) |
| 00:27.39 | velix | Wow, the Alexa TTS is really nice. But I think, they cannot be used outside the Amazon cosmos. |
| 01:11.23 | *** join/#asterisk sinaowolabi (~Sina@105.112.179.12) |
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| 02:04.32 | *** join/#asterisk DannyA (~DannyA@cpe-74-64-125-9.nyc.res.rr.com) |
| 02:05.02 | DannyA | hey all. i've been googline for hours and i can't figure this out. how does redirect via ARI work? |
| 02:05.38 | DannyA | if i have two channels that are bridged, and i want to redirect one of them to a new extension (which will Dial a number), can i use redirect via ARI to do that? |
| 02:09.33 | [TK]D-Fender | AMI is certainly doable |
| 02:10.28 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Channels+REST+API |
| 02:10.35 | [TK]D-Fender | ARI should be as well |
| 02:10.51 | DannyA | but what is the endpoint parameter? |
| 02:11.42 | [TK]D-Fender | I don't know ARI well enough |
| 02:11.47 | [TK]D-Fender | (AKA at all) |
| 02:12.14 | DannyA | how would u do it through AMI? |
| 02:12.53 | [TK]D-Fender | Redirect |
| 02:13.15 | DannyA | right, but what are the parameters? |
| 02:13.22 | [TK]D-Fender | Go read the instructions |
| 03:10.39 | DannyA | @[TK]D-Fender got it to work with AMI. thanks! |
| 03:10.51 | [TK]D-Fender | glad to hear... |
| 03:41.37 | *** join/#asterisk jmordica (uid18332@gateway/web/irccloud.com/x-fzxwhmkqtbdqiazw) |
| 03:43.34 | jmordica | Anyone here that is interested in helping track down the cause of high load spikes on a fairly large asterisk setup? Willing to pay for some good consulting/direction. |
| 03:44.56 | Samot | What is a "fairly large" setup? |
| 05:06.54 | jmordica | A set fronted by kamailio and approx 150 simultaneous calls per asterisk server with some recording, queues, fastagi (node) and odbc for realtime |
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| 12:11.18 | velix | Can I test manager commands from the asterisk CLI Ã |
| 12:11.22 | velix | Can I test manager commands from the asterisk CLI ? |
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| 15:38.38 | velix | FORBIDDEn. I am going mad. |
| 15:39.23 | *** part/#asterisk Nivex (nivex@triton.nivex.net) |
| 15:42.24 | Samot | On a register or an invite? |
| 15:44.10 | velix | On Register. I'm going up. I've informed the professor to kick the IT in the ass. |
| 15:44.16 | velix | giving up* |
| 15:44.44 | velix | It's not my fault. My PBX and the backup line are working perfectly stable. |
| 15:45.22 | velix | dialplan programming really makes fun. |
| 16:20.11 | velix | https://bpaste.net/CSKQ <-- That's the debug log of a working call (echo). It uses RTP port 27476, doesn't it? Just curious: is it UDP or TCP port? The transport for this peer has been set to TCP. |
| 16:27.16 | velix | Left call worked (echo), right one didn't (no echo at all): https://www.diffchecker.com/Opmuz9LB |
| 16:28.38 | velix | For me, it looks totally identical. |
| 16:36.25 | velix | I've set defaultexpiry=900 but it gets ignored. Still retries 120 |
| 16:40.54 | velix | "sip show settings" also shows 900 ... |
| 16:41.02 | velix | Where do the 120 come from? |
| 16:41.45 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 16:46.06 | velix | Ahm this comes from the other PBX. |
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| 17:39.57 | velix | I don't see any reason, why the echo service fails sometimes. |
| 17:40.18 | velix | I really think of running a second asterisk on another port... |
| 17:40.37 | velix | That'll solve my problem |
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| 18:35.35 | Testeree | why is codec_opus distributed as binary module yet? |
| 18:59.22 | jmordica | Samot: did you see my message responding to you about server setup? Sorry I was offline and may have missed it |
| 19:26.45 | *** join/#asterisk DannyA (~DannyA@cpe-74-64-125-9.nyc.res.rr.com) |
| 19:27.38 | DannyA | hey all. if Tom calls me, and I blind transfer hime to Joe, Joe will see Tom's phone number as his caller ID. When I do an attended transfer to Joe, Tom's number does NOT come up as the caller ID. is there a way to change that? |
| 19:28.05 | DannyA | (this is all via PJSIP -> trunk channels) |
| 19:28.50 | igcewieling | DannyA: not really. In an attended transfer YOU are the person calling. |
| 19:29.27 | DannyA | ah. hmm. can i somehow access Tom's caller ID and set it as the callerid(num) variable? |
| 19:30.00 | igcewieling | if you are using trunks, you can limit the passed callerid in the trunk config. |
| 19:30.14 | DannyA | what do u mean by limit? |
| 19:30.26 | igcewieling | are you using Asterisk or are you using FreePBX? |
| 19:30.31 | DannyA | asterisk |
| 19:30.54 | igcewieling | ah, nevermind then. Yes, you can Set the CALLERID(num) in the dialplan. |
| 19:32.01 | DannyA | right, but what do i set it to? so here's my flow. Tom calls a verizon DID -> Asterisk -> FollowMe -> Pete answers -> Attended Transfer to Joe (extension 123 which does a PJSIP/+1212-call-joey) |
| 19:32.10 | DannyA | how do i get Tom's caller_id to set that in the dialplan |
| 19:33.25 | igcewieling | get it at the start of the dialplan |
| 19:34.08 | igcewieling | Set(SAVEDCID=${CALLERID(num)) then just before the dial out Set(CALLERID(num)=${SAVEDCID}) |
| 19:34.28 | DannyA | right but that's global, that wouldn't be per-call, right? |
| 19:34.47 | DannyA | if i have multiple people calling this DID simultaneously... |
| 19:34.56 | igcewieling | What? No, CallerID is per call (sometimes per channel) |
| 19:35.49 | DannyA | so if the DID enters in the dialplan at Location A, and the attended transfer happens in the dialplan at Location B, if i set a variable in Location A, and access it in Location B, that will be unique to that particular caller? |
| 19:36.00 | igcewieling | yes |
| 19:36.03 | DannyA | ahhh |
| 19:36.05 | DannyA | ok let me try that now |
| 19:36.16 | igcewieling | except for situations where there are Local/ channels involved, then it gets complicated. |
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| 19:59.53 | DannyA | @igcewieling ok, it's definitely getting lost somewhere |
| 20:00.10 | DannyA | and from the logs, Local/ channels ARE involved, though im not dialing a Local/ channel directly |
| 20:00.29 | igcewieling | Try setting it in the pre-dial handler. |
| 20:00.43 | igcewieling | I think followme uses local channels. |
| 20:00.48 | DannyA | would it help to show my dialplan? |
| 20:01.24 | igcewieling | I'm not going to debug dialplan. |
| 20:02.47 | DannyA | np |
| 20:11.57 | DannyA | @igcewieling got it working! needed to prefix the variable with __ so it flowed all the way through |
| 20:14.01 | *** join/#asterisk thansen (~thansen@192.74.130.86) |
| 20:31.55 | DannyA | how do i call goto from ael to go to a context in the extensions.conf file/ |
| 20:49.21 | jmordica | Iâm not able to find any reviews or benchmarking on sangoma transcoding cards vs letting onboarding cpu handle transcoding. Should it really help cpu load when transcoding g711 - g722 with around 150 simultaneous calls? |
| 20:54.54 | *** join/#asterisk zx81 (~zx81@181.197.170.181) |
| 20:54.58 | igcewieling | jmordica: I doubt transcoding ulaw/g722 is a very cpu intensive task. you can run "core show translation recalc 10" to see transcoding costs. |
| 20:55.46 | jmordica | Ok thanks. Is there a transcoder that supports opus as well or no? |
| 20:56.04 | zx81 | Hi all, quick question - is there any way in an AGI to record the audio but detect DTMF at the same time? I'm wanting to do "Please say or type your ZIP Code" |
| 20:56.19 | igcewieling | no idea. I use Sangoma transcoding cards to do ulaw/g729 transcoding. |
| 20:56.21 | zx81 | I see that AGI Record can be terminated by a DTMF but don't see a way to get it |
| 20:56.25 | igcewieling | Since 2009 or so. |
| 20:56.44 | zx81 | alternatively is there a way for Asterisk to recognize DTMF in an audio file? |
| 20:57.01 | zx81 | oh no that wouldn't work because the audio wouldn't contain the DTMF |
| 20:57.14 | zx81 | so yeah just the first one - any way to get the DTMF that stopped the recording? |
| 20:57.34 | igcewieling | zx81: why do you need that information? |
| 20:57.44 | zx81 | In normal Asterisk Record App you can get it in the record results var but don't know if this is available in AGI |
| 20:57.54 | zx81 | doing a covid 19 support app for gov |
| 20:58.02 | igcewieling | Are you using the record app ion the AGI? |
| 20:58.04 | zx81 | need to plot cases against zip code |
| 20:58.07 | zx81 | yeah |
| 20:58.19 | igcewieling | then it works the same, just get the variable. |
| 20:58.21 | zx81 | I guess I could use the dialplan one via exec |
| 20:58.25 | zx81 | oh cool |
| 20:58.27 | zx81 | thanks man |
| 20:58.38 | igcewieling | when I said "use record app" I meant using exec. |
| 20:58.41 | zx81 | yeah |
| 20:58.45 | zx81 | I was typing that |
| 20:58.51 | zx81 | I realized when you said it |
| 20:58.59 | zx81 | I wasn't using exec but I'll just change over |
| 20:59.01 | zx81 | thanks man |
| 21:00.57 | igcewieling | here is part of the AGI code I used to get PINs. https://pastebin.com/qVTtbr0n |
| 21:22.36 | jmordica | Anyone here that is interested in helping track down the cause of high load spikes on a fairly large asterisk setup? Willing to pay for some good consulting/direction. Asterisk realtime with odbc, AEL dialplan, fronted by kamailio, and some fastagi (nodejs) |
| 21:41.27 | *** join/#asterisk sa02irc (~mbax@155-079-043-212.ip-addr.inexio.net) |
| 21:50.18 | velix | Sorry for asking, but how stable is chan_pjsip? I mean, are there any knows bugs? |
| 21:52.37 | Samot | Sigh. |
| 21:52.42 | Samot | It's the primary driver for ASterisk. |
| 21:53.07 | Samot | Chan_SIP hasn't been worked on by Digium since 2014. It's marked OK for removal by 2023. |
| 21:53.21 | Samot | PJSIP has been in use since Asterisk 12. |
| 21:53.24 | Samot | We're on Asterisk 17. |
| 21:53.53 | *** join/#asterisk zapata (~zapata@2a02:1748:fad4:7260:c0f5:82e4:5a15:1a70) |
| 21:55.16 | velix | I've tried all iterations today. More than 12 hours again. I'm always getting "No response received from 'sip:pbx.univer.city:5060' on registration attempt to 'sip:student01@192.168.103.10', retrying in ..." |
| 21:55.42 | [TK]D-Fender | velix, And you're showing the result and not the attempt |
| 21:55.44 | velix | Have to wait until tomorrow. I dropped the professor a message to get money for consulting. |
| 21:58.04 | velix | That was the latest iteration of 48 in the last hour: https://bpaste.net/4WBA |
| 21:59.45 | *** part/#asterisk wyoung (~wyoung@wesleyy.com) |
| 22:06.59 | Samot | outbound_proxy=sip:pbx.univer.city:5060 <-- Wrong format. |
| 22:07.05 | Samot | outbound_proxy=sip:pbx.univer.city:5060;\lr |
| 22:07.14 | Samot | outbound_proxy=sip:pbx.univer.city:5060\;lr |
| 22:07.19 | Samot | That last one |
| 22:08.52 | velix | uh?! hmm okay, let me try. Can I make chan_pjsip more verbose? I've set core to a high debug level, but pjsip only seems to have a history and a logger? |
| 22:15.50 | sibiria | 'pjsip set logger on' |
| 22:16.01 | sibiria | is about the most you can get out of it, i think |
| 22:19.49 | velix | Yeah, running this already ;) |
| 22:20.10 | velix | I've really read all the help manuals, sourcecode, wiki and *.conf.sample during the week on this. |
| 22:20.41 | velix | But \;lr was new to me (and didn't help) ;) |
| 22:20.56 | velix | Ah, it's for "loose-routing". |
| 22:21.59 | file | if you want TCP you'd also need to specify that in the server URI... \;transport=tcp |
| 22:22.32 | file | well, outbound proxy... and you probably also want server_uri to be whatever you're registering to... |
| 22:23.42 | velix | 192.168.103.10 is the server behind pbx.univer.city. When I'm on the university network, I directly can access 192.168.103.10 by UDP and TCp. |
| 22:27.23 | velix | This confuses me since hours: chan_pjsip: REGISTER sip:pbx.univer.city:5060;transport=tcp | chan_sip: REGISTER sip:192.168.103.10:5060 |
| 22:28.18 | file | change server_uri? |
| 22:28.51 | velix | Did so (iteration 23-28), then it tried to connect to 192.168.103.10, which is on my localhost somewhere. It didn't use the outbound proxy. |
| 22:29.29 | velix | Let me try with the new additions. |
| 22:55.18 | *** join/#asterisk lambda (~xiretza@213-47-232-21.cable.dynamic.surfer.at) |
| 23:01.17 | *** join/#asterisk yuljk (~yuljk@unaffiliated/yuljk) |
| 23:16.29 | velix | Highly interesting. I think, I'm getting into details, why the other PBX makes trouble: it suddenly changes the realm. |
| 23:17.12 | velix | other PBX suddently sends: Proxy-Authenticate: Digest realm="192.168.36.1" |
| 23:17.22 | velix | Of course, chan_sip doesn't have an answer for this. |
| 23:23.38 | *** join/#asterisk sinaowolabi (~Sina@169.159.108.246) |
| 23:28.27 | *** join/#asterisk tafa2 (~tafa2@t.ldn.dsrtnet.com) |
| 23:41.50 | Cyrillax | So let me see if i understand the topology here |
| 23:42.39 | Cyrillax | The university has multiple PBX systems on it's internal network. Is this pbx.univer.city proxy new or old? And you're trying to get outside asterisk installations to connect to the internal PBX via the proxy, yes? |
| 23:52.06 | Cyrillax | Is 192.168.103.10 the PBX inside the university network? Because you said that it tried to connect to your "localhost" somewhere. (Guessing you meant localnet?). If you have the same local network subnet as the PBX behind the proxy, that's going to cause you a lot of pain. |