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| 09:41.13 | ruied_ | Hello. Trying to make a calling user to go to jump to a dialplan context (to go to voicemai when press a key), added "context=QueueTest" in the queue and set a QueueTet context in the dialplan, but I can't make the calling queued party to ju,p to the QueueTest. |
| 09:41.51 | ruied_ | the user called and was sent to a queue |
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| 12:40.06 | [TK]D-Fender | ruied, show us the actual full configs and attempt |
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| 13:10.48 | ruied_ | [TK]D-Fender, I have figured out. It is working... Thanks! |
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| 13:33.53 | *** join/#asterisk qakhan (~qakhan@50-204-254-11-static.hfc.comcastbusiness.net) |
| 13:34.29 | qakhan | hi all. i am getting Exceptionally long voice queue length queuing to Local on * 13.18.0 |
| 13:34.40 | qakhan | there is delay in voice |
| 13:40.51 | Samot | That means Asterisk can't find the ;2 part of the Local file. |
| 13:45.31 | qakhan | its local/channel |
| 13:45.41 | qakhan | what could be cause of it |
| 13:46.05 | Samot | That means Asterisk can't find the ;2 part of the Local file. |
| 13:46.33 | Samot | You originate a call on a Local channel.. |
| 13:46.37 | Samot | That's leg ;2 |
| 13:47.00 | Samot | It calls, let's say a SIP channel, when that SIP channel answers it creates leg ;1 |
| 13:47.16 | Samot | It needs to reference ;2 leg |
| 13:47.23 | Samot | It needs to do crap with it. |
| 13:47.33 | Samot | If it can't find it, it can't finish and things break |
| 13:47.43 | Samot | Or it delays and you get what you are getting. |
| 13:47.49 | Samot | I just dealt with this last week |
| 13:50.09 | qakhan | were you able to fix it? |
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| 15:06.36 | qakhan | Samot what did you do to fix the issue |
| 15:09.46 | Samot | Well in my case it was the load on the system. |
| 15:09.59 | Samot | I had complete timeouts/drops |
| 15:10.07 | Samot | You have delay. |
| 15:10.17 | qakhan | yes. |
| 15:10.18 | Samot | Which means it's find it, just taking time to do it. |
| 15:10.35 | Samot | That could still be related to resources on the system. |
| 15:12.14 | qakhan | system resources are good enough. 25% cpu and 11% ram in use |
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| 15:12.40 | qakhan | now i am getting Autodestruct on dialog |
| 15:13.15 | [TK]D-Fender | means nothing |
| 15:15.14 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:15.39 | Samot | And when does this happen? |
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| 15:16.01 | Samot | How many active calls are on the system when this is happening? |
| 15:16.33 | Samot | I mean in my case it was when the system hit about 1900ish. |
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| 15:17.24 | qakhan | at this time i have 11 calls |
| 15:20.54 | igcewieling | Samot: do you recall the load average you had when those issues appeared? |
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| 15:21.52 | igcewieling | heh, you answered |
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| 15:57.38 | qakhan | what are the step i need to take |
| 15:57.46 | qakhan | to resolve this issue |
| 16:03.49 | Samot | I don't know. |
| 16:03.53 | Samot | Is it happening all the time? |
| 16:04.00 | Samot | Or just with certain calls? |
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| 16:18.25 | qakhan | it is happening on some calls since this morning |
| 16:20.42 | igcewieling | qakhan: what is the system load average? |
| 16:22.52 | igcewieling | Asterisk 11 (if you are going to tell me to upgrade, don't bother saying anything). Has anyone had a situation where outbound sendfax never uses T.38, even though the "z" option to SendFax is enabled? |
| 16:28.31 | qakhan | system resources are good enough. arounf 25% cpu and 11% ram in use |
| 16:29.36 | igcewieling | run the "w" command. I'm looking for the system load from the first line "12:29:07 up 23 days, 17:51, 5 users, load average: 0.60, 0.38, 0.38" |
| 16:29.47 | igcewieling | see where it says "load average"? |
| 16:31.06 | igcewieling | where are you getting those percentages from? |
| 16:31.42 | Samot | Also. |
| 16:31.49 | Samot | This isn't a system level resource issue. |
| 16:32.04 | Samot | Those Local channels are in *Asterisk* memory |
| 16:33.13 | qakhan | igcewieling here load average: 1.02, 0.54, 0.32 |
| 16:33.24 | Samot | Originate -> Open Local Channel Leg 2 -> Dial -> Answer -> Open Local Channel Leg 1 and then find the associated Leg 2 of the Local channel. |
| 16:33.26 | Samot | OK.. |
| 16:33.33 | Samot | So that is 5, 10 and 15 minutes. |
| 16:33.48 | Samot | So you were at 100% load in the last 5 minutes |
| 16:33.59 | Samot | 50% 10 minutes ago and 32% 15 minutes ago |
| 16:34.17 | Samot | So in 15 minutes you climbed 70% in usage. |
| 16:34.21 | Samot | er load |
| 16:34.35 | qakhan | Samot it issue is only on incoming queue calls |
| 16:34.58 | Samot | So it makes a local channel for the callers in the queue. |
| 16:35.05 | qakhan | how are you calculating the load? |
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| 16:35.10 | Samot | When the agent is connected, what I said before still applies. |
| 16:35.23 | Samot | 12:33:13 PM <qakhan> igcewieling here load average: 1.02, 0.54, 0.32 <-- |
| 16:35.27 | Samot | 1.0 = 1 CPU |
| 16:35.34 | Samot | Or one core. |
| 16:36.40 | Samot | How many CPUs/Cores and RAM does this system have? |
| 16:37.52 | igcewieling | qakhan: that load average looks good. |
| 16:37.53 | qakhan | 8 CPU and 8GB Ram |
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| 17:35.50 | igcewieling | does anyone know if chan_pjsip still allows invalid keywords like chan_sip does. |
| 17:44.45 | file | invalid keywords? |
| 17:46.26 | igcewieling | file: things like this in sip.conf. It doesn't generate and error and I rely on that. example: account_name=Acme Explosives |
| 17:46.44 | igcewieling | I saw a note pjsip was more strict in the format of the config file. |
| 17:47.30 | file | correct, for the .conf file it won't allow that - the object fails to get created |
| 17:47.34 | file | same applies to options |
| 17:47.47 | rmudgett | Unknown options were never allowed by chan_pjsip. If you try the object won't get created. |
| 17:48.35 | rmudgett | Anything dealing with sorcery or the config framework will not allow invalid options. |
| 17:49.02 | igcewieling | Thanks. I'll have to update my scripts. Does the same apply when using Realtime? |
| 17:49.04 | file | (those being the current way of doing configuration) |
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| 17:49.15 | igcewieling | A typical peer for me: https://pastebin.com/gsEWfZhi |
| 17:49.20 | file | realtime is more lenient, unknown columns are skipped, but per-option validation applies |
| 17:50.12 | igcewieling | I might have found a possible reason to use realtime. 8-| |
| 17:51.55 | igcewieling | Does the realtime problem of needing a "sip reload" for realtime peers to show up still exists when using pjsip? |
| 17:52.07 | igcewieling | not using registration, just peers with static IPs |
| 17:52.31 | file | CLI commands and stuff don't require a reload to have it pull from realtime |
| 17:52.48 | file | you do need a reload for some things, because they inherently have state and without being told or constantly querying we don't know |
| 17:53.47 | igcewieling | *nod* When I realized that I wrote a script to export the realtime database to a chan_sip.conf and stopped using realtime |
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| 20:23.17 | rrittgarn | with app_confbridge, is there anyway to turn off announcements (announce_join_leave) during a conference. Either by dialplan or other means? |
| 20:24.32 | rrittgarn | use case is that customer wants notifications during call setup if someone joins early, but once the call is going, users that did announce themselves, leave interrupting the meeting |
| 20:25.02 | rrittgarn | I know i can set up a specific user profile based on time, but that doesn't solve the issue of people that got on early, dropping out early |
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| 20:47.27 | syadnom | hoping for a quick answer on dahdi. I need dahdi because my asterisk kit uses it for conferencing, I want to run asterisk in LXC and this is the only hickup. Is there an LXC compatible dahdi option that doesn't require me to install something on the host and pass it through? |
| 20:55.46 | igcewieling | syadnom: Confbridge does not require dahdi |
| 20:55.49 | rrittgarn | app_confbridge doesn't need DAHDI, but i think app_meetme did? can you use app_confbridge? |
| 20:55.55 | rrittgarn | damn beat out by three seconds |
| 20:56.39 | syadnom | I don't control the 'suite' so I think I'm stuck on meetme |
| 21:03.22 | rrittgarn | I think you can technically install dahdi in a VM without needing physical hardware |
| 21:03.33 | rrittgarn | you just wouldn't try to configure it it |
| 21:06.38 | syadnom | in LXC I don't have kernel headers, can't compile |
| 21:16.44 | rrittgarn | did you build the container? |
| 21:19.02 | syadnom | template from proxmox. |
| 21:19.13 | syadnom | ie, this is on proxmox ve v5.1 |
| 21:28.06 | rrittgarn | i dunno then, i'm not big enough into containers... i would say if you can't install dahdi, then you have to change meetme to confbridge, if you can't do either you're kind of screwed |
| 21:32.26 | rrittgarn | perhaps switch from a container to a kvm vm? |
| 21:35.49 | syadnom | rrittgarn, I already have a kvm container but it's slow in comparison. kvm>lxc is the goal here. |
| 21:36.10 | *** join/#asterisk tzafrir (~tzafrir@62-90-199-247.barak.net.il) |
| 21:36.30 | syadnom | the kvm container is inexplicably 3-5x slower. takes 25+ minutes to compile asterisk while the lxc container takes 6 minutes. |
| 21:36.53 | syadnom | and this is on a server sitting in a datacenter far away, I have no option but kvm or lxc.. |
| 21:54.06 | *** join/#asterisk chiggins (~chiggins@unaffiliated/chiggins) |
| 22:07.59 | chiggins | Would anyone be able to offer a hand helping me get a Twilio SIP trunk working with my Asterisk setup? My two issues so far are 1) When I call in, my internal device rings and I can answer but there's no audio or anything, and 2) I'm unable to call out. |
| 22:08.43 | chiggins | For the most part I followed this guide https://www.twilio.com/docs/sip-trunking/sample-configuration#asterisk |
| 22:10.17 | file | I swear every day someone has a problem with Twilio SIP trunks |
| 22:10.32 | file | never have I seen such an upstream provider with so many people having problems |
| 22:11.09 | chiggins | D: |
| 22:11.20 | chiggins | That's sad to hear |
| 22:22.50 | *** join/#asterisk Major_Lurker (~Andrew@110-232-243-127.cust.dcsi.net.au) |
| 22:25.24 | chiggins | In terms of fireall, all I need open are UDP ports 5060 and 10000-20000? |
| 22:25.47 | chiggins | Part of me is thinking I'm having firewall issues but if the above is true then it shouldn't be my firewall I think |
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| 23:11.13 | Samot | chiggins: When you say there is no audio, do you mean completely or just incoming? |
| 23:18.21 | chiggins | Samot: Totally no audio on either end |
| 23:19.02 | Samot | Is the PBX behind NAT? |
| 23:21.09 | chiggins | Yu |
| 23:21.12 | chiggins | p |
| 23:26.16 | Samot | Do you have the local network setup on the PBX? |
| 23:26.24 | Samot | In the correct SIP driver config? |
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| 23:37.31 | chiggins | SIP driver should be good, I was able to use the same one to call around within the PBX no problem |
| 23:37.40 | chiggins | And local network setup, not sure what you mean by that |
| 23:54.25 | Samot | I mean Asterisk needs to know the difference between what is WAN (external) and what is LAN (internal) |
| 23:54.31 | Samot | Or local |
| 23:54.53 | Samot | It needs to know what external IP address to use for things like signaling and SDP. |
| 23:55.05 | Samot | Contact headers.. |